| 00:03.22 | *** part/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
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| 00:59.12 | jpsharp | since every call is handled as at least 1 separate thread, Asterisk will use the heck out of a multicore/multiprocessor system. |
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| 02:40.57 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 05:05.35 | ChannelZ | It's oh so quiet |
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| 06:06.49 | mirela666 | ~help |
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| 06:13.55 | gavimobile | im looking for a list of all the sounds file names from /var/lib/asterisk/sounds/en/ its not comfortable to view the list from the terminal. anyone know where I can see a list of sounds? |
| 06:17.26 | deo | what machine are you using gavimobile |
| 06:17.33 | deo | if windows you can use winscp |
| 06:17.47 | gavimobile | deo: pc with asterisk 1.8 |
| 06:18.14 | gavimobile | deo: I was looking for a website or documentation with a list |
| 06:18.15 | deo | you can view it into another pc / |
| 06:18.18 | gavimobile | can't find anything on google |
| 06:18.20 | deo | you can try winscp |
| 06:18.30 | gavimobile | im using putty |
| 06:19.28 | ChannelZ | ls -l > look-at-all-the-sounds.txt |
| 06:19.39 | ChannelZ | pico look-at-all-the-sounds.txt |
| 06:19.39 | gavimobile | ChannelZ: thanks |
| 06:19.39 | deo | you can always view it using putty but its not comfortable |
| 06:20.01 | gavimobile | than ill send it to /var/www/html/sounds.html |
| 06:20.04 | gavimobile | :-D |
| 06:20.12 | deo | :D |
| 06:22.33 | gavimobile | deo: I know I can view it in putty |
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| 07:35.55 | kaldemar | gavimobile: you'll find a list of sounds and their contents as text in sounds/core-sounds-en.txt in source packages. |
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| 07:36.58 | polysics | hi |
| 07:37.38 | polysics | can multiple Variable values in an Originate be specified by separating them with a comma in 1.8? |
| 07:38.25 | polysics | or do I need multiple Variable: lines? |
| 07:38.59 | polysics | ie. Variable: var1=1,var2=2 or Variable: var1=1 \n Variable: var2=2 |
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| 07:41.57 | kaldemar | polysics: what does "manager show command Originate" tell you about the Variable header? |
| 07:42.04 | gavimobile | kaldemar: thanks! I already made myself a nice html but ill know for next time |
| 07:47.05 | polysics | Channel variable to set, multiple Variable: headers are allowed. |
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| 07:47.39 | polysics | my issue is that we are generating the originate from code, and the keys are in a Hash |
| 07:47.49 | polysics | thus, I can currently only have one Variable |
| 07:48.00 | kaldemar | time to change to code then. |
| 07:48.08 | polysics | I will patch the code but needed a qcuik way to test the SIP header the provider gave us :-) |
| 07:48.18 | polysics | and yes, if you are interested, commas work |
| 07:48.26 | polysics | it's undocumented so code has to change |
| 07:48.40 | kaldemar | they do? interesting. |
| 07:54.21 | kaldemar | seems the commas/pipes always have worked. |
| 07:54.43 | Digweed | I've got a problem with Asterisk 1.4 working with SRV records. The machine is able to resolve the SRV domain, Asterisk shows one of the correct A records but afterwards tries to use the SRV domain again to register. Instead of the A record. Anyone a clue? |
| 07:54.50 | Digweed | srvlookup is set to yes of course |
| 07:55.02 | Digweed | The Linux is able to do the SRV lookup as well |
| 07:55.06 | Digweed | *box |
| 07:57.00 | Digweed | This is basically what happens |
| 07:57.01 | Digweed | Sep 4 13:04:24] NOTICE[9623]: chan_sip.c:7521 sip_reg_timeout: -- Registration for 'username@srv.record.domain.net' timed out, trying again (Attempt #1) |
| 07:57.02 | Digweed | -- ast_get_srv: SRV lookup for '_sip._udp.srv.record.domain.net' mapped to host sipproxy110.domain.net, port 5060 |
| 07:57.02 | Digweed | [Sep 4 13:04:24] WARNING[9623]: chan_sip.c:2921 create_addr: No such host: srv.record.domain.net |
| 07:57.02 | Digweed | [Sep 4 13:04:24] WARNING[9623]: chan_sip.c:7599 transmit_register: Probably a DNS error for registration to username@srv.record.domain.net, trying REGISTER again (after 20 seconds) |
| 07:58.13 | Digweed | The strange thing is, even without those SRV records, the box is not able to register to a normal sip proxy A record. It only works when using the IP behind it. DNS resolving works fine though. |
| 07:58.26 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
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| 08:01.52 | wdoekes | Digweed: looks like the resolving works, but then the result is ignored, using the original host as A-record instead. but you're using 1.4... I'm not sure you can trust srv record resolving there |
| 08:02.11 | wdoekes | but using sipproxy110.domain.net as host should work in 1.4 |
| 08:02.20 | Digweed | Normally I would have updated Asterisk right away hah, but this is a production machine I cannot upgrade at the moment :( |
| 08:02.36 | Digweed | No, using the A record does not work, it only works when using the IP behind it. |
| 08:03.04 | wdoekes | does it give the same error (without the srv part)? |
| 08:03.08 | Digweed | Correct. |
| 08:04.15 | wdoekes | and dnsmgr is enabled or disabled? |
| 08:05.08 | wdoekes | (and did you srvlookup=no in the A-record case?) |
| 08:05.42 | gavimobile | if asterisk 1.8 is working for me, would it be silly to upgrade to asterisk 11? |
| 08:05.51 | Digweed | dnsmgr, where do i set that? |
| 08:07.19 | Digweed | ah wait, the config file |
| 08:07.22 | Digweed | let's check |
| 08:07.37 | Digweed | it's set to yes |
| 08:07.54 | Digweed | so enabled |
| 08:09.32 | ChannelZ | gavimobile: yes |
| 08:09.39 | Digweed | and indeed, when setting srvlookup to no, it still cannot use the a record... |
| 08:09.46 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 08:10.13 | Digweed | wdoekes ^ |
| 08:10.44 | wdoekes | you can try disabling the dnsmgr (at your own risk) |
| 08:11.05 | Digweed | what would the risk be? i am remotely logged in via ssh, i wont use that right |
| 08:11.25 | wdoekes | you won't break ssh nor asterisk |
| 08:11.41 | wdoekes | but it could be that record lookups take more time and cause stallage |
| 08:12.20 | wdoekes | (better fix than dnsmgr is to run a copy of bind on localhost) |
| 08:13.26 | Digweed | I set dnsmgr to no, same problem after a reload. |
| 08:13.35 | Digweed | run a copy of bind on localhost |
| 08:13.43 | wdoekes | won't help with your problem |
| 08:13.50 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 08:13.57 | wdoekes | only help if the lack of dnsmgr causes lookup timeout issues |
| 08:14.00 | Digweed | hm ok, i really wonder why asterisk is messing with me ;) |
| 08:14.01 | Digweed | ok |
| 08:14.12 | Digweed | Asterisk 1.4.21.1 |
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| 09:16.49 | gavimobile | google voice is ALIVE |
| 09:16.50 | gavimobile | ! |
| 09:18.51 | k610 | was it down ? |
| 09:21.25 | ectospasm | I used it earlier yesterday. Of course, I was only setting my voicemail greeting |
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| 09:38.22 | Matthias | hi @all |
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| 09:49.33 | nextime | hello |
| 09:49.54 | nextime | is there a way to tell CURL() function to follow redirections? |
| 10:00.26 | Greenlight | I've got my CDR settings to batch, and use the defaults 100 records and 300 seconds. However looking at the database it's not inserting records 100 at a time - could there be any reason for this? |
| 10:01.18 | ectospasm | Greenlight: are you actually generating 100 records in five minutes? |
| 10:01.43 | Greenlight | Waaaaay more |
| 10:01.59 | Greenlight | 100 records every 15 seconds |
| 10:02.36 | Greenlight | My concern is that the "missing" records are not being written, but if it's something else im happy |
| 10:02.48 | Greenlight | Like im seeing 40-70 written at once, in bulk |
| 10:03.06 | ectospasm | yeah, batch processing of CDRs is wonky, I don't know how to answer that question. |
| 10:05.52 | Greenlight | Wonky as in how often it writes out, or wonky as in it looses CDR's ? |
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| 10:07.27 | ectospasm | Greenlight: yes |
| 10:07.38 | Greenlight | Both ? |
| 10:07.52 | ectospasm | both, afaict |
| 10:08.14 | ectospasm | ...it's especially bad if Asterisk should crash before the CDR is written to the backing store |
| 10:08.17 | Greenlight | Damn that sucks, would be nice if the documentation mentioned that it didn't bloody work. I use these for billing |
| 10:08.28 | Greenlight | Yea, I realised that, and wasn't too worried |
| 10:08.49 | Greenlight | So I loose 100 cdr's - not a big deal, we're doing 300k+ a day |
| 10:09.06 | Greenlight | But I loose 20%-40% that is a biiig deal |
| 10:09.09 | ectospasm | and to say batch CDR processing doesn't work may not be entirely accurate, I think a great deal depends on the backing store |
| 10:09.32 | Greenlight | Hmm... I'm writing to both local mysql and remote mssql |
| 10:09.50 | Greenlight | I've not actually checked the local mysql stored yet, only the mssql as that's where we do the billing from |
| 10:10.30 | ectospasm | I'd check and make sure they match. If you're lucky maybe it's only the MSSQL that isn't matching now. |
| 10:10.59 | Greenlight | This is particularly concerning... |
| 10:12.14 | Greenlight | The numbers match - 31k so far today on both |
| 10:12.59 | ectospasm | I dunno, I don't really deal with CDRs much |
| 10:13.02 | Greenlight | Does anyone else use cdr_adaptive_odbc to write to mysql in batch mode ? |
| 10:18.24 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:20.22 | Greenlight | Phew. Just downloaded CDR's from my wholesale provider. The numbers match! |
| 10:20.45 | Greenlight | So guess it's flushing out the batch more frequently than every 100 records. Panic over. |
| 10:21.00 | ectospasm | ...yeah, I forgot about that being the possibility |
| 10:21.50 | Greenlight | Yea - I dont care if that's the case. It's the possibity of £1000/day of missing CDR's that concerned me :) |
| 10:30.06 | ectospasm | understandable (-; |
| 10:30.29 | Greenlight | heh, yea. Anyways - thanks for your help, much appriciated. |
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| 11:15.48 | T-1000 | Hi there :) |
| 11:16.12 | T-1000 | Anyone knows when http://packages.asterisk.org/ will be back up? it seems down and mirrors are not responding too |
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| 11:47.40 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 11:47.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.1 (2012/08/30), 1.8.15.1 (2012/08/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 11:56.19 | n3hxs | \o/ yawn |
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| 12:16.25 | carrar | *Y*A*W*N* |
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| 12:44.05 | WoLFjuh | can someone confirm packages.asterisk.org is down? |
| 12:44.14 | wdoekes | it is, so is reviewboard |
| 12:44.43 | wdoekes | and more.. 14:19 < kpfleming> maybe digium is having network troubles |
| 12:44.56 | WoLFjuh | sysadmins with monitoring checks on yum or apt are having a bad time :) |
| 12:45.46 | carrar | http://www.downforeveryoneorjustme.com/packages.asterisk.org |
| 12:46.16 | *** join/#asterisk DelphiWorld (~VoCloud@openvpn/user/DelphiWorld) |
| 12:46.21 | DelphiWorld | hi asteriskers |
| 12:46.26 | DelphiWorld | is svn server down ? |
| 12:46.36 | WIMPy | No. Everything. |
| 12:47.03 | carrar | Asterisk has gone Chapter 11 and shut down |
| 12:47.04 | DelphiWorld | WIMPy: i'm trying to checkout but it's not connecting |
| 12:47.11 | carrar | err Digium |
| 12:47.12 | carrar | heh |
| 12:47.41 | DelphiWorld | svn: OPTIONS of 'http://svn.digium.com/svn/asterisk/trunk': could not connect to server (http://svn.digium.com) |
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| 12:51.38 | jkroon | hi guys, when using g729 in a pass-through fashion - is there any way to set it up to negotiate g729 or g279a, but NOT g729b? Specifically I'd like to try and get rid of silence suppression/vad ... |
| 12:52.19 | bulkorok | jkroon: I change the sdp-header... |
| 12:52.43 | bulkorok | but it's not asterisk I use... I don't know exactly how to manage this in asterisk... |
| 12:54.42 | *** join/#asterisk Sicelo__ (~sicelo@unaffiliated/sicelo) |
| 12:54.50 | jkroon | bulkorok, what header can you change there? |
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| 12:55.34 | jkroon | granted, I haven't messed with the SDP much, but IIRC it basically says something about g729 supporting annexb. |
| 12:55.48 | *** join/#asterisk chuckf_ (~chuckf@ubuntu/member/chuckf) |
| 12:55.48 | bulkorok | right... let me check |
| 12:56.02 | jkroon | now if ast has some way to manipulate the SDP on receive/transmit that's perfect, but I don't recall such a mechanism. |
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| 12:56.47 | DelphiWorld | back |
| 12:56.53 | DelphiWorld | svn: Server sent unexpected return value (504 Gateway Time-out) in response to OPTIONS request for ' http://svn.digium.com/svn/asterisk/trunk' |
| 12:56.53 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:57.04 | bulkorok | jkroon: a=fmtp:18 annexb=no |
| 12:57.17 | DelphiWorld | hey bulkorok |
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| 12:57.29 | bulkorok | hi DelphiWorld |
| 12:57.45 | DelphiWorld | bulkorok: any idea about asterisk svn not working ? |
| 12:57.51 | jkroon | bulkorok, sweet thanks. |
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| 12:58.04 | bulkorok | DelphiWorld: no... |
| 12:58.04 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:58.13 | WIMPy | DelphiWorld: It's all down. |
| 12:58.14 | DelphiWorld | hi malcolmd |
| 12:58.25 | DelphiWorld | WIMPy: ahhhh you say is up dude, no ? |
| 12:59.10 | WIMPy | No. It's not jut SVN. |
| 12:59.22 | DelphiWorld | WIMPy: ;) |
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| 13:02.54 | bulkorok | sth is going on today in irc... |
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| 13:03.47 | DelphiWorld | why bulkorok |
| 13:04.36 | bulkorok | sunstorm!?... dont know ... I dropped three times today... |
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| 13:05.46 | DelphiWorld | crap... svn die... all down |
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| 13:14.15 | WIMPy | DelphiWorld: Use the chance to work on something that works. |
| 13:14.22 | DelphiWorld | WIMPy: lol ? |
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| 13:14.49 | WIMPy | Not really |
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| 13:23.48 | DelphiWorld | hi mihamina |
| 13:23.53 | DelphiWorld | mihamina: is Madagascar west or east africa ? |
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| 13:27.45 | phpboy | lol |
| 13:27.52 | phpboy | that's a random question |
| 13:27.57 | phpboy | I'm in South Africa |
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| 13:27.58 | phpboy | :P |
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| 13:33.50 | DelphiWorld | phpboy: no, is not random |
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| 13:40.18 | gavimobile | when using google voice as trunk do I need to port forward any special ports in my router for incomming calls? |
| 13:40.50 | gavimobile | like port 5222? |
| 13:41.03 | DelphiWorld | gavimobile: i belieuv no |
| 13:41.18 | DelphiWorld | oh..... fuck... :( |
| 13:41.23 | gavimobile | DelphiWorld: so what's this business about it using port 5222 |
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| 13:41.49 | DelphiWorld | gavimobile: is XMPP port. asterisk is a client, not a server. a server may need port forwarding, a client i belieuv no |
| 13:42.09 | DelphiWorld | gavimobile: maybe netanyahu dont want you to use google voice. |
| 13:42.25 | gavimobile | DelphiWorld: lol |
| 13:42.33 | DelphiWorld | gavimobile: be carful |
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| 14:15.52 | DelphiWorld | any fix for that svn things ;( |
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| 14:56.03 | gavimobile | folks I can't seem to receive incoming calls via google voice. im not sure if it's a google voice setting misconfiguration or a dialplan misconfiguration |
| 14:56.56 | pabelanger | without a debug log, neither can we |
| 14:58.34 | gavimobile | pabelanger: http://pastebin.com/fHvh74Hy |
| 14:59.19 | blitzrage | gavimobile: you're using 1.6.2 -- you need to use Asterisk 11 beta |
| 14:59.32 | blitzrage | chan_motif fixes many of hte issues |
| 14:59.35 | gavimobile | blitzrage: nope, im using asterisk 1.8 |
| 14:59.45 | gavimobile | blitzrage: I am however able to make outgoing calls |
| 15:00.15 | pabelanger | that trace show nothing related to jabber |
| 15:00.23 | pabelanger | also, looks like you are having routing issues |
| 15:01.12 | gavimobile | I've heard that before but I haven't expirenced any routing issues |
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| 15:01.18 | gavimobile | your talking about the retransmission? |
| 15:01.50 | pabelanger | yes |
| 15:01.51 | pabelanger | Retransmitting #3 (NAT) to 174.141.212.106:10260: |
| 15:02.55 | [TK]D-Fender | I fail to see anything about an incoming call in there... |
| 15:03.17 | gavimobile | here's a debug with some details from the beginning of the call http://pastebin.com/fJUMqyRY |
| 15:04.20 | gavimobile | [TK]D-Fender: that should mean there is a google voice setting config error. does it mean anything when it says Starting Gtalk/110-5fbe at motekPC,gavimobile@gmail.com,1 failed so falling back to exten 's' |
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| 15:04.20 | gavimobile | <PROTECTED> |
| 15:04.55 | [TK]D-Fender | ok, and where is the backup for that? |
| 15:06.08 | gavimobile | [TK]D-Fender: when I make the call with set debug off it shows a few lines.. from here http://pastebin.com/ESeFDxch |
| 15:06.16 | gavimobile | I see the call is going out |
| 15:06.32 | gavimobile | and I know for a fact its going out |
| 15:06.42 | [TK]D-Fender | gavimobile, It is telling you to your face that you have a dialplan error. WHERE IS THE DIALPLAN? |
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| 15:07.53 | adeel | ~itsplist-ca |
| 15:07.53 | infobot | itsplist-ca is, like, Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 15:07.55 | gavimobile | [TK]D-Fender: isn't there a command which I can run from the cli which will allow me to output the dialplan in a neater format? |
| 15:08.10 | gavimobile | I can also copy the dialplan as is |
| 15:08.17 | [TK]D-Fender | Command? this is EXTENSIONS.CONF. |
| 15:08.26 | gavimobile | nevermind ok just a sec |
| 15:09.30 | gavimobile | [TK]D-Fender: http://pastebin.com/ZN9Jn3bi |
| 15:09.45 | DelphiWorld | pabelanger, any clue when the svn repo will be up ? |
| 15:10.24 | pabelanger | what repo? |
| 15:10.36 | DelphiWorld | pabelanger: svn.digium.com |
| 15:11.13 | pabelanger | *shurgs* |
| 15:11.18 | [TK]D-Fender | gavimobile, "dialplan show @motekPC" |
| 15:11.30 | gavimobile | [TK]D-Fender: THAT'S THE COMMAND |
| 15:12.02 | gavimobile | [TK]D-Fender: http://pastebin.com/EJVc7fyg |
| 15:12.13 | DelphiWorld | what hapened pabelanger |
| 15:12.18 | pabelanger | I don't know |
| 15:12.20 | pabelanger | you tell me |
| 15:12.22 | pabelanger | also |
| 15:12.29 | pabelanger | mjordan says to check email |
| 15:12.49 | mjordan | if you're subscribed to the -users or -dev list, that is |
| 15:12.53 | mjordan | if not, I can't help you :-) |
| 15:13.24 | DelphiWorld | mjordan: i'm not regged, only to the business list;) |
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| 15:16.01 | [TK]D-Fender | gavimobile, First, set your gtalk context to the actual context the specifically contains the target extensions, not just one in an INCLUDE chain |
| 15:16.58 | gavimobile | [TK]D-Fender: correct me if im wrong, in my case that should be googlein |
| 15:17.04 | gavimobile | ? |
| 15:17.18 | DelphiWorld | Qwell, what was hapening to your irc ? |
| 15:17.51 | Qwell | DelphiWorld: Was it actually connecting? |
| 15:17.51 | [TK]D-Fender | gaveAre you telling me YOU don't know specifically where YUO wanted it to go? |
| 15:18.36 | Qwell | DelphiWorld: I was having IPv6 issues. It wasn't even showing itself connecting. |
| 15:18.49 | gavimobile | no that's not what im saying. I just need a clarification of weather the context of gtalk.conf should be connected to incomming or outgoing calls |
| 15:19.14 | gavimobile | im still 100% of the concept of contexts. I do know they need to match though |
| 15:19.21 | gavimobile | not 100%* |
| 15:20.21 | DelphiWorld | Qwell: connecting |
| 15:20.28 | DelphiWorld | and disconnecting with cause of flooding |
| 15:20.42 | [TK]D-Fender | call comes in matching some section in a device driver config it goes to a context it sepcifies |
| 15:20.55 | DelphiWorld | i dont belieuv Qwell is a flooder ;) |
| 15:20.57 | Qwell | DelphiWorld: how long ago? for how long? |
| 15:21.12 | DelphiWorld | Qwell: i'm here one our ago... every 5min or so |
| 15:21.28 | [TK]D-Fender | When I do Dial(SIP/john), Asterisk does not care what "context=" in [john]. that is when HE places calls. |
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| 15:22.23 | [TK]D-Fender | is in* |
| 15:22.38 | gavimobile | [TK]D-Fender: oh!!! so user@motekPC? |
| 15:22.46 | gavimobile | rather than just user |
| 15:22.54 | [TK]D-Fender | .... huh? |
| 15:22.59 | [TK]D-Fender | what is this "user" business? |
| 15:23.20 | [TK]D-Fender | your entry in gtalk.conf points to a CONTEXT for where to send calls that are from that account. SET IT |
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| 15:24.34 | Blue_Ice | hi, is packages.asterisk.org yum/rpm repo just down for the moment, or is it deferred? |
| 15:24.34 | gavimobile | [motekPC] |
| 15:24.44 | gavimobile | rather than [general] |
| 15:25.00 | Qwell | Blue_Ice: It may be down. We are experiencing issues with some community services right now. |
| 15:25.00 | gavimobile | I think that's what you mean |
| 15:25.43 | Blue_Ice | Qwell: ok thanks! Don't need it very urgent, just wanted to know if there was any hope on recovery. I'll try again in a few days |
| 15:26.07 | Qwell | Blue_Ice: It shouldn't be more than a few hours or so, I'm sure. |
| 15:29.12 | DelphiWorld | Qwell: svn is down too |
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| 15:40.20 | youjelly | Salute |
| 15:41.25 | DelphiWorld | salut youjelly |
| 15:41.29 | DelphiWorld | youjelly: english here |
| 15:41.56 | youjelly | DelphiWorld: I know, oldbro here, just different nick heh |
| 15:42.03 | DelphiWorld | ;) |
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| 15:47.06 | DelphiWorld | Qwell: finaly svn up |
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| 15:55.35 | carrar | hahaha http://tnw.to/i4o7 |
| 15:56.12 | carrar | "Give people an iPhone 4S, tell them it’s an iPhone 5. This is what happens [Video]" |
| 15:57.52 | gavimobile | what would this mean? Auto fallthrough, channel 'Gtalk/+110-39a8' status is 'UNKNOWN' |
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| 16:07.43 | leifmadsen | gavimobile: means dialplan fell through to no more priorities and hung up |
| 16:09.59 | gavimobile | leifmadsen: thanks |
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| 16:19.46 | gavimobile | ITS ALIVE |
| 16:19.51 | gavimobile | [TK]D-Fender: beat that :-p |
| 16:20.04 | gavimobile | [TK]D-Fender: j/k you know you're the man! |
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| 16:28.47 | nahamu | if I |
| 16:29.25 | nahamu | am moving offices, and my new T1 provider is asking me if I need NI1 or NI2, is there an easy way to figure out from the asterisk machine what I'm currently getting? |
| 16:31.05 | nahamu | nevermind, I found it in the order from the previous provider. |
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| 16:34.36 | DelphiWorld | nahamu: is nahum arabic ? |
| 16:34.55 | nahamu | hebrew |
| 16:35.18 | nahamu | nice old testament name :) http://en.wikipedia.org/wiki/Nahum |
| 16:35.18 | DelphiWorld | nahamu: pretty similar;-) |
| 16:35.39 | DelphiWorld | we have this name here nahamu |
| 16:35.54 | nahamu | what does it mean in Arabic? |
| 16:37.19 | DelphiWorld | nahamu: hmmmmmmmmm. do you know the words "naama" ? |
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| 16:37.58 | nahamu | I know pretty much 0 Arabic. In Hebrew "Na'ama" means something like "pleasant" |
| 16:39.03 | gavimobile | I call on the phone and I see an incoming call on my google talk client. Once I click answer on google talk the google talk client appears connected, though I don't hear anything on it. Meanwhile my phone keeps ringing as if the call wasn't answered. When I finally hang up the call on the phone the google talk client shows a message that the call ends. Any ideas on what's going on? |
| 16:40.02 | DelphiWorld | right right arabic too nahamu :-P |
| 16:40.12 | gavimobile | To fix it, I have to add bindaddr=<the_ip_address_of_the_server_running_asterisk> to [general] in gtalk.conf I've tried both my external ip and my local. when using external the phone doesn't ring, when using local, it acts the same as if bindaddr=0.0.0.0 and does the same thing as my question |
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| 16:51.18 | Blue_Ice | Qwell: thanks, seems to be fixed (yum repo) |
| 17:01.04 | Qwell | Blue_Ice: Yeah, sorry. Something just happened to that server, and I don't think we even know what yet. |
| 17:03.23 | DelphiWorld | Qwell: thank you for making svn up |
| 17:03.46 | Qwell | DelphiWorld: I didn't do anything. |
| 17:03.52 | DelphiWorld | Qwell: :) |
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| 17:15.38 | DelphiWorld | astcanary.c |
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| 17:15.41 | DelphiWorld | what is this file ? |
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| 17:18.34 | DelphiWorld | what pjlib is used for now ? |
| 17:18.48 | DelphiWorld | i see my asterisk is checking out pjsib;) |
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| 17:36.40 | pgrace | Is there a way to refer to a sip configuration setting in the dialplan? Like if I wanted to check on the default accountcode for a particular SIP account? |
| 17:37.51 | navaismo | What means this error--> ERROR: could not insert 'dahdi': Exec format error |
| 17:38.18 | navaismo | and dmesg show --> dahdi: version magic '3.1.9 mod_unload ARMv6 p2v8 ' should be '3.1.9 preempt mod_unload ARMv6 ' |
| 17:39.04 | pgrace | navaismo: your asterisk server has a kernel compiled with different options than the module. You need to recomile the dahdi modules. |
| 17:41.44 | pgrace | In case anyone wonders about the answer to my question, I figured it out myself: the built-in function SIPPEER. So, ${SIPPEER(peer,accountcode)} will return the accountcode for that peer. |
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| 18:17.40 | HappyPsychoD | Good day, I have a GSM gateway which when full can return SIP code 408, 503 or 600. If I have an asterisk based PBX what code do I need it to return to have the PBX failover to the next rule. |
| 18:18.45 | navaismo | hmm a groupcount can help |
| 18:19.00 | navaismo | but others can give you more help |
| 18:20.30 | HappyPsychoD | This is a switchvox PBX I am not seeing a groupcount option. |
| 18:21.41 | HappyPsychoD | We have two of these gateways at 8 sims each, to give us a total of 16. Right now I have rules to split load between the two. |
| 18:21.47 | navaismo | falling back to my second sentence |
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| 18:24.00 | *** join/#asterisk egonzalez_ergio (~egonzalez@186.137.1.3) |
| 18:25.54 | egonzalez_ergio | Hi! Some one can explain me how to send a fax with the free fax extention from Digium (receive works fine!) |
| 18:26.35 | jpsharp | I don't know how you'd do it in switchvox, but you can pass the "g" option to the Dial command and that will cause the system to continue through the dialplan after Dial returns, then you can check the SIP response code. |
| 18:27.35 | jpsharp | egonzalez_ergio: Dial() a number, use the SendFax application with an appropriate TIF file |
| 18:30.48 | HappyPsychoD | I'm guessing the 'g' maps to the 'Is Final' option in switchvox. None of the rules are final and we have others to catch the call (direct land lines). |
| 18:31.21 | jpsharp | I've never touched switchvox, so I don't know how to translate to it. |
| 18:31.37 | egonzalez_ergio | jpsharp, there are some proter emulator/connector for cups? |
| 18:31.42 | egonzalez_ergio | printer |
| 18:32.07 | jpsharp | egonzalez_ergio: I don't know, I've never looked for one. |
| 18:35.27 | jpsharp | You'd have to have a printer driver on the the source machine is able to capture the destination fax number. |
| 18:40.53 | nextime | so, it seems there's no way to make CURL() follow redirects in a reliable manner |
| 18:41.11 | nextime | i have to use SHELL(wget...) instead |
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| 18:53.59 | ryan42 | grrghhh |
| 18:54.10 | ryan42 | client's system went ganky today |
| 18:54.35 | ryan42 | nothing seems to have changed on the server but "Error parsing JSON" when attempting to get messages |
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| 18:55.08 | MLNoah | I'm trying to create a call queue to ring only devices that aren't in use, and apparently I'm not doing something right. Asterisk 10.4.0, have callcounter=yes set in the general section of sip.conf. The queue is set up for strategy=ringall, ringinuse=no, static members defined as member=SIP/peer within queues.conf. SIP peer can make an outbound call, but queue show still shows the queue |
| 18:55.08 | MLNoah | member as (Not in use) and the queue tries to ring them. |
| 18:55.21 | MLNoah | Any suggestions as to what I have misconfigured? |
| 18:56.04 | MLNoah | or is there a simpler/easier way to have incoming external calls not do Call Waiting without completely disabling Call Waiting on the endpoints? |
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| 19:25.38 | MLNoah | alas, sip.conf works better with callcounter=yes instead of calcounter=yes |
| 19:26.00 | jpsharp | counting calories? |
| 19:26.41 | MLNoah | at least my asterisk won't get fat, since it's watching wat it eats? |
| 19:26.47 | MLNoah | *what |
| 19:26.54 | MLNoah | that kind of typing day, it seems... |
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| 21:21.10 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.8.0 (2012/09/13), 1.8.16.0 (2012/09/13), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 21:31.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:34.14 | *** join/#asterisk Sidrov (~sid@AC9E0F51.ipt.aol.com) |
| 21:34.19 | Sidrov | hello all |
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| 21:37.45 | jpsharp | <PROTECTED> |
| 21:39.29 | Sidrov | anyone can help me with some usa IAX-2 providers ? |
| 21:39.55 | Qwell | Sidrov: Just get a SIP provider, IMO |
| 21:40.04 | Qwell | ~itsplist-us |
| 21:40.04 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 21:40.45 | Sidrov | :) |
| 21:40.50 | Sidrov | thanks |
| 21:41.01 | Sidrov | ~itsplist-ru |
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| 22:06.07 | waxstone | Hi All. Is is possible to have a conference hosted by me with asterisk(3 SIP users) make a call "as a group" to an external conference hosted by someone else? |
| 22:06.32 | WIMPy | Sure. |
| 22:06.54 | WIMPy | But You have a good chance for an annoying echo that way. |
| 22:07.15 | waxstone | sweet, can you give me some leads so I can research how to do this? |
| 22:07.30 | WIMPy | Just do it. |
| 22:07.49 | WIMPy | Asterisk doesn't care what the channels are connected to. |
| 22:07.50 | waxstone | a simple call() ? |
| 22:08.11 | WIMPy | What's that? |
| 22:08.23 | WIMPy | You can call one conf and transfer it to the other. |
| 22:08.47 | WIMPy | If you want something automatic there are several ways to originate calls. |
| 22:09.15 | waxstone | auto would be best |
| 22:10.50 | waxstone | well since I know its possible Ill try to figure it out. thanks WIMPy |
| 22:10.54 | WIMPy | call files, vial CLI 'channel originate' via AMI originate or via dialplan Originate(). |
| 22:12.13 | waxstone | thanks |
| 22:15.57 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 22:15.59 | tompaw | Hi. |
| 22:16.11 | tompaw | With 10.8, is DAHDI needed for MeetMe()? |
| 22:16.55 | leifmadsen | yes |
| 22:16.57 | leifmadsen | with all versions |
| 22:17.10 | leifmadsen | use ConfBridge() to not require DAHDI for timing |
| 22:17.51 | leifmadsen | MeetMe() uses audio mixing components of DAHDI on all versions, and will continue to into the future. If you can't use DAHDI for timing, then use ConfBridge() (or if you need video conf, new features, etc) |
| 22:17.53 | tompaw | leifmadsen: thanks, I'll stick to MeetMe(), as I had some very bad experience with ConfBridge this spring (h/w crashes of server) |
| 22:18.03 | leifmadsen | just saying |
| 22:18.12 | tompaw | much appreciated, thanks :-) |
| 22:21.07 | tompaw | Hm... if app_meetme has XXX in menuselect, does it mean it's unavailable, or force selected? |
| 22:21.32 | WIMPy | unav |
| 22:21.44 | WIMPy | See the bottom for missing dependencies. |
| 22:21.49 | leifmadsen | what WIMPy said |
| 22:21.53 | leifmadsen | likely missing res_timing_dahdi |
| 22:22.38 | tompaw | only says depends on dahdi(E), gonna try and find that res_ |
| 22:22.51 | leifmadsen | well res_timing_dahdi is dependent on dahdi |
| 22:23.03 | leifmadsen | if you've just installed dahdi, you need to ./configure && make install asterisk again |
| 22:23.56 | tompaw | it's way too late for work apparently. leifmadsen: just realised that, I ran configure prior to installing dahdi :/ |
| 22:30.38 | tompaw | leifmadsen: I'm looking at the PDF from your *Con 2011 about the call centres. What I'm deploying now is pretty much your non-distributed setup with sql cluster and web gui. |
| 22:31.01 | tompaw | The only difference is my agents have the ability to add a 3rd party to the call using that web gui. |
| 22:31.12 | tompaw | So every call is in fact a MeetMe conference. |
| 22:31.47 | WIMPy | I'd expect ConfBridge to scale better. |
| 22:32.55 | tompaw | As I recall, MeetMe was the only stable solution back in Oct 2011 when we built it up. ConfBridge() was literally killing two different servers we tested the solution on. |
| 22:33.40 | WIMPy | The versions in 1.8 and 10 are VERY different. |
| 22:33.47 | youjelly | kernel =>2.6 ?? |
| 22:33.49 | tompaw | So basically, the way I have it set up now, is that if an agent calls the customer using the web gui, a new MeetMe conference is created using the agent ID, and both him and the customer are dialed into that conference. |
| 22:34.02 | tompaw | youjelly: yes |
| 22:34.23 | WIMPy | But for 3-way calls you could eben use ChanSpy. |
| 22:34.32 | WIMPy | even |
| 22:34.33 | youjelly | never had issues with confbridge |
| 22:34.42 | WIMPy | That would only run on demand then. |
| 22:35.11 | WIMPy | used it right from the start as well. |
| 22:35.39 | tompaw | WIMPy: sounds good, let me show you what I got if you're interested |
| 22:35.42 | WIMPy | But I did have a few issues in 1.8. |
| 22:35.56 | tompaw | Well, it was 1.8 back then. |
| 22:36.08 | tompaw | But it got me seriously scared as there was no errors, memory dumps, nothing. |
| 22:36.22 | tompaw | Just unresponsive asterisk for like 20 minutes and then bang, hw restart. |
| 22:36.25 | WIMPy | As mentioned: That was a different story back then. |
| 22:36.54 | tompaw | what's the best pastebin for asterisk dialplan? |
| 22:36.56 | *** part/#asterisk WolWid (WolWid@pD9ED48E5.dip.t-dialin.net) |
| 22:37.06 | WIMPy | ~pb |
| 22:37.06 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:37.16 | WIMPy | without pastebin.com |
| 22:37.28 | youjelly | WIMPy: I'm having this strange issue using IPKall trinks |
| 22:37.33 | youjelly | trunks* |
| 22:37.47 | youjelly | DTMF works fine on every phone cept for iphones |
| 22:38.19 | youjelly | tried relaxdtmf etc |
| 22:38.34 | youjelly | toggling the value fixes it sometimes |
| 22:38.35 | WIMPy | Looks like you know where to find the issue. |
| 22:39.01 | WIMPy | You mean iphones calling via PLMN/PSTN to you? |
| 22:39.06 | youjelly | its weird, sometimes it works with relaxdtmf set to on sometimes with dtmf = inband |
| 22:39.17 | youjelly | but the problem occurs only with iphones |
| 22:39.24 | youjelly | yes |
| 22:39.47 | WIMPy | I heard they were not made to be used as phones. |
| 22:39.58 | youjelly | Lol |
| 22:41.09 | youjelly | I mean sometimes on other phones too, but with speaker phone on (its conferencing) so that is understandable, might be happening due to echo |
| 22:41.21 | youjelly | but very rare |
| 22:41.29 | youjelly | on iPhones its almost all the time |
| 22:51.28 | tompaw | WIMPy: http://pastebin.pl/65977 that's how I have it done |
| 22:51.49 | tompaw | this is assuming a customer is dialing in and hitting an Agent Queue |
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