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| 04:05.11 | whitebook | Can someone give me a hand with format_mp3? |
| 04:05.22 | whitebook | not sure why I cant playback mp3 files |
| 04:11.20 | [TK]D-Fender | Show us |
| 04:11.23 | [TK]D-Fender | ~pb |
| 04:11.23 | 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. |
| 04:11.24 | [TK]D-Fender | ^^^ |
| 04:16.29 | *** join/#asterisk moneylotion (~moneyloti@unaffiliated/moneylotion) |
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| 04:17.45 | whitebook | Unable to open '/sounds/declined' (format (ulaw)): No such file or directory |
| 04:18.13 | whitebook | am I supposed to use Playback('/sounds/declined', 'mp3') |
| 04:18.38 | [TK]D-Fender | no |
| 04:18.55 | [TK]D-Fender | first those quotes alone are not legal |
| 04:19.31 | [TK]D-Fender | And * will pick the best format available automatically. You never pass an extension to playback apps |
| 04:19.51 | whitebook | ok I will remove quotes |
| 04:20.46 | whitebook | I see |
| 04:20.53 | whitebook | my quotes were the problem |
| 04:21.01 | whitebook | not sure why I thought I needed them |
| 04:22.11 | whitebook | does it work with mp3 files over http as well or only local files? |
| 04:23.05 | [TK]D-Fender | Don't think standar playback will work for that. Maybe via MP3Player() |
| 04:23.42 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 04:29.24 | whitebook | Is there any vlc or mod_vlc type thing in Asterisk? |
| 04:32.43 | [TK]D-Fender | nope |
| 04:33.12 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 05:13.56 | wyoung | Afternoon |
| 05:14.50 | wyoung | What are some good strategories for stopping people from port scanning you then trying to brute force their way into asterisk? Also they seem to like to ring all default extensions which is annoying |
| 05:15.39 | wyoung | I have had to put in place a queue and menu at night otherwise my phone rings at all hours |
| 05:28.07 | wyoung | I have fail2ban setup (so they cannot try to brute force me) but they can still ring my number and attempt to exploit asterisk :\ |
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| 06:14.36 | drmessano | disallow anonymous SIP |
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| 10:55.30 | pa | ChinaRoby.. is that crap? |
| 10:55.46 | pa | better atcom? or are both about the same? |
| 10:55.57 | pa | i guess both use the same tiger chip, right? |
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| 11:47.54 | pa | by the way, is the openvox b100e supported by asterisk via dahdi? |
| 11:50.57 | WIMPy | That's what they say. |
| 11:51.39 | *** join/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 11:53.23 | WIMPy | And they also work with Linux drivers. |
| 11:54.16 | pa | maybe it's worth trying then.. 80 euros is not crazy |
| 11:54.27 | pa | will do and let you know :) |
| 11:54.43 | pa | by the way, how does asterisk play with google voice numbers these days? |
| 11:54.52 | pa | can it receive calls placed to google voice numbers? |
| 11:55.16 | WIMPy | Does it still exist? |
| 11:56.55 | pa | hm.. i thought it's something new.. |
| 11:57.00 | pa | but for now only for US |
| 11:57.19 | pa | (but i guess i can connect to it also from abroad, given that i can get one somehow) |
| 11:59.38 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:02.52 | WIMPy | No, the new one is hangouts. |
| 12:08.49 | pa | right, but i guess that this is complementary, no? i mean: https://support.google.com/chat/answer/187927?hl=en |
| 12:09.59 | WIMPy | Maybe infobot has something... |
| 12:10.02 | WIMPy | ~googlevoice |
| 12:10.02 | infobot | For information on setting up calls with google using Asterisk 1.8, see https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google, or see the External Services chapter: http://ofps.oreilly.com/titles/9780596517342/ch17.html |
| 12:10.23 | WIMPy | Hmm. Not that up to date. |
| 12:10.52 | *** join/#asterisk dotbill (~dotbill@vpn-91-238-221-33.lon.vorboss.net) |
| 12:11.57 | dotbill | hi all - anyone around that has knowledge of PJSIP_HEADER for reading the destination channel in a pre dial handler? |
| 12:16.49 | wyoung | hi WIMPy |
| 12:17.05 | wyoung | drmessano: but I needs ppl to call me! |
| 12:17.25 | wyoung | just not people who are trying to break into my asterisk server |
| 12:17.38 | wyoung | those ppl can GTFO |
| 12:18.06 | WIMPy | You want to accept calls from everywhere? |
| 12:19.11 | WIMPy | And there's only one to stop people from trying to brek in to services and that's not being connected to the internet. |
| 12:19.11 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 12:20.09 | WIMPy | And even if you allow guest calls, do you have a catch-all extension for them? |
| 12:20.59 | WIMPy | allows guest calls, but so far no one ever tried a valid extension. Almost always the usual UK numbers. |
| 12:27.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
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| 12:34.11 | pa | maybe file knows. IIRC he wrote the asterisk module for google voice |
| 12:34.33 | file | Google Voice continues to exist, apparently, but who knows for how long |
| 12:36.04 | pa | they focus on hangout now, but i think it makes sense if they would also offer a real phone number for an hangout account, as they do now |
| 12:36.16 | pa | (just in the US, yes. better than nothing) |
| 12:37.07 | pa | but maybe i can try to ask around what's the destiny of all that voip |
| 12:37.36 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 12:38.10 | file | Google will be Google. |
| 12:45.50 | dotbill | Does anyone know of a way to put a "Name" into the To header of an INVITE to an internal extension using PJSIP - it only has the username@IP at the moment |
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| 13:05.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:32.34 | mew007 | hello all. I'm totally noob to asterisk, but I have to know is particular phone online (turn on and has IP-address so you can ping it). How can I check it via 'asterisk -rx ""' ? |
| 13:35.23 | *** join/#asterisk mastahh|work (~mastahh@89.238.175.130) |
| 13:35.26 | mastahh|work | Hello all, |
| 13:35.29 | mew007 | in the db asterisk, table cdr you can see records like this: 2015-09-10 08:17:43;911;office;DAHDI/6-1;SIP/27-0000076e;Queue;workgroup;124;119;ANSWERED;3 |
| 13:35.30 | [TK]D-Fender | you can see if the peer it is supposed to register to has indeed been registered to: |
| 13:35.36 | [TK]D-Fender | sip show peer X |
| 13:35.42 | mastahh|work | I was wondering if anyone has installed Asterisk on Hyper-V before and if there are any issues in doing so? |
| 13:39.29 | mew007 | [TK]D-Fender: thank you! Line 'asterisk -rx "sip show peer X"' works like a charm |
| 13:44.10 | mew007 | and what does "Expire : 55" mean in output? |
| 13:45.04 | [TK]D-Fender | IIRC that's the life span before it needs to re-reg |
| 13:45.07 | mew007 | Status : Unmonitored |
| 13:45.22 | *** join/#asterisk stefan27 (~stefan27@212.247.4.149) |
| 13:45.26 | [TK]D-Fender | that is saying you don't have qualify enabled for it |
| 13:46.32 | mew007 | 3 times a got different Expire values: 33, 55, -1... |
| 13:48.50 | stefan27 | Asterisk through chan_sip sends a 200 OK SDP to a peer with attribute Session-Expires: 600;refresher=uas\r\n , but 5 minutes later asterisk sends an re-INVITE (same call-id, and now with refresher=uac), is this really correct behaviour? |
| 13:50.55 | stefan27 | if peer and asterisk agreed that asterisk (the uas) should do the refreshing, why does asterisk all of a sudden wants the uac to be the refresher? I did set session-refresher=uac in sip.conf, but that's not supposed to change the refresher mid-call, if uas started out as refresher? |
| 13:52.25 | stefan27 | Or does asterisk _become_ the uac when sending the re-invite? (Im not a sip-expert) |
| 13:53.03 | stefan27 | I thought the role of uac and uas was something persistent during a call |
| 13:58.03 | *** join/#asterisk ks3 (~ks3@205.142.197.104) |
| 13:58.30 | rogers | Can anyone recommend a utility to go through asterisk sip debug logs and display in a call flow format? |
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| 14:04.34 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 14:05.31 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-jgrjzmamrdnpgixc) |
| 14:05.31 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:10.44 | *** join/#asterisk posixninja (~posixninj@ec2-54-214-51-133.us-west-2.compute.amazonaws.com) |
| 14:11.04 | mastahh|work | Has anyone been able to find a suitable IM and Presence option to bolt on to Asterisk? |
| 14:12.59 | [TK]D-Fender | "option" = vague |
| 14:13.04 | jvhester | stefan27: From the perspective of the SIP RFC, uac and uas depends on who is requesting and who is responding even in the same call |
| 14:13.37 | mastahh|work | [TK]D-Fender, well with Cisco you have Cisco Jabber. Microsoft has Lync. Asterisk has ? I have investiagted Openfire Spark but it looks like it has not been support for a few years. |
| 14:14.38 | [TK]D-Fender | Forget presence as far as reporting your status. it isn't meant for that |
| 14:14.51 | [TK]D-Fender | other useopenfire IIRC |
| 14:17.14 | jvhester | rogers: idk if wireshark can parse Asterisk SIP debug logs but that is the standard tool for SIP call flow diagrams. |
| 14:18.05 | rogers | ok thanks |
| 14:19.37 | mastahh|work | Also, I have yet to find a supported Attendant Console that works with Asterisk...? |
| 14:21.34 | [TK]D-Fender | There have been dozens of them |
| 14:21.44 | [TK]D-Fender | FOP2 for example |
| 14:22.12 | mastahh|work | if you could care to recommend a few for me to investigate, that would be appreciative |
| 14:24.54 | *** join/#asterisk c0rnoTa (~c0rnoTa@mail.fibex.su) |
| 14:26.17 | *** join/#asterisk reveal (reveal@2401:7000:0:7::e376:fc6) |
| 14:28.45 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:29.24 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:30.10 | [TK]D-Fender | You've already got 1.... |
| 14:30.32 | *** part/#asterisk c0rnoTa (~c0rnoTa@mail.fibex.su) |
| 14:31.11 | mastahh|work | You're correct. But if I need to go in to a management meeting, I literally cannot go in to it with my eggs in one basket |
| 14:31.52 | Chainsaw | mastahh|work: Take us into the management meeting. Have them join #asterisk and we will clarify. |
| 14:31.55 | reveal | good morning |
| 14:32.05 | mastahh|work | Haha Chainsaw - maybe I will :) |
| 14:32.14 | reveal | [TK]D-Fender: hello dear sir |
| 14:32.26 | [TK]D-Fender | http://www.asteriskexchange.com/listings/324 |
| 14:32.38 | [TK]D-Fender | http://www.dialplate.com/ |
| 14:32.46 | mastahh|work | Thank you [TK]D-Fender |
| 14:32.51 | [TK]D-Fender | or any of a pile of others you get from a 3 word google search |
| 14:32.55 | [TK]D-Fender | https://www.google.ca/#q=asterisk+attendant+console |
| 14:33.02 | [TK]D-Fender | Not exactly hard to find |
| 14:33.36 | reveal | [TK]D-Fender: what would cause asterisk to terminate a call as soon as its picked upm could it be bad codecs or something more serious |
| 14:33.45 | [TK]D-Fender | reveal: Anything |
| 14:33.52 | mastahh|work | In all fairness, I approached a few companies I had dealt with in the past to see if they offered Asterisk support. Attendant Console is not at the forefront of the solution in my opinion, it's a nice to have feature |
| 14:34.07 | [TK]D-Fender | reveal: Guessing would be a waste of time. |
| 14:34.42 | reveal | [TK]D-Fender: needles |
| 14:34.57 | Ice_Strike | Looking through the doc, from what I can see there is no option to enable/disable inbound Queue? |
| 14:35.10 | reveal | I have this log to go on http://pastebin.com/C12BYSY3 |
| 14:35.40 | *** join/#asterisk dwery (~dwery@nslu2-linux/dwery) |
| 14:35.59 | *** join/#asterisk CeBe (~CeBe@xd9beaf21.dyn.telefonica.de) |
| 14:36.26 | [TK]D-Fender | Ice_Strike: huh? |
| 14:36.46 | [TK]D-Fender | Ice_Strike: The words "disable queue" don't mean anything in *.... |
| 14:37.00 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 14:41.20 | [TK]D-Fender | ] agi-OSDagent_conf.agi,agentmon: Failed to execute '/var/lib/asterisk/agi-bin/agi-OSDagent_conf.agi': Permission denied |
| 14:41.36 | [TK]D-Fender | You are failing to even execute the stuff you have in your dialplan.... |
| 14:41.58 | reveal | problem is |
| 14:42.03 | reveal | i can chmod 777 that |
| 14:42.09 | reveal | and it still errors |
| 14:42.12 | [TK]D-Fender | and is then proceeding to deliberately and very clearly hangup |
| 14:42.13 | [TK]D-Fender | -- Executing [28601000@osdial:2] Hangup("DAHDI/i2/9116-82", "") |
| 14:42.25 | Ice_Strike | Aha |
| 14:42.29 | [TK]D-Fender | You sent it to an AGI it has no permission to run... and then hangup |
| 14:43.49 | reveal | i think i figured out some of the problem too |
| 14:43.51 | Ice_Strike | Looking at the GUI - there is On/Off button. If Enabled - it prevent Agent logged to that Queue and callers wont be placed in the queue. |
| 14:44.04 | [TK]D-Fender | GUI is NOT supported here, |
| 14:44.11 | [TK]D-Fender | And that entire concept is an invention. |
| 14:44.18 | reveal | <PROTECTED> |
| 14:44.45 | [TK]D-Fender | reveal: doesn't matter |
| 14:44.54 | [TK]D-Fender | reveal: Your AGI ffialed to execute at all |
| 14:44.58 | [TK]D-Fender | failed* |
| 14:46.10 | reveal | http://pastebin.com/6459Ccbz |
| 14:46.15 | reveal | same error |
| 14:46.19 | reveal | even with chmod 777 |
| 14:47.03 | reveal | shouldnt be a perm error i would imagine |
| 14:47.04 | Ice_Strike | [TK]D-Fender I understand. I guess the logic in the dial plan to check the boolian value from the database, if enabled then use Queue() or else skip it. |
| 14:47.33 | [TK]D-Fender | reveal: Not that I see. |
| 14:47.52 | [TK]D-Fender | reveal: I do not see the faile you supposedly fixed for this |
| 14:48.00 | [TK]D-Fender | or that it has access through the whole path |
| 14:48.00 | reveal | what |
| 14:48.28 | reveal | [2015-09-10 09:43:15] agi-OSDagent_conf.agi,agentmon: Failed to execute '/var/lib/asterisk/agi-bin/agi-OSDagent_conf.agi': Permission denied |
| 14:48.31 | [TK]D-Fender | Sorry, missed a line in there |
| 14:48.49 | reveal | and at the bottom youll see that 777 |
| 14:48.50 | [TK]D-Fender | check the whole path... |
| 14:48.58 | [TK]D-Fender | and verify who * is running as |
| 14:49.15 | reveal | stat -c '%U %G %A %a %n' /var/lib/asterisk/agi-bin/ |
| 14:49.16 | reveal | asterisk asterisk drwxr-x--- 750 /var/lib/asterisk/agi-bin/ |
| 14:49.54 | [TK]D-Fender | check the rest of the path |
| 14:49.57 | [TK]D-Fender | not just that file |
| 14:50.02 | [TK]D-Fender | and prove who * is actually running as |
| 14:50.18 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 14:50.57 | reveal | var 755 root root var/lib 755 root root var/lib/asterisk 755 asterisk asterisk |
| 14:51.21 | reveal | var/lib/asterisk/agi-bin 750 asterisk asterisk |
| 14:51.23 | [TK]D-Fender | [10:50][TK]D-Fenderand prove who * is actually running as |
| 14:52.03 | reveal | you sort of have me lost here then |
| 14:52.11 | reveal | are you asking me to see if asterisk is running as asterisk |
| 14:52.13 | reveal | or another user |
| 14:53.28 | reveal | asterisk:x:493:491:Asterisk PBX:/var/lib/asterisk:/bin/bash |
| 14:54.17 | [TK]D-Fender | "core show settings" |
| 14:54.53 | reveal | ok |
| 14:55.25 | reveal | anything specific or pastebin it |
| 14:57.10 | reveal | http://pastebin.com/FsL3B7Y4 |
| 14:59.32 | [TK]D-Fender | <PROTECTED> |
| 14:59.35 | [TK]D-Fender | both blank |
| 14:59.38 | [TK]D-Fender | NOT good |
| 15:00.43 | reveal | no bueno? |
| 15:01.49 | [TK]D-Fender | [10:59][TK]D-FenderNOT good <--------------------- |
| 15:02.48 | reveal | yes |
| 15:03.29 | reveal | i mean whats the negative side of that |
| 15:03.51 | reveal | unless that is causing my permission errors |
| 15:03.54 | reveal | and etc |
| 15:13.39 | [TK]D-Fender | lack of user = nobody |
| 15:13.45 | [TK]D-Fender | fix your basics |
| 15:16.35 | reveal | ok where can i read to do that |
| 15:17.34 | [TK]D-Fender | asterisk.conf and your init script that starts * |
| 15:21.09 | reveal | weird |
| 15:21.11 | reveal | asterisk.conf shows |
| 15:21.12 | reveal | ;runuser = asterisk ; The user to run as. |
| 15:21.12 | reveal | ;rungroup = asterisk ; The group to run as. |
| 15:21.39 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-dileidsclmcymqxh) |
| 15:21.49 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-pjirnvuaixqomwes) |
| 15:21.58 | [TK]D-Fender | looks commented out to me... |
| 15:22.02 | [TK]D-Fender | ; <-------------------- |
| 15:25.02 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:28.43 | *** join/#asterisk meatballsoda (41d13d72@gateway/web/freenode/ip.65.209.61.114) |
| 15:29.00 | meatballsoda | hey, I'm building an iOS app and using asterisk on the backend |
| 15:29.14 | meatballsoda | But the calls don't seem to be working on Verizon, is there something I'm doing wrong? |
| 15:31.18 | [TK]D-Fender | meatballsoda: We don't know you haven't shown us a call |
| 15:32.32 | meatballsoda | [TK]D-Fender: what info do you need? |
| 15:32.42 | [TK]D-Fender | THE CALL |
| 15:32.56 | meatballsoda | How can I show you the call? |
| 15:33.03 | [TK]D-Fender | The actual evidence of what your system wwas doing and what reaction it got |
| 15:33.18 | [TK]D-Fender | meatballsodaHow can I show you the call? <- * CLI |
| 15:33.31 | meatballsoda | Everything works as normal (in terms of my server) just the end user using verizon doesn't hear anything |
| 15:33.40 | meatballsoda | Apparently Verizon blocks SIP |
| 15:33.46 | [TK]D-Fender | That's the FIRST detail you'v just dropped on us |
| 15:33.49 | meatballsoda | So I wanted to know if it was a widespread problem/known issue |
| 15:34.12 | [TK]D-Fender | generally if they block SIP then not having audio wouldn't be the outcome |
| 15:34.19 | [TK]D-Fender | it's stop the negotiation entirely |
| 15:34.22 | [TK]D-Fender | it'd* |
| 15:34.30 | [TK]D-Fender | Look at the call. |
| 15:34.52 | [TK]D-Fender | "core set verbose 10", "sip set debug on" |
| 15:34.55 | [TK]D-Fender | ~pb |
| 15:34.56 | 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. |
| 15:34.58 | [TK]D-Fender | ^^^ your friend |
| 15:37.02 | meatballsoda | it seems like its a normal call and all, we just can't hear anything |
| 15:37.07 | meatballsoda | it works on all other carriers and on wifi |
| 15:37.10 | meatballsoda | just not verizon LTE |
| 15:37.44 | [TK]D-Fender | you need to provide a proper complete description of what your call is going over, from, and to |
| 15:38.11 | [TK]D-Fender | And we still need to see the call to prove what was even negotiated |
| 15:40.07 | [TK]D-Fender | Or could simply leave..... |
| 15:40.31 | [TK]D-Fender | Today seems full of people who have nothing to show.... |
| 15:40.58 | [TK]D-Fender | "Mr. Mechanic, why doesn't my car work?" |
| 15:41.13 | [TK]D-Fender | "Let's look under the hood an see what's wrong then ..." |
| 15:41.19 | [TK]D-Fender | <PROTECTED> |
| 15:44.17 | cmendes0101 | I've seen people post about Verizon blocking SIP with ALG. He was probably having that issue |
| 15:45.07 | *** join/#asterisk [373K] (~Adium@frontdoor.fiveninety.373k.net) |
| 15:45.12 | [TK]D-Fender | Guessing is for chumps. So is quitting without looking. |
| 15:51.47 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 15:52.53 | reveal | [TK]D-Fender: LOL |
| 15:54.47 | [TK]D-Fender | What, I have to exert even the slightest bit of thought or effort? SCREW THIS! |
| 15:56.08 | reveal | [TK]D-Fender: it seems my issue isnt asterisk related, its call center related |
| 15:57.05 | reveal | although when my dahdi extension was 0116 i told them to change it to 9116 bc asterisk might see it as an international # |
| 15:57.35 | Dovid | hello all |
| 15:58.11 | reveal | and as far as people ragequitting tk |
| 15:58.16 | reveal | i dont thnk some people can handle your straight up no bullshit answers |
| 15:58.38 | reveal | which common sense says: if you have problem show us the problem dont tell us the problem |
| 15:59.09 | [TK]D-Fender | "I use verizon, what's my problem"? is not valid. You can sit and guess all day and it's a waste of time. If someone doesn't have enough IQ to realize that then they have a more serious problem than just their call |
| 16:00.10 | [TK]D-Fender | reveal: Your problem seems to be a pretty clear permissions issue |
| 16:00.20 | [TK]D-Fender | [11:21][TK]D-Fenderlooks commented out to me... |
| 16:00.32 | [TK]D-Fender | I haven't heard anything about you fixing what I already pointed out... |
| 16:03.48 | [TK]D-Fender | Have you? |
| 16:11.10 | [TK]D-Fender | resumes breathing |
| 16:11.13 | reveal | the issue is that asterisk is the pbx but its being managed by a program called osdial |
| 16:11.40 | [TK]D-Fender | And I pinned down something very specific. |
| 16:11.43 | [TK]D-Fender | Didi you FIX it? |
| 16:11.48 | reveal | ive raised the concerns you pointed out to me and they replied thats normal |
| 16:12.47 | [TK]D-Fender | Can also prove what things are running as by calling System() and having it output the user into a file. |
| 16:13.22 | reveal | right |
| 16:13.39 | [TK]D-Fender | And try copying the agi to another folder and running it from there |
| 16:13.59 | reveal | accord to osdial support which is the same as vici if youre familiar with that / in the core settings is not blank the / indicates its normal |
| 16:14.51 | [TK]D-Fender | Do the rest |
| 16:15.12 | [TK]D-Fender | and PC it all up afterwards |
| 16:15.14 | [TK]D-Fender | PB* |
| 16:15.45 | reveal | huh |
| 16:16.11 | [TK]D-Fender | copy it to another folder.try running it from there |
| 16:16.23 | [TK]D-Fender | It said it's a permissions thing |
| 16:16.26 | reveal | ok |
| 16:16.31 | [TK]D-Fender | so change where you put it and try running it from somewhere else |
| 16:16.36 | [TK]D-Fender | and see what happens |
| 16:19.07 | reveal | [TK]D-Fender: could it also be a permission error not only from /var/lib but /var/run/asterisk/asterisk.pid |
| 16:20.05 | reveal | the perms on /var/run/asterisk/asterisk.pid is 644 root root |
| 16:21.08 | [TK]D-Fender | No, that's an AGI permissions error |
| 16:21.47 | reveal | ahh |
| 16:22.23 | [TK]D-Fender | It's very explicit. |
| 16:22.36 | [TK]D-Fender | No permission on that GI for what it is, or where it is based on the * user |
| 16:22.48 | [TK]D-Fender | Take that for what it says and change the circumstances |
| 16:25.56 | dan_j | Hi. I'm trying out pjsip for the first time and used the conversion script to migrate my existing sip.conf file to pjsip format. |
| 16:26.06 | dan_j | Any idea why I'm seeing this at asterisk startup? |
| 16:26.08 | dan_j | <PROTECTED> |
| 16:26.19 | dan_j | happens for all the endpoint entries. |
| 16:34.21 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 16:34.31 | dan_j | It seems the conversion script is faulty as external_media_address doesnt appear to be an option for an endpoint |
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| 16:44.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 18:07.04 | Synthase_ | D-Fender keeps it entertaining in here. |
| 18:09.25 | *** join/#asterisk bulkorok (~Adium@92.206.232.134) |
| 18:11.08 | [TK]D-Fender | schadenfreude... |
| 18:12.52 | jvhester | gesundheit |
| 18:14.23 | *** join/#asterisk [red] (~red@binarii.tk) |
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| 18:20.37 | Synthase_ | Working on creating a ring-answer page for classrooms utilizing Digium phones w/ DPMA, but I appear to be missing something. SIP-info header is ignored, phones ring and function as a normal call. Relevant configs: http://hastebin.com/raw/sesejifihi |
| 18:24.46 | [TK]D-Fender | no <> |
| 18:24.53 | [TK]D-Fender | Don't add extra junk |
| 18:25.02 | [TK]D-Fender | AND no "" |
| 18:25.14 | [TK]D-Fender | double fail there.... |
| 18:26.09 | Synthase_ | Going off of Malcolm's post here. I'll edit and give it a go. http://forums.digium.com/viewtopic.php?p=209105 |
| 18:27.56 | Synthase_ | No change with SIPAddHeader(Alert-Info: testalert) |
| 18:28.34 | Synthase_ | chan_sip for the record |
| 18:31.24 | [red] | let's say i wanted to store the output of "core show channels verbose" into a database like mysql, any suggestions of how to do that, if it's even possible? |
| 18:31.36 | robmal | AMI |
| 18:32.10 | [red] | and then what, parse the output using an outside script? |
| 18:32.20 | robmal | Yes. |
| 18:32.25 | WIMPy | wonders what the use culd be |
| 18:32.52 | robmal | Verbose history of channels, d'oh. |
| 18:33.14 | [red] | no, more like management of channels |
| 18:33.33 | [red] | for example, let's say i want an account with a specific account code to only be allowed 2 channels at the same time |
| 18:33.57 | robmal | Uhm, there are better ways to do that. |
| 18:34.01 | WIMPy | 'core show function lie GROUP' |
| 18:34.06 | WIMPy | like |
| 18:34.32 | [red] | i love you guys |
| 18:34.34 | [red] | <# |
| 18:34.36 | [red] | <3 |
| 18:34.45 | robmal | No homo. |
| 18:35.09 | [red] | definitely :) |
| 18:35.12 | robmal | :-) |
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| 19:47.26 | reveal | i love this channel |
| 19:47.42 | reveal | specifically [TK]D-Fender only bc hes a no bullshit type of guy |
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| 19:53.27 | Milenco | yeah [TK]D-Fender and WIMPy rule :> |
| 19:53.59 | Milenco | just did my dCAP exam today btw, hoping for the best now.. |
| 19:54.27 | WIMPy | Do you still have to configure analog shit for that? |
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| 20:02.06 | Milenco | yeah WIMPy |
| 20:02.20 | Milenco | had to configure analog phone for internal use |
| 20:02.43 | Milenco | and pri as a trunk |
| 20:04.30 | WIMPy | When I set up a box containing both an analog and a BRI card, both Digium, I found myself unable to find a dahdi configuration that would let me use the same dialplan for both. That was quite a |
| 20:04.37 | WIMPy | bummer :-( |
| 20:05.31 | Milenco | hmm, had no issues here |
| 20:05.59 | Milenco | used an analog and pri digium card, ran dahdi_genconf and everything was configured correctly |
| 20:06.04 | Milenco | hardware-wise that is |
| 20:06.42 | WIMPy | Well, you used it as trunk. I tried to connect phones to both. |
| 20:07.00 | Milenco | ah, no havent done it that way |
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| 20:07.10 | Milenco | was that for the dcap or unrelated? |
| 20:07.27 | WIMPy | I didn't do a dcap. |
| 20:07.53 | Milenco | ah:) |
| 20:08.17 | Milenco | apparently it can take up to two weeks before i get my result |
| 20:08.49 | Milenco | ah well, we'll see. multiple choice part i'm feeling confident about, it's gonna be a close one for the practical.. |
| 20:08.56 | WIMPy | Looks liek there aren't that many locations left to get a dcap. |
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| 21:43.07 | marceloamorim | hello guys, there is some trick to use transcode on asterisk? |
| 21:44.25 | mjordan | no |
| 21:45.03 | mjordan | Asterisk will transcode automatically if it detects that two channels aren't using the same format. |
| 21:45.07 | marceloamorim | using native format -> ulaw , writeformat -> gsm, readformat -> gsm ( the writetranscode: yes (gsm@8000) -> (slin@8000) -> (ulaw@8000), the readtranscode is yes (ulaw@8000) -> (slin@8000) -> (gsm@8000) |
| 21:45.12 | marceloamorim | but there is no audio |
| 21:48.02 | mjordan | Why do you think the issue is one of transcoding? No audio can occur for a lot of other reasons. |
| 21:48.39 | marceloamorim | because if I use ulaw without transcode there is no problem |
| 21:49.14 | mjordan | No idea then. GSM transcoding to ULAW works great here. |
| 21:50.19 | igcewieling | marceloamorim: what EXACTLY aree you trying to transcode to/from? |
| 21:52.34 | marceloamorim | actually, I have one * with all my gateways(trunks) ( all using ulaw ), and I'll have all other * in another vmware with all sip users. |
| 21:54.09 | marceloamorim | this machine with my users I was trying to set gsm or ilbc on the sip users, I call from my user using the gsm to this sip trunk to my * (gateway) |
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