| 00:00.55 | navaismo | is that -->Intel 82574L |
| 00:06.52 | [TK]D-Fender | yes |
| 00:07.57 | navaismo | thanks |
| 00:08.08 | navaismo | see you |
| 00:12.22 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 00:23.54 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 00:30.12 | *** join/#asterisk Cubber (~ronny@cpe-74-71-254-190.twcny.res.rr.com) |
| 00:30.12 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 00:30.12 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 00:30.12 | *** join/#asterisk Rac-on (jasper@bambi.rac-on.nl) |
| 00:30.12 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 00:30.12 | *** join/#asterisk Martin` (martin@shell.octocore.net) |
| 00:30.12 | *** join/#asterisk paul_andrew (~admin@ec2-50-17-118-113.compute-1.amazonaws.com) |
| 00:30.12 | *** join/#asterisk eureka^ (~vyatta@vpnc.sfd.asgdc.net) |
| 00:30.13 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 00:30.28 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 00:50.08 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.138) |
| 00:58.49 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 01:04.11 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 01:15.17 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xgrtjghxnjoeeijd) |
| 01:17.05 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:22.24 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:28.07 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:34.31 | *** join/#asterisk Bradada (~Bradada@59-120-137-34.HINET-IP.hinet.net) |
| 01:38.08 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.110) |
| 01:38.14 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 02:00.14 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 02:00.14 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:06.15 | vlad_sta_ | Question: Does anyone know why this message appears in CLI: "WARNING[10646][C-00000020]: chan_sip.c:16326 check_auth: username mismatch, have <CUS103847120943710>, digest has <CUS103847120943891>"? |
| 02:08.50 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 02:13.43 | *** join/#asterisk radic (~radic@dslb-094-216-252-124.pools.arcor-ip.net) |
| 02:19.38 | *** join/#asterisk Spengler1 (~Spengler1@pool-98-117-215-165.bltmmd.fios.verizon.net) |
| 02:21.32 | *** join/#asterisk phunguy (santas@dhcp.i-p.org.uk) |
| 02:36.46 | *** join/#asterisk kopilo (~kopilo@crm.anat.org.au) |
| 02:37.32 | kopilo | hey I have a system running asterisk 1.4 (debian) and I want to upgrade it to 1.6 is there anything I should be aware of, such as differences in configuration? |
| 02:40.49 | [TK]D-Fender | kopilo: 1.6 is dead. All THREE branches of it |
| 02:40.58 | kopilo | ahh |
| 02:40.59 | [TK]D-Fender | kopilo: The quick warning is "don't" |
| 02:41.19 | kopilo | so is there an upgrade from 1.4? |
| 02:41.49 | kopilo | also thank you |
| 02:41.54 | [TK]D-Fender | There are changes between major branches, but if you were on full 1.4 spec it shouldn't bee too much unless you're doing a lot of odd bits |
| 02:42.19 | [TK]D-Fender | read the "upgrade.txt" for the details on the major parameter changes, syntax requirements, etc |
| 02:42.58 | kopilo | I inherited the system but as far as complication all the last person did was includes |
| 02:43.01 | [TK]D-Fender | EOl for using the pipe char as a delimiter, minor app parameter changes. new timing options, etc |
| 02:43.11 | kopilo | thanks |
| 02:43.42 | [TK]D-Fender | I would generally recommend going straight to * 11 as it is about as stable as the next thing and is LTS as 1.8 is |
| 02:43.51 | kopilo | oh wait now I see what the problem i |
| 02:43.53 | kopilo | is |
| 02:43.55 | kopilo | "Asterisk 1.6.2.9-2+squeeze6" |
| 02:44.20 | kopilo | yeah I'm going to have to work out how to get 1.8 |
| 02:44.28 | kopilo | or 11 |
| 02:47.59 | kopilo | also I would like some clarification on something, I was told by a supplier that running asterisk and samba on the same box causes conflicts, is there any validity to this? |
| 02:48.52 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 02:49.19 | [TK]D-Fender | Directly, no |
| 02:49.30 | kopilo | thank you |
| 02:49.42 | [TK]D-Fender | However is something eats up needed bandwidth or ties up the CPU the wrong way and well.. this is "voice" |
| 02:49.51 | kopilo | yes |
| 02:50.29 | [TK]D-Fender | Describe your realistic use of the overall machine... |
| 02:51.13 | kopilo | 5 person office, file storage and voip |
| 02:51.29 | [TK]D-Fender | I'd QoS it for sure |
| 02:51.37 | kopilo | yeah |
| 02:52.10 | kopilo | in addition it is one of the only 5 cabled devices, everything else is on wifi (yes it makes me hate my life) |
| 02:53.58 | [TK]D-Fender | That can be a good thing depending how you set it up |
| 02:54.48 | [TK]D-Fender | Put the WiFi on a 2nd NIC in your server so that the voip path is clear and file serving off it doesn't hit the wire |
| 02:57.28 | kopilo | nods |
| 02:57.55 | kopilo | it has also been recommended that I run the server in bridgemode but I'm unsure about that |
| 02:58.10 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:59.28 | [TK]D-Fender | Yup, I was thinking about that. |
| 02:59.50 | [TK]D-Fender | No IP on the 2nd NIC in that case |
| 02:59.59 | kopilo | yeah |
| 03:00.06 | [TK]D-Fender | And if keeps trafic to the server direct |
| 03:00.10 | kopilo | I was thinking of dropping it into DMZ |
| 03:00.31 | [TK]D-Fender | kopilo: This is making the best out of a difficult situation. |
| 03:00.39 | [TK]D-Fender | DMZ not really required |
| 03:00.41 | kopilo | yeah |
| 03:00.47 | [TK]D-Fender | just a question of taking a load off it. |
| 03:00.57 | kopilo | nods |
| 03:01.33 | [TK]D-Fender | I'd like to ask "how much damage will those 5 people do" ... but I suspect you're already afraid |
| 03:01.50 | kopilo | well there are only 3 channels |
| 03:01.57 | kopilo | which are used at any one time |
| 03:02.38 | kopilo | but in reguards to files, we have had everything from 100GB movies (actually legitmately part of work) |
| 03:02.46 | kopilo | sorry have to go to lunch |
| 03:02.47 | kopilo | bbs |
| 03:05.48 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 03:10.34 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 03:15.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.53) |
| 03:30.32 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 03:30.45 | linocisco | hi all |
| 03:31.58 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.141) |
| 03:32.09 | linocisco | i m using cisco SAP502G and cisco 7942G which is also converted to sip. I would like to view telephone directory (name+extension) of all users on phone display screen . How can I configure? |
| 03:37.53 | kopilo | bk |
| 03:40.00 | linocisco | kopilo, bk? |
| 03:42.21 | kopilo | back :) |
| 03:48.34 | [TK]D-Fender | linocisco: * has nothing to do with that |
| 03:48.39 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 03:49.04 | kopilo | is currently compiling latest |
| 03:49.10 | *** join/#asterisk deo (~deo@58.71.19.178) |
| 03:49.23 | [TK]D-Fender | The 7942 support LDAP IIRC, the 502... no idea |
| 03:49.29 | [TK]D-Fender | Go read the docs for the specs |
| 03:49.35 | linocisco | [TK]D-Fender, can phone book feature on all IP phones can connect to asterisk server and retrieve and list all users on ip phone's display?? |
| 03:50.11 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 03:50.57 | kopilo | most voip phones I've seen don't have this 'phonebook' feature |
| 03:51.12 | [TK]D-Fender | linocisco: the SERVER yes.. but there is no STANDARD |
| 03:51.28 | [TK]D-Fender | linocisco: So go find out what your PHONE uses... because this has nothing to do with Asterisk |
| 03:51.47 | kopilo | alternatively you could make an interanet site with a phonebook listing? |
| 03:51.49 | [TK]D-Fender | linocisco: It may use XML. It may use LDAP. |
| 03:51.59 | [TK]D-Fender | Or it may use something else. |
| 03:52.18 | linocisco | kopilo, i dont know how to |
| 03:52.48 | [TK]D-Fender | http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps10499/qa_c67-552213-00.html |
| 03:52.51 | kopilo | was more of a question for D-Fender :D |
| 03:52.55 | [TK]D-Fender | Go read about your product |
| 03:53.20 | [TK]D-Fender | Q. Does the Cisco SPA 502G support XML or Lightweight Directory Access Protocol (LDAP) corporate directory? |
| 03:53.22 | [TK]D-Fender | A. Yes, the SPA 502G supports both LDAP and Cisco XML corporate directory structures. |
| 03:53.45 | kopilo | ooo if asterisk supports xml could you make a directory listing rss feed? |
| 03:54.05 | kopilo | guess it would just take marshalling data if that |
| 03:54.47 | [TK]D-Fender | Asterisk will have nothing to do with this |
| 03:55.09 | kopilo | :/ |
| 03:56.04 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-mukqxohlmsrdpgub) |
| 04:01.37 | igcewieling | heh, I have 9 contacts on my cell phone. I really DO hate people. |
| 04:01.39 | linocisco | what is 3 way calling, CITI, is agent ? |
| 04:01.59 | linocisco | igcewieling, how did u configure to get listed on your phone? |
| 04:02.02 | igcewieling | linocisco: English is not your first language, is it? |
| 04:02.10 | linocisco | igcewieling, no |
| 04:02.26 | linocisco | igcewieling, anyway 3 way calling vs conferencing? |
| 04:03.04 | igcewieling | linocisco: 3=way calling is the ability to add a 3rd party to your call, it is a common feature. Conferencing generally allows a much larger number of people and has more management features. |
| 04:09.47 | linocisco | igcewieling, follow me = meet me conferencing? |
| 04:10.01 | linocisco | igcewieling, thanks for your clarification |
| 04:12.23 | [TK]D-Fender | That isn't a properly formed questionno, followme is followme. |
| 04:12.26 | [TK]D-Fender | It is no confrencing |
| 04:12.29 | [TK]D-Fender | not* |
| 04:12.37 | *** join/#asterisk andross (~andrew@2610:1c0:0:2:101:a57a:5ac7:1688) |
| 04:12.48 | [TK]D-Fender | Folowme tries to dial one thing.. when it doesn't answer it goes to the next |
| 04:12.56 | [TK]D-Fender | Meetme = conferencing. |
| 04:13.06 | [TK]D-Fender | And is not a "phone" feature |
| 04:13.13 | [TK]D-Fender | This is Asterisk doing the work |
| 04:15.12 | kopilo | asterisk has been the least painful thing I have ever compiled, with the exception of my own code |
| 04:16.50 | igcewieling | kopilo: now you have the easy part done. 8-) |
| 04:18.01 | kopilo | no joke :D |
| 04:18.16 | kopilo | luckily I have done the hard part before |
| 04:18.48 | kopilo | actually I really should have read the makefile to know where it was installed to |
| 04:20.25 | *** join/#asterisk yoojin (~yoojin@68-113-101-213.static.leds.al.charter.com) |
| 04:21.13 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 04:21.33 | kopilo | >.> where has asterisk been installed x.x |
| 04:22.43 | kopilo | # asterisk -V |
| 04:22.43 | kopilo | Asterisk 11.3.0 :D |
| 04:22.53 | linocisco | hi all |
| 04:23.01 | linocisco | i have created conference bridge |
| 04:24.15 | linocisco | i have created conference bridge but conferencing is not ok . whenever I dialed to that extension, call declined |
| 04:26.03 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.53) |
| 04:26.16 | kopilo | just asking a sanity check, how many channels do you have? |
| 04:27.14 | [TK]D-Fender | You're right ... not sane at all :) |
| 04:27.39 | [TK]D-Fender | "declined" is a pretty grey term so far |
| 04:27.46 | [TK]D-Fender | Which is where really looking at the call comes in handy |
| 04:27.55 | [TK]D-Fender | Espcially when asking us why something didn't work. |
| 04:28.05 | [TK]D-Fender | By like ... SHOWING us the failure ... or somesuch :) |
| 04:28.08 | linocisco | [TK]D-Fender, on soft phone , it was displayed that call rejected |
| 04:28.23 | [TK]D-Fender | Forget what the phone says. |
| 04:28.32 | [TK]D-Fender | What does ASTERISK say with SIP DEBUG enabled? |
| 04:30.22 | [TK]D-Fender | PASTEBIN is your friend... |
| 04:30.23 | [TK]D-Fender | ~pb |
| 04:30.24 | 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. |
| 04:32.22 | kopilo | ~paste2 |
| 04:32.27 | kopilo | :/ |
| 04:34.31 | linocisco | [TK]D-Fender, mine is asterisk 1.4 |
| 04:34.37 | linocisco | [TK]D-Fender, how to set sip debug? |
| 04:34.47 | [TK]D-Fender | "sip debug on" <- |
| 04:35.14 | [TK]D-Fender | And get off 1.4 No more security updates not more bug fixes. It's EOL |
| 04:35.28 | *** part/#asterisk yoojin (~yoojin@68-113-101-213.static.leds.al.charter.com) |
| 04:35.54 | kopilo | xD |
| 04:36.01 | kopilo | and buggy |
| 04:37.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:39.20 | kopilo | is 1.6 EOL too? |
| 04:39.31 | [TK]D-Fender | yes |
| 04:39.33 | kopilo | :D |
| 04:39.37 | [TK]D-Fender | All 3 actual branches |
| 04:39.49 | kopilo | about time >.> |
| 04:40.04 | [TK]D-Fender | [22:40][TK]D-Fenderkopilo: 1.6 is dead. All THREE branches of it |
| 04:40.18 | [TK]D-Fender | I only told you this exact same thing ... 2 hours ago |
| 04:40.20 | kopilo | :D |
| 04:40.23 | kopilo | yes |
| 04:40.32 | kopilo | I'm not having a good day at work mmk |
| 04:40.55 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 04:51.20 | igcewieling | We use 1.8.21.0, Cert Asterisk 1.8, and 11.2 |
| 04:51.45 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 04:54.04 | linocisco | [TK]D-Fender, my asterisk on QNAP is really crappy. CLI is not shown with asterisk -r |
| 04:54.25 | linocisco | [TK]D-Fender, but QNAP is the only device i need to use. |
| 04:54.53 | [TK]D-Fender | If you can't get to CLI then your setup is broken |
| 04:56.46 | linocisco | [TK]D-Fender, no i m using asterisk.apk like plug in |
| 04:57.04 | [TK]D-Fender | Not anything we support here |
| 04:57.12 | [TK]D-Fender | If it's broken, contact the author |
| 04:57.18 | [TK]D-Fender | Then again ... we've been through this before |
| 04:57.36 | [TK]D-Fender | Get to * CLI. If you can't then you've got a real problem |
| 05:00.05 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 05:01.28 | linocisco | [TK]D-Fender, u have been through this before . what is "this"? this= QNAP ? |
| 05:01.36 | [TK]D-Fender | Yes |
| 05:01.49 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 05:05.53 | ChannelZ | I guess the Q is debatable |
| 05:08.40 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 05:22.05 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 05:31.23 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 05:32.23 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 05:32.48 | *** part/#asterisk deo (~deo@203.177.214.75) |
| 06:01.39 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 06:11.17 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 06:13.15 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:15.07 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 06:16.07 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:90a9:da14:d252:6e0a) |
| 06:19.54 | kopilo | does that mean you support Rpi? :P |
| 06:29.20 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 06:57.10 | eureka^ | just spent the last 4 hours replacing our core SAN |
| 06:57.11 | eureka^ | @_@ |
| 06:59.46 | kopilo | x.x |
| 07:04.36 | izbushka | hi |
| 07:05.46 | izbushka | here is my telephony scheme http://www.zimagez.com/zimage/-23042013-100356.php Is it possible to transfer incoming call to line #24 from SIP/11 to line #21 using Coral functionality? |
| 07:06.10 | izbushka | I can do it using asterisk, but in that case 2 FXO lines busy during transfer |
| 07:07.32 | eureka^ | have you looked at the documentation for the coral device |
| 07:08.04 | izbushka | It can transfer but how asterisk should do this? |
| 07:08.26 | eureka^ | how does it implement its transfer? |
| 07:08.32 | izbushka | so i can transfer from #21 to #22 for exaple |
| 07:09.04 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:9dc8:e0fb:21e0:c827) |
| 07:10.39 | izbushka | Well, from #21 to #22 call transfer by hanging up line for a sec then dial target number |
| 07:11.00 | izbushka | #21,#22 - analog phones |
| 07:20.53 | *** join/#asterisk chris-NB (~chris@212095007073.public.telering.at) |
| 07:29.39 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 07:30.41 | *** join/#asterisk fredericve (~fes@host-212-68-194-46.brutele.be) |
| 07:33.06 | fredericve | Hello friends! what are the recommendations in terms of music on hold usage on asterisk 1.8 and up? use mp3 or not? Is it possible to have a shared stream for all calls that are on hold without using mp3? what is the recommended way to implement that? |
| 07:38.23 | *** join/#asterisk k610 (~K610@cred02.epid.ucl.ac.be) |
| 07:39.47 | andross | i would recommend downloading copyright free classical music from wikipedia |
| 07:42.55 | fredericve | andross: thanks, but regardless which music you use, what is the recommended way to implement things? I read there are licensing issues with the use of mp3 formats. |
| 07:43.27 | andross | use wav files |
| 07:45.10 | andross | moh should allow you to throw a few in a dir and play them in sequence so the hold music doesnt get repetitive |
| 07:51.08 | fredericve | thanks. that is useful. I assume this is the mode=files? |
| 08:10.59 | *** join/#asterisk lkthomas (~lkthomas@n218250151072.netvigator.com) |
| 08:11.01 | lkthomas | hey guys |
| 08:11.24 | lkthomas | any alternative command to show how many channel being use? originally I could use g729 show license |
| 08:12.04 | fredericve | core show channels |
| 08:13.54 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 08:14.45 | lkthomas | no, I want to show the license |
| 08:14.53 | lkthomas | liecnesed channel |
| 08:18.07 | hrolf | is there any function or application that I can use in Asterisk to execute mkdir? |
| 08:18.30 | hrolf | I did try exten => 111,1,SHELL(mkdir myfolder) |
| 08:18.52 | hrolf | but it fails that 'no application SHELL for extension ....' |
| 08:18.56 | izbushka | system() |
| 08:19.02 | izbushka | hrolf, ^^ |
| 08:19.23 | hrolf | izbushka: Okay, also how would I do it in AMI? |
| 08:23.41 | ChannelZ | lkthomas: if g729 is really loaded, 'g729 show licenses' should still work |
| 08:24.38 | hrolf | How can I exec the mkdir using AMI? |
| 08:24.53 | izbushka | what is AMI? :) |
| 08:25.04 | hrolf | Asterisk Manager Interface |
| 08:25.12 | ChannelZ | if you've written an AMI app, why not just do it through whatever environment you write the app in? |
| 08:25.36 | hrolf | ChannelZ: It is FastAGI on remote system |
| 08:26.01 | hrolf | ChannelZ: I want to save my self from having to get a shell and then doing. |
| 08:26.01 | hrolf | it |
| 08:26.35 | ChannelZ | ok so it's AGI, not AMI. |
| 08:26.58 | ChannelZ | Use exec |
| 08:28.54 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 08:29.31 | hrolf | ChannelZ: Yep sorry my mistake |
| 08:30.08 | hrolf | ChannelZ: Exec System "mkdir myfolder" like this? |
| 08:30.09 | hrolf | rite |
| 08:30.12 | hrolf | ? |
| 08:30.41 | ChannelZ | presumably |
| 08:31.28 | ChannelZ | you may or may not have to specify the path to mkdir, I'm not sure what sort of environment the system app runs under |
| 08:32.38 | *** join/#asterisk Schreda (~Schreda@80-110-33-15.static.surfer.at) |
| 08:32.47 | Schreda | Hey everybody |
| 08:32.51 | *** join/#asterisk srp_ (~sandeep@115.119.115.26) |
| 08:33.03 | ChannelZ | Hi, Dr. Nick! |
| 08:33.44 | Schreda | I would have a question I use a Point to Point ISDN configuration. The extensions numbers works just fine but how to handle it in case that a person from pstn want to call the number without extensions....? |
| 08:34.08 | srp_ | Hi, I wanted to know how to access menuselect so that i can enable the cdr_odbc and other modules when i install asterisk from pre-built packages available for my distro rather than compiling from source.. |
| 08:34.23 | Schreda | I want to forward the call in case of no extension number to my default extension number 0 |
| 08:34.29 | Schreda | any idea? |
| 08:38.05 | ChannelZ | Not positive on ISDN but typically if someone calls you from the outside world, it will come into the extension matching the DID |
| 08:38.31 | ChannelZ | If there is really "no extension", the s extension usually matches |
| 08:38.51 | ChannelZ | srp_: you can't |
| 08:39.41 | ChannelZ | srp_: look for separate module packages, otherwise you wll have to build the whole thing from source (not terribly difficult but anyway..) |
| 08:41.18 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 08:44.22 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 08:44.54 | srp_ | ChannelZ: oh.. okay.. the problem is that I am running asterisk on a virtualized server which gives me error: "Illegal instruction (core dumped)". Someone on the mailing list suggested to try prebuilt packages.. |
| 08:45.18 | srp_ | ChannelZ: I mean when i compile from source, it gives me this error |
| 08:45.30 | srp_ | ChannelZ: but built packages are not giving me this error |
| 08:47.03 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:47.08 | ChannelZ | weird |
| 08:48.51 | srp_ | ChannelZ: can this be bacause of this issue? https://issues.asterisk.org/jira/browse/ASTERISK-20128 |
| 08:48.52 | LieutPants | [ASTERISK-20128] [Status: Open] [Assigned: kmoore] Virtualized asterisk.org 1.8.14.0 no longer runs in a KVM virtualized environment. Compiles without error, but fails with Illegal instruction on launch Regression since 1.8.13.0 Last good 1.8.12.2 - https://issues.asterisk.org/jira/browse/ASTERISK-20128 |
| 08:49.23 | srp_ | LieutPants: yep.. you gave me this yesterday.. still struggling with the same problem.. :-\ |
| 08:50.03 | srp_ | LieutPants: the thing is that it works fine with prebuild stuff.. so I think there is some problem with my disctibution / environment under which i'm trying to compile.. |
| 08:54.10 | Schreda | so does this mean I cant forward the call to a specific number in case of no extension is pressed? |
| 09:01.24 | Schreda | any idea? |
| 09:01.46 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 09:02.12 | srp_ | Schreda: I don't know if i'm right but did you see the WaitExten application? |
| 09:03.50 | Schreda | yes but in my case I want that in case no extension is pressed I automatical will be forwarded to extension 0... If someone is triing to call the phone from outside with extension all is already working except in case of no extension thatn the call wont come through |
| 09:05.45 | *** join/#asterisk k610 (~K610@cred02.epid.ucl.ac.be) |
| 09:08.57 | Schreda | any idea? |
| 09:08.59 | srp_ | Schreda: I think you need to put the "run-this-if-no-extension-pressed" part in a lower priority in the dial plan |
| 09:10.31 | Schreda | right but how to do it? |
| 09:15.01 | srp_ | Schreda: In the dial plan, you have the priorities.. example: exten => 555,1,Answer Here 1 is the priority.. |
| 09:15.23 | srp_ | Schreda: 1 has highest priority and 2 a lil lesser and so on.. |
| 09:15.37 | Schreda | I know and mostly I use the n (next) |
| 09:18.38 | Schreda | working thanks |
| 09:18.39 | Schreda | :d |
| 09:18.41 | Schreda | :D |
| 09:20.39 | srp_ | Schreda: no prob :) |
| 09:24.02 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:29.39 | *** join/#asterisk k611 (~K610@cred.epid.ucl.ac.be) |
| 09:40.45 | *** part/#asterisk Schreda (~Schreda@80-110-33-15.static.surfer.at) |
| 10:01.55 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:03.34 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:07.04 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 10:09.48 | *** join/#asterisk ForresGeek (~lee@31.114.7.59) |
| 10:19.24 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-061-084.pools.arcor-ip.net) |
| 10:20.49 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@62.140.132.24) |
| 10:25.24 | *** join/#asterisk CommaCrazy (1000@87.250.37.53) |
| 10:28.24 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 10:28.28 | *** part/#asterisk Matthias (~Matthias@195.16.243.99) |
| 10:29.09 | *** join/#asterisk Matthias (~Matthias@195.16.243.99) |
| 10:29.30 | *** part/#asterisk Matthias (~Matthias@195.16.243.99) |
| 10:29.31 | *** join/#asterisk Matthias (~Matthias@195.16.243.99) |
| 10:30.06 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 10:32.14 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 10:33.12 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 10:34.34 | *** part/#asterisk Matthias (~Matthias@195.16.243.99) |
| 10:37.28 | *** join/#asterisk Matthias (~Matthias@195.16.243.99) |
| 10:41.37 | *** join/#asterisk din3sh (2988fa23@gateway/web/freenode/ip.41.136.250.35) |
| 10:41.47 | din3sh | helo all |
| 10:42.11 | din3sh | am having lots of these [WARNING[26906] channel.c: Codec mismatch on channel SIP/201-00000006 setting write format to slin from alaw native formats 0x8 (alaw)] |
| 10:42.24 | din3sh | what does it mean? |
| 10:44.13 | CommaCrazy | hi all, how can I raise the volume on a outgoing gahdi/zaptel trunk |
| 10:44.16 | CommaCrazy | Asterisk 1.6.0.10-FONCORE-r40 |
| 10:44.17 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 10:44.24 | CommaCrazy | dahdi* |
| 10:48.08 | CommaCrazy | din3sh, probably got one codec on one side and another codec on the other side |
| 10:48.48 | din3sh | can this lead to issues whereby callers cannot hear people they called? |
| 10:48.57 | CommaCrazy | yes |
| 10:49.16 | din3sh | how to fix this? |
| 10:50.01 | CommaCrazy | set the same codec on both sides, and if you control only one side, on that one side set multiple codes and just keep changing the order until you hit the right codec |
| 10:50.42 | din3sh | the calls are coming through an E1 line |
| 10:50.49 | din3sh | to sip extensions |
| 10:53.34 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:53.40 | din3sh | the codec is settling for slin? |
| 10:54.20 | CommaCrazy | one sec |
| 10:54.30 | *** join/#asterisk ForresGeek (~lee@31.114.7.59) |
| 11:00.06 | CommaCrazy | which codec did u put (guessing alaw) |
| 11:00.07 | CommaCrazy | ?? |
| 11:04.56 | CommaCrazy | anyone used dahdi set hwgain and swgain?? |
| 11:07.03 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 11:08.24 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 11:13.12 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 11:13.58 | din3sh | commaCrazy: yes , alaw |
| 11:14.10 | izbushka | is it possible to send FLASH signal to FXO from asterisk (to transfer a call)? |
| 11:15.07 | din3sh | actually in trying to fix the issue, i have set alow=alaw,g729,ulaw,gsm,slin |
| 11:15.09 | din3sh | does not work |
| 11:15.30 | kaldemar | izbushka: you could try app Flash |
| 11:15.44 | izbushka | din3sh, right sintax is one codec per line |
| 11:16.50 | kaldemar | izbushka: not quite. you can use more than one nowadays. even negations like allow=!all,alaw,g729... |
| 11:17.15 | izbushka | hmm, thanks |
| 11:17.58 | izbushka | kaldemar, is Flash an app? "core show applications like flash" shows nothing.. Or it is just extention s,n,flash() ? |
| 11:18.24 | din3sh | yes kaldemar |
| 11:18.29 | din3sh | am having lots of these |
| 11:18.41 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 11:18.46 | din3sh | [WARNING[26906] channel.c: Codec mismatch on channel SIP/201-00000006 setting write format to slin from alaw native formats 0x8 (alaw)] |
| 11:18.49 | *** join/#asterisk PickledParrot (~aj@79.135.102.10) |
| 11:19.08 | din3sh | CommaCrazy: i've used rxgain, txgain in dahdi |
| 11:19.27 | *** join/#asterisk bkruse (~Adium@24.42.181.58) |
| 11:20.23 | kaldemar | izbushka: it is an application. seems that you don't have it loaded/installed. |
| 11:20.46 | izbushka | ok, i'll try to find it, thank you |
| 11:25.09 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 11:25.42 | kaldemar | izbushka: how did you install asterisk? |
| 11:28.38 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 11:31.04 | izbushka | kaldemar, i'm using gentoo linux, so i installed it by 'emerge asterisk' :) that's no problem, i'll find that app. But i think it works only with FXO cards and i have viop gateway device for FXO lines. but i'll try |
| 11:32.11 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:32.29 | kaldemar | izbushka: in that case it's no use for you. |
| 11:33.15 | izbushka | that's sad.. |
| 11:47.43 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 11:52.02 | *** join/#asterisk obalci (~Orkun_BAL@31.131.20.201) |
| 11:54.37 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:55.46 | *** part/#asterisk obalci (~Orkun_BAL@31.131.20.201) |
| 11:56.07 | *** join/#asterisk bombev (~User@PPPoE-Static-40-132.UnicsBG.Net) |
| 11:58.59 | *** join/#asterisk terrayi (~terrayi@203-167-189-215.dsl.clear.net.nz) |
| 11:59.56 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 11:59.56 | terrayi | hello? could somebody help ? |
| 12:00.39 | wdoekes | ~ask |
| 12:00.40 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:01.10 | terrayi | I have tdm2400p card installed working alright |
| 12:01.28 | terrayi | but suddenly a couple days ago it is not accepting any incoming call |
| 12:02.09 | terrayi | from dahdi_monitor i see some rx noise if i call from my mobile |
| 12:02.22 | terrayi | it rings once or two then it just hangs up |
| 12:02.47 | terrayi | on CLI asterisk does not show any message at all |
| 12:03.09 | terrayi | I can make outgoing calls and i can hear sound both ways though |
| 12:04.11 | terrayi | exhausted :( |
| 12:09.21 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 12:09.40 | terrayi | so in short fox port is seemed to be able to listen on incoming ring.. for a short period of time tho but asterisk is not picking up. outgoing call is fine. |
| 12:09.46 | terrayi | fox=fxo |
| 12:14.33 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:26.56 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 12:27.16 | *** join/#asterisk tparcina (~tomo@cisco15.fesb.hr) |
| 12:33.06 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 12:33.34 | danfromuk | Hi, with queues, is it possible to play the position announcements immediately on join? |
| 12:35.57 | [TK]D-Fender | danfromuk, Nope |
| 12:38.42 | *** join/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 12:41.43 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 12:44.35 | *** part/#asterisk terrayi (~terrayi@203-167-189-215.dsl.clear.net.nz) |
| 12:45.53 | danfromuk | Might be time for me to start looking at the code. |
| 12:48.13 | danfromuk | Is there an announcement which can be played to the caller on join other than the position announcement? |
| 12:49.49 | [TK]D-Fender | danfromuk, I suspect this would be a relatively easy patch. Set flag to false on load. The moment it wants to play MoH it checks the flag and a new parameter for this and if false plays the announcement. Flag gets set true so normal MoH will kick in |
| 12:50.14 | [TK]D-Fender | danfromuk, No, there is nothing triggered "on join" specifically |
| 12:50.40 | danfromuk | Ok. perfect. Thanks |
| 12:55.30 | *** join/#asterisk serafie (~erin@nat/digium/x-dulxlauorrmwjumw) |
| 12:56.27 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 12:58.16 | *** join/#asterisk linocisco (~linocisco@61.4.77.10) |
| 12:58.59 | linocisco | on my cisco IP phone 7942G, missed call are blinking and how to clear them? |
| 13:00.03 | danfromuk | View the missed call log? Its not an asterisk issue. |
| 13:00.44 | linocisco | danfromuk, no. to clear missed calls |
| 13:01.13 | din3sh | u press "clear" :P |
| 13:01.28 | danfromuk | No idea. Again, this is not asterisk related. |
| 13:01.38 | danfromuk | Ask someone in a cisco channel. |
| 13:02.03 | danfromuk | The answers you get here will probably not help. |
| 13:02.19 | din3sh | press the directory button, then press the oft key clear below missed calls |
| 13:03.54 | din3sh | cisco forums do not offer any help if you've set up cisco to work with asterisk. they do not support 3rd party pbx :D |
| 13:04.17 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:04.17 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:04.19 | danfromuk | Thats true! |
| 13:04.23 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 13:04.44 | danfromuk | Shame. I like the physical style of the cisco phones. |
| 13:05.23 | [TK]D-Fender | linocisco, Go download your phone's manual already.... |
| 13:05.53 | linocisco | din3sh, there is no clear button for that |
| 13:06.02 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 13:06.06 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:07.35 | din3sh | i have a cisco 7945 in front of me, 7942 should be similar |
| 13:07.44 | [TK]D-Fender | http://tinyurl.com/cj4n876 <---- |
| 13:08.49 | din3sh | if you are using gareth's presence patch, use the proper softkey file |
| 13:09.04 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:10.14 | linocisco | din3sh, what is gareth's presence patch? |
| 13:10.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:10.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:10.41 | din3sh | nevermind :) |
| 13:20.39 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:29.40 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-imvqgrvgmvrbwjbn) |
| 13:29.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:30.10 | Katty | morning |
| 13:30.26 | din3sh | gd evening |
| 13:30.30 | PickledParrot | afternoon |
| 13:36.45 | *** join/#asterisk Xaviertoor (~Xaviertoo@189.52.87.66) |
| 13:39.58 | Xaviertoor | Anybody have a script that get "variable dialstatus". I have a application that make calls with php and I need dialstatus of calls! |
| 13:42.31 | Xaviertoor | Pastebin of my simple application http://pastebin.com/CRx6tYcY |
| 13:43.53 | [TK]D-Fender | Xaviertoor, That originates a call. That does not track anything though |
| 13:45.11 | [TK]D-Fender | Xaviertoor, It does not know what it creates and Dialstatus is created only after a dial that is not answered. You might as well deal with that in the dialplan right after the dial |
| 13:48.05 | Xaviertoor | It will be necessary to use MySQL or just for marking the dialplan solves the problem? |
| 13:48.50 | [TK]D-Fender | Xaviertoor, You showed a simple originate. There isn't even an entire idea in there. |
| 13:48.58 | [TK]D-Fender | Xaviertoor, It does not explain what it is doing. |
| 13:49.11 | [TK]D-Fender | Xaviertoor, Three is no probelm. We have no idae what you're DOING. |
| 13:51.34 | Xaviertoor | I need to originate calls on PHP and return Dialstatus if the call, was answered, missed or occupied. |
| 13:53.39 | Xaviertoor | With this app I can now make calls to two phone numbers, now I need to return the status of the connection. |
| 13:56.33 | [TK]D-Fender | Xaviertoor, make your dialplan send something back. forget about trying to directly grab a channel variable |
| 13:57.18 | [TK]D-Fender | Xaviertoor, AMI is not sitting around where you can just poll it immediately. You need to be able to identify the call that it created in the first place |
| 13:57.45 | [TK]D-Fender | So go make your dialplan communicate back. |
| 14:01.54 | *** join/#asterisk k611 (~K610@cred.epid.ucl.ac.be) |
| 14:02.29 | Xaviertoor | Ok |
| 14:03.08 | Xaviertoor | Look my dialplan exten => _XXXXXXXXXX,1,Dial(SIP/telecom/55${EXTEN},60,mtT) |
| 14:04.11 | [TK]D-Fender | Xaviertoor, make it contact your script on answer, or after a failed dial. |
| 14:05.51 | Xaviertoor | I do not know where to start |
| 14:06.13 | Xaviertoor | I will study |
| 14:14.47 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 14:17.46 | igcewieling | Xaviertoor: exten => _XXXXXXXXXX,n,Noop(DIALSTATUS is ${DIALSTATUS}) |
| 14:18.06 | [TK]D-Fender | igcewieling, He's poriginating a call via AMI and wants to track progess |
| 14:18.13 | [TK]D-Fender | igcewieling, its WAY more than just a boring call |
| 14:18.28 | igcewieling | [TK]D-Fender: Ah, the poor sod. Trying to do brain surgery before he can stitch up a wound. |
| 14:18.31 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:20.06 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:20.58 | *** join/#asterisk Gaiax (bef280c5@gateway/web/freenode/ip.190.242.128.197) |
| 14:27.00 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:28.12 | *** join/#asterisk navaismo (~navaismo@189.191.250.167) |
| 14:30.26 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 14:37.32 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:43.22 | *** join/#asterisk Gaiax (bef280c5@gateway/web/freenode/ip.190.242.128.197) |
| 14:47.16 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:47.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:54.25 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 15:00.44 | *** join/#asterisk cchhat01 (~cchhat01@207-237-28-173.c3-0.elm-ubr2.qens-elm.ny.cable.rcn.com) |
| 15:08.54 | *** join/#asterisk butthurtface (~Butthurtf@38.122.108.2) |
| 15:11.34 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:48.51 | *** join/#asterisk navaismo (~navaismo@189.191.250.167) |
| 15:49.21 | Kobaz | has anyone seen any problems with long-running AMI connections where you're still connected and can send commands, but you stop receiving events? |
| 15:50.26 | igcewieling | Kobaz: I've heard reports of that (maybe from you) but I've not experienced it. We used to have a long running AMI connection to monitor CEL events and did not have that problem that I'm aware of. |
| 15:50.33 | Kobaz | probably me |
| 15:51.26 | igcewieling | We DID experience the connection dropping occasionally and until we updated our script to handle that error condition bad things happened. |
| 15:51.43 | Kobaz | which error condition? |
| 15:51.51 | igcewieling | Kobaz: the connection dropping. |
| 15:51.54 | igcewieling | you want the code? |
| 15:52.08 | Kobaz | oh |
| 15:52.15 | Kobaz | as in like your tcp connection goes byebye? |
| 15:52.48 | igcewieling | http://pastebin.ca/2364578 |
| 15:52.50 | *** part/#asterisk Matthias (~Matthias@195.16.243.99) |
| 15:52.54 | *** join/#asterisk Matthias (~Matthias@195.16.243.99) |
| 15:52.58 | Kobaz | i handle tcp reconnection |
| 15:53.19 | Kobaz | and i handle when ping/pong stop working, then i reconnect |
| 15:53.39 | Kobaz | what is difficult to handle is when you are connected, ping/pong works, you even send a userevent and get it back... but you don't get anything else |
| 15:53.51 | Kobaz | so the script thinks everything is fine and dandy |
| 15:53.57 | Kobaz | but you're missing thousands of events |
| 15:54.00 | igcewieling | disconnect and reconnect when you don't get a response. |
| 15:54.08 | Kobaz | i do |
| 15:54.12 | igcewieling | Kobaz: we only care about CEL events and didn't send any commands |
| 15:54.25 | Kobaz | so the problem is your manager session becomes isolated |
| 15:54.33 | Kobaz | what i think i need to do |
| 15:54.34 | igcewieling | this was before we logged all CEL to a database. |
| 15:54.40 | Kobaz | is have another script sending userevents |
| 15:55.05 | Kobaz | and have my event manager get the ping response back |
| 15:55.27 | Kobaz | so if you send a 'ping' from the outside and don't get it, then you force a reconnect |
| 15:56.22 | Kobaz | i feel like as time goes on |
| 15:56.41 | Kobaz | i have to write more and more code outside of asterisk, since I don't trust asterisk itself |
| 15:56.47 | Kobaz | pretty soon i'll have my own asterisk |
| 15:57.25 | Kobaz | for instance, i no longer trust core restart when convenient, since it tends to deadlock, and then you have a broken asterisk waiting to restart |
| 15:57.36 | Kobaz | so i wrote my own as an external script |
| 15:58.07 | igcewieling | you have to exit your asterisk session and reconnect, you'll then see CLI events again while the restart is pending. |
| 15:58.28 | Kobaz | the events and the restart are two separate issues |
| 15:58.46 | igcewieling | Kobaz: I was referring to CLI output, not ami events |
| 15:58.53 | Kobaz | oh |
| 15:58.54 | Kobaz | yeah |
| 15:58.55 | Kobaz | no i know that |
| 15:58.59 | Kobaz | it pauses your cli session |
| 15:59.04 | igcewieling | annyoing, stupid, and silly, but that is the way Asterisk seems to work |
| 15:59.06 | Kobaz | but it also deadlocks asterisk core in certain cases |
| 15:59.23 | igcewieling | I've not seen than in years, but we don't do a lot of restarts |
| 15:59.39 | *** join/#asterisk jblack (~jblack@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 15:59.42 | Kobaz | well i know why it pauses the cli.. it's becuase it uses your own cli thread to do the mutex waits on channel counts |
| 16:00.03 | Kobaz | igcewieling: i restart nightly... i was seeing it once every two weeks |
| 16:00.23 | *** part/#asterisk Matthias (~Matthias@195.16.243.99) |
| 16:00.24 | igcewieling | Any time I do a restart gracefully the big boss freaks out because the graphs look different 8-) |
| 16:00.37 | Kobaz | and i never tracked down the bug since it's one of those things that just must work, so i did a workaround and 'fixed' the problem |
| 16:00.41 | jmetro | i like "when convenient" |
| 16:00.53 | Kobaz | i like it to.. when it doesn't deadlock :/ |
| 16:01.14 | Kobaz | *too |
| 16:01.16 | igcewieling | jmetro: I don't. "when convenient" for us means "sometime after 11pm" because of call volume. |
| 16:01.29 | *** join/#asterisk Matthias (~Matthias@195.16.243.99) |
| 16:01.44 | igcewieling | gracefully will reject calls and they will failover to one of the other servers in our "cluster" |
| 16:01.58 | jmetro | ah clustered |
| 16:02.35 | Kobaz | i started retarting nightly before i had fixes in for the sip+replaces reference leaks |
| 16:02.38 | igcewieling | jmetro: I use the term loosely |
| 16:02.44 | Kobaz | so my asterisk was using 2 gigs of ram after 4 days of uptime |
| 16:02.59 | Kobaz | and then the oom killer would kick in at some point at the worst time imaginable |
| 16:03.11 | igcewieling | When we used Digium cards we restarted weekly, since switching to Sangoma about 5 years ago we've not had to do that. |
| 16:03.19 | Kobaz | and now... restarting nightly just works so well i've kept it in |
| 16:03.36 | Kobaz | dont have to worry about any memory leaks going unchecked and wrecking the machine |
| 16:05.20 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 16:06.00 | Kobaz | this is the kobaz-has-been-having-a-terrible-week-of-problems-and-must-rant lunch hour |
| 16:06.37 | Kobaz | currently trying to diagnose a problem where people's phones ring but they can't pick up the call |
| 16:09.35 | leifmadsen | Kobaz: oh... I feel like I've run into that a long time ago |
| 16:09.41 | leifmadsen | I forget what I changed :\ |
| 16:10.47 | Kobaz | leifmadsen: i've seen this problem only once before |
| 16:10.53 | Kobaz | and it was after a new cisco router got put in |
| 16:11.10 | igcewieling | Kobaz: we had a bad MONTH recently. 8-( |
| 16:11.12 | Kobaz | with the router in, no one could make any phone calls |
| 16:11.25 | Kobaz | and the phone calls were lan-only |
| 16:11.40 | Kobaz | like polycom to polycom just didn't work |
| 16:11.43 | igcewieling | Kobaz: let me guess "no sip fixup" fixed it. |
| 16:11.44 | Kobaz | take the router out, and everything worked fine |
| 16:12.08 | Kobaz | putting in the old router is what fixed it.. i don't know heh... i wasn't involved in the configuring of it |
| 16:12.18 | Kobaz | i just said take this pos out of here, it's wrecking the phone system |
| 16:12.43 | Kobaz | why would a router affect local to local calls that are not using the default gateway? |
| 16:13.55 | igcewieling | Kobaz: a real Cisco IOS router or a PoS fake Cisco aka "Linksys" |
| 16:14.13 | igcewieling | Kobaz: maybe DHCP server was enabled on the router? |
| 16:14.16 | Kobaz | it was a ios router... an asa5505 |
| 16:14.24 | eirirs | asa are not ios router |
| 16:14.36 | Kobaz | it runs ios |
| 16:14.39 | Kobaz | a stripped down version |
| 16:14.53 | jmetro | like junipers? |
| 16:14.56 | igcewieling | We used to have a problem with customers putting rogue DHCP servers on their LANs |
| 16:16.30 | Kobaz | pretty sure dhcp was off |
| 16:16.39 | Kobaz | so here was the problem with the cisco |
| 16:16.50 | Kobaz | start up asterisk, everything was fine for 2-3 minutes |
| 16:16.54 | Kobaz | then every single phone went unreachable |
| 16:17.07 | Kobaz | 30 seconds later they came back... and then every 2-3 minutes that cycle repeats |
| 16:17.39 | Kobaz | while the phones were reachable, if you made a phone call to another phone: asterisk would direct the call to another polycom just fine, it would ring the polycom |
| 16:17.59 | Kobaz | but then when you go to answer the call, the call would be stuck ringing, asterisk would never get the response to the invite from the polycom |
| 16:19.59 | jmetro | ciscos have a lot of funny options that are in a default mode that makes 0 sense |
| 16:20.55 | Kobaz | it's not nearly that bad with this current problem |
| 16:21.08 | Kobaz | it's just intermittent, and the phones don't go unreachable |
| 16:21.30 | igcewieling | sounds like the router had the same IP as the asterisk server. i.e. when the ARP entry gets refreshed things break |
| 16:22.12 | igcewieling | jmetro: like ip subnet-zero defaulting to off for so long |
| 16:22.43 | jmetro | igcewieling: like broadcasting SSID's being off by default and having it be called "Guest mode" |
| 16:22.57 | igcewieling | jmetro: ah, never used their wireless stuff |
| 16:23.04 | jmetro | igcewieling: be happy. |
| 16:23.14 | igcewieling | jmetro: wireless is evil. |
| 16:23.38 | jmetro | igcewieling: Evil Unnecessary Complications + Evil Random Failures = cisco wireless |
| 16:24.06 | igcewieling | I FINALLY installed WiFi at my apartment but only because it would be a huge hassle to get my android phone internet access via non-3G without it. |
| 16:24.15 | Kobaz | igcewieling: oh. you know... could have been |
| 16:24.26 | igcewieling | network cable along the base of the walls may not be PRETTY, but it is reliable and it is fast. |
| 16:24.35 | Kobaz | i like cable |
| 16:24.42 | Kobaz | i wired my whole house for gigabit |
| 16:24.47 | Kobaz | two jacks in every room |
| 16:24.48 | jmetro | Kobaz: +1 |
| 16:24.50 | Kobaz | one for data, one for poe |
| 16:24.58 | Kobaz | four jacks behind the tv |
| 16:25.09 | jmetro | <PROTECTED> |
| 16:25.14 | Kobaz | i know |
| 16:25.21 | igcewieling | I like to move too much to buy a house |
| 16:25.24 | Kobaz | right now i'm running in-the-wall wiring for surround sound |
| 16:25.43 | Kobaz | i had the fun of running wire through a wall that had blown-insulation |
| 16:25.46 | Kobaz | that was lots of fun |
| 16:26.04 | igcewieling | Kobaz: I've done that. Should be band under international treaty as torture. |
| 16:26.14 | igcewieling | banned too |
| 16:26.30 | igcewieling | try doing that with thicknet. |
| 16:29.22 | Kobaz | k |
| 16:29.23 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 16:29.28 | Kobaz | off to go look at a riding mower |
| 16:32.35 | *** join/#asterisk FonJockey (~FonJockey@70-10-5-180.pools.spcsdns.net) |
| 16:32.49 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 16:34.34 | carrar | Get a DEER |
| 16:35.40 | drmessano | Don't get a Bosch |
| 16:44.14 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:45.18 | igcewieling | Get a lawn care company. |
| 16:47.23 | drmessano | ~boschlawnmowers |
| 16:47.23 | infobot | http://www.voip-info.org/wiki/view/Bosch+lawnmowers .. "Same to you, buddy!" |
| 16:48.07 | drmessano | Long gone.. sadly |
| 16:50.57 | jmetro | ~whatareboshlawnmowers |
| 16:51.00 | jmetro | ? |
| 16:51.13 | jmetro | wait why is that even a bot macro |
| 16:52.46 | drmessano | Because a bot had posted some spam to the wiki.. and someone commented on it "Go f*** yourself". The original ad was removed, but the comment remained.. for 5 years |
| 16:54.27 | *** join/#asterisk italorossi (~italoross@201.76.144.193.digi.com.br) |
| 16:58.29 | jmetro | Nice XD |
| 17:01.38 | igcewieling | OT: We are looking for a MySQL guru to move our replicate MySQL "cluster" to a real MySQL NBD cluster. |
| 17:07.22 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 17:19.32 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 17:19.49 | EmleyMoor | tzafrir about? |
| 17:20.22 | tzafrir | yup |
| 17:20.31 | *** join/#asterisk jsjc (~Adium@23.Red-83-42-71.dynamicIP.rima-tde.net) |
| 17:22.08 | EmleyMoor | tzafrir: What's the story behind the backout of fix_xmpp_19532 in Debian? |
| 17:22.52 | *** join/#asterisk Xaviertoor (~Xaviertoo@189.52.87.66) |
| 17:24.07 | tzafrir | Sadly the ftp-masters would not accept any major changes besides the security fix |
| 17:24.36 | tzafrir | Which also meant dropping a trivial patch to enable an extra minor architecture |
| 17:24.48 | EmleyMoor | The ftp-masters of what exactly? |
| 17:27.18 | EmleyMoor | (are they a Debian thing?) |
| 17:32.42 | EmleyMoor | Anyway, I've put it back in locally now - was just looking for the story behind it - was the freeze of wheezy relevant to the backout? |
| 17:33.39 | cusco | tzafrir: in wheezy? |
| 17:34.19 | tzafrir | Yes. They said that it could be re-introduced in a stable update (for 7.0.1) |
| 17:35.02 | EmleyMoor | Ah, right... |
| 17:35.16 | tzafrir | ftp-masters are the people in charge of the archive - the packagesin each release. |
| 17:35.40 | tzafrir | Right now Wheezy is frozen - nothing goes automatically from Unstable to Testing |
| 17:42.59 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:43.00 | *** join/#asterisk jsjc (~Adium@23.Red-83-42-71.dynamicIP.rima-tde.net) |
| 17:44.01 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 17:44.16 | lauris | hi |
| 17:45.40 | lauris | is it possible to execute a dialplan after call is put into Queue() but before dialing to SIP/* interface? |
| 17:47.46 | navaismo | hmm not sure before but try the membermacro/gosub |
| 17:47.58 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:48.04 | igcewieling | lauris: all dialplan apps "block" |
| 17:48.09 | lauris | these are executed after member answers |
| 17:48.21 | lauris | both membermacro and agi |
| 17:48.25 | lauris | as well as gosub |
| 17:49.05 | jmetro | Lauris there is the M option in queue |
| 17:49.32 | jmetro | which executes a macro to the dialed party |
| 17:52.15 | lauris | can't find M option in core show application Queue |
| 17:54.42 | *** join/#asterisk jsjc (~Adium@23.Red-83-42-71.dynamicIP.rima-tde.net) |
| 17:55.43 | [TK]D-Fender | lauris, what do you want to do there? |
| 17:56.29 | lauris | I want to be able set auto-answer feature on agent basis |
| 17:57.11 | cusco | hu? |
| 17:57.18 | [TK]D-Fender | lauris, Then use Local channels as members instead |
| 17:57.34 | lauris | I know this way, but it produces lot's of log output as well as ringnoanswers |
| 17:57.50 | lauris | because when you send queued call to Local/* you don't now peer's state |
| 17:57.57 | lauris | is there a solution for this problem? |
| 17:58.06 | cusco | what's wrong with ringnoanswer? |
| 17:58.10 | lauris | can I somehow replicate SIP/100 state to Local/100 ? |
| 17:58.21 | [TK]D-Fender | <lauris> because when you send queued call to Local/* you don't now peer's state <- you do if you set the State Interface like you're supposed to |
| 17:58.25 | lauris | assume you have 20 agents associated with a queue |
| 17:58.31 | lauris | 15 of them are on the line |
| 17:58.36 | lauris | 5 are notinuse |
| 17:58.57 | cusco | in the queue members in our realtime db, there is a field 'state_interface' |
| 17:58.59 | lauris | when Local is used, then all of 20 dialed in a row |
| 17:59.00 | [TK]D-Fender | lauris, STATE DEVICE <------ |
| 17:59.03 | cusco | that w set to SIP/whatever |
| 17:59.25 | lauris | so I should set state_interface to SIP/... ? |
| 17:59.33 | [TK]D-Fender | lauris, clearly |
| 17:59.36 | lauris | cool!!! |
| 17:59.53 | leifmadsen | +1 |
| 18:00.01 | [TK]D-Fender | lauris, * has a few nifty ways around things. |
| 18:00.02 | lauris | so using Local wasn't that bad idea :) |
| 18:00.22 | [TK]D-Fender | lauris, Local is a very necessary evil. |
| 18:01.13 | lauris | going to check if it works ;-) |
| 18:02.06 | lauris | should I write there SIP/... |
| 18:02.09 | lauris | or SIP/.../n ? |
| 18:02.53 | cusco | SIP/peer |
| 18:03.57 | [TK]D-Fender | lauris, "/n" is a Local thing to prvent a bridging of the extent channels. This is where you want the dialplan in the middle to continue afterwards, etc. Which for me is almost always. |
| 18:04.37 | cusco | also what we do, we use leastrecent instead of ringall, and in Local dialplan dial only for 3 seconds, so people must use their phone's auto-answer |
| 18:05.22 | lauris | cool. it's working |
| 18:05.35 | lauris | <PROTECTED> |
| 18:06.02 | navaismo | I have this weird behavior of one digium phone not responding with the INVITE challenge, and the call goes to limbo for some seconds then the phone get the busy tones. SIP debug here--->http://pastebin.com/FrptD3S8 |
| 18:06.13 | cusco | [TK]D-Fender: where can I read more about the /n |
| 18:06.48 | lauris | how would you recommend importing queue log into database? |
| 18:07.03 | lauris | afaik there are two ways: odbc or importing queue_log with a script |
| 18:08.45 | cusco | I use mysql addon |
| 18:09.08 | lauris | res_mysql ? |
| 18:09.13 | cusco | that |
| 18:09.34 | leifmadsen | always use odbc |
| 18:09.47 | leifmadsen | the mysql stuff is old and outdated, and not maintained |
| 18:10.06 | cusco | I know |
| 18:10.23 | lauris | to me odbc looks a little bit old fashioned way of doing things ;) |
| 18:10.26 | lauris | isn't it ? |
| 18:10.49 | cusco | what is that /n that you folks where refering to? is that in the member => SIP/peer/n ? |
| 18:11.18 | navaismo | is that a deja vĂș with different persons involved o god the matrix jump again |
| 18:11.26 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:11.49 | navaismo | cusco, http://www.voip-info.org/wiki/view/Asterisk+local+channels |
| 18:13.18 | cusco | thanks navaismo |
| 18:13.33 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:16.32 | *** join/#asterisk jsjc (~Adium@23.Red-83-42-71.dynamicIP.rima-tde.net) |
| 18:20.39 | Katty | hello my asterisk does not work at all how to fix?? srs answers only pls is urgent thx. |
| 18:20.56 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:24.07 | igcewieling | Katty: feed it stewed squirrel. |
| 18:24.48 | Katty | WHAT?! |
| 18:24.57 | Katty | fetches thor's hammer, waves it at igcewieling |
| 18:25.15 | cusco | Katty: you can't lift it up |
| 18:25.16 | cusco | :P |
| 18:25.28 | Katty | i can if it's from Target ^_^ |
| 18:25.30 | cusco | why do you say it doesn't work? |
| 18:25.47 | igcewieling | a troll for a troll |
| 18:25.47 | Katty | thor's hammer works nicely, i hear. |
| 18:25.51 | Katty | let's test it on igcewieling |
| 18:26.05 | cusco | from target? |
| 18:26.10 | igcewieling | grabs his sourdough shield of might |
| 18:26.10 | cusco | nobody can lift it upo but thor |
| 18:26.12 | Katty | it's a department store. |
| 18:26.17 | cusco | ow |
| 18:26.18 | Katty | with a Children's toy aisle. |
| 18:26.24 | cusco | right |
| 18:26.31 | Katty | igcewieling: well that's just not fair :< |
| 18:26.40 | Katty | igcewieling: i can't destroy something that took THAT much effort |
| 18:26.51 | igcewieling | indeed |
| 18:26.54 | Katty | puts down hammer and volunteers pizza skills. |
| 18:27.08 | igcewieling | though it isn't really much effort |
| 18:30.56 | jmetro | i think sourdough pizza might be nasty. |
| 18:31.27 | igcewieling | jmetro: it isn't great, but isn't terrible either, don't don't usually let it rise long enough to get very sour. |
| 18:31.50 | igcewieling | one of my very few hobbies is baking |
| 18:32.48 | jmetro | I like cooking in general. I like making sauces. |
| 18:33.16 | jmetro | i make a pretty great lemon garlic for chicken or seafood |
| 18:33.34 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 18:33.47 | igcewieling | baking (other than bred-like stuff) is more scientific than savory cooking and I like that |
| 18:35.36 | jmetro | it is. |
| 18:39.10 | dlynes | Katty, how'd you break your asterisk? |
| 18:39.30 | dlynes | Katty, It went nutty from watching too many live squirrel feeds? :o |
| 18:46.37 | jmetro | katty: I put the sauce on the asterisk but its not cheesy enough? whare to get more chees for phones |
| 18:47.33 | igcewieling | jmetro: try grandstream phones, they are plenty cheeezy. |
| 18:48.35 | jmetro | igcewieling: i had grandstream but tried to dial in for more pizza and the dtmf broke |
| 18:49.24 | igcewieling | jmetro: they are working as designed. |
| 18:50.19 | *** join/#asterisk Jouva (~jouva@pool-96-245-167-3.phlapa.fios.verizon.net) |
| 18:50.39 | jmetro | igcewieling: thats like blizzards "Working as intended" |
| 18:54.33 | Jouva | So I'm still currently on 1.8 but I have a security related question: I've been seeing people attempting to make calls to various numbers, obviously trying to make free outgoing calls from the system. Obviously I've at least been smart enough to force calls having to go through a context only authenticated SIP devices can reach. But how can I prevent these other systems from contacting via |
| 18:54.33 | Jouva | SIP? |
| 18:54.54 | Jouva | The only one that should be contacting the system via SIP is the DID provider |
| 18:55.03 | Jouva | Or an actual phone on our system |
| 18:55.42 | jmetro | well obviously only inbound calls are coming from the SIP provider so only his calls go into the correct context |
| 18:55.46 | jmetro | the rest are just dumping into your dead-end |
| 18:56.53 | igcewieling | Jouva: if the attacker is getting far enough into your system to make outgoing calls then you have a design problem in your asterisk config |
| 18:57.15 | Jouva | Oh no like I said they're ATTEMPTING to but not able to |
| 18:57.44 | jmetro | I see things like that all the time, full length international numbers hit the console and get dead-ended |
| 18:57.51 | Jouva | I just want to stop seeing them even get to a point where it says that their call was rejected cause the extension wasn't found |
| 18:57.53 | igcewieling | Jouva: Ah. Yes, that happens all the time. If they are attempting to register, then you can use fail2ban to block them. |
| 18:58.38 | igcewieling | if not, you may need to enable sip debug or hack asterisk to show the IP of the attacker for non-registration attempts |
| 18:58.58 | igcewieling | though as far as I'm concerned if they are getting to the dialplan you have a security issue. |
| 18:59.20 | Jouva | [Apr 23 13:55:36] NOTICE[26135]: chan_sip.c:22622 handle_request_invite: Call from '' (94.23.59.135:5071) to extension '011972595637212' rejected because extension not found in context 'default'. |
| 18:59.26 | igcewieling | they should be blocked by your sip config before the calls get to the dialplan, unless you are doing something like federated sip setup. |
| 18:59.27 | Jouva | I'm just getting that, or similar |
| 18:59.44 | igcewieling | Jouva: They should not be getting that far. |
| 18:59.45 | Jouva | Right. That's why I'm here. What might I be doing wrong? |
| 19:00.13 | igcewieling | Jouva: do you have allowguest=no and always authreject=yes |
| 19:00.34 | igcewieling | though with THAT message, you can use fail2ban to block them |
| 19:00.35 | Jouva | I think allowguest is yes but I forget why if I do, let me check |
| 19:01.08 | Jouva | this is all sip.conf right? |
| 19:02.05 | igcewieling | correct, in [general] |
| 19:03.11 | Jouva | Ok. Neither of those are explicitly set |
| 19:03.36 | igcewieling | Jouva: what we do is simply block all access from the internet (except for a few IPs) using iptables, though that is not workable for most people. |
| 19:03.53 | Jouva | Yeah that won't work for me |
| 19:04.23 | Jouva | Of interest but might not be at all related (and I might not fully understand them) I have: canreinvite=no limitonpeer=yes type=friend insecure=invite,port |
| 19:09.09 | igcewieling | Jouva: you should read the Asterisk Book. |
| 19:09.10 | igcewieling | ~book |
| 19:09.10 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:10.36 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 19:11.49 | Jouva | Is there a way to get that free other than that website? |
| 19:12.11 | ageis | is there any way to randomize extension numbers? |
| 19:12.14 | Jouva | Like a .pdf or kindle reader app compatible version? |
| 19:12.43 | jmetro | i believe the epub is available on there |
| 19:13.14 | jmetro | ageis random extensions sound like a good way to forget everything. |
| 19:14.05 | igcewieling | and they provide NO extra security. |
| 19:14.06 | *** join/#asterisk jsjc (~Adium@41.Red-79-146-120.dynamicIP.rima-tde.net) |
| 19:14.18 | igcewieling | maybe you are calling the sip.conf peers "extensions". Classic n00b mistake. |
| 19:15.05 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:15.39 | jmetro | sip peernames are easist to just use Mac-Extension-Department |
| 19:15.42 | jmetro | or some combination thereof |
| 19:16.06 | jmetro | like 0f0f0f0f0f0f-100-Sales |
| 19:16.40 | igcewieling | I like just using the mac |
| 19:17.22 | *** join/#asterisk derjanni (~derjanni@ip-178-202-27-28.unitymediagroup.de) |
| 19:17.31 | derjanni | http://www.kammerath.net/click-to-call.html |
| 19:17.36 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:17.38 | derjanni | ^ how to do click to call with PHP and asterisk |
| 19:17.50 | derjanni | here's the source code: https://code.google.com/p/click-to-call-webapp/ |
| 19:18.02 | jmetro | Mac alone can get confusing , and can be a terrible mistake if you get into softphones / duplicates / etc.. |
| 19:18.16 | jmetro | like when 3com duplicated macs. |
| 19:18.53 | igcewieling | jmetro: heh, softphones are evil and now allowed |
| 19:19.08 | igcewieling | s/now/not |
| 19:19.20 | cusco | why not |
| 19:20.04 | igcewieling | because all softphones suck |
| 19:20.30 | jmetro | the 3cx softphones arent bad. |
| 19:20.35 | jmetro | Xlite is terrible nonsense. |
| 19:21.40 | Jouva | softphones are only good if you REALLY insist on access to the phone system on the go, or basic testing and diagnosing to take the hardware out of the equation |
| 19:21.48 | derjanni | what r ur experiences with softphones on wifi? |
| 19:21.51 | Jouva | Or if you're just that cheap |
| 19:21.56 | derjanni | I tried CSipSimple on my RAZR and it was horrible |
| 19:25.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:30.24 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:31.05 | *** part/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 19:31.59 | *** join/#asterisk italorossi (~italoross@201.76.144.193.digi.com.br) |
| 19:32.37 | *** join/#asterisk gtTuna (Tuna@d149-67-253-75.try.wideopenwest.com) |
| 19:32.55 | gtTuna | ugh...has anyone tried to do SIP trunking to a LG IPECs IP-PBX before? |
| 19:35.42 | *** join/#asterisk jsjc (~Adium@41.Red-79-146-120.dynamicIP.rima-tde.net) |
| 19:35.48 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:36.29 | *** join/#asterisk FonJockey (~FonJockey@108.107.236.239) |
| 19:38.17 | ageis | SIPdroid is good, prefer Jitsi |
| 19:47.40 | *** join/#asterisk FonJockey (~FonJockey@108.107.236.239) |
| 19:49.03 | igcewieling | Does anyone else announce an update/upgrade to customers and deliberately not do it in order to find out who the complainers are before scheduling the actual upgrade? |
| 19:49.55 | CheeseMouse | igcewieling, don't you think they'll eventually catch on? |
| 19:50.06 | CheeseMouse | cf. "boy who cried wolf" |
| 19:50.20 | igcewieling | CheeseMouse: we don't do announced upgrades often enough for that to happen. |
| 19:50.39 | CheeseMouse | In that case, I like the idea. |
| 19:51.42 | igcewieling | We have at least 200 customer left to migrate and based on the first 2 batches of 25 customers, they will bitch and scream about stuff totally unrelated to the changes which fixes to be delayed because of the "noise" reports. |
| 19:51.59 | igcewieling | s/fixes/causes fixes/ |
| 20:10.10 | *** join/#asterisk gtTuna (Tuna@d149-67-253-75.try.wideopenwest.com) |
| 20:18.28 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 20:26.21 | *** join/#asterisk FonJockey (~FonJockey@107.31.21.237) |
| 20:36.53 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 20:39.57 | *** join/#asterisk navaismo (~navaismo@189.191.250.167) |
| 21:01.10 | *** join/#asterisk din3sh (29885740@gateway/web/freenode/ip.41.136.87.64) |
| 21:02.42 | din3sh | hello all |
| 21:03.26 | din3sh | I have sip peers with disallow=all allow=alaw,g729 |
| 21:03.54 | din3sh | am still getting "Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write" on calls from PSTN (dahdi) |
| 21:06.35 | igcewieling | din3sh: do you have a g729 license? |
| 21:06.51 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:09.59 | din3sh | yes |
| 21:14.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:19.08 | igcewieling | din3sh: do a "core show translation" and make sure g729 does not have a - in it. |
| 21:19.38 | igcewieling | sln is an intermediate transcoding format |
| 21:24.12 | din3sh | can this indicate a random one-way audio problem? |
| 21:25.20 | din3sh | btw, what if i have MOH in .gsm format in an incoming queue, then transfered to a sip peer in that queue? |
| 21:29.54 | *** join/#asterisk bananapie (~david@76-10-164-164.dsl.teksavvy.com) |
| 21:30.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:30.16 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:30.18 | bananapie | Is there an option I can pass to ReceiveFax() or a variable I can set to tell it not to negotiate t.38 for a particular call? |
| 21:30.59 | bananapie | wait, nevermind |
| 21:37.36 | *** join/#asterisk FonJockey (~FonJockey@107.31.21.237) |
| 21:39.56 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 21:47.01 | *** join/#asterisk andy09usa (~Andrey@unaffiliated/andy09usa) |
| 21:53.27 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 21:54.17 | *** join/#asterisk wolfmitchell (~wolfmitch@botters/wolfmitchell) |
| 21:54.42 | *** join/#asterisk pbxbrian (~pbxbrian@unaffiliated/brian98) |
| 22:01.39 | *** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net) |
| 22:01.59 | x1user | How can I login/logout sip account with some code? |
| 22:02.56 | sp00kz | compile it ^^ |
| 22:11.08 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 22:24.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:36.27 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:36.27 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:37.10 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-imvqgrvgmvrbwjbn) |
| 22:54.01 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-231-146.ph.ph.cox.net) |
| 22:54.09 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:54.09 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:57.54 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 23:10.34 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 23:16.56 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 23:24.33 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 23:24.33 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:34.22 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:34.22 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 23:39.40 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 23:57.00 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |