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| 01:14.36 | forgotmynick | hey. i have an issue where some of those working from home are getting disconnected from the internet (on their end, nothing to do with asterisk) but sometimes it takes them 5-10 minutes to get connected again by which time the call has already dropped. is there a way we can move the caller back into a queue if the agent has lost connection? once the lockdown is over we'll turn it off but we're getting a lot of |
| 01:14.36 | forgotmynick | complaints and the IT manager is saying it can't be done but he doesn't actually have any knowledge on asterisk because we have been outsourcing for over a decade until February this year (contract dropped because of cost cutting) |
| 01:15.23 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
| 01:18.51 | Samot | forgotmynick: You can probably use the g Dial() option. |
| 01:19.09 | Samot | It will send the caller to the next priority in the dialplan when the called channel hangs up |
| 01:20.02 | Samot | But thinking about it, it will depend on the DIALSTATUS. |
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| 09:13.41 | IamTrying | https://paste.ubuntu.com/p/wrvxjGCCD6/ - I am getting this error when i submit the pincode to join the meeting. |
| 09:32.05 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
| 09:35.31 | IamTrying | Does the ring bell to anyone ? |
| 09:37.07 | pchero_work | Iamnacho: you didn't add the sip address to the Dial() |
| 09:38.06 | pchero_work | Or you don't have the pjsip endpoint jigasi. |
| 09:38.17 | *** join/#asterisk derPlexus (~plexus@ip-176-198-128-59.hsi05.unitymediagroup.de) |
| 09:38.30 | pchero_work | I think you don't have the endpoint, check your endpoints using `pjsip show endpoints` |
| 09:38.40 | pchero_work | or `pjsip show endpoint jigasi` |
| 09:38.51 | *** join/#asterisk DarkPonay (~Darkponay@laubervilliers-658-1-211-207.w90-63.abo.wanadoo.fr) |
| 09:39.12 | pchero_work | Wait, you are using the chan_sip and chan_pjsip at the same time. |
| 09:39.23 | *** join/#asterisk spatel (~spatel@pool-96-237-230-175.bstnma.fios.verizon.net) |
| 09:39.54 | pchero_work | That causes this problem. Use 1 channel driver for the SIP.\ |
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| 11:03.59 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
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| 11:16.15 | IamTrying | pchero: this do not show anything (pjsip show endpoint jigasi). how do i verify for sure am i using chan_sip or chan_pjsip , this is like rocket science confusing. |
| 11:16.27 | IamTrying | pchero_work: |
| 11:17.20 | IamTrying | https://paste.ubuntu.com/p/7FDNGBhCTn/ - pchero_work |
| 11:28.12 | pchero_work | IamTrying: Could you please share the result of `module show like chan_`? |
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| 11:29.44 | *** mode/#asterisk [+o bford] by ChanServ |
| 11:30.26 | IamTrying | https://paste.ubuntu.com/p/Ms6JZnbYHC/ - pchero_work line 37 |
| 11:30.50 | pchero_work | Yes you have chan_sip module enabled. |
| 11:30.52 | *** join/#asterisk samwierema (~samwierem@195.240.143.134) |
| 11:31.35 | pchero_work | Add the `noload => chan_sip.so` to the /etc/asterisk/modules.conf |
| 11:31.51 | *** join/#asterisk irrgit (~ch33se@192.241.175.183) |
| 11:32.04 | pchero_work | Then restart the Asterisk. |
| 11:32.34 | pchero_work | And check the loaded module again to ensure the chan_sip has not loaded. |
| 11:36.13 | IamTrying | https://paste.ubuntu.com/p/7Jn3FC6RZV/ - pchero_work line 68 confirms now chan_sip disabled. |
| 11:37.16 | IamTrying | [ oldState=Unregistered; newState=RegistrationState=Registering; reasonCode=-1; reason=null] - problem SIP register is failing |
| 11:38.25 | IamTrying | https://paste.ubuntu.com/p/nwwq8J9KtS/ - line 84 |
| 11:38.32 | pchero_work | Yes, that shows you've disabled chan_sip. Now you need to migrate the chan_sip.conf to pjsip.conf |
| 11:38.34 | pchero_work | https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip |
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| 12:35.38 | IamTrying | https://paste.ubuntu.com/p/htxzSTdpWn/ - pchero_work line 87 added. But SIP Phone is unable to register saying "No matching endpoint found" |
| 12:38.12 | file | your configured name is "user2020" but you are using "jigasi" |
| 12:41.26 | IamTrying | file: no for the moment jigasi i am not using. i am using PortSip softphone to register as user2020 with password: 1234 |
| 12:42.41 | Samot | OK so then you need do to Dial(PJSIP/user2020) |
| 12:42.54 | IamTrying | https://paste.ubuntu.com/p/qSjxTVzwvV/ - line 116 file |
| 12:43.37 | file | "dtmfmode" is not a valid configuration option |
| 12:43.38 | IamTrying | Samot: line 116 i mean is failing where user2020 has to register as user1 and make a test call to PJSIP/jigasi |
| 12:43.41 | file | the option is dtmf_mode |
| 12:43.43 | IamTrying | ok removed |
| 12:43.47 | IamTrying | ok |
| 12:44.14 | Samot | Where is the endpoint for jigasi? |
| 12:45.22 | IamTrying | https://paste.ubuntu.com/p/fZrrg6HBtG/ - Samot. its in another debian 10 linux |
| 12:45.58 | file | rfc2833 is not a valid option for that either, rfc4733 is |
| 12:46.06 | file | when the configuration is loaded it tells you what is wrong |
| 12:46.55 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
| 12:49.04 | Samot | Wait, wait. |
| 12:49.21 | Samot | IAmtrying: You need an _endpoint_ on this system for jigasi |
| 12:49.32 | Samot | So that you can do Dial(PJSIP/jisagi) |
| 12:49.39 | Samot | Otherwise there's no endpoint to dial from. |
| 12:50.21 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 12:50.29 | IamTrying | https://paste.ubuntu.com/p/3rY3mD94gq/ - please see line 121 file (strange). Where i have to add that Samot? |
| 12:51.54 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
| 12:53.32 | file | it is output when Asterisk starts |
| 12:53.42 | file | if you are using something like systemd to start Asterisk, then it would have gone to the log file |
| 12:56.00 | *** join/#asterisk elguero (~miguel323@c-73-238-205-3.hsd1.nh.comcast.net) |
| 12:57.10 | IamTrying | for the moment DTMF deactivated so that Registration with chan_pjsip first works - file |
| 12:59.34 | file | Samot / igcewieling |
| 12:59.36 | file | gah |
| 12:59.41 | file | you two might like a change I just put up, https://gerrit.asterisk.org/c/asterisk/+/14419 |
| 13:00.00 | Samot | Don't like change. |
| 13:00.07 | Samot | Scary. |
| 13:00.39 | Samot | Oh, that is good though. |
| 13:00.39 | IamTrying | after remove of line 104 https://paste.ubuntu.com/p/3rY3mD94gq/ user2020 has registered success over chan_pjsip - file |
| 13:01.11 | IamTrying | Now from user2020 i have to dial in to jigasi. that is failing as there is no DTMF to enter pincode |
| 13:02.10 | Samot | You need to show things |
| 13:02.14 | Samot | Telling us is pointless. |
| 13:03.00 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
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| 13:45.26 | IamTrying | https://paste.ubuntu.com/p/WKnKfWw5Jc/ - i am getting the ERROR line 5. user2020 is registered success but jigasi still return: RegistrationStateChangeEvent[ oldState=Authentication Failed; newState=RegistrationState=Unregistered; reasonCode=0; reason=User has canceled the authentication process.] |
| 13:49.51 | *** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-qvcqtkadcxstykni) |
| 13:49.52 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 13:52.07 | IamTrying | 2020-05-19 13:41:21.902 SEVERE: [51] org.jitsi.jigasi.ServerSecurityAuthority.log() Wrong username or password for provider:ProtocolProviderServiceSipImpl - I get this when using chan_pjsip |
| 13:55.12 | Samot | K, I'm going to say it again.. |
| 13:55.22 | Samot | You nee an ENDPOINT ON THE SYSTEM |
| 13:55.23 | Samot | [May 19 13:41:55] ERROR[7666]: chan_pjsip.c:2469 request: Unable to create PJSIP channel - endpoint 'jigasi' was not found |
| 13:55.44 | Samot | It can't Dial() PJSIP/jigasi because there's not endpoint jigasi to Dial THROUGH |
| 13:57.52 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 14:11.56 | IamTrying | https://paste.ubuntu.com/p/RWBjBzttCT/ - Samot, does it not include end piont [jigasi] the way i have added already in pjsip.conf or it needs to be added somewhere else while using chan_pjsip only? (confusing) |
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| 14:17.46 | Samot | host=dynamic |
| 14:17.55 | Samot | That's not a valid setting. |
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| 14:42.56 | IamTrying | I have to try further its ok from Asterisk but something is not correct in my config. thank you guys |
| 14:42.58 | IamTrying | This hackers started to attack my server from India: 103.145.13.16 |
| 14:44.52 | Samot | Is this directly on the Internet? |
| 14:44.58 | IamTrying | I have to move to my CentOS because in Debian i cant block and they are like bombing my console. i will go to CentOS Debian is horrible, cant work |
| 14:45.02 | IamTrying | Yes Samot |
| 14:45.14 | IamTrying | in my Asterisk console, i have like millions of attack |
| 14:45.22 | Samot | Wait, wait? |
| 14:45.30 | Samot | Debian can't work? |
| 14:45.33 | IamTrying | problem is i am not Debian guy. CentOS |
| 14:45.35 | Samot | It's iptables basically. |
| 14:45.40 | Samot | It's Linux. |
| 14:45.50 | IamTrying | systemctl restart netfilter-persistent; |
| 14:46.37 | IamTrying | Its too much in Debian. i will go in CentOS. Its so strange, i just had a brand new Debian and Indian hackers like bombing me to death because i am new in Debian. wow!! |
| 14:47.27 | file | it takes minutes for a public SIP server to start getting traffic |
| 14:55.04 | *** join/#asterisk samwierema (~samwierem@195.240.143.134) |
| 15:03.44 | Samot | So turn off netfilter and use iptables |
| 15:03.53 | Samot | It's not really that hard. |
| 15:06.48 | Samot | Let's also be clear about something. The hackers do not care or know that you are new to Debian. In fact they won't know it's Debian until they are in it. |
| 15:07.05 | Samot | All they know is they found a server on the public Internet that is butt ass naked open. |
| 15:09.34 | IamTrying | https://paste.ubuntu.com/p/HKRsVtDKzb/ - why my /etc/sysconfig/iptables not working cat /etc/iptables/rules.v4. do we not have systemctl start netfilter-persistent? |
| 15:10.36 | *** join/#asterisk ChkDigit (~u388mw@207-195-34-73.prna.static.sasknet.sk.ca) |
| 15:13.48 | Samot | What's not working? |
| 15:14.21 | Samot | Nothing shows up with you do iptables --list |
| 15:20.40 | *** join/#asterisk miralin (~Thunderbi@195.209.246.194) |
| 15:23.42 | igcewieling | maybe he fell asleep |
| 15:23.54 | drmessano | Wow |
| 15:24.49 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
| 15:25.09 | drmessano | Everyone knows Debian doesn't have firewall because you must make your own, from these: |
| 15:25.17 | drmessano | https://usercontent.irccloud-cdn.com/file/U88CYsbg/IMG_2543.PNG |
| 15:26.01 | Samot | Man, I remember building Debian firewalls in Arts & Crafts. |
| 15:27.07 | drmessano | Everyone knows this is listed on any comparison between CentOS and Debian |
| 15:27.30 | drmessano | * Must make own fiarwell |
| 15:37.18 | drmessano | 11:06:49 <Samot> Let's also be clear about something. The hackers do not care or know that you are new to Debian. In fact they won't know it's Debian until they are in it. |
| 15:37.21 | drmessano | WRONG |
| 15:37.36 | drmessano | When you put a Debian system online with Asterisk on it and no firewall |
| 15:37.42 | drmessano | It ping 255.255.255.256 |
| 15:39.18 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
| 15:49.50 | *** join/#asterisk cresl1n (uid299068@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:49.50 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 16:21.08 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 16:21.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 16:21.16 | wonderworld | all grandstream phones are unable to get audio from 0800 |
| 16:21.55 | wonderworld | (and probably other diversions, havent tested yet) |
| 16:26.12 | igcewieling | make sure you have directmedia disabled |
| 16:27.02 | *** join/#asterisk wonderworld (~simon@unaffiliated/wonderworld) |
| 16:27.22 | wonderworld | igcewieling: thank you i will try |
| 16:27.56 | wonderworld | it was already set to no |
| 16:38.20 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
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| 16:48.51 | *** mode/#asterisk [+o rmudgett] by ChanServ |
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| 16:55.09 | sibiria | we run an external script off of Monitor() (using MONITOR_EXEC) in order to mix the two legs of the call together, and sometimes we see that asterisk apparently loses the channel during or after that script ends, and never reaches the 'h' extension |
| 16:56.43 | sibiria | there's no obvious pattern to when this happens but we've observed that some calls this manifests on have an odd log message from pbx_spool popping up: pbx_spool.c: Outgoing PJSIP/+123456789@TRUNKNAME: DelayedRetry |
| 16:58.07 | sibiria | when this happens it *appears* asterisk mismanages RTP audio somehow. we keep seeing RTCP reports on the channel but the audio from that point on is just silence |
| 16:59.44 | sibiria | these calls are initiated using call files. they use neither 'MaxRetries' nor 'RetryTime' in the call file |
| 16:59.57 | sibiria | this is on 16.5.1 |
| 17:00.03 | sibiria | any ideas on how to try nail down the problem? |
| 17:00.11 | igcewieling | Why are you not using MixMonitor? |
| 17:00.21 | sibiria | because we haven't migrated the platform over to it yet |
| 17:00.40 | igcewieling | look at adding or removing /n from the channel in the call file. |
| 17:01.32 | sibiria | you suggest we add a line break at the end of the call file? |
| 17:01.32 | igcewieling | Make sure you create the file outside of /var/spool/asterisk/outgoing, but on the same filesystem, then mv it to the correct location. |
| 17:01.39 | sibiria | yeah the call files aren't the problem |
| 17:01.44 | sibiria | they're fine and we manage them atomically |
| 17:02.12 | igcewieling | no that would be \n. see https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Optimization |
| 17:02.16 | sibiria | (i.e. we create them outside the spool dir and we move them into the spool dir) |
| 17:02.44 | igcewieling | If you are using spool files, I assume you are using Local/ channels. |
| 17:02.56 | sibiria | we're not |
| 17:03.39 | sibiria | i don't even know if the problem with "DelayedRetry" popping up in the middle of an ongoign call is related to the problem with the channel disappearing |
| 17:03.42 | igcewieling | If you are running Monitor on the local channel and not using /n then it might not work right. |
| 17:04.05 | sibiria | it works for 99.9% of the calls. we're seeing a few outliers |
| 17:06.40 | sibiria | if i understand "DelayedRetry" correctly, this happens when the call file's "RetryTime" timer elapses |
| 17:07.01 | sibiria | which should not happen with the default "MaxRetries" of 0 |
| 17:07.13 | sibiria | and especially not during an answered call |
| 17:25.11 | Zombie | I contacted a Company willing to connect my Asterisk instance to the outside world. |
| 17:25.31 | Zombie | They want $10/monthly and they are. |
| 17:28.36 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 17:29.31 | *** join/#asterisk Blashyrkh (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 17:29.52 | drmessano | Heh |
| 17:30.16 | Zombie | This is for my Residential Phone Service. |
| 17:31.00 | drmessano | You make it sound like you signed up into some big contractual agreement. Pick a provider, create an account, throw some cash into it. Done |
| 17:31.40 | drmessano | I have like 3 ITSPs for home. I pay less than $5 a month for the DIDs |
| 17:32.38 | Zombie | Am, I getting ripped off? |
| 17:33.23 | drmessano | You stated you contacted a company willing to connect your Asterisk to the outside world |
| 17:33.31 | Zombie | Yes. |
| 17:33.33 | Zombie | I did, |
| 17:33.55 | Zombie | The Company wants $10 monthly for the service. |
| 17:34.00 | *** join/#asterisk chandoo (~chandoo@pool-100-1-166-161.nwrknj.fios.verizon.net) |
| 17:34.16 | Zombie | Is that too much? |
| 17:34.19 | sibiria | for what service? |
| 17:34.24 | drmessano | You make it sound like they are coming out to plug you into something. Dude, you go to a provider, create an account, make a trunk, throw some money in, pick out what you want |
| 17:34.29 | sibiria | giving you a public phone number that can dial into your asterisk setup? |
| 17:34.39 | sibiria | or allowing you to terminate calls from it to phones on the public grid? |
| 17:34.41 | Zombie | Yes. |
| 17:34.46 | drmessano | This is like saying "I contacted a company willing to provide email for me" |
| 17:35.00 | sibiria | there are DID providers that offer numbers for like $1/month |
| 17:35.42 | Zombie | Well they said they offer 911 Servixce |
| 17:35.45 | Zombie | Well they said they offer 911 Service |
| 17:35.52 | drmessano | Residential SIP service is SELF-SERVE and mostly Ala Carte. You don't contact someone who is willing to do anything. |
| 17:35.54 | Zombie | Or 999 Service |
| 17:36.07 | Zombie | Oh, |
| 17:36.29 | Zombie | Its not too late to go with someone else if you think I'm getting into a financially bad deal. |
| 17:37.07 | drmessano | I donât know how I can make this analogy any more clear |
| 17:37.17 | Zombie | I get that! |
| 17:38.06 | drmessano | If you have an email account and you acquired it by contacting someone who is willing to supply email to your home, it's either 1985 or youre doing it wrong. |
| 17:38.22 | drmessano | So that alone tells me you did something weird |
| 17:38.55 | drmessano | Getting SIP to your Asterisk box is as simple as creating an email account |
| 17:40.17 | Zombie | The Same Company who provides me my Primary E-mail is offering this. |
| 17:40.34 | Zombie | They were my ISP at one point about 5 years ago |
| 17:40.48 | drmessano | Well, that says it all |
| 17:41.06 | sibiria | the service is surely good. it just sounds a bit expensive for what it does, assuming actual minute fees are applied as well |
| 17:41.35 | Zombie | They said their residential service is 500 minutes/monthly. |
| 17:41.45 | drmessano | It sounds like you're buying phone service from a full-stack provider not an ITSP |
| 17:42.09 | drmessano | SO you're going to pay accordingly. |
| 17:42.18 | Zombie | Full Stack? |
| 17:43.04 | Samot | Zombie You have a PBX |
| 17:43.14 | Zombie | Yes I do. |
| 17:43.21 | Samot | That means Residential Service is the _wrong_ service. |
| 17:43.26 | Zombie | They support Asterisk. |
| 17:43.28 | Samot | Because the expectations is an ATA device. |
| 17:43.30 | Samot | OK |
| 17:43.54 | Samot | Residential VoIP service is like your POTS line. |
| 17:44.05 | Samot | You'll have 3 calls, they'll provide you voicemail |
| 17:44.12 | Samot | They control your CallerID |
| 17:44.29 | Samot | As to what you can send, etc. |
| 17:45.07 | drmessano | On the other hand |
| 17:45.12 | drmessano | Maybe this is what he needs |
| 17:45.16 | Zombie | They told me they will be very flexible what they will let my Asterisk server do, the only caveat was international calling outside the US. |
| 17:45.27 | Zombie | Canada and US are Fine. |
| 17:45.39 | drmessano | Nice little residential voip account |
| 17:46.00 | drmessano | Tiny little box to make Asterisk go beep and boop |
| 17:46.08 | Samot | Let your Asterisk server do? |
| 17:46.11 | Samot | Like what? |
| 17:46.20 | Samot | Are they going to let you have more calls than other resi users? |
| 17:46.21 | Zombie | Do it's thing. |
| 17:46.29 | Samot | Sigh. |
| 17:46.37 | Samot | OK. |
| 17:46.51 | drmessano | Zombie: Do you understand the difference between running a mail server and someone supplying you a mailbox on a server? |
| 17:47.00 | Zombie | Yes I do. |
| 17:47.07 | Samot | That's what Resi service is. |
| 17:47.09 | drmessano | THIS here is someone giving you a mailbox |
| 17:47.11 | Samot | They are the PBX. |
| 17:47.16 | Samot | Essentially. |
| 17:47.24 | Samot | So you'll have a voicemail account. |
| 17:47.24 | drmessano | This is the SIP version of that |
| 17:47.35 | Samot | When it times out, it will go to their voicemail account. |
| 17:47.49 | drmessano | Which is why I said, if it's not SELF-SERVE, ALA Carte, you did it wrong |
| 17:47.55 | Zombie | They were not going to host anything for me. |
| 17:48.00 | Samot | Dude. |
| 17:48.03 | drmessano | If it's "We give you a number for $10 a month with voicemail", it's a "mailbox" |
| 17:48.04 | Samot | Remember your POTS line? |
| 17:48.08 | Zombie | Yes. |
| 17:48.15 | Samot | All the features it had? |
| 17:48.22 | Zombie | Yes. |
| 17:48.24 | Samot | Or didn't have. |
| 17:48.25 | Samot | OK |
| 17:48.26 | drmessano | Ok, I explained it 4 different ways.. Getting nowhere |
| 17:48.31 | Samot | Residential VoIP IS THAT |
| 17:48.34 | drmessano | Good luck |
| 17:48.40 | Samot | Just because you have a PBX doesn't matter. |
| 17:48.48 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:114b:7f59:bc7d:1581:4646:bba2) |
| 17:48.56 | Samot | The service provides ALL FEATURES. |
| 17:49.12 | Samot | You will have voicemail WITH them. |
| 17:49.16 | Samot | Because that is expected. |
| 17:49.33 | Samot | You will be limited to 3 calls. |
| 17:49.38 | Samot | Because that's what Resi service is. |
| 17:50.06 | drmessano | 3 is generous |
| 17:50.19 | Samot | Call Waiting |
| 17:50.22 | drmessano | Usually I see 2, which is one concurrent, and one channel so "Call Waiting" works |
| 17:50.31 | Samot | Sorry |
| 17:50.34 | Samot | 3 Way calling. |
| 17:50.45 | Samot | It's pretty standard feature. |
| 17:50.55 | drmessano | Yeah, on the $24.95 stuff |
| 17:51.04 | Samot | Heh |
| 17:51.14 | Samot | well there's a contract involved. |
| 17:51.17 | Samot | So yeah. |
| 17:51.17 | drmessano | Callcentric, Flowroute, etc all do 2 channels for their little "residential DID" thing |
| 17:52.01 | Samot | Yup. |
| 17:52.12 | Samot | I was referring to the context of a local telecom provider. |
| 17:52.17 | Samot | Comcast's, ATT, etc. |
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| 17:54.17 | drmessano | Right |
| 17:55.09 | Samot | I do agree though, a Flowroute would have been better. |
| 17:55.39 | drmessano | As soon as he said "They supply my email" |
| 17:55.49 | drmessano | and "i contacted them" |
| 17:56.51 | drmessano | The "ISP provided email box" analogy seemed fairly simple |
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| 17:58.00 | drmessano | If Mikos is providing you an email account and personal website, let him keep the SIP stuff |
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| 20:24.49 | Kobaz | sooooooooooooooooooooooooooo |
| 20:25.19 | Kobaz | If you have a carrier that's saying, we'll shut you down if you don't pay your bill, and the customer can't |
| 20:25.23 | *** join/#asterisk wonderworld (~wonderwor@unaffiliated/wonderworld) |
| 20:25.33 | Kobaz | get in touch with anyone to actually pay the bill... what recourse does the customer have? |
| 20:25.41 | Kobaz | in the past, we've contacted the FCC and they do jack |
| 20:25.52 | Samot | Wait... |
| 20:26.01 | Samot | The customer can't pay because they can't find someone to pay? |
| 20:26.09 | Kobaz | well the carrier isn't responding |
| 20:26.23 | Kobaz | they are dragging because they don't want to lose the account |
| 20:26.32 | Kobaz | and would rather fsck up the customer |
| 20:27.04 | Kobaz | long story short, they found out the customer is leaving, tacked on hundreds in bogus charges |
| 20:27.07 | Samot | OK so the carrier contacted their end user and said '"Pay or get cut off" |
| 20:27.13 | Kobaz | and the customer is like f it, pay it and be done |
| 20:27.24 | Samot | Now the customer is trying to pay but can't get a single person at the carrier to respond? |
| 20:27.28 | Kobaz | correct |
| 20:27.30 | Kobaz | 'carrier' |
| 20:27.39 | Kobaz | probably one guy in the basement reselling vitelity |
| 20:28.13 | Samot | So if they are leaving why are they concerned about being shut off? |
| 20:28.18 | Kobaz | DIDs |
| 20:28.24 | Samot | Toll Frees? |
| 20:28.31 | Kobaz | non-tf |
| 20:28.36 | Samot | Local DIDs can't be stopped. |
| 20:28.44 | Kobaz | well |
| 20:28.49 | Kobaz | legally, i agree |
| 20:28.51 | Samot | So have they started the porting process? |
| 20:28.53 | Kobaz | but... technically |
| 20:29.00 | Kobaz | you can disconnect the number and say byebye |
| 20:29.06 | Samot | Well first, the guy has not control. |
| 20:29.08 | Kobaz | and enter into a 3 year legal battle |
| 20:29.16 | Samot | If he is a reseller then he is not the carrier. |
| 20:29.24 | Kobaz | yeah |
| 20:29.32 | Samot | The carrier may inform him as a courtesy... |
| 20:29.58 | Kobaz | yeah |
| 20:30.05 | Kobaz | we'll do that, contact the actual tier1 |
| 20:30.16 | Samot | Hang on. |
| 20:30.22 | Samot | Because you'll be ignored. |
| 20:30.27 | Kobaz | hehe |
| 20:30.28 | Samot | They won't talk to you. |
| 20:30.33 | Kobaz | yes they will, i have my contacts |
| 20:30.46 | Samot | OK then they would be doing something shady. |
| 20:30.56 | Samot | So has this customer started the porting process? |
| 20:30.59 | Kobaz | well no, they will give me the phone number to call |
| 20:31.23 | Kobaz | nothing shady about looking up the did and saying 'call department X, talk to Y' tell them what's going on |
| 20:31.54 | Kobaz | porting, i need to find out |
| 20:31.59 | Kobaz | lots of layers here |
| 20:32.05 | Samot | Providing a non-customer information about another customer's account is shady. |
| 20:32.21 | Samot | They need to submit a port request. |
| 20:32.26 | Samot | Otherwise they got dick |
| 20:32.32 | Kobaz | yeah, definitely will be doing that |
| 20:32.36 | Kobaz | it's not customer information, it's carrier information |
| 20:32.38 | Samot | They need to start it and prove it was rejected |
| 20:32.42 | Kobaz | right |
| 20:33.02 | Samot | Because they could submit the port and it be completed in 24 hours. |
| 20:33.07 | Kobaz | right |
| 20:33.08 | Samot | Or up to two weeks. |
| 20:33.35 | Samot | So the customer has records of attempting to contact the provider to pay their bill. |
| 20:34.06 | Samot | They could send a certified payment via mail to show it was paid. |
| 20:34.21 | Kobaz | yeah sounds like a plan |
| 20:35.55 | wonderworld | hey, what happened to [TK]D-Fender? haven't seen him for a while, and he used to be here nostop over a decade. |
| 20:36.56 | igcewieling | wonderworld: I think he finally took a break. |
| 20:38.07 | Samot | He was online last night |
| 20:38.12 | Samot | Just not in here. |
| 20:39.32 | Samot | But from what I recall he works at a place that requires workers to be there for his job to mean something. |
| 20:39.52 | Samot | With most places on lockdown and not open, there may be no workers to deal with. |
| 20:41.36 | Kobaz | ah |
| 20:41.43 | wonderworld | i see. I hope he is well. he helped me a lot over the years and i liked his attitude |
| 20:41.57 | drmessano | blinks |
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| 20:42.15 | Samot | Yes, the bulging veins in forward attitude is always appreciated. |
| 20:42.31 | Samot | s/forward/forehead/ |
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| 21:06.59 | wonderworld | vodafone is a joke. they force customers to switch to voip but they don't offer T.38 because it has "bad interoperability". |
| 21:08.43 | drmessano | Bad interop with what? The 70s? |
| 21:11.58 | Samot | Well, Vodafone is German. |
| 21:12.34 | Kobaz | what's the new version of an Adtran NV644 (Quad PRI) |
| 21:12.47 | Kobaz | TA 908e is 2x PRI... |
| 21:12.56 | Kobaz | There's no replacement for the NV644 on their docs |
| 21:13.06 | Kobaz | probably wind up calling tomorrow |
| 21:15.34 | wonderworld | yes, courts and local administration treat FAX as an secure and official way of transmission, so nearly all offices have stupid FAX |
| 21:16.24 | drmessano | I wasn't knocking the need for fax |
| 21:16.36 | Kobaz | hah, secure |
| 21:17.21 | drmessano | I was knocking saying T.38 had "bad interop". How can something meant to shoehorn old tech into new tech have bad interop with the original tech |
| 21:17.46 | Kobaz | it kind of does |
| 21:17.49 | Kobaz | T.38 has its issues |
| 21:17.53 | Kobaz | because it IS interop |
| 21:17.57 | wonderworld | they don't specify it any further |
| 21:18.03 | Kobaz | and not equivalent tech |
| 21:18.23 | drmessano | So T.38 is worse than using G711/RTP? |
| 21:18.24 | Kobaz | you're convering a very timing-specific and very analog data interface to a very digital one |
| 21:18.35 | Kobaz | no, don't use g711 |
| 21:19.06 | wonderworld | This is the complete information on Fax from the official interface specification: |
| 21:19.06 | drmessano | Tell that to ALL the providers that don't support T.38 and only offer G711 and G729 as options |
| 21:19.08 | wonderworld | For Group 3 fax transmissions, at least the Passthrough mode (T.30 via G.711 A-law) must be supported. Group 4-fax is not supported according to the service description. The use of ITU-T.38 for FAX Transmission often leads to interoperability problems. |
| 21:19.13 | Kobaz | drmessano: haha, right |
| 21:19.15 | wonderworld | this will be fun |
| 21:19.21 | drmessano | That was my point |
| 21:19.49 | drmessano | If the options are G711/RTP or T.38, and they are disallowing T.38 because of "Bad interop", what in the actual fsck |
| 21:19.52 | Kobaz | wonderworld: faxing in general leads to interoperability problems |
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| 21:21.11 | drmessano | haha |
| 21:21.16 | drmessano | <PROTECTED> |
| 21:21.52 | drmessano | I'm glad Germany didn't win the war |
| 21:23.00 | Kobaz | heh |
| 21:23.06 | Kobaz | that would have been bad for a number of reasons |
| 21:23.11 | Kobaz | not just faxing |
| 21:23.16 | Kobaz | even the germans agree |
| 21:23.42 | igcewieling | Faxes are deceptively evil. |
| 21:23.46 | drmessano | Now, I wouldn't be against Snom phones on every desktop |
| 21:24.06 | drmessano | and maybe having a name like dieter |
| 21:24.18 | drmessano | But I draw the line at barring T.38 |
| 21:26.02 | wonderworld | for me it sounds like: buy our PBX, buy our service plan. it will work. otherwise: good luck. |
| 21:26.25 | igcewieling | That is what I'd expect from Vodafone |
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| 21:27.15 | Samot | 5:18:36 PM <Kobaz> no, don't use g711 <<- What would you use instead for fax? |
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| 21:27.21 | Samot | You can't compress it. |
| 21:27.34 | Kobaz | Samot: t.38 |
| 21:27.43 | Samot | T.38 is not a codec. |
| 21:27.50 | Samot | It's an encaspulation. |
| 21:28.04 | Kobaz | Right, i'm saying don't JUST use g711, sorry |
| 21:28.07 | Samot | G711 with T.38 encapsulation |
| 21:28.25 | Samot | But the fallback to T.38 failing or not being there is G711 |
| 21:28.45 | Samot | As T.38 isn't done until the other side says it can do T.38 |
| 21:28.47 | Kobaz | People i've talked to people who are like, just use g711, don't bother with t.38. And it's fine and dandy until it's not |
| 21:28.56 | drmessano | ^ that |
| 21:29.11 | Kobaz | Look it works! and then.... Look, it's completely broken! |
| 21:29.24 | wonderworld | look this 14 page fax only has 2 pages now |
| 21:29.28 | Samot | What is to say the other side is IP? |
| 21:29.30 | drmessano | Yep, Tuesday vs Thursday basically |
| 21:29.43 | Samot | Guys, FAX has always been best effort. |
| 21:29.46 | Kobaz | right |
| 21:29.48 | Samot | Even on POTS. |
| 21:29.53 | Samot | So let's have some realism here. |
| 21:29.55 | Kobaz | people don't even get THAT |
| 21:30.06 | Kobaz | they are like, BUT IT MUST GO THROUGH |
| 21:30.13 | Kobaz | send a fscking priority mail |
| 21:30.20 | Kobaz | if you want guaranteed.. |
| 21:30.27 | Kobaz | take a dump in a box and mark it guaranteed |
| 21:30.32 | Samot | But then you need a certified copied. |
| 21:30.49 | Samot | Facsimile |
| 21:31.00 | Samot | A direct duplicate of the original copy. |
| 21:31.09 | drmessano | Samot: Do you refuse to support T.38 because of bad interop? |
| 21:31.12 | wonderworld | i would move them to virtual fax service, but vodafone holds their fax number hostage for the next 2 years |
| 21:31.25 | Samot | I do T.38 |
| 21:31.38 | drmessano | Ok, so why are you arguing the other side of this? |
| 21:31.42 | Samot | But I have a fax platform. It has its own ATA |
| 21:31.54 | Samot | Because if T.38 fails |
| 21:31.59 | Samot | G711 is what is used. |
| 21:32.01 | wonderworld | PAP2T ? |
| 21:32.05 | Samot | No. |
| 21:32.06 | drmessano | So? |
| 21:32.19 | Samot | I was commenting on the "Don't use G711" statement. |
| 21:32.25 | drmessano | ..... pl |
| 21:32.28 | drmessano | ..... ok |
| 21:32.31 | Samot | Because even with T.38, it's still G711. |
| 21:33.15 | Samot | wonderworld: The device talks analong FAX protocols to the fax machine. It communicates with the fax server over HTTPS |
| 21:33.36 | Samot | So even if the Internet is down you can send faxes from the machine, it will spool themand then send them out. |
| 21:33.57 | Samot | When the Internet comes up. Then it will pull all the received faxes waiting at the server. |
| 21:34.01 | wonderworld | i see. a nice device! |
| 21:34.23 | drmessano | Kobaz: I think we were on the same page without being oddly specific |
| 21:34.37 | wonderworld | i am sure i can build something, but i hate to spend my time with this crap. |
| 21:34.59 | Samot | Yes, as am I. I do T.38 support for trunking customers so they can use it on their PBX. |
| 21:35.32 | Samot | Build a fax solution? |
| 21:35.45 | drmessano | I guess for now on I will make sure I refer to it as "T.38-less G711" vs "T.38 encapsulated G711" so the codec nazis don't get upset |
| 21:35.46 | wonderworld | build something that works 80% of the time |
| 21:35.47 | drmessano | Jeez |
| 21:36.01 | Samot | Sayz the grammar nazi. |
| 21:36.32 | Samot | drmessano: It's OK, we're fine people on both sides. |
| 21:37.14 | Samot | wonderworld: Build what? |
| 21:38.24 | wonderworld | a soulution for my customer to be able to fax without the availability of T.38 |
| 21:39.23 | igcewieling | Time to call Analog Man! |
| 21:39.40 | igcewieling | Saving faxes one POTS line at a time. |
| 21:39.45 | jsmith | wonderworld: Are you more concerned about inbound or outbound faxing, or both? |
| 21:39.55 | wonderworld | both |
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| 21:40.21 | wonderworld | i can buy hardware, they have the budget, but i am limited to vodafone t.30 |
| 21:40.51 | drmessano | Sangoma had that fax box thing |
| 21:40.59 | drmessano | Not sure how well it worked |
| 21:41.05 | wonderworld | i'll have a look |
| 21:41.36 | wonderworld | i ordered a PAP2T for testing. maybe it will work but i doubt it. |
| 21:41.49 | Kobaz | facts box |
| 21:41.54 | Kobaz | the world needs one of those right now |
| 21:43.04 | drmessano | The PAP2T is ancient |
| 21:43.23 | Kobaz | speaking of ancient... i've never used iaxmodem, and i've been intersted in trying it out at some point |
| 21:43.44 | drmessano | and it won't work any better than any other T.38-less-free-unencumbered-decapsulated-g711 connection |
| 21:44.58 | wonderworld | well at least PAP2T is not from the 70s |
| 21:45.03 | drmessano | The only way any of the PAP2's worked was the T.38 on the PAP2T, but we're back to T.38-encapsulated-g711-for-the-lawyers-TM |
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| 21:54.14 | wonderworld | another nice feature: |
| 21:54.30 | wonderworld | Vodafone does not offer STUN functionality and STUN messages to the STUN server of a third party must not influence the communication between the device and the Vodafone SBC. |
| 21:56.56 | wonderworld | Encryption of the signaling via TLS and the voice channel via sRTP is not supported for Vodafone Business VoiP. |
| 21:58.06 | wonderworld | ok, i ranted enough. if you have the choice, don't give them money. |
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| 23:38.14 | Zombie | I am going to see if they won't let me wait on this |
| 23:39.21 | Samot | Who is going to wait on what? |
| 23:39.58 | Zombie | On the Trunk connection to the VoiP Provider I talked too today... |
| 23:40.10 | Zombie | I think I may be shooting myself in the foot. |
| 23:43.43 | Zombie | A local ISP with a Voip division wants to offer me a Trunk for $10 for 500 minutes. |
| 23:43.47 | Zombie | a month. |
| 23:44.09 | Samot | That's not a trunk. |
| 23:44.30 | *** join/#asterisk nickvsnetworking (~nickvsnet@203.176.184.10) |
| 23:44.33 | Samot | That is basic phone service which will be equivalent to a POTS service. |
| 23:44.55 | Zombie | I'm not sure I fully understand everything the guy explained to me. |
| 23:45.09 | Samot | What did he explain? |
| 23:46.07 | arcarius | Just use Twilio Elastic SIP Trunking. $0.0045/min origination, $0.007/min termination. Nice API too. |
| 23:46.24 | arcarius | (I am not a Twilio employee, just a fan) |
| 23:46.42 | Zombie | Basically one Phone number, 500 minutes a month, they even probive their own Asterisk specific SIP Boxes. |
| 23:46.49 | Zombie | $10 |
| 23:47.00 | Samot | So it's basic phone service. |
| 23:47.24 | Zombie | They wouldn't handle Voicemail,thats all me, |
| 23:47.45 | Samot | What is their own "Asterisk specific SIP box"? |
| 23:48.40 | Zombie | Well they sent me a link to a Virtual Box Downloadable Asterisk Image that they say they offer to some small businesses |
| 23:48.56 | Zombie | They sent me a Service form. |
| 23:50.27 | Samot | HAHAHA |
| 23:50.37 | Samot | So they offered you an Asterisk distro. |
| 23:50.41 | arcarius | Wouldn't surprise me if it is just Asterisk installed on Ubuntu. Meh. |
| 23:50.43 | Zombie | Yes. |
| 23:50.44 | joepublic | is their virtualbox image provided as an RPM? |
| 23:51.01 | Zombie | ... no... |
| 23:51.14 | joepublic | I am, then, havin trouble picturing you doing anything with it |
| 23:51.29 | Samot | It's an image. |
| 23:51.36 | Samot | A prebuilt VM for VirtualBox |
| 23:51.36 | arcarius | Zombie, you probably know more about RPM than they do. |
| 23:51.43 | Samot | Import into VirtualBox. |
| 23:52.04 | Samot | arcarius: Let's not go that far. |
| 23:52.16 | Zombie | I know a scary amount about RPM, yes. |
| 23:52.17 | Samot | After all the RPM issues he had |
| 23:52.36 | arcarius | *giggle* |
| 23:52.48 | Samot | Yes, it was scary. |
| 23:55.10 | Samot | Scary amount of knowledge would mean that the amount of knowledge you have scares people. |
| 23:55.20 | Samot | As in makes them afraid, alarms them. |
| 23:55.26 | Samot | It's not actually a _good thing_ |
| 23:55.31 | arcarius | Enough knowledge to be dangerous? |
| 23:56.53 | Zombie | Well look, |
| 23:57.37 | Zombie | I Would rather look before I leap. |
| 23:57.58 | Samot | ...... |
| 23:58.00 | Zombie | and if you think this is too expensive, or a bad idea, I'm willing to listen. |
| 23:58.11 | Samot | That is amusing. |
| 23:58.31 | Samot | You have a PBX, anything other than a SIP trunking service is a poor choice. |
