| 00:00.04 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fdzhaghbywwrgali) |
| 00:00.40 | Tagor | navaismo: all calls go to exten=>firstDID,1,... |
| 00:03.05 | navaismo | then you need to add other did |
| 00:03.48 | igcewieling | Tagor: Unless you are not telling us something (using freepbx or other GUI, etc) the call should come in and match an exten line for that DID. |
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| 00:04.30 | igcewieling | If that is not the case then your provider is likely sending all calls to that one DID. You can confirm this by looking to the To: info in sip debug |
| 00:04.33 | WIMPy | That obviousely depends on the provider and the general setup of the "trunk". |
| 00:05.00 | igcewieling | Tagor: what is your register => line (remove ONLY the password) |
| 00:11.29 | Tagor | igcewieling: 'To' does contain the right number: To: "00000002 00000002"<sip:00000002@company.mttm.voipit.nl>; but in * it goes to extension 00000001: -- Executing [00000001@mttm:1] |
| 00:11.54 | Tagor | navaismo: there's just one register line for all numbers |
| 00:12.20 | igcewieling | Tagor: do you use a /regexten on the register? |
| 00:12.32 | igcewieling | (that is why I wanted to see the register. |
| 00:12.44 | igcewieling | Tagor: put the entire invite packet on a pastebin |
| 00:13.28 | Tagor | igcewieling: you're right, I feel so stupid now :D |
| 00:13.33 | Tagor | igcewieling: thanks! |
| 00:13.42 | igcewieling | 8-) |
| 00:22.40 | Tagor | igcewieling: hmm, I tought removing the extension would fix the problem. But it still doesn't work. Here's my new register line: register => 0000001@company.mttm.voipit.nl@89.146.63.27:5060 |
| 00:22.54 | Tagor | igcewieling: when I call 000002 it goes to extension 000001 |
| 00:23.05 | igcewieling | pastebin the invite |
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| 00:32.00 | Tagor | igcewieling: http://pastebin.com/uXDaS0ye |
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| 00:35.50 | s[X] | lo all, Curiously has anyone played with Oreka/Orkaudio ? |
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| 00:47.01 | igcewieling | [2013-03-05 01:27:52] NOTICE[30850]: chan_sip.c:20325 handle_request_invite: Call from '000000001' to extension 's' rejected because extension not found in context 'mttm'. |
| 00:48.01 | igcewieling | Tagor: INVITE sip:s@my.ip.add.ress SIP/2.0 |
| 00:48.25 | igcewieling | there is where the extension s is coming from. Maybe your provider does not support sending calls to a specific DID on an account. Many don't |
| 00:50.48 | Tagor | igcewieling: I did not add 00000001,s,... because I want DID 00000002 go to 00000002,s,.... However To: "000000002 000000002"<sip:000000002@company.mttm.voipit.nl> does contain the right number. Is there a way to get that string in a var? |
| 00:50.55 | igcewieling | if you don't find a better solution you might try in exten => s, use SIP_HEADER to extract the To: information and Goto the TN extracted |
| 00:51.38 | igcewieling | When asterisk does not receive DNIS (a dialed number) it uses "s" as the extension -- it indicates the number was not in the actual invite |
| 00:51.53 | Tagor | igcewieling: ok, thanks, I'll give them a call tomorrow :) |
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| 01:47.05 | beefcafe | what should I do to run asterisk 11 as non-root? right now "service asterisk start" shows it started but in fact it didn't. using sudo before that works |
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| 03:27.34 | zerohalo | beefcafe: check out http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-13-SECT-4.html |
| 03:28.37 | beefcafe | thanks zerohalo |
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| 05:58.37 | chang33 | hi, i got the message =====> chan_sip.c:22807 handle_response_invite: just did sched_add waitid(264) for sip_reinvite_retry for dialog 2f44e651679edb625070b7d06f870809@10.0.110.31:5060 in handle_response_invite |
| 05:58.50 | chang33 | and the sip call dropped |
| 05:59.47 | chang33 | is it a bug or related to settings, i'm using asterisk 11.2.1 |
| 06:00.22 | [TK]D-Fender | Sounds like an attempt to reinvite is failing (typically NAT related) |
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| 06:01.21 | chang33 | the two sip phones are in the same subnet |
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| 06:04.55 | [TK]D-Fender | ChaAnd * itself? |
| 06:06.04 | chang33 | sorry, i didn't get it |
| 06:08.40 | [TK]D-Fender | And * itself? |
| 06:12.45 | chang33 | i would happen only in a video call |
| 06:12.58 | chang33 | it would |
| 06:13.32 | [TK]D-Fender | Where. Is. Asterisk. In. This. Picture. |
| 06:14.10 | chang33 | all in the same subnet |
| 06:15.45 | [TK]D-Fender | I'd go look at the complete call.... |
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| 06:23.41 | [TK]D-Fender | And ... it's checkout time.. later all |
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| 07:58.42 | schmidts | good morning |
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| 08:15.56 | aurs | Any suggestions for alternatives to gotoiftime? I'm seeing something similar to this: http://forums.digium.com/viewtopic.php?t=19241 |
| 08:16.30 | aurs | ...and I cannot find anything in the changelogs that indicates that the problem is fixed in newer releases |
| 08:18.15 | schmidts | aurs maybe you should search in den asterisk issues if there has been an official issue for this problem, cause if not, there will not be a fix for it |
| 08:21.47 | aurs | true |
| 08:22.21 | schmidts | aurs depending on the issue described in the forum, have you tried this with some more delay? |
| 08:22.59 | schmidts | i mean what happens if the first call comes in, 30 minutes after the start time? does it work then, or is also the first call wrong? |
| 08:23.05 | aurs | schmidts, I am not 100% sure of the pattern in it. But I do get some calls in my "opening hours" dialplan, that should go in "closed" dialplan some times |
| 08:23.28 | aurs | schmidts, no, I think you have to be within the first 2-3 minutes, but I don't have any good data on this |
| 08:24.08 | aurs | schmidts, I do however have 4 calls now that I'm going to search my verboselogs for |
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| 08:28.51 | aurs | schmidts, and when I check the first of the 4 calls: user error :D |
| 08:33.35 | schmidts | ;) |
| 08:34.25 | schmidts | if this really only happens 2 or 3 minutes after the start time then its possible that some kind of event has not been fired to update the time, but thats something i have to take a look at |
| 08:34.51 | aurs | I've seen that happen before, so I was.. what is it called.. I jumped to conclusions |
| 08:35.31 | aurs | but the last time I saw that bug was on a 1.4 installation, so it might be different now |
| 08:36.11 | schmidts | aurs not necessary ;) if nobody noticed a bug in this section its possible that this part of code is unchanged since then |
| 08:36.14 | aurs | the verboselog from the forum thread shows it clearly. I'll let you know if I can reproduce it on 1.8 |
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| 08:41.02 | schmidts | aurs i have checked the code for gotoiftime and it uses ast_tvnow which gives back the actual time so there is no problem with some code part which waits for anything |
| 08:41.35 | aurs | ok. going to do a quick test now |
| 08:41.42 | aurs | in T-4 minutes :D |
| 08:45.48 | aurs | interesting |
| 08:46.46 | aurs | ok, we might have a bug here schmidts |
| 08:46.55 | aurs | just going to check my logs |
| 08:47.33 | schmidts | have you enabled warnings? do you see this warning: "Using current time to evaluate" |
| 08:49.26 | schmidts | aurs if possible please add this before the GotoIfTime: Noop(Testtime: "${TESTTIME}" strlen: ${LEN(${TESTTIME})}) |
| 08:50.03 | schmidts | cause i dont know why this is there but there is a way to set the seconds value using this TESTTIME var for a gotoIftime |
| 08:50.32 | aurs | schmidts, ok. I can do that on my next test. just cross-checking with timestamp from logs now |
| 08:54.11 | schmidts | aurs i think i have found the problem, or maybe where it comes from :D |
| 08:54.14 | schmidts | if (!(i->minmask[tm.tm_hour * 2 + (tm.tm_min >= 30 ? 1 : 0)] & (1 << (tm.tm_min >= 30 ? tm.tm_min - 30 : tm.tm_min)))) |
| 08:54.34 | schmidts | this is the code part which does the check if the minute matches |
| 08:56.32 | aurs | schmidts, I had 0800-0945 defined as "day" in my test now. and I jumped to "day" until 0946 |
| 08:56.38 | aurs | schmidts, so it is off by one? |
| 08:57.15 | aurs | schmidts, I'm also having a hard time understanding that line of code :D |
| 08:57.17 | schmidts | aurs i havent really understand what this if does exactly |
| 08:57.19 | schmidts | :D |
| 08:57.24 | aurs | lol |
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| 08:58.46 | schmidts | aurs ok i think i understand it now. minutes are saved as a bitmask and you need 30 bit for one hour ;) |
| 08:59.03 | WIMPy | Or rather 64 bits per two hours. |
| 08:59.22 | WIMPy | That certainly looks interesting. |
| 08:59.34 | schmidts | WIMPy http://doxygen.asterisk.org/asterisk1.8/pbx_8c.html#75160d417c742f4c432dbccbc6d14d4a have fun :P |
| 08:59.46 | WIMPy | So do you get per minute accuracy on a 64 bit system? |
| 09:00.01 | WIMPy | doen't think so. |
| 09:00.31 | aurs | the server I did the test on is on a 64bit os |
| 09:00.35 | aurs | if that matters |
| 09:00.58 | aurs | has more skills of finding bugs than fixing them :D |
| 09:02.27 | WIMPy | No. It's 2x32bit hour. |
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| 09:03.35 | schmidts | WIMPy exactly its nsigned int minmask [48] |
| 09:03.39 | schmidts | +u |
| 09:04.12 | WIMPy | It might just be about the end time being inclusive. |
| 09:04.43 | WIMPy | So -0946 menaing -09:46:59? |
| 09:05.04 | schmidts | seconds are not checked ;) |
| 09:05.07 | aurs | WIMPy, I wouldn't bet on it, but yeah, it might be |
| 09:05.28 | aurs | but I think that I've seen this happen with more than 1 min wrong |
| 09:05.33 | aurs | (meaning 2) |
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| 09:06.28 | indra | Can pridialplan,prilocaldialplan be set on a group? or is it a global setting? |
| 09:06.56 | WIMPy | Per channel |
| 09:07.46 | aurs | I can do another test. when I tested now, I got wrong profile on times: 09:45:06, 09:45:20, 09:45:58 |
| 09:07.55 | aurs | and correct on 09:46:29 |
| 09:08.02 | aurs | (end time was 0945) |
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| 09:15.38 | schmidts | aurs which would be correct so far |
| 09:28.51 | aurs | schmidts, GotoIfTime("08:00-09:45,*,*,*?day") should return false after 0945 but it returns true |
| 09:29.10 | WIMPy | define "after". |
| 09:29.30 | aurs | WIMPy, I use the same logic as in 00/24 |
| 09:29.42 | aurs | :) |
| 09:30.32 | WIMPy | 09:45:something is 09:45, neither before nor after. |
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| 09:31.14 | aurs | select * from cdr where calldate > '2013-03-05 09:45'; |
| 09:32.22 | WIMPy | Now do that on a field that doesn;t contain seconds. |
| 09:32.57 | aurs | I see |
| 09:33.51 | WIMPy | Everything depends on definitions. |
| 09:33.56 | aurs | agree |
| 09:36.06 | aurs | but I'm saying that the logical thing would be to have 0845-0945 actually mean 084500-094459... not sure if I describe it clearly, but.. |
| 09:36.15 | aurs | from docs: If the current time is within the |
| 09:36.15 | aurs | given time specification, the channel will continue at <labeliftrue>. |
| 09:36.22 | aurs | "within the given time specification" |
| 09:37.19 | aurs | but let me see if I can get more than 1 min "wrong" |
| 09:37.22 | WIMPy | It would make sense to argue that "within" implies inclusive. |
| 09:37.37 | aurs | if it is just 1 minute I can add +1 to all end times, no problem |
| 09:37.51 | aurs | breaks my internal logic, but it would work at least :D |
| 09:37.58 | WIMPy | sign error |
| 09:38.02 | aurs | so |
| 09:38.10 | aurs | when a shop has opening hours 08-16 |
| 09:38.22 | aurs | you expect them to have open until 1601? |
| 09:38.33 | aurs | doesn't make sense to me |
| 09:38.56 | aurs | or rather 1659 |
| 09:39.01 | aurs | ;) |
| 09:39.03 | WIMPy | No, if they only give hours, no minutes I'd expect them to be open until 16:59:59. |
| 09:39.30 | aurs | hehehe |
| 09:40.02 | aurs | now you're just trolling sir ;) |
| 09:40.03 | WIMPy | Open until 16, so it's open AT 16, ot in that hour. |
| 09:40.31 | WIMPy | But off course we know they will close at 15:58, because they are lazy :-) |
| 09:41.12 | WIMPy | No, I'm just taking the logic route without social error correction. |
| 09:41.35 | aurs | "until" does not include |
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| 09:42.07 | WIMPy | Correct. My mistake. |
| 09:42.14 | aurs | or... |
| 09:42.40 | aurs | now I'm confused WIMPy :P |
| 09:42.43 | WIMPy | Maybe not. |
| 09:43.24 | WIMPy | definitions |
| 09:43.33 | aurs | but I will do some more tests and see if this "off by 1" is consistent, or if I can get more than 1 minute "wrong" (according to me) :D |
| 09:43.34 | WIMPy | Evil stuff. |
| 09:43.43 | WIMPy | And languages are far too often too vague. |
| 09:44.53 | aurs | anyway, when I read the doc on gotoiftime, I expect 0800-1600 to return false at 1600 |
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| 09:45.36 | aurs | but if it is consistent I'm happy. *back to testing* |
| 09:45.49 | WIMPy | I still think that "within" is more explicit than "until" in saying that it's an inclusive range. |
| 09:46.06 | WIMPy | Definitely not intuitive, though. |
| 09:53.22 | aurs | I would say that it should be <= start time and < end time |
| 09:53.27 | aurs | no |
| 09:53.34 | WIMPy | "between" would be exclusive. Putting a mixed inclusive/exclusive range in to human language seems harder. |
| 09:53.43 | aurs | make that >= start time |
| 09:55.07 | aurs | yeah, see your point |
| 10:03.43 | aurs | seems pretty consistent |
| 10:04.26 | ectospasm | ghost75: sorry I missed you. I wouldn't bank on me paying attention to IRC when you need me, better ask your questions to everyone in the channel. |
| 10:05.58 | ghost75 | just wanted to ask if you done a post in debian forum about bluetooth |
| 10:07.15 | ghost75 | http://forums.debian.net/viewtopic.php?f=7&t=63892 |
| 10:08.17 | skrusty | morning |
| 10:08.42 | ghost75 | is this you ectospasm? |
| 10:16.55 | ectospasm | That is me. |
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| 10:19.02 | ghost75 | did you only update kernel? |
| 10:19.22 | ghost75 | i have squeeze with new kernel but bluetooth device doesnt connect (pairing is ok) |
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| 10:31.06 | ectospasm | ghost75: yeah, by the time I fixed it I had installed wheezy. I'm on sid now, and don't remember the last time I had that particular problem. |
| 10:32.17 | ghost75 | so maybe bluez is the problem (which is a crap anyway) |
| 10:34.58 | Bradada | Need a simple answer, could asterisk do IVVR only by ifself? |
| 10:44.05 | ectospasm | Bradada: Asterisk can do an IVR "by itself" |
| 10:44.50 | Bradada | Sorry, I mean video IVR. |
| 10:47.00 | ectospasm | first I've heard of something like that. Wouldn't you be better off with a web page and web browser? |
| 10:47.06 | *** part/#asterisk grummund (~user@unaffiliated/grummund) |
| 10:50.19 | *** join/#asterisk Menti (9c23dea5@gateway/web/freenode/ip.156.35.222.165) |
| 10:51.05 | Menti | Hi guys, is there anybody here to answer a little question? i googled it but i can't find the answer.... :/ |
| 10:52.10 | schmidts | Menti just ask the question ;) |
| 10:52.11 | ghost75 | no |
| 10:52.26 | ectospasm | ~ask |
| 10:52.26 | 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. |
| 10:52.30 | Bradada | ectospasm: Okay, I'll think about that, anyway thanks. |
| 10:52.44 | schmidts | btw does anybody knows how to make the tilde "~" on a mac? :P |
| 10:53.08 | aurs | schmidts, mark it and press cmd+c :P |
| 10:53.18 | ectospasm | heh |
| 10:53.19 | schmidts | aurs thx :D |
| 10:53.24 | aurs | lol |
| 10:53.45 | aurs | schmidts, but yes I do, but only on norwegian keyboard layout |
| 10:54.55 | Menti | well, i installed Asterisk following the Asterisk definitive guide, and i made a netstat -tuplv... i see some asterisk services (iax, sip etc.....) i tried to register a sip account but i don't get any traffic on port 5060 |
| 10:56.00 | Menti | all the config of the softphone and accounts was made following the "quick start guide" from Asterisk main web btw |
| 10:56.10 | schmidts | Menti any firewall in between your sip client and the server like iptables? |
| 10:56.19 | aurs | Menti, you should be able to see the traffic even if asterisk is not running, so it sounds like a client issue? |
| 10:56.40 | aurs | or network |
| 10:57.44 | Menti | there's only windows firewall here in this computer, Asterisk is installed in a RaspberryPi, |
| 10:59.55 | Menti | i tried Zoiper and sJphone and i didn't get any traffic of any of it (tcpdum running on the Raspberry Pi) |
| 11:01.34 | aurs | Menti, does a icmp ping show in your tcpdump on the pi? |
| 11:01.42 | aurs | ...if you ping it |
| 11:05.15 | Menti | aurs, O__O i cant make a ping (with windows) to pi's IP, but.... i'm connected via SSH with the pi, this is quite strange :S |
| 11:20.09 | aurs | Menti, might be firewalled, but you should see it in tcpdump anyway (the ping) |
| 11:20.35 | aurs | Menti, so perhaps your tcpdump is running on the wrong interface? |
| 11:24.00 | Menti | aurs, tcpdum is running on the pi, and i tried to ping the pi from my windows. the pi is at home and i'm at university, i'll try to avoid the firewall, and the tcpdump is running on the correct interface (eth0) |
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| 11:36.36 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:46.02 | Menti | aurs, no, really can't ping my router, should i put the router "naked" ? i mean....should i open all the traffic? |
| 11:51.50 | *** join/#asterisk Neo31 (~nexus31@unaffiliated/neo31) |
| 11:55.27 | Neo31 | hello, (sorry for the newbie question) I have an intranet asterisk server and I need to connect it to PSTN so intranet users could recieve/emit calls from and to PSTN. what hardware architecture do I need to use with the asterisk server? (does this do what I need? http://www.amazon.com/OBi100-Telephone-Adapter-Service-Bridge/dp/B004LO098O/ref=sr_1_1?ie=UTF8&qid=1362483827&sr=8-1&keywords=ata ) thanks |
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| 12:07.46 | *** join/#asterisk jkroon (~jkroon@dsl-244-34-204.telkomadsl.co.za) |
| 12:07.56 | jkroon | hi guys, picked up a segfault in pbx_dundi |
| 12:08.18 | jkroon | in rescomp, both a and b are being passed in as NULL, any ideas? |
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| 12:34.34 | ghost75 | https://issues.asterisk.org/jira/browse/ASTERISK-13145 since which versions this is fixed |
| 12:35.59 | *** join/#asterisk davlefouAMD (~david@197.15.50.19) |
| 12:36.10 | schmidts | ghost75 hmm if i read it right, but i only looked fast about the topics and the bottom, it isnt fixed and released by now |
| 12:36.51 | ghost75 | somebody here wrote that it was working in 11 (not sure which version) |
| 12:38.37 | schmidts | ghost75 atleast there is a patch in there for the 11 version and if a part of this patches were going into the code base you could see it anywhere in the issue itself |
| 12:41.28 | schmidts | ok now i remember this stuff but imho it hasent been submitted, cause the review for this code was too big |
| 12:41.57 | ghost75 | the patch is quite big |
| 12:42.43 | schmidts | yes and matt jordan wrote on review board that gareth should split it up into smaller pieces, but this never happend |
| 12:43.34 | ghost75 | (13:07:41) eirirs: I got it working by patching 1.6 , but upgraded to 11, and BLF still works, no change, and feature buttons still don't work |
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| 13:00.43 | jkroon | hi guys, found a segfault in pbx_dundi. gdb for the win |
| 13:00.52 | jkroon | ability to print stack frames are AWESOME |
| 13:01.03 | jkroon | with or without debug info |
| 13:03.16 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.100.176) |
| 13:08.50 | *** join/#asterisk Tagor (~Tagor@s55978a13.adsl.online.nl) |
| 13:08.52 | Tagor | Is there a way to change the Contact-header so that it sends 'number@my.ip' instead of 'asterisk@my.ip'? |
| 13:10.32 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:10.33 | ghost75 | mwi_from ? |
| 13:13.10 | schmidts | Tagor do you want allways the same number or a different one? Cause if its allways the same you can set the user_agent value in sip.conf |
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| 13:22.54 | Tagor | schmidts: It's always the same, so I can just set user_agent=1234567? |
| 13:24.33 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-nklfmgmeuphppnbn) |
| 13:25.15 | aurs | Tagor, is "callerid=1234567" in sip.conf what you're looking for? |
| 13:34.13 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 13:37.38 | Tagor | aurs: tried it, but doesn't work |
| 13:37.40 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 13:38.44 | [TK]D-Fender | Tagor, show us |
| 13:42.34 | *** join/#asterisk leifmadsen (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:42.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:42.36 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 13:44.27 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 13:45.07 | Tagor | [TK]D-Fender: hold on, I'm waiting for the log from them. They say the Contact-header they receive contains 's@my.ip.add.ress', Asterisk shows a Contact-header that contains 'asterisk@my.ip.add.ress'. They want me to send a Contact-header that contains '1234567@my.ip.add.ress' |
| 13:46.03 | [TK]D-Fender | Tagor, Are you referring to a CALL or a REGISTER? |
| 13:46.08 | [TK]D-Fender | Tagor, and ... SHOW US |
| 13:46.11 | danfromuk | Does anyone know of a non-java based voip packet speed tester? Like this but not java. http://myspeed.visualware.com/indexvoip.php |
| 13:46.59 | Tagor | [TK]D-Fender: I don't have the logs yet, they said they will send them in a minute. I'm referring to a REGISTER |
| 13:48.21 | [TK]D-Fender | Tagor, Who is "they". Isn't this YOUR server? |
| 13:52.43 | Tagor | [TK]D-Fender: The SIP provider. Just got the logs, here's the REGISTER that they receive: http://pastebin.com/yVvvRR6i |
| 13:53.23 | Tagor | [TK]D-Fender: Line 11 should be '<sip:00000001@my.ip>' |
| 13:53.44 | [TK]D-Fender | Tagor, Where is YOUR SIP debug? You don't need them to show you this. And your register is not specifying the EXTENSION on the end. |
| 13:54.16 | [TK]D-Fender | Tagor, REGISTER => user:pass@hotst/IFYOUDON'TPUTYOUREXTENSIONHEREYOU'LLSENDTHEM"S" |
| 13:55.10 | *** join/#asterisk serafie (~erin@nat/digium/x-iasscynfakqnxeut) |
| 13:55.13 | Tagor | [TK]D-Fender: I have multiple DIDs on 1 SIP trunk. Someone here told me I shouldn't define the extension |
| 13:55.40 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.100.176) |
| 13:55.53 | Tagor | [TK]D-Fender: when I define the extension, is there a way to route the calls based on the DID that was called? |
| 13:56.05 | [TK]D-Fender | Tagor, this is your REGISTER. that is the ONLY place to set it |
| 13:56.15 | Tagor | [TK]D-Fender: or should I extract it from the To-header? |
| 13:56.19 | [TK]D-Fender | Tagor, Set it right and look at the CALL they send you |
| 13:57.58 | Tagor | [TK]D-Fender: Sorry, I don't get it. I have one SIP trunk (1 register line) with 10 DIDs (000001-000010). When I use 'register => ,,,,,/000001 all calls go to extension 000001 |
| 13:58.23 | [TK]D-Fender | Tagor, Last time : Look. At. The. Call. |
| 13:59.14 | Tagor | [TK]D-Fender: I did, but as far as I can see only the To-header contains the dialed DID |
| 13:59.29 | [TK]D-Fender | Tagor, ... so that's the answer |
| 13:59.41 | hal-j | morning folks |
| 13:59.41 | [TK]D-Fender | Tagor, get GET it and use the number |
| 13:59.48 | [TK]D-Fender | go* |
| 14:00.02 | [TK]D-Fender | Tagor, "core show function SIP_HEADER" |
| 14:00.12 | Tagor | [TK]D-Fender: Ok, so there's no easier way than extracting it from the To-header? |
| 14:00.29 | *** join/#asterisk leifmadsen (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:00.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:01.58 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 14:01.59 | [TK]D-Fender | Tagor, There is not. You have to do this in the dialplan. many providers like CallCentric, etc work like this. |
| 14:02.23 | Tagor | [TK]D-Fender: Ok, many thanks! |
| 14:03.47 | as001 | Hello I tried to activate asterisk 11 built in http server but I can't see anything in browser with http://127.0.0.1:8088 .Which folder does that http server search for ? |
| 14:04.23 | as001 | I try to open it from localhost (same machine where asterisk is running) |
| 14:04.27 | [TK]D-Fender | as001, What are you hoping to use it for out of curiousity? |
| 14:04.52 | as001 | I try to follow web rtc instructions. |
| 14:04.59 | *** join/#asterisk hbonath (~hbonath@rrcs-98-102-153-237.central.biz.rr.com) |
| 14:05.10 | as001 | it requires ws subfolder |
| 14:18.13 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
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| 15:17.02 | hal-j | for those curious (which I'd imagine is somewhere between 0 and 1) - I got incoming GV working again last night |
| 15:17.59 | hal-j | won't _swear_ it was the upgrade to 11, since it didn't work immediately after the update either, but after fiddling with GV settings (on google's side) and returning all settings but one (global spam protection) back to their original state, then restarting asterisk, it worked |
| 15:18.42 | hal-j | but since it worked previously at some point (with global spam protection enabled at the time) I suspect none of the above was actually an issue, but [something] ended up being reset during the dance |
| 15:22.43 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
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| 15:40.17 | Ice_Strike | Yo |
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| 15:49.56 | *** join/#asterisk Yeretik (~Yeretik@nixe.com) |
| 15:50.33 | Yeretik | Can i use asterisk to videoconferences with the app_meetme module? |
| 15:50.55 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:50.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:51.17 | Yeretik | I mean, enable video in a meeting room |
| 15:54.43 | Yeretik | Can anybody tell me if it's possible enable conference rooms in asterisk with video support ? |
| 15:55.15 | [TK]D-Fender | yES, IN "FOLLOW THE SPEAKER" ONLY |
| 15:55.31 | [TK]D-Fender | And I'd recommend ConfBridge for this. |
| 15:55.49 | Yeretik | follow the speaker ? |
| 15:56.20 | Yeretik | it's a mode? |
| 15:56.31 | [TK]D-Fender | No, it's a fact of life |
| 15:56.44 | [TK]D-Fender | as in no multi-party, multi-window |
| 15:56.57 | [TK]D-Fender | Everyone gets the video of ONLY the person talking |
| 15:57.44 | Yeretik | It depends of the client... No? |
| 15:57.52 | [TK]D-Fender | No |
| 15:58.22 | *** join/#asterisk alexv_ (~alexv@173.231.102.254) |
| 15:58.24 | _Corey_ | Digium had some multi-party thing going on with Dialogic a while ago... not sure if it's still around but it looked decent. (Commercial though) |
| 16:01.20 | Yeretik | Hmmm |
| 16:04.53 | *** join/#asterisk navaismo (~navaismo@189.191.199.181) |
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| 16:05.06 | alexv_ | Does anybody know a way to modify To: header to reflect the domain name? Thank you. |
| 16:08.54 | *** join/#asterisk polysics (~Adium@host183-69-dynamic.2-87-r.retail.telecomitalia.it) |
| 16:09.55 | polysics | what's the correct syntax to dial a specified priority using a Local channe, please?ù |
| 16:10.36 | [TK]D-Fender | polysics, There is no such thing. It starts at 1. Always |
| 16:11.37 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:13.06 | *** join/#asterisk x1101 (~x1101@cpe-72-224-119-84.maine.res.rr.com) |
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| 16:15.41 | polysics | ah ok, thanks |
| 16:21.56 | *** join/#asterisk Uthark (~Uthark@190.0.58.186) |
| 16:21.59 | BludSuckingFiend | ls -alh |
| 16:22.05 | BludSuckingFiend | err... oops |
| 16:22.29 | BludSuckingFiend | not the worst thing I could have typed into the wrong window I guess |
| 16:22.45 | x1101 | hello everyone. I have what is probably an newbie question about agents/groups/queues setup |
| 16:24.07 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 16:25.29 | x1101 | I am trying to tidy up some config for a call center with 20+ agents and 60+ queues using groups, and the doc is a bit confusing. |
| 16:25.52 | *** join/#asterisk ghost75 (~trechber@dslb-092-075-062-069.pools.arcor-ip.net) |
| 16:26.20 | BludSuckingFiend | how so? |
| 16:27.59 | x1101 | What I want is to define a group in my agents.conf (I've done this, and Asterisk throws erros when trying to load it) and put that group in queue with priority 1 (ring any member avaliable ) and another with priority 2 (ring any member avalible if all group 1 agents are busy). The doc I'm finding is a bit unclear as to how to do this |
| 16:29.15 | BludSuckingFiend | and what's the error? |
| 16:29.42 | x1101 | ERROR[26091]: channel.c:4866 ast_get_group: Syntax error parsing group configuration 't1' at 't1'. Ignoring. |
| 16:30.10 | x1101 | I am running 1.4.31 (and upgrading isn't really an option at this time) |
| 16:31.11 | BludSuckingFiend | can you pastebin your agent config? (sanitized of course) |
| 16:31.49 | BludSuckingFiend | what is t1? |
| 16:32.18 | BludSuckingFiend | because groups can only be numbers |
| 16:32.23 | BludSuckingFiend | assuming t1 is a group |
| 16:34.38 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 16:35.56 | x1101 | BludSuckingFiend: that would do it then |
| 16:36.09 | x1101 | I didn't see anywhere that noted groups could *only* be numbers |
| 16:36.29 | x1101 | I am working on pastebin'ing my config and sanatizing it, but got called away, back in a mo |
| 16:36.46 | BludSuckingFiend | ok. |
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| 16:41.04 | Qwell | ~upgrade asterisk |
| 16:41.04 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 16:41.21 | Qwell | x1101: Not upgrading also isn't an option. |
| 16:41.48 | BludSuckingFiend | Eh, sometimes not upgrading is the only option. |
| 16:42.20 | BludSuckingFiend | There are situations where it's not practical |
| 16:43.13 | Qwell | http://www.asterisk.org/downloads/security-advisories says otherwise. |
| 16:43.16 | x1101 | Qwell: well, if you can put AgentCallbackLogin back in, and convince my management team that an upgrade is worthwhile I would be overjoyed to do it. Otherwise, Im stuck where I am |
| 16:43.28 | Qwell | AgentCallbackLogin is crap. Don't use it. |
| 16:45.04 | BludSuckingFiend | everything has vulnerabilities. The trick is being aware and reducing the surface area |
| 16:45.30 | x1101 | Qwell: crap or no, our call center flow is based around it. |
| 16:45.30 | BludSuckingFiend | If you can adequately protect the version you're on given known vulnerabilities, then "if it aint broke, don't fix it" |
| 16:45.54 | x1101 | BludSuckingFiend: http://pastebin.com/sENDqBzw << relevant portion of agents.conf and queues.conf |
| 16:45.56 | Qwell | x1101: You're going to need to hire a consultant then. We can't help you here. |
| 16:46.00 | Greenlight | BUt at the point people are here asking for assistance, then it "is broke" ? |
| 16:46.02 | BludSuckingFiend | I recently upgraded from 1.8.1 to 11 because we were able to. |
| 16:46.19 | BludSuckingFiend | Doesn't mean anything is broke Greenlight |
| 16:46.29 | BludSuckingFiend | people looking for advice don't need to upgrade |
| 16:46.34 | hal-j | keeping up to date with anything actively used in production, is a two edged sword |
| 16:46.40 | Greenlight | Yea, I see people all day loggin in here and saying "Hi guys, no problems, everything's working today!" |
| 16:46.41 | Qwell | BludSuckingFiend: They do if they want help here. |
| 16:46.55 | hal-j | if you do it regularly, you're doing it a lot |
| 16:47.13 | kaldemar | x1101: try a numeric group. |
| 16:47.15 | hal-j | but the flip side is that most upgrades are evolutionary not revolutionary… and only small config changes are needed. they're MUCH easier upgrades |
| 16:47.19 | BludSuckingFiend | No, they really don't. If someone comes in asking for config assistance as x1101 did, version is irrelevant if nothing has changed |
| 16:47.25 | Greenlight | It's like complaining that Office 2012 doesn't work on Windows 95 |
| 16:47.30 | Qwell | BludSuckingFiend: The code he's using *no longer exists*. |
| 16:47.47 | BludSuckingFiend | You mean that Agents.conf no longer supports groups? |
| 16:48.01 | hal-j | but at the same time, not a single person who's ever managed production systems of any kind, has had the resources to keep everything 100% up to date :D |
| 16:48.05 | BludSuckingFiend | because that was his question. Did that fix your issue btw x1101 ? |
| 16:48.53 | x1101 | Qwell: its folks like you that give FLOSS such a bad name. both BludSuckingFiend and kaldemar offered me assistance (which looks like the correct answer) rather than giving me shit about being stuck on an old version of software. Do you think I *want* to be ~3yrs out of date? Our work flow relies on a specific function that the Asterisk developers decided to remove because it could be, and I quote 'duplicate in the dialplan'. |
| 16:49.03 | x1101 | BludSuckingFiend: I am going to check now. Thank you for trying to help! |
| 16:49.09 | BludSuckingFiend | Qwell: I can see if he said something was broken how being on the latest version would matter. But you're making a fuss over someone that just asked a general configuration question |
| 16:49.27 | BludSuckingFiend | np x1101 |
| 16:49.28 | Qwell | x1101: That's great. It sucks you're stuck. But we can't spend the time here helping with old crap. Other people also require help. |
| 16:49.42 | *** part/#asterisk polysics (~Adium@host183-69-dynamic.2-87-r.retail.telecomitalia.it) |
| 16:52.12 | x1101 | Qwell: There was no traffic in the channel when I asked the question, and I havent seen anything since other than BludSuckingFiend and kaldemar trying to help, and you being an ass about versions. So Im sorry, but you're point is invalid. |
| 16:53.20 | Greenlight | Supporting outdated versions just leads to confusion all around, and quite often wastes peoples time, I think that's the point he's trying to make |
| 16:53.55 | Qwell | The point is simple. Follow the rules of the channel. |
| 16:53.57 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 16:54.01 | BludSuckingFiend | Greenlight: I understand his point and it would be valid if the question was about some bug or something. This was just about config file syntax, the version wasn't really relevant |
| 16:58.36 | Qwell | BludSuckingFiend: In some configs, his syntax is now valid. |
| 16:59.59 | BludSuckingFiend | Qwell: If you want to refuse to answer questions for that reason fine... but there's no need to interfere with someone else trying to assist. |
| 17:03.31 | Qwell | Feel free to help him in private messages. |
| 17:04.38 | BludSuckingFiend | I'll assist wherever I please, unless you intend to eject me for helping people :) |
| 17:08.28 | Greenlight | BludSuckingFiend: And what if you choose to assist with an outdated version, or feature, or syntax, which someone else then sees (this is a public channel) and tries to use or thinks is valid? Rules are there for a reason. The number of times I've seen the same people come in time and time again regarding a bug for an outdated version is astounding. |
| 17:10.01 | BludSuckingFiend | Then at least it would be less of a waste of chat lines than chastising someone for being stuck on an old version unnecessarily |
| 17:11.18 | BludSuckingFiend | This whole conversation is kind of pointless so I'm going to drop it. |
| 17:11.23 | drmessano | So how do you justify helping someone for 30 mins, an hour, 3 hours, when their issue is resolved in an update or two release that's 2 years newer than the 4 year old version they are running? |
| 17:11.36 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 17:12.03 | Greenlight | I hear #windows95 is in need of people to help with support ... :) |
| 17:14.02 | drmessano | TBH, 1.4.31 is NOT the most recent of the now-extinct 1.4 branch, it's 1.4.44. So "Help me with an issue on an unsupported branch where I am not even updated to the latest release, and don't give me no shit!" ??? |
| 17:14.30 | drmessano | FOSSPP |
| 17:15.09 | drmessano | drops the microphone |
| 17:18.37 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.80) |
| 17:25.50 | Greenlight | Is there much overhead using a frame hook on all channels? The hook itself is only checking for a single frame type, and ignoring the rest. Just wondering if it's worth having the hook destroy itself, or if it'll not make much different. Sytem has around 400-500 channels at any one time. |
| 17:25.56 | *** join/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 17:26.07 | *** part/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 17:38.07 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 17:39.05 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 17:39.17 | kuku | IS there a way to identify if hte call was hanged up by the trunk or by the extension ? |
| 17:40.02 | Qwell | kuku: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=20185363 |
| 17:40.26 | Qwell | I think. |
| 17:42.35 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 17:46.56 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 17:49.46 | igcewieling | kuku: yes. Excatly how depends on your version of Asterisk |
| 17:50.25 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-agdpfhaigbucrphq) |
| 17:51.38 | *** join/#asterisk camerin (hoax@newelite.bshellz.net) |
| 17:51.56 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 17:52.17 | jmetro | is it possible to use a variable with a variable? |
| 17:52.30 | Qwell | jmetro: ${${VAR}} ? |
| 17:53.05 | jmetro | Yes, so essentially, i can decide what variable to use by passing it to the sub. like ${${arg1}} and arg1 would contain a variables name |
| 17:53.16 | Qwell | sure, why not? |
| 17:53.24 | jmetro | Some languages dont work like that =) Thanks |
| 17:55.54 | kuku | I have version 1.8 |
| 17:57.21 | igcewieling | kuku: put the "g" option on Dial. The dialplan will continue to the next dialplan priority if the DEST hangs up. If the SOURCE hangs up then the dialplan will jump to exten => h in the dialplan. |
| 17:57.53 | igcewieling | in Asterisk 11 you can make hangup handlers for each leg of the call. |
| 17:58.08 | kuku | Is there a way to add sound to the recording so we know who hanged up ? |
| 17:58.42 | kuku | Or are you saying after I add a g, then I can add the beep and hangup to it ? |
| 17:58.51 | igcewieling | the recording? |
| 17:59.18 | igcewieling | yes, to play a beep to the caller when the destination hangs up |
| 18:03.19 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 18:03.26 | jmetro | hm.. further question |
| 18:03.51 | jmetro | ${ARG1} is 127 and i want this to evaluate to "jmetro-Title" |
| 18:04.23 | jmetro | I have// 127=jmetro |
| 18:04.38 | jmetro | and Verbose(${${ARG1$}}-Title) |
| 18:05.02 | [TK]D-Fender | $variables can't start with a digit last I checked |
| 18:05.31 | jmetro | SO i would have to do ${name${ARG1}} and then name127 = jmetro |
| 18:05.33 | [TK]D-Fender | "core show function EVAL" <- |
| 18:06.23 | *** part/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 18:06.35 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 18:09.02 | jmetro | hm |
| 18:09.12 | jmetro | ${name${arg1}} still evals to "" |
| 18:09.52 | kuku | igcewieling: What about inbound calls - how do I found out who hanged up in that situation ? |
| 18:10.06 | igcewieling | (12:57:21 PM) igcewieling: kuku: put the "g" option on Dial. The dialplan will continue to the next dialplan priority if the DEST hangs up. If the SOURCE hangs up then the dialplan will jump to exten => h in the dialplan. |
| 18:10.41 | kuku | inbound calls. |
| 18:12.46 | igcewieling | define "inbound call" all your calls have an inbound leg |
| 18:13.09 | igcewieling | when a call comes into asterisk it is an inbound call. does not matter where it comes from. therefore all your calls are inbound calls. |
| 18:13.31 | kuku | it comes from a trunk, hits a ring group, an extension picks up |
| 18:13.32 | igcewieling | now, how do you define "inbound call" |
| 18:13.48 | kuku | that is how i define inbound call |
| 18:13.55 | igcewieling | you'll have to handle that in your dialplan, since the distinction is entirely artificial |
| 18:14.20 | igcewieling | I do it by checking ${CHANNEL(peername)} to get the name of the peer and compare it against a list of "trunk" peers. |
| 18:15.17 | igcewieling | you could also put something in the sip.conf peer entry like setvar=IS_TRUNK=yes and check to see if IS_TRUNK is equal to "yes" |
| 18:17.33 | kuku | and it wont be there if it goes through the trunk on the outbound call ? |
| 18:19.54 | *** join/#asterisk Tagor (~Tagor@s55978a13.adsl.online.nl) |
| 18:20.06 | Tagor | Can extension names contain a dot? |
| 18:21.11 | [TK]D-Fender | "core show function EVAL" <- |
| 18:21.20 | igcewieling | Tagor yes, as long as you don't put it in a pattern (i.e. _ prefix) |
| 18:21.35 | [TK]D-Fender | [.] <- |
| 18:37.00 | jmetro | D-Fender so you are saying i should eval the statement rather than just stick it in there like i have it |
| 18:44.15 | [TK]D-Fender | jmetro, What do you think? </psych101> |
| 18:51.18 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 18:57.37 | mcf3782 | I need some help diagnosing some log messages on my Asterisk 1.8.3.2 system. It appears that someone is getting through the firewall to my PBX, but I'm not finding enough log information to give the firewall guys enough information to go on. |
| 18:58.06 | mcf3782 | A sample of the log messages and CLI output is in this pastebin file… http://pastebin.com/Txbn6nVd |
| 18:59.25 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.149.47) |
| 18:59.56 | mcf3782 | This system has one POTS line plugged into a Digium TDM400 card, and one Google Voice number configured as SIP connected to it. |
| 19:00.32 | mcf3782 | I can't figure out how/where this traffic is originating, and why the log messages don't show an IP address for where the traffic is coming from. |
| 19:03.12 | *** join/#asterisk Defraz (~Defraz@67.60.210.130) |
| 19:04.23 | navaismo | so you have the allowguest=yes and the context allowed for guest can dial out? bad bad |
| 19:06.54 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 19:07.53 | mcf3782 | navaismo - gtalk.conf does have "allowguests=yes" in it. |
| 19:08.17 | mcf3782 | I will go read up on that. |
| 19:13.03 | *** join/#asterisk nanoha-sama (~nanoha-sa@d216-232-208-44.bchsia.telus.net) |
| 19:13.04 | *** join/#asterisk TriJetScud (~TriJetScu@d216-232-208-44.bchsia.telus.net) |
| 19:13.37 | Tagor | [TK]D-Fender: Thanks again! |
| 19:17.08 | navaismo | mcf3782, that setting isn't for gtalk is for your sip configuration, if you want to see origin IP you need to check the source of chan_sip.c and add/change the (%s) or update to asterisk version >1.8.5 |
| 19:18.33 | navaismo | and you will see--> chan_sip.c:25184 handle_request_invite: Call from '' (10.0.1.102:5060) to extension '25' rejected because extension not found in context 'guest'. |
| 19:19.06 | mcf3782 | navaismo - the only place in /etc/asterisk where "allowguests=yes" is in gtalk.conf. It doesn't apper in sip.conf, not even commented out. |
| 19:20.15 | mcf3782 | Thanks for the pointer about upgrading to 1.8.5. I know this system is way behind on being on current code. I just can't seem to get caught up. :( |
| 19:20.19 | [TK]D-Fender | mcf3782, You should explicitly set it to "no" |
| 19:20.45 | [TK]D-Fender | mcf3782, You should be able to jump up to the latest 1.8 release without having to change anything else... |
| 19:21.07 | mcf3782 | [TK]D-Fender - explicitly set it to "no" in sip.conf? |
| 19:21.34 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 19:21.57 | mcf3782 | Is the implication of that, that without it explicitly being set to "no"; that it defaults to "yes"? |
| 19:22.35 | [TK]D-Fender | mcf3782, Certainly looks like it... |
| 19:22.55 | igcewieling | mcf3782: the default for allowguest= and alwaysauthreject= is different in different Asterisk versions. |
| 19:23.27 | mcf3782 | I love consistency. ;) |
| 19:29.35 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 19:34.38 | igcewieling | mcf3782: Yup! new options usually default to whatever won't break stuff, once the option has been around for a while, if it is important to have it enabled, then it is enabled by default. |
| 19:34.59 | mcf3782 | :) |
| 19:35.05 | igcewieling | This is why you read the UPGRADE-*.txt files, they tell you about such issues. |
| 19:36.46 | mcf3782 | I'm glad you remember everything you read in UPGRADE-*.txt files for every application. :) I'm lucky if I can remember today, what I had for lunch yesterday. :-) |
| 19:37.15 | mcf3782 | speaking of lunch….does anyone know what I had for lunch yesterday? I don't want to have the same thing today…. ;) |
| 19:38.17 | igcewieling | mcf3782: I reread the UPGRADE-*.txt files about 3 times per year. |
| 19:38.59 | igcewieling | same for "core show applications" and "core show functions". I also read all three things any time I upgrade to a major new version of Asterisk. |
| 19:39.10 | igcewieling | oh, and read the sip.conf.sample files. too 8-) |
| 19:39.18 | *** join/#asterisk marksaitis (~marksaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 19:40.49 | navaismo | you are such a nerd |
| 19:42.52 | mcf3782 | 1.8.20.1 - here I come. |
| 19:43.22 | mcf3782 | coffee++ |
| 19:54.02 | igcewieling | navaismo: you have to be when you work in telecom. I also sometimes read the tarrifs. 8-) |
| 19:58.43 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:59.35 | mcf3782 | I hope app_swift still works with 1.8.20.1. :/ |
| 20:00.32 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 20:02.01 | hal-j | hmm — Any quick thing I should check to see why I'm seeing flapping like this: http://pastebin.com/3kRM6PYs |
| 20:02.37 | hal-j | the client is an iPad running linphone, plugged in and open (so it's not going to sleep) — that doesn't mean iOS isn't doing something odd though |
| 20:04.13 | hal-j | not sure if it's coincidental or not that of the 4 instances I picked, 3 are 11 seconds, gone-to-return |
| 20:09.55 | BludSuckingFiend | wifi is pretty notorious for unstable pings |
| 20:10.24 | BludSuckingFiend | Other than that... I'd run a ping from the asterisk box |
| 20:10.45 | BludSuckingFiend | to make sure it's the real ping and not just the SIP messages getting delayed |
| 20:17.16 | hal-j | yeah, everything is on wifi at the moment |
| 20:18.44 | [TK]D-Fender | hal-j, Increase your qualify time |
| 20:20.40 | hal-j | nah, looks like it's apple being silly |
| 20:20.43 | hal-j | is shocked |
| 20:21.21 | hal-j | ping times to everything else on the network is ~4ms. iPad is 300 one second, 1500+ the next |
| 20:21.51 | BludSuckingFiend | maybe your ipad is lazy |
| 20:22.21 | BludSuckingFiend | I am curious now though... I'll check ping times to mine |
| 20:22.23 | [TK]D-Fender | Qualify != Ping |
| 20:22.37 | [TK]D-Fender | Not the same. Stop comparing them |
| 20:22.39 | BludSuckingFiend | no, [TK]D-Fender , but it is closely related |
| 20:22.48 | BludSuckingFiend | they both measure response time |
| 20:22.53 | [TK]D-Fender | ... |
| 20:23.03 | [TK]D-Fender | they do not scale |
| 20:23.26 | BludSuckingFiend | they are both affected by network latency |
| 20:24.19 | [TK]D-Fender | Ping is not always the actual letncy. Every device has it's own proirity levels and processing capabilities. |
| 20:24.19 | Penguin | [TK]D-Fender: its |
| 20:24.32 | [TK]D-Fender | Then imagine that variably worse for OPTIONS. |
| 20:24.33 | hal-j | [TK]D-Fender: increase the time or increase the frequency |
| 20:24.44 | BludSuckingFiend | You're missing the point [TK]D-Fender |
| 20:25.02 | hal-j | it's a valid point. just because ICMP is delayed doesn't mean everything is |
| 20:25.06 | BludSuckingFiend | if you check ICMP response and it does the same thing as the SIP OPTIONS.... then you know it is network latency |
| 20:25.08 | hal-j | ICMP ping is typically a low priority |
| 20:25.08 | [TK]D-Fender | hal-j, Increasing the qreuency just gives you more chances to fail. |
| 20:25.21 | [TK]D-Fender | hal-j, if they are local, then there is little need |
| 20:25.22 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:25.26 | hal-j | they're local |
| 20:25.36 | [TK]D-Fender | hal-j, So just increase the time |
| 20:26.01 | [TK]D-Fender | hal-j, Since it's not like you're using them to keep a NAT'd port open |
| 20:26.10 | *** join/#asterisk polysics (~Adium@host22-155-dynamic.14-87-r.retail.telecomitalia.it) |
| 20:26.15 | [TK]D-Fender | frequency* |
| 20:26.23 | polysics | hi there |
| 20:26.55 | polysics | Dial(Local/1234@context/n) results in a weird pbx.c: No application '' for extension error |
| 20:27.06 | polysics | the context and extension exist, etc |
| 20:27.25 | hal-j | hm, looks like I've got qualify=yes and qualifyfreq=60 |
| 20:27.58 | hal-j | I'll set the "yes" to something more than the 2s "yes" seems to translate to |
| 20:31.24 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 20:32.26 | [TK]D-Fender | polysics, you've typo'd your extension. |
| 20:32.55 | [TK]D-Fender | hal-j, Freq isn't the real problem... TTR is |
| 20:33.08 | polysics | [TK]D-Fender: yeah, thanks. It was TOO weird :-) |
| 20:33.30 | [TK]D-Fender | polysics, This is usually a rather evident double comma type error |
| 20:33.59 | polysics | extra comma before priority label |
| 20:34.42 | hal-j | [TK]D-Fender: arguably, the real problem is my attempt to use the iPad as a desk phone |
| 20:38.00 | *** join/#asterisk vastersk (~vinscentp@124.6.136.142) |
| 20:39.11 | [TK]D-Fender | hal-j, it's a cool option actually.... |
| 20:39.20 | vastersk | hi all! need some ideas ... how can i improve my sound quality ... hearing static, and crackling noise on the phones |
| 20:39.40 | [TK]D-Fender | vastersk, Start by thinking about what's on both ends of the call first.... |
| 20:39.43 | hal-j | it's cool, yes |
| 20:39.49 | [TK]D-Fender | vastersk, And follow through by telling us. |
| 20:41.37 | vastersk | we're both hearing static on both ends ... outbound or inbound |
| 20:42.01 | BludSuckingFiend | How are the calls coming in? PRI? Analog cards? SIP/VoIP provider? |
| 20:42.09 | BludSuckingFiend | and what kind of phones? Analog, SIP, etc? |
| 20:42.13 | *** part/#asterisk polysics (~Adium@host22-155-dynamic.14-87-r.retail.telecomitalia.it) |
| 20:42.14 | vastersk | sorry, voip |
| 20:42.27 | vastersk | using ip phones |
| 20:42.28 | hal-j | static and crackling sounds an awful lot like analog |
| 20:42.45 | hal-j | so if it was me, I'd be looking at wherever you're falling off of digital |
| 20:42.58 | BludSuckingFiend | that's what I was thinking |
| 20:43.35 | BludSuckingFiend | jitter and loss on IP sounds very different from static and crackling |
| 20:44.28 | vastersk | could it be on the server's side? |
| 20:46.36 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:46.38 | BludSuckingFiend | If it's all VoIP, it's possible if there's a lot of load on the PBX for some reason. |
| 20:46.54 | BludSuckingFiend | You could do "uptime" at the console and make sure the load average isn't too high |
| 20:46.56 | [TK]D-Fender | vastersk, Models. Services. Names. |
| 20:51.08 | *** join/#asterisk Ice_Strike (Ice_Strike@87.115.83.87) |
| 20:53.02 | drmessano | You could also provide even LESS info than you have |
| 20:54.32 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
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| 21:09.15 | *** join/#asterisk billybutts138 (~william@xnat-52.csumb.edu) |
| 21:10.15 | billybutts138 | is someone around who can help me with an odic issue? |
| 21:10.19 | billybutts138 | *odbc |
| 21:19.06 | [TK]D-Fender | checkout time, BBL |
| 21:27.19 | *** join/#asterisk KNERD (~KNERD@24.175.249.177) |
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| 21:44.21 | vastersk | sorry, busy, checking settings on the server... got <[TK]D-Fender> polycom phones, with pri card also in the server... |
| 21:47.33 | navaismo | he back later |
| 22:01.49 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 22:04.53 | billybutts138 | can someone help me with and odbc issue? it is making asterisk hang |
| 22:08.45 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:08.46 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:11.15 | navaismo | billybutts138, ask and pb the logs and configs, then wait for an answer |
| 22:11.19 | navaismo | ~ask |
| 22:11.19 | 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. |
| 22:12.12 | billybutts138 | awesome thanks! sorry, I'm a noob |
| 22:13.22 | navaismo | and pb means pastebin |
| 22:13.24 | Ice_Strike | Why this wont work |
| 22:13.25 | Ice_Strike | sudo apt-get install ncurses-devel |
| 22:13.51 | navaismo | i think for debian/ubuntu its libncurses-dev |
| 22:14.00 | Ice_Strike | Hmm |
| 22:14.21 | Ice_Strike | Yea thats work |
| 22:14.27 | Ice_Strike | Damn CentOS and Ubunto! |
| 22:14.32 | pabelanger | $ sudo apt-get install libncurses5-dev |
| 22:14.32 | navaismo | you can check it with: aptitude search ncurses |
| 22:14.33 | Ice_Strike | I have to remembers the difference :/ |
| 22:14.34 | pabelanger | yar |
| 22:14.47 | Ice_Strike | Ubuntu* |
| 22:15.36 | Ice_Strike | compiling kernal :P |
| 22:19.14 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.120) |
| 22:22.50 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 22:23.14 | billybutts138 | So I am having an issue with ODBC and a database connectivity. I have an asterisk dial plan that prompts a user to spell out a last name with the keypad, then a SQL search is performed, and the user is transferred to the person of their choice. After asterisk is idle for around 35+ minutes, this doesn't work anymore. Instead, asterisk hangs for around 15 minutes. I think it is trying to reconnect, but I am not sure how to set this up to do this |
| 22:23.15 | billybutts138 | gracefully. Following this is my pastebins: |
| 22:23.15 | billybutts138 | http://pastebin.com/HqD0x7ua |
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| 22:31.09 | billybutts138 | when this is happening, if I run an "odbc show" on the asterisk CLI, it returns absolutely nothing |
| 22:39.59 | navaismo | billybutts138, im not familiar with odbc, but what you describe sound like a deadlock, which asterisk version? |
| 22:40.36 | billybutts138 | 1.8.2 |
| 22:40.54 | billybutts138 | more specifically |
| 22:41.05 | billybutts138 | Asterisk 1.8.20.0 built by mockbuild @ buildvm-26.phx2.fedoraproject.org on a i686 running Linux on 2013-01-18 19:52:25 UTC |
| 22:43.21 | navaismo | that version is the latest of the 1.8 branch? |
| 22:45.12 | billybutts138 | it was installed via YUM, and there are no more updates at the moment |
| 22:45.18 | igcewieling | 1.8.20.1 I think. I don't specifically remember any ODBC related changes in the changelog |
| 22:45.19 | navaismo | I see |
| 22:45.47 | billybutts138 | i just had the issue and was running and odbc trace |
| 22:45.48 | igcewieling | billybutts138: Asterisk is one of VERY few programs I install from source. |
| 22:45.50 | billybutts138 | let me pastbin it |
| 22:46.12 | navaismo | igcewieling, +1000 |
| 22:46.22 | billybutts138 | http://pastebin.com/8FdyqUB1 |
| 22:46.37 | billybutts138 | unfortunately, i wasn't in charge of how to install it |
| 22:46.44 | billybutts138 | and now i am stuck using this version |
| 22:47.47 | igcewieling | I don't think it matters for this specific problem, but you'll regret not installing from source soon enough. |
| 22:47.48 | billybutts138 | i did notice that this machine is using an older version of ODBC version 2.2.14 |
| 22:48.34 | billybutts138 | thanks for the heads up igcewieling! i'll make sure to never do this again lol |
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| 23:47.19 | vastersk | will the pri and analog card contribute cracking noise on the ip phone? |
| 23:47.43 | KNERD | analog |
| 23:48.27 | vastersk | with fxo and fxs connection? |
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