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| 00:21.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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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| 07:43.56 | Guest9232 | Hi, a general question I hope someone can answer. Is the ${CALLERID(num)} plucked from the From header or the Contact header? I believe maybe the PAI is the highest priority? |
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| 08:04.12 | wdoekes | Guest9232: (for chan_sip.so) depends on the trustrpid= setting. it is never taken from the Contact though (I would guess this last statement applies to chan_pjsip.so as well) |
| 08:08.27 | Guest9232 | Thank you wdoekes. The issue I am having is that the From header seem to be rewrited and I have no clue why. This is the sip dialogue https://pastebin.com/kzFZuG0v |
| 08:08.47 | Guest9232 | Call from 0747585000 to 0747840990 which ends up in a queue and it then rings the agent catherine.achola |
| 08:09.10 | Guest9232 | But the from number is being rewritten as 254747840990 |
| 08:09.14 | Guest9232 | When sent to the agent |
| 08:21.32 | wdoekes | first step in debugging would be to check that before calling the Queue() you'd see the expected ${CALLERID(num)} -- add a NoOp and check the Asterisk verbose logs |
| 08:23.04 | wdoekes | if the callerid(num) is already 254747840990, you'd have to work backwards from there (the source device) -- if it's 0747585000, work your way forward and check the config of the queue/agent/destination device |
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| 08:35.26 | Guest9232 | wdoekes: we nooped the callerid(num) as pos 1 in the dialplan |
| 08:35.33 | Guest9232 | And it outputted the 254747840990 as well |
| 08:36.00 | Guest9232 | I do not know how much backwards there is from the top of the dialplan. |
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| 08:55.46 | Rac-on | Guest9232: there is sip-traffic ingress into the asterisk, have a look at that |
| 08:56.46 | Rac-on | otherwise it might be in your peer-definition |
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| 09:23.01 | Guest9201 | I lost connection, just wanted to say that the peer was incorrectly configured. I must have overlooked it |
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| 09:37.32 | Rac-on | Guest9201: that makes sense: '10:56 < Rac-on> otherwise it might be in your peer-definitio' |
| 09:37.35 | Rac-on | ;-) |
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| 09:51.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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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| 11:51.39 | [J]oules | so the question becomes how to tell if an endpoint solicits for mwi |
| 11:53.34 | Samot | You look at the device's settings for MWI. |
| 11:53.47 | Samot | Which you can tell to check or not check for MWI. |
| 11:56.09 | [J]oules | what should it be? |
| 11:56.32 | Samot | First, do you even know the difference of Unsolicited vs Solicited MWI? |
| 11:56.37 | [J]oules | no |
| 11:57.02 | [J]oules | and in 14 there is another new option |
| 11:57.21 | [J]oules | Aggregate MWI |
| 11:57.38 | Samot | So you don't know the difference but you seem to understand what the comment in the ticket means? |
| 11:58.01 | [J]oules | its quite obvious when Richard says to use solicited |
| 11:58.14 | [J]oules | but do i know the diff, no |
| 11:59.08 | Samot | Unsolicited means the endpoint *did not* request the MWI update. |
| 11:59.22 | Samot | Unsolicited = NOTIFY |
| 11:59.23 | [J]oules | ok |
| 11:59.31 | Samot | Solicited = SUBSCRIBE |
| 11:59.44 | [J]oules | let me open gui to 1 phone |
| 12:00.13 | Samot | Aggregate MWI is an Unsolicited option. |
| 12:00.31 | Samot | The phones are "Solicited" |
| 12:00.46 | Samot | It means the device is SUBSCRIBEing and ASKING for the MWI |
| 12:01.11 | Samot | Unsolicited is when a VM is left and Asterisk SENDS a NOTIFY to the device to update MWI |
| 12:01.16 | [J]oules | SUBSCRIBE for MWI on my grandstream = No |
| 12:01.20 | Samot | oK |
| 12:01.30 | Samot | So you're phone is not checking for MWI updates. |
| 12:01.36 | [J]oules | so that sort of goes against what he said |
| 12:01.38 | Samot | But Asterisk is sending them. |
| 12:01.50 | [J]oules | ok |
| 12:01.55 | [J]oules | and that means? |
| 12:02.07 | [J]oules | .. but asterisk is sending them.. meaning? |
| 12:02.20 | Samot | It means, your phone won't get MWI until it's told. |
| 12:02.25 | Samot | Because it's not asking for it |
| 12:02.27 | Samot | Or checking for it. |
| 12:02.36 | [J]oules | ok |
| 12:02.49 | [J]oules | well thats the default in grandstream |
| 12:02.54 | [J]oules | no |
| 12:03.08 | Samot | Then thats dumb. |
| 12:03.13 | [J]oules | so if that is the default, what should the setting be in pjsip advanced |
| 12:03.16 | Samot | That's generally a setting that has to be turned off. |
| 12:03.22 | [J]oules | really? |
| 12:03.24 | [J]oules | hmm |
| 12:03.33 | [J]oules | no, it was like that |
| 12:03.33 | Samot | But I don't use Grandstream. |
| 12:03.39 | Samot | So it just could be a shitty phone |
| 12:03.48 | [J]oules | i just had to factory default it 2 days ago |
| 12:03.57 | [J]oules | yes they are sort of shitty |
| 12:04.09 | [J]oules | i am looking at digium phones |
| 12:04.23 | [J]oules | the disti said they are solid like polycom |
| 12:04.37 | [J]oules | and made by the same manufacturer |
| 12:04.37 | sekil | I have one client having grandstream and only she is complaining about one way audio..grr |
| 12:04.50 | [J]oules | thats nt the phone sekil |
| 12:04.55 | [J]oules | thats the network |
| 12:05.27 | sekil | don't think so |
| 12:05.31 | [J]oules | unless that phone is using ipv6 then its something else |
| 12:05.40 | Samot | Wow. |
| 12:05.41 | [J]oules | since there is no uncle nat with ipv6 |
| 12:05.42 | Samot | Not. |
| 12:05.56 | Samot | No NAT does not = zero audio issues. |
| 12:06.09 | [J]oules | one way audio yes |
| 12:06.12 | Samot | No. |
| 12:06.15 | Samot | It doesn't |
| 12:06.17 | [J]oules | thats what he is saying |
| 12:06.30 | [J]oules | i think thats what he's saying |
| 12:06.35 | Samot | I'm telling you |
| 12:06.35 | Samot | Not him |
| 12:06.36 | [J]oules | or she's saying |
| 12:06.45 | [J]oules | ok |
| 12:06.56 | Samot | 8:05:31 AM <[J]oules> unless that phone is using ipv6 then its something else |
| 12:06.56 | Samot | 8:05:42 AM <[J]oules> since there is no uncle nat with ipv6 |
| 12:07.13 | [J]oules | so lets go back to issue at hand |
| 12:07.14 | Samot | ^^^ That. Just because there is no NAT doesn't mean it makes audio issues go away. |
| 12:07.19 | sekil | true |
| 12:07.59 | [J]oules | ok Samot , tell us what causes the one way audio issue |
| 12:08.09 | sekil | come to think of it..I'm the culprit...I said I wouldn't support grandstreams...and I'm doing it.. |
| 12:08.13 | Samot | sekil: Is the phone remote to the PBX? |
| 12:08.18 | sekil | no |
| 12:08.26 | Samot | So it's on the same local network? |
| 12:08.40 | sekil | ocassional one way audio when calling some ata that's connecting panasonic |
| 12:08.47 | [J]oules | that for sure takes uncle nat out of the equation |
| 12:08.54 | sekil | and no issues with that ata or other ip phones.. |
| 12:09.18 | Samot | So, the people having one way audio issues are on analog devices that connect to an ATA? |
| 12:09.31 | Samot | Or is this a Panasonic PBX? |
| 12:10.00 | sekil | ata and gs in same network...panasonic connected to ata fxs |
| 12:10.24 | Samot | Is it a Panasonic phone or a Panasonic PBX? |
| 12:10.30 | sekil | pbx |
| 12:10.30 | Samot | It makes a difference. |
| 12:10.33 | Samot | OK |
| 12:10.45 | Samot | So if you unplug that PBX and put a phone in that ATA, zero issues? |
| 12:11.15 | sekil | I have no idea...all I know..when I call from PSTN to that ATA (to the receptionist analogue phone) no issues afaik |
| 12:11.36 | sekil | also I didn't receive complains from other ppl having ip phones (not gs) |
| 12:11.41 | Samot | So the audio issues only exist on the PBX? |
| 12:11.47 | Samot | The Panasonic PBX? |
| 12:12.09 | sekil | well..hard to tell..when I'm calling from pstn to that ata/pbx ...I didn't have issues |
| 12:12.24 | sekil | only this one calling from inside from gs |
| 12:12.31 | Samot | But when you call from the Grandstream to the PBX via the ATA? |
| 12:12.37 | sekil | yeah |
| 12:12.50 | Samot | OK |
| 12:12.54 | sekil | she says they can't her unless she dials more than once.. |
| 12:12.56 | Samot | So my question still stands... |
| 12:12.57 | sekil | for some reason.. |
| 12:13.03 | sekil | hear her |
| 12:13.12 | Samot | 8:10:45 AM <Samot> So if you unplug that PBX and put a phone in that ATA, zero issues? |
| 12:13.31 | Samot | Does the GS have issues when the device is just a old analog phone vs the PBX? |
| 12:13.32 | sekil | Fixed There was one-way audio after hold/unhold call when there is incoming call in outgoing call status |
| 12:13.41 | Samot | OK |
| 12:13.42 | sekil | Samot: I heard you ..and I don't know.. |
| 12:13.45 | sekil | will test it out.. |
| 12:13.47 | Samot | You have an SDP issue. |
| 12:13.47 | sekil | makes sense |
| 12:13.59 | sekil | this message is from some newer fw |
| 12:14.07 | Samot | If putting the call on hold and taking it off hold fixes the audio issue.. |
| 12:14.07 | sekil | other than one which is currently on the phone |
| 12:14.13 | Samot | You have an SDP issue in the call setup. |
| 12:14.34 | sekil | Samot: yeah..I'm not sure if that's the case...just looking at the release notes of newer fw |
| 12:14.43 | Samot | Meaning that the IP set for SDP/media is wrong |
| 12:14.49 | Samot | OR SDP flow is wrong |
| 12:14.51 | sekil | Samot: probably yeah |
| 12:15.02 | Samot | Going on Hold/off Hold is a new INVITE |
| 12:15.06 | Samot | New SDP |
| 12:15.06 | sekil | Samot: means mostly that I'm stupid... |
| 12:15.12 | sekil | Samot: for breaking my own rules |
| 12:15.22 | Samot | Which then corrects to the proper IP and boom you have two way audio. |
| 12:16.01 | Samot | So you need to run debugs on the Asterisk server for when the GS calls the PBX |
| 12:16.13 | Samot | See what the SDP is using for an IP |
| 12:16.21 | Samot | Replicate the problem. |
| 12:16.43 | Samot | Watch the debug so you can see the INVITES and the changes in the SDP when the hold happens. |
| 12:17.30 | Samot | 8:08:00 AM <[J]oules> ok Samot , tell us what causes the one way audio issue |
| 12:18.00 | Samot | ^^ I just did. It's an SDP issue during the initial INVITE. Nothing to do with IPv6/IPv4 or NAT. |
| 12:18.01 | sekil | Samot: yeah |
| 12:18.06 | [J]oules | so the one line for the problem is ip address? |
| 12:18.07 | Samot | Outside of what could be in the phone settings. |
| 12:18.13 | sekil | Samot: actually just did a test call from pstn |
| 12:18.16 | sekil | Samot: no issues |
| 12:18.28 | Samot | This is an issue with the GS making the call |
| 12:18.31 | Samot | Not the PBX |
| 12:18.32 | sekil | Samot: so it's probably phone related only |
| 12:18.34 | sekil | Samot: yep |
| 12:18.46 | sekil | Samot: I just want to isolate first.. |
| 12:18.56 | Samot | Follow what I gave you |
| 12:19.05 | [J]oules | sekil: i take it you're not too happy with grandstream |
| 12:19.17 | Samot | Either "sip set debug on" or "pjsip set logger on" to watch the SIP debg. |
| 12:19.24 | sekil | [J]oules: I heard terrible stories about it |
| 12:20.04 | sekil | when I did this PBX migration...they had this 2160 |
| 12:20.11 | sekil | so I said what a hell ..should work.. |
| 12:20.19 | sekil | was I wrong |
| 12:21.12 | [J]oules | yesterday a school we provide telephone services for brought down a gs 1615 which would freeze right after booting up. Its like 7 months old. GS issued an rma and we replaced the phone a few hours later. But who looses? Us, we can't charge the client for this. |
| 12:21.42 | *** join/#asterisk gruetzkopf (gruetzkopf@bnc.dont-follow.us) |
| 12:22.06 | [J]oules | so Samot are you saying SUBSCRIBE for MWI on a phone s/b Yes ? |
| 12:22.23 | sekil | [J]oules: do you want to have subscription? |
| 12:22.40 | [J]oules | this sounds dumb, but I have no real idea |
| 12:22.59 | sekil | [J]oules: basically pbx like software (* or FS) does NOTIFY's on its own by default |
| 12:23.04 | [J]oules | my gs1630 was set to no |
| 12:23.15 | sekil | [J]oules: when having proxy in the middle you should subscribe |
| 12:23.37 | [J]oules | proxy in the middle? |
| 12:23.44 | sekil | yes |
| 12:23.45 | Samot | JFC. |
| 12:23.48 | Samot | Dude it's simple |
| 12:23.51 | Samot | I explained it. |
| 12:24.02 | Samot | Either the phone will check, if told, for MWI |
| 12:24.16 | Samot | It will SUBSCRIBE to the voicemail for updates. |
| 12:24.59 | Samot | The VOICEMAIL SYSTEM can also send NOTIFYs when there isn't an active subscription so the phone knows it has MWI. |
| 12:25.15 | [J]oules | ok well i am still a bit lost here. We've been having problems with pjsip dropping registrations. Created an issue with digium/asterisk. The fellow replied back last night saying we should have mwi set to solicited on pbx. |
| 12:25.51 | [J]oules | so if that is set to yes in pbx , then i take it SUBSCRIBE for MWI s/b Yes on sip phones |
| 12:26.02 | Samot | In short, only have Asterisk send MWI NOTIFYs if the endpoint SUBSCRIBEs |
| 12:26.25 | [J]oules | and what setting is that Samot ? |
| 12:26.34 | Samot | Well that's the rub. |
| 12:26.42 | Samot | Digium is all about Asterisk. |
| 12:26.42 | [J]oules | huh?: |
| 12:26.57 | Samot | You have FreePBX which means those settings are controlled by FreePBX. |
| 12:27.04 | [J]oules | right |
| 12:27.05 | Samot | Therefore, you have no real control over them. |
| 12:27.14 | Samot | Which, to Digium, isn't their issue. |
| 12:27.20 | [J]oules | oh so there is no setting? |
| 12:27.30 | [J]oules | that doesnt make sense |
| 12:27.43 | Samot | There are ways to set up PJSIP endpoints to get unsolicited MWI |
| 12:27.46 | [J]oules | fpbx as auto/unsolicited/solicited |
| 12:27.55 | Samot | Most PBXes are. |
| 12:28.03 | Samot | Or voicemail systems. |
| 12:28.10 | Samot | How would you know you had new VM? |
| 12:28.21 | Samot | If you weren't subscribing and nothing was being pushed? |
| 12:28.25 | Samot | You'd never have MWI |
| 12:28.37 | [J]oules | i get a blinking red led on phone |
| 12:28.47 | [J]oules | or a number in bria at bottom left |
| 12:28.48 | Samot | BECAUSE THE PBX IS SENDING IT |
| 12:28.53 | [J]oules | yes |
| 12:28.55 | [J]oules | correct |
| 12:28.58 | [J]oules | agreed |
| 12:28.58 | Samot | Even though the phone ISN'T ASKING |
| 12:29.05 | [J]oules | right |
| 12:29.20 | Samot | What that ticket reply is stating is "DON'T SEND IF NOT ASKED FOR" |
| 12:29.24 | [J]oules | and with that logic what should the setting be in phone |
| 12:29.30 | Samot | Which means, change the default behavior. |
| 12:29.41 | [J]oules | default of? |
| 12:29.49 | Samot | Oh FFS. |
| 12:29.58 | Samot | Asterisk. |
| 12:30.05 | [J]oules | well you're not being clear |
| 12:30.12 | [J]oules | what setting???? |
| 12:30.17 | Samot | GO READ |
| 12:30.23 | [J]oules | solicited ? |
| 12:30.28 | Samot | This is covered in the wiki. |
| 12:30.41 | [J]oules | what wiki.. fpbx? |
| 12:30.48 | Samot | JFC. |
| 12:30.54 | Samot | Who did you open the issue with? |
| 12:30.58 | Samot | Digium, right? |
| 12:31.01 | [J]oules | yes |
| 12:31.02 | Samot | Who does what? |
| 12:31.10 | Samot | Asterisk. |
| 12:31.11 | [J]oules | but I dont use straight asterisk |
| 12:31.17 | [J]oules | as you are well aware |
| 12:31.20 | Samot | That's not their concern. |
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| 12:31.25 | [J]oules | right |
| 12:31.31 | [J]oules | its my concern |
| 12:31.45 | Samot | 8:26:35 AM <Samot> Well that's the rub. |
| 12:31.46 | Samot | 8:26:42 AM <Samot> Digium is all about Asterisk. |
| 12:31.46 | Samot | 8:26:43 AM <[J]oules> huh?: |
| 12:31.46 | Samot | 8:26:57 AM <Samot> You have FreePBX which means those settings are controlled by FreePBX. |
| 12:32.04 | Samot | Digium told you what needs to be done. |
| 12:32.09 | [J]oules | yes |
| 12:32.14 | [J]oules | change to solicited |
| 12:32.20 | Samot | However, you can't easily do it because you don't have control over the settings that are being used. |
| 12:32.29 | [J]oules | what?? |
| 12:32.36 | [J]oules | of course you can |
| 12:32.41 | Samot | FreePBX generates those files. |
| 12:32.45 | Samot | And those settings. |
| 12:32.50 | [J]oules | in pjsip ext. /advanced |
| 12:32.57 | Samot | What setting? |
| 12:33.10 | [J]oules | do you have any pjsip ext's Samot ? |
| 12:33.29 | [J]oules | yes or no |
| 12:33.38 | Samot | I'm making sure you actually looked. |
| 12:34.08 | Samot | Because the only MWI setting you've mentioned is "Aggregated" which would be wrong. |
| 12:35.10 | [J]oules | https://usercontent.irccloud-cdn.com/file/0MNHjjUq/mwi.png |
| 12:35.18 | Samot | Good. |
| 12:35.32 | Samot | So now if you change that to Solicited, you need to update your phone. |
| 12:35.42 | Samot | Otherwise you will no longer receive MWI on it. |
| 12:35.46 | [J]oules | so the default needs to be solicited. find. What about aggregate MWI? |
| 12:36.01 | [J]oules | s/find/fine |
| 12:36.10 | Samot | I'm not going to tell you what you can read in the tooltips. |
| 12:36.23 | Samot | That's what those little question marks are for. |
| 12:36.56 | [J]oules | brb time to feed the birds |
| 12:45.00 | [J]oules | unless i am blind, there doesn't appear to be an MWI setting in vvx300 |
| 12:45.37 | [TK]D-Fender | There is a long chain of precedent for "blind" |
| 12:46.13 | [TK]D-Fender | So that's always a fair bet |
| 12:52.10 | [J]oules | message center/Subscription Address |
| 12:52.26 | [J]oules | thats it? |
| 12:54.58 | Samot | I guess you should read their docs or at least to help tips in the GUI. |
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| 14:46.48 | friend18 | Hello guys, I would know if is possible to configure using XML provisioning D65 Digium phone in a way I can set the label and function for the four programmable keys |
| 14:47.35 | friend18 | I was able to configure just the standar options, but I need to use two keys, one for park and one for unpark calls, and both labels in not english language and not same "transfer" label |
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| 19:49.25 | [J]oules | Samot: i did read the help guide on right side, but what server are they talking about? message center server |
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