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| 00:09.29 | qloogkm[m] | Does anybody know the difference between rrmemory and rrordered in queues? |
| 00:10.40 | qloogkm[m] | No matter how I test, it just looks like both are plain old round robin.... docs aren't really clear on the difference either :/ |
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| 00:15.39 | [TK]D-Fender | rrmemory - round robin with memory, remember where we left off last ring pass |
| 00:15.39 | [TK]D-Fender | ; rrordered - same as rrmemory, except the queue member order from config file |
| 00:15.39 | [TK]D-Fender | ; is preserved |
| 00:15.46 | [TK]D-Fender | Seems pretty clear |
| 00:17.52 | qloogkm[m] | [TK]D-Fender: thank you, but that's what is confusing me |
| 00:17.57 | Akari | In other words, rrordered always starts at Stacy and goes to Kim and then Susan. rrmemory keeps track of whether Stacy, Kim, or Susan answered the last call, and directs the next call accordingly. |
| 00:18.26 | [TK]D-Fender | One clearly says "remember where LEFT OFF" |
| 00:18.27 | qloogkm[m] | OK so rrordered is like linear? |
| 00:19.58 | Akari | It's like linear but the order changes on each call. |
| 00:20.15 | Akari | You can't see linear by looking at a single call. |
| 00:20.18 | Akari | Err |
| 00:20.19 | Akari | Round robin |
| 00:20.53 | qloogkm[m] | meaning that you would have to have multiple calls coming in at the same time to see the difference? |
| 00:21.19 | Akari | Stacy, Kim, Susan; Kim, Susan, Stacy; Susan, Stacy, Kim; Stacy, Kim, Susan |
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| 00:22.43 | Akari | Linear would be Stacy, Kim, Susan; Stacy, Kim, Susan; Stacy, Kim, Susan; Stacy, Kim, Susan; |
| 00:24.04 | qloogkm[m] | and rrmemory is |
| 00:24.38 | Akari | rrmemory depends on who answered the call. |
| 00:24.39 | qloogkm[m] | Stacy, Kim (answered); Susan, Stacy(answered); Kim, Susan, Stacy; |
| 00:24.56 | Akari | Yeah |
| 00:25.01 | qloogkm[m] | So rrordered is round robin for where the call STARTS |
| 00:25.44 | Akari | rrordered is the "proper" round robin that's good for distributing calls evenly over your group. |
| 00:26.45 | qloogkm[m] | Because it makes more passes over the same agents? |
| 00:28.07 | Akari | rrordered is good if everyone's supposed to pick the call up first hit. rrmemory is good if people are away from their desk because it tends to ring people who are away less. |
| 00:28.19 | Akari | For instance if Susan is away from her desk, it would ring her half as much |
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| 00:28.50 | qloogkm[m] | rrmemory would completely skip somebody like Susan who is unavailable, until the next time it comes around, hence ringing them less |
| 00:29.46 | qloogkm[m] | OK I think I get this |
| 00:31.16 | Akari | rrmemory isn't something you would need to use a lot I don't think |
| 00:31.23 | qloogkm[m] | Yeah |
| 00:31.39 | [TK]D-Fender | depends on how evenly you want calls to go around |
| 00:31.39 | qloogkm[m] | I have to explain the difference to a training class today... :) |
| 00:32.45 | qloogkm[m] | Thanks [TK]D-Fender and Akari, that explanation was very helpful |
| 00:33.00 | Akari | You kind of need a big group of callers to see the effects |
| 00:33.17 | Akari | Or people manning the phones |
| 00:34.01 | qloogkm[m] | rrordered will end up with all agents having "about" the same number of calls at the end of the day |
| 00:34.06 | qloogkm[m] | right? |
| 00:34.27 | Akari | If they all picked up first time, yes. |
| 00:34.51 | qloogkm[m] | OK great! A giant knot has been pulled from my brain :) |
| 00:34.53 | Akari | rrmemory would skip people for a first ring if they didn't answer a ring that fell through, essentially. |
| 00:35.04 | qloogkm[m] | Got it |
| 00:35.14 | Akari | That's probably a better way to put it. |
| 00:35.43 | qloogkm[m] | Thank you so much. Very cool. |
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| 05:46.36 | chezidek | possible to send SMS via SIP w/ asterisk? provider supports it but i can't find any good docs |
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| 06:01.26 | igcewieling | chezidek: "SMS" is a generic term these days. You need to know the exact protocol name. |
| 06:01.58 | igcewieling | SIMPLE or XMPP or something like that, then you know the terms to google. |
| 06:02.55 | chezidek | hmm. ok. my provider is voip.ms, but they don't provide much documentation except that it's "supported" |
| 06:03.05 | chezidek | but, that is helpful, i'll try and shotgun it |
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| 10:12.35 | Ahmadi | Hi |
| 10:13.08 | Ahmadi | Is it anyway that i log all SIP Packets that come in/out in/from my asterisk? |
| 10:13.16 | Ahmadi | is there * |
| 10:21.08 | ixyd | Ahmadi: when using chan_sip: sip set debug?! |
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| 10:30.12 | wyoung | Ahmadi: yes |
| 10:30.33 | wyoung | wireshark, tcpdump, anything that uses libpcap |
| 10:31.19 | wyoung | also you can type in sip set debug on if you just want sip headers |
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| 12:50.02 | wasanzy | hi |
| 12:50.46 | wasanzy | exten => 1,Goto(bima-login,s,1) this gives me Label missing trailing ')' |
| 12:53.40 | wasanzy | I found the error. I didn;t set the priority? |
| 13:19.54 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 13:20.51 | wasanzy | Set(check_member=${SHELL(grep ${CALLERID(num)} ${QUEUE_LIST})}) |
| 13:21.10 | wasanzy | is there something wrong with the way am using the grep? seems not to be working |
| 13:21.58 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 13:23.29 | wasanzy | welcome TK]D-Fender |
| 13:29.03 | [TK]D-Fender | is there something wrong with the way am using the grep? seems not to be working <- I would consider actually looking at what is happening. |
| 13:29.06 | [TK]D-Fender | (like always) |
| 13:29.14 | [TK]D-Fender | because once again, we aren't |
| 13:30.27 | wasanzy | I dont understand. "I would consider actually looking at what is happening" |
| 13:31.03 | wasanzy | Set("SIP/outgoing2-0058d95a", "check_member=") in new stack |
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| 13:31.18 | wasanzy | GotoIf("SIP/outgoing2-0058d95a", "0?agent:client") in new stack |
| 13:31.28 | wasanzy | that is from the logs |
| 13:32.08 | [TK]D-Fender | So ... Where do we see you checking what you're feeding that function? |
| 13:32.39 | wasanzy | GotoIf($["${check_member}" != ""]?agent:client) |
| 13:33.05 | [TK]D-Fender | those labels mean nothing and that is not a proper check |
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| 13:34.53 | wasanzy | https://paste.linux.community/view/a3920e9b |
| 13:35.42 | wasanzy | how do I check then? I wanted a condition that, if the Callerid is not found in the list, the flow shld go to client.... |
| 13:36.45 | [TK]D-Fender | <[TK]D-Fender> So ... Where do we see you checking what you're feeding that function? |
| 13:36.46 | [TK]D-Fender | ^ |
| 13:39.10 | wasanzy | are you talking about log? |
| 13:40.04 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> So ... Where do we see you checking what you're feeding that function? <----------------------------------- |
| 13:40.09 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:40.44 | wasanzy | QUEUE_LIST=/etc/asterisk/bima/agents.txt |
| 13:40.53 | wasanzy | that contains a list of phone numbers |
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| 14:08.59 | NiugeS | hi all.. i've been trying for the better part of a day to try and get ODBC working with asterisk under a new FreePBX distro installation. I can't connect and have tried and reviewed so many websites i'm at a loss.. i'm fairly new to asterisk etc.. but have checked the odbc.ini files etc.. tried granting privleges etc.. i also followed this https://wiki.asterisk.org/wiki/display/AST/Getti |
| 14:08.59 | NiugeS | ng+Asterisk+Connected+to+MySQL+via+ODBC but after i create the user i'm unable to connect to the mysql and database ERROR 1045 (28000): Access denied for user 'asterisk'@'localhost' (using password: YES) |
| 14:09.13 | NiugeS | any ideas what i'm doing wrong? i can't connect remotely as the connection is refused.. |
| 14:12.24 | [TK]D-Fender | Clearly not a value user/pass |
| 14:12.30 | [TK]D-Fender | TThatt error means what it says |
| 14:22.24 | NiugeS | hmmm i copied and pasted the instructions.. |
| 14:22.44 | NiugeS | apart from 'replace_with_strong_passowrd'; |
| 14:25.25 | Samot | How can you run the FreePBX Distro and have a ng+Asterisk+MySQL setup? |
| 14:27.19 | [TK]D-Fender | ... you configure ODBC |
| 14:27.34 | [TK]D-Fender | I fail to see the issue with doing so |
| 14:28.43 | Samot | Yeah, this is sounding more like a source install. |
| 14:29.31 | wasanzy | I have changed the condition to this: GotoIf($["${check_member}" == ${CALLERID(num)}]?agent:client) |
| 14:31.31 | NiugeS | Samot [TK]D-Fender just going over everythign one more time.. noticed changing CREATE USER 'asterisk'@'%' CREATE USER 'asterisk'@'localhost' allows me to login using that user... |
| 14:36.01 | *** join/#asterisk verify (~NiugeS@37.77.180.234) |
| 14:36.09 | verify | sorry got disconnected.. |
| 14:37.44 | *** join/#asterisk verify (~NiugeS@37.77.180.234) |
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| 14:43.02 | verify | Samot apologies but i assumed the setup was already there.. i've followed the instructions and i'm able to connect to the database.. but not sure how to the recompilation as it fails according to instructions.. in simple terms, several years ago we had someone install a PBX for us and we were able to connect excel to CDR via an ODBC connection.. i'm trying to replicate that but obviously with |
| 14:43.02 | verify | very little success.. |
| 14:43.42 | Samot | Either you installed the Distro which does all this for you on install or you installed from source which requires *you* to configure all of this. |
| 14:44.29 | Samot | Which was it? The first means you went in and messed with things, the second means you just missed steps in the source install process. |
| 14:45.14 | verify | I installed the FreePBX distro. |
| 14:45.22 | verify | And obviously messed things up. |
| 14:46.00 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 14:46.02 | Samot | Then you shouldn't have needed to create MySQL user, one already exists. |
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| 14:46.26 | Samot | The ODBC files would have already been written out with proper information. |
| 14:46.33 | NiugeS | the problem i was facing (and still am) is that I can't connect via ODBC mysql connector |
| 14:47.12 | Samot | OK, so 1) Not an Asterisk issue. 2) Installed FreePBX Distro = #freepbx for support. |
| 14:47.29 | NiugeS | lol.. and i specifically didn't go in there as i thought it was an asterisk issue.. |
| 14:47.30 | NiugeS | sorry |
| 14:48.01 | Samot | MySQL failures are not Asterisk related. |
| 14:48.10 | NiugeS | gotcha.. |
| 14:49.09 | NiugeS | thanks for the thelp |
| 14:49.12 | NiugeS | *help |
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| 15:21.47 | wasanzy | I got this https://paste.linux.community/view/16df62d6 in the logs now |
| 15:22.13 | wasanzy | that is after changing the line to: GotoIf($["${check_member}" = "${CALLERID(num)}"]?agent:client) |
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| 15:23.11 | wasanzy | from the logs, check_member is set the the CallerID so I don't know why the if statement not working |
| 15:24.00 | Peiniger | 1478704915.7823 how is the `7823` portion of this call id deteremined? |
| 15:25.42 | Samot | wasanzy: You could use NoOp to log your progress in the call. |
| 15:26.13 | Samot | That way you can track what variables are being set and what value it is. |
| 15:26.42 | wasanzy | <PROTECTED> |
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| 15:27.16 | wasanzy | is there something wrong with how am doing the gotoif? |
| 15:28.06 | igcewieling | I don't see the callerid num in that noop |
| 15:29.00 | igcewieling | ah, in the pastebin. |
| 15:29.12 | wasanzy | igcewieling: you mean in the Gotoif? GotoIf("SIP/outgoing2-0058dd7d", "0?agent:client") ?? |
| 15:29.24 | igcewieling | the 0? means "expression evaluated to false" [s@bima-start:3] GotoIf("SIP/outgoing2-0058dd7d", "0?agent:client") in new stack |
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| 15:29.48 | wasanzy | igcewieling: ok |
| 15:30.01 | igcewieling | are you expecting it to evaluate to true? |
| 15:30.02 | wasanzy | then there is something wrong with the expression |
| 15:30.13 | wasanzy | yes am expecting it to evaluate to true |
| 15:30.51 | igcewieling | check the two variables in a noop right before the expression line. |
| 15:31.28 | wasanzy | igcewieling: Ok |
| 15:37.34 | wasanzy | NoOp("SIP/outgoing2-0058dddc", ""Callerid:"+233271520077 "and check_number:" +233271520077 |
| 15:38.52 | wasanzy | so from the above debug info, this GotoIf("SIP/outgoing2-0058dddc", "0?agent:client") shld have been result to true |
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| 15:44.30 | Samot | Show a new call. |
| 15:48.08 | wasanzy | ok |
| 15:50.52 | wasanzy | https://paste.linux.community/view/24fc5742 |
| 15:51.01 | wasanzy | that is a new call |
| 15:52.08 | Samot | Show the updated context dialplan. |
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| 15:52.16 | igcewieling | notice you have extra spaces |
| 15:53.03 | igcewieling | -- Executing [s@bima-start:3] NoOp("SIP/outgoing2-0058de61", ""Callerid:"+233271520077 "and check_number:" +233271520077") |
| 15:53.31 | wasanzy | https://paste.linux.community/view/8e3920dd dialplan |
| 15:53.54 | wasanzy | I put the space there |
| 15:53.57 | igcewieling | how about you fix the spacing issue and try another test call, just so we know. |
| 15:54.53 | Samot | same => n,GotoIf($["${check_member}" = "${CALLERID(num)}"]?agent:client) |
| 15:55.04 | Samot | ^ Or use the proper syntax.. |
| 15:55.07 | Samot | == |
| 15:55.18 | igcewieling | = and == are the same in Asterisk |
| 15:57.42 | wasanzy | https://paste.linux.community/view/98eb4c4e |
| 15:57.49 | wasanzy | the space is no more there |
| 15:58.32 | igcewieling | no quotes so we can see that I see. |
| 15:58.49 | igcewieling | in any case, it looks like it should be working. |
| 15:58.57 | igcewieling | I don't know why it isn't. |
| 15:59.45 | wasanzy | you mean no quotes in the NoOp() ? |
| 16:03.40 | Samot | Change = to != just to see if it will flip. |
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| 16:05.53 | wasanzy | Samot: yea seem to be working that way |
| 16:06.14 | Samot | So then it's not seeing those two values as the same. |
| 16:10.21 | igcewieling | might want to try FILTER() to remove any non-number characters when comparing |
| 16:10.49 | wasanzy | Somot: maybe |
| 16:10.57 | wasanzy | igcewieling: ok |
| 16:11.04 | Samot | Not maybe, thats whats happening. |
| 16:11.15 | Samot | Your statement was only true because they didn't match. |
| 16:11.45 | Samot | Then statement to only be true if they match, fails. |
| 16:11.55 | Samot | s/Then/The |
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| 18:28.07 | twanny796 | I need to install version 12 instead of FreePBX 13.0.188.8, which img should I download |
| 18:28.09 | twanny796 | ? |
| 18:28.19 | twanny796 | FreePBX |
| 18:29.43 | [TK]D-Fender | My guess.... a version taht is older than 130.0.188.8 |
| 18:34.17 | [TK]D-Fender | Which one ... is naturally not a topic for here |
| 18:36.47 | twanny796 | [TK]D-Fender: thanks, long time since we spoke! |
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| 19:24.25 | TandyUK | ok, so we've been having problems where one of our isps uplink is gettng saturated, and were getting 4-10% sporadic packet loss |
| 19:24.53 | TandyUK | is there anything i can do at either phone or asterisk ends to make it more aggressive at identifying a dropped packet and re-transmitting it |
| 19:25.13 | TandyUK | for the most part were using TCP/TLS for sip and SRTP for audio |
| 19:25.47 | TandyUK | the point where the packet loss is occuring is totally out of my control |
| 19:34.54 | rrittgarn | when using features.conf to initiate mixmon, shouldn't TOUCH_MIXMONITOR_OUTPUT or some other variable get set to the file name? Or did that functionality move/go away moving to 13? |
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| 19:40.29 | rrittgarn | upgraded to 13 - back in 11 i had a hangup handler that would email me the recording once done - but now i see it isn't triggering because I'm looking for that particular variable - and on a Dump_chan it isn't showing up - so wondering if there's something I'm missing in the configuration? perhaps in features.conf? |
| 19:49.52 | mjordan | rrittgarn: TOUCH_MIXMONITOR_OUTPUT is set on the peer channel in the bridge |
| 19:51.00 | mjordan | rrittgarn: using app_dumpchan, I would expect channel variables to still be dumped out. What comes out after Variables:? |
| 19:51.59 | mjordan | rrittgarn: Keeping in mind that TOUCH_MIXMONITOR_OUTPUT is set on the peer channel, not on the initiator channel |
| 19:52.08 | mjordan | which means your hangup handler needs to be on the peer in the bridge |
| 19:53.29 | mjordan | TandyUK: Natively, Asterisk won't ever retransmit audio. Your only real option is to put a JITTERBUFFER on the channel on the packet loss side; Asterisk will interpolate audio if it attempts to read audio from said JITTERBUFFER and none is there |
| 19:53.39 | mjordan | That won't really *fix* the problem, but it may make it less severe. |
| 19:54.45 | rrittgarn | matt: I do see variables, but i was looking on the 'local' side of the call, not the remote party - not sure how i had it working before if this has always been the behavior |
| 19:55.41 | rrittgarn | dialing out using option X, and dialing my feature code - i see the file get created, etc. but in dumpchan nothing mix mon related - at least not in 13 - i can run the same test on an 11 box to be sure i'm not crazy (which is a possibility) |
| 19:56.38 | Samot | TandyUK: Is the connection only used for voice or do you have data on it too? |
| 20:03.51 | rrittgarn | @mjordan - should I see "any" mixmon related variables on the originating channel (channel that presses the feature keys) ? |
| 20:19.12 | qakhan | hi all, i am originating call from manager. i want to get OriginateResponse of the call. its not generating the response. here is my php script http://pastebin.com/wcwXLKqv |
| 20:23.49 | Samot | [Response] => Success |
| 20:23.54 | Samot | ^ Wouldn't that be the response? |
| 20:24.31 | Samot | You successfully executed an originate. |
| 20:28.21 | qakhan | Samot no. i want to get OriginateResponse which contain this info https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_OriginateResponse |
| 20:30.32 | [TK]D-Fender | I don't see you setting up an event handler |
| 20:31.12 | TandyUK | Samot: connection is purely for voice |
| 20:31.20 | TandyUK | the data connection has been fine lol |
| 20:33.10 | qakhan | [TK]D-Fender can you please give me some example for event handler |
| 20:33.26 | [TK]D-Fender | GOOGLE <- |
| 20:33.39 | [TK]D-Fender | You clearly haven't even LOOKED at what php-agi methods you are even using |
| 20:33.56 | qakhan | i did |
| 20:34.04 | [TK]D-Fender | you failed pretty hard |
| 20:34.20 | [TK]D-Fender | I spend all of 30 seconds having NEVR done this and see you not setting those up |
| 20:34.21 | qakhan | you are right |
| 20:35.02 | [TK]D-Fender | $res = $asm->Events('OriginateResponse'); <- what part of this says it'll sit around LISTENING and waiting? |
| 20:35.07 | [TK]D-Fender | it could take TIME. |
| 20:35.59 | qakhan | $asm->Events() is function in phpagi-asmanager.php |
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| 20:40.16 | [TK]D-Fender | http://eder.us/projects/phpagi/ |
| 20:40.22 | [TK]D-Fender | Events (line 323) |
| 20:40.22 | [TK]D-Fender | Enable/Disable sending of events to this manager |
| 20:40.45 | [TK]D-Fender | That is a dumb function that it tells you you pass "yes" or "no" to ... to ENABLE even GETTING them |
| 20:41.00 | [TK]D-Fender | it does NOTHING about actually listening for something specific |
| 20:41.18 | [TK]D-Fender | And you've done a terrible job just reading the instructions they provide you as to whatt that command even means |
| 20:43.39 | qakhan | so i should not use this AMI? |
| 20:44.24 | [TK]D-Fender | "this AMI"? |
| 20:44.28 | [TK]D-Fender | What does that even mean? |
| 20:44.57 | [TK]D-Fender | That command doesn't sit around LISTENING for a specific response event to happen |
| 20:45.02 | [TK]D-Fender | READ THE DAMN MANUAL |
| 20:46.02 | qakhan | i mean phpAGI 2 |
| 20:47.34 | [TK]D-Fender | Still a broken question. |
| 20:47.41 | [TK]D-Fender | It CAN listen for events |
| 20:48.19 | [TK]D-Fender | You have just called something inappropriately and seem incapable of even reading what those methods do before feeding them junk values and expecting results. |
| 20:48.25 | [TK]D-Fender | Go hire a programmer |
| 20:54.48 | TandyUK | hands [TK]D-Fender a cookie :) |
| 20:55.13 | [TK]D-Fender | spins up some Limp Bizkit... |
| 20:55.40 | TandyUK | break stuff ;) |
| 20:56.15 | [TK]D-Fender | Nookie <- |
| 20:56.29 | [TK]D-Fender | :p |
| 21:24.16 | *** join/#asterisk theseb (~user@96-91-58-241-static.hfc.comcastbusiness.net) |
| 21:25.11 | theseb | can i ask a general Google Hangouts (YouTube Air) VOIP question?....i noticed sometimes when a new person talks there is some static |
| 21:25.30 | theseb | it seems the voip software needs a few seconds to adjust to new talker.....sound familiar? any way to minimize? |
| 21:25.32 | [TK]D-Fender | We don't use or support them here |
| 21:25.38 | [TK]D-Fender | I'm pretty sure Google has their own |
| 21:25.50 | theseb | [TK]D-Fender: but does that problem plague all voip systems like asterisk? |
| 21:26.09 | theseb | [TK]D-Fender: i.e. noise when switch talkers |
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| 21:33.23 | [TK]D-Fender | No. |
| 21:33.30 | [TK]D-Fender | There is no "switch talkers" |
| 21:33.44 | [TK]D-Fender | Static is not something that should exist in a digital call |
| 21:33.57 | [TK]D-Fender | unless it's at the mic level |
| 21:34.06 | [TK]D-Fender | packs up to head home |
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| 23:49.24 | dmasiero | Howdy. Any help on using SIPAddHeader to modify a 200 OK for 11.6? |
| 23:51.33 | infernix | does asterisk still care about a 1000Hz kernel? |
| 23:51.48 | infernix | will I run into any issues if i recompile at 100Hz |
| 23:52.02 | infernix | or 250? |
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