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| 15:07.22 | AL13N_lappy | ok, there's a very weird thing... i have 2 computers that both have the same version of linphone (which i use for work on a PBX at work). but, if using my asterisk, i call each other, both ends don't hear each other, BUT, there is DIRECT (and bidirectional) traffic over port 7078 to each other , so the directmedia seems to work, I actually tested by making noise in both and i see the UDP payload of 7078 go up if that end sends audio... |
| 15:07.22 | AL13N_lappy | . |
| 15:08.01 | AL13N_lappy | i know this seems unrelated to asterisk, but it's quite weird, it seems no network issue and both are sending audio... but both don't hear each other? |
| 15:08.23 | AL13N_lappy | is there some kind of asterisk flag that says it should NOT use direct media or something? |
| 15:09.46 | drmessano | There is, actually |
| 15:11.03 | AL13N_lappy | is there a way to see this in the SIP traffic? |
| 15:12.05 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Configuration_res_pjsip |
| 15:12.13 | drmessano | direct_media directive |
| 15:12.25 | drmessano | I would disable it.. Never seen it work in 15 years |
| 15:12.45 | AL13N_lappy | oh, i've set direct_media to yes |
| 15:12.47 | AL13N_lappy | hmm |
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| 15:47.40 | AL13N_lappy | drmessano: well, it seems you were right, it's kind of weird that direct media would not work well, it's kind of unintuitive... |
| 15:48.36 | drmessano | Yep |
| 15:48.49 | drmessano | Don't be so surprised |
| 15:54.26 | theGoat | nothing surprises me with asterisk anumore |
| 15:54.28 | theGoat | ..anymore |
| 15:54.51 | drmessano | Not sure about "anymore" |
| 15:55.19 | drmessano | Direct Media is kind of a flawed concept in most network topologies |
| 15:55.28 | drmessano | So no point trying to fight to make it work |
| 15:55.35 | drmessano | Just let Asterisk handle the media, done |
| 15:55.53 | drmessano | That's not new, and not Asterisk's fault |
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| 17:15.53 | [sr] | howdy, |
| 17:16.35 | [sr] | im looking for a framework that can give me the events when a call is performed or received, ive tried asternet but, the event is fired but no information is received about the caller id and so on |
| 17:16.43 | [sr] | this is for .NET development |
| 17:17.04 | [sr] | does anyone has a clue on this? I'm trying with AMI 1.3, maybe a but for this AMI version? |
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| 18:37.16 | sibiria | AMI works, or you can just do it directly in your asterisk dial plan |
| 18:37.18 | sibiria | a lot simpler |
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| 19:11.37 | igcewieling | What might cause these errors when the peers are able to register just fine. chan_sip.c:27796 handle_request_subscribe: Failed to authenticate device "323" <sip:323@10.0.0.254>;tag=993E32E9-FF3F15B0 for SUBSCRIBE |
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| 20:03.44 | igcewieling | *sigh* |
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| 20:54.04 | [sr] | sibiria: but did you ever done it with .net, catching the events? |
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| 21:15.13 | sibiria | no i don't write c#, but that's less relevant in this case. you can use any language you want to receive e.g. a REST request or to take action in the dial plan via AGI |
| 21:23.50 | [sr] | hum in case, i want to catch the events, of new call's |
| 21:23.59 | [sr] | and not place call's |
| 21:24.47 | [sr] | well i''m checking AMI, i have to build a tcp listener, it has all events i need |
| 21:24.55 | [sr] | just have to find a way to filter them |
| 21:44.04 | sibiria | whether you place calls or receive calls it can go through a dial plan |
| 21:44.18 | sibiria | easier to generate an event from there than hooking into the AMI stream and analyzing the log |
| 21:58.30 | [sr] | thanks |
| 21:58.51 | [sr] | well, after some search and search, i dound that asternet has a property "fireallevents" that is = false by default! |
| 22:03.06 | [sr] | dound=found |
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