| 00:26.03 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:26.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:45.44 | siegmarb | register => "number":"password"@sip.easybell.de:5064/number~90, sip show registry shows Refresh 75 which is ok, after several minutes, the Refresh increases to 585(600) - is this an expected behavior? |
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| 11:35.57 | td34 | hi all |
| 11:36.01 | td34 | just installed asterisk |
| 11:36.07 | td34 | g711 isn't in the codecs |
| 11:36.12 | td34 | is it under a different name? |
| 11:39.12 | file | ulaw and alaw |
| 11:45.10 | td34 | they are also not there |
| 11:45.20 | td34 | Is there anyway for me to include new codecs? |
| 11:49.20 | file | they're there as codecs, how are you checking? and as for including new codecs - it's possible... yes... has to be added to the core, and if transcoding is desired then a codec module written for it |
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| 12:14.27 | *** join/#asterisk nibbier (~quassel@geekbox.info) |
| 12:16.17 | nibbier | how would one usually manage dialgroups? setting them in extensions does not seem good, as they are stored in the sql backend anyway. can they comfortably be managed via the CLI? |
| 12:17.18 | file | nibbier: are you using a GUI such as FreePBX? if so - that's a concept of theirs, including how they store/manage things |
| 12:17.39 | nibbier | nope, no GUI |
| 12:18.12 | file | Asterisk doesn't store things in a sql backend by default |
| 12:19.13 | nibbier | file: okay, I just compiled what I got from asterisk.org and edited mainly sip.conf and extensions.conf. when I say "database show" in the CLI I get quite some entries |
| 12:19.42 | file | it will persist some data there by default, but it's not in regards to any dial groups |
| 12:19.58 | file | like it will store inbound registrations from devices so if you restart Asterisk they can still be reached |
| 12:20.24 | nibbier | I see various /SIP/Registiry/... but also all my former and current /dialgroup/... |
| 12:21.21 | file | oh geez, there's actually a func_dialgroup |
| 12:21.41 | file | 10 years old, that's why I didn't know it existed |
| 12:23.05 | nibbier | hehe.... I'm migrating some 12 year old asterisk right now... it used FreePBX (which I want to get rid of) and uses dialgroups, so I assumed that's the way to go. the easy add and delete functions seemed useable, in the end it makes it to the persistant db just in the concatinated form SIP/123&SIP/456... |
| 12:23.34 | file | it's not something I've ever seen anyone use, or if they have they haven't mentioned it |
| 12:23.37 | nibbier | file: how would you make sure that on incoming connections a specific set of phones ring? |
| 12:24.08 | file | depends on how it needs to be controlled... |
| 12:24.21 | file | it can be done in extensions.conf directly |
| 12:24.37 | nibbier | mainly it's very static, just one "callgroup" has one changing member (mobile of the technican on duty) |
| 12:24.41 | file | you can use the DB dialplan functions and AMI/CLI commands to store information in the AstDB like func_dialgroup does |
| 12:24.47 | file | you can use func_odbc to use an outside database |
| 12:24.48 | td34 | file |
| 12:24.57 | td34 | i'm looking in the linphone codecs part |
| 12:25.01 | td34 | no mention of ulaw or g711 |
| 12:25.15 | file | td34: I have no idea about linphone. |
| 12:25.30 | nibbier | file: AstDB is the one "database show" dumps out? |
| 12:25.41 | td34 | ok - at the moment I am getting a Subscriber absent |
| 12:25.44 | file | nibbier: yes - func_dialgroup is just a wrapper over it which also stores it in memory |
| 12:26.06 | nibbier | file: okay, thanks a lot! I'll have a look into this and most likely write my own code around it. |
| 12:27.15 | td34 | file, it's showing as "Unspecified" in the sip show peers |
| 12:27.18 | td34 | interesting.... |
| 12:27.25 | file | td34: then it hasn't registered or the registration was rejected |
| 12:28.12 | td34 | ok |
| 12:28.16 | td34 | ahh!!! |
| 12:28.19 | td34 | i think i understand |
| 12:28.41 | td34 | however, in the sip.conf i have the config secret=password |
| 12:30.49 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:31.21 | [TK]D-Fender | td34, just because you have that doesn't means something else might not be wrong, or that the request never made it to * in the first place |
| 12:33.21 | td34 | Well when I try and do an Invite |
| 12:33.30 | td34 | im getting a 401 Anauthorised back |
| 12:33.39 | td34 | but with verbosity 3 i see nothng on the sip server... |
| 12:34.29 | td34 | the odd thing is that I then said an invite to ring someone and it says 100 trying then.... 503 service unavilable |
| 12:35.56 | td34 | it's driving me insane |
| 12:36.37 | file | verify that traffic is reaching Asterisk, do "sip set debug on" and see if the traffic appears at Asterisk |
| 12:37.17 | td34 | ye it does |
| 12:37.36 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-vnzduutpnxldpknr) |
| 12:38.51 | td34 | i've got an unauthorised |
| 12:39.20 | file | it will also tell you the SIP peer/user that was matched, if that is correct then it's likely some configuration on the linphone side |
| 12:39.43 | td34 | actually..... |
| 12:39.46 | td34 | right... |
| 12:39.56 | td34 | from asterisk, i get a 503 service unavailable back |
| 12:40.16 | td34 | it doesn't send back auth error, that must be linphone crappiing it's pants |
| 12:40.23 | td34 | so the question is.... why isn't linphone registering... |
| 12:49.24 | td34 | so asterisk isn't even seeing the registration messages by the looks of things... |
| 12:52.43 | [TK]D-Fender | you only see actual call attempts, never any attempts to register? |
| 12:52.54 | td34 | looks like it |
| 12:54.05 | td34 | when sending a message request |
| 12:54.19 | td34 | it should be sip:<your extension>@thesipserver right? |
| 12:57.44 | [TK]D-Fender | INVITE <- |
| 12:58.23 | [TK]D-Fender | Actually, sorry... just "REGISTER blah...." |
| 12:58.48 | [TK]D-Fender | is going cold-turkey on coffee and some brain damage is to be expected... |
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| 13:05.45 | td34 | linphone is specifying local loopback networks only |
| 13:06.02 | td34 | i think this is why it wont register |
| 13:07.00 | [TK]D-Fender | Are you running linphone on your actual server box? |
| 13:08.36 | td34 | noope |
| 13:08.39 | td34 | linphone on a pc |
| 13:08.41 | td34 | connected to asterisk |
| 13:10.42 | [TK]D-Fender | So why would it be pointing to a loopback? |
| 13:11.05 | [TK]D-Fender | show us |
| 13:15.20 | td34 | i have no idea |
| 13:15.29 | td34 | proxy account = |
| 13:15.42 | td34 | your sip identity: sip:bob@10.1.1.1 (10.1.1.1 is the sip server) |
| 13:15.55 | td34 | sip proxy address sip10.1.1.1;transport=udp |
| 13:16.03 | td34 | registration duration :3600 |
| 13:16.06 | td34 | transport : udp |
| 13:18.09 | td34 | it never registers itself (showing as offline in sip show peers) |
| 13:18.12 | td34 | but it still decides to register |
| 13:20.57 | [TK]D-Fender | ? |
| 13:21.05 | [TK]D-Fender | what do you mean "decides to register"? |
| 13:21.23 | td34 | sorry |
| 13:21.25 | td34 | tired |
| 13:21.27 | [TK]D-Fender | <td34> it never registers itself (showing as offline in sip show peers) <- this just means it hasn't been successful yet. It doesn't mean attempts aren't coming in |
| 13:21.59 | td34 | it never registers itself (i'm lookin in wireshark and there is no registration messages) but when you try and make a phonecall it will send an invite. |
| 13:22.11 | [TK]D-Fender | Show us actual debug and "sip show settings" from CLI |
| 13:22.11 | td34 | however i am getting keep alives ... |
| 13:22.26 | [TK]D-Fender | Show everything |
| 13:23.32 | *** join/#asterisk juvenal (~juvenal@191.19.245.171) |
| 13:23.41 | td34 | it can't be the asterisk server |
| 13:23.50 | td34 | because the linphone isn't sending any registration messages. |
| 13:25.21 | td34 | FUCK |
| 13:25.26 | td34 | ITS BECAUSE THERE WAS NO DEFAULT ROUTE ON THE BOX |
| 13:25.29 | td34 | even though its on the same lan.... |
| 13:25.30 | td34 | WTF |
| 13:27.12 | td34 | allowing multiple codecs in asterisk |
| 13:27.19 | td34 | is it codec1&codec2 |
| 13:27.23 | td34 | or codec1,codec2? |
| 13:28.28 | file | , |
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| 13:51.31 | dnit | Hi |
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| 16:24.02 | dnit | Hi, how can make an agent hear the ringtone of a channel once the dial has been initiated. |
| 16:25.41 | dnit | Hi, how can I make an agent hear the ringtone of a channel once the dial has been initiated. |
| 16:43.53 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:43.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 17:25.21 | flujan | Hello Guys, I am trying to execute a macro on a Queue before bridge. It is possible to execute in on the calling channel instead of the called channel? |
| 17:33.00 | [TK]D-Fender | "core show application queue" <- |
| 17:34.50 | [TK]D-Fender | if it's there the instructions will tell you |
| 17:35.25 | [TK]D-Fender | the alternative is to look at WHAT you doing to dial those members. |
| 17:36.04 | [TK]D-Fender | DIRECT called devices don't give you more than Queue() lists. |
| 17:39.24 | flujan | [TK]D-Fender: thanks for the tip. |
| 17:39.43 | [TK]D-Fender | Local Channel <- |
| 17:39.47 | [TK]D-Fender | FTW |
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| 19:14.10 | Kobaz | [TK]D-Fender: meep |
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| 19:51.21 | klow | hello, I am trying to mitigate the issue of iOS sip phones becoming unreachable. I'm thinking about setting qualityfreq=5 globally on chan_sip . A) is this even possible B) will that actually keep SIP dialog pinging every 5 secs . I understand this will majorly affect mobile battery life but have decided it is worth it until I get something else going |
| 19:52.32 | klow | (I am working on a push notification implementation but need a stop-gap measure) |
| 19:52.55 | klow | iOS keeps implementing more stringent battery saving features that stop wifi antenna and make SIP very difficult esp for inbound calls |
| 19:56.33 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 20:08.04 | drmessano | klow: Which client? |
| 20:09.02 | drmessano | klow: Is this linphone or zoiper? |
| 20:09.05 | *** join/#asterisk mou (~donwillia@188.228.46.39) |
| 20:12.21 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 20:12.46 | ModFather | hi all does anyone know if asterisknow has any support channel to freenode? |
| 20:14.17 | [TK]D-Fender | it doesn't |
| 20:14.28 | [TK]D-Fender | it's functionall the FreePBX ISO recently |
| 20:14.42 | [TK]D-Fender | whatever parts they can't support in #freepbx you can ask here |
| 20:14.45 | [TK]D-Fender | heads out |
| 20:15.06 | ModFather | i just downloaded AsteriskNOW 10.13 |
| 20:15.17 | ModFather | from my network i can call just fine |
| 20:15.23 | ModFather | from another network i cannot hear |
| 20:15.29 | ModFather | and it saying poor quallity network |
| 20:17.34 | wabbits | maybe its a poor quality network. there is a *nix utility called mtr that can help you see the state of things. |
| 20:17.55 | wabbits | or the good old nat/firewall issue. |
| 20:18.01 | wabbits | ModFather |
| 20:19.00 | ModFather | i have disabled the firewalls |
| 20:19.05 | ModFather | from both networks |
| 20:19.13 | ModFather | and i have set the asterisknow server to the DMZ |
| 20:23.34 | wabbits | ok then investigate the other option. |
| 20:24.57 | ModFather | i dont think its an MTR issue |
| 20:24.57 | ModFather | but even if it is, how it should be fixed? |
| 20:25.05 | wabbits | What do you think MTR means? |
| 20:25.15 | drmessano | What is an MTR issue? |
| 20:25.22 | drmessano | O.o |
| 20:25.40 | wabbits | I will leave you in the more than capable hands of drmessano. |
| 20:26.03 | drmessano | wabbits: hell no |
| 20:26.17 | drmessano | wabbits: You brought her, you poke her |
| 20:26.17 | wabbits | grabs keys |
| 20:26.17 | drmessano | Rules are rules |
| 20:26.34 | klow | no its actually a branded stretto client, which is counterpath bria but we have white-labelled it |
| 20:27.05 | drmessano | klow: Can't they shoot you a new build? |
| 20:27.05 | drmessano | They've got BRIA working just fine in 10.3.x |
| 20:27.41 | ModFather | wabbits mtr, its the tracing tool that you can see latency and ping times over networks |
| 20:27.46 | klow | sorry actually I'm getting my apps confused, we are in *process* of rolling out branded but we are using "stretto" from app store |
| 20:28.04 | drmessano | klow: It's simply not going to work |
| 20:28.19 | drmessano | It goes in the background and dies in 10.3.x |
| 20:28.46 | drmessano | Unless they've updated it recently |
| 20:28.52 | klow | are you saying you think they fixed this is "bria" but not "stretto" / |
| 20:29.23 | klow | im referring not to white-label just the off the shelf stretto |
| 20:29.27 | drmessano | Im a beta tester.. Followed the whole process. Bria is working 100% |
| 20:29.44 | drmessano | No clue where the Stretto code is |
| 20:29.45 | drmessano | But |
| 20:30.12 | drmessano | Lets look |
| 20:30.38 | klow | ya, thanks for any info. this is huge .. like not a pet project |
| 20:30.47 | drmessano | Go to Preferences |
| 20:30.55 | drmessano | Do you have "Background Support"/ |
| 20:30.57 | drmessano | Do you have "Background Support"? |
| 20:30.57 | klow | its #1 prio for me. working with counterpath has been pretty painful, and we paid $$$$ for this platform and have struggled to roll it out |
| 20:31.02 | klow | in stretto app? |
| 20:31.04 | drmessano | Yes |
| 20:31.06 | drmessano | Stretto |
| 20:31.25 | klow | let me see. .that might be a setting that is only available to be provisioned on the backend |
| 20:31.46 | wabbits | ModFather plus jitter and loss. " ModFather: and it saying poor quallity network" |
| 20:32.08 | wabbits | grabs wallet and glasses. Heads toward door. |
| 20:32.19 | ModFather | wabbits yes the bria client, saying poor quallity network |
| 20:32.29 | drmessano | I guess there basically are no menus in Stretto still |
| 20:32.45 | drmessano | so klow look for a background support option in whatever tool you use |
| 20:33.44 | drmessano | Options should/could be "Default" and "Enhanced" |
| 20:34.46 | klow | ah yes I see this, this is new to me |
| 20:34.50 | ModFather | Sending to 74.108.69.34:61555 (no NAT) |
| 20:34.52 | ModFather | i see that |
| 20:35.01 | klow | it is in stretto |
| 20:35.04 | klow | doh |
| 20:35.25 | drmessano | klow: If default doesnt work, try enhanced, obviously |
| 20:36.19 | drmessano | I've actually had success with default |
| 20:36.19 | drmessano | Enhanced is more aggressive, but hey, if it works and default doesnt.. whatever |
| 20:36.19 | drmessano | klow: Are you using TCP or TLS ? |
| 20:37.02 | klow | TLS always |
| 20:37.31 | drmessano | Great.. Just wanted to be sure you're not using god-awful UDP |
| 20:50.57 | drmessano | klow: My bad earlier, I didn't realize they had version bumped Stretto. Glad they got the new code in there. Going through the various Bria betas, the improvement in the end was literally not-working-at-all > working. |
| 21:02.17 | *** join/#asterisk dnit (71c11f6e@gateway/web/freenode/ip.113.193.31.110) |
| 21:02.36 | dnit | Hi how can I confbirdge a channel which is in dialing state with an agent ? |
| 21:03.50 | *** join/#asterisk u0m3 (~u0m3@188.26.47.163) |
| 21:03.53 | dnit | Confbridge(channel, agent) is making the channel drop |
| 21:04.16 | klow | drmessano: ya, this hit us hard and we were not ahead of the curve. lots of missed calls |
| 21:04.46 | drmessano | klow: Bria is my backup.. and I had no idea it wasn't working until I needed it |
| 21:04.57 | drmessano | Then I dug in and found the issues |
| 21:05.21 | drmessano | Apparently iOS 10.0.x was one challenge, and 10.3.x added more |
| 21:05.57 | klow | Ya. I'm in the process of setting up push notifications with our branded stretto which will be nice to have going |
| 21:06.41 | klow | we would have just used groundwire more for this since they built push notifications into their system, but the SRTP info goes to a 3rd party which is a dealbreaker for us |
| 21:06.55 | klow | but this reminds me |
| 21:07.04 | klow | we *definitely* need a new built of our branded stretto bc it is quite old |
| 21:11.58 | drmessano | Ah |
| 21:23.30 | dnit | <PROTECTED> |
| 21:25.38 | dnit | Basically I want the agent to listen to the early media for that channel |
| 21:25.38 | dnit | Unfortunately I dont know which agent is it going to be when I dial. |
| 22:45.46 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:45.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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