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| 01:19.43 | *** 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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| 06:20.52 | bittis | hey guys, random question, is it possible under any scenario for asterisk 13 to create 2 cdr records with the same uniqueid ? |
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| 08:02.03 | bittis | for some reason cdr records get created with the same uniqueid more than once if one provider fails and then tries to call another provider, and a last one gets created which says answered although the call was not answered, and duration is 0 |
| 08:03.03 | bittis | this call started via an "originate" request |
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| 10:16.13 | joerg9811 | Hello everyone. Is there somebody with experience with Asterisk 15 and FreePBX 14 in connection with Deutsche Telekom sip-trunk.telekom.de? |
| 10:21.32 | joerg9811 | I want to get the sip-trunk work. Either with config-files or from the webgui... It would be nice, if someone knows how to do it. Google was no help. |
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| 11:32.04 | Dovid | @Tzafrir: Do you have handy a URL to upgrade the firmware on an Asterbank? This is what dahdi_hardware is showing. From what I remember the last time I installed one of these I need to update the firmware on it: |
| 11:32.04 | Dovid | usb:001/004 xpp_usb- e4e4:1160 Astribank-modular no-firmware |
| 11:34.57 | tzafrir | Dovid, any chance you don't have fxload on that system? |
| 11:35.13 | tzafrir | Latest dahdi should have that firmware |
| 11:35.24 | tzafrir | What version of dahdi is it? |
| 11:36.07 | Dovid | tzafrir: I have fxload. I have dahdi-2.11.1 installed |
| 11:36.26 | Dovid | I remember last time I had this issue and a firmware update fixed it |
| 11:36.46 | Dovid | I dont have xpp_fxloader. |
| 11:38.43 | Dovid | sorry I do. this is what I get: https://pastebin.com/qtTjRPYw let me look @ the script |
| 11:42.34 | Dovid | @tzafrir: line 470 has # We have a potential astribank |
| 11:42.34 | Dovid | astribank_is_starting -a |
| 11:42.52 | Dovid | is that a mistake or is astribank_is_starting -a supposed to be there? |
| 11:46.15 | tzafrir | Dovid, not sure what may cause this. /usr/sbin not in PATH? |
| 11:47.29 | Dovid | tzafrir: seems it was never built on thsi box. on my old box it has it. an mlocate turned up empty for astribank_is_starting. seems it was not built |
| 11:50.35 | Dovid | tzafrir: seems I didnt have libusb. All good now. |
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| 12:05.03 | bittis | guys anyone here working with Asterisk 12 or higher, originating calls and managed to figure out how CDR records are being created? |
| 12:07.50 | stefan27 | try the wiki documentation for the first part... https://wiki.asterisk.org/wiki/display/AST/Home There are several ways to give instructions to asterisk to make a call. Perhaps https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Command+Reference "AMI ACtions" suit your needs |
| 12:09.29 | stefan27 | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerAction_Originate But you'd need to know how the dialplan works and getting an AMI client working with configs and stuff; not sure what your context is |
| 12:09.34 | stefan27 | (I never used asterisk's built in CDR) |
| 12:10.14 | bittis | the originate part works, been going through the CDR specification of 12 to figure out why i some times get 2 CDR records and why sometimes one |
| 12:10.45 | stefan27 | Aha, if you have a specific question, try posting an extensive problem description on a forum |
| 12:11.00 | stefan27 | or directly here (I'm unlikely to be able to direct you) |
| 12:11.00 | bittis | when a call originates, then once party a picks up a cdr is created, a channel is established there, then a call to the second party is placed, another channel gets created, and once answered bridged |
| 12:11.26 | bittis | if answered then n all is good, one cdr record created |
| 12:11.48 | bittis | if not answered though then i end up with 2 records |
| 12:14.06 | bittis | then this could be because channels never got bridged we end up with 2 cdr records rather than 1 |
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| 12:54.39 | stefan27 | if this is the only exception where you are not happy with the cdr module, it sounds like you could handle it as an exceptional case in whatever client you have that views/processes the cdrs... Otherwise google it, I never used that module |
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| 13:09.59 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
| 13:10.09 | Samot | bittis: It changed in v12 |
| 13:10.22 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification |
| 13:11.29 | *** join/#asterisk LunaLovegood (~alice@75.98.139.193) |
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| 13:26.11 | bittis | is the CEL description a more comprehensive guide? will go through it |
| 13:26.16 | bittis | Samot: Thank you |
| 13:30.31 | Samot | CDR and CEL are two different things. |
| 13:33.40 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-vwphfeffbqcrxskp) |
| 13:33.48 | LunaLovegood | I have all my IP phones in subnet 172.19.0.0/16 + VLAN 60, which has no Internet access. So the RTP packets for all outside calls must pass through Asterisk. Is there a way I can use res_pjsip's direct_media=yes but only when it's between phones in the same subnet? |
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| 13:39.04 | LunaLovegood | I suppose one way to acheive that would be to run two instances of Asterisk; one with direct_media=on that manages the local phones, and one that connects the other server to the outside, with direct_media=no. But that seems overly complicated. |
| 13:40.20 | Samot | direct_media has a few options. |
| 13:40.41 | LunaLovegood | I don't want to use NAT or the like though. |
| 13:41.19 | Samot | What? |
| 13:41.28 | LunaLovegood | About half the calls are internal, so I figured it'd be preferable if their RTP was handled by the L2 switches rather than the Asterisk server. |
| 13:41.40 | Samot | That's not how that works. |
| 13:41.56 | Samot | direct_media=yes makes Asterisk "proxy" the media. |
| 13:41.57 | LunaLovegood | 'config show help res_pjsip endpoint direct_media' only has yes or no |
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| 13:42.15 | Samot | The L2 switches have nothing to do with RTP |
| 13:42.29 | sekil | you mean phone from server 1 rtp to phone from server 2? |
| 13:42.32 | Samot | The devices involved and Asterisk have something to do with it. |
| 13:42.40 | Samot | Ie. the two phones. |
| 13:43.39 | LunaLovegood | Two phones in my subnet can send RTP to each other directly through the L2 switch, after Asterisk takes care of the SIP/SDP stuff |
| 13:44.01 | LunaLovegood | but only if I use direct_media=yes |
| 13:44.04 | sekil | LunaLovegood: yes |
| 13:44.06 | LunaLovegood | AFAIK |
| 13:44.52 | LunaLovegood | But if I use direct_media=yes, the phones will try to send RTP directly to the ITSP, and that won't work since the phone VLAN is not connected to the Internet. |
| 13:45.02 | LunaLovegood | for external calls I mean. |
| 13:45.17 | sekil | LunaLovegood: direct_media=no is needed on a trunk to ITSP |
| 13:45.37 | sekil | LunaLovegood: and Asterisk will proxy rtp then |
| 13:45.43 | LunaLovegood | exactly |
| 13:45.55 | Samot | OK I'm going to try this again. |
| 13:46.01 | LunaLovegood | but I DON'T want it to proxy between two local phones on my local subnet |
| 13:46.03 | Samot | direct_media makes Asterisk "proxy" the media. |
| 13:46.24 | [TK]D-Fender | that sounds backwards |
| 13:46.26 | Samot | So that when it's needed for like voicemail, IVR, and other Asterisk side stuff in the call it can put itself back in the media path. |
| 13:47.02 | sekil | still has 1.2 versions somewhere with can_reinvite |
| 13:47.32 | Samot | Putting in a second Asterisk box is going to be more problems than answers. |
| 13:47.42 | Samot | Because now you've got two boxes in the call path |
| 13:48.07 | Samot | Both of which are B2BUA's and do stuff |
| 13:50.15 | [TK]D-Fender | that is indeed backwards |
| 13:50.33 | [TK]D-Fender | ;direct_media=yes ; Determines whether media may flow directly between |
| 13:50.33 | [TK]D-Fender | <PROTECTED> |
| 13:50.43 | LunaLovegood | Pretty sure that 'direct_media=yes' is the one that allows phones to send RTP to each other directly without being proxy'd by ASterisk, not the reverse. |
| 13:50.57 | [TK]D-Fender | LunaLovegood, indeed |
| 13:52.04 | Samot | Asterisk by default tries to redirect the RTP media stream to go directly from the caller to the callee. Some devices do not support this (especially if one of them is behind a NAT). The default setting is YES. If you have all clients behind a NAT, or for some other reason want Asterisk to stay in the audio path, you may want to turn this off. |
| 13:52.06 | LunaLovegood | Yeah, and what I want is the 'direct_media=yes' behaviour when it's an internal call between two phones in the same subnet, but I want 'direct_media=no' when the call involves the ITSP (extarnal call). |
| 13:52.18 | [TK]D-Fender | LunaLovegood, correct |
| 13:52.39 | LunaLovegood | Because my VoIP VLAN has no route to the Internet. |
| 13:52.45 | Samot | Additionally this option does not disable all reINVITE operations. It only controls Asterisk generating reINVITEs for the specific purpose of setting up a direct media path. If a reINVITE is needed to switch a media stream to inactive (when placed on hold) or to T.38, it will still be done, regardless of this setting. Note that direct T.38 is not supported. |
| 13:53.09 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Configuration_res_pjsip |
| 13:55.04 | LunaLovegood | There's a 'local_net' option, but it says it's for NAT purposes. |
| 13:55.12 | Samot | How does Asterisk get calls from the PSTN? |
| 13:55.17 | Samot | Over the Internet? |
| 13:55.28 | LunaLovegood | In my case? through the ITSP. |
| 13:55.38 | Samot | How does it get from the ITSP to Asterisk? |
| 13:55.57 | Samot | How does a call get from Asterisk to the ITSP? |
| 13:56.01 | LunaLovegood | Through a 1 Gbps Internet connection |
| 13:56.04 | Samot | OK |
| 13:56.22 | Samot | So then your Asterisk box is on two subnets? |
| 13:56.42 | Samot | Because it just can't be on the VoIP VLAN that has no Internet route. |
| 13:56.48 | LunaLovegood | Yeah, it has access to the internet, and to the VoIP vlan for the SIP phones. |
| 13:58.06 | LunaLovegood | The Asterisk box is not a router though, and I don't want it to be one. The VoIP VLAN shouldn't have Internet access. |
| 13:58.30 | sekil | LunaLovegood: that's pretty much how I do here.. |
| 13:58.40 | Samot | OK |
| 13:58.44 | Samot | I get that |
| 13:58.55 | Samot | What does that have to do with direct_media? |
| 13:59.15 | Samot | The peer between Asterisk and the ITSP has to have direct_media=no |
| 13:59.23 | LunaLovegood | yes |
| 13:59.29 | Samot | So.. |
| 13:59.36 | Samot | If direct_media is yes for all the endpoints |
| 13:59.40 | Samot | But not for the trunk.. |
| 13:59.43 | Samot | Guess what happens? |
| 13:59.48 | LunaLovegood | That works? |
| 13:59.49 | LunaLovegood | cool |
| 13:59.52 | LunaLovegood | didn't think of that |
| 13:59.55 | Samot | When endpoint A calls end point B it's direct_media |
| 14:00.06 | sekil | LunaLovegood: all phones have yes...that's default I think |
| 14:00.10 | Samot | No. |
| 14:00.11 | Samot | FFS. |
| 14:00.15 | Samot | It's default to yes |
| 14:00.18 | Samot | Not "all phones" |
| 14:00.20 | Samot | All PEERS |
| 14:00.26 | sekil | right |
| 14:00.44 | Samot | You don't add direct_media=no to a peer, Asterisk assumes "yes" |
| 14:00.52 | sekil | yeah |
| 14:01.16 | sekil | LunaLovegood: so all RTP between phones is direct |
| 14:01.22 | sekil | LunaLovegood: and one to ITSP is proxied... |
| 14:01.23 | Samot | Also, when A puts B on hold guess what happens? |
| 14:01.30 | sekil | LunaLovegood: I guess that's what you want.. |
| 14:01.34 | Samot | Asterisk gets involved. |
| 14:01.38 | LunaLovegood | yes thanks |
| 14:01.45 | Samot | Because how do you think MoH plays back? |
| 14:02.10 | sekil | does pjsip have concept of multiple UAs like in FS? |
| 14:02.15 | Samot | Asterisk has to be part of the media stream. |
| 14:02.53 | Samot | PJSIP supports multiple contact for peer. |
| 14:03.05 | Samot | Like most SIP servers do. |
| 14:03.08 | LunaLovegood | But Asterisk can just do another INVITE and point the RTP to itself when it wants to play MOH |
| 14:03.17 | Samot | Right |
| 14:03.21 | Samot | That's a re-INVITE |
| 14:03.30 | sekil | I meant socket pair per profile...like listen=1.2.3.4:5060..listen=5.6.7.8:5060 on other |
| 14:04.19 | Samot | When A puts a call on hold, A is sending a re-INVITE |
| 14:04.31 | Samot | It's telling B that media is only one way... |
| 14:04.34 | Samot | From A to B |
| 14:04.52 | Samot | Normally if A and B were in a direct call to each other, it would be dead air |
| 14:05.01 | sekil | or local MoH |
| 14:05.03 | Samot | Because that re-INVITE hits Asterisk... |
| 14:05.18 | Samot | Asterisk updates the SDP to B |
| 14:05.27 | Samot | Saying "Hey, your media is from ME now |
| 14:05.44 | Samot | sekil: Not many IP phones have local MoH |
| 14:05.56 | Samot | Since that's what a PBX they connect to is for. |
| 14:05.59 | sekil | Samot: Gigaset, linksys, cisco.. |
| 14:06.19 | sekil | not sure of others |
| 14:07.12 | sekil | Panasonic haven't |
| 14:08.06 | Samot | I'm looking at my Cisco SPA514G.. |
| 14:08.11 | Samot | I dont see a spot for MoH |
| 14:08.48 | sekil | Samot: it goes too-too-too every n seconds |
| 14:09.02 | sekil | Samot: by itself |
| 14:09.09 | Samot | That's not MoH |
| 14:09.17 | sekil | Samot: if send-only / or receive-only |
| 14:09.20 | sekil | comes to phone |
| 14:09.43 | sekil | Samot: it's not dead air too |
| 14:09.50 | Samot | So. |
| 14:09.57 | Samot | You're on your Cisco SPA phone. |
| 14:09.58 | sekil | Samot: Gigaset even has nice music.. |
| 14:10.17 | Samot | OK. |
| 14:10.48 | Samot | Never once in over a decade of doing this have I had someone go "I need my IP phone to do MoH directly" |
| 14:10.50 | sekil | Samot: you can't upload your own file...if that's what you mean... |
| 14:11.03 | sekil | Samot: but you can provision moh server too.. |
| 14:11.13 | Samot | You could. |
| 14:11.21 | Samot | But again, this is what a PBX is FOR |
| 14:11.43 | sekil | Samot: I thought it was for transferring calls etc |
| 14:12.06 | Samot | Direct phone to phone calling is not a thing in the business world. |
| 14:12.48 | sekil | Samot: when you have larger operation and hosted..having a moh server is better |
| 14:12.50 | Samot | Most providers that know what they are doing for "hosted voice" and give you a phone turn off SIP URI dialing to stop that. |
| 14:13.08 | Samot | They want that phone going through their system |
| 14:13.20 | Samot | Just like you want your phones in your office going to your PBX |
| 14:13.25 | Samot | So you can control THEM |
| 14:14.00 | Samot | Putting a phone on Bob's desk that lets Bob get/make calls directly to other endpoints makes all the stuff on the PBX to control Bob's calls pointless. |
| 14:14.05 | Samot | Bob can get around it. |
| 14:15.20 | sekil | Samot: what do you mean? signalling is always to server... |
| 14:15.26 | sekil | Samot: RTP can be direct... |
| 14:15.32 | Samot | Not with SIP URI dialing. |
| 14:15.36 | LunaLovegood | But if the phones are on a network or vlan with no Internet access, then he can only call within the office anyway. |
| 14:15.41 | sekil | Samot: ah... |
| 14:15.48 | Samot | Which.. |
| 14:16.01 | Samot | By the way is how the phone calls Asterisk anyways.. |
| 14:16.17 | Samot | It's just forced to use what is in the Host/Proxy setting for that SIP account on the phone. |
| 14:16.25 | sekil | Samot: yeah |
| 14:16.46 | Samot | Enabling "SIP URI Dialing" on those phones allows from a direct call from that Phone to another endpoint.. |
| 14:16.50 | sekil | I provision phones to talk to server[s] only.. |
| 14:17.00 | Samot | It doesn't use the Host/Proxy domain |
| 14:17.05 | Samot | Because you're entering it. |
| 14:17.18 | sekil | nowadays there's a provisioning field to allow SIP only from servers |
| 14:17.28 | Samot | Yes, I know. |
| 14:17.46 | sekil | I know you know....rethorical |
| 14:17.49 | Samot | I work with/at ITSPs/LECs. |
| 14:21.59 | *** join/#asterisk dan_j (sid21651@gateway/web/irccloud.com/x-sxfmkpefamggdnxh) |
| 14:22.24 | Samot | Just because something like Asterisk _can_ do something doesn't mean it _should_ do it in most practical deployments. |
| 14:23.06 | Samot | Like you _can_ put Asterisk (or any PBX) directly on the Internet but it's not something you general _should_ do. |
| 14:23.12 | Samot | generally |
| 14:23.19 | sekil | absolutely.. |
| 14:23.35 | Samot | So yes some IP phones _can_ do MoH.. |
| 14:23.51 | Samot | But in 99.9% of the practical deployments they do not. |
| 14:24.06 | Samot | Because they shouldn't be. |
| 14:24.41 | sekil | when you have proxy and local office...for inter-calls who care about moh |
| 14:24.51 | sekil | for outside calls you have moh from gw/pbx |
| 14:25.03 | sekil | and it makes sense to have it there |
| 14:25.07 | Samot | They care. |
| 14:25.10 | Samot | Sorry... |
| 14:25.12 | sekil | ok |
| 14:25.25 | sekil | so you do a moh server |
| 14:25.25 | Samot | Clients care if there is no MoH between local calls |
| 14:25.49 | Samot | Clients care about a lot of little things. |
| 14:26.10 | sekil | depends on office size.. |
| 14:26.11 | Samot | They also except things. |
| 14:26.15 | Samot | No, it doesnt. |
| 14:26.21 | Samot | Again, I work for ITSPs |
| 14:26.32 | Samot | I deploy service to a lot of different places. |
| 14:26.42 | sekil | I am an ITSP too... |
| 14:26.42 | Samot | I've seen all sorts of setup requests.. |
| 14:26.48 | Samot | 3 or 30 users |
| 14:26.53 | Samot | They care if there is MoH. |
| 14:27.14 | sekil | so if you have like 3 people in the same room |
| 14:27.18 | Samot | I've had clients with 4 users that cared about things more than the clients with 40 users. |
| 14:27.23 | sekil | they don't even use phones |
| 14:27.27 | Samot | Office size does not mean they don't care. |
| 14:27.32 | sekil | to talk between themselves |
| 14:27.47 | Samot | What does that have to do with it? |
| 14:27.51 | sekil | they need phones to call out.. |
| 14:27.55 | Samot | OK. |
| 14:28.04 | Samot | So they won't be calling each other. |
| 14:28.10 | Samot | Or putting each on hold |
| 14:28.15 | Samot | What if they transfer a call? |
| 14:28.45 | sekil | from the outside line..caller gets moh |
| 14:28.51 | Samot | OK. |
| 14:28.55 | sekil | from pbx/gw/whatnot |
| 14:29.25 | sekil | like I said...my clients hosted pbx have inter-calls via proxy.. |
| 14:29.41 | sekil | so they don't have moh right now..I should do some moh server btw.. |
| 14:30.06 | sekil | but for outbound/inbound to PSTN or wherever moh is generated from PBX cluster |
| 14:30.29 | Samot | So when A calls B in the office, they don't go to the PBX? |
| 14:30.35 | sekil | no |
| 14:30.41 | Samot | You have a proxy route them back to each other? |
| 14:30.43 | sekil | no need at all |
| 14:30.45 | Samot | And then what? |
| 14:30.53 | sekil | yeah |
| 14:30.54 | Samot | Have the proxy route to the PBX for voicemail? |
| 14:30.57 | sekil | sure |
| 14:31.14 | Samot | OK. |
| 14:31.21 | Samot | So they have a PBX.. |
| 14:31.28 | Samot | They just don't use it locally |
| 14:31.40 | sekil | in/out outside calls end up on pbx |
| 14:31.46 | Samot | They just don't use it locally |
| 14:31.51 | sekil | and that's where is dialplan.. |
| 14:32.18 | sekil | transferring/pickups/blah for local calls..is done via proxy or phones themselves |
| 14:32.35 | sekil | proxy can't do much there but mangle SIP... |
| 14:33.10 | sekil | transfers are done via phones themselves...on outbound...PBX kicks in to help with REFER |
| 14:33.29 | sekil | this is SIP anyway.. |
| 14:33.34 | sekil | should be like this I guess.. |
| 14:33.41 | Samot | Not really. |
| 14:34.01 | Samot | Putting a proxy in an office to do that is not something normally done. |
| 14:34.16 | Samot | You basically bypass the PBX for local calls. |
| 14:34.25 | Samot | Where is call forwarding done? |
| 14:34.27 | sekil | well yes and no.. |
| 14:34.33 | Samot | What about Follow-Me? |
| 14:34.48 | Samot | What about call recording? |
| 14:35.07 | sekil | user can set 302 on phone itself |
| 14:35.17 | Samot | Sigh |
| 14:35.35 | sekil | right now I don't do interaction with proxy..but I could |
| 14:35.36 | Samot | So one of the biggest ways phones are hacked is the answer. |
| 14:35.42 | sekil | so it doesn't reach phone |
| 14:36.18 | sekil | follow-me is call park? |
| 14:36.29 | sekil | I implement it with the PBX |
| 14:36.50 | sekil | call recordings for local calls I don't do right now..but SIP REC and rtp proxy can do it... |
| 14:36.50 | Samot | Almost all the fraud happening from clients with IP phones/devices are because they got hacked and someone setup CF to redirect calls to an International localation. |
| 14:37.08 | sekil | yeah..but there are safe guards on proxy/pbx/etc |
| 14:37.19 | Samot | All this could be handled via Asterisk |
| 14:37.20 | sekil | like no cf to international and such.. |
| 14:37.23 | Samot | The actual PBX. |
| 14:37.49 | sekil | so you want to do that like in the 60s |
| 14:37.56 | sekil | smart central device... |
| 14:38.00 | sekil | and stupid endpoints... |
| 14:38.23 | Samot | What is the proxy? |
| 14:38.28 | Samot | A central device? |
| 14:38.29 | sekil | opensips |
| 14:38.31 | sekil | ah |
| 14:38.41 | Samot | So now if your clients have issues.. |
| 14:38.48 | Samot | You have the proxy at their site. |
| 14:38.51 | Samot | And their hosted PBX |
| 14:38.56 | sekil | no |
| 14:39.10 | sekil | only phones are on premises.. |
| 14:39.33 | Samot | So then this is actually a standard hosted voice deployment |
| 14:39.49 | sekil | right |
| 14:40.15 | Samot | Except for the fact you never send "local" calls to the PBX |
| 14:40.23 | Samot | That's where you miscommunicated. |
| 14:40.31 | Samot | "local" as in on the Local NETWORK |
| 14:40.49 | Samot | Vs. Internal calling. |
| 14:40.52 | sekil | so A calls B |
| 14:40.59 | Samot | "Locally" |
| 14:41.01 | Samot | But it's not |
| 14:41.02 | sekil | where A and B are regged via proxy.. |
| 14:41.07 | Samot | It's still using the Internet |
| 14:41.16 | sekil | ah...no Internet here :) |
| 14:41.21 | sekil | that's what I didn't explain.. |
| 14:41.30 | sekil | I mostly have L1/L2 to customers.. |
| 14:41.30 | Samot | How can it be hosted and have no Internet? |
| 14:41.38 | sekil | so it's on a closed networks |
| 14:41.40 | Samot | .... |
| 14:42.19 | sekil | well hosted can be off Internet.. |
| 14:42.35 | Samot | It's still on a WAN |
| 14:42.41 | Samot | Be it public or private. |
| 14:42.55 | sekil | well..that's semantics.. |
| 14:43.03 | Samot | OK. |
| 14:43.06 | sekil | what is a WAN nowadays... |
| 14:43.54 | Samot | Well I dislike games and conversations where the goal posts move during them. |
| 14:44.10 | sekil | back when we had serial links only we had LAN/WAN/MAN... |
| 14:44.22 | sekil | nowadays with MPLS VPNs...VPLSs.. |
| 14:44.33 | sekil | you practically have one big LAN |
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| 14:49.42 | Samot | A MPLS network is considered a WAN network |
| 14:49.48 | Samot | Not a LAN network. |
| 14:50.10 | Samot | Because it's generally connecting multiple LANs over a large area together. |
| 14:50.23 | sekil | I meant as a customer viewpoint |
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| 15:37.22 | drmessano | Never heard of MPLS being considered a LAN |
| 15:37.54 | drmessano | "One big LAN" literally contradicts the notion of WAN and LAN |
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| 15:58.30 | twitchnln | morning everyone |
| 16:04.36 | twitchnln | anyone in here used iaxmodems in production environment? |
| 16:08.13 | [TK]D-Fender | lots of people have |
| 16:10.03 | twitchnln | I have a client that sends about 10k-15k faxes a day, I need to spec a chassis to run 23 iaxmodems, was wondering what others would recommend. |
| 16:11.47 | twitchnln | having difficulty determining how much overhead/io each modem will require. |
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| 17:31.06 | Samot | 10k in faxes? |
| 17:31.23 | Samot | You mean total pages or total fax calls? |
| 17:31.44 | twitchnln | fax calls |
| 17:32.12 | Samot | In a 24 hr period? |
| 17:32.22 | twitchnln | apparently when communicating with doctors hippa makes faxing best solution, yep |
| 17:32.36 | twitchnln | yes thats daily |
| 17:33.30 | Samot | They are using 400+ channels, minimum per hour ? |
| 17:34.00 | Samot | I have doctors offices |
| 17:34.20 | twitchnln | currently they are using rocketport and 23 analog modems |
| 17:34.32 | Samot | So 23 lines? |
| 17:34.35 | twitchnln | i'm trying to move them out of the dark ages |
| 17:35.00 | Samot | How does one do 10k calls over 23 channels? |
| 17:35.18 | Samot | The maths dont make it possible. |
| 17:36.14 | twitchnln | thats 434 calls per channel per day |
| 17:36.38 | twitchnln | 434 1 page faxes doesn't add up to 1440 minutes |
| 17:36.57 | Samot | 1 fax page is roughly 1 min |
| 17:37.17 | twitchnln | 1 to 1.5 |
| 17:37.45 | Samot | 23 per minute at one page means 33k+ every 23 hours |
| 17:37.58 | Samot | With. 23 faxes going at all times |
| 17:38.06 | twitchnln | which is 3x what he's sending |
| 17:38.28 | Samot | So they do this 2r hours straight? |
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| 17:39.15 | twitchnln | usually the fax jobs run during business hours, though some of the businnesses are on the other coast, so their hours are a bit different |
| 17:39.20 | Samot | All at one page? |
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| 17:39.36 | twitchnln | never more than 2 |
| 17:39.42 | twitchnln | pages |
| 17:39.47 | Samot | Just trying to digure out the load. |
| 17:39.55 | twitchnln | yeah, thats where i'm at |
| 17:41.41 | Samot | But IAX modems are probably a bad call |
| 17:41.55 | twitchnln | what would you recommend? |
| 17:42.10 | Samot | That software got its first update in 7 yeas 2 years ago almost. |
| 17:42.39 | Samot | T.38 based atas |
| 17:42.49 | Samot | Fxs gateway banks |
| 17:43.08 | Samot | Something that has support and current |
| 17:43.21 | Samot | Since this ia medical abd HIPPA |
| 17:43.41 | Samot | damn phone |
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| 17:49.32 | twitchnln | if I use atas or fxs channel bank, that would mean that i end up continuing to support a 20 year old rocketport on his computer that connects to all the analog modems |
| 17:49.49 | twitchnln | i'm trying to work the analog out of the mixture |
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| 19:54.37 | *** topic/#asterisk by bford -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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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| 20:11.49 | jeffspeff | can tcp and udp both bind to the same port? |
| 20:15.20 | [TK]D-Fender | same port NUMBEr |
| 20:15.26 | [TK]D-Fender | each has their own #'s |
| 20:15.44 | [TK]D-Fender | udp 55 has nothng to do with tcp 55 |
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| 23:17.50 | kfife | What's the way to associate channels with a given call (e.g. in AMI)? |
| 23:17.58 | kfife | For example |
| 23:18.22 | kfife | For example does each channel have a primary key associated with it that I can use to link them up? |
| 23:19.08 | kfife | Or find the status of all bridges? |
| 23:21.18 | kfife | If the latter, what about non-bridged calls, for example to an IVR? |
| 23:21.44 | kfife | A nudge in the right direction much appreicated. |
| 23:22.30 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:23.06 | [TK]D-Fender | kfife, A "call" would be 2 given channels that are bridged for instance. There is a value in the channel dumps showing bridged channels to tie them together |
| 23:23.31 | [TK]D-Fender | You can also add variables to channels you originate to make them easier to track |
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| 23:24.02 | kfife | Love it. |
| 23:25.36 | kfife | [TK]D-Fender: : Thank you! Bridge ID! |
| 23:25.43 | kfife | You da mensch! |
| 23:28.02 | kfife | [TK]D-Fender: what about a single-ended call? Such as an IVR call. Those would just not have a bridge ID? |
| 23:28.30 | [TK]D-Fender | correct, and the last app, and current location in the dialplan should make it pretty obvious where they are |
| 23:29.02 | kfife | Where do I send the flowers? |
| 23:29.28 | kfife | As a token of thanks to [TK]D-Fender? |
| 23:31.17 | [TK]D-Fender | Those tend to arrive wilted. |
| 23:32.34 | centrex | I'm sure he takes whole bitcoins |
| 23:34.22 | kfife | what about KodakCoin? Do you take taht? You can mine them with a pocket calculator |
| 23:34.31 | kfife | so far... |