| 01:34.58 | *** join/#asterisk-dev klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 03:10.25 | *** join/#asterisk-dev klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 04:04.58 | *** join/#asterisk-dev klow (~textual@c-67-161-106-89.hsd1.wa.comcast.net) |
| 04:32.19 | *** join/#asterisk-dev F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 04:50.09 | *** join/#asterisk-dev klow (~textual@c-67-161-106-89.hsd1.wa.comcast.net) |
| 04:57.41 | *** join/#asterisk-dev bof22 (~Thunderbi@185.13.183.107) |
| 06:29.57 | *** join/#asterisk-dev klow (~textual@c-67-161-106-89.hsd1.wa.comcast.net) |
| 06:38.07 | *** join/#asterisk-dev pchero_work (~pchero@109.70.54.56) |
| 06:43.54 | *** join/#asterisk-dev CELYA_ (~Thunderbi@LStLambert-657-1-90-73.w80-13.abo.wanadoo.fr) |
| 07:01.21 | *** join/#asterisk-dev jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 07:11.28 | *** join/#asterisk-dev oej (~oej@h87-96-134-129.cust.se.alltele.net) |
| 07:11.29 | *** mode/#asterisk-dev [+o oej] by ChanServ |
| 07:19.09 | *** join/#asterisk-dev klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 07:28.57 | *** join/#asterisk-dev hehol (~hehol@gatekeeper.loca.net) |
| 07:30.18 | *** join/#asterisk-dev file (~file@asterisk/developer-and-muffin-lover/file) |
| 07:30.18 | *** mode/#asterisk-dev [+o file] by ChanServ |
| 08:07.34 | *** join/#asterisk-dev CELYA_1 (~Thunderbi@LStLambert-657-1-90-73.w80-13.abo.wanadoo.fr) |
| 08:11.23 | *** join/#asterisk-dev kippi (~kippi@host-4.dxi.eu) |
| 10:17.23 | *** join/#asterisk-dev CELYA_1 (~Thunderbi@lcy44-2-88-161-121-174.fbx.proxad.net) |
| 12:01.34 | *** join/#asterisk-dev WIMPy (~wimpy@85.183.95.26) |
| 12:07.08 | *** join/#asterisk-dev acidfu_ (~acidfoo@198-48-218-90.cpe.pppoe.ca) |
| 13:35.03 | file | falls over |
| 13:35.59 | gtjoseph | file: you've been awfully wobbly lately |
| 13:36.09 | file | just like the British pound! |
| 13:36.14 | file | too soon? |
| 13:36.20 | gtjoseph | nah |
| 13:36.46 | file | very good |
| 13:48.56 | jkroon | guys, i'm sitting with a sip profiling patch in my local tree which someone suggested I submit ... it basically times how long asterisk (chan_sip) takes to process a single SIP frame and logs the duration in micro-seconds, and logs an error for >100ms. |
| 13:49.44 | jkroon | not sure if there would be more interest in something like that, the particular use-case I used it for was figuring out why I get a backlog on incoming SIP packets on UDP, I've since determined it's the fault of astdb on busy disks so don't really care about it any more ... |
| 13:53.01 | file | provided it's configurable and doesn't introduce any performance issues itself... |
| 13:54.45 | *** join/#asterisk-dev bof23 (~Thunderbi@185.13.183.107) |
| 13:55.13 | jkroon | gettimeofday() calls mostly... |
| 13:55.32 | jkroon | file, would it really be useful? or just something that's there "just in case"? |
| 13:55.47 | file | it might be useful to someone... chan_sip isn't really our focus these days |
| 13:56.01 | jkroon | it is on a critical path and my findings was back then that for other than REGISTER things gets done in <100us for the most part. |
| 13:56.33 | file | if you'd like to contribute it though it'd certainly get looked at |
| 13:56.46 | jkroon | and putting astdb onto tmpfs (which carries other risks) that processing time comes down from >150ms to <750us. |
| 13:57.44 | jkroon | would need to be made a bit more configurable. all trip points and everything is hard-coded at the moment. thus my resistance, but I have been asked now. thus why I'd like to know if it'll be generally considered useful before going through the cleanup efforts. |
| 14:19.52 | *** join/#asterisk-dev mjordan (mjordan@nat/digium/x-mmfkxenpcjlmdqgy) |
| 14:19.52 | *** mode/#asterisk-dev [+o mjordan] by ChanServ |
| 14:32.22 | *** join/#asterisk-dev rmudgett (rmudgett@nat/digium/x-enqrpzpjsstvyevy) |
| 14:47.00 | file | tzafrir, meep |
| 14:58.18 | *** join/#asterisk-dev newtonr (RustyNewto@nat/digium/x-eduxsjnmxecyyjpu) |
| 14:58.18 | *** mode/#asterisk-dev [+o newtonr] by ChanServ |
| 15:05.41 | file | the future doesn't exist! |
| 15:36.22 | file | achoo. |
| 15:50.39 | *** join/#asterisk-dev oej (~oej@h87-96-134-129.cust.se.alltele.net) |
| 15:50.39 | *** mode/#asterisk-dev [+o oej] by ChanServ |
| 16:15.21 | file | gtjoseph, only your newly added tests failed overnight - this is goooooood |
| 16:15.33 | gtjoseph | cool |
| 16:15.51 | gtjoseph | hopefully I can figure out the ari test in the next day or so. |
| 16:15.57 | tzafrir | file, hi |
| 16:16.22 | file | tzafrir, I was just curious if you didn't make the autoconf changes the regeneration of the configure script one commit for a reason, or really all of it |
| 16:16.58 | tzafrir | They seemed like separate changes |
| 16:17.40 | file | k |
| 16:17.41 | tzafrir | The configure script itself is generated, as thus if this change collides with anything: I don't really care |
| 16:18.48 | file | usually that stuff is squashed down to 1 thing which encompasses all that is required for the new addition, that's why I was wondering |
| 16:21.34 | *** join/#asterisk-dev cazorla19 (~cazorla19@37.113.30.58) |
| 16:23.23 | cazorla19 | Hi, guys! Let me ask one newbie-developer question. |
| 16:23.56 | file | cazorla19, go ahead |
| 16:24.20 | cazorla19 | How do you actually probe and test your code? Is there any Asterisk server simulator where I can receive some test calls? |
| 16:25.06 | file | there's tools (SIPp and pjsua) and the Asterisk testsuite, which can do a lot of things |
| 16:25.17 | file | when developing initially most people probably just use real phones or softphones |
| 16:26.22 | file | what exactly is being tested sort of determines how |
| 16:29.00 | cazorla19 | So, I have one goal. I'd like to make AGI-based module. Basic theme - speech recognition and reverse response to user. So, I have to call to Asterisk by myself, but I need an environment which could receive this and use my module in configuration. |
| 16:29.41 | file | the SIPp application allows you to give it a scenario which can call in and also send a provided pcap based RTP stream |
| 16:30.25 | cazorla19 | Can I configure in SIPp environment a dialplan with my module? |
| 16:30.38 | file | no, SIPp is a SIP testing tool |
| 16:30.41 | file | it doesn't know Asterisk |
| 16:31.02 | file | you'd have to configure Asterisk separately, and within the testsuite such a thing would look like this: https://github.com/asterisk/testsuite/tree/f2185d0e1a370b0786eb7af700301f3210a7f078/tests/channels/SIP/rfc2833_dtmf_detect |
| 16:31.25 | file | that test calls into Asterisk and sends an RTP stream containing DTMF and tests it |
| 16:36.05 | cazorla19 | yeah, that's necessary to configure Asterisk separately, but where will be environment at that case? |
| 16:36.46 | file | you'd have to run Asterisk somewhere, in the case of the testsuite that is local |
| 16:36.52 | cazorla19 | I mean what should play the physical server role? Could it be any remote test environment? |
| 16:40.06 | file | cazorla19, so in the case of the testsuite it runs both Asterisk and SIPp locally where it is run like I said - if you don't use the testsuite then it's up to you how you want to do it |
| 16:41.05 | file | there's no free infrastructure to do what you want though, it's up to you |
| 16:41.17 | cazorla19 | got you |
| 16:41.28 | cazorla19 | but still one misunderstanding |
| 16:43.24 | cazorla19 | use case: make calls to asterisk --> get an answer --> take a redirection to magic Asterisk client by dialplan |
| 16:43.50 | cazorla19 | how can I use testsuite and sipp at that use case? how could it make development easier? |
| 16:43.53 | file | I don't know what you mean by "take a redirection to magic Asterisk client by dialplan" |
| 16:44.05 | cazorla19 | may be not easier, but more clear |
| 16:44.30 | cazorla19 | I mean rust redirect the call to client |
| 16:44.33 | cazorla19 | just* |
| 16:46.15 | cazorla19 | generally there is a hardware problem; I could have virtual environment for Asterisk server, but I don't have any real phones to connect them |
| 16:46.18 | file | writing those types of tests allows you to reliably test different situations to make sure it works, so that if you change things it will (hopefully) catch it |
| 16:46.47 | cazorla19 | how can be hardware problem solved? |
| 16:47.07 | file | SIPp doesn't require hardware or a phone... it acts as one based on the scenario you give it |
| 16:47.45 | cazorla19 | So, is it possible to send to SIPp scenario audio file with speech? |
| 16:47.58 | file | yes, you can provide it a packet capture that contains RTP and it will play it back |
| 16:48.41 | file | that is what the test I linked to does. |
| 16:48.49 | cazorla19 | But RTP is a mostly streaming multicast protocol, and audio file is just a data container. Where is the bond? |
| 16:49.05 | file | you have to create a packet capture with RTP, that is what SIPp accepts |
| 16:49.40 | cazorla19 | OK. And testsuite is just a SIPp integration tool? |
| 16:50.04 | file | the Asterisk testsuite contains many things, it uses SIPp to accomplish some things |
| 16:50.22 | cazorla19 | indeed, documentation should help |
| 16:50.57 | cazorla19 | and one more question, is it possible to call Asterisk directly from desktop? Like Skype |
| 16:51.09 | file | Skype is a closed protocol |
| 16:51.14 | cazorla19 | yeah, sure |
| 16:51.16 | file | there exist SIP softphones which can be used |
| 16:51.52 | cazorla19 | SIP softphones which support calls to Asterisk, correct? |
| 16:52.07 | file | SIP is an open protocol and Asterisk supports it, so yes |
| 16:54.25 | cazorla19 | And SIPp...could it provide Asterisk client imitation in its scenario? I mean when someone calls to host through Asterisk and host should answer something |
| 16:54.45 | file | it could... |
| 16:54.52 | file | it can act as a sender or receiver of calls |
| 16:56.38 | cazorla19 | Ok, finally got you. Thanks so much for your answers and patience. |
| 16:57.05 | file | you're welcome |
| 17:15.13 | *** join/#asterisk-dev klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 17:32.01 | *** join/#asterisk-dev CELYA_ (~Thunderbi@2a01cb0583d31d00e4bdfb9caee386bf.ipv6.abo.wanadoo.fr) |
| 17:34.22 | *** join/#asterisk-dev pchero (~pchero@109.70.54.56) |
| 18:10.12 | seanbright | when dialing PJSIP channels like PJSIP/endpoint/sip:foo@bar |
| 18:10.25 | seanbright | does chan_pjsip look up 'bar' in /etc/hosts? |
| 18:15.32 | *** join/#asterisk-dev klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 18:28.22 | file | seanbright, that depends on the configuration - if an outbound proxy is set it won't, but if not then it'll do standard resolution |
| 18:28.26 | file | SRV and A in 13 |
| 18:28.31 | file | NAPTR/SRV/AAAA/A in master |
| 18:39.51 | seanbright | gotcha |
| 18:39.54 | seanbright | thanks |
| 18:49.49 | seanbright | file: getting PJLIB_UTIL_EDNS_NXDOMAIN |
| 18:50.11 | seanbright | nevermind, i'll figure it out |
| 18:50.46 | file | neverminded |
| 18:53.59 | coreyfarrell | file: I've been looking at some of the format.c code, do you remember why format->interface doesn't hold a reference (in ast_format_create_named). |
| 18:54.12 | file | nope |
| 18:54.56 | file | actually, I don't think it's an ao2 object |
| 18:55.05 | file | the interface itself that is |
| 18:56.15 | coreyfarrell | oh I see that it's two objects now, thanks |
| 18:56.40 | seanbright | so pjsip's resolver doesn't look at /etc/hosts |
| 18:56.46 | seanbright | so... yay? |
| 19:00.06 | *** join/#asterisk-dev klow (~textual@96.81.150.137) |
| 19:24.49 | *** join/#asterisk-dev klow (~textual@96.81.150.137) |
| 20:17.41 | file | Or tacos. |
| 20:28.27 | *** join/#asterisk-dev piling (~piling@unaffiliated/piling) |
| 20:42.10 | *** join/#asterisk-dev piling (~piling@unaffiliated/piling) |
| 21:13.49 | *** join/#asterisk-dev klow (~textual@96.81.150.137) |
| 22:12.17 | *** join/#asterisk-dev acidfu_ (~acidfoo@198-48-218-90.cpe.pppoe.ca) |
| 22:43.22 | *** join/#asterisk-dev klow (~textual@96.81.150.137) |
| 23:01.42 | *** join/#asterisk-dev jsmith (~jsmith@fedora/jsmith) |
| 23:01.43 | *** mode/#asterisk-dev [+o jsmith] by ChanServ |
| 23:02.12 | *** join/#asterisk-dev Juggie (~Juggie@unaffiliated/juggie) |