| 00:19.20 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:19.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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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| 00:51.22 | drmessano | Nobody is packaging up to the second releases for any distro I am aware of |
| 00:51.29 | drmessano | That isn't FreePBX, of course |
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| 02:42.35 | igcewieling | Asterisk is one of the very few things I build from source. |
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| 05:55.39 | OS-24589 | hey everyone - I have a situation here - an old install of asterisk (Elastix 2.x something on Centos 5.11) The admin tried to update all of the elastix modules and wound up in a broke state basically what I can discern is that dahdi module is not loading (and not able to load ' module chan_dahdi.so could not be loaded') |
| 05:56.12 | OS-24589 | any help to force load? CentOS 5.11 is deprecated for years now, so no repos to update from |
| 05:56.55 | Samot | Elastix is dead. |
| 05:57.23 | OS-24589 | yeah - I don't care about keeping elastix working - just getting phones back before the clients get here in the morning |
| 05:57.36 | Samot | Start installing a new box. |
| 05:57.41 | OS-24589 | ./sigh |
| 05:57.58 | Samot | You can either try to fix the broken box... |
| 05:58.05 | Samot | Or you'll have to replace it. |
| 05:58.28 | Samot | I'd go for the latter because if you're using Elastix 2.x on CentOS 5...everything is outdated. |
| 05:58.52 | OS-24589 | replace is the path - we are already on that but looking for a bandaid to get phones back for the immediate need (tomorrow) while hardware is on the way |
| 05:59.04 | OS-24589 | The uptime on this box is over 3 years |
| 05:59.22 | Samot | There is no bandaid. |
| 05:59.27 | Maliuta_ | OS-24589: extract the dialplans and move to something new |
| 05:59.53 | Maliuta_ | you can't triage something that broken, you go straight to major surgery |
| 06:00.16 | OS-24589 | yup - backed it all up already...was hoping for a quick hitter...thanks! |
| 06:00.30 | Samot | Well... |
| 06:00.35 | Samot | How did you back it up? |
| 06:00.54 | Samot | Because the "Backup/Restore" option in Elastix (really FreePBX at this point) will not work. |
| 06:01.06 | Maliuta_ | if you have a full OS backup you could restore to a known state |
| 06:01.28 | Maliuta_ | but seriously it's still probably not worth the pain |
| 06:01.30 | OS-24589 | no, just asterisk backups of the dialplan, configs, voicemail, tftp etc |
| 06:01.44 | Samot | How did you back them up?! |
| 06:01.58 | OS-24589 | cron job - I setup years ago |
| 06:02.16 | Maliuta_ | sounds faster to install a new box and adapt the configs/dialplan to new * version |
| 06:02.27 | OS-24589 | yep...off to do that... |
| 06:02.45 | Maliuta_ | even on my crappy home DSL connection a fresh minimal install is done in less than an hour |
| 06:03.01 | OS-24589 | any recommendations for underlying OS? I'm more partial to deb than rpm |
| 06:03.13 | Samot | What are you going to do? |
| 06:03.26 | OS-24589 | build clean and adapt |
| 06:03.31 | Samot | Are you going to replace Elastix with FreePBX or a clean install of Asterisk? |
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| 06:03.56 | OS-24589 | probably freepbx |
| 06:04.06 | Samot | OK, so then the Distro. |
| 06:04.11 | Samot | Which will install it's own OS. |
| 06:04.41 | Samot | What was this box running, Asterisk wise, before it died? |
| 06:05.03 | OS-24589 | 11.25 I think... |
| 06:05.35 | Samot | Well that's a silver lining at least. |
| 06:05.40 | OS-24589 | yup 11.25 |
| 06:05.58 | OS-24589 | well...I dunno - thats what I see...don't know what it was before he started this mess |
| 06:06.14 | Samot | OK.. |
| 06:06.54 | lorsungcu | this is sort of what the migration tool was made for |
| 06:07.14 | OS-24589 | :) |
| 06:07.26 | Samot | Yeah. But he has broken stuff |
| 06:07.40 | Samot | I dont know how that will impact. |
| 06:07.49 | lorsungcu | i dont think it wil |
| 06:07.55 | lorsungcu | it mostly/only moves dialplan |
| 06:08.00 | OS-24589 | This is an old client of mine, I set it up years ago, handed it off 2 years ago and the new admin doesn't know/understand/care linux/asterisk/non M$ anything |
| 06:08.35 | Samot | How many users on the box? |
| 06:08.59 | OS-24589 | so....I have no idea how broken in what states for how long.... |
| 06:09.26 | OS-24589 | 12 desks , 3 conference rooms and a couple of conference bridges... only a couple of ring groups |
| 06:09.38 | lorsungcu | you could recreate that in 30 minutes |
| 06:09.51 | OS-24589 | no IVR tree... yeah I'm sucking down the distro now |
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| 06:11.36 | Maliuta_ | OS-24589: you need to pick better people to hand stuff off to :P |
| 06:11.56 | OS-24589 | lol yeah - well, an old client is new again :D |
| 06:12.04 | Maliuta_ | I handed off one client when I got sick a while back - now that guy won't give them back |
| 06:12.22 | OS-24589 | Well, these guys called me out of semi-retirement to take it back |
| 06:12.27 | Maliuta_ | I don't begrudge him that, they are good to do stuff for |
| 06:16.01 | OS-24589 | ok - off to rebuild a box now - thanks all |
| 06:17.15 | Samot | Another case of an ignored box coming back to bite someone in the ass. |
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| 09:12.05 | blah_321 | hey, is this a support chat ? |
| 09:14.42 | aiksa[LV] | kind of. - users helping users |
| 09:14.56 | blah_321 | nice thanks for replay me :) |
| 09:15.28 | aiksa[LV] | but no guranatees or anything |
| 09:15.50 | aiksa[LV] | consider this a real-time version of stackexchange |
| 09:17.04 | blah_321 | it OK i'm always expecting the worst and hoping for the best |
| 09:24.24 | blah_321 | i got 2 elastix voip connected with sip truck, i have a remote ext connected to one of the elastix i can call from the remote ext to local ext, from local ext to remote ext, from remote ext to local home phone (example 03-9453-825) all work greate, but when i try to call from remote ext to cellular (mobile phone 052-4531234) the call connected i can hear but cannot hear me - one way audio |
| 09:26.13 | blah_321 | after 30 sec the call hangup |
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| 09:34.53 | aiksa[LV] | so probably that call goes through some kind of trunk? |
| 09:35.25 | aiksa[LV] | i would check 2 things: that trunk configuration; and externip and localnet settings in sip.conf |
| 09:35.34 | aiksa[LV] | and NAT rules in router |
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| 10:57.32 | blah_321 | aiksa[LV]: in sip.conf i need to speicfy remote ext public ip ? |
| 11:00.58 | aiksa[LV] | you are using chan_sip rather than pjsip, right? |
| 11:01.15 | aiksa[LV] | in that case - yes it might help a lot |
| 11:01.40 | aiksa[LV] | SIP normally indicates the path back to itself in SIP message |
| 11:02.11 | aiksa[LV] | if you have not specified external_ip, it will indicate the ip you have on your box in internal network |
| 11:02.26 | aiksa[LV] | and your trunk provider will not know where to send the traffic back. |
| 11:03.08 | aiksa[LV] | this is on thing: another thing is that it will use that to indicate proper return path for media (RTP traffic) |
| 11:03.29 | aiksa[LV] | but your router should also know how to handle this |
| 11:04.01 | aiksa[LV] | that is - allow UDP packets carrying RTP traffic to get back to your PBX |
| 11:04.15 | aiksa[LV] | CLI > core rtp set debug on |
| 11:04.28 | aiksa[LV] | helps a lot to see if you are getting traffic back. |
| 11:05.18 | aiksa[LV] | https://www.voip-info.org/wiki/view/NAT+and+VOIP |
| 11:05.34 | aiksa[LV] | quite old, but still relevant |
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| 14:24.03 | EvilCoder | Hello people |
| 14:24.11 | EvilCoder | I got a question regarding IAX |
| 14:24.21 | EvilCoder | Or well, not a question. But a bug |
| 14:25.18 | EvilCoder | I'm trying to attach/tag an accountcode on incoming IAX traffic, but when I do it for more then 1 iax connections. They all just grep the first accountcode, which is the first accountcode found... So other locations are having the same accountcode.. |
| 14:25.41 | EvilCoder | Anybody has a clue? |
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| 14:33.54 | *** join/#asterisk EvilCoder (~mvdworp@firewall.office.utelisys.net) |
| 14:34.03 | EvilCoder | ? |
| 14:34.07 | EvilCoder | Nobody has a clue? |
| 14:39.24 | Samot | So you have multiple IAX trunks.. |
| 14:39.35 | Samot | And each trunk has it's own accountcode? |
| 14:45.13 | igcewieling | EvilCoder: set up a dummy trunk as the last in iax.conf and see if it always uses the account code of the last peer in iax.conf. I've seen a bug where all incoming iax connections were always matching the last peer. |
| 14:45.42 | igcewieling | Oh, and almost nobody uses IAX anymore so not a lot of people can help with it. |
| 14:46.01 | EvilCoder | Ah, is there an replacement for the iax? |
| 14:46.23 | EvilCoder | If so, are there improvements in the new way of connecting boxes together? |
| 14:46.23 | igcewieling | It is called "SIP" |
| 14:46.26 | EvilCoder | uh |
| 14:46.31 | EvilCoder | IAX is an connection between servers |
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| 14:47.14 | EvilCoder | I've never used sip between internal servers, but hmkay. |
| 14:47.37 | igcewieling | The IAXy would disagree with "iax is for servers" thing. |
| 14:49.12 | igcewieling | EvilCoder: Just about everyone uses sip between Asterisk servers. IAX was built to solve 2 issues that I know of. 1) reduce overhead by trunking and 2) nat issues. "1" doesn't usually matter anymore and "2" was solved for most situations long ago. |
| 14:49.44 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:fc00:983e:f8fb:c5cb) |
| 14:49.50 | EvilCoder | Hmm, clear. So time to migrate to SIP then eventually. |
| 14:49.56 | Samot | Wait. |
| 14:50.05 | Samot | And each trunk has it's own accountcode? |
| 14:50.16 | EvilCoder | Yes, but I think I found out why that did not work. |
| 14:50.24 | EvilCoder | Thats not an option on a iax tunnel |
| 14:50.36 | EvilCoder | So i'll just put in on a context, and set the accountcode there. |
| 14:51.02 | EvilCoder | Or atleast whenever I view the iax tunnel from the asterisk CLI it does not even bother to show up |
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| 14:51.25 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:52.31 | EvilCoder | Uh, the same happens with contexts, like wth |
| 14:52.36 | Samot | accountcode= is a viable IAX trunk setting. |
| 14:52.46 | Samot | EvilCoder: Show an actual call. |
| 14:53.29 | EvilCoder | And on that ip, there is an accountcode set to =ramada |
| 14:53.32 | EvilCoder | and |
| 14:53.34 | EvilCoder | context =ramada |
| 14:53.40 | EvilCoder | But it goes to context [casa] |
| 14:53.41 | EvilCoder | x) |
| 14:53.54 | Samot | Do you have a trunk that goes to [casa]? |
| 14:53.57 | EvilCoder | yes |
| 14:54.02 | Samot | OK |
| 14:54.04 | EvilCoder | The first one (on top of the .conf doc) |
| 14:54.08 | Samot | So you're trunks are matching. |
| 14:54.23 | EvilCoder | Nope, different ips, different users, and different contexts |
| 14:54.23 | Samot | You're sending a call fro PBX A to PBX B |
| 14:54.30 | Samot | On PBX B you have multiple IAX peers |
| 14:54.37 | Samot | All expecting calls from PBX A |
| 14:54.41 | EvilCoder | Well, actually that is point A, but yes |
| 14:55.03 | Samot | So each trunk between PBX A and PBX B is on their own IPs? |
| 14:55.08 | EvilCoder | Yes |
| 14:55.17 | Samot | You need to show an actual call |
| 14:55.23 | Samot | So we can see how this is matching |
| 14:55.27 | Samot | asterisk -rvvvvvvvv |
| 14:55.30 | EvilCoder | The only thing which matches, is on both B sides (so the second boxes, have the same uplink name) - but this does not matter, already tested ths |
| 14:55.32 | Samot | iax set debug on |
| 14:55.33 | EvilCoder | The only thing which matches, is on both B sides (so the second boxes, have the same uplink name) - but this does not matter, already tested this |
| 14:55.34 | igcewieling | EvilCoder: Is the last peer setup for context casa> |
| 14:55.43 | EvilCoder | No |
| 14:55.51 | EvilCoder | It greps the first accountcode it see's |
| 14:55.54 | EvilCoder | context* |
| 14:55.58 | Samot | Show the configs |
| 14:55.59 | igcewieling | cool! you're not running into that issue. |
| 14:55.59 | Samot | Show a call |
| 14:56.08 | Samot | Let's see what issue you are actually having... |
| 14:56.10 | EvilCoder | Any way I can share my desktop? |
| 14:56.16 | Samot | No |
| 14:56.21 | Samot | ~pb |
| 14:56.22 | infobot | somebody said pastebin was 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. |
| 14:56.25 | igcewieling | Download PuTTY and use scrollback |
| 14:56.33 | EvilCoder | wait let me just past it here, there is no pw anyway |
| 14:56.41 | Samot | ~pb |
| 14:56.41 | infobot | somebody said pastebin was 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. |
| 14:56.44 | Samot | Not here. |
| 14:56.45 | Samot | There |
| 14:56.51 | EvilCoder | [general] |
| 14:56.51 | EvilCoder | autokill=yes |
| 14:56.51 | EvilCoder | ; TYPES: |
| 14:56.51 | EvilCoder | ; friend = both sides (incoming and outgoing) |
| 14:56.51 | EvilCoder | ; peer = send calls to |
| 14:56.54 | EvilCoder | ; user = auth an incomming call |
| 14:56.56 | EvilCoder | [uplink-casa400-ams] |
| 14:56.59 | EvilCoder | type=friend |
| 14:57.01 | EvilCoder | accountcode=casa400 |
| 14:57.02 | Samot | FFS. |
| 14:57.04 | EvilCoder | host=172.24.0.60 |
| 14:57.06 | EvilCoder | permit=172.24.0.60 |
| 14:57.07 | Samot | STOP |
| 14:57.09 | igcewieling | kick 'em |
| 14:57.09 | EvilCoder | user=uplink-to-tc2 |
| 14:57.11 | EvilCoder | qualify=yes |
| 14:57.12 | *** part/#asterisk igcewieling (~ewieling@ip98-170-215-112.pn.at.cox.net) |
| 14:57.14 | EvilCoder | context=casa |
| 14:57.16 | EvilCoder | description=in/out for casa400. |
| 14:57.19 | EvilCoder | [uplink-ramada-ams] |
| 14:57.19 | Samot | STOP |
| 14:57.21 | EvilCoder | type=friend |
| 14:57.24 | EvilCoder | accountcode=ramada |
| 14:57.26 | EvilCoder | host=172.24.0.42 |
| 14:57.29 | EvilCoder | permit=172.24.0.42 |
| 14:57.31 | EvilCoder | user=uplink-to-tc2 |
| 14:57.34 | Samot | Completely useless |
| 14:57.34 | EvilCoder | qualify=yes |
| 14:57.36 | EvilCoder | context=ramada |
| 14:57.39 | EvilCoder | description=in/out for ramada. |
| 14:57.41 | EvilCoder | oh |
| 14:57.43 | Samot | You're going to have to do this all over again in the right spot. |
| 14:57.44 | EvilCoder | Sorry |
| 14:57.46 | EvilCoder | Saw that too late |
| 14:57.50 | Samot | Please follow instructions. |
| 14:57.52 | Samot | No. |
| 14:58.02 | Samot | I told you to use pastebin before you said "I'll post it here" |
| 14:58.16 | Samot | And I said no, and gave the pastebin link again. |
| 14:58.20 | Samot | Please put this on pastebin |
| 14:58.31 | EvilCoder | Yes, I will do it properly now, my apologizes. |
| 14:59.42 | EvilCoder | Link to my iax.conf is : https://pastebin.com/8gV2LGeY |
| 15:00.45 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 15:00.45 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 15:02.06 | *** join/#asterisk krzie (~k@openvpn/community/support/krzee) |
| 15:02.57 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 15:03.10 | Samot | Where is the call showing this hitting the wrong context/trunk? |
| 15:03.28 | *** join/#asterisk OS-24589 (~Moj0@unaffiliated/moj0) |
| 15:03.41 | OS-24589 | good morning (evening) all |
| 15:03.43 | EvilCoder | Sec, let me also put that on pastebin. Moment |
| 15:03.57 | *** join/#asterisk matrix1233 (~matrix123@197.3.205.70) |
| 15:04.00 | *** join/#asterisk dan_j_ (sid21651@gateway/web/irccloud.com/x-nvoiposukprjbrig) |
| 15:04.08 | Samot | Why do you have permit= in this? |
| 15:04.17 | Samot | host= with the IP is all you need. |
| 15:04.23 | Samot | If it don't match that host, it don't work. |
| 15:04.42 | EvilCoder | Ah I didn't even notice that. Thanks, will remove that. |
| 15:04.47 | Samot | The only time you need permit/deny is when the host is dynamic. |
| 15:05.02 | EvilCoder | It used to be, so no chance to remove it yet |
| 15:05.49 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 15:05.52 | EvilCoder | The call information is : https://pastebin.com/3v2Nb50E |
| 15:06.14 | *** join/#asterisk robinak (~quassel@unaffilated/robink) |
| 15:06.21 | OS-24589 | Another interesting problem for you all - fresh install FreePBX Distro (x64) with a TDM 410 and a wildcard TE121 card. Stateside PRI coming into it. Things *seem* fine, but when making an othbound call, upon passing the call to connect asterisk has a segfault and leaves the channel open but no voice moving |
| 15:06.34 | *** join/#asterisk FatalFUUU (~none@vps.esdf.uk) |
| 15:06.47 | OS-24589 | inbound calls seem to be fine |
| 15:06.55 | Samot | OS-24589: #freepbx |
| 15:06.57 | Samot | Go there. |
| 15:07.19 | OS-24589 | will do , came here cuz asterisk is the segfault..but it's their distro :) |
| 15:07.21 | Samot | EvilCoder: This was the casa trunk? |
| 15:07.22 | EvilCoder | I also have used the TE121 card, and works fine on my asterisk |
| 15:07.35 | EvilCoder | Samot, it came from the Ramada trunk |
| 15:07.44 | EvilCoder | But it arrived on the Casa400 destination on the main uplink |
| 15:08.02 | *** join/#asterisk Martin` (~martin@shell.ipv6.octocore.net) |
| 15:08.16 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-cadmqoezaojgddzv) |
| 15:08.38 | EvilCoder | Very strange issue, especially that it just completely ignores the destination context set in the iax configuration |
| 15:08.50 | *** join/#asterisk file_ (uid178970@asterisk/developer-and-muffin-lover/file) |
| 15:08.50 | *** mode/#asterisk [+o file_] by ChanServ |
| 15:08.55 | OS-24589 | the cards were great - everything was working smashingly until an upgrade failed - which resulted in a full clean rebuild |
| 15:09.40 | *** join/#asterisk miralin1 (~Thunderbi@195.19.212.24) |
| 15:09.46 | *** join/#asterisk troyt_ (~troyt@c-24-11-28-185.hsd1.ut.comcast.net) |
| 15:10.16 | *** join/#asterisk Rac-on_ (jasper@bambi.rac-on.nl) |
| 15:11.03 | EvilCoder | You can also ref the IP on top of the pastebin, so you can see its really the right ip+context+config |
| 15:12.18 | *** join/#asterisk yoink (~yoink@unaffiliated/yoink) |
| 15:13.28 | Samot | Change the order. |
| 15:15.49 | EvilCoder | that won't help, because Asterisk sorts them alphabetic |
| 15:16.03 | EvilCoder | Tried, and no results, also when doing iax show peers you can still see the casa one on top. |
| 15:16.14 | Samot | You can try authing each peer. |
| 15:16.23 | Samot | With their own username/secret |
| 15:17.08 | EvilCoder | Well not really, its a production system right now. And this should also work? |
| 15:17.25 | Samot | I don't know. |
| 15:17.35 | Samot | I don't use IAX because..well SIP |
| 15:17.50 | EvilCoder | And does SIP allow you to set the contexts properly? |
| 15:17.56 | *** join/#asterisk marlinc (~marlinc@1.0.0.127.13.204.167.185.in-addr.arpa) |
| 15:18.03 | Samot | Of course. |
| 15:18.34 | EvilCoder | You say it if there are absolutely no bugs, but after having so many issues with Asterisk there will probably come something new |
| 15:18.38 | Samot | IAX was an answer to when SIP wasn't plausible. |
| 15:18.44 | Samot | Like 15 years ago. |
| 15:19.02 | Samot | I peer Asterisk boxes with SIP all over. |
| 15:19.10 | EvilCoder | Hmm very well then |
| 15:19.17 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 15:19.20 | EvilCoder | Will install test-environment, and use SIP trunking instead |
| 15:19.25 | Samot | I have an entire network of Kamailio and Asterisk that use SIP peers to send calls between them. |
| 15:20.12 | EvilCoder | Yes, but I also have phones connected on the SIP channel and gsm gateways. So I would loved to seperated them from each other |
| 15:22.14 | Samot | I have phones on SIP channels on these boxes. |
| 15:22.26 | Samot | I also have outbound calls going out from the boxes.. |
| 15:25.39 | *** join/#asterisk igcewieling (~ewieling@ip98-170-215-112.pn.at.cox.net) |
| 15:26.26 | EvilCoder | I've posted it to the issueboard of Asterisk |
| 15:27.29 | EvilCoder | Thats not my point Samot, my point is that I want it to be seperated. As it has been for over years now. |
| 15:28.38 | *** join/#asterisk matrix1233 (~matrix123@197.3.205.70) |
| 15:40.22 | Samot | I don't know what you mean by "separated" |
| 15:40.28 | Samot | Because all mine are. |
| 15:41.47 | igcewieling | EvilCoder: what version of Asterisk? |
| 15:42.13 | EvilCoder | Ast 13.8/cert4 |
| 15:42.38 | igcewieling | why not turn off trunking? |
| 15:42.58 | EvilCoder | What do you mean with turn off trunking? |
| 15:43.15 | igcewieling | You said earlier account codes don't work on trunks, is that correct? |
| 15:43.26 | EvilCoder | On iax tunnels, yes |
| 15:43.35 | igcewieling | I've never heard of an IAX tunnel. |
| 15:43.40 | Samot | The IAX trunk did not match the IP |
| 15:43.49 | *** join/#asterisk miralin (~Thunderbi@194.8.128.115) |
| 15:43.54 | EvilCoder | Yes it does, lol |
| 15:43.55 | Samot | He sent the call from a specific IP, the wrong trunk context was loaded. |
| 15:44.19 | EvilCoder | Ooh, you mean like not match the right trunk |
| 15:44.23 | igcewieling | that must have been after I evacuated the flooding. |
| 15:44.24 | EvilCoder | Sorry, yes thats correct |
| 15:44.26 | Samot | Right. |
| 15:44.49 | Samot | You sent the call from specific IP, even though Asterisk allowed the call because the trunk had it listed in the host.. |
| 15:44.58 | Samot | It used the wrong trunk context to send the call to. |
| 15:45.16 | EvilCoder | Correct |
| 15:45.50 | Samot | Why I suggested to use a username/secret in addition to the host |
| 15:46.02 | Samot | So when it does come in, it will have to AUTH on the proper trunk. |
| 15:46.11 | Samot | See if it sends the call to the right context. |
| 15:46.17 | Samot | If now, I would suggest SIP. |
| 15:46.39 | Samot | Because honestly, IAX is just the walking dead. |
| 15:46.44 | Samot | It's dead but doesn't realize it. |
| 15:48.09 | EvilCoder | I will change it in the future, yes. But for now the answer stays No. |
| 15:48.28 | EvilCoder | Changing that would mean to change a lot of other stuff, its not just asterisk which needs to be changed. |
| 15:48.30 | Samot | OK. |
| 15:48.41 | Samot | So for now you have calls loading on the wrong contexts. |
| 15:48.44 | EvilCoder | That requires maintenance windows, nightly work *ugh* |
| 15:48.48 | Samot | So you'll need to adjust for that. |
| 15:49.21 | EvilCoder | Well, I did make a ticket on issues.asterisk.org |
| 15:49.42 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 15:50.44 | Samot | OK |
| 15:50.55 | Samot | I wouldnt hold my breath on that. |
| 15:51.06 | EvilCoder | Yeh, i figured. But we'll see. |
| 15:51.12 | EvilCoder | Otherwise, f* it. |
| 15:51.17 | EvilCoder | And i'll leave it for now. |
| 15:51.36 | Samot | Doesnt this mess with your calls? |
| 15:53.27 | *** part/#asterisk OS-24589 (~Moj0@unaffiliated/moj0) |
| 15:58.14 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 16:06.07 | *** join/#asterisk matrix1233 (~matrix123@197.3.205.70) |
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| 16:13.52 | *** join/#asterisk nickgaw (~nick@adsl-108-78-239-15.dsl.hstntx.sbcglobal.net) |
| 16:15.08 | nickgaw | Hi, I just upgraded from asterisk 14.3.0 to 14.4.0 and had to recompile everything from scratch again is there not an upgrade source method where I can just download the patches or the sources from git and just recompile what has changed rather then the entire thing all over again to save time? |
| 16:16.21 | Samot | Upgrading = recompiling with new version |
| 16:17.16 | Samot | If you are installing from source. |
| 16:17.25 | nickgaw | yes but just what has changed not everything else probably most of the stuff I already have from the old version. |
| 16:17.34 | Samot | If you have a package then it's how you would update a package for the OS you're on. |
| 16:17.59 | Samot | Look at the change log. |
| 16:17.59 | nickgaw | So they don't distribute patches for the sources? |
| 16:18.11 | Samot | They release a new version. |
| 16:18.21 | Samot | You install it when you want to. |
| 16:18.28 | nickgaw | yesI have |
| 16:19.00 | nickgaw | kind of like rsync for source code where I just fetch the new parts and recompile them and relink the binaries. |
| 16:19.02 | Samot | Source installs basically work the same way. |
| 16:19.11 | Samot | If I install Apache 2.4.4 from source.. |
| 16:19.26 | Samot | When Apache 2.4.5 comes out, I recompile from the source. |
| 16:19.59 | Samot | It takes like 10 minutes to recompile. |
| 16:20.01 | Samot | If that. |
| 16:20.23 | nickgaw | yes but you have to recompile everything even parts you already have I am running this asterisk installation on a virtual private server with a single core so it takes a long time to recompile everything. |
| 16:20.42 | Samot | OK. |
| 16:21.03 | Samot | I've compiled Asterisk on a 1CPU server, it doesn't take much resources. |
| 16:21.32 | nickgaw | I am trying to save time in upgrading asterisk so you are saying the only way to upgradecompletely from source is to download untar and recompile everything all over again? |
| 16:21.36 | Samot | Because what you are basically saying is anything you have to (re)compile on the server is going to take up resources and hose the server. |
| 16:21.48 | Samot | It takes less than 10 minutes to do this. |
| 16:21.56 | Samot | How much time do you need to save on top of that? |
| 16:22.43 | nickgaw | not a lot just trying to save as much as possible on this virtual private server it took around 10 or 15 minutes as I added the --with-pjsip-bundled option. |
| 16:23.23 | nickgaw | Next time I will time it as I would like to know how long it takes. |
| 16:23.44 | Samot | So do it at night or when the server isn't in use that much. |
| 16:24.12 | EvilCoder | @Samot, currently billing from the other side |
| 16:24.18 | EvilCoder | So not really atm |
| 16:24.32 | Samot | Well --with-psjip-bundle it still has to reach out and download the pjsip library |
| 16:24.53 | Samot | EvilCoder: So the calls aren't landing at the wrong destination? |
| 16:25.05 | Samot | Even though they are using the wrong context? |
| 16:26.29 | nickgaw | It is really no big deal I was just asking if there are methods for saving time in rebuilding asterisk. |
| 16:26.43 | Samot | If you build from source.. |
| 16:26.54 | Samot | You have to recompile when installing new versions. |
| 16:27.06 | Samot | If you install from a package, you're still doing the same thing. |
| 16:27.19 | *** join/#asterisk Samael28 (~Samael28@79.110.128.128) |
| 16:27.27 | nickgaw | yes true but do you understand what I was trying to do? |
| 16:28.07 | Samot | Yes but it's not going to happen. |
| 16:28.26 | EvilCoder | Samot, no i'm just pushing them out to the dahdi channel. They all eventually go there, i was just hoping of adding some macro's to it, such as check outgoing caller-id. And adding some cdr information. But I'll try to do that from the other end before they go to the uplink-asterisk server. |
| 16:28.27 | Samot | Not unless you can 100% guarantee that underlying code hasn't changed for bug fixes and updates. |
| 16:29.28 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 16:29.29 | Samot | So all the calls comes in different trunks but go out to the PSTN on the same one? |
| 16:31.23 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 16:32.07 | nickgaw | I was trying to only recompile the changes and relink the asterisk binary rather then the entire source tree again. |
| 16:32.07 | nickgaw | but yes I totally get your points about recompiling all over again from source as I like compiling from source as you get the latest version. |
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| 16:35.02 | Samot | Here's the issue with that. |
| 16:35.14 | Samot | 1) Where would the other source files be coming from? |
| 16:35.39 | Samot | 2) People, in some cases, don't keep source files on the server. |
| 16:35.48 | Samot | Once they are compiled and installed... |
| 16:36.23 | Samot | 3) When recompiling the existing version because of errors, you still have to "make clean" so the previous make information is cleared out. |
| 16:36.45 | Samot | It is just easier and faster to download the new tarball and install from there. |
| 16:37.54 | Samot | So you would still need the 13.14.0 source on the system to be used for the "changes-only" recompile because it still needs all those source files for a recompile. |
| 16:39.19 | nickgaw | ok now I understand your points I was thinking of things like the kernel where you can patch between the differences and just then recompile the new sources so you could not download or produce a patch from 14.3.0 to 14.4.0 and then repatch the old sources? |
| 16:39.44 | Samot | Why would you? |
| 16:39.48 | Samot | Honestly? |
| 16:39.59 | Samot | They release 14.3.0 |
| 16:40.19 | Samot | Between 14.3.0 and 14.4.0 they release fixes and updates. |
| 16:40.47 | Samot | I believe the confusion is this, the term "upgrade" |
| 16:41.03 | Samot | Yes, you are "upgrading" because you are going from 14.4.0 to 14.4.1 |
| 16:41.12 | Samot | But really, you're installing a NEW version. |
| 16:41.17 | Samot | OVER the existing one. |
| 16:41.19 | nickgaw | good point. they would have to release patches for every release between every numbered release to allow that system to work properly which would be a mess. |
| 16:42.12 | Samot | This isn't like some web app that says "Here are the 6 new files that were changed for the updates, just replace the current ones" |
| 16:42.42 | *** part/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 16:42.44 | nickgaw | After I finish my second read threw of the asterisk book if I would like people to help me out do either you or others offer paid work on asterisk setting it up and configuration to my needs? |
| 16:42.51 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 16:43.35 | Samot | I don't solicit on IRC. |
| 16:43.40 | *** join/#asterisk matrix1233 (~matrix123@197.3.205.70) |
| 16:43.53 | Samot | I'm on here to be part of the community and help out that way. |
| 16:44.01 | Samot | Not to find new clients. |
| 16:45.06 | nickgaw | ok just wonderingas I am sure you have lots of clients and I understand about why you would not want to look for clients on IRC as I am sure lots of people want services done. |
| 16:45.18 | Samot | Yes they do. |
| 16:45.29 | Samot | But they come in here for community (free) help. |
| 16:46.06 | Samot | There are plenty of outlets to find freelance techs that are experienced with Asterisk. |
| 16:46.40 | Samot | Or perhaps someone will see your requests in here and PM you about it. |
| 16:46.53 | nickgaw | From asterisk 11 to14 should the basic configuration file syntax be valid other then the new features like setting up extensions and voicemails work ok? |
| 16:47.02 | Samot | Sure. |
| 16:47.08 | nickgaw | from setting up a brand new installation. |
| 16:47.16 | Samot | But 12 introduced a bunch of new stuff. |
| 16:47.19 | Samot | Like PJSIP |
| 16:47.31 | Samot | Changes to Call Event Logging, CDRs, etc. |
| 16:48.19 | nickgaw | but I should be able to follow the book and get a working system like voicemail boxes and phone extensions as well as the automated attendent and or an IVR menu? |
| 16:52.56 | nickgaw | I understand that you are not looking for new clients but once I finish reading the book if I should want to pay someone where is a good place to look for workers? |
| 16:54.13 | Samot | The book should get you through... |
| 16:55.58 | nickgaw | So what you are saying is first finish the book then if I need help come back and ask? |
| 16:56.11 | Samot | Yes. |
| 16:56.17 | Samot | Try and when you have issues, ask. |
| 16:57.22 | nickgaw | I have three numbers I ported to a sip provider can one trunk service all three numbers if I have two on one asterisk system and one on another? |
| 16:58.50 | nickgaw | Like two numbers on the virtual private server and the one testing number on my laptop that I want to use for testing should all three be able to be called? |
| 16:59.12 | *** part/#asterisk nickaugust (~nick@nickaugust.com) |
| 16:59.28 | nickgaw | This way I can test out things on one number and if they work then move the information to the production system? |
| 17:06.19 | *** join/#asterisk vstemen (~vince@ip72-202-187-29.ok.ok.cox.net) |
| 17:10.40 | nickgaw | It looks like asterisk does not do paralell makes when given -j9 on my I7 laptop. |
| 17:12.26 | *** join/#asterisk vstemen (~vince@ip72-202-187-29.ok.ok.cox.net) |
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| 18:28.24 | igcewieling | considers letting my domain lapse and simply not have a personal e-mail address. |
| 18:28.32 | igcewieling | Can't spam what does not exist. |
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| 18:34.40 | Samot | Yes you can. |
| 18:34.46 | Samot | Just doesn't go anywhere. |
| 18:34.56 | Samot | You can mail a lot of stuff to a non-existing address. |
| 18:35.23 | nickgaw | yes you can but I am careful with doing that. |
| 18:39.49 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
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| 20:20.50 | P2E | ok, I finally have an ATA112 on the way (I think) - is there anything I should do to make sure it's not counterfeit? any good resources for this? |
| 20:20.50 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:35c7:1f72:8fb7:f09e) |
| 20:28.03 | *** join/#asterisk matrix1233 (~matrix123@197.3.205.70) |
| 20:32.06 | igcewieling | counterfit what? |
| 20:34.19 | P2E | oops, SPA112 |
| 20:34.31 | P2E | sorry, was reading something and typing at the same time |
| 20:34.56 | igcewieling | I have no idea how one might tell if it is fake or not. |
| 20:36.24 | P2E | me neither; that's why I asked |
| 20:55.11 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
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| 21:11.11 | Samot | Does the cover say Cisco? |
| 21:11.58 | Samot | There generally aren't "counterfeit" ATAs. |
| 21:14.21 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
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| 21:34.00 | P2E | back when we discussed purchasing, it was advised against purchasing one on ebay because most of them are |
| 21:35.06 | P2E | I don't anticipate I will receive a counterfeit unit, but checking the case cover is something I can do, didn't know if there are any other things |
| 21:35.13 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 21:35.50 | Samot | ... |
| 21:35.52 | Samot | No |
| 21:36.39 | Samot | Old Linksys ATAs, like the PAP2, are known for being locked by a provider. |
| 21:40.07 | P2E | ok. well if there isn't much danger, I won't do much checking |
| 21:41.07 | Samot | The only way it would be bogus is if you recieved a different model. |
| 21:48.51 | P2E | ok. I guess the cisco things that were most counterfeited were routers and the like? |
| 21:49.26 | Samot | Not that I am aware of |
| 21:49.39 | Samot | They are kinda hard to fake |
| 21:50.13 | Samot | Mainly due to their being a "Cisco way" of doing things. |
| 21:52.54 | Samot | Hey file: In regards to the call recording we were talking about in #freepbx and the limit on recording outbound channels... |
| 21:53.16 | Samot | It was noted as an Asterisk limitation but should have been resolved in 13, is X-Rob correct in that? |
| 21:53.21 | Samot | Or was it something resolved in 14? |
| 22:11.25 | file | limit on recording outbound channels? the behavior of things based on where MixMonitor is placed was made more sane |
| 22:11.41 | Samot | OK. |
| 22:11.50 | file | that was done in 13+ |
| 22:11.54 | file | well, technically 12+ |
| 22:11.54 | Samot | Yeah the limit of I call someone over the PSTN |
| 22:12.18 | Samot | When I transfer them to someone else, the recording is stopped because it was on my channel and not the "outbound" channel. |
| 22:12.42 | file | provided MixMonitor is started on the outbound channel, it follows it |
| 22:12.49 | Samot | Got it. |
| 22:13.07 | file | wherever MixMonitor is placed it follows through transfers |
| 22:13.56 | Samot | So as long as I place it on the outbound channel, i can transfer it between all the extensions... |
| 22:13.58 | Samot | Cool. |
| 22:44.18 | *** part/#asterisk kharwell (kharwell@nat/digium/x-rwtizhuptocakaae) |
| 23:31.08 | *** join/#asterisk CRCinAU (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 23:31.33 | CRCinAU | ok - so I'm putting together a 'Lenny' extension. |
| 23:32.10 | CRCinAU | I use BackgroundDetect to play the sound file.... but I'm trying to trap if nothing is said to play something else. |
| 23:32.25 | CRCinAU | ie maybe BackgroundDetect falls through and I can play something else, then loop. |
| 23:37.16 | *** join/#asterisk matrix1233 (~matrix123@197.0.130.40) |
| 23:44.00 | *** join/#asterisk chendy (~alexc@183.14.135.209) |