| 00:09.25 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-45-30-44.hsd1.in.comcast.net) |
| 00:10.34 | *** join/#asterisk lorsungcu (sid65806@gateway/web/irccloud.com/x-panmetfgvlgqryok) |
| 00:11.00 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-qjrudxdaijtbhhmt) |
| 00:21.12 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:21.12 | *** 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 |
| 00:29.58 | *** join/#asterisk scgm11_ (~scgm11@r186-52-183-212.dialup.adsl.anteldata.net.uy) |
| 00:31.08 | *** join/#asterisk Iamnacho (~Iamnacho@70.171.163.5) |
| 00:40.34 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:59.24 | *** join/#asterisk Kunsi (felix@unaffiliated/kunsi) |
| 01:41.09 | *** join/#asterisk [\\\] (~triplesla@unaffiliated/imsaguy) |
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| 02:20.17 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 02:29.29 | snadge | can anyone cast their mind back to the asterisk 1.6 days, and recall if there were issues with dnsmgr_lookup taking longer than it should? |
| 02:29.56 | snadge | i've checked general dns performance seems to be okay.. queries coming back in 0-4ms |
| 02:30.52 | snadge | forgot to upgrade this box to 11 like the rest of them.. even though 11 is equally out of date ;) |
| 02:32.13 | [TK]D-Fender | I recall all sorts of DNS issues |
| 02:32.19 | [TK]D-Fender | where it was always best to have a local server |
| 02:33.04 | snadge | yeah i could probably just put dnsmasq on it and redirect queries to local, as a bandaid if need be |
| 02:33.47 | [TK]D-Fender | or.... you could jsut upgrade |
| 02:33.56 | [TK]D-Fender | because that's woefully out of date |
| 02:34.08 | snadge | yeah thats the plan.. outside of hours |
| 02:34.22 | snadge | im not doing it whilst people are using it :P |
| 02:35.23 | snadge | 310 active sip peers.. 12 calls.. thats nothing |
| 02:37.29 | snadge | if i update /etc/resolv.conf i dont have to do anything to asterisk do i? |
| 02:37.39 | carrar | fail them over to your backup server |
| 02:37.47 | snadge | you're funny carrar ;) |
| 02:37.59 | carrar | heh |
| 02:38.19 | carrar | sad, but funny yes |
| 02:38.46 | snadge | we have daily backups, snapshot of the entire vm.. but thats not exactly convenient to fail people over to... no standby server or anything sensible like that |
| 02:39.23 | carrar | should test that |
| 02:39.24 | snadge | i guess i could make one.. but its probably just easier to wait until people aren't using it in about 5 hours |
| 02:39.37 | snadge | and recompile asterisk |
| 02:41.32 | snadge | after all, this pbx has been in production with this configuration.. for.. gee.. i dont know.. the oldest file timestamp i can see is.. Jan 13 2000 |
| 02:41.47 | snadge | so whats that.. 17 years? ;) |
| 02:42.08 | snadge | im sure it can wait another 5 hours |
| 02:45.31 | snadge | ok.. on a redhat/centos based system you can use "rpm -qi basesystem" .. which reveals Fri 25 Mar 2011.. that seems more reasonable |
| 03:55.24 | *** part/#asterisk DexDeadly (~DexDeadly@pool-71-175-51-23.phlapa.fios.verizon.net) |
| 04:20.01 | wyoung | snadge: sudo apt-get install bind9 |
| 04:20.27 | wyoung | oh wait, you are using an old version of asterisk, that must mean you are using Redhat or Centos |
| 04:21.14 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 04:52.29 | *** join/#asterisk DexDeadly (~DexDeadly@pool-71-175-51-23.phlapa.fios.verizon.net) |
| 04:54.15 | DexDeadly | Hey all, so I have installed the chan-sccp plugin to get my Cisco 7970 to work with BLFs, etc. It works great. however playing around I saw that doing a sccp message device the screen only shows 4 characters. I have the download of the module and want to take a stab at finding out why its cutting it off. The bottom part its displaying clearly can display longer as it is the same section |
| 04:54.16 | DexDeadly | of the screen where other messages are displayed. My question would be where would these files located once you do a make install? |
| 05:33.14 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 05:41.47 | *** join/#asterisk bl3nto (~bl3nto@78.134.210.254) |
| 05:47.40 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 05:48.16 | Dirk23 | [TK]D-Fender: good morning. Is there a good tutorial for connecting my Peers (SIP Client and Speaker) to my finaly running Asterisk? |
| 05:48.35 | [TK]D-Fender | Every guide |
| 05:48.41 | Samot | Dirk23: It's all over the place |
| 05:48.41 | [TK]D-Fender | HowSIP is SIP |
| 05:48.43 | Samot | We told you this |
| 05:49.01 | [TK]D-Fender | ALL the same thing |
| 05:49.16 | Samot | Connecting a SIP client to Asterisk has been something that's been happening for almost 20 years. |
| 05:49.23 | Samot | It honestly hasn't changed that much. |
| 05:50.43 | Dirk23 | ok.... i have NO Experience and i dont know what to do at all in those configs. In the asterisk wiki they tell me to backup the old sip.conf and create a empty one. Yesterday you (Samot) where laughting about that almost empty sip.conf. |
| 05:50.51 | Dirk23 | so what was wrong with it? |
| 05:52.18 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 05:52.56 | Samot | Have you even attempted to register the device? |
| 05:53.40 | Dirk23 | i havent tried, i wanted to test it today. So shall i use the sip.conf from tutorial? |
| 05:53.50 | [TK]D-Fender | go do it |
| 05:54.03 | [TK]D-Fender | why are you asking to even try? |
| 05:54.47 | drmessano | I think he wants permission to set up Asterisk |
| 05:54.50 | drmessano | Granted |
| 05:54.56 | Dirk23 | because you confused me yesterday and i thougth i made everything wronfg |
| 05:54.57 | carrar | proceed |
| 05:55.05 | Dirk23 | drmessano: no |
| 05:55.12 | Dirk23 | carrar: thnx |
| 05:55.54 | carrar | heh |
| 05:56.05 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 05:56.09 | drmessano | np |
| 05:56.36 | drmessano | Please let me know if you need a 30 day extension |
| 05:56.48 | Dirk23 | do i need to reload the configs somehow after changing them? |
| 05:56.54 | [TK]D-Fender | yes |
| 05:57.07 | [TK]D-Fender | "reload" for most at once |
| 05:57.27 | Dirk23 | ok, fine. How do i reload them? |
| 05:57.40 | [TK]D-Fender | just1looking4u2 |
| 05:57.43 | [TK]D-Fender | ug |
| 05:57.47 | [TK]D-Fender | wtf |
| 05:57.57 | [TK]D-Fender | <[TK]D-Fender> "reload" for most at once |
| 05:58.30 | Dirk23 | <Dirk23> ok, fine. How do i reload them? |
| 05:59.45 | Samot | What is this for? |
| 05:59.45 | Dirk23 | do i need to get into the asterisk cli, do i need to restar the service? |
| 05:59.47 | [TK]D-Fender | "reload" <- THE FUCKING THING IN QUOTES |
| 05:59.53 | drmessano | "reload" |
| 05:59.54 | [TK]D-Fender | YOU TYPE THE FUCKING WORD. |
| 06:00.05 | drmessano | reload <press the enter key> |
| 06:00.06 | [TK]D-Fender | YES * CLI |
| 06:00.09 | Dirk23 | ah! |
| 06:00.13 | Samot | I want to know what this is for... |
| 06:00.20 | Samot | Is this some commercial deployment? |
| 06:01.02 | Samot | Or did you just think it would be cool to talk to your friend over a ceiling speaker? |
| 06:01.34 | Dirk23 | Samot: its just for fun and to have you employed |
| 06:01.43 | Dirk23 | of course not! |
| 06:01.59 | Samot | What do you men "to have you employed"? |
| 06:02.06 | Samot | What do you mean "to have you employed"? |
| 06:02.17 | drmessano | Dirk23: You know NONE of us talking are paid to be here, right? |
| 06:02.37 | drmessano | We're all community members |
| 06:02.41 | Dirk23 | drmessano: really, i never thougth so |
| 06:02.47 | Samot | What do you mean "to have you employed"? |
| 06:02.49 | [TK]D-Fender | Which means if you are lazy or waste our time ... you are just wasting our time. |
| 06:02.54 | drmessano | ^ that |
| 06:03.04 | Dirk23 | ok, thnx for your community help. |
| 06:03.19 | drmessano | You're not serious, are you? |
| 06:03.27 | Dirk23 | i will not ask anything again. I thought that is what a community is for |
| 06:03.33 | drmessano | Did you honestly think Digium just pays people to IRC all day? |
| 06:03.43 | Dirk23 | drmessano: of course not |
| 06:03.43 | [TK]D-Fender | Well.. they do.. a little bit :) |
| 06:03.49 | [TK]D-Fender | anyway... go try |
| 06:03.54 | Samot | Dirk23: Don't even go there. Not after you wasted over an hour of my time last night. |
| 06:04.09 | Samot | Of me trying to help you. |
| 06:04.10 | Dirk23 | sip client connected successfully. Now i need to configure my speaker |
| 06:04.18 | [TK]D-Fender | so .. you should have reloaded your configs. |
| 06:04.24 | [TK]D-Fender | where are you at now? |
| 06:04.33 | Samot | Which we told you MULTIPLE times last night. |
| 06:04.45 | Dirk23 | i have a ZOIPER connected to asterisk |
| 06:04.59 | [TK]D-Fender | ok |
| 06:05.44 | Dirk23 | https://pastebin.com/Rh1giVC2 |
| 06:06.03 | Dirk23 | do i configure the number of a phone only in those [] |
| 06:06.05 | [TK]D-Fender | that's 1 peer |
| 06:07.51 | Dirk23 | so, Speaker is connected to |
| 06:08.08 | [TK]D-Fender | to ....? |
| 06:08.16 | Dirk23 | asterisk |
| 06:08.28 | [TK]D-Fender | How is Zoiper connected? |
| 06:08.59 | Dirk23 | what you mean by "how"? I added an SIp Account |
| 06:10.08 | Dirk23 | The wiki now tells me i can make a call now, but i cant |
| 06:10.44 | [TK]D-Fender | You jsut said you have TWO things connected |
| 06:10.50 | [TK]D-Fender | And showed a config with ONE peer <- |
| 06:11.08 | [TK]D-Fender | You should not have to 2 things saying "Hi, I'm JOHN" |
| 06:11.17 | [TK]D-Fender | ONE of them is JOHN, the other ... should be someone else |
| 06:11.36 | Dirk23 | yes, the peer comes second. i need to see how to connect my pbx with a sip account. That will take a lone time to figure that out. I wanted to test call the seaker |
| 06:11.36 | [TK]D-Fender | Also, you've shown only sip.conf so far |
| 06:11.46 | [TK]D-Fender | that doesn't mean you can process ANY call at all. |
| 06:12.02 | [TK]D-Fender | So how are BOTH "connected" if you only have 1 peer? |
| 06:12.02 | Dirk23 | ok, wait |
| 06:12.06 | Dirk23 | nono |
| 06:12.15 | drmessano | Dirk23: You need ONE PEER for the Speaker.. ONE PEER for Zoiper |
| 06:12.19 | [TK]D-Fender | <Dirk23> so, Speaker is connected to |
| 06:12.22 | drmessano | Thats TWO |
| 06:12.25 | [TK]D-Fender | <Dirk23> i have a ZOIPER connected to asterisk |
| 06:12.30 | Dirk23 | https://pastebin.com/MBXvRRx9 |
| 06:12.30 | drmessano | You have ONE |
| 06:12.32 | [TK]D-Fender | CLARITY. Find it fast. |
| 06:12.36 | drmessano | ONE + ONE = TWO |
| 06:12.43 | [TK]D-Fender | better |
| 06:12.51 | Dirk23 | yes, i know |
| 06:12.57 | Dirk23 | but i cant call the speaker |
| 06:12.59 | [TK]D-Fender | Also don't use the same passwords |
| 06:13.05 | Dirk23 | ok |
| 06:13.06 | [TK]D-Fender | things will eventually start fucking up on you |
| 06:13.20 | Samot | Dirk23: Do you have dialplan written to call the speaker? |
| 06:13.33 | [TK]D-Fender | [TK]D-Fender> Also, you've shown only sip.conf so far |
| 06:13.33 | [TK]D-Fender | <[TK]D-Fender> that doesn't mean you can process ANY call at all. |
| 06:13.40 | Dirk23 | no i have no dialplan |
| 06:13.42 | [TK]D-Fender | All call processing = dialplan = extensions.conf |
| 06:13.45 | [TK]D-Fender | then you can't do anything |
| 06:13.50 | Samot | Then you can't process calls. |
| 06:14.12 | [TK]D-Fender | Every single step your call is expected to take = dialplan |
| 06:14.42 | Dirk23 | ok |
| 06:14.48 | [TK]D-Fender | "You're John and you want to dial 12345"? Well you'd better have that defined with the steps you expect it to take |
| 06:15.06 | [TK]D-Fender | dialpan = 90% of Asterisk. |
| 06:15.10 | Dirk23 | kk |
| 06:58.14 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 06:59.20 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 07:06.39 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 07:12.48 | *** join/#asterisk slima (~slima@unaffiliated/slima) |
| 07:13.11 | slima | Hello Is there any 'virtualhost' funcionality in asterisk 11? |
| 07:19.16 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:19.40 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 07:21.01 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:24.09 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:28.36 | drmessano | virtualhost? |
| 07:28.38 | drmessano | How? |
| 07:29.25 | Dirk23 | Samot: my Peers dont connect anymore. In CLi see: |
| 07:29.26 | Dirk23 | [Apr 28 09:28:45] NOTICE[1315]: chan_sip.c:28377 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 6001 |
| 07:30.29 | Dirk23 | sip.conf https://pastebin.com/g879AUB3 |
| 07:30.59 | Dirk23 | extension.conf: https://pastebin.com/vGK7ewyx |
| 07:31.28 | Dirk23 | whats wrong now? |
| 07:32.24 | Dirk23 | hmm.... Zoiper is connected again now... but Speaker still says timeout |
| 07:32.52 | Dirk23 | ah, now its registered again.... |
| 07:38.25 | *** join/#asterisk jkroon (~jkroon@165.255.161.211) |
| 07:47.06 | Alblasco1702 | Dirk23, <[TK]D-Fender> Also don't use the same passwords. |
| 07:47.32 | Dirk23 | Alblasco1702: they are not the same |
| 07:48.02 | Dirk23 | but thnx. The Problme is solved, my peers are all connected |
| 07:48.02 | Alblasco1702 | Dirk23, i can't see that on your sip.conf |
| 07:48.12 | Dirk23 | Alblasco1702: read carefully |
| 07:48.20 | Dirk23 | the passwords are not the same |
| 07:48.56 | Alblasco1702 | yes i saw it now 1 number is diffrent srry |
| 07:49.10 | Dirk23 | yes |
| 07:49.26 | Dirk23 | thnx |
| 07:54.56 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:06.27 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 08:15.41 | *** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es) |
| 08:36.46 | *** join/#asterisk DanB (~DanB@clt-195.192.205.172.ip-anschluss.net) |
| 09:02.39 | Dirk23 | great.... my PBX Support-Hotline told me that he's not into PBX (because they do routers and PBX) and the ONE colleague who knows Stuff about that PBX, is allway busy. Thats what i call Support.... |
| 09:03.58 | Dirk23 | i added a SIP-Provider in my physical PBX, the SIp Account is created in Asterisk (sip.conf) and my extensions allows all unmbers to call each other. But i cant see my physical PBX trying to connect to Asterisk. |
| 09:06.04 | Dirk23 | my physical PBX can reach Asterisk |
| 09:27.56 | *** join/#asterisk BakaKuna (~BakaKuna@145.129.205.133) |
| 09:53.20 | *** join/#asterisk DanB__ (~DanB@clt-195.192.205.172.ip-anschluss.net) |
| 10:00.22 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 10:12.34 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 10:23.44 | slima | drmessano: like in apache, freeradius etc. |
| 10:23.56 | slima | different configurations, one instalation |
| 10:57.30 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 10:58.47 | *** join/#asterisk pawiecki (~pawiecki@router.dir.pl) |
| 11:06.42 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 11:09.11 | pawiecki | Hi. Not really an Asterisk question, but maybe someone will be able to give me a hint. I have a Grandstream GXW410X PSTN Gateway and Asterisk on Intel NUC. Problem is, that this GW is connected to the PSTN via 4 analog lines. I've setup the GW to use a single sip account for all 4 accounts (lines), so the calls are being routed in and out * with no problem, but I can not get it to pass the caller id for incomming call. |
| 11:37.46 | carrar | heh |
| 11:38.01 | carrar | And how does caller ID work on PSTN analog lines? |
| 11:38.27 | Samot | You generally don't set it |
| 11:38.29 | Samot | The carrier does |
| 11:38.36 | carrar | well depends |
| 11:38.48 | carrar | if it's sent, it's sent between the first and second ring |
| 11:38.55 | carrar | which means |
| 11:39.15 | carrar | You need to wait till the second ring before doing anythign with the call so as to grab the caller ID |
| 11:39.44 | carrar | IF indeed you are getting CallerID from th PSTN and your gateway is passing it |
| 11:47.40 | Samot | Well I guess the first question would be, is the callerid making it to the GS. |
| 11:47.46 | pawiecki | thanks guys. I'm reading about it, as I'm not really experienced in analog lines. It looks like it may be either turned off by the carrier or I have incorrect CID scheme in my GW settings. I'll try to verify that next. |
| 11:50.56 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 11:50.58 | Haris | hello all |
| 11:51.31 | carrar | - There is no such nick all |
| 11:51.44 | Haris | ? |
| 11:52.11 | carrar | Oh look, storm troopers |
| 11:52.25 | carrar | https://www.osburn.com/stormtrooper_in_shibuya.jpg |
| 11:52.26 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:52.55 | Haris | I can't find a setting in MicroSIP for silence suppression. I guess I'll have t let my logs get filled with one liner |
| 11:56.36 | carrar | You just asked that in FreeBPX |
| 11:57.03 | Samot | Yeah, SOP. |
| 11:57.24 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-qxlrwwhbxeshnkfy) |
| 11:57.27 | carrar | the answer will be the same |
| 11:58.35 | forgotmynick | hello |
| 11:58.41 | carrar | HARRO |
| 11:58.44 | forgotmynick | :D |
| 11:58.48 | carrar | How are you? |
| 11:58.57 | forgotmynick | i'm alright thanks how are you |
| 11:59.07 | carrar | I am great, though getting late here |
| 11:59.17 | carrar | any fun plans for the weekend? |
| 11:59.21 | forgotmynick | go to sleep then? after you answer or someone answers my question |
| 11:59.31 | carrar | damn!! |
| 11:59.35 | forgotmynick | yeah im salty |
| 12:01.02 | forgotmynick | so we want to put in around 11 voip phones in our office but we want to run the asterisk server externally. how, if it's possible, could we ensure extensions can ring each other directly/not use the internet? |
| 12:01.26 | Samot | You can't. |
| 12:01.40 | Samot | Unless you call the phones directly on the network. |
| 12:02.01 | Samot | i.e. Bypass Asterisk |
| 12:02.26 | carrar | the RTP could bypass Asterisk |
| 12:02.30 | carrar | but not the SIP |
| 12:02.38 | *** join/#asterisk bl3nto (~bl3nto@78.134.210.254) |
| 12:02.43 | Samot | "directly/not use the Internet" |
| 12:02.46 | forgotmynick | iax? |
| 12:02.49 | Samot | If Asterisk is in the cloud.. |
| 12:02.55 | Samot | That's still INTERNET |
| 12:02.58 | forgotmynick | samot that's just a pissing contest |
| 12:03.21 | Samot | If you PBX is in the cloud... |
| 12:03.28 | carrar | How would a phone know about all the other phones? |
| 12:03.34 | Samot | SIP URI |
| 12:03.42 | carrar | if your phones support that |
| 12:03.44 | Samot | You would have to dial <exten>@<ip> |
| 12:04.02 | Samot | Well he's asking how not to go over the Internet to call other devices on the LAN |
| 12:04.16 | Samot | The B2BAU system (Asterisk) is in the cloud. |
| 12:04.33 | carrar | move the cloud into the office |
| 12:04.39 | Samot | That's the only way |
| 12:04.42 | carrar | hope it doesn't rain |
| 12:04.54 | Samot | To keep full functionality of the PBX during calls. |
| 12:05.20 | Samot | Because Phone <--> Phone means no VM or FollowMe or anything that Asterisk would do to the call. |
| 12:05.26 | carrar | You could run 2 PBX's |
| 12:05.41 | carrar | it always tries the internal to the office PBX first |
| 12:05.51 | Samot | That's more than needed. |
| 12:06.02 | Samot | Either bring the PBX in house |
| 12:06.11 | Samot | Or except your calls will go over the Internet to the PBX and back |
| 12:07.24 | forgotmynick | i understand thanks |
| 12:07.45 | carrar | Should try to remember what your nick is |
| 12:08.14 | forgotmynick | haha we share this account between 20 or so people |
| 12:08.38 | carrar | why? |
| 12:08.55 | forgotmynick | i thought maybe there was a way to redirect extension to extension calls with asterisk saying the IP is 192.168.xxx.yyy |
| 12:09.00 | forgotmynick | because we're salty |
| 12:09.05 | carrar | Access to the internet is only allowed via 1 account? xWTFx |
| 12:09.35 | carrar | but asterisk is in the cloud |
| 12:09.43 | carrar | so to get that answer |
| 12:09.49 | carrar | You'd have to go to the cloud |
| 12:10.18 | forgotmynick | i didn't mean to avoid asterisk all together, i meant to have the extensions eventually communicate directly with each other without transmitting the call through the internet and back |
| 12:10.30 | carrar | scroll back |
| 12:10.36 | carrar | RTP can be direct |
| 12:10.43 | carrar | SIP can't in your setup |
| 12:11.03 | carrar | but that still requires internet access |
| 12:11.10 | carrar | because the SIP sets up the call |
| 12:12.05 | carrar | You might need to read up on how a sip call works |
| 12:12.16 | carrar | that will probably clear things up a bit |
| 12:12.39 | Samot | I don't know where you think the RTP can be directed. |
| 12:12.48 | forgotmynick | http://blog.davidvassallo.me/2013/10/02/enabling-direct-rtp-streams-between-sip-phones-in-asterisk/ |
| 12:12.51 | Samot | His phone is going to send everything to Asterisk. |
| 12:13.00 | forgotmynick | thanks for pointing me in the right direction |
| 12:13.08 | Samot | Asterisk will decide if it needs to stay in the path for the full call or only when media is needed. |
| 12:13.09 | carrar | np |
| 12:13.21 | Samot | Based on directmedia=yes or no |
| 12:13.57 | carrar | samot we're saying the same thing |
| 12:14.13 | carrar | RTP can be direct |
| 12:14.16 | Samot | But it still goes to Asterisk |
| 12:14.19 | carrar | no |
| 12:14.27 | Samot | How does it not? |
| 12:14.27 | carrar | it does not if you configure it not too |
| 12:14.43 | carrar | Asterisk can tell the phones to talked directly to each other |
| 12:14.46 | Samot | How does Phone A know where to send it's RTP? |
| 12:14.49 | carrar | but changing the SDP headers |
| 12:14.49 | Samot | Right |
| 12:15.02 | carrar | AKA the result is what he wants |
| 12:15.07 | Samot | But it still has to go to Asterisk for it to do that |
| 12:15.24 | carrar | the SIP goes through Asterisk |
| 12:15.25 | carrar | yes |
| 12:15.28 | carrar | not the RTP |
| 12:15.47 | carrar | because in the SDP headers the SIP tells the phones where to send the RTP |
| 12:16.02 | carrar | so RTP does not need to be sent to Asterisk |
| 12:16.15 | Samot | Right.. |
| 12:16.24 | Samot | Until the signal says to. |
| 12:16.26 | Samot | Like hold |
| 12:16.33 | Samot | Asterisk puts itself back in the RTP |
| 12:16.38 | carrar | yeah that wasn't the question |
| 12:16.53 | Samot | But it's part of the answer. |
| 12:17.01 | carrar | well it will work |
| 12:17.04 | Samot | "Asterisk will redirect the RTP" |
| 12:17.09 | Samot | ^^ That's not specific. |
| 12:17.11 | carrar | cause if sip doesn't work, the call won't work in the first place |
| 12:17.16 | Samot | It doesn't tell you when or how it does it. |
| 12:17.24 | carrar | but the call will be "direct" |
| 12:17.30 | Samot | Leads to the expectation it never touches it, which isn't true. |
| 12:18.08 | carrar | if you're on hold you're not really having a direct call at that point |
| 12:18.16 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:18.35 | *** join/#asterisk Milos_ (~Milos@pdpc/supporter/student/milos) |
| 12:18.41 | Samot | If I call two endpoints directly |
| 12:18.46 | Samot | And I place the call on hold.. |
| 12:18.58 | Haris | ok |
| 12:19.02 | Samot | The SDP is updated to sendonly or recvonly on the proper devices. |
| 12:19.07 | Samot | There would be no music. |
| 12:19.07 | Haris | can silence suppression be disabled in asterisk ? |
| 12:19.15 | Haris | manually, via config |
| 12:19.17 | Haris | http://forums.asterisk.org/viewtopic.php?p=67178 |
| 12:19.37 | Haris | perhaps it can help |
| 12:19.39 | Samot | Stop man. |
| 12:19.40 | Samot | Stop |
| 12:19.52 | Samot | Stop asking Asterisk based questions when you're running FreePBX |
| 12:20.22 | Haris | Dude, asterisk and freepbx are two things. I haven't asked how to config asterisk via fpbx for this |
| 12:20.29 | Haris | When I do, you can stop me |
| 12:20.43 | Haris | you need to stop from stopping me when I ask asterisk related Qs in here |
| 12:20.49 | forgotmynick | Haris, samot hasn't taken his medication today |
| 12:20.51 | Haris | asterisk != freepbx |
| 12:21.03 | Samot | So this isn't for a FreePBX box? |
| 12:21.09 | Samot | This is for a pure Asterisk box? |
| 12:21.20 | Haris | this is for one. but this is also for my learning of asterisk on boxes where there's no fpbx |
| 12:21.32 | Haris | I don't have fpbx on all my asterisk boxes |
| 12:21.43 | Samot | So why are you asking this question in #freepbx? |
| 12:21.44 | Haris | its on two of them |
| 12:21.58 | Samot | Stop asking questions in two channels at once that don't related to one of those f'ing channels. |
| 12:21.59 | Haris | because I'm not asking if fpbx can configure asterisk for it |
| 12:22.04 | Haris | I'm asking if asterisk can be configured for it |
| 12:22.15 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample |
| 12:22.19 | Samot | ^^^ I thought you READ THE DOCS |
| 12:22.38 | forgotmynick | carrar should we call someone? |
| 12:22.50 | Samot | First off.. |
| 12:23.06 | Samot | Haris has been a help vampire for over a year. |
| 12:23.24 | Haris | this sample file does not show asterisk config for silence suppression |
| 12:23.28 | Samot | Right. |
| 12:23.30 | Samot | Because?! |
| 12:23.33 | Haris | it just has a reference in comments |
| 12:23.41 | carrar | forgotmynick, who would you call? |
| 12:23.43 | Samot | Silence Suppression is done on the DEVICE |
| 12:24.10 | Haris | there was a forum topic which mentioned it may be configurable at asterisk level. That is why I'm asking if asterisk has such a setting |
| 12:24.52 | Samot | From 9 years ago |
| 12:25.15 | Samot | And it claims to use a setting that isn't covered any in docs now. |
| 12:25.20 | Samot | Here's the thing.. |
| 12:25.39 | Samot | Asterisk doesn't remove deprecated settings unless they really really have to be removed. |
| 12:25.55 | Samot | so perhaps 9 years ago that was a setting, I don't recall it, but now it doesn't exist. |
| 12:26.05 | Samot | So it either never was a proper setting or it was completely removed. |
| 12:26.08 | sekil | hello |
| 12:26.10 | Haris | just saying "this setting doesn't exist now" would have been enough |
| 12:26.14 | Haris | hey sekil |
| 12:26.22 | Haris | thank you. I appreciate it |
| 12:26.31 | Haris | please be calm |
| 12:26.43 | file | that option has never existed |
| 12:26.52 | Samot | That's what I thought. |
| 12:27.29 | *** join/#asterisk scgm11_ (~scgm11@r186-52-183-212.dialup.adsl.anteldata.net.uy) |
| 12:27.51 | Haris | "that option has never existed" <--- that much is also k |
| 12:29.18 | sekil | Haris: what do you mean? |
| 12:30.46 | Samot | That was directed at me, I'm pretty sure. |
| 12:33.07 | *** join/#asterisk scgm11_ (~scgm11@186.52.66.9) |
| 12:35.33 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 12:35.38 | sekil | Samot: ah.. |
| 12:37.09 | Samot | We have a bit of history. See I spent almost 4 to 5 months and HOURS of my FREE time to help Haris setup Asterisk for a WebRTC solution that his company could sell to banks. He could never grasp the core concepts needed to do this. Including almost three weeks of trying to help him install Asterisk right. |
| 12:38.24 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-iwunypoxigpaacnf) |
| 12:38.28 | Haris | with respect to this .. the thing is, it came out only at the end .. that fpbx has webrtc client configured only in UCP. it doesn't allow for webrtc to function with other clients .. through its setup of asterisk |
| 12:38.30 | sekil | Samot: right |
| 12:38.53 | Haris | if that had come out earlier or in the beginning, that would have stopped right there at the start |
| 12:38.56 | Samot | Haris: We told you that in the beginning. |
| 12:39.01 | Samot | It did. |
| 12:39.17 | Haris | only when the webrtc fucntionality was not working after all the trying, that bit of info came out |
| 12:39.17 | Samot | What you were ACTUALLY DOING with it didn't come out until after. |
| 12:39.26 | Samot | Either way.. |
| 12:39.52 | Haris | secondly, I never asked anyone to give me a complete tutorial on asking a Q. I only ask for specific stuff |
| 12:39.53 | Samot | I tend to have issue when I use my _free_ time to help someone because they have to make a paycheck on it. |
| 12:40.05 | Samot | You were told numerous times to go learn this stuff |
| 12:40.11 | Samot | To hire someone that could help your directly |
| 12:40.16 | Samot | And here we are a year later. |
| 12:40.16 | Haris | but, I appreciate every bit of time I consumed |
| 12:40.38 | sekil | Haris: not according to your statements though... |
| 12:40.39 | Haris | that project was dropped. so no sales were made |
| 12:40.46 | Samot | Right |
| 12:40.51 | Haris | no commercial activity was ever conducted |
| 12:40.53 | sekil | Haris: please be calm is such a bs sentence.. |
| 12:40.54 | Haris | based on that |
| 12:40.55 | Samot | The project was dropped because _you_ couldn't make it work |
| 12:41.09 | Haris | nope |
| 12:41.14 | Haris | not because of that |
| 12:41.24 | Haris | that was a feature, which was let go, at that point in time |
| 12:41.48 | Haris | but anyway. I'm not sure if any of the support provided here is not being used in commercial activity all across the globe |
| 12:42.03 | Samot | It is. |
| 12:42.13 | Samot | People come in here all the time for help |
| 12:42.25 | file | WebRTC is a fascinating thing, people don't realize how much you truly have to learn to make a solution off of it and support it. |
| 12:42.28 | Samot | Because they are being paid for a service they don't understand how to run or support themselves. |
| 12:42.40 | Samot | @file: Very true |
| 12:42.47 | Samot | Kinda like VoIP. |
| 12:43.19 | file | VoIP is simpler, WebRTC is harder because of everything involved (ICE and DTLS-SRTP), and the black box that is the browser at times |
| 12:43.20 | Haris | they were trying to be a market pioneer in offering a cutting edge service, in another part of the world |
| 12:43.20 | Samot | There is more to selling VoIP/Voice services than having an Asterisk box, a provider and some billing software. |
| 12:43.30 | Haris | %s/service/feature |
| 12:43.57 | Haris | Samot: There's alot to alot of which most of the time. but human beings don't accept that .. at face value |
| 12:44.05 | Haris | especially there, where people try all sorts of things |
| 12:44.07 | Haris | all the time |
| 12:44.34 | sekil | does * support RFC 4579 for conferences nowadays? |
| 12:44.44 | *** join/#asterisk rwb (~Thunderbi@75-150-110-170-NewEngland.hfc.comcastbusiness.net) |
| 12:46.00 | file | no |
| 12:46.50 | stefan27 | Yeah... we have had so many issues with WebRTC and the components that uses it |
| 12:47.37 | stefan27 | With the latest versions of asterisk, most of the issues that remain are on the client side |
| 12:48.23 | stefan27 | It's hard to provide crash-logs, find the cause of bad audio quality, or our latest issue was that chrome's latest Automatic Gain Control feature seemed to bug out for some clients |
| 12:48.49 | stefan27 | (the volume kept increasing through-out a call) |
| 12:49.30 | Samot | Well when there are multiple browser that consume the market shares.. |
| 12:49.44 | Samot | Generally by the same individuals.. |
| 12:50.00 | Samot | And browser basically send updates out constantly.. |
| 12:50.27 | Samot | Running a web service like WebRTC over them tends to be a bit rough to keep up with. |
| 12:50.37 | stefan27 | When letting users install other softphones that support encryption, we can tell them to carry out the testing phase without encryption to be able to trace everything better, but with WebRTC DTLS cant be shut off |
| 12:52.52 | Samot | Yeah.. |
| 12:53.05 | Samot | I followed FreePBX's steps. |
| 12:53.11 | Samot | They make a separate user. |
| 12:53.37 | file | yeah, supporting both under a single endpoint is difficult because the SDP is different between the two |
| 12:53.52 | Samot | Well separate device.. |
| 12:54.01 | Samot | And just route incoming calls to ring both devices. |
| 12:54.07 | file | theoretically we could have two streams in the SDP - but how that would be handled by implementations is a bit ... unknown |
| 12:55.39 | *** join/#asterisk jkroon_ (~jkroon@165.16.204.40) |
| 12:55.45 | stefan27 | I read an interesting white-paper/science article by someone talking about how to test audio quality in webrtc; but that scenario abstracted away the issue with microphone capture, because they had a fixed audio source pumping audio through some artificial peerconnection and then measuring various things related to audio quality. That article had no practical use for me; our biggest problem |
| 12:55.45 | stefan27 | is when someone says "I experience bad audio quality with WebRTC" |
| 12:56.03 | stefan27 | there are like 10 unknowns and no way to get measure points for all of them |
| 12:56.12 | stefan27 | it's the most time consuming trouble shooting sessions ever |
| 12:56.22 | *** join/#asterisk newtonr (~newtonr@173-21-147-197.client.mchsi.com) |
| 12:56.22 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 12:56.23 | file | it's all well and good, until it's not |
| 12:56.32 | Samot | Honestly, I rarely get WebRTC requests. |
| 12:57.24 | stefan27 | All we can do is treat it like black-boxes and tell em like "Try a different headset", browser, PC or ask if the problem is exclusive to some call directions or destinations but it's always a puzzle |
| 12:57.57 | Samot | It's like a softphone. |
| 12:58.00 | stefan27 | which is nearly impossible, it's not like we can ask them to connect from a different network on the fl |
| 12:58.06 | Samot | It's running on a system doing a lot of other things... |
| 12:58.32 | Samot | All of which can impact the browser and the WebRTC call since WebRTC clients are JS based. |
| 12:59.31 | Samot | It's not like a physical device such as a phone, who's CPU, memory and resources are allocated for one thing.. |
| 12:59.34 | Samot | Being a phon. |
| 12:59.37 | Samot | Being a phone. |
| 12:59.58 | stefan27 | Yeah; frankly I have no idea how that works... When chrome carries out VoIP tasks does it get CPU priority over other things your PC does? (On Windows or Mac) |
| 13:00.11 | Samot | Or Firefox |
| 13:00.13 | Samot | Or Safari |
| 13:00.19 | Samot | Or Edge/Explorer |
| 13:00.21 | drmessano | It doesnt |
| 13:00.33 | drmessano | Chrome has no loyalty to any app |
| 13:00.35 | drmessano | or site |
| 13:00.45 | Samot | At some point browsers may care about WebRTC |
| 13:01.38 | Samot | Then, of course, you have the fact the client has to go through Apache. |
| 13:01.58 | Samot | I'm sure in about 5 years WebRTC will be a regular thing.. |
| 13:02.10 | Samot | OR it will have died a horrible death. |
| 13:02.31 | jkroon_ | is there a sensible quide anywhere on setting up WebRTC? |
| 13:02.39 | Samot | Nope. |
| 13:02.51 | file | define "setting up WebRTC" |
| 13:03.16 | jkroon_ | well, i've got a bunch of SIP accounts currently, I'd like to enable users to log in on a web interface and use it as if it's their phone in some way ... |
| 13:03.21 | file | WebRTC itself is a technology in the browser, with the client being <something> |
| 13:03.49 | *** join/#asterisk XATRIX (~xatrix@185.76.80.126) |
| 13:03.52 | file | there's a wiki page that describes how to set it up in Asterisk |
| 13:04.04 | jkroon_ | ok, that would help already, will take a peek thanks. |
| 13:04.46 | jkroon_ | essentially for now it'll probably be outbound calls only. and it has nice potential for support type things, ie, click here to speak with us ... |
| 13:05.11 | jkroon_ | anyway, project for another day. was interesting reading the discussion above. thanks. |
| 13:05.11 | stefan27 | I wonder if that wiki-page is updated with the requirement to add "rtcp_mux=yes" or whichever, in sip.conf files? |
| 13:05.23 | XATRIX | Hi guys, can you advice ? Why my call goes directly to 'FollowMe' number, instead of rinning on SIP client? It should ring SIP client fist, and in case of SIP is disconnected then go to GSM FollowMe number. https://paste.fedoraproject.org/paste/CprXzze5Xs3spt4MyhKq1l5M1UNdIGYhyRLivL9gydE= |
| 13:06.33 | file | stefan27: I just did it |
| 13:06.57 | file | https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 entries updated and a warning added |
| 13:07.10 | [TK]D-Fender | XATRIX, #freepbx <- |
| 13:07.23 | XATRIX | sorry |
| 13:09.33 | stefan27 | Great; I would personally warn that even though the SIPml5-client-js-library is certainly cool and functional; it's not as simple as cloning their demo to get a robust webrtc softphone. |
| 13:09.46 | stefan27 | you're likely to run into a bunch of micro-issues! |
| 13:11.24 | file | stefan27: added an info section at the SIPML5 part |
| 13:11.50 | stefan27 | Does anyone know why updates to https://github.com/DoubangoTelecom/sipml5 are so sparse now? Project seemed really active 3 years ago |
| 13:12.28 | stefan27 | I hear that many people recommend using https://github.com/meetecho/janus-gateway instead between clients and asterisks |
| 13:13.05 | stefan27 | You had a long dialogue with the creator of the janus-gateway some years ago, remember that file? |
| 13:13.12 | file | nope |
| 13:13.22 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 13:13.23 | file | entirely possible, though |
| 13:13.49 | stefan27 | I think it was about how to make asterisk 13.1.0 more webrtc friendly |
| 13:14.06 | stefan27 | in the mailing lists |
| 13:14.57 | file | does a lot of stuff and thus does not recall |
| 13:16.35 | Haris | with the multi-core processors now coming in .. say desktops, one or two or some of those could be used for voice traffic |
| 13:16.52 | Haris | but it still requires alot of work |
| 13:17.04 | file | it's up to the browser. |
| 13:17.05 | Haris | for webrtc to be a normal thing on a pc |
| 13:18.04 | Samot | Other industries aren't going to conform to WebRTC |
| 13:18.17 | Samot | Until WebRTC requires them to. |
| 13:18.24 | [TK]D-Fender | <Haris> for webrtc to be a normal thing on a pc <- that really doesn't mean anything |
| 13:18.36 | [TK]D-Fender | It's up to the browser, and then every other rndom thing a user will do |
| 13:18.45 | Haris | I mean for it to be like a normal app that is used in every day usage |
| 13:18.59 | Samot | WebRTC is kinda young still.. |
| 13:19.02 | Haris | like a browser is used |
| 13:19.10 | Haris | or office apps |
| 13:19.16 | Samot | It hasn't saturated the market with use yet. |
| 13:19.29 | Samot | No one really knows if it's going to catch on over all.. |
| 13:19.30 | stefan27 | audio encoding is not expensive enough to prompt for specially dedicated hardware right? like there's specially dedicated hardware for decoding video? (No I dont know processor design) |
| 13:19.43 | Samot | Because it's too many factors right now to figure out stability. |
| 13:20.10 | Samot | PCs already do this |
| 13:20.17 | Samot | Softphones exist that do video and voice |
| 13:20.27 | Samot | Clearly the PCs can handle the codecs needed. |
| 13:21.04 | Samot | WebRTC is a browser based softphone client. |
| 13:21.17 | Haris | it has to catch on if they want calls via browser on mobiles |
| 13:21.29 | Samot | But unlike other softphone clients, it depends on multiple vendors for it's use. |
| 13:21.34 | Haris | to be a no-brainer for the average guy |
| 13:21.44 | Samot | Why? |
| 13:21.50 | Haris | its going towards globalism |
| 13:21.57 | Samot | Why do I need WebRTC on my mobile device when I have Bria? |
| 13:22.09 | Haris | multiple services stacked under one banner by the mega corps |
| 13:22.18 | Samot | Bria |
| 13:22.20 | Haris | hmm |
| 13:22.28 | Samot | Video, Voice, XMPP, SIP SIMPLE |
| 13:22.35 | Samot | Contacts, BLF, etc, etc. |
| 13:22.42 | Samot | WebRTC is a SOFTPHONE |
| 13:22.55 | Samot | Like every other softphone on the market. |
| 13:23.13 | Samot | It just uses a browser and HTTP requests to do stuff. |
| 13:23.14 | Haris | more like .. its a tech or capability in browser used by softphone |
| 13:23.36 | file | yes, it's tech that can be used to make a softphone |
| 13:23.40 | file | it can also do other things |
| 13:23.47 | Samot | Yes. |
| 13:23.50 | Samot | So can other softphones. |
| 13:24.04 | file | no |
| 13:24.11 | file | :D |
| 13:24.19 | file | WebRTC, the technology, can do more |
| 13:24.27 | Haris | with multi-core devices sold, it may become the norm in far off time to come |
| 13:24.38 | Haris | but not with 4-8 core machine |
| 13:24.42 | Haris | average machine |
| 13:24.43 | Samot | Haris: It has to be adopted at some point. |
| 13:24.45 | file | data channels are a useful thing, Ubiquiti uses them for example to allow remote access to their Unifi controller |
| 13:24.54 | Samot | For voice. |
| 13:25.02 | Samot | As a viable voice option. |
| 13:26.22 | file | for voice I remain unconvinced personally |
| 13:26.31 | Samot | ^^ Yup. |
| 13:26.42 | Samot | WebRTC may be used for a bunch of other stuff as a standard.. |
| 13:26.48 | Samot | But right now for voice..no one sees it. |
| 13:27.05 | file | I've found the quality I get from stuff that has been doing it for much longer greater than what I get from WebRTC |
| 13:27.28 | Samot | Well in an office... |
| 13:27.30 | file | no echo, no background noise, no weird gain wonkiness |
| 13:27.40 | Samot | For people connecting on their mobile devices.. |
| 13:27.45 | Samot | Something like Bria is more viable. |
| 13:27.46 | Haris | it'd be interesting to see how they function on desktops with 24 core CPUs |
| 13:27.52 | file | if you control the hardware fully then sure |
| 13:27.56 | file | which is what stefan27 probably does |
| 13:28.03 | stefan27 | in an office why would you want a softphone anyways, with a dedicated table phone that never shuts off, never plugs out microphone, it just works 24/7 |
| 13:28.36 | stefan27 | (some people do want it but I don't make enough calls to want one) |
| 13:28.51 | Samot | Well that's the thing.. |
| 13:28.53 | stefan27 | I answer like a few calls a week, I'm happy with my SNOM table phone |
| 13:28.58 | Haris | land line ? use it alot with ISPs, vendors, etc etc |
| 13:29.20 | Samot | If WebRTC becomes the "Well I don't warrant enough use to have a softphone app" it's not going anywhere. |
| 13:29.54 | Haris | softphone may not work on every platfor |
| 13:29.56 | Samot | stefan27: I'm talking about offices that have agents on the road.. |
| 13:29.56 | Haris | platform |
| 13:30.00 | Haris | but browsers do work on most |
| 13:30.04 | Samot | But they want connected for communication. |
| 13:30.16 | Haris | browsers work across any platform |
| 13:30.27 | stefan27 | yeah that's one perk of webrtc, no installation so you get very free seating |
| 13:30.32 | Haris | I should use the word 'any' sparingly here |
| 13:30.33 | Samot | Bria works on iOS, Android, Windows and Mac |
| 13:30.42 | file | to me it's about the quality and experience |
| 13:30.46 | stefan27 | if it runs in chrome environments without bad network rules that is |
| 13:30.46 | Samot | There is a reason Bria is the top softphone client for businesses. |
| 13:31.12 | Dirk23 | Samot: FYI. i got everything up and running! Thnx for all your help and so |
| 13:31.21 | Samot | stefan27: I'm talking about more than calls. |
| 13:31.26 | file | wonders what he's started |
| 13:31.34 | Samot | Presence, BLF, IM |
| 13:31.55 | stefan27 | my mobile devices always run out of power because Im sloppy with charging :( |
| 13:31.57 | Samot | These are things that offices will want for their mobile users or telecommuting users |
| 13:32.05 | Haris | me too |
| 13:32.08 | stefan27 | but yes, I like bria in those odd cases |
| 13:32.16 | Samot | WebRTC does not do that yet. |
| 13:32.21 | Haris | want a phone that doesn't need charging in a week's time :D |
| 13:32.35 | Samot | So it being adopted by offices that need a client to support all those things won't happen. |
| 13:32.52 | file | WebRTC will never do those, the client built on top of it would |
| 13:33.02 | stefan27 | It's beyond my competence to judge this, but I hear WebRTC takes security very seriously, so you might get safer encryptions with it than with say zoiper' encryption? |
| 13:33.03 | [TK]D-Fender | Can I get that in corn-flower blue? |
| 13:33.03 | Samot | Therefore the providers to those offices will not adopt it quickly. |
| 13:33.27 | stefan27 | (zoiper being an arbitrary mentioned softphone) |
| 13:34.37 | Samot | stefan27: It's HTTP |
| 13:34.37 | stefan27 | but judging security would mean you'd have to know the complete audio and signaling paths from end devices |
| 13:34.42 | Samot | It uses TLS certs. |
| 13:34.49 | Samot | Encryption is as powerful as your TLS |
| 13:35.09 | Samot | First, RTP doesn't get encrypted. |
| 13:35.11 | Samot | At all |
| 13:35.12 | Samot | Ever. |
| 13:35.19 | Samot | It gets encapsulated |
| 13:35.29 | Haris | they'r building encryption related chips in or near the cpu |
| 13:35.30 | stefan27 | when we call out from our webrtc softphone, we go via a PSTN gateway provider, and then routed to some other network, i have no idea how hard or easy it is to eavesdrop those networks |
| 13:35.34 | Haris | in desktops |
| 13:35.37 | stefan27 | so it might be irrelevant if our uplink is well encrypted? |
| 13:35.48 | Samot | Well there is no encryption on the PSTN |
| 13:35.49 | Samot | At all. |
| 13:36.04 | stefan27 | but how hard is it to eavesdrop on calls? |
| 13:36.18 | Samot | That's relevant |
| 13:36.21 | Samot | To a lot of factors. |
| 13:36.30 | Samot | Including the person(s) trying to easedrop |
| 13:37.10 | stefan27 | well sure, the person might be carrying a hidden microphone someone put in his pocket |
| 13:37.42 | Haris | PSTN doesn't need encryption, unless someone is directly hooked into your TELCO |
| 13:37.44 | Samot | Sure so that would require them to be next to the person making the call |
| 13:37.48 | Haris | that'd be the govt or the corp |
| 13:37.49 | Samot | And only get one side of the call. |
| 13:37.55 | *** join/#asterisk scgm11_ (~scgm11@r186-52-248-245.dialup.adsl.anteldata.net.uy) |
| 13:38.15 | Samot | What? |
| 13:38.28 | Samot | There's no encryption because, copper. |
| 13:38.36 | Samot | Analog |
| 13:40.44 | stefan27 | I've come incorrectly come to use the term PSTN "for everything outside our own VoIP-world" |
| 13:40.58 | stefan27 | are the mobile networks counted under PSTN? |
| 13:41.46 | Haris | no |
| 13:41.58 | Haris | traditional land line |
| 13:42.20 | stefan27 | so if PSTN is just the set of analog land lines and the interconnecting devices |
| 13:42.24 | Samot | Public Switched Telephone Network |
| 13:42.33 | Samot | EVERYTHING is on the PSTN that is a CALL |
| 13:42.47 | Samot | VoIP, Copper, Mobile |
| 13:42.56 | Samot | It's how calls get routed over the world |
| 13:43.28 | Samot | The only time you bypass the PSTN is when you have your own enclosed network between the caller and the callee |
| 13:43.52 | stefan27 | right, there's something which is defined by a shared address-space of numbers, e.g. +4642342 and +1342348324823 |
| 13:43.54 | Samot | Or in the case of SIP and SIP-to-SIP call. |
| 13:44.08 | stefan27 | but is that really PSTN? I'd like to see some rigorous definitions |
| 13:44.16 | Haris | its all monitored by the govt or the folks behind it. no getting around them .. yet |
| 13:44.18 | Samot | 9:42:26 AM S<Samot> Public Switched Telephone Network |
| 13:44.36 | Samot | It's how carriers pass calls between each other |
| 13:45.55 | Samot | "Mobile" is a delivery method |
| 13:45.59 | Samot | Like VoIP |
| 13:46.03 | Samot | Or copper. |
| 13:47.33 | stefan27 | I can see how "Public" makes sense since the whole world shares the same address space and +467123712371 doesnt need a context do define the destination, but "Switched Telephone Network" doesn't tell me anything |
| 13:49.17 | stefan27 | I know one can talk about packet-switched or circuit-switched networks, but what's that word "Switched" doing there in PSTN |
| 13:50.09 | stefan27 | (uh this chain of thought got out of hand) |
| 13:50.18 | Samot | It was created in like 1940 |
| 13:50.30 | Samot | "Switched" means switching between carriers. |
| 13:51.03 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 13:51.14 | *** join/#asterisk Dugroin (~ygancberg@2a02:2788:2b4:1eef:9639:a306:2371:877b) |
| 13:51.43 | Samot | As a carrier if you want your subscribers to call or get calls to/from people on other carriers, it has to be routed |
| 13:51.58 | Samot | The two carriers need to be able to talk to each other and send the call between them. |
| 13:52.40 | Dugroin | Hello everybody.. I have a little question.. maybe someone can help me. I would like to change the sound played to the caller, when he dials.. I would like to set up there a sound like 'welcome to blablabla, etc...' does some one knows how to do that ? |
| 13:54.40 | [TK]D-Fender | clarify the call flow you're loking for exactly. |
| 13:54.49 | Samot | Well actually before the 1940's really, the North American Number Plan was created in the 40's. |
| 13:55.36 | [TK]D-Fender | Dugroin, If you're referring to * processing an incoming call, that's all your dialplan. You're the one picking the apps to run against that call. |
| 13:55.56 | Dugroin | The call comes from a sip channel.. what I want is that the caller (placing the call) hears a 'waiting music', instead of the normal ring tone... |
| 13:56.09 | [TK]D-Fender | Dugroin, So if yuo want a recording, then go record it and use Playback() or Background() if you are looking to use it in an IVR |
| 13:56.28 | [TK]D-Fender | what is causing the ringing? |
| 13:56.34 | [TK]D-Fender | the call has arrivied at your server |
| 13:56.43 | [TK]D-Fender | what are you doing that is passing back ringing exactly? |
| 13:56.49 | *** join/#asterisk foo (~foo@unaffiliated/foo) |
| 13:57.20 | foo | I have someone in Vienna who is going to send a text to a number. Before I ask him to do that, anyone happen to know what average costs may be for him to send a text to an American number? |
| 13:58.29 | [TK]D-Fender | nope |
| 13:58.39 | [TK]D-Fender | ask the telco |
| 13:59.13 | foo | I'm beginning to wonder if people using SMS for an onboarding process is a good idea. It made sense in the US... but I thought anyone else texting a US number has to pay more $ (even though it's only < 5 text messages). |
| 13:59.20 | foo | Or, I wonder if I could buy phone numbers in different parts of the world. hm |
| 13:59.27 | foo | Probably |
| 14:00.21 | Dugroin | d-fender, could I get somewhere if I try something like Set(CHANNEL(musicclass)=blablabla) N |
| 14:00.23 | Dugroin | ? |
| 14:01.15 | igcewieling1 | Dugroin: in the dialplan. |
| 14:01.18 | Rac-on | foo: getting foreign numbers on sip isnt that hard. but certain country's have very strict rules, like being registered at the local chamber-of-commerce or even having an office-address in the region you want a number in |
| 14:01.41 | Dugroin | igcewieling1, what do you mean :-( ? |
| 14:01.58 | [TK]D-Fender | Dugroin, I asked what you are doing that is CAUSING ringing. |
| 14:02.04 | [TK]D-Fender | Dugroin, what APP? |
| 14:02.08 | foo | Rac-on: I see. Since this specific app reaches a worldwide audience, I'm thinking having alternatives (eg. e-mail / text a US number) is best and people can decide what they want to do |
| 14:02.12 | igcewieling1 | Dugroin: looks like you you need to learn Asterisk. |
| 14:02.25 | *** join/#asterisk kharwell (kharwell@nat/digium/x-fjwvnxetqgmrkdpc) |
| 14:02.25 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:02.25 | foo | Rac-on: This is for an onboarding process, I thought SMS made it as easy as possible (and it does in US). Appreciate your trip |
| 14:02.28 | foo | tip * |
| 14:02.41 | Dugroin | igcewieling1, indeed... any help ? |
| 14:02.47 | igcewieling1 | ~book |
| 14:02.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:02.57 | igcewieling1 | there you go. |
| 14:03.23 | [TK]D-Fender | [TK]D-Fender> Dugroin, I asked what you are doing that is CAUSING ringing. |
| 14:03.23 | [TK]D-Fender | <[TK]D-Fender> Dugroin, what APP? |
| 14:04.20 | Haris | your PSTN, as with the rest of your infrastructure is monitored |
| 14:04.26 | Haris | heavily |
| 14:04.32 | Dugroin | euh.. d-fender... how can I find that out ? The phone rings.. because there is a 'Goto' to it.. :-( |
| 14:04.46 | *** join/#asterisk |ance|ott (~|ance|ott@208.69.10.6) |
| 14:04.55 | Haris | there's no way your going to get that call route to get the bad guys out without a fight |
| 14:04.58 | [TK]D-Fender | Dugroin, Goto doesn't cause ringing |
| 14:05.01 | [TK]D-Fender | DIAL rings things |
| 14:05.05 | [TK]D-Fender | QUEUE rings things |
| 14:05.47 | [TK]D-Fender | When you get an incoming call YOU are the reason your caller hears ringing after it arrives, typically because YOU call OUT to something else and the ringing is PASSED on. |
| 14:06.07 | [TK]D-Fender | You should not be unaware of what steps your caller is actually going through |
| 14:06.34 | Dugroin | I have access to the asterisk logs, if that is what you mean... |
| 14:06.46 | [TK]D-Fender | No. |
| 14:06.50 | [TK]D-Fender | We shouldn't need "logs" |
| 14:06.54 | [TK]D-Fender | this is YOUR server |
| 14:06.59 | [TK]D-Fender | what are YOU doing with the call? |
| 14:07.19 | [TK]D-Fender | that caller is hearing ringing for a REASON. |
| 14:08.03 | Dugroin | when the call arrive, it is send in a diaplan... and it this dialplan, there is a 'Goto' which makes the phone ring on one side and the caller hearing a ring sound on the other side... |
| 14:08.07 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 14:09.11 | [TK]D-Fender | No. |
| 14:09.16 | [TK]D-Fender | Goto() has nothing to do with ringing |
| 14:09.25 | igcewieling1 | Dugroin: Goto does NOT cause a phone to ring. It does NOTHING except jump to another place in the dialplan. |
| 14:09.39 | igcewieling1 | Perhaps you should read the Asterisk book. |
| 14:09.50 | stefan27 | happy friday everyone; im off to drink beer |
| 14:10.03 | [TK]D-Fender | This is looking to be the case... |
| 14:14.38 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 14:15.07 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:15.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:18.26 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:18.26 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:19.14 | *** join/#asterisk miralin (~Thunderbi@178.34.161.211) |
| 14:32.32 | pawiecki | Dugroin: To make things a bit simpler: 1. log into your asterisk server, 2. enter the CLI (asterisk -vvvr), 3. make the call, 4. hangup, 5. check step-by-step what has happened in the CLI and try to understand it. If you are unsure, inside that same CLI, enter 'core show application Goto' or 'core show application dial' and so on. Step by step, learn those things and understand what is actually happening. |
| 14:37.39 | *** join/#asterisk Wimdev (~wim@ip-94-140-174-105.reverse.destiny.be) |
| 14:37.45 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 14:38.21 | Wimdev | hi, has anyone used a letsencypt cert for asterisk tls? |
| 14:38.49 | Wimdev | when i set the tlscertfile and reload sip i get TLS/SSL error loading cert file. |
| 14:39.44 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 14:40.15 | Wimdev | do i need to concatenate the cert.pem with the letsencrypt intermediate certificate? |
| 14:40.27 | igcewieling1 | Wimdev: Welcome to the hell which is TLS. |
| 14:40.33 | Wimdev | :) |
| 14:40.41 | igcewieling1 | Did you load the cert into your phone? |
| 14:41.05 | Wimdev | im not even there yet. im just trying to get asterisk to listen on TLS |
| 14:41.39 | igcewieling1 | post the error message. |
| 14:42.23 | Samot | Let's Encrypt requires port 80 access to the server. |
| 14:44.35 | Wimdev | Samot, that part went ok |
| 14:45.09 | Wimdev | igcewieling1, i figured id move the certificates to /etc/asterisk and now i no longer have the error.... |
| 14:45.24 | Wimdev | tcp 0 0 0.0.0.0:5061 0.0.0.0:* LISTEN 112 61874881 1608/asterisk |
| 14:46.00 | Wimdev | i did try a !cat /old/cert/path from asterisk cli just to see if it could actually read the file |
| 14:46.18 | Wimdev | the default path is /etc/letsencrypt/live/domain/ |
| 14:52.30 | Wimdev | anyway, should have waited 5 more minutes to ask in here i guess :-). |
| 14:53.00 | igcewieling1 | In my opinion, if you have to load the CA cert or intermediate cert into all the devices anyway when using discount CAs, then why not generate your own. |
| 14:54.11 | Samot | Wimdev: Show your peer settings from sip.conf for a peer trying to use TLS |
| 14:56.30 | Wimdev | i have it working now. could register a Yealink to it |
| 14:56.41 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-isesdajlgehctciy) |
| 14:56.41 | *** mode/#asterisk [+o rmudgett] by ChanServ |
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| 16:03.41 | forgotmynick | Samot are you feeling better buddy? |
| 16:03.59 | Samot | I was feeling fine before. |
| 16:04.17 | forgotmynick | there's no reason to hide the tantrum you had before, it happens |
| 16:04.37 | Samot | I wasn't aware I was hiding something. |
| 16:04.53 | forgotmynick | you're not very perceptive then are you |
| 16:05.07 | Samot | Pardon? |
| 16:08.25 | [TK]D-Fender | Would be nice if that description was coherent phrase to phrase, but I suppose I'm being too picky |
| 16:14.03 | Samot | I thought there'd be more. |
| 16:14.09 | Samot | shrugs. |
| 16:15.33 | [TK]D-Fender | How would one hide a previous tantrum? Scrub IRC logs? Is that where this is assumed to be going? |
| 16:16.07 | [TK]D-Fender | Then hiding the hiding... |
| 16:16.17 | Samot | Hence my statement that I wasn't aware I was hiding something. |
| 16:16.24 | Samot | It was in the open. |
| 16:16.38 | Samot | I mean it was no where near "tantrum" levels for me. |
| 16:16.42 | Samot | But yeah it happened. |
| 16:16.48 | [TK]D-Fender | I figured I'd start with the first failure before building on it :) |
| 16:16.59 | [TK]D-Fender | For you know ... context. |
| 16:17.06 | Samot | Psssh. |
| 16:17.15 | Samot | context is something in dialplan. |
| 16:17.24 | [TK]D-Fender | And voicemail |
| 16:17.32 | Samot | And well, everything else. |
| 16:18.08 | Samot | I'm curious has to how my perception skills where called into question. |
| 16:20.48 | Samot | Because I apparently didn't perceive me doing something openly as hiding. |
| 16:25.55 | Samot | Well that was all very anti-climatic. |
| 16:40.40 | *** join/#asterisk dexta_ (~D3XTA@86.178.237.39) |
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| 17:40.28 | TheGallopingFox | how long will the 13 branch be supported for? |
| 17:42.01 | igcewieling1 | TheGallopingFox: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 17:42.43 | TheGallopingFox | thanks |
| 17:42.54 | TheGallopingFox | 2020-10-24 wow nice |
| 17:43.27 | igcewieling1 | LTS branches are supported for a long time. |
| 17:44.04 | TheGallopingFox | i try and do everything LTS these days, including kernels |
| 17:44.53 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 17:45.27 | TheGallopingFox | and i guess the upgrade to 15 will be pretty smooth anyhow |
| 17:46.44 | TheGallopingFox | from a fresh install of 13, i have these errors in the log with nothing configured at all http://sprunge.us/IMQY |
| 17:47.02 | TheGallopingFox | to stop those errors would i need to disable some modules? |
| 17:58.19 | [TK]D-Fender | well it's saying you're missing config files... so "nothing configured at all" is your actual problem there. |
| 17:58.36 | [TK]D-Fender | [Apr 27 23:06:08] ERROR[16845] config_options.c: Unable to load config file 'res_parking.conf' <- probably one you WILL want to fix |
| 17:59.08 | [TK]D-Fender | Many of the rest are used much less rarely. Take your pick on how you want to handle these |
| 17:59.26 | [TK]D-Fender | Either noload-ing them or providing an appropriate config |
| 18:05.32 | *** join/#asterisk Alex_Bkash (7c06eeee@gateway/web/freenode/ip.124.6.238.238) |
| 18:17.06 | TheGallopingFox | [TK]D-Fender: would it be better to copy over all the sample configs to /etc/asterisk then just configure each config, ie, disable them |
| 18:17.33 | TheGallopingFox | if they are not required |
| 18:18.17 | TheGallopingFox | then asterisk will not complain about missing configs |
| 18:18.38 | [TK]D-Fender | If they are not required then disabling is perfectly valid and more efficient in the end |
| 18:18.41 | [TK]D-Fender | just be sure of it |
| 18:18.43 | TheGallopingFox | <PROTECTED> |
| 18:19.04 | TheGallopingFox | ok thanks |
| 18:28.56 | *** join/#asterisk Iamnach0 (~Iamnacho@ip70-171-163-5.om.om.cox.net) |
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| 18:45.35 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 18:48.32 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 18:57.42 | mub | How bad of an idea is it to have 3+ asterisk servers behind a NAT? My provider (les.net) says not to do this, but it's working fine so far.. |
| 18:58.17 | mub | they're all sharing one WAN IP, but it seems my NAT settings are good |
| 19:09.06 | [TK]D-Fender | Each server should be bound to its own different SIP port # and RTP range |
| 19:09.11 | [TK]D-Fender | aside from that you should be OK |
| 19:20.28 | mub | Oh snap, that's a great idea |
| 19:20.32 | mub | thank you [TK]D-Fender |
| 19:20.52 | tuxd00d | Iâm trying to help out a company that is having trouble sending calls to their main provider (which looks like it is running a SipWise product). Outgoing calls are rejected due to what looks like unsatisfied proxy auth although subsequent INVITES inlcude âProxy-Authorizationâ. Calls work to an alternate provider. Incoming calls are fine. http://pastebin.ca/3804234 |
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| 19:30.45 | *** join/#asterisk bidk (~bipsen@85.184.162.81) |
| 19:32.02 | bidk | Struggeling a bit trying to "migrate" and old installation.. Gets this in my log: |
| 19:32.18 | bidk | pbx.c: No application 'SetMusicOnHold' for extension (from-external-custom, s, 3) |
| 19:32.36 | tuxd00d | bidk: FreePBX? |
| 19:33.22 | bidk | Yup |
| 19:33.42 | bidk | known issue ? |
| 19:34.21 | bidk | I am not so experienced in asterisk/freepbx - but I had the setup running on a really old version :-) |
| 19:34.39 | tuxd00d | This isnât the place for FreePBX questions⦠your issue isnât with Asterisk, itâs with FreePBX configuration. Iâm sorry. |
| 19:35.11 | bidk | That's okay... |
| 19:35.13 | *** part/#asterisk bidk (~bipsen@85.184.162.81) |
| 19:47.09 | Samot | tuxd00d: They aren't authorizing the call properly |
| 19:47.20 | tuxd00d | The client or the provider? |
| 19:47.23 | Samot | tuxd00d: Asterisk is responding to the 407 with auth digest.. |
| 19:47.34 | Samot | And they are kicking that back. |
| 19:47.41 | tuxd00d | So send clear text? |
| 19:47.44 | Samot | So yeah, check their registration |
| 19:47.54 | Samot | What do you mean? |
| 19:47.55 | Samot | No. |
| 19:48.05 | Samot | It's responding properly. |
| 19:48.16 | Samot | They just don't like the stuff being sent. |
| 19:48.20 | Samot | Could be a bad password |
| 19:48.30 | Samot | Auth name could be wrong.. |
| 19:48.49 | tuxd00d | SIP Reg uses the same password, but Iâll have the provider double check. |
| 19:48.57 | tuxd00d | and same username |
| 19:49.20 | tuxd00d | Thanks Samot |
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| 22:09.54 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 22:10.36 | UncleKiwi | hi people hope all is well - i am trying to make my FXO device dial a number and it dont see to be having any joy |
| 22:11.20 | UncleKiwi | chan_sip.c:22885 handle_response_invite: Received response: "Forbidden" from |
| 22:11.48 | UncleKiwi | it's a grandstream HT-503 |
| 22:12.04 | UncleKiwi | it is registered as a peers |
| 22:12.06 | UncleKiwi | peer |
| 22:12.18 | UncleKiwi | it take incomming calls ok |
| 22:12.26 | UncleKiwi | from the pstn |
| 22:13.11 | UncleKiwi | but when i try to make a call out over pstn using this FXO device asterisk returns the message above *chan_sip |
| 22:13.16 | Samot | Need to see more |
| 22:13.35 | Samot | An actual attempt |
| 22:13.45 | UncleKiwi | mm ok |
| 22:15.49 | UncleKiwi | exten => _77,n,dial(SIP/pstn2/500) |
| 22:16.01 | UncleKiwi | i just created that for testing it |
| 22:16.26 | Samot | Ok |
| 22:16.47 | Samot | Show a call attempt with a sip debug. |
| 22:19.13 | UncleKiwi | whats the best pastebin thing to use |
| 22:19.42 | Samot | ~pb |
| 22:19.43 | infobot | methinks pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:22.06 | UncleKiwi | https://pastebin.com/0WyQvcAE |
| 22:25.11 | Samot | Seriously? |
| 22:25.21 | UncleKiwi | ahaha |
| 22:25.44 | UncleKiwi | sorry |
| 22:25.47 | UncleKiwi | what did i do ? |
| 22:26.02 | Samot | Look at your pastebin |
| 22:26.39 | UncleKiwi | opps |
| 22:26.41 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fjwvnxetqgmrkdpc) |
| 22:26.43 | UncleKiwi | wait |
| 22:27.16 | UncleKiwi | https://pastebin.com/3KLWYX0h |
| 22:28.24 | UncleKiwi | that was someone else paste |
| 22:28.52 | Samot | Your GS is blocking the call |
| 22:29.00 | UncleKiwi | hmm thanks |
| 22:29.23 | UncleKiwi | hmm i wonder why |
| 22:46.21 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 23:16.29 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-45-30-44.hsd1.nj.comcast.net) |
| 23:28.49 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 23:46.23 | UncleKiwi | do i register the fxo as a peer or friend |
| 23:47.02 | UncleKiwi | the grandstream fxo device wants to register to asterisk and it does so correctly |
| 23:47.15 | UncleKiwi | i just dont seem to be able to make calls out over it |
| 23:47.38 | UncleKiwi | i am just attempting to make it dial 31 for example |
| 23:48.37 | UncleKiwi | the peer or friend name is pstn2 |
| 23:49.17 | UncleKiwi | i made an extention 77 that when i dial from another phone will cause Dial(SIP/pstn2/31) |
| 23:49.24 | UncleKiwi | that should work right ? |
| 23:50.23 | UncleKiwi | i am going to drink some beer because this is stressful |
| 23:51.35 | UncleKiwi | is the format important Dial(SIP/31@pstn2) |
| 23:52.56 | UncleKiwi | as you said Samot the gs is blocking |
| 23:53.10 | UncleKiwi | i have a cisco spa3102 here |
| 23:53.15 | UncleKiwi | i know that will play the game |
| 23:53.22 | UncleKiwi | i might test that |
| 23:58.45 | *** join/#asterisk elguero (~miguel323@74-95-21-41-Connecticut.hfc.comcastbusiness.net) |
