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| 11:01.03 | wasanzy | Hello |
| 11:02.08 | wasanzy | I have installed asterisk on ubuntu server but the full log file doesn't seem to be writing transactions. I have this in logger.conf full => notice,warning,error,verbose |
| 11:02.13 | wasanzy | what else can be done? |
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| 11:54.16 | sotoz | What is the difference between a ChannelHangupRequest and a ChannelDestroyed ARI event? When is it safe to wrap up a call in a stasis app that receives these events? |
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| 12:46.04 | wasanzy | hello |
| 12:46.08 | wasanzy | sorry my net went off |
| 12:46.20 | wasanzy | I have installed asterisk on ubuntu server but the full log file doesn't seem to be writing transactions. I have this in logger.conf full => notice,warning,error,verbose |
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| 13:23.55 | jes_ | Good day to all. |
| 13:26.19 | jes_ | I am attempting to transfer a call from one open channel to another. I can successfully transfer by using one direct when I am only dealing with one channel. However, if I attempt a redirect from A to B where both our open then A attempts to reconnect to B and B gives a busy signal. I'd like to connect the two without a bridge so that asterisk is no longer involved in the call. Is this possible? |
| 13:26.31 | jes_ | I am using astersk 14.6 with ARI. |
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| 13:29.48 | jes_ | This way channel is freed |
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| 13:34.25 | wasanzy | I have installed asterisk on ubuntu server but the full log file doesn't seem to be writing transactions. I have this in logger.conf full => notice,warning,error,verbose |
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| 13:46.20 | jes_ | Maybe what you want is higher verbose level. Try full => notice,warning,error,verbose(5) and see if that logs what you want |
| 13:50.27 | wasanzy | ok |
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| 14:03.08 | Samot | 5 is a little high |
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| 14:07.16 | jes_ | That is a good point. Pick a number and walk it up or down to determine what level you need for the information you want logged. If you are attempting to diagnose a problem then always good to turn logging back down after the problem is solved. |
| 14:07.49 | Samot | The left. |
| 14:07.54 | Samot | They left. |
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| 14:08.11 | jes_ | Ah I see. Oh well. Hopefully they realize that bit. |
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| 14:50.59 | mcargile | So I am getting a warning whenever a local channel hangs up from my h extension in Asterisk 13 and I am trying to figure out which function is throwing it. Here is the pastebin |
| 14:51.00 | mcargile | https://pastebin.com/pjXL7WDJ |
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| 14:59.11 | Georbe | Hi. I need a help configuring (fine tuning) a queue in my setup. |
| 14:59.18 | Georbe | I have a queue with 3 Agents with ringall strategy. Let's say 1 caller joins the queue. All extensions of the 3 Agents are ringing. And let's say that the 1st agents answeres the call. |
| 14:59.26 | Georbe | While he is speaking, a second caller joins the queue. Now the extensions of the other two Agents are ringing. And then, the 1st agent completes the call and hungups. |
| 14:59.34 | Georbe | I want to configure the Asterisk so the call from the second caller will ring on all three Agents (As it is configured now, the extension of the 1st Agent doesn't ring.) |
| 14:59.41 | Georbe | Thank you. |
| 15:02.17 | [TK]D-Fender | If it is curently ringing those 2 members and the 3rd guy hangs up his call it will not start calling out to him |
| 15:02.28 | [TK]D-Fender | You need to wait until the next round of distribution |
| 15:03.07 | Georbe | OK. That's how it is currently done . |
| 15:03.18 | Georbe | Is there a way I can fix this? |
| 15:03.47 | [TK]D-Fender | It isn't broken |
| 15:03.49 | [TK]D-Fender | that's how it works |
| 15:04.07 | Georbe | I was thinking to play with pick up groups, but currently I don;t want to do it that way. |
| 15:04.45 | [TK]D-Fender | That doesn't just make the newly freed member get called |
| 15:05.52 | Mr_Pleb_Mgoo | the only way you'll get around it is to not use "skip busy agents" and make sure the phones support call waiting, but then the agents will ring even during calls so... take your pick i guess |
| 15:06.38 | Mr_Pleb_Mgoo | maybe you could experiment with the queue timeouts and retries... |
| 15:07.13 | Mr_Pleb_Mgoo | annnd i thought this was the freepbx channel, whoops. /relevance |
| 15:07.27 | Georbe | That't a good solution Mr Pleb Mgoo (skip busy agents), but with this solution, callers, will never wait in MoH. |
| 15:09.57 | [TK]D-Fender | sure they will |
| 15:10.01 | Georbe | In that way, in stead of having 3 Agents, it will appear as we have 6 Agents |
| 15:15.37 | [TK]D-Fender | And your busy agents phones will constantly be beeping them while they talk to other people |
| 15:15.52 | Georbe | that's right |
| 15:16.22 | [TK]D-Fender | tune your dialing timeout and learn patience |
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| 15:31.19 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 15:50.55 | lvlinux | t38_udptl_nat=no in PJSIP sample file says "Whether NAT support is enabled on UDPTL sessions" What exactly does that do? Is it the same thing as the way Asterisk handles NAT with SIP calls or ?? |
| 15:54.02 | file | it means that outgoing UDPTL packets will go to the learnt source IP address and port |
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| 15:58.42 | DanQuinney | Does anyone know if it's possible to stop * trying to lookup the domain portion of the To header of an INVITE? eg. To: <sip:441202901901@sip.a.e164.org.uk> |
| 16:00.56 | Samot | No. |
| 16:01.06 | Samot | That's the actual destination |
| 16:02.02 | Samot | And Asterisk isn't looking it up |
| 16:02.07 | Samot | That is what is sent to it |
| 16:02.13 | Samot | That is the TO header coming from the provider. |
| 16:02.26 | DanQuinney | Indeed it is Samot |
| 16:02.39 | lvlinux | file: thanks---so it's basically like force_rport for RTP? |
| 16:02.41 | Samot | They are the ones using the FQDN. |
| 16:02.50 | file | no, it's more like rtp_symmetric |
| 16:02.54 | lvlinux | ah k |
| 16:04.27 | DanQuinney | the reason I ask Samot is it opens up a way for users to DOS a * server, too many requests with broken DNS in the To header cause * to hang until the DNS requests timeout. |
| 16:04.53 | mcargile | * isn't looking anything up |
| 16:05.05 | mcargile | that is coming either from your * settings or from your provider |
| 16:05.06 | Samot | Well, not true. |
| 16:05.15 | Samot | It doesn't look up inbound stuff. |
| 16:05.29 | Samot | But it looks up the FQDNs it has in its configures. |
| 16:05.37 | Samot | To make sure they are responding. |
| 16:05.49 | file | unless PJSIP is doing it internally, I don't think our side touches To |
| 16:06.05 | Samot | If there is broken DNS, either it's the DNS being used or the zone for the record being sent. |
| 16:06.16 | mcargile | Right but DNS manager does at a set interval. |
| 16:06.22 | Samot | DanQuinney: Do you have SRV lookups enabled? |
| 16:07.12 | DanQuinney | They're not explicitly disabled |
| 16:07.21 | Samot | It's a setting. |
| 16:07.24 | Samot | It's either on or off |
| 16:07.28 | Samot | By default it's off |
| 16:07.32 | Samot | I believe |
| 16:08.05 | *** part/#asterisk mcargile (~mcargile@rrcs-97-76-33-146.se.biz.rr.com) |
| 16:08.07 | DanQuinney | then they're disabled, we've not enabled them |
| 16:08.28 | Samot | Oh wait, you use PJSIP |
| 16:08.34 | Samot | Which doesn't have this setting. |
| 16:08.47 | Samot | Which means it either isn't doing them or doing it automatically |
| 16:09.14 | Samot | I will say this again. |
| 16:09.17 | DanQuinney | NO |
| 16:09.19 | DanQuinney | don't |
| 16:09.36 | DanQuinney | we're not going around the PJSIP vs chan_sip thing again |
| 16:09.37 | Samot | PJSIP is NOT provider level ready |
| 16:09.47 | Samot | Nope. |
| 16:09.51 | Samot | That's it. |
| 16:09.56 | Samot | You've made your choice. |
| 16:10.49 | DanQuinney | Yes, that's very true, and one I'm going to stick with. |
| 16:12.53 | Samot | Well then, get a provider with working DNS |
| 16:13.04 | Samot | Host sip.a.e164.org.uk not found: 2(SERVFAIL) |
| 16:13.16 | Samot | There's not an A, SRV or NAPTR record for that domain. |
| 16:13.23 | Samot | Completely internal. |
| 16:13.29 | Samot | If it exists. |
| 16:13.31 | Samot | So.. |
| 16:13.31 | DanQuinney | That was actually my typo, it's sip.e.e164.org.uk |
| 16:13.55 | DanQuinney | which, is now fixed |
| 16:14.06 | Samot | Same. |
| 16:14.14 | Samot | I can't pull records for that FQDN |
| 16:14.22 | Samot | Which means I couldn't send traffic to it. |
| 16:16.59 | DanQuinney | Yes, by fixed, I mean their DNS servers now reject all requests to it, previously due to a misconfiguration it was timing out, causing some issues |
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| 17:28.55 | Kobaz | anyone know why a polycom 331 would be randomly sending back 603 decline |
| 17:28.58 | Kobaz | it's not in DND or anything |
| 17:29.39 | Kobaz | file: [12:09:37] <Samot> PJSIP is NOT provider level ready |
| 17:29.42 | Kobaz | file: thoughts? |
| 17:30.13 | file | I've given my thoughts 3 times before. |
| 17:30.35 | Kobaz | scrolls around |
| 17:31.04 | Samot | Provider level is one of those areas I took from file's previous statement of "PJSIP hasn't been proven in all environment so you may be finding stuff new that no one else has" |
| 17:31.26 | Kobaz | i didn't see that one |
| 17:31.36 | Samot | That was like last week when we had this talk |
| 17:31.43 | Kobaz | ah |
| 17:31.54 | file | By sheer fact that it hasn't been around as long as chan_sip it hasn't been exposed to as much |
| 17:31.59 | Kobaz | right |
| 17:32.01 | Kobaz | makes sense |
| 17:32.01 | Samot | ^^^ |
| 17:32.11 | Samot | My whole point the first time around was... |
| 17:32.15 | Kobaz | that i figure, but i was wondering if anyone was seeing glaring issues pointing to DO-NOT-USE (tm( |
| 17:32.28 | Kobaz | that was basically me with asterisk 1.8 |
| 17:32.48 | Kobaz | it was such a major set of changes that basically everything i tried to do that was regular in my day-to-day, broke asterisk |
| 17:32.54 | Samot | End users shouldn't be unwilling beta testers |
| 17:33.22 | Kobaz | this end user is a willing beta tester |
| 17:33.29 | Samot | Thats fine. |
| 17:33.33 | Samot | You've made that choice. |
| 17:33.35 | Samot | Go for it |
| 17:33.43 | Kobaz | speaking of testing |
| 17:33.43 | Samot | But the guy paying me for phone service.. |
| 17:33.50 | Kobaz | right... |
| 17:33.57 | Samot | Isn't paying me to unwillingly be testing PJSIP |
| 17:33.58 | Kobaz | any diea on the random 603 from a polycom? |
| 17:34.06 | Samot | The server is declining it |
| 17:34.20 | Kobaz | the internal server on the polycom, yes |
| 17:34.24 | Kobaz | i gathered that |
| 17:34.42 | Kobaz | i've seen this back in like early firmware 3.x |
| 17:34.57 | Kobaz | if the phone was in the middle of a reg renew, it would 603 new calls |
| 17:35.13 | Kobaz | that got fixed in 3.3 and i haven't seen it since, until now |
| 17:35.13 | Samot | Oh IT is sending 603. |
| 17:35.21 | *** part/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 17:35.21 | Kobaz | yes, polycom -> asterisk... is 603 |
| 17:35.33 | Samot | Wait. |
| 17:35.37 | Samot | Who is sending the 603? |
| 17:35.41 | Kobaz | polycom -> asterisk |
| 17:35.44 | Samot | OK |
| 17:35.52 | Kobaz | asterisk -> polycom (INVITE) polycom->asterisk 603 |
| 17:35.55 | Samot | So Asterisk sends a message to the polycom and it declines it.. |
| 17:36.00 | Kobaz | right |
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| 17:37.19 | Samot | So it's the odd case of a call hitting it during re-REG? |
| 17:37.36 | Kobaz | in this case it doesn't seem so |
| 17:37.44 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 17:37.55 | Samot | Well how often does the polycom send back a 603? |
| 17:37.58 | Kobaz | i was just throwing that in as a historical reference... that was the previous issue in spip 3.1/3.2 days |
| 17:38.06 | Samot | What is the issue now? |
| 17:38.15 | Kobaz | it's like almost all the time now |
| 17:38.24 | Kobaz | let me do some more testing and get some stats |
| 17:38.25 | Samot | Can you replicate and show a debug? |
| 17:38.37 | Samot | Full SIP debug so the full messages are there |
| 17:38.44 | Kobaz | right |
| 17:38.48 | Kobaz | debuggerate |
| 17:39.54 | Kobaz | so there's some voicemail waiting |
| 17:39.57 | Kobaz | do de do |
| 17:44.00 | Kobaz | https://pastebin.ca/3869690 |
| 17:45.26 | Kobaz | 192.168.51.150 is the polycrum |
| 17:45.42 | Kobaz | decline! no soup for you |
| 17:46.51 | Samot | No man. |
| 17:46.57 | Samot | I want to see how it's processed in Asterisk |
| 17:47.12 | Kobaz | why would that matter? |
| 17:47.22 | Samot | Because that's how Asterisk handles it |
| 17:47.29 | Kobaz | right |
| 17:47.32 | Samot | It shows it before it leaves/after it comes in |
| 17:47.50 | Kobaz | it shows what leaving? |
| 17:48.30 | Kobaz | what's the context? |
| 17:51.50 | Samot | So is this an outside call? |
| 17:52.00 | Kobaz | it's via B2BUA |
| 17:52.01 | Samot | Or from another device? |
| 17:52.10 | Samot | It's all via the B2BUA |
| 17:52.20 | Kobaz | but it occurs from either coming in on a trunk, or from another device |
| 17:52.22 | Kobaz | it's all SIP |
| 17:52.24 | Samot | I'm asking was this a call from the PSTN or from another device on the box |
| 17:52.29 | Samot | OK |
| 17:53.37 | Samot | http://knowledgebase-iframe.polycom.com/kb/viewContent.do;jsessionid=CB11A605C32B91B2830AF5D1EEF8A9CA?externalId=15206&sliceId=1 |
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| 17:54.03 | Samot | Are there two lines on this phone? |
| 17:54.07 | Kobaz | yeah |
| 17:54.31 | Kobaz | Okay, i think i see the problem |
| 17:54.33 | Samot | IS 3.3.4 the latest for that phone? |
| 17:54.38 | Samot | Yeah, it's confused. |
| 17:54.45 | Kobaz | this phone is configured for forwarding |
| 17:54.50 | Kobaz | but it's not forwarding |
| 17:54.53 | Samot | It's seizing the wrong line it looks like. |
| 17:55.00 | Kobaz | so it's declining the call instead... is what it looks like |
| 17:55.13 | Samot | It's not even making it to the forward |
| 17:55.17 | Kobaz | it should be sending a 302 redirect |
| 17:55.21 | Samot | The phone never gets that far to process it |
| 17:55.22 | Kobaz | right, yeah the poly isn't sending anything |
| 17:55.23 | Kobaz | right |
| 17:55.30 | Samot | Just like that bug says |
| 17:55.37 | Samot | It's probably seizing the wrong line |
| 17:55.47 | Samot | And not matching |
| 17:56.17 | Kobaz | i can update to 4.x |
| 17:56.26 | Kobaz | i generally dont touch firmware unless there's an issue |
| 17:56.33 | Kobaz | there's a much newer one |
| 17:57.21 | Samot | Well |
| 17:57.30 | Samot | If the 331 can support it the UCS is at 5.6 |
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| 17:58.30 | Samot | I think it should |
| 17:58.44 | Samot | Those things have support until the end of 2019 |
| 17:58.45 | Kobaz | ResolutionUpgrade to UCS 3.3.2 or later. |
| 17:58.51 | Samot | I understand that |
| 17:58.54 | Samot | You're on 3.3.4 |
| 17:58.56 | Kobaz | 3.3.4 meets that critera, just saying |
| 17:58.59 | Kobaz | haha yeah i know, it's old |
| 17:59.09 | Kobaz | i'm upgrading, i'm upgrading |
| 17:59.09 | Samot | Doesn't mean the bug didn't get reintroduced. |
| 17:59.33 | Samot | But I'm not sure how far up the chain 331's go |
| 17:59.58 | Samot | I think they can get on 5 |
| 18:00.19 | Samot | But the SoundPoints are done phones. Sadly. |
| 18:00.33 | Kobaz | yeah |
| 18:00.42 | Kobaz | that's okay,. vvx is pretty nice finally.. with 411's new cpu |
| 18:00.51 | Kobaz | the original 400 were pretty, but laaaaagy |
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