| 00:02.28 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:02.28 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:09.03 | ^rage^ | http://dpaste.org/wi3Eq/raw/ |
| 00:11.56 | ChannelZ | their machines are getting hammered |
| 00:12.00 | ChannelZ | Im sure |
| 00:19.43 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:19.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:20.08 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 00:21.55 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 00:35.39 | *** join/#asterisk andy09usa (~Andrey@audotov.com) |
| 00:54.24 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 00:56.42 | *** join/#asterisk coreyf1513 (~cfarrell@ws2.cfware.com) |
| 01:00.09 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 01:18.06 | ghost75 | cant touch this |
| 01:21.29 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 01:43.38 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 01:46.20 | *** part/#asterisk ghost75 (~ghost75@dslb-088-064-062-218.pools.arcor-ip.net) |
| 01:51.07 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 02:04.02 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.25.167) |
| 02:19.13 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-76-226.se.biz.rr.com) |
| 02:21.17 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 02:30.01 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 02:30.26 | *** join/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 02:30.54 | slicknick5181 | I need some help with DTMF in=-call features Asterisk 1.8 |
| 02:36.21 | slicknick5181 | is anybody active right now? |
| 02:48.53 | *** part/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 03:03.42 | ChannelZ | meh |
| 03:13.35 | *** join/#asterisk Nautilus (~Username@d192-24-50-173.try.wideopenwest.com) |
| 03:13.39 | *** part/#asterisk Nautilus (~Username@d192-24-50-173.try.wideopenwest.com) |
| 03:14.02 | *** join/#asterisk Nautilus (~Username@d192-24-50-173.try.wideopenwest.com) |
| 03:14.04 | *** part/#asterisk Nautilus (~Username@d192-24-50-173.try.wideopenwest.com) |
| 03:19.21 | *** join/#asterisk jvwjgames (180ab988@gateway/web/freenode/ip.24.10.185.136) |
| 03:19.35 | jvwjgames | hello i need help with my asterisk server |
| 03:20.40 | jvwjgames | hello is anyone there |
| 03:23.10 | jvwjgames | i need help please :( |
| 03:26.50 | jvwjgames | is anyone here i am having problems with my Asterisk Server |
| 03:29.44 | jvwjgames | hello |
| 03:29.46 | *** part/#asterisk jvwjgames (180ab988@gateway/web/freenode/ip.24.10.185.136) |
| 03:29.54 | *** join/#asterisk jvwjgames (180ab988@gateway/web/freenode/ip.24.10.185.136) |
| 03:29.59 | jvwjgames | hello |
| 03:30.25 | jvwjgames | I NEED HELP WITH MY ASTERISK SERVER PLEASE |
| 03:31.41 | Dovid | anyone see the results of: mtr 216.81.59.173 ? |
| 03:31.57 | Dovid | jvwjgames: What do you need help with? |
| 03:32.51 | jvwjgames | hi my asterisk server is working i can register phone but they can't call eachother |
| 03:33.12 | Dovid | what happens when you try ? |
| 03:33.18 | Dovid | what do you see in the Asterisk console? |
| 03:33.46 | jvwjgames | [Feb 9 20:32:33] NOTICE[11841]: chan_sip.c:23272 handle_request_invite: Call from '1001' (192.168.1.254:65248) to extension '1001' rejected because extension not found in context 'default'. |
| 03:34.01 | jvwjgames | opps sorry |
| 03:34.36 | ChannelZ | is that your problem? |
| 03:35.09 | jvwjgames | hold on |
| 03:35.21 | jvwjgames | let me paste my resaults in a website paster |
| 03:36.45 | jvwjgames | http://www.pastit.dotcloud.com/226?format=raw+ |
| 03:36.55 | jvwjgames | http://www.pastit.dotcloud.com/226?format=raw |
| 03:37.06 | jvwjgames | sorry the + at the end is not supposed to be there |
| 03:37.22 | ChannelZ | looks like you have permissions problems probably |
| 03:38.28 | ChannelZ | IE if you have a sound set installed (demo-congrats.ulaw or such really exists in your sounds/en dir) then the user Asterisk is running over likely doesn't have permissions to read them |
| 03:39.04 | jvwjgames | do you have team viewer |
| 03:39.08 | ChannelZ | Not sure what the //'s are about in your log file either |
| 03:39.18 | ChannelZ | no |
| 03:39.22 | jvwjgames | cause i need help with configure ing my dial plan |
| 03:40.13 | jvwjgames | i just installed asterisk using the opensuse 1 click installer |
| 03:40.25 | jvwjgames | and it can as broken |
| 03:40.36 | jvwjgames | *Came* |
| 03:42.37 | ChannelZ | well I dunno about that installer or package. In normal-world sounds usually live in /var/lib/asterisk/sounds/en/ |
| 03:42.56 | jvwjgames | do you have |
| 03:42.57 | jvwjgames | ssh |
| 03:43.18 | jvwjgames | cause i will let you ssh into my server |
| 03:43.54 | ChannelZ | yes but are you unable to check this for yourself? Or you just want someone to go in and do it all for you? |
| 03:44.45 | jvwjgames | not it all i just need help withthe begin after that i can do it |
| 03:53.16 | jvwjgames | just help with the begining then i can do the rest |
| 06:14.34 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 07:01.39 | *** join/#asterisk alx0 (~alx@host.alx0.ru) |
| 07:14.46 | *** join/#asterisk tamiel (~tamiel@AToulouse-257-1-18-211.w90-5.abo.wanadoo.fr) |
| 07:26.11 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 07:32.59 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 08:21.24 | *** join/#asterisk Alex_Bkash (cbdf5c4a@gateway/web/freenode/ip.203.223.92.74) |
| 08:24.03 | Alex_Bkash | hello. I have two IAX2 trunk to same client PC. each trunk uses different nic in the client PC. How can i distribute calls in these two IAX2 trunk based on available bandwidth??? |
| 08:28.54 | Alex_Bkash | hello. I have two IAX2 trunk to same client PC. each trunk uses different nic in the client PC. How can i distribute calls in these two IAX2 trunk based on available bandwidth??? |
| 08:32.23 | ChannelZ | I think you'd have to write that yourself. Maybe use the group counting functions to keep track and choose accordingly |
| 08:35.37 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 08:36.17 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 08:36.42 | Alex_Bkash | okk ChannelZ |
| 08:51.56 | Alex_Bkash | ChannelZ i am Create IAX2 trunk within a openvpn tunnel. how can i check if one tunnel is down and use failover? |
| 09:40.03 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 09:42.21 | Alex_Bkash | i am Create IAX2 trunk within a openvpn tunnel. how can i check if one tunnel is down and use failover? |
| 09:55.57 | *** join/#asterisk ^rage^ (~mts@213.21.4.41) |
| 10:10.18 | *** join/#asterisk chendy (~chatzilla@113.116.62.69) |
| 10:36.57 | *** join/#asterisk evilman_home (kvirc@95-29-217-180.broadband.corbina.ru) |
| 10:42.53 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-176-240.red.bezeqint.net) |
| 10:44.15 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 11:01.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:15.31 | ChannelZ | Man. Shit-tons of SIP drivebys this weekend. |
| 11:37.30 | *** join/#asterisk ghost75 (~ghost75@dslb-088-066-176-035.pools.arcor-ip.net) |
| 11:54.25 | ghost75 | the size of chan_sip.so depends on how much options were selected before compile? |
| 11:56.10 | ghost75 | my compiled one is 4mb whereas the one from debian has only 600kb |
| 11:57.19 | WIMPy | Did you enable debugging options? What about optimisation? |
| 12:00.17 | ghost75 | didnt change anything in menuselect |
| 12:01.54 | ghost75 | wanted to replace only this single file with blf patch, but asterisk didnt work after |
| 12:09.22 | WIMPy | That's what I said yesterday. Usually you can't just copy over modules from another build of the same version. |
| 12:10.17 | ghost75 | can i findout which options the debian version used during compile? |
| 12:10.20 | WIMPy | So remove the package install and do the whole thing. |
| 12:14.50 | ghost75 | the asterisk-dev package from debian has only header files lol |
| 12:15.16 | Gugge | that is usually what the -dev packages in ubuntu has |
| 12:15.43 | ghost75 | how i compile without .c files then |
| 12:18.10 | WIMPy | Obviousely not |
| 12:22.44 | ghost75 | what this messages say during patch: Hunk #23 succeeded at 7284 (offset 23 lines). |
| 12:24.37 | WIMPy | The code to be patched was found 23 lines away from the expected position. |
| 12:25.25 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 12:46.27 | ghost75 | WIMPy by optimizing you mean to manually select the needed packages? |
| 12:50.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 12:52.02 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 12:52.04 | WIMPy | No. Compiler options. |
| 12:52.24 | ghost75 | no idea |
| 12:59.28 | ghost75 | now i found, DONT_OPTIMIZE is not enabled |
| 13:00.21 | ghost75 | do i need to select cpu somewhere? |
| 13:07.28 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 13:16.19 | ghost75 | <PROTECTED> |
| 13:16.24 | ghost75 | i have ncurses installed |
| 13:17.17 | *** join/#asterisk ptruman (522e89a8@gateway/web/freenode/ip.82.46.137.168) |
| 13:18.22 | ptruman | Hello. |
| 13:18.54 | ptruman | Any managed to get Google Talk on *11 to work nicely? I've just upgraded from 10.8 to 11 (latest download from last night) and just fired up GTalk |
| 13:19.15 | ptruman | I get call notifications in & out, but no audio. |
| 13:19.30 | ptruman | icesupport is on |
| 13:22.21 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:f185:ab1f:d72d:8a51) |
| 13:23.58 | ptruman | look like rtp is misbehaving (rtp set debug on shows activity for local things (like voicemail) but NOTHING for Gtalk) |
| 13:25.55 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 13:30.07 | ghost75 | is this for a fully working * or are these just additional selections: http://pastebin.com/wLYUB6Cy |
| 13:31.23 | ptruman | default compile, but with a forced used of openssl |
| 13:31.48 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 13:38.35 | ghost75 | what happens if i place this file in asterisk src path, will it add those options or replace all with it |
| 13:47.33 | ghost75 | aha just figured this file contains all which to remove |
| 13:55.47 | *** join/#asterisk din3sh (~din3sh@41.136.80.219) |
| 13:56.02 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 13:57.58 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 14:03.11 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:08.13 | ptruman | anybody? |
| 14:08.16 | ptruman | bueller? :) |
| 14:13.13 | ghost75 | i was just wondering what iksemel is in compile options |
| 14:17.16 | din3sh | ghost75: you got your cisco phone working finally? |
| 14:17.24 | ghost75 | yes |
| 14:18.15 | ghost75 | i copied menuseelct.makeopts file to source and get errors like this: The existing menuselect.makeopts file did not specify |
| 14:18.16 | ghost75 | <PROTECTED> |
| 14:19.28 | ghost75 | depends on: iksemel |
| 14:19.40 | ghost75 | do i need to install iksemel to compile asterisk with gtalk ? |
| 14:22.31 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:25.20 | *** join/#asterisk sipman (~slane@71-14-128-129.dhcp.ftwo.tx.charter.com) |
| 14:25.46 | *** part/#asterisk sipman (~slane@71-14-128-129.dhcp.ftwo.tx.charter.com) |
| 14:26.51 | *** join/#asterisk sipman (~slane@71-14-128-129.dhcp.ftwo.tx.charter.com) |
| 14:27.03 | *** part/#asterisk sipman (~slane@71-14-128-129.dhcp.ftwo.tx.charter.com) |
| 14:32.12 | ghost75 | install_prereq install <- do i really need to install that all? 182 packages! |
| 14:37.13 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:38.55 | din3sh | locate the iksemel package |
| 14:40.44 | ghost75 | when i run make menuselect and do save, will it remove everything with XXX automatically what was previously select in menuselect.makeopts? |
| 14:46.15 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 14:52.01 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-176-240.red.bezeqint.net) |
| 14:52.03 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 14:54.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:54.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:56.23 | *** join/#asterisk mattsl (~user@174.sub-97-233-242.myvzw.com) |
| 15:03.58 | ptruman | damnit. even gone to asterisk svn trunk now (having seen some xmpp fixes listed) and it's still now working |
| 15:04.01 | ptruman | not working even |
| 15:06.00 | ghost75 | i installed libspandsp-dev and res_fax_spandsp still complains |
| 15:12.22 | din3sh | why dont you install all the packages |
| 15:12.28 | din3sh | less hassle |
| 15:12.38 | ghost75 | too much crap |
| 15:12.43 | ptruman | has anyone got xmpp audio with gtalk working? |
| 15:13.12 | ghost75 | what is chan_alsa used for? |
| 15:13.41 | din3sh | yum install spandsp |
| 15:14.17 | ghost75 | it was libspandsp-dev |
| 15:23.02 | ghost75 | poor atom compiling now |
| 15:31.05 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-176-240.red.bezeqint.net) |
| 15:39.07 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:39.07 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:41.01 | *** join/#asterisk tamiel (~tamiel@AToulouse-257-1-65-5.w90-5.abo.wanadoo.fr) |
| 15:44.26 | ghost75 | its irritating that my compiled chan_sip.so has 4mb and the one from debian 650kb, how this can be possible? |
| 15:45.46 | prgmrchris | statically compiled? |
| 15:45.57 | prgmrchris | not stripped? |
| 15:46.32 | ghost75 | stripped? static? |
| 15:47.01 | prgmrchris | those are things i would check that could be the reason its so much bigger |
| 15:47.33 | prgmrchris | run ldd chan_sip.so and file chan_sip.so |
| 15:47.39 | prgmrchris | see what they say |
| 15:50.40 | ghost75 | http://pastebin.com/EiuTz3GL |
| 15:51.51 | prgmrchris | so its dynamic with same linked libs |
| 15:52.10 | prgmrchris | what about running file chan_sip.so against both modules |
| 15:53.02 | ghost75 | against both modules? |
| 15:57.34 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 15:59.27 | ghost75 | like so? http://pastebin.com/HaE44KhM |
| 16:14.54 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 16:16.57 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.191.222) |
| 16:17.17 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.191.222) |
| 16:19.03 | prgmrchris | ghost75: run the command "file /usr/lib/asterisk/modules/chan_sip.so" and "file /home/trechber/asterisk-1.8.14.0/channels/chan_sip.so" |
| 16:20.42 | ghost75 | oh didnt know thats a cmd |
| 16:21.17 | WIMPy | You don't know the file magic? ;-) |
| 16:21.46 | ghost75 | ELF 32-bit LSB shared object, Intel 80386, version 1 (SYSV), dynamically linked, not stripped |
| 16:21.49 | ghost75 | both ^^ |
| 16:22.01 | ghost75 | not |
| 16:22.05 | ghost75 | one says stripped |
| 16:22.39 | ghost75 | whats that sort of striptease |
| 16:23.01 | WIMPy | Strip without tease. |
| 16:23.23 | ghost75 | is teasing me a lot |
| 16:25.42 | prgmrchris | is it the debian one thats stripped? |
| 16:25.49 | prgmrchris | if so theres your size problem most likely |
| 16:26.19 | prgmrchris | you can run "strip /home/trechber/asterisk-1.8.14.0/channels/chan_sip.so" then check the size again see how much it changes |
| 16:26.25 | WIMPy | It can't make that much of a difference, can it? |
| 16:26.34 | prgmrchris | just a guess |
| 16:26.53 | prgmrchris | i thought it was more likely huge due to static compilation but stripping can be some of it too |
| 16:27.27 | WIMPy | It surely makes much more of a difference than I would have expected. |
| 16:27.34 | ghost75 | yes debian is stripped |
| 16:27.44 | prgmrchris | ghost75: so strip yours and see how much smaller it gets |
| 16:27.49 | WIMPy | From 2.2M to 871K for me. |
| 16:27.50 | prgmrchris | WIMPy: yeah i know its surprising sometimes |
| 16:27.53 | prgmrchris | yeap |
| 16:27.54 | ghost75 | is this just making smaller file or is it faster after? |
| 16:28.02 | prgmrchris | smaller |
| 16:28.07 | prgmrchris | removes debugging symbols in the binary |
| 16:30.29 | ghost75 | 764kb now |
| 16:31.01 | ghost75 | 4,2mb before |
| 16:33.09 | prgmrchris | :) |
| 16:40.36 | ghost75 | hmmm apt-get remove asterisk didnt remove the moduels |
| 16:46.38 | ghost75 | ok asterisk is started |
| 16:46.42 | ghost75 | now i can haz blf ? |
| 16:53.54 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 16:55.38 | din3sh | ghost75:https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 16:55.58 | ghost75 | thats the reason i compiled |
| 16:56.36 | din3sh | ? |
| 16:57.01 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 16:57.06 | ghost75 | u need to compile asterisk in order to use patch |
| 16:58.08 | din3sh | you need to patch asterisk before compiling you mean? |
| 16:58.12 | din3sh | I did |
| 16:58.31 | din3sh | running some 100 cisco phones with SIP firmware 9.x.x |
| 16:58.37 | ghost75 | of course it needs |
| 16:58.48 | ghost75 | i had installed asterisk from debian repository |
| 17:00.02 | din3sh | ah ok |
| 17:00.07 | din3sh | didnt compile from source |
| 17:00.12 | ghost75 | i think i miss g711 codec |
| 17:05.12 | din3sh | apply the patch |
| 17:05.22 | din3sh | you let me know how it works out |
| 17:09.35 | ghost75 | it works :) |
| 17:09.39 | ptruman | where does chan_motif pick up it's ip from? gtalk.conf had a bindaddr, and chan_motif doesn't? |
| 17:10.05 | ptruman | hacking down my xmpp logs I'm seeing internal IPs in the messaging which may explain my lack of audio |
| 17:10.12 | din3sh | mine too |
| 17:10.24 | din3sh | but i have these crazy publish requests on my cli |
| 17:10.58 | ghost75 | do you have mwi working? |
| 17:11.28 | din3sh | yes |
| 17:11.53 | ghost75 | only that red light on handset goes on when new voicemail or? |
| 17:12.31 | din3sh | yup only when there is new vm |
| 17:12.59 | ghost75 | why is then there this buggymwi parameter |
| 17:13.26 | ghost75 | i didnt set it |
| 17:14.22 | ghost75 | and somewhere is written that qualify=yes should not be used for presence |
| 17:15.27 | din3sh | i have set qualify=no |
| 17:15.45 | ghost75 | i have yes and works so far |
| 17:20.47 | din3sh | i received tons of these [Feb 7 09:19:06] NOTICE[17686] chan_sip.c: Failed to authenticate device <sip:58971ecd3a3a@192.168.3.240>;tag=58971ecd323a0256259fd365-5c17add9 |
| 17:21.43 | ghost75 | how often |
| 17:22.39 | din3sh | every 5 secs |
| 17:22.52 | ghost75 | one phone every 5 ? |
| 17:22.53 | din3sh | multiply by the number of phones I have |
| 17:22.57 | din3sh | yeah |
| 17:23.05 | din3sh | cli cluttered |
| 17:23.20 | ghost75 | i dont get any |
| 17:24.11 | din3sh | which xml file you used? |
| 17:24.36 | ghost75 | somebody here from hawai but i changed it a bit |
| 17:25.08 | ghost75 | i used featureid 21 |
| 17:25.50 | ghost75 | and <featureOptionMask>1</featureOptionMask> |
| 17:29.39 | ghost75 | din3sh: which asterisk? |
| 17:30.46 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:30.46 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:31.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 17:32.38 | ptruman | somewhat unhelpful :\ |
| 17:32.48 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 17:34.27 | din3sh | ghost75:1.8.14 |
| 17:34.36 | ghost75 | same |
| 17:37.19 | din3sh | have a look at my xml http://pastebin.com/S6xsUAcZ |
| 17:37.38 | ghost75 | http://pastebin.com/LfW8xy1v |
| 17:38.37 | din3sh | using 1.8.14 i uncovered another issue |
| 17:38.39 | din3sh | https://issues.asterisk.org/jira/browse/ASTERISK-19610 |
| 17:38.54 | din3sh | affects 1.8.x till 1.8.18 |
| 17:39.06 | ghost75 | maybe try |
| 17:39.07 | *** join/#asterisk bytemaster (~erw@host81-150-217-168.in-addr.btopenworld.com) |
| 17:39.08 | ghost75 | <!-- Force short registration timeout to keep NAT connection alive --> <timerRegisterExpires>180</timerRegisterExpires> |
| 17:39.32 | din3sh | for some incoming numbers through dahdi, E1, fast dtmf keys do not work |
| 17:39.35 | WIMPy | Weren't you using TCP? |
| 17:39.37 | *** join/#asterisk mattsl (~user@84.sub-97-190-201.myvzw.com) |
| 17:39.49 | ghost75 | yes cisco uses tcp |
| 17:40.15 | WIMPy | Pretty senseless to re-register at all then. |
| 17:42.02 | ghost75 | i think the file was from kikohnl |
| 17:42.54 | din3sh | I dont have NAT |
| 17:42.55 | ghost75 | dsp.c stopped being able to decode a quick DTMF sequence starting in 1.8.10rc1. 50ms tone, 50ms silence does not work anymore. <- this is when pressing buttons? |
| 17:42.59 | din3sh | am not using TCP |
| 17:43.02 | din3sh | using udp |
| 17:43.10 | ghost75 | try tcp |
| 17:43.26 | din3sh | tcp had some issues when i tested |
| 17:43.35 | din3sh | i dont remember which issues though |
| 17:44.05 | ghost75 | <transportLayerProtocol>2</transportLayerProtocol> |
| 17:44.06 | ghost75 | this right? |
| 17:44.28 | din3sh | dsp.c stopped being able to decode.... i became nuts trying to figure why a call from S.Africa couldnt key in dtmf correctly |
| 17:44.44 | din3sh | happens if dtmf is keyed rapidly, asterisk doesnt understand it |
| 17:45.52 | WIMPy | Is it done in Asterisk or in dahdi? |
| 17:47.22 | ghost75 | <PROTECTED> |
| 17:47.28 | ghost75 | this my phone does on boot on cli |
| 17:48.34 | din3sh | <transportLayerProtocol>2</transportLayerProtocol> ---> udp |
| 17:48.42 | din3sh | <transportLayerProtocol>4</transportLayerProtocol> ---> tcp |
| 17:48.43 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-176-240.red.bezeqint.net) |
| 17:48.54 | ghost75 | ok |
| 17:49.01 | din3sh | WIMPy: asterisk |
| 17:49.19 | ghost75 | with udp my phone doesnt register |
| 17:49.44 | WIMPy | din3sh: That part can be configured IIRC. |
| 17:50.01 | din3sh | have you set udpbindaddr= in sip.conf? |
| 17:50.27 | din3sh | WIMPy: how? |
| 17:50.44 | ghost75 | yes 0.0.0.0 and other phones are registering over udp |
| 17:51.01 | WIMPy | If I'd remember where I read about it... |
| 17:51.07 | din3sh | tcp works for you? |
| 17:51.15 | ghost75 | yes |
| 17:51.23 | ghost75 | maybe u get those errors because of udp |
| 17:51.36 | din3sh | smtimes my 7911 would not receive any calls even registered |
| 17:51.39 | din3sh | on tcp |
| 17:51.48 | din3sh | have to test |
| 17:51.57 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.191.222) |
| 17:52.09 | din3sh | i've already deployed 100 phones , udp in production |
| 17:52.13 | WIMPy | That might be because of (missing) reuse of connections. |
| 17:52.35 | ghost75 | i read that tcp is recommended with cisco |
| 17:52.39 | WIMPy | There have been fixes to the tcp stuff. |
| 17:54.32 | din3sh | i have another problem, havent debugged though |
| 17:54.50 | din3sh | directed pickup smtimes hangs up calls after 30secs |
| 17:55.37 | ghost75 | when u pickup call from other extension? |
| 17:56.23 | ghost75 | btw is there way to remote boot those phones |
| 17:56.41 | ghost75 | putty i cannot connect to phone |
| 17:57.35 | din3sh | you can reboot the phones |
| 17:57.39 | din3sh | using sip notify |
| 17:58.04 | ghost75 | by sending sip packet to it? |
| 17:58.15 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 17:58.35 | din3sh | sip notify cisco-reset $EXTEN |
| 17:59.10 | ghost75 | do you have ssh working? |
| 17:59.36 | din3sh | nope |
| 18:03.57 | din3sh | why is tcp better than udp? |
| 18:05.18 | WIMPy | Who says so? |
| 18:05.27 | ghost75 | maybe not better but maybe working |
| 18:06.27 | ghost75 | notify works |
| 18:06.45 | *** join/#asterisk BrokenArrow (~Lp@unaffiliated/brokenarrow) |
| 18:07.14 | din3sh | just asking WIMPy |
| 18:07.34 | ghost75 | i read also that non cisco phones are making cisco phones not work -.- |
| 18:07.40 | WIMPy | Then my answer is: Not |
| 18:17.10 | din3sh | <ghost75> i read also that non cisco phones are making cisco phones not work -.- |
| 18:17.23 | din3sh | not really, i have a blend of various phone brands |
| 18:20.01 | ghost75 | time is not showing correctly |
| 18:21.07 | ghost75 | Central Standard/Daylight Time <- where is that |
| 18:21.38 | *** join/#asterisk MxG (quassel@london.atplanet.co.za) |
| 18:22.41 | din3sh | no idea :p |
| 18:22.50 | ghost75 | i think us |
| 18:23.01 | ghost75 | 12:22pm |
| 18:30.10 | ghost75 | language seems to be us only on sip |
| 18:30.40 | din3sh | you cant change the Locale? |
| 18:30.53 | ghost75 | cisco says that sip is us only |
| 18:43.37 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 18:50.23 | *** join/#asterisk WebSprocket (~RaptorXI@static-159-253-74-115.karoo.kcom.com) |
| 18:51.42 | WebSprocket | Hey, Im having an issue with one way audio with an ATA outside the network, I have the ports forward for SIP and RTP but I can hear them but they cannot hear me |
| 18:52.23 | WIMPy | The the issue is at the other end. |
| 18:53.03 | WebSprocket | using a Zyxel P-2812HNU-F1 |
| 18:53.14 | WebSprocket | didn't used to have a problem, it only just started happening |
| 18:55.12 | WebSprocket | I have noted int he log i get this a lot though Peer '132' is now Lagged. (2174ms / 2000ms) |
| 18:56.42 | [TK]D-Fender | Your phone is talking longer than the qualify timeout to respond. |
| 18:56.55 | [TK]D-Fender | so either bump the time or remove its use |
| 19:10.39 | ghost75 | din3sh: do you have xml directory? |
| 19:13.44 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-176-240.red.bezeqint.net) |
| 19:13.48 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:18.22 | *** join/#asterisk tamiel (~tamiel@AToulouse-257-1-65-5.w90-5.abo.wanadoo.fr) |
| 19:19.10 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:19.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:25.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:25.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:26.38 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 19:26.46 | *** part/#asterisk WebSprocket (~RaptorXI@static-159-253-74-115.karoo.kcom.com) |
| 19:27.56 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 19:38.21 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 19:41.01 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 19:41.02 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:41.50 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 19:58.11 | ghost75 | whats going on here |
| 19:58.45 | WIMPy | No early media any more, |
| 19:58.47 | file | stuff and things. |
| 20:00.08 | ghost75 | my apache lists inc files as clear text lol |
| 20:16.09 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 20:18.23 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 20:25.44 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 20:36.22 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 20:39.53 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:49.19 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:14.46 | *** join/#asterisk nightrid3r (~kvirc@62.205.67.108) |
| 21:22.53 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 21:26.20 | *** join/#asterisk Dovid (~Dovid@float-gw.core-fw-t.mypbxmanager.net) |
| 21:39.29 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:39.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:55.41 | *** join/#asterisk ghost75 (~ghost75@dslb-088-064-048-050.pools.arcor-ip.net) |
| 21:57.02 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 21:58.50 | *** join/#asterisk mjordan (~mjordan@216.186.152.216) |
| 21:58.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:08.48 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 22:14.04 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 22:16.36 | *** join/#asterisk bigd1234567890 (~dburgess@adsl-99-183-176-49.dsl.hstntx.sbcglobal.net) |
| 22:16.59 | bigd1234567890 | can anyone help me to add a delay before dialing pstn? |
| 22:17.30 | bigd1234567890 | i'm getting error message from pstn saying "your call cannot be completed as dialed". i see the error about 50 percent of the time |
| 22:18.18 | ChannelZ | put a w in before |
| 22:18.33 | ChannelZ | like Dial(DAHDI/1/w5551212) |
| 22:18.40 | bigd1234567890 | where does it go? i'm running asterisknow |
| 22:18.53 | bigd1234567890 | version 2.0.2 |
| 22:18.59 | ChannelZ | well for that I have no idea |
| 22:19.20 | bigd1234567890 | the above command goes in which file? |
| 22:19.38 | ChannelZ | well it's extensions.conf as part of the dialplan but all of that is generated by the GUI so you're a slave to how the GUI does things. |
| 22:20.12 | ChannelZ | You'd have to hunt around and see if they let you add prefix digits or something along those lines |
| 22:20.56 | bigd1234567890 | ah, kk. that gives me something to look at |
| 22:21.24 | ChannelZ | sorry I have no idea what it might be called in there or where. I don't do GUIs |
| 22:26.33 | *** join/#asterisk eirirs (eirirs@dalvik.ping.uio.no) |
| 22:27.04 | eirirs | anyone been into an asterisk with two internal lan ips and one external ? |
| 22:27.27 | [TK]D-Fender | yes |
| 22:28.35 | eirirs | it's ... I have two subnets 10.0.0.* and 10.0.3.* , with phones at both subnets... asterisk at 10.0.0.* net, and port 5060 is being forwarded by the router out to NAT, BUT 10.0.3.* are from a VPN connection with another NAT behind it ... so if I try with a phone from 10.0.3.* it will never get any reply from PBX because of the port forwarding, so I had to create another ip address only for |
| 22:28.41 | eirirs | internal usage |
| 22:29.02 | eirirs | tries to let phone register at internal IP #2, but pbx keep trying to answer by internal ip #1 and fails... |
| 22:29.28 | eirirs | lol hope anybody understood what I tried to write |
| 22:29.52 | eirirs | can I have multiple localnet= lines in sip conf? |
| 22:30.09 | eirirs | or do I have to cheat with lowering the /24 numbers? :p |
| 22:30.33 | [TK]D-Fender | [17:29]eirirscan I have multiple localnet= lines in sip conf? <- yes |
| 22:30.36 | WIMPy | No. I didn;t get whete the NATtings happen. |
| 22:30.57 | [TK]D-Fender | eirirs: And your descrption of where NAT is actually involved is very poorly worded and unclear |
| 22:31.01 | WIMPy | Usually you can just list the RFC1918 Block in localnets. |
| 22:31.25 | eirirs | I agree |
| 22:31.45 | eirirs | I've got PBX at 10.0.0.0/24 , and got 10.0.3.0/24 linked by VPN |
| 22:31.46 | WIMPy | Might be a good idea to draw a (ASCII) picture. |
| 22:33.01 | eirirs | router is forwarding 10.0.3.0/24 packets from PBX to outside IP instead of sending it to VPN |
| 22:33.10 | eirirs | because of the port 5060 forward rule I guess |
| 22:33.30 | eirirs | .. um that's my guess why the phone never gets any answer from PBX, that's |
| 22:33.59 | [TK]D-Fender | eirirs: Sounds like you didn't specify the local subnet and probably didn't set up your routes properly |
| 22:34.00 | eirirs | so thats why I made a second 10.0.0.0/24 address for my PBX, that don't have a forward rule put on it on router |
| 22:34.35 | eirirs | phone register on the second address, but pbx keep trying to answer to the first address |
| 22:35.06 | eirirs | guess I'm doing something very wrong :) what would be a ideal setup if one have multiple subnets, with VPN ? |
| 22:35.24 | WIMPy | Usually nothing special. |
| 22:36.16 | eirirs | if I tell the phone to use TCP, pbx says peer is unreachable, if I tell phone to use UDP, pbx says unauthorized |
| 22:36.40 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 22:36.40 | eirirs | guess thats problem with backroute |
| 22:36.43 | WIMPy | That's something entirely different. |
| 22:36.44 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 22:36.56 | bigd1234567890 | ChannelZ->I figured out where to leave the wait under the GUI for asterisknow |
| 22:37.08 | [TK]D-Fender | eirirsI've got PBX at 10.0.0.0/24 , and got 10.0.3.0/24 linked by VPN <- I am not hearing a Nat involved/not involved clarification here |
| 22:37.17 | [TK]D-Fender | eirirs: Nor are we seeing configs or debug |
| 22:38.08 | bigd1234567890 | CahnnelZ->under edit DAHDI channel, Outbound Dial Prefix field i set to 'ww' and it is now waiting correctly..outbounds work fine |
| 22:38.10 | WIMPy | Seems rather clowdy to me. Give us the full picture. And dont mix networking issues with application level issues. |
| 22:38.31 | eirirs | router have static route for port 5060 for the PBX's LAN ip, I'm wondering that causes the problem |
| 22:39.02 | WIMPy | The FULL picture. |
| 22:39.25 | eirirs | http://pastie.org/6112742 |
| 22:39.27 | eirirs | thats the debug |
| 22:39.35 | eirirs | when phone is on TCP |
| 22:40.12 | WIMPy | Looks like you firewalled yourself off. |
| 22:41.19 | ChannelZ | bigd1234567890: cool glad you found it |
| 22:41.33 | eirirs | http://pastie.org/6112754 |
| 22:41.37 | eirirs | that's when it's in UDP mode |
| 22:43.12 | eirirs | do it look like firewall issue? |
| 22:44.41 | WIMPy | You gave too little information to answer that. But in the first exaple it's pretty clear. |
| 22:44.59 | [TK]D-Fender | eirirs: Those are just OPTIONS packets, not actual calls. Your description is STILL bad. |
| 22:45.10 | [TK]D-Fender | eirirs: is ther a NAT across your VPN bridge or not? |
| 22:45.26 | [TK]D-Fender | eirirs: Do not give us a story about this. it is a yes/no answer |
| 22:45.44 | eirirs | [TK]D-Fender: NAT - router - VPN - router - NAT |
| 22:46.02 | WIMPy | And wehre is Asterisk and where is the phone? |
| 22:46.08 | eirirs | at both ends |
| 22:46.19 | [TK]D-Fender | eirirs: Terrible path description again. |
| 22:46.36 | eirirs | 10.0.0.0/24 NAT (and pbx and working phones) - router - VPN - router - NAT (10.0.3.0/24 and not working phone) |
| 22:46.38 | [TK]D-Fender | eirirs: I do not see * in there. Ho wis "NAT" separate from "router"? |
| 22:47.10 | WIMPy | So you have the phones in one net and Asterisk in another and both are NATted to some VPN? |
| 22:47.20 | eirirs | it's a nat to nat VPN |
| 22:47.29 | eirirs | I can access either hosts from both sides fine |
| 22:47.43 | WIMPy | That sucks. |
| 22:47.53 | WIMPy | But that also means that the other LAN is NOT local. |
| 22:48.19 | eirirs | I can ping local ips from the other side of VPN |
| 22:48.29 | [TK]D-Fender | eirirsI can access either hosts from both sides fine <- then it is not NAT |
| 22:48.38 | [TK]D-Fender | NAT ffunnel all through a common IP |
| 22:48.39 | WIMPy | Where is the NAT then? |
| 22:49.08 | eirirs | guess the NAT isn't visible for the phone - pbx |
| 22:49.24 | WIMPy | That's not how NAT works. |
| 22:51.32 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 22:51.49 | eirirs | ah |
| 22:52.16 | eirirs | when I removed the static portforward from PBX's local LAN ip to the internet in the router, bang, phone registered to PBX fine |
| 22:52.26 | eirirs | at port 5060 |
| 22:52.43 | WIMPy | Where is the Internet involved? |
| 22:52.50 | eirirs | PBX and out to ISP trunk |
| 22:53.14 | eirirs | for ISP, PBX is behind a NAT |
| 22:53.55 | WIMPy | I guess there's a chance there really is a NAT there. |
| 22:54.52 | eirirs | the router have both NAT to internet and VPN to the other subnet running |
| 22:56.32 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 22:56.45 | [TK]D-Fender | eirirs: chances ar the VPN is *NOT* NAT'd |
| 22:56.46 | eirirs | when I put static port 5060 to the internet, then I can't use port 5060 to VPN I guess, so I have to set up a secondary LAN ip for the PBX |
| 22:57.08 | [TK]D-Fender | eirirs: And stop guessing. You also haven't shown a call attempt |
| 22:57.19 | eirirs | phone don't work |
| 22:57.22 | [TK]D-Fender | Or your configs |
| 22:57.41 | WIMPy | You probably forwarded the port at the wrong point. |
| 22:57.53 | eirirs | but as soon as I remove the port 5060 static forward to internet, phone registers and works fine, but then the trunk out to ISP PBX are down |
| 22:58.44 | [TK]D-Fender | eirirs: We don't see these "forwards" you're doing or know what equipment you're using. |
| 22:58.53 | [TK]D-Fender | eirirs: Do you actually want help? You've shown us nothing. |
| 22:59.58 | eirirs | ip nat inside source static udp 10.0.0.2 5060 <ip to ISP's PBX> 5060 |
| 23:00.07 | eirirs | that line causes the problem, in the cisco 1841 IOS router |
| 23:00.20 | eirirs | I remove it, phone register to the PBX fine |
| 23:00.41 | WIMPy | You need to make sure the rule only affects traffic from the WAN interface. |
| 23:00.55 | WIMPy | Or you find out why you need it at all. |
| 23:00.56 | eirirs | but VPN go to the same WAN interface |
| 23:01.12 | eirirs | I need it for the trunk to the ISP PBX |
| 23:01.17 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.51 | WIMPy | You need it because the NAT doesn't do it by itself. |
| 23:02.12 | WIMPy | And the VPN traffic doesn;t come from the WAN *interface*. |
| 23:02.24 | WIMPy | It's just a transport. |
| 23:02.41 | WIMPy | That's the idea of tunneling. |
| 23:04.34 | eirirs | what info and configs would you like me to show ? |
| 23:05.38 | WIMPy | None. Ask someone who knows how to configure Ciscos. |
| 23:06.23 | eirirs | when I try phone in TCP mode, it looks like it don't help that I remove the NAT line. |
| 23:06.58 | eirirs | it says unreachable in sip show peers |
| 23:07.01 | eirirs | and keep retransmitting |
| 23:07.19 | [TK]D-Fender | eirirs: Where is the call attempt? |
| 23:07.30 | [TK]D-Fender | eirirs: You are not showing actual comms coming FROM the phone |
| 23:08.36 | eirirs | I do not understand? do you mean I should lift the phone and press any numbers to make any call? |
| 23:08.50 | jzaw | eirirs, is <ip to ISP's PBX> an ip on an interface on the cisco which i presume is your router/firewall? |
| 23:09.09 | [TK]D-Fender | moves on to more productive matters |
| 23:09.11 | eirirs | jzaw: global internal ip |
| 23:09.24 | eirirs | inside global ip |
| 23:09.43 | jzaw | but its on the cisco for it to be natting? and routed to and from |
| 23:10.08 | eirirs | yes |
| 23:10.13 | jzaw | k np |
| 23:10.19 | eirirs | between pbx and isp pbx |
| 23:10.59 | *** join/#asterisk Devon_ (~Devon_@adsl-184-32-199-39.mia.bellsouth.net) |
| 23:17.37 | *** join/#asterisk darkskiez_ (~dz@fsf/member/darkskiez) |
| 23:20.10 | eirirs | looks like I have to conclude that I need a secondary LAN ip address for my PBX, and that's my first quesition, is it as simple as just set it up in ifconfig and make phone register to that IP, and no more additional config needed? |
| 23:20.53 | WIMPy | You need to get your routing right anyway. |
| 23:21.19 | WIMPy | Even for that workaround. |
| 23:21.34 | eirirs | I tested with other services at other servers, ssh on port 22 to lan ip stopped working, only ssh to port 22 to internet ip works if I have a static port forward up |
| 23:21.58 | *** join/#asterisk MxG (~quassel@london.atplanet.co.za) |
| 23:22.26 | WIMPy | Find out how to configure the forwarding correctly. |
| 23:22.49 | ChannelZ | With so little working, at least your VPN is secure. |
| 23:23.04 | jzaw | eirirs, another question .. i dont speak cisco but ...... nat source .... should that maybe nat destination 10.x.x.x |
| 23:23.07 | eirirs | lol |
| 23:23.14 | jzaw | forgive any cisco ignorance |
| 23:26.59 | *** join/#asterisk pcAngel (yoink@S0106602ad072bdf0.vc.shawcable.net) |
| 23:28.28 | pcAngel | Hi guys, I'm working on making two asterisk servers redundant on a master/slave type model -- is it possible to forward registrations from one asterisk server (slave) to the second (master), or to have it so that when a peer connects to the slave, the slave connects to the master in order to register for inbound calls? |
| 23:28.39 | *** join/#asterisk Devon_ (~Devon_@adsl-184-32-199-39.mia.bellsouth.net) |
| 23:31.15 | pcAngel | Ie - Peers 1/2/3/4/5 need to be able to call each other; S1 is the slave and S2 is the master. If peer 1/2 are resolving dns to S1(slave) and peer 3/4/5 are resolving to S2 (master) --- peers 1/2 would betold to re-register to master, or S1 would register to S2 using the peers credentials, so that if a call from P3/4/5 is directed to P1/2 then it would send to P1@S1 and then to |
| 23:31.15 | pcAngel | P1@myclient |
| 23:32.04 | pcAngel | the end effect is that I want the two servers to look and act like one |
| 23:32.14 | WIMPy | Do you want fail-over or do you want load balancing? |
| 23:32.40 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:32.42 | pcAngel | fail-over, but the part I'm concerned about is while DNS is in the process of changing |
| 23:32.58 | pcAngel | and possibly to implement RR DNS |
| 23:33.17 | WIMPy | That doesn't fit together. |
| 23:33.41 | WIMPy | With fail over a change would happen if the former master failed. |
| 23:34.08 | pcAngel | Right, but at the moment I'm manually updating the DNS as well as updating a mysql row indicating which server is the master |
| 23:34.30 | pcAngel | The two servers have a set of mysql databases on master-master replication |
| 23:34.48 | pcAngel | and in the db I indicate which server is master |
| 23:34.58 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 23:35.04 | pcAngel | if a client has issues with one server (ie latency) but not the other, I update the DB and DNS to have them resolve solely to the other server |
| 23:35.48 | WIMPy | Sounds more like some advanced load balancing. |
| 23:35.54 | pcAngel | And from there I don't want the client to have to refresh for the changes to be active, anyone still on the new slave server would hopefully be able to call the people on the new master |
| 23:36.10 | pcAngel | Yeah it's kind of manual client site-based load balancing |
| 23:36.38 | WIMPy | Pretty easy. If you can't reach a peer, try the other server next. |
| 23:37.47 | pcAngel | kk. I know I can do that by calling through dialplans, and then the other side of it is my predictive dialer script which forwards calls to queues that include members assigned to that dialer |
| 23:37.59 | pcAngel | The members are currently set up as SIP/clientId-stationId |
| 23:38.31 | pcAngel | which I could change to Local/stationId@clientId_clientTable |
| 23:38.45 | pcAngel | which could handle the failover |
| 23:38.51 | pcAngel | Is that the best solution? |
| 23:39.17 | WIMPy | You don't have to. |
| 23:39.36 | WIMPy | Just diel the peer and if the Dial() fails try again via the other server. |
| 23:40.02 | pcAngel | Right but in the context of a queue member |
| 23:40.28 | pcAngel | What's the best way to accomplish that, to keep one active queue on the master that will distribute to all peers, between both servers? |
| 23:40.40 | pcAngel | I kind of changed the rules on you :p sorry for not being clear |
| 23:41.02 | WIMPy | You'd have to have all queues on only one server. |
| 23:41.23 | pcAngel | I just figure there must be some kind of elegant solution to this, ie peer 100-1 connects to slave, then slave connects to 100-1@master and forwards the calls, or slave says go away, re register to 100-1@master |
| 23:41.38 | WIMPy | You can chek your db to find out if you queue the call yourself or if you have to send it to the other server. |
| 23:42.29 | pcAngel | right I could probably add the peer as SIP/100-1@master |
| 23:43.12 | pcAngel | in that case I don't think it'd keep track of whether or not the peer is available, though |
| 23:43.46 | WIMPy | That's two different things. |
| 23:43.53 | WIMPy | Or isn;t it about queues but about agents? |
| 23:44.19 | pcAngel | it's both, but having agents able to dial each other is easy to work out by failover dial commands |
| 23:44.30 | pcAngel | which leaves the queues to sort out |
| 23:44.34 | WIMPy | yes |
| 23:44.58 | WIMPy | The queues themselves should be almost as easy. |
| 23:45.19 | WIMPy | But states of agents wil be some issue. |
| 23:45.47 | pcAngel | My AMI scripts remove the user from queues within a few seconds of a call being answered by them, but there's still a few second delay where if they aren't reported as unavailable, then multiple calls can be sent to them |
| 23:46.22 | pcAngel | which I've had to resolve by auto-rejecting secondary calls that were set to AA, within my webphone API, because I have autofill=yes |
| 23:46.43 | pcAngel | so I can keep doing that, if there's not a more elegant solution :o |
| 23:46.53 | WIMPy | I'm not in to queues. |
| 23:47.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:49.50 | pcAngel | ok. I appreciate your help by the way. Do you know if there is there a way for "slave" to register to "master" as peer 100-1, 100-2 and 100-3, once those peers connect to 'slave'? that would make master have those slaves appear as present. |
| 23:50.17 | WIMPy | Not easily. |
| 23:50.51 | WIMPy | You could try xmpp device states. But Nothing I even looked at. |
| 23:51.26 | pcAngel | if the call gets sent to 100-1 on master, it would forward to slave, likely using a backup peer number like 100-1Bridge, then I would have to have a default context that strips Bridge from the peer ID and sends the call to that peer |
| 23:51.35 | pcAngel | Ok I'll read up on that |
| 23:52.43 | WIMPy | What do you mean by "backup peer number" and "Bridge"? |
| 23:52.53 | pcAngel | I think it may be best to just re-write the server discovery code in my CRM. Right now, if they refresh their screen, it will update them on the correct server to connect to, but if it changes mid-server, it won't reregister |
| 23:53.46 | WIMPy | You just need a context that allows the servers to call any peer. |
| 23:56.02 | pcAngel | if the slave only registered to the master once, though, then it wouldn't be able to detect how any active calls there are from the master to the slave, for a given peer on the slave |
| 23:56.12 | pcAngel | **the master wouldn't be able to detect that |
| 23:56.34 | WIMPy | Do you need that? |
| 23:56.34 | pcAngel | which means that the queues would not know the device state for a peer that is on the slave |
| 23:56.43 | pcAngel | yes |
| 23:57.22 | WIMPy | Ah, that way. Ok. That would be a possibility. |
| 23:58.23 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 23:58.31 | WIMPy | You might be able to use the group functions. But you'd have to do it for all calls then. |
| 23:58.54 | *** join/#asterisk ChannelZ (channelz@burner.com) |