| 00:00.57 | artyx | Hey, stupid question time... I've set up permission restrictions to inhibit/deny external (sip from a non 10.xx.yy.zz/24 ip) , but I can still see attempts made to dial some bad numbers (011...) They (teh caller) is getting an error tone, Is this normal behavior, or does this mean they are subveerting extension masks |
| 00:01.00 | WIMPy | I'm never sure if all of them are r/w there are some obvious duplicates. Maybe they are just aliases, even. |
| 00:01.07 | Kobaz | yay that worked |
| 00:01.14 | Kobaz | RDNIS="" |
| 00:01.19 | Kobaz | no more diversion header |
| 00:01.45 | WIMPy | I haven't tried to work that out. That's certainly a lack of documentation/clarification. |
| 00:05.11 | ghost75 | kikohnl: where is extension/pass in the file? |
| 00:09.09 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 00:09.47 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:09.48 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:11.18 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 00:15.05 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 00:15.41 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:91b8:7e52:91aa:f22) |
| 00:19.38 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:91b8:7e52:91aa:f22) |
| 00:23.31 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:91b8:7e52:91aa:f22) |
| 00:24.40 | kikohnl | ghost75: look for 6300 that is my extension, the reset of the config is sip.conf and extensions.conf |
| 00:27.44 | ghost75 | is there no password for 6300 somewhere? |
| 00:28.31 | ghost75 | or is line1 the main one? |
| 00:43.52 | kikohnl | Line1 is the top right button on the physical phone |
| 00:44.02 | ghost75 | yes i know |
| 00:44.26 | ghost75 | just looking for password of extension |
| 00:45.03 | kikohnl | <authPassword>1234</authPassword> |
| 00:45.18 | kikohnl | the 1234 should match what is in sip.conf |
| 00:46.47 | ghost75 | so i have to put line1 button always like the extension of the phone |
| 00:49.03 | kikohnl | under <sipLines> you put a section for each "Line appearance" you want. |
| 00:49.22 | ghost75 | that was not my question :) |
| 00:51.20 | kikohnl | you have to associate an extension with the "Line" buttons |
| 00:51.38 | ghost75 | yes that was it |
| 00:53.13 | ghost75 | the extension of the phone has to be a line button |
| 00:53.14 | ruben231 | hi guys can i block particular nummber on asterisk..? coming from PSTN or mobile line from calling my asterisk |
| 00:56.05 | tzanger | ruben231: just put the CID match in |
| 00:56.37 | tzanger | exten => 9995551212/5558675309,1,Goto(vm,1) |
| 00:56.55 | tzanger | as an example to prevent 5558675309 from ringing your 9995551212 phone nmber, it'll just go to voicemail |
| 00:57.05 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 01:00.27 | kikohnl | ghost75: what version of firmware did you end up with ? |
| 01:00.39 | ghost75 | same as you have |
| 01:02.22 | kikohnl | There is an exploit for this version, Cisco has a "pre-release" version available and a real version out soon, |
| 01:03.28 | kikohnl | there is a but in their SSH that allows specific data packets to open the phone up to anyone, but it seems you have to have real access to the phone for this to happen |
| 01:04.07 | ghost75 | what is ssh used for? |
| 01:04.17 | kikohnl | http://www.cisco.com/en/US/customer/products/csa/cisco-sa-20130109-uipphone.html |
| 01:04.45 | ghost75 | the forward button also not works, somebody else said |
| 01:04.52 | kikohnl | you can do lots of things, reset the phone, view logs, view debug info |
| 01:05.44 | ketas | mmm hm, where's "proper" caller id i can use... |
| 01:06.00 | ghost75 | need cisco account to open the link |
| 01:08.14 | kikohnl | the phone also has a web interface |
| 01:08.26 | ghost75 | i saw yes |
| 01:08.30 | ghost75 | but readonly i think |
| 01:08.56 | kikohnl | yes, the web interface is read only |
| 01:10.34 | kikohnl | I saved the web page from Cisco as a pdf, and tried to send it to you via IRC, |
| 01:11.16 | ghost75 | thx, dont know if my client supports |
| 01:11.43 | kikohnl | yep, the send just timed out |
| 01:12.44 | *** join/#asterisk jsjc (~Adium@33.Red-2-136-105.dynamicIP.rima-tde.net) |
| 01:13.36 | ghost75 | i found i have cisco account |
| 01:14.10 | kikohnl | Cool |
| 01:16.21 | *** join/#asterisk mattsl (~user@c-50-147-66-240.hsd1.tn.comcast.net) |
| 01:17.53 | ghost75 | The file or application you are trying to access may require additional entitlement |
| 01:17.55 | ghost75 | loooooooooool |
| 01:18.03 | kikohnl | You will also need to change the timezone in the SEP*.cnf file |
| 01:18.29 | kikohnl | I use to work for a Cisco Partner, and my access is still partially working |
| 01:19.22 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 01:20.51 | ghost75 | https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 01:22.52 | kikohnl | I"ve not seen that before :-) |
| 01:23.09 | ghost75 | have to read first what that actually is for |
| 01:33.20 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.164) |
| 01:46.51 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 01:49.14 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 01:58.17 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.25.167) |
| 02:05.44 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 02:12.29 | *** join/#asterisk mail323 (~mail323@c-98-254-82-94.hsd1.fl.comcast.net) |
| 02:21.25 | mail323 | What is the simplest method to send calls between different asterisk systems using IP address (no password)? |
| 02:23.46 | ketas | iax? |
| 02:26.30 | mail323 | IAX seems even more of a pain than SIP |
| 02:28.45 | mail323 | Right now with SIP I get a "couldn't call exten@host" error and no INVITE is issued |
| 02:29.39 | ketas | hm, can't help much, i only have one asterisk for now |
| 02:29.51 | lvlinux | i think the simplest is IAX2---config is pretty much same as SIP but only one port used between the machines |
| 02:30.44 | lvlinux | just put a host context in the iax.conf file on each server and you should be done |
| 02:31.35 | Penguin | http://pastebin.com/Ag7tknm2 |
| 02:31.44 | Penguin | http://pastebin.com/UUxJtERA] |
| 02:31.47 | Penguin | http://pastebin.com/UUxJtERA |
| 02:32.04 | lvlinux | exactly |
| 02:32.20 | mail323 | lvlinux: Sure but how can I make asterisk place the call instead of giving a generic "couldn't call" error? |
| 02:33.00 | [TK]D-Fender | mail323: You aren't showing us your actual failed call. How are we expected to know what was done wrong? |
| 02:33.10 | mail323 | Penguin: Pretty much that's what I have except for no user or secret |
| 02:36.01 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 02:39.18 | mail323 | Ok it's a codec issue, but very strange that sip debug didn't show anything. |
| 02:40.48 | [TK]D-Fender | Correct, you haven't shown anything.... |
| 02:41.53 | mail323 | [TK]D-Fender: I have shown EVERYTHING that asterisk is supplying: "couldn't call exten@host" Nothing more, nothing less. SIP debug on both ends did not show an INVITE. |
| 02:42.27 | [TK]D-Fender | mail323: That is no a pastebin of your dialplan and the complete call attempt at verbose 10, and sip debug enabled |
| 02:43.16 | [TK]D-Fender | mail323: You pasted one line. Not any of the backup of the bits that caused it. |
| 02:51.35 | *** part/#asterisk ghost75 (~trechber@dslb-178-002-159-194.pools.arcor-ip.net) |
| 03:02.10 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 03:08.55 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 03:08.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 03:09.12 | kuku | http://pastebin.ca/2311084 << any ideas why one side of the call gets dropped right away after I bridge ? |
| 03:09.43 | kuku | sorry, http://pastebin.ca/2312024 |
| 03:13.41 | *** join/#asterisk dpilon (~nobody@c-50-138-178-238.hsd1.ct.comcast.net) |
| 03:13.54 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 03:15.19 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 03:15.20 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 03:18.47 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 03:31.16 | *** join/#asterisk hbee (~hbee@87-100-155-185.bb.dnainternet.fi) |
| 03:37.02 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 03:40.42 | lvlinux | can someone point me to a page somewhere talking about/explaining setting outgoing callerid? (on outgoing sip calls specifically fi it makes any difference) |
| 03:45.48 | [TK]D-Fender | lvlinux: "core show function CALLERID" |
| 03:48.34 | lvlinux | wow thanks---didn't know that type of info was built-in. |
| 03:51.59 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.164) |
| 03:57.20 | *** join/#asterisk serafie1 (~erin@76.73.167.231) |
| 04:05.16 | *** join/#asterisk CunningPike (~CunningPi@d206-116-73-41.bchsia.telus.net) |
| 04:22.52 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 04:27.43 | *** join/#asterisk gg608f (~Adium@187.207.35.79) |
| 04:37.38 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 05:08.24 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 05:16.31 | *** join/#asterisk din3sh (~din3sh@41.136.81.209) |
| 05:23.43 | din3sh | how does callbackextension work in ast 1.8? |
| 05:27.48 | igcewieling | din3sh: I'm not aware of any significant changes in that regard between 1.4 and 1.8 |
| 05:29.39 | igcewieling | the functionality of /whatever in a register line should be replicated by the callbackextensoin= in the peer for 1.6 and later |
| 05:30.47 | din3sh | euhh actually how does this work in 1.4? |
| 05:30.51 | din3sh | never tried it |
| 05:33.02 | igcewieling | The effect of register => username:secret@host/1234 should cause incoming calls to be routed to extension 1234. Without the /1234 calls will be routed to extension "s". |
| 05:33.42 | igcewieling | I do not know all the gory details of exactly HOW that works. If understand it correctly, the server you are registering to has to accept the request for 1234 |
| 05:33.54 | din3sh | hmm |
| 05:34.03 | din3sh | not exactly what am looking for then |
| 05:34.19 | igcewieling | in 1.6+ I believe you can put the registration info in the peer config, but I've never used it that way. |
| 05:34.24 | igcewieling | What are you looking to do?\ |
| 05:34.31 | din3sh | call back on busy |
| 05:34.51 | din3sh | trying to make "Call Completion Supplementary Services" work |
| 05:34.55 | din3sh | not working so far |
| 05:35.36 | igcewieling | "core show application retrydial" may not be exactly what you are looking for, but check it out anyway |
| 05:35.48 | igcewieling | I've never used CCS |
| 05:36.34 | igcewieling | din3sh: have you checked the Asterisk Book to see if there is any info there. voip-info.org might be helpful, but be SURE to verify what it tells you -- it is often wrong |
| 05:37.02 | din3sh | followed this |
| 05:37.03 | din3sh | https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29 |
| 05:37.04 | ectospasm | voip-info tends to have a lot of old info |
| 05:37.31 | ectospasm | ... evidenced by pages saying "New in 1.2!" |
| 05:37.41 | din3sh | voip-info shows callback using call file in ...asterisk/outgoing |
| 05:37.58 | igcewieling | Yup! the sample configs included with Asterisk are good to look at too. the recent Asterisk book may include the info. asteriskdocs.org I think. |
| 05:38.20 | igcewieling | bbiaw |
| 05:49.36 | din3sh | having a look at it |
| 05:51.06 | Kobaz | schnow |
| 05:55.03 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 05:55.10 | din3sh | there's 2 new columns in CDR mysql table in ast 1.8, peeraccount & linkedid |
| 05:55.26 | din3sh | i added them to my db but no value is recorded |
| 06:01.06 | *** join/#asterisk mattsl (~user@c-76-123-231-159.hsd1.tn.comcast.net) |
| 06:11.51 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 06:14.27 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 06:28.24 | *** join/#asterisk hegars (~hegars@061092248178.ctinets.com) |
| 06:40.39 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.164) |
| 06:44.02 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 07:00.37 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 07:24.57 | *** part/#asterisk mail323 (~mail323@c-98-254-82-94.hsd1.fl.comcast.net) |
| 07:41.48 | *** join/#asterisk dgeary2 (~david@2001:5c0:1000:a::417) |
| 07:42.31 | dgeary2 | any tips for troubleshooting sip/tls connections? |
| 07:43.18 | dgeary2 | seeing 'Encrypted Alert' in Wireshark |
| 07:43.54 | dgeary2 | sflphone is reporting 'Host unreachable' |
| 07:45.42 | dgeary2 | using asterisk 1.8 |
| 07:54.15 | *** join/#asterisk ^rage^ (~mts@213.21.4.41) |
| 07:54.46 | ^rage^ | hi! anybody using AsyncAGI? |
| 08:00.20 | *** part/#asterisk izx (~karthick@shellium/member/karthick87) |
| 08:01.25 | ketas | ffs, someone is dosing my asterisk now :P |
| 08:03.19 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 08:06.11 | ketas | my 1mbit upstream can't handle so many calls :) |
| 08:07.47 | ketas | 15 active ulaw channels |
| 08:10.52 | ketas | so much of wait forever guest calls |
| 08:12.05 | ketas | i didn't realize that those things never hang up |
| 08:16.58 | ^rage^ | using asterisk on TP-LINK TL-MR3020 with chan_dongle and 4 gsm modems ))) |
| 08:23.38 | ketas | hahaha |
| 08:28.16 | ketas | nice disk filler too |
| 08:39.08 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 08:40.01 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 08:47.26 | ketas | haa, so this wasn't very good idea |
| 09:03.40 | *** join/#asterisk evilman_home (kvirc@2.92.139.179) |
| 09:31.38 | *** join/#asterisk sorhas (~lakee@178-164-147-109.pool.digikabel.hu) |
| 09:37.49 | ChannelZ | yawns |
| 09:51.27 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 09:55.02 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-159-194.pools.arcor-ip.net) |
| 10:34.36 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.17.63) |
| 10:37.31 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 10:38.10 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 10:50.29 | *** join/#asterisk dgeary2 (~david@2001:388:e001:1e00:98f9:edec:345:ad33) |
| 11:01.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:02.26 | *** join/#asterisk tamiel (~tamiel@AToulouse-257-1-18-211.w90-5.abo.wanadoo.fr) |
| 11:31.06 | *** join/#asterisk joshuahh (~joahahua@27.32.48.149) |
| 11:32.00 | joshuahh | hello, looking at buying possibly a couple of IP phones to connect to my asterisk server.. I am wondering if anyone can recommend a phone they are very happy with.. more looking for your feedback.. |
| 11:32.26 | joshuahh | i was looking at a Cisco 7940 |
| 11:32.42 | ectospasm | I'm biased, but Digium phones (D40, D50, D70) are my favorite. |
| 11:32.59 | joshuahh | you work for them ? |
| 11:33.03 | ectospasm | yep |
| 11:33.05 | ectospasm | (-; |
| 11:33.29 | ectospasm | Digium phones are the only ones on the market that are specifically designed to work with Asterisk. |
| 11:34.17 | ectospasm | They can work with other things (they're SIP phones), but the lionshare of their features are only available with the Digium Phone Module for Asterisk (DPMA, free with Digium phones) |
| 11:34.24 | ectospasm | joshuahh: what version of Asterisk are you using? |
| 11:34.27 | joshuahh | http://www.ebay.com.au/sch/i.html?_from=R40&_sacat=0&_nkw=digium&LH_PrefLoc=1 |
| 11:34.39 | joshuahh | :( |
| 11:35.10 | ectospasm | I wouldn't suggest buying from eBay. You won't be eligible for support (since no authorized reseller or distributor is authorized to sell via eBay). |
| 11:36.02 | ectospasm | http://store.digium.com/products.php?category_id=200 |
| 11:36.19 | ectospasm | That's the high end price; you're likely to find cheaper deals from a local reseller. |
| 11:36.28 | ghost75 | cisco if you have lots of time lol |
| 11:37.19 | joshuahh | ghost75: do any of us have time :( |
| 11:37.30 | ghost75 | btw is there any reason to use tcp instead udp on phone? |
| 11:38.06 | WIMPy | Bad firewalls. |
| 11:38.21 | joshuahh | http://www.digium.net.au/ <-- i guess these are the guys to go with |
| 11:38.23 | joshuahh | lets see.. |
| 11:39.38 | ectospasm | RTP never makes sense over TCP. What're you gonna do, retransmit a lost frame? Pfft! |
| 11:40.00 | joshuahh | http://voip-australia.net/digium-d40-voip-phone-1teld040lf-p-11730.html?zenid=4a392fcee3693f5691deaed598e337ac |
| 11:40.33 | ectospasm | link is borken |
| 11:41.08 | joshuahh | http://voip-australia.net/digium-d40-voip-phone-1teld040lf-p-11730.html ? |
| 11:42.09 | ghost75 | <PROTECTED> |
| 11:42.47 | WIMPy | The issue with Digium phones is that it seems you cannot configure all features without DPMA, but the availability of that module is somewhat restricted. |
| 11:43.27 | ectospasm | WIMPy: if you have a Digium phone, it's not restricted. |
| 11:43.33 | ectospasm | of course, it's not open source |
| 11:43.37 | WIMPy | The features aren't bad, but I find the user interface a rather annoying. |
| 11:43.49 | joshuahh | WIMPy - what phone do you use? |
| 11:43.49 | ectospasm | how so? |
| 11:44.02 | WIMPy | That's the point. How do I get the module to work with *MY* Asterisk? |
| 11:45.18 | WIMPy | In a lot of ways. Use of yes/no keys, line 1 to locese dialogs for a start. |
| 11:45.41 | ectospasm | WIMPy: DPMA only works with certain versions of Asterisk, because they had to rewrite large chunks of it for the phones. |
| 11:45.49 | WIMPy | And the call handling suffers quite a bit when using more than one account. |
| 11:46.10 | ghost75 | hmm the phone doesnt register |
| 11:46.52 | WIMPy | As ist's a binary module, I suspect it will only work if you installed Asterisk from the compatible package. |
| 11:47.34 | ectospasm | You can install Asterisk from source, but it has to be Certified Asterisk (i.e. 1.8.15-cert1), 10-digiumphones, or 11 |
| 11:47.37 | WIMPy | Usually you can't just copy over a module from another build even of the same version. |
| 11:48.11 | WIMPy | is on TRUNK |
| 11:48.11 | ectospasm | No, because APIs/ABIs change |
| 11:48.43 | ectospasm | ...not sure why you'd want to track that for production... |
| 11:48.44 | WIMPy | So big are the cances to get that module working? |
| 11:48.54 | WIMPy | +h |
| 11:49.10 | ghost75 | who needs really encrypted phones? |
| 11:49.11 | WIMPy | Because it's the best version available. |
| 11:49.16 | ectospasm | I don't know if we'll ever release a module for trunk. Maybe when trunk forks into a branch |
| 11:49.56 | WIMPy | For me Asterisk is work in progress, not something stable. |
| 11:50.00 | ghost75 | whooohooo my cisco phone is ringing |
| 11:50.24 | ^rage^ | i prefer cisco SPA504 |
| 11:50.28 | WIMPy | Anyway: How big are the chances to get a binary module working with *YOUR* build of a supported version? |
| 11:50.50 | ectospasm | pretty good, if you've got the right version for that module (-; |
| 11:51.01 | ectospasm | of course, depending on the module and the Asterisk version. |
| 11:51.42 | WIMPy | So that module is less picky about compatibility than others? |
| 11:53.01 | joshuahh | if you had to choose, would you buy any of these? |
| 11:53.03 | joshuahh | http://www.ebay.com.au/itm/CISCO-IP-Phone-7940-Neutral-Bay-/221185830979?pt=LH_DefaultDomain_15&hash=item337fb3f043&_uhb=1#ht_500wt_1288 |
| 11:53.11 | joshuahh | http://www.ebay.com.au/itm/Cisco-IP-Phone-7961-/221185831618?pt=LH_DefaultDomain_15&hash=item337fb3f2c2&_uhb=1#ht_500wt_1156 |
| 11:53.15 | joshuahh | or |
| 11:53.16 | joshuahh | http://www.ebay.com.au/itm/Cisco-Unified-IP-Phone-7945G-Used-Colour-screen-VoIP-Excellent-condition-/251222436829?pt=LH_DefaultDomain_15&hash=item3a7e062bdd&_uhb=1#ht_500wt_1288 |
| 11:53.17 | joshuahh | :) |
| 11:53.36 | ghost75 | i recommend 7941 if you dont need 6 line buttons |
| 11:53.40 | WIMPy | So far I prefer the old Snom phones. 320/360/370. |
| 11:53.57 | ghost75 | the 7940/7960 are quite old |
| 11:54.06 | WIMPy | thinks that line buttons are a bad concept. |
| 11:54.18 | WIMPy | Phones should have call buttons. |
| 11:54.39 | ghost75 | speeddialbutton? |
| 11:55.02 | WIMPy | No. current calls. |
| 11:55.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 11:55.32 | WIMPy | To be able to easyliy manange multiple calls with only one touch of a button. |
| 11:56.13 | WIMPy | Having said that the call handling on the Digium phones is the best I've seen on VOIP phones so far, but definitely not ideal. |
| 11:58.04 | ghost75 | you mean switch the call with a button? |
| 11:58.25 | WIMPy | Yes |
| 11:58.50 | ectospasm | actually, handling multiple calls makes the most sense of any phone I've used. But again, I'm biased. |
| 11:58.54 | ghost75 | i think most people use it to see if somebody else is talking and to transfer call |
| 11:59.49 | WIMPy | It doesn;t seem possible to get BLF on the Digium phones without teh DPMA module. |
| 12:00.00 | ectospasm | I handle up to three queue calls at once. Digium phones make it easy to navigate (and that is a feature of their firmware, IIRC. No DPMA required). |
| 12:00.23 | WIMPy | Correct. |
| 12:00.57 | ectospasm | WIMPy: it's possible, but you need Contacts XML magic, and the appropriate subscription hints |
| 12:01.02 | WIMPy | But it could be easier. |
| 12:01.20 | WIMPy | Is that documented anywhere? |
| 12:01.21 | ectospasm | right, but folks could use DPMA as well |
| 12:01.48 | ectospasm | WIMPy: https://wiki.asterisk.org/wiki/display/DIGIUM/Contacts |
| 12:02.13 | WIMPy | Let me guess. That's not linked from anywhere? |
| 12:02.34 | ectospasm | It's on the official Asterisk Wiki |
| 12:02.50 | ectospasm | I have it in my history |
| 12:03.02 | WIMPy | Just like I just found firmware updates by accident while browsing the downloads. I haven't found a link on any page. |
| 12:03.46 | ectospasm | well, downloads is the place to get it. Not sure if there's an announce list for phone firmware/DPMA |
| 12:04.13 | WIMPy | I dind't even find a link there. |
| 12:04.37 | WIMPy | If you look at phones.digium.com you get the impressin the phones are more or less unsupported. |
| 12:05.26 | ectospasm | That's more for the App Engine API |
| 12:05.40 | WIMPy | If you dig deep, you can find bits of intormation scattered around. But as a potential buyer, you probably won;t do so. |
| 12:05.41 | ectospasm | I'd never been to phones.digium.com until right now |
| 12:06.02 | WIMPy | Yes, that seems to be the only obvious but of information. |
| 12:07.58 | ectospasm | Hmmm... most of them should hit here: http://www1.digium.com/en/products/asterisk/phones |
| 12:08.09 | ectospasm | ...it has links to downloads, and the Asterisk Wiki |
| 12:08.47 | ghost75 | http://pastebin.com/zuYhWfrJ <- why this rings on my extension 14 ? |
| 12:09.36 | ectospasm | ghost75: need more input, from that output it should dial SIP/11 |
| 12:09.47 | ghost75 | yes i dialed 11 but it rings on 14 |
| 12:10.21 | ectospasm | is SIP/11 forwarded to 14? The output doesn't indicate that |
| 12:10.25 | WIMPy | ectospasm: firmware is mentined for phones.digium.com. I don't see a way how to find it. |
| 12:11.37 | ectospasm | It's in the DPMA documentation. |
| 12:11.40 | WIMPy | And I can't find the contacts document you just posted anywhere, either. But I see some information on provisioning there which I will take a look at now. |
| 12:12.20 | ghost75 | in dialplan there is only the dial cmd |
| 12:12.25 | WIMPy | Maybe not the best place to put it. I already know I can;t use it. |
| 12:12.26 | ectospasm | WIMPy: it's in any of the DPMA sections. |
| 12:12.50 | ghost75 | _1X,1,Dial(SIP/${EXTEN},20,tTkKxX) |
| 12:14.32 | ectospasm | ghost75: pastebin the entire CLI output |
| 12:14.35 | WIMPy | And the other thing is that DPMA is only available for local phones anyway. |
| 12:15.31 | ectospasm | well, you can connect to a remote Asterisk system, but you've got to plug in the IP address of the remote server. I dunno why you'd think mDNS would be transmitted over the WAN |
| 12:15.37 | ghost75 | hey i think i know why |
| 12:15.54 | WIMPy | I don't. |
| 12:15.56 | ghost75 | on extension 14 i configured line button for 11 |
| 12:16.28 | ghost75 | i thought this was just displaying calls, not ringing there |
| 12:16.37 | ectospasm | what kind of phone is it? |
| 12:16.47 | ghost75 | cisco :) |
| 12:17.51 | ghost75 | http://pastebin.com/qFka6qtY <- i did something here wrong looks |
| 12:19.26 | ectospasm | is that for phone 14? |
| 12:19.46 | ectospasm | That's having line 2 register as authName 11 |
| 12:20.45 | ^rage^ | think about some strange thing: asterisk + zeromq |
| 12:20.52 | ghost75 | yes thats for the single line2 button of ext 14 |
| 12:21.56 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.160.253) |
| 12:22.24 | ectospasm | you should probably just have that line key subscribe to SIP/11, otherwise that phone will be getting 11's calls |
| 12:23.11 | ghost75 | only how is the question |
| 12:25.00 | ectospasm | I don't know much about anything Cisco, so I can't help you. |
| 12:28.44 | *** join/#asterisk BrokenArrow (~Lp@unaffiliated/brokenarrow) |
| 12:40.19 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 12:44.27 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:d033:44aa:139c:d103) |
| 12:46.34 | *** join/#asterisk nicknam1232 (d9494764@gateway/web/freenode/ip.217.73.71.100) |
| 12:51.59 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 12:56.38 | *** join/#asterisk dgeary2 (~david@2001:388:e001:1e00:98f9:edec:345:ad33) |
| 12:58.19 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:07.26 | ghost75 | the messages uri is a extension or in featuremap? |
| 13:14.50 | ghost75 | wow mwi works |
| 13:25.38 | ghost75 | i think the line buttons on this phone are kind of useless |
| 13:36.52 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 14:07.08 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:07.08 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:18.52 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.25.65) |
| 14:41.15 | WIMPy | has converted his phone book to a contacts.xml. After presig the contacts button I can stare at "loading contacts list" for about a minute, then I'm back to the idle screen. Do I have too many? |
| 14:49.36 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:58.49 | WIMPy | yes |
| 15:10.06 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:21.21 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 15:21.34 | *** join/#asterisk dgeary2 (~david@2001:388:e001:1e00:98f9:edec:345:ad33) |
| 15:27.45 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 15:31.56 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.98.93) |
| 15:38.54 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:38.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:40.11 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 15:54.57 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 15:56.44 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 15:57.48 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 15:59.08 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 16:00.41 | *** join/#asterisk andy09usa (~Andrey@audotov.com) |
| 16:15.19 | pabelanger | well that was painless. Dahdi upgrade from 2.5.1 to 2.6.1 via packaging |
| 16:15.40 | pabelanger | works well in my testing environment, now to do production |
| 16:21.58 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 16:31.49 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 16:32.36 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 16:45.26 | WIMPy | Is there any way to reload te contacts on the Digum phones other than by "Reconfigure" and dinking 2 or 3 pots of coffee? |
| 16:53.45 | *** join/#asterisk din3sh (~din3sh@41.136.85.30) |
| 17:04.12 | WIMPy | traceroute 216.81.59.173 |
| 17:30.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 17:34.46 | *** join/#asterisk mun24 (~chatzilla@173.248.193.85) |
| 17:36.00 | mun24 | I am trying to link lync serverwith trixbox. Please help |
| 17:40.44 | *** join/#asterisk thecoda (~kevin@027b608d.bb.sky.com) |
| 17:41.38 | thecoda | Can anyone advise on how I'd track down why I'm not detecting a phone answered over a pots connection? |
| 17:49.03 | [TK]D-Fender | thecoda: Versus? |
| 17:49.15 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 17:50.02 | thecoda | I dial the thing, it rings, I pick it up, it still appears to be ringing front he perspective of my SIP phone |
| 17:50.12 | thecoda | s/front/from |
| 17:50.53 | [TK]D-Fender | What does your CLI show? |
| 17:50.55 | thecoda | What's the best way to debug exactly what's going on in my DAHDI card & drivers? |
| 17:54.32 | mun24 | I am getting this error in wireshark 590 destination unreachable for tunk |
| 17:54.56 | thecoda | There's nothing revealing on the CLI, I've asked here before… I need to see what's going on at a lower level |
| 17:55.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:56.01 | [TK]D-Fender | thecoda: Show us the call, and your configs |
| 17:57.05 | thecoda | You'll have to wait until the mrs is off the phone :) |
| 17:58.56 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:59.35 | thecoda | http://pastebin.com/5TXh3jm0 |
| 18:01.21 | thecoda | http://pastebin.com/A6iN5iNR |
| 18:03.32 | thecoda | Not importing dahdi-channels.conf, it looks to be using a deprecated configuration |
| 18:04.34 | thecoda | And because the book doesn't show it being imported either :) |
| 18:29.47 | *** join/#asterisk ghost75 (~ghost75@dslb-088-066-184-139.pools.arcor-ip.net) |
| 18:35.49 | igcewieling | chan_dahdi.conf is the Asterisk config file for dahdi. there are no others. You can #include a_file.conf if you want. that is what people do with dahdi-channels.conf, but Asterisk will NEVER read it unless you #include it |
| 18:41.55 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-76-226.se.biz.rr.com) |
| 18:47.04 | thecoda | I didn't want to include it, so I haven't |
| 18:48.15 | thecoda | Though I *did* make sure that I'd manually copied any relevant setting from that file into chan_dahdi and set up the remainder according to the book + whatever Google could tell me about UK-specific configuration |
| 18:49.33 | thecoda | My problem is that I can dial out over that dahdi line, but asterisk doesn't detect the call being answered |
| 18:49.38 | thecoda | It just keeps on ringing |
| 18:52.04 | *** join/#asterisk nightrid3r (~kvirc@62.205.84.115) |
| 19:07.32 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:10.47 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 19:10.49 | ghost75 | my extensions doesnt like to subscribe, looking now since 1h why |
| 19:12.51 | ghost75 | is there something to be enabled on the phone itself? |
| 19:15.16 | [TK]D-Fender | to what? |
| 19:15.25 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:26.34 | din3sh | ghost75:still with the cisco? |
| 19:27.50 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 19:28.10 | *** part/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 19:28.43 | *** join/#asterisk rdm (rdm@unaffiliated/qubix) |
| 19:34.38 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6cef::8899) |
| 19:34.41 | cusco | howdy folks |
| 19:35.06 | cusco | is there some json stuff in asterisk |
| 19:35.14 | cusco | to output lines in json or so? |
| 19:38.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:40.10 | *** join/#asterisk CunningPike (~CunningPi@d206-116-73-41.bchsia.telus.net) |
| 19:43.09 | ghost75 | din3sh: that is working more or less |
| 19:44.03 | ghost75 | but no luck with blf yet |
| 19:47.10 | ghost75 | in asterisk everything should be enabled |
| 19:55.55 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 19:57.26 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:57.26 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:08.05 | *** part/#asterisk ghost75 (~ghost75@dslb-088-066-184-139.pools.arcor-ip.net) |
| 20:12.31 | *** join/#asterisk ghost75 (~quassel@dslb-088-066-184-139.pools.arcor-ip.net) |
| 20:27.03 | Nugget | was inspired by that joke traceroute to make "telnet macnugget.org" version 2.0 |
| 20:30.05 | WIMPy | That seems to be missing some CRs or the right terminal config. |
| 20:32.17 | Nugget | possibly. |
| 20:32.40 | Nugget | I only tested from unix |
| 20:43.06 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:91b8:7e52:91aa:f22) |
| 20:50.01 | *** join/#asterisk ghost75 (~ghost75@dslb-088-064-062-218.pools.arcor-ip.net) |
| 20:53.42 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 21:02.08 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 21:21.42 | *** join/#asterisk andy09usa (~Andrey@audotov.com) |
| 22:06.45 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:11.17 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 22:11.53 | kuku | How can I stress test asterisk ? |
| 22:11.57 | kuku | Can I make test calls somehow ? |
| 22:13.09 | WIMPy | Just connect it to the internat and wait. |
| 22:13.28 | WIMPy | s/nat/net/ |
| 22:19.22 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 22:20.41 | kuku | hehe |
| 22:38.38 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 22:48.01 | ChannelZ | I dunno, it sort of is the internat |
| 22:48.41 | WIMPy | thought about that as well. |
| 22:49.35 | *** part/#asterisk dym (~patrick@intrepid.openroot.de) |
| 22:51.10 | ghost75 | downloads the internat |
| 22:58.54 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:58.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:01.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:07.51 | ghost75 | qualify=yes makes problems with presence? |
| 23:08.17 | WIMPy | What makes you think so? |
| 23:09.37 | ghost75 | http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones |
| 23:12.48 | ChannelZ | CRISCO! |
| 23:13.11 | ghost75 | they subscribe only once or what |
| 23:13.14 | ChannelZ | but notice it says "sip TCP peer" |
| 23:14.09 | ghost75 | yes it uses tcp |
| 23:14.19 | ChannelZ | It's probably more pointless, but I'd be interested to know why. |
| 23:14.36 | Nugget | You have to use sip/tcp on the ciscos in order to use the presence patch |
| 23:15.01 | *** part/#asterisk sethrogaine (~Stian@svr3.st0len.co.za) |
| 23:15.09 | ghost75 | no patch installed |
| 23:15.17 | Nugget | You'll need to do that. |
| 23:15.48 | ChannelZ | How much was this phone again? :) |
| 23:16.08 | Nugget | Cisco phones and asterisk are a total pain in the ass. |
| 23:17.13 | kikohnl | I'd rather use a Cisco than a Polycom :-) |
| 23:18.12 | ghost75 | 20 bucks phone |
| 23:18.34 | ChannelZ | You'd rather use a Cisco, they just don't make it that easy to actually do it. |
| 23:19.54 | ghost75 | after i removed the qualify option, the phone subscribed |
| 23:19.55 | kikohnl | before the company went under, we were a Cisco partner, 7945's were $95 new |
| 23:20.10 | kikohnl | partner pricing for internal use |
| 23:27.38 | *** join/#asterisk rdm (rdm@unaffiliated/qubix) |
| 23:40.56 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 23:41.18 | *** join/#asterisk thecoda (~kevin@027b608d.bb.sky.com) |