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| 02:24.35 | Get_The_Fish | Hello all, I have an Asterisk 13.5.0 using PJSIP. Asterisk is not replying to SIP INFO packets used by my carrier as a keep alive during long calls. Is there any way to fix this? |
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| 03:34.52 | Get_The_Fish | So the carrier sends a SIP INFO packet, and Asterisk is just ignoring it. Under the spec I believe Asterisk is supposed to be replying. |
| 03:45.50 | Get_The_Fish | At least replying with an ACK. |
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| 04:16.54 | Get_The_Fish | Here is a packet trace of chan_pjsip on Asterisk 13.5.0 not sending ACK messages to an endpoint in reply to a SIP INFO packet during a SIP session: http://pastebin.com/KdtjUSTP |
| 04:21.00 | Get_The_Fish | I believe that this is a bug in Asterisk 13.5.0's implementation of PJSIP, chan_pjsip running against PJPROJECT 2.4. The SBC is known to be an ACME Packet SBC, and is using SIP INFO packets as a "zombie" SIP session detection system. |
| 04:33.16 | X-Rob | file: re that packet trace above - looks like you need to manually tell pjsip to care about INFO packets. https://trac.pjsip.org/repos/wiki/FAQ#info-method is what I found |
| 04:34.05 | X-Rob | an dyou're looking at line 249 onwards |
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| 06:09.26 | juanmapalad | !ping |
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| 06:27.04 | ChannelZ | #pong |
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| 07:09.55 | aiksa[LV] | Hi |
| 07:10.12 | aiksa[LV] | Anyone could hel around with Stasis devices state update through ARI? |
| 07:10.47 | aiksa[LV] | I have ARI connection to asterisk 13.5.0 |
| 07:10.55 | aiksa[LV] | ARI connection works |
| 07:11.33 | aiksa[LV] | when I try to update device I am not supposed to update (like Custom:1) - I get ARI exception; everything as expected |
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| 07:12.04 | aiksa[LV] | now if try to update "Stasis:1" to any state; I get 200 OK response |
| 07:12.12 | aiksa[LV] | empty message body |
| 07:12.29 | aiksa[LV] | but CLI> devstate list shows nothing |
| 07:12.40 | aiksa[LV] | any ideas? |
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| 07:22.59 | juanmapalad | what do i need to study if i want to learn asterisk? im a biginner |
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| 07:46.27 | ChannelZ | juanmapalad: |
| 07:46.29 | ChannelZ | ~book |
| 07:46.29 | 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 |
| 07:46.32 | ChannelZ | ~primer |
| 07:46.32 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
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| 09:18.28 | aiksa[LV] | Is it okay that ari websocket deosnt provide devicestate events which are triggered on Custom:XX device from CLI? |
| 09:18.49 | aiksa[LV] | or should I ask this in -dev ? |
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| 09:25.08 | aiksa[LV] | it's even stranger than that: |
| 09:25.18 | aiksa[LV] | deviceStates set through ARI |
| 09:25.25 | aiksa[LV] | does not shop up on CLI |
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| 09:25.46 | aiksa[LV] | through CLI> devicestate show |
| 09:26.19 | aiksa[LV] | and vice versa - ARI calls to list device state does not list Custom device state |
| 09:27.41 | aiksa[LV] | Is it supposed to work like that or Am I missing something here? |
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| 10:49.54 | cyford | ChannelZ, nice primer :) |
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| 12:25.22 | juned | Hi all |
| 12:25.41 | juned | how to write request and response in specific file |
| 12:25.48 | juned | I want to log information |
| 12:26.12 | juned | i.e exten => h,n,Set(RESPONSE=${CURL(http://192.168.1.111/telecom/V1/response.php,x=30}) |
| 12:26.37 | [TK]D-Fender | "core show application system", "core show function SHELL" |
| 12:27.10 | juned | okay let me check this quickly |
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| 13:10.59 | aiksa[LV] | any idea why pjsip acts like this on 13.5.0: |
| 13:11.02 | aiksa[LV] | <PROTECTED> |
| 13:11.05 | aiksa[LV] | asterisk: ../src/pj/os_core_unix.c:1254: pj_mutex_lock: Assertion `mutex' failed. |
| 13:11.08 | aiksa[LV] | Aborted (core dumped) |
| 13:11.34 | aiksa[LV] | this happens upon receivieng OPTIONS package from older version of asterisk |
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| 13:36.54 | juned | [TK]D-Fender: System command worked well for me |
| 13:36.57 | juned | Thanks :) |
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| 13:39.41 | [TK]D-Fender | You're welcome |
| 13:40.19 | zaf | so, i've got digium phones, and have been using SIP rather than PJSIP so far. If I want the BLF lights, etc, do I NEED to be PJSIP? |
| 13:40.42 | WIMPy | no |
| 13:41.38 | zaf | also not using DPMA |
| 13:42.20 | WIMPy | Not neccessary, either. |
| 13:42.52 | zaf | hmm, then i'm not sure why i'm not seeing lights on the entries... know of a page that details that? |
| 13:44.08 | WIMPy | The wiki for the phones would be a good place too look for xml examples. |
| 13:45.13 | newtonr | zaf, https://wiki.asterisk.org/wiki/display/DIGIUM/Smart+BLF |
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| 13:50.16 | Ice_Strike | WIMPy o/ |
| 13:51.13 | WIMPy | Ice_Strike: What's up? |
| 13:52.22 | Ice_Strike | Chillin |
| 13:52.24 | Ice_Strike | You? |
| 13:53.04 | WIMPy | Not much ATM, either. |
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| 13:55.25 | Ice_Strike | Have you ever used Kamailio? |
| 13:56.04 | WIMPy | No |
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| 14:35.31 | jms77 | Is it possible to share registrations between Asterisks using odbc/realtime configuration, specifically when using PJSIP as the channel driver? In the extconfig.conf file there appears to be the capability to do something like that using sippeers and sipregs when using chan_sip, Is there similar functionality for pjsip? |
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| 14:50.06 | newtonr | jms77, there is documentation about pjsip realtime here: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
| 14:50.37 | newtonr | the wiki page needs to be updated as mentioned here https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 , "2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py - adds the ps_systems, ps_globals, ps_transports, and ps_registrations tables. Adds several new columns for ps_endpoints, ps_contacts, and ps_aors." |
| 14:51.37 | newtonr | Based on https://issues.asterisk.org/jira/browse/ASTERISK-23807 I believe reloads will be required to recognize changes to ps_registrations |
| 14:53.03 | file | yes, outbound registrations require a reload |
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| 15:09.42 | phormulate | hello |
| 15:10.27 | phormulate | having an issue with asterisk 11.19 neither tlsbindaddr nor tlsbindport seem to work, I'm stuck with default 5060/5061, any ideas? |
| 15:20.51 | jms77 | thanks newton. I think I may be misunderstanding some terminology here. I have a more general question of "Can asterisk do this" |
| 15:21.39 | jms77 | Where I have two asterisk boxes sharing a database and multiple clients are registering to the separate servers |
| 15:22.10 | jms77 | At some point, an outside app will start a call using ari to generate channels |
| 15:24.51 | jms77 | If the endpoint hasn't registered on the particular asterisk that was chosen to originate the call, then my understanding is that it won |
| 15:25.05 | jms77 | wont be able to successfully place that call |
| 15:25.54 | jms77 | If I was reading correctly ps_registrations are different then registrations a peer will send to asterisk? |
| 15:26.12 | jms77 | And am I able to share that peers registrations somehow between the asterisk boxes |
| 15:27.06 | newtonr | phormulate, does your configuration work properly in previous versions? |
| 15:28.48 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-egfnvqgsjvurhraa) |
| 15:29.18 | phormulate | can't use em, need oauth2 patches |
| 15:29.27 | phormulate | can only use 11.x |
| 15:29.59 | newtonr | phormulate, sorry I mean previous versions of 11 |
| 15:30.31 | phormulate | the patches only work for 11.18-19, so I'm stuck with those |
| 15:30.37 | phormulate | neither version worked |
| 15:30.42 | phormulate | stuck on default ports |
| 15:31.35 | newtonr | jms77, peers is chan_sip terminology. ps_registrations is for the registration objects which are for outbound registration (asterisk to some remote registrar) |
| 15:32.04 | jms77 | gotcha |
| 15:32.11 | *** part/#asterisk juned (~juned@202.131.119.122) |
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| 15:32.58 | newtonr | jms77, you configure inbound registration to an endpoint through AOR and AUTH objects associated with the endpoint |
| 15:33.59 | newtonr | jms77, probably worth going through here https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip before you try to plan out how to do what you want. |
| 15:35.41 | qakhan | i am trying to install asterisk-addon-1.6.2.4 |
| 15:36.16 | qakhan | when i select mysql cdr and app there are errors http://pastebin.com/YAF2RRsp |
| 15:36.24 | jms77 | alright will do thanks |
| 15:38.13 | newtonr | phormulate, so i just searched JIRA, this may be the issue: https://issues.asterisk.org/jira/browse/ASTERISK-24428 |
| 15:38.14 | qakhan | i have install asterisk 13.5 |
| 15:38.25 | qakhan | with CentOS 6.7 |
| 15:39.24 | file | qakhan, addons were moved into Asterisk a LONG time ago - they can be selected in "make menuselect" |
| 15:39.31 | file | they aren't separate anymore |
| 15:40.00 | *** join/#asterisk jms77 (80677c8d@gateway/web/cgi-irc/kiwiirc.com/ip.128.103.124.141) |
| 15:40.07 | [TK]D-Fender | qakhani am trying to install asterisk-addon-1.6.2.4 <- mistake. |
| 15:40.20 | [TK]D-Fender | the only branches to even be considering are 11 & 13 |
| 15:40.42 | qakhan | @file you mean in asterisk make menuselect |
| 15:40.47 | file | yes |
| 15:41.02 | qakhan | [TK]D-Fender i have installed 13.5 |
| 15:41.07 | phormulate | is I can hardcode in the tls port |
| 15:41.08 | phormulate | ok |
| 15:41.11 | phormulate | will try |
| 15:41.19 | phormulate | thing is, multiple ports ugh |
| 15:41.22 | phormulate | instances |
| 15:41.23 | phormulate | etc |
| 15:41.53 | phormulate | I guess I could multihome the interface |
| 15:42.01 | phormulate | maybe vlan idk |
| 15:42.10 | phormulate | so painful |
| 15:42.51 | *** join/#asterisk sekil (~sekil@78.24.109.191) |
| 15:50.29 | phormulate | externtlsport=foo tlsbindaddr=0.0.0.0:foo don't work for me sadly |
| 15:53.36 | qakhan | [TK]D-Fender asterisk-addon-1.6.2.4 is the latest version on asterisk |
| 15:53.42 | qakhan | website |
| 15:53.51 | [TK]D-Fender | qakhan: Because it stopped existing after 1.6.2 |
| 15:54.50 | qakhan | so how can i install app_addon_sql_mysql, cdr_addon_mysql and res_config_mysql |
| 15:55.19 | [TK]D-Fender | Everything you've got is in the main tarball |
| 15:55.33 | [TK]D-Fender | Most direct DB stuff has been phased out. |
| 15:55.39 | [TK]D-Fender | You should be using ODBC for all of that |
| 15:55.43 | newtonr | phormulate, sip.conf right? i'm tired and can't remember if you were talking about TLS for manager or chan_sip |
| 15:55.46 | [TK]D-Fender | or external use via AGI, etc |
| 15:56.20 | qakhan | we were discussing the realtime queue_log 2 days ago. |
| 15:56.32 | phormulate | ya, sip.conf newtonr |
| 15:56.43 | phormulate | pre external pjsip 11.x newtonr |
| 15:56.53 | qakhan | this the problem, i could not install app_addon_sql_mysql, cdr_addon_mysql and res_config_mysql |
| 15:57.49 | qakhan | and therefore i had queue_log realtime problem. but asterisk-addon-1.6.2.4 works fine wit asrerisk 11.19 and queue_log perfectly |
| 15:58.09 | [TK]D-Fender | Everything is ODBC now |
| 15:58.13 | [TK]D-Fender | Go set it up |
| 15:59.00 | qakhan | so cdr also work through ODBC? |
| 15:59.12 | newtonr | phormulate, huh. IDK. you might post your configs and netstat output on a mailing list post for others to look over. i haven't seen anyone else with this issue so hopefully it is just a misconfiguration. |
| 16:00.25 | phormulate | will do |
| 16:00.32 | phormulate | first coffee and breakfast |
| 16:00.33 | phormulate | bbiab |
| 16:00.34 | phormulate | ty |
| 16:00.43 | [TK]D-Fender | [11:58][TK]D-FenderEverything is ODBC now |
| 16:04.32 | qakhan | [TK]D-Fender ok thanks |
| 16:08.01 | *** join/#asterisk RobertLaptop (~rmiddle@74.112.203.154) |
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| 16:17.33 | ChannelZ | cyford: thanks |
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| 16:27.08 | zaf | ok, i'm confused. for presence subscription, does each extension need to be configured to list every other extension that will receive hints for it? |
| 16:28.07 | zaf | in other words, in the dialplan, for extension 1000, do i need to have something like "exten => 1000,hint,SIP/1000&SIP/1002&SIP/1002" ? |
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| 16:33.47 | file | zaf, a hint is a mapping of an extension number to a device (or devices), it's not a permission list |
| 16:33.58 | file | without it Asterisk has no idea that '1000' is a device |
| 16:35.01 | zaf | ok, so i just need SIP/1000 as the mapping, right? |
| 16:42.58 | [TK]D-Fender | to watch SIP/1000, yes |
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| 16:59.53 | zaf | still can't figure out why i'm not getting lights on for the D70 phones.. these are the relevant pastes if anything shows up as incorrect - http://fpaste.org/260760/78059714/ |
| 17:05.59 | [TK]D-Fender | If you put the hints in a context each then your phone can't look at all of them, only the one you pointed it to |
| 17:07.03 | zaf | hmm, so maybe i should use subscribehints? |
| 17:07.08 | Get_The_Fish | Hello, I've identified an issue with chan_pjsip against PJPROJECT 2.4 on Asterisk 13.5.0 where Asterisk is not conforming with SIP specs. Asterisk is not replying to SIP INFO packets. See http://pastebin.com/KdtjUSTP, line 249 is the first SIP INFO packet. |
| 17:07.49 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 17:08.09 | [TK]D-Fender | zaf: You should make a single context with all the hints and INCLUDE it in each of your device-specific contexts so they all have access |
| 17:08.36 | zaf | [TK]D-Fender, will try that, thanks |
| 17:11.36 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 17:17.18 | Get_The_Fish | My understanding is that Asterisk should be replying to all SIP INFO packets with a 200 OK. |
| 17:19.31 | Get_The_Fish | Also note that the above referenced platform has dtmf_mode=info configured for this endpoint, so I believe that Asterisk should be replying to these packets. |
| 17:19.40 | [TK]D-Fender | Get_The_Fish: Just add the issue to the tracker |
| 17:20.17 | Get_The_Fish | You think my description and documentation of the issue is sufficient to keep it from getting rejected? |
| 17:22.20 | *** join/#asterisk sekil (~sekil@78.24.109.191) |
| 17:33.44 | [TK]D-Fender | If you include all the debug information and configs, version info etc it'll be fine |
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| 18:05.55 | Get_The_Fish | So I am getting an endless loop using Apple's Safari 8.0.8 on Mac 10.10.5 of certificate denial and requests to sign a certificate when going to issues.asterisk.org |
| 18:06.07 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
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| 18:07.12 | [TK]D-Fender | Go post that up as well then |
| 18:07.19 | Get_The_Fish | lol |
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| 18:29.32 | Get_The_Fish | So when posting an issue should I just leave the pastebin link, which will expire, or should I paste in what I have in pastebin? |
| 18:30.45 | WIMPy | You can attach text files there. |
| 18:32.24 | Get_The_Fish | WIMPy: I dont see that capability |
| 18:32.53 | Get_The_Fish | WIMPy: Nevermind, it's not in the edit screen but when you view it you can. |
| 18:42.45 | zaf | [TK]D-Fender, no dice so far.. i'm wondering if i have my subscribe_to incorrect in my contacts xml |
| 18:42.50 | Get_The_Fish | Great, so its a duplicate of an unresolved issue. Boo. |
| 18:43.07 | [TK]D-Fender | zaf: Do you see it subscribing to them? |
| 18:43.14 | zaf | [TK]D-Fender, no |
| 18:43.26 | [TK]D-Fender | zaf: This IS with SIP debug enabled, right? |
| 18:43.52 | zaf | no, just showing subscriptions. turning debug on now to see what i can see |
| 18:44.09 | [TK]D-Fender | That's what I meant by "have you looked" |
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| 19:10.25 | zaf | [TK]D-Fender, watching debug, i'm only seeing what appears to be the phone subscribing to itself |
| 19:13.16 | [TK]D-Fender | Looks like you didn't add the right contacts on the phone... |
| 19:13.59 | zaf | the contact buttons can dial each other when i press them |
| 19:19.03 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:26.12 | [TK]D-Fender | Show us some of those subscription attempts so we can verify |
| 19:26.40 | zaf | k, rebooting a phone now, will paste some fresh ones |
| 19:31.24 | zaf | http://fpaste.org/260818/40790274/ |
| 19:39.05 | zaf | here'ss a much more complete one, since i may have not pasted the right entries http://fpaste.org/260820/90723144/ |
| 19:39.16 | [TK]D-Fender | yup, sure looks like you added a contact for the phone itself there... |
| 19:43.06 | zaf | do you mean the phonebook xml file may be wrong? |
| 19:43.48 | *** join/#asterisk othebirony (~bchen@50-205-197-158-static.hfc.comcastbusiness.net) |
| 19:44.08 | [TK]D-Fender | yup |
| 19:44.22 | [TK]D-Fender | The phone is asking for what it is asking for, and the request doesn't look sane.... |
| 19:46.01 | *** join/#asterisk andrew42 (4077862e@gateway/web/freenode/ip.64.119.134.46) |
| 19:46.54 | andrew42 | Hi everyone. I've noticed that there haven't been any updated RPMs for centos/rhel 6. I couldn't find anything in the documentation for the repos saying they had been moved or discontinued. Anyone know what is doing on? |
| 19:47.25 | zaf | [TK]D-Fender, this is my phonebook - http://fpaste.org/260823/44079109/ |
| 19:52.56 | *** part/#asterisk othebirony (~bchen@50-205-197-158-static.hfc.comcastbusiness.net) |
| 19:53.02 | [TK]D-Fender | Not familiar with specific format but it does feel "right" |
| 19:53.18 | zaf | yeah, i know this is getting digium-specific :/ |
| 19:54.10 | [TK]D-Fender | Well evey phone is "specific" really, I just don't have any experience with that one specifically |
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| 20:20.15 | Ice_Strike | Which is faster for phone provisioning? FTP or HTTP? |
| 20:25.49 | [TK]D-Fender | No logical difference |
| 20:28.44 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:29.12 | [TK]D-Fender | packs up to head home |
| 20:30.36 | Ice_Strike | Ah ok |
| 21:01.33 | Synthase_ | zaf: DPMA? |
| 21:10.02 | zaf | Synthase_, no |
| 21:12.30 | Synthase_ | zaf: Trying to setup BLF on the phones, or have I missed something? |
| 21:15.07 | zaf | Synthase_, correct |
| 21:15.15 | zaf | contacts show up, but the indicators don't operate |
| 21:15.32 | Synthase_ | Verified the context of your SmartBLF XML file? |
| 21:15.44 | Synthase_ | s/context/syntax/ |
| 21:16.00 | zaf | Synthase_, i believe so. i have a digium guy on the phone now going over it |
| 21:17.15 | Synthase_ | The contact ID's between contacts XML and the SmartBLF file need to match, can be a little tricky. |
| 21:17.27 | Synthase_ | If Andrey, tell him hello |
| 21:17.39 | zaf | it is |
| 21:19.46 | WIMPy | Multiple files? |
| 21:19.52 | zaf | Synthase_, yeah, they do match. |
| 21:19.58 | zaf | multiple files, yes |
| 21:20.34 | Synthase_ | WIMPy: |
| 21:20.49 | WIMPy | Why multiple files? Sounds - err - unfortunate. |
| 21:20.49 | Synthase_ | Yes, contacts and smarblf directives are two files |
| 21:21.01 | Synthase_ | *SmartBLF |
| 21:21.22 | Synthase_ | ¯\_(ã)_/¯ It's how D-phones work without DPMA |
| 21:21.41 | WIMPy | Special features? |
| 21:22.30 | Synthase_ | Correct. https://wiki.asterisk.org/wiki/display/DIGIUM/Smart+BLF |
| 21:22.58 | WIMPy | I mean, you need more than BLF? |
| 21:23.29 | zaf | nah, just getting the contacts in there and lighting properly |
| 21:23.53 | WIMPy | The normal contacts are all you need for that. |
| 21:25.20 | zaf | how would you do normal contacts for blf without smart_blf? |
| 21:25.53 | WIMPy | Just add a "subscribe_to=" to the contacts. |
| 21:26.53 | zaf | isn't that considered part of the smart blf function? |
| 21:27.29 | WIMPy | NFI. But it definitely works just in the contacts.xml. |
| 21:29.01 | WIMPy | I thought SmartBLF is the fancy stuff beyond normal SIP subscriptions. |
| 21:30.06 | zaf | andrey saying that without DPMA it wont work |
| 21:30.46 | WIMPy | I took it from the samples in the wiki and it surely works. |
| 21:31.33 | zaf | hmm |
| 21:31.46 | zaf | what do you use for subscribe_to? |
| 21:31.54 | zaf | just the SIP endpoint name? |
| 21:32.03 | WIMPy | The extension with the hint. |
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| 21:32.51 | WIMPy | You need to hafe hintes configured in your dialplan for subscriptions to work. |
| 21:32.56 | WIMPy | v |
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| 21:35.03 | WIMPy | "hintes"? An I starting to learn french? |
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| 21:42.52 | Synthase_ | It's contagious |
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