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| 05:18.27 | *** topic/#asterisk-dev is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Check out our blog! blogs.asterisk.org -=- Follow on Twitter at @AsteriskDev |
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| 11:30.08 | coreyfarrell | is it possible to setup PJSIP in a way that a remote SIP trunk uses a different endpoint for inbound vs outbound calls? specifically I want Dial(PJSIP/18605551212@trunk-out) to create a channel 'PJSIP/trunk-out-0000000' but then have inbound calls create PJSIP/trunk-in-0000000. |
| 11:33.32 | file | sure, how it is matched on coming in is up to you |
| 11:39.09 | coreyfarrell | so given https://gist.github.com/coreyfarrell/8a639577ead3a1fb806ef948c14436a9 as the starting point would I create copies of the type=aor and type=endpoint named `trunk-out`, but no copy of the type=identify? that way `trunk-out` would only match for the outbound calls? |
| 11:39.44 | file | yes |
| 11:40.07 | file | you could also remove the aor from trunk if it is only being used for inbound |
| 11:42.46 | coreyfarrell | interesting. so the aor is for outbound, identify is for inbound and endpoint is like a peer? |
| 11:43.04 | file | yes |
| 11:43.16 | coreyfarrell | great. thanks for the info. |
| 11:43.20 | seanbright | coreyfarrell: re: atomics - that is fantastic |
| 11:44.15 | seanbright | but now i have to return this ultrasparc i bought |
| 11:44.21 | coreyfarrell | so one thing that tripped me up switching from chan_sip to chan_pjsip is the dial strings. previously I used `Dial(SIP/trunk/8605551212)` but this format doesn't work with PJSIP. seems like `Dial(PJSIP/8605551212@trunk)` is the required format? |
| 11:44.40 | seanbright | yeah |
| 11:44.50 | coreyfarrell | C or GCC atomics don't support ultrasparc? |
| 11:45.04 | seanbright | i was making a joke... a hilarious one |
| 11:45.33 | seanbright | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 11:45.34 | file | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels is the supported formats |
| 11:45.45 | seanbright | jinx |
| 11:46.00 | coreyfarrell | as far as the GCC atomics I view those as legacy.. the C atomics are better in a couple ways and support a couple more methods |
| 11:46.36 | seanbright | i agree |
| 11:47.36 | seanbright | can't wait to switch from c99 to c18 |
| 11:47.45 | coreyfarrell | when I made the change to support only GCC or C atomics I only kept the GCC atomics because of some supported platforms having older GCC without support for the C atomics |
| 11:48.16 | coreyfarrell | centos 6 probably doesn't support C atomics, I don't know about centos 7. |
| 11:48.49 | coreyfarrell | what does c18 bring that you can't wait for? |
| 11:49.45 | seanbright | i was making another joke - because we won't ever switch |
| 11:50.52 | coreyfarrell | ah sorry I'm in rough shape right now, been troubleshooting a problem with some 10 year old perl code |
| 11:50.57 | seanbright | i think c18 is just a 'fixed' version of c11 |
| 11:51.22 | seanbright | i love perl |
| 11:52.03 | coreyfarrell | I'm going to assume that's another joke |
| 11:52.18 | seanbright | now you're getting it :D |
| 11:57.48 | seanbright | i have a pbx in production where i used perl to build all of the configuration from the db |
| 11:58.12 | seanbright | it's awful |
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| 12:33.21 | coreyfarrell | the project I'm working on right now is about 3300 lines of perl that controls asterisk with AMI and until a couple months ago the only test it had was to verify each module could be loaded. I now have 96% code coverage through integration testing. |
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| 12:49.51 | seanbright | nice |
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| 15:57.49 | seanbright | for the streams experts out there, if you could give https://gerrit.asterisk.org/c/asterisk/+/13672 a glance that would be super helpful |
| 16:25.50 | seanbright | so i set up an endpoint to accept ulaw & g722, made an offer of ulaw & g722 |
| 16:26.36 | seanbright | i had a pcap of all ulaw packets, changed the first to a g729 packet (hexl-mode ftw), and sent that with sipp |
| 16:26.56 | seanbright | and the packet is dropped as desired |
| 16:27.20 | seanbright | changed the single g729 packet to a g722 one, ran the same test, and it changes the channel's format |
| 16:27.37 | seanbright | so i know that it "works" but i am not sure if it is correct |
| 16:31.27 | file | it is |
| 16:33.04 | seanbright | sweet |
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