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| 00:25.05 | volga629 | in asterisk 10.5 is this valid options echocancel=yes echocancelwhenbridged=yes for sip.conf or need echo cancellation hardware ? |
| 00:25.57 | volga629 | I am trying improve little bit quality for far branch where latency about 180 |
| 00:26.16 | volga629 | and they using vpn tunnel to reach asterisk |
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| 00:34.17 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 00:34.28 | pyther | Hi |
| 00:35.16 | pyther | In my dial plan I want to forward a call onto my cell phone, but DIAL(SIP/15554443232@flowroute,20) is not working, the console shows it bridges the call, by my cell phone never rings |
| 00:35.35 | pyther | FollowMe works as expected, but I don't want to press 1 to answer the call |
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| 00:51.58 | leifmadsen | the syntax should really be Dial(SIP/flowroute/<number_to_call>) |
| 00:52.05 | leifmadsen | with [flowroute] being configured in sip.conf |
| 00:52.35 | leifmadsen | if the cell never rings, have you validated that it doesn't go to voicemail or something? |
| 00:52.46 | leifmadsen | Dial() certainly works fine, especially if you got FollowMe() to work |
| 00:57.27 | leifmadsen | volga629: not sure how echo cancellation helps with latency and call quality. I believe you want to enable a jitterbuffer (which will help with some call quality -- it'll increase latency slightly though, which you can't decreased except with a better network connection) |
| 01:01.11 | volga629 | sorry didn't explain clear, yes for branch office I thought enable QOS Enable DSCP for video and audio |
| 01:02.06 | volga629 | echo cancellation is for SIP trunk out, because currently we not using any PRI or so |
| 01:03.09 | volga629 | so the question I need modify switches for qos and phones on remote end ? |
| 01:03.47 | *** join/#asterisk chris_n (~Chris@nc-63-162-197-228.sta.embarqhsd.net) |
| 01:04.11 | volga629 | this what i have on switch: canlswitch01#sh voice vlan type auto Best Local Voice VLAN-ID is 300 Best Local VPT is 1 Best Local DSCP is 46 (default) |
| 01:04.33 | volga629 | I set same value on phone |
| 01:07.19 | volga629 | I believe this value Audio RTP DSCP it just prioritize the RTP traffic on switch ? |
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| 01:29.17 | pyther | leifmadsen: thanks for the response, sadly, I've ran out of time to troubleshoot the issue for now. It seems like asterisk shows that it bridges the call, but the call does not seem to get bridged correctly. I've had some occasion when the cell would ring, but when I'd pick up neither party could communicate. |
| 01:35.10 | leifmadsen | sounds like a NAT issue to me then |
| 01:35.22 | leifmadsen | runs off to bed |
| 01:37.45 | volga629 | rtpkeepalive= what recommended value in NAT environment ? |
| 01:38.01 | leifmadsen | that really has nothing to do with nat |
| 01:38.14 | leifmadsen | it's to destroy channels that aren't receiving RTP |
| 01:38.17 | leifmadsen | regardless of NAT |
| 01:39.51 | volga629 | I just going through all sip setting trying find which parameter I can adjust to improve quality, latency is about 180. I the phone him self I can set only Qos |
| 01:40.07 | volga629 | On |
| 01:40.41 | volga629 | this from sip.conf ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open |
| 01:50.30 | volga629 | thank you for explanations I still need think what exactly need to do, to try minimize this trouble |
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| 02:30.14 | chris_n | does anyone know if Fax for * will work with * 10? |
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| 02:52.22 | volga629 | is anybody know what is mean ccss.c:936 ast_set_ccbs_available_timer: 0 is an invalid value for ccbs_available_timer. Retaining value as 4800 ? |
| 02:54.01 | volga629 | Set("SIP/101-00000067", "CALLCOMPLETION(ccnr_available_timer)=") in new stack look like warning from here |
| 02:54.52 | WIMPy | Yes. |
| 02:54.59 | WIMPy | Where is the value? |
| 02:55.05 | jpsharp | chris_n: Both Free Fax for Asterisk from digium as well as the SpanDSP based stuff will work with *.10 |
| 02:55.41 | jgowdy | free fax for asterisk? link to that? |
| 02:56.04 | jgowdy | nm |
| 02:56.36 | jpsharp | I should have said "Fax For Asterisk" from digium, rather than specifying only the Free version. |
| 02:58.22 | chris_n | tnx jpsharp |
| 02:59.03 | volga629 | yes this the question which file I need set ccbs_available_timer = 4800;ccnr_available_timer = 7200 |
| 02:59.45 | volga629 | This my current trouble: I just going through all sip setting trying find which parameter I can adjust to improve quality, latency is about 180. I the phone him self I can set only Qos |
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| 03:02.56 | volga629 | On |
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| 06:13.41 | bagataway | hello hello… looking for a little help with a fire drill project |
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| 06:14.37 | bagataway | any suggestions on the best call log reporting tools? |
| 06:17.23 | jkroon | WIMPy, removing that check for zlen<3 seems to have fixed the problem ... |
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| 07:01.57 | schmidts | good morning |
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| 07:09.55 | youjelly | morning |
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| 08:11.15 | gavimobile | 1700 numbers and 1800 numbers in my country are 9 digits total.. will this work? exten => _1[7-8]00NXXXXX,1,; call the queue |
| 08:11.15 | gavimobile | <PROTECTED> |
| 08:11.26 | gavimobile | oops |
| 08:11.30 | gavimobile | , no good |
| 08:12.13 | gavimobile | also ) replace with } |
| 08:14.06 | gavimobile | hrm... circuit busy |
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| 08:14.37 | gavimobile | found ya |
| 08:15.11 | gavimobile | nope |
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| 09:36.56 | afournier | is it difficult to add a systemd layer to oe-core ? |
| 09:37.18 | afournier | not to oe-core but next to oe-core |
| 09:37.57 | afournier | is it too heavy ? |
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| 10:11.26 | WIMPy | jkroon: Oh, so it was just a bug then, eh? |
| 10:12.06 | afournier | oops wrong chan |
| 10:12.28 | WIMPy | Apart from the question why that check was in the code at all, I mostly still wonder how it could go unnoticed for so long and off course what the difference on that line may be. |
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| 10:14.22 | oquidave | i want to implement a server-side supported "call transfer" . i've read that i've to uncomment the line blindxfer => #1 in [featuremap] section of the features.conf file...now when a user(10)1 calls user2(20) and they pickup...for user2 to transfer the call to user3(30), they dial #130 is that it? |
| 10:17.07 | oquidave | i've tried that but it doesn't seem to work |
| 10:18.15 | oquidave | and yes i set the flags...exten => s,1, Dial(SIP/${ARG1},tT,30);enable call transfer when caller sends 1# DTMF codes |
| 10:20.08 | oquidave | wait, i might have mixed up the order of arguments in teh Dial() app |
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| 10:45.39 | mathi | hi |
| 10:46.25 | mathi | is it possible to execute dialplan after the caller hung up? |
| 10:50.39 | bulkorok | mathi: check h-extension |
| 10:52.15 | mathi | bulkorok, thanks, I also found Hangup handlers |
| 10:55.56 | *** join/#asterisk dexteruk (~dexteruk@77.70.92.171) |
| 10:57.36 | dexteruk | having problems with CLID on sip termination my ISP says that i need to set the From header to read From: "anonymous" sip:anonymous@ but all i get is From: "anonymous" sip:asterisk@ |
| 10:57.43 | dexteruk | how can i change this |
| 10:57.50 | dexteruk | i have changed the systemname |
| 10:58.13 | mathi | I read: "As always, hangup handlers, like the h extension, need to execute quickly because they are in the hangup sequence path of the call leg. Specific channel driver protocols like ISDN and SIP may not be able to handle excessive delays completing the hangup sequence. |
| 10:58.26 | mathi | how much time does that give to me precisely? |
| 10:58.28 | dexteruk | i have set the CALLERID to both name and number |
| 10:58.38 | dexteruk | when i set the number the header looks ok |
| 11:09.53 | dexteruk | id From: "number" sip:number@ip |
| 11:10.16 | dexteruk | can anyone tell me how i can replace asterisk with anonymous |
| 11:10.52 | mcolombo | dexteruk : of course |
| 11:10.54 | mcolombo | ;exten => _X.,n,Set(CALLERID(all)= Anonymous <anonymous@anonymous.invalid>) |
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| 11:11.48 | mcolombo | this line is in my asterisk 10.9 |
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| 11:14.00 | dexteruk | i just tried that and the from line From: "anonymous@anonymousinvalid" <sip:asterisk@XXXXXXXX> |
| 11:15.33 | dexteruk | ah it might be a problem with the version of asteisk im running |
| 11:15.39 | dexteruk | as it not 10 its 1.6 |
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| 11:54.09 | jkroon | WIMPy, looks like it, however, I am now getting proper FCS errors |
| 11:55.04 | WIMPy | Hmm. So it was not the real issue then. |
| 11:55.25 | jkroon | not sure if there is more differences between wctdm24xxp and wcb4xxp, and it raises the question of why isn't the hdlc handling abstracted into another module? |
| 11:55.33 | WIMPy | Have you tried to get the lines swapped? |
| 11:55.41 | jkroon | WIMPy, no, but they just put the old hardware backed and everything is working again. |
| 11:55.58 | WIMPy | Yes. would seem like a good idea. |
| 11:55.58 | jkroon | i'm going on leave so this will have to go further from the 8th onwards. |
| 11:56.11 | jkroon | old hardware being the server, not the ISDN equipment. |
| 11:56.36 | WIMPy | Ok, well. I guess we all have to wait then. |
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| 12:21.23 | jkroon | afraid so :p |
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| 12:56.35 | brian98 | anyone alive! |
| 12:56.35 | brian98 | ? |
| 12:57.08 | bulkorok | alive, yes... |
| 12:57.22 | brian98 | Hi 1 |
| 12:57.37 | brian98 | Can you help me with this line |
| 12:57.37 | brian98 | exten => _X.,n,ExecIf($[${PRIVACY} = id ]?Set{CALLERID(num)}=UNKNOWN ) |
| 12:57.53 | brian98 | I'm not getting the syntax 100% correct. |
| 12:58.11 | brian98 | [Sep 28 13:57:55] WARNING[28283]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 12:58.33 | [TK]D-Fender | brian98, Because when you ran it ${PRIVACY} was BLANK |
| 12:58.45 | brian98 | ok |
| 12:58.50 | [TK]D-Fender | brian98, Thus elaving empty space on the left side of the = |
| 12:58.54 | [TK]D-Fender | leaving* |
| 12:58.59 | brian98 | one mo |
| 12:59.20 | brian98 | I did test this.. |
| 12:59.26 | brian98 | <PROTECTED> |
| 12:59.27 | brian98 | [Sep 28 13:59:08] WARNING[28291]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 12:59.29 | tzanger | jkroon: pong |
| 12:59.40 | [TK]D-Fender | Set{CALLERID(num)}=UNKNOWN <- next, what does execif say about the parameters following the "?" |
| 13:00.10 | [TK]D-Fender | You also have a { with a $ in front, so that's clearly wrong too... I'd look REAL close at this again.... |
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| 13:00.33 | brian98 | ok |
| 13:00.34 | brian98 | thanks |
| 13:00.53 | brian98 | I'll go back to the drawing board for a moment |
| 13:00.57 | jkroon | tzanger, sup? |
| 13:00.58 | brian98 | thanks for time. |
| 13:01.37 | tzanger | jkroon: you had BRI questions? |
| 13:01.54 | brian98 | TK- Got it working |
| 13:02.04 | jkroon | tzanger, hehe, you could put it that way. i think i got a little greying since yesterday this time. |
| 13:02.21 | [TK]D-Fender | brian98, You just need to break it down 1 little piece at a time... |
| 13:02.33 | brian98 | Sometimes you need to step away for a cup of tea :) |
| 13:02.36 | jkroon | tzanger, are you familiar with Telkom's ISDN2 vs ISDN2a? |
| 13:02.45 | tzanger | jkroon: no, sorry |
| 13:03.19 | tzanger | jkroon: I received brain damage doing the BRI stuff for the B400M from Digium. That's why sruffel was mentioning me :-) |
| 13:03.20 | jkroon | ok, as I understand it these are NT1 and NT2 units (ISDN2 only provides two S/T ports, the ISDN2a provides two S/T ports, and two analog ports). |
| 13:03.41 | jkroon | ah ok :). we'll he's got the case number if you'd like to take a look? |
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| 13:05.29 | jkroon | if you need someone to get more brain damage I'm guessing I'm it. Seems I'm one of a handful of suckers still dealing with BRIs. |
| 13:05.42 | oquidave | hello,i badly need to install the chan_mobile.so module on my asterisk built and compiled from source |
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| 13:06.35 | oquidave | when i "make menuselect" and select "add-ons" i can see it with 3 XXX "XXX chan_mobile" meaning it's missing a dependency |
| 13:07.44 | oquidave | i've cross checked and i've bluez packages, libbluetooth but not "libbluetooth-dev" |
| 13:08.10 | jkroon | oquidave, check what does ./configure output |
| 13:08.14 | WIMPy | jkroon: You're surely not alone. We have 11.6M BRIs here. |
| 13:08.31 | jkroon | it should list there why it thinks chan_mobile is not available. |
| 13:08.49 | jkroon | WIMPy, alone != handful :p. |
| 13:09.44 | jkroon | but yea, i know there are many, many BRIs still out there but I'd like to know how many people are using it with Telkom's BRIs, and specifically with the ISDN2a units, which I'm convinced is what's causing the crap. |
| 13:10.26 | WIMPy | I uess we will find out weh you're back. |
| 13:10.45 | WIMPy | +g |
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| 13:11.22 | oquidave | jkroon> what should i look out for ...because there's alot of output... |
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| 13:12.41 | jkroon | no clue :). you will need to read I'm afraid. Or be sure to pass whatever the appropriate --with-mobile option is, which should then hopefully stop if it can't enable chan-mobile with an appropriate reason. |
| 13:12.48 | oquidave | > jkroon> otherwise it says |
| 13:12.49 | oquidave | > checking for ba2str in -lbluetooth... yes |
| 13:12.51 | oquidave | > checking bluetooth/bluetooth.h usability... yes |
| 13:12.52 | oquidave | > checking bluetooth/bluetooth.h presence... yes |
| 13:12.54 | oquidave | > checking for bluetooth/bluetooth.h... yes |
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| 13:13.30 | jkroon | WIMPy, yea, we need to figure this one out. |
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| 13:17.00 | oquidave | jkroon> it says...configure: WARNING: unrecognized options: --with-mobile |
| 13:17.30 | oquidave | jkroon> but anywhere it's not raising any flags that suggest a problem with chan_mobile |
| 13:18.16 | jkroon | ok, so in the menuopts files (can't remember their exact names) find the chan_mobile option and figure out what the dependencies are and check them. |
| 13:18.39 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qwqtlygebpwawtbz) |
| 13:18.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:19.59 | *** join/#asterisk dwayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 13:21.11 | oquidave | jkroon> menuopts file? i don't think am familiar with them! |
| 13:21.59 | jkroon | menuselect* files |
| 13:23.46 | jkroon | MENUSELECT_DEPENDS_chan_mobile=BLUETOOTH |
| 13:24.01 | jkroon | ok, so you need to figure out why it thinks BLUETOOTH isn't available. |
| 13:24.18 | oquidave | jkroon am doing my best... |
| 13:26.49 | jkroon | don't apologize to me, it makes no difference to me whether you get it right or not, it does make a difference to you. |
| 13:27.20 | jkroon | i've only ever compiled chan_mobile once, and it "just worked", i canned it because audio sync was problematic, but have been wanting to give it another shot for a while now. |
| 13:27.20 | oquidave | jkroon i've recompiled asterisk...it's now not showing the XXXs...so |
| 13:31.16 | *** join/#asterisk n3hxs (~n3hxs-5@TOROON63-3096776761.sdsl.bell.ca) |
| 13:35.41 | oquidave | jkroon i can see the lines "[CC] chan_mobile.c -> chan_mobile.o" and "[LD] chan_mobile.o -> chan_mobile.so" does this mean it's successfully installed? |
| 13:43.41 | oquidave | jkroon it's installed thanks |
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| 13:57.15 | bulkorok | hey... If I use VoiceMail() with realtime and odbc-storage... after recording the message I get WARNING[11707]: app_voicemail.c:3457 retrieve_file: SQL Get Data error! coltitle=label |
| 13:57.23 | bulkorok | [SELECT * FROM voicemessages WHERE dir=? AND msgnum=?] |
| 13:57.42 | bulkorok | somebody got an idea what is going on!? |
| 13:58.19 | bulkorok | hopes every friday that somebody can help... |
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| 14:03.22 | vader-- | morning |
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| 15:12.04 | *** join/#asterisk stugster (~aaaa@office.considerit.co.uk) |
| 15:13.36 | stugster | does the new verison of Asterisk support multiple registrations to the same sip extension? |
| 15:15.56 | *** join/#asterisk rking (~rking@unaffiliated/rking) |
| 15:16.50 | malcolmd | i think i know what you're asking, and the answer is no; |
| 15:17.02 | rking | Hi guys. Would you recommend Asterisk for a total "user" as an alternative to Skype? What I mean is, I just want to talk to my coworkers over my computer, I don't really want to spend N weeks diving into a new config system; I just want off Skype. |
| 15:17.15 | rking | malcolmd: How did you know what I was typing? Do you have some special IRC client? |
| 15:17.27 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:17.28 | ChannelZ | Asterisk is a beast compared to Skype |
| 15:17.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:17.35 | malcolmd | rking - hah, that was to stugster, i should have prefixed ;) |
| 15:17.37 | rking | (j/k) |
| 15:17.38 | ChannelZ | They are not at all really equivilant |
| 15:17.43 | [TK]D-Fender | rking, Go use Google Voice or something then |
| 15:18.11 | rking | ChannelZ: Yeah, I know that much — I tried setting it up back in the day (2005), and was thrown down to the ground, so I don't know if it's worth trying again or not. |
| 15:18.28 | ChannelZ | It's not really a client-side application. |
| 15:18.31 | rking | [TK]D-Fender: Ugh. Out of the Microsoft frying pan into the Google fryer. |
| 15:18.39 | stugster | malcolmd, yeah, we're looking to set up two different SIP phones using the same registration ID... |
| 15:18.49 | rking | ChannelZ: It seems like all of the open-source VoIP solutions require a server setup. |
| 15:19.14 | malcolmd | stugster: not a good idea; they'll contend their registrations |
| 15:19.23 | ChannelZ | Well if you're crafty you can call client-to-client |
| 15:19.35 | rking | Yeah, that's what I want to do. |
| 15:19.46 | rking | For conference calls we can keep using stupidSkype. |
| 15:19.56 | ChannelZ | You just have to contend with your own firewalls on each side |
| 15:20.10 | rking | I thought about doing like arecord | nc -l -p 8001 |
| 15:20.23 | rking | Firewalls I can easily poke through with ssh port forwarding. |
| 15:20.45 | rking | But what client would you recommend? |
| 15:21.24 | pabelanger | stugster: you need a SIP proxy then |
| 15:21.41 | ChannelZ | Well SIP is UDP and depends on each side knowing their own external IPs so SSH tunneling might not work so hot |
| 15:22.26 | ChannelZ | But any SIP softphone would take the place of your Skype client (video depending) and then you could technically just call each other by IP or hostname |
| 15:22.54 | ChannelZ | Or what's that free SIP provider... |
| 15:23.03 | [TK]D-Fender | rking, Or sign up with an open SIP proxy like ekiga.net |
| 15:23.05 | ChannelZ | is it Ekiga where you can get an account? |
| 15:23.08 | ChannelZ | yeah |
| 15:23.19 | rking | OK, cool. Thanks. |
| 15:23.20 | [TK]D-Fender | rking, that could do for you |
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| 15:35.02 | Greenlight | Hiya all. Just noticed some errors appearing - "WARNING[3696]: app_queue.c:5319 dump_queue_members: Could not create persistent member string, out of space". Getting this about 20 times a minute. My thinking is that its an issue related to the Asterisk sqllite database, is this correct? |
| 15:36.56 | Greenlight | I've plenty of resources on server, and there is plenty disk space. Is there a setting somewhere which limits the size of the db? |
| 15:37.44 | leifmadsen | I doubt it... |
| 15:38.38 | Greenlight | Other than the error I'm seeing no ill effects, but I reckon that if I restartted Asterisk I'd loose my queue memberships. What might cause this error ? |
| 15:39.10 | leifmadsen | lose* |
| 15:39.15 | leifmadsen | df -h |
| 15:39.18 | leifmadsen | reports lots of HD space? |
| 15:39.42 | leifmadsen | Greenlight: please specify all resources you've looked at |
| 15:39.49 | leifmadsen | memory? swap? disk? |
| 15:40.10 | Greenlight | Filesystem Size Used Avail Use% Mounted on |
| 15:40.10 | Greenlight | tmpfs 12G 141M 12G 2% /dev/shm |
| 15:40.11 | Greenlight | 931G 846G 86G 91% /mnt/recordings |
| 15:40.23 | Greenlight | Hmm that didn't paste correctly |
| 15:40.46 | leifmadsen | how long is the string you're entering into the DB? |
| 15:41.09 | Greenlight | I guess it would be the member name |
| 15:41.20 | leifmadsen | it's probably a lot longer than just the membername |
| 15:41.24 | Greenlight | The size hasn't changed since I started getting the error |
| 15:41.37 | Greenlight | How could I see what Asterisk is trying to save to the DB? |
| 15:42.19 | Greenlight | (In regards disk space I've >300Gb on / and >3Tb on /var. I've > 11Gb free memory |
| 15:42.22 | Greenlight | ) |
| 15:42.53 | leifmadsen | ya talking in #asterisk-dev about it -- seems like you're hitting a length limit for the data input in a single query/string |
| 15:43.48 | Greenlight | I'll continue conversation over there then |
| 15:44.48 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 15:46.30 | kuku | https://issues.asterisk.org/jira/secure/Dashboard.jspa << is this the office bug database? |
| 15:46.39 | *** join/#asterisk oquidave (~oquidave@41.223.84.35) |
| 15:46.46 | Qwell | office bug? |
| 15:46.56 | kuku | official - sorr |
| 15:46.57 | kuku | y |
| 15:47.02 | kuku | Wow - I can't type. |
| 15:47.08 | Qwell | it is, yes |
| 15:48.49 | oquidave | hello am having a challenging writing a context for my inbound mobile calls...i've configured chan_mobile.so module to work with a usb bluetooth dongle and an android phone and everything seems to be fine. i've also correctly configured the chan_mobile.conf file in /etc/asterisk. My only challenge now is writing a context for inbound mobile calls. Can anyone show me a sample? Many thanks |
| 15:49.39 | WIMPy | Make a call and see what it's looking for. |
| 15:49.53 | WIMPy | Unless it's configurable, "s" seems to be a good guess. |
| 15:50.17 | kuku | When I pass variables thourh asterisk manager - are they global ? |
| 15:50.38 | [TK]D-Fender | oquidave, ... |
| 15:50.39 | [TK]D-Fender | ~book |
| 15:50.39 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:50.42 | [TK]D-Fender | ^^^ |
| 15:50.44 | WIMPy | kuku: Depends on how you do it. |
| 15:50.59 | kuku | fputs($socket, "Variable: foti1=custom/foti_".$ext."_1\r\n" ); |
| 15:51.30 | oquidave | [TK]D-Fender> i've the book. it makes no mention of chan_mobile.so module |
| 15:51.43 | *** join/#asterisk evilman_home (kvirc@2.92.91.10) |
| 15:51.53 | kuku | I'm thinking that issue where the chanspy gets attached to the wrong channel might be because when multiple people use it at the same time, the variables get written over.... |
| 15:51.54 | oquidave | <WIMPy> let me try what you've told me... |
| 15:52.17 | WIMPy | kuku: IIRC you can leave the channel blank to set globals. The Gloabal function should be another way. |
| 15:52.18 | oquidave | am using |
| 15:52.19 | oquidave | [incoming-mobile] exten=>s,1, Dial(SIP/30,30,tT) |
| 15:54.19 | kuku | WIMPy: so the way I set it now, it will not be a global one - correct? |
| 15:55.03 | WIMPy | kuku: I didn't see anything about a channel. |
| 15:55.44 | *** join/#asterisk d_preston215 (~chatzilla@50-73-214-237-philadelpia.hfc.comcastbusiness.net) |
| 15:56.12 | [TK]D-Fender | oquidave, You jsut asked about DIALPLAN. Dialplan has nothing specifically to do with chan_mobile. |
| 15:56.14 | d_preston215 | What is the best option for realtime queue monitoring and reporting? |
| 15:56.15 | kuku | Now I'm confused |
| 15:56.55 | [TK]D-Fender | kuku, Setting a variable over AMI sets it in a channel. Not "everywhere" |
| 15:57.42 | [TK]D-Fender | d_preston215, No such thing as "best". There are tons of web-based reporting tools, AMI-hook-in live status viewers, etc |
| 15:58.00 | kuku | [TK]D-Fender: But which channel ? the "channel" that the AMI session is in ? |
| 15:58.34 | [TK]D-Fender | kuku, AMI has nothing implicitly to do with a call channel |
| 15:58.47 | [TK]D-Fender | kuku, You should really be looking at the intructions for that function call. |
| 15:58.55 | d_preston215 | Ok. Which tools are the most used for it? I just want a start point for finding products to evaluate. |
| 15:59.00 | kuku | [TK]D-Fender: so variables are assigned to each seperate AMI session ? |
| 15:59.15 | Greenlight | No, to channels |
| 15:59.32 | [TK]D-Fender | d_preston215, Very litte mention here for those trying multiple. Most maybe mention on or two and noone is collecting stats to give you |
| 15:59.51 | [TK]D-Fender | kuku, Variables do NOT belong to AMI sessions. they are in CHANNELQS |
| 16:00.05 | kuku | [TK]D-Fender: I have an issue where I run chanspy to a dummy context that plays a message ( that is specific the extension ). The issue is that either chanspy gets attached to the wrong extension, or the wrong extension gets passed to the context |
| 16:00.25 | d_preston215 | I don't want stats. I just want to know what tools people are using to collect that information and/or display it in realtime. |
| 16:00.27 | [TK]D-Fender | kuku, Stop and go read the instructions for the AMI command you are looking to use |
| 16:00.50 | [TK]D-Fender | d_preston215, Yuo asked what is best, and what everyone uses. |
| 16:00.50 | mcolombo | [TK]D-Fender : Can you help me with asterisk's re-invite? |
| 16:00.52 | [TK]D-Fender | THAT is a stat |
| 16:00.59 | [TK]D-Fender | There is no "best", and noone is keeping count |
| 16:01.10 | kuku | [TK]D-Fender: so if I have 3 seperate AMI calls ( 3 different logins ), and int he first one I set a variable to 1, 2nd to 2, and in the 3rd call I set it to 3 - do I now have three variables with or did the last call overwrite the other two ? |
| 16:01.22 | [TK]D-Fender | kuku, Variables do not exist in AMI |
| 16:01.42 | [TK]D-Fender | that is not a place for them. AMI manipulates call channels |
| 16:01.43 | navaismo | d_preston215, Queuemetrics & asternic here |
| 16:02.13 | [TK]D-Fender | Collect & display = stats. |
| 16:02.15 | kuku | [TK]D-Fender: I |
| 16:03.13 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 16:03.30 | kuku | [TK]D-Fender: I'm using Action: Originate. http://pastebin.ca/index.php |
| 16:03.31 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 16:04.16 | navaismo | wrong pb? |
| 16:04.46 | WIMPy | kuku: It's the Asterisk channel, you specify in the Channel: field. |
| 16:04.48 | autofsckk | hello, i have a little problem, i think its codecs missmatch http://pastebin.com/q1jYRyNk i have yealink phones for what i understand they are trying to use g729 and i have ulaw |
| 16:05.22 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 16:05.24 | [TK]D-Fender | autofsckk, You should be looking at the actual SIP DEBUG for your call and seeing what is negotiated. |
| 16:05.34 | kuku | WIMPy: but each time I runt hat function - it creates a new Local/9998@dummy channel - correct? |
| 16:05.46 | [TK]D-Fender | kuku, Your PB link is BAD. We see nothing. |
| 16:05.52 | d_preston215 | Thanks @navaismo |
| 16:05.56 | [TK]D-Fender | kuku, No AMI call data, and no dialplan to back it up |
| 16:05.58 | kuku | http://pastebin.ca/2228888 << sorry |
| 16:06.31 | [TK]D-Fender | kuku, DIALPLAN and CLI OUTPUT <- |
| 16:07.03 | kuku | http://pastebin.ca/2228892 << with dialplan |
| 16:07.03 | WIMPy | kuku: If you set it on originate it will be withing the channel that will be created. |
| 16:07.47 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:07.50 | [TK]D-Fender | kuku, it creates that var in the channel you ORIGINATE |
| 16:08.48 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 16:09.26 | kuku | Ok. Local/9998@dummy-72f4, Local/9998@dummy-de98m, Local/9998@dummy-72f4 < I can see it create its own channels for 9998 |
| 16:11.17 | *** join/#asterisk jsjc (~Adium@153.Red-83-52-137.dynamicIP.rima-tde.net) |
| 16:11.36 | kuku | Each extension has a pre-recorded message ( an agents own voice ). 60% of the time, when this "originate" message gest sent to AMI, the agent will not hear the message. So after further review, the message DOES get played - but on someone elses call. I already added "-" after the extension, so it doesnt confuse SIP/15 with SIP/115, but that didn't fix it. |
| 16:12.08 | kuku | I've been using asterisk since 2006 - and for the life of I can't figure out what is wrong here. |
| 16:12.45 | [TK]D-Fender | kuku, fputs($socket, "Data: SIP/".$ext."-,qB\r\n" ); <--- how do WE know what you're targeting? |
| 16:12.58 | kuku | I've troubleshot this thing for a few days and can't figure out whats wrong. when I look at the cli output - "app_chanspy.c: Attaching Local/9998@dummy-72f4;1 to SIP/43-0000e4d5" the call gets properly attached. |
| 16:13.11 | [TK]D-Fender | kuku, where do I trust that is a spcific channel and not just a device name at BEST once filled in? |
| 16:13.50 | kuku | I don't specify teh channel - I aspecify the extension. Example: SIP/15- |
| 16:14.15 | [TK]D-Fender | No such thing as "extension" there. Never use that term again. And you need to target the SPECIFIC CALL CHANNEL |
| 16:14.24 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 16:14.32 | [TK]D-Fender | no just a "device" or ANY channel matching a fractional name could get connected as is happening. |
| 16:14.38 | [TK]D-Fender | You are not being specificin your target. |
| 16:14.55 | kuku | ok. I would love to target the specific channel - as it will give me accurate "attachment" of chanspy. |
| 16:15.34 | kuku | The question is - how do I get the channel - if at the moment, I only know the extension? I can query for it - but I Can't find an AMI function that will return that data. |
| 16:15.48 | kuku | s/extension/device |
| 16:16.07 | [TK]D-Fender | You don't seem to be telling up what the complete call flow you're going through is. We don't know what or how you are deciding anything at all. |
| 16:16.21 | [TK]D-Fender | You are asking vage things about finding specific channels with NO criteria |
| 16:16.32 | [TK]D-Fender | vague* |
| 16:17.02 | *** join/#asterisk n3hxs (~n3hxs-5@TOROON63-3096776761.sdsl.bell.ca) |
| 16:17.05 | kuku | The agent makes a call - then they have an app running ontheir desktop with a button they presss. The app knows their extension. If they want to play their recording, it triggers this script which makes an AMI call. I pass the extension to that script. |
| 16:17.08 | [TK]D-Fender | We also don't know what is launching the AMI request. Based on what other information? |
| 16:17.29 | *** join/#asterisk YoMomma (~YoMomma@142.129.178.68) |
| 16:17.48 | [TK]D-Fender | kuku, Show more complete configs and debug. |
| 16:18.08 | kuku | with 60 agents on the call - its hard for me to get that for you now ( the cli ) |
| 16:18.28 | kuku | I can get you the config - 1 sec |
| 16:22.19 | kuku | [TK]D-Fender: here you go: http://pastebin.ca/2228894 |
| 16:22.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:23.53 | kuku | ?bdl=12345&phone=8475551212&exten=81&customer=5557&first_name=Michael&last_name=Smith << this is what triggers that AMI script |
| 16:29.50 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 16:31.40 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:37.22 | kuku | [TK]D-Fender: Any ideas what it could be ? |
| 16:38.33 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:39.51 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 16:40.52 | [TK]D-Fender | kuku, We see no actual calls. I can't proves what's hooking on to what or if what's being executed matech the little bits you have shown so far. |
| 16:41.22 | [TK]D-Fender | kuku, This lack of detail is not workable. There is no cshortchanging the process. Yuo want it fixed to you need real debug. All of it. Complete channel dump listings, etc |
| 16:41.59 | *** join/#asterisk evilman_home (kvirc@128-73-24-96.broadband.corbina.ru) |
| 16:42.18 | [TK]D-Fender | If it's hooking onto a "wrong" channel then you aren't being specific enough about what yuo are targeting and there are MULTIPLE targets that are close enough |
| 16:42.37 | [TK]D-Fender | Go really look at what's happening. |
| 16:43.27 | kuku | [TK]D-Fender: OK. Then how can I identify all avaialable channels for a given device ? |
| 16:43.30 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:43.37 | kuku | so I can pass the specific channel |
| 16:46.46 | [TK]D-Fender | kuku, How do YOU think you should know which one is "right" if there are multiple? Youa re asking US to be psychic for you. |
| 16:46.53 | [TK]D-Fender | kuku, You are not thinking this process through. |
| 16:47.44 | [TK]D-Fender | kuku, Go look at the circumstances of the failure and ask yourself why this could happen and based on the states of things, what that can tell you about how you should deal with it |
| 16:47.51 | kuku | [TK]D-Fender: the agent has only one channel at the same time. If they have two, then I will grab the first channel name -as they should be only on one call at a time. |
| 16:48.11 | [TK]D-Fender | "should" is Weak Sauce (tm). |
| 16:48.13 | kuku | The issue is that it gets attached to a totally different device, if I just pass the device. |
| 16:48.27 | [TK]D-Fender | Go look at the actual failures and be sure of the circumstances. |
| 16:50.18 | kuku | 235k of cli output |
| 16:52.23 | kuku | What bothers me is that if I test after hours - it works 100% |
| 16:55.08 | navaismo | kuku, because you have only one channel, try to find your complete channel to spy on |
| 16:56.12 | kuku | nobody can answer how I can list all channels for a given device |
| 16:56.28 | kuku | that would solve one of the potential questions and maybe even solve the issue. |
| 16:56.34 | navaismo | i.e, if you want to spy the call on peer 5001, try asterisk -rx 'core show channels verbose' | grep 5001, that show you channels with 5001 string in there |
| 16:56.54 | navaismo | want to be more specific: asterisk -rx 'core show channels verbose' | grep SIP/5001 |
| 16:57.17 | navaismo | and the response is something like: |
| 16:57.19 | navaismo | SIP/5001-00000002 from-internal 2525 2 Up MusicOnHold default 5001 00:03:16 |
| 16:57.24 | kuku | Beutiful ! |
| 16:57.33 | kuku | Thats all I needed - thank you. |
| 16:57.36 | navaismo | where SIP/5001-00000002 is the channel |
| 16:59.15 | kuku | I'll add a "-" at the end for the grep, to get SIP/15- ( b/c I have sip/115 as well ) |
| 17:00.04 | kuku | navaismo: can I get that information somehow through AMI ? |
| 17:00.20 | kuku | Since the script that I'm launching this from is a web server, which connecto to asterisk through ami. |
| 17:00.40 | navaismo | i guess the core show channel command exist in AMI too |
| 17:01.22 | navaismo | if not you can try: $channel=(exec "asterisk -rx 'core show channels verbose' | grep 5001"); |
| 17:06.31 | [TK]D-Fender | No need to shell out for it |
| 17:07.49 | kuku | ? |
| 17:09.02 | kuku | how else? |
| 17:12.16 | *** join/#asterisk Tim_Toady (~fuzzy@130.43.35.150.dsl.dyn.forthnet.gr) |
| 17:12.16 | [TK]D-Fender | kuku, You should really, really, look at the AMI command reference..... |
| 17:12.56 | kuku | http://www.voip-info.org/wiki/view/Asterisk+manager+API coreshowchannels isn't documented here. |
| 17:13.22 | [TK]D-Fender | asterisk.org, not some decript WIKi |
| 17:13.42 | navaismo | manager show command CoreShowChannels |
| 17:13.46 | [TK]D-Fender | And you are expecting a specific name. I said read what's there, not loko for one specific thing you expect it to be called. |
| 17:23.11 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
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| 17:37.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:39.17 | kuku | <PROTECTED> |
| 17:41.00 | navaismo | O_o |
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| 18:19.08 | BorjaGVO | Anyone can give me recommendation on High Vailability for Asterisk? |
| 18:19.12 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 18:19.22 | kuku | no asterisk :) |
| 18:19.45 | BorjaGVO | ?... |
| 18:20.34 | navaismo | look into the digium products and the hearbeat daemon too |
| 18:20.34 | BorjaGVO | well, it's related to asterisk....right? |
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| 18:24.47 | _Corey_ | BorjaGVO: You've asked a fairly open-ended question. Explain exactly what you're trying to achieve and you'll get better help. |
| 18:28.12 | BorjaGVO | As Corey suggested, my question was little open-ended. What I want is to set up a High Availability VoIP platfor based on Asterisk. I have Asterisk already running but now I want to know which extra elements do I need in order to build up the high availability enviroment. I heard about Linux HA and OpenSER but I'm running out of time and it would be excellent if some one that already implemented something alike can give me some gu |
| 18:28.14 | BorjaGVO | s |
| 18:30.07 | _Corey_ | So, what I'm suggesting you describe is what you want the "high availability" behavior to be... i.e. a hot standby server, etc |
| 18:30.49 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-43-209.user.veloxzone.com.br) |
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| 18:41.32 | pzn | how can I match a number in "extensions.conf" with a prefix, but an unknown number of digits? I mean, I need to match _123X or _123XX or _123XXX, or any number of X |
| 18:42.45 | anonymouz666 | _123. |
| 18:42.46 | anonymouz666 | ? |
| 18:43.39 | pzn | anonymouz666, thanks! |
| 18:44.06 | anonymouz666 | pzn: you don't want to match 123, right? |
| 18:44.19 | pzn | anonymouz666, right |
| 18:45.17 | anonymouz666 | wait. do you want exactly 123XXX? |
| 18:45.58 | pzn | anonymouz666, it worked allright! I need it to match 123x and 123xx and 123xxx and 123xxxx and any number of "x"... _123. worked ok |
| 18:46.09 | anonymouz666 | ok |
| 18:46.40 | *** join/#asterisk kikohnl (~keith@udp278187uds.hawaiiantel.net) |
| 18:47.20 | BorjaGVO | hot standby server, exactly |
| 18:47.56 | BorjaGVO | by high availability I mean not letting down the service...that if a server cannot take traffic, it gets holded by the other one |
| 19:00.19 | *** join/#asterisk aross42 (~aross@216.16.232.86) |
| 19:01.09 | *** join/#asterisk corretico (~luis@190.211.93.11) |
| 19:02.09 | navaismo | with what softphone can I send a sip uri without been registered? |
| 19:02.32 | Qwell | any? |
| 19:04.08 | navaismo | hmm, I have tried with zoiper, putting in the dial field my domain sip:mymail@mypbx.com but nothing happens |
| 19:06.16 | anonymouz666 | navaismo: that depends on the implementation not the SIP protocol. |
| 19:07.00 | anonymouz666 | most of configs have an option to enable dial without register. |
| 19:09.34 | navaismo | ok let me see that, I'm trying to get working the "call my mail", but if my phone is registered to some server of course the sip:mymail@mydomain never will work |
| 19:11.06 | brian98 | Hello - can you help me get this correct? |
| 19:11.07 | brian98 | ;exten => _X.,n,ExecIf($[${SIP_HEADER(Privacy):0} = id ]?Set(CALLERID(num))=UNKNOWN ) |
| 19:11.49 | brian98 | I have it working as two lines of code. ie setting the value from the header in a variable then checking it but can't get the syntax correct. |
| 19:14.01 | leifmadsen | what is wrong? |
| 19:14.06 | leifmadsen | what is the error? |
| 19:14.17 | jpsharp | navaismo: If you want someone to be able to reach you by dialing foo@foo.com, you either need to have your DNS set so foo.com resolves to the IP address of your asterisk box or have an SRV record pointing to the appropriate DNS entry. |
| 19:15.08 | brian98 | [Sep 28 20:14:59] WARNING[31095]: pbx.c:10020 pbx_builtin_setvar: Set requires an '=' to be a valid assignment. |
| 19:15.38 | Qwell | brian98: Your syntax is wrong. |
| 19:15.51 | Qwell | ExecIf(...?Set(CALLERID(num)=...)) |
| 19:15.54 | Qwell | move your ) |
| 19:15.58 | brian98 | ahhh |
| 19:16.00 | brian98 | feck |
| 19:16.01 | brian98 | :) |
| 19:16.02 | brian98 | thanks |
| 19:16.03 | leifmadsen | oh he has 2 )) |
| 19:16.05 | Qwell | That'll be $299.99. :) |
| 19:16.10 | leifmadsen | didn't catch it because it was on 2 lines |
| 19:16.14 | brian98 | send me your paypal details please |
| 19:16.14 | Qwell | leifmadsen: and I've had lunch beers. |
| 19:16.20 | Qwell | brian98: t'was a joke. |
| 19:16.24 | brian98 | Thanks guys! |
| 19:16.26 | brian98 | I know |
| 19:16.30 | navaismo | jpsharp, yes that part is resolved, now the issue is how to dial without a server |
| 19:16.34 | leifmadsen | my window was exactly at the width so the 2nd ) showed on another line |
| 19:16.51 | leifmadsen | Qwell: I suspect the problem might have been lack of lunch beers today |
| 19:16.56 | Qwell | leifmadsen: definitely |
| 19:16.58 | Qwell | or |
| 19:16.59 | Qwell | defiantly |
| 19:17.02 | leifmadsen | :D |
| 19:18.42 | jpsharp | navaismo: Oh, gotcha. Yes, I've been bitten by that same problem. Some SIP implmentations refuse to dialout unless registered, some require at least an outbound proxy to be set. |
| 19:20.08 | navaismo | yep |
| 19:20.51 | TechSmurf | how incredibly proactive |
| 19:21.05 | brian98 | qwell: I'm still not getting this to work. Is this correct? exten => _X.,n,ExecIf($[${SIP_HEADER(Privacy):0}=id]?Set(CALLERID(num))=UNKNOWN ) |
| 19:21.08 | TechSmurf | Digium just called to make sure I got dpma working |
| 19:21.22 | Qwell | TechSmurf: and? |
| 19:21.29 | Qwell | brdude: You didn't move your )... |
| 19:21.39 | TechSmurf | it's working fine |
| 19:21.45 | TechSmurf | I never contacted them for support. |
| 19:21.46 | Qwell | TechSmurf: You're welcome. |
| 19:21.49 | brian98 | flip |
| 19:21.49 | Qwell | really? O.o |
| 19:22.07 | TechSmurf | no, it turned out to be a firmware version issue with *now |
| 19:22.31 | TechSmurf | tossed latest firmwares in and the phone's been happy ever since |
| 19:22.39 | brian98 | ok |
| 19:22.46 | brian98 | exten => _X.,n,ExecIf($[${SIP_HEADER(Privacy):0}=id]?Set(CALLERID(num)=UNKNOWN )) |
| 19:22.51 | brian98 | works perfectly now - thanks |
| 19:22.52 | Qwell | brian98: that's better |
| 19:22.59 | brian98 | I'm braindead today |
| 19:23.02 | brian98 | not enough beers |
| 19:23.24 | TechSmurf | so, let me rephares... Digium just cold called me to make sure I got dpma working based solely on the fact that I downloaded it... |
| 19:23.37 | Qwell | TechSmurf: huh |
| 19:23.49 | TechSmurf | how incredibly proactive... |
| 19:24.10 | brian98 | one issue I am trying to resolve if the privacy id tag isn't there I get : ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 19:24.11 | brian98 | =id |
| 19:24.15 | pabelanger | money is a good motivator |
| 19:24.19 | brian98 | how can I get around that? |
| 19:24.48 | TechSmurf | Qwell: Thanks tho! |
| 19:24.49 | Qwell | brian98: Your syntax looks okay. You might do the old bash trick of using x's before a var or quoting both sides. It could be empty. |
| 19:25.14 | brian98 | Yes - the issue is if it's empty |
| 19:25.36 | Qwell | brian98: quote both sides then |
| 19:25.49 | brian98 | ok |
| 19:25.49 | Qwell | $["${SIP_HEADER(Privacy):0}"="id"] |
| 19:26.13 | brian98 | Thanks man |
| 19:27.01 | Qwell | You keep wracking up those $299.99 charges. ;) |
| 19:27.28 | brian98 | :) |
| 19:28.11 | jpsharp | If he doesn't pay, I'll be your collection's agent. |
| 19:29.36 | brian98 | send me your addresses |
| 19:29.40 | brian98 | there's a cheque in the post. |
| 19:31.20 | brian98 | though the postal service out of ireland is notoriously bad |
| 19:31.39 | jgowdy | "there's a cheque in the post" that statement makes me feel very American |
| 19:33.20 | *** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk) |
| 19:35.49 | navaismo | someone has blink running on Linux, Linphone cant send uris, jitsi sucks other options? |
| 19:37.58 | brian98 | You hear it a lot? |
| 19:40.57 | jgowdy | no, just that we'd say "the check's in the mail" :) |
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| 20:07.23 | *** join/#asterisk qmr (~qmr@gamma.qtech.us) |
| 20:07.31 | qmr | Any free SIP clients for Mac that are good? |
| 20:10.19 | navaismo | people said blink |
| 20:10.41 | qmr | k. would you mind terribly grabbing that and building it for me? I'll wait here.. |
| 20:10.59 | navaismo | for non crap-bian based system in linux sucks |
| 20:11.21 | qmr | I'm not sure I follow that last line |
| 20:13.00 | navaismo | blink seems to work out of the box only for debian/ubuntu |
| 20:13.24 | qmr | o |
| 20:13.43 | qmr | I found a build of it, but it crashes when I try to open the preferences page. So, that's great. |
| 20:17.35 | *** join/#asterisk ska (~ska@unaffiliated/ska) |
| 20:17.59 | navaismo | great the my conutry congress its trying to approve the hourly rate $7pesos ==> 0.54USD; I dont want to live here anymore |
| 20:19.04 | ska | Is there a simple test or set of tests that can determine if a VPS is a good candidate for a * server? mtr, ping? assume memory and cpu are sufficient. |
| 20:19.24 | Qwell | ska: If it has the characters "VPS", sequentially, in the name, it is a bad choice. :) |
| 20:20.47 | ska | That bad eh.. |
| 20:21.04 | Qwell | They all use custom kernels, broken packages of things, etc. |
| 20:21.16 | Qwell | It's not supportable or maintainable. |
| 20:22.08 | pabelanger | Install KVM yourself and profit |
| 20:22.56 | ska | Any particular packages that typcially lack or are broken so I can check? |
| 20:23.28 | Qwell | ska: all of them :p |
| 20:28.26 | jpsharp | Asterisk on VMWare isn't bad. |
| 20:29.12 | ska | jpsharp: I'll probably run on Xen or KVM,... at least for testing |
| 20:29.34 | ska | I don't think i need any zaptel drivers, this will be all IP based. |
| 20:30.52 | jgowdy | Personally I'd check that their kernel was getting a paravirtualized timer in dmesg, and that the kernel was new enough for timerfd |
| 20:31.01 | jgowdy | Not that I think virtualized Asterisk is a good idea |
| 20:31.26 | jpsharp | Its fine for small instances, but you certainly don't want to run a call center on it. |
| 20:33.21 | ska | <PROTECTED> |
| 20:34.15 | ska | jgowdy: kern: 2.6.32-5-686-bigmem #1 SMP 4-core |
| 20:34.47 | ska | I just need to be able to run 2 calls from it. |
| 20:34.50 | jgowdy | So I'd *guess* and *hope* that based on those two pieces of information, you can get good enough timing from timerfd |
| 20:35.14 | jgowdy | you're probably fine |
| 20:35.16 | ska | jgowdy: the only thing i'm not sure of is the module problem. |
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| 20:50.51 | brian98 | navaismo: have you had a bad experience with debian or are you just a yum sadist ? |
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| 20:59.56 | navaismo | brian98, naaah I only hate that pussy command called sudo, and hate those ubuntu lovers |
| 21:00.19 | brian98 | aha! |
| 21:00.20 | brian98 | lol |
| 21:00.46 | [TK]D-Fender | Yeah, real men run everything as root! |
| 21:00.53 | brian98 | haha |
| 21:01.06 | brian98 | while blindfolded drinking tequila |
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| 21:01.21 | navaismo | XD |
| 21:03.11 | navaismo | jokes apart i fell more comfortable with redhat based distros, I grown with it since fedora core. So don't start a flavor distro war |
| 21:03.31 | navaismo | s/fell/feel/ |
| 21:04.01 | brian98 | yum yum |
| 21:04.21 | brian98 | sorry just eating some pancakes. |
| 21:04.29 | navaismo | but in my past work used a lot debian so aptitude or apt-get its same for me |
| 21:06.03 | brian98 | Its quite funny that people are so polar when it comes to linux dists. |
| 21:06.13 | brian98 | My favourite is debian |
| 21:06.28 | brian98 | But I can use centos without a problem. |
| 21:06.55 | brian98 | My first distribution to get my hands dirty with was redhat |
| 21:09.10 | brian98 | Redhat 5 |
| 21:13.11 | Qwell | shudders |
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| 21:17.43 | jpsharp | What's the easiest way to let someone hack your RedHat box? Turn it on. |
| 21:18.34 | navaismo | fanboys...fanboys everywhere |
| 21:19.06 | jpsharp | I only use Linux because it is a cast iron pain in the ass to get it to build on NetBSD. |
| 21:19.18 | jpsharp | to get Asterisk to build, rather. |
| 21:20.00 | *** join/#asterisk twanny796 (~twanny@85.232.219.59) |
| 21:20.11 | kikohnl | try it on Solaris, fails to build all the time |
| 21:20.47 | jpsharp | I haven't used Solaris since I gave away my Ultra 30 and my Sparcserver 1000. |
| 21:21.22 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 21:22.14 | kikohnl | I'm running Solaris 10 on an HP desktop, sendmail & apache, running asterisk on a mac mini running ubuntu |
| 21:22.52 | kikohnl | asterisk fails to build due to ncurses being built with wide characters :-( |
| 21:24.15 | *** part/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 21:30.45 | anonymouz666 | OK, guys. 6:30pm. I declare open the weekend throughout the national territory. |
| 21:31.15 | navaismo | 16:31 here |
| 21:31.23 | kikohnl | 11:31 here |
| 21:31.28 | jgowdy | 14:31 |
| 21:31.39 | anonymouz666 | you live in the past |
| 21:31.42 | anonymouz666 | :P |
| 21:31.46 | jgowdy | California > * |
| 21:32.09 | kikohnl | Honolulu |
| 21:32.33 | jpsharp | It is always the weekend when you're retired :) |
| 21:33.05 | navaismo | XD |
| 21:33.26 | navaismo | honolulu sweet!!! |
| 21:34.09 | kikohnl | the company I worked for went out of business in january, and I'm still looking for work :-( |
| 21:35.25 | navaismo | are you looking in freelancer, odesk or elance?? |
| 21:36.09 | kikohnl | never saw those |
| 21:36.26 | kikohnl | thanks for the info !! |
| 21:37.02 | navaismo | take a look maybe you have chance to get some job there, im without job too and so far survived |
| 21:37.36 | jpsharp | Good luck with those. I gave up on them after the 10th time I got underbid by 75% by someone in India or Ukraine. |
| 21:37.37 | kikohnl | Off to Kauai for vacation tomorrow morning, got to get out of the city for a while |
| 21:39.16 | navaismo | yes i have losed bids for 30USD with asian guys for complete installations and configurations |
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| 21:39.31 | navaismo | but when you dont have any more money 30USD are better than 0 |
| 21:39.42 | kikohnl | wow, that's low |
| 21:40.18 | navaismo | very low, but in average people pay between 100 and 400 USD the installation |
| 21:41.48 | kikohnl | last asterisk install job I did, i never to paid for it, they say the bank won't cover it under the bankrupcy proceedings |
| 21:42.07 | jpsharp | the biggest problem I had, though, was getting lowballed and then the client coming back and saying "Hey, these chuckleheads screwed everything up. Can you fix it for your original bid amount?" |
| 21:42.52 | jpsharp | kikohnl: Cash up front :) |
| 21:43.46 | kikohnl | from now on yes! |
| 21:44.04 | jpsharp | Until I get a good relationship with someone, it is %50 upfront, 25% upon delivery, 25% upon final acceptance. |
| 21:44.30 | jpsharp | And if there's hardware involved, 100% of hardware costs upfront. |
| 21:45.09 | navaismo | yes those site have a system called milestone, first you receive one paid for start the job, then when finish the second |
| 21:45.32 | kikohnl | well, this was the previous company, so I thought it would be ok. |
| 21:47.47 | jpsharp | I'ma cynical and bitter old bastard now, but it has kept me from getting shafted a few times lately. |
| 21:48.10 | jpsharp | Well, not so old. |
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| 22:02.32 | newtonr | Anyone using 1.8.X - have you attempted playing G722 sound files , and did you run into any issues with it? |
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| 23:53.36 | qmr | So.. I made a couple users in the .conf file, connected w/o registering, and successfully made a video call. Now it's not working :( any idea what to check? |
| 23:53.54 | qmr | [Sep 28 23:52:49] WARNING[3466] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) shows up in the logs |
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| 23:59.25 | anthony[1] | I need help because suddenly Asterisk 1.8.15.1produces error messages when trying to call two SIP clients on my PC: WARNING[3378]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |