IRC log for #asterisk on 20180111

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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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06:20.52bittishey 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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08:02.03bittisfor 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
08:03.03bittisthis call started via an "originate" request
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10:13.40*** join/#asterisk joerg9811 (548e1ed4@gateway/web/freenode/ip.84.142.30.212)
10:16.13joerg9811Hello everyone. Is there somebody with experience with Asterisk 15 and FreePBX 14 in connection with Deutsche Telekom sip-trunk.telekom.de?
10:21.32joerg9811I 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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11:32.04Dovid@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:
11:32.04Dovidusb:001/004          xpp_usb-     e4e4:1160 Astribank-modular no-firmware
11:34.57tzafrirDovid, any chance you don't have fxload on that system?
11:35.13tzafrirLatest dahdi should have that firmware
11:35.24tzafrirWhat version of dahdi is it?
11:36.07Dovidtzafrir: I have fxload. I have dahdi-2.11.1 installed
11:36.26DovidI remember last time I had this issue and a firmware update fixed it
11:36.46DovidI dont have xpp_fxloader.
11:38.43Dovidsorry I do. this is what I get: https://pastebin.com/qtTjRPYw let me look @ the script
11:42.34Dovid@tzafrir: line 470 has # We have a potential astribank
11:42.34Dovidastribank_is_starting -a
11:42.52Dovidis that a mistake or is astribank_is_starting -a supposed to be there?
11:46.15tzafrirDovid, not sure what may cause this. /usr/sbin not in PATH?
11:47.29Dovidtzafrir: 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
11:50.35Dovidtzafrir: seems I didnt have libusb. All good now.
11:56.57*** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com)
12:05.03bittisguys anyone here working with Asterisk 12 or higher, originating calls and managed to figure out how CDR records are being created?
12:07.50stefan27try 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
12:09.29stefan27https://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
12:09.34stefan27(I never used asterisk's built in CDR)
12:10.14bittisthe 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
12:10.45stefan27Aha, if you have a specific question, try posting an extensive problem description on a forum
12:11.00stefan27or directly here (I'm unlikely to be able to direct you)
12:11.00bittiswhen 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
12:11.26bittisif answered then n all is good, one cdr record created
12:11.48bittisif not answered though then i end up with 2 records
12:14.06bittisthen this could be because channels never got bridged we end up with 2 cdr records rather than 1
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12:54.39stefan27if 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
13:01.13*** part/#asterisk joerg9811 (548e1ed4@gateway/web/freenode/ip.84.142.30.212)
13:09.59Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification
13:10.09Samotbittis: It changed in v12
13:10.22Samothttps://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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13:26.11bittisis the CEL description a more comprehensive guide? will go through it
13:26.16bittisSamot: Thank you
13:30.31SamotCDR and CEL are two different things.
13:33.40*** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-vwphfeffbqcrxskp)
13:33.48LunaLovegoodI 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?
13:37.09*** join/#asterisk sekil (~sekil@nat-73.net011.net)
13:39.04LunaLovegoodI 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.
13:40.20Samotdirect_media has a few options.
13:40.41LunaLovegoodI don't want to use NAT or the like though.
13:41.19SamotWhat?
13:41.28LunaLovegoodAbout 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.
13:41.40SamotThat's not how that works.
13:41.56Samotdirect_media=yes makes Asterisk "proxy" the media.
13:41.57LunaLovegood'config show help res_pjsip endpoint direct_media' only has yes or no
13:42.03*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
13:42.15SamotThe L2 switches have nothing to do with RTP
13:42.29sekilyou mean phone from server 1 rtp to phone from server 2?
13:42.32SamotThe devices involved and Asterisk have something to do with it.
13:42.40SamotIe. the two phones.
13:43.39LunaLovegoodTwo phones in my subnet can send RTP to each other directly through the L2 switch, after Asterisk takes care of the SIP/SDP stuff
13:44.01LunaLovegoodbut only if I use direct_media=yes
13:44.04sekilLunaLovegood: yes
13:44.06LunaLovegoodAFAIK
13:44.52LunaLovegoodBut 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.
13:45.02LunaLovegoodfor external calls I mean.
13:45.17sekilLunaLovegood: direct_media=no is needed on a trunk to ITSP
13:45.37sekilLunaLovegood: and Asterisk will proxy rtp then
13:45.43LunaLovegoodexactly
13:45.55SamotOK I'm going to try this again.
13:46.01LunaLovegoodbut I DON'T want it to proxy between two local phones on my local subnet
13:46.03Samotdirect_media makes Asterisk "proxy" the media.
13:46.24[TK]D-Fenderthat sounds backwards
13:46.26SamotSo 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.
13:47.02sekilstill has 1.2 versions somewhere with can_reinvite
13:47.32SamotPutting in a second Asterisk box is going to be more problems than answers.
13:47.42SamotBecause now you've got two boxes in the call path
13:48.07SamotBoth of which are B2BUA's and do stuff
13:50.15[TK]D-Fenderthat 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>
13:50.43LunaLovegoodPretty 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-FenderLunaLovegood, indeed
13:52.04SamotAsterisk 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.
13:52.06LunaLovegoodYeah, 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-FenderLunaLovegood, correct
13:52.39LunaLovegoodBecause my VoIP VLAN has no route to the Internet.
13:52.45SamotAdditionally 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.
13:53.09Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Configuration_res_pjsip
13:55.04LunaLovegoodThere's a 'local_net' option, but it says it's for NAT purposes.
13:55.12SamotHow does Asterisk get calls from the PSTN?
13:55.17SamotOver the Internet?
13:55.28LunaLovegoodIn my case? through the ITSP.
13:55.38SamotHow does it get from the ITSP to Asterisk?
13:55.57SamotHow does a call get from Asterisk to the ITSP?
13:56.01LunaLovegoodThrough a 1 Gbps Internet connection
13:56.04SamotOK
13:56.22SamotSo then your Asterisk box is on two subnets?
13:56.42SamotBecause it just can't be on the VoIP VLAN that has no Internet route.
13:56.48LunaLovegoodYeah, it has access to the internet, and to the VoIP vlan for the SIP phones.
13:58.06LunaLovegoodThe Asterisk box is not a router though, and I don't want it to be one. The VoIP VLAN shouldn't have Internet access.
13:58.30sekilLunaLovegood: that's pretty much how I do here..
13:58.40SamotOK
13:58.44SamotI get that
13:58.55SamotWhat does that have to do with direct_media?
13:59.15SamotThe peer between Asterisk and the ITSP has to have direct_media=no
13:59.23LunaLovegoodyes
13:59.29SamotSo..
13:59.36SamotIf direct_media is yes for all the endpoints
13:59.40SamotBut not for the trunk..
13:59.43SamotGuess what happens?
13:59.48LunaLovegoodThat works?
13:59.49LunaLovegoodcool
13:59.52LunaLovegooddidn't think of that
13:59.55SamotWhen endpoint A calls end point B it's direct_media
14:00.06sekilLunaLovegood: all phones have yes...that's default I think
14:00.10SamotNo.
14:00.11SamotFFS.
14:00.15SamotIt's default to yes
14:00.18SamotNot "all phones"
14:00.20SamotAll PEERS
14:00.26sekilright
14:00.44SamotYou don't add direct_media=no to a peer, Asterisk assumes "yes"
14:00.52sekilyeah
14:01.16sekilLunaLovegood: so all RTP between phones is direct
14:01.22sekilLunaLovegood: and one to ITSP is proxied...
14:01.23SamotAlso, when A puts B on hold guess what happens?
14:01.30sekilLunaLovegood: I guess that's what you want..
14:01.34SamotAsterisk gets involved.
14:01.38LunaLovegoodyes thanks
14:01.45SamotBecause how do you think MoH plays back?
14:02.10sekildoes pjsip have concept of multiple UAs like in FS?
14:02.15SamotAsterisk has to be part of the media stream.
14:02.53SamotPJSIP supports multiple contact for peer.
14:03.05SamotLike most SIP servers do.
14:03.08LunaLovegoodBut Asterisk can just do another INVITE and point the RTP to itself when it wants to play MOH
14:03.17SamotRight
14:03.21SamotThat's a re-INVITE
14:03.30sekilI meant socket pair per profile...like listen=1.2.3.4:5060..listen=5.6.7.8:5060 on other
14:04.19SamotWhen A puts a call on hold, A is sending a re-INVITE
14:04.31SamotIt's telling B that media is only one way...
14:04.34SamotFrom A to B
14:04.52SamotNormally if A and B were in a direct call to each other, it would be dead air
14:05.01sekilor local MoH
14:05.03SamotBecause that re-INVITE hits Asterisk...
14:05.18SamotAsterisk updates the SDP to B
14:05.27SamotSaying "Hey, your media is from ME now
14:05.44Samotsekil: Not many IP phones have local MoH
14:05.56SamotSince that's what a PBX they connect to is for.
14:05.59sekilSamot: Gigaset, linksys, cisco..
14:06.19sekilnot sure of others
14:07.12sekilPanasonic haven't
14:08.06SamotI'm looking at my Cisco SPA514G..
14:08.11SamotI dont see a spot for MoH
14:08.48sekilSamot: it goes too-too-too every n seconds
14:09.02sekilSamot: by itself
14:09.09SamotThat's not MoH
14:09.17sekilSamot: if send-only / or receive-only
14:09.20sekilcomes to phone
14:09.43sekilSamot: it's not dead air too
14:09.50SamotSo.
14:09.57SamotYou're on your Cisco SPA phone.
14:09.58sekilSamot: Gigaset even has nice music..
14:10.17SamotOK.
14:10.48SamotNever once in over a decade of doing this have I had someone go "I need my IP phone to do MoH directly"
14:10.50sekilSamot: you can't upload your own file...if that's what you mean...
14:11.03sekilSamot: but you can provision moh server too..
14:11.13SamotYou could.
14:11.21SamotBut again, this is what a PBX is FOR
14:11.43sekilSamot: I thought it was for transferring calls etc
14:12.06SamotDirect phone to phone calling is not a thing in the business world.
14:12.48sekilSamot: when you have larger operation and hosted..having a moh server is better
14:12.50SamotMost providers that know what they are doing for "hosted voice" and give you a phone turn off SIP URI dialing to stop that.
14:13.08SamotThey want that phone going through their system
14:13.20SamotJust like you want your phones in your office going to your PBX
14:13.25SamotSo you can control THEM
14:14.00SamotPutting 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.
14:14.05SamotBob can get around it.
14:15.20sekilSamot: what do you mean? signalling is always to server...
14:15.26sekilSamot: RTP can be direct...
14:15.32SamotNot with SIP URI dialing.
14:15.36LunaLovegoodBut if the phones are on a network or vlan with no Internet access, then he can only call within the office anyway.
14:15.41sekilSamot: ah...
14:15.48SamotWhich..
14:16.01SamotBy the way is how the phone calls Asterisk anyways..
14:16.17SamotIt's just forced to use what is in the Host/Proxy setting for that SIP account on the phone.
14:16.25sekilSamot: yeah
14:16.46SamotEnabling "SIP URI Dialing" on those phones allows from a direct call from that Phone to another endpoint..
14:16.50sekilI provision phones to talk to server[s] only..
14:17.00SamotIt doesn't use the Host/Proxy domain
14:17.05SamotBecause you're entering it.
14:17.18sekilnowadays there's a provisioning field to allow SIP only from servers
14:17.28SamotYes, I know.
14:17.46sekilI know you know....rethorical
14:17.49SamotI work with/at ITSPs/LECs.
14:21.59*** join/#asterisk dan_j (sid21651@gateway/web/irccloud.com/x-sxfmkpefamggdnxh)
14:22.24SamotJust because something like Asterisk _can_ do something doesn't mean it _should_ do it in most practical deployments.
14:23.06SamotLike you _can_ put Asterisk (or any PBX) directly on the Internet but it's not something you general _should_ do.
14:23.12Samotgenerally
14:23.19sekilabsolutely..
14:23.35SamotSo yes some IP phones _can_ do MoH..
14:23.51SamotBut in 99.9% of the practical deployments they do not.
14:24.06SamotBecause they shouldn't be.
14:24.41sekilwhen you have proxy and local office...for inter-calls who care about moh
14:24.51sekilfor outside calls you have moh from gw/pbx
14:25.03sekiland it makes sense to have it there
14:25.07SamotThey care.
14:25.10SamotSorry...
14:25.12sekilok
14:25.25sekilso you do a moh server
14:25.25SamotClients care if there is no MoH between local calls
14:25.49SamotClients care about a lot of little things.
14:26.10sekildepends on office size..
14:26.11SamotThey also except things.
14:26.15SamotNo, it doesnt.
14:26.21SamotAgain, I work for ITSPs
14:26.32SamotI deploy service to a lot of different places.
14:26.42sekilI am an ITSP too...
14:26.42SamotI've seen all sorts of setup requests..
14:26.48Samot3 or 30 users
14:26.53SamotThey care if there is MoH.
14:27.14sekilso if you have like 3 people in the same room
14:27.18SamotI've had clients with 4 users that cared about things more than the clients with 40 users.
14:27.23sekilthey don't even use phones
14:27.27SamotOffice size does not mean they don't care.
14:27.32sekilto talk between themselves
14:27.47SamotWhat does that have to do with it?
14:27.51sekilthey need phones to call out..
14:27.55SamotOK.
14:28.04SamotSo they won't be calling each other.
14:28.10SamotOr putting each on hold
14:28.15SamotWhat if they transfer a call?
14:28.45sekilfrom the outside line..caller gets moh
14:28.51SamotOK.
14:28.55sekilfrom pbx/gw/whatnot
14:29.25sekillike I said...my clients hosted pbx have inter-calls via proxy..
14:29.41sekilso they don't have moh right now..I should do some moh server btw..
14:30.06sekilbut for outbound/inbound to PSTN or wherever moh is generated from PBX cluster
14:30.29SamotSo when A calls B in the office, they don't go to the PBX?
14:30.35sekilno
14:30.41SamotYou have a proxy route them back to each other?
14:30.43sekilno need at all
14:30.45SamotAnd then what?
14:30.53sekilyeah
14:30.54SamotHave the proxy route to the PBX for voicemail?
14:30.57sekilsure
14:31.14SamotOK.
14:31.21SamotSo they have a PBX..
14:31.28SamotThey just don't use it locally
14:31.40sekilin/out outside calls end up on pbx
14:31.46SamotThey just don't use it locally
14:31.51sekiland that's where is dialplan..
14:32.18sekiltransferring/pickups/blah for local calls..is done via proxy or phones themselves
14:32.35sekilproxy can't do much there but mangle SIP...
14:33.10sekiltransfers are done via phones themselves...on outbound...PBX kicks in to help with REFER
14:33.29sekilthis is SIP anyway..
14:33.34sekilshould be like this I guess..
14:33.41SamotNot really.
14:34.01SamotPutting a proxy in an office to do that is not something normally done.
14:34.16SamotYou basically bypass the PBX for local calls.
14:34.25SamotWhere is call forwarding done?
14:34.27sekilwell yes and no..
14:34.33SamotWhat about Follow-Me?
14:34.48SamotWhat about call recording?
14:35.07sekiluser can set 302 on phone itself
14:35.17SamotSigh
14:35.35sekilright now I don't do interaction with proxy..but I could
14:35.36SamotSo one of the biggest ways phones are hacked is the answer.
14:35.42sekilso it doesn't reach phone
14:36.18sekilfollow-me is call park?
14:36.29sekilI implement it with the PBX
14:36.50sekilcall recordings for local calls I don't do right now..but SIP REC and rtp proxy can do it...
14:36.50SamotAlmost 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.
14:37.08sekilyeah..but there are safe guards on proxy/pbx/etc
14:37.19SamotAll this could be handled via Asterisk
14:37.20sekillike no cf to international and such..
14:37.23SamotThe actual PBX.
14:37.49sekilso you want to do that like in the 60s
14:37.56sekilsmart central device...
14:38.00sekiland stupid endpoints...
14:38.23SamotWhat is the proxy?
14:38.28SamotA central device?
14:38.29sekilopensips
14:38.31sekilah
14:38.41SamotSo now if your clients have issues..
14:38.48SamotYou have the proxy at their site.
14:38.51SamotAnd their hosted PBX
14:38.56sekilno
14:39.10sekilonly phones are on premises..
14:39.33SamotSo then this is actually a standard hosted voice deployment
14:39.49sekilright
14:40.15SamotExcept for the fact you never send "local" calls to the PBX
14:40.23SamotThat's where you miscommunicated.
14:40.31Samot"local" as in on the Local NETWORK
14:40.49SamotVs. Internal calling.
14:40.52sekilso A calls B
14:40.59Samot"Locally"
14:41.01SamotBut it's not
14:41.02sekilwhere A and B are regged via proxy..
14:41.07SamotIt's still using the Internet
14:41.16sekilah...no Internet here :)
14:41.21sekilthat's what I didn't explain..
14:41.30sekilI mostly have L1/L2 to customers..
14:41.30SamotHow can it be hosted and have no Internet?
14:41.38sekilso it's on a closed networks
14:41.40Samot....
14:42.19sekilwell hosted can be off Internet..
14:42.35SamotIt's still on a WAN
14:42.41SamotBe it public or private.
14:42.55sekilwell..that's semantics..
14:43.03SamotOK.
14:43.06sekilwhat is a WAN nowadays...
14:43.54SamotWell I dislike games and conversations where the goal posts move during them.
14:44.10sekilback when we had serial links only we had LAN/WAN/MAN...
14:44.22sekilnowadays with MPLS VPNs...VPLSs..
14:44.33sekilyou practically have one big LAN
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14:49.42SamotA MPLS network is considered a WAN network
14:49.48SamotNot a LAN network.
14:50.10SamotBecause it's generally connecting multiple LANs over a large area together.
14:50.23sekilI meant as a customer viewpoint
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15:37.22drmessanoNever heard of MPLS being considered a LAN
15:37.54drmessano"One big LAN" literally contradicts the notion of WAN and LAN
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15:58.30twitchnlnmorning everyone
16:04.36twitchnlnanyone in here used iaxmodems in production environment?
16:08.13[TK]D-Fenderlots of people have
16:10.03twitchnlnI 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.
16:11.47twitchnlnhaving difficulty determining how much overhead/io each modem will require.
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17:31.06Samot10k in faxes?
17:31.23SamotYou mean total pages or total fax calls?
17:31.44twitchnlnfax calls
17:32.12SamotIn a 24 hr period?
17:32.22twitchnlnapparently when communicating with doctors hippa makes faxing best solution, yep
17:32.36twitchnlnyes thats daily
17:33.30SamotThey are using 400+ channels, minimum per hour ?
17:34.00SamotI have doctors offices
17:34.20twitchnlncurrently they are using rocketport and 23 analog modems
17:34.32SamotSo 23 lines?
17:34.35twitchnlni'm trying to move them out of the dark ages
17:35.00SamotHow does one do 10k calls over 23 channels?
17:35.18SamotThe maths dont make it possible.
17:36.14twitchnlnthats 434 calls per channel per day
17:36.38twitchnln434 1 page faxes doesn't add up to 1440 minutes
17:36.57Samot1 fax page is roughly 1 min
17:37.17twitchnln1 to 1.5
17:37.45Samot23 per minute at one page means 33k+ every 23 hours
17:37.58SamotWith. 23 faxes going at all times
17:38.06twitchnlnwhich is 3x what he's sending
17:38.28SamotSo they do this 2r hours straight?
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17:39.15twitchnlnusually the fax jobs run during business hours, though some of the businnesses are on the other coast, so their hours are a bit different
17:39.20SamotAll at one page?
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17:39.36twitchnlnnever more than 2
17:39.42twitchnlnpages
17:39.47SamotJust trying to digure out the load.
17:39.55twitchnlnyeah, thats where i'm at
17:41.41SamotBut IAX modems are probably a bad call
17:41.55twitchnlnwhat would you recommend?
17:42.10SamotThat software got its first update in 7 yeas 2 years ago almost.
17:42.39SamotT.38 based atas
17:42.49SamotFxs gateway banks
17:43.08SamotSomething that has support and current
17:43.21SamotSince this ia medical abd HIPPA
17:43.41Samotdamn phone
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17:49.32twitchnlnif 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
17:49.49twitchnlni'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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20:11.49jeffspeffcan tcp and udp both bind to the same port?
20:15.20[TK]D-Fendersame port NUMBEr
20:15.26[TK]D-Fendereach has their own #'s
20:15.44[TK]D-Fenderudp 55 has nothng to do with tcp 55
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23:17.50kfifeWhat's the way to associate channels with a given call (e.g. in AMI)?
23:17.58kfifeFor example
23:18.22kfifeFor example does each channel have a primary key associated with it that I can use to link them up?
23:19.08kfifeOr find the status of all bridges?
23:21.18kfifeIf the latter, what about non-bridged calls, for example to an IVR?
23:21.44kfifeA 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-Fenderkfife, 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-FenderYou can also add variables to channels you originate to make them easier to track
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23:24.02kfifeLove it.
23:25.36kfife[TK]D-Fender: : Thank you!  Bridge ID!
23:25.43kfifeYou da mensch!
23:28.02kfife[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-Fendercorrect, and the last app, and current location in the dialplan should make it pretty obvious where they are
23:29.02kfifeWhere do I send the flowers?
23:29.28kfifeAs a token of thanks to [TK]D-Fender?
23:31.17[TK]D-FenderThose tend to arrive wilted.
23:32.34centrexI'm sure he takes whole bitcoins
23:34.22kfifewhat about KodakCoin?  Do you take taht?  You can mine them with a pocket calculator
23:34.31kfifeso far...

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