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| 00:25.04 | Sean-Der | Does anyone know of prebuilt/embedded device that would work well for Asterisk+analog ? |
| 00:25.27 | Sean-Der | I have looked at doing a mini atx and installing a FXO card, but it doesn't really scale |
| 00:26.20 | Sean-Der | It would be nice if I could just buy something, and then just image it |
| 00:28.34 | [TK]D-Fender | you ask for embedded and then say MAXT won't scale... what do you thing embedded will handle? |
| 00:29.58 | Sean-Der | [TK]D-Fender: Sorry I meant scale in terms of time, I will waste time building |
| 00:30.22 | [TK]D-Fender | You should probably be clearer on your on scale of production, etc |
| 00:30.23 | Sean-Der | As far as resource usage, I don't expect to see more than 2-4 channels at a time |
| 00:31.15 | [TK]D-Fender | http://www.rowetel.com/blog/?page_id=440 |
| 00:31.19 | [TK]D-Fender | on embedded solution |
| 00:31.22 | [TK]D-Fender | one* |
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| 00:31.31 | [TK]D-Fender | Xorcom has some as well |
| 00:31.37 | [TK]D-Fender | port mix will vary |
| 00:32.00 | Sean-Der | My immediate need would be maybe 5, with the intention of having 300-350 of them |
| 00:32.05 | Sean-Der | so not a lot, but more than I can handle |
| 00:32.17 | Sean-Der | Cool, thank you very much for your help |
| 00:32.26 | [TK]D-Fender | at that volume you could have a rPC mfg build them to spec... |
| 00:35.56 | Sean-Der | Xorcom looks like it could fit my exact need. Most of the products out there seem like they try to stay locked down and only give you some sort of web GUI (FreePBX or the companies home grown one) |
| 00:36.21 | Sean-Der | I have some custom dialplan, and have a few custom Apps that hit some things via cURL |
| 00:36.25 | Sean-Der | thank you so much |
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| 02:15.29 | *** join/#asterisk Egyptian[Laptop] (~Egyptian@unaffiliated/egyptian) |
| 02:16.34 | Egyptian[Laptop] | hi .. i want to use voip to gsm .. can someone help me with documentation? i do not know the keywords for google |
| 02:20.05 | [TK]D-Fender | Egyptian[Laptop]: SIP GSM gateway |
| 02:20.29 | *** join/#asterisk moshin (~mo@udp001682uds.hawaiiantel.net) |
| 02:25.14 | Egyptian[Laptop] | [TK]D-Fender: thats hardware? |
| 02:25.33 | [TK]D-Fender | Software doesn't magically pull microwaves out of the air..... |
| 02:26.33 | Egyptian[Laptop] | so i dont need documentation? |
| 02:26.59 | [TK]D-Fender | Your entire line of questioning is unclear |
| 02:27.10 | [TK]D-Fender | Please rephrase exactly what it is you'ree looking for |
| 02:28.21 | Egyptian[Laptop] | ok .. i want to make calls from my cell phone using voip to gsm. the gsm is in another country |
| 02:30.54 | [TK]D-Fender | where your cell talks to is irrelevant as long as it works. many cell phones can work using chan_dongle (GOOGLE) via Bluetooth |
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| 02:33.16 | Egyptian[Laptop] | [TK]D-Fender: that looks like a gsm modem and i got a few of those .. so all i need is a voice sim in that? (and the docs of course!) |
| 02:33.57 | [TK]D-Fender | You just said "from my cell phone". What happened to going THROUGH you phone for this? |
| 02:34.07 | [TK]D-Fender | You description is getting WORSE |
| 02:34.19 | Egyptian[Laptop] | ok let expand on the description :) |
| 02:34.26 | Egyptian[Laptop] | i am in country A with my cell phone |
| 02:34.34 | [TK]D-Fender | what DEVICE is your SIP going to be plugged into? |
| 02:34.45 | [TK]D-Fender | Yyour phone? Or some NEW piece of hardware you want us to recommend? |
| 02:34.49 | Egyptian[Laptop] | in country b i got asterisk and a gsm |
| 02:35.01 | [TK]D-Fender | a GSM is not a "thing" |
| 02:35.07 | [TK]D-Fender | Stop saying "my GSM" |
| 02:35.18 | [TK]D-Fender | You have SIM CARD, correct? |
| 02:35.55 | Egyptian[Laptop] | you do realise i konw practically nothing about sip/voip ? |
| 02:36.14 | [TK]D-Fender | Do you have no clue what you plug a SIM c ard into? I never said ANYTHING about voip |
| 02:37.10 | [TK]D-Fender | Your description is not making it any clearer as to what hardware you are looking to plug this SIM card into |
| 02:37.19 | [TK]D-Fender | what is that device? |
| 02:37.24 | Egyptian[Laptop] | ok so in country b i got a gsm phone AND i got vodafone/huawei gsm modems that look like these chan_dongles you talk about |
| 02:37.47 | Egyptian[Laptop] | are chan_dongles and gsm modems the same thing? |
| 02:37.55 | [TK]D-Fender | no |
| 02:38.06 | [TK]D-Fender | forget I even mentioned that |
| 02:38.27 | [TK]D-Fender | be clear about exactly what hardware is where |
| 02:40.14 | Egyptian[Laptop] | why cant i use the gsm phone in country A to do a voip call to the voip server in country b that will automatically forward the call out to the gsm device? |
| 02:40.37 | Egyptian[Laptop] | is trying to understand and learn |
| 02:41.59 | [TK]D-Fender | Your GSM phone in country * has a PROVIDER in country A that is giving it INTERNET ACCESS? |
| 02:42.18 | [TK]D-Fender | Your GSM phone in country A has a PROVIDER in country A that is giving it INTERNET ACCESS? |
| 02:42.36 | Egyptian[Laptop] | yes country a has a gsm phone with 3g access |
| 02:42.40 | Egyptian[Laptop] | 3g internet |
| 02:43.37 | [TK]D-Fender | Then that phone does not matter yet. Install a SIP client on it. that will talk SIP to your server |
| 02:43.58 | [TK]D-Fender | For that part your server requires INTERNET ACCEESS, nothing more. |
| 02:44.29 | [TK]D-Fender | If you want to take that SIP call in from your Phone in "A"... and the4n go out ANOTHER GSM provider, you'll need to have * talk to some piece of HARDWARE to do so. |
| 02:45.01 | [TK]D-Fender | * could use a CELL PHONE using "chan_dongle" as the way * talks to it. Or you could be a special dedicated gateway device |
| 02:45.01 | Egyptian[Laptop] | ok .. the server then will be country B .. and it will be connected to a gsm device |
| 02:45.28 | [TK]D-Fender | could buy* |
| 02:47.02 | Egyptian[Laptop] | [TK]D-Fender: thats cool .. now i am looking at google images and the chan_dongle looks like very much like a huawei 3g usb dongle |
| 02:47.20 | Egyptian[Laptop] | ah .. this page http://wiki.e1550.mobi/doku.php?id=introduction says chan_dongle is an Asterisk channel driver for Huawei UMTS/3G USB modems (dongles). |
| 02:47.49 | [TK]D-Fender | no.... |
| 02:47.58 | [TK]D-Fender | Chan_dongle is a BLUETOOTH DRIVER |
| 02:48.11 | [TK]D-Fender | it has NOTHING to do with Huawei |
| 02:48.26 | [TK]D-Fender | It is the means by which * can talk to your CELL PHONE |
| 02:48.32 | [TK]D-Fender | and use THAT to go out |
| 02:49.03 | [TK]D-Fender | Asterisk Server -> Bluetooth Adapter _. cheap shit cell-phone |
| 02:49.13 | [TK]D-Fender | Asterisk Server -> Bluetooth Adapter -> cheap shit cell-phone |
| 02:49.58 | [TK]D-Fender | Actually I may have picked the wrong driver there... |
| 02:50.04 | [TK]D-Fender | check also : chan_mobile |
| 02:50.38 | [TK]D-Fender | Either way, each should be clear as to what devices they are compatible with. |
| 02:52.57 | smkelly | file: hi |
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| 02:57.04 | Egyptian[Laptop] | [TK]D-Fender: thank you very much for the help. i believe i now have some direction |
| 02:57.21 | Egyptian[Laptop] | as arnie once threatened "i will be back" ;) |
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| 03:21.02 | WIMPy | Err. No. |
| 03:21.14 | WIMPy | Chan_mobile is the Bluetooth thing. |
| 03:21.30 | WIMPy | Chan_dongle is for "modems". |
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| 04:01.19 | Fwny | Has anyone tried to get ice support working with external calls being routed to phones behind a NAT? |
| 04:01.34 | Fwny | I'm trying to do it but asterisk doesn't seem to be generating candidates |
| 04:01.52 | Fwny | when it passes through the INVITE |
| 04:02.40 | Fwny | using 11.7.0 |
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| 05:28.16 | AsterRack | Hello All, I'm trying to get a home pbx up and running. When I dial out i get Call from 'robert' (72.*.*.*:5060) to extension '2814091***' rejected because extension not found in context 'public'. Here is sip.conf and extensions.conf http://pastebin.com/raw.php?i=PDBqypz5 |
| 05:29.03 | AsterRack | I seem to be stuck on the syntax needed to complete the outbound call |
| 05:32.49 | WIMPy | The config you posted doesn't match the message. |
| 05:33.07 | AsterRack | oops old one one sec |
| 05:35.46 | WIMPy | Good, because you don't want to use the default context. |
| 05:37.35 | AsterRack | context is supposed to be public, but im pretty sure i jacked up the config.....back to the drawing board |
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| 07:12.15 | AsterRack | Hello All, Im trying to get a home pbx set up, but i seem to be having issues with the outbound syntax. Im getting an error "chan_sip.c:22978 handle_response_invite: Failed to authenticate on INVITE to" sip.conf and extension.conf >>http://pastebin.com/raw.php?i=21iCDq2U |
| 07:15.31 | kaldemar | AsterRack: you don't have a peer by that name in your sip.conf |
| 07:26.55 | kaldemar | AsterRack: as in you're dialing by host name but you have a peer, you're just not using it. use the peer for dialing and add secret and fromuser parameters, then asterisk knows how to authenticate. |
| 07:28.19 | AsterRack | yeah working on that now i think |
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| 07:59.42 | Zogot | Ahoyhoy |
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| 09:23.14 | mfu | anybody got a idea why asterisk get an timeout after a few secs of a established tls connection? WARNING[23242] chan_sip.c: sip_send_keepalive to 83.125.8.71:5061 returned 0: Succes |
| 09:26.03 | Zogot | mfu: have you added a rtpkeepalive to the general config? |
| 09:26.16 | mfu | and after a few checks more...NOTICE[23242] chan_sip.c: Peer 'proxy.dus.net' is now UNREACHABLE! Last qualify: 12 and chan_sip.c: Probably a DNS error for registration to xxxxxxxxxxx@proxy.dus.net, trying REGISTER again (after 20 seconds) |
| 09:26.21 | mfu | yep |
| 09:26.25 | mfu | Zogot: yep |
| 09:28.35 | mfu | http://pastebin.ca/2563151 |
| 09:30.26 | Zogot | mfu: no idea then, sorry. only been messing with it a few hours. Nothing to do with iptables or so? |
| 09:30.50 | mfu | Zogot: maybe.. its behind nat... |
| 09:31.02 | mfu | Zogot: but a normal (udp) connections works nice |
| 09:31.35 | mfu | Zogot: and the fireall/router dont tell any blocking connections or something |
| 09:31.59 | Chainsaw | mfu: I suspect you've found https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 09:33.27 | Chainsaw | mfu: Applying the patch will fix it. (There's other issues with it, but this one is the most glaring) |
| 09:34.06 | mfu | lets try, thx :) |
| 09:35.03 | Chainsaw | mfu: Are you using a self-signed certificate or a proper one? |
| 09:35.10 | mfu | on my side? |
| 09:35.18 | mfu | on my side yes, selfsigned |
| 09:35.22 | mfu | the other is a wildcard |
| 09:35.34 | Chainsaw | mfu: Ah, okay. As long as your side is self-signed you won't hit the other bug. |
| 09:35.36 | mfu | but a valid from geotrust |
| 09:35.59 | mfu | hm |
| 09:36.11 | mfu | the patch is from may 2012 |
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| 09:36.44 | mfu | i use 11.7 |
| 09:36.54 | mfu | thats very old and not fixed? |
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| 09:37.56 | *** join/#asterisk Bearish (~Bearish@80.250.214.230) |
| 09:37.58 | Chainsaw | mfu: I haven't been unsuccessful in getting Digium to fix it. |
| 09:38.14 | Chainsaw | mfu: As in, the patch is on there. It's obviously correct. I keep telling people here, in their official IRC channel, to apply it. |
| 09:38.22 | Chainsaw | mfu: If that doesn't make them care, I don't know what will. |
| 09:38.43 | mfu | maybe because only a little group is using tls? |
| 09:39.06 | mfu | 99% are using udp and dont need this.. but thx Chainsaw |
| 09:39.49 | Chainsaw | mfu: And if your certificate is at all valid, please apply this too: https://issues.asterisk.org/jira/browse/ASTERISK-17727 |
| 09:40.00 | Chainsaw | mfu: Talking about obviously correct one-liner... |
| 09:42.24 | mfu | ok it works... |
| 09:42.26 | mfu | :) |
| 09:42.33 | mfu | now a ebuild patch :D |
| 09:42.47 | Chainsaw | mfu: Please consider replying in the bug, so it gets applied upstream. |
| 09:43.05 | mfu | ok |
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| 09:43.45 | Chainsaw | mfu: ebuild patch? If you're using Gentoo you already have these if you emerge. |
| 09:44.07 | mfu | cant be true? |
| 09:44.28 | mfu | [ebuild R ] net-misc/asterisk-11.7.0 |
| 09:45.22 | mfu | pbx files # grep -r "ast_wait_for_input" * |
| 09:45.23 | mfu | pbx files # pwd |
| 09:45.23 | mfu | pbx files # |
| 09:45.30 | Chainsaw | mfu: 11.7.0 has those patches. |
| 09:45.49 | mfu | in /usr/portage/net-misc/asterisk/files? |
| 09:45.53 | mfu | dont see him |
| 09:46.07 | Chainsaw | mfu: Part of gentoo-asterisk-patchset-3.7.tar.bz2 |
| 09:46.15 | mfu | hmmm |
| 09:46.20 | Chainsaw | mfu: I really am very sure too. Look in the Changelog. |
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| 09:53.02 | mfu | Chainsaw: you got right |
| 09:53.09 | mfu | Chainsaw: hmmmmmmm |
| 09:54.14 | mfu | ok.. so on my testpbx that patch wasnt active because its no gentoo |
| 09:54.37 | mfu | but on the live gentoo sys it must be active... so its another problem |
| 09:54.57 | mfu | Chainsaw: any other idea? |
| 09:55.42 | Chainsaw | mfu: Well if your TLS stack doesn't crash out right on start, it's not the chaining support. |
| 09:55.56 | Chainsaw | mfu: Generally if it doesn't hold a connection, it's the lack of blocking. |
| 09:56.12 | mfu | nope, the connection established perfect |
| 09:56.19 | Chainsaw | mfu: If it still won't work after that, you'd need to trace it and see if you can see signs of NAT issues. |
| 09:56.23 | mfu | i can call in and out a few secs after.. |
| 09:57.18 | mfu | so lets see if there is some tls/ssl debug on asterisk |
| 09:57.36 | Chainsaw | mfu: Okay. |
| 10:03.57 | Bearish | Can somebody explain contexts to me, like i'm five? Why even bother, when in the end you just include everything in everything? |
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| 10:06.14 | Chainsaw | Bearish: It's how I determine what you're allowed to do. |
| 10:06.33 | Chainsaw | Bearish: If you're calling from a phone inside the office... you can call internationally. |
| 10:06.45 | Chainsaw | beardy: If you're some random Romanian dude on the internet, I probably shouldn't let you do that. |
| 10:06.52 | Chainsaw | Bearish: That's what contexts are for. |
| 10:14.12 | Bearish | Chainsaw: thanks |
| 10:16.03 | Bearish | Chainsaw: so basically, i'm supposed to have a "int-phones" context and then include it where I see fit? |
| 10:16.43 | Chainsaw | Bearish: Well, contexts are very personal choices. It depends on how *you* run your business. |
| 10:22.41 | mfu | Chainsaw: chan_sip.c: SIP call-id changed from '6f55a5fa3f0228f808504b726c84370d@192.168.3.6:5060' to '0228cfab4fe4681b1b3a638b42a05a24@my_external_ip:5061' |
| 10:23.19 | mfu | Chainsaw: is this normal? seems like asterisk uses 5060 and 5061 fpr this trunk? |
| 10:23.56 | Chainsaw | mfu: I would need to see that in context, rather then as just a single line. |
| 10:24.38 | mfu | mom plz |
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| 10:27.52 | mfu | Chainsaw: seems like asterisk using the port from the srv record?.. can i /msg you the pastebin url? |
| 10:28.08 | Chainsaw | mfu: I'd rather you shared it here, so others can give their input as well. |
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| 10:28.34 | mfu | of course, but dont know if on debug level4 there is any secure info :) |
| 10:28.40 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:28.53 | Chainsaw | mfu: It will never show passwords. |
| 10:29.07 | mfu | http://bpaste.net/show/170066/ :) |
| 10:30.39 | mfu | but.. dont see the great goal for the tls problem on proxy.dus.net |
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| 10:32.20 | Chainsaw | mfu: You're going to need someone well-versed in NAT issues, because that's what this one looks like at first glance. |
| 10:32.32 | Chainsaw | mfu: (My kit is on an externally reachable IP) |
| 10:33.12 | mfu | hm |
| 10:33.29 | mfu | you mean the complete conenctions or only the tls? |
| 10:33.36 | mfu | because normal udp works nice |
| 10:33.48 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 10:46.11 | mfu | Chainsaw: ok.. maybe i put the pbx in the dmz and allow only traffic from/to the 2 trunks... so this problem should be solved |
| 10:46.40 | Chainsaw | mfu: Hope that does it for you. |
| 10:46.44 | mfu | on the route/iptables there is nothing to see and all modules for sip are unloaded |
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| 11:57.02 | matrix1233 | hello |
| 11:57.15 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 11:57.42 | mfu | Chainsaw: same problem... |
| 11:57.58 | mfu | Chainsaw: maybe comment out "localnet"? |
| 11:58.09 | matrix1233 | i have asterisk 11.07 and i wanna use a remote database with res_config_mysql but i have alway ERROR[18205]: res_config_mysql.c:1577 mysql_reconnect: MySQL RealTime: Failed to connect database server |
| 11:58.56 | matrix1233 | with my local databaseserver its work but with the remote i can't connect |
| 11:59.27 | matrix1233 | i have tested also with mysql -uuser -hremoteserver -p and its work |
| 12:00.52 | matrix1233 | any one cat tel me what is the problem ? |
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| 13:10.41 | matrix1233 | alo |
| 13:10.55 | Zogot | alloallo |
| 13:15.47 | matrix1233 | hello Zogot |
| 13:16.09 | matrix1233 | matrix1233i have asterisk 11.07 and i wanna use a remote database with res_config_mysql but i have alway ERROR[18205]: res_config_mysql.c:1577 mysql_reconnect: MySQL RealTime: Failed to connect database server |
| 13:16.09 | matrix1233 | 13:07 <#asterisk> matrix1233with my local databaseserver its work but with the remote i can't connect |
| 13:16.09 | matrix1233 | 13:08 <#asterisk> matrix1233i have tested also with mysql -uuser -hremoteserver -p and its work |
| 13:18.02 | Zogot | matrix1233: the remote database is configured to allow remote connections? |
| 13:18.21 | Zogot | by default that isn't the case, which is why you could connect to your localhost ( the default for mysql is only allow connections from local ) |
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| 13:21.11 | mfu | Chainsaw: so boring with tls... |
| 13:21.34 | Chainsaw | mfu: Your wireshark-powered listening station fell silent? |
| 13:22.42 | matrix1233 | it configured to bind 0.0.0.0 and also i tested mysql -uuser -hremoteserver -p and its work |
| 13:26.03 | matrix1233 | Zogot: have some idea for the problem |
| 13:26.08 | matrix1233 | ? |
| 13:26.34 | Zogot | matrix1233: did you google info about allowing remote connections for MySQL? |
| 13:27.23 | Zogot | matrix1233: http://www.cyberciti.biz/tips/how-do-i-enable-remote-access-to-mysql-database-server.html |
| 13:27.50 | Zogot | You will want your bind to be the IP of the server and you will need to grant rights for users to certain databases |
| 13:27.53 | matrix1233 | Zogot: yes i do all what i found on google ... and really i cant understand the problem . |
| 13:28.12 | matrix1233 | Zogot: its not a problem for database and i am sure that its ok |
| 13:28.41 | matrix1233 | Zogot: i bind 0.0.0.0 and i added the correct grant .. |
| 13:29.26 | Zogot | matrix1233: then try connecting to it from another tool or a quick bit of code, to ensure that it isn't a misconfiguration. |
| 13:29.50 | Zogot | I don't know enough about asterisk yet to know if that is the problem, but Failed to connect when you are accessing remotely and it works locally, says to me its this :p |
| 13:29.55 | matrix1233 | i tested to connect to remote database with the command mysql -h remote host and its work |
| 13:30.07 | Zogot | ah ok |
| 13:30.49 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 13:31.42 | matrix1233 | Zogot: when i set the config to localhost on asterisk its work and weh i set the config to remote database i have a problem |
| 13:32.16 | matrix1233 | Zogot: ERROR[18205]: res_config_mysql.c:1577 mysql_reconnect: MySQL RealTime: Failed to connect database server .... (err 1045). |
| 13:33.35 | matrix1233 | any one can tel me some idear ? |
| 13:34.59 | Zogot | matrix1233: try adding a dbsock config line |
| 13:35.06 | Zogot | to your mysql.sock |
| 13:35.19 | matrix1233 | Zogot: ok |
| 13:35.20 | Zogot | dbsock = /var/lib/mysql/mysql.sock |
| 13:36.12 | matrix1233 | its added |
| 13:36.16 | matrix1233 | Zogot: [Jan 20 14:35:57] ERROR[27193]: res_config_mysql.c:1577 mysql_reconnect: MySQL RealTime: Failed |
| 13:36.25 | matrix1233 | Zogot: i have the same problem |
| 13:36.47 | matrix1233 | Zogot: how can i trace or debug perhaps there is one think that i cant see |
| 13:37.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
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| 13:41.57 | Bearish | hmmm, i can't seem to get SQLite CDRs working, tables are created, ids are incrementing, but values are empty |
| 13:41.58 | Bearish | https://www.dropbox.com/s/qyx1h9lcshpzfmi/Screenshot%202014-01-20%2017.41.44.png |
| 13:43.20 | Bearish | it doesn't want to grap CDR variables from values=>'${CDR(start)}','${CDR(clid)}','test' |
| 13:43.28 | Bearish | any ideas why? |
| 13:45.12 | Bearish | And it's all fine in CSV database |
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| 13:49.51 | matrix1233 | Zogot: no idea ? :p |
| 13:51.18 | Zogot | matrix1233: sorry man, only been messing with asterisk for a few hours |
| 13:54.27 | mfu | Chainsaw: the only point i see is, the "Call-ID" Header includes a "@127.0.0.1".. on other trunks (without tls) there is the hostname of the peer... |
| 13:55.02 | mfu | Chainsaw: you know if the Call-ID Header does any critical impact on the tls problem? |
| 13:55.19 | mfu | exit |
| 13:55.26 | mfu | ups, fc :) |
| 13:57.15 | mfu | chan_sip.c:15099 sip_reg_timeout: -- Registration for 'xxx@proxy.dus.net' timed out, trying again (Attempt #3) |
| 13:57.20 | mfu | Really destroying SIP dialog '469ab4a232aff5470d3c2b82352c3faf@127.0.0.1' Method: REGISTER |
| 13:57.22 | mfu | and so on |
| 13:57.43 | [TK]D-Fender | mfu: pastebin the complete SIP debug |
| 14:05.34 | Ibrahim22 | Does someone have a good resource where i can learn about the structure of CDR's |
| 14:07.15 | *** join/#asterisk bitwize1 (~S2TS@h87-96-213-2.dynamic.se.alltele.net) |
| 14:08.25 | [TK]D-Fender | ~book |
| 14:08.25 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:08.27 | [TK]D-Fender | ^^^^ |
| 14:09.33 | mfu | [TK]D-Fender: http://pastebin.ca/2565169 |
| 14:09.42 | mfu | there is no nat |
| 14:10.30 | [TK]D-Fender | mfu: disable core debug, leave verbose 10, and sip debug only |
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| 14:29.18 | mfu | [TK]D-Fender: http://dpaste.com/1561191/ better? :) |
| 14:31.59 | [TK]D-Fender | mfu: I don't see the 2nd reg attempt with that timeout related to it. |
| 14:32.23 | [TK]D-Fender | [Jan 20 15:28:16] WARNING[8895] chan_sip.c: Probably a DNS error for registration to 000387250902@proxy.dus.net, trying REGISTER again (after 20 seconds) |
| 14:32.32 | [TK]D-Fender | I'm also wondering on the full string used for this |
| 14:34.15 | mfu | [TK]D-Fender: http://dpaste.com/1561202/ maybe now? |
| 14:34.57 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-wkyiqhjkshydcybi) |
| 14:34.58 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:35.51 | [TK]D-Fender | Still only see a single registration attempt in there and t's getting ack'd 200 OK |
| 14:36.29 | mfu | hmm |
| 14:36.35 | *** join/#asterisk pietro (~pietro@78-134-22-165.v4.ngi.it) |
| 14:36.37 | pietro | hello |
| 14:36.52 | mfu | is there a way to set sip debug to on default in conf? |
| 14:37.36 | mfu | got it.. mom |
| 14:37.44 | pietro | someone of you knows a SIP phone that supports ICE ? |
| 14:37.54 | pietro | (an hard one, no softphone) |
| 14:38.55 | WIMPy | The Snoms have settings for ICE. |
| 14:40.00 | mfu | [TK]D-Fender: the log is to big.. dont get only sip for debug on |
| 14:41.27 | pietro | WIMPy: I know, but stands SIP firmware doesn't support ICE at the moment, (is enabled for UC/Lync only) |
| 14:42.35 | pietro | s/stands/standard/ |
| 14:42.41 | WIMPy | Huh? I have the option to turn it on. |
| 14:42.47 | *** join/#asterisk vlad_starkov (~vlad_star@93.191.18.82) |
| 14:43.33 | pietro | WIMPy: yes, but the option doesn't affects nothing. |
| 14:43.53 | WIMPy | Strange |
| 14:44.48 | WIMPy | The wiki only has a note about reliability. |
| 14:46.37 | *** part/#asterisk bitwize1 (~S2TS@h87-96-213-2.dynamic.se.alltele.net) |
| 14:48.03 | mfu | [TK]D-Fender: you git right |
| 14:48.06 | mfu | *got |
| 14:48.45 | [TK]D-Fender | ? |
| 14:49.03 | mfu | there is no attempt to register the peer |
| 14:49.07 | mfu | but why... |
| 14:49.34 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:49.35 | mfu | he established on restart perfect.. bit after the first time.. there comes a timeout |
| 14:50.51 | mfu | there comes only a "CSeq: 108 REGISTER" |
| 14:50.53 | mfu | without hsotname |
| 14:51.00 | mfu | *hostname |
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| 15:04.35 | Bearish | http://serverfault.com/questions/568524/empty-asterisk-12-cdrs-in-sqlite3-database Any ideas guys? I really don't want to use MySQL |
| 15:05.21 | *** join/#asterisk MauriceM_ (~MauriceM_@66-193-40-64.static.twtelecom.net) |
| 15:05.37 | file | CDRs were completely redone in 12, it is possible that the way the module is grabbing the values got broken |
| 15:06.33 | file | mjordan, ^^^ |
| 15:07.37 | mjordan | sighs |
| 15:08.14 | file | I knew you were going to do that. |
| 15:08.25 | [TK]D-Fender | Bearish: that looks like a useless site to post that to... you should use the asterisk mailing lists |
| 15:08.42 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 15:08.47 | mjordan | and go ahead and file a bug report on issues.asterisk.org |
| 15:08.53 | [TK]D-Fender | Bearish: of if you've got a good basis for it being an actual bug, then to the tracker |
| 15:09.41 | mjordan | both clid and start are canonical CDR values, so those should have snagged something. |
| 15:09.46 | mjordan | ergo: bug. |
| 15:10.02 | file | oh, wait a second |
| 15:10.29 | file | mjordan, what if batching is enabled? |
| 15:10.33 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 15:10.46 | mjordan | nah, that still works. |
| 15:10.51 | mjordan | (it should) |
| 15:11.05 | mjordan | batching just changes when it get dispatches to the backend. The object passed to the backend is the same. |
| 15:11.44 | mjordan | unfortunately, the problem is res_sqlite3_custom decided to be weird and interpret its custom values differently than the other custom backends |
| 15:11.51 | file | of course |
| 15:12.39 | mjordan | I'm going to end up back in CDR hell this week anyway, may as well pull the rip cord. |
| 15:12.57 | Bearish | Alright, I'll bugreport it, meanwhile I guess I'll have to use mysql :( |
| 15:13.01 | Bearish | thanks guys |
| 15:13.08 | *** part/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:13.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:13.59 | mjordan | Bearish: just an FYI, in general, not many of us pay attention to ServerFault or StackOverflow for Asterisk questions |
| 15:14.55 | *** join/#asterisk serafie (~erin@nat/digium/x-iqqojfnmgfyjnnjr) |
| 15:16.40 | cusco | hey folks |
| 15:17.19 | cusco | say,, I connected a new PRI line to the digium card... can I make asterisk read that new span (just changed in chan_dahdi.conf) without dropping ongoing calls via dahdi? |
| 15:18.28 | WIMPy | You could try module reload chan_dahdi. |
| 15:18.48 | cusco | well that won't work because its in use |
| 15:18.56 | [TK]D-Fender | cusco: No. |
| 15:19.02 | cusco | ok... |
| 15:19.18 | WIMPy | That makes a clear not then, I guess. |
| 15:19.37 | WIMPy | or no |
| 15:19.43 | cusco | not then if xor else and |
| 15:20.14 | Bearish | mjordan: https://issues.asterisk.org/jira/browse/ASTERISK-23162 |
| 15:21.11 | WIMPy | But there was some hotplugging support, IIRC. Maybe it only works if the final stage was preconfigured. |
| 15:21.33 | mjordan | Bearish: thanks |
| 15:21.43 | cusco | what does that mean? WIMPy ? final stage? |
| 15:21.58 | cusco | ow, if the new span was already on chan_dahdi.conf ? |
| 15:22.01 | cusco | right.. |
| 15:22.16 | WIMPy | yes |
| 15:22.45 | WIMPy | nOT TOO USEFULL, i GUESS. |
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| 15:22.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:22.50 | WIMPy | oops |
| 15:23.09 | cusco | ok thanks |
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| 16:37.37 | mjordan | Bearish: feedback for you on ASTERISK-23162. I think it might be your config - I ran with the latest out of the SVN 12 branch and got expected values. |
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| 16:42.24 | kolko | Hello. Can i ask here about exctension. I use "Set(SIP_Authorization=${SIP_HEADER(Authorization)})". This param is ok when user in sip.conf, but when i use MySQL - this param is empty. I doesn't found where this param is set. |
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| 16:48.22 | [TK]D-Fender | kolko: Headers are part of a call, not part of a peer. This should have no impact. Show us your exact code and complete call output. |
| 16:51.07 | newtonr | kolko, Are your peers configured identically between the sip.conf and database configurations? Perhaps the call where your database peer is queried doesn't do authorization. As [TK]D-Fender said, you'll want to post a call log including the SIP trace and at least verbose output. |
| 16:51.41 | newtonr | kolko, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:52.14 | kolko | 5 minutes, please |
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| 16:55.11 | teddy_salad | list |
| 16:56.33 | teddy_salad | sry |
| 16:56.43 | kolko | http://pastebin.com/cze9QgVc it's enough? |
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| 16:58.12 | kristoffelloos | Hi, has anyone excperiances with voip traffic over powerline with asterisk, if so are these good or a absolute don't!? |
| 16:58.55 | [TK]D-Fender | kolko: Where the SIP DEBUG requested? |
| 17:00.08 | [TK]D-Fender | kristoffelloos: The medium is unimportant. The jitter/packet-loss, speed, bandwidth, etc is. |
| 17:05.44 | kolko | http://pastebin.com/nkpjFsqc this? |
| 17:07.08 | kolko | Authorization section in 23 line. |
| 17:08.05 | [TK]D-Fender | kolko: externalauth=yes <- sip.conf. In MySQL I see it as NULL |
| 17:08.35 | [TK]D-Fender | kolko: Already not the same |
| 17:09.28 | [TK]D-Fender | kolko: Your mailbox format is also not the same (missing the context). You do not seem to have done a good job making these comparable entries |
| 17:10.42 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 17:11.32 | kolko | externalauth - this i alter to table for as experimental, when not helped - set this to null |
| 17:13.24 | kolko | update table - did not help |
| 17:14.30 | iggi | Hello again, I'm having an odd problem with a Follow Me extension. When I dial it, it calls out to a cell phone (as it is supposed to), but if the user does not answer, it does not send it to the voicemail even though that is the setting. It just hangs up on the caller. |
| 17:15.55 | kolko | You think, what user from mysql does not make Authorization and exctension run without Authorization context? |
| 17:18.14 | [TK]D-Fender | kolko: Also depends if your realtime entries are CACHED or not... |
| 17:18.19 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:18.47 | [TK]D-Fender | iggi: Show us |
| 17:18.50 | [TK]D-Fender | ~pb |
| 17:18.50 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:18.52 | [TK]D-Fender | ^^^^ |
| 17:18.54 | kolko | After set fase username in database: http://pastebin.com/gjvbXj7u |
| 17:19.12 | iggi | [TK]D-Fender, What do you need to see? Config? |
| 17:19.20 | iggi | or log |
| 17:19.30 | [TK]D-Fender | iggi: both |
| 17:19.31 | kristoffelloos | Are we at the point today that we can say you can use Asterisk in a VMware ESX or are we going to have issue in production for timing?? |
| 17:20.13 | [TK]D-Fender | kristoffelloos: I hear ESXi tends to have the best results for virtualization |
| 17:20.53 | kristoffelloos | You heard that, but can someone in this chatroom confirm me this in real practice? |
| 17:21.18 | kolko | About cached: only rtcachefriends='yes' in sip.conf |
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| 17:25.30 | [TK]D-Fender | kolko: Which means it's not likely to poll your DB for those "updates" you just did |
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| 17:27.28 | kolko | as it may relate to the absence of SIP_HEADRE? |
| 17:27.37 | kolko | sip_header* |
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| 17:36.18 | kolko | on auth from sip.conf in sip log doesn't exist lines from 35 to 70 in http://pastebin.com/nkpjFsqc maybe this part erase Authorization? |
| 17:41.28 | [TK]D-Fender | I'm not seeing consistent comparable call debug from both scenarios here especially where the database entry at least properly mirrors the text-based config |
| 17:42.36 | iggi | [TK]D-Fender, Log: http://pastebin.com/Nw7uYGKh |
| 17:42.39 | iggi | working on config. |
| 17:43.45 | [TK]D-Fender | iggi: no need |
| 17:43.50 | iggi | ok |
| 17:44.09 | [TK]D-Fender | iggi: that is FREEPBX Followme. This is a call processing issue, not an Asterisk issue and needs to be taken up in #freepbx |
| 17:44.21 | iggi | Ok, thanks |
| 17:49.12 | kolko | http://pastebin.com/0rkZGAta - this is correct auth from sip.conf. i dont know sip protocol to full undeastand all debug. In Mysql version Authorisation header is set in SUBSCRIBE, here he set in INVITE |
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| 18:13.26 | kolko | probably, username is set in Auth header in check_auth_result check_user_full. in chan_sip.c comment above function: "This is used on first invite (not re-invites) and subscribe requests" - maybe this is answer? with mysql in subscribe - header is ok. in text conf CSeq: 1 INVITE, but in mysql: CSeq: 2 INVITE |
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| 20:08.50 | boom^time | I'm trying to determine if my ITSP is correct in doing this. Is a SIP 503 Response the correct response for a disconnected number? |
| 20:09.38 | boom^time | Because asterisk is marking it as congestion, which is not the case. Trying to get it to be marked as a bad number. |
| 20:11.01 | pabelanger | boom^time, 503 is usually used for a temp outage, so the number is good but there server is down |
| 20:11.31 | pabelanger | 488 Not Acceptable Here would likely be a better response for a disconnected number |
| 20:11.57 | pabelanger | either way, if they are using 503, you can just add dialplan and deal with it |
| 20:12.29 | boom^time | Unfortunately not, I'm using AMI and handling Reason codes on failures |
| 20:12.47 | boom^time | both 480 and 503 return a failure and reason code of 8 |
| 20:13.04 | pabelanger | for what, dial? |
| 20:13.17 | boom^time | So I can't distinguish between a true congestion and a disconnected number |
| 20:13.17 | boom^time | Yes |
| 20:13.18 | pabelanger | Originate a local channel and add the logic there |
| 20:14.18 | boom^time | That creates all sorts of complications for me. Since the code that follows the calls doesn't piece together all of the zombie channels created and combined in that case |
| 20:14.36 | boom^time | no to mention the CDR processor |
| 20:15.17 | pabelanger | so, what are you expecting back for a disconnected number? |
| 20:15.38 | boom^time | a reason code of 1 |
| 20:15.49 | pabelanger | what is returning that? |
| 20:16.10 | boom^time | no such extension or number |
| 20:16.15 | pabelanger | OriginateResponse? |
| 20:16.17 | boom^time | Yes |
| 20:16.24 | pabelanger | Then you need to hack asterisk then |
| 20:16.33 | boom^time | That's what I'm currently doing |
| 20:16.37 | boom^time | *attempting* |
| 20:16.43 | pabelanger | or have your ITSP return what asterisk expects for code 1 |
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| 20:17.05 | pabelanger | well, that seems like the long way around of doing things vs updating your application code |
| 20:17.13 | boom^time | Yeah, the chain of events between SIP responses and the AMI Reason codes is a complicated one |
| 20:17.47 | boom^time | Maybe not, changing one constant vs rewriting automated calling/cdr processing daemons |
| 20:18.44 | boom^time | Plus I've already had to hack up the version of asterisk I'm using so the desire to keep it pure is already gone. |
| 20:19.32 | pabelanger | well, you are talking about modifying ast_pbx_outgoing_exten() or ast_pbx_outgoing_app() to give the specific reason you need. Which is basically main/pbx.c, not a trivial task since a lot of code depends on it. |
| 20:20.07 | pabelanger | But if you originate a local channel, and just modify the cause code, you'd likely save your self a ton of work |
| 20:28.57 | mjordan | boom^time: are you using Asterisk 11? |
| 20:29.02 | boom^time | 11.5 |
| 20:29.18 | mjordan | you should be able to get the technology specific hangup cause of the channel using HANGUPCAUSE |
| 20:29.30 | mjordan | through a hangup handler |
| 20:30.37 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_HANGUPCAUSE |
| 20:30.44 | boom^time | Hmm, I'm going to look into that. |
| 20:30.57 | mjordan | once you have the hangup cause of the outbound channel, you can make whatever decision you want on how you want to hangup the inbound channel |
| 20:31.31 | mjordan | this page has some examples of using it: |
| 20:31.31 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause |
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| 21:15.08 | rrittgarn | so sometime last week i posed a question and then got distracted and didn't make it back to see the answers. I've found an issue where if i have a call set up between two callers (A -> B) and i use the AMI to redirect channel A to "anywhere" else, say caller C, I don't get a CDR entry. I get all the CEL entries in my DB but no cdr entry. I've seen other people reference bugs similar. The question is: What can I do t |
| 21:19.21 | rrittgarn | i found an unresolved bug from 1.8 from 2011 on Jira for it, but not sure if that's just because its now an old version or what |
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| 21:22.02 | roramirez | hello |
| 21:22.16 | roramirez | i little problem with cel and pgsql |
| 21:23.28 | roramirez | i configure the cel.conf accord to cel_pgsql.conf, i do reload but when ran "cel show status" return "CEL Logging: Enabled |
| 21:23.28 | roramirez | CEL Event Subscriber: CEL Custom CSV Logging |
| 21:23.29 | roramirez | " |
| 21:25.10 | roramirez | anybody can give a hand? |
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| 21:26.11 | deltaray | When someone calls me using caller id blocking (*67) the call goes straight to voicemail and seems to skip the Dial line. |
| 21:26.32 | deltaray | Is this something that Asterisk is doing or could it be that my SIP phone (Linksys SPA941) is blocking it? |
| 21:26.43 | deltaray | the caller id shows up as "Anonymous" |
| 21:27.52 | [TK]D-Fender | checkout time, BBAIB |
| 21:28.08 | navaismo | roramirez, is the cel_pgsql module loaded? |
| 21:28.28 | navaismo | deltaray, show us the complete cli output when that happens |
| 21:30.38 | deltaray | cli output? |
| 21:30.48 | deltaray | you mean the log? |
| 21:31.46 | navaismo | weel both works if are complete |
| 21:32.33 | navaismo | cli-->command line interface, here talking about cli is for the output when you connect to asterisk via asterisk -r |
| 21:32.38 | navaismo | ~cli |
| 21:32.38 | infobot | cli is probably a Command Line Interface, the best form of interface around, of course Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction Common Language Infrastructure (See mono or .net) |
| 21:32.49 | navaismo | ¬¬ |
| 21:32.50 | roramirez | let me check navaismo |
| 21:32.57 | navaismo | ah you rock bot |
| 21:33.17 | deltaray | navaismo: thanks. I see now a line in my log "Got SIP response 406 "Not Acceptable" back from 192.168.1.144:5060" |
| 21:33.22 | deltaray | That's one of my SIP phones. |
| 21:33.33 | deltaray | Interesting. |
| 21:34.24 | navaismo | well that is not necessary about your issue, that is a sip client telling that it cant handle the request i.e. the codec is not supported |
| 21:34.58 | roramirez | navaismo: yep, is loaded |
| 21:35.05 | deltaray | Its only happening on Anonymous calls though from the outside. |
| 21:35.32 | deltaray | Like if I dial *67 and the number of my asterisk server on my phone, it goes straight to voicemail. |
| 21:35.38 | deltaray | I mean from my cell phone |
| 21:36.09 | deltaray | Nevermind, I see other people talking about this same issue with Linksys phones, there is an option to block anonymous calls. |
| 21:36.27 | navaismo | ok show us the complete sip debug when it happens |
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| 21:36.53 | roramirez | where i say to asterisk used backend postgresql for cel? |
| 21:38.34 | roramirez | i am restart asterisk and worked |
| 21:38.43 | roramirez | is very stranger |
| 21:39.26 | deltaray | Its working now after changing the "Block ANC Setting:" on the phone to no. |
| 21:43.28 | roramirez | thanks navaismo |
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| 21:43.43 | navaismo | np |
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| 22:00.15 | rrittgarn | I've found an issue where if i have a call set up between two callers (A -> B) and i use the AMI to redirect channel A to "anywhere" else, say caller C, I don't get a CDR entry. I get all the CEL entries in my DB but no cdr entry. I've seen other people reference bugs similar. The question is: What can I do to help fix this bug? I can reproduce it 100% of the time on version 11r3 and 11.6. Could a Dev point me in the |
| 22:02.11 | navaismo | while(!answer){ask} |
| 22:03.37 | navaismo | all we have passed trough that, maybe if you create a new bug in the jira page get more attention from the "versed" people |
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| 22:09.23 | [TK]D-Fender | rrittgarn: So NO CDR at all for any leg? |
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| 22:15.51 | rrittgarn | correct TK |
| 22:16.01 | rrittgarn | only CEL events |
| 22:16.53 | [TK]D-Fender | rrittgarn: Firstupgrade to current, then post a bug report with backup |
| 22:17.46 | WIMPy | The deprecation of macros was very very evil. |
| 22:18.51 | rrittgarn | current for 11 is 11.8.0-rc1 correct? |
| 22:19.05 | rrittgarn | or just 'asterisk-11-current' in the lovely list of files? |
| 22:19.11 | [TK]D-Fender | <PROTECTED> |
| 22:19.28 | [TK]D-Fender | Through RC might not be bad to test again |
| 22:20.12 | rrittgarn | tell me which one you'd like me to test on (i can do both) and i'll collect whatever logs i can to be helpful. |
| 22:20.47 | rrittgarn | i can't be the only person experiencing this bug... unless the AMI is really that under-utilized... |
| 22:22.01 | [TK]D-Fender | I'd use the RC |
| 22:22.12 | navaismo | Is this error an configuration error? chan_dahdi.c:2004 dahdi_r2_write_log: Chan 72 - Protocol error. Reason = Invalid Multi Frequency Tone, R2 State = Seize ACK Received, MF state = DNIS Digit Transmitted, MF Group = Forward Group I, CAS = 0x0C |
| 22:23.55 | navaismo | s/an/a/ |
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| 22:24.01 | navaismo | pfff |
| 22:28.35 | rrittgarn | TK: I'll build and test this version after hours tonight and report back tomorrow. say it fails, what can i log to show its failing? easy to show when its doing something it shouldn't but hard to show you its not doing something somewhere... |
| 22:29.41 | [TK]D-Fender | rrittgarn: have it output the uniqueID during the call, show a call before & after and the glaring gap in the middle I guess |
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| 22:30.12 | rrittgarn | kk. Thanks for the help. |
| 22:31.52 | newtonr | rrittgarn, According to the bug tracker, it is a known issue. You can post a me too, along with your debug on https://issues.asterisk.org/jira/browse/ASTERISK-18733 if you want. CEL was designed to solve some of the issues that can't be solved with CDR, so using CEL in the meantime is the workaround to the various cases where CDR has undefined behavior. |
| 22:35.00 | newtonr | Well, workaround and likely final solution, unless someone goes in to overhaul CDR, which is unlikely. If you can show that the bug is a regression and the behavior worked in a previous version it would be much more likely to get fixed. |
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| 22:42.23 | rrittgarn | yeah i saw that bug, but wasn't sure it was still 'active' or however you want to deem it since its all the way back to 1.8 and the last updated is june of 2012 |
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| 22:56.17 | ipengineer | in app_confbridge does a user profile have to be declared for every participating user or can we create a "default" user profile that multiple people can use when they dial in? |
| 22:57.08 | WIMPy | Have ypu taken a look at the sample config? |
| 22:57.34 | ipengineer | WIMPy: Yes, I looked online as well in the wiki |
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| 22:59.32 | WIMPy | The definitions in the config file are like templates. |
| 22:59.53 | WIMPy | You can use them for as many users as you like |
| 23:00.09 | WIMPy | And you can also change them per user via dialplan functions. |
| 23:01.37 | ipengineer | Ok. Thats what I was curious of. I just wasn't sure if the same user with the same PIN could join multiple times.. Now in the instance 2 users join in with the same pin/profile I guess start_muted applies to each caller independently. One user unmuting will not effect the other with the same pin. |
| 23:02.03 | WIMPy | Correct. |
| 23:02.12 | WIMPy | In theory. |
| 23:02.23 | ipengineer | Ok, that makes sense. Thanks |
| 23:02.34 | WIMPy | I have had the effect that it affected both users. But only once per call. |
| 23:02.50 | ipengineer | Ok |
| 23:02.55 | WIMPy | Nt sure what's going on there. |
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| 23:05.06 | ipengineer | My only other thought would be if confbridge supports Realtime now in v12.. I knew it used to not |
| 23:05.14 | navaismo | WIMPy, do you have an advice for this error: chan_dahdi.c:2004 dahdi_r2_write_log: Chan 72 - Protocol error. Reason = Invalid Multi Frequency Tone, R2 State = Seize ACK Received, MF state = DNIS Digit Transmitted, MF Group = Forward Group I, CAS = 0x0C |
| 23:07.03 | WIMPy | Nope. No idea about smoke signals :-) |
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| 23:12.39 | navaismo | me too haha |
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