| 00:18.42 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 00:18.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.1 (2018/06/11), Standard: 15.4.1 (2018/06/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:19.19 | *** join/#asterisk u0m3 (~u0m3@86.124.245.34) |
| 00:21.52 | *** join/#asterisk mhache (~mhache_@CPE00005e000105-CM00fc8d221b00.cpe.net.cable.rogers.com) |
| 00:55.23 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
| 01:05.36 | *** join/#asterisk rpifan (~rpifan@2600:1:c148:e4b4:9dd:33c7:986b:f686) |
| 01:06.59 | *** join/#asterisk mhache (~mhache_@198.164.250.208) |
| 03:21.02 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 04:06.33 | *** join/#asterisk jamesaxl (~James_Axl@41.140.181.49) |
| 04:18.59 | *** join/#asterisk miralin (~Thunderbi@91.237.94.68) |
| 05:33.05 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:33.32 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 06:15.43 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 06:56.16 | *** join/#asterisk waldo323 (~waldo323@75-151-31-89-Michigan.hfc.comcastbusiness.net) |
| 07:19.19 | *** join/#asterisk jkroon (~jkroon@165.16.203.57) |
| 07:35.34 | *** join/#asterisk miralin (~Thunderbi@91.237.94.68) |
| 07:49.56 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 07:58.46 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 08:05.02 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 08:08.21 | *** join/#asterisk Oatmeal (Suzeanne@gateway/vpn/privateinternetaccess/suzeanne) |
| 08:34.39 | *** join/#asterisk Jackmor (95c89756@gateway/web/freenode/ip.149.200.151.86) |
| 08:35.35 | Jackmor | Dears I have been using asterisk kolmisoft for a longtime , now itsw crashing every 4-5 hours when i run dmesg it says segfault error 6 in ps |
| 08:35.58 | Jackmor | i have been searching and ready for almost 1 month , i even paid for people in freelance.com and they never helped |
| 08:36.08 | Jackmor | anyone can help ? |
| 08:42.36 | *** join/#asterisk jamesaxl (~James_Axl@41.140.181.49) |
| 08:44.20 | Jackmor | ? |
| 08:58.47 | *** join/#asterisk Sepultura (~Sepultura@unaffiliated/sepultura) |
| 08:59.05 | Jackmor | anybody alive ?> |
| 09:01.22 | Jackmor | <PROTECTED> |
| 09:15.45 | *** join/#asterisk jamesaxl (~James_Axl@41.140.156.186) |
| 09:40.03 | *** join/#asterisk totalizator (~totalizat@degu.pl) |
| 09:42.01 | totalizator | hi guys; any idea why my recordings in /var/spool/asterisk/monitor/ are suddenly created with new pwermissions - -rw-rw---- instead of -rw-r--r-- |
| 09:42.16 | totalizator | this way my sftp user can't read them |
| 09:43.19 | totalizator | this used to work and I can't understand what has changed |
| 09:47.03 | Jackmor | segfault at 6dd20000001a ip 00006dd34530b94a sp 00006dd22954fa10 error 4 in libc-2.12.so[6dd345293000+18a000] |
| 10:06.24 | *** join/#asterisk war9407 (war@pool-70-106-236-148.clppva.fios.verizon.net) |
| 10:07.50 | wdoekes | Jackmor: that info is by far not enough to tell you anything useful. you want a real backtrace, that can be obtained if you ensure that asterisk writes core files when it crashes |
| 10:08.01 | *** join/#asterisk DanB_ (~DanB@clt-195.192.205.215.ip-anschluss.net) |
| 10:09.23 | wdoekes | after configuring asterisk (init scripts) properly, you can `kill -SEGV` it to test that a core file is indeed written and readable |
| 10:10.33 | wdoekes | and then you wait for a new core file of an actual crash, use gdb and extract the backtrace (google will know how) |
| 10:13.16 | wdoekes | however, you'll need to run a recent asterisk to get any support here. "kolmisoft asterisk" does not sound familiar. it sounds like a modified/old version of asterisk. so that probably limits the usefulness of any collected backtrace |
| 10:16.09 | *** join/#asterisk war9407 (war@pool-70-106-236-148.clppva.fios.verizon.net) |
| 10:17.03 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 10:20.13 | Jackmor | error 4 in libc-2.12.so[6dd345293000+18a000] |
| 10:20.23 | Jackmor | crashing asterisk every 4 hours |
| 10:20.28 | Jackmor | whats wrong there |
| 10:24.27 | wdoekes | did you read what I wrote, Jackmor? |
| 10:47.51 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 10:47.52 | *** join/#asterisk gerhard_ (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 10:49.42 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 10:51.01 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 10:55.12 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
| 10:55.29 | Jackmor | i think ive done it |
| 10:55.39 | Jackmor | where is the backgtrace located |
| 11:02.10 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:10.22 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 11:26.40 | *** join/#asterisk SebastienThiry (~Thunderbi@232.190-246-81.adsl-dyn.isp.belgacom.be) |
| 11:31.28 | *** join/#asterisk miralin1 (~Thunderbi@194.8.128.66) |
| 11:31.38 | wdoekes | Jackmor: in gdb, once you've located the core file |
| 11:31.52 | wdoekes | google for 'how to get a backtrace from gdb' |
| 11:34.26 | Jackmor | i think ddos attack |
| 11:34.28 | Jackmor | and syn flood |
| 11:34.32 | Jackmor | makin this |
| 11:43.05 | *** join/#asterisk vt (quasselcor@unaffiliated/vt) |
| 11:48.53 | *** join/#asterisk sibyakin (~sibyakin@188.162.238.135) |
| 11:53.57 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 11:57.21 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 12:06.42 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:20.02 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:22.00 | *** join/#asterisk mhache (~mhache_@198.164.250.208) |
| 12:26.12 | *** join/#asterisk DannyA (~DannyA@cpe-74-64-127-27.nyc.res.rr.com) |
| 12:27.47 | DannyA | morning all...i was here yesterday asking about my issues with setting up a polycom ip phone on my home network using twilio sip. the issue was that outgoing calls always worked fine, but incoming calls only worked for the first few minutes after the device booted up. so today i decided to try to bypass my router, and plug the phone directly into my cable modem (which is itself a router with ports on back). so incoming calls wor |
| 12:27.47 | DannyA | k, and it says its registered with the sip server, but i can't make any outgoing calls, only a fast busy signal. any ideas what that means? |
| 12:35.18 | [TK]D-Fender | DannyA, means it doesn't think it's registered |
| 12:35.39 | DannyA | when i go to "Server Status" it says Registered, but i guess that may be a red herring |
| 12:35.56 | [TK]D-Fender | or your dialplan isn't set right |
| 12:36.09 | [TK]D-Fender | and what you're dialing doesn't match |
| 12:36.22 | DannyA | ....but it works to dial out when i plug it right back into my home router |
| 12:36.29 | DannyA | without changing anything |
| 12:37.16 | [TK]D-Fender | well because you're doing this direct we don't have normal means of getting debug |
| 12:37.29 | [TK]D-Fender | we don't support phones on that level without Asterisk involved |
| 12:41.28 | DannyA | understood. one last question: the issue im having with my TPLink router, where i guess the UDP connection is not being kept alive, preventing me from getting incoming calls, is there any way that software such as asterisk would be able to "solve" that, or it completely depends on the router hardware? |
| 12:45.19 | [TK]D-Fender | that's application level |
| 12:45.33 | [TK]D-Fender | when means either your provider sending one, or your device |
| 12:46.06 | [TK]D-Fender | But you confirm being able to receive calls which is what this would normally affect |
| 12:46.21 | [TK]D-Fender | keep-alives aren't normally required to be able to make outbound calls |
| 12:46.30 | [TK]D-Fender | esp if you get incoming |
| 12:49.07 | DannyA | ...right, which is why its so strange that outgoing suddenly doesn't work |
| 12:54.51 | [TK]D-Fender | can't really help with that... |
| 12:56.15 | Samot | Sigh |
| 12:56.36 | Samot | It's Twilio, do they expect the numbers to be formatted a certain way? |
| 12:59.56 | DannyA | oh hi :-) |
| 13:00.11 | DannyA | no. its not even reaching them. and again, it works fine as is when plugged into the tp link |
| 13:01.08 | Samot | How do you know it's not reaching them? |
| 13:02.25 | DannyA | theres no record of the call in their logs, no errors or anything. just an immediate fast busy signal when i dial any number. |
| 13:02.26 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:b1b2:875a:33ba:a35e) |
| 13:02.56 | Samot | OK that doesn't mean it is not reaching them. |
| 13:03.18 | Samot | First, Twilio requires outbound calls be sent in E.164 format, so the + has to be there. |
| 13:03.23 | [TK]D-Fender | I'm suspecting phone dialplan |
| 13:03.26 | Samot | Is the phone adding the plus? |
| 13:03.46 | DannyA | yes |
| 13:03.59 | vt | I was wondering if you knew any project that'd use so-called Mobile Wifi-calling (which is the possibility to authenticate a SIM card to an Internet IP base station via IPsec/EAP and then make calls via SIP (VoLTE)) to make outgoing calls using a SIM-card reader and Asterisk on a Linux server ? |
| 13:04.39 | Samot | vt: Uhm. You can connect Asterisk to a GSM gateway |
| 13:05.19 | Samot | DannyA: What is the host/server you have programmed in the phone? |
| 13:05.33 | vt | Samot: I don't need GSM network coverage with a GSM gateway, right ? |
| 13:05.44 | Samot | Yes, yes you do. |
| 13:05.47 | Samot | That's the point of it |
| 13:05.56 | Samot | GSM = Mobile Data |
| 13:06.19 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 13:07.10 | DannyA | <mytwiliodomain>.sip.us1.twilio.com |
| 13:07.54 | vt | Samot: yes, but my point is using the possibility to encapsulate GSM data (at least 4G and up, which are IP based, I don't know if any operator propose SIGTRAN for 3G over Internet) over the Internet |
| 13:08.07 | vt | and the protocol used in thoses IP-calls is plain SIP |
| 13:08.18 | Samot | With Asterisk? |
| 13:08.20 | Samot | No. |
| 13:08.37 | Samot | Asterisk needs to use Chan_Mobile or a GSM gateway it can connect to via SIP. |
| 13:08.43 | vt | so I can use for example a SIM card from country A in country B, and receive incoming calls over the Internet without roaming fees |
| 13:09.07 | Samot | No. |
| 13:09.12 | Samot | The GSM card is still a SIM card. |
| 13:09.28 | DannyA | Samot: i take that back, i dont know if its adding the +, but if it wasn't, how would it work when plugged into the TPLink? |
| 13:09.31 | Samot | A US SIM card on a Canadian GSM network is still roaming. |
| 13:09.51 | Samot | DannyA: Does the phone get an IP address when you connect it to the modem directly? |
| 13:09.54 | vt | chan_mobile is too high level, you use bluetooth-audio to make the call. I was thinking about Asterisk talking directly SIP to the mobile phone provider over Internet |
| 13:10.05 | Samot | vt: It can't. |
| 13:10.17 | Samot | vt: That would require the mobile provider to talk SIP |
| 13:10.18 | DannyA | yes |
| 13:10.27 | Samot | vt: You need a gateway that takes the SIP to GSM and vice versa. |
| 13:10.53 | vt | Samot: most of them do now, and the feature is embedded in Android and iPhone devices |
| 13:11.12 | Samot | What do you mean? |
| 13:11.16 | vt | anyway thank you, I guess someone with sufficent technical knowledge will implement this one day |
| 13:11.19 | Samot | You mean using your mobile data to make a SIP call? |
| 13:11.35 | Samot | Uhm. |
| 13:11.42 | Samot | It will take more than knownledge. |
| 13:11.46 | Samot | It will take more than knowledge. |
| 13:11.54 | Samot | It will take an infrastructure for it. |
| 13:12.10 | vt | Samot: No, I mean that operator for some years already, propose the feature « Wifi-calling ». It's 3GPP 29.234 I think. It's the possibility to do 4G directly over an IPsec tunnel |
| 13:12.41 | Samot | You are confusing Data with Voice in this case. |
| 13:12.56 | file | it's VoLTE. |
| 13:13.03 | vt | you can have absolutely no GSM coverage and just a Wifi-connection and your iPhone or Android device is able to receive and make calls |
| 13:13.13 | Samot | Right |
| 13:13.15 | Samot | I get that |
| 13:13.27 | Samot | That's because the phone is *changing* networks. |
| 13:13.39 | vt | Samot: voice is just a SIP session in 4G. It used to be some SS7 GSM packets in 3G, but now everything converges |
| 13:14.49 | Samot | VoLTE is Voice over LTE |
| 13:14.57 | Samot | Not VoIP ie VoIP over IP |
| 13:15.07 | Samot | Voice over IP |
| 13:15.30 | file | it uses some of the same standards underneath, ie SIP |
| 13:15.37 | Samot | Some |
| 13:15.50 | Samot | Which requires infrastructure to be in place for it. |
| 13:15.55 | file | I doubt anyone would implement the stack required to do such a thing on a computer, though |
| 13:16.00 | file | (acting as a phone) |
| 13:16.04 | file | the network side already exists |
| 13:17.05 | vt | file: one could grab the Android source code (which already implements it) and adapt it ? You just need a SIM card (with openct ?) reader to get the key for the tunnel, which also use support standards (EAP, IPsec) |
| 13:17.43 | file | I don't know IMS and VoLTE enough to say whether it is actually possible, I'm just saying I don't think anyone would do it |
| 13:20.26 | Samot | As far as I see it, if I have a mobile device with VoLTE on it that means my mobile number can receive/make calls ove the LTE or Wifi. |
| 13:20.42 | Samot | How does that work with my number that exists on the Asterisk server? |
| 13:20.49 | Samot | How does the phone know to connect to the Asterisk server? |
| 13:21.08 | file | wrong direction |
| 13:21.17 | Samot | Either direction. |
| 13:21.20 | file | vt is referring to Asterisk acting as the phone and connecting to the mobile provider |
| 13:21.28 | Samot | So a peer. |
| 13:22.46 | vt | It seems that they are some project (openlte, openvolte). Some projects involve using Asterisk in the core network of VoLTE mobile providers. But it's still embryonic |
| 13:23.22 | vt | « In one of our recent deployments of VoLTE & IMS in a mobile network, weâve replaced several components of our platform with the latest LTS release of Asterisk. » |
| 13:23.38 | file | yes, within the network |
| 13:24.00 | Samot | Right |
| 13:24.04 | Samot | Asterisk doesn't do IMS |
| 13:24.12 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 13:24.14 | Samot | That sounds like Kamailio+Asterisk |
| 13:24.22 | vt | I guess, wait some years and see, if some group of people with technical knowledge starts an open-source project for it |
| 13:25.28 | Samot | "weâve replaced several components of our platform with the latest LTS release of Asterisk." <-- That could mean "We use Asterisk for voicemail now" |
| 13:28.51 | [TK]D-Fender | Or they printed out the source code and are using it as a monitor stand... |
| 13:31.31 | Samot | Well |
| 13:32.16 | Samot | If you want Asterisk to talk to a GSM provider so you can be in Country B while the PBX and SIM card are in Country A with the mobile provider you use the current solution for that. |
| 13:32.29 | Samot | Asterisk + SIP-GSM Gateway |
| 13:36.21 | vt | Samot: yes, but I have no infrastructure in country A, just a SIM-card when I used to live there :) anyway, I understand no solution actually exists, but it's promising for the future |
| 13:37.05 | Samot | OK. |
| 13:37.11 | Samot | So where would the SIM card be then? |
| 13:37.15 | Samot | In another country? |
| 13:38.20 | theGoat | i am trying to compile dahdi on a fresh install of ubuntu 18.04 and keep getting these errors: https://pastebin.com/rv9b6vMc. has anyone gotten it to compile on 18.04 yet? |
| 13:38.53 | vt | yes, I've moved from there, but still has a lot of contacts from there and compagny that calls me (important calls). Now, I'm using gammu-smsd with a USB modem to receive calls notification, and I call them back with a SIP provider (so from another number). |
| 13:40.49 | Samot | vt: OK so what would the SIM card connect to? It needs to talk to a tower. |
| 13:40.52 | vt | On some another topic, maybe you know why, in the Monitor() documentation, I read « The channel's input and output voice packets are logged to files » and below, I see « file_format - optional, if not set, defaults to wav » |
| 13:41.01 | Samot | If your provider doesn't have a tower there, they have to use someone elses. |
| 13:41.20 | vt | what if the voice packets are in G.711 or G.729, why does Monitor() does not record the stream as-is ? |
| 13:41.44 | Samot | It takes that and records it to wav |
| 13:41.54 | vt | Samot: the tower is on the Internet. The SIM cards doesn't need to talk to a physical tower over radio, it doesn't care about the radio part |
| 13:42.13 | Samot | But who's tower is it? |
| 13:42.17 | Samot | It's not the providers. |
| 13:42.18 | vt | Samot: strangely enough, Monitor() gives me PCM data while the call is set-up using G.729 |
| 13:42.49 | vt | Samot: it's the provider's tower. Mobile phone providers now provides Internet access to their "tower" |
| 13:43.10 | Samot | So your provider has a presence in this country? |
| 13:45.02 | vt | Samot: no, but Internet packets can usually crosses international boundaries transparently |
| 13:45.25 | Samot | vt: I get that but that's not how providers work between each other. |
| 13:45.41 | Samot | There's still a transport/interconnection that has to happen |
| 13:45.57 | file | it's a feature of IMS, an encrypted ipsec tunnel over the public internet from the phone back to the provider so even if there is no network it can connect |
| 13:46.14 | Samot | Right |
| 13:46.19 | Samot | So this isn't going to be an Asterisk thing. |
| 13:46.53 | vt | I've used many times, but on my phone. I took dumps of network traffic, they provides IP endpoints announced on the whole Internet |
| 13:47.58 | Samot | theGoat: You may be experiencing a bug. |
| 13:48.18 | vt | Samot: the tunnel/SIM authentification part is not going to be part of Asterisk, but it that third-party software can then allow Asterisk/PJsip to do the SIP talking and manage the call |
| 13:48.19 | Samot | theGoat: Ubuntu 18.04 is like 2 months old. DADHI-Tools hasn't had an update in 2 years. |
| 13:48.49 | Samot | vt: Right. Which means any B2BAU can be used. Like FreeSWITCH |
| 13:49.05 | Samot | Because it's not the key component to make this happen. |
| 13:49.37 | theGoat | hmm..... i think there is a package that can be installed with apt, i'll have to give that a try |
| 13:50.11 | Samot | declaration of function âinit_timerâ; did you mean âinit_timersâ? [-Werror=implicit-function-declaration] |
| 13:50.11 | Samot | <PROTECTED> |
| 14:01.04 | *** join/#asterisk GoldenBear (~gb@104.223.31.252) |
| 14:04.28 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ykdyefnrhfurcdcn) |
| 14:04.28 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:06.16 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-ryecdrzmbnyqfcqr) |
| 14:06.16 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:15.29 | vt | Samot: it seems that Monitor() and MixMonitor() uses ast_audiohook(), and the audiohook interface to Asterisk provides only decoded audio stream, so it can't record the G.729 as-is, without any decoding/re-encoding |
| 14:16.07 | *** join/#asterisk cresl1n (uid299068@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:16.07 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:22.03 | vt | theGoat: I don't know about Ubuntu, but on debian, I install the package dahdi-source and do "m-a a-i dahdi" |
| 14:39.53 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 14:51.09 | Samot | vt: I'm not sure what your issue is? You want to record a g729 call in g729 format? |
| 14:51.15 | Samot | How would you listen to it? |
| 15:02.47 | *** join/#asterisk jkroon (~jkroon@165.16.204.170) |
| 15:32.40 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 15:36.28 | vt | Samot: with sox ? |
| 15:37.58 | Samot | Sure. I guess. |
| 15:39.17 | igcewieling | I didn't know sox supported g729 |
| 15:39.39 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 15:39.39 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 15:41.10 | Samot | I don't think it does either. |
| 15:41.17 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-saoxhuntdrwrtdxh) |
| 15:41.51 | igcewieling | I don't normally worry about it. Once I got a hardware transcoding card all my g729 problems went away. |
| 15:43.05 | *** join/#asterisk kharwell (kharwell@nat/digium/x-hmnnmtrpvkpwblhw) |
| 15:43.05 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:43.17 | Samot | I've never really had g729 problems. |
| 15:43.28 | Samot | Well that's not true. I had one when I got to the LEC. |
| 15:43.38 | Samot | But the solution was to dump g729 |
| 15:44.51 | igcewieling | I was never able to totally prevent transcoding to/from g729. We will violate our contract with Verizon if we switched to all ulaw. |
| 15:45.10 | Samot | We didn't have that issue. |
| 15:45.14 | Samot | We had our own network. |
| 15:45.19 | igcewieling | hopefully that will go away when the new connection to VZ is in place. |
| 15:45.47 | igcewieling | Samot: define "our network" you mean an SS7/SIP network or a data network or what? |
| 15:45.54 | Samot | Yes. |
| 15:46.00 | Samot | To both. |
| 15:46.36 | igcewieling | We mainly use XO to get to/from the customer and Verizon (SIP) and Level3 as the upstream voice carriers |
| 15:46.37 | Samot | The only thing ATT/VZ(Qwest/CL) did for us was x-connects and the wiring of the last/first mile. |
| 15:47.05 | Samot | We also owned our own blocks of NPA-NXX-X's. |
| 15:47.31 | igcewieling | *nod* that is handy. |
| 15:47.40 | Samot | Well that's what CLECs do. |
| 15:47.50 | igcewieling | there is more than one kind of clec |
| 15:48.01 | Samot | Sure. |
| 15:48.03 | Samot | We were Tier 1 |
| 15:50.09 | igcewieling | Management keep talking about becoming a facilities based CLEC, but it doesn't give us much more than XO for the amount of money we'd have to spend. |
| 15:53.51 | Samot | They became both Facilities / Resale when they bought the ITSP I was with. |
| 15:57.27 | Samot | Well, if they filed to be a Resale CLEC vs a Facilities CLEC then they knew they were going to pay more. |
| 15:57.46 | Samot | Especially if their main market was "local" |
| 16:02.42 | Samot | Even our LD rates, because we had to route out other LECs for out of state, was insanely low. I think I max rate for standard LD was like $0.00045. |
| 16:04.46 | Samot | In contrast, our "Resale" markets the lowest we could get was around $0.001 |
| 16:06.48 | Samot | Not to mention our DS0/DS1/DS3 costs in our facilities regions was stupid cheap. |
| 16:13.59 | Samot | So these 3-4 customers you have down due to VZ's outage, it's just voice? |
| 16:14.29 | jpsharp | That's my trick for the next few months. Setting up a CLEC/homing tandem provider. |
| 16:15.02 | Samot | Few months? |
| 16:15.13 | Samot | So you've already started? |
| 16:15.29 | jpsharp | Yeah. |
| 16:15.41 | jpsharp | The paperwork and filings are already in place. |
| 16:16.09 | Samot | Cool, that's the biggest PITA of it. |
| 16:16.51 | jpsharp | The Boss has people to do that. I just gotta do the tech side of it. |
| 16:17.30 | Samot | You doing both? |
| 16:17.43 | Samot | Facilities and Non-Facilities? |
| 16:18.08 | jpsharp | A mix of both. |
| 16:18.50 | jpsharp | Trying to do as much of it over IP as I can so I dont have to order eleventy thousand circuits. |
| 16:19.08 | Samot | Well depending on the service you provide, there are still circuits. |
| 16:19.57 | Samot | Unless you're just doing voice and letting someone else do the data. |
| 16:21.40 | jpsharp | I know I'll have to order data circuits, but I'd rather order and manage a GigE circuit than a dozen DS3s and their assorted terimnation gear. |
| 16:22.00 | Samot | Oh for sure. |
| 16:23.05 | Samot | We were doing as much GigE for new backhauls as possible when I left the LEC. |
| 16:23.31 | Samot | It was just most those DS3's where in place for years and needed to be migrated. |
| 16:23.57 | jpsharp | The only thing I'm scratching my head about and need to find some assistance with is doing all the SS7 stuff over IP. And the boss wants 2-way SMS too. |
| 16:24.09 | Samot | Well I hate SS7. |
| 16:24.39 | Samot | As for the SMS, you'll need an SMPP setup. |
| 16:25.20 | jpsharp | I'm familiar with doing ss7 over T1s with linksets and all that jazz. |
| 16:27.58 | Samot | I only really dealt with the SS7 side of things when I absolutely had to. |
| 16:30.53 | jpsharp | it'll be fun and exciting, they said! |
| 16:31.47 | *** join/#asterisk ih8wndz (jwpierce3@srv001.trnkmstr.com) |
| 16:32.04 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 16:32.22 | Samot | Well, look for MSPs in your area. |
| 16:33.33 | Samot | I think part of the success of the CLEC was how sales/installs/on-site were handled. |
| 16:34.43 | Samot | A very aggressive outside sales setup. The MSPs bundled us for the data/voice as part of the services they provided to the customer (PBX/Network/etc) and they did all the installs. |
| 16:48.47 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 16:53.30 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 16:57.30 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 17:05.00 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 17:40.12 | *** join/#asterisk user03 (~user03@unaffiliated/user03) |
| 17:48.47 | robl^ | Has anyone here had a chance to try the new A-series Digium phones? I'm trying to get some feedback / impressions on those versus say the D Series or the latest Polycom VVX for a new deplyment with roughly ~20-25 phones. Nothing too elaborate feature wise, but wanting intuitive and pleasant end-user experience |
| 17:50.13 | *** join/#asterisk miralin (~Thunderbi@194.8.128.66) |
| 19:51.12 | *** join/#asterisk captain118 (uid167508@gateway/web/irccloud.com/x-ynpcwjgzakauljbj) |
| 21:03.18 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:18.03 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
| 21:37.13 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-wnegnozhoevgqtme) |
| 22:08.43 | *** part/#asterisk user03 (~user03@unaffiliated/user03) |
| 22:09.43 | *** join/#asterisk Micc (~micster@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 22:09.58 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 22:53.52 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
| 22:54.58 | *** part/#asterisk kharwell (kharwell@nat/digium/x-hmnnmtrpvkpwblhw) |
| 23:28.05 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |