| 00:00.20 | autofsckk | ha |
| 00:00.39 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 00:08.35 | *** join/#asterisk kaushal (~kaushal@14.97.219.51) |
| 00:13.17 | kaushal | sorry got disconnected yesterday |
| 00:13.30 | kaushal | Any clue about http://fpaste.org/PftW/ ? |
| 00:15.33 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 00:16.59 | kaushal | debug mode here -> http://fpaste.org/Ovz1/ |
| 00:22.45 | WIMPy | That doesn't tell much. |
| 00:22.54 | WIMPy | What do you hear if you call there? |
| 00:25.25 | kaushal | WIMPy: i do not get any call |
| 00:26.00 | WIMPy | Yes, but it says there are tones or announcements. |
| 00:34.11 | ferdna | how do you validate the dial plan in cli? |
| 00:34.57 | WIMPy | reload it and watch for errors |
| 00:35.14 | WIMPy | But that won't do much checking. |
| 00:36.04 | ferdna | WIMPy, and config file? |
| 00:36.10 | ferdna | there is a way to validate config file |
| 00:38.06 | WIMPy | Which (part of the) config? |
| 00:38.20 | ferdna | WIMPy, i think sip.conf |
| 00:38.37 | ferdna | what do you pass to asterisk -C? /sip.conf? |
| 00:38.51 | WIMPy | chan_sip usually doesn't moan about errors in it config. |
| 00:39.23 | WIMPy | And -C takes the asterisk.conf. |
| 00:39.42 | ferdna | i have no asterisk.conf |
| 00:44.15 | carrar | try installing Asterisk |
| 00:50.31 | ferdna | carrar, i already created one |
| 00:50.51 | carrar | Did you do make install, make config |
| 00:51.10 | carrar | and read the readme files |
| 00:51.16 | ferdna | yes |
| 00:51.38 | carrar | then your file lies in /etc/asterisk/asterisk.conf |
| 00:51.53 | ferdna | yes |
| 00:52.07 | carrar | <ferdna> i have no asterisk.conf |
| 00:52.26 | ferdna | <ferdna> carrar, i already created one |
| 00:52.34 | carrar | You don't create it |
| 00:52.38 | ferdna | yes i do |
| 00:52.39 | carrar | it's already created FOR you |
| 00:53.08 | ferdna | yeah i copied the example that comes with the instalation |
| 00:53.22 | carrar | you don't need to copy it |
| 00:53.27 | carrar | it puts it in the right place for you |
| 00:53.33 | carrar | it sets it up for you |
| 00:54.03 | carrar | if you are having to create these by hand you did not install it per the docs |
| 00:54.28 | carrar | which is fine if thats what you wanted to do, but doesn't sounds like it since you couldn't find it |
| 00:55.01 | carrar | and if that file is missing |
| 00:55.10 | carrar | who knows what else is |
| 00:57.12 | ferdna | how do i know asterisk loaded my stuff? |
| 00:57.45 | carrar | define stuff |
| 00:58.04 | carrar | you can check to see if your stuff is there |
| 00:58.20 | carrar | You should however start by reading this |
| 00:58.22 | carrar | ~book |
| 00:58.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 01:28.31 | pigpen | I have a question if anyone is around. |
| 01:28.52 | pigpen | So, if I name my sip peers as lets say ACME_201 |
| 01:29.02 | pigpen | polycom actually |
| 01:29.17 | pigpen | and I dial from ACME_201 -> ACME_202 |
| 01:29.48 | pigpen | how would I get the CID to display 201 on the display of ACME_202 instead of ACME_201 ? |
| 01:30.25 | pigpen | ie: users are stupid. they will probably try to dial each other by attempting to punch in the ACME_ part. |
| 01:36.44 | *** join/#asterisk Turingi (~devon@unaffiliated/devon-hillard/x-7250961) |
| 01:37.01 | Turingi | Have you been successful in installing asterisk under ubuntu 11.10 (oneiric)? |
| 01:43.11 | pigpen | me? |
| 01:46.46 | Turingi | pigpen: there are no quirks so far with 11.10 and asterisk? |
| 01:47.25 | Turingi | using these instructions now https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 01:48.05 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 01:49.11 | pigpen | Yeah, sorry, I don't use Ubuntu |
| 01:49.19 | pigpen | never have, never will. |
| 01:49.46 | pigpen | I use Gentoo compiled from scratch |
| 01:50.11 | pigpen | good luck though. |
| 02:12.00 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 02:17.03 | Naikrovek | wowwww polycom ip 321s are only $80 now with adapter. |
| 02:17.06 | Naikrovek | jeepers |
| 02:18.02 | Naikrovek | gonna get 335s anyway |
| 02:54.29 | SeRi | Naikrovek, I finally got my 501 back to factory :P old but works. |
| 02:54.33 | SeRi | p3nguin, you in? |
| 03:03.36 | Naikrovek | SeRi: nice |
| 03:04.45 | *** join/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 03:06.02 | SeRi | :) |
| 03:06.23 | SeRi | p3nguin, please msg me when you are in. I need your info again :/ |
| 03:06.30 | SeRi | lost it. :( |
| 03:41.18 | carrar | pigpen, you need to switch to LFS |
| 03:41.30 | carrar | why go half way with gentoo :) |
| 03:47.05 | carrar | SeRi, get that Polycom configured yet? |
| 03:47.26 | Naikrovek | he did |
| 03:47.35 | SeRi | carrar, yes sr :) |
| 03:47.42 | carrar | via FTP? |
| 03:47.51 | SeRi | yes sr :) |
| 03:47.56 | carrar | most excellent |
| 03:48.17 | carrar | usign the default polycom login or custom login sent via DHCP? |
| 03:48.41 | SeRi | I was stuck trying to do it via tftp than I found dout that updates are done via ftp. |
| 03:48.45 | SeRi | using default |
| 03:48.49 | Naikrovek | daylight savings change tonight? |
| 03:48.54 | SeRi | working on creating a xml file |
| 03:48.58 | carrar | make sure not to leave ftp open to the public |
| 03:49.01 | SeRi | Naikrovek, yes |
| 03:49.08 | carrar | that default login/pass is on the list of scanners |
| 03:49.10 | SeRi | carrar, no its all internal |
| 03:49.23 | SeRi | I changed all that |
| 03:49.43 | carrar | Are you also doing DHCP via your Asterisk/FTP serer? |
| 03:50.02 | SeRi | yes sr. my asterisk behind another router |
| 03:50.06 | SeRi | so its dual nated |
| 03:50.09 | carrar | you can send a login/pass that you want you polycom to use when it connects via FTP |
| 03:50.18 | carrar | from your dhcp server |
| 03:50.32 | SeRi | ah! cool. |
| 03:50.52 | carrar | You can make it the same for all polycoms or give each polycom their own unique login/pass |
| 03:51.02 | carrar | anything is better then using the default login |
| 03:51.14 | carrar | as well as default directory |
| 03:51.32 | SeRi | carrar, I got it setup all unique. |
| 03:51.33 | carrar | polycom really are great phones |
| 03:51.38 | SeRi | carrar, do you know a good source to learn how to configure it via the cfg files and xml files? |
| 03:51.44 | carrar | yes |
| 03:51.48 | carrar | the admin guide |
| 03:52.01 | SeRi | looks like ill have to download it :) |
| 03:52.02 | carrar | I've slept with it |
| 03:52.06 | SeRi | LOL |
| 03:52.33 | carrar | Just skim through it |
| 03:52.39 | carrar | so you know what it can do |
| 03:52.57 | carrar | or at least you'll remember, oh yeah, I saw that feature int eh guide and then you can go back and do it |
| 03:53.07 | SeRi | cool. |
| 03:53.26 | SeRi | so far I have worked with the webui but is very limited and it does not do what I wanted to do |
| 03:53.44 | carrar | yeah the polycom web interface is pretty bad |
| 03:54.09 | carrar | Just turn that feature off and never look back |
| 03:54.33 | SeRi | lol. yeap. I learned that |
| 03:55.17 | carrar | and of course use split file for both bookrom and the sip app |
| 03:55.20 | carrar | files |
| 03:55.25 | carrar | bootrom |
| 03:55.35 | carrar | phear my typing skillZ |
| 03:55.54 | SeRi | lol |
| 03:56.23 | SeRi | got it. All I am missing is the damn base :( |
| 03:56.26 | carrar | on some of the older phones you need to symlink the default name to the split files till they get the newer code loaded that looks for split files first |
| 03:56.30 | SeRi | it wont stand! LOL |
| 03:57.24 | carrar | sym link stuff like sip.ld to 2345-11605-001.sip.ld |
| 03:57.36 | carrar | till it loads it |
| 03:57.40 | carrar | then remove it |
| 03:57.44 | carrar | (sym link) |
| 03:57.52 | carrar | same with bootrom.ld |
| 03:57.56 | carrar | anyways |
| 03:57.59 | SeRi | how do I split the files? |
| 03:58.06 | carrar | download them |
| 03:58.09 | SeRi | right now its all in the tftp root |
| 03:58.18 | carrar | there is split & combined downloads |
| 03:58.22 | SeRi | ah! |
| 03:58.25 | SeRi | I have the combined |
| 03:58.41 | SeRi | remove it? |
| 03:58.46 | carrar | I would |
| 03:59.09 | carrar | split files are smaller |
| 03:59.13 | carrar | and only for specific phones |
| 03:59.17 | carrar | so they load faster |
| 04:00.25 | SeRi | ok ok cool |
| 04:00.28 | SeRi | working on it |
| 04:02.01 | carrar | 501 split files are 2345-11500-040.bootrom.ld & 2345-11500-040.sip.ld respectivly |
| 04:02.20 | SeRi | ok |
| 04:02.24 | SeRi | I just got it. |
| 04:02.38 | SeRi | so remove the files I got in my tftp and upload this ones right? |
| 04:02.42 | carrar | thats specificied also in the admin guide or notes |
| 04:02.51 | carrar | yup |
| 04:03.02 | carrar | and check your ftp logs to see what its downlaoding |
| 04:03.11 | SeRi | ok |
| 04:03.16 | carrar | should match up |
| 04:03.43 | carrar | if it never requests 2345-11500-040..... then you need to upgrade the bootrom and or sip app |
| 04:03.58 | carrar | as the older software just asks for sip.ld or bootrom.ld |
| 04:04.02 | SeRi | I got the latest bootrom |
| 04:04.32 | carrar | well the latest for tha tmodel |
| 04:04.35 | carrar | since it EOL |
| 04:04.40 | SeRi | yes |
| 04:05.29 | SeRi | ok all the files are in place |
| 04:05.37 | carrar | check the ftp logs also |
| 04:06.03 | Naikrovek | carrar: the 4.0 web interface is much nicer, or so the screenshots say |
| 04:06.20 | carrar | I haven't ventured into 4 land yet |
| 04:06.31 | Naikrovek | i wonder if someone who is a polycom reseller would share the firmware with me? ANYONE |
| 04:06.38 | carrar | for the sip app anyways |
| 04:06.59 | Naikrovek | a lot of my phones are unsupported in 4.0 so i'll have to be weary when i upgrade |
| 04:07.09 | carrar | same here |
| 04:07.16 | carrar | older phones work great and are cheaper |
| 04:07.22 | carrar | and customers are happy with them |
| 04:07.26 | SeRi | carrar, remove all of the ld files except 2345-11500-040.sip_317.ld? |
| 04:07.46 | carrar | well 2345-11500-040.sip_317.ld should really be 2345-11500-040.sip.ld |
| 04:08.11 | carrar | as your ftp logs probably indicate |
| 04:08.16 | carrar | that you looked at |
| 04:08.18 | carrar | right? |
| 04:08.53 | SeRi | actually it does not matter it loads sip_317.ld just fine |
| 04:09.26 | carrar | You might be callign it out in the 1st config file |
| 04:09.43 | carrar | NAC,cfg |
| 04:09.45 | carrar | . |
| 04:09.57 | carrar | what do you have for files in MAC.cfg |
| 04:10.16 | SeRi | 000000000000.cfg |
| 04:10.31 | carrar | ahh see I remove all the default crap |
| 04:10.34 | SeRi | pb comming up |
| 04:10.38 | carrar | only leave in there whats used |
| 04:10.42 | carrar | and needed |
| 04:11.11 | SeRi | carrar, yea I am not that experience. I want to do that so I can have a clean tftp server :) |
| 04:11.17 | carrar | I stayed with the polycom defailt filename of sip.ld |
| 04:11.58 | SeRi | carrar, http://pastebin.com/XxPNZbAy |
| 04:12.35 | carrar | yeah I jsut have |
| 04:12.38 | carrar | <APPLICATION APP_FILE_PATH="sip.ld" ...... |
| 04:12.58 | carrar | phone knows what model it is and what file to grab after that |
| 04:13.13 | SeRi | is it safe for me to do the same? |
| 04:13.29 | carrar | You've checked the logs right? |
| 04:13.38 | carrar | and what do they tell you? |
| 04:14.15 | carrar | try removing all the files in your FTP directory to see what it asks for |
| 04:14.29 | SeRi | carrar, one problem... there is no logs :( This is an embedded system and it can not write ftp logs to / |
| 04:14.57 | carrar | You need to log |
| 04:15.06 | carrar | otherwise you are blind to whats going on |
| 04:15.43 | carrar | embedded system is fine once you know what you're doing and have tested everythign |
| 04:15.51 | carrar | however NOT for learning |
| 04:16.36 | carrar | even so, you want logs regardless |
| 04:16.50 | carrar | How you gonna know whats happen |
| 04:17.03 | SeRi | let me see if I can enable them one sec |
| 04:17.14 | carrar | log to a USB key |
| 04:17.39 | carrar | or send your logs out via UDP to a remote syslog server or syslog-ng |
| 04:18.30 | carrar | remote syslog servers that are locked down extremely tight are nice to have |
| 04:18.39 | SeRi | working on it |
| 04:19.07 | carrar | though probably not necessary for a home user :) |
| 04:23.08 | carrar | vsftpd is a nice FTP server to use |
| 04:23.36 | carrar | https://security.appspot.com/vsftpd.html |
| 04:24.01 | SeRi | Thats what I am using carrar |
| 04:24.07 | SeRi | I just enable xferlog |
| 04:24.12 | carrar | w00t |
| 04:24.13 | SeRi | let me see if it takes. one sec |
| 04:24.52 | carrar | may need to create the file "touch /var/log/xferlog" (as root) |
| 04:26.45 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 04:33.38 | SeRi | carrar, I got loging working :P |
| 04:33.57 | *** join/#asterisk radic (~radic@dslb-178-002-225-163.pools.arcor-ip.net) |
| 04:34.42 | carrar | if you remove all your files you can see what files it requests |
| 04:34.47 | carrar | good to know |
| 04:35.21 | SeRi | ok so remvoe all the files from the ftp root and wait and see what it asks for... |
| 04:35.28 | carrar | yeah |
| 04:35.30 | SeRi | remove* |
| 04:35.33 | carrar | keep them of course |
| 04:35.37 | carrar | but jsut move them |
| 04:35.39 | SeRi | yes sr. |
| 04:36.14 | Naikrovek | or you could just look at the ftp log to see what it requests |
| 04:36.34 | carrar | I would do that next |
| 04:36.39 | carrar | add all your files back |
| 04:36.44 | carrar | and watch again |
| 04:36.51 | carrar | this is just a learning thing at this point |
| 04:37.35 | SeRi | ok |
| 04:38.18 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 04:38.23 | SeRi | phone restarting |
| 04:38.40 | SeRi | phone uploaded log file |
| 04:39.25 | SeRi | OK DOWNLOAD: Client "10.30.3.144", "/000000000000.cfg", 1423 bytes, 342.19Kbyte/sec |
| 04:39.35 | SeRi | FAIL DOWNLOAD: Client "10.30.3.144", "/2345-11500-040.sip_317.ld", 26280 bytes, 89.38Kbyte/sec |
| 04:40.02 | SeRi | OK DOWNLOAD: Client "10.30.3.144", "/2345-11500-040.sip_317.ld", 2462135 bytes, 491.32Kbyte/sec |
| 04:40.12 | Naikrovek | the log it uploads will be delayed; it doesn't upload it in real time |
| 04:40.25 | Naikrovek | you moved the files, so it's not downloading anything |
| 04:41.01 | carrar | anyways, keeping playing and trying things |
| 04:41.06 | carrar | logs are good :) |
| 04:41.29 | SeRi | Naikrovek, I put the fiels back |
| 04:41.44 | SeRi | the phone is loading applications |
| 04:41.58 | SeRi | man a backlight would be nice :P |
| 04:42.09 | carrar | would be nice on a lot of phones |
| 04:42.19 | SeRi | running app = sip_317.ld |
| 04:42.20 | carrar | there is a mod some guy did on his Cisco phone once |
| 04:42.34 | carrar | some blue light |
| 04:42.42 | carrar | had to take his phone completely apart |
| 04:42.43 | SeRi | OK DOWNLOAD: Client "10.30.3.144", "/phone1_317.cfg", 11969 bytes, 16055.60Kbyte/sec |
| 04:43.22 | SeRi | I can live without it :) |
| 04:43.52 | carrar | now in your MAC.cfg if you change things and reboot you will see what it does |
| 04:44.43 | SeRi | carrar, ok the phone asks for /bootrom.ld /0004f202a5ca.cfg /phone1_317.cfg /sip_317.cfg |
| 04:44.59 | carrar | I'm guessing your bootrom is old |
| 04:45.10 | carrar | you probably didn't upgrade that |
| 04:45.18 | SeRi | o and /0004f202a5ca-phone.cfg /0004f202a5ca-directory.xml |
| 04:45.33 | carrar | read up on those in the admin guide |
| 04:45.46 | SeRi | carrar, It is on the latest for the 501 EROL which is 4.1.4 |
| 04:45.57 | carrar | yes |
| 04:46.11 | SeRi | EOL* |
| 04:46.25 | carrar | or 4.2.1, not sure |
| 04:46.29 | carrar | had to look at the matrix |
| 04:46.43 | carrar | have |
| 04:46.53 | carrar | 4.1.4 is probably fine |
| 04:47.06 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 04:47.48 | SeRi | cool |
| 04:48.32 | carrar | I don't see 2345-11500-040 in 4.2.1 |
| 04:50.00 | SeRi | is 4.2.1 the latest? |
| 04:50.24 | carrar | no |
| 04:50.38 | carrar | the latest won't work on your phone |
| 04:51.00 | carrar | I think 4.1.4 may be the latest that supports your phone |
| 04:51.06 | carrar | I haven't looked |
| 04:51.17 | SeRi | cool |
| 04:54.40 | SeRi | carrar, you know where I can get a base for this phone? a desktop base... |
| 04:59.29 | carrar | should have came with one |
| 04:59.31 | carrar | try polycom or ebay |
| 04:59.47 | SeRi | less d00d this thing is so old is nto even funny. It has move with me like 0 times |
| 04:59.53 | SeRi | I lost the base God knows wehn |
| 05:00.04 | SeRi | 10* |
| 05:00.19 | SeRi | ill try that |
| 05:00.25 | SeRi | playing with the config now |
| 05:06.03 | SeRi | can I use vi to edit the xml file? |
| 05:06.11 | SeRi | I dont want to mess it up. |
| 05:06.15 | carrar | I would |
| 05:12.31 | Naikrovek | you don't have a base at all, or just a wall-mount base |
| 05:12.48 | SeRi | Naikrovek, I dont have one at all |
| 05:12.57 | Naikrovek | oh |
| 05:12.58 | SeRi | the phone does not stand :( |
| 05:13.07 | Naikrovek | that would suck |
| 05:13.09 | SeRi | lol |
| 05:13.12 | SeRi | It does |
| 05:13.48 | Naikrovek | actually the phone would rock without a base, but saying 'that would rock' would not communicate what i was trying to say |
| 05:14.24 | SeRi | does this seem about right? [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|[11XXT] |
| 05:16.30 | carrar | no |
| 05:16.34 | carrar | read the admin guide |
| 05:16.41 | carrar | or use the default till you figure out why |
| 05:37.15 | SeRi | looks like I am set. |
| 05:37.31 | SeRi | overwhelming but got my default values in |
| 05:37.52 | SeRi | well what I think is necessary for now |
| 05:46.21 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 06:18.11 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 06:49.08 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 07:11.50 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:13.18 | *** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net) |
| 07:13.35 | SeRi | man I am loving this polycom. so simple to configure using provisioning! |
| 07:15.04 | *** join/#asterisk FainaUkraina (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 07:24.32 | *** join/#asterisk dokg (Matthew@224.86-240-81.adsl-dyn.isp.belgacom.be) |
| 07:24.38 | dokg | hi |
| 07:29.01 | SeRi | hola |
| 07:31.01 | dokg | how do I allow external calls to connect to the asterisk server? |
| 07:42.25 | ChannelZ | as in "anyone"? |
| 07:47.09 | dokg | yes, I am readin the documentation and read about how to connect internal phones to the server, but how does the server receives calls from anyone anywhere ? |
| 07:47.25 | ChannelZ | I guess that depends on what you mean |
| 07:47.48 | ChannelZ | If you want to receive calls from the public phone network, you generally get a SIP account from an ITSP |
| 07:48.33 | dokg | and how do I make the link between this account and the server and asterisk? |
| 07:48.41 | ChannelZ | If you want to receive SIP-to-SIP calls from anyone on the net, then you've got to allowguest=yes in your sip.conf and handle things appropriately -- see README-SERIOUSLY.bestpractices.txt |
| 07:49.33 | ChannelZ | In the case of an ITSP you add a peer/peers for them in sip.conf based on your account info and their specific requirements |
| 07:49.36 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 07:50.38 | dokg | what do you mean in case of an ITSP? |
| 07:51.19 | ChannelZ | If you are using a VoIP provider to make/receive phone calls. ITSP = Internet Telephony Service Provider |
| 07:52.37 | dokg | and in terms of hardware, do I need to buy anything to connect the ITSP to the Asterisk server in order to use the SIP |
| 07:53.06 | ChannelZ | Just a decent internet connection |
| 07:53.44 | ChannelZ | If you want to interface to the phone system via T1/E1/ISDN/etc yourself then you'd need additional hardware |
| 07:54.20 | dokg | thank you, then I have a last question, is: how do I simulate an external device? Because for now I don't have a VoIP provider, SIP account, etc. And I wish to build a prototype first |
| 07:54.59 | ChannelZ | You can use local SIP devices or softphones |
| 07:55.46 | dokg | do you have any to recommend ? |
| 07:55.52 | *** join/#asterisk johaidi (~johaidi@94.30.149.10) |
| 07:56.54 | ChannelZ | well there are a ton of SIP phones. Softphones like Zopier, Bria, Blink... |
| 07:57.07 | ChannelZ | depends on your platform |
| 07:57.27 | dokg | well asterisk requires Linux anyway? |
| 07:57.54 | dokg | you mean I could install a softphone an another computer connected locally to the asterisk server rmaybe? |
| 07:58.10 | ChannelZ | Yes but the softphones can run on any platform. Yes |
| 07:58.58 | dokg | thanks:) |
| 07:59.41 | ChannelZ | sure |
| 08:01.05 | dokg | is it possible to get the phone number of the caller (in the case of a call through the public phone network) |
| 08:01.19 | ChannelZ | Yes, to a certain degree |
| 08:01.33 | dokg | what do you mean? are there limitations, |
| 08:01.50 | ChannelZ | It's not a "secure" system, it's very easy to lie and transmit any number you choose |
| 08:03.27 | dokg | what happens if someone calls with these private numbers (not shown) or through internet with programs like skypeOut? |
| 08:03.58 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 08:04.18 | ChannelZ | You get whatever you get. Could be nothing |
| 08:04.45 | ChannelZ | I don't think Skype Out lets the user screw with it so chances are you'll get a good number from them |
| 08:05.15 | johaidi | you will get the number most likely |
| 08:07.36 | dokg | I have a database of people with their phone numbers. I would like to identify someone with the number he calls from. unfortunately if he uses a soft like skype I will get the number of skype, not the user. and if the user has a private number, won't be able to identify as well |
| 08:08.35 | ChannelZ | such is life |
| 08:08.41 | johaidi | in other words if u dont have the number in the database you wont be able to identify |
| 08:09.06 | ChannelZ | But it's easy to write a script to look up phone numbers in your own database and attach any name you choose. I do this |
| 08:09.15 | johaidi | or you could upgrade it with HMI and get the name no matter what the number is |
| 08:09.19 | *** join/#asterisk god-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 08:10.06 | dokg | but then I can ask them for their numbers. the only remaining problem is to distinguish a number from a soft like skype, and a real user number. Imagine I attach the skype number to a user. any other user calling with skype will be identified as the person attached to the skype number. I don't have to screw with this |
| 08:10.46 | ChannelZ | I don't think so |
| 08:10.58 | dokg | how do you handle that ? |
| 08:10.59 | ChannelZ | If someone has Skype Out, they have their own phone number don't they? |
| 08:11.14 | dokg | I use Skype to make phone calls, the number starts with +4.... |
| 08:11.15 | johaidi | let people to update database on unknown incoming numbers |
| 08:11.16 | *** join/#asterisk irroot (~irroot@197.172.27.198) |
| 08:11.17 | dokg | not my number |
| 08:11.39 | ChannelZ | oh. I guess you only get one if you have Skype In then. |
| 08:11.44 | ChannelZ | Well as I said it's not in your control |
| 08:12.38 | *** join/#asterisk godmachine (~godmachin@unaffiliated/godmachine-x6) |
| 08:12.45 | dokg | is it possible with asterisk to distinguish a call from a mobile, usual phone, and softs like skype or any other call through internet network ? |
| 08:13.04 | ChannelZ | It's the same way if someone is in an office building with a pool of 5 phone lines they all share; You don't necessarily know what person from that office is calling you if every person there doesn't have their own number |
| 08:13.16 | ChannelZ | Not really, no. |
| 08:13.23 | dokg | omg that might be a problem as well |
| 08:14.02 | ChannelZ | It's how the phone system has worked for decades |
| 08:14.14 | johaidi | indeed, cant see a problem there |
| 08:14.52 | johaidi | dokg you know how they differ? |
| 08:15.08 | johaidi | mobile always starts with some number, usual with other, skype - other |
| 08:15.25 | ChannelZ | You can discern "internet" calls in the sense that they don't follow normal phone number conventions, and come into Asterisk via a different path and can be handled totally separately |
| 08:15.56 | irroot | dokg if you peered with the providers through something like SS7 then the calls from them will come to you on the peered link so even with ported number it will come in on the carrier connected too but this will require a link to all carriers |
| 08:17.18 | dokg | irroot, could you tell me what you suggest without all these terms specific to telephony?))) |
| 08:17.52 | ChannelZ | It'd be like having to have a separate internet connection to every single service provider individually |
| 08:18.18 | ChannelZ | Unrealistic suffice to say |
| 08:20.54 | dokg | so when someone calls, I'll get the number, but as it might come from a pool of lines or from internet, I can still ask the user to confirm that this is his number, if not he'll enter the number he wants |
| 08:21.22 | ChannelZ | What is it you're trying to do? When your phone rings, does it really matter if the person is calling you from a cell phone or a landline or a helicoptor? |
| 08:21.31 | irroot | here there only 4 carriers and they not well interconnected there less than 10 data centers where its possible to get connected and there is one ITSP offering this as a service but for countries with many providers its not plausible we also have to conduct peering with all the voip providers to share calls at lower cost and not go via carriers |
| 08:22.09 | dokg | ChannelZ, no, I only want to identify the person according to his number. usually companes give to people an ID, and they type it in, in my case it's not possible |
| 08:22.43 | ChannelZ | You can do this, it's just not 100% reliable |
| 08:22.52 | dokg | i don't need 100% |
| 08:23.08 | ChannelZ | I could call you right now and make it look like you called yourself |
| 08:23.48 | dokg | spoofing |
| 08:23.56 | ChannelZ | Yes |
| 08:24.48 | dokg | there is no advantage for someone doing this in my case |
| 08:24.50 | ChannelZ | The phone system is actually quite "virtual" in this regard, every phone number does not equate to a single physical wire going from one place to another |
| 08:27.01 | johaidi | dokg cant you give your operators work on unknown incoming calls? |
| 08:27.08 | johaidi | they could update your db |
| 08:27.53 | dokg | johaidi, yes that's what I am going to do. |
| 08:31.57 | dokg | ChannelZ, how does skype differ with softphones like zopier, bria, blink, ... ? |
| 08:32.11 | ChannelZ | Skype is propriatary |
| 08:32.38 | ChannelZ | Their own protocols, codecs, and network |
| 08:32.46 | dokg | i can't simulate my system using skype? |
| 08:32.48 | ChannelZ | The others are all generic SIP compliant phones |
| 08:33.10 | dokg | I see, skype is not working with SIP protocol? |
| 08:33.16 | ChannelZ | Not really no. There aren't good ways to interface Skype with Asterisk |
| 08:33.29 | ChannelZ | Skype does have a SIP for Skype but it's an extra service you have to buy |
| 08:38.08 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 08:42.32 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 08:43.06 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:57.10 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 08:58.57 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 09:38.54 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 09:39.26 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 09:49.19 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 09:54.37 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 09:55.41 | *** join/#asterisk pietro (~pietro@88-149-224-253.dynamic.ngi.it) |
| 10:20.06 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 10:22.40 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 10:31.50 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 11:16.32 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 11:16.51 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 11:18.29 | *** join/#asterisk dokg (Matthew@224.86-240-81.adsl-dyn.isp.belgacom.be) |
| 11:26.44 | dokg | what is an "unauthenticated" call exactly as written in the oreilly book? |
| 11:33.01 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:38.34 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
| 11:42.10 | *** join/#asterisk ketas-ts (~ketas@82.131.20.203.cable.starman.ee) |
| 11:46.32 | *** join/#asterisk pietro (~pietro@88-149-224-134.dynamic.ngi.it) |
| 11:55.41 | Dovid | hi all |
| 11:57.06 | dokg | when I enter "sip show peers" in the CLI of Asterisk, I have the error "No such command". I typed "module show" to check wether the module is loaded, and chan_sip.so is in the list |
| 11:57.13 | dokg | can anyone help? |
| 11:57.41 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:57.46 | oej | write "sip" then press tab - what happens? |
| 11:58.06 | dokg | No such command 'sip -' |
| 11:58.27 | oej | Then something went wrong when the SIP channel was loaded. Which version are you using? |
| 11:58.48 | dokg | 1.8.7.1 |
| 11:58.55 | Dovid | j #asterisk-dev |
| 11:59.23 | dokg | the file seems ok |
| 11:59.26 | dokg | sip.conf |
| 12:00.30 | dokg | can the name of the device contain an underscore ? |
| 12:00.59 | Dovid | how hard it would it be to modify asterisk logging so that in logger.conf say I can set "full => notice,warning,error,debug,verbose,dtmf,fax|debug=0,agi-debug=9,sip-debug-on=yes. basicall i want SIP and AGI debuging but only for syslog and NOT for local files. |
| 12:04.24 | *** join/#asterisk cerberus_za (~coert@8ta-151-150-227.telkomadsl.co.za) |
| 12:10.23 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 12:21.33 | oej | Stop asterisk and start with "asterisk -cvvvv" and you will propably find an error message. |
| 12:27.50 | dokg | oej, I recompiled asterisk and now it works |
| 12:28.01 | dokg | oej, I read your message only after recompiling sorry |
| 12:30.31 | oej | No problem. If it works all is fine |
| 12:35.05 | dokg | hm, how can exit the CLI ? |
| 12:35.55 | WIMPy | gives dokg one guess. |
| 12:37.04 | dokg | exit doesn't work because I didn't start asterisk with -r flag. on the other hand, when I try to start asterisk with -r flag it says it's unable to connect to remote asterisk |
| 12:37.30 | dokg | WIMPy, ^ |
| 12:37.42 | WIMPy | If you start it in the foreground, there's no exit. |
| 12:38.02 | kaldemar | you need to stop it, not detach from the CLI. |
| 12:38.12 | kaldemar | core stop now |
| 12:38.14 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 12:38.31 | dokg | ok, so i'm trying to launch it in background ... I guess it's specific to ubuntu |
| 12:39.20 | WIMPy | Just start asterisk without parameters, then connect to it with rasterisk. |
| 12:39.39 | dokg | o that works thank you! |
| 12:48.44 | pigpen | I have a question if anyone is around. |
| 12:48.52 | pigpen | So, if I name my sip peers as lets say ACME_201 |
| 12:49.02 | pigpen | polycom actually |
| 12:49.10 | pigpen | and I dial from ACME_201 -> ACME_202 |
| 12:49.19 | pigpen | how would I get the CID to display 201 on the display of ACME_202 instead of ACME_201 ? |
| 12:49.35 | pigpen | ie: users are stupid. they will probably try to dial each other by attempting to punch in the ACME_ part. |
| 12:50.55 | *** join/#asterisk eicto (~morgoth@eicto.broker.freenet6.net) |
| 12:51.15 | pigpen | I have messed around with the polycom config with no luck. I have also messed around with the sip registration info (mine is in realtime, not sip.conf) with no luck. |
| 12:51.23 | WIMPy | that's called the dialplan. That maps extensions to functions, like dialling devices. |
| 12:51.35 | pigpen | Or is this just the way it is? |
| 12:51.49 | WIMPy | And for the peers you can definde a callerid. |
| 12:52.09 | pigpen | Oh, so just strip out the unwanted CID info in the dial plan? |
| 12:52.37 | kaldemar | pigpen: you'd have to take extra measures to get the phones to display the device name anyway. set callerid prameter for the devices in sip.conf. |
| 12:52.37 | WIMPy | No, set the CID you want in sip.conf. |
| 12:53.26 | eicto | Hi, I how to call queues from extesions ? |
| 12:53.46 | eicto | I tried Dial(SIP/NNNN) but it return acl.c: Cannot connect |
| 12:54.04 | eicto | I tried Queue() but it stall without call |
| 12:54.14 | kaldemar | core show application queue |
| 12:54.45 | pigpen | Ah, so currently I have a generic cidname defined like "Work Room", just define it further like "Work Room <201>" ? |
| 12:55.03 | WIMPy | yes |
| 12:55.14 | eicto | kaldemar, I tried application Queue, it stall with unknown reason |
| 12:55.31 | pigpen | Will the other still show up on screen (such as ACME_201) |
| 12:55.33 | pigpen | ? |
| 12:55.39 | eicto | when I call queue from phone it works fine |
| 12:57.08 | WIMPy | pigpen: Depends on the phonw what it will display. But usually number and name. |
| 12:58.54 | eicto | exten => da,n,Queue(3200,,,30) looks fine ? |
| 12:59.06 | pigpen | k. I'll give it a try. My realtime table has "callerid" as a field, but may try to add "cid_number" |
| 12:59.29 | kaldemar | eicto: what are you trying to do as opposed to calling the queue from a phone? |
| 13:00.18 | kaldemar | eicto: if 30 is supposed to be a timeout, you need one more comma, see the syntax. |
| 13:01.43 | eicto | I trying to initiate dial to queue from extention |
| 13:02.38 | kaldemar | eicto: rephrase that please. |
| 13:03.03 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 13:03.24 | eicto | I have the extention, and trying to dial some numbers from it |
| 13:03.34 | eicto | It calls well with dial |
| 13:03.43 | eicto | but with queue it fail |
| 13:03.51 | eicto | i found that now: |
| 13:03.54 | kaldemar | extension as in a an extension in your dialplan? |
| 13:04.05 | eicto | Exceptionally long voice queue length queuing to Local/dialalerts@DialAlerts-b1bb;1 |
| 13:04.13 | eicto | and all start to be slow |
| 13:04.49 | kaldemar | you need to pastebin your dialplan and CLI output of a call. |
| 13:05.45 | eicto | I making call with callfile |
| 13:06.21 | eicto | seems Queue operate much different from Dial |
| 13:06.34 | eicto | may be there is way to call queue using Dial() ? |
| 13:12.34 | eicto | kaldemar, I have to reformat my extention to be able to run from cli and back soon |
| 13:13.26 | *** join/#asterisk pietro (~pietro@88-149-226-114.dynamic.ngi.it) |
| 13:18.27 | eicto | kaldemar, how to initiate call from CLI ? |
| 13:19.04 | WIMPy | 'channel originate' |
| 13:25.13 | *** join/#asterisk singler (~singler@84.15.181.227) |
| 13:28.10 | pigpen | WIMPy, got it. I just had to define the callerid field from 'Name' to 'Name <201>' |
| 13:28.30 | pigpen | much easier, this way I don't have to modify a mess of db's |
| 13:28.42 | pigpen | thanks for the nudge (you too kaldemar ) |
| 13:32.09 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:34.21 | eicto | it is possible to channel originate same as described in call file there http://pastebin.com/EW60Wj8K ? |
| 13:34.41 | eicto | It show me help on syntax I assumed as correct |
| 13:34.56 | *** join/#asterisk DennisG (~dennisg@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 13:41.47 | WIMPy | Did you write "extension"? |
| 13:42.05 | WIMPy | And you cannot set callerID. |
| 13:42.52 | eicto | ah thanks |
| 13:43.37 | eicto | so I getting the warning channel.c: Exceptionally long voice queue length queuing to Local/dialalerts@DialTest-d256;1 |
| 13:44.01 | eicto | as sideeffect it make asterisk to use much resources |
| 13:44.23 | eicto | nothing in console output after originate channel] |
| 13:44.25 | WIMPy | I have no idea what you're trying to do. But I do have doubts it makes sense. |
| 13:44.51 | eicto | I tryiing to call queue from the extention |
| 13:45.09 | eicto | and it not dial, |
| 13:45.57 | WIMPy | That stoll doesn't make sense to me. |
| 13:46.08 | WIMPy | Did you read the |
| 13:46.10 | WIMPy | ~book |
| 13:46.10 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 13:46.12 | WIMPy | ? |
| 13:46.24 | WIMPy | Or another one. |
| 13:47.13 | eicto | Yes I readed parts of that and docs |
| 13:47.46 | eicto | The problem I have that it makes Dial() well, but Queue() alsmost kill instance |
| 13:49.35 | dokg | the following dialplan code should trigger a voice saying "main menu", but I don't hear any sound: Background(main-menu) |
| 13:50.21 | WIMPy | Did you answer the call? |
| 13:50.56 | eicto | WIMPy, it does not call |
| 13:51.19 | eicto | just stall asterisk, drops after timeout |
| 13:51.23 | WIMPy | That was for dokg |
| 13:51.48 | WIMPy | But in that case the question applies to you as well. |
| 13:52.13 | dokg | WIMPy, it's an example from the book, I called extension 201, which does a GoTo to extension "start", in extension start you have this background + WaitExten(5). and I can press 1 or 2, which will successfully trigger a voice saying "one" and "two" respectively |
| 13:52.58 | WIMPy | do you have a file named main-menu.*? |
| 13:53.23 | dokg | where should I look for this ? it isn't specified in the book it seems, I can't find it in sound directory |
| 13:53.27 | Dovid | besides for Nagio what is the "other software" that is like Nagios that I can use to monitor asterisk? I forget the name |
| 13:54.55 | WIMPy | Never mind, there seems to be a main-menu by defult. |
| 13:55.25 | dokg | why don't I hear it ? ( |
| 13:55.33 | dokg | the (hello-world) sound worked fine |
| 13:57.05 | WIMPy | Then the file may be missing. |
| 13:57.09 | WIMPy | is AFK |
| 13:57.28 | dokg | how can I check if the file is missing or not ? |
| 14:02.12 | dokg | I see in /var/lib/sounds there's a sound hello-world, but no main-menu sound, could have been deleted :( |
| 14:06.01 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:09.51 | *** join/#asterisk master_of_master (~master_of@p57B52B22.dip.t-dialin.net) |
| 14:18.25 | *** join/#asterisk mathi (Matthew@18.84-240-81.adsl-dyn.isp.belgacom.be) |
| 14:19.13 | SeRi | guys is there a way to have asterisk use another carrier for fail over in case there is no internet? like to use my gsm gateway? I am using GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN},failover) but that waits for the system to dial out and retunr a status.... aby other way? |
| 14:19.34 | SeRi | any* |
| 14:19.38 | DennisG | yes SeRi. just use 2 times dial |
| 14:19.59 | DennisG | like exten => s,n,Dial(SIP/number) |
| 14:20.32 | DennisG | that row 2 times. if the SIP carrier fails than it will use the other Dial (for the gsm gateway) |
| 14:20.51 | SeRi | ah! yes clever. lol |
| 14:20.57 | SeRi | time for testing :) |
| 14:21.06 | DennisG | good luck! |
| 14:21.10 | SeRi | Thanks |
| 14:22.00 | eicto | What is background reason to have Queue() application instead of Dial() ? |
| 14:22.19 | eicto | that Queue() looks broken |
| 14:22.31 | DennisG | Queue is another application then Dial |
| 14:22.56 | eicto | yes |
| 14:23.00 | DennisG | Dial can be used to dial to 1 SIP-account. Queue can be used to call to multiple SIP-accounts (eg a whole department) |
| 14:23.13 | eicto | Dial can be used to dial several Sips |
| 14:23.14 | eicto | also |
| 14:23.22 | cusco | dial can also be used to dial several peers |
| 14:23.29 | cusco | however queue has more features |
| 14:23.36 | eicto | as I can dial from external to sip/queue why i can't do it from internal ? |
| 14:23.41 | DennisG | that's true but with Queue() you have more support for that feature ;) |
| 14:24.10 | eicto | The problem that Queue() complet not work for me |
| 14:24.17 | eicto | it overload asterisk |
| 14:24.20 | cusco | hu? |
| 14:24.30 | DennisG | like to much load? |
| 14:24.35 | eicto | http://pastebin.com/EW60Wj8K |
| 14:24.52 | eicto | channel.c: Exceptionally long voice queue length queuing to Local/dialalerts@DialTest-5e29;2 |
| 14:24.59 | eicto | I getting that |
| 14:25.01 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:25.19 | eicto | I tried to setup options to that Queue() and etc, but it still fails |
| 14:25.34 | eicto | if I place Dial(SIP/num) there it dials fine |
| 14:25.38 | cusco | I don't think that is a problem with queue() |
| 14:25.42 | cusco | but I'm not suire |
| 14:25.50 | *** join/#asterisk pietro (~pietro@88-149-224-154.dynamic.ngi.it) |
| 14:25.52 | DennisG | look at the documentation of queue() |
| 14:25.56 | DennisG | you use it incorrectly |
| 14:26.12 | eicto | I feel blind |
| 14:26.29 | eicto | because I not understand what is incorrect |
| 14:26.53 | eicto | On other hand, is there any way to use it correct in my case |
| 14:27.16 | DennisG | eicto, https://wiki.asterisk.org/wiki/display/AST/Application_Queue |
| 14:27.26 | eicto | I readed that several times |
| 14:27.57 | DennisG | you need to create a queue in queues.conf and add the agent 3200 |
| 14:29.12 | eicto | I have queue in queue.conf |
| 14:29.26 | eicto | but where to add agent ? to queue 3200 ? agent 3200 ? |
| 14:29.42 | DennisG | in the queues.conf queue context ;) |
| 14:30.59 | eicto | still not understand |
| 14:31.04 | eicto | I have working queue |
| 14:31.06 | eicto | 3200 |
| 14:31.13 | eicto | I can call it from phone |
| 14:31.18 | eicto | and it works |
| 14:31.25 | eicto | why I need to change queue itself ? |
| 14:31.30 | DennisG | ooh oke i thought that it even didn't work |
| 14:31.37 | DennisG | whats then the problem? |
| 14:31.53 | eicto | when I call it from that extention it not works |
| 14:32.27 | eicto | not called by Queue() and then show that warning about Exceptionally long voice queue length queuing to Local/dialalerts@DialTest-5e29 |
| 14:33.19 | eicto | so I think to avoid using of Queue() and find a way to Dial(SIP/3200) or something like that (redirection from normal peer to 3200 ? ) |
| 14:34.05 | DennisG | so you have problems with the queue application itself? |
| 14:34.13 | DennisG | like high cpu load if you call it? |
| 14:34.20 | eicto | yes |
| 14:34.36 | eicto | and there is loop in debug3 about "Goto, Set" |
| 14:34.54 | DennisG | try to recompile it.. queue should not have a very high load |
| 14:34.59 | eicto | core debig 3 mode i mean |
| 14:35.12 | DennisG | i can even run like 100 queues on the same machine |
| 14:35.14 | eicto | seems it will not help |
| 14:35.30 | eicto | because there endless loop in Set/Goto |
| 14:35.31 | DennisG | which asterisk version do you use? |
| 14:35.48 | eicto | no any problem when I call that queue from external |
| 14:36.05 | eicto | Asterisk 1.8.5.0 |
| 14:36.39 | DennisG | sorry have to go.. |
| 14:36.49 | eicto | I think queue() can't be used from originate call |
| 14:38.15 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:40.29 | Madkiss | Hi there. I seem to recall that asterisk has an OpenAIS plugin, can somebody tell me what exactly that thing does? I.e. can I run a "clustered asterisk" with master/slave-clients and the like? |
| 15:07.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:12.59 | eicto | Still have that Queue() not working... http://pastebin.com/pnaVmcGj |
| 15:13.20 | eicto | Can anybody help ? Looks like bug in asterisk, but i not sure |
| 15:14.52 | [TK]D-Fender | Funny fail on link. use another PB site |
| 15:17.24 | eicto | [TK]D-Fender, what yoou'll suggest ? |
| 15:17.35 | [TK]D-Fender | ~pb |
| 15:17.35 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:18.28 | eicto | http://jsfiddle.net/oceog/qdwDe/ |
| 15:18.48 | eicto | I placed here :) |
| 15:20.35 | eicto | [TK]D-Fender, http://pastecode.com/dG |
| 15:20.52 | [TK]D-Fender | Show the compete call attempt |
| 15:21.13 | eicto | It says nothing, or you want from log ? |
| 15:21.30 | [TK]D-Fender | CLI only |
| 15:21.39 | [TK]D-Fender | verbose 10 |
| 15:23.56 | eicto | [TK]D-Fender, moment, it spammed console |
| 15:27.16 | eicto | [TK]D-Fender, http://pastecode.com/dH |
| 15:27.26 | eicto | it very spam immediatly |
| 15:28.49 | [TK]D-Fender | <PROTECTED> |
| 15:28.57 | [TK]D-Fender | Look at this REALLY closely |
| 15:29.08 | [TK]D-Fender | infinite loop <- |
| 15:29.15 | [TK]D-Fender | 2 = goto 1 |
| 15:29.56 | eicto | I updated with top of that output (was hard to catch) |
| 15:29.57 | eicto | http://pastecode.com/dI |
| 15:32.00 | eicto | [TK]D-Fender, may be there any other way to call queue from extension ? |
| 15:32.26 | eicto | It seems I can't submit bug to Asterisk mantis |
| 15:34.26 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 15:38.50 | eicto | btw It changing called peer sometime, |
| 15:45.30 | eicto | ah looks bug in freepbx :) |
| 15:45.36 | [TK]D-Fender | eicto: Ok,, you don't seem to understand |
| 15:45.40 | [TK]D-Fender | your goto is broken. |
| 15:45.42 | [TK]D-Fender | Completely broken |
| 15:45.56 | [TK]D-Fender | Stemp 2 is jumping to step 1 immediately and all it does is goes in circles |
| 15:52.22 | eicto | Yes, but it is not mine goto |
| 15:52.35 | eicto | I found it in files generated by freepbx |
| 15:53.45 | [TK]D-Fender | what version? |
| 15:58.02 | eicto | freepbx ? |
| 15:58.24 | [TK]D-Fender | yes |
| 16:00.43 | p3nguin | <eicto> [TK]D-Fender, moment, it spammed console <--- It isn't spam when you ask for it. |
| 16:01.53 | eicto | 2.9.0 |
| 16:02.06 | eicto | [TK]D-Fender, asterisk 1,8.5 |
| 16:02.36 | eicto | strnage that it is works when call from external phone |
| 16:03.30 | wdoekes2 | eicto: some variable is unset, causing it to jump to ",1" <- 1 instead of "somewhere,1" |
| 16:04.36 | eicto | http://www.freepbx.org/forum/freepbx/users/freepbx-looping-forever-when-agi-script-forwards-to-a-queue |
| 16:04.45 | eicto | It seems i found same problem |
| 16:19.05 | *** join/#asterisk nicola_pav (~chatzilla@62.219.238.196) |
| 16:19.59 | nicola_pav | hello. i have an audiocodes mediant 1000. i connected it with asterisk via sip trunk. i am having trouble with the caller id. the call goes from asterisk --> audiocodes --> PRI --> hidden caller id |
| 16:20.08 | nicola_pav | i read that myabe i need to upgrade the firmware |
| 16:21.01 | nicola_pav | anyone familiar with links so i can download audiocodes firmware for mediant 1000? or maybe anyone familiar what should i enable in the audio to enable the caller id? |
| 16:24.44 | *** join/#asterisk francisvgarcia (~networker@190.122.115.211) |
| 16:25.01 | francisvgarcia | Hi, guys |
| 16:27.51 | francisvgarcia | I was trying to set the externaddr option in sip.conf to a dyndns hostname, but I got this error chan_sip.c: Invalid address for externaddr keyword: |
| 16:28.10 | wdoekes2 | externhost= |
| 16:28.19 | wdoekes2 | externaddr/externip takes an ip |
| 16:34.08 | francisvgarcia | wdoekes2 that solved my issue thanks, what happens was that I was having one way audio issues and I generated a blank config |
| 16:34.38 | francisvgarcia | now after i reloaded asterisk I have this output chan_sip.c:29771 peer_iphash_cb: Empty address |
| 16:42.15 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 16:42.51 | *** join/#asterisk m29poff (~m29poff@96.255.90.79.rev.sfr.net) |
| 16:43.49 | m29poff | Hello, is it possible to launch a script when a phone registers ? I've googled about that and found nothing. |
| 16:44.32 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 16:46.37 | p3nguin | francisvgarcia: Pastebin your entire sip.conf |
| 16:46.53 | [TK]D-Fender | externhost != externaddr |
| 16:46.56 | [TK]D-Fender | Wrong parm |
| 16:48.28 | WIMPy | m29poff: Listen on AMI |
| 16:51.04 | mathi | can anyone explain me what the following does? exten => s,1,Set(TIMEOUT(digit)=30) I understand that you can give a variable a value, but give a function a value?? does this make any sense? |
| 16:51.43 | WIMPy | That's half of what functions are there for. |
| 16:51.50 | mathi | WIMPy, I've never seen this in java or php |
| 16:52.25 | m29poff | WIMPy: I thought about that. I'd like to find a more elegant way than listening to AMI |
| 16:53.02 | SeRi | p3nguin! |
| 16:53.08 | SeRi | please be here |
| 16:53.10 | SeRi | lol |
| 16:53.12 | WIMPy | m29poff: If you find another way, it surely won;t be more elegant. |
| 16:53.28 | mathi | what happens with the value 30 exactly? |
| 16:53.48 | WIMPy | 'core show function TIMEOUT' |
| 16:54.46 | [TK]D-Fender | mathi: You sit waiting for input. After the first digit comes in if you wait 30 seconds without dialing more digits it will assume you're finished |
| 16:56.06 | mathi | it's like calling a setter method |
| 16:57.18 | [TK]D-Fender | Or an array element |
| 17:00.10 | SeRi | [TK]D-Fender, do you know a good source for asterisk programing? I am trying to ping out and if it returns 1 go here if returns 0 go to there... |
| 17:02.11 | [TK]D-Fender | SeRi: "core show function SYSTEM" |
| 17:02.20 | [TK]D-Fender | SeRi: Or AGI |
| 17:02.31 | SeRi | [TK]D-Fender, Thank you very much ill look in to it. |
| 17:03.23 | p3nguin | or SHELL() |
| 17:03.44 | SeRi | p3nguin! I been looking for you since Friday. LOL |
| 17:03.58 | p3nguin | That's wasn't so long ago. |
| 17:04.10 | [TK]D-Fender | SHELL, that's what I meant... |
| 17:04.11 | SeRi | I know... just been dramatic :P |
| 17:04.47 | SeRi | Thanks [TK]D-Fender. p3nguin Ill look in to it. |
| 17:05.11 | SeRi | p3nguin, msg me please. |
| 17:23.00 | SeRi | thinking out loud is there a way to tell if the peer is register if is not failover? |
| 17:24.04 | ChannelZ | huh? |
| 17:25.03 | [TK]D-Fender | SeRi: regexten + chanisavail |
| 17:26.12 | SeRi | [TK]D-Fender, Thanks |
| 17:26.52 | SeRi | ChannelZ, I am trying to failover to a gsm gateway when my ITSP is not online for what ever reason |
| 17:27.25 | SeRi | like no Internet which has happen a few times now... |
| 17:28.18 | [TK]D-Fender | SeRi: then your request is backwards |
| 17:28.45 | [TK]D-Fender | seri you register to your ITSP, not the other way around. |
| 17:28.58 | [TK]D-Fender | SeRi: Peers don't register. They get registered to. |
| 17:29.42 | SeRi | Looks like I did had it backwards. sorry. |
| 17:30.08 | SeRi | Thats what I am trying to do. If I am not register than failover to gsm. |
| 17:32.17 | WIMPy | Enable qualify. That should take care of that for you. |
| 17:33.13 | SeRi | WIMPy, I do have them enable |
| 17:33.33 | SeRi | how can I tell it if unavailable to failover using qualify? |
| 17:33.41 | WIMPy | Then Dial should fail immediately when the peer is unreachable. |
| 17:34.14 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:34.15 | WIMPy | You can also use DEVICE_STATE. |
| 17:34.23 | SeRi | Yes that what happens... |
| 17:38.53 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:43.13 | francisvgarcia | wow, do u know any way to exclude all that begging with ; on cat |
| 17:43.46 | p3nguin | Don't cat. Use grep. |
| 17:44.09 | p3nguin | grep -v ^\; file |
| 17:44.20 | francisvgarcia | my sip.conf is too large due to I used the default sample |
| 17:44.53 | p3nguin | No need for cat there. |
| 17:45.09 | ChannelZ | I'm more of a dog person |
| 17:46.07 | p3nguin | grep -v ^\; sip.conf |
| 17:47.07 | francisvgarcia | this command only removes the ;. I only want to see the relevant config with does not start with ; |
| 17:47.24 | p3nguin | grep -v ^\; sip.conf <----- this |
| 17:50.08 | ChannelZ | grep -v ^\\s\*\; sip.conf |
| 17:52.32 | ChannelZ | Though not sure how you'd get it to not show blank lines. Hmm. |
| 17:53.34 | p3nguin | In addition to lines starting with a semi-colon, what does does your command remove? |
| 17:53.51 | ChannelZ | indented comments |
| 17:54.17 | WIMPy | |grep . |
| 17:54.47 | ChannelZ | It's not necessarily ideal.. but the dist sip.conf has tons of explanation over multiple lines in some cases |
| 17:54.53 | p3nguin | I suppose sed, awk, or perl would get rid of blank lines. |
| 17:55.01 | p3nguin | I noticed that. |
| 17:55.10 | ChannelZ | What WIMPy said would work |
| 17:55.14 | ChannelZ | grep -v ^\\s\*\; sip.conf |grep . |
| 17:55.27 | ChannelZ | super compact |
| 17:55.38 | *** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net) |
| 17:58.22 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:58.24 | WIMPy | cut might be a good idea. You could also remove comments from lines with settings. |
| 18:02.57 | *** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net) |
| 18:07.19 | francisvgarcia | grep -v ^\\s\*\; sip.conf that one worked fine but leaves many blank lines |
| 18:11.51 | *** join/#asterisk oej_ (~olle@ns.webway.se) |
| 18:12.25 | ChannelZ | read back a few lines |
| 18:12.42 | ChannelZ | grep -v ^\\s\*\; sip.conf |grep . |
| 18:23.31 | mathi | can anyone explain what is the 4169671111 number ? |
| 18:23.32 | mathi | [incoming] ; a DID coming in on a channel with context=incoming (PRI, SIP, or IAX) |
| 18:23.36 | mathi | exten => 4169671111,1,Goto(main_menu,s,1) |
| 18:24.15 | ChannelZ | That's the extension |
| 18:24.30 | mathi | in this case is it a phone number? |
| 18:24.32 | ChannelZ | IE an incoming call to 4169671111 |
| 18:24.39 | ChannelZ | Presumably |
| 18:24.53 | mathi | you mean an incoming call FROM 4169671111 ? |
| 18:25.01 | ChannelZ | Most ITSPs send calls to the extension matching the DID the call came in on |
| 18:25.05 | ChannelZ | No, to. |
| 18:25.53 | SeRi | under the dial plan defaults can I set default dial plan to execute every time somebody dials out? for example this http://pastebin.com/iRBBRVtE |
| 18:26.40 | ChannelZ | Lets say you have a DID from your provider 5551112222... When someone calls that number, your ITSP sends a call to you TO extension 5551112222 so you know what number the person calling you dialed and can handle the call accordingly |
| 18:26.53 | mathi | ChannelZ, it would be the number of linked to my Asterisk server ? |
| 18:27.07 | ChannelZ | mathi: yeah you could define it that way |
| 18:28.11 | mathi | ChannelZ, the VoIP provides me this number, then people with will call this number with their phones and it will call the script on my server ? |
| 18:28.40 | ChannelZ | SeRi: Not really, you just need to make each of your dialout extensions do what you want. |
| 18:28.56 | SeRi | ChannelZ, Thank you very much. |
| 18:29.25 | ChannelZ | mathi: Basically yes |
| 18:29.52 | mathi | then there is another entry in the same file, but without number: |
| 18:29.53 | mathi | [from-pstn] ; an analog line that has context=from-pstn (typically a DAHDI channel) |
| 18:29.57 | mathi | exten => s,1,Goto(main_menu,s,1) |
| 18:30.04 | mathi | here it is "s" extension |
| 18:30.18 | ChannelZ | Yes |
| 18:30.19 | mathi | why there is no number ? |
| 18:30.50 | Freeaqingme | mathi, hi. Long time no see ;) |
| 18:30.54 | ChannelZ | If you have an analog ATA for instance, it doesn't send the call to any extension. 's' sort of means "no extension" (or some say 'start') |
| 18:31.25 | mathi | Freeaqingme, I was abroad for 6 months:) |
| 18:32.00 | mathi | ChannelZ, why not ? |
| 18:32.22 | ChannelZ | Why not what |
| 18:33.54 | mathi | ChannelZ, how will the call trigger "s" extension? I understand with DID, I have a number |
| 18:34.04 | ChannelZ | It depends on the type of channel |
| 18:34.20 | *** join/#asterisk bluregard (~mattbrei@c-98-228-3-34.hsd1.il.comcast.net) |
| 18:34.28 | ChannelZ | SIP generally (but not always!) comes in to a specific extension, but an analog line has no such concept and winds up in 's' |
| 18:34.36 | bluregard | good afternoon all |
| 18:35.15 | bluregard | has anyone used the Asterisk::AMI perl module in here? |
| 18:35.52 | ChannelZ | Think of it like a URL. if I go to http://amazon.com that's like extension 's', the user wants to contact amazon but doesn't know what specifically... whereas http://amazon.com/myaccount is like extension 'myaccount', the user was more specific about what they wanted |
| 18:37.05 | mathi | ChannelZ, why would someone want to distinguish an analog call (s) or call over IP ? (sorry if I don't use the terms properly :-)) |
| 18:37.13 | ChannelZ | If you have a DID through an ITSP, they (again, 99% of the time but not always) are configured to send the call to an extension matching the DID. If you had an analog card hooked directly to an analog phone line, it would come in to extension 's' |
| 18:38.57 | bluregard | can't you default to 's' if the incoming context doens't have an ext matching the did? |
| 18:39.24 | ChannelZ | Yes and no |
| 18:39.28 | mathi | ChannelZ, why would you want a seperate extension for call over internet and calls through PSTN ? |
| 18:39.45 | ChannelZ | If they provide no extension, it goes to 's' - if they provide an extension that doesn't exist, you can match that with the 'i' exten |
| 18:39.59 | ChannelZ | mathi: you don't necessarily. |
| 18:40.04 | bluregard | yeah, didn't think about 'i' |
| 18:40.39 | ChannelZ | mathi: What I'm saying is that it's different depending on HOW you receive the call.. via SIP over IP, or if you have physical phone lines connected to your system via an analog adapter... or even a T1/E1 |
| 18:41.11 | mathi | ChannelZ, but if they call on your SIP number, how can they go to "i" ? |
| 18:41.27 | ChannelZ | If they specify an extension that doesn't exist |
| 18:42.09 | mathi | ChannelZ, but in the case above the extension was the SIP number, so how can they redirect on my server if they call another number |
| 18:42.34 | ChannelZ | You're getting confused - and it is a bit confusing |
| 18:42.46 | bluregard | mathi: do you have the O'reilly asterisk book? |
| 18:42.53 | mathi | I'm reading it :) |
| 18:43.21 | bluregard | best thing you can do is build an asterisk box while reading it. |
| 18:43.31 | ChannelZ | If you have a phone number through an ITSP and that is all, all incoming calls should wind up on an extension matching your phone number |
| 18:43.43 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 18:44.26 | ChannelZ | However you could also accept SIP-to-SIP calls -- that is, a call that didn't come from the public telephone system, but over the internet from another SIP-compliant device; your phone number is not involved. |
| 18:44.38 | ChannelZ | In that case they can try to call any extension they can make up, or none at all. |
| 18:45.18 | ChannelZ | But you can turn that off if you don't need/want "anonymous" SIP calls and not worry about it |
| 18:46.07 | mathi | yea I get it now ... I understand how a SIP phone can call a specific extension, I have already configured phones calling each other and extensions. I'm just confused when it comes to calls over the Internet. As long as I don't share my server details, they can call only one extension, which is my SIP number? But i'm new to this SIP thing |
| 18:46.47 | ChannelZ | if you do "allowguest = no" in sip.conf, they can't call you at all |
| 18:47.02 | ChannelZ | Only calls from peers matched in your sip.conf will be allowed |
| 18:47.16 | ChannelZ | (IE your ITSP, or your local phones you've setup, etc.) |
| 18:47.31 | mathi | in my case the only incoming calls I'll have are external calls, so I need to allow it |
| 18:47.54 | ChannelZ | well, no... |
| 18:48.48 | mathi | no? |
| 18:49.09 | ChannelZ | If you are signed up to a VoIP service who is providing you with a DID (a phone number) those are 'authenticated' calls |
| 18:49.33 | ChannelZ | You have to configure a peer for them in sip.conf to accept the calls (and to place calls out) |
| 18:50.15 | ChannelZ | allowguest is for SIP calls that come in from unmatched peers |
| 18:51.09 | mathi | ChannelZ, anybody may call |
| 18:51.36 | ChannelZ | But you're confusing a SIP call with someone calling a regular phone number |
| 18:52.32 | ChannelZ | Anyone may call your phone number, sure |
| 18:52.37 | mathi | ok, probably I don't need SIP at all? |
| 18:52.58 | ChannelZ | you do |
| 18:53.05 | mathi | I may only receive calls from a regular phone number |
| 18:53.15 | mathi | so why do I need SIP? |
| 18:53.24 | ChannelZ | Yes but do you have an analog card in your computer with a phone line physically plugged into it? |
| 18:53.46 | ChannelZ | A TDM card or some other ATA? |
| 18:53.51 | mathi | not yet.. |
| 18:54.32 | ChannelZ | The difference is whether or not you are going to interface with the phone system directly at your location, or use an ITSP as a bridge |
| 18:55.19 | mathi | I have two options here? |
| 18:55.32 | ChannelZ | At minimum :) |
| 18:55.40 | mathi | so which one should I go for? |
| 18:55.44 | ChannelZ | Let's go back a little |
| 18:56.40 | mathi | ok:) |
| 18:57.17 | ChannelZ | An ITSP is a middle-man who handles the actual call origination/termination on the public phone system. You get a phone number, but that phone number actually routes to THEIR hardware. They then "convert" that call to SIP and send it to you over the internet. |
| 18:57.43 | ChannelZ | The reverse is also true, if you want to place a call, you talk over the net to your ITSP who then bridges it onto the actual phone system |
| 18:58.11 | mathi | yes clear |
| 18:58.55 | ChannelZ | The other way to do it is for YOU to run the hardware and have the physical phone lines terminating on YOUR machine... like plugging a phone into your wall, except you want to plug Asterisk into it. |
| 18:59.36 | mathi | ok, this is done with ISDN right? |
| 18:59.39 | ChannelZ | In that case you need to buy extra hardware, an ATA in the case of analog phone lines, to interface the actual phone lines to your computer. |
| 18:59.59 | ChannelZ | It could be ISDN too yes, but it's the same thing, you need to buy extra hardware to do it |
| 19:00.38 | mathi | in the first case, i'll have data transmitted with SIP protocol, and in the latter case, what protocol will be used? |
| 19:00.42 | ChannelZ | Could be another high capacity PRI like a T1 or E1. Digital interface cards vs analog interface cards, but same idea. |
| 19:01.09 | ChannelZ | In the latter, there's no protocol you need to worry about really |
| 19:01.43 | ChannelZ | DAHDI is the interface between Asterisk and telephony hardware, so your calls come in/go out on DAHDI channels rather than SIP channels |
| 19:03.30 | mathi | ChannelZ, if someone calls with to my phone number, and I'm using ITSP, do you consider that as an SIP call ? |
| 19:03.52 | *** join/#asterisk johaidi (~johaidi@94.30.149.10) |
| 19:04.14 | ChannelZ | From your point of view, yes |
| 19:04.38 | mathi | you said earlier: <ChannelZ> But you're confusing a SIP call with someone calling a regular phone number |
| 19:04.46 | mathi | looks like you didn't consider as an SIP call |
| 19:05.51 | ChannelZ | The distinction I was trying to make was between a SIP call coming to you from your ITSP as a result of someone dialing your phone number, versus a "SIP-to-SIP" call coming from some random person on the internet |
| 19:06.52 | SeRi | ChannelZ, I have this: http://pastebin.com/1qbB92nb but asterisk is telling me that my context in ExecIf is deprecated... Can you take a look and see what I have wrong. I thought I was following it the right way. |
| 19:06.56 | ChannelZ | That is, you could setup a softphone on your computer right now and call me via SIP over the net, the phone system completely un-involved. |
| 19:08.50 | ChannelZ | SeRi: your ExecIf syntax is just wrong |
| 19:09.13 | mathi | ChannelZ, if I accept calls from anybody, how can I add the peers in sip.conf? I can't, thus I need to allow unauthenticated calls, though, you told me not |
| 19:09.54 | ChannelZ | mathi: right, only if you WANT to receive unauthenticated calls though. And you have to be careful about this when constructing your dialplan |
| 19:10.31 | ChannelZ | mathi: I thought you were perhaps confusing an unathenticated SIP call with someone you don't know calling your actual phone number, if you were receiving those calls via SIP from an ITSP |
| 19:10.38 | mathi | ChannelZ, why wouldn't you want to? only in the case where you want only local calls in your office. Maybe I missed a point here? |
| 19:12.26 | ChannelZ | SeRi: I think you want something more like ExecIf($["${SIPPEER(${PEERCHECK1},status)}"="OK"]?Set(TRUNKCHECK=1)) |
| 19:12.57 | ChannelZ | mathi: I don't know.. some people don't want to accept SIP calls from just anyone over the net |
| 19:13.08 | ChannelZ | mathi: It's up to you really |
| 19:13.16 | SeRi | ChannelZ, Thank you very much. I was looking threw the book and trying to understand my mistake. |
| 19:13.26 | mathi | ChannelZ, I don't understand, it's like receiving a call from someone's phone |
| 19:14.28 | ChannelZ | mathi: yes, but we're talking about the Internet here... lots of kiddies out there trying to exploit your system, to try and make actual phone calls on your dime. |
| 19:15.03 | ChannelZ | Some people don't want to mess with it and just shut off external SIP, by denying anonymous calls or even via a firewall |
| 19:16.00 | mathi | ChannelZ, so to go back with what was confusing... I was saying that if I accept unauthenticated calls (and I don't get incoming calls locally, only externally PSTN or over internet (SIP)) then I will never go to "i" extension that you suggested, because they can only dial my number to connect to the asterisk server |
| 19:17.15 | ChannelZ | If you accept unathenticated calls via SIP, any number of things might happen. 1. If the person calling didn't specify an extension, it would try to go to extension 's'. If they _did_ provide an extension but it was one that you didn't configure, it will try to go to extension 'i' (invalid). |
| 19:17.48 | ChannelZ | 3. If they provided an extension that you did have configured, it would go to that extension. |
| 19:19.15 | ChannelZ | If you are receiving public phone calls via an ITSP they should (generally) all be directed to an extension matching your phone number, as 99% of ITSPs are configured to do (so you have a means to figure out what number someone dialed to call you if you had 10 different phone numbers through them.. like different departments of a business for example) |
| 19:19.40 | mathi | here comes the confusing, ready?:)) how can they call another extension than the extension labelled by my number ? Let's say my number(provided by ISTP) is "4169671111", how can they reach my server other than calling that number which will trigger the following extension: exten => 4169671111,1,Goto(main_menu,s,1) |
| 19:19.58 | ChannelZ | In that specific case, they can't. |
| 19:20.11 | mathi | well, that's what I was saying for so long :P |
| 19:20.46 | ChannelZ | Yes but then we talked about anonymous SIP (allowguest) |
| 19:21.28 | ChannelZ | Which is where it seemed like you were a bit confused between the two |
| 19:22.12 | mathi | ChannelZ, maybe i've misread but in the book it was written that SIP might replace PSTDN, though you tell me that people might prefer to refuse calls coming from the internet. It's kind of contradictory |
| 19:22.28 | ChannelZ | no |
| 19:22.54 | ChannelZ | "from the internet" versus "from a traditional phone" |
| 19:23.15 | ChannelZ | They both might come to you OVER the internet (via an ITSP) but they are two different things. |
| 19:24.15 | ChannelZ | Let me put it this way.... the ITSP is a bridge from the traditional phone system to the internet |
| 19:24.20 | mathi | yes |
| 19:24.58 | ChannelZ | But you can also make "internet-to-internet" calls that don't involve your ITSP or the traditional phone system at all. |
| 19:25.44 | ChannelZ | So two people similarly equipped with SIP phones or softphones can call each other anywhere in the world for just the normal costs of their internet connections. |
| 19:25.48 | ChannelZ | Just like sending an email |
| 19:26.08 | mathi | but if I use ITSP and convert call to SIP, I have no way anymore to detect if it's from Internet ? It will never go to "s" extension in this case? Thus I won't be able to unauthorized these "kids" trying to exploit the system |
| 19:26.11 | ChannelZ | That is the distinction I was making. |
| 19:26.29 | ChannelZ | If you block anonymous SIP calls yes |
| 19:27.00 | mathi | how is this done ? |
| 19:27.12 | ChannelZ | allowguest=no in sip.conf |
| 19:28.10 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
| 19:28.25 | mathi | ChannelZ, but how will I receive external calls in this case? as if I use ITSP, all calls will be using SIP and I need allow anyone |
| 19:28.39 | ChannelZ | no |
| 19:28.47 | ChannelZ | You would need SIP but you accept calls from your ITSP. |
| 19:28.54 | SeRi | well it worked. With some adjustment of my typos etc.... Just unpluged my internet and went out threw my gsm. Thanks for the point outs chalcedony |
| 19:28.55 | mathi | aaaahhhh |
| 19:29.02 | SeRi | I mean ChannelZ |
| 19:29.04 | SeRi | damn it |
| 19:29.05 | SeRi | lol |
| 19:29.07 | SeRi | sorry |
| 19:29.21 | mathi | ChannelZ, it's the ITSP that detects PSTN call or SIP call, and send the information to me ? |
| 19:29.39 | ChannelZ | Yes |
| 19:30.02 | mathi | ChannelZ, we found the point that was contantly confusing)))) |
| 19:30.20 | ChannelZ | heh |
| 19:30.28 | ChannelZ | SeRi: glad you got it working |
| 19:30.53 | mathi | ChannelZ, i need to eat, I will come back later |
| 19:30.59 | mathi | thanks a lot for your help!!! |
| 19:31.00 | ChannelZ | me too |
| 19:31.08 | mathi | bon appétit |
| 19:31.10 | ChannelZ | Sure. I should have just scribbled out an illustration |
| 19:31.56 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 19:33.37 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 19:34.58 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 19:44.39 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 19:58.20 | *** join/#asterisk beccara (~beccara@mail.ubergroup.co.nz) |
| 20:00.32 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:d809:c1d7:4710:855e) |
| 20:03.32 | mathi | ChannelZ, are you here? |
| 20:07.33 | *** join/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
| 20:09.12 | lhfnet | Hello, I have problems trying to configure an MFC r2 with dahdi and asterisk 1.8, the command dahdi_genconf detect the E1 as ccs but when I change it to cas the card light shutdowns but dahdi said that it is ok |
| 20:52.22 | [TK]D-Fender | mathi: Yout ITSP is a user. They get the call from the PSTN and bridge in a cal to you via SIP (in this case). The call comes from them... with the callerID of the PSTN call that started this all. |
| 20:57.08 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 20:57.35 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 21:02.03 | ChannelZ | mmmm burrito |
| 21:07.04 | mathi | [TK]D-Fender, yes I got it now :) |
| 21:07.23 | *** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net) |
| 21:07.24 | mathi | when talking about ISTP provider and VoIP provider, are these the same thing ? |
| 21:09.34 | SeRi | ITSP (Internet Telephony Service Provider) |
| 21:09.49 | SeRi | VoIP (Voice Over IP) |
| 21:09.55 | mathi | i know |
| 21:10.16 | SeRi | Two different things in context but people usually use them together |
| 21:10.20 | mathi | are we talking about the same provider when people say "ISTP provider" and "VoIP provider" |
| 21:10.45 | SeRi | Yes |
| 21:11.55 | mathi | thanks:) |
| 21:12.54 | SeRi | Some of the more experience guys will tend to disagree because they like to be exact when using the two. most us of who use them the same way are those who are still learning. like me :) |
| 21:13.09 | SeRi | of us* |
| 21:13.55 | mathi | I don't know wether I should use ITSPSIP or DAHDI, how do I choose the best solution in my case? On what does it depend |
| 21:14.19 | mathi | *ITSP/SIP |
| 21:14.39 | mathi | ChannelZ, maybe you can advice on that? |
| 21:14.55 | ChannelZ | sorry not paying attention. reading... |
| 21:15.11 | SeRi | Dahdi is not a provider. at least that I know of. |
| 21:15.18 | mathi | ChannelZ, about the two possible ways you described to me earlier :) |
| 21:15.50 | ChannelZ | It just depends on how involved you want to get.. having to buy/run (possibly expensive) telephony equipment yourself, vs a good internet connection |
| 21:15.51 | mathi | I know, the question is, should I have a ISTP/SIP provider, or should I plug extra hardware myself and use DAHDI |
| 21:16.24 | ChannelZ | You're probably going to spend a few hundred bucks for an interface card if you do it yourself, depending |
| 21:16.37 | mathi | ChannelZ, I want the minimal cost, because I have to deploy the system at several workplaces |
| 21:17.11 | ChannelZ | It's not necessarily a question we can answer for you. |
| 21:17.22 | SeRi | ^^ :) |
| 21:17.40 | [TK]D-Fender | [16:09]SeRiITSP (Internet Telephony Service Provider) <- Saying ITSP Provider is redundant |
| 21:18.24 | SeRi | jus defining it. |
| 21:18.38 | mathi | ChannelZ, how good should the internet connection be ? |
| 21:18.42 | [TK]D-Fender | SeRi: Wasn't your mistake, was copying the breakdown |
| 21:18.45 | SeRi | but yes it is redundant like saying NIC Card |
| 21:18.53 | [TK]D-Fender | Indeed |
| 21:18.54 | ChannelZ | Depends on how many simultaneous calls you need to make/receive |
| 21:19.03 | mathi | ChannelZ, not more than 5 |
| 21:19.09 | ChannelZ | Latency is a bit more important than massive bandwidth |
| 21:19.27 | ChannelZ | And reliability.. if you have bad access that doesn't work more often than it does, probably not a good idea :) |
| 21:19.29 | [TK]D-Fender | mathi: For your needs depends what you want and trust. If your internet is unreliable maybe you'll want a direct PSTN link. |
| 21:20.02 | [TK]D-Fender | mathi: Also depends on costs. What equipment you already have or are planning on getting and a bunch of other ideas. |
| 21:20.13 | SeRi | [TK]D-Fender, I would like to thank you again. The Polycom has been solid so far. I have the xml files and config files set for me and I have to say its very simple. |
| 21:20.29 | [TK]D-Fender | SeRi: You're welcome... |
| 21:20.50 | [TK]D-Fender | SeRi: Was just a little bit of a fight to get past that lock. |
| 21:21.24 | SeRi | yea and to think that its been like that for over 5+ years... lol silly... |
| 21:21.31 | mathi | internet connections usually restart every 36 hours, so when it restarts nobody may call... |
| 21:22.17 | ChannelZ | Anyone on Comcast business? |
| 21:22.19 | [TK]D-Fender | mathi: Resets are an issue, jitter, latency, ISP blocking (assholes), PL, etc |
| 21:22.23 | SeRi | I am ChannelZ |
| 21:22.53 | ChannelZ | Do you know what modem you have? |
| 21:22.54 | SeRi | I have my remote office in CB |
| 21:23.49 | mathi | [TK]D-Fender, what can ISP block that would bother me? |
| 21:24.03 | SeRi | ChannelZ, is a SMC piece of crap that is broken most of the tme |
| 21:24.14 | ChannelZ | mathi: SIP traffic :) |
| 21:24.15 | SeRi | Its actually a SMC/Modem/Router |
| 21:24.40 | SeRi | I have convince them to let me have bridge mode on it because its so problematic |
| 21:25.00 | ChannelZ | SeRi: yeah.. I think my "downstream power" is low but I'm not sure what's normal, never payed attention before. Wondered what other people are seeing. |
| 21:25.05 | mathi | ChannelZ, if I have a ISTP, I keep my ISP for internet connection? ISTP is only managing the convert PSTN call to SIP ? |
| 21:25.36 | [TK]D-Fender | mathi: Correct |
| 21:25.50 | SeRi | ChannelZ, you ever heard of dslrports.com? |
| 21:26.00 | ChannelZ | yeah |
| 21:26.10 | mathi | how much will an ISTP charge me average ? |
| 21:26.10 | ChannelZ | mathi: right |
| 21:26.29 | SeRi | go there and got o the ComCast Direct forum. Let ComcastSteve help you |
| 21:26.35 | ChannelZ | Usually some dollars a month, plus cents/min for calls |
| 21:26.35 | SeRi | ChannelZ, ^^ |
| 21:27.08 | ChannelZ | I need to just call them, was just curious. Seems like at least once or twice a day my connection will drop for a minute |
| 21:27.30 | ChannelZ | I have problems with someone else in the neighborhood hooking shitty equipment up and it screws everyone around here up |
| 21:28.14 | SeRi | That sucks. My experience with tech supports has been very bad so I go there for help and the are very helpful and fast. |
| 21:28.26 | mathi | ChannelZ, how do I find sucha provider? should it be a company in my country ? |
| 21:28.32 | SeRi | go cut there cable :P |
| 21:28.56 | ChannelZ | I haven't had to call them much but they've been responsive when I have. Problem is they always want to send someone out to my house, and I'm never here |
| 21:29.06 | ChannelZ | If I knew who it was, I would :/ |
| 21:29.13 | ChannelZ | mathi: what country? |
| 21:29.17 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 21:29.19 | mathi | belgium |
| 21:30.05 | ChannelZ | ah.. well yes there should be some, Google |
| 21:30.39 | ChannelZ | Though for instance if you wanted to make/receive calls in the US cheaply, you could get an account from a US ITSP too |
| 21:32.41 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 21:33.59 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:38.15 | mathi | ChannelZ, what card would I need in the case I go for dahdi? I won't have more than 5 calls at the same time |
| 21:38.40 | ChannelZ | depends what technology, you said ISDN before? |
| 21:39.24 | mathi | I guess I don't need ISDN |
| 21:39.39 | ChannelZ | Just POTS then? |
| 21:40.00 | mathi | I think so |
| 21:40.22 | ChannelZ | http://www.digium.com/en/products/hardware/analog |
| 21:40.26 | mathi | I just need to accept external calls to my Asterisk server, I'll have just a single number |
| 21:41.26 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 21:42.11 | ChannelZ | I don't know what the telcos are like in Belgium, but at least in the US, unless you use a LOT of minutes, it's usually much cheaper to use an ITSP |
| 21:42.59 | mathi | ChannelZ, but over the months you will pay more using ITSP. an analog card, you buy only once and that's all you pay, no ? |
| 21:43.11 | ChannelZ | I pay $4/mo plus $.01/min for calls.. I don't use my home phone that much so I maybe pay $5/mo average, compared to over $35 for a single POTS line from the phone company with Caller ID |
| 21:43.28 | ChannelZ | Well, you're still paying the telco for the phone lines |
| 21:44.21 | mathi | uhm, I don't know .. do I?) I don't have a phone |
| 21:44.51 | ChannelZ | well you'd have to get phone lines if you wanted to buy a card and do it yourself - it doesn't work by magic :) |
| 21:46.39 | mathi | ChannelZ, I'll explain you what I need to do: I need to install an IVR system at doctors workplaces, people will book appointments through IVR, but it still should be possible to call the doctor drectly. So I need to connect their phone to Asterisk somehow, in case caller has for example urgent problem |
| 21:47.16 | mathi | that's why I thought in this case dahdi/analog card would best fit ? if not, I should ask them to change their phones as well ... |
| 21:47.20 | mathi | is that right? |
| 21:47.42 | ChannelZ | Yeah you probably don't want them to have to be involved in the technology side on their end |
| 21:48.27 | mathi | yes, and for me it will be one more hassle if there is a provider, one more thing to handle |
| 21:49.03 | ChannelZ | If a person is to call you to make these automated appointments, but then you need to be able to connect them to a doctors office, you'd need as many phone lines/channels to make additional outgoing calls as there might be people doing that simultaneously. |
| 21:49.53 | ChannelZ | The nice thing about doing it through an ITSP is they have boatloads of channels so you just pay for what you use instead of having to have 15 phone lines, or pay for a T1/E1 when most might be idle most of the time. |
| 21:50.51 | ChannelZ | But you'd really have to do the math based on what you think your needs are, how much telcos cost in your area, versus how much an ITSP would charge, etc. |
| 21:51.59 | ChannelZ | There's pros and cons to both really |
| 21:52.07 | ChannelZ | BRB bathroom break |
| 21:53.44 | mathi | ChannelZ, why would I need additional outgoing calls? |
| 22:00.22 | mathi | you meant additional ingoing calls ? |
| 22:02.25 | ChannelZ | No, you said you wanted the ability for the person calling in to be connected to the doctors office directly? |
| 22:03.46 | mathi | ChannelZ, yes, if urgent need then yes. but anyway, the doctor can only answer to one call at the time, that's why I didn't understand why you were suggesting several telephone lines |
| 22:04.55 | ChannelZ | You need one line/channel for the person calling in to your IVR, then another line/channel for it to dial back out to the doctor if they hit that. |
| 22:05.44 | mathi | ChannelZ, well the minimum here is 4 port: http://www.digium.com/en/products/hardware/analog |
| 22:06.01 | mathi | so I can connect 4 lines to it |
| 22:06.27 | ChannelZ | Right, but if 2 people call in and both hit 'talk to the doctor', all 4 channels are then in use. |
| 22:07.16 | ChannelZ | Unless you're wanting to install a separate Asterisk system in every office, in which case 'call the doctor' is local to the machine... but that would assume SIP phones.. |
| 22:07.30 | mathi | i thought if i have one line in the Asterisk, several people can use the IVR system |
| 22:07.37 | ChannelZ | I guess I was assuming you wanted to centralize the IVR for multiple offices, maybe a bad assumption on my part |
| 22:07.47 | *** join/#asterisk blerp (~blerp@S0106000c42bcfc93.cg.shawcable.net) |
| 22:08.04 | ChannelZ | If you only have one physical phone line into the system, only one person could call in |
| 22:08.18 | mathi | (I don't plan to centralize) |
| 22:08.48 | ChannelZ | ok.. well in that case you need a card with as many channels as phone lines the particular office has (to keep the same capacity) |
| 22:09.24 | ChannelZ | But then you either need additional channels to interface with their existing in-office PBX, or to connect their existing analog phones directly to Asterisk, or have to switch them all over to SIP phones internally and build a network for them |
| 22:09.33 | mathi | why would that assume SIP phones? I can use the analog card to plug the phone in |
| 22:09.36 | ChannelZ | (if they don't already have a data network in-office) |
| 22:10.08 | ChannelZ | Yes you can use analog phones.. but now you need a bigger capacity card with as many channels as phones you need. |
| 22:10.38 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:10.49 | ChannelZ | If there are 10 phones in the office (receptionist, phones in several offices, a lobby phone, etc.) you need a card with at least 10 ports, plus however many more for the phone lines to the outside world |
| 22:11.01 | mathi | so if I want to be able to have 5 people calling at the same time into the Asterisk server, it will use 5 ports ??? |
| 22:11.09 | ChannelZ | Yes |
| 22:11.51 | mathi | and what do I need to plug into the port for the outside world? |
| 22:11.52 | ChannelZ | ...and 5 phone lines through the telco |
| 22:12.01 | mathi | oh my god |
| 22:12.43 | [TK]D-Fender | That's right... the math IS that simple :) |
| 22:12.44 | ChannelZ | These doctors offices presumably already have 1 or more phone lines going to them though, I'd assume |
| 22:13.04 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 22:13.17 | mathi | why ? I rather assume that they just have one phone line, as they can answer only one call |
| 22:13.42 | ChannelZ | That might be true |
| 22:14.06 | mathi | but indeed now i'm busy with an office having 4 doctors, I have no idea about their phone system |
| 22:14.23 | ChannelZ | But I'd think not, what happens if someone is calling in to make an appointment and a doctor or nurse has to call out to check a prescription? |
| 22:15.02 | mathi | doctors work all differently |
| 22:15.43 | ChannelZ | Sure, but I'm not sure I'd go to a doctor with only one phone line! An emergency nightmare |
| 22:16.19 | ChannelZ | I am a small business with only 2 employees and we have 4 lines |
| 22:17.52 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:19.14 | mathi | ChannelZ, still with the ITSP, problem with reliabilty, that's not acceptable when a patient wants to make an appointment. However, the Asterisk IVR system will have anyway to connect to a webservice to book the appointment as the calendar is online ... |
| 22:19.40 | *** join/#asterisk beccara (~beccara@mail.ubergroup.co.nz) |
| 22:20.06 | ChannelZ | Well, the world is not perfect. |
| 22:22.21 | mathi | ChannelZ, so what would you do in my case. If I understand, drawbacks of Analog card=expensive,need to install many phone lines to support ingoing simulatenous calls; and of SIP=need to change their phones to SIP phone, reliability |
| 22:22.43 | ChannelZ | All I can say is that any doctors office or clinic I've ever been in has had a PBX of some sort.. probably with at MINIMUM 2 phone lines to the outside world, and half a dozen internal extensions |
| 22:23.21 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:23.52 | mathi | you libe in America |
| 22:24.19 | mathi | *live |
| 22:24.31 | mathi | maybe the offices have more advanced technology |
| 22:24.36 | ChannelZ | The offices surely already have external phone lines; So to replace any existing system you'd need a card with at least as many ports, PLUS as many internal phones as they have to re-use as much of their existing infastructure as possible |
| 22:25.19 | ChannelZ | If this is a village doctor with 1 phone line and no other phones in the office, it obviously becomes an easier setup :) |
| 22:26.56 | mathi | ChannelZ, this card with 4 ports costs already $500+ |
| 22:27.05 | ChannelZ | Yup |
| 22:27.28 | ChannelZ | There are some cheaper clones and other companies but I can't speak to them, no experience with them |
| 22:28.02 | ChannelZ | Analog telephony sucks |
| 22:28.06 | mathi | ChannelZ, do you use ITSP in your business? |
| 22:28.09 | ChannelZ | But there's a cost involved either way |
| 22:28.37 | mathi | ChannelZ, well it is more reliable than SIP, why do you say it sucks |
| 22:28.41 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 22:29.27 | ChannelZ | Not at the moment; I have 4 POTS lines using a TDM800P for incoming/outgoing, and then about 12 SIP phones for local extensions |
| 22:30.14 | [TK]D-Fender | mathi: of SIP=need to change their phones to SIP phone, reliability <- no |
| 22:30.17 | ChannelZ | I might switch over to VoIP for my incoming/outgoing when we move next year, not sure. I DO use an ITSP for my home phone though |
| 22:30.34 | [TK]D-Fender | mathi: What you choose to use for phones has no relation to how you connect to the PSTN |
| 22:31.15 | mathi | [TK]D-Fender, I was talking about drawbacks, and I wanted to say NO reliability |
| 22:31.42 | mathi | [TK]D-Fender, by the way, did you read? any comments on which solution you would go for for my docto'rs office ? ) |
| 22:31.42 | [TK]D-Fender | mathi: "change to SIP phones" <- not required |
| 22:32.15 | mathi | [TK]D-Fender, why not required ?? If I have ITSP how do I connect the usual phone to my Asterisk server ?? |
| 22:32.29 | [TK]D-Fender | mathi: Your requirements are a little vague for me |
| 22:33.00 | mathi | I didn't get that: <[TK]D-Fender> mathi: "change to SIP phones" <- not required |
| 22:33.05 | ChannelZ | I used SIP phones in my office because it was a startup, I had to buy all the phones no matter what. If I had an existing infastructure, usually analog phones, I'd have to have bought a larger capacity card to connect all the phones to * --- nevermind my connection to the outside world |
| 22:33.59 | [TK]D-Fender | mathi: What don't you get? You said that you understood that you will HAVE to change your phones. I'm saying this is incorrect |
| 22:34.20 | *** join/#asterisk F2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 22:34.58 | mathi | [TK]D-Fender, if I go for ITSP, how do I connect the usual phone to the asterisk server? that's what I don't get. I thought I *had* to change their phone to SIP phone |
| 22:35.09 | F2Knight | I created a script for converting sip.conf to asteriskRT if anyone is interested. http://www.enetworkservices.net/wordpress/sip-2-mysql-db |
| 22:35.34 | ChannelZ | mathi: if they already have analog phones, you'd need an ATA (one of those TDM cards) |
| 22:35.50 | [TK]D-Fender | mathi: Same kind of TDM card can let you plug in PHONES depending on wath modules you get. Or you could use ATA's for regular phones. |
| 22:36.05 | [TK]D-Fender | mathi: Linksys PAP2T-NA for example |
| 22:36.27 | mathi | yes, but if I go for ITSP it would be to avoidf buying a card! |
| 22:36.28 | F2Knight | mathi, depending on how many lines they need you could also use something like spidermux |
| 22:36.44 | [TK]D-Fender | mathi: A card is ONE of the options I gave you. |
| 22:36.48 | ChannelZ | mathi: only the need for the EXTERNAL lines |
| 22:37.52 | ChannelZ | There are two halves to this system. One is your connection from Asterisk to the PSTN, the other is the internal phones connected to Asterisk |
| 22:39.01 | ChannelZ | Your Asterisk-PSTN connection could be via Internet using an ITSP; No telephony hardware needed for that. But the "local" side, the local phones, either need an analog interface card in your Asterisk OR SIP phones that talk over Ethernet to it. |
| 22:39.18 | ChannelZ | Which is why I said there's some cost either way, whether in ATAs or new phones. |
| 22:39.33 | F2Knight | or softphones that work on a computer. |
| 22:39.45 | ChannelZ | Sure |
| 22:40.03 | ChannelZ | But, not really ideal for the laymen |
| 22:40.53 | F2Knight | I don't know ChannelZ, I have had a lot of success with people understanding the softphone, and liking it more then the 'hardphone' |
| 22:41.16 | mathi | ChannelZ, it looks much less expensive than a card!! http://www.voipsupply.com/linksys-pap2t-na |
| 22:42.05 | F2Knight | mathi, and you can get them even cheaper then that if you look a little |
| 22:42.11 | ChannelZ | mathi: yes but if you have 10 phoens you need 5 of those. |
| 22:42.34 | F2Knight | how many phone mathi |
| 22:42.35 | ChannelZ | If you only have one or two in a location then yeah that's cheaper |
| 22:43.17 | mathi | I want 1 phone, and at least 2-3 to the external world |
| 22:44.06 | ChannelZ | assuming VoIP for the external world, then you just need 1 |
| 22:44.26 | [TK]D-Fender | matYou have 4 doctors you want all sharing the same PHONE? |
| 22:44.27 | F2Knight | whats this 2-3 to the external world? phones or ability to call 2-3 numbers at the same time.? |
| 22:44.59 | ChannelZ | Incoming for an IVR. Presumably only one outgoing call if there is only one physical phone :) |
| 22:45.07 | mathi | only one doctor is wanting to work with me out of the 4 |
| 22:45.24 | mathi | 2-3 calls to IVR system |
| 22:45.28 | mathi | simultaneously |
| 22:45.41 | ChannelZ | mathi: but the other 3 still have to be able to receive/make calls...right? |
| 22:45.45 | F2Knight | okay so you need 1 trunk with 3 channels. |
| 22:45.49 | mathi | ChannelZ, yes, omg is that a problem? :) |
| 22:46.07 | [TK]D-Fender | mathi: 1 PAP2 it is then.... |
| 22:46.10 | ChannelZ | Well unless they all have discreet lines already |
| 22:46.10 | F2Knight | well it will be a problem if all 3 want to use the phone and you only have 1 phone :) |
| 22:46.29 | [TK]D-Fender | F2Knight: Only 1 want's to work with him he says... |
| 22:46.37 | mathi | I guess they have different lines, as they work independantly |
| 22:46.40 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 22:46.50 | ChannelZ | ok then yeah you can forget about them |
| 22:47.08 | F2Knight | then if only 1 wants to work with them then he only needs to make one call .. |
| 22:47.41 | mathi | [TK]D-Fender, actually I don't even need 2 ports, one is enough I guess |
| 22:47.42 | F2Knight | mathi, I assume you are a doctor in the office? |
| 22:48.05 | [TK]D-Fender | mathi: There is (was?) a 1-port model but it was only a little bit cheaper Not worth skimping on |
| 22:48.09 | mathi | F2Knight, no I created his online calendar, and I want to make an IVR system that interacts with it |
| 22:48.24 | F2Knight | mathi, ahh okay. |
| 22:48.31 | ChannelZ | The 3102 has 1 FXO and 1 FXS |
| 22:48.56 | F2Knight | mathi, Your asterisk box... does it reside at the clients location? |
| 22:49.27 | mathi | [TK]D-Fender, would you suggest me to go for ITSP for ingoing calls to IVR or to invest in a card ? I guess i'll need to test the reliability of the SIP service |
| 22:49.35 | mathi | F2Knight, yes |
| 22:49.55 | F2Knight | mathi, reason I ask is that if you have built a calendring app, you presumably are hosting this service out side of the office.. and if so you could use your own asterisk server, and just provide them with the pap2 and have it connect to your box |
| 22:50.44 | mathi | F2Knight, an Asterisk server at my own premises you mean ? |
| 22:50.50 | F2Knight | mathi, where is your 'calendaring app' housed at? |
| 22:51.36 | ChannelZ | The only problem with doing that is now the system is dependent on 2 people's own internet connections (plus the ITSP really..) |
| 22:52.29 | F2Knight | ChannelZ, but if this 'app' is centearlized... it is probably being used by multiple places right mathi? |
| 22:52.57 | F2Knight | and that being the case, then that location is required to be up. |
| 22:53.01 | mathi | F2Knight, I can't have my own Asterisk server, it's too expensive (not only hardware, but also when you consider availability 24/7, reliability). if the internet connection fails at the client's place, i'll blame the client |
| 22:53.02 | ChannelZ | That was the impression I got earlier |
| 22:53.44 | F2Knight | but your app is stored where.? the app that they access is located where exactly? |
| 22:53.51 | ChannelZ | But then we went down the road of having to have enough capacity to make outgoing calls back to the doctors office, and then that wasn't acceptable, etc.... |
| 22:54.33 | mathi | online calendar is hosted at a webhosting company |
| 22:54.43 | F2Knight | okay so this hosting company... |
| 22:54.51 | F2Knight | do you have a dedicated box? |
| 22:55.00 | F2Knight | how many people do you have using this 'app' |
| 22:55.27 | ChannelZ | this is getting too cyclical for me |
| 22:55.33 | mathi | for now it is a server which is hosting other web applications, shared server |
| 22:55.33 | F2Knight | because this is where you would idealy want to put your 'asterisk' server.. they have the redundant internet already |
| 22:55.58 | mathi | yes, but the server needs special hardware, I don't know any hosting company who provides this |
| 22:56.12 | F2Knight | okay so if you put a single asterisk server there.. commodity PC even, the server needs no special hardware. |
| 22:56.40 | F2Knight | I have 10 asterisk boxes and all are single 1U rackmount servers with nothing but an ethernet interface. |
| 22:57.26 | ChannelZ | There's 50 ways to set this up |
| 22:57.28 | mathi | so you are using ITSP to make this work |
| 22:57.37 | F2Knight | Yep |
| 22:58.03 | mathi | what is an asterisk box? you mean server? |
| 22:58.21 | ChannelZ | yah |
| 22:58.25 | F2Knight | I have one spidermux at an office providing 'pots' service to them , instead of using pap2's and I have small offices that use pap2s |
| 22:58.52 | ChannelZ | F2Knight: ahh thanks I couldn't remember the names of those things |
| 22:59.29 | F2Knight | the spidermux? |
| 22:59.41 | ChannelZ | yeah. of course their website is dead. |
| 22:59.43 | SeRi | thinks mathi's head is spinning at 1000rpm. |
| 22:59.55 | mathi | right... :)))) |
| 22:59.56 | ChannelZ | I think there was another multiplexor someone was in here talking about months ago |
| 23:00.02 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 23:00.10 | F2Knight | spidermux.com? I was just on it lastnight |
| 23:00.25 | ChannelZ | Doesn't work from here, right now anyway |
| 23:00.28 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:00.31 | mathi | dead here too |
| 23:00.32 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 23:00.34 | SeRi | you can setup an asterisk server in an amazon cloud for nearly free |
| 23:00.44 | ChannelZ | Their DNS seems dead |
| 23:01.32 | SeRi | all you have to pay for is a public faciing IP which is like two dollars a month or so. you have to use there basic OS |
| 23:01.58 | SeRi | its like 800mhz vm with 256MB of ram or something of that nature |
| 23:02.17 | F2Knight | umm traceroute dies right out of a comcast loop for me.. maybe a router/switch is out |
| 23:02.57 | mathi | I'm kind of lost now |
| 23:03.00 | ChannelZ | If this doctor literally only has one phone and one phone line, you can do this all over the Internet (assuming he has an internet connection) but just make him get a SIP phone. You can get a decent one for not much more than a decent ATA |
| 23:03.34 | ChannelZ | The Asterisk server can be somewhere else and handle multiple clients so you don't have to buy separate comptuers for each office. |
| 23:03.41 | mathi | well I was discussing to install a server fanless at the doctor's workplace, but now I read there's an alternative to host a server at a hosting company .... |
| 23:03.42 | F2Knight | ChannelZ, is right.. even the grandstream GXP2010 are a decent affordable phone |
| 23:04.02 | SeRi | grandstream sip phones can go for like 25+ dollars |
| 23:04.11 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 23:04.57 | ChannelZ | His SIP phone logs in to your Asterisk; His phone number goes to your ITSP which sends you calls over the net; Your IVR does whatever it does, or if they press 0 to talk to the doctor (or whatever), it calls his phone at his office over the net. |
| 23:04.57 | F2Knight | I personally use a Grandstream 2010, nice phones lots of BLF and 'enterprize' features. |
| 23:04.57 | ChannelZ | Done. |
| 23:05.35 | ChannelZ | He can keep his analog phone line at his office as a backup in case of emergency or the system goes down. |
| 23:05.41 | F2Knight | basicly mathi, the internet becomes a really long phone line. |
| 23:05.45 | SeRi | I have a budget Tone 100 that serve me well. It cost me nothing :) It was free but I think there in ebay for like 15 dollars |
| 23:06.05 | ChannelZ | OR you can let him keep his analog phone and his phone line, and just place a new outgoing call from your Asterisk server to his phone number. |
| 23:06.10 | SeRi | now I have a Polycom IP 501 |
| 23:06.20 | ChannelZ | It's just the bandwidth and cost of the additional call. |
| 23:07.13 | ChannelZ | Which is now back to where we were 3 hours ago or whenever that was :P |
| 23:07.26 | F2Knight | ChannelZ, is right again.. if your only providing a 'service' of your app.. then you do not even need to provide him 'phone service' the only reason to replace his phone or anything like that is if you wanted the residual income form providing phone service |
| 23:07.53 | F2Knight | sorry I had to ask what was going on |
| 23:08.06 | SeRi | residual income is the best income to have when in volume :) |
| 23:08.35 | F2Knight | SeRi, even if not in volume. just 'extra' on the side ":) |
| 23:08.47 | SeRi | agree :) |
| 23:09.21 | mathi | ChannelZ, so why didn't you go for SIP and instead buy a $700 card if it is so simple with internet |
| 23:09.57 | ChannelZ | Because I had shitty internet at my first office and didn't want to have to deal with it while I was busy running the rest of the business |
| 23:09.59 | F2Knight | becuase we have our own T1 circuits and terminate and orginate our own calls |
| 23:10.25 | SeRi | because I am a pimp. |
| 23:10.43 | F2Knight | we SIP out our LD calls, and bring in a few DID's from other states and area codes |
| 23:10.47 | ChannelZ | When we move next year I will probably just put in a dedicated DSL or something for VoIP and go that route. Not sure yet, have to run the numbers on cost and see if it would be beneficial. |
| 23:10.51 | SeRi | just felt like joing the because* lol |
| 23:10.59 | SeRi | out for a while. fam is calling :) |
| 23:11.01 | SeRi | cya guys |
| 23:11.06 | ChannelZ | bye |
| 23:11.16 | F2Knight | later SeRi |
| 23:11.28 | F2Knight | ChannelZ, what is your line of work? |
| 23:11.39 | ChannelZ | Video post production |
| 23:11.52 | F2Knight | ahh. nice. |
| 23:12.00 | ChannelZ | http://idolum.com |
| 23:12.36 | ChannelZ | ...which apparently is being slow, speaking of dead websites. grrph |
| 23:12.46 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 23:12.48 | F2Knight | lol yea i just noticed |
| 23:13.05 | ChannelZ | We lost a hard drive last week and it's still been resyncing the RAID. Logging in to see what's going on.. |
| 23:13.27 | F2Knight | looks like FDCServers.net.. |
| 23:13.45 | F2Knight | which seems to be a comcast switch maybe. has some heavey lag |
| 23:14.06 | ChannelZ | It's a colo box but yes at FDC. Wonder if they're getting attacked |
| 23:14.33 | F2Knight | I am doing an MTR on it and keep getting excess of 300ms responses |
| 23:14.48 | F2Knight | with a worst rating of 980ms to respond |
| 23:15.12 | ChannelZ | hmm. I'm pinging the box directly and it's ~30ms max |
| 23:15.46 | mathi | so I could rent a VPS server, put Asterisk on it, and that's it in terms of hardware ? |
| 23:15.48 | F2Knight | your in denver, |
| 23:15.54 | F2Knight | I am in Portland and Miami |
| 23:16.07 | F2Knight | both sides are choking at FDC |
| 23:16.15 | ChannelZ | now it's responding quickly. meh |
| 23:16.40 | ChannelZ | actually it's up and down. They're having issues. |
| 23:17.29 | mathi | ChannelZ, I understood it correctly? |
| 23:17.39 | ChannelZ | mathi: yes |
| 23:18.00 | F2Knight | mathi, you coudl vps it... but you would have to make sure your not sharing too much processor with other hosts... not all vps providers are equal. |
| 23:19.17 | mathi | I will not rent a single server for all the doctors, too risky in case it would fail |
| 23:19.27 | mathi | so a few doctors, and a small server |
| 23:19.52 | mathi | e.g. a small server for max 5 doctors |
| 23:20.22 | F2Knight | A college of mine provide services for about 25 docotors in about 5 offices. using our servers. they are all on one server actually. |
| 23:21.54 | mathi | well depends what kind of services. I'm talking here about the ability for patients to make appointments with their doctors. it's crucial for the doctor's business |
| 23:22.35 | F2Knight | mathi, that however has nothing to do with you providing phones for the doctor.. that is strictly your application. |
| 23:23.11 | F2Knight | I assume you are doing the appointment setting on line. |
| 23:23.25 | mathi | I didn't get you sorry |
| 23:23.28 | F2Knight | and as such your server will have to be on line. |
| 23:23.36 | mathi | the appointments are done in asterisk through the ivr system |
| 23:23.51 | ChannelZ | mathi: if you do it as a VPS you can just freeze a backup image once in awhile.. if the thing dies, you start up another one somewhere else and switch over |
| 23:24.21 | F2Knight | mathi, so I am client1, I can not set my appointment on line? I have to do it over the phone? |
| 23:24.24 | WIMPy | Last time I was there, Belgium was in Europe. You most probably want BRIs there. |
| 23:24.59 | mathi | F2Knight, there is also an internet site, but I don't understand how that question is related |
| 23:25.18 | F2Knight | it is related to understand how you expect things to flow. |
| 23:25.30 | F2Knight | it is related because it is the core of your 'model' |
| 23:25.32 | mathi | it's getting late i guess... :) |
| 23:25.48 | mathi | WIMPy, I concluded that ITSP was probably the best solution |
| 23:25.49 | F2Knight | it is related because your not a phone provider but an application provider |
| 23:26.23 | mathi | F2Knight, I know, but I can't just provide the service without giving them the tools to use it |
| 23:26.37 | F2Knight | so I am client 1 |
| 23:26.45 | mathi | patient 1 |
| 23:26.45 | mathi | yes |
| 23:26.56 | F2Knight | what number do you envision me calling to make an apointment |
| 23:27.14 | mathi | uhm, the doctor's number? |
| 23:27.35 | F2Knight | my own doctors number? A toll free number? some other number provided by the doctor for the sole use of making appointments? |
| 23:27.44 | mathi | each doctor has a number |
| 23:27.56 | mathi | oh... is that a problem if I have a central asterisk server hosted ? |
| 23:28.24 | F2Knight | now ... this 'number' for each doctor is it a specaial number just for the appointments.. or is it there primary number? |
| 23:28.55 | mathi | I should discuss it but it should be their primary number |
| 23:29.07 | mathi | for sure actually |
| 23:29.52 | F2Knight | IF it is there primary number.. then they will have to have a phone system at there location.. that can forward the call to your server. |
| 23:30.27 | F2Knight | You can setup a toll free number for this if you do not want your clients having to pay toll charges. |
| 23:30.45 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:31.08 | F2Knight | or you have to provide them with phone servers.. for each client.. (not practical at all) a better way would be having them forwarding the call to you. |
| 23:31.55 | F2Knight | OR you provide the doctors with a toll free number that the clients can call directly and enter the doctors office number or something to identify them ... or a different number for each doctor. |
| 23:32.09 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 23:32.35 | F2Knight | but if you are going to have them having the call come to there office, then you have to process the calls either at there office or have them forwarded you have no other options. |
| 23:33.31 | mathi | mm what do I need to forward the call ? |
| 23:33.50 | F2Knight | if you choose to provide a phone system to each of your clients you will require retrainging all of them, and mimicking what it is they are use to doing, or purchasing additional hardware for each site to handle what you need to and integrating it with there existing systems. either way your looking at a complext route |
| 23:34.02 | F2Knight | you need a number to forward the calls to! |
| 23:34.23 | F2Knight | they will have to have a phone system that already supports either an IVR |
| 23:34.56 | F2Knight | or the receptionist will have to forwad the call to you, usually done by programing an extension on there system |
| 23:35.19 | F2Knight | at that point the calls get directed to your own server that you can manage and control |
| 23:36.05 | mathi | so you mean anyway I need a server client side if I want to redirect the call to my hosted centralized server? |
| 23:36.25 | mathi | *doctor side |
| 23:36.32 | F2Knight | failing to wnat to do that you will have to setup servers in every doctors office and be managing there phone system for them, and I can assure you that you will get less then 15% of those doctors to sign up by that method. better to forward the calls to your service and not get in to the phone business. |
| 23:37.14 | F2Knight | no . you need for the doctors office to have a way to forward/transfer a call to you. |
| 23:37.22 | *** join/#asterisk JamesWatt (JamesWatt@unaffiliated/jameswatt) |
| 23:37.44 | F2Knight | if they already have a phone system, then that system would have to be reprogramed. |
| 23:38.20 | WIMPy | Do any of the ITSPs tell you where a call was forwarded from? |
| 23:38.31 | mathi | well a lot of doctors forward their calls to a company with phone operators. how is this usually done ? I don't think a lot of doctors have a phone system |
| 23:38.37 | F2Knight | if they just have regular phone lines that ring directly to real phones they can not use your service with out providing the patiatin a direct phone number to call (to your system) |
| 23:39.22 | WIMPy | They probably do the forwarding in the switch. |
| 23:39.29 | F2Knight | mathi, most people with out a phoen system have no IVR, and they forward ALL calls to an answering service. |
| 23:40.05 | F2Knight | for your service to work. you have to either A) provide a number for patients to call your system directly on behalf of there doctors. |
| 23:40.33 | F2Knight | b) have doctors that have phone systems on site that they can have reprogramed to forward an ivr menu item to your system |
| 23:40.45 | F2Knight | c) rething your entire business stradagy |
| 23:41.13 | mathi | i think for b) there are more options |
| 23:41.14 | F2Knight | d) encure large costs managing and replacing all the offices phone sytems for them |
| 23:41.42 | F2Knight | which is why youre here. |
| 23:41.48 | mathi | i think there are other ways to forward a call |
| 23:41.50 | WIMPy | You don't need a client side phone system to do forwarding. |
| 23:42.08 | mathi | i hope so ... |
| 23:42.28 | F2Knight | nope to forward a call your provider must allow you to forward it. |
| 23:42.42 | F2Knight | there is no other option, |
| 23:42.52 | F2Knight | to transfer a single call |
| 23:43.09 | WIMPy | Forward or transfer? |
| 23:43.18 | F2Knight | transfer actually |
| 23:43.40 | F2Knight | he has an SaS app that he is providing |
| 23:43.56 | F2Knight | but requires DTMF services to set appointments |
| 23:44.05 | WIMPy | Transfer might not be available. |
| 23:44.10 | WIMPy | sas? |
| 23:45.04 | F2Knight | he does not exactly want to become a phone service provider, so he has to make solution to work around the tech issues.. namely he has to run his own phone system and provide dial in numbers for all his doctors offices to connect to him. tht number would be what the office would transfer too. |
| 23:45.19 | F2Knight | SaS = Software as Service |
| 23:45.29 | WIMPy | aye |
| 23:45.43 | WIMPy | Ok, not a big deal so far, is it? |
| 23:46.03 | F2Knight | no, but he doesnt' understand the basic way even POTS work |
| 23:46.26 | WIMPy | That seems to be true for most ppl here. What's the issue? |
| 23:46.47 | F2Knight | he org. was talkign about puting asterisk at clients offices and handling there phone calls for them just to get the IVR to connect to his service.. expensive route. |
| 23:46.54 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 23:47.16 | F2Knight | what he needs to do is host is own asterisk server, at his colo (which he already has a webserver at) |
| 23:47.22 | F2Knight | take sip trunking in |
| 23:47.23 | WIMPy | Complicated. |
| 23:47.40 | F2Knight | and provide a dialin access number for his 'clients' |
| 23:47.46 | WIMPy | And the side effects might not go down too well. |
| 23:47.46 | F2Knight | 3 steps |
| 23:47.59 | mathi | WIMPy, which side effects ? |
| 23:48.20 | F2Knight | remember WIMPy, hes not in the phone business at all |
| 23:48.23 | WIMPy | Ok, I asked that earlier: Do ITSPs tell you were a call was forwarded from? |
| 23:48.44 | F2Knight | ITSp will tell you where the call orginated from |
| 23:48.54 | WIMPy | The side effects of using Asterisk instead of a PBX or just a plain phone line. |
| 23:49.24 | F2Knight | well his clients may have just POTS in there offices, or they may have a PBX of some type, (unknown) |
| 23:49.24 | WIMPy | Yes, but he needs to know who forwarded the call, not who was calling, right? |
| 23:49.34 | F2Knight | thats actually easy. |
| 23:49.45 | F2Knight | infact .. that is actually the easiest part |
| 23:49.45 | WIMPy | POTS seems highly unlikely. |
| 23:49.48 | mathi | I need to know the patient number also! |
| 23:50.04 | mathi | that might be a problem? |
| 23:50.05 | WIMPy | Only if you get the information or have enought numbers. |
| 23:50.28 | F2Knight | the MOST easiest solution is for him to provide an 800 number or some type of number for each of his clients. |
| 23:50.43 | WIMPy | mathi: Generelly no. But I'm not sure about ITSPs. |
| 23:50.47 | F2Knight | you can always request the caller enter there phone number |
| 23:50.52 | mathi | F2Knight, I can't do that, you know the doctors are referenced on the internet with their phone numbers |
| 23:50.53 | WIMPy | How many clients? |
| 23:51.07 | WIMPy | Numbers don't come for free in larger quantities. |
| 23:51.20 | F2Knight | right now he has 1 |
| 23:51.50 | WIMPy | Well, with one, the selection os pretty easy :-) |
| 23:52.00 | mathi | I have 1 doctor now, but I can have several small VPS servers handling max 5 doctors, they aren't expensive |
| 23:52.04 | F2Knight | but he could easily put an IVR on a toll free number and say for Dr. Enkels office press 1., for Dr. Mathews Press 2 etc. |
| 23:52.06 | *** join/#asterisk bulletrtr_ (48a00c18@gateway/web/freenode/ip.72.160.12.24) |
| 23:52.28 | mathi | that will be complicated if one day I have 20 doctors ... |
| 23:52.36 | WIMPy | Yes, but that should already be known. |
| 23:52.50 | F2Knight | if you have 20 doctors you can have 20 numbers |
| 23:52.53 | F2Knight | 1 for each of them |
| 23:53.03 | F2Knight | then no need for the for xxx press 1. |
| 23:53.19 | F2Knight | but with only a small number of doctors.. this is a better 'cost' option |
| 23:53.24 | WIMPy | Or use extended DDI. |
| 23:54.05 | F2Knight | WIMPy, I think that would only work over SIP orgination .. wouldnt work on pots .. (think little old laddy calling ) |
| 23:54.20 | mathi | F2Knight, what do I need to pay in case I use ITSP and have several numbers? I thought I could use as many channels as I wish |
| 23:54.21 | mathi | ) |
| 23:54.42 | ChannelZ | depends on the ITSP |
| 23:54.58 | WIMPy | I use that all the time in the PSTN. |
| 23:55.01 | ChannelZ | Some just charge whatever you use, some want to place artificial limits on simultaneous channels |
| 23:55.12 | F2Knight | mathi, that all depends on your provider. |
| 23:56.03 | F2Knight | no with the pstn you only get 1 channel at a time per line |
| 23:56.26 | WIMPy | I thought it was about numbers, not channels. |
| 23:56.47 | F2Knight | It is abount numbers. but he asked about channels.. like I said ... doesnt understand much |
| 23:56.49 | WIMPy | And forget about POTS. No except your granny uses that. |
| 23:57.03 | WIMPy | no one |
| 23:57.08 | F2Knight | granny is the one thats calling the doctor to make an apointment :) |
| 23:57.19 | ChannelZ | to an office with only one phone line |
| 23:57.20 | WIMPy | That's ok. |
| 23:57.23 | mathi | F2Knight, I can't have another number for doctors (the easy solution you were talking about). So the only way is to forward/transfer (don't know the difference) the call to my asterisk server. And I didn't really understand what I need to do here. Anyway, I need the patient's number, plus the doctor's number of course to load the scripts according to the doctor called. I wonder if I can have these two numbers |
| 23:57.32 | WIMPy | But the doctor won't, so that's ok. |
| 23:57.58 | WIMPy | mathi: Sure. |
| 23:58.02 | ChannelZ | I missed the part about these multiple phone numbers and what that has to do with anything |
| 23:58.05 | WIMPy | At least on the PSTN. |
| 23:58.42 | mathi | ChannelZ, well if I have a centralized server, each doctor has a number that needs to forward/transfer(?) to my asterisk server |
| 23:58.46 | WIMPy | He needs to know who forwarded the call. |
| 23:58.54 | F2Knight | mathi, you will probably NOT get the callers phone number, if the call was forwarded to you. SO you will have to prompt the caller to enter it. |
| 23:59.02 | mathi | omg |
| 23:59.13 | ChannelZ | so? |
| 23:59.15 | WIMPy | Surew you get the callers number. |
| 23:59.27 | F2Knight | only from the number calling you! |
| 23:59.31 | F2Knight | he wants this... |
| 23:59.54 | WIMPy | He wants both, and that's what he'll get. |