| 00:12.52 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 00:15.17 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 00:39.25 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 00:52.33 | *** join/#asterisk italorossi (~textual@187.61.182.39) |
| 01:12.36 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 01:13.48 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 01:14.07 | tapout | what's another provider in the category of voip.ms ? I love voip.ms, i just want to try another provider along side |
| 01:21.04 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:29.36 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 01:29.59 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 01:39.09 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 02:03.36 | *** part/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 02:05.31 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-17.phlapa.fios.verizon.net) |
| 02:17.17 | *** join/#asterisk cmendes0101 (~cmendes01@pool-98-112-76-215.lsanca.fios.verizon.net) |
| 02:19.41 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 02:26.49 | *** join/#asterisk brdude (~brdude@24.7.76.160) |
| 02:29.43 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:36.36 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 02:37.31 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:43.17 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 02:58.27 | *** join/#asterisk cmendes0101 (~cmendes01@pool-98-112-76-215.lsanca.fios.verizon.net) |
| 03:06.35 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 03:34.54 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 04:09.08 | *** join/#asterisk wma (~wma@203.81.80.94) |
| 04:09.14 | wma | hello |
| 04:09.50 | wma | pls help me something |
| 04:12.27 | jpsharp | You have to say what you need help with. |
| 04:13.45 | wma | yes bro |
| 04:13.55 | wma | The outgoing call is stuck in queue after forwarding to another number. It was not hang up automatically after forwarding and call is still remain in queue. I would like to know how to hangup the remain calls in queue after forwarding to another number in Asterisk |
| 04:16.14 | jpsharp | You can use the "soft hangup" command. |
| 04:17.01 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 04:17.08 | wma | yes bro |
| 04:17.32 | wma | but there is no staff in office we make call forwarding |
| 04:17.50 | wma | in that case after 2 call forwarding there is no more call bro |
| 04:17.51 | wma | :( |
| 04:19.23 | *** join/#asterisk cmendes0101 (~cmendes01@pool-98-112-76-215.lsanca.fios.verizon.net) |
| 04:20.08 | wma | how can i hang up automatically bro |
| 04:24.04 | jpsharp | You can set a maximum call length. |
| 04:27.30 | wma | i c. bro where |
| 04:27.34 | wma | asterisk.conf? |
| 04:30.23 | wma | is there any other solution bro |
| 04:40.09 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-ipxbnnxftfbmwblb) |
| 04:54.28 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 05:02.15 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 05:14.44 | *** join/#asterisk deo_ (~dnepangue@58.71.19.178) |
| 05:43.34 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 05:54.04 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:59.50 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 06:03.46 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:08.35 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 06:11.19 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:20.28 | *** join/#asterisk moy_ (~moy@CPE185933fe7837-CM185933fe7834.cpe.net.cable.rogers.com) |
| 06:24.41 | *** join/#asterisk kannan (~chatzilla@122.164.175.135) |
| 06:47.04 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:47.29 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 06:48.31 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 06:53.56 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:56.25 | kannan | hi, I want to limit a call to X seconds and use L option for this. How can we re direct the channel to an IVR when the 'L' message is played? |
| 06:58.56 | kaldemar | why do you want to use the L() option for that? |
| 07:01.32 | kannan | kaldemar, basically, in a custom call card, i need to xfer to a payment IVR when low balance |
| 07:03.39 | kannan | shorting of running a cron that checks Db entries (with channel details) every second, is there any other way to go for this? |
| 07:03.40 | kaldemar | and then? continue the call? |
| 07:03.43 | kannan | yes |
| 07:03.49 | kannan | contnue the call |
| 07:04.03 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:04.42 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:04.43 | kannan | if i use feature map, then the dtmf will be enabled even before low balance is played. the user does not want that to happen |
| 07:07.54 | kannan | if i put in a Meetme with background AGI , then i cannot use sip (we have only voip, no ydm) |
| 07:07.56 | kaldemar | i don't think you can do that with dialplan alone. if you use L(), both channels are hung up when the time hits. |
| 07:08.26 | kannan | kaldemar, thanks. now i need to think of the way that can acheive this. |
| 07:08.51 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 07:11.26 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-pideklcqwbwlgbve) |
| 07:11.37 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:fde6:ffbf:fb37:ffff) |
| 07:11.45 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-tenfprvlssyztvfb) |
| 07:16.05 | kannan | if i put the call into a meetme, then i can mute and unmute channels, or re-direct the chanels , we will need to trigger a background AGI script thats counts seconds and knows the channels' values |
| 07:16.46 | kannan | this will be possible? only how to run the agi as a separate background script when we dial |
| 07:17.59 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 07:24.09 | kannan | if i originate call with AMI, then i can also execute an AGI on connect? |
| 07:35.17 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 07:40.20 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 07:42.41 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:42.44 | schmidts | good morning |
| 07:45.47 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:47.53 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 07:59.57 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 08:30.00 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 08:31.01 | kresp0 | good morning schmidts |
| 08:31.48 | *** join/#asterisk oquidave (~oquidave@41.223.84.35) |
| 08:33.32 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 08:47.09 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 08:59.07 | *** join/#asterisk smast (~chatzilla@seepzgw.mumbai.geodesic.com) |
| 08:59.36 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 09:02.29 | *** join/#asterisk Vince-0 (c4d7bcf4@gateway/web/freenode/ip.196.215.188.244) |
| 09:02.50 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 09:08.27 | smast | when i start asterisk on virtual machine /sbin/asterisk -cvv output: 'illegal instruction' |
| 09:08.36 | smast | how to fix this bug ? |
| 09:11.47 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 09:11.49 | *** join/#asterisk chris-NB (~chris@195.230.164.108) |
| 09:14.56 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 09:16.38 | kaldemar | smast: cross-compile gone wrong? |
| 09:25.20 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:31.18 | *** join/#asterisk kasanop (~kasanop@95-27-144-8.broadband.corbina.ru) |
| 09:35.09 | smast | yes i think |
| 09:54.55 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a10b9.pool.mediaWays.net) |
| 10:00.23 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 10:05.31 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:11.02 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:21.50 | *** join/#asterisk C3r0 (~root@95.211.107.250) |
| 10:22.46 | *** join/#asterisk angryuser (~Angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:24.38 | angryuser | hello my provider uses D chan for E1 number 26, in system.conf i have hardhdlc=26 in zapata i have channel =>1-25, however on boot i have [Oct 11 12:24:27] ERROR[8047]: chan_dahdi.c:8973 mkintf: Channel 16 is reserved for D-channel. |
| 10:24.45 | angryuser | chan_dahdi.c:14265 build_channels: Unable to register channel '1-25' |
| 10:24.50 | angryuser | Why N* |
| 10:25.32 | angryuser | Why the hell he is using channel 16 when i have 26 in system.conf ? |
| 10:28.56 | schmidts | angryuser isnt there a param to set the d-channel? |
| 10:29.49 | schmidts | angryuser like this: http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf |
| 10:39.16 | angryuser | schmidts: looks like d chang is hardcoded, i had to use the trunkgroups to move it |
| 11:01.07 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 11:03.10 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:05.11 | bulkorok | somebody got experience with fax for asterisk module? I have a number where fax will be recieveid with fax_spandsp but not with fax_digium |
| 11:06.38 | coppice | this is why people use spandsp :-) |
| 11:08.47 | bulkorok | I know... |
| 11:08.53 | bulkorok | I wish I could... |
| 11:09.12 | coppice | huh? |
| 11:10.34 | bulkorok | I have to check all the scripts etc. when spandsp will be used... |
| 11:10.41 | bulkorok | error messages and so... |
| 11:12.31 | bulkorok | it's so strange... I get P30EVN_PAGE_END, P30EVN_DOC_END, STAT_SES_COMPLETE, P30EVN_COMPLETE and asterisk tells me that there is a fax_failure_protocol_error |
| 11:12.53 | bulkorok | third try was successfull... |
| 11:23.51 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 11:23.53 | *** join/#asterisk kannan (~chatzilla@123.238.229.114) |
| 11:38.29 | *** join/#asterisk italorossi (~textual@189.124.163.167) |
| 11:49.01 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 11:51.18 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:51.19 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:02.55 | ectospasm | bulkorok: is this Fax for Asterisk (FAX) or Free Fax for Asterisk (FFA)? |
| 12:03.39 | coppice | FFA == Fax for Asterisk |
| 12:04.40 | ectospasm | coppice: no, the licence key for Free Fax for Asterisk is FFA... for Fax for Asterisk it's FAX.... |
| 12:05.43 | *** join/#asterisk italorossi (~textual@201.76.151.245.digi.com.br) |
| 12:05.58 | ectospasm | s/nce/nse/ |
| 12:07.21 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
| 12:08.48 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
| 12:09.28 | bulkorok | I have a test-server with RESFFA and a live server with RESFAX and RESFFA but this shouldnt matter I suppose... |
| 12:09.50 | bulkorok | I tried at least on the test server where the third try was successful... |
| 12:09.54 | coppice | bulkorok: you have scripts specific to Fax for Asterisk? |
| 12:10.08 | bulkorok | I think so... |
| 12:10.46 | bulkorok | well not on the receiving side... |
| 12:11.30 | bulkorok | I check the FAXOPT(status) string... when it's not SUCCES the script is just writing a different log... |
| 12:11.31 | coppice | I thought the resources were supposed to be transparent. they crippled the original spandsp module code to dumb it down to match the commercial code, supposedly for that reason |
| 12:12.21 | bulkorok | I hope that too... I have to check my send-script... |
| 12:13.07 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:13.27 | *** part/#asterisk deo (~dnepangue@58.71.19.178) |
| 12:13.48 | bulkorok | I use the FAXOPT status, statustr and error codes foor logging and fax2mail infos to the customer... |
| 12:15.50 | bulkorok | is res_fax_spandsp talking to the AMI too?! |
| 12:16.38 | coppice | they are supposed to be interchangeable |
| 12:18.43 | bulkorok | The FAXSTATUSSTRING channel variable is not used by the Open Source Asterisk trunk app_fax module |
| 12:19.31 | bulkorok | this is what I have to check... |
| 12:21.02 | bulkorok | uh... I miss the fax-messages in CLI with spandsp... |
| 12:21.18 | bulkorok | but faxopt status and statustr is used... |
| 12:22.09 | *** join/#asterisk chris-NB (~chris@195.230.164.110) |
| 12:24.03 | *** join/#asterisk oquidave (~oquidave@41.223.84.34) |
| 12:25.41 | coppice | I wonder why they only provided such limited functionality in fax for asterisk? surely the Commetrex code it uses has more functionality than they expose? |
| 12:27.37 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 12:27.52 | bulkorok | my problem is we have a java-application that is parsing the AMI-output to the fax-info lines... and if spandsp is not sending them out I can not use it... |
| 12:28.52 | coppice | spandsp provides lots of info. the question is how much they crippled it |
| 12:30.54 | bulkorok | here we go: Event: ReceiveFAXStatus |
| 12:31.07 | bulkorok | I hope that they contain all that will be needed... |
| 12:32.43 | bulkorok | ah... res_fax_spandsp still has the Use Count-Bug I suppose... |
| 12:38.27 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 12:39.52 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 12:39.57 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 12:41.39 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 12:41.44 | elliot98 | politely enters |
| 12:41.51 | zamba | when looking at sip show channels.. how can i interpret what i'm reading? |
| 12:42.07 | elliot98 | does Asterisk do updtl/t38 -> DahdI/PRI translation? |
| 12:42.38 | jmetro | Zamba> read the top of the columns. |
| 12:43.30 | jmetro | First peer is "source" second peer is "destination" |
| 12:43.53 | zamba | first peer is an ip, and the last peer is the name of that peer |
| 12:43.56 | zamba | so not really source/destination |
| 12:44.08 | zamba | the corresponding name of that ip, i mean |
| 12:44.16 | jmetro | see, you understand it =p |
| 12:44.19 | [TK]D-Fender | It can't be because * is not a PROXY |
| 12:44.37 | zamba | 192.168.111.21 conf-rolf 740c6b8d7cdc093 0x0 (nothing) No Tx: ACK conf-rolf |
| 12:44.41 | zamba | stuff like that makes no sense |
| 12:44.43 | elliot98 | or does T38 need both end nodes to be SIP? |
| 12:45.02 | jmetro | conf...conference. |
| 12:45.06 | zamba | 172.31.205.42 +4933818904111 5c9e73705dd9cc8 0x8 (alaw) No Tx: ACK LYNC_TRUNK |
| 12:45.10 | zamba | yeah, that's my name of the peer |
| 12:45.31 | zamba | i have no idea which of my peers has established that last phone call |
| 12:45.44 | leifmadsen | elliot98: https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway |
| 12:47.14 | leifmadsen | elliot98: so to answer your earlier question: yes, asterisk can pass from the PSTN to a T.38 endpoint via the gateway method, but that is in Asterisk 10 and later |
| 12:47.34 | leifmadsen | otherwise, everything needs to be T.38 in order to simply perform t.38 passthrough |
| 12:47.39 | *** join/#asterisk aross42 (~aross@99.224.40.85) |
| 12:47.51 | leifmadsen | (or of course, asterisk can be the end point and answer the fax itself, if you want fax to email or something |
| 12:48.32 | elliot98 | leifmadsen: thanks! |
| 12:49.05 | *** join/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 12:49.08 | mcolombo | hi all |
| 12:49.54 | zamba | how can i destroy a stuck channel? |
| 12:50.06 | leifmadsen | channel request hangup |
| 12:50.27 | [TK]D-Fender | zamba, Those are not channel's to "kill" |
| 12:50.33 | zamba | 192.168.111.18 (None) 61b4fab77058953 0x0 (nothing) No Init: OPTIONS wlan-TEST |
| 12:50.35 | zamba | i have this |
| 12:50.36 | [TK]D-Fender | zamba, and that command will not work for them |
| 12:50.42 | zamba | and 192.168.111.18 has been down for over a day |
| 12:50.44 | [TK]D-Fender | zamba, Ignore it |
| 12:50.49 | zamba | hm, ok |
| 12:50.57 | WIMPy | Can someone recommend a DECT or CAT-IQ base that can handle more than 2 simultaneous calls? |
| 12:51.08 | zamba | but still.. the question remains.. how can i figure out who's calling or has been called by +4933818904111? |
| 12:51.44 | [TK]D-Fender | zamba, CDR <- |
| 12:51.51 | [TK]D-Fender | zamba, logs |
| 12:52.39 | leifmadsen | zamba: if you're looking in 'sip show channels' it is normal to see things living there for a while. Perfectly normal. |
| 12:53.27 | mcolombo | can anybody help me for enable SIP 181 response? |
| 12:53.33 | zamba | i see now how it's connected |
| 12:55.34 | jmetro | Hm...how many open channels can asterisk support? or is it only dependent on hardware |
| 12:55.59 | oquidave | hello, can someone enlight me again on asterisk condition branching which follows the syntaxt GotoIf(expression? destination1:destination2)...i thought that the dialplan will go to destination1 if the expression evaluates to true and destination2 otherwise, correct? |
| 12:56.18 | [TK]D-Fender | oquidave, Correct |
| 12:59.41 | oquidave | [TK]D-Fender: okay then i've a problem with this expression same=>n,GotoIf($[${device_state}="NOT_INUSE"] ? call:hangup) ...when i verbose the device state is "NOT_INUSE" but asterisk drops same=>n(hangup),Hangup() instead of same=>n(call),Dial(SIP/30,30) in my dialplan!!! |
| 13:00.27 | leifmadsen | oquidave: you need to wrap ${device_state} in double quotes too |
| 13:00.28 | [TK]D-Fender | oquidave, QUOTES are your prboem. they are LITERAL characters and the variable on the LEFT side of the = does not have them as well |
| 13:00.53 | leifmadsen | oh, and I suspect you expect that to be a function name, do ${DEVICE_STATE(device)} |
| 13:01.17 | leifmadsen | oquidave: plus remove the spaces around the ? |
| 13:01.33 | [TK]D-Fender | Also never put extra spaces. |
| 13:01.39 | leifmadsen | oquidave: basically, the GotoIf() isn't the problem -- it's your expression |
| 13:02.01 | mcolombo | [TK]D-Fender : can you help me with SIP 181 response? |
| 13:02.05 | leifmadsen | your expression is always evaluating false, since they will never match |
| 13:02.32 | *** join/#asterisk Pegasus_RPG (~chatzilla@p5B3EFF83.dip.t-dialin.net) |
| 13:02.35 | oquidave | leifmadsen: why would i have to add quotes like "${device_state}" and yet variables are writen like ${device_state} without quotes? |
| 13:02.45 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:02.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:02.47 | leifmadsen | because quotes are literal in the dialplan |
| 13:02.49 | [TK]D-Fender | mcolombo, No, and do not target people at random like that. Anyone who has seen your question and has something to contribute will tell you... |
| 13:02.55 | leifmadsen | which means foo = "foo" will never match |
| 13:03.03 | leifmadsen | only "foo" = "foo" will match |
| 13:03.06 | leifmadsen | and foo = foo |
| 13:03.21 | leifmadsen | you could just have easily prefixed with the letter 'x' or something |
| 13:03.27 | [TK]D-Fender | oquidave, Because what you put on the RIGHT side of the = has quotes. |
| 13:03.37 | leifmadsen | x${DEVICE_STATE} = xfoo |
| 13:04.04 | [TK]D-Fender | same=>n,GotoIf($[${device_state}=NOT_INUSE]?call:hangup) <- would work |
| 13:04.04 | oquidave | leifmadsen: okay so i can also remove the quotes in NOT_INUSE ie the Right side, right |
| 13:04.04 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:04.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:04.05 | oquidave | [TK]D-Fender: OKAY |
| 13:04.12 | oquidave | i see |
| 13:04.19 | leifmadsen | oquidave: I prefer to wrap in quotes in case the variable is ever null, which will cause the whole thing to fail |
| 13:04.29 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 13:04.29 | leifmadsen | $[ = foo] |
| 13:05.01 | leifmadsen | is very invalid |
| 13:05.01 | leifmadsen | dialplan will just baulk at that |
| 13:05.01 | leifmadsen | which is why $["${myvar}" = "value"] is much better |
| 13:05.17 | oquidave | leifmadsen: i can see wisdom in that now |
| 13:07.13 | oquidave | time to test |
| 13:08.15 | *** part/#asterisk Pegasus_RPG (~chatzilla@p5B3EFF83.dip.t-dialin.net) |
| 13:09.07 | leifmadsen | I've noted this to add to the book |
| 13:09.22 | oquidave | leifmadsen: i can also see why i should remove the spaces No such label ' call' in extension 's' in context 'subDeviceState'...so asterisk doesn't strip the white space! |
| 13:09.52 | leifmadsen | syntax: GotoIf($["${myvar}" = "foo"]?true_label:false_label) |
| 13:10.33 | leifmadsen | don't use spaces except around the evaluator |
| 13:10.33 | leifmadsen | when doing numbers, don't use quotes, and prefix with a 0 |
| 13:11.13 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-dnwholjkcywkzghg) |
| 13:11.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:12.19 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:12.41 | wdoekes | leifmadsen: and don't use unescaped quotes in your vars |
| 13:12.47 | leifmadsen | +1 |
| 13:13.12 | leifmadsen | heck, just don't quotes in the vars at all :D |
| 13:13.19 | oquidave | leifmadsen: so GotoFi($[${EXTEN}=030]?call:hangup) would refer to extension 30? |
| 13:13.21 | leifmadsen | just don't quotes....? |
| 13:13.40 | leifmadsen | oquidave: no, that is a comparison operator, I mean when doing math operations |
| 13:14.13 | oquidave | leifmadsen: yeap but i mean am comparing extension 30 |
| 13:14.13 | leifmadsen | Set(result=$[0${myvar} + 5]) |
| 13:14.17 | leifmadsen | I know |
| 13:14.29 | leifmadsen | you don't need to prefix with zero on comparison operations, use quotes |
| 13:14.32 | [TK]D-Fender | oquidave, No, not good for that comparison |
| 13:14.52 | [TK]D-Fender | oquidave, Also better when you spell GotoIf right :) |
| 13:15.02 | WIMPy | \why do you want to compare extensions? That's what ther dialplan does for you. |
| 13:15.24 | leifmadsen | the example is bad, ya :) |
| 13:16.08 | oquidave | okay |
| 13:17.09 | leifmadsen | better is: exten => 123,1,Hangup() |
| 13:17.31 | oquidave | leifmadsen: just out curiosity, does the asterisk syntax have some sort of looping function because ive not seen it sofar! |
| 13:17.40 | leifmadsen | yes |
| 13:17.46 | leifmadsen | While() and EndWhile() |
| 13:17.49 | leifmadsen | there is documentation on this |
| 13:18.31 | oquidave | leifmadsen: thanks i'll google it up |
| 13:20.36 | *** join/#asterisk bchia (~Adium@nat/digium/x-crusorzjxrrhmueb) |
| 13:20.53 | [TK]D-Fender | ~book |
| 13:20.53 | 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://www.asteriskdocs.org/ or see ~buybook |
| 13:20.56 | [TK]D-Fender | oquidave, ^ |
| 13:21.05 | [TK]D-Fender | And the WIKI @ asterisk.org |
| 13:21.06 | *** join/#asterisk amessina (~amessina@h-66-166-108-205.chcgilgm.static.covad.net) |
| 13:21.27 | [TK]D-Fender | oquidave, Before wondering if such things exist, go read the full list of applications and functions.... |
| 13:23.02 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:29.01 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 13:32.34 | kannan | working on a custom call card, i need to use a low balance warning (like the L with Dial), but instead of hanging up the channels, the agent must be on a meetme room, while the calling in party is transferred to a payment IVR.. |
| 13:32.48 | kannan | wondering how i can achieve it |
| 13:34.53 | Katty | hello my asterisk does not work at all how to fix??? it is urgent plz answer thx. |
| 13:35.11 | wdoekes | pour some water on it |
| 13:35.22 | schmidts | Katty feed it with some boiled chicken |
| 13:36.45 | kresp0 | Katty: try pressing "*" key harder |
| 13:37.04 | chuckf | Katty: pick up the receiver before dialing. |
| 13:37.18 | schmidts | have you rebootet your phone first? |
| 13:37.48 | Katty | fixed! thx |
| 13:37.53 | kresp0 | nice :D |
| 13:38.21 | kresp0 | How to ask questions The Smart Way: http://www.catb.org/esr/faqs/smart-questions.html |
| 13:38.45 | oquidave | kannan: what's your setup? |
| 13:39.05 | kannan | asterisk 1.8.x, voip only, no TDM |
| 13:39.07 | oquidave | kannan: what operating system what asterisk version |
| 13:39.18 | kannan | ubuntu 11.10 server |
| 13:39.30 | [TK]D-Fender | wonders how the OS version plays into that at all.... |
| 13:40.07 | oquidave | oh sorry i meant katty |
| 13:40.14 | chuckf | [TK]D-Fender: the answer to that is why you'll stay with this converstation |
| 13:40.29 | oquidave | but it seems his already sorted! |
| 13:40.33 | kannan | if i call the agent AND a Local/extension , which runs an AGI counting the seconds , after which it can spawn an AMI script to re-direct the channels (the details to be stored in a table) , will this be good solution? |
| 13:41.50 | [TK]D-Fender | kannan, Spawning AGI seems pointless. AGI is for processing an actual call and since the spawned channel is independant the only way it could interact with the original is via AMI. AMI is not AGI thus an actual Asterisk channel adds no value and lots of overhead |
| 13:42.43 | Katty | oquidave: it runs on windows hold on i will get you the version |
| 13:42.50 | Katty | oquidave: it says 95 |
| 13:43.20 | oquidave | Katty: asterisk on windows? |
| 13:43.31 | WIMPy | Katty: Did you do an upgrade? |
| 13:43.34 | Katty | no i'm tinkering with after dark screensavers |
| 13:43.38 | Katty | why on earth would i run asterisk on windows |
| 13:43.51 | Katty | WIMPy: i'll upgrade you in a minute |
| 13:44.12 | WIMPy | Sounds exciting. In what way? |
| 13:44.35 | kannan | [TK]D-Fender, then i cannot see how I can count the seconds of the call, except by a cron every second. |
| 13:44.40 | carrar | Katty, you ported asterisk to windows!!!?? |
| 13:44.40 | Katty | WIMPy: CYBERMEN |
| 13:44.44 | carrar | nice |
| 13:44.48 | Katty | carrar: no, but i ported your mom. |
| 13:44.56 | carrar | OH THATS HOT++ |
| 13:44.59 | Katty | ^_^ |
| 13:45.02 | Katty | hugs carrar |
| 13:45.05 | Katty | how'rechu dear |
| 13:45.18 | carrar | hugs Katty back and dribbles espresso on her back |
| 13:45.31 | WIMPy | Katty: Does it mean, I will become virtual? |
| 13:45.57 | carrar | everyone here is cirtual! |
| 13:46.00 | carrar | virtual even |
| 13:46.11 | Katty | you will become a .....Virtual Boy |
| 13:46.20 | Katty | wonders if anyone gets it |
| 13:46.24 | Katty | am i showing my age? |
| 13:46.26 | Katty | nevermind |
| 13:46.34 | carrar | ref tag pls |
| 13:46.43 | WIMPy | As long as that doesn't mea real female I guess that might be ok. |
| 13:47.12 | Katty | http://3.bp.blogspot.com/_xe4ykozpQD0/TTeBmDoPVGI/AAAAAAAABPA/hdI0Pcb8rC4/s1600/virtual-boy.jpg <- REF |
| 13:47.16 | [TK]D-Fender | kannan, Then you need to think a little harder |
| 13:47.18 | carrar | Katty's transation is outside of the IRC Dialog!! |
| 13:47.28 | carrar | transaction even |
| 13:48.11 | Katty | i'm guessing i'm the only one to have played a virtual boy. |
| 13:48.27 | carrar | yes |
| 13:48.40 | WIMPy | Maybe you shoudl try real boys? |
| 13:48.48 | carrar | nice bed btw |
| 13:48.54 | carrar | put some sheets on that |
| 13:49.07 | Katty | that's not my bed sweety |
| 13:49.11 | carrar | SURE!!!! |
| 13:49.15 | Katty | my bed has a big fluffy feather blanket and a duvet on it |
| 13:49.16 | kannan | [TK]D-Fender, a clue or pointer in the right direction , please ? :) |
| 13:49.26 | Katty | and japanese cherry blossom artwork on the wall |
| 13:49.30 | Katty | may have a photo on the phone |
| 13:49.31 | carrar | PICS!! |
| 13:49.59 | Katty | hmm, no. |
| 13:50.06 | Katty | i will take one later, if i remember to |
| 13:50.26 | [TK]D-Fender | kannan, You only care at the point where you have to intervene with the channel. Spawn a completely external script with the channel to target. Sleep on it. Then if it's active hijack it. |
| 13:51.44 | Katty | carrar: http://www.jojospring.com/images/Winter%20Plum%205-piece%20Oil%20Canvas%20Art%20Set.jpg <- that's the artwork |
| 13:51.49 | [TK]D-Fender | kannan, It can sleep just fine taking up virtually no resources until it has to do its dirty work. You could have it check on some longer interval to see if it is no longer required (5min?) so it doesn't sleep forever. |
| 13:52.15 | [TK]D-Fender | kannan, This is basic programming. You had BETTER have these kind of instincts if you're going to be taking on a project like this. |
| 13:53.33 | carrar | yeah |
| 13:54.11 | kannan | [TK]D-Fender, thanks, again, as usual you have helped with the solution |
| 13:54.30 | tzanger | Katty: that's a terrible picture, someone's cut it all up on you |
| 13:54.33 | Katty | everytime someone says solution i think of something with a pH balance |
| 13:54.52 | Katty | tzanger: i like it ^_^ |
| 13:54.56 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:55.02 | tzanger | :-) |
| 13:55.16 | tzanger | one more day in denver, then I can get back home |
| 13:55.27 | carrar | Katty, so is: http://pics.osburn.com/photo/34287/original |
| 13:55.36 | tzanger | my daughter moved in with me and I had to go out here the same week :-/ |
| 13:55.43 | carrar | I've got better pics someplace |
| 13:55.52 | chuckf | if someone signs up for a dropbox account with your email address, how much do you mess with them? |
| 13:56.05 | kannan | does a2b provide this (the ability to re-charge over IVR on low balance) out of the box ? |
| 13:58.03 | [TK]D-Fender | kannan, Not supported here... and you're best asking them directly.. |
| 13:58.17 | kannan | oh , ok |
| 13:59.09 | kannan | I can finish this feature fine now anyhow |
| 14:03.10 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:08.13 | bulkorok | somebody git a quick answer for this error message when receiving fax with spandsp: WARNING T.30 Non-ECM carrier not found |
| 14:08.29 | bulkorok | the fax is received successfully at the end... |
| 14:10.08 | *** join/#asterisk alexscott (~alexscott@37.122.202.194) |
| 14:17.11 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:17.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:18.51 | leifmadsen | a warning is not an error :) |
| 14:20.36 | *** join/#asterisk Jouva (~jouva@static-98-111-131-98.phlapa.fios.verizon.net) |
| 14:21.30 | *** join/#asterisk srini (~Srinivasa@182.73.123.46) |
| 14:21.34 | srini | hi all! |
| 14:21.48 | srini | Is there way to set the warning off in the CLI? |
| 14:22.48 | leifmadsen | change logger.conf to not log them on the console => line |
| 14:23.05 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 14:23.21 | SuperNull | Hey all .. any support for H248 ? |
| 14:23.37 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 14:24.48 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:26.01 | Qwell | SuperNull: no |
| 14:26.43 | SuperNull | ty. that was easy :) |
| 14:26.52 | Jouva | Hey folks. We've got some Aastra 6755i phones and just started setting up Asterisk 1.8.10.1 on Ubuntu 12.04. I've used these phones before and setup Asterisk before to do BLF successfully, but for some reason, over here it's not working properly. We DO have the HOLD notification working, but the server never seems to want to indicate that it's ringing or on a call. As soon as it takes a call |
| 14:26.53 | Jouva | off hold, it says it's "idle". |
| 14:26.58 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:27.37 | Qwell | Jouva: callcounter enabled? |
| 14:27.45 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:28.10 | Jouva | Qwell: Yep. Let me copy the general section of the sip.conf and a sample user (since they're all the same except for caller ID, account name and password) |
| 14:28.19 | Qwell | ~pastebin |
| 14:28.19 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:28.31 | Jouva | I know I know :P was just going there |
| 14:29.52 | *** join/#asterisk rolandow (~pi@92.68.81.83) |
| 14:29.59 | rolandow | hi! |
| 14:30.07 | Qwell | ~hello |
| 14:30.07 | infobot | Howdy Bub |
| 14:30.11 | rolandow | i want to setup my asterisk server as dhcp server as well |
| 14:30.22 | rolandow | so i can seperate the phones connected to it, from our internet traffic |
| 14:30.27 | rolandow | to make sure that people won |
| 14:30.35 | rolandow | won't plugin their computer to the asterisk server |
| 14:30.40 | rolandow | i want to block DNS requests |
| 14:30.52 | rolandow | when people find out they cannot surf, they will probably call the IT department :) |
| 14:30.54 | Jouva | http://pastebin.com/L23Yz1A3 |
| 14:31.06 | rolandow | however, i would like *some* requests to pass, for example ntp.org, and firmware upgrades |
| 14:31.29 | Qwell | rolandow: sounds like a general Linux question. |
| 14:31.34 | rolandow | does anybody know a DNS server/caching with the ability to control (with wildcards perhaps?) what DNS requets it'll answer to |
| 14:31.41 | rolandow | uhm.. |
| 14:31.42 | rolandow | true |
| 14:31.57 | rolandow | on the other hand i thought maybe people have the same scenario here :) |
| 14:32.08 | srini | Why do we generally get :Unable to write frame to channel Local/8600051@default-8e9b,2? Something do with SIP carrier? |
| 14:32.25 | Jouva | rolandow: Quick and dirty way COULD involve simply setting up ntpd on the dhcp/asterisk server to answer NTP requests, as well as TFTP for only the config and firmware files |
| 14:32.41 | Jouva | How to set that up you'd need to look up yourself, but that might be the answer |
| 14:33.53 | Jouva | Qwell: I got the pastebin link above with the config we are using. |
| 14:34.08 | Qwell | Jouva: looks about how I'd have expected it |
| 14:34.26 | Qwell | hrm, is callcounter a general option though? |
| 14:34.40 | Qwell | yes, yes it is. |
| 14:35.06 | Jouva | Maybe not? I had THOUGHT that practically any setting that can apply to a phone can be put in general, minus stuff like username and password |
| 14:35.20 | Qwell | many, but not all. in this case, you're fine |
| 14:35.46 | Jouva | Qwell: One thing of note is that while this IS on the same LAN and we're using 10.x.x.x IPs, the phones are currently on 10.0.0.x but the Asterisk server is on something like 10.100.200.x |
| 14:36.09 | Jouva | We had read of an account where somebody had issues with BLF on different subnets, but I couldn't see WHY that would be an issue |
| 14:37.13 | Qwell | err, type=friend in general? |
| 14:37.21 | Jouva | Hmmm I can remove that |
| 14:37.33 | Jouva | There IS a peer it connects to |
| 14:37.33 | Qwell | I very highly doubt that's your problem. Weird though |
| 14:37.52 | Jouva | I could remove it since pretty much everything IS a friend |
| 14:38.35 | Jouva | Now recently when I switched the general section from saying canreinvite=no to canreinvite=yes, it looks like a softphone stopped working |
| 14:40.27 | Jouva | But what I found odd was that when a phone went from placing a call on hold to being back on the line, Asterisk said it told a subscribed phone that the phone in a call was now "idle" |
| 14:40.27 | [TK]D-Fender | None of those have any impact |
| 14:40.31 | srini | ok.... Now when I am trying to make a call using SIP, i am getting " Received response: "Forbidden" from '"M0112008480000000007"" ... Is it normal that UID coming in the scene? |
| 14:40.57 | *** join/#asterisk becca_r (~becca_r@12.25.151.59) |
| 14:41.05 | becca_r | Good morning |
| 14:41.31 | Jouva | And I should be more specific about "stopped working". I meant that the two way communications wasn't working. Hard phone couldn't hear the soft phone, but soft phone could hear the hard phone |
| 14:42.46 | becca_r | Is there a best way to do a realtime lookup using multiple matchvalues? For example, I am doing a schedule lookup which has schedule exceptions. When looking up the schedule exceptions, I would like to lookup based off the schedule and the date. |
| 14:43.33 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 14:43.49 | [TK]D-Fender | srini, Fix your auth. |
| 14:44.51 | srini | [TK]D-Fender, register => user:password@host/user is what I have set... in the sip show registry, I can see it registered also... |
| 14:45.21 | [TK]D-Fender | srini, And you have stated a terminating response to a request we don't see. |
| 14:46.23 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 14:48.16 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:52.11 | *** join/#asterisk AcidBurn (AcidBurn@per18-2-88-165-49-175.fbx.proxad.net) |
| 14:55.10 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-gopixejzuskkfayh) |
| 15:03.14 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 15:03.36 | srini | [TK]D-Fender, http://paste.lisp.org/+2U8W |
| 15:03.45 | rolandow | ok it seems that dnsmasq is the answer |
| 15:03.57 | rolandow | if anybody was interested in the answer to my question ;-) |
| 15:04.35 | rolandow | Jouva: yes.. an own ntp daemon could be nice too ... i haven't dived into TFTP yet, but as it seemed hard to setup |
| 15:06.06 | [TK]D-Fender | srini, You aren't even looking at that call with SIP debug enabled, we don't see a registration that could be considered related to it... which shouldn't matter anyway. You aren't even really looking at your problem yet. |
| 15:06.51 | [TK]D-Fender | rolandow, funny... tftp is a one minute job for most. As to what you PUT in there... that is another matter. |
| 15:08.17 | rolandow | [TK]D-Fender: yes.. i mean what to put in there :) |
| 15:08.23 | rolandow | to create the correct config files |
| 15:08.53 | *** join/#asterisk Phr0zen (Phr0zen@blk-224-135-210.eastlink.ca) |
| 15:09.09 | Phr0zen | I am wondering if asterisk 1.8 is sipconnect compliant? |
| 15:09.22 | Katty | hi. |
| 15:09.26 | Katty | how are you Phr0zen |
| 15:09.36 | [TK]D-Fender | Phr0zen, And that is ...? |
| 15:09.41 | Phr0zen | not bad, and yourself |
| 15:09.47 | Katty | good thanks (= |
| 15:10.12 | Phr0zen | http://www.sipforum.org/sipconnect |
| 15:10.28 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:10.48 | [TK]D-Fender | Phr0zen, Yes, that is a SIP terminator like any other ITSP out there. |
| 15:10.53 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:11.03 | [TK]D-Fender | Phr0zen, Their service is nothing magically incompatible |
| 15:11.14 | Phr0zen | no, not a service, compliance standards |
| 15:12.40 | Phr0zen | basically my telco is asking me if the asterisk server is sipconnect compliant....all i could find on google with regards to this and asterisk was a patch (2010?) that got submitted |
| 15:13.16 | [TK]D-Fender | Phr0zen, news to me... I'd call Digium on this one... |
| 15:14.09 | Phr0zen | https://reviewboard.asterisk.org/r/1515/ |
| 15:14.24 | Phr0zen | that was what i saw on google |
| 15:14.31 | Phr0zen | yea, i guess asking digium is the way to go |
| 15:14.47 | srini | [TK]D-Fender, Sorry... here is the sip debug http://pastie.org/5035741 |
| 15:15.18 | [TK]D-Fender | Phr0zen, http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,143/Itemid,75/ <--- |
| 15:16.27 | Phr0zen | yea, that looks like appliances though |
| 15:16.41 | [TK]D-Fender | Phr0zen, And what impact does that have on anything? |
| 15:17.20 | Phr0zen | nothing specific about 1.8 |
| 15:17.40 | [TK]D-Fender | Phr0zen, Do you think Asterisk is becoming LESS compliant? |
| 15:18.01 | Phr0zen | lol, no.. not sure why you would jump to that conclusion |
| 15:18.28 | Phr0zen | I am just looking for something somewhere say this (perhaps on their website) |
| 15:18.32 | [TK]D-Fender | Phr0zen, What about EARLIER versions? Is that to say their list of other vendors products means that an earlier or later revision of each is NOT compliant and only the specific one listed is? |
| 15:18.49 | Phr0zen | fender, what are you going on about |
| 15:19.00 | [TK]D-Fender | <Phr0zen> lol, no.. not sure why you would jump to that conclusion <- because you implied that something NEWER might not be because they don't say Asterisk 1.8 |
| 15:19.37 | Phr0zen | look, different versions may have different bugs, code, whatever... so i am just looking for confirmation. i dont like assumptions |
| 15:19.54 | Phr0zen | so while i may agree with you, while that may make sense, i want proof |
| 15:19.57 | Phr0zen | so to speak |
| 15:20.51 | [TK]D-Fender | Phr0zen, Then go ask SIPConnect. They are the ones who collect the proof. They are the ones who cert it. They should ahve the listing to give you. For every version. Of every product they've encountered from every vendor |
| 15:21.26 | [TK]D-Fender | Phr0zen, Apparently only R13 & R14 of Broadsoft is "compliant" by that thought process |
| 15:22.15 | Phr0zen | seriously dude, lay off the bath salts |
| 15:22.26 | Phr0zen | you are being quite rude for no reason |
| 15:22.42 | [TK]D-Fender | Phr0zen, I'm just making sure that your demands of prrof are reasonable. |
| 15:23.06 | Phr0zen | and given the sheet you linked, yes, by that sheet only r13 and r14 are listed under broadsoft |
| 15:24.09 | [TK]D-Fender | Phr0zen, I'm not seeing anyone else they've sertified say more than the bare minimum they list. It's their list. If you want proof and that's the kind of list they make then maybe that means that your expectation is not going to be met and that their process isn't thorough an you aren't going to find a pulished answer |
| 15:24.30 | Phr0zen | all i am saying is that normally if there is an accepted "compliance" then most vendors/providers tend to list that... like a resume |
| 15:24.39 | [TK]D-Fender | And dangit the typing skills just start draining away.... |
| 15:25.19 | [TK]D-Fender | Phr0zen, That is a very small compliance list considering all the vendors and models out there. Makes me wonder how much their approval means tot he market at large. |
| 15:25.44 | [TK]D-Fender | AKA "nearly useless metric" (?) |
| 15:25.45 | Phr0zen | yea exactly, that is why it threw me for a loop. never saw nor heard of it until now |
| 15:26.06 | Phr0zen | i figure its bs, but they are asking for it. |
| 15:26.28 | [TK]D-Fender | so you're the one chasing imaginary dragons :) |
| 15:26.40 | Qwell | There are many providers out there. If they are making you jump through hoops...pick a different one. |
| 15:26.48 | [TK]D-Fender | hands Phr0zen some pepper to go with those salts. |
| 15:26.50 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 15:27.04 | [TK]D-Fender | There, full condiment set! |
| 15:27.08 | Phr0zen | yea, i am just gonna put "you tell me" in there |
| 15:27.20 | Phr0zen | or something equally vague |
| 15:29.29 | [TK]D-Fender | Phr0zen, Here is their "full members" list who paid the $7500 : http://www.sipforum.org/component/option,com_fullmember/Itemid,195/ |
| 15:30.49 | [TK]D-Fender | Phr0zen, Amazing who didn't make their "cert" list given the products they sell. |
| 15:30.59 | [TK]D-Fender | Phr0zen, I wouldn't put much stock in thenm |
| 15:31.07 | [TK]D-Fender | (SIP Connect) |
| 15:31.13 | srini | [TK]D-Fender... |
| 15:32.44 | [TK]D-Fender | srini, Remote-Party-ID: "M0112035320000000017" <sip:0000000000@192.168.0.2>;privacy=off;screen=no |
| 15:32.48 | [TK]D-Fender | srini, that isn't a UUID |
| 15:32.53 | [TK]D-Fender | srini, that's RPID <- |
| 15:33.29 | srini | [TK]D-Fender, And the "Forbidden" is .... |
| 15:33.31 | [TK]D-Fender | srini, Next, do NOT pask your call like that. I am not debugging this blind. |
| 15:33.48 | [TK]D-Fender | mask* |
| 15:33.54 | srini | [TK]D-Fender, sorry to bother! that way! |
| 15:33.56 | [TK]D-Fender | srini, unfiltered. |
| 15:34.04 | Phr0zen | fender: yea, i wasn't putting much stock in it lol |
| 15:39.39 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:42.54 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:43.35 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:43.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:46.38 | *** part/#asterisk Phr0zen (Phr0zen@blk-224-135-210.eastlink.ca) |
| 15:46.44 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:01.50 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:07.44 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 16:08.56 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 16:08.58 | cusco | hi |
| 16:10.04 | cusco | using a remote sip peer on our system, when a call comes in trough them, using features.conf to transfer, the person from that peer location, canno hear us, we hear them. we have nat = yes |
| 16:15.22 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 16:16.41 | AkkerKid | Hi all! Does anyone know how to get the previous ${EXTEN} variable after a Goto() statement? |
| 16:16.56 | WIMPy | Save it |
| 16:17.23 | AkkerKid | ...without editing anything on or before the Goto()? |
| 16:17.34 | WIMPy | not |
| 16:25.45 | *** join/#asterisk FireAndIce (~administr@175.100.134.66) |
| 16:26.02 | FireAndIce | Hi everyone!! |
| 16:27.43 | FireAndIce | I'm new to asterisk.. I've it running on my local machine.. |
| 16:27.49 | FireAndIce | But the problem is I need to generate SIP uri for soft phone.. How do I do that? |
| 16:28.10 | FireAndIce | I'm trying to establish a connection between 2 android phones using IMSDroid.. |
| 16:28.17 | WIMPy | define "generate" |
| 16:28.43 | FireAndIce | Please help.. |
| 16:29.02 | pabelanger | ~book |
| 16:29.02 | 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://www.asteriskdocs.org/ or see ~buybook |
| 16:29.05 | WIMPy | And SIP is P2P you can call one SIP phone directely from another. |
| 16:29.08 | pabelanger | FireAndIce, Start here^ |
| 16:30.24 | FireAndIce | WIMPy, I need to create username@domain address for each sip user.. Do I have to register with a ISTP service to do that.. or can it be done through asterisk.. |
| 16:30.32 | FireAndIce | Please spare me for asking silly questions.. |
| 16:30.40 | FireAndIce | I'm new to asterisk and SiP.. |
| 16:30.58 | FireAndIce | pabelanger, infobot I'll surely check the book out.. |
| 16:31.13 | WIMPy | If you have an asterisk running, you call call your users with extension@yeourAsteriskserver |
| 16:32.08 | FireAndIce | WIMPy, I got the extension part, but not "yourAsteriskserver" |
| 16:32.09 | carrar | FireAndIce, you don't need a ISTP |
| 16:32.22 | carrar | You can call each other through your asterisk box if they both register to it |
| 16:32.41 | WIMPy | If your server has an FQDN, you can use that or the IP otherwise. |
| 16:34.08 | *** join/#asterisk qakhan (~qakhan@173-166-183-225-washingtondc.hfc.comcastbusiness.net) |
| 16:34.10 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 16:34.17 | qakhan | hi all |
| 16:34.51 | carrar | hi!! |
| 16:34.53 | FireAndIce | carrar, Thanks for informing that I dont need an ISTP.. |
| 16:34.53 | qakhan | is there any thing to detect eight my call picked up by person or voice mail |
| 16:35.27 | FireAndIce | WIMPy, I've learnt some new terms, I'll check it out.. |
| 16:35.37 | WIMPy | Depends on the type of VM and th destination. |
| 16:36.00 | carrar | qakhan, you can use the ring time |
| 16:36.07 | carrar | if it's constant |
| 16:36.35 | qakhan | but what about if phone powered off and call goes to VM |
| 16:36.44 | carrar | then it's not constant |
| 16:37.05 | WIMPy | qakhan: What destination? What VM? |
| 16:37.11 | carrar | You can have the person answering to press a key |
| 16:37.16 | carrar | heh |
| 16:37.32 | WIMPy | In the PSTN and PLMN VM is usually implemented via CF so you get a notification about that. |
| 16:37.33 | qakhan | WIMPy i have written a script to call out |
| 16:37.55 | [TK]D-Fender | qakhan, "core show application AMD" <- |
| 16:38.00 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 16:38.52 | qakhan | [TK]D-Fender i am using 1.4.38 |
| 16:39.20 | [TK]D-Fender | qakhan, App is the same... |
| 16:40.34 | zamba | what's it called when someone dials in and instead of seeing their phone number, you see their name? |
| 16:40.41 | zamba | what's the english term for that? |
| 16:40.48 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:41.01 | [TK]D-Fender | zamba, There is no term for seeing a name INSTEAD of a number |
| 16:41.07 | zamba | [TK]D-Fender: ok |
| 16:41.10 | [TK]D-Fender | zamba, What you see identifying the caller is CALLERID |
| 16:41.20 | [TK]D-Fender | zamba, Name & number are separate halves to it |
| 16:41.40 | zamba | we're using polycom sip phones and we have a problem that we only see numbers.. even when those numbers are defined as a contact locally on the phone |
| 16:41.55 | zamba | should this job be asterisk'? |
| 16:42.01 | paulc | SBC sends call to Asterisk, which answers, interacts a bit, then Dials back to SBC, where the media gets bridged via a re-invite. Calls drop after 1800 seconds. Change sip.conf session-expires to a big number and calls drop after 3600 (the SBC's expiry timer) |
| 16:42.02 | [TK]D-Fender | Hit an arrow key |
| 16:42.03 | zamba | rewriting callerid? |
| 16:42.08 | paulc | First time I've come across this kind of issue and I'm feeling a bit lost.. would a different value of session-timers (like refuse?) help, or session-refresher (because some change was made between 10.8 and 10.9?) |
| 16:42.17 | zamba | [TK]D-Fender: huh? |
| 16:42.18 | [TK]D-Fender | zamba, No, you probably are not using the phone right |
| 16:42.20 | [TK]D-Fender | ^ |
| 16:42.32 | zamba | [TK]D-Fender: could very well be :) |
| 16:42.42 | [TK]D-Fender | zamba, First look at your call. Then look at your phone. |
| 16:43.11 | zamba | [TK]D-Fender: we're also a bit puzzled by that first configured line, since that takes up the first slot in the speed dial list.. we don't see the point in that, as that only gives us the dialtone |
| 16:43.30 | zamba | and you can get the dialtone the same way by just taking the phone off the hook or pressing new call or whatever |
| 16:43.39 | zamba | [TK]D-Fender: are you familiar with polycom? |
| 16:43.50 | [TK]D-Fender | zamba, that isn't a spot on the speed dial list... that is a LINE KEY |
| 16:44.10 | zamba | yeah, i know it's a line key.. but i don't understand why it needs to be there |
| 16:44.18 | zamba | and why you can't remove it |
| 16:44.28 | [TK]D-Fender | zamba, You need to choose what you want to dial out of. Guess you've never used a business phone anywhere else in your life based on this... |
| 16:44.39 | WIMPy | You can't? |
| 16:44.45 | zamba | [TK]D-Fender: we only have one line |
| 16:44.57 | [TK]D-Fender | zamba, that's YOU. These phones support multiple |
| 16:45.13 | [TK]D-Fender | zamba, What if you want to juggle between multiple calls? |
| 16:45.18 | [TK]D-Fender | zamba, LINE KEYS |
| 16:45.24 | zamba | hm, ok |
| 16:45.58 | zamba | so what you're saying is, if someone dials in to that extension while it's busy, then the phone won't dial, right? |
| 16:46.17 | zamba | .. i mean.. i guess that's a duh moment, but :) |
| 16:46.17 | [TK]D-Fender | zamba, Depends how you configured your registration(s) on it |
| 16:46.27 | [TK]D-Fender | zamba, Oh, very much so... |
| 16:46.28 | zamba | i only have one |
| 16:46.50 | [TK]D-Fender | zamba, Yes well even a shitty 8 year old IP 300 can juggle *10* calls at a time |
| 16:47.00 | *** join/#asterisk JunK-Y (~junky@pdpc/supporter/active/junk-y) |
| 16:48.06 | [TK]D-Fender | zamba, How many calls you want to handle, and how depends on your model and how much of it you choose to use |
| 17:00.56 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 17:14.14 | *** join/#asterisk mobile_gordita (~Robert@66-87-95-82.pools.spcsdns.net) |
| 17:15.16 | zamba | [TK]D-Fender: it's a 650 |
| 17:16.29 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 17:17.25 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.64.195.dsl.dyn.forthnet.gr) |
| 17:17.34 | [TK]D-Fender | zamba, So what is your previous use of the term "one line" supposed to mean regarding the actual size of your overall setup? |
| 17:17.43 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:18.01 | zamba | [TK]D-Fender: well.. i've never been used to working with several lines, so i guess that's where my original question originated from |
| 17:18.17 | [TK]D-Fender | <zamba> [TK]D-Fender: we only have one line <-- clarify.... |
| 17:18.20 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:18.31 | zamba | [TK]D-Fender: there's only one registration active for this phone |
| 17:19.05 | zamba | but we can of course add more registrations |
| 17:19.07 | [TK]D-Fender | zamba, Very few people need more than 1. Doesn't describe how they want to use their line keys however.... |
| 17:19.24 | zamba | well.. to use that line key is just in the way |
| 17:19.31 | zamba | it's taking up one speed dial position |
| 17:19.38 | zamba | since it'll never be used |
| 17:19.51 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 17:19.52 | [TK]D-Fender | zamba, I used to reserve 3 line keys to a single registraion and restrict each key to handling 1 call at a time. Therefor each linekey represented a single call appearance. |
| 17:20.18 | zamba | but what is a line compared to a registration? |
| 17:20.24 | zamba | can you have several lines over one registration? |
| 17:20.39 | zamba | i guess that's what i haven't fully grasped yet |
| 17:20.45 | [TK]D-Fender | Single registration VS how many CALLS acn be represented on how many buttons <- |
| 17:21.22 | [TK]D-Fender | if I have 3 calls going on, how do I jump between them? that is the point of line keys...s o I'm not hitting the context keys on the phone all the time and scrolling through them with the arrows, etc. |
| 17:21.30 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:21.32 | zamba | ok |
| 17:21.35 | [TK]D-Fender | Also gives something else to light up when that call comes in. |
| 17:21.53 | zamba | but do you have one registration per line? |
| 17:22.02 | [TK]D-Fender | Drop the term "line". |
| 17:22.17 | [TK]D-Fender | LINE-KEY is one thing. REGISTRATION is another. |
| 17:22.31 | [TK]D-Fender | You associate LINE-KEYS to your REGISTRATIONS. |
| 17:25.42 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 17:31.23 | dwayne | is there anyone w/ an ISDN switch that would allow me to send them a test call? I'm interested in seeing if I put a custom SIP header (User-to-User) in my outbound call, the carrier will convert the contents to an ISDN UUI information element |
| 17:33.06 | *** join/#asterisk navaismo (~navaismo@189.191.10.146) |
| 17:33.38 | Kobaz | dwayne: is that something that carriers can do? |
| 17:34.18 | dwayne | Kobaz, I don't know if they do it automatically, on a carrier-to-carrier basis, or at all |
| 17:34.32 | dwayne | that's partly why I want to try it |
| 17:34.49 | Kobaz | that would be pretty cool if you can tack on some oob info |
| 17:34.59 | Kobaz | then you can do like exten to exten dialing over the pstn |
| 17:36.00 | dwayne | http://tools.ietf.org/html/draft-johnston-sipping-cc-uui-09 |
| 17:36.44 | dwayne | I haven't combed through that RFC so I may be missing something obvious |
| 17:37.45 | srini | Thanks for all the help here! I resolved all the issues... dialling out happlily! |
| 17:38.33 | *** part/#asterisk srini (~Srinivasa@182.73.123.46) |
| 17:42.33 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.174) |
| 17:45.32 | Kobaz | Invoice Description: 30273 Intel Core i3-540 i3540 30274 Intel Core i3-540 i3540 |
| 17:45.34 | Kobaz | do de do |
| 17:45.35 | Kobaz | sexy |
| 17:46.08 | *** join/#asterisk sickgrinder (~sickgrind@rrcs-97-76-33-146.se.biz.rr.com) |
| 17:46.10 | jmetro | an i3? |
| 17:46.18 | Kobaz | for a hosted dedicated server |
| 17:46.22 | Kobaz | 8 gigs of ram |
| 17:46.25 | Kobaz | not bad for 150 a month |
| 17:46.39 | Kobaz | bandwidth included |
| 17:47.45 | *** join/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
| 17:48.09 | Kobaz | 3.02ghz |
| 17:48.12 | Kobaz | ubiquityservers |
| 17:48.30 | Kobaz | switching from my other provider that doesn't have remote reboot and private ipkvm |
| 17:48.39 | MLNoah | is there a way to change the mailbox associated with a SIP peer on the fly (either by having the end-point request a specific MWI mailbox or whatever) without having to use realtime? i'm trying to get MWI working in a hotdesking environment where the endpoints are Cisco SPA5xx phones, Asterisk 11. |
| 17:49.16 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 17:56.04 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 17:56.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:57.36 | [TK]D-Fender | MLNoah, I'd probably do it by symlinking the targets instead |
| 17:57.43 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 17:58.20 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 18:01.20 | MLNoah | [TK]D-Fender: is there a good document to get me pointed in the right direction? |
| 18:02.22 | [TK]D-Fender | MLNoah, "man ln" <- |
| 18:02.50 | MLNoah | oh, so you're saying basically create a symlink within the file system to map what the sip peer is checking to the extension's file system dir from voicemail.conf |
| 18:04.19 | *** part/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 18:09.21 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-180.mobile.uci.edu) |
| 18:09.50 | MLNoah | thanks for the pointer |
| 18:11.10 | *** join/#asterisk odenkos (~odenkos@unaffiliated/odenkos) |
| 18:16.03 | gusto | hey |
| 18:16.11 | gusto | are there some americans out there? |
| 18:17.17 | [TK]D-Fender | gusto, No, we got the last of them .... |
| 18:17.41 | navaismo | gusto, You mean US people? Im an American by continent US != America |
| 18:18.13 | leifmadsen | navaismo: +1 |
| 18:18.25 | carrar | You mean the new China? |
| 18:18.25 | leifmadsen | there are lots of North and South Americans :) |
| 18:18.37 | gusto | however |
| 18:18.54 | navaismo | http://d24w6bsrhbeh9d.cloudfront.net/photo/2636622_700b.jpg |
| 18:19.13 | gusto | i am shocked about that lies that are now around on TV that some i do not know what state in the US said that Huawei and ZTE do espionage or what |
| 18:19.56 | gusto | i mean ... american companies are very famous of doing backdoors in their hw, so ... ??? WTF? |
| 18:20.02 | [TK]D-Fender | gusto, First you expess shock ... then tell us you're not sure who it is that shocked you exactly. When yuo find your taget, let us know. |
| 18:20.34 | gusto | no idea ... does anybody of you there believe this? |
| 18:20.57 | carrar | gusto, why is that shocking? |
| 18:21.08 | AkkerKid | I almost consider that common sense |
| 18:21.36 | AkkerKid | of course manufacturers are going to put backdoors and other unwanted things secretly in their product |
| 18:21.46 | gusto | because i never thaught that someone could spread such a conspiracy theory that successful ... i thaught that ppl have learned some lesson from 9/11 |
| 18:22.01 | carrar | How do you know it's not true? |
| 18:22.05 | gusto | w8 |
| 18:22.23 | carrar | Have you reverse engineered Huawei devices? |
| 18:22.40 | gusto | but the issue is that they are chinese manufacturers so ... when american products have backdoors in it too, then whats the treat? |
| 18:23.01 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 18:23.05 | carrar | tricker treat |
| 18:23.06 | [TK]D-Fender | gusto, You said you don't know WHICH STATE. This means you don't even know who you think this random comment is coming from. |
| 18:23.07 | gusto | i mean ... US engineered products have backdoors ... now chinese ones may have backdoors too, so what is the problem? |
| 18:23.20 | Kobaz | even people have back doors |
| 18:23.26 | gusto | yes |
| 18:23.37 | [TK]D-Fender | gusto, ind a clue. Seriously. You are thorwing FUD around left, right, and center. |
| 18:23.38 | AkkerKid | budum tsh. |
| 18:24.02 | [TK]D-Fender | throwing* |
| 18:24.04 | [TK]D-Fender | gah |
| 18:24.46 | carrar | gusto, you really should pay more attention to China |
| 18:25.03 | gusto | i do not care about china |
| 18:25.30 | carrar | obviosuly |
| 18:25.32 | gusto | there are ppl out there that say that chinese products are insecure because they have backdoor, but american companies do that too, so where is the security issue? |
| 18:25.42 | drmessano | The issue isn't with Huawei specifically |
| 18:26.08 | gusto | when there is someone around watching you watch p0rn ... do you care if he/she sits in peking or NY? |
| 18:26.22 | carrar | US has not recorse with Huawei because China does not play fair |
| 18:26.32 | carrar | s/not/no/ |
| 18:26.55 | gusto | well. there should not be any hardware with backdoors, but why do this idiots on TV say then, that the problem is somewhat "only" with ZTE or Huawei |
| 18:27.28 | drmessano | The issue is that mobile phone networks are the new 1800s telegraph, 1900s wireline telephone.. and no american company produces a product that even close to competes with the Huawei option.. and now we have an entire America (f*ck, yeah!) communication network built by the chinese |
| 18:27.51 | gusto | well ... somehow as even CISCO would be popular by playing fair :-D LOL |
| 18:28.13 | [TK]D-Fender | drmessano, Well.... the railways were already done.... |
| 18:28.29 | drmessano | and it happened faster than congress could regulate the shit out of it or put some other hold on it, because we couldn't stand for the chinese supplying our whole new modern PSTN |
| 18:28.37 | carrar | They sare still building railways today!! |
| 18:28.40 | carrar | are |
| 18:29.02 | gusto | so what is the problem then? |
| 18:29.29 | drmessano | American companies dropped the ball on the mobile network build out, and aren't even in place for the next generation.. and the public is just now realizing it, and suddenly don't like it. |
| 18:29.35 | drmessano | That's the issue |
| 18:29.41 | gusto | are the railways in US somehow "insecure" only because they were built by "untrusted" slaves or some immigrants? |
| 18:30.07 | carrar | meanwhile |
| 18:30.21 | carrar | please take this to #conspiracy |
| 18:30.41 | drmessano | No, but China is the big red enemy, incredibly evil, and all these hundreds of thousands of cell sites powered by chinese equipment *must be a bad thing somehow because it has to be* |
| 18:30.52 | gusto | well, they should be taking that to #conspiracy or better - conspiracy TV channel and not on the main news |
| 18:31.18 | carrar | gusto, did you see this on your 5PM news? |
| 18:31.24 | gusto | no |
| 18:31.26 | gusto | on youtube |
| 18:31.33 | carrar | or on ENTERTAINMENT NEWS |
| 18:31.43 | carrar | AKA Fox and CNN |
| 18:31.44 | drmessano | It is news.. It's news because the public had no idea that the mobile carriers were betraying them by building out 3G and 4G on chinese hardware, and OMG WE'RE SURROUNDED!! CHINA IS ON MY PHONE!!! |
| 18:32.41 | drmessano | Expect more pieces on Dateline and 20/20 about this until the american public discovers Sam Walton was CHINESE TOO |
| 18:32.52 | drmessano | Then we'll be all over that |
| 18:32.58 | carrar | Will this effect my weekend BBQ? |
| 18:33.05 | drmessano | It already has |
| 18:33.07 | carrar | WTF |
| 18:33.14 | AkkerKid | His real name was Sam Wonton! |
| 18:33.24 | carrar | You know |
| 18:33.29 | carrar | we could have taken China |
| 18:33.31 | drmessano | If I call you to tell you how I want my ribs cooked, CHINA IS TEH BACKBONE... |
| 18:33.52 | carrar | but the president at the time said no |
| 18:34.56 | carrar | Korean War was harsh |
| 18:34.57 | gusto | isnt the real problem some xenophobia in the US? i have the impression that they think that all chinese are evil and when someone who is now some kind of leader or founder of a chinese telco company must be some communist spy, because he served in the chinese army (like a lot of other chinese ppl did) |
| 18:35.10 | Kobaz | drhonk|afk: medium/rare |
| 18:35.14 | Kobaz | drmessano: |
| 18:35.19 | tzanger | this was all done right around election time to get everyone good and riled up |
| 18:35.24 | drmessano | Lets face it.. As much as we dog out Chinacom phones, knock-off iPad chargers and cables, cheap USB wireless and bluetooth adapters, and the rest of the crap we buy dirt cheap on ebay, the shit is better than half of the name-brand crap, and 10% of the price. They already won |
| 18:35.24 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-dakwholrxgmjqouk) |
| 18:35.30 | gusto | i wonder what US catholics do when they find out that the current pope was to hitlerjugend |
| 18:36.51 | drmessano | Who wants to laugh at someone for buying an ATA or IP phone off ebay when chinacom "crap" is powering our mobile phone network, and who-knows-what-else-that-sits-in-a-rack-making-other-stuff-work |
| 18:37.07 | gusto | drmessano: the price they sell it off does not have anything to do with the quality / production costs |
| 18:37.19 | drmessano | I never said it did |
| 18:37.36 | carrar | I'm inside of your IP PHone stealing all your packets |
| 18:38.17 | gusto | ppl. i once bought an apple mighty mouse for 50 EUR ... and then it stopped working so i had to disassemble it to clean it and when i opened it up, i was shocked by the hardware i discovered in there. that was the same crap as in a mice for 5 EUR ... and from there on i only trust logitech |
| 18:38.48 | AkkerKid | you think logitech does any different? |
| 18:38.52 | gusto | yes |
| 18:38.58 | gusto | logitech has better hw in there |
| 18:39.04 | AkkerKid | I guess you just believe what you want |
| 18:39.09 | gusto | no |
| 18:39.19 | gusto | i never had a broken logitech mice |
| 18:39.22 | drmessano | Hell, I just bought a USB+5-in-1 Card reader for my iPhone/iPad, something that Apple doesn't even make, for $5 on ebay. I suspect I will get $5 out of it, since I am not running my ventilator or artifical heart off of it. |
| 18:39.43 | gusto | however, my last mice i bought would be 5 years ago, so i do not know what's now, but until back then it was so |
| 18:40.12 | AkkerKid | gusto, there's more to an assembled piece of hardware than the part numbers on the chips |
| 18:40.59 | drmessano | I remember someone telling me that they would never buy a Foxconn barebones machine, because it's cheap chinese crap... but they own an iPad, iPhone, and Macbook |
| 18:41.02 | drmessano | UMMMMMMMMM...... |
| 18:41.13 | drmessano | UHHHHHHH.... |
| 18:41.18 | rgsteele | So, I inherited an ugly dialplan, and am attempting to clean it up a little. Can I get some thoughts on the following, please: http://pastie.org/5036701 |
| 18:41.40 | AkkerKid | What do you think chip manufacturers do with the chip that don't perform up to a final requirement? THrow them away? |
| 18:41.53 | rgsteele | Functionally equivalent, for the most part? Could anything be done better? |
| 18:42.36 | [TK]D-Fender | rgsteele, Stop jumping to another exten, just make those direct priorities |
| 18:43.05 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 18:43.54 | [TK]D-Fender | rgsteele, exten => s,n,GotoIf($[${DIALSTATUS} != CHANUNAVAIL] & $[${DIALSTATUS} != CONGESTION]]?s-${DIALSTATUS},1) <--- bracket & expression FAIL. |
| 18:44.44 | [TK]D-Fender | rgsteele, Nexte it's a good idea to do Congestion() with a timeout like 5 and then hangup. |
| 18:45.05 | rgsteele | [TK]D-Fender: It's untested, I'm sure there are one or two syntactical typos - I was going for general look & feel before getting down to brass tacks. |
| 18:45.18 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 18:45.57 | [TK]D-Fender | rgsteele, prio witihin s, not jumping out. |
| 18:46.24 | [TK]D-Fender | rgsteele, Not a fan of globals for options either... better to pass as parms to a macro, etc |
| 18:46.47 | rgsteele | [TK]D-Fender: Yeah, I figure I'll probably be piece-mealing this instead of trying to overhaul the whole thing in one swipe. |
| 18:47.43 | rgsteele | [TK]D-Fender: I don't think I'm following - jumping out with the GotoIf's, you mean? |
| 18:48.03 | [TK]D-Fender | yes |
| 18:48.10 | [TK]D-Fender | s-insanity |
| 18:50.24 | rgsteele | How would you do the equivalent of line 3 (on the bottom) with direct prios? I mean, if it's chanunavail or congestion, I want to try the second gateway; otherwise, it bails. |
| 18:51.22 | [TK]D-Fender | Yes well you jump out to another exten... just make that another priority |
| 18:51.43 | [TK]D-Fender | s(congestion) |
| 18:51.45 | [TK]D-Fender | etc |
| 18:52.06 | rgsteele | Well, because CANCEL/NOANSWER/BUSY aren't congestion. |
| 18:52.18 | rgsteele | I don't want to assume congestion if that's not the most appropriate choice. |
| 18:53.11 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 18:53.20 | *** join/#asterisk mokmeister (~mokmeiste@109.76.82.142) |
| 18:54.32 | rgsteele | [TK]D-Fender: Or am I just being dense? Not an uncommon occurrence, so a fair question :) |
| 19:01.00 | *** join/#asterisk bmg505 (~leon@196-209-71-4.dynamic.isadsl.co.za) |
| 19:03.30 | rgsteele | [TK]D-Fender: If I'm not off-base, I'd love something more succinct than the expression you noted above. |
| 19:04.24 | [TK]D-Fender | rgsteele, s(cancel), s(or), s(however), s(many), s(of), s(these), s(you), s(need) |
| 19:05.08 | [TK]D-Fender | If you can goto some Frankendialplan extension, then you can just make proper labels like everyone else... |
| 19:09.14 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 19:09.47 | rgsteele | [TK]D-Fender: Oh, I misunderstood what you meant. But yes, that would definitely be beneficial. |
| 19:10.37 | rgsteele | Anything else you can think of? I mean, functionally, it seems equivalent to you, then? (ignoring the lack of labels for the moment) |
| 19:11.43 | [TK]D-Fender | rgsteele, well I don't know wverything you are expecting it to do. I don't know that this is a macro/gosub, etc or not... don't see what else you have in there... |
| 19:12.01 | [TK]D-Fender | rgsteele, So it's down to basic cleanliness for the part I do see |
| 19:12.08 | rgsteele | I just wanted to clean up the GotoIf's everywhere. So, cutting it down to a single GotoIf from four was a start. |
| 19:13.15 | rgsteele | But, cleanliness means nothing if the results differ, and it's hard to simulate congestion after dialing a real external number without a bunch of hackery. |
| 19:13.31 | rgsteele | [TK]D-Fender: Thanks for the input, btw |
| 19:15.26 | [TK]D-Fender | rgsteele, Well one by one you seem to haev almost all the possible values covered anyway... you could simply have done them consistently as priorities instead of having : exten => noanswer,1 then exten => s-CANCEL,1 |
| 19:16.15 | [TK]D-Fender | rgsteele, you did 2 ways of jumpin based on the very same output var, but never "really" changing scope as far as processing is concerned (not just jumping to a radically different place in the dialplan |
| 19:17.12 | rgsteele | Yeah, like I said it'll probably take a few sweeps of incremental changes before it becomes palatable :) |
| 19:19.16 | drmessano | Does anyone know how I can implement G.729 legally on an Asterisk 11 RC1 install since Digium has completely neglected us early adopters? |
| 19:19.22 | drmessano | </trollbait> |
| 19:23.36 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:23.36 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:25.10 | [TK]D-Fender | drmessano, In a "grey" scope I would thing that you could consider unused licenses for one version and use one of "those" builds of codecs out there within the qty allowed and I doubt anyone should really have a fit over it... |
| 19:25.34 | [TK]D-Fender | drmessano, Like saying "I'm paying for the right, but your physical product needs to be substituted temporarily" |
| 19:25.50 | [TK]D-Fender | drmessano, VERY light-grey I figure. |
| 19:28.22 | drmessano | I thought about that as well... especially with a majority of my endpoints using G729 and little transcoding involved |
| 19:28.41 | drmessano | I may just do that |
| 19:29.39 | drmessano | I'm gonna start a rumor that g.729 is deprecated in Asterisk 11 and maybe that will speed things up |
| 19:31.07 | drmessano | I can't imagine it being much longer though.. Astricon is in 11 days. I'm sure we're pushing for the big 11.0.0 for then |
| 19:31.41 | drmessano | I'm just impatient. 11 RC1 is out, so Asterisk 10 is now old and busted to me |
| 19:33.18 | [TK]D-Fender | 11 steps to go.. |
| 19:35.30 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 19:37.17 | leifmadsen | drmessano: actually now that RC1 is out, the commercial modules should not be far behind |
| 19:37.31 | Qwell | leifmadsen: he knows |
| 19:38.47 | leifmadsen | Qwell: ok |
| 19:40.37 | *** part/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 19:46.08 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 19:46.38 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 19:50.28 | drmessano | @leifmadsen: That's the LIE that Qwell has told me. |
| 19:50.35 | leifmadsen | drmessano: indeed |
| 19:50.37 | Qwell | No, you're wrong. |
| 19:50.40 | gusto | ah |
| 19:50.40 | Qwell | I've told you lots of lies. |
| 19:50.44 | drmessano | lol |
| 19:50.52 | gusto | LTE came around to the discussion, how interesting |
| 19:51.06 | drmessano | Ask me no questions, I will tell you no lies |
| 19:52.08 | gusto | you are behaving here like small children |
| 19:52.36 | Qwell | gusto: You are correct. |
| 19:52.47 | jaytee | I'm trying to provision a Cisco 7960 for Asterisk. When the phone boots it first says "Configuring VLAN" then it switches to "Configuring IP" and then it quickly displays "TFTP P0S3-08-3-00.loads" and "Verifying load" and then back to "Configuring VLAN", "Configuring IP" and then it shows "Requesting configuration" and then "Resetting". This process loops over and over again. |
| 19:53.19 | gusto | i am in germany i can not hear LTE any more ... i have a headache from it, i do not watch TV any more ... i can not take it any more ... that LTE bullshit is being marketed too aggressively |
| 19:53.30 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:53.31 | Qwell | gusto: Where did LTE come from? |
| 19:53.56 | jaytee | If I do a tcpdump -i eth1 tftp on my Asterisk system I can see traffic from the Cisco phone to my Asterisk system |
| 19:54.37 | gusto | Qwell: i do not care, i wish it to be gone |
| 19:57.05 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 20:01.36 | drmessano | gusto: You sound angry. Have you tried herbal tea? |
| 20:01.47 | gusto | no |
| 20:01.59 | gusto | but i am thinking about making some tea |
| 20:02.35 | drmessano | I think that's a good idea. Maybe some Chamomile |
| 20:02.46 | gusto | but there are some things that are making me happy |
| 20:03.12 | gusto | drmessano: are you from USA? |
| 20:03.16 | drmessano | You hit a kid while driving to town yesterday? |
| 20:03.20 | drmessano | Yes, I am |
| 20:03.53 | Jouva | Qwell: FYI, my BLF issue seemed to resolve itself today. I've admittedly had a copy of the sip.conf file on the network that wasn't always copied up to the asterisk server and instead edited directly, but I swear I looked at the general section and a phone extension side by side and they looked identical |
| 20:04.20 | Jouva | Qwell: But it's also possible that while we were working on getting the phones and Asterisk on its own subnet, that we added some firewall rules that fixed this all |
| 20:04.27 | gg608f | hi all, anyone knows how to see if Asterisk 10.9 has fix for http://code.google.com/p/sipml5/source/detail?r=373330 |
| 20:04.35 | gg608f | Rev 373330 |
| 20:04.37 | gusto | i have two soft toys that make me happy when i look at them, one of them is a BSD devil beastie |
| 20:04.50 | gg608f | im trying to implement Websockets |
| 20:05.14 | Jouva | there's no revision 373330 |
| 20:05.24 | Jouva | it only goes up to revision 115 |
| 20:07.04 | gg608f | if you open http://svn.digium.com/svn/asterisk/trunk/ you will see Revision 374876, so I assume I will get the fix in this one |
| 20:07.13 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 20:07.50 | Jouva | you linked to revision 373330 for the sipml5 code, which only goes up to revision 115 |
| 20:08.12 | gg608f | oh k |
| 20:08.16 | gg608f | thanks Jouva |
| 20:08.22 | Jouva | no prob |
| 20:08.29 | Jouva | And I've gotta go anyway |
| 20:10.54 | drmessano | gusto: The BSD Devil thingo is cute. Imagine if they sold one to all of the 7 people that use BSD |
| 20:11.24 | gusto | i have three of them |
| 20:11.36 | gusto | mine are made in east germany |
| 20:13.07 | carrar | HEY |
| 20:13.11 | carrar | 8 people |
| 20:13.15 | carrar | I use FreeBSD |
| 20:13.26 | jpsharp | I use NetBSD. So 9. |
| 20:14.04 | gusto | well |
| 20:14.11 | rrittgarn | AMI question: How do i pass a value to a sub in the dial plan? Setting up a "Click to Record" esq function. |
| 20:14.25 | drmessano | Maybe double-digits someday for BSD. That's awesome |
| 20:14.36 | drmessano | I hope adoption picks up soon |
| 20:14.36 | carrar | maybe |
| 20:14.54 | carrar | back in the day FreeInternet.Com was 99.999999% FreeSBD |
| 20:14.56 | kikohnl | I use FreeBSD for some things, just not asterisk |
| 20:15.02 | carrar | hundreds of machines |
| 20:15.13 | drmessano | BSD is good for more than compiling BSD components? |
| 20:15.17 | drmessano | Daaaaayum |
| 20:16.03 | gusto | BSD is OK, but the hardware support is a problem, i can not use FreeBSD or some other on any of my laptops (i have only two) |
| 20:16.16 | jpsharp | I have two Linux VMs. One for Asterisk, one for Minecraft. Everything else is NetBSD. |
| 20:16.20 | gusto | on PC's i have, but i have no desktop PC here |
| 20:16.27 | jmetro | bsd is good for zfs, and even then that can work on linux now. |
| 20:16.35 | drmessano | Yeah |
| 20:16.37 | carrar | FreeBSD always worked great for laptops I had |
| 20:16.41 | gusto | well, i have freebsd here as well (on a virtualbox) |
| 20:17.02 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:17.06 | gusto | depends on the drivers you need |
| 20:17.11 | drmessano | Ok, and I guess there is FreeNAS |
| 20:17.14 | Qwell | I used a BSD once. Then I realized it was no longer 1983. |
| 20:17.20 | carrar | haha |
| 20:17.31 | carrar | SHouldn't you be warming up milk or something |
| 20:17.35 | gusto | on my netbook it would work, but without the fan control, it's not usable, because w/o fancontrol the fan is too loud |
| 20:18.41 | *** join/#asterisk nightrid3r (~kvirc@94-227-155-32.access.telenet.be) |
| 20:19.47 | carrar | haha there are a lot of "stop making fun of" videos on youtube |
| 20:20.04 | carrar | Guess they don't need one for freebsd |
| 20:22.28 | jpsharp | LEAVE BSD ALONE! |
| 20:22.47 | gusto | should i call my devil? |
| 20:23.08 | gusto | he is on the bed ready to go |
| 20:25.23 | gusto | my soft toy has connections to huawei and ZTE and thus to the chinese army :-D LOL |
| 20:28.21 | *** join/#asterisk beefcafe (~beefcafe@unaffiliated/beefcafe) |
| 20:28.57 | beefcafe | hi, is 11.0 near release? |
| 20:30.07 | [TK]D-Fender | beefcafe, Next Eventually ... SHARP |
| 20:30.36 | gusto | cold coffee |
| 20:30.53 | beefcafe | hm? |
| 20:31.12 | gusto | i am angry because of some austrian economists |
| 20:31.24 | beefcafe | according to the wiki, 11.0 will release on oct. 17? |
| 20:31.47 | gusto | but i did not discover america, the austrian school is known to be suggestion-resistent |
| 20:32.02 | gusto | beefcafe: well, why do you need that release? |
| 20:32.07 | tapout | what's another provider in the category of voip.ms ? I love voip.ms, i just want to try another provider along side |
| 20:32.43 | beefcafe | gusto: because I'm about to install a new copy |
| 20:32.44 | Qwell | beefcafe: Where do you see that? |
| 20:32.58 | beefcafe | Qwell: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 20:33.07 | Qwell | 2012-10-xx (tentative) |
| 20:33.09 | gusto | beefcafe: well, i use 1.8, that's fine |
| 20:33.14 | lvlinux | i need it for google voice/motif support |
| 20:33.21 | beefcafe | Qwell: First Release of Asterisk from Branch 3rd Wednesday of October |
| 20:33.26 | gusto | what is .ms for a TLD? |
| 20:33.48 | beefcafe | gusto: does 1.8 work well with google voice? |
| 20:33.53 | Qwell | gusto: Montserrat |
| 20:33.55 | carrar | montserrat |
| 20:34.06 | Qwell | beefcafe: *shrug* just a goal, invented months ago |
| 20:34.27 | beefcafe | Qwell: alright. |
| 20:34.36 | gusto | who cares about google voice, but i think yes, there is chan_gtalk |
| 20:34.43 | lvlinux | 1.8 was real sketchy for me w GV |
| 20:35.00 | Qwell | Wait for 11 if you want to use Google Voice stuff. |
| 20:35.11 | Qwell | You could always just get the RC and test it... |
| 20:35.12 | beefcafe | I do, because I'm not in the US atm |
| 20:36.40 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 20:36.46 | gusto | well, i do not understand what that has to do with google voice stuff, ha? |
| 20:36.49 | beefcafe | Qwell: That's what I plan to do. Is it possible to use only GV w/ IPv6 and all other stuff w/ IPv4? |
| 20:37.20 | beefcafe | gusto: free calls, ofc |
| 20:37.22 | gusto | asterisk and IPv6 is a problem, according to my experiences |
| 20:38.02 | beefcafe | so far I only need the GV to go over IPv6, if possible at all |
| 20:38.52 | beefcafe | I intend to use all extentions and trunks over IPv4. |
| 20:38.57 | gusto | i do not see the problem in the possibility, that would go, but i for example tried more times to set udpbindaddr to :: and so on, but it never worked |
| 20:39.02 | Qwell | file: IPv6 + chan_motif? |
| 20:39.28 | tzanger | motif? wow at least go chan_qt or something :-) |
| 20:39.30 | Qwell | gusto: Get a real OS. :) |
| 20:39.36 | gusto | and to freebsd it's even worse, because there when you use IPv6, IPv4 does not work any more |
| 20:39.39 | file | unsupported at the moment |
| 20:39.46 | Qwell | beefcafe: ^ |
| 20:39.56 | file | and GV doesn't do IPv6 to begin with |
| 20:40.06 | gusto | so so |
| 20:40.08 | Qwell | file: surprising |
| 20:40.13 | beefcafe | Qwell: yes. IPv6 would be used only for getting google to connect my calls. all other thing should go over IPv4. Is it feasible? |
| 20:40.17 | gusto | not surprising |
| 20:41.10 | beefcafe | file: It doesn't? GV works fine over IPv6 in browers. |
| 20:41.11 | xSmurf | hey guys, a little off topic, but does anyone here have experience with old ACN IRIS-3000 softphones?! I got it flashed and it can receive calls just fine, but when I try to make a call all I see in the logs is "Using SIP RTP TOS bits" and "Using SIP RTP mark 5" over and over again |
| 20:41.35 | file | browser != XMPP + Jingle |
| 20:42.06 | jaytee | anyone have a set of working config files for a Cisco 7960 they would be willing to share (masking out or changing passwords or other sensitive info of course)? |
| 20:42.40 | beefcafe | file: fair enough. do you know if google authenticates geoip info on XMPP + Jingle? |
| 20:42.48 | file | it doesn't |
| 20:43.20 | beefcafe | that sounds much better. thanks |
| 20:45.37 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-22-148.broadband.tenet.odessa.ua) |
| 20:48.47 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:48.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:48.52 | *** join/#asterisk lorsungcu (~anonymous@75-146-46-85-Minnesota.hfc.comcastbusiness.net) |
| 20:49.30 | gg608f | hi jaytee |
| 20:49.35 | gg608f | i think I do 1 sec |
| 20:51.45 | gusto | heh |
| 20:51.53 | gusto | someone from ukraine was here :-D |
| 20:52.00 | gusto | odessa is btw. not that bad city out there |
| 20:53.21 | xSmurf | no one as any idea?? :( |
| 20:53.34 | gusto | about what? |
| 20:53.56 | xSmurf | see a couple of lines above? |
| 20:54.01 | gusto | yes |
| 20:54.27 | xSmurf | I wished I had more info, but this is the only things that appears in the logs |
| 20:57.01 | gusto | welll |
| 20:57.15 | gusto | TOS is only about the QoS |
| 20:57.41 | gusto | but no idea if your other network hardware cares about QoS because very few do |
| 20:58.07 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-22-148.broadband.tenet.odessa.ua) |
| 20:58.18 | xSmurf | just turned on debugging, phone is booting I'll see if I get anything more from this |
| 21:00.06 | xSmurf | not a whole lot more |
| 21:02.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:03.45 | xSmurf | essentially http://pastie.org/private/zxrzoocjivomkioxylz2tw |
| 21:07.23 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 21:08.38 | *** join/#asterisk Galen (~Galen@rrcs-76-79-170-42.west.biz.rr.com) |
| 21:17.53 | *** join/#asterisk Galen (~Galen@rrcs-76-79-170-42.west.biz.rr.com) |
| 21:18.06 | xSmurf | it would be really nice to get this working, my community space could really use a real sip phone instead of some half baked unlocked ATA :/ |
| 21:21.32 | *** join/#asterisk nightrid3r (~kvirc@94-227-155-32.access.telenet.be) |
| 21:23.07 | Qwell | xSmurf: How much is your time worth? Just buy something decent. |
| 21:24.02 | xSmurf | we're freaking poor :/ |
| 21:27.47 | *** join/#asterisk nightrid3r (~kvirc@94-227-155-32.access.telenet.be) |
| 21:30.13 | *** join/#asterisk nix8n82-phone (~AndChat@231.sub-174-253-142.myvzw.com) |
| 21:39.13 | JunK-Y | mooo! |
| 21:39.47 | jpsharp | Quack |
| 21:49.31 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 21:51.29 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 21:55.50 | Galen | Hello, does anyone know of a way to pull a running config file off of asterisk? I have a file that got interrupted midsave and is now 1/10 of what is currently running on the my server. I have not reloaded the server since the interrupted save occured. |
| 21:59.17 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 22:00.52 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 22:01.13 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-dnwholjkcywkzghg) |
| 22:01.30 | jmetro | if its a dialplan you can use dialplan save |
| 22:08.29 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 22:11.37 | Galen | I tried the dialplan save command and it gave me the following:I can't save dialplan now, see 'extensions.conf' example file. |
| 22:11.38 | Galen | Command 'dialplan save ' failed. |
| 22:19.02 | navaismo | the other is dump the ouput of dialplan show and then edit by hand the dialplan in that archive |
| 22:19.19 | navaismo | asterisk -rx 'dialplan show' > temdialplan.conf |
| 22:20.20 | JerJer | Galen: check your conf files into svn / git |
| 22:20.22 | JerJer | :) |
| 22:20.35 | JerJer | then when this happens again, svn up :) |
| 22:21.08 | JerJer | not if :) |
| 22:22.01 | navaismo | You need static=yes and writeprotect=no in order to use the dialplan save command |
| 22:23.40 | Galen | Yeah, that is what it looks like. I had removed those lines from my dialplan so I will start working on the export and rebuild. |
| 22:23.53 | tzanger | JerJer: re |
| 22:24.04 | Galen | I'll also back it up for next time. |
| 22:24.32 | JerJer | tzanger: meep meep |
| 22:24.45 | tzanger | sup, long time |
| 22:25.12 | JerJer | in the Silly Valley now (no more frozen michigan winters :) |
| 22:25.41 | tzanger | haha nice |
| 22:25.58 | tzanger | I'm out in denver at the moment (just a business trip) |
| 22:26.07 | JerJer | schweet |
| 22:26.23 | JerJer | i'm headed to washdc end of month for a site build - gonna be fun |
| 22:26.46 | tzanger | if you say so :-) |
| 22:45.21 | JerJer | only 50 phones |
| 22:45.38 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 22:52.34 | *** join/#asterisk rx4change (~joshua@65.112.207.3) |
| 22:52.54 | rx4change | can anyone tell me if something funky happens with outbound tls registrations |
| 22:53.04 | rx4change | I register with standard UDP, the SIP packet looks good |
| 22:53.22 | rx4change | just add the tls:// on front on register string and the contact header host gets nulled out |
| 22:53.38 | rx4change | shows 1234567890@(null) |
| 22:54.17 | JerJer | is there only one TLS connection on that IP address? |
| 22:54.56 | JerJer | i know one cannot have more than one TLS session per IP address |
| 22:55.55 | rx4change | at this point there is just the one |
| 22:56.04 | rx4change | the rest of the REGISTER message appears perfect |
| 22:56.35 | rx4change | it's just that the Contact: line has the host (appears normall in UDP) appear as (null) |
| 22:57.49 | *** join/#asterisk iprouteth0 (ccf60469@gateway/web/freenode/ip.204.246.4.105) |
| 23:13.09 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a10b9.pool.mediaWays.net) |
| 23:29.42 | rx4change | interestingly, somewhat buggy behavior - if you don't have tlsenable explicitly on, but try to register to a server with tls:// you get the null contact header |
| 23:29.43 | rx4change | odd |
| 23:29.58 | rx4change | set tlsenable=yes and you're good |
| 23:32.15 | JerJer | rx4change: that smells like a bug needing to be submitted |
| 23:32.20 | iprouteth0 | rx4change: could you elaborate? Sip w/ TLS? Feel like I'm coming into a conversation somewhere in the middle |
| 23:38.00 | rx4change | Trying to do an outbound registration via tls... so tls:// |
| 23:38.14 | rx4change | in sip.conf if you don't have tlsenable=yes, it attempts to register |
| 23:38.27 | rx4change | but it sets the contact header host to (null) |
| 23:38.40 | rx4change | this is on 1.8.8.2, so no guarantee it's an issue in 10 or 11 |
| 23:38.57 | rx4change | but I did a rather large amount of hair pulling on this |
| 23:39.29 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 23:43.23 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 23:43.59 | Dovid | can anyone help me get variables from an AGI? i am using php agi. when i do the get_variable and I do an agi debug I see it yet i can not seem to get in to a variable. |
| 23:44.57 | iprouteth0 | I've yet to work with SIP w/TLS |
| 23:45.36 | iprouteth0 | I'd love to try it with my new Digium D40 I won (yay!), but my cisco phones definitely won't support it |
| 23:50.02 | ChannelZ | Dovid: are you trying to get a channel variable, or what |
| 23:50.26 | Dovid | ChannelZ: Yes |
| 23:50.53 | Dovid | ChannelZ: <SIP/vm1-5744-0000000e>AGI Rx << GET VARIABLE DIALSTATUS |
| 23:50.53 | Dovid | <SIP/vm1-5744-0000000e>AGI Tx >> 200 result=1 (CANCEL) |
| 23:50.58 | Dovid | so Asteirsk is getting it |
| 23:51.02 | Dovid | but PHP does not |
| 23:52.38 | ChannelZ | perhaps the PHP wrapper you're using is just broken then. |
| 23:52.47 | ChannelZ | You're looking at the result as an array, right? |
| 23:54.21 | ChannelZ | Or are using the second argument to get_variable to tell it to return the value only I guess (just looking at the online docs for what I think you're using) |
| 23:55.49 | Dovid | ChannelZ: Yes. It seems like PHP is not sendig it back correctly. if I do print_r($my_var) then I get code is HAN and data is GUP |
| 23:55.51 | Dovid | strange |
| 23:56.24 | Dovid | ChannelZ: http://pastebin.com/98hS26XD |
| 23:56.50 | Dovid | ChannelZ: http://pastebin.com/WmySVfiZ |
| 23:56.52 | Dovid | thats the php code |
| 23:59.28 | ChannelZ | I dunno, guess it depends if 'evaluate' is getting/parsing the data right or not. |
