| 00:07.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.69) |
| 00:39.03 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 00:40.16 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 00:42.34 | *** join/#asterisk Nickinator (~Nickinato@14-201-136-222.static.tpgi.com.au) |
| 00:49.28 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 00:56.09 | *** join/#asterisk bandroidx (~bandroidx@unaffiliated/bandroid) |
| 01:09.07 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 01:23.27 | *** join/#asterisk Minotaur01 (~minotaur0@S010660735c150ce1.hm.shawcable.net) |
| 01:25.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:37.46 | *** join/#asterisk Nickinator (~Nickinato@14-201-136-222.static.tpgi.com.au) |
| 01:44.00 | *** join/#asterisk Nickinator (~Nickinato@14-201-136-222.static.tpgi.com.au) |
| 01:49.18 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 01:50.35 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 02:15.01 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 02:23.20 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 03:06.42 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 03:19.17 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 03:19.48 | ChannelZ | huh. Never noticed until today but the stock ulaw echo-test sound sounds like grap |
| 03:19.54 | ChannelZ | s/grap/crap/ |
| 03:20.29 | floren | hi bob |
| 03:24.42 | ChannelZ | Hello floren |
| 03:25.01 | j4jackj | Hi ChannelZ |
| 03:25.18 | j4jackj | Next to me is a Haiku machine, compiling FFmpeg. |
| 03:27.08 | ChannelZ | Facebook sucks. Why the hell can't I search my own timeline for something I said.. or anyone else's for that matter? |
| 03:27.47 | igcewieling | ChannelZ: You are not the customer, you are the product. |
| 03:27.47 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 03:28.45 | igcewieling | I've been slowly reducing my online footprint and moving my data offshore. I doubt it will do a lot of good, but I have to try. |
| 03:29.26 | igcewieling | Swiping the debit card is so much easier than cash, but I'm going to stay strong! |
| 03:29.27 | j4jackj | ChannelZ: I use Haiku forl a laugh. |
| 03:32.08 | ChannelZ | I use ffmpeg for a cry. |
| 03:37.20 | ChannelZ | I miss BeOS. It was fun to program for. |
| 03:38.05 | Penguin | I tried to install that on my old world Mac a long time ago, but never was able to get it done. |
| 03:38.47 | ChannelZ | I had a BeBox, and then later ran the Intel version |
| 03:41.10 | j4jackj | ChannelZ: get Haiku OS today. |
| 03:42.06 | ChannelZ | Too many other things to play with first. Sitting here trying to finish my Asterisk primer on my website |
| 03:53.56 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 03:55.48 | *** join/#asterisk hjf (hjf@unaffiliated/hjf) |
| 03:57.01 | hjf | why does asterisk insist in transcoding calls? i have a device that can do g729 and g722. i have disallow=all and allow=g722,g729,alaw,ulaw |
| 03:57.24 | ChannelZ | codec negotiation is not really a negotiation. |
| 03:57.46 | hjf | hm |
| 03:57.57 | hjf | but the call comes in from the sip provider, as g729 |
| 03:58.22 | hjf | and when it Dial(SIP/hardphone) and i pick up, it tries to talk g722 to the phone |
| 03:58.51 | ChannelZ | Each leg of the call is evaluated separately. |
| 03:59.40 | hjf | i also tried: |
| 03:59.43 | hjf | Set(SIP_CODEC=g729) |
| 03:59.48 | hjf | before Dial(...) |
| 04:00.06 | hjf | is that the right syntax? |
| 04:00.52 | ChannelZ | I've never heard of the SIP_CODEC var (which doesn't necessarily mean it doesn't exist) |
| 04:02.15 | hjf | i see mentions to SIP_CODEC, _SIP_CODEC and even __SIP_CODEC |
| 04:04.39 | ChannelZ | hmm interesting, seeing it in the source. Do you get a NOTICE about it? |
| 04:07.03 | hjf | NOTICE[100173][C-00000015]: chan_sip.c:7243 try_suggested_sip_codec: Changing codec to 'g729' for this call because of ${SIP_CODEC} variable |
| 04:08.43 | igcewieling | SIP_CODEC SIP_CODEC_INBOUND and SIP_CODEC_OUTBOUND are pretty useless. |
| 04:09.07 | igcewieling | For example, you'll need to transcode to provide early audio ringback, can't do that if yo don't have a g729 license |
| 04:09.47 | hjf | yes but transcoding sucks. bad. |
| 04:09.47 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 04:10.01 | igcewieling | then stop trying to do something you will fail at doing |
| 04:11.07 | igcewieling | We solved the whole G729 issue by buying 360 channels of g729 hardware transcoding goodness |
| 04:11.30 | igcewieling | has not been an issue since and I can move on to interesting stuff |
| 04:13.00 | igcewieling | I spent weeks trying to avoid transcoding, using SIP_CODEC_INBOUND and _OUTBOUND, etc. None of it eliminated the issue for us. |
| 04:13.00 | hjf | well i guess life is good when you have money to burn |
| 04:14.29 | Penguin | SIP_CODEC is valid. I use it for fax calls. |
| 04:14.40 | igcewieling | We saved money in the long run. Happy customers and we did not need to exceed 5Mbps of VoIP traffic (we get charged extra if we use more than 5Mbps) |
| 04:14.50 | Penguin | ;same => n,Set(__SIP_CODEC=ulaw); |
| 04:14.50 | Penguin | same => n,Set(__SIP_CODEC_INBOUND=ulaw); |
| 04:14.51 | igcewieling | Penguin: depends on the Asterisk version |
| 04:15.01 | Penguin | It's valid in this version or I wouldn't use it. |
| 04:15.31 | hjf | i wonder why i can see caller id from a POTS line to my sip provider, but when i dial out from the SIP provider i get a PRIVATE NUMBER |
| 04:15.50 | igcewieling | Penguin: maybe SIP_CODEC is an alias for SIP_CODEC_OUTBOUND or something. I think the _INBOUND and _OUTBOUND were added in 1.8 or 10, not sure. It is mentioned in the changlog |
| 04:16.08 | Penguin | SIP_CODEC is supposed to work for both inbound and outbound with just one var. |
| 04:16.27 | igcewieling | Penguin: you do ulaw fax? |
| 04:16.28 | hjf | i called the reseller guy and he tells me that "caller ID can't be set from VoIP calls" |
| 04:16.46 | Penguin | If I fax, I have to use ulaw because my ITSP does not have T.38. |
| 04:16.55 | igcewieling | Penguin: that sucks. |
| 04:17.10 | ChannelZ | huh, I can't get SIP_CODEC to do anything. |
| 04:17.13 | igcewieling | one of our wholesalers does T.38, one doesn't. |
| 04:18.40 | igcewieling | ChannelZ: I found it is great at causing customer service calls. 8-| check chan_sip.c, it is handled there. |
| 04:18.47 | Penguin | It will only switch the codec on the current channel, so if you are already using g729 between the ITSP and asterisk and set the thing to switch to g729 on the inbound call, nothing will change. |
| 04:20.22 | igcewieling | looks to be around line 7232 in 11-rc1 |
| 04:20.23 | Penguin | If you use the proper inheritance, i.e. underscores, you should be able to force the codec after a Dial(). |
| 04:20.39 | igcewieling | ..er.. 11.5-rc1 |
| 04:21.17 | Penguin | <hjf> Set(SIP_CODEC=g729) <------- __SIP_CODEC |
| 04:21.24 | Penguin | or at least _SIP_CODEC |
| 04:21.36 | Penguin | SIP_CODEC will not change the codec AFTER a Dial(). |
| 04:21.51 | ChannelZ | igcewieling: I see it in the source but it doesn't seem to do anything for me, I don't even get the notices. I'll fool with it later, off to play some Portal 2 |
| 04:22.37 | Penguin | I don't recall if I used it in 1.4, but it works correctly in 1.8. |
| 04:22.43 | igcewieling | goes back to setting up his own mailserver |
| 04:27.41 | *** join/#asterisk Nickinator (~Nickinato@14-201-136-222.static.tpgi.com.au) |
| 04:36.44 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 04:41.26 | hjf | could it be that the telco is blocking caller id from voip to avoid CID spoofing? |
| 04:44.02 | Penguin | Could be. |
| 04:44.21 | hjf | i hate that |
| 04:44.42 | hjf | cause i dont pick up calls from hidden numbers |
| 04:44.47 | Penguin | Not all telcos will allow people to randomly set an outgoing caller id number. |
| 04:44.54 | hjf | and i know a few people who don't either |
| 04:45.13 | hjf | hidden numbers are ALWAYS spam calls |
| 04:45.29 | hjf | my bank, for example, when it tries to sell me something it calls as a hidden number |
| 04:45.31 | Penguin | Are you talking about incoming calls or calls that you send out? |
| 04:45.57 | hjf | but when it tries to make me pay (like, i'm late to pay my credit card) their number is clearly visible |
| 04:46.24 | hjf | Penguin: i call from a telco to my voip line, and i can see the number. but if i call from my voip line to the telco, it shows up as hidden |
| 04:46.49 | Penguin | How are you connecting your asterisk to your telco? |
| 04:48.22 | hjf | via a voip provider |
| 04:49.54 | igcewieling | it is unlikely the receiving telco is doing anything with the callerid. chances are you are not sending the callerid you are required to send when handing the call to your ITSP. |
| 04:51.02 | igcewieling | our wholesaler, for example, gets upset if we don't send the number in E.164 format (i.e. +12129182000) |
| 04:51.31 | Penguin | hjf: You can't have both. You either use an ITSP or you hook the telco's line to your asterisk. |
| 04:51.56 | hjf | Penguin: what? |
| 04:52.03 | igcewieling | the other wholesaler will simply bill the call to the default TN on our account which means we can't bill it back to the customer. |
| 04:52.42 | igcewieling | Vitelity, on the other hand, requires you to set the outgoing CallerID via your account portal page and not when you send the call. |
| 04:52.45 | Penguin | hjf: You either use an ITSP or you hook the telco's line to your asterisk via some hardware interface on-premise. Which are you doing? |
| 04:53.35 | hjf | Penguin: ok, i'll try to be as verbose as possible |
| 04:53.37 | Penguin | It's not going to be both. |
| 04:53.44 | hjf | i have two phones in my desk right now |
| 04:53.58 | hjf | one is an analog phone connected to the telco (Telecom Argentina) |
| 04:54.10 | hjf | it has an rj11 jack going to the wall |
| 04:54.28 | hjf | the other phone is a voip phone that's registered to my asterisk server |
| 04:54.36 | hjf | which in turn is registered to an ITSP |
| 04:54.58 | hjf | when I call my ITSP number from Telecom AR i see my Telecom AR number in the IP phone display |
| 04:55.12 | Penguin | If asterisk is not connected to a physical phone line, don't say you are connected to the telco. Because you aren't. |
| 04:55.27 | igcewieling | hjf: pastebin the invite from the call from Asterisk to your ITSP. |
| 04:55.49 | hjf | when I call my Telecom AR number from the IP phone, i see the no caller ID info in my analog phone |
| 04:56.09 | Penguin | Pastebin your sip.conf and extensions.conf |
| 04:56.28 | igcewieling | suspects a hinky invite |
| 04:56.30 | Penguin | Callerid is set in dial plan and allowed to pass or not pass by the settings in sip.conf |
| 04:56.35 | hjf | 1 sec |
| 04:57.22 | hjf | exten => _X.,1,Dial(SIP/${EXTEN}@provider) |
| 04:57.32 | hjf | exten => _X.,n,Hangup() |
| 04:57.52 | Penguin | First of all, that's not the best syntax. SIP/peername/extension |
| 04:58.11 | Penguin | Second, where is the callerid data that you're sending? |
| 05:00.30 | hjf | http://pastebin.com/hkbngdgU |
| 05:02.04 | Penguin | Your conf is all fucked up. |
| 05:02.19 | Penguin | I'd tell you what is wrong, but you'd just ignore it and not fix it. |
| 05:02.22 | Penguin | Therefore, I probably won't bother. |
| 05:03.07 | hjf | ok |
| 05:06.04 | Penguin | I will tell you to remove the callerid value from your ITSP peer entry and set a callerid number in the dial plan. |
| 05:14.19 | hjf | exten => _X.,1,Set(CALLERID(num)=+54....) |
| 05:14.29 | hjf | exten => _X.,n,Dial(SIP/provider/${EXTEN}) |
| 05:15.00 | Penguin | Make the call and see what happens. |
| 05:15.39 | hjf | still appears as uknown number |
| 05:16.09 | hjf | i tried with +54, 54, and 0054, and with no 54 |
| 05:16.33 | Penguin | You may need to set sendrpid=yes on the itsp peer entry. |
| 05:16.42 | hjf | in argentina area codes can have 2, 3 or 4 digits |
| 05:16.54 | hjf | recently my city was moved from 4 digits to 3 |
| 05:18.14 | Penguin | Some ITSPs don't allow sending a callerid number. Those ITSPs usually make you set your callerid number in the customer portal. |
| 05:18.25 | hjf | http://www.telecom.com.ar/operacionesmayoristas/upload/contratos/OIR_28122011.pdf |
| 05:18.31 | hjf | will have to read that i guess |
| 05:18.35 | hjf | maybe the format is buried |
| 05:18.45 | hjf | there's no option to set caller id in the web portal |
| 05:18.58 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:21.51 | igcewieling | gives DNS a stern talking to |
| 05:23.51 | hjf | wow i don't understand anything in that document i posted |
| 05:23.59 | hjf | it mentions something about the "B number" |
| 05:24.06 | hjf | and the PQR, whatever that is |
| 05:24.25 | hjf | the format is +54-PQR-NUMBER |
| 05:24.33 | igcewieling | call them and ask |
| 05:24.35 | hjf | PQR seems to be an operator number |
| 05:24.54 | hjf | lol |
| 05:25.00 | hjf | what's from telecom wholesale |
| 05:25.08 | hjf | you don't "call" them |
| 05:25.28 | hjf | you show up there, in a suit. with a bunch of lawyers |
| 05:25.49 | hjf | and your telecommunications license, lol |
| 05:25.51 | *** join/#asterisk Nickinator (~Nickinato@14-201-136-222.static.tpgi.com.au) |
| 05:26.09 | hjf | (costs $20.000 and takes 5 years to get) |
| 05:26.44 | igcewieling | sounds like you need to call them up and cancel your account |
| 05:27.24 | igcewieling | Any ITSP which refused to support their service would soon be my former ITSP |
| 05:27.38 | hjf | you mean the ITSP? i tried contacting them and a drone just answered "not possible" |
| 05:28.04 | igcewieling | hjf: you called them up and said "My number is always showing as private"? |
| 05:28.29 | Penguin | You're sending phone calls from asterisk to an ITSP. We don't care about anything beyond that ITSP. |
| 05:28.50 | Penguin | The only thing you have control over is the "link" between asterisk and the ITSP. |
| 05:28.56 | hjf | yeah i told them i wanted my number to appear when i place a call through them and they told me "no, voip calls don't show a caller id number" |
| 05:29.11 | igcewieling | hjf: then you should cancel your account. |
| 05:29.28 | hjf | because that's obviously bullshit? |
| 05:29.58 | igcewieling | They may not permit sending your number out, but it is unlikely they are PROHIBITED from doing so. |
| 05:30.38 | hjf | yes, the document i'm reading from telecom says very clearly that a call without ANI WILL be dropped |
| 05:30.51 | igcewieling | There must be more than one ITSP in your country. |
| 05:31.12 | igcewieling | hjf: unless the document is from your ITSP it means nothing. |
| 05:31.38 | Penguin | Maybe this ITSP is a guy with asterisk and a couple FXO-SIP devices. |
| 05:31.49 | Penguin | That would be a good reason he can't send your callerid to the PSTN. |
| 05:31.57 | hjf | hehe |
| 05:32.12 | hjf | well they offer several local numbers to choose from |
| 05:32.29 | hjf | i doubt he has FXOs in every city |
| 05:37.09 | hjf | also |
| 05:37.13 | hjf | they're very expensive |
| 05:37.52 | hjf | i think it was sometihng about 7 cents (ARS, 1 cent USD) a minute for a local call |
| 05:38.25 | hjf | i'm pretty sure it's a LOT less with the telco |
| 05:40.03 | igcewieling | Vitelity (USA) is $0.0273 to +54, more to mobile, of course. |
| 05:40.20 | igcewieling | +54 mobile is $0.11955 |
| 05:40.28 | ChannelZ | Why is mobile more |
| 05:40.50 | igcewieling | ChannelZ: you must be in the USA or Canada |
| 05:41.01 | ChannelZ | yes |
| 05:41.23 | igcewieling | In MOST of the world calling mobile phones is more expensive than calling landlines. also in most of the world it is free to receive calls on your mobile. |
| 05:42.25 | hjf | CPP |
| 05:42.26 | igcewieling | The USA / Canada is unusual in that calling cell phones costs the same as calling landlines and that cell phones don't have their own area code and that receiving calls on your cell costs you |
| 05:42.29 | hjf | calling party pays |
| 05:42.44 | hjf | it used to be like that here in argentina |
| 05:42.52 | hjf | liek, in 1993 |
| 05:43.07 | hjf | mobile phones had the 0767 area code |
| 05:43.19 | hjf | but no one wanted them because it seemed dumb to pay to receive calls |
| 05:43.38 | hjf | so they changed it to CPP and now the caller pays for all of the call |
| 05:43.48 | igcewieling | The vitelity mobile rates is for +54-9 |
| 05:44.39 | hjf | the 9 is special |
| 05:44.46 | hjf | it means it's a mobile phone |
| 05:44.52 | hjf | for example |
| 05:44.54 | igcewieling | hjf: the mobile carriers in the USA convinced the regulators it was "unfair" to have cell phones in their own area code or some such silliness |
| 05:45.19 | hjf | take for example a number from buenos aires, area code 11 |
| 05:45.21 | ChannelZ | I guess I don't see either way as particularly advantageous over the other |
| 05:45.27 | hjf | 11-5555-1212 |
| 05:45.39 | hjf | if it's a mobile phone and i'm calling from a land line |
| 05:45.51 | hjf | i call (011)-15-55551212 |
| 05:46.02 | hjf | dial 011 if i'm outside 011 |
| 05:46.04 | igcewieling | in case you are interested http://www.itu.int/dms_pub/itu-t/oth/02/02/T02020000090001PDFE.pdf |
| 05:46.11 | hjf | now, if i'm outside argentina i dial |
| 05:46.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.148) |
| 05:46.21 | hjf | +5491155551212 |
| 05:46.35 | hjf | if 5555-1212 were a land line i'd dial |
| 05:46.41 | hjf | +541155551212 |
| 05:46.45 | hjf | no 9 |
| 06:52.40 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 07:00.03 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 07:31.17 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:36.51 | *** join/#asterisk CeBe (~CeBe@port-92-206-21-1.dynamic.qsc.de) |
| 07:45.41 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 08:06.09 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.192) |
| 08:06.19 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 08:34.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.205) |
| 08:43.07 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:11a4:58bd:ba06:e38f) |
| 08:53.01 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 09:05.09 | *** join/#asterisk apb1963 (~apb1963@174.134.98.138) |
| 09:18.54 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 09:25.56 | *** join/#asterisk TimeRider (~steve@2.219.208.88) |
| 09:26.52 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.205) |
| 10:05.46 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 10:06.53 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 10:10.11 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-149-086.pools.arcor-ip.net) |
| 10:10.15 | *** part/#asterisk ghost75 (~trechber@dslb-178-002-149-086.pools.arcor-ip.net) |
| 10:10.45 | WIMPy | X3.64 |
| 10:10.46 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 10:10.59 | WIMPy | oops |
| 10:13.25 | apb1963 | x11 |
| 10:13.32 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-69-23-217.home.otenet.gr) |
| 10:20.29 | ChannelZ | XXX |
| 10:25.48 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 10:43.43 | *** join/#asterisk vlad_starkov (~vlad_star@193.33.185.132) |
| 10:52.25 | *** join/#asterisk Nickinator (~Nickinato@14-201-136-222.static.tpgi.com.au) |
| 11:08.12 | *** join/#asterisk Blashyrkh (~chatzilla@194-166-36-32.adsl.highway.telekom.at) |
| 11:08.45 | Blashyrkh | i have just compiled 11.2 on a raspberry pi, but on make config it tells me we couldnot install init scripts for your operating system |
| 11:12.42 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 11:14.09 | ChannelZ | Guess it can't figure out what environment it's running in |
| 11:14.23 | ChannelZ | What sort of distro are you running on it? |
| 11:14.41 | Blashyrkh | latest raspbian |
| 11:14.57 | Blashyrkh | i guess i can use the debian thing |
| 11:17.55 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 11:18.05 | Blashyrkh | i know its unrelated |
| 11:18.09 | Blashyrkh | but in the makefile the |
| 11:18.33 | Blashyrkh | if [ -f /etc/debian_version |
| 11:18.39 | Blashyrkh | is the -f a check if the file exists? |
| 11:26.22 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 11:27.31 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 11:30.21 | Blashyrkh | okay i just told the makefile to treat is as debain |
| 11:30.25 | Blashyrkh | looks to work |
| 11:30.31 | Blashyrkh | seems to work |
| 11:49.33 | ChannelZ | sorry wandered off, but good glad its working |
| 11:57.34 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 12:12.25 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 12:25.41 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 12:36.48 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 13:02.45 | *** join/#asterisk shailender123 (7aa1f164@gateway/web/freenode/ip.122.161.241.100) |
| 13:08.20 | shailender123 | exit |
| 13:21.16 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.7.213) |
| 13:25.47 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 13:29.35 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 13:36.52 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.7.213) |
| 13:39.36 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 14:09.44 | *** join/#asterisk passthru (~ramon@177.158.168.85) |
| 14:14.00 | *** join/#asterisk Minotaur01 (~minotaur0@199.119.232.214) |
| 14:25.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 14:27.01 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:56.36 | *** join/#asterisk barbosa2 (~juliano.b@177.41.76.61) |
| 15:02.46 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 15:11.17 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:11.58 | *** join/#asterisk Minotaur01 (~minotaur0@S010660735c150ce1.hm.shawcable.net) |
| 15:15.25 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.7.213) |
| 15:29.12 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-69-23-217.home.otenet.gr) |
| 15:43.03 | *** join/#asterisk CoreyIrwin (~coreyirwi@69.165.255.164) |
| 15:44.16 | *** join/#asterisk monsterco (40e76515@gateway/web/freenode/ip.64.231.101.21) |
| 15:45.20 | monsterco | Hi everyone - I think I need something like SalesForce to connect to my Asterisk - I don't have much experience with SalesForce but I am looking for a CRM that can be easily deployed and can connect to Asterisk with all features enabled. My other requirement is that it has a mobile app for it. Of course I would love it to be open-source with a strong community. Does anything come to mind? thanks |
| 15:49.07 | [TK]D-Fender | vtiger |
| 15:49.17 | [TK]D-Fender | or sugarcrm |
| 15:49.22 | [TK]D-Fender | there are * addons for it |
| 15:51.11 | monsterco | what's the major difference between the two? |
| 15:51.37 | [TK]D-Fender | listed on their sites. |
| 15:58.48 | *** join/#asterisk Alex36 (uid12585@gateway/web/irccloud.com/x-clwoxvkvwcosrwcr) |
| 16:19.39 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 16:23.50 | *** join/#asterisk Alex_Bkash (b4eab263@gateway/web/freenode/ip.180.234.178.99) |
| 16:36.25 | igcewieling | We constantly have issues with database performance on SugerCRM. Looks like the schema was designed by a total DB n00b |
| 16:37.14 | jeev | wonderful |
| 16:37.20 | jeev | i've got a client movinv to sugar soon |
| 16:46.52 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.202) |
| 16:57.37 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 16:58.59 | igcewieling | jeev: keep throwing hardware at the problem until it goes away. |
| 16:59.52 | igcewieling | jeev: obviously EVERYONE doesn't have our issues or nobody would use SugarCRM. |
| 17:05.36 | apb1963 | Is there a technical reason my phone company can't assign me a 12 digit DID? Is it inherent in the switch, or is it just by convention or what? |
| 17:07.13 | apb1963 | I remember reading about a study done years ago stating that 7 digits was the most anyone could remember, and then they added 3 more... was that the reason numbers in the US are 10 digits? Or is it a physical hardware limitation of some kind? |
| 17:09.06 | [TK]D-Fender | ~nanpa |
| 17:09.06 | infobot | extra, extra, read all about it, nanpa is North American Numbering Plan Administration; the organization responsible for administering the integrated telephone numbering plan serving 19 North American countries. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. http://www.nanpa.com/ |
| 17:09.09 | [TK]D-Fender | ^ |
| 17:09.09 | igcewieling | apb1963: I guess it would depend on what country you are in. |
| 17:09.27 | [TK]D-Fender | nubers are what your administrative area defined' |
| 17:09.39 | apb1963 | So it's by convention |
| 17:10.01 | igcewieling | apb1963: in some countries you can likely get a 12 digit DID, but not in the USA. |
| 17:10.06 | [TK]D-Fender | You don't just invent any old number you want. |
| 17:10.20 | [TK]D-Fender | How would a system built over half a century ago know how to route calls? |
| 17:10.34 | igcewieling | [TK]D-Fender: no idea, but Europe figured it out. |
| 17:10.57 | jeev | igcewieling, keep throwing hardware? |
| 17:11.06 | jeev | igcewieling, how big is your db? |
| 17:11.17 | igcewieling | jeev: yes. I think we are on our third server now, each faster than the last. |
| 17:11.25 | apb1963 | so then it's a hardware limitation |
| 17:11.26 | jeev | igcewieling, specs? |
| 17:11.51 | igcewieling | jeev: I do not know, The extent of my involvement is mostly telling sales that I am NOT the person responsible for their slow Sugar stuff. |
| 17:12.08 | igcewieling | I also added a few indexes to help speed up queires about a year ago. |
| 17:12.12 | jeev | igcewieling, ok.. hmm, issue with these guys is that they dont necessarily have a server room |
| 17:12.20 | jeev | i guess i can go tower |
| 17:12.23 | jeev | if they need it.. |
| 17:12.32 | igcewieling | give me a min I'll see if I can find the IP |
| 17:12.35 | jeev | they do have fiber though, can easily scale it to a hundred megs, okay thanks |
| 17:13.01 | apb1963 | I'm presuming that a system built half a century ago has no software.... also the fact that the pics I've seen of switches show they're 9 track reel memory devices with lots and lots of relays. |
| 17:13.51 | jeev | igcewieling, salesforce burns my a$$. |
| 17:13.52 | jeev | i hate it. |
| 17:14.52 | igcewieling | Total of 6 processors activated (28800.67 BogoMIPS). DB is about 400M |
| 17:15.07 | igcewieling | hard to tell there are multuiple dbs |
| 17:15.29 | igcewieling | some are not used anymore in our system |
| 17:15.40 | igcewieling | jeev: do you know anything about database design? |
| 17:15.46 | jeev | no. |
| 17:15.48 | apb1963 | back'em up and delete'em? |
| 17:16.17 | jeev | so that's a single processor 6 core 2.4ghz ? |
| 17:16.45 | igcewieling | jeev: To me hardware is something people install and then tell me the IP of, I |
| 17:16.53 | apb1963 | lol |
| 17:17.03 | jeev | heh |
| 17:17.35 | apb1963 | I've built "several" computers starting with an empty case. |
| 17:17.39 | igcewieling | I've never even been within 2,000 miles of the server and access it a few times a year to look at some stupid complaint |
| 17:17.40 | jeev | i'm not a big guy with mysql but i know you've gotta have fast clock speed, 6 core 2.4ghz wouldn't be better th an a 4 core 3.4ghz, i've only seen it utilize a core, i'm not a pro with multithreading |
| 17:17.54 | igcewieling | apb1963: Last system I built was in 2004 |
| 17:18.13 | jeev | igcewieling, i've spent 80k on servers (for myself) in the past 9 months, 48k on client |
| 17:18.14 | jeev | :/ |
| 17:18.22 | jeev | not that client |
| 17:18.31 | igcewieling | Around that time I decided I "just want it to work" and not spend days figuring out stupid hardware stuff. |
| 17:18.45 | jeev | i love servers.. i had to upgrade my servers cause i'm adding more space. |
| 17:18.57 | apb1963 | I almost built one about 3 weeks ago... then realized I had an old hand-me down Dell from my brother sitting here, that only needed a hard drive. |
| 17:20.58 | igcewieling | jeev: with us, new server arrives, operations runs an auto-install DVD I built, server gets moved to customer and connected to a portable IP KVM, I set up networking on the box, then techs take it from there. |
| 17:21.52 | apb1963 | The Dell is what my asterisk now runs on. |
| 17:25.07 | apb1963 | So, how would one go about getting a block of DID's if one wanted to offer a service similar to what IPKALL offers? |
| 17:25.45 | igcewieling | apb1963: you sign a contract and write a check |
| 17:26.04 | apb1963 | kinda vague |
| 17:26.12 | igcewieling | though why you would want to offer a service like IPKALL confuses me. |
| 17:26.23 | igcewieling | apb1963: your question was kinda vague |
| 17:27.02 | apb1963 | OK..... who would I contact? My phone company? |
| 17:27.30 | apb1963 | NANPA? Do I need a license? |
| 17:27.40 | igcewieling | If you want connectivity to the PSTN either get one or more PRIs from your phone company, or use a SIP wholesaler like Verizon Business or Level 3 |
| 17:28.07 | igcewieling | with a PRI you are limited to having numbers in the local exchange where the PRI is located |
| 17:29.15 | apb1963 | ok, this is informative |
| 17:29.16 | igcewieling | as for needing a license, you should contact a telecom lawyer |
| 17:29.51 | jeev | igcewieling, yea i've done the stupid auto dvd stuff for redhat, i am not a big fan of redhat but i set it up, so far 600 servers have come online because of it, same specs. |
| 17:29.59 | igcewieling | The company I work for is a licensed CLEC in several states, but I think that is for our ILEC Resale stuff, not the VoIP stuff. Not sure though. |
| 17:30.00 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 17:30.10 | apb1963 | I see |
| 17:30.37 | apb1963 | That's basically what I was wondering.... about the voip stuff. |
| 17:31.21 | igcewieling | apb1963: I suspect your issue will be how to calculate taxes rather than licensing |
| 17:32.38 | igcewieling | apb1963: I'm pretty sure IPKALL does NOT use VoIP connectivity to the PSTN. |
| 17:32.57 | apb1963 | I'd rather have a tax issue than a licensing issue... and that's important of course, but at the moment I'm interested in the technical details of how to put it together. |
| 17:33.39 | apb1963 | Yeah, they probably use a PRI because all their area codes are in the same state |
| 17:33.49 | igcewieling | The only way I can think of for them to offer these free PSTN services is by being a CLEC and collecting termination fees. So the likely have PRIs or similar TDM interconnect, but then you run into issues with 911 services./ |
| 17:34.34 | apb1963 | So you have to be a CLEC to collect? |
| 17:34.56 | igcewieling | apb1963: Unless your telco REALLY loves you and wants to share revenue |
| 17:36.57 | apb1963 | I can feel the love every time I call to complain. |
| 17:37.34 | apb1963 | They've got a dossier on me the CIA would be proud of. |
| 17:38.08 | apb1963 | And all in just 2 short years! |
| 17:38.53 | apb1963 | So to collect fees, I'd need a license as a CLEC? |
| 17:39.25 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.202) |
| 17:40.02 | igcewieling | apb1963: contact a telecom lawyer |
| 17:40.04 | apb1963 | Everything is a mountain to climb. Isn't there any flat land anywhere? |
| 18:02.02 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.202) |
| 18:17.59 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 18:25.43 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 18:35.51 | vlad_starkov | Question: Recommend me please a good SIP client with reliable TLS support, for OS X. |
| 18:40.36 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.202) |
| 18:41.31 | bacobart | vlad_starkov: blink? |
| 18:41.41 | bacobart | http://icanblink.com/ |
| 18:42.49 | vlad_starkov | bacobart: it works terribly with TLS (at least when there is NAT) |
| 18:43.22 | bacobart | ok. then i dont know:P |
| 18:43.30 | vlad_starkov | bacobart: :) |
| 18:54.00 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:57.51 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 19:02.18 | CIDname | anyone looking for a good source for CNAM? |
| 19:12.08 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 19:15.01 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 19:15.16 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.15) |
| 19:16.12 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 19:24.55 | *** join/#asterisk nam3l3zz (~quassel@86-46-203-155-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 19:26.40 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 19:30.04 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 19:30.18 | ChannelZ | Define 'good' |
| 19:30.36 | pensmit | Does meetme work on asterisk 11? |
| 19:30.38 | pensmit | by default |
| 19:31.01 | ChannelZ | It's not built by default but it does work if you do |
| 19:31.16 | ChannelZ | ConfBridge is the better way to go |
| 19:31.41 | pensmit | hmm |
| 19:31.44 | pensmit | how can I add it |
| 19:31.47 | pensmit | doing some testing |
| 19:31.55 | [TK]D-Fender | Install DAHDI first like has always been required |
| 19:31.57 | ChannelZ | well you need to have DAHDI built and installed first |
| 19:32.03 | pensmit | it is |
| 19:32.36 | pensmit | Is it an option in menuselect? |
| 19:32.53 | ChannelZ | yes under Applications |
| 19:33.05 | pensmit | ok so im going to have to recompile and install then it sounds like |
| 19:33.07 | ChannelZ | have to scroll down to the extended group |
| 19:33.12 | ChannelZ | yes |
| 19:33.56 | pensmit | ok...is there a quick way to do that...clean the source I have or something or should I start with a fresh unpack of asterisk? |
| 19:34.07 | pensmit | I'm certainly no compiling guru |
| 19:34.17 | pensmit | ...or any kind of guru for that matter |
| 19:34.19 | pensmit | :) |
| 19:34.58 | ChannelZ | Well if you have a completely configured system you want to maintain, get a source tarball from whatever version you're running. |
| 19:35.48 | ChannelZ | I assume you're running a package now |
| 19:37.11 | pensmit | no |
| 19:37.25 | ChannelZ | you build from source? |
| 19:37.29 | pensmit | i did |
| 19:37.41 | ChannelZ | Is that build still sitting around? just use it |
| 19:37.41 | pensmit | following the definitive guide |
| 19:37.50 | pensmit | that's what i'm asking |
| 19:38.41 | ChannelZ | If you have DAHDI installed, just ./configure and then make menuselect |
| 19:38.53 | ChannelZ | turn on app_meetme, save, and then 'make' |
| 19:38.56 | foamz | hi |
| 19:39.06 | foamz | does anyone know of any good GSM gateways? |
| 19:39.48 | foamz | for terminating SIP/VoIP calls into GSM calls |
| 19:41.33 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 19:44.14 | pensmit | You are the man ChannelZ |
| 19:44.21 | pensmit | I owe you my first born child |
| 19:50.44 | ChannelZ | uh, you can keep it :P |
| 19:51.11 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.15) |
| 20:04.16 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:06.19 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 20:07.21 | igcewieling | ChannelZ: you should have taken it, you can always turn around and sell it. |
| 20:13.01 | ChannelZ | http://cdn.iofferphoto.com/img3/item/516/709/651/94SUvintage-social-posters-images-collection-200+-on-disk.jpg |
| 20:13.56 | ChannelZ | Oops wrong one... http://images.somethingawful.com/inserts/articlepics/photoshop/09-19-03-hygiene/Goo_baby.jpg |
| 20:14.49 | *** join/#asterisk kombi (~kombi@port-213-160-14-18.static.km-it.de) |
| 20:21.23 | pensmit | What's the default port of the ami? |
| 20:22.12 | igcewieling | pensmit: look in manager.conf.sample |
| 20:23.00 | *** join/#asterisk rneese (~rneese@pool-71-112-4-15.pitbpa.east.verizon.net) |
| 20:23.06 | rneese | afternoon |
| 20:23.26 | rneese | has anyone compiled debian armhf pkgs for asterisk ? |
| 20:23.36 | pensmit | thanks igce |
| 20:24.11 | ChannelZ | someone was here last night who built on raspberry pi |
| 20:24.17 | ChannelZ | but from source |
| 20:24.31 | rneese | ahh I pref pkgs |
| 20:24.40 | rneese | then you can load multi machines |
| 20:24.50 | rneese | dont want to just clone |
| 20:24.57 | rneese | also easier to keep update |
| 20:25.19 | rneese | is there a asterisk deb repo nowdays ? |
| 20:25.22 | ChannelZ | except that apparently no one has made one or maintains it. |
| 20:25.34 | rneese | ahh ok |
| 20:26.32 | rneese | well was thinking of a small arm based home pbx based on the digium asterisk gui and asterisk |
| 20:27.03 | rneese | I know the gui is not maintained but I am going to start workign on it to keep it growing |
| 20:29.38 | igcewieling | rneese: if you use packages you are pretty much on your own as far as support on this channel |
| 20:30.21 | rneese | you think they woudl support pkgs if it makes a standard install |
| 20:30.33 | rneese | thay woudl be easier to track and maintain |
| 20:31.06 | rneese | and make installs easier /faster |
| 20:31.36 | rneese | and if split out the modules into pkgs you could upgrade modules as needed |
| 20:31.46 | rneese | but just a debian users world |
| 20:32.47 | rneese | but ok then that counts asterisk out.. have to go look something else |
| 20:32.50 | *** part/#asterisk rneese (~rneese@pool-71-112-4-15.pitbpa.east.verizon.net) |
| 20:34.45 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 20:51.11 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.76) |
| 20:51.39 | pensmit | Anyone know why meetme list shows some accounts as unmonitored and how to fix that? |
| 21:00.19 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 21:03.59 | pensmit | Figured it out |
| 21:04.00 | pensmit | thanks |
| 21:19.10 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 21:25.57 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 21:28.16 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 21:29.54 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 21:52.35 | *** join/#asterisk Spengler1 (~spengler@pool-98-117-213-86.bltmmd.fios.verizon.net) |
| 21:53.05 | Spengler1 | whats the best way to set a blf to light up when a caller is in a call queue? |
| 22:06.41 | *** join/#asterisk Freeaqingme_ (~Freeaqing@91.214.168.110) |
| 22:08.30 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 22:11.55 | *** join/#asterisk seanbright (~sean@207.114.53.29) |
| 22:12.43 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 22:13.57 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 22:18.34 | *** join/#asterisk pbxbrian (~pbxbrian@unaffiliated/brian98) |
| 22:18.34 | *** join/#asterisk andy09usa (~Andrey@unaffiliated/andy09usa) |
| 22:18.34 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 22:18.39 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 22:18.39 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 22:18.39 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 22:18.39 | *** join/#asterisk Merlin (merlin@evendata.net) |
| 22:18.39 | *** join/#asterisk NET||abuse (~luke@86.6.148.64) |
| 22:18.39 | *** join/#asterisk tparcina (~tomo@161.53.170.46) |
| 22:19.32 | *** join/#asterisk CoreyIrwin (~coreyirwi@69.165.255.164) |
| 22:19.32 | *** join/#asterisk CeBe (~CeBe@port-92-206-21-1.dynamic.qsc.de) |
| 22:19.32 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 22:19.32 | *** join/#asterisk bipolar (~bipolar@204.186.46.94) |
| 22:19.32 | *** join/#asterisk zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
| 22:19.32 | *** join/#asterisk carbinemonoxide (~carbine@outonjupiter.com) |
| 22:19.32 | *** join/#asterisk defswork (~andy@141.0.50.105) |
| 22:19.32 | *** join/#asterisk CIDname (~CIDname@c-76-30-153-192.hsd1.tx.comcast.net) |
| 22:19.32 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 22:19.32 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 22:19.44 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:19.44 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 22:19.44 | *** join/#asterisk apb1963 (~apb1963@174.134.98.138) |
| 22:19.44 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 22:19.44 | *** join/#asterisk floren (~floren@unaffiliated/floren) |
| 22:19.44 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-jpcttcrgquidfexp) |
| 22:19.44 | *** join/#asterisk ncrollo (~nolhay@d233-64-78-227.dim.wideopenwest.com) |
| 22:19.44 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 22:19.44 | *** join/#asterisk carrar (~tim@osburn.jp) |
| 22:19.44 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 22:19.44 | *** join/#asterisk jalewis (~jlewis@solo.atlantic.net) |
| 22:19.44 | *** join/#asterisk Who-m3 (john@my.virt-host.com) |
| 22:29.15 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.76) |
| 22:37.49 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.76) |
| 22:45.02 | Freeaqingme_ | who here is using NAPTR/SRV records for locating sip servers? why/why not? |
| 22:46.17 | CeBe | Freeaqingme_: what do you mean with "locating"? |
| 22:47.00 | Freeaqingme_ | well, allowing one to determine what SIP server to contact based on a domain name. |
| 22:47.01 | ChannelZ | I use SRV |
| 22:48.05 | Freeaqingme_ | reason I'm asking is that it seems really easy to implement, yet I hardly see anybody using it (I never see: telephone: username@domain.tld, but always a plain pstn nr) |
| 22:48.39 | ChannelZ | Well SIP-to-SIP is not particularly widely used, that I've ever seen. |
| 22:49.02 | ChannelZ | 95% of the people out there have no idea what it even means. |
| 22:49.29 | Freeaqingme_ | it's a good thing enum exists then :P |
| 22:50.15 | Freeaqingme_ | ChannelZ, I guess it isn't widely used. too bad(?) |
| 22:53.42 | j4jackj | MASSIVE NETSPLIT |
| 22:57.26 | igcewieling | Asterisk's SRV support is primitive at best. |
| 22:57.31 | j4jackj | Heh. probably |
| 22:57.57 | Freeaqingme_ | igcewieling, got any url on that? |
| 22:58.03 | j4jackj | In other news, I got Opus compiled on Haiku. |
| 22:58.19 | ChannelZ | What constitutes non-primitive SRV handling? |
| 22:59.04 | igcewieling | ChannelZ: when using SRV, Dial will not try other servers if the first one fails. |
| 22:59.29 | ChannelZ | Does dialing by hostname? |
| 22:59.50 | igcewieling | I doubt it. |
| 23:00.31 | ChannelZ | More of a limitation of core then, nothing to do with SRV. |
| 23:00.47 | Freeaqingme_ | still a good one to be aware of |
| 23:00.57 | Freeaqingme_ | although ideally the first one is always available |
| 23:01.00 | igcewieling | That is what I said, Asterisk's SRV support, not SRV stuff in general |
| 23:01.05 | ChannelZ | Perhaps pjsip will have some better brains for this |
| 23:01.19 | Freeaqingme_ | I hope that the resolution stuff isn't hardcoded in pjsip |
| 23:01.37 | Freeaqingme_ | I mean, if it is, it's a good example of code duplication? |
| 23:02.11 | igcewieling | I use an overly complex AEL script to handle outbound SRV dialing rather than relying on Asterisk's SRV support |
| 23:02.19 | ChannelZ | I mean it has nothing to do with SRV records, at all. If I dial SIP/fart.com and fart.com resolves 5 A records, Asterisk still won't do anything about it. |
| 23:03.01 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 23:03.32 | Freeaqingme_ | it's not in pjsip I think, looks like it's in main/dnsmgr.c ast_get_ip_or_srv() |
| 23:03.33 | igcewieling | ChannelZ: Multiple A records were not designed for "failover". SRV is. |
| 23:52.53 | ChannelZ | Anyone doing BLF on Crisco phones? |
| 23:59.40 | *** join/#asterisk dgeary2 (~debian@pa49-187-88-50.pa.nsw.optusnet.com.au) |
