| 00:01.35 | WIMPy | BTW: The subscription stuff seems to be fully ok again. |
| 00:16.16 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-okccstbnlemvepno) |
| 00:52.52 | fling | Hello! |
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| 01:06.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 03:22.38 | MrTAP | Is it possible to have someone take a look at a long standing bug with sip and realtime: https://issues.asterisk.org/jira/browse/ASTERISK-17523 |
| 03:36.35 | *** join/#asterisk din3sh (~din3sh@41.136.87.157) |
| 03:37.00 | din3sh | mrning all |
| 03:48.49 | bipul | Morning din3sh |
| 04:01.18 | saint_ | din3sh: morning ? where the hell are you located ? |
| 04:01.45 | saint_ | Ha.. i guess i could have used whois ... |
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| 05:44.03 | din3sh | saint_: indian ocean, its 09:42 am |
| 05:44.05 | din3sh | :P |
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| 06:16.10 | ChannelZ | just listened to Deep India |
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| 06:36.07 | beardy | beardy_ is listening to: Metallica – Ride the Lightning |
| 06:36.22 | beardy | Morning it is. |
| 06:36.50 | *** join/#asterisk kchehab (~ck@bba425084.alshamil.net.ae) |
| 06:38.34 | kchehab | [TK]D-Fender if two extensions are behind one router the call will not path using SRTP and it will bass without it , i paste my debug on http://pastebin.com/zCCpzZmD |
| 06:39.40 | kchehab | i can find 488 Not Acceptable Here but i can call another extension having same sofphone config on aother network ,please help |
| 06:40.18 | kchehab | as i can call my trunk using srtp too |
| 06:44.57 | [TK]D-Fender | no idea. |
| 06:45.05 | [TK]D-Fender | I don't do TLS/STRP |
| 06:45.20 | [TK]D-Fender | Though you somehow assumed I knew |
| 06:45.47 | *** join/#asterisk verybigbread (~ryan@50.31.0.5) |
| 06:46.04 | [TK]D-Fender | I do notice that the outbound leg does NOT haev the same AES header that your inbound call showed which does not lead me to believe that it is set up right however |
| 06:46.17 | [TK]D-Fender | a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:h6VOpQ6ANPapbOx+wMUIEucDjIHVyKyXA//sUUUo |
| 06:46.18 | [TK]D-Fender | a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:mmeqVfRaiyswtFWZ4QEydnb2RYYdhizv3tOfJHCn |
| 06:46.24 | [TK]D-Fender | No crpto when you call the other ext. |
| 06:47.04 | kchehab | [TK]D-Fender but srtp are allowed on both softphones |
| 06:47.05 | [TK]D-Fender | And your trunk may work... but that may be because *'s call to it does NOT have encryption and doesn't care. I'm betting that your other ext DOES have it enforced and * isn't sending it so it gets refused. |
| 06:47.17 | [TK]D-Fender | And i doubt the * side is right for your 2nd ext |
| 06:47.46 | kchehab | [TK]D-Fender what do you mean by * side ? |
| 06:48.02 | [TK]D-Fender | Asterisk side |
| 06:48.05 | [TK]D-Fender | What is unclear there? |
| 06:48.35 | kchehab | [TK]D-Fender now i set in the sip_additional encyption=yes under the two accounts |
| 06:48.41 | kchehab | and it works thanks alot for help |
| 06:52.57 | kchehab | [TK]D-Fender a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:h6VOpQ6AN and a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:mmeqV should be the same key right ? |
| 06:53.20 | [TK]D-Fender | kchehab: I wouldn't believe so as * is a B2BUA, not a proxy |
| 06:53.57 | kchehab | [TK]D-Fender AES_CM_128_HMAC_S is it 128 bit encyption ? |
| 06:54.38 | [TK]D-Fender | It does give that impression |
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| 09:10.40 | vassilux | hi, I have a cluster with digium r-series. After change(write) a file my on the drbd device the file goes to read-only mode. I checked the drbd status and two nodes are on UpToDate. I know it is not the palce for my question but maybe someone has an idea |
| 09:11.24 | vassilux | sorry for my faults , just wake up :-) |
| 09:25.38 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 09:27.23 | EmleyMoor | I'm trying to get fax receiving to work - and have had only limited success so far. I want to get t38modem to work, but I have little confidence in it as I haven't been able to send faxes through it either. I have tried direct receiving in Asterisk but it doesn't actually seem to receive anything... I'm on 1.8 - any advice on this? |
| 09:29.26 | ChannelZ | I've had success with Fax For Asterisk but that's it |
| 09:30.33 | [TK]D-Fender | EmleyMoor: t38modem should do its thing. * 10+ have * support T.38 gateway function as well so you could use a more traditional iaxmodem, etc. |
| 09:31.26 | [TK]D-Fender | EmleyMoor: If you have the opportunity, see if you can get a word in with coppice. He would probably be of great assistance to you |
| 09:31.33 | EmleyMoor | [TK]D-Fender: Hmmm... iaxmodem already OK for sending, so that interests me... and anyway, I have no problem with sending over my PSTN line |
| 09:31.51 | EmleyMoor | (apart from the fact I want to make it "free calls only") |
| 09:33.25 | EmleyMoor | [TK]D-Fender: I'll see if I can get t38modem working, but the problem it was logging before was hard to fathom |
| 09:34.29 | [TK]D-Fender | EmleyMoor: What are you using for PSTN exactly? |
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| 09:42.16 | [TK]D-Fender | And on that not, it's way past my bedtime. Back later |
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| 10:04.02 | niklaswe | any idé about this.. I can call to my other sip-users ans talk, but when I calling my conference-number I got connected but it doesnt play any playback sound.. |
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| 10:31.30 | din3sh | is it good or bad practice to clear cache regularly in an * box? i.e. echo 3 > /proc/sys/vm/drop_caches |
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| 11:13.49 | ectospasm | din3sh: you mean filesystem cache? Should be OK, but I don't recommend doing it during production. |
| 11:15.22 | niklaswe | hmm I dont get playback to work :( I have open up the ports in my firewall, When I trying to call my conference, I got connected but nothing happens.. I can call other invidual sip-users. |
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| 11:36.00 | din3sh | ectospasm: i have set a cron job to run on sundays |
| 11:36.33 | din3sh | actually i have had to run it on production box also, out of 4gb, 30mb was free |
| 11:36.37 | din3sh | :D |
| 11:36.54 | ectospasm | that shouldn't be a problem, if most of it is cache |
| 11:37.17 | ectospasm | I doubt you'll be a better memory manager than the kernel. |
| 11:39.30 | din3sh | memory usage is supposed to be constant on the system? |
| 11:41.21 | ectospasm | over time the OS/kernel will cache used files in memory, to speed accesses. As soon as a process needs more space, the kernel will remove or otherwise page the cached files out to make room. |
| 11:41.46 | ectospasm | din3sh: yes, it will appear constant if the uptime gets long enough |
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| 11:42.04 | UnixDev | how can I get the name of the channel who is calling the channel in the dial plan? |
| 11:42.08 | ectospasm | ...as long as most of the usage is cache (see "free -m") I wouldn't worry about it. |
| 11:42.57 | ectospasm | UnixDev: see ${CHANNEL(name)} |
| 11:43.25 | ectospasm | for more details about the CHANNEL function, see "core show function CHANNEL" |
| 11:44.07 | UnixDev | ectospasm: in the dial plan, how can I get the name of the channel that called the one thats executing, $(CHANNEL(name)) gives me the name of my channel, i want to know who called me |
| 11:44.41 | ectospasm | CHANNEL(peername) maybe? |
| 11:45.29 | ectospasm | is the calling channel SIP, DAHDI, or some other technology? |
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| 11:46.03 | ectospasm | UnixDev: ^ |
| 11:46.18 | UnixDev | sip peer name is not right |
| 11:46.34 | UnixDev | I'm thinking I'm going to have to pass this from the parent channel |
| 11:46.45 | UnixDev | basicly i need to write back to a parent channel from a child channel |
| 11:46.54 | kaldemar | UnixDev: ${MASTER_CHANNEL(CHANNEL(name))} |
| 11:47.24 | kaldemar | the MASTER_CHANNEL function can be used for both setting and getting variables on the master channel. |
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| 11:48.01 | UnixDev | kaldemar: you are a GENIUS!!! |
| 11:48.10 | UnixDev | thank you!! |
| 11:48.27 | kaldemar | np |
| 11:49.09 | ectospasm | TIL |
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| 14:52.47 | igcewieling | UnixDev: It is often helpful to browse the output of "core show applications" and "core show functions" saves a LOT of time. |
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| 16:04.49 | Neptu_laptop | Hej, Just trying to understand a bit how to get a sip trunc with 200 output lines. Do you know if breveke is a server like asteriks or is a carrier? |
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| 16:40.47 | Zaintel | hello |
| 16:41.23 | Zaintel | any asterisk developer here |
| 16:42.25 | Zaintel | i need a paid installtion of Astrisk for my company |
| 16:42.27 | Zaintel | can anyone |
| 16:42.29 | Zaintel | ? |
| 16:42.44 | Zaintel | please PM me |
| 16:43.52 | *** join/#asterisk vlad_starkov (~vlad_star@wn2nat52.beelinegprs.ru) |
| 16:44.27 | igcewieling | the asterisk developers do not generally do paid consulting |
| 16:45.04 | igcewieling | voip-info.org may list companies which do paid consulting and installation. |
| 16:45.40 | igcewieling | Zaintel: where are you (or at least the server) located? |
| 16:46.02 | WIMPy | And noone will answer without any description anyway. |
| 16:48.35 | Zaintel | i need to transfer DID calls to SIP URI via extension |
| 16:48.46 | Zaintel | but i am not so expert for the installtion |
| 16:48.57 | Zaintel | can u please refer some one |
| 16:49.14 | Zaintel | to install the Server in US datacenter remotely |
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| 17:21.25 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 17:22.49 | izx | We are using linux operating system for our call centre and we have asterisk server.. For windows we are using xlite.. For linux we are using twinkle.. Is there anyother better package than twinkle?? Coz twinkle has lot of problems.. |
| 17:29.02 | psilikon | izx: sflphone, kiax, zoiper |
| 17:31.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 17:34.37 | izx | psilikon: Is it possible to configure twinkle via command line?? And save the configuration somewhere to autostart.. |
| 17:38.25 | MrTAP1 | what's the best way to get a bug acknowledged that's been sitting in the queue for ~ 2 years? |
| 17:38.27 | MrTAP1 | https://issues.asterisk.org/jira/browse/ASTERISK-17523 |
| 17:42.07 | *** join/#asterisk AnotherDeadBody (~quassel@d-65-175-244-102.cpe.metrocast.net) |
| 17:42.59 | AnotherDeadBody | hello all. i need to use custom contex to control extention to outbound routes. no matter what i do , it always passes the call out the first route. is there a tutorial on this that is complete? or does anyone do this? im using asterisk now |
| 17:43.48 | WIMPy | Are you using a GUI? Which one? |
| 17:44.18 | AnotherDeadBody | yes, asterisknow ( freebpx ) |
| 17:44.29 | *** join/#asterisk vlad_starkov (~vlad_star@wn2nat52.beelinegprs.ru) |
| 17:44.36 | WIMPy | Then ask in #freepbx where to do what you want. |
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| 17:47.04 | izx | WIMPy: Is that for me >> "Are you using a GUI? Which one?" |
| 17:47.35 | WIMPy | no |
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| 17:52.54 | din3sh | zzzZZZZZ |
| 17:53.31 | din3sh | WIMPy: digium says might be a problem with my 4th generation card |
| 17:53.39 | din3sh | taking down the server on monday |
| 17:54.09 | WIMPy | Doesn't sound good. |
| 17:54.26 | WIMPy | Was it a PCIe card? |
| 17:54.36 | din3sh | yup |
| 17:55.51 | Zaintel | PCI Express |
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| 17:57.02 | WIMPy | That doesn't explain why it got worse gradually. |
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| 18:16.19 | igcewieling1 | hugs his sangoma cards |
| 18:21.19 | psilikon | izx: Not sure never tried it , I think zoiper has a conf/ini file you can edit. |
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| 18:35.38 | coreyf1513 | izx: what distribution are you trying to use for the call centre clients? |
| 18:36.37 | izx | <PROTECTED> |
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| 18:40.02 | coreyf1513 | have you asked #ubuntu? they might know what will work best in ubuntu, they know what you are dealing with for GUI, kernel, audio subsystem versions, etc. so they might know what packages are most stable on ubuntu |
| 18:40.21 | din3sh | WIMPy: actually digium wont even troubleshoot a 4th generation card |
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| 18:40.58 | igcewieling1 | the best Asterisk packages for Asterisk is compiling from source. |
| 18:41.33 | igcewieling1 | *sigh* the best Asterisk packages for Ubuntu is compiling Asterisk from source. 8-| |
| 18:41.54 | coreyf1513 | igcewieling1: he was asking about softphones to connect with asterisk :) |
| 18:41.55 | igcewieling1 | din3sh: reboot every night. What is what we did when we used Digium cards. (this was a long time ago) |
| 18:42.12 | igcewieling1 | Ah, in that case "all softphones suck" 8-) |
| 18:43.06 | din3sh | reboot eveyr night? argh |
| 18:44.07 | igcewieling1 | din3sh: depending on system load. at some higher volume sites we rebooted every night, at the lower volume sites we rebooted once a week. Used a cron job. We stopped using Digium cards around 2003 or so. I assumed they had fixed such issues by the 4th gen cards. |
| 18:45.23 | din3sh | what was your actual problem? |
| 18:45.24 | igcewieling1 | I don't normally dis Digium cards since I don't have experience with modern ones -- only the old ones |
| 18:45.39 | igcewieling1 | din3sh: in our case analog ports randomly stopped working on TDM400P cards. |
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| 18:46.42 | igcewieling1 | din3sh: did you try a different motherboard or tried moving the card into a different slot? |
| 18:46.57 | din3sh | i have 15 productin servers running with digium cards |
| 18:47.11 | ponyofdeath | hi, I am having issues with asterisk 11.2.1 and google voice stop to receive calls after what seems like an timeou / relogin into google talk |
| 18:47.11 | din3sh | 1st time I'am encountering such problem |
| 18:47.22 | igcewieling1 | din3sh: same on all servers? |
| 18:47.27 | igcewieling1 | (same problem that is) |
| 18:47.28 | din3sh | npe |
| 18:47.33 | din3sh | single span ones |
| 18:47.38 | din3sh | work without problem |
| 18:47.44 | din3sh | this is a double span one |
| 18:48.24 | igcewieling1 | if you end up having to replace the card, I recommend Sangoma. The driver build process is a bit of a hassle and their support has declined recently, but I still recommend them. |
| 18:48.54 | igcewieling1 | I'd leave that as a last resort -- better to try to get your existing stuff working. |
| 18:49.06 | din3sh | yeah |
| 18:49.24 | din3sh | i expected the installation to be trouble-free |
| 18:49.34 | din3sh | brand new DL380 Gen7 |
| 18:49.36 | igcewieling1 | There is some kernel linux kernel option to disable pcix (or is it pcie) powr management -- worth trying |
| 18:50.34 | igcewieling1 | I'm lazy, I'd rather throw money at a problem than throw weeks of time at a problem. |
| 18:52.33 | carrar | You can give me $5000 to go into your bios and turn it off |
| 18:52.34 | igcewieling1 | din3sh: your server doesn't have the Intel Corporation 82574L Gigabit Network onboard LAN chipset does it? |
| 18:52.47 | igcewieling1 | lspci will show you |
| 18:53.12 | igcewieling1 | carrar: I was referring to replacing the card. 8-| |
| 18:53.28 | carrar | oh, then just $5000 is fine then |
| 18:53.33 | carrar | heh |
| 18:53.53 | carrar | to help you get into the mood of "throwing money" |
| 19:02.21 | din3sh | igcewieling1:thernet controller: Broadcom Corporation NetXtreme II BCM5709 Gigabit Ethernet (rev 20) |
| 19:02.30 | din3sh | igcewieling1:ethernet controller: Broadcom Corporation NetXtreme II BCM5709 Gigabit Ethernet (rev 20) |
| 19:04.02 | *** join/#asterisk vlad_starkov (~vlad_star@wn2nat37.beelinegprs.ru) |
| 19:07.17 | Weezey | I'm playing with sip notify and it seems to be sending out messages from the IPv6 address and so the phones don't know how to reply. How can I tell sip to only be ipv4? |
| 19:07.40 | Weezey | (it is currently bindaddr=0.0.0.0 and I need public and private ipv4) |
| 19:16.25 | *** join/#asterisk Ice_Strike (Ice_Strike@31.185.170.171) |
| 19:16.27 | Ice_Strike | Hello =) |
| 19:17.16 | Weezey | Ice_Strike: 'sup? |
| 19:17.39 | Ice_Strike | Chillin |
| 19:17.43 | Ice_Strike | You? |
| 19:18.04 | Weezey | same, both kids napping, time for some * |
| 19:19.33 | WIMPy | din3sh: What does that mean "they won't troubleshoot"? |
| 19:19.36 | Ice_Strike | Good stuff |
| 19:19.43 | *** join/#asterisk mjordan (~mjordan@216.186.152.216) |
| 19:19.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:20.42 | Weezey | yup, and just made a breakthrough with my remote voicemail notification problem |
| 19:20.47 | igcewieling1 | Weezey: are you SURE? bindaddr=0.0.0.0 should limit Asterisk to only IPv6 |
| 19:21.44 | Weezey | igcewieling1: it definitely sends out my sip notify commands via the IPv6 address |
| 19:21.51 | Weezey | everything else works just fine though. |
| 19:22.11 | Weezey | I changed it to the public IP and the internal phones look fine. |
| 19:22.15 | Weezey | just need to test inbound |
| 19:22.28 | igcewieling1 | report it as a bug, bindaddr=0.0.0.0 is documented to tell asterisk to only use IPv4. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html |
| 19:22.30 | Weezey | but since * has canreinvite=no, I'd assume I'm good. |
| 19:22.41 | coreyf1513 | Weezey: what do you get for netstat -antup|grep asterisk (run this from root) |
| 19:22.45 | igcewieling1 | canreinvite=no is the old 1.4 option |
| 19:23.02 | igcewieling1 | it may not even work in recent versions of Asterisk |
| 19:23.04 | Weezey | what should it be for 1.8 or > ? |
| 19:23.16 | [TK]D-Fender | directmedia=no |
| 19:23.25 | Weezey | igcewieling1: looks like it's already a bug: https://issues.asterisk.org/jira/browse/ASTERISK-16545 |
| 19:23.30 | igcewieling1 | Weezey: that directmedia= sounds like you did not read the UPGRADE*.txt files. |
| 19:23.43 | din3sh | WIMPy: maybe the 4th generation card has some fault, useless to troubleshoot in that case |
| 19:23.43 | Weezey | but never resoled |
| 19:23.49 | Weezey | resolved. |
| 19:24.36 | Weezey | anyway, it's only for sip notify, call flow has always worked correctly for my setups. |
| 19:25.13 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 19:27.20 | *** join/#asterisk ashd (~ashleyd@94-194-155-253.zone8.bethere.co.uk) |
| 19:32.21 | ponyofdeath | I am having this same issue looks like http://forums.digium.com/viewtopic.php?f=1&t=85007 anyone any ideas? |
| 20:06.40 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 20:09.26 | jmls | evening. |
| 20:09.51 | jmls | I've used exchange with asterisk before to create a unified vm type thingy |
| 20:10.04 | jmls | however, has anyone used office365 for the same purpose ? |
| 20:11.57 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 20:32.09 | ghost75 | office yuck |
| 20:37.58 | jmls | ghost75 : ya |
| 20:38.01 | jmls | not my call |
| 20:38.32 | *** join/#asterisk ipconfeng (~ipconfeng@pool-71-185-231-211.phlapa.fios.verizon.net) |
| 20:38.39 | [TK]D-Fender | office365 & Office 2013 licensing = FUBAR'd |
| 20:38.47 | jmls | mine is not to reason why, mine is but to do and die. |
| 20:38.54 | [TK]D-Fender | Not touching until some serious corrections get made. |
| 20:39.09 | [TK]D-Fender | jmls: We still hear you ... you're only half-way there ;) |
| 20:39.10 | ipconfeng | can someone tell me how I can export a list of sip peer IPs to a text file? |
| 20:39.25 | jmls | what, the reasoning or the dying ? ;) |
| 20:39.37 | [TK]D-Fender | ipconfeng: asterisk -rx "sip show peers" > atextfile |
| 20:40.57 | ipconfeng | D-Fender: is there any way to filter only the IP address list? |
| 20:41.09 | [TK]D-Fender | ipconfeng: "man grep" |
| 20:42.06 | ipconfeng | been messing with it all day. I just cant get it right. guess ill keep at it. Thanks! |
| 20:42.12 | ectospasm | ipconfeng: the appropriate application of sed, awk, or perl should do you wonders. |
| 20:42.50 | ectospasm | or even GNU cut |
| 20:43.29 | ectospasm | cut might be the simplest for you |
| 20:44.55 | ipconfeng | Thanks all! |
| 20:56.59 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 21:05.16 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 21:07.28 | *** part/#asterisk mjordan (~mjordan@216.186.152.216) |
| 21:10.47 | Ice_Strike | What is the reasonable approach to send command data from browser to daemon that run AMI.. also browser recieve data (example call answer, hangup, etc) |
| 21:11.41 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 21:11.50 | Ice_Strike | For Predictive dialer |
| 21:12.15 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 21:14.27 | [TK]D-Fender | Whatever you want |
| 21:14.31 | [TK]D-Fender | This is your back-end |
| 21:14.43 | [TK]D-Fender | And your daemon that's in front of AMI as well apparently |
| 21:18.36 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 21:22.23 | Ice_Strike | Do you mean run two daemons? |
| 21:22.32 | Ice_Strike | One that run AMI |
| 21:22.43 | Ice_Strike | and other daemon to listen the events from a browser |
| 21:23.03 | Ice_Strike | I could use Json or XML to send and recive data between |
| 21:24.19 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 21:24.22 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 21:25.06 | [TK]D-Fender | Ice_Strike: You described already having something else between AMI. |
| 21:26.06 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 21:33.08 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
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| 22:06.14 | igcewieling1 | One common method is to use AJAX of some form to run a script on the web server which connects to AMI, issues the command and sends the result back to the browser either as JSON or as javascript. |
| 22:07.32 | igcewieling1 | the transactional model of http doesn't mesh well with the long term connection model of AMI, but you can do it. |
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| 23:39.13 | *** join/#asterisk moe613 (46371f24@gateway/web/freenode/ip.70.55.31.36) |
| 23:39.48 | moe613 | Hello. I am working on a project, I want to know if Asterisk is the correct backend. The idea is (on a VPS server) 1. Make outgoing call using google voice account to Number 1 2. 1. Make outgoing call using google voice account to Number 2 3. Connect two calls together on server side. Is asterisk a good solution for this? |
| 23:42.34 | ectospasm | moe613: your numbering is confusing. what's "Number 1 2. 1." mean? |
| 23:42.50 | moe613 | Sorry. Let me try again. |
| 23:43.08 | moe613 | Hello. I am working on a project, I want to know if Asterisk is the correct backend. The idea is (on a VPS server) |
| 23:43.18 | moe613 | 1. Make outgoing call using google voice account to Number 1 |
| 23:43.40 | moe613 | 2. Make outgoing call using google voice account to Number 2 |
| 23:43.41 | moe613 | 3. Connect two calls together on server side. Is asterisk a good solution for this? |
| 23:49.47 | [TK]D-Fender | Sure |
| 23:50.16 | [TK]D-Fender | Only thing you've completely out is determining who to do this for, and triggered how. |
| 23:50.21 | [TK]D-Fender | But that aside, sure |
| 23:50.39 | ectospasm | I concur |
| 23:50.45 | ectospasm | should be relatively simple. |
| 23:50.52 | ectospasm | no special hardware needed |
| 23:52.34 | Neptu_laptop | a question im looking for a trunk that can give me about 2dolar price per line + calls can someone recomend any? |
| 23:52.40 | Neptu_laptop | i need to setup 200 |
| 23:53.16 | Neptu_laptop | If for a life quest so all participants needs to be updated as soon as possible |
| 23:53.37 | Neptu_laptop | so i need a lot of out lines and 0 in basically |
| 23:53.55 | moe613 | Thanks guys, I will get on it. We have a lamp server to control incomming requests from an app. |
| 23:54.00 | ectospasm | what kind of trunk are you looking for, Neptu_laptop? |
| 23:54.11 | [TK]D-Fender | Neptu_laptop: " 2dolar price per line + calls" <- rephrase this and be clearer |
| 23:54.18 | ectospasm | I'd think you'd find a better deal with SIP |
| 23:54.47 | ectospasm | ...and with SIP the concept of "lines" doesn't really exist, just available bandwidth and how you use it. |
| 23:54.53 | Neptu_laptop | a sip trunk to connect to asterisk they are givin me a price like 5 dolar per line * 200 lines and i feel s a bit expensive |
| 23:55.15 | ectospasm | by "lines" do you mean simultaneous calls? |
| 23:55.41 | Neptu_laptop | ectospasm, well i need to be able to reach 200 cellphones same time so i need a gateway someway, just guessing |
| 23:56.12 | ectospasm | Asterisk might not be the best solution to blast 200 channels simultaneously. |
| 23:56.17 | Neptu_laptop | but yes is 200 concurrent calls |
| 23:56.40 | Neptu_laptop | ectospasm, need to play a recoding only not big load |
| 23:57.40 | Neptu_laptop | the major problem is to find a sip trunk cheap enough because is expensive... kills the fun |
| 23:57.53 | *** join/#asterisk nc11235 (~textual@pool-72-64-103-45.dllstx.fios.verizon.net) |
| 23:57.59 | ectospasm | you should find a SIP provider that gives you a flat rate per minute per call (like 1.5¢/min, as an example), and let your limit be your bandwidth |
| 23:58.34 | ectospasm | there are ITSPs out there that do that |
| 23:58.48 | ectospasm | maybe a low maintenance fee |
| 23:58.55 | ectospasm | ...plus your usage |
| 23:59.03 | Neptu_laptop | ITSPs |
| 23:59.06 | Neptu_laptop | let me google that |
| 23:59.13 | ectospasm | Internet Telephony Service Provider |
| 23:59.22 | WIMPy | Flatrate per munite? What does that mean? |
| 23:59.30 | WIMPy | ~itsp |
| 23:59.30 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 23:59.32 | [TK]D-Fender | <PROTECTED> |
| 23:59.47 | [TK]D-Fender | internet* |
| 23:59.48 | ectospasm | WIMPy: forgive me, I am tired. |