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| 01:19.13 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.5 (2017/12/22), Standard: 15.1.5 (2017/12/22); 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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| 08:05.30 | znoteer_ | tests his connection |
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| 08:30.07 | babak | Hi, when |
| 08:31.19 | babak | I use SetCallerPres(prohib) in dial plan I receive not founding application, any module I need load ? |
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| 14:30.57 | matt_ | hi, I have a weird issue, on some, not all, phones when they are sent a 180 Ringing I dont hear any ringing tone |
| 14:31.53 | matt_ | its weird because it seems to be effected by what remote network I am dialing through, so some mobile networks work but others I dont hear anything untill the line has been answered |
| 14:32.24 | matt_ | whats even more weird is that homer is showing its sending a 180 to the phone |
| 14:32.58 | matt_ | if I set inband_progress it works, but I dont see why its not working with the 180's |
| 14:34.31 | Samot | 180 Ringing isn't a thing that makes the phones ring. |
| 14:34.37 | Samot | It's a state. |
| 14:35.55 | matt_ | Samot: ahh ok, so that might be right then? and the phones that are ringing have just got some code in them that makes them do that? |
| 14:36.04 | Samot | You send an INVITE the other side sends back a "100 Trying" which means "We're trying to find this user" |
| 14:36.33 | Samot | Which should be followed by "180 Ringing" which means, "We found the user and we are sending the call to it" |
| 14:36.38 | matt_ | Samot: ok, 100 trying dosn't mean the remote phone is ringing tho right? |
| 14:36.45 | Samot | No. |
| 14:36.56 | Samot | It means the other side is trying to find a location for the phone. |
| 14:37.18 | Samot | The next reply to follow is either a 18X message or an error message. |
| 14:37.46 | matt_ | ok, which is when the audio is being setup? like for a 183? |
| 14:37.55 | Samot | The 180 Ringing means they found the location and are trying it. |
| 14:38.22 | Samot | The other side can then follow that up by a 183 Session Progress which is the phone handling the incoming call... |
| 14:38.43 | Samot | 180 Ringing alone or a 180 Ringing followed by a 183 Session Progress are normal replies. |
| 14:40.38 | matt_ | so should I be using inband_progress on all endpoints that are handsets? |
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| 14:43.00 | Samot | So you're calling destinations on the PSTN via Asterisk and not getting ring back? |
| 14:46.47 | matt_ | Samot: yea, theres a couple of asterisk boxes linked with iax2 trunks between the handset and the provider and the outgoing provider used reinvites to other networks |
| 14:47.18 | Samot | OK then it's up to Asterisk to send that information back to you. |
| 14:47.22 | matt_ | so calling different pstn networks might result in getting reinvited to different equipment |
| 14:47.51 | Samot | If Box1 is sending a call to Box2 then Box2 is sending the call to the PSTN.. |
| 14:48.00 | matt_ | intresting, I think, when it dosn't work the 180 Ringing contains a SDP payload but when it does its 0 length |
| 14:48.02 | Samot | You have to make sure it gets back from Box2 to Box1 |
| 14:48.27 | matt_ | Samot: homer is showing a 180 message being sent to the phone |
| 14:49.13 | Samot | OK |
| 14:49.24 | Samot | That is telling the phone the other side is being tried. |
| 14:49.34 | Samot | It's now up to Asterisk to tell the device there is ring back |
| 14:49.53 | Samot | You're hopping this call through _two_ Asterisk boxes. |
| 14:50.00 | Samot | Which is basically 4 channels. |
| 14:50.19 | Samot | Plus you're using IAX2 so I'm not sure how that handles early media. |
| 14:52.42 | matt_ | Samot: ok, I have a sim card in a GoIP-1 box, i'm just setting it up so I can use it as a test |
| 14:53.19 | matt_ | because, belive it or not, all of my phones are now sip based, including my mobile |
| 14:53.32 | Samot | OK. |
| 14:53.49 | Samot | Your phone only cares about the Asterisk box it is connected to. |
| 14:54.06 | Samot | If that Asterisk box is trunked to the PSTN or another system, it doesn't know or care. |
| 15:01.23 | matt_ | Samot: ok, just rang it and I could hear ringing after I turned off inband_progress |
| 15:01.41 | matt_ | intresting, i'm seeing both 180 and 183 going to the phone and 180 contains an sdp payload |
| 15:01.59 | matt_ | might just reboot the phone to make it re register |
| 15:03.01 | Samot | 180 is when early media can happen |
| 15:03.10 | matt_ | ok |
| 15:03.10 | Samot | A 183 isn't required but recommended. |
| 15:03.23 | Samot | In fact a 180 OR a 183 can follow a 100 |
| 15:03.48 | Samot | It can be 100 -> 180 or 100 -> 180 -> 183 or 100 - > 183 |
| 15:04.12 | matt_ | ok, that last call was 100 -> 183 -> 180 |
| 15:04.17 | Samot | OK. |
| 15:04.27 | Samot | Because you have two boxes. |
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| 15:04.44 | Samot | The 183 could follow the 180 almost immediately. |
| 15:05.13 | Samot | So Box 2 could be getting a 183 sending it to Box 1 and that is sending a 180 to your phone |
| 15:05.24 | Samot | Again, you're hopping this through two Asterisk boxes. |
| 15:05.33 | Samot | That's two channels per box. |
| 15:06.11 | Samot | Phone opens a SIP channel on Box1 then creates an IAX2 channel with the Dial() to call Box2 |
| 15:06.27 | Samot | Box2 has an IAX2 channel that answers the call from Box1 |
| 15:06.41 | Samot | And then creates a SIP channel to Dial() the provider. |
| 15:06.51 | Samot | You're going from SIP to IAX2 to SIP |
| 15:06.59 | Samot | Why? I don't know. |
| 15:07.21 | matt_ | it used to be because of nat but, most of this stuff is on a private network now |
| 15:07.38 | Samot | Connecting Asterisk boxes via IAX2 is very early 2000's |
| 15:07.42 | matt_ | so I dont have to use iax anymore, it was just setup like that |
| 15:07.48 | Samot | NAT isn't an issue any more. |
| 15:07.56 | Samot | Unless you have some really, really shitty routers. |
| 15:08.00 | matt_ | lol ok |
| 15:08.14 | matt_ | I could change it all |
| 15:08.17 | Samot | IAX2 solved a problem 10 years ago that no longer exists today |
| 15:08.35 | Samot | Part of the reason IAX2 has been community supported/driven only for quite some time. |
| 15:09.59 | matt_ | humm, i took off inband progress but asterisk is still sending a 183 |
| 15:10.28 | Samot | OK |
| 15:10.33 | Samot | You're still not listening.. |
| 15:10.39 | Samot | Those are SIP messages. |
| 15:10.46 | Samot | Replies actually to the INVITE |
| 15:10.58 | Samot | They are showing the STATE of the SIP transaction. |
| 15:11.13 | Samot | It has nothing to do with the actual media. |
| 15:11.37 | matt_ | I was tring to reproduce the problem, when it happens I only see a 180 after 100 |
| 15:11.38 | Samot | SDP can be sent in 180/183 messages thus allowing for early media in the call before it is "setup/answered" |
| 15:12.10 | Samot | Well you need to watch the call from all sides. |
| 15:12.20 | Samot | You need to see how the provider is responding to the INVITE |
| 15:12.41 | Samot | And then how Box2 is sending that response back to Box1 and then to the phone. |
| 15:13.01 | Samot | If you're only looking at Box1 then you are only seeing the call between Box1 and Box2 |
| 15:13.03 | matt_ | ok |
| 15:13.04 | Samot | Not the PSTN |
| 15:13.14 | Samot | The call to the PSTN exists only on Box2 |
| 15:13.42 | Samot | What the provider sends back to the INVITE goes there. |
| 15:13.47 | Samot | Not to the phone. |
| 15:13.48 | matt_ | I havn't been looking at the pstn part at all, all of this is between the phone and the asterisk box its registered to |
| 15:13.51 | Samot | It's not going to "pass-through" |
| 15:14.36 | Samot | Again, this is a "two-part" call. |
| 15:14.43 | Samot | Phone to Asterisk Box1 |
| 15:14.52 | Samot | Then Asterisk Box 1 to Asterisk Box 2 |
| 15:14.59 | Samot | Then Asterisk Box 2 to PSTN |
| 15:15.27 | Samot | If the phone gets a 503 Error it's because Asterisk Box 2 is messed up |
| 15:16.00 | Samot | And Box 1 can't call it. |
| 15:16.18 | Samot | Not that the call hit the PSTN and the provider sent it. |
| 15:17.18 | matt_ | erm.., this is out going calls? |
| 15:17.45 | matt_ | so when I call a number I dont hear the ring ring while the other phone is ringing |
| 15:18.01 | matt_ | just silence untill the line is answered at the far end and I get audio |
| 15:18.26 | matt_ | so the first thing I hear after dialing the number is 'hello?! ...' |
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| 16:53.17 | Samot | matt_: I had to run out |
| 16:53.29 | Samot | matt_: But you're missing what I was saying. |
| 16:54.02 | Samot | matt_: You make a call from your phone, the phone sends the INVITE to the Asterisk box it is "registered to" |
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| 16:54.52 | freebs | did you actually run though? |
| 16:55.13 | Samot | matt_: That Asterisk box accepts the call and processes it based on your dialplan. Which in this case is matching your digits and going "Oh I need to execute a Dial() for this" |
| 16:55.34 | Samot | matt_: That Dial is opening an IAX2 channel to your _other_ Asterisk box. |
| 16:56.05 | Samot | matt_: rinse/repeat same process on the other Asterisk box but this time a Dial() over a SIP channel to your Provider (aka the PSTN) |
| 16:57.29 | Samot | matt_: That calll from your phone to the PSTN number is going through two Asterisk boxes and since Asterisk is a B2BUA it essentially is answering and dialing the call twice. |
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| 20:49.14 | Kobaz | hmmm |
| 20:49.29 | Kobaz | never really monkied with this during a call.. is there a way to get VOLUME to apply to a call in progress |
| 20:49.34 | Kobaz | ie. varset in ami |
| 20:50.30 | Kobaz | ooo |
| 20:50.34 | Kobaz | never mind... it does work |
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