| 00:00.07 | Samot | With "core set verbose 10" |
| 00:00.09 | warewolf | sec, redacting |
| 00:00.20 | warewolf | oh, hold on then I'll have to redo a call |
| 00:00.32 | warewolf | all I've got is the XML blob from XMPP. |
| 00:01.54 | warewolf | https://gist.github.com/warewolf/0ac389a22aaa170c8867e2c5d9cbe113 is the XML blob, lemme call my google voice number from work, sec |
| 00:06.41 | Samot | Also, I did misspeak. In a sense. |
| 00:07.18 | Samot | You are correct, Google Voice isn't going anywhere. It's still the voice platform for Google. GTalk, however, is dead. |
| 00:07.43 | Samot | GTalk is how the third-party stuff is done, like Motif via the XMPP Federation. |
| 00:14.12 | warewolf | alright, full detail with redacting personal specifics |
| 00:14.13 | warewolf | https://gist.github.com/warewolf/17e89bae2dbbdc0035c8085254b9e376 |
| 00:14.58 | warewolf | that <nick:nick xmlns:nick="http://jabber.org/protocol/nick">Work</nick:nick> there, is the name I have in google for my office desk phone |
| 00:15.10 | Samot | xabean*CLI> -- Executing [s@incoming-gvoice:4] Set("Motif/+14105551212-ea0d", "CALLERID(all)=+14105551212") in new stack |
| 00:15.29 | warewolf | is that overwriting the name field? |
| 00:15.40 | Samot | That is the full callerid |
| 00:15.48 | warewolf | okay. |
| 00:15.50 | Samot | CALLERID(name) is the name only variable. |
| 00:16.04 | Samot | CALLERID(num) or CALLERID(number) is the number only field |
| 00:16.13 | warewolf | lemme switch that over then. |
| 00:16.16 | Samot | CALLERID(all) is the full CallerID header. |
| 00:17.11 | Samot | So a full callerid would be CALLERID(full)=Name <number> |
| 00:17.33 | Samot | er all not full |
| 00:18.25 | warewolf | okay, apparently by default asterisk stuffs (what it thinks is like a SIP endpoint) the <ses:session initiator="this here"> into the name field. |
| 00:18.45 | warewolf | so I get +14105551212@voice.google.com/srvenc-crcMU7nYUdaeiZwWoaOBU2wipUIZqJnM in the name field, which is crazy unweildy. |
| 00:19.03 | warewolf | and the whole point of the "stripcrazy" stuff in the dialplan documented in the asterisk wiki. |
| 00:20.19 | Samot | I really don't know. |
| 00:20.27 | warewolf | okay, thanks. |
| 00:20.34 | Samot | But as I said Motif/Jingle is XMPP Federation |
| 00:20.45 | Samot | It's not longer supported for GTalk. |
| 00:20.57 | warewolf | can you do me a favor? |
| 00:20.58 | Samot | A call is not really a 1-to-1 XMPP chat |
| 00:21.14 | Samot | https://gsuiteupdates.googleblog.com/2017/03/updates-in-g-suite-to-streamline-hangouts-and-gmail.html |
| 00:21.18 | warewolf | if I ask question "X", can you not provide answers to question "Y" ? |
| 00:21.20 | Samot | ^^ I'll put that right here. |
| 00:21.24 | Samot | Dude. |
| 00:21.38 | Samot | You're asking questions about something that is most likely broken. |
| 00:21.39 | warewolf | I'm not asking about google voice. I'm asking about asterisk. |
| 00:21.45 | Samot | It's not sending your callerid name |
| 00:21.53 | Samot | That's not an asterisk issue. |
| 00:22.18 | Samot | It's not sending the callerid name is a format that is recognized. |
| 00:22.20 | warewolf | ok, you're not listening to what I'm saying |
| 00:22.33 | warewolf | the data is in the XMPP stanza sent to asterisk as a client. |
| 00:22.44 | warewolf | https://gist.github.com/warewolf/17e89bae2dbbdc0035c8085254b9e376#file-gistfile1-txt-L13 line 13. |
| 00:23.03 | warewolf | see "Work" there? That's basically the equivlant of the caller ID name field. |
| 00:23.23 | warewolf | I want to figure out what to hack in asterisk to leverage that data from the XML, and stuff it into the caller ID name field. |
| 00:23.44 | warewolf | I appreciate the basic troubleshooting of the caller id all v.s. name v.s. number stuff |
| 00:23.58 | Samot | 8:22:17 PM <Samot> It's not sending the callerid name is a format that is recognized. <-- See there.... |
| 00:24.11 | warewolf | yes. |
| 00:24.12 | Samot | GTalk/GV is not sending the callerid in a format Asterisk is understanding.. |
| 00:24.14 | warewolf | correct. |
| 00:24.15 | Samot | Not XMPP |
| 00:24.23 | Samot | So.. |
| 00:24.36 | Samot | Is this something that just started to happen? |
| 00:24.41 | warewolf | it's in the XML over the XMPP connection |
| 00:24.48 | Samot | I know. |
| 00:25.03 | Samot | It may not understand those attributes. |
| 00:25.06 | warewolf | which is fine! |
| 00:25.12 | warewolf | I want to make it understand those attributes! |
| 00:25.16 | Samot | Did this just start happening? |
| 00:25.18 | warewolf | no. |
| 00:25.23 | Samot | When did it start? |
| 00:25.24 | warewolf | I literally just two days ago got this working |
| 00:25.33 | Samot | OK |
| 00:25.40 | Samot | So now listen to what I am saying. |
| 00:25.41 | warewolf | this is not some magic change on the google side that broke caller id name |
| 00:25.50 | Samot | It could be. |
| 00:26.09 | Samot | My whole point about this no longer being supported. |
| 00:26.12 | Samot | See... |
| 00:26.15 | warewolf | this is obviously something that asterisk doesn't support *as is currently* and I'm trying to figure out how to make it work |
| 00:26.25 | Samot | Listen to me. |
| 00:26.32 | Samot | Since you are new to this.. |
| 00:26.54 | Samot | Motif has, for it's life time, been updated constantly because Google keeps changing stuff. |
| 00:27.08 | Samot | Google makes a change, Motif needs a patch |
| 00:27.18 | Samot | Constant thing. |
| 00:27.32 | Samot | People have had their stuff break with GV constantly. |
| 00:27.40 | Samot | Google has pulled the plug. |
| 00:27.56 | Samot | The way Motif connects/communicates with GV is no longer supported. |
| 00:28.00 | Samot | Eventually will go away. |
| 00:28.08 | Samot | So you *could* hack Asterisk... |
| 00:28.10 | Samot | But why? |
| 00:28.17 | warewolf | it sounds like you're trying to convince me that this isn't worth my time |
| 00:28.25 | Samot | BINGO |
| 00:28.38 | warewolf | (and quite possibly yours, since you chimed in) |
| 00:28.45 | Samot | Perhaps. |
| 00:28.54 | warewolf | alright. |
| 00:28.58 | Samot | Or perhaps this is a break from the drudge of coding I'm doing. |
| 00:29.11 | Samot | And now I have warm fuzzies.. |
| 00:29.25 | warewolf | have another beer. More warm fuzzies. |
| 00:29.33 | Samot | Because I *helped* you understand this is a waste of time. |
| 00:29.39 | Samot | It's why I'm here. |
| 00:29.40 | warewolf | haha |
| 00:29.46 | warewolf | negative, it's not a waste of time. |
| 00:30.05 | warewolf | "this is broken, I'd like to fix it" is how all open source comes into existance. |
| 00:30.12 | Samot | Sometimes "support" is telling someone "No that's dumb. Don't do it." |
| 00:30.20 | warewolf | nah, I get it. |
| 00:30.23 | Samot | And I'd be all over it |
| 00:30.31 | Samot | If Google wasn't going to shut it all down. |
| 00:30.51 | Samot | Having something like that is only as good as to what it connects to |
| 00:31.05 | Samot | When you make it work and Google shuts it down or makes another change... |
| 00:31.09 | Samot | You have a shit burger. |
| 00:31.14 | warewolf | don't care that google is banging the war drums for gtalk/motif/google voice/hangouts/allo/whatever-the-hell-else-comes-next dieing |
| 00:31.35 | Samot | OK. |
| 00:31.45 | warewolf | here, have a hypothetical: lets say you're the primary dev of chan_motif or res_xmpp |
| 00:31.54 | Samot | OK. |
| 00:32.20 | Samot | res_xmpp is not just for Google. |
| 00:32.24 | Samot | It's used for other things. |
| 00:32.32 | warewolf | and you point me "Here, asterisk ver 23.2.1, line 57 has an example of setting channel variables, in res_xmpp.c, and there's already functions elsewhere that extract data from the received XML" |
| 00:32.43 | warewolf | know what I'd do? |
| 00:32.49 | warewolf | I'd go hack it to do what I want. |
| 00:33.22 | Samot | Well..sure. |
| 00:33.24 | warewolf | then I'd go subscribe to asterisk-dev{elopers,patches,users,whatever} and say "I figured out X. I'm not an asterisk expert, but here's the 80% that got me what I desired" |
| 00:33.27 | Samot | If you want to do that go head. |
| 00:33.30 | Samot | But.. |
| 00:33.54 | Samot | As the developer of res_xmpp, in this hypothetical, I work on Asterisk. |
| 00:34.11 | Samot | Which means I follow a set of rules/guidelines. |
| 00:34.31 | Samot | So if it's not on the list, I'm not heart broken because res_xmpp is there for XMPP not GV |
| 00:34.32 | warewolf | you're forgetting that you have the power of saying "Thanks, but we're going to reject your patch." |
| 00:34.46 | warewolf | and that's 100% fine |
| 00:34.49 | Samot | You want to patch it.. |
| 00:34.51 | Samot | Go for it |
| 00:35.15 | Samot | But again.. |
| 00:35.33 | Samot | Making your radio pick up station WXYT doesn't help when WXYT goes off the air. |
| 00:35.45 | warewolf | so what's the rule/guideline where you activly tell someone not to work on something because in your opinion it's not worth the time? |
| 00:36.17 | Samot | Well |
| 00:36.48 | Samot | It all depends on my mood at the time. |
| 00:36.56 | warewolf | gotcha. |
| 00:37.48 | Samot | Just note... |
| 00:38.11 | Samot | That while I have been providing advice/opinions you're making a big deal over.. |
| 00:38.17 | Samot | I still helped you with your issue. |
| 00:38.32 | Samot | I didn't say "It's broken, you're screwed" |
| 00:38.49 | Samot | I helped you understand what the issue is. |
| 00:39.01 | Samot | And then advised that your solution to fix it was bonkers. |
| 00:40.25 | warewolf | one man's trash is another man's treasure |
| 00:40.36 | Samot | Very true. |
| 00:40.47 | warewolf | my fiancee is in hawaii, and I'm in virginia. The only chance I get to hear from her is over the phone. |
| 00:40.57 | Samot | OK. |
| 00:41.01 | warewolf | I previously used a sip trunking provider, and the call qualtiy was so bad my fiancee yelled at me. |
| 00:41.03 | Samot | flowroute.com |
| 00:41.05 | warewolf | yep |
| 00:41.12 | warewolf | I literally just ditched them |
| 00:41.27 | Samot | So.. |
| 00:41.28 | warewolf | so I switched over to xmpp/motif |
| 00:41.33 | Samot | You went to this. |
| 00:41.37 | Samot | OK. |
| 00:41.37 | warewolf | and the call quality went through the roof |
| 00:41.49 | warewolf | I'm more invested than you are. I get that. |
| 00:42.01 | warewolf | but please stop trying to step on someone else's dream? |
| 00:42.05 | Samot | OK |
| 00:42.09 | Samot | Go for it. |
| 00:42.13 | Samot | I never said not. |
| 00:42.17 | Samot | I said "Why would you" |
| 00:42.57 | Samot | If you think this is the best option then you either have to accept the flaws in it. |
| 00:43.06 | Samot | Or you need to invest time into fixing it. |
| 00:43.15 | Samot | Digium doesn't take requests |
| 00:43.28 | Samot | If they are going to make it work, they will |
| 00:43.38 | Samot | If Jingle is third party, it's up that guy. |
| 00:44.16 | Samot | So.. |
| 00:44.26 | Samot | Here's a question for you... |
| 00:44.38 | Samot | Why don't you ship her an ATA or an IP phone |
| 00:44.48 | warewolf | I did! |
| 00:44.52 | Samot | Then she can be registered directly to Asterisk like you |
| 00:44.52 | warewolf | she never plugged it in. |
| 00:45.05 | warewolf | I even bought her a brand new google pixel phone! |
| 00:45.08 | Samot | So the easiest and most practical solution is that |
| 00:45.13 | warewolf | she still uses her samsung galaxy s5. |
| 00:45.25 | Samot | Have her get Bria or something. |
| 00:45.27 | warewolf | (which has battery life problems) |
| 00:45.30 | warewolf | anyhooo |
| 00:45.40 | warewolf | we're back to x/y again |
| 00:46.01 | warewolf | I'm gonna go deep dive through source and try to figure this out. |
| 00:46.17 | Samot | OK. |
| 00:46.31 | warewolf | I do appreciate your time, thank you. |
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| 01:56.43 | k-man | I need to get 2 outdoor rated door phones for my new warehouse, anyone have any experience with any they can recommend? I just want the kind with one button |
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| 02:24.09 | mltngpot | I have a user whos voicemail is showing -1 new messages in the cli |
| 02:25.01 | mltngpot | I am using 1.8.32.2, I have cleared out the /var/lib/asterisk/voicemail/default/xxx directory |
| 02:26.51 | mltngpot | I am new to asterisk and have been unable to figure out how to fix that problem |
| 02:28.41 | Samot | mltngpot: well...this is a bad introduction. |
| 02:29.11 | Samot | 1.8 is dead. But we can give it a shot |
| 02:29.13 | mltngpot | fair enough |
| 02:29.55 | Samot | Show the directiory contents |
| 02:30.01 | mltngpot | it is an inheritance. I am going to upgrade it once I know enough |
| 02:30.09 | Samot | And the cli outout showing the -1 |
| 02:30.24 | Samot | ~pb |
| 02:30.24 | infobot | i heard pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 02:30.49 | Samot | Need 10 mins. Mobile right now |
| 02:31.21 | mltngpot | no worries, you are helping me out of a bind |
| 02:31.59 | Samot | Just post the links..look in a nit |
| 02:32.01 | Samot | Bit |
| 02:32.29 | mltngpot | https://pastebin.ca/3864724 |
| 02:41.50 | Samot | Console output? |
| 02:43.50 | mltngpot | updated |
| 02:44.36 | Samot | Where's the output of Asterisk telling you there is -1 voicemail. |
| 02:44.39 | Samot | I don't see it. |
| 02:45.50 | mltngpot | I just put it on the bottom, Ill put up a new pastbin |
| 02:46.11 | mltngpot | https://pastebin.ca/3864728 |
| 02:49.34 | Samot | Not sure. |
| 02:49.48 | Samot | I recall this being an issue but I can't remember this fix. |
| 02:50.40 | mltngpot | ok, can I create a new voicemail box and have it dump things in there instead of his assigned extension? |
| 02:51.36 | Samot | yes |
| 02:52.02 | Samot | but that's going to require a few changes to the dialplan but that's not too hard. |
| 02:52.36 | mltngpot | so just copy that config line, put in the new mailbox number in voicemail.conf, change the sip.conf to point there |
| 02:53.15 | Samot | extensions.conf |
| 02:53.26 | Samot | You might need to do a little reading up. |
| 02:53.28 | Samot | ~book |
| 02:53.28 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:53.46 | mltngpot | I have a lot of reading up todo |
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| 02:57.44 | mltngpot | thank you so much |
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| 07:15.36 | gavimobile | i have an issue where all of my trunks disconnect and dont reconnect unless i restart asterisk using from the cli "core restart now". i've attached my log which shows normal behavior, however at 13:40:14] something happened where the message log shows something was reset. https://pastebin.com/MEfuWeD6 |
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| 08:49.23 | pawiecki | Hi I have * version 11.20, with default register expiry settings, 'sip show peer A' shows Sess-Expires: 1800 secs / Min-Sess: 90 secs. Phone is Yealink T21P, with timer set to 180 secs. Now this sip peer loses registration randomly, but usually right before it should re-register. Any ideas? |
| 08:51.29 | pawiecki | I have the same problem with Grandstream GXW4232, where only a single sip peer loses registration, and the rest ~30 work just fine. |
| 08:52.07 | pawiecki | Two different devices, the same *, and this weird behaviour. |
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| 08:59.54 | pawiecki | I figured maybe 'ignoreregexpire = yes' might do the trick (testing), but I still don't know what is the cause of this problem. |
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| 10:35.11 | bluez_ | hmm how do you access and set global variables in extensions.lua? |
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| 11:05.38 | cervajs2 | <PROTECTED> |
| 11:06.14 | DivideBy0 | cervajs2: those are devices that were created by or subscribed to via ARI |
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| 11:07.50 | cervajs2 | any tips howto "ARIficate" devices created in i.e. pjsip.conf ? |
| 11:09.16 | file | you might want to explain your goal first |
| 11:09.23 | cervajs2 | my goal is dynamic routing of agents. i need check i agents device is in_use |
| 11:10.30 | file | then you can either do a GET of the specific device to get the device state, or you can subscribe using https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Applications+REST+API#Asterisk13ApplicationsRESTAPI-subscribe to get events as their device state changes |
| 11:10.31 | DivideBy0 | you can subscribe to all events related to device states: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Applications+REST+API#Asterisk13ApplicationsRESTAPI-subscribe |
| 11:10.45 | DivideBy0 | come on file, give a guy a chance to copy and paste! :) |
| 11:10.56 | cervajs2 | using node.js. i will listen for event deviceChange but i need bootstrap when application starts |
| 11:11.02 | file | DivideBy0: the floor is yours then :D |
| 11:11.10 | DivideBy0 | breakdances |
| 11:11.17 | file | fancy |
| 11:11.26 | file | cervajs2: there is no route to get the device state of EVERYTHING on the system |
| 11:12.11 | cervajs2 | file: its missing or its by design? |
| 11:12.26 | file | noone has written such a thing |
| 11:12.50 | file | and there are other queue implementations which work fine without it |
| 11:13.09 | cervajs2 | file: with AMI its possible run https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_DeviceStateList and then catch events right? |
| 11:13.30 | file | yes? if that's what it says |
| 11:16.14 | cervajs2 | file: other method is - somewhere get device names - then run /deviceStates/{deviceName} |
| 11:16.27 | file | yes |
| 11:16.41 | file | you have to know the device names regardless |
| 11:17.02 | cervajs2 | file: last question. if works /deviceStates/{deviceName} why /deviceStates is tied to ARI? |
| 11:17.16 | file | because noone wrote it to return EVERYTHING? |
| 11:17.45 | file | or has contributed a patch to do so |
| 11:17.49 | cervajs2 | file: ok :) thanks for answers |
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| 11:50.06 | infernix | i set up an SPA941 on a remote site with static port forwards and configuring the phone to use the external IP for sip sessions; it's successfully registering and calls work fine to and from asterisk (which is on a public IP) and its public sip trunks |
| 11:50.50 | infernix | however, asterisk keeps audio ip local in a sip show channel X, despite having the same codecs, having nat=no directmedia=yes directrtpsetup=yes |
| 11:51.13 | infernix | every call, whether in or out, is a 'simple_bridge' basic-bridge |
| 11:51.41 | infernix | so wireshark shows the rtp media traffic going from sip phone <> asterisk server <> public sip provider |
| 11:52.21 | infernix | i know it works fine when a call is coming from the sip trunk and is forwarded back to a PSTN number on the same sip trunk (e.g. asterisk gets out of the media path in that case) |
| 11:52.40 | infernix | but for the life of me I cannot get it out of the media path with this linksys |
| 11:53.20 | infernix | codec is g729 on both channels, |
| 11:56.20 | infernix | sip show peer shows DirectMedia : Yes for both the phone and sip trunk |
| 11:56.53 | bluez_ | hmm how do i set a call duration timeout? TIMEOUT(absolute)=5 seems to be from dialing, not from connect |
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| 12:02.52 | bluez_ | nvm got it |
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| 12:16.47 | bluez_ | hmm no way to delete a global variable totally |
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| 12:27.42 | [TK]D-Fender | infernix, Show us the call |
| 12:28.20 | infernix | it might be this: [Sep 7 14:27:02] WARNING[2693][C-00000003]: chan_iax2.c:1239 jb_warning_output: Resyncing the jb. last_delay 0, this delay -2369, threshold 1000, new offset 2369 |
| 12:28.37 | infernix | i don't use iax at all though |
| 12:28.43 | infernix | and i don't think jitterbuffer is enabled by default? |
| 12:30.44 | [TK]D-Fender | infernix, Show us the call |
| 12:31.05 | infernix | redacting the sip debug, hang on |
| 12:32.48 | [TK]D-Fender | First rule do NOT fuc with the evidence |
| 12:33.08 | [TK]D-Fender | I don't care if you PW & PM the link. |
| 12:33.20 | [TK]D-Fender | But do NOT fuck with the evidence |
| 12:34.51 | infernix | just replacing my mobile number |
| 12:35.56 | infernix | https://bpaste.net/show/11c7519556b0 |
| 12:36.21 | infernix | but i may have found it already |
| 12:36.32 | infernix | jbenable=yes somewhere |
| 12:36.46 | infernix | yep |
| 12:36.49 | infernix | that was it |
| 12:37.26 | cervajs2 | file: will you accept minor patch for ast13 pjproject to change PJSIP_MAX_PKT_LEN from 6000 to 32000 as this is in ast15? |
| 12:37.49 | cervajs2 | file: or can you change it on your own? |
| 12:38.24 | file | um, it already was |
| 12:38.27 | file | https://gerrit.asterisk.org/#/c/6034/ |
| 12:40.55 | cervajs2 | file: great, tnx |
| 12:42.15 | infernix | jbenable is global? |
| 12:43.06 | infernix | and why would asterisk choose to create a local bridge with a jitterbuffer if directmedia is possible between peer & user? |
| 12:45.27 | infernix | anyway, i think I can do without |
| 13:04.04 | Samot | Because Asterisk doesn't truly "proxy" the media. |
| 13:04.13 | Samot | It will also interject itself when needed |
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| 14:13.11 | pawiecki | Hi, I've posted earlier, but now more people are online: Hi I have * version 11.20, with default register expiry settings, 'sip show peer A' shows Sess-Expires: 1800 secs / Min-Sess: 90 secs. Phone is Yealink T21P, with timer set to 180 secs. Now this sip peer loses registration randomly, but usually right before it should re-register. Any ideas?I have the same problem with Grandstream GXW4232, where only a single sip peer loses registration, and the r |
| 14:13.13 | pawiecki | est ~30 work just fine. Two different devices, the same *, and this weird behaviour. |
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| 14:37.24 | lagzilla | Is it possible to have AMI generate a test call with audio? |
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| 14:57.02 | Samot | Sure. |
| 14:57.32 | Samot | You can issue an Originate command that will pickup a local channel and issue a Dial() to a destination |
| 14:58.02 | Samot | Be it a local phone/device peer or a trunk peer to another system or PSTN |
| 14:58.33 | Samot | Then you can do a Playback() or something along those lines when the call is answered and bridged. |
| 14:59.14 | Samot | But that's a real call. |
| 14:59.20 | Samot | You can't simulate it. |
| 14:59.43 | sekil | hi |
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| 16:08.09 | jameswf | ahhh the old "channel originate Local/1001 application Playback tt-monkeys" |
| 16:08.19 | jameswf | best thing ever. |
| 16:10.04 | jameswf | This seems like a good ARI example..... monkey roulette. Pull a list of endpoints and randomly dial one every minute or two with screaming monkeys |
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| 16:57.30 | summerholidays | hi all, |
| 16:57.30 | summerholidays | i have 2 asterisks servers communicating on a standard setup, (no private ips, no custom ports, etc |
| 16:57.30 | summerholidays | when i dial a call from AST 1 to AST 2, i get all signalling, except when the phone is answered |
| 16:57.30 | summerholidays | AST 2 starts trying to "retransmit" the connect 200 message .. and i always see RETRANSMITTING.. it never actually arrives at AST1 |
| 16:57.30 | summerholidays | what could be the cause please? |
| 16:57.31 | summerholidays | please take into mind that all other messages are received/sent fine.. just when it gets to answer |
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| 17:14.40 | [TK]D-Fender | SIp/netowrking screwup |
| 17:14.47 | [TK]D-Fender | very clearly |
| 17:14.51 | [TK]D-Fender | things aren't getting answered |
| 17:15.07 | [TK]D-Fender | which is either because they are trying to talk to the wrong place or something isn't making it there |
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| 19:58.31 | tuxd00d | I have one location, using a Ubiquity EdgeMAX router that likes to use the same port numbers for phones. https://pastebin.ca/3864993 (Ext 402 and 405). Protocol (UDP, TCP, TLS) doesnât seem to matter. SIP ALG is disabled. Latest firmware. They never had issue with their Vonage service. Currently registering to Asterisk 12, but will be moving to 14. Has anyone else experience this? Was it fixable? |
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| 21:34.01 | dan_j | tuxd00d: vonage may be using a vpn to get around any issues with blocked/used ports. or it could simply be a config issue within asterisk? |
| 21:34.39 | tuxd00d | I thought they might be as well, but the config on the phones does not have one configured. And the router does not either. |
| 21:37.41 | tuxd00d | On this server, which has been running for 3 years, with hundreds of endpoints, only this location is having this particular issue. So Iâm leaning towards the router as the cause. But Iâm not ruling out an asterisk misconfiguration. Iâve just never seen a router assign two or more devices the same port mapping. |
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| 22:44.28 | nny | having to install freepbx from distro and garbage SangomaOS. Someone kill me |
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| 23:20.20 | summerholidays | getting retransmitted on a connection stage of a call.. no reason why .. any one got any ideas please ? no firewalls involved. |
| 23:20.33 | [TK]D-Fender | SHow us actual debug |
| 23:20.54 | [TK]D-Fender | ~pb |
| 23:20.54 | infobot | i guess pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:20.55 | [TK]D-Fender | ^^^ |
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