| 00:09.07 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-ctkiaplfcpzysxfd) |
| 00:13.21 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 00:28.35 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 00:48.40 | *** join/#asterisk bwdunn (~textual@cpe-76-185-83-193.tx.res.rr.com) |
| 00:50.38 | bwdunn | Hi everyone. I've trying to do a long overdue SwitchVox to FreePBX. Because my old SIP provider isn't available after hours or on weekends, I cannot get their help. I have decided to move the lines to FlowRoute. My question is this - can anyone tell me how I might go about forwarding all calls coming into the old system over to the new temporary number? |
| 00:51.54 | bwdunn | Everything I have tried results in a fast busy. I've been at this since Friday morning and I'm hoping for a suggestion. |
| 00:53.07 | bwdunn | Calls on the old system come into an IVR, but when I put an instruction to dial an extension that's the new temp phone number, it dies with a fast busy. |
| 00:57.25 | bwdunn | I'm trying to send to an external number either NXXNXXXXXX or 1NXXNXXXXXX and the old system tells me the number doesn't match any call rules - which makes no sense at all. |
| 00:57.28 | bwdunn | (to me) |
| 00:59.23 | carrar | You can open up a switchvox support ticket for help on switchvox |
| 01:00.10 | carrar | but sounds like you're not forwarding correctly out of switchvox, or you peer setup is not correct (assuming your new freepbx is setup correctly, which also sounds doubtfull) |
| 01:00.48 | carrar | Can you make a call from a extension on switchvox to a extension on freepbx? |
| 01:01.09 | carrar | to verify your peer is setup correct |
| 01:04.23 | bwdunn | I don't have support with them and haven't for quite a while or I would. :-( |
| 01:04.38 | bwdunn | Yes - FreePBX seems to be working fine. |
| 01:04.48 | bwdunn | When I set it up to send to my cell phone, it works. |
| 01:04.49 | carrar | setup your freepbx system as a voip provider and then choose that when seting the call out with your outgoign call rules |
| 01:05.29 | bwdunn | Woah... that sounds like a good idea, but I think it's way above my hear. Is that simple? |
| 01:05.41 | carrar | well thats exactly what you are trying to do isn't it? |
| 01:06.09 | carrar | VOIP Provider --> swithcvox --> freebsd |
| 01:06.14 | carrar | err |
| 01:06.24 | carrar | VOIP Provider --> switchvox --> freepbx |
| 01:06.30 | bwdunn | Hmmm... that's what I would like end up with I think. I think what I was trying to do is to send the call back through my provider to the new provider to the new system. |
| 01:07.02 | carrar | why not link your switchvox directly to your freepbx system |
| 01:07.03 | bwdunn | That makes sense to me. I've read years ago that AIX is fairly simple. Is that how I should set it up? |
| 01:07.26 | carrar | no |
| 01:07.28 | carrar | see above |
| 01:07.51 | bwdunn | Use the FreePBX as a voip provider... that's where I get lost |
| 01:08.19 | carrar | You can send calls out any provider you wish |
| 01:08.41 | carrar | in switchvox it will look like a "provider" but you're really using it as a PBX peer |
| 01:08.47 | bwdunn | Okay... I'm looking at this idea now. They are on the same network so it makes a lot more sense than what I was trying to do. |
| 01:09.04 | carrar | I'm not sure the switchvox PEER setting will work with another none switchvox system |
| 01:09.15 | carrar | hence using the voip provider section |
| 01:09.36 | carrar | yup |
| 01:09.40 | carrar | keep the call local |
| 01:09.55 | bwdunn | One of the reasons I'm switching is because there are so many problems with getting SV to do certain things. Hands are tied. |
| 01:09.58 | carrar | that way you can also forward over extensions as you move them over as well |
| 01:10.29 | carrar | What are you trying to do with SV thats not working? |
| 01:10.31 | bwdunn | One thing I don't get yet - one the FreePBX side, how do I setup the ability to accept peering from the SV? |
| 01:10.56 | carrar | I've not use FreePBX, only Asterisk |
| 01:11.05 | carrar | this is not a freepbx channel |
| 01:11.13 | carrar | but I am sure it can do it |
| 01:11.29 | bwdunn | The latest was I couldn't use SoftPhones on the existing "physical phone" extensions. Works fine with FB. |
| 01:11.53 | bwdunn | Yeah - Okay - I will play around with this idea. That makes a lot of sense. |
| 01:12.07 | carrar | You'll want to setup a "peer" in the sip.conf that authenticates on IP |
| 01:12.20 | carrar | so it accepts any DID |
| 01:12.52 | carrar | and point that into a incoming context where you'll match on incoming DID's then send off to where ever you need |
| 01:13.06 | bwdunn | Gotcha. I ran Asterisk years ago but couldn't keep up with it so I went to SV. The box is old and I hate paying their fees, so it's time to move on. |
| 01:13.22 | carrar | yeah SV gets expensive |
| 01:13.36 | bwdunn | Some of this seems a little familiar. I will dig into it. I appreciate your help a lot. I was doing it very backwards. |
| 01:13.37 | carrar | but it's a decent system with one of the best GUI's |
| 01:13.42 | carrar | if you stay current |
| 01:13.47 | bwdunn | Agreed. |
| 01:14.13 | carrar | though if you've got some bug good luck on ever getting it fixed |
| 01:14.27 | carrar | I still have open bug reports that have not been resolved |
| 01:14.28 | bwdunn | Right. |
| 01:14.36 | bwdunn | So have we. |
| 01:14.49 | bwdunn | For clients. Never cared enough for us. :-) |
| 01:18.40 | [TK]D-Fender | <bwdunn> One thing I don't get yet - one the FreePBX side, how do I setup the ability to accept peering from the SV? <- trunk just like any other |
| 01:18.42 | [TK]D-Fender | SIP is SIP is SIP |
| 01:20.40 | [TK]D-Fender | Look up any other trunk |
| 01:20.43 | [TK]D-Fender | rules are the same |
| 01:21.35 | bwdunn | Yeah - I'm looking into doing that now. I have forgotten so much since I started with SV. |
| 01:21.42 | [TK]D-Fender | you either ID by IP (peer), or username (friend), you either actually auth (secret) or accept without challenge (insecure=port,invite), etc |
| 01:21.45 | [TK]D-Fender | All the same |
| 01:22.01 | [TK]D-Fender | bwdunn> Yeah - I'm looking into doing that now. I have forgotten so much since I started with SV. <- then go goolge up a sample for anyone else |
| 01:22.52 | bwdunn | I've actually downloaded the configs from the SV and I'm going to use that and what I have on the FB as a model. I think I see how this can work. Shouldn't be too bad - I hope. Makes sense now. |
| 01:24.24 | bwdunn | What I've found on Google is about using IAX as a peer for the SV. Any reason to avoid that if I run into issues with this: |
| 01:24.25 | bwdunn | ? |
| 01:24.45 | carrar | use g711u |
| 01:24.50 | carrar | and just SIP |
| 01:25.33 | bwdunn | Gotcha. I think the issue from what else I've read is that AIX is going to get complicated if it's not another SV. |
| 01:26.00 | bwdunn | When I was first learning this many years ago AIX was supposed to be the way to do peering with another system. No longer the case? |
| 01:26.17 | carrar | Just a staight SIP to SIP link |
| 01:26.39 | [TK]D-Fender | IAX isn't magic either |
| 01:26.44 | [TK]D-Fender | Neither is SV |
| 01:26.51 | [TK]D-Fender | it's jsut another closed-source GUI for Asterisk |
| 01:26.59 | carrar | but since both boxes are on the same LAN you don't need to both with IAX |
| 01:27.04 | carrar | both=bother |
| 01:27.07 | [TK]D-Fender | Asterisk is the motor underneat ... and it doesn't not CARE about pretty web-pages |
| 01:27.15 | [TK]D-Fender | -not |
| 01:27.21 | [TK]D-Fender | #quintuplenegative |
| 01:27.25 | carrar | hah |
| 01:27.44 | carrar | #thischannelneedsmorehashtags |
| 01:33.59 | *** join/#asterisk axisys (~axisys@unaffiliated/axisys) |
| 01:34.53 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-mbmyalncxsntvrji) |
| 01:58.07 | *** join/#asterisk jetlag (~jetlag@pool-71-168-203-134.cmdnnj.east.verizon.net) |
| 02:00.14 | *** join/#asterisk jjrh (~jjrh@2607:f0b0:1:6e12:7cbd:8735:196c:c3df) |
| 02:23.10 | *** join/#asterisk jab416171 (~jab416171@c-76-27-96-12.hsd1.ut.comcast.net) |
| 02:38.13 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 03:11.49 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 03:20.01 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 03:21.22 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 03:47.49 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 04:02.19 | *** join/#asterisk MarcoZink (~MarcoZink@201.141.206.216) |
| 04:46.48 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 04:53.10 | *** join/#asterisk bwdunn (~bwdunn@cpe-76-185-83-193.tx.res.rr.com) |
| 05:02.04 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 06:19.30 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 07:03.12 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 07:19.14 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 07:22.50 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 07:32.27 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 07:41.12 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-iutsudwrtjimhyuh) |
| 08:17.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:24.24 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 08:31.52 | *** join/#asterisk K0HAX (~K0HAX@c-75-72-143-131.hsd1.mn.comcast.net) |
| 08:48.33 | *** join/#asterisk Takapa (vegard@junior.svanberg.no) |
| 08:59.26 | *** join/#asterisk danjenkins (firrre@gateway/shell/firrre/x-aiqwlmbiajylcytz) |
| 09:28.46 | *** join/#asterisk Acilim (~cimbom@unaffiliated/acilim) |
| 09:30.01 | Acilim | Hello all, i am abit confused about ha and load balancing of asterisk. do anyone recommend any solutions about it |
| 09:42.13 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 09:44.23 | *** join/#asterisk CeBe1 (~CeBe@a81-14-224-229.net-htp.de) |
| 10:15.09 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 11:13.26 | *** join/#asterisk bulkorok (~Adium@92.206.232.166) |
| 11:35.14 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 11:57.30 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 11:59.38 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:45.45 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 12:53.51 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:bdef:e063:66a0:db6a) |
| 13:13.14 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 13:18.22 | *** join/#asterisk bkruse (~Adium@68.235.235.117) |
| 13:23.43 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:42.09 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 13:47.22 | ketas | it's fun what people say if you answer call but emit ringing tone there |
| 13:47.50 | ketas | "hey, it's ringing, there's no way someone will hear me" |
| 13:50.28 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 14:13.25 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:35.17 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 14:38.59 | *** join/#asterisk u0m3 (~u0m3@92.80.92.123) |
| 14:40.01 | *** join/#asterisk u0m3 (~u0m3@92.80.92.123) |
| 14:40.51 | *** join/#asterisk u0m3 (~u0m3@92.80.92.123) |
| 15:00.45 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 15:07.55 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:59.31 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 16:18.46 | *** join/#asterisk bkruse (~Adium@68.235.235.117) |
| 16:35.12 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 16:51.18 | *** join/#asterisk zerkalo (myricae@ny1.hashbang.sh) |
| 17:49.43 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-isdqtdrbebkqizpf) |
| 18:01.33 | *** join/#asterisk bkruse (~Adium@68.235.235.117) |
| 18:25.05 | *** join/#asterisk ruied (~ruied@bl7-215-11.dsl.telepac.pt) |
| 18:53.10 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5E0B3.dip0.t-ipconnect.de) |
| 19:31.15 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 19:31.56 | *** join/#asterisk dustinm- (~dustinm@105.ip-167-114-152.net) |
| 19:32.16 | *** join/#asterisk Tenhi_0 (~tenhi@178.18.241.180) |
| 19:32.29 | *** join/#asterisk Milenco (~Milenco@milenco.net) |
| 19:35.35 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 19:40.45 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 19:42.53 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 19:57.57 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 20:05.01 | *** join/#asterisk mjordan (mjordan@nat/digium/x-liqshhduelskdzfc) |
| 20:05.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:09.08 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 20:36.03 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 21:13.14 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 21:50.31 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 22:08.26 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 22:24.08 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:25.24 | *** join/#asterisk ruied (~ruied@bl7-215-11.dsl.telepac.pt) |
| 22:40.26 | SpeakerToMeat | Question, is there a "home" voip provider (1 or 2 channel outgoing + did) that you can recommend, that has equiparable prices to magicjack/ooma, with an easy end user interface for normal operations, with unlimited us calls, and which would let me use it as trunk with asterisk with sip, tls and srtp? |
| 22:42.19 | mjordan | That's a list of requirements. |
| 22:44.24 | SpeakerToMeat | yes. |
| 22:50.27 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:52.20 | *** join/#asterisk EOIP (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 22:53.24 | file | you are unlikely to find many, if any, with tls/srtp |
| 23:06.38 | SpeakerToMeat | file: So far I've found one. In hong kong (no unlimited calling though). Not expensive, but the UI they use is not very friendly, and... well something on their service feels not entirely right. |
| 23:06.52 | *** join/#asterisk MarcoZink (~MarcoZink@148.240.31.173) |
| 23:07.11 | SpeakerToMeat | Maybe I can not require tls and srtp if I can make it work with vpn. And get a decent vpn provider. |
| 23:07.35 | SpeakerToMeat | I wonder if rtp and sip will survive vpn... Maybe if I can achieve a low enough latency |
| 23:07.45 | [TK]D-Fender | UI? |
| 23:07.49 | [TK]D-Fender | We're talking about an ITSP. |
| 23:07.54 | [TK]D-Fender | SIP doesn't have a UI |
| 23:08.26 | SpeakerToMeat | [TK]D-Fender: Yeah but, you need to do some stuff like check balance, pay monthly fees or recharge, and in some cases maybe check logs (if you dont want to check them in your pbx) |
| 23:08.35 | [TK]D-Fender | vpn isn't going to hit your latency unless your router and bandwidth are garbage |
| 23:08.59 | SpeakerToMeat | Hmmm. |
| 23:13.04 | SpeakerToMeat | Leaving srtp aside, is there a decently cheap and reliable itsp you like? (general question) |
| 23:13.31 | [TK]D-Fender | vitelity |
| 23:13.33 | [TK]D-Fender | voip.ms |
| 23:13.33 | SpeakerToMeat | Geared thoward home (no need for a pletora of services, basically termination, origination/did and that's it) |
| 23:13.51 | [TK]D-Fender | flowroute.com |
| 23:13.51 | SpeakerToMeat | Thanks, I'll check them |
| 23:17.49 | SpeakerToMeat | Is there a diference between itsp and sip trunking? is the idea basically the same? |
| 23:18.27 | [TK]D-Fender | ~itsp |
| 23:18.28 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 23:19.13 | SpeakerToMeat | Yes, but sip trunking as I'vee seen some services refered to, is it basically a different name? or different service |
| 23:19.44 | [TK]D-Fender | no |
| 23:19.48 | [TK]D-Fender | SIP is a PROTOCOL |
| 23:19.53 | [TK]D-Fender | ITSP is a VoIP provider |
| 23:20.07 | [TK]D-Fender | Without getting nit-picky about the protocol |
| 23:20.25 | [TK]D-Fender | Although the vast majority use SIP |
| 23:20.39 | [TK]D-Fender | "SIP Trunk" is a crappy term |
| 23:20.56 | [TK]D-Fender | Trunk implies a "line" concept which inherently limits connctions |
| 23:21.05 | [TK]D-Fender | SIP is not bound by that concept |
| 23:21.10 | SpeakerToMeat | Yeah |
| 23:21.23 | SpeakerToMeat | ANyhow, thanks for the recommendations |
| 23:21.25 | [TK]D-Fender | Hence the term is looked down on by many |
| 23:21.39 | SpeakerToMeat | Yeah it's good to get the right term |
| 23:21.40 | file | I use Callcentric, they have various options |
| 23:22.33 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 23:24.06 | SpeakerToMeat | file: Looks good. Is service stable and reliable so far? (I do kinda know why this a stupid question, but, I ask nonetheless, sorry) |
| 23:24.21 | file | been fine for me |
| 23:24.28 | SpeakerToMeat | Ok, thanks |
| 23:36.14 | *** join/#asterisk italorossi (~Adium@177.193.104.232) |
| 23:56.30 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:57.49 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 23:58.58 | *** join/#asterisk bkruse (~Adium@68.235.235.117) |
| 23:59.39 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |