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| 00:41.11 | snadge | ok.. maybe someone can humour me.. the documentation for voicemail.conf says that you can set tz = blah .. where blah is defined in [zonemessages] |
| 00:41.41 | snadge | so i go into voicemail.conf .. look.. and tz is set to "eastern" which corresponds to the zonemessage entry of "America/New_York|'vm-received' Q 'digits/at' IMp" |
| 00:41.56 | snadge | great.. so why are the voicemail messages in utc format |
| 00:42.35 | snadge | never mind the fact that new york time isn't what i want.. im more concerned by the fact that it completely ignores that and is in UTC |
| 00:43.22 | snadge | i wonder if theres a way i can interrogate these values from the asterisk console? |
| 00:43.47 | snadge | will it be in the asterisk db? |
| 00:49.37 | snadge | ive also checked the voicemail context does not override the global timezone setting |
| 00:50.32 | snadge | i don't know why i bother.. i'll just reply to the customer.. the timezone is in UTC, can't be changed.. deal with it |
| 01:26.12 | Samot | Whats the localtime file have in it? |
| 01:27.46 | Samot | For the server. |
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| 01:54.50 | consolejazz | hi. I signed up with a SIP trunk provider today and so far just have it configured with a softphone client on my Android device. By having my own home PBX with asterisk what kind of different in reliability and quality will I see? |
| 01:56.18 | consolejazz | The SIP client on Android is obviously using wifi, so I'm thinking the voice clipping observed by the other party is a result of latency from wifi and also, possibly, because I'm not configured with a home PBX and operating directly with SIP service |
| 01:56.33 | consolejazz | just trying to figure out that second half of that assumption.. |
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| 02:49.28 | snadge | Samot, localtime is correct... i've just noticed it might be a feature of asterisk |
| 02:50.38 | snadge | the notification email comes through with the correct timestamp.. but the raw voicemail message.. eg msg0000.txt contains within origdate=Thu Mar 31 02:24:24 AM UTC 2016 |
| 02:51.31 | snadge | so the actual problem is.. the web interface (user portal) does not respect the timezone setting.. and presumably asterisk stores the raw message in UTC, and then the voicemail application itself compensates for that based upon the global or extension timezone |
| 02:51.52 | snadge | its also proprietary a shit.. and not asterisk's problem ;) |
| 02:51.58 | snadge | a/and |
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| 06:04.42 | EllenorL | night! |
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| 10:04.24 | NirS | morning all |
| 10:04.58 | Chainsaw | Hello NirS. |
| 10:05.31 | NirS | Hi Chainsaw |
| 10:06.08 | NirS | say, are you familiar with an issue with chan_sip in recent versions ? |
| 10:06.25 | NirS | it appears that it locks up completely and not responding to SIP requests |
| 10:06.32 | NirS | happens once every few hours |
| 10:06.43 | Chainsaw | "recent versions" is not very numeric. What branch? |
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| 10:22.37 | catphish | is there a limit on the length of a parameter passed to an app? some of my extremely long dial strings (713 characters) appear truncated |
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| 10:31.23 | Zogot | NirS: i think we're having the same problem also |
| 10:31.46 | NirS | Zogot |
| 10:31.56 | Zogot | what version you on? |
| 10:32.14 | Zogot | we are 13.3.2 |
| 10:32.20 | NirS | 12.8.2 |
| 10:32.23 | NirS | and 13.3.2 |
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| 10:33.00 | Zogot | ours isn't every few hours, but our system isnt in heavy use yet, it happens once a day |
| 10:33.37 | Zogot | where it fails to have any incoming requests, core reload either freezes or does work and doesnt solve the problem |
| 10:33.51 | Zogot | the only solution we've had is to service asterisk restart, nothing special noticed in logs either |
| 10:34.20 | NirS | same here |
| 10:34.22 | NirS | really odd |
| 10:35.05 | Zogot | we are in the process of building the latest version of asterisk, since the centos rpm isn't kept upto date anymore |
| 10:36.16 | NirS | I'm not sure if that's worth while |
| 10:36.22 | NirS | 12.8.2 is the latest 12 release |
| 10:36.35 | NirS | and it's very much inlined with the release of 13 |
| 10:36.48 | NirS | so, I'm not entirely sure what affect that will have |
| 10:36.52 | Zogot | hmm |
| 10:37.04 | NirS | Hmmmm. |
| 10:37.09 | NirS | hold on |
| 10:37.39 | NirS | ok |
| 10:37.51 | NirS | 12.8 is almost a year old |
| 10:37.57 | NirS | scratch my remark |
| 10:37.58 | NirS | althought |
| 10:38.05 | NirS | although, the weird thing is this |
| 10:38.15 | NirS | we have FreePBX systems running with that version, no problems |
| 10:38.35 | NirS | when it works with A2Billing, it causes this issue |
| 10:39.28 | Zogot | might be this NirS? https://issues.asterisk.org/jira/browse/ASTERISK-25468 |
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| 10:41.42 | NirS | hmmmm.... does seem similar in nature |
| 10:41.53 | NirS | but I need to recompile Asterisk to get the "core show locks" command |
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| 11:23.54 | yellowdog | Hello, i have problem with RasPBX. do you know if there are a channel for that? I have problem with chan_dongle solution where asterisk floods me with 500 e-mails. :) |
| 11:26.16 | yellowdog | Title Incoming SMS from +4612345678 Sender asterisk PBX <myemail@gmail.com> Message: 2016-03-31 11:28:46 - dongle0 - +4612345678: - from +4612345678- from +4612345678 |
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| 11:35.16 | catphish | does a GPL G729 ship with asterisk? |
| 11:37.47 | catphish | oh, it doesn't, but others provide such a package |
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| 14:42.26 | zeorin | Hello all, I've got a quick question. I haven't much time... I'm sure this is in the docs somewhere, sorry for not having checked them first. |
| 14:43.23 | zeorin | Is it possible to use have Asterisk answer a call and forward it to another phone number... But the number depends on the region the caller is in? |
| 14:44.28 | zeorin | And then if the number that the call has been forwarded to doesn't answer, could Asterisk try a backup number, and another backup after that if necessary? |
| 14:44.40 | zeorin | And if it runs out of backup numbers, just record a message? |
| 14:45.09 | zeorin | I'm helping out a small non-profit and they're currently paying a fortune to the phone company for only half of the above functionality... |
| 14:45.41 | catphish | zeorin: definitely possible |
| 14:47.06 | zeorin | I assume it'd need to be connected to an ISDN phone line? |
| 14:47.30 | zeorin | Or at least two normal phone lines (one to take the call, another to forward)? |
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| 15:03.40 | Samot | zeorin: Yes it's possible. It's all dialplan. |
| 15:14.24 | yellowdog | where is the eximconfiguration for asterisk? I have problem that exim are sending thousends of email to me :( |
| 15:16.50 | zeorin | catphish & Samot: thanks for your input. I'll feedback to the non-profit shortly. I'm a dev and systems admin myself, so I'm pretty sure that I can figure it out from the docs if they decide to go for it. |
| 15:17.33 | Samot | yellowdog: There is no Exim configuration for Asterisk. |
| 15:17.49 | yellowdog | aha! something raspbx team cooked up? |
| 15:18.05 | Samot | yellowdog: What do you mean? Exim is just an MTA |
| 15:18.09 | catphish | zeorin: to be honest, it sounds like you just need a dialplan with a series of Dial() commands, each with a timeout, followed by a Play() to play a message, very basic setup |
| 15:19.03 | yellowdog | samot: i´m running raspbx and chan_dongle (cellphone to asterisk) are flooding me with email when activating exim with email smarthouse. |
| 15:19.14 | catphish | zeorin: and yes, it'll need at least 2 channels, one for the inbound, and one for outbound, can all be done with SIP though (no physical lines needed), but also possible with ISDN, or normal phone lines (i wouldn't recommend analog lines to anyone) |
| 15:19.21 | yellowdog | i think its trying to send me a sms via email |
| 15:19.26 | yellowdog | Title Incoming SMS from +4612345678 Sender asterisk PBX <myemail@gmail.com> Message: 2016-03-31 11:28:46 - dongle0 - +4612345678: - from +4612345678- from +4612345678 |
| 15:19.43 | Samot | OK, sounds like an issue with chan_dongle. |
| 15:20.33 | Samot | This isn't an MTA issue, Exim or otherwise. |
| 15:20.40 | Samot | It's doing what's it's told to do, send email. |
| 15:20.56 | Samot | It's doing what it's told to do, send email. |
| 15:21.34 | yellowdog | ahh okey thanks |
| 15:21.45 | yellowdog | so it´s chan_dongle who are responsible... |
| 15:21.59 | Samot | Or what every is handling the SMS. |
| 15:22.21 | Samot | Or what ever is handling the SMS. |
| 15:22.25 | yellowdog | yeah |
| 15:25.25 | Samot | zeorin: You're going to need to do a bit of pattern matching as well. |
| 15:25.51 | Samot | And also handle the response from the DIAL() application to roll over to the next number. |
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| 15:27.51 | Samot | zeroin: The other consideration is what will you do with calls that go to voice mail? Because those calls will come back with an ANSWERED status. |
| 15:29.17 | Samot | zeorin: Will you want them to acknowledge the call before it's connected to them? |
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| 15:47.03 | consolej1zz | For single line telephone setup at home looking to install asterisk on raspberry pi 2 and configure it for SIP trunk. Plan to connect an ATA device but not sure which one.. Obi's often come up in my research. There's also Cisco SPA series. Does anyone have experiene with these devices and could help recommend something |
| 15:47.09 | consolej1zz | ? |
| 15:50.08 | [TK]D-Fender | SPA is a good choice |
| 15:50.50 | consolejazz | Okay. Even something like the entry level SPA122? http://amzn.com/B00684H2EY . By the way, not going to be doing any faxing |
| 15:51.19 | [TK]D-Fender | Sure |
| 15:52.07 | [TK]D-Fender | This is just for you to plug your home phone(s) into |
| 15:52.11 | consolejazz | [TK]D-Fender: there's that model, SPA112 ata with router, then there's SPA112 2 port ATA http://amzn.com/B00684PN54 . |
| 15:52.25 | consolejazz | Should I go for the router version? |
| 15:52.40 | [TK]D-Fender | You should really lok at those |
| 15:52.44 | [TK]D-Fender | its the SAME unit |
| 15:53.10 | [TK]D-Fender | The difference is one has a different shitty description than the other :) |
| 15:53.21 | consolejazz | wtf. I swore the tech spec was different. Hm |
| 15:53.50 | [TK]D-Fender | at the PICTURE.... it's the same picture and has the word " router" right there |
| 15:54.11 | [TK]D-Fender | And yes ... cheap Amazon-homed companies with terrible descriptions |
| 15:55.05 | consolejazz | Sure does. Got me thinking there was an error on their part now |
| 15:55.32 | [TK]D-Fender | Use better retailers + MFG site to validate model #'s |
| 15:55.51 | consolejazz | Altex.com ? |
| 15:56.05 | consolejazz | Really need to ween myself off Amazon.. |
| 15:57.33 | consolejazz | data sheet from Cisco site: https://goo.gl/2fFwv7 |
| 15:57.57 | consolejazz | note the rear of the unit. it's the sans router model |
| 15:59.35 | consolejazz | ah, SPA112 is sans router, SPA122 with router |
| 16:00.24 | [TK]D-Fender | Hard to know what to trust from those retailers... |
| 16:00.34 | [TK]D-Fender | Try to buty from somewhere reputable... |
| 16:00.51 | consolejazz | so yea, that retailer goofed. I've heard Altex is good, looking for others |
| 16:00.53 | [TK]D-Fender | Pics show one othing, description another... stuff just broken.... |
| 16:01.16 | [TK]D-Fender | Where are you locatted? |
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| 16:02.24 | consolejazz | Montreal, however, have mail drop box across border |
| 16:03.21 | [TK]D-Fender | How many ports do you need? |
| 16:08.04 | consolejazz | Just one would suit me actually |
| 16:08.47 | [TK]D-Fender | PM <- |
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| 17:45.18 | nunopereira | Hello |
| 17:45.24 | nunopereira | I'm trying to implement a configuration where the outgoing calls should be sent from a fixed IP (say 1.1.1.1), but that IP may not be available when asterisk starts |
| 17:45.30 | nunopereira | I'm binding to address 0.0.0.0. But when the host has the fixed IP present in the machine, the outgoing calls are sent using the primary host (say 1.1.1.2) |
| 17:45.36 | nunopereira | Is it possible to force the outgoing calls to be established from the 1.1.1.1 IP? |
| 17:50.28 | [TK]D-Fender | Bind to itt |
| 17:52.27 | nunopereira | itt? what's that? |
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| 17:57.24 | [TK]D-Fender | it |
| 17:57.33 | nunopereira | I can't |
| 17:57.35 | [TK]D-Fender | You want to force if from that IP? |
| 17:57.37 | [TK]D-Fender | Bind to thatt IP |
| 17:57.56 | nunopereira | but that IP may not be available when asterisk is started |
| 17:59.43 | nunopereira | and it didn't work when the IP was available |
| 18:00.32 | [TK]D-Fender | * isn't going to clean up if your setup changes |
| 18:01.00 | [TK]D-Fender | * can claim the IP in the SIP headers, but the packet will source from the interface your routing table chooses based on the bind. |
| 18:01.49 | nunopereira | so, to force an outgoing address, I have to bind to that IP, right? |
| 18:05.06 | [TK]D-Fender | for the actual source IP on the packet thatt is up to you and your routing table |
| 18:07.32 | nunopereira | what do you mean by that? |
| 18:09.10 | [TK]D-Fender | Your core networking stack choposes what interface to source thigns from |
| 18:09.20 | nunopereira | ok |
| 18:09.20 | [TK]D-Fender | If you use ALL... it will chose for you by whatever means it does so |
| 18:10.01 | nunopereira | it seems that I can't do what I want |
| 18:10.55 | WIMPy | So what exactely do you want? |
| 18:11.47 | nunopereira | force an outgoing calls address that may not be available when asterisk starts |
| 18:12.28 | WIMPy | And what do you expect Asterisk to do during that time? |
| 18:15.31 | nunopereira | when the asterisk host don't have that IP, it doesn't sends calls, and so it's not a problem |
| 18:16.43 | WIMPy | Then you shouldn't start it until the network is up. |
| 18:16.58 | nunopereira | when the IP is available (managed by a corosync/pacamaker cluster), it should start to send calls, with the IP |
| 18:18.08 | nunopereira | but the network is up, and the host has another IP, in the same subnet |
| 18:18.36 | WIMPy | Ok |
| 18:18.45 | WIMPy | Then you shouldn't start it until the network works the way you need it. |
| 18:19.07 | nunopereira | I was trying to have asterisk always up, and just be certain that it's up |
| 18:19.26 | nunopereira | but, as I was saying, asterisk can't do like that |
| 18:20.36 | nunopereira | actually, what you're suggesting is what is implemented |
| 18:21.06 | nunopereira | i.e., pacemaker manages both the IP and the asterisk, and when the IP moves, the asterisk moves too |
| 18:21.35 | nunopereira | but that's 2 resources to manage (IP and asterisk) |
| 18:22.01 | nunopereira | instead of just 1 |
| 18:22.08 | nunopereira | but I have to stick to it |
| 18:31.42 | nunopereira | thank you WIMPy and [TK]D-Fender |
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| 18:43.20 | reign99 | Has anyone had issues getting Polycom phones to use TCP vs UDP by chance? |
| 18:48.14 | fireant327 | just installing the new iso in vmware player now on Ubunto. SEcond attempt it was just staying on a black screen :-| |
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| 18:53.38 | fireant327 | it gets stuck at VMWAre tools installation. ILl start again tomorrow far to many beers last night |
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| 18:59.01 | fireant327 | stuck on SHMZ 6.5 now. |
| 18:59.02 | fireant327 | FFS |
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| 19:10.16 | linux4life | any idea why the message-summary event won't list the actual message summary???? http://pastebin.com/Q9Sq5ePZ |
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| 19:24.22 | fireant_ | Moving my phone system from 3CX to Asterisk :-| |
| 19:24.35 | fireant_ | pretty new to *NIX as well... |
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| 20:02.00 | fireant_ | head spinning |
| 20:02.45 | file | linux4life, "Subscription-State: terminated" |
| 20:07.09 | linux4life | file, yea I noticed that. trying to figure out why now... :-( yea link and asterisk realtime not playing nice.... |
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| 22:06.49 | snadge | we had a couple of our hosted pbxs go to lunch yesterday.. but not all of them, the pattern was kinda random.. had me scratching my head |
| 22:07.36 | snadge | then I found out a major peering link went to lunch at around the same time.. and im theorising that the pbxs that were affected, were ones that had a higher percentage of peers which use the affected route |
| 22:08.29 | snadge | so the symptom was basically that asterisk was running.. but the console was doing absolutely nothing.. this is on a servers with say.. 100 - 200 qualified peers |
| 22:08.56 | snadge | then after a period of time of nothing.. would spew a whole bunch of peer timeout messages |
| 22:09.36 | snadge | i'm of the painful understanding that this is just one of many reasons why chan_sip -> pjsip |
| 22:11.34 | snadge | is there anything besides that, that can be done to mitigate the issue of multiple peers timing out affecting all of them.. turning qualify off? not changing qualify to 5000 from the default of 2000.. finding a new job that isn't voip related? :P |
| 22:12.56 | WIMPy | Te new job thing seems to be the sane option. |
| 22:13.03 | snadge | putting a kamailio proxy in front of it? .. hehe |
| 22:13.18 | WIMPy | Did you stagger the qualifys? |
| 22:13.43 | snadge | im not familiar with any of the qualification options, besides the per extension setting |
| 22:24.51 | wyoung | I am allowed 6 concurrent channels to my SIP provider. How does one play back a message when the limit has been reached when an internal phone attempts to dial to an external line? |
| 22:25.02 | wyoung | call-counter or something? |
| 22:26.12 | WIMPy | Or check what happens if it fails. |
| 22:26.50 | wyoung | yes I could max the channels out I guess |
| 22:27.01 | wyoung | That would cost a $1AUD |
| 22:27.21 | wyoung | per attempt |
| 22:27.34 | wyoung | I think I can afford that :) |
| 22:27.44 | WIMPy | ? |
| 22:27.49 | *** join/#asterisk mafairnet (~mafairnet@187.153.107.35) |
| 22:27.51 | WIMPy | No test numbers? |
| 22:27.58 | wyoung | no idea |
| 22:28.36 | wyoung | I looked on their site with no luck and I tried to explain what I want to do over the phone but the tech supports Indianess got in the way |
| 22:29.19 | mafairnet | hi, does anyone have experienced this message "ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10" when making call with video |
| 22:29.19 | mafairnet | I'm using Asterisk 13 + Chan_SIP + FreePBX + SIPML5 + Chrome. |
| 22:30.06 | WIMPy | There is a chance there's nothing unique happening, of course. |
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| 22:47.18 | wyoung | WIMPy: of corse |
| 22:47.24 | wyoung | course* |
| 22:47.40 | wyoung | now to get SIP_HEADER(To) :/ |
| 22:49.15 | wyoung | s,1,NoOp(SIP_HEADER(To)) just prints SIP_HEADER(To) |
| 22:49.30 | wyoung | I think I am doing it wrong |
| 22:49.50 | wyoung | ${SIP_HEADER(To)} ? |
| 22:50.22 | WIMPy | That looks like something. |
| 22:51.46 | wyoung | exten => s,n,GotoIf($[SIP_HEADER(To) == "XXXXXXXXX"]?XXXXXXXXX) |
| 22:51.48 | wyoung | doesn't seem to work |
| 22:52.18 | WIMPy | Now you're missing the ${} again. |
| 22:52.30 | wyoung | $[] doesn't do that? |
| 22:52.37 | WIMPy | no |
| 22:58.56 | wyoung | ok I got it, apaparantly == is wrong :/ ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting '-' or '!' or '(' or '<token>'; |
| 22:59.32 | WIMPy | Err, right |
| 23:00.38 | wyoung | and SIP_HEADER(To) is set to ""Caller ID"<sip:tonumber@domain.tld>" |
| 23:01.35 | wyoung | I need to use cut? or is there a way to get the localpart ? |
| 23:02.08 | WIMPy | Cut is your friend. |
| 23:02.12 | wyoung | any way to test cut? |
| 23:04.11 | wyoung | Set(TO=${CUT(CUT(CUT(SIP_HEADER(To),@,1),<,2),:,2)}) ? |
| 23:05.07 | wyoung | How can I test that against a pattern? |
| 23:05.33 | WIMPy | toot easy to misread, but doesn't look too bad. |
| 23:05.42 | WIMPy | There's Regex(). |
| 23:05.50 | WIMPy | Maybe you should use an AGI for the test. |
| 23:05.51 | wyoung | :D <3 regex |
| 23:07.51 | wyoung | if I use set in [incoming] context does it stay in this context only or is it global? |
| 23:08.01 | wyoung | any reversed words I should avoid? |
| 23:09.00 | WIMPy | Like what? |
| 23:09.16 | WIMPy | The only reserved context is [defailt] AFIK. |
| 23:09.39 | wyoung | woooo!!!! my cut pattern worked |
| 23:09.52 | WIMPy | \o/ |
| 23:09.53 | wyoung | WIMPy: I mean setting a variable |
| 23:10.10 | wyoung | You deserve a raise WIMPy |
| 23:20.39 | wyoung | WOOOO!!! All working now! I can ring different extensions based on the TO header. |
| 23:21.02 | wyoung | Now to figure out queues :S |
| 23:21.26 | wyoung | It would really be nice if the background music and the anouncement were overlayed |
| 23:22.05 | wyoung | lower background music volume slightly and play accouncement, would be awesome |
| 23:24.26 | WIMPy | Yes. Would be nice. But not happening unless you saved the file that way. |
| 23:31.04 | wyoung | I might write an app to do it |
| 23:31.22 | wyoung | splicing channels is easy right? |
| 23:32.59 | WIMPy | I wouldn't dare to look on to that. |
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