| 00:00.12 | navaismo | yep but my question goes to use somthing like: exten => s,1,GotoIfTime(09:00-16:59,Mon,fri,*,*?SomeContext,1)///exten => s,1,GotoIfTime(09:00-16:59,tue,thu,*,*?SomeContext,1) |
| 00:00.14 | phix | Thu even |
| 00:00.37 | navaismo | for more than one date with same settings if present the case |
| 00:00.49 | phix | navaismo: not sure, i know you cant use commar there, maybe the | will ork |
| 00:00.51 | phix | work |
| 00:01.11 | phix | your should google or [TK]D-Fender it :) |
| 00:01.15 | navaismo | let say: sun &tue mon&wed thu&sat |
| 00:01.49 | phix | not sure if that syntax will work as a time cannot be both sunday and tuesday at the same time |
| 00:01.50 | navaismo | just asking if that app use special separator than '- ' |
| 00:02.06 | phix | Logical expression speaking that is |
| 00:02.24 | [TK]D-Fender | there is no "multiple different" options |
| 00:02.28 | phix | http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime |
| 00:02.30 | [TK]D-Fender | There is single or range. |
| 00:02.32 | navaismo | i think my english isnt helping here |
| 00:02.38 | [TK]D-Fender | This is not complicated |
| 00:02.56 | navaismo | yes as i wrote i finally doit per day |
| 00:02.58 | phix | just redundant, not complicated :) |
| 00:03.06 | phix | alot of repeat in lines, but it gets the job done |
| 00:03.07 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 00:03.16 | navaismo | not complicated but asking for faster ways |
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| 00:03.29 | phix | faster or more compact way of writing it? |
| 00:03.46 | [TK]D-Fender | navaismo: There is none |
| 00:03.50 | navaismo | same for me |
| 00:03.55 | [TK]D-Fender | navaismo: Multiple lines. Do it. |
| 00:04.21 | navaismo | yep |
| 00:04.26 | TheRedOctober | Hi all. I am keep getting TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 on a CUCM sip trunk that _was_ working...no problems with tcp/ip, has anyone seen such an issue before? |
| 00:04.30 | navaismo | basically gotoiftime only work with ranges |
| 00:06.12 | navaismo | thx all |
| 00:06.16 | navaismo | time to go |
| 00:07.46 | phix | [TK]D-Fender: so, recording voice when I press the record button |
| 00:09.59 | [TK]D-Fender | phiI'm sorry, you forgot to phrase that in the form of a question </trebek> |
| 00:10.06 | [TK]D-Fender | phix: I'm sorry, you forgot to phrase that in the form of a question </trebek> |
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| 00:19.17 | phix | [TK]D-Fender: yeah question was above that a few lines or so, that was a reminder :) |
| 00:21.01 | [TK]D-Fender | phiOh you thought I was going to scroll for miles hoping to find something resembling "coherenace"! |
| 00:21.12 | [TK]D-Fender | coherence* |
| 00:21.15 | [TK]D-Fender | gah |
| 00:21.24 | datruth | the sip.conf file makes me sleepy |
| 00:21.49 | [TK]D-Fender | [18:55]phixI got me a snom 320, it has a record button, I push said record button and in my logs it tells me recording isn't enabled, URL me please <- www.google.com |
| 00:21.52 | [TK]D-Fender | There you go! |
| 00:22.30 | phix | <3 |
| 00:28.30 | phix | [TK]D-Fender: ok do I set monitoring in queue? or do I need to call the command explisively? or can i get it to record whenI press the record button on my snom? |
| 00:28.40 | phix | what is the usual way about doing this> |
| 00:28.52 | phix | http://www.voip-info.org/wiki/view/MixMonitor looks interesting but it doesnt really help me |
| 00:31.37 | [TK]D-Fender | queues.conf has it's own automatic recording option. |
| 00:51.22 | phix | [TK]D-Fender: I noticed |
| 00:52.09 | phix | so what do I use? I tried featuremap and set *1 to record, added automon to DYNAMIC_FEATURES or whatever the var is, also set wWxX in the option to dial, still not recording |
| 00:52.26 | phix | even tried the *9 applicationmap example, still doesnt work |
| 00:53.00 | phix | voip-info.org seems to be very out of date, I am using 1.6, any ideas how to do this? |
| 00:53.56 | phix | or where I should be looking to do this? my snom 320 has a dedicated record button, my snom 300 doesn't I want both to beable to record |
| 00:54.00 | jdoe | phix: did you rebind the transfer default? |
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| 00:56.48 | phix | automon => *1 ; One Touch Record |
| 00:56.59 | phix | in [featuremap] context |
| 00:57.06 | [TK]D-Fender | phix: that is a dial option. do not assume that your agent will have dial options |
| 00:58.01 | phix | SIP and DAHDI honor t and T |
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| 00:58.25 | phix | are yousaying they may not honor w, W, x or X? |
| 00:58.42 | [TK]D-Fender | phiWhere do you see those options being chosen? |
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| 00:59.03 | phix | [TK]D-Fender: when I call Dial() |
| 00:59.12 | [TK]D-Fender | Then you should have them |
| 00:59.18 | phix | but i dont |
| 00:59.38 | phix | I press the key sequence and nothing happens, nothing is recorded and nothing in log files |
| 01:00.07 | phix | [TK]D-Fender: so you know of a better example config to use? |
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| 01:00.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:00.16 | phix | you know, one that actually works |
| 01:00.26 | phix | hai mjordan! |
| 01:01.08 | [TK]D-Fender | phiMaybe we should look at what youre actually doing... |
| 01:01.33 | jdoe | phix: I got feature codes working with no particular tweaking (other than changing the default record bind) |
| 01:03.24 | phix | jdoe: pastebin example? |
| 01:03.34 | phix | [TK]D-Fender: you want me to paste bin it? |
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| 01:09.26 | phix | [TK]D-Fender: example? |
| 01:09.39 | [TK]D-Fender | yes |
| 01:10.59 | phix | you has an example that works? |
| 01:11.52 | [TK]D-Fender | phix: show us your problem. |
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| 05:12.50 | tully` | Hey all, it's to my understanding that when you spawn an EAGI function you can access the audio in file descriptor 3 of the process id of the program spawned. Does anyone know what would cause the fd to not be created or why I can't access it? |
| 05:14.49 | tully` | I'm using the phpagi scripts and i'm checking to see if the file exists, opening it, and then testing to see if the file has actually been openend. The file exist check works, but when i test the var I use fopen on it returns null |
| 05:16.58 | tully` | here's the code and result http://pastebin.com/gWTesBNm |
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| 07:12.20 | schmidts | good morning |
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| 08:05.27 | ChannelZ | or something |
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| 08:34.42 | aberrios_ | when setting up an R850 should I have a file /etc/ha.d/ha.cf ? Node complaining it cant be found but cant find a reference to it in the manual |
| 08:36.48 | ChannelZ | uhmm... what? |
| 08:37.58 | *** part/#asterisk gajini (~root@61.12.17.171) |
| 08:39.08 | aberrios_ | ChannelZ: Setting up an R850 failover switch. Unfortunately its saying there's no heartbeat from one node and the other node is intermittant. Both nodes are complaining they cant find the aforementioned file. |
| 09:04.13 | aberrios_ | it seems the second node is not pinging the watchdog on the R850 |
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| 09:17.23 | awk | Hi, ManagedEngine has a product called VQManager that monitors SIP, H323, etc. I only need SIP monitoring, what it does is installs via a BIN file on a linux box, and captures all traffic on port 5060 and it's RTP stream. The issue is its becoming end of life, and I don't seem to see another product as good that monitors packet loss, complete sip trace of each part of the call, MoS, etc. Any suggestions what I can look at I, see IPTego has a product but it do |
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| 10:50.06 | ruben23 | hi guys any idea on this error log during i install asterisk-addon on my existing asteriks server-------->http://pastebin.com/fqwvmJzk <-----------------help guys |
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| 10:52.10 | krotos | hi all :) |
| 10:53.37 | krotos | i'have a curiosity, in a public telefony enviorment, such as ss7, when a new number is originated how it's "announced" / routed? |
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| 11:11.19 | mechbangirc | can I use a proxy b/w asterisk and a phone? |
| 11:11.39 | dym | b/w ? |
| 11:11.44 | dym | bandwidth? |
| 11:11.45 | mechbangirc | between |
| 11:11.57 | dym | jesus christ |
| 11:12.02 | dym | b/w is between? |
| 11:12.07 | dym | in what world? |
| 11:12.28 | ruben23 | hi guys any idea..? |
| 11:12.34 | mechbangirc | is there any other world? |
| 11:12.45 | dym | quite obviously |
| 11:12.53 | dym | but yeah mechbangirc - why shouldnt you? |
| 11:13.23 | kaldemar | mechbangirc: of course you can use a proxy. |
| 11:13.37 | schmidts | krotos it depends on the number range, if the whole range is routed "short" then you dont have to touch it by hand. if it is a new block or only routed long then you have to set a routing entry by hand or through a provisionial interface |
| 11:14.10 | dym | cheers kaldemar :D |
| 11:14.32 | mechbangirc | I run an asterisk based switch and often get complaints from customers regarding call not being established or not reached at all. Mostly they are behind some kind of proxy. I dont know much about proxies. |
| 11:15.13 | schmidts | mechbangirc i guess you mixed some kind of proxis. a typical proxy for users is a http proxy which doesnt belong to a sip proxy |
| 11:15.42 | dym | ahhhh |
| 11:15.43 | schmidts | if they really mean a sip proxy then it shouldnt make a difference to your system |
| 11:15.56 | dym | that sounds more like some sort of nat issue |
| 11:15.59 | krotos | schmidts: thankyou for reply! I mean a new block, how it's routed /signaled. My curiosity was "if a new block was assigned, and i call a number inside this new block, how my mobile operator say where to route the call" |
| 11:16.13 | mechbangirc | schmidts: how about a nat gateway for all kind of traffic? |
| 11:16.36 | dym | thought so |
| 11:16.37 | mechbangirc | they are not surely running sip aware proxy like opensips |
| 11:17.32 | schmidts | krotos i dont know how this works in other countries but here in austria it works as stupid as it could be. you have to tell your carrier which have interconnects to other carriers to provision this routing, and normaly this provisioning to other carriers is a fax or email to enable routing for this number range to the carriers routing code |
| 11:18.23 | schmidts | mechbangirc as dym said, thats a typical nat issue. make sure you have set nat=yes in your sip config for these peers and also enable qualify=yes to keep nat ports open |
| 11:19.24 | dym | schmidts: i would keep my nat port open for you anytime |
| 11:19.25 | dym | (: |
| 11:19.47 | mechbangirc | I have that already, I am sure it is their side "bad routing" or something. I want to know some solid reason to satisfy them. |
| 11:19.50 | schmidts | krotos typically you will get a date after the new block should be reachable through all source networks but that depends on how the interconnects between these carriers work |
| 11:19.50 | krotos | schmidts: ok, i think is similar here ( in Italy), it was onlly a curiosity. before you reply to my answer, i thinked that was something like a routing table and announcement (BGP ) such as IP network |
| 11:20.32 | dym | mechbangirc: well, running your own telephony system you should at least be able to pinpoint something like that. |
| 11:20.56 | schmidts | krotos aaahh ok in italy it could allready be different cause as far as i know they allready have to use IP voice interconnects instead of normal ones. but there is iirc no global database like enum for routing |
| 11:21.21 | schmidts | it would be very nice to have such thing but even in italy they arent using something like this allready |
| 11:21.27 | mechbangirc | dym, thats not very encouraging. |
| 11:22.04 | kaldemar | mechbangirc: NAT itself is no problem, but a SIP ALG (application level gateway) in a router can cause headaches. better to disable those entirely if such exist on the signaling path. |
| 11:22.19 | dym | mechbangirc: Well, face the facts |
| 11:23.06 | mechbangirc | kaldemar: I do disable ALGs all the time. I think something to do with UDP ports being closed before response could reach, not sure though. |
| 11:23.33 | schmidts | mechbangirc atleast you can ask your customers to also enable nat keep alive packets |
| 11:24.18 | mechbangirc | schmidts: and that would be in phones right? or in proxy configuration? |
| 11:24.26 | krotos | schmidts: ahahahah , so every new block is configured "manually"...is like '80 years :P |
| 11:26.52 | ruben23 | hi guys any idea on this error log during i install asterisk-addon on my existing asteriks server-------->http://pastebin.com/fqwvmJzk <-----------------help guys |
| 11:27.58 | schmidts | mechbangirc dont mix up proxy with nat, but yes its a typical phone setting |
| 11:28.23 | schmidts | krotos the whole PSTN network is like from the 80's ;) |
| 11:28.28 | krotos | schmidts: i confirm you that in italy we have started to use IP Voice interconnected to PSTN network...but the problem in italy was that the xDSL interconnection is not too good.. |
| 11:28.37 | mechbangirc | schmidts I got it |
| 11:28.46 | mechbangirc | thanks |
| 11:28.59 | *** part/#asterisk mechbangirc (~mechbangi@mbl-82-58-67.dsl.net.pk) |
| 11:30.46 | *** join/#asterisk j0b (~jb@unaffiliated/j0b) |
| 11:30.51 | j0b | howdy |
| 11:31.40 | dym | and to you sir |
| 11:32.15 | j0b | so, here is a question, we have connected two pbx:es together, we recive a call that goes to context 162, then we call the other pbx with dial(sip/number) |
| 11:32.38 | j0b | somehow the callerid that is presented to the person that is called |
| 11:32.41 | j0b | is 101 |
| 11:33.01 | j0b | [101] |
| 11:33.02 | j0b | context=intern |
| 11:33.02 | j0b | type=friend |
| 11:33.02 | j0b | username=101 |
| 11:33.02 | j0b | secret=welcome |
| 11:33.04 | j0b | host=dynamic |
| 11:33.06 | j0b | nat=no |
| 11:33.09 | j0b | disallow=all |
| 11:33.10 | dym | heard of nopaste? |
| 11:33.11 | j0b | allow=alaw |
| 11:33.14 | j0b | why is that |
| 11:33.16 | dym | very annoying |
| 11:33.18 | j0b | can i send a callerid when i dial? |
| 11:33.23 | j0b | dym: yeah, sorry man |
| 11:33.24 | dym | ~nopaste |
| 11:33.25 | infobot | it has been said that nopaste is http://nopaste.snit.ch/ or http://pastebin.com/, or preferably http://pastebin.ca/ |
| 11:33.31 | j0b | next time:) |
| 11:33.51 | dym | you can always set a callerid |
| 11:33.59 | dym | or manipulate the sip headers |
| 11:34.03 | j0b | mm but why does it become 101 |
| 11:34.05 | j0b | i have like |
| 11:34.21 | dym | what else would it become? |
| 11:34.22 | j0b | exten => 162,1,Set(CallerIDnum=${CALLERID(num)}) |
| 11:34.23 | j0b | exten => 162,2,Dial(SIP/6341@101) |
| 11:34.44 | j0b | 101 is the account that the other pbx is registered to |
| 11:34.53 | j0b | so, we call out that way |
| 11:35.07 | j0b | hm |
| 11:35.21 | j0b | could i send like $CallerIDnum when i dial |
| 11:35.39 | j0b | i have fought this battle for at least 6 hours now |
| 11:36.26 | j0b | because, if a register a software phone directly to this pbx, i see the correct number calling |
| 11:36.47 | j0b | but not when forwarding call to my provider |
| 11:36.57 | j0b | sighs |
| 11:37.23 | schmidts | j0b sorry i dont understand what exactly you want to do |
| 11:37.30 | j0b | sorry |
| 11:37.32 | j0b | :( |
| 11:37.39 | schmidts | btw your set is just wrong if you want to set the outgoing callerid |
| 11:37.54 | j0b | i want to set the outgoing caller id |
| 11:38.09 | schmidts | SET(CALLERID(num)=123) |
| 11:38.21 | j0b | but i want it to be dynamically |
| 11:38.26 | j0b | so |
| 11:39.43 | schmidts | and you are sure that your provider allows you to set the callerid manually? |
| 11:39.46 | j0b | someone calls 6333 -> pbx1 -> pbx2 (that one i pasting configs for above), pbx2 calls back to pbx1 to a number we specify |
| 11:40.10 | j0b | we have control over pbx2 |
| 11:40.20 | schmidts | ok |
| 11:40.32 | j0b | but when pbx2 calls pbx1 it shows 101 as caller id |
| 11:40.39 | j0b | calls back even |
| 11:40.50 | j0b | we just want it to proxy our calls |
| 11:40.56 | j0b | this for logging and such |
| 11:41.09 | j0b | s/proxy/forward/ |
| 11:41.25 | kaldemar | j0b: where is the caller id 101? on pbx1? |
| 11:41.55 | j0b | 101 is a sip-account on pbx2 that pbx1 is registered to |
| 11:42.28 | kaldemar | j0b: where are you looking at the caller id when it is 101? on pbx1? |
| 11:42.47 | j0b | yes |
| 11:42.52 | j0b | correct |
| 11:43.12 | kaldemar | j0b: if you don't control pbx1, there's nothing you can do about it. |
| 11:43.47 | j0b | so i want to send someone calls 6333 -> pbx1 -> pbx2 where "someone calls" is a number |
| 11:43.54 | j0b | from where its calling from |
| 11:44.03 | j0b | so i want to keep that caller id |
| 11:44.11 | j0b | all the way back to pbx1 |
| 11:44.19 | schmidts | j0b this will not work if pbx1 allways set 101 for an incoming call |
| 11:44.31 | j0b | ah allright |
| 11:44.32 | j0b | hm |
| 11:44.34 | schmidts | j0b maybe you can show (pastebin) a sip trace of a call from pbx2 back to pbx1 |
| 11:45.03 | j0b | schmidts: how do i trace (be patient with me, have used asterisk for 8 hours) |
| 11:45.07 | j0b | :( |
| 11:45.15 | schmidts | :D |
| 11:45.24 | kaldemar | CLI commands: "core set verbose 10" "sip set debug on" |
| 11:45.39 | schmidts | thx kaldemar |
| 11:45.55 | kaldemar | you get to CLI with "asterisk -r" from shell. |
| 11:45.57 | j0b | allright done |
| 11:46.00 | j0b | so now i call |
| 11:46.24 | kaldemar | now make a call and pastebin all output. |
| 11:47.03 | j0b | ok |
| 11:47.08 | j0b | pasebin and all that |
| 11:47.11 | j0b | just a sec:) |
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| 11:48.08 | j0b | http://pastebin.com/C0Ld257Y |
| 11:48.10 | j0b | kaldemar: schmidts |
| 11:48.13 | j0b | :) |
| 11:48.51 | j0b | its rining in my phone, 6341 |
| 11:48.58 | kaldemar | in the first invite you have From: "6939" <sip:101@10.42.0.38>;tag=AI7E785BC50B8E2223 |
| 11:48.59 | j0b | and shows 101 |
| 11:49.16 | kaldemar | that means they have set caller id number as 101. the name part is 6939. |
| 11:49.24 | j0b | ah |
| 11:49.30 | j0b | ahhhhh |
| 11:49.39 | j0b | got damn |
| 11:49.48 | kaldemar | if you want to set 6939 as the number when dialing back, put Set(CALLERID(num)=${CALLERID(name)}) before the dial. |
| 11:50.32 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:50.56 | j0b | JESUS |
| 11:50.58 | j0b | fuck |
| 11:51.03 | j0b | it works |
| 11:51.05 | j0b | oh man |
| 11:51.15 | j0b | kaldemar: thank you very much |
| 11:51.19 | j0b | sighs |
| 11:51.40 | j0b | i thought is was num and not name |
| 11:51.51 | j0b | awsome |
| 11:51.56 | j0b | sighs again |
| 11:51.57 | kaldemar | no problem. |
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| 11:57.20 | j0b | kaldemar: is it usual like that, this is an astra pbx, why does it set name as callerid and not num |
| 11:57.23 | j0b | ?:) |
| 11:58.37 | kaldemar | j0b: quite odd |
| 11:58.54 | j0b | mhm |
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| 13:07.36 | catphish | i'm a little confused about something |
| 13:08.15 | catphish | i'm looking at 2 servers with very similar config, specifically they both have [logfiles] asterisk.log => debug, warning, error, notice, verbose |
| 13:09.09 | catphish | however one seems to log what it's executing: [Mar 15 12:57:04] VERBOSE[9637] pbx_realtime.c: -- Executing [nnnnnn@incoming:1] Set("SIP/peer4-48b2a2-0000ab51", "CHANNEL(musicclass)=xxxx") |
| 13:09.14 | catphish | whereas the other doesn't |
| 13:12.03 | [TK]D-Fender | catphish, What does "similar config" tell us? |
| 13:12.57 | [TK]D-Fender | catphish, You should probably be showing us the configs on the side that isn't doing what you expect, along with actual call debug (and that means SIP debug as well) |
| 13:13.19 | catphish | well what i'm interested in is what isn't being logged |
| 13:13.45 | catphish | i wondered if there was anything to control whether what is being executed gets logged or not |
| 13:14.01 | [TK]D-Fender | check your CLI log & debug levels, show us the actual configs for logging, show us that it is processing an actual call that it is not logging, etc |
| 13:14.21 | ruben23 | guys any idea how to setup asterisk-addon to an existing asterisk install....any suggestions..? |
| 13:14.41 | catphish | [TK]D-Fender: what the config i posted not sufficient? |
| 13:14.52 | catphish | is there other config that affects logging? |
| 13:15.05 | [TK]D-Fender | ruben23, follow the instructions in the tarball and install it |
| 13:15.46 | [TK]D-Fender | catphish, show the complete config. Show that * loaded it on start. Confirm your CLI verbose and debug levels |
| 13:18.38 | catphish | [TK]D-Fender: it's likely the verbosity level |
| 13:18.53 | catphish | is that configurable from a config file? |
| 13:20.55 | [TK]D-Fender | Not that I know of. |
| 13:21.52 | ruben23 | [TK]D-Fender: yes i did, and i get this error on installing--------------------->http://pastebin.com/fqwvmJzk |
| 13:22.33 | catphish | [TK]D-Fender: thanks, interesting if something that gets logged to a file can only be configured from the console |
| 13:22.50 | catphish | will check back later and see if anyone knows more |
| 13:22.52 | catphish | thanks |
| 13:25.13 | [TK]D-Fender | ruben23, And what ar you you installing that over? |
| 13:25.52 | [TK]D-Fender | ruben23, And what are you installing that over? |
| 13:26.00 | ruben23 | [TK]D-Fender: to create asterisk cdr on mysql |
| 13:26.14 | [TK]D-Fender | ruI didn't ask "why". What version of * <- |
| 13:33.08 | cusco | hey |
| 13:33.39 | cusco | is it possibple to playback() something while doing something else that takes time, and then call stop that playback, and carry on? |
| 13:33.50 | cusco | or any other thing besides playback |
| 13:35.06 | cusco | catphish: you can set the verbosity level at startup in asterisk.conf or set it via cli: core set verbose 15 |
| 13:35.07 | kaldemar | what do you mean by something else? |
| 13:35.36 | cusco | kaldemar: in this case... it runs ${SHELL()} and takes about 12 seconds |
| 13:35.54 | kaldemar | StartMusicOnHold/StopMusicOnHold |
| 13:36.14 | cusco | ow.. cool! |
| 13:36.18 | cusco | THANKS |
| 13:36.19 | cusco | oops |
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| 13:44.43 | catphish | cusco: thanks! |
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| 13:48.22 | ruben23 | <PROTECTED> |
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| 13:49.21 | [TK]D-Fender | ruGot an answer to the question I had to ask twice? |
| 13:49.28 | [TK]D-Fender | ruben23, Got an answer to the question I had to ask twice? |
| 13:53.17 | ruben23 | <PROTECTED> |
| 13:53.30 | ruben23 | sorry |
| 13:53.32 | [TK]D-Fender | ruben23, I did not ask that. |
| 13:54.22 | catphish | i have someone complaining that a SPA303 with a default dialplan refuses to dial *8 :( |
| 13:54.23 | ruben23 | [TK]D-Fender: i installed it on an asterisk ver: ---> asterisk-1.4.27.1 |
| 13:54.38 | catphish | they can dial *7 but not *8 - as if its getting picked up as something special |
| 13:56.51 | ruben23 | [TK]D-Fender: did i asnwer your question..? |
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| 13:59.49 | [TK]D-Fender | ruben23, What makes you think you can take 1.6.2 addons and put them on 1.4? |
| 14:00.13 | kaldemar | catphish: is it the SPA303 that refuses to dial or does the call not get into dialplan to an extension that matches *8? |
| 14:00.23 | [TK]D-Fender | ruben23, Do you put 2012 Ford Mustang parts in your 1967 Shelby Cobra? |
| 14:01.08 | catphish | kaldemar: the phone refuses to dial it |
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| 14:04.35 | ruben23 | [TK]D-Fender:ok ill match it with the same version |
| 14:05.31 | j0b | would be greatful for any tip regarding a CDR-log analyzer, like php and webstuff |
| 14:05.42 | j0b | :) |
| 14:06.03 | j0b | i have the schizzle in a postgresql database |
| 14:08.56 | j0b | seems like queue log analyzer wants mysql |
| 14:10.54 | [TK]D-Fender | j0b, There is another queue log one that is PGSQL based actually... lemme see if I can drudge up the name for you |
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| 14:11.57 | j0b | http://www.cdr-stats.org/download/download-cdr-stats/ |
| 14:12.06 | j0b | [TK]D-Fender: cool |
| 14:14.11 | j0b | i guess that asterisk are able to log to two different databases at the same time |
| 14:14.29 | j0b | if i load both pgsql.so and mysql.so |
| 14:14.32 | j0b | or sqlite |
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| 14:17.48 | [TK]D-Fender | j0b, http://www.asteriskguru.com/tools/queue_stats.php |
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| 14:18.08 | [TK]D-Fender | j0b, Correct you can log to multiple interfaces simultaneously |
| 14:19.21 | j0b | [TK]D-Fender: cool, thank you very much:) |
| 14:19.44 | [TK]D-Fender | j0b, I implemented it once for one of my clients and they were fairly happy with it. |
| 14:20.02 | j0b | nice |
| 14:20.02 | [TK]D-Fender | j0b, There are a few nice paid ones as well if your needs expand |
| 14:20.32 | j0b | nah, this is just for ONE number at our company, the IT-support number:p |
| 14:20.38 | j0b | so i guess we will manage:D |
| 14:22.15 | [TK]D-Fender | j0b, its not just how many queues or agents, but also the kinds of reports you'll get with the package. Take a look and if you're happy out of the gate with it then there you go. |
| 14:29.27 | j0b | [TK]D-Fender: great, im sure i will be happy with it:) |
| 14:31.09 | catphish | does asterisk have any handling of the extension *8 by default? |
| 14:31.23 | catphish | without it being listed as an extension? |
| 14:31.25 | kaldemar | catphish: yes, it's the pickup extension defined in features.conf. |
| 14:32.01 | catphish | kaldemar: is it the default if features.conf is absent? |
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| 14:32.09 | kaldemar | catphish: it is the default. |
| 14:32.30 | catphish | interesting |
| 14:34.56 | catphish | thanks |
| 14:36.18 | catphish | kaldemar: it looks like my *8 extension isn't being matched |
| 14:36.27 | catphish | and its falling back to asterisk's default callpickup |
| 14:40.40 | [TK]D-Fender | catphish, set it to something else |
| 14:41.06 | catphish | just trying that |
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| 14:51.57 | one47 | Anyone got time to discuss a potential crash in ast_readaudio_callback - Found in 1.6.2.22, but probably still applies in newer 1.8/10 code? I think I have a solution, but want to know if it is appropriate to raise as an "issue" |
| 14:52.30 | Qwell | one47: Unless you can reproduce with 1.8, no, you shouldn't open an issue. |
| 14:52.59 | one47 | okay. I don't have 1.8 to reproduce it, but think it is still a concern for 1.8 and 10 users. |
| 14:53.12 | one47 | what should I do? mainling list? |
| 14:53.15 | Qwell | well, you're more than welcome to report it once you're able to reproduce |
| 14:53.16 | one47 | *mailing |
| 14:55.09 | one47 | LOL. So if I can see a potential failure to ao2 refcount an object in the 1.8 code, I should keep it a secret? |
| 14:55.35 | Qwell | No, I'm saying that if you can show that it's an issue in 1.8, you should report it. Talking about 1.6.2 is futile, as bugs won't even be looked at. |
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| 14:56.25 | one47 | Agree, and acknowledge. I know there is a bug in 1.8, but can only personally demonstrate in in 1.6.2 - Suggestions on how to proceed? I don't want to keep it a secret |
| 14:59.34 | [TK]D-Fender | oneI'd say post it on the mailing list and hope someone adopts and tests it for 1.8 to submit for you |
| 14:59.41 | [TK]D-Fender | one47, I'd say post it on the mailing list and hope someone adopts and tests it for 1.8 to submit for you |
| 14:59.47 | Qwell | If you can show the flaw in the code, great, report it. |
| 15:00.05 | one47 | Thanks to both. I will start typing up an email :) |
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| 15:40.21 | cusco | hi |
| 15:41.00 | cusco | I tested on two different dialplans StartMusicOnHold() after Answer(), one works the other I hear no audio, tho on CLI I still read: -- Started music on hold, class 'aguarde', on SIP/150-00002e1d |
| 15:41.51 | cusco | the dialpan where it doesn't work: http://paste.debian.net/hidden/9e293433/ |
| 15:42.42 | cusco | how can I find out why i'm not listening to the MOH? |
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| 15:44.54 | kaldemar | cusco: look at the CLI when you make a call. |
| 15:45.09 | cusco | I did |
| 15:45.28 | kaldemar | StopMusicOnHold(); might have something to do with it. or an invalid class for musiconhold. |
| 15:45.36 | kaldemar | show the dialplan that does work |
| 15:45.42 | cusco | ok hold |
| 15:46.31 | cusco | http://paste.debian.net/159852/ |
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| 15:47.12 | cusco | Wait(3), then Hangup() it plays the moh for 3 secs |
| 15:48.06 | kaldemar | in your first dialplan, you don't have anything between StartMusicOnHold and StopMusicOnHold or Hangup that would take practically any time. |
| 15:48.18 | kaldemar | unless Queue_log does. |
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| 15:48.51 | scubes13 | anyone ever setup polycom provisioning on legacy (301/501/etc) using FTPS? |
| 15:49.06 | tm1000 | scubes13: yes |
| 15:49.08 | cusco | kaldemar: SHELL does take about 12 secs |
| 15:49.09 | tm1000 | err |
| 15:49.13 | tm1000 | not FTP-S |
| 15:49.35 | kaldemar | cusco: sure, but you execute StopMusicOnHold before it, don't you? |
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| 15:49.43 | scubes13 | tm1000 yey! oh…bummer :P |
| 15:50.02 | tm1000 | not sure if they support SSL |
| 15:50.18 | cusco | no |
| 15:50.19 | cusco | err |
| 15:50.21 | cusco | let me check |
| 15:50.23 | cusco | but no |
| 15:50.44 | cusco | ahhh |
| 15:50.47 | cusco | dumb me |
| 15:50.58 | cusco | lol |
| 15:51.04 | cusco | thanks for the tip kaldemar |
| 15:51.32 | kaldemar | cusco: why are you setting CHANNEL(musicclass) if you're giving the class as an argument to StartMusicOnHold? |
| 15:51.37 | scubes13 | having to provision phones remotely… and a bit antsy about using standard FTP :( |
| 15:52.25 | cusco | kaldemar: I thought it could be the problem |
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| 16:05.05 | cusco | !nan |
| 16:05.16 | Qwell | cusco: you broke shido. good job. |
| 16:14.17 | cusco | thanks. I don't know what shido is tho |
| 16:14.21 | cusco | a minor fault in judo? |
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| 16:34.15 | Micc | I'm trying to remember the nick name of a guy that is in here sometimes. Something like ChrisSydney. He lived in Austrelia I believe. |
| 16:34.27 | Micc | Anyone remember him? |
| 16:34.48 | Qwell | Micc: ChrisInSydney? |
| 16:34.54 | Qwell | -NickServ- Last seen : Mar 14 16:21:27 2012 (1 day, 00:13:11 ago) |
| 16:36.11 | Micc | yeah, thats him |
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| 17:26.08 | faithlove | heya. any great API for asterisk? |
| 17:27.24 | Qwell | faithlove: rephrase your question |
| 17:27.34 | chuckf | nope, but an awesome one exists |
| 17:33.31 | sgtpepper | guys... I'm having the following issue. Even with progressinband=never I'm getting early audio from a peer |
| 17:33.41 | sgtpepper | I've a pastebin with the log (if needed) |
| 17:34.14 | sgtpepper | I want to replace that early audio (saying that the route have'nt got credit) with a ring... because the call eventually gets rerouted |
| 17:34.47 | sgtpepper | prematuremedia=no also on all peers |
| 17:34.51 | sgtpepper | and r on the dial line |
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| 17:39.51 | faithlove | i'm planning to start a project based on PHP which is based on asterisk. so, i'm wondering if there is something already made like a Class or anything like that which i could use... i'm not experienced with asterisk, so i'm just asking for your oppinions |
| 17:40.12 | faithlove | *PHP project, sorry |
| 17:40.45 | kaldemar | faithlove: what would the outcome of your project do? |
| 17:42.02 | faithlove | kaldemar: it would be a small web app for a call center. they want to get raports, to search for inbound/outbound calls (source, destination), date range, listen calls |
| 17:42.41 | kaldemar | that won't be a small app. |
| 17:44.01 | faithlove | kaldemar: don't think of something very elaborated. it's for a friend of mine :) he asked me if i can help him with this as i did some PHP coding before. so it'll be like a small grid with some pagination. something like that |