| 00:34.00 | *** join/#asterisk xrobau (~xrobau@2001:4830:121e:ffff:44ae:cf3e:1ea1:dfa0) |
| 01:16.39 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 01:26.31 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:32.43 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:45.15 | *** join/#asterisk matrix1233 (~matrix123@197.3.255.218) |
| 01:53.06 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 02:01.29 | *** join/#asterisk brokensyntax (~quassel@45.62.254.31) |
| 02:06.40 | *** join/#asterisk boris_t (~boris_t@212.119.243.183) |
| 02:35.33 | *** join/#asterisk matnav (~matnav@cpe-24-27-45-27.austin.res.rr.com) |
| 03:09.43 | *** join/#asterisk iheartlinux (~jwpierce3@mail.johnthecomputerguy.com) |
| 03:10.14 | iheartlinux | is it not possible to put a variable inside sip.conf [context]? |
| 03:24.10 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
| 03:33.53 | *** join/#asterisk matrix1233 (~matrix123@197.3.255.218) |
| 03:37.07 | WIMPy | You can set channel variables, if that's what you want. |
| 03:44.01 | iheartlinux | WIMPy: I was/am trying to use a central "account" variable for both sip.conf and extensions.conf. |
| 03:44.35 | WIMPy | Doing what? |
| 03:45.30 | iheartlinux | or include rather. i.e. [${device}] in sip.conf, & SIP/${device} in extensions.conf |
| 03:46.41 | WIMPy | I have no idea what that variable thing is supposed to do there. |
| 03:47.45 | WIMPy | What are you trying to accomplish? |
| 03:50.15 | iheartlinux | I want a central include, lets say accounts, to be included "#include accounts" in both sip.conf and extensions.conf, that will allow me to manage the "sip devices"/extensions/whatever centrally |
| 03:50.53 | WIMPy | And how would you use that? |
| 03:51.15 | matnav | gets popcorn |
| 03:51.18 | iheartlinux | this works fine for extensions.conf i.e. SIP/${device} |
| 03:51.26 | WIMPy | There is users.conf, but that's not really recommended. |
| 03:52.11 | iheartlinux | but when you put a variable in sip.conf inside square brackets, it gets ignored |
| 03:52.24 | WIMPy | I don't see how variables would help. Unless it was an array, which is something Asterisk doesn't know. |
| 03:53.04 | WIMPy | Just write your own config file and generate asterisk configs from that, like many of us do. |
| 03:53.54 | iheartlinux | well, aparently it doesn't in this case. I'll just have to maintain sip.conf seperately |
| 03:54.50 | iheartlinux | I wouldn't need an array in this case. just a common variable |
| 03:55.16 | iheartlinux | a variable common to both sip and extensions |
| 03:56.53 | WIMPy | What's the benefit of using the same variable name instead of the same peer name? |
| 04:02.49 | iheartlinux | I decided to name my devices randomly using "pwgen" i.e. [J2NMLn1HTnMfhFk7] & SIP/J2NMLn1HTnMfhFk7. Probably not much benefit other than being able to identify without using comments and also to be able to switch out peers slightly easier. Really just for organizational/readability purposes |
| 04:05.02 | iheartlinux | I manage many asterisk boxes (for small businesses). I just wanted to make my life easier as I jump from box to box |
| 04:05.28 | matnav | sounds complicated.. |
| 04:06.05 | WIMPy | Well, just generate the config. Makes life easier in many other ways as well. |
| 04:06.41 | iheartlinux | no not really. if I use a "random string" instead of an extension number. I can switch phones and their locations remotely instead of having to go to the location to reprogram the phone |
| 04:06.56 | matnav | ah I see |
| 04:07.01 | matnav | I never thought about that |
| 04:07.08 | matnav | Asterisk newb here. |
| 04:07.52 | matnav | would it be the same for programmable buttons and what not? |
| 04:07.53 | WIMPy | You could even integrate phone provisioning. |
| 04:07.55 | iheartlinux | Me too, started the beginning of the year. :) |
| 04:08.31 | matnav | we just go in and change everything. That's kind of the service we offer. |
| 04:09.01 | iheartlinux | WIMPy: I not that advanced yet... :) |
| 04:09.48 | matnav | and implment auto OUI on the switches to automaticlly get phones on the correct vlan too |
| 04:09.57 | matnav | in a perfect world.. |
| 04:10.52 | iheartlinux | I do small business, vps', no vlans, |
| 04:12.01 | matnav | so do you setup on-premise pbx servers? |
| 04:12.09 | iheartlinux | nope |
| 04:12.11 | matnav | k |
| 04:12.16 | matnav | what do you use for traffic shaping? |
| 04:13.11 | iheartlinux | honestly, haven't had a problem with traffic. probably due to the nature of my customers |
| 04:14.59 | matnav | lucky you |
| 04:15.11 | matnav | that Level 3 the other day though |
| 04:15.12 | matnav | omg |
| 04:16.33 | iheartlinux | reading about it now |
| 05:09.11 | *** join/#asterisk lankanmon (~LKNnet@2607:fea8:d20:239::a) |
| 05:22.39 | *** join/#asterisk matrix1233 (~matrix123@197.2.13.84) |
| 05:57.28 | *** join/#asterisk Iamnacho (~Iamnacho@ip24-252-4-195.om.om.cox.net) |
| 05:59.12 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 06:10.59 | *** join/#asterisk matrix1233 (~matrix123@197.2.13.84) |
| 06:22.06 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-yfyencuyuvagxfui) |
| 06:22.45 | *** join/#asterisk hsmiths (uid95325@gateway/web/irccloud.com/x-puobbvhbxkwpqfov) |
| 06:43.57 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 06:51.16 | *** join/#asterisk hsmiths (uid95325@gateway/web/irccloud.com/x-zzeublgpwcdcmgre) |
| 06:51.18 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-noupapicatlfttok) |
| 07:00.28 | *** join/#asterisk tirej (~tirej@95.9.123.3) |
| 07:02.01 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:06.39 | tirej | hi everyone |
| 07:12.22 | *** join/#asterisk matsk (~matsk@vpnbl.borderlight.net) |
| 07:22.52 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:33.14 | *** join/#asterisk zahyl (~quassel@185.12.1.186) |
| 07:35.23 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 07:35.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 07:44.06 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 07:57.42 | *** join/#asterisk miralin1 (~Thunderbi@195.19.212.23) |
| 07:58.10 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 08:04.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:19.27 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 08:24.26 | *** join/#asterisk matrix1233 (~matrix123@197.2.13.84) |
| 08:37.50 | *** join/#asterisk ruied (~ruied@192.61.136.95.rev.vodafone.pt) |
| 08:56.27 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 09:00.25 | Rasputin3711 | What is the recommended configuration for: max 40 cps, 90 ext, ulaw/ulaw, without transcoding,recording,cdr,etc, only calls point-to-point? |
| 09:06.13 | Samot | Uhm. |
| 09:06.16 | Samot | 40 CPS? |
| 09:06.45 | Samot | What is generating 40 CPS? |
| 09:07.24 | Samot | How do 90 users generate 40 CPS? |
| 09:09.23 | tuxd00d | Perhaps he means, total simultaneous calls. |
| 09:10.02 | Rasputin3711 | yep,tsc |
| 09:10.26 | Samot | BIG difference. |
| 09:10.29 | Samot | HUGE |
| 09:12.02 | tuxd00d | Rasputin3711: What do you mean by ârecommended configurationâ? Are you asking what type of hardward is needed? |
| 09:12.20 | Rasputin3711 | ~ cpu,disk,ram |
| 09:12.43 | seik0 | Hi! does digium have statistics on downloads of old versions of Asterisk? |
| 09:12.54 | Samot | You probably want about 6 cores. |
| 09:13.03 | Samot | Or vCPUs. |
| 09:13.10 | Samot | And about 4GB or so of RAM. |
| 09:13.39 | tuxd00d | Samot: I think that is overkill⦠heâs not doing any transcoding. |
| 09:13.52 | Samot | 90 concurrent calls? |
| 09:13.59 | tuxd00d | 40 |
| 09:14.03 | Samot | er 40 concurrent calls with 90 users. |
| 09:14.25 | Samot | I also have a feeling we're not getting the full scope. |
| 09:14.55 | Rasputin3711 | 40 it's a maximum |
| 09:15.08 | tuxd00d | Rasputin3711: Are you going to be doing this on a dedicated machine, or as a virtual server? |
| 09:15.19 | Rasputin3711 | the dedicated machine |
| 09:15.58 | Rasputin3711 | average 10-20 concurrent calls |
| 09:16.06 | tuxd00d | Then any machine should be fine. Asterisk isnât a resource hog. |
| 09:16.17 | seik0 | you really do not need 6 cores and 6ram for 40 simple concurrent calls and 100 users |
| 09:16.22 | seik0 | *4gb ram |
| 09:16.59 | Samot | Did you guys ever stop to think about growth? |
| 09:17.12 | Samot | Or various other items that could impact things? |
| 09:17.17 | seik0 | wait, I have just this configuration, but with hard DB connection and monitoring recording and voicemail |
| 09:17.28 | seik0 | he asked for minimal |
| 09:17.40 | Samot | He asked for recommended |
| 09:17.44 | Samot | That's not minimal. |
| 09:17.45 | seik0 | not a configuration, but 40 concurrent calls and 100 users |
| 09:17.49 | tuxd00d | Is this machine only going to be running Asterisk, or will other services such as MySQL be running? |
| 09:18.00 | Rasputin3711 | only asterisk |
| 09:18.12 | Samot | So no voicemail? |
| 09:18.21 | Samot | No email notifications? |
| 09:18.22 | Rasputin3711 | nope |
| 09:18.59 | Samot | Quad core, 2GB RAM. |
| 09:19.15 | tuxd00d | Rasputin3711: Asterisk will work well, but you may also want to look at Kamailio or openSIPS |
| 09:19.47 | tuxd00d | Kamailo and openSIPS only do what youâre asking for, not much more. |
| 09:19.51 | Samot | I'm not even sure he needs that. |
| 09:19.54 | Rasputin3711 | For that case do i need it? |
| 09:20.31 | tuxd00d | Asterisk has a better community than those other tow, so if you need help, Asterisk may be the better option. |
| 09:20.37 | seik0 | I have server with 2 running asterisk (for dahdi, simple and stable, and main, more complicated), about 40 concurrent calls, monitoring recording, heavy DB usage (DB on other machine). It has 2GB ram, 1 core (1 cpu), slow HDD |
| 09:20.45 | Rasputin3711 | I do not have any external links |
| 09:20.55 | seik0 | and sometimes it works slow |
| 09:21.07 | Samot | Yes, sometimes it will be. |
| 09:21.12 | Samot | That's a bit low for that. |
| 09:21.31 | seik0 | CPU is now enough |
| 09:21.43 | seik0 | *not |
| 09:22.16 | seik0 | in fact, asterisk does not need much RAM |
| 09:22.37 | tuxd00d | It doesnât need much of anything.... |
| 09:23.01 | seik0 | cpu for transcoding |
| 09:23.02 | tuxd00d | Itâs only gets busy with tasks such as recording and voicemail. |
| 09:23.06 | seik0 | hdd for writing |
| 09:23.21 | Samot | Or setup or tear down. |
| 09:24.23 | tuxd00d | Rasputin3711: The answer isâ¦. it depends. |
| 09:24.57 | tuxd00d | Any quad-core with 4+ GB or RAM should work fine. |
| 09:25.05 | *** join/#asterisk whizzi (~whizzi@185.21.52.255) |
| 09:25.25 | Samot | Even 2GB |
| 09:25.37 | Rasputin3711 | What profit will i get from Kamailio or openSIPS? |
| 09:25.40 | Samot | If all you are doing is calls. |
| 09:26.02 | tuxd00d | Rasputin3711: Kamailio and openSIPs only do SIP registration and routing. |
| 09:26.15 | Samot | For what you are doing? Probably not much. |
| 09:26.34 | Rasputin3711 | I don't have routing. |
| 09:26.40 | tuxd00d | They would be in place of Asterisk |
| 09:26.46 | tuxd00d | Call routing. |
| 09:26.57 | Samot | And zero media control |
| 09:27.01 | Rasputin3711 | and call routing too |
| 09:27.06 | tuxd00d | But, like I said, finding help with them is hard. |
| 09:27.13 | Samot | Outside of proxying the media |
| 09:27.43 | Rasputin3711 | it will be isolate system |
| 09:27.58 | Samot | What is this for? |
| 09:28.13 | Samot | What is the purpose? |
| 09:31.03 | Rasputin3711 | Internal call system, 1 tel per room(between rooms ~ 15-20m), 90 people consult each over |
| 09:32.20 | Rasputin3711 | ~65% don't have a computer and we cann't use a chat or other tools. |
| 09:32.28 | tuxd00d | Rasputin3711: You donât want any calling to or from the outside world? |
| 09:32.41 | Rasputin3711 | yep |
| 09:33.04 | Samot | Yes, you do? Or no, no outside calls? |
| 09:33.30 | Rasputin3711 | Calls only between internal extensions |
| 09:33.55 | tuxd00d | You donât even need a server with some phones, if you only want internal calls. |
| 09:35.42 | Samot | See now... |
| 09:35.47 | Samot | Kamailio makes more sense. |
| 09:35.58 | Samot | As long as all the phones are configured properly and the same. |
| 09:36.08 | tuxd00d | Yes, if it were my project, Iâd use Kamailio. |
| 09:36.24 | Samot | If it's all internal and it's just calls.. |
| 09:36.36 | Samot | Use Kamailio. |
| 09:36.54 | tuxd00d | And old P4 can handle 100,000âs of calls |
| 09:36.56 | Samot | You can get away with 1x1 or 2x2 with CPUxRAM. |
| 09:37.16 | Samot | Just have to make sure the endpoints are configured the same. |
| 09:37.25 | tuxd00d | Jump over to #Kamailio |
| 09:37.31 | Samot | So they always offer up the same codecs in the same order in the SDP. |
| 09:37.55 | tuxd00d | Or limit the phones to only one codec |
| 09:38.10 | Samot | That would fall under the same codec in the same order. |
| 09:38.12 | Samot | 1 |
| 09:38.17 | tuxd00d | Or limit Kamailio to only one codec |
| 09:38.27 | Samot | You can't. |
| 09:38.44 | Samot | The phones must have at least one matching codec. |
| 09:38.46 | tuxd00d | oh yeah, no transcoding. |
| 09:38.48 | Samot | Or the call will fail. |
| 09:38.54 | Samot | Or media control. |
| 09:39.14 | Samot | Media does not flow through Kamailio. |
| 09:39.24 | Rasputin3711 | Kamalio without Asterisk? |
| 09:39.33 | Samot | You can use an RTP proxy like RTPProxy or RTPEngine. |
| 09:39.35 | tuxd00d | Which is why it can hadel so many calls at once on crappy hardware |
| 09:39.36 | Samot | Yes. |
| 09:39.49 | Samot | Kamailio without Asterisk. |
| 09:39.53 | tuxd00d | Rasputin3711: Yes, just Kamailio by itâs self. |
| 09:40.00 | Samot | You're not using any media driven applications. |
| 09:40.04 | Samot | You don't need Asterisk. |
| 09:41.27 | Samot | Honestly, you could probably manually program the phones to add contacts with direct URI dialing. |
| 09:41.35 | Samot | So if I'm 102 and you're 103 |
| 09:42.05 | Rasputin3711 | 90* 90 = 8100 it will be fun) |
| 09:42.07 | Samot | I'm at 192.168.1.34 and you're at 192.168.1.45 I can program a contact in my phone to dial 103@192.168.1.45 |
| 09:42.26 | Samot | No Asterisk, no Kamailio.. |
| 09:42.31 | Samot | Just the two phones. |
| 09:42.34 | Samot | And the call will work. |
| 09:43.35 | tuxd00d | Rasputin3711: Youâd want to have a server for provisioning your phones via HTTP or FTP or something. That way you can configure them via text files on one server instead of configuring each phone by hand. |
| 09:43.57 | Rasputin3711 | I need a second server) |
| 09:43.58 | tuxd00d | Just look for a provisioning guide for your phone. |
| 09:44.00 | Samot | No. |
| 09:44.05 | Samot | You don't need a second server. |
| 09:44.16 | Samot | What kind of phones? |
| 09:44.21 | tuxd00d | Nah, the provisioning server is just an HTTP or FTP service |
| 09:44.32 | Samot | You will need to have Apache or TFTP or FTP |
| 09:44.34 | Rasputin3711 | yealink |
| 09:44.37 | Samot | I'd go with TFTP over FTP. |
| 09:44.55 | tuxd00d | Yeahlinks provision well from HTTP |
| 09:44.57 | Samot | Well, if you want to have provisioning server... |
| 09:45.09 | Samot | You need something that can support HTTP, TFTP or FTP |
| 09:45.21 | tuxd00d | You just set your DHCP server to reference the provisioning server URL |
| 09:45.55 | tuxd00d | Rasputin3711: http://www.yealink.com/Upload/T2X/2014102/Yealink_SIP-T2_Series_T19P_T4_Series_IP_Phones_Auto_Provisioning_Guide_V72_1.pdf |
| 09:46.16 | Samot | So now you can use the same server as Kamailio/Asterisk (which ever you choose) |
| 09:46.34 | Samot | But now you will need to install Apache or Pure-FTP or configure TFTP to run... |
| 09:46.59 | Samot | So a 1GB/1CPU server may or may not be the answer now. |
| 09:47.14 | tuxd00d | Page 30 talks about DHCP option to set the provisioning URL. In this configuration, you donât have to touch any phones. Youâd only have to have a configuration file for it. |
| 09:47.32 | Samot | Yes for TFTP. |
| 09:47.33 | tuxd00d | Samot: 1/1 is still pleanty |
| 09:47.48 | Samot | As long as things are configured right. |
| 09:47.57 | tuxd00d | True |
| 09:47.57 | Rasputin3711 | I think in our case, static conf will be more robust. |
| 09:48.04 | Samot | Install Apache and then let it just spawn 256 child processes. |
| 09:48.10 | Samot | Your RAM will drop like a rock. |
| 09:48.20 | tuxd00d | Yep⦠:P |
| 09:48.31 | tuxd00d | Rasputin3711: What do you mean by âstaticâ? |
| 09:48.53 | Samot | I think he means IP addresses. |
| 09:49.06 | Samot | DHCP options only work if you're using DHCP. |
| 09:49.21 | tuxd00d | Thatâs true. |
| 09:49.46 | tuxd00d | But having a DHCP server running does make network and phone management much easier. |
| 09:50.15 | Samot | That is dependant of the network deployment. |
| 09:50.23 | tuxd00d | A lot of cheap routers can do DHCP options⦠but usually I have them run dd-WRT. |
| 09:50.43 | Rasputin3711 | static ip address |
| 09:50.53 | Samot | All my devices are sent out preconfigured. |
| 09:51.02 | Samot | Because, like, I test them. |
| 09:51.08 | tuxd00d | Static is fine.. just extra work when devices are changed out. |
| 09:51.15 | Samot | Extra work? |
| 09:51.18 | Samot | It's 3 minutes. |
| 09:51.25 | Samot | Log into the phone, add the provisioning URL.. |
| 09:51.36 | Samot | Save, restart, configured. |
| 09:51.45 | tuxd00d | For a 90 fines, sure, youâre good. |
| 09:51.58 | Samot | I'd use provisioning for 90 phones, myself. |
| 09:52.08 | tuxd00d | I manage almost 100,000 phonesâ¦. |
| 09:52.20 | Samot | But adding a provisioning URL to a phone when you swap it is not hard work. |
| 09:52.29 | Samot | The initial setup is the PITA. |
| 09:53.17 | Samot | But I add the provisioning url to all the phones I ship. |
| 09:53.38 | Samot | I make sure the phone pulls the config and registers... |
| 09:54.04 | tuxd00d | There is nothing wrong with the way you want to do it |
| 09:54.23 | Samot | But I also manage multiple custom templates for clients. |
| 09:54.38 | Samot | And the URL changes based on the client. |
| 09:55.52 | tuxd00d | Rasputin3711: Itâs all just a matter of preference. : |
| 09:55.54 | tuxd00d | :) |
| 09:59.18 | dan_j | What storage method is recommended for voicemail when using multiple servers? I'm in the middle of switching to IMAP from Mysql, but i'm not happy with it. It seems to slow down the startup of asterisk. |
| 09:59.45 | Samot | Are you storing the voicemail on each server or do you have a centralized vm server? |
| 10:10.10 | Samot | ?? |
| 10:27.31 | dan_j | Samot: Sorry. Had a phone call. With both imap and mysql, its a centralized vm server. |
| 10:28.08 | dan_j | mysql with a slave. for imap, dovecot in a redundant pair |
| 10:28.18 | Samot | Well centralized makes it easier but honestly, I don't user either option. |
| 10:28.39 | Samot | I use a centralized vm server but just standard Asterisk vm. |
| 10:31.26 | dan_j | I'm not happy with either option really which is why i thought i'd see what method is most used/recommended. |
| 10:31.29 | Samot | If the user just wants their VM in their email, it just gets emailed to them. |
| 10:31.43 | Samot | Honestly, I have yet to hear someone recommend IMAP for VM. |
| 10:32.09 | Samot | I've always just used the standard Asterisk vm. |
| 10:32.18 | Samot | Just route calls to the server when needed. |
| 10:32.59 | dan_j | MYSQL/IMAP is just the storage backend. It's still asterisk's built-in VM system. |
| 10:33.00 | Samot | Use returns a 486, route userb@vm. 408, route useru@vm |
| 10:33.08 | Samot | I just use files. |
| 10:33.30 | dan_j | Files isn't an option for me since Ive got redundant asterisk boxes too. |
| 10:33.40 | dan_j | Automatic failover if one box dies |
| 10:33.48 | Samot | rsync does wonders. |
| 10:35.00 | *** join/#asterisk J0hnSteel (~J0hnSteel@92.55.116.125) |
| 10:37.23 | Samot | So why are you changing from MySQL to IMAP for storage |
| 10:37.24 | Samot | ? |
| 10:38.22 | dan_j | a) I dont like having to handle such a large mysql db. |
| 10:38.37 | dan_j | b) I want my users to be able to manage their messages using an email client. |
| 10:39.00 | dan_j | But I dont like how slow asterisk is to start up when using imap. |
| 11:00.12 | Samot | You mean keep the ones they want and delete the ones they don't via email? |
| 11:00.41 | Samot | Why wouldn't you just email them the VM and delete it after it's emailed? |
| 11:01.07 | Samot | They want to manage their VM on their email server, there they go. |
| 11:01.33 | Samot | That's how I manage my VM via IMAP. I email my VMs to my email account, that's handled via IMAP. |
| 11:04.06 | Samot | I'll even create email accounts for my clients they can connect to via IMAP and just email the VMs to that. |
| 11:04.28 | Samot | Mainly because I'm in the mindset that my VM server is not an email server. |
| 11:05.11 | dan_j | The problem with that is when the user's mailbox is full and won't accept emails. |
| 11:05.33 | Samot | Again, I'll create email accounts for them. |
| 11:05.51 | Samot | So then I can say "No limit" |
| 11:06.01 | Samot | And if the user's mailbox is full... |
| 11:06.07 | Samot | They can't get ANY emails. |
| 11:06.16 | Samot | So its a bigger issue than just getting their VM. |
| 11:06.44 | Samot | If they are using IMAP with a limit, they need to manage their emails better. |
| 11:08.02 | dan_j | Hmm. interesting solution. Provide an email server with imap, have asterisk email the recordings to the imap server and delete the messages. But if someone wanted to access voicemail over the phone AND via imap, I can't delete the messages from asterisk. |
| 11:08.15 | Samot | OK. |
| 11:08.16 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 11:08.21 | Samot | So then keep them there. |
| 11:08.41 | Samot | Asterisk VM via IMAP is a PITA. |
| 11:09.00 | Samot | When you can just email the VMs to an email account that supports IMAP. |
| 11:09.44 | Samot | Either way, management of messages whether they be VM or email is a USER responsability. |
| 11:10.01 | Samot | Either way, management of messages whether they be VM or email is a USER responsibility. |
| 11:10.07 | dan_j | I hear you |
| 11:10.13 | Samot | It's my job to manage how I keep those messages. |
| 11:10.20 | Samot | No matter the storage method. |
| 11:10.48 | Samot | And since my email server is cPanel.. |
| 11:10.54 | Samot | Daily backups.. |
| 11:11.02 | Samot | Oh and cPanel supports clustering... |
| 11:11.08 | Samot | So now all my emails are replicated. |
| 11:12.49 | Samot | The only downside is that you can't delete from email and also delete from VM server if you are storing both. |
| 11:13.13 | Samot | But that can be mitigated with minor user education. |
| 11:14.05 | Samot | But then again, you can then delete it from the VM server while keeping it in email for records. |
| 11:14.17 | Samot | So it does have a benefit as well. |
| 11:14.56 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 11:16.41 | *** join/#asterisk nanoha-sama (znc@van-app-svr.ad.v10networks.ca) |
| 11:26.36 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 11:41.59 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 11:45.53 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 11:48.16 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:57.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:01.57 | *** join/#asterisk matrix1233 (~matrix123@197.2.13.84) |
| 12:09.14 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 12:13.54 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 12:19.01 | *** join/#asterisk Oatmeal (~Suzeanne@99-103-96-198.lightspeed.iplsin.sbcglobal.net) |
| 12:23.02 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 12:47.49 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
| 12:47.56 | [sr] | hi |
| 12:47.58 | [sr] | hi WIMPy |
| 12:56.12 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:59.35 | *** join/#asterisk boris_t (~boris_t@212.119.243.183) |
| 13:25.43 | WIMPy | Hi [sr]! It's been some days... |
| 13:45.33 | Kunsi | hm, i've got an asterisk server behind NAT (but port forwardings for SIP/SIPS/RTP). it should register to another asterisk instance, connected directly to internet (no nat). what should i set 'nat' setting in sip.conf to? already tried 'yes' and 'force_rport', both get me to call outbound, but 'sip show registry' always shows 'request sent' - sip debug shows me it sends OPTIONS and REGISTER to remote |
| 13:45.36 | Kunsi | asterisk, but i don't see any answers. any tips? maybe firewall related? |
| 13:45.46 | [sr] | WIMPy: yap... been away from life! |
| 13:45.52 | [sr] | WIMPy: how r u? |
| 13:46.31 | WIMPy | Too many things to do, as usual. |
| 13:47.27 | WIMPy | Kunsi: Yiu need to enable nat on the other side, usually. |
| 13:50.22 | Kunsi | other side has nat enabled |
| 13:50.42 | *** join/#asterisk matrix1233 (~matrix123@197.2.13.84) |
| 13:53.37 | Samot | If your box is registering to another system that is not behind NAT, then your peer for that connection should have nat=no |
| 13:55.34 | *** join/#asterisk pawiecki (~pawiecki@router.dir.pl) |
| 13:57.02 | [TK]D-Fender | Kunsi, there are a FEW settings you have to make, not just "nat" |
| 13:57.10 | [TK]D-Fender | Kunsi, And you should be looking at the full request |
| 13:58.37 | Kunsi | wait, i'll nopaste some stuff |
| 13:59.20 | pawiecki | i have NOTICE "Call from '' <Grandstream_GXW4224_IP>:<port> to extension <some_number> rejected[...] not found in context 'default'." And now I don't understand why this show up. First of all, from field is empty. Second of all, GW has no FWD's on any account/port. What am I missing here? |
| 14:00.12 | [TK]D-Fender | Itt's sending a # to you and you don't have a match in your dialplan for it. |
| 14:00.17 | [TK]D-Fender | pawiecki, ^ |
| 14:00.30 | [TK]D-Fender | pawiecki, And you should NEVER be using a context named [default] at all |
| 14:00.50 | Kunsi | sip debug: http://paste.debian.net/857818/ |
| 14:00.54 | Kunsi | sip.conf http://paste.debian.net/857819/ |
| 14:01.05 | [TK]D-Fender | pawiecki, that one is default fallback for other things and should never exist |
| 14:01.57 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-raneicvdthhzgboj) |
| 14:01.57 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:02.11 | pawiecki | [TK]D-Fender: I do not have 'default' context, because of security concerns, like you said. But why would it send # to me? Every registered port on GW has SIP user ID and Auth ID set up correctly. |
| 14:03.09 | [TK]D-Fender | Kunsi, Reliably Transmitting (NAT) to 92.222.104.42:5061: Contact: <sip:3397@172.22.72.25:5061> |
| 14:03.25 | pawiecki | but it shows in field "from" empty value - ''. I don't get that. |
| 14:03.36 | [TK]D-Fender | Kunsi, You are contacting a WAN IP and telling them your return contact isa PRIVATE IP. You have failed to set up your WAN IP and localnets properly in your config |
| 14:04.01 | Kunsi | do i need to set more than localnet and externip? |
| 14:04.43 | [TK]D-Fender | Prevent re-invites as well... |
| 14:04.46 | [TK]D-Fender | and "nat" |
| 14:04.56 | [TK]D-Fender | 4 setttings right there |
| 14:05.03 | [TK]D-Fender | then there are the ones related to your peer |
| 14:05.05 | [TK]D-Fender | (s) |
| 14:05.18 | [TK]D-Fender | to make sure * know what it can and cannot trust |
| 14:05.33 | [TK]D-Fender | pawiecki, go look what you put in your GW |
| 14:05.59 | Kunsi | you mean, 'canreinvite=no', and 'nat'-setting, yes? |
| 14:06.30 | Kunsi | (first in global template, second in peer config) |
| 14:07.48 | [TK]D-Fender | directmedia <- |
| 14:08.04 | [TK]D-Fender | "canreinvite" was phased out in 1.6 over half a decade ago |
| 14:08.13 | Kunsi | oh |
| 14:21.26 | *** join/#asterisk puzzled_ (~puzzled@2001:982:1097:1::1:5) |
| 14:25.57 | Samot | Have there been any changes to AMI in recent updates? |
| 14:26.14 | Kunsi | hm, i think i need to debug my firewall |
| 14:27.03 | Kunsi | asterisk now sends REGISTER with correect ip set, but i don't recieve any answers |
| 14:28.06 | Samot | I'm having an issue where I see the AMI user login successfully but the following Originate command is returned "permission denied" but I don't see the Originate command in the debug or logs at all. |
| 14:28.17 | Samot | Then I see a successful logout. |
| 14:28.23 | Kunsi | ⦠but tcpdump shows data coming in on udp 5060 ⦠strange |
| 14:28.44 | Samot | Just the Originate in the middle, I never see it on the server but the remote side is getting the XML error response. |
| 14:30.54 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 14:31.05 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 14:33.48 | WIMPy | Samot: What are you using? AMI doesn't do XML. |
| 14:34.05 | Samot | Sorry, let me rephrase. |
| 14:34.31 | WIMPy | And you know that AMI users have permissions? |
| 14:34.35 | Samot | Using the mini-HTTP server, amxml |
| 14:34.43 | Samot | Full permissions. |
| 14:35.09 | Samot | https://www.irccloud.com/pastebin/2fYSi4Qc/ |
| 14:35.18 | Kunsi | hmm, think i found the error ⦠asterisk doesn't listen on 5061 tcp |
| 14:35.52 | Samot | Kunsi: It will if you tell it to. |
| 14:36.06 | Samot | https://www.irccloud.com/pastebin/MMGxnQha/ |
| 14:36.15 | Samot | Originate permissions allowed. |
| 14:36.39 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 14:37.13 | Samot | I've got an Asterisk 13.7 box doing this just fine. I setup a new box, did all the configuration and settings the same... |
| 14:37.31 | Samot | And the Originate command portion of it is the only thing I'm having an issue with. |
| 14:38.54 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 14:38.58 | Samot | I don't even see the HTTP request for it.. |
| 14:39.20 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 14:39.22 | Samot | But we get back the AMI response in XML format. |
| 14:39.31 | Samot | As we should. |
| 14:42.02 | Samot | So I'm wondering.. |
| 14:43.22 | Samot | Would the permission denied come back if the session couldn't be related to the session created during the login? |
| 14:43.57 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 14:46.02 | Samot | Or if perhaps any old deprecated functions/commands might have finally been removed. This script they use is kinda old so it may need to be reworked. |
| 14:46.39 | Samot | Because I can initiate a call by going to the web server url in a browser via the amxml and originate a call. |
| 14:46.44 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 14:47.42 | *** join/#asterisk matrix1233 (~matrix123@197.2.13.84) |
| 15:00.42 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 15:04.47 | scv | hm |
| 15:04.54 | scv | definitely encountering some leak |
| 15:07.18 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-emkhoghxwmbhdqel) |
| 15:11.39 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-mllaruxtaxylxjsj) |
| 15:11.52 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ixyadnecghwveanb) |
| 15:13.30 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 15:20.59 | *** join/#asterisk nunne (sid38499@gateway/web/irccloud.com/x-nlrapscbpyxpawyy) |
| 15:28.53 | *** join/#asterisk chadxz (cmcelligot@nat/digium/x-qekvqpjjdthlnepu) |
| 15:30.40 | *** part/#asterisk chadxz (cmcelligot@nat/digium/x-qekvqpjjdthlnepu) |
| 15:36.02 | *** join/#asterisk dan_j (sid21651@gateway/web/irccloud.com/x-xyjoudiilmjcgayw) |
| 15:36.13 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-itrpxarpzficvdkb) |
| 15:38.55 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 15:43.41 | *** join/#asterisk gregs (sid160074@gateway/web/irccloud.com/x-sygzqjesggvmuymm) |
| 15:52.35 | scv | sigh |
| 15:52.50 | scv | rebuild with MALLOC_DEBUG and now asterisk crashes on startup |
| 15:52.53 | scv | just my luck -_- |
| 15:53.49 | scv | <PROTECTED> |
| 15:53.49 | scv | Segmentation fault |
| 15:53.51 | scv | lovely |
| 15:58.13 | *** join/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 16:12.09 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 16:30.00 | *** join/#asterisk axisys (~axisys@unaffiliated/axisys) |
| 16:31.47 | *** join/#asterisk Oatmeal (~Suzeanne@99-103-96-198.lightspeed.iplsin.sbcglobal.net) |
| 16:45.48 | *** join/#asterisk ChkDigit (~u388mw@208.77.142.15) |
| 16:57.16 | scv | fun times, some areas seem to be mixing ast_* and regular allocator functions |
| 16:57.28 | scv | maybe that's relevant to why its crashing |
| 16:57.48 | *** join/#asterisk matrix1233 (~matrix123@197.2.13.84) |
| 17:02.21 | scv | yep that was it, got it building and reporting memory properly with patches |
| 17:03.20 | *** join/#asterisk ctcx (~user@unaffiliated/ctcx) |
| 17:06.39 | ctcx | Does * has default enabled presence and SIP SIMPLE? |
| 17:06.46 | ctcx | Using asterisk 11 |
| 17:12.02 | ctcx | [TK]D-Fender, robink: Does * has default enabled presence and SIP SIMPLE? |
| 17:12.34 | [TK]D-Fender | There is no "default", and * is not a presense server |
| 17:12.53 | [TK]D-Fender | it does not accept it from devices, itt only broadcast's its own usage |
| 17:13.20 | [TK]D-Fender | SIP messaging goes through dialplan apps @ functions which are documented on the WIKI |
| 17:18.45 | Samot | You need to install a presence server if you're looking for that feature. |
| 17:19.16 | Samot | As for SIP SIMPLE, you want it just configure it. WIKI covers it, like TK said. |
| 17:20.10 | ctcx | Ok, this time I'll try better to search. |
| 17:20.21 | ctcx | Before annoying mr TK again. |
| 17:21.57 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:22.35 | ctcx | Is this right https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5242984 ? Because, I thought SMS as such was a slightly different thing... |
| 17:23.13 | ctcx | Different than sip message since SMS forcibly implies a mobile carrier paid service |
| 17:27.06 | [TK]D-Fender | * is not a "SIP server" is is a telephony engine across multiple protocols |
| 17:27.15 | [TK]D-Fender | many apps server more than 1 technology |
| 17:27.18 | [TK]D-Fender | serve* |
| 17:28.03 | igcewieling | ctcx: Asterisk's app_sms expects to work with modems and carriers which support sms. If you are referring to the generic term for "SMS" aka "messaging" that would be handled outside of Asterisk |
| 17:28.52 | igcewieling | I'm not aware of any carriers in the USA which support SMS in the way app_sms expects. |
| 17:30.10 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend?src=search |
| 17:30.41 | ctcx | Already had found, reading... |
| 17:32.22 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:33.17 | Kunsi | hm, how can i set asterisk log directory? logger.conf only specifies file, not path to file |
| 17:34.07 | [TK]D-Fender | asterisk.conf <- |
| 17:36.07 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
| 17:36.54 | Kunsi | ah, i see |
| 17:39.29 | ctcx | I need to learn to also read the comments as well. |
| 17:39.40 | ctcx | I only focused in article contents. |
| 17:40.18 | ctcx | [TK]D-Fender: I'm sorry for being disturbing; I'm slow learner, but I'm seriously working on that. |
| 17:40.58 | ctcx | Just a last doubt regarding this. By chance, auth_message_requests default value is "no"? |
| 17:41.13 | ctcx | I.e., if this directive is not specified. |
| 17:41.26 | igcewieling | ctcx: the sip.conf.sample might tell you. |
| 17:42.09 | ctcx | AAahhh, ok. Thanks. |
| 17:43.41 | ctcx | With these settings done, is it now just a matter of using a client supporting SIP SIMPLE messages? Such as (I'm guessing) 3cx, pidgin, zoiper... |
| 17:44.27 | [TK]D-Fender | * will process them as your dialplan tells it to. |
| 17:45.30 | ctcx | So it depends how I set the dialplan file... |
| 17:47.42 | *** join/#asterisk u0m3 (~u0m3@5-12-199-171.residential.rdsnet.ro) |
| 17:52.02 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 18:00.30 | [TK]D-Fender | There is a function& applications for both sending and parsing out received messages. Thre are very clear sample of thisto dreceive |
| 18:00.34 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 18:00.35 | [TK]D-Fender | All processing = dialplan |
| 18:02.14 | ctcx | igcewieling: I don't have any sip.conf.sample file... |
| 18:02.36 | igcewieling | ctcx: then go download the asterisk source and get one. |
| 18:02.53 | igcewieling | the sample config files are a very important part of Asterisk's documentation |
| 18:03.01 | ctcx | !!!???? |
| 18:03.06 | ctcx | So it should be there!! |
| 18:03.09 | ctcx | WTH... |
| 18:03.16 | ctcx | I have problems. |
| 18:03.17 | igcewieling | ctcx: unless you did something silly and install from an RPM |
| 18:03.30 | ctcx | igcewieling: I'm using Elastix distro. |
| 18:03.37 | igcewieling | ctcx: sucks to be you. |
| 18:03.44 | ctcx | Why? |
| 18:03.58 | igcewieling | because it can't be supported here. |
| 18:04.02 | igcewieling | ~freepbx |
| 18:04.03 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:04.19 | igcewieling | that also applies to Elastix. |
| 18:04.28 | WIMPy | ~elastix |
| 18:04.28 | infobot | well, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 18:05.14 | igcewieling | and as you now know, you are missing a lot of important information by not having the .sample files. I refer to them at least a few times a week to see documentation for a setting or to found out that the default is for a setting. |
| 18:05.21 | ctcx | Ok, I apologize to everybody here, but I think it's also a misunderstanding. I just mentioned that, but *I'm not asking for support for that*. |
| 18:05.37 | ctcx | Though Asterisk is also included in Elastix. |
| 18:05.42 | ctcx | But anyway. |
| 18:06.01 | WIMPy | I don't think they'd even support elastix in #freepbx. |
| 18:06.11 | ctcx | Elastix entire community sucks anyway. |
| 18:06.25 | igcewieling | ctcx: get the asterisk source, even if you don't build asterisk from it. |
| 18:06.28 | [TK]D-Fender | <ctcx> So it should be there!! <- go download the * tarball |
| 18:06.30 | ctcx | But again, I apologize and NOT asking for support for that. |
| 18:06.38 | [TK]D-Fender | xxthis has nothing to do with what some distro chooses to include |
| 18:06.55 | [TK]D-Fender | ctcx, this has nothing to do with what some distro chooses to include |
| 18:06.57 | ctcx | And yes, I'm just downloading the source. |
| 18:06.59 | igcewieling | ctcx: you sort of are. You need help because your distro didn't include the sample configs. |
| 18:07.15 | [TK]D-Fender | Just go download it |
| 18:07.25 | ctcx | I'm doing it right now. |
| 18:07.40 | [TK]D-Fender | not having already doesn't make downloading a tarball and viewing now hard |
| 18:07.53 | scv | boo |
| 18:08.22 | ctcx | I know, I know, I already understood. |
| 18:08.22 | *** join/#asterisk Mango45 (~Mango45@d142-59-245-8.abhsia.telus.net) |
| 18:08.32 | [TK]D-Fender | last one wasn't for you.... |
| 18:09.35 | ctcx | I meant yours sir, not scv's |
| 18:10.26 | ctcx | Anyway, I was lent this test server, so probably owner deleted the file accidentally or the like... |
| 18:10.39 | [TK]D-Fender | No, distro's don't come witth the samples |
| 18:11.00 | [TK]D-Fender | They don't expect you to configure anything yourself directly, that's the entire point of GUI's |
| 18:11.38 | ctcx | Mm... |
| 18:11.43 | ctcx | ok, will take it into account. |
| 18:11.49 | Mango45 | I have a directrtpsetup question if anyone has time. |
| 18:11.52 | Mango45 | Call path is carrier -> Asterisk, no NAT -> Asterisk, NAT -> Phone, same NAT |
| 18:11.57 | Mango45 | Currently the Asterisk behind NAT proxies audio. Can I get it get it to go straight to the phone? I have directrtpsetup=yes and directmedia=yes set for all peers. |
| 18:12.47 | [TK]D-Fender | Phone is the weakest link, and you have the RTP ports FORWARDED to your NAT'd *. That spells likely failure |
| 18:13.29 | Mango45 | I do not have any RTP ports forwarded, and no SIP ALG. |
| 18:13.55 | Mango45 | "Phone is weakest link", so if the phone could figure out the public IP, would audio go direct to it? |
| 18:14.13 | [TK]D-Fender | clarify your sertup for that 2nd * |
| 18:15.06 | Mango45 | The 2nd Asterisk and the phones are all behind the same router, with no ports forwarded. All (relevant) peers have directrtpsetup=yes and directmedia=yes. The 2nd Asterisk knows its public IP via externhost. |
| 18:15.29 | [TK]D-Fender | that 2nd * should have RTP as well as SIP forwarded to it like normal |
| 18:15.51 | Mango45 | That wouldn't make audio go direct to the phone though, would it? |
| 18:16.04 | Mango45 | No SIP forward required due to registration. |
| 18:16.19 | [TK]D-Fender | No, it wouldn't, but that's what you are normally required to do just for that * to work properly |
| 18:17.14 | Mango45 | I assume RTP works due to * sending RTP, and the router then allowing it to come back in the other direction. |
| 18:17.51 | ctcx | ";auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests." "End to end security" ..... I guess like SSL/TLS? |
| 18:18.14 | [TK]D-Fender | ctcx, Whatever auth that peer is defined for |
| 18:18.27 | [TK]D-Fender | at a minimum that means it'll challenge with the secret, etc |
| 18:18.44 | *** join/#asterisk viro (~viro@unaffiliated/viro) |
| 18:21.37 | igcewieling | Mango45: don't expect to make directmedia work with NAT is involved. |
| 18:25.07 | Mango45 | igcewieling: Difficult or completely impossible? |
| 18:25.27 | ctcx | Doc says auth_message_requests is by default enabled. Does it mean not including this line at all will take it as "yes"? |
| 18:26.22 | igcewieling | Mango45: I've never heard of anyone having success with it when using NAT. |
| 18:26.52 | igcewieling | ctcx: correct |
| 18:27.13 | ctcx | Ah, so it does have a "default value". |
| 18:27.57 | Mango45 | Oh well, I can't imagine * adds much latency. |
| 18:32.17 | [TK]D-Fender | "Doc says auth_message_requests is by default enabled." <- sorta says it rightt there |
| 18:34.25 | Mango45 | omg I'm an idiot |
| 18:34.46 | Mango45 | Set(__DYNAMIC_FEATURES=Transfer6#Transfer7) |
| 18:34.51 | Mango45 | That would be why direct audio wasn't working |
| 18:39.24 | Mango45 | Now it works...direct to the caller! Not even the carrier is proxying audio! |
| 18:45.14 | igcewieling | Mango45: In *both* directions? |
| 18:45.58 | Mango45 | Yes sir. |
| 18:46.33 | *** join/#asterisk matrix1233 (~matrix123@197.0.236.98) |
| 18:49.47 | igcewieling | any thing interesting happening when you do this in the CLI: rtp set debug on |
| 18:49.54 | Mango45 | Nothing at all. |
| 18:49.58 | igcewieling | impressive. |
| 18:50.23 | Mango45 | SIP SHOW CHANNEL [channel id] confirms the audio IP is of the terminating carrier. |
| 19:02.30 | igcewieling | Mango45: which version of Asterisk? |
| 19:09.38 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:33.10 | *** join/#asterisk viro (~viro@unaffiliated/viro) |
| 19:55.15 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 19:57.42 | *** join/#asterisk matrix1233 (~matrix123@41.230.68.28) |
| 20:02.50 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 20:19.14 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 20:20.12 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:20.12 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:22.06 | *** join/#asterisk matrix1233 (~matrix123@41.230.68.28) |
| 20:49.15 | *** join/#asterisk monsterco (~monsterco@TOROON474AW-LP130-02-1177735178.dsl.bell.ca) |
| 20:52.00 | *** part/#asterisk monsterco (~monsterco@TOROON474AW-LP130-02-1177735178.dsl.bell.ca) |
| 20:56.20 | *** join/#asterisk viro (~viro@unaffiliated/viro) |
| 20:57.23 | *** join/#asterisk matrix1233 (~matrix123@41.230.68.28) |
| 21:00.05 | *** join/#asterisk axisys (~axisys@unaffiliated/axisys) |
| 21:33.20 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 21:39.31 | *** join/#asterisk NirS (~Nir_s@87.69.126.26) |
| 22:01.12 | Mango45 | igcewieling: Sorry for the delay. 11.7.0. |
| 22:12.22 | *** join/#asterisk yulius (~yulius@201.170.106.136.dsl.dyn.telnor.net) |
| 22:21.31 | *** join/#asterisk ctcx (~user@unaffiliated/ctcx) |
| 22:21.46 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 22:25.00 | ctcx | In the sip.conf file, does # mean a working line instead of a comment? |
| 22:25.09 | ctcx | Or is it just another comment as well? |
| 22:25.25 | Samot | Include. |
| 22:25.53 | Samot | #extensions_custom.conf <-- inside extensions.conf would include that file and it's dialplan. |
| 22:28.12 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 22:28.13 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 22:29.07 | ctcx | Aaahh. |
| 22:30.33 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:30.38 | [TK]D-Fender | Samot, close |
| 22:31.02 | ctcx | Anyway, for all editings I should not use sip.conf nor extensions.conf, and instead use the _custom files, right? |
| 22:31.22 | ctcx | (I tried installing the freepbx module, but I guess it's not working as I was expecting...) |
| 22:37.24 | [TK]D-Fender | "the module"? |
| 22:37.27 | [TK]D-Fender | Which module? |
| 22:37.51 | *** join/#asterisk viro (~viro@unaffiliated/viro) |
| 22:43.27 | ctcx | [TK]D-Fender http://wiki.freepbx.org/pages/viewpage.action?pageId=1048598 |
| 22:44.09 | ctcx | But I cannot edit the direct files because now they say "do not edit these auto generated by freepbx" |
| 22:45.26 | [TK]D-Fender | that is general install instructions, not a "module" |
| 22:45.38 | [TK]D-Fender | And indeed you are not supposed to modify the base files |
| 22:45.51 | [TK]D-Fender | You need to learn what FP generates and what you can hook into |
| 22:46.02 | [TK]D-Fender | And that is normally only in the _custom files |
| 22:46.20 | ctcx | Ah, ok. |
| 22:49.21 | ctcx | I'm trying to do this https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend (the first comment), but after a long google search I cannot tell difference between "sip_custom.conf" and "sip_general_custom.conf", nor tell which one to use... |
| 22:50.40 | ctcx | Could someone help a bit? |
| 22:56.56 | [TK]D-Fender | Those file names mean nothing. |
| 22:56.59 | [TK]D-Fender | what's IN them does. |
| 22:57.07 | *** join/#asterisk celer (~celer@pc1.robertwaltersjp-unet.ocn.ne.jp) |
| 22:57.16 | [TK]D-Fender | You need to understand how the base configs actually work |
| 22:57.34 | [TK]D-Fender | And you don't sue EITHER for that |
| 22:57.38 | [TK]D-Fender | use* |
| 22:59.21 | [TK]D-Fender | MessageSend is a DIALPLAN application. |
| 22:59.24 | [TK]D-Fender | that is extensions.conf |
| 22:59.29 | [TK]D-Fender | not sip.cofn |
| 23:01.06 | ctcx | [TK]D-Fender: er... sir, the comment I pointed to does clearly mention that |
| 23:01.33 | Samot | Well you do need sip.conf |
| 23:01.43 | [TK]D-Fender | <PROTECTED> |
| 23:01.51 | [TK]D-Fender | You put that in EXTENSIONS.CONF, not SIP.CONF |
| 23:01.59 | ctcx | Obviously I was not meaning the MessageSend command, I meant the first 2 lines |
| 23:02.00 | [TK]D-Fender | You neeed to learn the very basics of * first |
| 23:02.23 | [TK]D-Fender | From that comment, then yes |
| 23:02.39 | ctcx | And I have been doing all day along. |
| 23:02.43 | Samot | You have to have accept_outofcall_message=yes and outofcall_message_context=<context> |
| 23:02.54 | ctcx | ^ |
| 23:02.56 | ctcx | Yes, yes |
| 23:03.00 | Samot | To call on the context in the dialplan. |
| 23:03.23 | ctcx | But those 2 are not in extensions.conf because they're not dialplan |
| 23:03.31 | ctcx | They are for sip.conf |
| 23:03.41 | Samot | You set them in the peer configuration. |
| 23:03.41 | ctcx | Except I cannot use directly sip.conf |
| 23:03.53 | Samot | That's accepting the outofcall messages. |
| 23:03.59 | Samot | That's how you get it in there. |
| 23:04.22 | ctcx | Oh, no.... |
| 23:04.24 | Samot | You can add them into the general config or by peer. |
| 23:05.02 | ctcx | Aaahhh, if choosing to add them to config file, should it be the sip_general_custom one instead of the sip_custom one indeed? |
| 23:05.14 | Samot | If you want them in your general Chan_SIP settings you need to use the Extra Settings option in the Chan_SIP settings page. |
| 23:06.07 | Samot | ctcx: You can either add it to the GENERAL settings, which will apply to all peers. |
| 23:06.16 | Samot | Or you can add it to the peer/trunk that will be accepting the calls. |
| 23:06.26 | ctcx | Mm..... |
| 23:06.28 | ctcx | thanks. |
| 23:06.30 | Samot | It's an inbound message it needs a context to handle it in the dialplan. |
| 23:09.27 | ctcx | Samot, [TK]D-Fender: by any chance, and if I ever manage to make messages work, does * keep all message text history somewhere? |
| 23:10.10 | [TK]D-Fender | not unless you put it somewhere |
| 23:10.26 | [TK]D-Fender | It gets sent to the dialplan. Everything that happens is your job |
| 23:12.06 | Samot | Right. |
| 23:12.43 | Samot | You can save the information to a csv file, database, your choice. |
| 23:15.24 | ctcx | So if I'm chatting with another SIP user through some client, all the messages are being "stored" in the dialpan? |
| 23:17.27 | [TK]D-Fender | no |
| 23:17.38 | [TK]D-Fender | Ther is no storage that you don't CODE |
| 23:17.49 | [TK]D-Fender | You need to stop trying to fly and learn how to walk |
| 23:18.19 | [TK]D-Fender | Dialplan != storage. Dialplan = CALL PROCESSING PROGRAMMING INSTRUCTIONS |
| 23:18.25 | [TK]D-Fender | EXTENSIONS.CONF <- |
| 23:18.32 | [TK]D-Fender | Go read the book |
| 23:20.02 | ctcx | So I'd need to code an entire thing for message to store somewhere... *specially* if a message is sent to an offline user for that user to receive it once logging in. |
| 23:20.43 | [TK]D-Fender | Not "specifically". Everything. Every singe step. |
| 23:21.22 | [TK]D-Fender | And there is never any proff that a user actually got the message |
| 23:24.20 | ctcx | Guess I cannot (or would turn impractical) use *'s SIP SIMPLE as full replacement for a chat server... |
| 23:24.33 | ctcx | Oh, no, I'd have indeed to rely on Openfire..... |
| 23:26.08 | ctcx | My great problem with this, just for the record, would be **duplicated users**!!! |
| 23:26.45 | ctcx | 2 different user databases, each one with exactly the same users. |
| 23:27.44 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 23:27.44 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 23:28.49 | ctcx | Because if trying to send a text message through SIP with * to an offline user I guess message is simply lost, right? |
| 23:29.24 | [TK]D-Fender | Yo never know if they DO get it |
| 23:29.29 | [TK]D-Fender | Forget about "offline" |
| 23:29.39 | [TK]D-Fender | "SUCCESS - Successfully passed on to the protocol handler, but delivery has not necessarily been guaranteed." <- See this? |
| 23:29.46 | [TK]D-Fender | Even SUCCESS doesn't mean anything |
| 23:30.04 | [TK]D-Fender | Yay, an IP packet was SENT. |
| 23:30.11 | [TK]D-Fender | Recieved? Who knows |
| 23:31.14 | ctcx | What happens to the message/IP packet if receiving was not successful? |
| 23:31.50 | [TK]D-Fender | You don't seem to be listening |
| 23:31.53 | [TK]D-Fender | YOU NEVER KNOW. |
| 23:32.21 | [TK]D-Fender | YOU. WILL. NEVER. KNOW. THAT. THEY. GOT. THE. PACKET. |
| 23:32.28 | [TK]D-Fender | Is it becoming clearer? |
| 23:34.42 | ctcx | Yes, sorry. |
| 23:35.00 | ctcx | Guess I wanted to do impossible: |
| 23:35.00 | ctcx | Guess I cannot (or would turn impractical) use *'s SIP SIMPLE as full replacement for a chat server... |
| 23:35.12 | ctcx | For the reasons described above. |
| 23:35.29 | ctcx | I'd need to think in more solutions... if there are. |
| 23:35.36 | ctcx | Thanks very much for your help, and patience. |
| 23:35.54 | ctcx | I know I already used it all out. |