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| 00:20.24 | MaliutaLap | WIMPy: sounds high tech to me |
| 00:20.41 | MaliutaLap | WIMPy: probably faster than Australian internet access too |
| 00:21.11 | MaliutaLap | Katty: have you tried turning it off and on again? |
| 00:24.21 | Katty | I turned the coffee pot on |
| 00:24.29 | Katty | That's a good first step I hear |
| 00:40.18 | [TK]D-Fender | http://tinyurl.com/zzxkhmc |
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| 02:40.47 | drmessano | Seriously |
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| 14:48.32 | hexanol | I'm having a weird issue on asterisk 13 |
| 14:48.44 | hexanol | not sure if it's really related to asterisk or not, but wondering if someone had a similar issue |
| 14:48.58 | hexanol | the issue is, sometimes, when trying to establish a TCP connection to the AMI on 127.0.0.1:5038, the TCP connection doesn't establish succesfully |
| 14:49.14 | hexanol | i.e. the client receives a connection timeout error |
| 14:49.29 | hexanol | looking at lsof / netstat and the likes, I see that the client socket is stuck in state SYN_SENT, but it seems like asterisk actually accepted the connection (since the socket/file descriptor shows up in the output of lsof) and it's in the CLOSE_WAIT state |
| 14:49.35 | hexanol | there's no message in asterisk logs whatsover |
| 14:49.43 | hexanol | someone has already seen this kind of issue or knows what could cause this ? |
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| 15:34.54 | qakhan | hi all, How does holdtime get calculated for a queue? |
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| 15:45.31 | hexanol | qakhan: each time a call is answered by a queue member, the holdtime value for that queue is updated |
| 15:45.47 | hexanol | with a formula that looks like "new_queue_holdtime = 0.75 * queue_holdtime + 0.25 * call_holdtime" |
| 15:55.27 | igcewieling | hexanol: In earlier Asterisk versions that is configured in /etc/asterisk/logger.conf |
| 16:00.14 | qakhan | hexanol queue_holdtime is current holdtime? |
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| 16:10.35 | hexanol | qakhan: yes |
| 16:11.14 | qakhan | what is 0.25? |
| 16:12.04 | hexanol | it's part of the formula |
| 16:13.41 | qakhan | 0.75 is second? |
| 16:14.59 | KNERD | In the asterisk docs it briefly mentions the function SIP_HEADER(name,number) but nowhere does it include what the valid names to put in the name field are. Where can thos ebe had? |
| 16:17.04 | [TK]D-Fender | ~rfc |
| 16:17.10 | [TK]D-Fender | ~siprfc |
| 16:17.10 | infobot | from memory, siprfc is http://www.faqs.org/rfcs/rfc3261.html |
| 16:17.13 | [TK]D-Fender | ^^^^ |
| 16:17.27 | [TK]D-Fender | You can put whatever you want in there |
| 16:17.30 | [TK]D-Fender | including "FRED" |
| 16:17.35 | igcewieling | KNERD: you can pretty much add any header. |
| 16:17.42 | hexanol | qakhan: I'll give you an example. If the current hold time for queue A is 5 seconds and a member of that queue just answered a call, and that call stayed in the queue 20 seconds before being answered, than the new hold time for the queue will be "0.75 * 5 seconds + 0.25 * 20 seconds = 8.75 seconds" |
| 16:17.56 | [TK]D-Fender | It's only a matter of if the other side will care about whatttever non-standard looking thing you shove in there |
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| 16:18.51 | KNERD | hexanol: I want cmd SIP_HEADER, not SipAddHeader |
| 16:19.19 | KNERD | thus I am trying to read an element of the header |
| 16:19.33 | igcewieling | KNERD: Contact: is a valid header and on most packets I think. |
| 16:19.48 | igcewieling | as is pretty much any header you'd see with "sip set debug on" |
| 16:20.23 | KNERD | okay thanks |
| 16:20.37 | KNERD | but a list would be nice to see |
| 16:20.54 | qakhan | thanks hexanol |
| 16:21.25 | igcewieling | KNERD: then go read all the SIP RFCs. 8-| |
| 16:21.47 | KNERD | i sure will thanks |
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| 16:28.58 | igcewieling | *sigh* I hate regular expressions. |
| 16:33.04 | KNERD | then use irregualr |
| 16:34.19 | [TK]D-Fender | uses illogical operators : X MAYBE Y |
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| 17:22.55 | KNERD | awesome |
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| 17:46.25 | jrun | in order to keep up with the LTS in the 13 series would it suffice to point hour working branch to origin/13.11? |
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| 17:47.16 | jrun | is that always keeps updated with the current LTS (tag 13.11.2 i believe) in the series? |
| 17:48.05 | file | no |
| 17:48.20 | file | there is no tag or branch which is the current release of the LTS series |
| 17:48.30 | jrun | so tag->13.11.2 != branch->remotes/origin/13.11 |
| 17:48.49 | jrun | oh, i thought tag->13.11.2 was |
| 17:48.51 | file | well, there is the specific tag |
| 17:49.05 | file | however 13.12 will exist eventually, and 13.12.0 off that |
| 17:50.06 | jrun | then i guess my question would where does the release tar files come from? there is a seperate repo for the patches applied to the current LTS? |
| 17:50.37 | file | the 13 branch receives the changes, when a new release is made if it's the first one then a branch of 13 is created, 13.11 |
| 17:50.47 | file | then a tag is created off that release branch |
| 17:51.05 | file | the release branch exists so we can pull any specific changes in for a subsequent release off of it (such as the ones to make 13.11.2) |
| 17:51.38 | file | and the release tarball is created from that tag |
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| 19:38.29 | jrun | i wanna give opus a try via pjsip. that requires, i think, --enable-sound for pjsipproject but enabling that will break asterisk build. anyone here with the same experiment? |
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| 20:01.44 | scv | hello hi |
| 20:01.49 | kubrrick | hello |
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| 20:02.12 | scv | somebody want to peek at a backtrace, having a weird deadlock |
| 20:02.38 | scv | https://paste.ee/p/bcnda |
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| 20:04.51 | file | what version of PJSIP? |
| 20:05.19 | scv | bundled, using 13.11.2 |
| 20:06.14 | file | scv, doesn't look like it |
| 20:06.29 | scv | hm |
| 20:06.29 | file | that backtrace shows the symbols coming from a shared library |
| 20:06.40 | scv | it should be .. |
| 20:06.42 | scv | lemme se |
| 20:06.43 | scv | see |
| 20:07.04 | file | we build it into our own module, we don't install it |
| 20:07.17 | scv | indeed i am pulling system pjsip |
| 20:07.20 | scv | one sec |
| 20:07.29 | scv | pjproject-2.4.5-r0 |
| 20:07.52 | file | hum, I don't know if there's been anything since then which would contribute |
| 20:08.10 | scv | should i give it a shot with bundled? |
| 20:08.23 | file | it'd be nice as a data point |
| 20:08.29 | scv | let me rebuild |
| 20:12.04 | *** join/#asterisk Toerkeium (~Miranda@190.2.55.250) |
| 20:12.09 | Toerkeium | hello guys |
| 20:14.47 | Toerkeium | I'm facing proglems when connecting to my ITSP -I guess-, so I've found how to debug their particular IP, but I cannot understand the result. Anyone could give me a hand? this is asterisk debug http://pastebin.ca/index.php |
| 20:15.44 | Toerkeium | any idea? |
| 20:15.53 | rymax99 | that's just a link to pastebin |
| 20:15.55 | rymax99 | but i'd assume http://pastebin.ca/3718694 ? |
| 20:16.13 | Toerkeium | oh, sorry |
| 20:16.18 | Toerkeium | http://pastebin.ca/3718694 |
| 20:16.23 | Toerkeium | yes |
| 20:16.32 | Toerkeium | how did you knew that? |
| 20:16.33 | Toerkeium | heh |
| 20:16.45 | rymax99 | they call me ryan the magician |
| 20:16.51 | Toerkeium | hehe |
| 20:17.06 | Toerkeium | Recent posts :P |
| 20:18.22 | Toerkeium | I get an error "Circuit busy" at the asterisk console |
| 20:18.48 | Toerkeium | not sure if that's because they have any trouble or I did something wrong |
| 20:19.32 | Penguin | Are you registered to the ITSP? Most of them won't allow you to send calls inbound if you are not first registered. |
| 20:19.54 | Toerkeium | they don't require to be registered |
| 20:20.01 | Penguin | What ITSP is it? |
| 20:20.21 | Toerkeium | it's called CrossFone, from Argentina, doubt you know it |
| 20:20.36 | Toerkeium | they give you firewall access instead of requiring registering |
| 20:21.02 | Penguin | uh what? |
| 20:21.05 | Penguin | That doesn't even make sense. |
| 20:21.24 | Toerkeium | you can't register the trunk |
| 20:21.40 | Toerkeium | don't know why, but is not required |
| 20:22.03 | Penguin | OK, so they don't allow SIP registration or they don't require SIP registration? Those are to different things. |
| 20:22.43 | Toerkeium | they don't require, not sure if they allow |
| 20:22.57 | Toerkeium | but the trunk was working a couple of hours ago |
| 20:23.13 | Penguin | In that case, the circuit could be busy. |
| 20:23.26 | *** part/#asterisk rymax99 (~Ryan@unaffiliated/rymax99) |
| 20:23.32 | Toerkeium | I was trying to identify any problem so when I can get them to answer me, tells them |
| 20:23.53 | Toerkeium | the pastebin I shown doesn't say nothing? |
| 20:24.32 | Toerkeium | I mean, how can I see if it's a timeout or if they are sending any response code or something? |
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| 20:26.13 | Toerkeium | -- SIP/CrossFone-000000b7 is circuit-busy |
| 20:26.22 | Toerkeium | that's the error friendly message I get |
| 20:28.55 | Toerkeium | I've set a wrong IP address as a trunk, and it gives the same error |
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| 21:05.33 | rymax99 | anyone seen solar around |
| 21:09.46 | scv | file: ok, rebuilt my pkg and updated my test instance |
| 21:09.57 | scv | if i reproduce i'll get the bt |
| 21:14.32 | file | Kk |
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| 21:28.37 | flujan | Hello. I am trying to use the leastrecent strategy on my queue. I am using dynamic agents on a table. Every time a agent join the queue, asterisk deliveries a call. Ignoring the other agents . I noticed that the new logged in agent does not have the âlast wasâ info on the queue show. So asterisk is delivering direct to him. |
| 21:29.06 | flujan | It is possible to join a queue with the 0 on the least was info? or does it have a longest idle strategy to delivery calls? |
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