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| 02:01.05 | volga629 | Is there are goodway to check if voicemail has being left and find recording file |
| 02:01.10 | volga629 | from dialplan |
| 02:02.35 | volga629 | I see like |
| 02:02.37 | volga629 | <PROTECTED> |
| 02:02.37 | volga629 | <PROTECTED> |
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| 02:03.13 | volga629 | under s-CHANUNAVAIL |
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| 05:33.30 | igcewieling | run dumpchan after the voicemail command or run dumpchan in a hangup handler |
| 05:33.47 | igcewieling | See if there are any variables set which might be useful. |
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| 13:16.35 | seanbright | volga629: check the variable VM_MESSAGEFILE |
| 13:24.11 | seanbright | volga629: a bug from a few years ago that dealth with JABBER_STATUS - https://gerrit.asterisk.org/c/asterisk/+/5286 |
| 13:26.40 | Samot | Well that's all FreePBX dialplan he was showing... |
| 13:27.01 | Samot | So throwing in variables to check might be a bit complicated. |
| 13:27.19 | seanbright | ah, sorry for the noise |
| 13:28.09 | volga629 | I dropped use JABBER_STATUS and from script getting API call to ejabberd that report properly then set STATUS |
| 13:29.57 | volga629 | voicemail it being a bit complicated to add override on proper channel status. Final goal is if voicemail recording left then convert to base64 and delivery via Jabber Send |
| 13:31.32 | volga629 | I found this ${HASVMSTATUS} * hasnewvoicemail(), but not sure if it check all new voicemails or only latest |
| 13:35.49 | volga629 | I can use https://wiki.asterisk.org/wiki/display/AST/VoiceMail+Channel+Variables |
| 13:36.02 | volga629 | to construct message |
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| 13:36.37 | volga629 | but need proper spot in macro-vm |
| 13:37.06 | seanbright | you might have better luck in #freepbx |
| 13:37.18 | seanbright | if that is still a channel |
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| 13:49.21 | Samot | Yup. |
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| 15:37.23 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.36.0 (2020/09/03) 16.13.0 (2020/09/03) Standard: 17.7.0 (2020/09/03); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 15:45.10 | Alex69 | still I think I'll stay with FreePBX, the web ui is pretty powerful for things like reporting, updating, managing... |
| 15:47.25 | Samot | Sure. |
| 15:47.48 | Samot | But if you bypass all the things that FreePBX does, like your contexts do, then how is that helpful? |
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| 15:53.58 | Alex69 | you still have a clean way of looking at logs, auto updating the software |
| 15:54.08 | Alex69 | email notifications when something goes wrong |
| 15:54.19 | Alex69 | dashboard to checkout who called who |
| 15:54.24 | Alex69 | dashboards to checkout who called who |
| 15:54.50 | Alex69 | clean way to look at the basic configurations |
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| 15:58.03 | Samot | Alex69: Yes you do. |
| 15:58.32 | Samot | Alex69: However , exten => 001,1,Dial(PJSIP/001) is skipping a lot of things. |
| 15:59.11 | Samot | Like CallerID, setting certain CDR variables or other variables that might be used for tracking, logging and other features. |
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| 15:59.39 | Samot | You also skip the hangup process that is called to clean up things. |
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| 16:03.14 | Alex69 | I see, I am willing to take the risk that this won't be an issue since the benefit from the web ui that comes from using FreePBX seems to outweigh the issues that may arise. However, it might be indeed that the issues are so large that I might be forced to switch back to asterisk |
| 16:04.55 | Samot | I don't think you're really understanding what I'm saying. |
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| 16:35.00 | Alex69 | I think that you mean that the updating process may screw things up or that the logs/statistics might be not correct, I hope they will be |
| 16:35.37 | Alex69 | I'm trying to do the following pattern matching: exten => 00XX,1,Dial(PJSIP/00XX) |
| 16:36.04 | Alex69 | this one doesn't seem to work because I think the Dial() function does not accept it, is there a clean way to achieve this behavior? |
| 16:38.57 | Samot | No that doesn't work. |
| 16:39.12 | Samot | On a couple levels. |
| 16:39.20 | Samot | Matches need a _ at the start |
| 16:39.43 | Samot | exten => _00XX,1,Dial(PJSIP/${EXTEN}) |
| 16:45.42 | Alex69 | ${EXTEN} this will just copy paste the match? |
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| 17:32.20 | Kobaz | 00XX is a 'literal',,, and will match exactly the characters 00XX |
| 17:32.24 | Kobaz | which can be useful... |
| 17:32.33 | Kobaz | but if you want pattern matching, you need the _ |
| 17:32.36 | Kobaz | that's the difference |
| 17:33.34 | Kobaz | oh he's gone |
| 17:33.36 | Kobaz | whoops |
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| 17:51.32 | igcewieling | *sigh* 4 hours and the on-site techs have managed to plug in 1 phone. |
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| 18:51.39 | wayne47 | Still fighting with pjsip. Here is a trace of 5143 (SIP) calling 5147 (PJSIP). 5147 never rings. When 5147 calls 5143, the call goes through but I have one-way audio. 5147 worked fine on SIP. Suggestions on where to look next? |
| 18:51.40 | wayne47 | https://pastebin.pl/view/27ad5788 |
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| 19:11.24 | igcewieling | Stop running both at the same time. |
| 19:12.11 | igcewieling | [2020-09-05 16:47:02] ERROR[19139][C-0000002a] res_pjsip_header_funcs.c: This function requires a PJSIP channel. |
| 19:12.38 | igcewieling | are you doing something stupid like not autoloading modules or disabling modules you don't understand? |
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| 19:44.33 | wayne47 | I'm in the process of converting a working PBX from SIP extensions to PJSIP. As I can not get the PJSIP extensions to work at all, I'm can't take down the SIP extensions that are actually working. Did not autoload anything new, only change was to force rport to NO for Cisco phones. |
| 19:45.21 | wayne47 | Only "odd" thing is that SIP is on 5060 and PJSIP is on 5160 |
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| 19:45.49 | sibiria | are your channel formats correct? |
| 19:46.01 | sibiria | i.e. PJSIP/exten@trunk |
| 19:46.11 | igcewieling | then at least put 2 phones on PJSIP and test between them |
| 19:46.32 | wayne47 | I've tried that. They can call each other but no audio gets passed. |
| 19:46.51 | wayne47 | PJSIP phones seem never to send audio, but can receive. |
| 19:48.55 | igcewieling | *shrug* It isn't a PJSIP issue. |
| 19:49.19 | wayne47 | Just wondering where to look |
| 19:51.29 | igcewieling | I'd look at NAT and localnet settings |
| 19:51.37 | wayne47 | Not using NAT |
| 19:52.24 | igcewieling | I didn't say you were using it. If it is configured wrong nat settings could cause problems with non-nat |
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| 19:53.08 | wayne47 | different between SIP & PJSIP? As the extension worked fine on SIP |
| 19:53.37 | igcewieling | they are totally different config files, there is no one-to-one conversion. |
| 19:54.14 | igcewieling | my point is that PJSIP isn't the problem since PJSIP works for other people. |
| 19:54.38 | wayne47 | I'm sure it's my issue, just trying to figure out how to debug the issue |
| 19:55.53 | igcewieling | pastebin the output of "pjsip show endpoint" for your two PJSIP endpoints |
| 19:56.09 | igcewieling | well pjsip show endpoint the-endpoint-name |
| 19:56.52 | igcewieling | also "pjsip show transport the-transport-you-use |
| 19:57.12 | igcewieling | on my system it is "pjsip show transport 0.0.0.0-tcp" |
| 19:58.57 | igcewieling | I'm not going to troubleshoot chan_sip<->chan_pjsip, if you want that, someone else will have to help you |
| 19:59.04 | wayne47 | Here's transport https://pastebin.pl/view/4f14a658 |
| 19:59.42 | igcewieling | do you really have localnet=A.B.C.0/255.255.254.0 in your transport? |
| 19:59.48 | wayne47 | redacted |
| 20:00.01 | igcewieling | don't do that. redact only passwords. |
| 20:00.24 | igcewieling | 172.16.7.1 looks wrong too. |
| 20:00.33 | wayne47 | THAT is right |
| 20:01.05 | igcewieling | no, that is supposed to be the public IP. |
| 20:01.27 | igcewieling | pastebin the actual transport setup and we'll see. |
| 20:01.54 | wayne47 | It's all private via VPNs. No traffic goes out on public IP other than the connection to the VOIP p[rovider |
| 20:02.42 | igcewieling | You still need the public IP as the external address if the server is going to call the provider via NAT. |
| 20:03.20 | igcewieling | you need localnet set for each network which is directly accessable without NAT |
| 20:03.52 | wayne47 | server calls provider on SIP on public IP. Extensions (both SIP and PJSIP) all use VPNs and use 172.16.7.1 |
| 20:04.40 | igcewieling | right, but you are telling Asterisk to use 172.16.7.1 as the public IP when sending calls to your provider. |
| 20:05.05 | igcewieling | What network blocks will have phones on them? |
| 20:05.18 | igcewieling | You want only those listed as local nets in the transport config. |
| 20:05.42 | igcewieling | I still don't see a pastebin for the endpoint configs |
| 20:06.02 | wayne47 | Providers: |
| 20:06.04 | wayne47 | sip show registry |
| 20:06.04 | wayne47 | 16:04:12 |
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| 20:08.08 | wayne47 | Here's 5147: https://pastebin.pl/view/431b7804 |
| 20:09.21 | igcewieling | allow only ulaw as the codec. you can' tuse g729 or g721 |
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| 20:12.20 | wayne47 | what about alaw? |
| 20:13.04 | igcewieling | Are you in the USA or CANADA? |
| 20:13.09 | wayne47 | us |
| 20:13.14 | igcewieling | the no alaw |
| 20:13.43 | wayne47 | will do |
| 20:13.50 | igcewieling | Once it is working, then I'd enable g722 and any video codecs your phones might support. |
| 20:14.36 | wayne47 | Progress - I now have audio from extensions. But 5143(SIP) still can not call 5147(PJSIP). Straight to voicemail |
| 20:14.56 | igcewieling | comment out any local net or external ip config options or fix the settings |
| 20:15.06 | wayne47 | Thanks, I figured phones negotiated the codecs with the PBX |
| 20:15.39 | igcewieling | in theory yes, in practice it rarely works as expected. |
| 20:17.09 | wayne47 | Got it |
| 20:19.01 | igcewieling | Run on the PBX box from a shell, what is the output of: wget -q -O - "http://help.nyigc.net/ip/?script=1" |
| 20:19.10 | igcewieling | that should be your external IP for pjsip |
| 20:20.16 | igcewieling | local net settings should be networks which don't require NAT to get to |
| 20:20.33 | wayne47 | that returns the correct IP but the extensions will never see traffic unless it comes from 172.16.7.1 |
| 20:21.49 | wayne47 | yes, my local nets are 172.16.7.0/24 and RealIP/23 (correct net masks) |
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| 20:23.47 | igcewieling | no, that is not a localnet for this. |
| 20:24.30 | igcewieling | localnets are the local networks behind NAT. your real IP is not behind NAT |
| 20:24.43 | wayne47 | So just kill em all? |
| 20:25.42 | igcewieling | The 172.16.7.x network is not behind nat from the standpoint of the server, so that one should stay, and you should add any remote networks to local nets if there is a VPN between them and no NAT |
| 20:27.52 | wayne47 | pulled em both - no change in behavior. The server should not thing anything is behind NAT |
| 20:27.57 | wayne47 | ..think... |
| 20:28.57 | igcewieling | then comment out your external IP and localnet settings and see if it helps. just remember you have to restart Asterisk when you make changes to the transport. |
| 20:29.18 | igcewieling | if you didn't do that then you didn't apply the change. |
| 20:32.45 | wayne47 | restarted as required. OK, oddity. 5147 has voicemail enabled 5347 does not. 5347 can not call 5147 (like it's on dnd, but it's not). 5147 CAN call 5347 |
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| 20:37.08 | wayne47 | I'll mess with it from here, might have to destroy the extension (I converted it from SIP via the GUI). Thanks again for the Codec help |
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| 20:44.36 | Samot | Contact: 5147/sip:5147@198.11.63.248:50890 << Has that changed? |
| 20:44.41 | Samot | Or is it still a private IP? |
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| 21:20.45 | beta2 | exit |
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