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| 01:19.51 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); 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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| 10:56.21 | pchero_work | Does Asterisk support iSAC seems like it doesn't but want to clear. :) |
| 10:56.34 | pchero_work | iSAC codec. |
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| 15:34.08 | SKaReCRoW | Anyone in here have experience with the famous getting disconnected after 15 minutes issue with RTP timeout? |
| 15:34.35 | Samot | It's called NAT. |
| 15:34.41 | Samot | And a poor configuration. |
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| 15:35.00 | igcewieling | SKaReCRoW: session-timers can cause issues at 20 mins, but I don't think at 15 mins. |
| 15:36.00 | SKaReCRoW | [2018-11-14 10:26:24] NOTICE[23973]: chan_sip.c:29574 check_rtp_timeout: Disconnecting call 'SIP/FlowRoute-00000070' for lack of RTP activity in 31 seconds |
| 15:36.24 | SKaReCRoW | My bosses phone keeps dropping calls after anywhere between 14 and 17 minutes. |
| 15:37.34 | Samot | 10:34:36 AM <Samot> It's called NAT. |
| 15:37.57 | Samot | Asterisk is saying "I got no audio flow from this peer for 31 seconds." |
| 15:38.25 | Samot | Which means the RTP packets from Flowroute's media servers are not making it back to Asterisk. |
| 15:38.29 | SKaReCRoW | But it always drops in the middle of a conversation. |
| 15:38.42 | Samot | I get that. |
| 15:39.02 | Samot | Your PBX is behind NAT, right? |
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| 15:39.57 | SKaReCRoW | Behind a crappy FiOS router. |
| 15:41.18 | Samot | Ok but still NAT |
| 15:41.42 | Samot | Does this happen on inbound calls only, outbound or both? |
| 15:42.30 | SKaReCRoW | Incoming calls just suddenly started working once in a while about a week ago. They wouldn't even get inside before that. They would all go to the failover number. I didn't change anything at all in any config, and suddenly they don't go to my cell anymore, but the caller gets an error message supplied by Asterisk instead. That part doesn't matter though, as we're only interested in using it for ou |
| 15:42.36 | SKaReCRoW | tgoing right now. |
| 15:43.13 | Samot | Ok. So this is happening on outbound calls? |
| 15:43.21 | SKaReCRoW | For 2 days I was able to get incoming calls to ring my Aastra 57i. That took a few hours of searching and trying different configs to get working, then it stopped working without any more changes. |
| 15:43.55 | Samot | Internal calls are fine? |
| 15:44.38 | SKaReCRoW | Actually my phone can't reach my bosses phone, but his can call mine. His Aastra 53i says NO SERVICE on the screen, but otherwise seems to work fine. |
| 15:45.07 | Samot | And the phones and PBX are on the same network? |
| 15:47.36 | Samot | Because this is sounding like a combo of issues. All network or config related |
| 15:48.45 | SKaReCRoW | Yes, both on same network. |
| 15:49.55 | Samot | Then you have few problems. |
| 15:50.19 | Samot | 1 your router/firewall sucks and/or isnt configures right |
| 15:50.48 | Samot | 2 you have internal issues that are not related to your router/firewall. |
| 15:52.39 | SKaReCRoW | 3 I have a boss with issues |
| 15:54.33 | SKaReCRoW | I appreciate you trying to help, but my boss just gave me a job that's gonna put me on the road for about the next 5 hours. Hopefully you, or someone else willing to help, will be on when I get back. |
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| 16:32.37 | malaphus | I'm setting up an IVR and everything is going fine, I create call files in the outbound directory and asterisk calls them as expected, but I've noticed that when calling an internal extension this way, if the user presses # the asterisk voice says "transfer" and waits for a number to transfer to, is there an easy way to disable this feature so that # gets passed properly to my AGI script like other digits? |
| 16:33.31 | Samot | You have to be careful with # |
| 16:34.09 | Samot | In some cases it is how you "send" DTMF that is entered |
| 16:35.19 | Samot | But if while on a call # is initiating a transfer, that is the In-Call features. |
| 16:35.27 | Samot | Look at features.conf |
| 16:41.04 | malaphus | Thanks |
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| 17:30.24 | *** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.1 (2018/11/14), Security Only: 15.6.2 (2018/11/14); 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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| 17:47.29 | muAdmDev | just wanted to say thanks for yesterdays input! after reading up on hints, states and subscriptions I was able to solve my problem already today |
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| 17:57.37 | trmg | I have a silly dial plan question. So, I have a dial plan entry that's basically a non-supervising Caller ID number readback. It works exactly as intended when using a standard DID, but when using a toll-free DID it seems to just loop until either a) the carrier drops the non-suped call, or b) the caller hangs up. Both DIDs are through Anveo Direct FWIW. Anything I can do to preven it from continually |
| 17:57.40 | trmg | looping? Here are the two identical entries: https://ghostbin.com/paste/x96dd |
| 17:58.05 | trmg | s/preven/prevent |
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| 18:15.03 | TheHonorableKitt | any idea what would cause this? |
| 18:15.04 | TheHonorableKitt | WARNING[11077]: chan_sip.c:4131 retrans_pkt: Timeout on 1485005030-1349686388-554543974 on non-critical invite transaction. |
| 18:17.01 | [TK]D-Fender | failure to get a response |
| 18:17.34 | TheHonorableKitt | please elaborate :) |
| 18:17.36 | [TK]D-Fender | either due to improper networking, setup to tell your system where to tell the other side to reach your server at (NAT, etc), etc |
| 18:17.52 | TheHonorableKitt | ah ok |
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| 19:02.46 | malaphus | Anyone know why I'd be getting 'Unable to register extension 'sw_31_1' priority 12 in 'ext-ivr', already in use by aliased extension 'sw_31_-1'' during an ael reload? This seems to be related to a simple switch () { ... } block, but everything looks fine as far as I can tell |
| 19:03.13 | malaphus | If I continue to reload, the 31 increments to 32, then 33, etc |
| 19:04.42 | [TK]D-Fender | Because it sees a duplicate. |
| 19:05.52 | malaphus | I understand that part heh, none of the switch cases are duplicate though, i'm using case 1, case 2, case -1, and default, no others |
| 19:06.16 | [TK]D-Fender | You're asking why something is happening and not showing us |
| 19:07.55 | malaphus | Sorry, my switch just contains case 1: break, case 2: break, case -1: break, and default, it seems as if the 1 and -1 are being parsed as the same perhaps |
| 19:09.15 | [TK]D-Fender | Debugging in pieces is a waste of time |
| 19:09.55 | [TK]D-Fender | If you're not prepared to show us the actual full context and load sequence doen't expect us to be lucky enough to find the error or attemtp to spend the time to try to. |
| 19:10.23 | [TK]D-Fender | heads out for a few minutes, BRB |
| 19:13.20 | malaphus | I was unclear, I know what's causing the issue, I just don't know why that particular thing is a problem. the AEL parser is converting my switch cases for "case 1" and "case -1" to two individual extensions, in this case "sw_31_1" and "sw_31_-1", so obviously using the case value as the last part of the auto-generated extension |
| 19:14.04 | malaphus | My question is, unrelated to the code itself, is why asterisk things sw_31_1 is the same extension as sw_31_-1? This worked fine in older versions, I don't see anything in 16.x changelogs about no longer supporting hyphens, etc |
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| 19:20.32 | rmudgett | The '-' in extens is 'eye-candy' and is ignored. This is so you can use pretty extens as exten = 256-555-1212,1,NoOp() |
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| 19:23.08 | rmudgett | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching#PatternMatching-OrderofPatternMatching |
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| 21:08.19 | sibiria | are there any performance- or stability-related changes (or anything else compelling) introduced in asterisk 14, 15 or 16? |
| 21:08.26 | sibiria | something not back-ported to ast 13 |
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| 22:16.24 | AsteriskRoss | With the latest GIT 13 commit, does anyone else have a problem with asterisk not restarting when running 'core restart gracefully'? |
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| 22:48.16 | life_of_e | Hi all, may I ask SIP/Asterisk related hardware questions here? |
| 22:51.29 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:58.07 | sibiria | life_of_e: that sounds like the usual type of questions in here |
| 22:58.41 | life_of_e | sibiria: Thanks, I always like to make sure |
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| 23:13.07 | life_of_e | So to the room: my question is about FXS/FXO network devices like the old Linksys SPA3102 to use for POTS as a trunk. Longer question here: https://pastebin.com/x4bMD3qZ Short question, looking for a supported suggestion for an FXO adapter that won't break the bank but supports POTS trunking and inbound CallerID. |
| 23:14.15 | life_of_e | I know the SPA3102 isn't supported by Cisco anymore so I was looking for something that still has support/updates at minimum unless there isn't much of a reason to do this. |
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| 23:16.28 | setham | Have you look at http://www.grandstream.com |
| 23:16.29 | setham | ? |
| 23:17.05 | setham | *looked |
| 23:17.38 | life_of_e | setham: That was the name I was trying to remember when I wrote the pastebin. I had heard that some Grandstream stuff wasn't all that great but I am no expert to know if that's true or just someone being excessively negative. |
| 23:17.58 | setham | I have a couple and they work as intended |
| 23:18.17 | setham | http://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors |
| 23:20.07 | setham | I have a box running asterisk, I only use CLI |
| 23:20.24 | life_of_e | setham: thank you for the link. I see the HT813 has the FXO port. What unit do you have? |
| 23:20.26 | setham | And if I need one I would get one of those |
| 23:21.59 | setham | 2x HandyTone (286) old and 1 HT812 |
| 23:22.24 | setham | two PAP2 and a coupleof SPA3102 |
| 23:23.11 | setham | (personal, not business) |
| 23:23.23 | life_of_e | so you're using the linksys fxo since the ht812 is fxs only? |
| 23:23.36 | setham | It depends where |
| 23:23.56 | setham | I use it for family and friends over several countries |
| 23:24.55 | setham | Sorry, not HT286, the 386 |
| 23:25.26 | setham | Those are connected to the local POTS |
| 23:25.32 | [TK]D-Fender | Capable of performing a hook switch for call waiting <- no ATA I've ever seen supports this |
| 23:25.38 | setham | The HT812 is just FXS |
| 23:25.53 | [TK]D-Fender | Using a DAHDI card in your server can. |
| 23:26.15 | [TK]D-Fender | "Capable of somehow passing DID data to Asterisk, manual configuration is fine, the phone number won't change." <- there is no such thing as DID on analog |
| 23:26.34 | [TK]D-Fender | (in practical rel-world applications) |
| 23:26.43 | life_of_e | I underss |
| 23:26.45 | setham | What do you mean there is not DID? |
| 23:27.13 | [TK]D-Fender | Your line doesn't see WHY it is ringing. |
| 23:27.20 | setham | Sorry read CID |
| 23:27.39 | setham | Read DID, thought of CID |
| 23:28.00 | life_of_e | I understand, I just want Asterisk to know which POTS line is ringing so it can route accirdingly |
| 23:28.46 | [TK]D-Fender | so the only solution I can see for the rest requires a card in your server to handle the line, and feature-codes to be able to trigger the hook-switch to deal with the call-waiting scenario |
| 23:29.33 | setham | But if you have an ATA for each line you will know what line is ringing |
| 23:30.19 | life_of_e | My server is a Raspberry Pi so no cards. :) |
| 23:30.21 | [TK]D-Fender | I've yet to see one that will support the call-waiting requirement |
| 23:30.37 | [TK]D-Fender | So if you actually want that aspect to happen ..... |
| 23:30.42 | life_of_e | I did stumble across a hack for the SPA3102 that would simulate a hook switch with a fast hangup answer |
| 23:31.10 | life_of_e | I actually coded it into my dial plan in anticipation of having one but I never got around to getting the 3102. |
| 23:32.18 | life_of_e | As for the DID, I was thinking about devices that had more than one FXO (like the Vega which has 4 minimum) |
| 23:32.40 | [TK]D-Fender | don't use the term "DID". |
| 23:32.45 | setham | The GrandStream GXW4104 has 4 fxo |
| 23:33.00 | life_of_e | D-Fender: Fair enough |
| 23:33.05 | [TK]D-Fender | it's just knowing which line the call was coming from through whatever you have it plugged into |
| 23:34.41 | life_of_e | setham: I saw the GXW as I was browsing the rest of the website. I suppose that's an option, too, though certainly more than the HT813 :) but it might have more features, too. It doesn't have FXS ports which I thought was interesting. I like the fail-over feature of the HT813 |
| 23:36.58 | *** join/#asterisk setham (~textual@unaffiliated/setham) |
| 23:40.20 | *** join/#asterisk K0HAX_ (~michael@gateway/tor-sasl/k0hax) |
| 23:40.22 | life_of_e | So the Granstream might actually support hook flash |
| 23:40.48 | life_of_e | In theory it supports sending the Flash signal (16) |
| 23:41.11 | life_of_e | It already supports flash and hook-flash over the FXS port so if it accepts SIP Flash it would do it. |
| 23:41.22 | [TK]D-Fender | * doesn't |
| 23:41.33 | life_of_e | No? Bummer |
| 23:43.21 | *** join/#asterisk paulgrmn (~paulgrmn@162.219.176.22) |
| 23:53.25 | life_of_e | D-Fender: The manual says otherwise |
| 23:53.44 | *** part/#asterisk kharwell (kharwell@nat/digium/x-klqwdszbnyznoseo) |
| 23:54.00 | life_of_e | "Send Hook Flash to PSTN: If Yes, hook flash will be sent to PSTN upon receiving flash event from RFC2833 or SIP INFO" |
| 23:54.34 | [TK]D-Fender | What manual is that from? |
| 23:54.41 | life_of_e | Page 63 of the HT813 Administration Guide |
| 23:54.47 | [TK]D-Fender | Now read what I said again |
| 23:55.45 | life_of_e | You were talking about a card and that you hadn't seen an ATA capable of it |
| 23:56.11 | life_of_e | However, this one claims it can |
| 23:56.13 | [TK]D-Fender | <[TK]D-Fender> * doesn't <- |
| 23:56.43 | life_of_e | where are you inserting doesn't? |
| 23:57.21 | [TK]D-Fender | * = Asterisk |
| 23:57.41 | life_of_e | Oh, I did not understand the use of * as an abbreviation |
| 23:58.14 | [TK]D-Fender | That is the very name of that symbol and the reson it's the logo |
| 23:58.31 | life_of_e | Well yes but it was also being used here as a correction to a typo |
| 23:58.36 | [TK]D-Fender | Useful bit going forward |
| 23:59.36 | *** join/#asterisk TheHonorableKit- (~TheHonora@gateway/tor-sasl/thehonorablekitt) |