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| 01:36.09 | Mango45 | [05:31] <Mango45> Is session-expires ignored if session-timers is set to accept? |
| 01:36.15 | Mango45 | Answer: No it is not. |
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| 07:00.10 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.2 (2016/09/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:33.50 | Rasputin3711 | Set(GROUP()) - counts channels? |
| 08:40.05 | ChannelZ | not exactly but it can in part be used for that |
| 08:40.48 | ChannelZ | GROUP sets the current channel in a group.. and then you can use GROUP_COUNT() to see how many channels are in that group |
| 08:42.01 | Rasputin3711 | exten => _101!,1,Macro(limit-group,101,1) |
| 08:42.01 | Rasputin3711 | exten => _102!,1,Macro(limit-group,102,2) |
| 08:42.27 | Rasputin3711 | macro: |
| 08:42.28 | stefan27 | Yeah, so if your incoming calls goes through dialplan doing Set(GROUP()=mygroup34) then you could query ${GROUP_COUNT(mygroup34)} to see current number of incoming channels |
| 08:42.33 | Rasputin3711 | <PROTECTED> |
| 08:42.33 | Rasputin3711 | <PROTECTED> |
| 08:42.33 | Rasputin3711 | <PROTECTED> |
| 08:42.56 | Rasputin3711 | It will be correct? 101 max to one incomming call? |
| 08:45.43 | stefan27 | should work but try it |
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| 08:54.02 | jkroon | hi all, transcoding problems with asterisk getting stuck after g729 inadequate licenses - do i take that up direct with digium support or discuss here? |
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| 09:02.30 | ChannelZ | well probably them if your licenses aren't working |
| 09:04.30 | ChannelZ | does 'g729 show licenses' say plausible things about how many you have? |
| 09:05.45 | ChannelZ | or 'show g729' maybe.. I forget |
| 09:06.02 | ChannelZ | (I don't have any so I don't have any g729 modules loaded) |
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| 09:09.01 | jkroon | ChannelZ, client is running out of licenses, but asterisk shouldn't start reporting more active calls than active channels (eg, 140 active channels with 800 active calls) |
| 09:09.20 | jkroon | once we hit that upper 72 limit it's bye bye, need to restart asterisk to recover. |
| 09:09.27 | jkroon | i've sent an email to support@digium.com |
| 09:10.22 | ChannelZ | oh I see |
| 09:11.01 | ChannelZ | I misread the 'stuck' part of your query |
| 09:11.15 | jkroon | hehe, thanks for reading :) |
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| 09:48.51 | jkroon | Error in my_thread_global_end(): 18 threads didn't exit <-- something to worry about? |
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| 12:00.12 | Samot | Sounds like youre closing the db comnection before threads are closed. |
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| 16:33.29 | jkroon | Samot, asterisk -rx "core restart when convenient" ... |
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| 16:37.13 | igcewieling | jkroon: I used to have similar problems, but we threw money at it and solved the issue. 8-) |
| 16:41.32 | jkroon | igcewieling, i'm assuming you're referring to the codec issue and not the mysql thread stuff? |
| 16:41.57 | igcewieling | jkroon: yes. I was partially joking. |
| 16:42.33 | jkroon | :) ... it's been a long day of staring at BGP routing tables ... unfortunately my sense of humour has been somewhat eroded. |
| 16:42.36 | igcewieling | we bought a couple of sangoma transcoding cards -- expensive, but less so than dealing with codec BS. Sounds like you have licenes, but they are not being released, is that correct? |
| 16:43.23 | jkroon | they're working mostly, but once you've tried to exceed the limit you're screwed. calls doesn't get released. the codecs doesn't get released. |
| 16:43.38 | jkroon | so core show channels will show higher call count than channel count ... |
| 16:43.58 | jkroon | and even after everybody has hung up their calls it still shows max licenses in use. |
| 16:44.34 | drmessano | Which version of Asterisk? |
| 16:50.33 | drmessano | I guess I am on ignore |
| 16:53.14 | freebs | i think so... |
| 16:55.41 | drmessano | Doesnt surprise me |
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| 17:16.45 | danielyk | Hi, our username for an pjsip trunk is an email-adress. How can I type @-character correct for the parser? |
| 17:16.48 | Samot | Could not use g729. You know, it's like 2016. |
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| 17:26.19 | jameswf | it is still 1987 in some parts of the world |
| 17:26.57 | WIMPy | Not too bad considering most are stuck in 1992. |
| 17:27.13 | danielyk | Hi, our username for an pjsip trunk is an email-adress. How can I type @-character correct for the parser? |
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| 17:41.47 | drmessano | I don't really see G729 taking off without a G729 Pro iPhone app |
| 17:43.30 | WIMPy | Is G.729 no longer #asterisks darling? |
| 17:44.19 | drmessano | G729 is hugely popular amongst mom's basement ITSPs |
| 17:44.55 | drmessano | I use it because I don't have a basement |
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| 17:45.17 | drmessano | But for the most part, bandwidth has become so much less expensive |
| 17:45.30 | Samot | And bigger. |
| 17:45.46 | drmessano | I've stopped deploying ATAs forced to G729 |
| 17:46.03 | drmessano | Because even crappy home internet has no problem with G711 |
| 17:47.38 | Samot | Oh yeah |
| 17:49.03 | Samot | Speaking of basement ITSPs.. |
| 17:49.15 | Samot | I find one of your gems when going through my bookmarks. |
| 17:49.17 | Samot | http://imgur.com/ykmkfaL |
| 17:49.29 | drmessano | YESH! |
| 17:49.38 | drmessano | ...and i;m only 14!!! |
| 17:50.13 | drmessano | Dad sells German Shepherd Puppies and wanted to offer something new, so he asked me to learn Trixsterisk |
| 17:50.21 | drmessano | Here I am! |
| 17:51.06 | drmessano | I wonder what happened to that guy |
| 17:51.19 | drmessano | Probably changed his mind and started selling Pokemon GO maps |
| 17:51.20 | Samot | hahah |
| 17:52.18 | drmessano | "So my Dad bought a NUC, right..." |
| 17:53.17 | Samot | Sometimes you do wonder what happened to people and what they were "attempting" to do. |
| 17:53.34 | Samot | Like I wonder if Haris ever got *anything* working. |
| 17:53.49 | Samot | Or he's just not here anymore because they fired him. |
| 17:55.24 | Samot | Or the kid last week in #freepbx that decided that last Friday was a good day to learn Asterisk/FreePBX to interview for a job this week to manage a system. |
| 17:55.58 | Samot | Oh and that job including updating an FreePBX 2.10/Asterisk 1.8 system. |
| 17:56.10 | Samot | So that's where he wanted to learn. |
| 17:56.12 | Samot | woooo. |
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| 18:03.14 | drmessano | It never ends, I guess |
| 18:05.44 | Samot | It's not like cramming for a test. |
| 18:05.56 | Samot | With questions you'll probably never be asked again. |
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| 18:24.36 | drmessano | Lunch is really tasty cooked over a Samsung phone |
| 18:24.48 | drmessano | Tastes a little metallic, but wow did it heat up fast |
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| 18:34.35 | *** join/#asterisk dp (~gms8994@freyr.expansesoftware.com) |
| 18:35.24 | dp | I'm trying to find a call recording that should exist, but I can't seem to find through the FreePBX web interface; is there a way I can look directly in the database to find the call information? |
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| 18:56.25 | danielyk | I received res_pjsip_outbound_registration.c:878 handle_registration_response: Fatal response '403' received from 'sip:tel.t-online.de:5060' on registration attempt to 'sip:012345678987@tel.t-online.de:5060', stopping outbound registration |
| 18:57.04 | danielyk | Does Asterisk send REGISTER any more? |
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| 19:02.13 | danielyk | Can I set "forbidden_retry_interval" to send REGISTER-Requests also if an 403 is received? |
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| 20:12.34 | Samot | tuxdood: yes. |
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| 20:26.20 | Samot | tuxd00d: yes. |
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| 20:53.38 | igcewieling | polycom phone assembly -> phone model info for anyone who might find it helpful. I created it for an internal project. http://pastebin.ca/3727635 |
| 20:55.20 | robmal | <3 Thanks! |
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| 21:18.25 | danielyk | I use PJSIP and Asterisk 13 and migrate currently from chan_sip. In REGISTER-Request there is Contact: <sip:0123456788@192.168.1.12:5060>. How can Asterisk use the public IP instead of the private IP? |
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| 21:20.18 | danielyk | @file Can you say something to this theme? |
| 21:21.07 | *** join/#asterisk J0hnStee- (~J0hnSteel@92.55.116.125) |
| 21:22.24 | file | have you configured it for being behind NAT? |
| 21:25.24 | danielyk | Not yet, I have no static public IP. I am able to set local_net, but cannot set external_media_address and external_signaling_address because of the dynamic IP |
| 21:25.47 | file | they have to be set in order to have an IP address to put in the signaling |
| 21:26.44 | danielyk | How can I do this with my dynamic IP (if my router lost the connection to the ISP, I get an new address) |
| 21:27.06 | file | it accepts a hostname and will periodically refresh |
| 21:27.26 | *** join/#asterisk ketas (~ketas@200-62-46-176.dyn.estpak.ee) |
| 21:28.09 | danielyk | Oh, thats brilliant. I am suprised. So I set local_net to 192.168.1.0/24 and external_media_address & external_signaling_address to my dyndns service. Right? |
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| 21:28.33 | file | yes |
| 21:28.42 | danielyk | Thank you so much!!! |
| 21:45.49 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 21:46.30 | danielyk | @file I get now: [2016-10-12 23:45:58] WARNING[5232]: pjproject:0 <?>: tsx0x7f15d4032 ...Error sending Response msg 200/INVITE/cseq=1 (tdta0x7f15d406dd70): Operation not permitted |
| 21:50.20 | [TK]D-Fender | SOunds like you're prevented from creating the packet which smells like a firewall issue |
| 21:51.04 | danielyk | I am running iptables in front of asterisk. I will deactivate and try it again |
| 21:51.47 | [TK]D-Fender | I'd recommend LOOKING at it before just deactivating |
| 21:52.04 | [TK]D-Fender | Never an excuse to leave your eyes shut and blindly swat at switches |
| 21:52.53 | danielyk | I am sMy firewall rules are thought out and tested |
| 21:53.05 | danielyk | *My firewall rules are thought out and tested |
| 21:53.18 | [TK]D-Fender | And in the real-world mistakes happen |
| 21:53.29 | [TK]D-Fender | And things you didn't think about come into play |
| 21:53.41 | danielyk | Yes you are right |
| 21:54.28 | danielyk | If I remove external_media_address and external_signaling_address from transports I do not get this warning message |
| 21:55.01 | danielyk | the firewall should not have something to do with that, because I only want to set the correct public contact header |
| 21:55.58 | [TK]D-Fender | we aren't seeing what it's trying to put onto the networking stack yet. No SIP debug to examine |
| 21:56.11 | [TK]D-Fender | And no firewall to dump to compare it to. |
| 21:56.31 | file | and you may also want to check the startup log to see if it was fine with the values provided |
| 21:57.07 | danielyk | I think the values should be fine. I checked it with pjsip show transport 0.0.0.0-udp |
| 21:57.26 | danielyk | I have not seen something in the startup log |
| 21:59.12 | danielyk | My provider wirtes following in his technical specification: General procedure for REGISTER Message answered with a 403: General a 403 is an Indication that the user is not provisioned within the HSS. Nevertheless if 403 (Forbidden) has been received as a response to a REGISTER request, a further registration attempts shall be done after 15 sec. In case further 403 responses received with the same URI in the Contact header field |
| 21:59.12 | danielyk | <PROTECTED> |
| 21:59.35 | danielyk | I must wait 30-60 minutes to send you an relevant sip trace |
| 22:09.00 | *** join/#asterisk danielyk_ (~danielyk@p5B21B789.dip0.t-ipconnect.de) |
| 22:13.29 | danielyk_ | REGISTER sip:tel.t-online.de:5060 SIP/2.0 |
| 22:13.29 | danielyk_ | Via: SIP/2.0/UDP 91.33.xxx.xxx:45060;rport;branch=z9hG4bKPj95e320b1-bbcc-4a87-9287-62eac286911d |
| 22:13.29 | danielyk_ | From: <sip:0123456789@tel.t-online.de>;tag=71922be9-f13f-4fdc-a05b-3a640776677c |
| 22:13.29 | danielyk_ | To: <sip:0123456789@tel.t-online.de> |
| 22:13.31 | danielyk_ | Call-ID: 4a52d382-aa09-4fb6-97b6-e366c3d8ed1e |
| 22:13.31 | danielyk_ | CSeq: 6339 REGISTER |
| 22:13.33 | danielyk_ | Contact: <sip:0123456789@91.33.xxx.xxx:45060> |
| 22:13.33 | danielyk_ | Expires: 480 |
| 22:13.35 | danielyk_ | Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER |
| 22:13.35 | danielyk_ | Max-Forwards: 70 |
| 22:13.37 | danielyk_ | User-Agent: FPBX-13.0.188.9(13.11.2) |
| 22:13.37 | danielyk_ | Content-Length: 0 |
| 22:14.16 | danielyk_ | res_pjsip_outbound_registration.c:878 handle_registration_response: Fatal response '403' received from 'sip:tel.t-online.de:5060' on registration attempt to 'sip:0123456789@tel.t-online.de:5060', stopping outbound registration |
| 22:18.03 | danielyk_ | That refers to the note in the technical specifications of my ISP. So I have added added the public hostname to the transports section, get also an 403 Forbidden and the WARNING from asterisk |
| 22:18.29 | *** join/#asterisk monsterco (~monsterco@70.50.209.133) |
| 22:23.12 | danielyk_ | @file Do you see a mistake? |
| 22:29.49 | Samot | danielyk: Please do flood the channel with multi-line pastes. |
| 22:29.52 | Samot | ~pb |
| 22:29.52 | infobot | i guess 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. |
| 22:29.58 | Samot | ^^ Use that. |
| 22:30.15 | *** part/#asterisk monsterco (~monsterco@70.50.209.133) |
| 22:31.12 | Samot | danielyk_: We would need to see the entire attempt. You REGISTER they send back a 401 Unauthorized you send back REGISTER with your creds and it accepts or rejects it. |
| 22:31.16 | danielyk_ | Sorry, I will use the service in the future |
| 22:32.48 | Samot | Has the provider told you why they are returning a 403 Forbidden to your requests? |
| 22:33.14 | danielyk_ | "In case further 403 responses received with the same URI in the Contact header field REGISTER requests are allowed with a random delay of 30- 60 minutes." |
| 22:33.25 | Samot | That's not why. |
| 22:33.40 | Samot | That's a result of multiple 403's. |
| 22:33.59 | danielyk_ | my contact header was always the same, because I used my private IP |
| 22:34.24 | Samot | Registrations don't care if you come from the same IP all the time. |
| 22:34.29 | Samot | Or they shouldn't. |
| 22:34.39 | Samot | That makes no sense whatsoever. |
| 22:34.41 | danielyk_ | General a 403 is an Indication that the user is not provisioned within the HSS. |
| 22:34.59 | danielyk_ | That says the specs, but I do not know what "HSS" means |
| 22:35.14 | Samot | Have you actually engaged their support? |
| 22:35.42 | Samot | They are returning that response for a reason. You need to figure out why and what you or they can do to resolve it. |
| 22:36.03 | danielyk_ | Yes, I tried, but the company is tooo big to get an good answer |
| 22:36.47 | Samot | So no one in that company can tell you why they would return a 403 to your requests? |
| 22:38.41 | danielyk_ | My asterisk and the provider works really fine together till my ip changes (because the router e.g. rebooted). Then asterisk sends his REGISTER and get an 403 and say "stopping outbound registration". Then I can wait 30 minutes and send new REGISTER and it works. |
| 22:39.03 | Samot | Well then that's a provider issue. |
| 22:39.58 | danielyk_ | I get no one to the telephone, because the normal supporters do not know anything from SIP... |
| 22:40.32 | Samot | So a SIP provider has a support team that knows nothing about SIP? |
| 22:40.36 | danielyk_ | I do not think it is a provider problem, Telekom Germany is too big for that. And with chan_sip it works great... |
| 22:40.38 | Samot | Time to change providers. |
| 22:41.13 | danielyk_ | They have their experts but these experts are not at the phone... |
| 22:42.19 | Samot | Well then it must be in how PJSIP is configured for that peer. |
| 22:44.23 | danielyk_ | I think that the provider do not let me re-register because I use the same contact header as before the ip changes. So I tried to set external_media_address and external_signaling_address to my dyndns hostname. When asterisk now receives an incoming INVITE, following warning comes up: [2016-10-12 23:45:58] WARNING[5232]: pjproject:0 <?>: tsx0x7f15d4032 ...Error sending Response msg 200/INVITE/cseq=1 (tdta0x7f15d406dd70): |
| 22:44.24 | danielyk_ | <PROTECTED> |
| 22:45.33 | danielyk_ | Without these both values pjsip works fine (until these 30 minutes delay after ip changes) |
| 22:49.28 | Samot | Well you're contact shouldn't contain the private IP. |
| 22:49.49 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 22:52.06 | danielyk_ | Yes, so I added the two values. The REGISTER is with the correct Contact. But Asterisk inbound and outbound calls do not work any more. That sounds like an bug... |
| 22:52.30 | Samot | Well show your pjsip.registrations.conf |
| 22:52.48 | Samot | Then show an actual full REGISTER attempt from start to finish. |
| 22:53.19 | danielyk_ | with the private or public IP? |
| 22:53.40 | Samot | And the pjsip.transports.conf |
| 22:53.50 | Samot | With the one that works but you have no aduio. |
| 22:53.52 | Samot | audio. |
| 23:03.36 | *** join/#asterisk artisangoose (~textual@66.119.2.181) |
| 23:03.45 | danielyk_ | http://pastebin.ca/3727944 |
| 23:05.13 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 23:06.25 | Samot | That's the entire register? |
| 23:06.43 | Samot | I thought you said it registered but there was no audio? |
| 23:07.09 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ijvqdircvnlgajxb) |
| 23:07.26 | danielyk_ | Not now, I have to wait 30 minutes |
| 23:07.46 | danielyk_ | I send you also a trace from an invite |
| 23:11.47 | danielyk_ | That is the INVITE, I cannot find an Forbidden in my terminal, but the call does not worked http://pastebin.ca/3727945 |
| 23:12.39 | file | what happens if you don't set external_media_address? |
| 23:13.18 | danielyk_ | It works fine till I lost my public IP |
| 23:13.59 | danielyk_ | I can call inbound/outbound without problems (It works like chan_sip) |
| 23:14.20 | file | then that's the problem of why that error comes up, and PJSIP may not support the scenario under which you are using it fully (with an IP that is changing) |
| 23:15.46 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 23:15.59 | danielyk_ | Asterisk have to work if I lost my public IP. |
| 23:16.47 | file | https://issues.asterisk.org/jira/browse/ASTERISK-26458 |
| 23:22.45 | danielyk_ | I have read the issue but do not think that this is my problem |
| 23:23.26 | file | it's part of the problem |
| 23:25.33 | danielyk_ | My problem is that asterisk does not use my external ip correctly (otherwise it would function; it does his job great with the private ip). The issues sounds like, the external ip is not fetched correctly. That is not so, the SDPs contains the right, public ips. |
| 23:25.43 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 23:26.10 | danielyk_ | Have you looked into the INVITE? There where my public and private IP in one telegram... |
| 23:26.45 | file | your invite contains the hostname in the SDP, which may be the cause of the weirdness with sending response |
| 23:27.59 | danielyk_ | Yes, i think that could be the reason for the WARNING during incoming calls |
| 23:28.21 | danielyk_ | Is that a bug or something else? |
| 23:41.57 | *** part/#asterisk danielyk_ (~danielyk@p5B21B789.dip0.t-ipconnect.de) |
| 23:48.40 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |