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| 00:22.38 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 02:34.07 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 07:02.07 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 09:17.20 | Guest9301 | Hi, is there a way not to playback DTMF tones to a incoming call? When the agent recieves a call and uses the dialpad for various reason, the loud and inhospitable dtmf tones is poisoning the customer. |
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| 09:28.06 | Guest9301 | Maybe I can use inband on the agent sip configuration. |
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| 09:45.30 | armin | hi, i'm trying to send a FAX via t.38 between 2 asterisk instances. however, i get this behaviour here from one of the asterisks in place: https://paste.fedoraproject.org/paste/TkzlHLmgYgqOYXRWHv052g/raw - anyone of you got a hint what i could look into further? |
| 09:46.21 | Guest9301 | armin: the A part does not ACK the 200 OK, thus the re-transmissions |
| 09:47.40 | armin | Guest9301: ah, right, thanks. |
| 09:48.02 | armin | Anything further I could investigate? Sorry, VoIP noob here. |
| 09:52.47 | Guest9301 | You should post the full sip transaction |
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| 09:53.00 | Guest9301 | the content of each package |
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| 10:02.40 | armin | Guest9301: http://base.m2m.pm/bla3.pcap |
| 10:02.51 | armin | Guest9301: i had to strip anything other than SIP for obvious reasons. |
| 10:02.59 | armin | Guest9301: but the call flow remains. |
| 10:03.26 | armin | Guest9301: also since you're probably a helpful person: have you considered a permanent nickname and idling in here? :) |
| 10:05.57 | Guest9301 | Where is this trace taken from ? A or B? |
| 10:07.08 | Guest9301 | I'd prefer output from the asterisk CLI with verbosity an sip debug activated. |
| 10:08.27 | Guest9301 | asterisk -rvvvv and sip set debug on |
| 10:09.40 | Guest9301 | Pref on the originator, meaning A. |
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| 11:32.19 | armin | Guest9301: that's A, the originating side of the communication. |
| 11:44.18 | armin | Guest9301: https://paste.fedoraproject.org/paste/XmlKubnfTD0V9zyLYC56tA |
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| 11:49.01 | Guest9301 | ah ok, you have PJSIP. |
| 11:50.29 | Guest9301 | I cannot say why the 200 OK is not ACKed. I think someone else must consult you in this, since my experience on PJSIP is vague at most. |
| 11:57.54 | armin | Guest9301: still, many thanks for your kind help! |
| 11:58.02 | armin | Guest9301: did you consider a permanent nick? |
| 11:59.22 | Guest9301 | However I cannot see that T38 is negotiated in the SDP blocks |
| 12:01.47 | Samot | Does the other side allow/support T.38? |
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| 13:07.26 | armin | Samot: good question. |
| 13:07.45 | Samot | Also a rather important one. |
| 13:07.52 | Samot | T.38 needs to be on both sides. |
| 13:07.58 | armin | Samot: ty!!! |
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| 13:23.45 | Guest9301 | Samot: so the pcap would indicate that T38 is not enabled on either side. Since the originator does not present T38 in the SDP? |
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| 13:27.20 | Samot | The originator never starts T.38 |
| 13:27.39 | Guest9301 | Huh? INVITE -> (Hi I want to send a FAX) |
| 13:27.42 | Samot | The other side has to say "Yup, I can do T.38" and a RE-Invite is done to update the SDP for T.38 |
| 13:27.48 | Guest9301 | Ah ok |
| 13:27.58 | Samot | T.38 is a codec encapsulation. |
| 13:28.26 | Samot | It's done during the offer/accept SDP negotiation. |
| 13:28.28 | Guest9301 | Fax shouldn't exist in 2018 |
| 13:28.42 | Samot | Fax isn't going anywhere any time soon. |
| 13:28.48 | Samot | People just need to accept that. |
| 13:28.50 | Samot | Move on. |
| 13:29.00 | Guest9301 | You don't say. I've been waiting 15 years for the FAX to die. And I thought it was close 15 years ago |
| 13:29.31 | Samot | Not until the Medical and other industries like that agree on a more suitable solution. |
| 13:29.39 | Samot | This is why T1's are not dying anytime soon. |
| 13:30.19 | Guest9301 | We already started terminating ISDN on a national level. Operators are sending our termination notices with SIP replacements. |
| 13:30.26 | Guest9301 | Been doing so for the last two years |
| 13:30.55 | Samot | SIP is not considered "Mission Cricitial" |
| 13:31.03 | Samot | SIP is not considered "Mission Critical" |
| 13:31.07 | Samot | T1's are. |
| 13:31.16 | Samot | T's have MTTRs |
| 13:31.23 | Samot | SLAs |
| 13:31.35 | Samot | SIP does not. |
| 13:32.27 | Samot | As your phone provider (SIP) I cannot guarantee service SLA's or MTTRs because I can't control when your Cable Internet goes down or comes back up. |
| 13:32.31 | Samot | Not my stuff. |
| 13:32.56 | Samot | As your phone provider (PRI/T1) I can because that is my stuff. |
| 13:35.12 | Samot | I have a doctors office that uses SIP/T.38 for their faxing but they can't use my digital fax platform. |
| 13:35.38 | Samot | Because their faxes cannot be stored on a third party server that doesn't meet HIPAA compliance. |
| 13:35.55 | Samot | So I have to have an ATA sit there connected to their fax system. |
| 13:37.22 | Samot | Gmail/GoDaddy and most email services are not HIPAA compliant. And emailing a medical record that sits on those servers is bad. |
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| 13:59.04 | sekil | hates faxes |
| 13:59.26 | armin | too |
| 14:00.43 | Samot | I hate PRIs and FXS based PBX systems. But I still gots to deal with them. It's the job. |
| 14:02.56 | sekil | PRIs I can handle.. |
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| 14:21.19 | SunnyB | Hi. I have an old PBX connected to * via PRI (chan_dahdi). When an extension on the PBX calls an outside number via *, the PBX it will hang up if the number does not answer in 30seconds. Is there a way for * to answer the call and play some music towards the PBX until the remote end answers instead ? |
| 14:25.29 | [TK]D-Fender | Answered in #freepbx |
| 14:26.20 | igcewieling | For some reason I thought Asterisk 13 fixed the issue with tel: URLs generate this error "sip/reqresp_parser.c:931 get_name_and_number: can not parse name and number from sip header.". Must be a later version of Asterisk |
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| 14:55.58 | mahafyi | hello. i would like to inquire about reliable VoIP service providers in USA, who have APIs for provisioning also and support re-sellers (i.e small VoIP services). |
| 14:59.04 | mahafyi | anyone familiar with https://www.inteliquent.com ? |
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| 17:46.18 | starfall | I've been reading about the asterisk api and I was wondering if there was a way to create extensions through it. |
| 17:46.59 | starfall | Or is it more like I need to directly connect to the DB and create them myself? |
| 17:49.38 | [TK]D-Fender | Depends what you mean by "extensions" |
| 17:52.52 | starfall | [TK]D-Fender: I mean like sip users who can register and place calls to others |
| 17:54.03 | [TK]D-Fender | extensions = extensions.conf = diaplan |
| 17:54.17 | [TK]D-Fender | For SIP Peers you could DB thos. |
| 17:54.29 | [TK]D-Fender | And there is no "API" |
| 17:54.37 | starfall | I'm working on a app which, on new account creation, create an associated sip account on a Asterisk PBX server which would be used to make VoIP calls to others on that PBX. |
| 17:55.17 | starfall | So I'm trying to figure out what would be the best way to do that |
| 17:56.01 | starfall | Some forum posts suggest that astrisk doesn't have an api call which could accomplish this, so I was curious if that is still the case. |
| 17:56.14 | starfall | excuse me, not api |
| 17:56.52 | starfall | ARI |
| 17:58.40 | [TK]D-Fender | ARI isn't a tool for this |
| 17:58.42 | [TK]D-Fender | DB is DB |
| 17:58.47 | [TK]D-Fender | just shove records in there |
| 18:01.07 | starfall | Figured as much |
| 18:03.43 | starfall | I should prob just find how asterisk creates those accounts on the db and more or less do it that way, right? |
| 18:07.48 | [TK]D-Fender | * never creates them |
| 18:07.56 | [TK]D-Fender | You always do |
| 18:08.05 | [TK]D-Fender | ~book |
| 18:08.05 | 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 |
| 18:08.07 | [TK]D-Fender | ^^^ |
| 18:11.25 | starfall | I've been testing using freepbx, so I guess I kind of assumed that when I created an account using the GUI it was * handling putting that data into the DB |
| 18:17.50 | [TK]D-Fender | This channel does not support FreePBX. |
| 18:18.03 | [TK]D-Fender | ~#freepbx |
| 18:18.09 | [TK]D-Fender | ~freepbx |
| 18:18.09 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:18.30 | [TK]D-Fender | You'll need to ask. So every answer I have just given you ... does not apply at all. |
| 18:18.37 | starfall | I know, just explaining why what I said may have not made much sense |
| 18:19.28 | starfall | and the help has been very appreciated regardless! |
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| 20:43.05 | L|NUX | hello everyone |
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