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| 03:19.07 | KK4YQV | Hello all, I'm getting a 407 error when trying to setup SIP Trunking on my pbx. FreePBX 14.0.5.25. SIP Trunk - chan_pjsip. I do not see where my FreePBX is responding to the authentication request. From what I'm reading I need to edit pjsip.registration_custom.conf , but I'm not sure if this is correct. Any help or guidance would be appreciated. Thank you in advance. |
| 03:29.32 | rockman37 | KK4YQV: Can you post your current config and a debug trace with 'sip set debug on'? |
| 03:29.45 | [TK]D-Fender | Wrong debug command |
| 03:29.59 | [TK]D-Fender | "pjsip set logger on" |
| 03:30.33 | [TK]D-Fender | And do this in FreePBX's channel as that is where you will be able to get advice on how to use their GUI to fix settings as required |
| 03:30.38 | [TK]D-Fender | #freepbx <- |
| 03:31.12 | KK4YQV | [TK]D-Fender - Thanks! I didn't know they had a channel too. |
| 03:31.23 | KK4YQV | Heading there, thanks for thehelp |
| 03:32.19 | rockman37 | [TK]D-Fender: Oh yeah, thanks, I'm too used to chan_sip. |
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| 04:43.51 | wyoung | [TK]D-Fender: Is pjsip prefered over chan sip? |
| 04:43.59 | wyoung | Or is chan sip dead now? |
| 04:48.25 | Samot | Chan_SIP has been deprecated since 2014. |
| 04:49.10 | Samot | It's road mapped to be removed from Asterisk completely within the next 1-2 LTS releases. |
| 04:49.37 | Samot | Rumbles are it might be no-load in v17 as part of that road map. |
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| 10:15.30 | danjenkins | chan_sip isnt deprecated yet. its on extended support. but its on the path of being deprecated - it probably will never be removed but yes the plan long term that was agreed at devcon 2 years ago was to no-load it etc. I'd love to see it removed - it causes so much pain with people just starting with Asterisk as to what to use |
| 10:18.16 | rockman37 | Sigh, I guess I'll have to migrate. |
| 10:18.41 | danjenkins | pjsip performs extremely well now |
| 10:18.52 | danjenkins | you'll thank yourself! |
| 10:19.09 | rockman37 | Mm, although I've had so trouble with chan_sip. |
| 10:19.20 | rockman37 | It's the configuration of pjsip that I don't look forward to. |
| 10:19.46 | danjenkins | theres a pjsip wizard that lets you configure it in a pretty nice way |
| 10:19.57 | danjenkins | but to be perfectly honest, it doesnt take long to change over |
| 10:20.18 | rockman37 | I feel like a wizard would make reconfiguration a pain. |
| 10:20.28 | danjenkins | theres probably some videos on the astricon youtube channel to help you |
| 10:20.46 | rockman37 | I may as well sit down and do it some time. I've got upgrade from 15 to 16 anyway. |
| 10:21.02 | danjenkins | but in essence there are endpoints and those endpoints have aors and auths |
| 10:21.12 | danjenkins | none of this friend/peer nonsense any more |
| 10:21.22 | danjenkins | but i remember switching and it did take a little while |
| 10:21.31 | danjenkins | but i couldnt write chan_sip config today |
| 10:21.34 | rockman37 | To be honest if I have to watch a video to be able to configure something, there's probably something wrong with it (or its documentation). |
| 10:21.39 | danjenkins | pjsip makes sense, once youve switched |
| 10:21.43 | rockman37 | I'm sure I can work it out. |
| 10:22.27 | danjenkins | different people learn differently :) I like watching a video (asterisk from scratch by an asterisk trainer) when it comes to something new - just how i take information in |
| 10:22.54 | danjenkins | but if you do find docs lacking - let someone know and we can try and make them more thorough etc |
| 10:23.00 | rockman37 | Fair enough. |
| 10:23.12 | rockman37 | I'm reading them now. |
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| 10:27.33 | rockman37 | Mmkay, I guess the config makes some kind of sense. |
| 10:27.45 | danjenkins | :) |
| 10:29.08 | rockman37 | Can you explain server_uri and client_uri in type=registration? |
| 10:30.01 | danjenkins | erm, let me go and have a read |
| 10:31.11 | rockman37 | I'm honestly not sure if my provider is trunking or not. |
| 10:31.39 | danjenkins | how do you mean? |
| 10:32.35 | danjenkins | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_outbound_registration |
| 10:32.43 | danjenkins | client is how you want to show yourself as |
| 10:32.51 | danjenkins | and server is the uri you want to connect to |
| 10:35.44 | rockman37 | As in, it works, but apparently there is a distinction between trunking providers and non-trunking providers that I don't understand. |
| 10:36.14 | rockman37 | In chan_sip I don't think I'm setting what to show myself as. |
| 10:36.49 | danjenkins | i dont think you do from what i recall |
| 10:37.20 | danjenkins | yu probably dont even need to specify it if you dont want to |
| 10:38.20 | danjenkins | so theres a "registration provider" and then theres a provider who just sends invites to a designated URI and then yeah they need to be treated differently |
| 10:42.18 | rockman37 | Ah right. I do register with my provider. |
| 10:55.29 | rockman37 | So in chan_sip I can specify which extension a call comes in on in the register => line. How would I do the same with pjsip? |
| 10:58.41 | file | contact_user |
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| 10:59.45 | rockman37 | file: Thanks. |
| 11:01.12 | file | you can also try the line support, https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ which if the remote obeys the SIP RFC... will remove the need to do any sort of inbound IP based matching |
| 11:06.06 | rockman37 | Okay, cool. |
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| 11:17.59 | rockman37 | Huh, so the pjsip wizard isn't a wizard at all, but some kind of preprocessor. |
| 11:18.07 | rockman37 | Wish I'd known that earlier. |
| 11:22.19 | danjenkins | sorry rockman37 |
| 11:22.34 | rockman37 | Not your fault. |
| 11:26.10 | danjenkins | can be a tad confusing - there is also a tool that helps translate your sip.conf file over to pjsip.conf but in my experience it doesn't cover all it needs to etc and just confuses people more. depends how complicated your config is I guess |
| 11:27.06 | rockman37 | It's not complicated. I've got various registrations to two servers, and two phones. |
| 11:27.24 | rockman37 | I think I'll work with the wizard but I'm also thinking I should go to bed. |
| 11:38.13 | Chainsaw | rockman37: A well-rested magician is a good magician. |
| 11:39.02 | rockman37 | Chainsaw: Heh :) |
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| 13:46.34 | sibiria | what's the proper way to disable features that react to # and * DTMF? |
| 13:46.51 | sibiria | (so as to allow these extensions in the dial plan) |
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| 13:48.16 | Samot | You don't program features to use them. |
| 13:48.21 | sibiria | everything under the general and featuremap sections in features.conf is commented out |
| 13:48.32 | sibiria | but maybe there are defaults coming to life by the sections simply being there? |
| 13:48.33 | Samot | And what is the issue? |
| 13:49.20 | file | yeah, you'll need to be specific and give a scenario... |
| 13:49.23 | sibiria | that asterisk appears to capture * and # DTMF, denying me to use these in the dial plan |
| 13:49.39 | Samot | During a call? |
| 13:49.43 | sibiria | yes |
| 13:50.07 | file | are you placing a new call with * or # at the front? or are you doing DTMF features while inside of an existing call? |
| 13:50.13 | sibiria | file: the latter |
| 13:50.42 | file | and what happens? what's the console output? |
| 13:50.42 | sibiria | i simply want # and * to work in the dial plan as 0-9 does, but some feature is hijacking them and i'm not sure where |
| 13:50.46 | sibiria | none, actually... |
| 13:50.54 | sibiria | i'm a bit confused |
| 13:51.07 | Samot | Show a call that this happens on... |
| 13:57.25 | Samot | 5:15:33 AM <@danjenkins> chan_sip isnt deprecated yet. its on extended support. <-- Sorry just saw this but by that you mean that Asterisk/Digium no longer actively develop or support or do fixes on Chan_SIP and that everything that has been done since v14 on Chan_SIP is community driven and contributed as being "extended support"? |
| 13:58.50 | file | it's been community supported since 13 |
| 13:59.08 | danjenkins | thanks file |
| 13:59.20 | file | we (Digium/Sangoma) only get involved for reviews, and if a community change goes in that breaks something that isn't resolved by a community person |
| 14:00.06 | danjenkins | but because you know... its the default new people end up using a stack that isnt actively maintained by the project (sorry, inserting personal feelings about it ;) ) |
| 14:00.32 | Samot | Part of that problem is providers. |
| 14:00.48 | Samot | Even then with their Chan_SIP configs they are like from 1.6/1.8 |
| 14:00.49 | danjenkins | with their docs that show chan_sip config? |
| 14:00.55 | Samot | Correct. |
| 14:01.04 | file | plus chan_sip just has history with it across the internet |
| 14:01.30 | danjenkins | yup completely agree; ive been trying to influence any that i have dealings with to have pjsip config since devcon 2 years ago |
| 14:01.32 | Samot | Like I said most of them still have outdated Chan_SIP examples anyways because no one keeps them updated. |
| 14:01.50 | Samot | And by outdated I mean even for Chan_SIP standards. |
| 14:02.59 | Samot | Hell, some have examples that contain settings that don't even exist in Chan_SIP but they probably found some weird blog that had them. The issue with Chan_SIP is it doesn't overly puke or not work with bad configs. |
| 14:03.14 | Samot | So most people have no clue some setting they have is garbage and doesn't do jack. |
| 14:04.29 | danjenkins | many people who know better still believe pjsip wont work for them |
| 14:05.04 | Samot | Well... |
| 14:05.24 | Samot | I have a theory about that. Many people only have SIP experience in Asterisk and thus Chan_SIP. |
| 14:05.50 | Samot | They actually don't know SIP outside of Asterisk so the fact that PJSIP is more SIP RFC compliant and follows standards makes people confused. |
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| 14:07.10 | Samot | Terms like AOR or Contacts throw them for a loop. |
| 14:07.31 | danjenkins | you mean friend doesnt mean anything? |
| 14:07.43 | danjenkins | sorry, feeling super bitchy today :) |
| 14:07.49 | Samot | Yes. |
| 14:08.18 | Samot | Or the fact that PJSIP is now letting Asterisk do things that pretty much every other SIP driver has been doing since the start of Asterisk, |
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| 14:18.13 | Samot | \ |
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| 14:29.06 | simbalion | Hi, my pbx seems to randomly fail on outgoing calls, the error says "rejected because extension not found in context..', but this happens randomly and tends to clear up after awhile. Is this a problem with the SIP provider? |
| 14:30.27 | sibiria | more likely a problem with your dial plan |
| 14:30.49 | simbalion | sibiria: I understand, but it works fine for days and then fails randomly. I haven't made any changes to the dialplan |
| 14:31.14 | sibiria | do you perhaps have something that regularly renders the dial plan, or do you query it from a database? |
| 14:31.41 | simbalion | sibiria: nope, it's a very static setup, I only have two extensions and it's all in /etc/asterisk/*.conf files |
| 14:32.59 | sibiria | and asterisk can read extensions.conf? |
| 14:33.14 | simbalion | sibiria: Yep |
| 14:34.07 | simbalion | I made some outbound calls just yesterday according to my call log, nothing's changed from then to today |
| 14:35.54 | simbalion | Seems to me the issue is probably the upstream provider |
| 14:36.00 | sibiria | i presume you've verified the extensions when this happens, so that they contain the context the error refers to and that said context has the extensions you want to reach |
| 14:37.38 | simbalion | ah wait, I was only entering a 9 digit number |
| 14:37.54 | simbalion | that was it. im too stressed lol. |
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| 15:23.38 | Kobaz | anyone have an idea why i would be getting odbc logs writing to /var/spool/asterisk |
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| 15:34.32 | jsmith | Kobaz: Do you have debug tracing turned on? |
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| 15:49.54 | Kobaz | yes, but... Trace = Yes TraceFile = /tmp/extodbc.log |
| 15:50.26 | Kobaz | this isn't trace data though |
| 15:50.47 | Kobaz | https://pastebin.com/g1s9MRrc |
| 15:50.59 | Kobaz | perhaps there's a seperate error log file |
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| 16:19.21 | zaf | continuation of the queues issue from friday: when queues stop taking calls, any other queue operations hang the cli, for example just tab completion on queue show <tab> will crash the cli |
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| 16:29.34 | Samot | zaf: What are you running? |
| 16:30.50 | sibiria | the */# DTMF problem i had earlier turned out to not be asterisk - they're being filtered away by all three providers' trunks for god knows what reason |
| 16:31.00 | sibiria | bastards |
| 16:34.19 | zaf | Samot, 13.24.1 |
| 16:35.37 | Samot | sibiria: What do you mean? |
| 16:35.57 | Samot | zaf: How many agents? When does this start to happen? |
| 16:36.45 | zaf | Samot, i've got the issue on a few different systems, all on same version. it happens randomly 0-4 times per day. 25-65 agents each |
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| 16:37.40 | Samot | What are the resources on the servers? |
| 16:37.58 | sibiria | Samot: the problem wasn't that some feature in asterisk hijacked * # DTMF as i initially thought |
| 16:38.09 | zaf | Samot, 4 xeon cores, 8gb ram |
| 16:38.18 | zaf | sorry, 4gb ram |
| 16:38.40 | Samot | zaf: Are you doing recordings? What is the average call volume? |
| 16:38.42 | zaf | load avg hovers around 0 |
| 16:38.58 | Samot | sibiria: How are they filtering this? |
| 16:39.20 | zaf | no recordings by default, some users may be recording things but we haven't seen a correlation between recording and queues dying |
| 16:39.34 | Samot | Call volume? |
| 16:39.38 | sibiria | Samot: not sure, but "by mistake" is probably the correct answer. it's odd that all three of them are doing it. i suspect it may be some common default as both * and # are commonly used to access special functions, but it's just a theory |
| 16:40.02 | Samot | But if they are "filtering" it how would people be able to use DTMF in your IVRs? |
| 16:40.12 | zaf | maybe 10-20 active calls at a time max |
| 16:40.44 | Samot | Has there been any correlation of the queues locking up with call volume? |
| 16:40.47 | sibiria | Samot: up until now we've only used digits 0-9 so it hasn't been a problem |
| 16:41.02 | sibiria | and we don't offer voicemail features or so |
| 16:41.07 | Samot | sibiria: So people are calling into your PBX and can't use * / #? |
| 16:41.16 | Samot | Over the PSTN? |
| 16:41.25 | sibiria | over the PSTN, both incoming and outbound calls |
| 16:41.38 | Samot | They are not the same. |
| 16:41.42 | zaf | Samot, yes in that it never happens at idle, always when queues are being used. i don't think there is a threshold that triggers it, but some decent amount of usage does seem to be needed for it to happen |
| 16:42.04 | Samot | zaf: Have you rolled back to see if this is a 13.24.1 issue? |
| 16:42.21 | zaf | doing that this very moment |
| 16:42.48 | zaf | i do have a system on 13.20.0 that does not have this issue that is running with a lot of queues |
| 16:43.48 | zaf | on friday i was under the impression that if a single queue would start to fail, that others would be fine, but it turns out that once one queue shows the behavior, the others do very quickly afterwards |
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| 17:28.00 | rrittgarn | having an issue finding the right combo of bridge/user profiles for confbridge. Just want a bridge that beeps when someone enters or leaves. I've set sound_join=beep and sound_leave=beep (on the bridge profile) and quiet=yes on the user, but no love. Pretty sure I'm missing something stupid simple. |
| 17:30.24 | rrittgarn | could anyone point out what I'm missing please and thanks? |
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| 17:32.17 | malcolmd | are you reversed on the use of quiet=yes? |
| 17:33.31 | rrittgarn | possibly? If i turn it off it asks name, and I don't want that particular prompt |
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| 17:47.36 | zaf | @Samot, well, reverting to 13.20.0 didn't solve the issue... |
| 17:49.29 | Samot | So what is different between the working and this? |
| 17:50.12 | zaf | so far i haven't been able to find one |
| 17:51.07 | Samot | So agent count, call volume, system setup all the same? |
| 17:55.42 | zaf | not exactly but as a matter of scale, yes |
| 17:56.03 | zaf | meaning all relatively similarly-sized |
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| 18:12.58 | jeffspeff | i'm running into an odd issue with one of our phones (polycom ip 335) showing a voicemail indication. initially when it was showing this, there was not a mailbox configured for the extension. pruning the realtime peer and rebooting the phone did not resolve the problem. i have created a voicemail box for the extension, reloaded voicemail pruned the peer and rebooted the phone by power-cycling yet the phone is still getting a voicemail notification |
| 18:12.58 | jeffspeff | even though there is nothing in the vmailbox folders. |
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| 19:11.16 | zaf | ok, so one difference i do see is that the systems in question have DPMA v 3.4.10 installed, but now I'm seeing the wiki only shows up to version 3.4.9 @ https://wiki.asterisk.org/wiki/display/DIGIUM/Changes |
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| 19:51.34 | zaf | @malcolmd, is there a reason why the wiki doesn't show dpma v3.4.10 |
| 20:04.20 | malcolmd | the responsible party hasn't gotten around to updating the wiki. the change in 3.4.10 is very minor and only improves things for freepbx |
| 20:08.37 | Kobaz | anyone know of any services like vitelity's old vmobile? |
| 20:08.50 | Kobaz | standard cell calls routed to a sip endpoint? |
| 20:27.47 | Samot | Not really and it was a specific endpoint. |
| 20:33.24 | Kobaz | yes |
| 20:33.29 | Kobaz | a specific endpoint that you pre-set |
| 20:33.41 | Kobaz | so all calls to go pub.lic.ip.here:5060 |
| 20:33.47 | Samot | Or got from them. |
| 20:33.50 | Samot | No |
| 20:34.00 | Kobaz | ? |
| 20:34.11 | Samot | You needed vmobile devices |
| 20:34.14 | Kobaz | right |
| 20:34.26 | Samot | Or sprint devices. |
| 20:34.31 | Kobaz | to send calls to a specific network, correct |
| 20:34.40 | Kobaz | and all calls to/from where via sip |
| 20:34.59 | Kobaz | i know how it worked (in general) |
| 20:35.07 | Kobaz | looking for something to replace it with |
| 20:35.13 | Kobaz | well, something that actually exists |
| 20:35.24 | Samot | I dont think thete ism it wasnt popular. |
| 20:35.35 | Samot | It was a PITA. |
| 20:35.43 | Kobaz | how so? |
| 20:35.43 | Samot | I had test devices. |
| 20:36.11 | Samot | The best feature was rolling between mobile/wifi networks |
| 20:36.39 | Samot | But that required their os on the phone. |
| 20:39.06 | Samot | It just wasn't popular. When Onvoy bought them, they let it ride but did nothing with it. |
| 20:39.31 | Samot | Voyant bought Onvoy and they have started trimming fat of crap that costs more than it makes. |
| 20:40.56 | Samot | Onvoy let Vitelity be Vitelity when they bought them. Voyant isn't doing that so much. They've already cut resale/wholesale programs or modified how they work so they aren't as lenient. |
| 20:44.03 | *** join/#asterisk kessius (c91620a6@gateway/web/freenode/ip.201.22.32.166) |
| 20:44.59 | Kobaz | ah |
| 20:45.47 | Kobaz | voyant seems like, meh |
| 20:45.56 | Kobaz | i tried getting an account directly with them |
| 20:46.01 | Kobaz | their sales team is awful |
| 20:46.08 | Kobaz | they have one job |
| 20:47.51 | Samot | They want volume now. |
| 20:48.11 | Kobaz | right |
| 20:48.13 | Kobaz | i have some volume |
| 20:48.17 | Kobaz | they didn't even get back to me |
| 20:48.24 | Kobaz | i left like 12 voicemails |
| 20:48.29 | Kobaz | and i finally called tech support |
| 20:48.45 | Kobaz | And they answered the phone and i asked uhh, can you send me to someone in sales? Sorry we don't have their phone number |
| 20:49.22 | Samot | Was this through Vitelity channels? |
| 20:49.27 | Kobaz | no |
| 20:49.30 | Kobaz | voyant directly |
| 20:49.45 | Samot | Can I ask what type of volume? |
| 20:50.02 | Kobaz | about 250k a month |
| 20:50.04 | Kobaz | not a huge amount |
| 20:50.27 | Kobaz | but not a tiny amount |
| 20:51.37 | Samot | No, I'm surprised they didn't respond. |
| 20:52.17 | Kobaz | and we're always growing |
| 20:52.32 | Kobaz | that's the thing, you get an account like mine and it's going to be a million a month in the future |
| 20:52.45 | Kobaz | little by little it goes up |
| 20:53.04 | Kobaz | and we're about to get in some large scale texting |
| 20:59.59 | Samot | Makes me wonder what they will do with other things like the vfax. |
| 21:00.16 | Kobaz | we have a number of vfax customers |
| 21:00.25 | Kobaz | voyant apparently has their own vfax platform |
| 21:00.30 | Kobaz | i got that much out of their sales team |
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| 21:06.38 | kessius | people ircs, I newbie in this asterisk, I need one manage web application, which sends and receives sms and logs, Someone knows api-sms-web FREE |
| 21:07.20 | Kobaz | you want to send sms for free? |
| 21:24.24 | kessius | people ircs, I newbie in this asterisk, I need one manage web application, which sends and receives sms and logs, Someone knows AGI-API-AMI-sms-web FREE |
| 21:24.46 | seanbright | you want to send sms for free? |
| 21:26.25 | kessius | seanbright, not, I do send sms for pstn |
| 21:30.43 | kessius | ircs freenode, Someone knows api or agi-ami-api manege sms-web |
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