| 00:08.38 | *** join/#asterisk CountRumford (ylorb@gave-jessica-and-ashlee-simpson-my.stickynapalm.com) |
| 00:08.50 | CountRumford | is this as-close as one can get to a telecom channel? |
| 00:08.54 | CountRumford | telecom pointed me in this direction |
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| 00:14.04 | *** mode/#asterisk [+o newtonr_] by ChanServ |
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| 00:19.15 | *** 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 |
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| 01:32.16 | Samot | Well apparently he was unhappy with his wait time and disconnected. |
| 01:32.24 | Samot | I'm sure he'll try back. |
| 01:33.08 | igcewieling1 | I wonder who mr/ms telecom is |
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| 08:10.12 | georgemp | anybody have any recommendations for a sip provider with DID in the US (and possibly India)? I've looked at callcentric (they don't seem to support tcp or srtp/tls), voip.ms (don't allow accounts from India), and Twilio. Looking to see if there are any other providers I should consider..thanks |
| 08:13.21 | sekil | if anyone is interested OpenSIPS summit is live at https://www.youtube.com/watch?v=anmyMC6Ovl8 |
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| 08:28.42 | Dovid | try Telnyx.com |
| 08:29.46 | Dovid | georgemp: try Telnyx.com. They have US numbers. Indian numbers are hard to come by due to regulations in India. I know Voxbone.com and tollfreeforwarding.com have India but only freephone numbers and the callers line must support intl. calling |
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| 08:44.49 | TandyUK | hey guys any homer users here? |
| 08:45.11 | TandyUK | confused trying to get it working... does homer have an agent to listen for HEP packets, or do we need to configure kamailio for that? |
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| 09:02.50 | georgemp | Dovid: thank you..will check those out |
| 09:17.34 | dan_j | Hi. I've got ringinuse set to 1 for all queues. I have a selection of DIDs. Each DID goes into it's own queue. All SIP phones are members of all queues. When a call comes in, an agent answers. When a 2nd call comes in for that queue, if the agent is still on the phone, it does not ring that agent. But if a 2nd call comes in for a different queue, it does |
| 09:17.34 | dan_j | ring. |
| 09:17.47 | dan_j | Any ideas why it might not be ringing? |
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| 09:43.52 | DanQuinney | TandyUK: homer runs its own kamailio install to capture the HEP packets |
| 09:46.01 | dan_j | My issue has been occuring since upgrading from v11 to v13. |
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| 11:08.05 | TandyUK | [2017-05-02 12:07:18] == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 11:08.06 | TandyUK | [2017-05-02 12:07:18] WARNING[23597]: tcptls.c:683 handle_tcptls_connection: FILE * open failed! |
| 11:08.23 | TandyUK | still getting loads of these, any clues as to what i can do to make asterisk give me more info about what that is relating to? |
| 11:08.45 | TandyUK | "some random TLS connection failed" is really unhelpful for debugging the cause |
| 11:10.00 | TandyUK | 100+ extensions are online and working perfectly, im having issues with 2 specific handsets, and having turn TLS/SRTP OFF on both ofthem, im suddenly getting MORE of these errors |
| 11:10.03 | TandyUK | go figure |
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| 12:21.36 | mrcirca | hello can someone tell me what is the limit of sip peer latency for monitoring? |
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| 13:33.15 | ramirezb | Hello people. It is my first time in this channel and I need some help concerning Asterisk behaviour. |
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| 13:35.36 | ramirezb | I deployed one month ago an Asterisk server in my office with their stored version which is the 13.9. There are several deployments in physical machines but I am dealing with the virtual ones (in openstack). |
| 13:36.06 | ramirezb | the problem is since 2 weeks ago the virtual machines restarts (only asterisk) with no reason. |
| 13:37.32 | ramirezb | and the logs don´t say anything. Just, the welcome message when you start the service. |
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| 13:37.48 | [TK]D-Fender | You'll need to find a core dump, etc to trace it by |
| 13:37.53 | [TK]D-Fender | ~collectdebug |
| 13:37.53 | infobot | well, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 13:38.14 | ramirezb | mmmmmm |
| 13:39.56 | ramirezb | I followed that reference https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information but the log doesnt give me any interesting information |
| 13:41.54 | ramirezb | for example, the last info in my log.txt before rebooting is the following |
| 13:42.43 | ramirezb | [May 2 15:05:26] DEBUG[3971] res_pjsip_session.c: Response is 486 Busy Here |
| 13:42.57 | ramirezb | [May 2 15:05:26] DEBUG[3986][C-0000000e] channel.c: Hanging up channel 'PJSIP/pbx_main-00000018' |
| 13:43.06 | ramirezb | [May 2 15:05:26] VERBOSE[3986][C-0000000e] app_dial.c: Everyone is busy/congested at this time (1:1/0/0) |
| 13:43.09 | ramirezb | thats all |
| 13:44.06 | [TK]D-Fender | doesn't say anything there |
| 13:44.17 | [TK]D-Fender | if * crashes there should be a core dump |
| 13:44.27 | [TK]D-Fender | make sure of that |
| 13:44.46 | [TK]D-Fender | taht is if * itself is the one thing exclusively rstarting on that box |
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| 13:45.21 | ramirezb | i see |
| 13:45.59 | ramirezb | i am going to check it know. btw, sorry if my questions is kind of naive. I am new in the asterisk world. |
| 13:46.07 | ramirezb | are* |
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| 13:50.36 | [TK]D-Fender | Nothing wrong with your quetions. You just need some actual evidence that * itself core dumped & crashed, not something else that just took it out without the ability to do so |
| 13:51.01 | ramirezb | when asterisk crashes |
| 13:51.12 | ramirezb | the core file where it is? |
| 13:52.08 | [TK]D-Fender | Hrm, not the link you need there... |
| 13:52.22 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 13:52.28 | [TK]D-Fender | Sorry about that. |
| 13:52.40 | [TK]D-Fender | I have seen the previous link thrown around and hadn't validated it |
| 13:52.52 | [TK]D-Fender | This should do it |
| 13:53.23 | ramirezb | Thanks |
| 13:53.26 | ramirezb | OMG |
| 13:53.36 | ramirezb | omw |
| 13:58.05 | ramirezb | i cannot find it |
| 13:58.13 | ramirezb | i am not sure if it is generated |
| 13:58.33 | ramirezb | the core file I mean |
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| 14:06.58 | ramirezb | I thing core is not generated because i cannot find it |
| 14:07.54 | [TK]D-Fender | It'd only get generated if * itself crashed in a "normal" way I believe |
| 14:08.07 | [TK]D-Fender | So perhaps it isn't * itself that is dying? |
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| 14:08.31 | [TK]D-Fender | (specificically) |
| 14:10.12 | ramirezb | mmmmmmmmmm |
| 14:11.41 | ramirezb | i thing asterisk is not crashing |
| 14:11.45 | ramirezb | just reloading |
| 14:11.51 | ramirezb | with no reason |
| 14:13.10 | [TK]D-Fender | reloading ... or actually completely restarting? |
| 14:13.13 | [TK]D-Fender | does it kill calls? |
| 14:14.30 | ramirezb | i´ve it in realtime with 20k endopints aprox. It does not kill calls but becomes unavailable during endpoints reloading. |
| 14:15.22 | ramirezb | it takes about 15 minutes for checking all the endpoints before enabling new calls again. |
| 14:15.54 | [TK]D-Fender | Ok, then we aren't talking a "crash" |
| 14:16.03 | [TK]D-Fender | which is what yuo said walking in. |
| 14:16.17 | [TK]D-Fender | if it is just doing a reload process... really can't help you there... |
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| 14:16.28 | ramirezb | mmm |
| 14:17.10 | ramirezb | that´s the reason because i am a bit anxious XD |
| 14:17.20 | ramirezb | because is too weird. |
| 14:18.04 | ramirezb | anyway, thank you so much for your support :) |
| 14:18.24 | ramirezb | if i can find the solution i´ll notice it. |
| 14:19.20 | Mr_Pleb_Mgoo | thats a decent amount of end points |
| 14:19.34 | Mr_Pleb_Mgoo | its not struggling for memory/RAM or anything? |
| 14:20.37 | ramirezb | i checked it and dont |
| 14:21.08 | ramirezb | this machine handles with 20~30 calls as much. |
| 14:21.10 | file | there's an issue, https://issues.asterisk.org/jira/browse/ASTERISK-26806 |
| 14:22.55 | ramirezb | thanks people |
| 14:23.29 | file | that's for .conf specifically, but changes went into later versions to improve the realtime case |
| 14:23.51 | ramirezb | i see |
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| 14:38.51 | dan_j | Hi. I've got a queue with ringinuse set to on, but when an agent answers a call from that queue, they dont receive any other calls from that queue until their call ends. |
| 14:38.58 | dan_j | But they do receive calls from other queues. |
| 14:39.32 | dan_j | Any ideas what could be causing that? |
| 14:39.35 | dan_j | They are SIP phones |
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| 14:50.24 | Samot | By "on" you mean "yes"? |
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| 15:51.55 | rrittgarn | join #asterisk-ari |
| 15:52.05 | rrittgarn | darn missing slashes... |
| 15:52.06 | rrittgarn | sorry |
| 15:53.51 | [TK]D-Fender | #emo |
| 15:54.17 | rrittgarn | (//_-) |
| 15:54.22 | TandyUK | can someone try connectiing to 109.169.6.122 on udp port 5060 |
| 15:54.41 | TandyUK | you should be blocked, but struggling to understand why im still seeing a load of failed logins |
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| 16:15.51 | igcewieling1 | what is the iptables setting you are using to block it? |
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| 18:07.02 | wolfmitchell | How would I be able to get rid of these messages spamming my console? https://screenshits.nofla.me/2017-05-01_18-07-36.txt |
| 18:07.57 | igcewieling1 | disabling message would be a good start |
| 18:08.01 | file | don't try to do something involving media in the dialplan on the Message channel? |
| 18:08.49 | [TK]D-Fender | "Doctor, doctor, it hurts when I raise my arm like this!" |
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| 18:37.49 | *** mode/#asterisk [-o DivideBy0] by DivideBy0 |
| 18:38.54 | klow | This is a weird one, has anyone ever ran into a situation where audio from the remote end begins to loop? Have a customer saying this is happening to him. Havent started digging in yet, my first intuition is softphone or audio driver on the PC itself vs an asterisk/RTP issue . Just wondering if anyone has experienced similar. |
| 18:39.22 | klow | they are saying it only happens with calls that are trunked in from PTSN (just a sipstation connection on our side) |
| 18:57.12 | wolfmitchell | igcewieling1, what config file is that set in? |
| 18:57.41 | wolfmitchell | file, pretty sure I'm not doing anything at all on the message channel |
| 18:57.59 | file | wolfmitchell: that'd be the only reason those messages would appear |
| 18:59.00 | wolfmitchell | file, how would I see what is doing anything on it then? |
| 19:00.30 | file | You can bump up the verbose |
| 19:00.43 | igcewieling1 | try adding accept_outofcall_message=no to sip.conf |
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| 20:18.31 | dan_j | How can I find out where this is coming from? I cant locate 'ringinuse/ignorebusy' in any of the queue conf files. Nor in realtime. |
| 20:18.32 | dan_j | app_queue.c:10961 load_module: No entries were found for ringinuse/ignorebusy in queue_members table. Using 'ringinuse' |
| 20:22.59 | Samot | ringinuse is by default yes |
| 20:23.09 | Samot | Unless otherwise told, it is used. |
| 20:23.30 | dan_j | Ah. I mis-read it. |
| 20:23.47 | dan_j | <PROTECTED> |
| 20:23.54 | dan_j | But they can receive calls from other queues |
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| 20:33.16 | Samot | You need it to be no |
| 20:33.47 | Samot | Or do you want him to ring while "in-use"? |
| 20:33.57 | Samot | it/them/whatever. |
| 20:36.08 | dan_j | I want it to ring while in use. the cli says ringinuse enabled, but it cannot receive two calls from the same queue. |
| 20:36.29 | dan_j | Other queues will ring that 'busy' peer, without a problem |
| 20:36.40 | Samot | What do they have set in them? |
| 20:37.17 | dan_j | Please can you explain what you mean? |
| 20:37.23 | dan_j | ringinuse=yes |
| 20:37.41 | Samot | So the queues that are doing what you want, that is set in them? |
| 20:38.32 | dan_j | Nope. I have set ringinuse=yes, but asterisk is only partially ringing-in-use. |
| 20:38.39 | dan_j | Let me explain again.... |
| 20:38.57 | dan_j | There are two queues (Queue 1 and Queue 2) |
| 20:39.21 | dan_j | There are 5 SIP peers. All peers are members of both Queues |
| 20:39.28 | dan_j | ringinuse=yes for both queues |
| 20:39.46 | dan_j | Call comes in for Queue 1, Peer 5 accepts the call |
| 20:40.08 | dan_j | 2nd call comes in for Queue 1, Peer 5 does not receive the call, even though ringinuse=yes |
| 20:40.14 | Samot | Like Chan_SIP peers? |
| 20:40.22 | Samot | Or PJSIP peers? |
| 20:40.28 | dan_j | 3rd call comes in, for Queue 2 this time, Peer 5 receives an incoming call notification |
| 20:40.30 | dan_j | PJSIP |
| 20:40.48 | Samot | https://www.irccloud.com/pastebin/EnukUHMJ/ |
| 20:40.54 | Samot | Ehm. |
| 20:40.58 | Samot | Line 8. |
| 20:41.10 | Samot | Unless that has changed. |
| 20:41.27 | dan_j | SIP channel driver = chansip + pjsip as far as im aware |
| 20:41.39 | Samot | No. |
| 20:41.45 | Samot | PJSIP is it's OWN driver |
| 20:41.59 | Samot | SIP is Chan_SIP |
| 20:42.18 | Samot | ie Dial(SIP/<exten>) vs Dial(PJSIP/<exten>) |
| 20:42.26 | dan_j | One sec. It says 'is able to report 'in use''. |
| 20:42.38 | Samot | Right |
| 20:42.41 | dan_j | In this case, I dont care what 'in use' state it's in. I want it to ALWAYS ring! |
| 20:43.34 | Samot | Remove the setting. |
| 20:43.34 | dan_j | This reads to me like a bug. It works fine if the 2nd call is coming from a different queue. But if the new call is from the same queue as the active call, the peer does not get called. |
| 20:43.40 | Samot | It's not needed unless you set it to no |
| 20:43.54 | Samot | Maybe. |
| 20:45.58 | Samot | When Peer 5 is on a call.. |
| 20:46.17 | Samot | queue show queue1 (where they're on the call) |
| 20:46.25 | Samot | Will show them 'inuse'? |
| 20:46.29 | dan_j | yes |
| 20:46.37 | Samot | But doing that on queue2 shows? |
| 20:46.50 | dan_j | Also inuse |
| 20:46.54 | dan_j | Which is correct |
| 20:46.56 | Samot | K |
| 20:46.57 | Samot | Right. |
| 20:47.01 | Samot | Just checking. |
| 20:47.57 | dan_j | It seems to be tracking it correctly. It just doesnt seem to be honouring the 'ringinuse' when the new call is coming from a queue that already has an active call with that peer. |
| 20:48.16 | dan_j | I'm going to try to run some tests tomorrow to try to work it out. |
| 20:50.34 | *** join/#asterisk bl3nto (~bl3nto@78.134.210.254) |
| 20:55.48 | Samot | And it doesn't matter which peer it is. |
| 20:55.56 | Samot | Always the same result? |
| 21:08.14 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 21:08.19 | dan_j | Yes |
| 21:10.12 | dan_j | Checking out for the night. Will see what I can find tomorrow in the debug output to explain it. |
| 21:22.01 | *** join/#asterisk scgm11_ (~scgm11@r186-49-13-160.dialup.adsl.anteldata.net.uy) |
| 21:30.24 | *** join/#asterisk rwb (~Thunderbi@65.183.151.239) |
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| 22:20.16 | rrittgarn | Anybody worked with large banks of SIMs/GSM texting that could recommend a hardware manufacturer? Seems like there are a few, but I'm unsure of quality |
| 22:25.45 | igcewieling1 | I suspect people needing to do a lot of txting use a service provider with a REST or JSON interface to do it for them.] |
| 22:32.58 | *** join/#asterisk scgm11_ (~scgm11@r186-49-13-160.dialup.adsl.anteldata.net.uy) |
| 22:37.37 | rrittgarn | yeah, I am aware, however there are limitations of what an SMS enabled SIP number can do, vs. an actual mobile number |
| 22:39.45 | *** join/#asterisk av0cad0 (~jordan@access-63-249-68-54.static.cruzio.com) |
| 22:42.35 | av0cad0 | Hello, I have a customer complaing about "ghost" calls that keep calling their phone extension. If I'm not seeing the invites on a call report/sip capture, is it safe to assume that a device on their network was compromised and is attempting to place calls to the phone? |
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| 23:15.49 | *** part/#asterisk kharwell (kharwell@nat/digium/x-dzvuxxefgubdpgwl) |
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| 23:54.09 | Download-Fritz | Hey again, does somebody happen to run an Asterisk server with PJSIP and has a transport binding to a local-link IPv6 address? |
| 23:54.27 | Download-Fritz | my global IPv6 works fine, though with the local-link, I get 'invalid argument' when trying to create the transport |
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