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| 06:01.13 | snadge | what am i missing here.. i've got insecure=port,invite and a permit line which includes the ip address that its coming from |
| 06:01.19 | snadge | but asterisk is still responding unauth |
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| 06:52.05 | drmessano | Pastebin the config |
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| 06:59.01 | snadge | http://pastebin.com/UwkAvdF4 .. thats the inbound trunk |
| 07:02.32 | drmessano | Your Deny needs to be after the Permit |
| 07:02.36 | drmessano | OOPS.. Before |
| 07:03.01 | drmessano | You need to Deny All then Allow the IP Mask |
| 07:03.21 | snadge | thats probably true.. i've been messing with it.. but i think i may have figured it out |
| 07:03.33 | drmessano | It's not probably true, it is true |
| 07:03.37 | snadge | the trunks name is 19616 .. so the fromuser for the inbound call, probably needs to be set to that |
| 07:03.53 | snadge | otherwise the call needs to be accepted in the default context |
| 07:03.55 | snadge | something like that yes? |
| 07:04.00 | snadge | which is denied by default |
| 07:04.31 | snadge | when the call hits the pbx.. with debug turned on.. it never finds 19616 or mentions it anywhere |
| 07:04.31 | drmessano | The peer has to match the fromuser, yes |
| 07:04.55 | snadge | such a noob mistake.. wowsers.. but this is an unfamiliar upstream provider |
| 07:05.03 | snadge | so i dont know how to tell them to set the fromuser for inbound calls |
| 07:05.13 | snadge | on our system, we just tick a box on the username in the billing account |
| 07:06.08 | drmessano | You can look at the SIP Debug and see what they are setting it to, then match appropriately |
| 07:07.03 | snadge | fromuser is set to the DID or the same as the inbound number |
| 07:07.11 | snadge | and thats not very useful :P |
| 07:08.18 | drmessano | So then use type=peer and stop trying to use type=user |
| 07:08.59 | snadge | can you still use a range of ip addresses for that? |
| 07:09.02 | snadge | or does it have to be a single ip |
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| 07:09.08 | drmessano | Yes you can |
| 07:11.11 | drmessano | With PJSIP you can. With chan_sip you'll need to define multiple peers. But you can shortcut it and template it, to some extent |
| 07:13.18 | snadge | upstream provider is parent company.. so.. on monday (friday afternoon now) i'll be able to try to get them to configure their end, to work with type=user .. as that is much easier |
| 07:13.49 | snadge | they have load balancing clusters with different ip ranges etc, at multiple locations |
| 07:14.11 | snadge | and the pbx uses chan_sip .. so it doesn't sound like much fun to go down that path |
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| 07:15.45 | snadge | they use freeswitch with kamailio in front of it.. fun times |
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| 08:09.53 | jay8232 | I have compiled both asterisk (13.11.0) and pjsip (trunk) with opus support, but I get error message "channel.c:5505 set_format: Unable to find a codec translation path: (jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|ulaw|alaw|slin) -> (opus)" when trying to setup a opus<->PCMA call. Am I missing some config stuff? |
| 08:10.17 | obrut | does anybody know, why digium dropped fax2asterisk support in asterisk 13 and later ? |
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| 08:11.00 | stefan27 | asterisk 13.11 does not support opus codec does it |
| 08:11.21 | stefan27 | Aha, sorry didn't read your message correctly |
| 08:11.54 | stefan27 | You might need an opus patch for asterisk still |
| 08:11.57 | jay8232 | stefan: Well I dunno. Just compiled with it. Using pjchan anyways.. root@jlinux-compiler:/jlinux/source/asterisk/13.11.0/asterisk-13.11.0# ./configure --help|grep opus --with-opus=PATH use Opus files in PATH |
| 08:12.19 | jay8232 | Ok so it only supports passthrough an no resampling? |
| 08:13.19 | drmessano | Correct |
| 08:13.25 | drmessano | Passthrough, not transcoding |
| 08:13.42 | drmessano | I'm not aware of Opus transcoding support |
| 08:13.49 | *** join/#asterisk bof23 (~Thunderbi@185.13.183.107) |
| 08:14.09 | drmessano | A current version, that is |
| 08:15.08 | jay8232 | Then there must be something wrong with either asterisk or pjsip. The sdp in party A contains opus, g722 and pcma. Party B contains pcma, pcmu and l16. Asterisk decided to accept opus and not the shared codec pcma |
| 08:15.21 | stefan27 | We got opus to work on asterisk 13.5 but then we disabled it... but we had to apply this custom patch https://github.com/meetecho/asterisk-opus |
| 08:15.31 | stefan27 | and that may not be compatible with asteisk version 13.11 |
| 08:15.39 | stefan27 | because it was designed for asterisk 11.2 |
| 08:15.43 | jay8232 | k |
| 08:16.21 | jay8232 | If asterisk went for the pcma codec instead of opus there wouldnt be a need of transcoding |
| 08:17.17 | stefan27 | is that asterisk under your control? you can force it to use pcma? |
| 08:18.18 | jay8232 | Yeah Im running it on a test-platform. I am guessing it went for opus because its first in the allow-list (because the audio quality is better), but the particular other remote party does not support it |
| 08:18.47 | jay8232 | I want to add support for opus (and amr-wb) while still supporting the old legacy devices and pcma |
| 08:20.12 | jay8232 | Btw, I'm not sure who I should report this to, but I found this "funny" bug: |
| 08:20.17 | jay8232 | res/res_pjsip_authenticator_digest.c static int verify(struct ast_sip_auth *auth, pjsip_rx_data *rdata, pj_pool_t *pool) if (authed == PJ_SUCCESS || authed == 171110) { // pjsip defines PJ_SUCCESS as 0 while asterisk defines it as 171110 |
| 08:20.42 | jay8232 | Had to change the digest authentication code to get it working between asterisk and pjsip :) |
| 08:41.34 | jkroon | hi all, i'm seeing a very strange one with core show channels |
| 08:41.55 | jkroon | 164 active channels with 347 active calls |
| 08:42.05 | jkroon | how is it possible to have more active calls than active channels? |
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| 09:21.28 | linjan | hello there! im using asterisk 13. is there any application/function to check if user is a member of specific queue? |
| 09:52.32 | scv | linjan: no application to do so afaik |
| 09:52.47 | linjan | :( |
| 09:54.02 | wyoung | hey gang |
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| 11:30.39 | zahyl | hi, does someone can help me with a problem i got with setting up an asterisk server? " Rejecting 'IP' due to a failure to pass ACL '(BASELINE)' "despite "permit=IP/255.255.255.0" in sip.conf . Thx |
| 12:03.04 | wyoung | zahyl: yes I can help |
| 12:04.20 | wyoung | zahyl: sorry, what I mean to say was, I am someone can help you setting up an asterisk server. Wow, you got an right, I am impressed |
| 12:04.46 | wyoung | most English speaking people get that wrong |
| 12:05.38 | wyoung | zahyl: what is your fill sip.conf (XXXX out the passwords please, and use hastebin or equiv to paste) |
| 12:05.57 | Samot | And the BASELINE ACL settings. |
| 12:06.10 | wyoung | yes |
| 12:06.13 | wyoung | what Samot said |
| 12:06.30 | Samot | Since that's what is failing. |
| 12:07.38 | wyoung | Samot: sip.conf was mentioned also |
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| 12:08.00 | Samot | Yes and he could have that in sip.conf |
| 12:08.04 | Samot | Or in acls.conf |
| 12:08.21 | Samot | I'm just saying, along with the sip.conf stuff make sure the baseline ACL settings are there too. |
| 12:08.34 | Samot | If they are in sip.conf, no problem. If they are not, post them as well. |
| 12:09.03 | wyoung | Samot: just cat /etc/asterisk/* > all_of_the_conf and paste that, just to be reaall sure |
| 12:09.31 | Samot | I'm just looking at the error. |
| 12:09.40 | Samot | It says the IP failed the BASELINE ACL check. |
| 12:09.54 | Samot | I'd look at the BASELINE ACL. |
| 12:11.54 | wyoung | Samot: he / she / shim / unspecified pasted their confs already? |
| 12:12.11 | Samot | Not that that I've seen. |
| 12:13.14 | wyoung | ok |
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| 12:28.18 | zahyl | wyoung: sry for my bad english. there it is my sip.conf : http://pastebin.com/LpH4dpjr |
| 12:28.39 | wyoung | zahyl: no it is fine :) |
| 12:31.34 | zahyl | i didn't even modifie the default acl.conf |
| 12:33.42 | zahyl | and here is theexact error message : http://pastebin.com/ZtMKvAdD |
| 12:33.58 | Samot | Fix your permit. |
| 12:34.11 | Samot | You have a /24 IP range but a /32 netmask. |
| 12:34.21 | Samot | Oh wait, read that wrong. |
| 12:34.23 | Samot | That's good. |
| 12:35.22 | Samot | What happens when you specify an IP? |
| 12:35.39 | Samot | Does the device on that IP still have ACL issues? |
| 12:35.56 | Samot | Show the BASELINE acl. |
| 12:36.28 | zahyl | yes, when i specify the 192.168.56.10 address i got the same error message |
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| 12:36.57 | Samot | That is still has a failure to pass the BASELINE acl? |
| 12:37.02 | Samot | Again, why aren't we looking at this ACL? |
| 12:37.31 | zahyl | the baseline acl is the default one, so it's empty |
| 12:38.30 | Samot | Have you tried adding the rules there? |
| 12:38.32 | Rasputin3711 | host=dynamic ; Lâutilisateur nâest pas associé à une IP fixe |
| 12:39.07 | Samot | Correct. |
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| 12:40.13 | [TK]D-Fender | defaultuser = user3 |
| 12:40.23 | [TK]D-Fender | Your register is coming in as 200@..... |
| 12:40.38 | [TK]D-Fender | this looks wrong since they are usuing the peername as the username |
| 12:40.42 | [TK]D-Fender | get rid of tthis |
| 12:44.34 | zahyl | host=static does-nt seems to be a good parameter |
| 12:45.31 | Samot | TK already pointed out the issue. |
| 12:45.49 | Samot | You're using the wrong username to connect. You're using the peer name instead of the username. |
| 12:54.24 | zahyl | thank you guys, the error message is gone |
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| 13:04.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:21.06 | vadiml | Hello, i wonder if i can assign channels from 2 diffrent spans to the same channel group |
| 13:21.42 | vadiml | so when on span is full the channels for the other span will be used automatically |
| 13:22.23 | [TK]D-Fender | you can |
| 13:22.36 | wyoung | hey [TK]D-Fender! |
| 13:22.40 | wyoung | how have you been? |
| 13:23.24 | [TK]D-Fender | meh |
| 13:23.52 | wyoung | oh |
| 13:24.04 | wyoung | The asterisk business hasn't been great? |
| 13:25.08 | wyoung | Rasputin3711: That's some nice unicode there |
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| 14:30.16 | vadiml | Hi folks, i need a litlle help with asterisk |
| 14:38.04 | wyoung | vadiml: ok |
| 14:38.14 | wyoung | what type of help do you need |
| 14:39.17 | wyoung | although please note that I am a programmer by trade, I cant help you with any emotional issues you may have with asterisk |
| 14:41.41 | vadiml | i wonder if i can assign channels from 2 diffrent spans to the same channel group |
| 14:41.52 | *** join/#asterisk jkroon (~jkroon@197.96.224.13) |
| 14:41.57 | vadiml | so when one span is full the channels for the other span will be used automatically |
| 14:42.19 | [TK]D-Fender | <vadiml> Hello, i wonder if i can assign channels from 2 diffrent spans to the same channel group |
| 14:42.19 | [TK]D-Fender | <vadiml> so when on span is full the channels for the other span will be used automatically |
| 14:42.19 | [TK]D-Fender | <[TK]D-Fender> you can |
| 14:42.20 | WIMPy | vadiml [TK]D-Fender already confirmed that. |
| 14:42.27 | [TK]D-Fender | I answered you the first time, and within 1 minute |
| 14:43.05 | vadiml | Sorry i missed th response... Thank you very much |
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| 15:03.47 | wyoung | hi |
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| 18:32.46 | nenadalm | Hi. I just wanted to start with asterisk using mobile_chan as described here: https://www.mattgibson.ca/installing-and-configuring-chan_mobile-for-bluetooth-presence-support-in-asterisk-16/. I added mac address of my bt adapter into chan_mobile.conf and then run "$ rasterisk" but when I type in "*CLI> module load chan_mobile.so" I get "[Sep 23 20:25:11] ERROR[23338]: chan_mobile.c:4718 load_module: No Bluetooth devices found. Not loading mo |
| 18:32.46 | nenadalm | phone via "bluetoothctl" so my adapter is definitely working). Any ideas? |
| 18:33.10 | nenadalm | (I restarted asterisk service after editing the config ofc) |
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