| 00:00.09 | showme | layer 3 switchs are "typically" used to connect larger groups together, for all purposes, they are a simplified router. |
| 00:07.46 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:10.45 | el3slave | showme, excellent, thank you sir |
| 00:14.19 | xxiao | is there an opensource ATA somewhere? |
| 00:22.55 | ChannelZ | what, like circuitboard plans? |
| 00:24.43 | xxiao | slic+ethernet+software |
| 00:26.46 | xxiao | http://item.taobao.com/item.htm?id=5931735617 |
| 00:26.55 | xxiao | $16? |
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| 01:17.47 | Valen | hmm, having a slow memory consumption issue with a piaf purple install inside KVM |
| 01:17.53 | Valen | running asterisk 1.8.3.3 |
| 01:18.18 | Valen | doesn't make it into swap but after a while it just stops taking calls |
| 01:21.08 | Valen | any suggestions? |
| 01:29.47 | leifmadsen | could be anything... you could start debugging by following the instructions on the asterisk wiki |
| 01:30.23 | leifmadsen | once you do that, I'd file an issue |
| 01:30.31 | leifmadsen | or look for other issues that are already open |
| 01:31.21 | Valen | are there any signifigant differences in behaviour in 32 vs 64 bit? |
| 01:31.45 | Valen | i know there shouldnt be but real life seems to disagree with theory too often |
| 01:37.56 | leifmadsen | there should be no differences |
| 01:38.18 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 01:38.19 | leifmadsen | I've used asterisk on both 32 and 64 bit... the differences tends to just be in the libraries and their availability |
| 01:42.31 | *** join/#asterisk remx (~remx@corporate.runecho.com) |
| 01:42.52 | remx | What does CDR stand for? |
| 01:48.12 | Kobaz | call detail record |
| 01:48.22 | Kobaz | or |
| 01:48.50 | Kobaz | crazy drinking rabits |
| 01:50.25 | Kobaz | cruddy darn racoons |
| 01:50.38 | justdave | ok, so I'm on Asterisk 1.8.4.1, and I have an incoming number that's on a PRI, and one that comes in via SIP, and a menu option on the IVR which sends the call into an appliance box |
| 01:50.40 | *** join/#asterisk Vicksters (~Vicksters@modemcable067.73-83-70.mc.videotron.ca) |
| 01:51.15 | justdave | if I call in via the PRI and dial the menu option for the appliance box, the call transfers, the log on asterisk says the box has answered, but the caller just hears ringing still. |
| 01:51.37 | justdave | if I call in via the number that comes in on SIP and do the same thing, the log entries look exactly the same, except that it actually connects the caller |
| 01:51.55 | justdave | the connection to the external box is via SIP |
| 01:52.16 | justdave | why would asterisk thing the call connected, and the caller on the PRI would continue to get a ring no answer? |
| 01:52.20 | justdave | think* |
| 01:52.23 | Vicksters | Hi, I'm having a problem with chan_dahdi. When I try "dahdi show status", I get "No DAHDI found. Unable to open /dev/dahdi/ctl: Permission denied". Permissions and ownership in /dev/dahdi look fine. Anyone has a clue? |
| 01:52.36 | Kobaz | Vicksters: what user is asterisk running as |
| 01:52.42 | Vicksters | asterisk |
| 01:52.43 | justdave | Vicksters: define "fine" :) |
| 01:52.48 | Kobaz | ls -al /dev/dahdi/ctl |
| 01:52.58 | Vicksters | crw-rw---- 1 asterisk asterisk 196, 0 2011-06-06 21:33 ctl |
| 01:53.08 | Kobaz | and what about /dev/dahdi |
| 01:53.31 | Vicksters | .. Well now you make me feel retarded. |
| 01:53.50 | Vicksters | Thanks. ;p |
| 01:53.51 | Kobaz | if something is complaining about permission denied, obviously something is wrong... |
| 01:55.07 | Vicksters | That was it. Thanks a lot Kobaz |
| 01:56.03 | Kobaz | heh |
| 01:56.43 | Kobaz | i have stuff in my startup asterisk script in /etc/init.d that sets the permissions on /dev/dahdi before starting asterisk |
| 01:56.55 | Kobaz | because with udev, all your permissions will be reset on reboot |
| 01:57.17 | Vicksters | oh.. now thats annoying. |
| 02:00.38 | Valen | turned off a few modules, seems to have taken memory usage from 30% after a restart to 5% |
| 02:00.49 | Kobaz | justdave: probably mismatched signalling on the pri, or the remote side is broken in some fashion... or you're getting errors on the t1/pri |
| 02:00.53 | Valen | see if it grows now |
| 02:04.37 | remx | Is real time replication of the cdr to remote servers possible? |
| 02:18.03 | leifmadsen | remx: what do you mean "real time" ? |
| 02:18.23 | leifmadsen | CDR only gets written at the time of call completion |
| 02:18.50 | Kobaz | chicken dogs and roosters |
| 02:18.52 | showme | you don't know call length until call completion.. |
| 02:19.14 | remx | leifmadsen: yes after the call is completed |
| 02:19.27 | remx | technically that's real time :) |
| 02:19.36 | Kobaz | remx: you can replicate using your database engine |
| 02:19.43 | Kobaz | ie, mysql/postgres have replication services |
| 02:19.46 | leifmadsen | remx: that's just how it works anyways :) |
| 02:19.58 | leifmadsen | you don't do anything other than enable it |
| 02:20.20 | leifmadsen | like Kobaz said, just use MySQL or PostgreSQL replication :) |
| 02:20.31 | leifmadsen | it works for my on MySQL by setting up master-master replication |
| 02:20.36 | remx | the remote server is a shared web hosting provider, I can't set it up as a slave. If from the top of your head there's not another method, I'll just resort to php and crons |
| 02:21.27 | remx | Kobaz: thanks for the cdr definition |
| 02:21.42 | Kobaz | crows ducks and rainbows |
| 02:21.57 | showme | why would you want CDR to go to a public location? One wrong oops by your ISP and the whole world has your call records. |
| 02:22.03 | Kobaz | remx: heh, np |
| 02:22.25 | showme | I mean web host, not isp.. |
| 02:23.53 | Kobaz | hmm |
| 02:24.17 | remx | This server running asterisk is 'locked' off from the outside and only the IPs of our trunk providers can get passed the our firewalls. I'd rather have a seperate entity holding some data for other staff to perform there duties |
| 02:25.36 | showme | your staff doesn't have access to the network? worst case, if you're trying to isolate the asterisk box, setup another mysql box on the lan for them. |
| 02:27.05 | showme | with your setup, the first time the ceo calls his mistress, there will be heads rolling :) |
| 02:28.14 | showme | maybe I'm just too security consious, but CDR should be very protected and privileged info. |
| 02:29.02 | showme | for all purposes, thats your customer list, and many other things, all rolled into one |
| 02:29.06 | remx | the telephone network is on its own network |
| 02:29.49 | showme | Yeah, I get that, but pushing data to a public server?? |
| 02:31.24 | remx | people hold bank details on shared web hosts |
| 02:31.41 | remx | it still passes pci compliancy and banks can still do business with you |
| 02:32.12 | remx | all servers that have routes to them by the internet are public |
| 02:32.23 | remx | I |
| 02:32.28 | showme | shared web hosts usually don't fall under being pci complient. |
| 02:32.31 | remx | I'll have to resort to using php |
| 02:32.39 | remx | I think you'll find they are |
| 02:33.14 | remx | unlimitedwebhosting.co.uk for example, my bank has approved them |
| 02:33.19 | showme | I do pci compliece auditing, not a chance on my watch. |
| 02:33.26 | remx | yes and so do i |
| 02:34.02 | showme | if a sql database has a direct exposure to the internet, its not complient. |
| 02:34.04 | remx | anyway thank you all for your time, appreciate it. take care |
| 02:37.04 | Kobaz | speaking of pci complience |
| 02:37.15 | Kobaz | i should set up my firewall to pass these stupid checks |
| 02:37.20 | Kobaz | i'm getting charged like 90 bucks a month |
| 02:37.42 | Kobaz | i don't run anything on my web server, and it wants me to open all these ports to stuff I don't even run |
| 02:37.59 | Kobaz | it's completely retarded |
| 02:39.07 | Kobaz | jhmichelle352: your gdb output still doesn't match anything |
| 02:39.08 | Kobaz | er |
| 02:41.41 | remx | Kobaz: if you have financial insurance, you'll pay a smaller premium if you isolate the server. there's more chance of theft internally than externally |
| 02:42.32 | *** part/#asterisk remx (~remx@corporate.runecho.com) |
| 02:44.02 | Kobaz | but these tests are just completely ridiculous |
| 02:45.16 | Kobaz | i would say the risk of internal theft is just about 0 since i'm the only one with access to any of it |
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| 02:57.01 | phix | hey hey |
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| 05:18.06 | Valen | remx could perhaps use triggers to make things nicer |
| 05:18.44 | Valen | Kobaz: to be compliant they want you to *open* ports? |
| 05:18.48 | Kobaz | yes |
| 05:18.53 | Kobaz | for their scanner to run |
| 05:19.10 | Kobaz | i block everything except for 443 |
| 05:19.16 | Kobaz | and i fail all their tests |
| 05:19.26 | Valen | could you not perhaps just change the message to reject on those ports rather than drop? |
| 05:19.27 | Kobaz | because they want to see the ssh version that's running, and what ftp server is running |
| 05:19.42 | Kobaz | i have a reject, so you get connection refused as if there is no service |
| 05:19.57 | Valen | i set mine for drop ;-> |
| 05:20.20 | Kobaz | anyway |
| 05:20.21 | Valen | I can understand their desire to ensure your using a current ssh |
| 05:20.22 | Kobaz | nap time |
| 05:20.26 | Valen | have fun |
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| 06:05.28 | schmidts | good morning |
| 06:05.47 | WIMPy | Moin |
| 06:06.38 | schmidts | how are you WIMPy? i hope you didnt eat any vegatables latly? ;) |
| 06:07.01 | WIMPy | Only frozen ones :-) |
| 06:07.16 | WIMPy | Except for onions. |
| 06:07.19 | schmidts | :D |
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| 06:26.40 | irroot | schmitdts WIMPy maybe i wont go to speciality german restaurant down the road :P |
| 06:27.39 | schmidts | irroot oh thats no problem, just dont eat any vegatables only if they are boiled to death :D |
| 06:28.11 | irroot | mmm kraut ?? hehe its pretty dead but is it deadly :D |
| 06:28.31 | coppice | Oh, the dilemma. Should I risk e-coli with veggie form the EU, or radiation with veggie from Japan. |
| 06:28.50 | ChannelZ | screw that, eat meat |
| 06:29.04 | irroot | coppice glow in the dark green tea ?? |
| 06:29.18 | coppice | what about Kobe beef? |
| 06:29.56 | irroot | yeah thats what i tell my girl and she aint a vegitarian either |
| 06:30.47 | coppice | my wife keeps checking the manufacturing date on japanese foods, to verify its before 11th march |
| 06:31.21 | schmidts | :D |
| 06:34.08 | Wiretap7 | lol |
| 06:34.46 | Wiretap7 | I eat japanese food every day and I'm not glowing yet |
| 06:35.32 | coppice | the sukiyaki sauce we bought last week was made in mid feb, so we could be getting close to the cutoff point |
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| 07:59.53 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 07:59.53 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4.2 (2011/06/04), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 08:45.13 | k3asd` | hi |
| 08:54.28 | tuxx- | ohai |
| 08:54.47 | irroot | lo |
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| 09:18.24 | Marioooooo | hellooooo |
| 09:19.58 | irroot | lo there hows luigi |
| 09:21.21 | Marioooooo | is there any te122 expert here? i have a simple question |
| 09:21.35 | WIMPy | ~ask |
| 09:21.35 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:23.22 | Marioooooo | is there a way to check if the echo cancel hardware module is enabled and is working fine on a te122 card? Thanks |
| 09:23.54 | WIMPy | dmesg|grep VPM |
| 09:25.02 | Marioooooo | [ 14.124952] wcte12xp 0000:05:00.0: Booting VPMADT032 |
| 09:25.03 | Marioooooo | [ 14.191977] wcte12xp 0000:05:00.0: VPM present and operational (Firmware version 125) |
| 09:25.15 | Marioooooo | :) |
| 09:25.25 | Marioooooo | thanks |
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| 09:26.00 | Marioooooo | what about system.conf and chan_dahdi.conf |
| 09:26.05 | cneb3000 | how do everyone |
| 09:27.12 | Marioooooo | in system.conf i disabled the sot echocancel : # echocanceller=mg2,1-15,17-31 |
| 09:27.13 | WIMPy | I think they don't matter, other that the SWEC modules will be loaded, but not used. |
| 09:27.25 | Marioooooo | *soft |
| 09:28.10 | Marioooooo | in chan_dahdi.conf i have : echocancel=yes |
| 09:28.10 | Marioooooo | echocancelwhenbridged=no |
| 09:28.19 | Marioooooo | that's all |
| 09:28.44 | Marioooooo | @ WIMPy thanks for your help |
| 09:28.47 | Marioooooo | appreciated |
| 09:28.56 | WIMPy | I think you should comment out the echocanceller= line. |
| 09:29.13 | WIMPy | Otherwise it loads the mg2 module and initialises it for every channel. |
| 09:29.44 | Marioooooo | yes I did it.. in system.conf there is the "#" |
| 09:29.55 | Marioooooo | # echocanceller=mg2,1-15,17-31 |
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| 10:53.39 | cusco | hi |
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| 11:08.04 | doolittlework | good day |
| 11:09.25 | doolittlework | How would one go about creating a webpage where one can adjust all the settings for a sip user and via the webpage populate the sip.conf file? |
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| 11:15.46 | doolittlework | looks like sip realtime config hope i am right |
| 11:17.12 | kaldemar | make an app that modifies sip.conf or the db if you use realtime and then tells asterisk to re-read configs. |
| 11:17.58 | kaldemar | nothing forces you to use realtime. you can as well use the config files. |
| 11:19.35 | WiretapMac | doolittlework: had a look at asterisk-gui or freepbx? |
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| 11:27.28 | dobby156 | hi |
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| 11:32.19 | dobby156 | does any one have an opinion over whether the MYSQL cmd or the ODBC func is better? We do currently use MYSQL as our database |
| 11:37.54 | irroot | im a fan of ODBC with asterisk mostly historic reasons i use ODBC+PgSQL |
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| 11:44.48 | Kobaz | i've had some issues with odbc, so i go with the direct drivers if possible, direct to pgsql |
| 11:45.35 | dobby156 | irroot: well we are a point right now where we can choose one or the other, all our DB's are mysql and that is unlikely to change anytime soon (ever) |
| 11:46.28 | dobby156 | we haev found myql cmds to be klunky till now, your saying ODBC is very much the same? |
| 11:46.36 | irroot | well ODBC+MySQL is a option too |
| 11:46.50 | irroot | can use both also so its versitile |
| 11:47.15 | irroot | func_odbc may be something you use sometimes in addition to others |
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| 11:50.51 | dobby156 | irroot: so are you recommending using the MYSQL cmd for most purposes over the odbc |
| 11:51.09 | dobby156 | irroot: we would rather use one or the other |
| 11:51.26 | dobby156 | irroot: like what are the advantages of each, thanks |
| 11:51.58 | irroot | func_odbc i find flexible and versitile use it often |
| 11:52.10 | dobby156 | irroot: thank you |
| 11:52.22 | dobby156 | ;D |
| 11:52.26 | irroot | i dont use MySQL so cant compare |
| 11:53.02 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 11:53.29 | irroot | E-Bola that is so last milenium suggest E-Coli |
| 11:53.44 | E-bola | I have an issue where the music on hold stutters and have very degraded sound when people are calling in via a SIP trunk, but when its via an IAX trunk it does not apear to happen |
| 11:53.51 | WIMPy | :) |
| 11:53.56 | E-bola | Im not sure how to debug, there's no errors or warnings in the console |
| 11:54.06 | E-bola | irroot: ohh ebola is much worse trust me :P |
| 11:54.13 | dobby156 | irroot: out of interest does the ODBC use a unified (SQL) syntax or the syntax of the specific database |
| 11:55.06 | irroot | ODBC Open DataBase Connector it is not a DB it is a abstraction layer so it will talk to MySQL |
| 11:55.32 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:55.42 | irroot | you still need xSQL with ODBC as a bridge between it and Asterisk |
| 11:56.04 | irroot | E-Bola check timing almost certain a issue |
| 12:03.04 | skrusty | is it possible from the AMI to find out which channel caused another channel to spawn? I.e. without them being bridged/linked? |
| 12:03.27 | skrusty | i.e. phone a calls b, but until they get linked, you can't tell that a caused b to ring |
| 12:03.32 | E-bola | irroot: ya... but what to check? What module im using? I'm using res_timing_pthread because the others give problems with iax under debian |
| 12:04.03 | E-bola | What i dont understand is how its only MOH thats affected actual conversations work fine on both sip and iax |
| 12:04.57 | irroot | what codecs you using on SIP/IAX ?? is it same ? |
| 12:05.01 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:05.15 | E-bola | Yep same, raw ulaw |
| 12:05.35 | E-bola | im on the process of converting the moh from mp3 to ulaw |
| 12:05.45 | E-bola | since thats the only thing i coudl think of |
| 12:06.11 | irroot | or slin ... i prefer raw slin moh |
| 12:06.29 | irroot | but for ulaw only ulaw will be best |
| 12:08.23 | WIMPy | skrusty: Isn't the unique ID available? |
| 12:11.34 | skrusty | yes, but that wont tell you which call spawned the channel...? they differ |
| 12:11.39 | doolittlework | please could someone please have a look and see where i am going wrong, trying to use realtime http://pastebin.com/WdCXrzft |
| 12:11.43 | skrusty | or am i missing something? |
| 12:13.08 | WIMPy | skrusty: I did call tracking some years ago and I think that part worked. But I can't remember what I had to do. |
| 12:13.42 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 12:13.47 | skrusty | god damn it! :) |
| 12:13.59 | skrusty | bloody windows |
| 12:14.11 | WIMPy | skrusty: I did call tracking some years ago and I think that part worked. But I can't remember what I had to do. |
| 12:14.15 | WIMPy | kill it |
| 12:15.05 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 12:15.06 | skrusty | :) |
| 12:15.08 | wdoekes2 | doolittlework: sip show peer <peername> load |
| 12:15.09 | *** join/#asterisk cneb3000 (~cneb3000@87.127.15.113) |
| 12:15.15 | wdoekes2 | without the load, realtime peers do not show up |
| 12:15.22 | skrusty | it just seems that the new uniqueid has no reference to the original channel :/ |
| 12:15.31 | skrusty | i understand it's much better in 1.8 |
| 12:15.37 | *** join/#asterisk voipforces (~chatzilla@69.165.199.195) |
| 12:15.42 | skrusty | but need to get it working on 1.6 first :) |
| 12:15.58 | skrusty | use realtime cache (i do anyway) |
| 12:16.13 | WIMPy | Both didn't exist, when I did it :-) |
| 12:16.18 | *** join/#asterisk crewnecktech (~crewneckt@98.100.29.105) |
| 12:16.18 | skrusty | hehe :) |
| 12:16.19 | wdoekes2 | (and it might be that you want 'sip show USER ..' instead for your sipusers config) |
| 12:16.29 | voipforces | Hi All, have a question about AMI originate, failed extxtension, PRI and disonnected... Details are here http://pastebin.com/6dgrAM3y |
| 12:16.56 | skrusty | WIMPy: it's enough to drive a man nuts :) |
| 12:16.56 | voipforces | Basically trying to find real noanswers from disconneted numbers |
| 12:17.01 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-zltbfsohalhxmxbo) |
| 12:17.39 | skrusty | have you checked PRI status codes? |
| 12:18.01 | wdoekes2 | cause 19? ${HANGUPCAUSE} ? |
| 12:18.04 | skrusty | im sure there is a signal for number not in use |
| 12:18.04 | irroot | HANGUPCAUSE ?? voipforces that is the one |
| 12:18.06 | WIMPy | voipforces: What is that REASON you're using there? Try HANGUPCAUSE. |
| 12:18.15 | voipforces | I checked ${REASON} and ${HANGUPCAUSE} |
| 12:18.32 | WIMPy | 1=unallocated number |
| 12:18.40 | voipforces | check line 49,50 and 67,68 in the paste bin |
| 12:18.43 | voipforces | they are the same |
| 12:19.16 | voipforces | WIMPy: HANGUPCAUSE is 0 in both case |
| 12:19.38 | hensema | hi there, does anybody have a good script to convert an mp3 to gsm format (for music on hold)? |
| 12:19.56 | *** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 12:20.03 | WIMPy | voipforces: That's clearly a phone that has been ringing and the call got disconnected due to timeout. |
| 12:20.06 | wdoekes2 | are you checking hangupcause after your own Hangup()? |
| 12:20.26 | irroot | mpg123 .... |sox .... |
| 12:20.29 | gavimobile | folks, im getting vm's by email via asterisk from asterisk@localhost. how can I change this please |
| 12:20.43 | skrusty | voicemail.conf? |
| 12:20.51 | voipforces | WIMPy: The first one no, I cann it from a land line 7057296882 or from the same PRI and get the "this number has been disconnected" message |
| 12:20.55 | wdoekes2 | gavimobile: your MTA config |
| 12:21.01 | cusco | WIMPy: hey, my telco replied that they have overlap active, but what I requested was to have a number with more than 9 digits comming in ISDN and that was not permitted by law |
| 12:21.06 | hensema | irroot: I've tried sox: sox Heartland.mp3 -t raw -r 8000 -s -2 -c 1 Heartland.gsm |
| 12:21.06 | cusco | so... what is overlap again? |
| 12:21.07 | irroot | mailcmd in voicemail.conf |
| 12:21.26 | hensema | irroot: but that gives me a raw pcm file, not a gsm file |
| 12:21.31 | gavimobile | irroot: in centos5 where is it located, |
| 12:21.31 | irroot | hensema need to use mpg123 and pipe it to sox |
| 12:21.37 | gavimobile | wdoekes2: where is that mta file located |
| 12:21.42 | WIMPy | voipforces: Well, if they tell you bullshit, you're screwed. |
| 12:21.55 | *** join/#asterisk bchia (~Adium@nat/digium/x-bockdvtmrottiqdy) |
| 12:22.00 | wdoekes2 | gavimobile: depends.. exim? postfix? sendmail? qmail? |
| 12:22.14 | gavimobile | sendmail wdoekes2 |
| 12:22.15 | kaldemar | cusco: when a dialed number is sent one digit at the time. as opposed to en block, when all digits are sent in a single message. |
| 12:22.33 | voipforces | WIMPy: Well, the PRI debug shows a difference: Cause: User alerting, no answer (19) for a normal hangup |
| 12:22.47 | voipforces | WIMPY: Cause: Normal Clearing (16) for a no answer |
| 12:22.49 | WIMPy | cusco: It just means that the destination number need not be sent completely in the first message, but can come later or be split into multiple messages. |
| 12:23.06 | voipforces | but I can not get that 16 or 19 cause code in the dialplan |
| 12:23.28 | cusco | ow... so that woudln't be needed... |
| 12:23.51 | cusco | I was looking for a way to dial the number and pass #999 at the end of it or so |
| 12:23.57 | cusco | and have asterisk read those 999 |
| 12:24.01 | wdoekes2 | gavimobile: I don't know sendmail config.. but you can probably use mailcmd like irroot suggested |
| 12:24.01 | WIMPy | voipforces: You should see 16 if YOU gave up waiting, 19 if your telco gave up waiting. Anyway the "user alerted" part says it has been ringing. |
| 12:24.11 | wdoekes2 | see: man 1 sendmail |
| 12:24.15 | doolittlework | wdoekes2: i keep on seeing peer not found using sip show peer 1000 load, how does one know if asterisk is indeed reading the data from the mysql database? |
| 12:24.22 | gavimobile | ok thanks folks |
| 12:24.26 | wdoekes2 | sip show user 1000 load |
| 12:24.32 | voipforces | WIMPy: ok misiinterpreted that part. |
| 12:24.42 | WIMPy | cusco: Looks like youtube isn't the only place where learn that you're in the wrong country. |
| 12:24.55 | irroot | mailcmd=/usr/sbin/sendmail -f asterisk@XXXXXX -t |
| 12:24.56 | WIMPy | +you |
| 12:24.59 | cusco | WIMPy: LOL |
| 12:25.08 | cusco | yea youtube tells me that sometimes |
| 12:25.15 | *** join/#asterisk benji (~benji@c-71-62-106-255.hsd1.va.comcast.net) |
| 12:26.01 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 12:26.09 | cusco | gavimobile: you m ay not need a mta at all, I use heirloom-mailx (former nail) and use my gmail configuration at it... (any smtp/pop/imap account will do) |
| 12:26.41 | doolittlework | wdoekes2: user laod not found |
| 12:26.57 | wdoekes2 | doolittlework: core set debug 5 res_config_mysql |
| 12:27.39 | coppice | its annoying when youtube presents you with a screen full of suggested videos, and every ones gives you "not available in your country" |
| 12:28.10 | benji | Hi all. I have a quick (I think) question. I'd like to make a pseudo conference room where three calls are connected and two of the people can hear each other but the third person can only be heard by one of the others. |
| 12:28.10 | WIMPy | coppice: Oh, yes. |
| 12:28.33 | gavimobile | thanks cusco |
| 12:28.44 | WIMPy | benji: Use ChanSpy() |
| 12:29.21 | wdoekes2 | WIMPy: does chanspy have a whisper-to-one-leg-only? |
| 12:29.25 | voipforces | WIMPy: thanks for the help/hints. |
| 12:29.35 | benji | WIMPy: thanks, that looks promising! |
| 12:29.44 | WIMPy | wdoekes2: Yes. The one, the other or both. |
| 12:29.54 | wdoekes2 | ok, didn't know that |
| 12:30.14 | WIMPy | You can even change that interactively. |
| 12:31.59 | WIMPy | Or was it none, one or both? But not the other? |
| 12:34.15 | skrusty | can i pass a SIP/xxx device or chanspy, or even a specific channel? OR does it have to be done with this odd prefix thing? |
| 12:34.55 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:35.03 | WIMPy | The prefix can b the whole channel anme, I think. |
| 12:35.27 | skrusty | ok cool |
| 12:35.29 | wdoekes2 | yes.. and then there's the e(enforce) option, to explicitly match only some accounts |
| 12:35.39 | skrusty | just tried it and it didn't work, so thought i would check :) |
| 12:36.20 | skrusty | doing it from an AMI call originate |
| 12:36.37 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 12:45.46 | doolittlework | wdoekes2: should i use users or peers? |
| 12:47.01 | wdoekes2 | I only use friends.. but you probably want either friend or peer |
| 12:48.25 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.83.116) |
| 12:50.02 | doolittlework | wdoekes2: could you please paste your extconfig and rea_mysql.conf please, i have been at it for hours but can not seem so get it to work, i know i am missing something |
| 12:52.06 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 12:52.10 | wdoekes2 | sipusers => odbc,osvpi_db,asterisk_sipfriend |
| 12:52.10 | wdoekes2 | sippeers => odbc,osvpi_db,asterisk_sipfriend |
| 12:52.10 | wdoekes2 | sipregs => odbc,osvpi_db,asterisk_sipreg |
| 12:55.11 | doolittlework | wdoekes2 not using mysql? |
| 12:55.34 | wdoekes2 | yes/no, not using res_mysql |
| 12:55.52 | doolittlework | ok i see |
| 12:56.07 | doolittlework | how do you talk to the mysql? |
| 12:57.01 | wdoekes2 | takes a red marker and circles the odbc part in the previous lines |
| 12:59.12 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:59.20 | doolittlework | lol |
| 12:59.57 | doolittlework | doolittlework: hits himselve |
| 13:03.59 | *** join/#asterisk DanFromUK (DanFromUK@2.27.2.10) |
| 13:04.07 | DanFromUK | hi |
| 13:05.52 | DanFromUK | Can any help? i've got a dialplan, with an incoming call, ive set CDR(accountcode), but when the call gets placed into a queue, the accountcode is lost. the queue_member is a local/ channel which dials out via a sip trunk based on the accountcode. any ideas how i can keep the accountcode inside the queue? |
| 13:08.47 | kaldemar | DanFromUK: show how you set the code and how you verify that it is lost. |
| 13:09.44 | irroot | Queue does do some funky cdr stuff |
| 13:10.11 | DanFromUK | <PROTECTED> |
| 13:10.39 | DanFromUK | in the Local extension, it performs an sql lookup |
| 13:10.42 | DanFromUK | - Found no rows [SELECT RateList FROM accounts WHERE accountname=''] |
| 13:10.49 | DanFromUK | works fine when i don't use the queue |
| 13:11.26 | DanFromUK | if i send the call straight to the Local extension, it works fine. |
| 13:12.11 | DanFromUK | Is there any way to set the cdr(accountcode) manually for a queue? |
| 13:13.46 | irroot | why not use astdb for this purpose |
| 13:15.00 | kaldemar | or func SHARED |
| 13:15.01 | DanFromUK | can you explain? |
| 13:16.34 | DanFromUK | alternatively, in the Local extension, is there any way of telling which queue the call is coming from? |
| 13:16.47 | DanFromUK | Local being one of the queue members. |
| 13:17.18 | irroot | <PROTECTED> |
| 13:18.16 | kaldemar | DanFromUK: see setqueuevar option in the sample queues.conf. |
| 13:19.18 | DanFromUK | kaldemar: cool. that looks perfect |
| 13:20.01 | DanFromUK | kaldemar: is setqueuevar available in 1.4? |
| 13:22.06 | *** join/#asterisk wonderworld (~ww@port-92-201-47-33.dynamic.qsc.de) |
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| 13:29.37 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:5ce6:4f07:90b5:8bbe) |
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| 13:35.10 | DanFromUK | does realtime mysql queue support 'setqueuevar'? |
| 13:38.16 | Corydon76-home | DanFromUK: The realtime support is completely independent from any backend driver.. So, yes. |
| 13:39.33 | DanFromUK | thanks, however it doesnt seem to be setting the QUEUENAME variable. i'm going to try to change 'yes' to '1' |
| 13:40.22 | Corydon76-home | That won't make a difference |
| 13:40.56 | DanFromUK | ur right, it didnt. |
| 13:41.17 | DanFromUK | any idea why its not setting ${queuename}? |
| 13:42.04 | Corydon76-home | Did you reload after changing the queue? |
| 13:42.47 | DanFromUK | the queue is in mysql so shouldnt need to |
| 13:42.59 | DanFromUK | but i did do this: realtime load queues name kesher_101 |
| 13:43.17 | DanFromUK | and the setqueuevar shows up |
| 13:44.24 | Corydon76-home | Yeah, that's because you're hooking into the same routine |
| 13:44.51 | DanFromUK | what do you mean? |
| 13:45.16 | Corydon76-home | That CLI line does exactly the same thing as the queue module |
| 13:45.34 | DanFromUK | so setqueuevar should be working. |
| 13:45.46 | Corydon76-home | but if the queue is already in memory, it won't change anything |
| 13:46.19 | DanFromUK | according to voipinfo, the queue settings are refreshed each time a new caller joins the queue |
| 13:46.47 | Corydon76-home | Ah, voipinfo. Sometimes right, good to get a clue, mostly wrong. |
| 13:47.00 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 13:47.13 | DanFromUK | ok, how do i reload the queue without reloading the whole of asterisk? |
| 13:47.17 | DanFromUK | queue reload? |
| 13:47.25 | Corydon76-home | module reload app_queue.so |
| 13:47.26 | DanFromUK | nope |
| 13:47.28 | DanFromUK | ah |
| 13:48.07 | Corydon76-home | or: queue reload parameters |
| 13:48.12 | DanFromUK | nope. still not working. |
| 13:48.57 | Corydon76-home | What version are you running? |
| 13:49.11 | DanFromUK | 1.4 |
| 13:49.22 | Corydon76-home | Ah, I'm looking at 1.8 |
| 13:49.31 | sxpert | farenheit 451 spotted |
| 13:49.58 | DanFromUK | is it supported in 1.4? |
| 13:50.10 | Corydon76-home | It should be, though the commands might be different |
| 13:50.21 | Corydon76-home | but if there's a bug, it won't be fixed now. |
| 13:50.40 | Corydon76-home | Security fixes only |
| 13:51.00 | irroot | unless you fix it yourself :P |
| 13:51.38 | *** part/#asterisk voipforces (~chatzilla@69.165.199.195) |
| 13:51.52 | Corydon76-home | Aha, setqueuevar is not supported in 1.4 |
| 13:52.02 | Corydon76-home | It's a new feature (either starting in 1.6.2 or 1.8) |
| 13:52.12 | DanFromUK | ah |
| 13:52.12 | DanFromUK | ok |
| 13:52.18 | DanFromUK | ill make a workaround |
| 13:52.28 | DanFromUK | is 1.8 ready for production yet? |
| 13:53.02 | Corydon76-home | Always test before deploying a new version. There are plenty of people using 1.8 in production. |
| 13:53.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:53.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.26 | DanFromUK | ok, i'll add it to the todo list |
| 13:53.37 | Corydon76-home | Morning, putnopvut |
| 13:54.19 | putnopvut | Corydon76-home: howdy, how goes? |
| 13:54.37 | Corydon76-home | Yawny |
| 13:55.01 | irroot | im yawny but after doing a days work :P |
| 13:55.03 | tzanger | morning |
| 13:55.12 | jaytee | hmmmm, no SendDTMF in AGI? |
| 13:56.05 | DanFromUK | thanks for your help |
| 13:56.32 | *** join/#asterisk remx (~remx@corporate.runecho.com) |
| 13:56.40 | Corydon76-home | jaytee: why would there be? |
| 13:57.13 | jaytee | Corydon76-home, dunno. |
| 14:03.45 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 14:04.17 | DanFromUK | if i add something to func_odbc, how can I reload it? |
| 14:04.38 | irroot | module reload func_odbc.so |
| 14:05.04 | DanFromUK | thanks |
| 14:07.35 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 14:08.08 | irroot | Big Hammer |
| 14:08.13 | DanFromUK | lol |
| 14:08.23 | Katty | what is big hammer??? |
| 14:08.34 | irroot | 16lb is about right |
| 14:08.38 | malcolmd | it's the solution |
| 14:09.26 | orn | Katty: Big Hammer is not the solution. You need to deliberately the config. |
| 14:09.46 | Katty | how to deliberate the config??? |
| 14:09.57 | orn | Just deliberately the whole thing. |
| 14:10.12 | Katty | how to do this plz??? |
| 14:10.15 | dr0ck | MAGIC TIPS |
| 14:11.20 | Katty | hello r u there?? |
| 14:11.27 | DanFromUK | ok, last question. If i've got a variable containing 'kesher_201', how I can strip out the 'kesher' bit? bearing in mind that is 'anylength_3digits' |
| 14:11.35 | Katty | oww side hurts |
| 14:11.38 | Katty | too much giggling |
| 14:12.05 | orn | DanFromUK: Do you want to strip the underscore as well? |
| 14:12.19 | irroot | Katty hi there |
| 14:12.30 | Katty | ohai ^_^ |
| 14:12.32 | Katty | hugs on irroot |
| 14:12.39 | irroot | we back in reality even slightly altered :P |
| 14:12.56 | *** part/#asterisk benji (~benji@c-71-62-106-255.hsd1.va.comcast.net) |
| 14:13.09 | irroot | thx :-* |
| 14:13.17 | Katty | snap, back to reality |
| 14:13.20 | Katty | oh there goes gravity |
| 14:13.31 | irroot | bitch when that happens ... |
| 14:13.42 | orn | DanFromUK: Anyway, I think you're looking for s,n,Set(VARIABLE=${CUT(VARIABLE,_,2)}) |
| 14:13.47 | irroot | redefines G |
| 14:14.20 | Katty | grooves with irroot |
| 14:14.59 | Katty | who else is awakes |
| 14:15.58 | irroot | define awake for any value of timezone |
| 14:16.21 | Katty | eyes pried open |
| 14:16.35 | Katty | that's what toothpicks are for ya know |
| 14:16.54 | *** join/#asterisk davlefou (~david@41.225.228.197) |
| 14:17.19 | orn | bad mental image |
| 14:17.27 | Katty | nods |
| 14:17.34 | irroot | conciders going to the german restaurant tonight do they serv e-coli on the side i wonder |
| 14:17.50 | Katty | :< |
| 14:18.00 | Katty | we don't have any german places here. what kind of food do they serve...besides beer |
| 14:18.27 | cneb3000 | sausages? |
| 14:18.47 | irroot | oh have not noticed any thing else :P there snitchels eisbien plenty sausage kraut meatloaf .... |
| 14:18.52 | orn | Schweinehaxe? |
| 14:19.04 | irroot | Shwabingers |
| 14:19.20 | Katty | frankfurters?! |
| 14:19.55 | Katty | i want to try some snitzel |
| 14:20.02 | irroot | http://www.jhblive.com/reviews/random/ich_bin_ein_eisbeiner:_schwabinger_stuben/24774 |
| 14:20.06 | luckman212 | I'm trying to make a ".call" file that will just dial an external# and connect it with the MusicOnHold() application.... is this possible? |
| 14:20.09 | drmessano | Spatzel, Red Cabbage.. mmmm |
| 14:20.23 | Katty | luckman212: yup pretty sure it is |
| 14:20.35 | Katty | luckman212: i've connected them to a call then played an audio file |
| 14:20.35 | luckman212 | Katty: I haven't been able to make it happen |
| 14:20.40 | Katty | luckman212: i don't see any reason why |
| 14:20.45 | Katty | luckman212: i'm sorry to hear that |
| 14:21.30 | luckman212 | Katty: any tips? |
| 14:21.59 | Katty | luckman212: double check your code |
| 14:22.10 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:22.12 | Katty | luckman212: also, tip your waitress |
| 14:22.26 | cneb3000 | luckman122: paste your code on pastebin.com :) |
| 14:23.23 | infernix | anyone familiar with voicexml? looking for a decent framework to work with; looked at some php frameworks but they were outdated |
| 14:27.13 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:29.54 | Katty | never heard of it before. what is it do |
| 14:29.58 | Katty | what's it do |
| 14:30.01 | Katty | grammar fail |
| 14:30.19 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:30.29 | Katty | hai |
| 14:31.17 | luckman212 | cneb3000: I just figured it out, thanks to http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out cheers :) |
| 14:31.28 | Katty | yay |
| 14:35.15 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:35.18 | irroot | off home |
| 14:35.29 | irroot | maybe latter |
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| 14:43.57 | *** part/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
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| 14:48.04 | doolittlework | can realtimesip be configured using res_odbc.conf connecting to a mysql database |
| 14:49.12 | bobb_WU | hey is anyone around to give me a hint? we're looking at a small issue with asterisk/dahdi/ emergency phones and how they go to fast busy when the other side hangs up |
| 14:51.16 | Deeewayne | bobb_WU, what's an emergency phone? |
| 14:51.41 | bobb_WU | we use something that we call a hotline. it auto dials down to our campus police (i work for a univeristy) |
| 14:51.56 | bobb_WU | the emergency phones have a button, loud speaker, and mic |
| 14:52.04 | Deeewayne | automatic ringdown? |
| 14:52.17 | bobb_WU | when you push the button, it connects, then the operator sends a tone to get the phones to hang up |
| 14:52.21 | bobb_WU | yes auto ringdown |
| 14:52.58 | bobb_WU | the reason i ask, a phone yesterday was not disconnected before the op hung up, and it sat there with a busy tone for a while- had to disconnect the cable to turn it off |
| 14:53.07 | bobb_WU | how do i make the line auto-reset in such a case? |
| 14:54.04 | Katty | eyes Deeewayne |
| 14:54.41 | Deeewayne | puts Katty on my shoulders and challenges Qwell to a chicken fight |
| 14:54.42 | Qwell | tickletackles Katty for eyeing Deeewayne |
| 14:54.58 | Katty | >.< |
| 14:55.11 | Katty | hugs Qwell |
| 14:55.52 | Deeewayne | bobb_WU, I haven't used auto ringdown in a while, but I think you'd have to send a command to your hotline phone to go onhook |
| 14:56.35 | Katty | has never heard of auto ringdown before |
| 14:56.50 | Deeewayne | once an auto ringdown phone is offhook it is in the answered state (I think) |
| 14:56.58 | Deeewayne | Katty, maybe 'batphone' ? |
| 14:57.06 | Katty | toooo the batphone!!! |
| 14:57.21 | bobb_WU | well that's the general idea- the ops dials a code that tells the phone to go on-hook |
| 14:57.26 | Deeewayne | I think it was referred to as batphone in one of the books |
| 14:57.27 | Qwell | Deeewayne: That would make a good cell provider name. |
| 14:57.30 | Qwell | BatMobile |
| 14:57.41 | Qwell | (it helps to pronounce Mobile the normal way) |
| 14:57.55 | bobb_WU | but what if they mess up? we have about 4 different styles of phones and 7 operators, so its not always going to be done correctly |
| 14:57.56 | Qwell | sort of the opposite of FileMobile |
| 14:58.08 | Katty | i totally read it the wrong way the first time |
| 14:58.15 | Katty | i think it's cuter the wrong way |
| 14:58.18 | Qwell | Katty: yes |
| 14:58.21 | Deeewayne | bobb_WU, are you making the hardware ? |
| 14:58.24 | Katty | :>> |
| 14:58.24 | Qwell | To the filemobile! |
| 14:58.30 | Katty | dundunDUN! |
| 14:58.36 | Katty | grabs purse, runs |
| 14:58.50 | bobb_WU | what do you mean? i built the servers and installed asterisk/dahdi so yes |
| 14:58.53 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:58.57 | Katty | hi tony |
| 15:03.06 | *** join/#asterisk fragileX (~bill@204.134.45.21) |
| 15:05.12 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-ebbfwlylsajqqpaw) |
| 15:06.00 | Deeewayne | bobb_WU, I thought you might be actually making the phones you are using |
| 15:06.06 | *** join/#asterisk irroot (~irroot@41.51.169.100) |
| 15:06.46 | fragileX | I have an asterisk server that has out going scratchy calls. new to the asterisk voip thing |
| 15:07.37 | bobb_WU | no they are production phones, designed for outdoor use. they even have a blue strobe when they are in use. kinda fancy |
| 15:07.52 | bobb_WU | i tried to append a Hangup() command at the end of my dial plan, but it doesn't seem to work |
| 15:08.24 | irroot | irrback |
| 15:08.31 | Deeewayne | bobb_WU, pastebin your dialplan |
| 15:09.26 | bobb_WU | exten=> _[1-4,6-8]xxx,1,Set(CALLERID(all)=${ODBC_CALLERID(${CHANNEL},${CONTEXT},${databaseName})}) |
| 15:09.26 | bobb_WU | exten=> _[1-4,6-8]xxx,2,Set(CALLERID(ani)=${ODBC_CALLERANI(${CHANNEL},${CONTEXT},${DatabaseName})}) |
| 15:09.26 | bobb_WU | exten=> _[1-4,6-8]xxx,3,NoOP(${CALLERID(ani)}) |
| 15:09.26 | bobb_WU | exten=> _[1-4,6-8]xxx,4,Dial(${ODBC_WILD(${EXTEN},${CONTEXT},${databaseName})}, ${ODBC_RINGTIME(${EXTEN},${CONTEXT},${databaseName})}) |
| 15:09.26 | bobb_WU | exten=> _[1-4,6-8]xxx,5,SipAddHeader(Diversion: <sip:${EXTEN}@10.0.0.9>;reason=no-answer;counter=1) |
| 15:09.28 | bobb_WU | exten=> _[1-4,6-8]xxx,6,Dial(SIP/siprelay/2600) |
| 15:09.32 | bobb_WU | exten=> _[1-4,6-8]xxx,7,Hangup() |
| 15:10.04 | Deeewayne | ~pastebin |
| 15:10.04 | infobot | [~pastebin] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:10.13 | Deeewayne | bobb_WU, you'll get kicked for that |
| 15:10.18 | bobb_WU | gotcha |
| 15:10.31 | bobb_WU | i'm sorta new to the whole irc thing... i see the infobot message though |
| 15:11.49 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:13.55 | Deeewayne | bobb_WU, make a web interface with giant hangup buttons that send hangup actions to AMI |
| 15:14.03 | irroot | anyone seen problems with FXO and polarity reversal that it does not dial out ?? |
| 15:14.27 | irroot | picks up line does not dial |
| 15:14.46 | Deeewayne | irroot, I use 'w' in my Dial string |
| 15:15.07 | irroot | ah so you seen this Deewayne ?? |
| 15:15.46 | irroot | got a report of this and it must be new in 1.8 thx |
| 15:16.12 | Deeewayne | maybe, I started using 'w' a couple of years ago and it solved my problem so I stuck w/ it. That was probably 1.2 |
| 15:16.32 | *** join/#asterisk thalesfp (~thalesfp@139.82.149.61) |
| 15:17.45 | irroot | could be a call progress thing too but there is no proper call progress for #ZA |
| 15:18.47 | bobb_WU | meh, looking for a more elegant solution to that. we don't always know when someone is using the emergency phones, and when they mess up.... |
| 15:18.59 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:20.19 | *** join/#asterisk coppice (~chatzilla@210.17.219.183) |
| 15:23.07 | thalesfp | Hello, Im having voice delay problems to brazil mobile calls, land line calls are perfectly. I using debian, asterisk 1.8 and g729 codec. What do can I do to minimize it? |
| 15:24.01 | ChannelZ | don't use cell phones... |
| 15:24.07 | ChannelZ | ain't much you can do about it |
| 15:24.29 | *** join/#asterisk Geek_Slack (~geekslack@CAcert/Assurer/GeekSlack) |
| 15:25.29 | *** join/#asterisk jkroon (~jkroon@dsl-241-240-125.telkomadsl.co.za) |
| 15:25.50 | thalesfp | ChanServ: but I need to use |
| 15:26.11 | thalesfp | ChanServ: maybe could be my termination? |
| 15:26.55 | thalesfp | ChannelZ: |
| 15:26.58 | thalesfp | maybe could be my termination? |
| 15:27.18 | ChannelZ | Mobile phones have quite a bit of delay. |
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| 15:31.04 | ChannelZ | Unless your ITSP is not in the same country and your calls are being routed all over the place, I'd say most of the delay is the limitation of the cell network. |
| 15:31.48 | thalesfp | ChannelZ: My ITSP is on USA and my server in Brazil. |
| 15:32.17 | bobb_WU | well, we got it to work- using g and goto commands |
| 15:32.37 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-vsdkcxixdffctutg) |
| 15:32.48 | bobb_WU | it loops back and continues to call down until the operator sends the correct hangup command |
| 15:36.40 | *** join/#asterisk cmendes0101 (~nn@pool-173-51-199-161.lsanca.fios.verizon.net) |
| 15:38.48 | cmendes0101 | I've been trying to use chanspy but I can not get it to work correctly. I provide the channel to listen on but am getting this warning. Channel is SIP using ULAW and so is the side that is initiating the chanspy. |
| 15:38.50 | cmendes0101 | WARNING[9901]: chan_sip.c:6083 sip_write: Asked to transmit frame type slin, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 |
| 15:40.51 | *** join/#asterisk dhorner_mb (~dhorner_m@184.18.40.39) |
| 15:45.07 | irroot | cmendes0101 you get this repeating endlessly ?? |
| 15:45.33 | cmendes0101 | irroot: yes, it just keeps outputting and nothing is heard |
| 15:45.44 | irroot | ok had same problem |
| 15:46.26 | irroot | fixed it with a small patch |
| 15:46.40 | cmendes0101 | irroot: patching chanspy module? |
| 15:46.51 | irroot | nope chan_sip |
| 15:48.02 | cmendes0101 | oh ok, what needed to be patched for it or do you have the patch? |
| 15:48.28 | Katty | i think i just pulled a muscle )= |
| 15:49.12 | *** join/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 15:49.23 | irroot | Katty i dare ask how |
| 15:49.29 | Katty | pushups |
| 15:49.31 | irroot | i have it |
| 15:49.46 | Katty | like by my left shoulderblade |
| 15:49.49 | irroot | pushup bra *duck* |
| 15:49.57 | Katty | too much padding |
| 15:50.06 | Katty | too hot for work during summer |
| 15:50.38 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 15:50.45 | irroot | cmendes0101 i haz it but cant figgure out where it is in the system |
| 15:50.51 | irroot | ill post it somewhere |
| 15:51.02 | Katty | locate 'filename |
| 15:51.04 | irroot | indeed K |
| 15:51.32 | irroot | got it trying to see if i put it on mantis/jira/reviewboard |
| 15:51.36 | irroot | its in SVN |
| 15:52.04 | ariel_ | tzafrir_laptop: I have a configuration issue with an Asterisk Bank in mixing E1 and T1 CAS. I would like to see if I can put the right info like pri_protocol/xbus-00/xpd-02 T1 CAS for the 2 ports I need. |
| 15:53.03 | tzafrir_laptop | ariel_, yes, that should generally be OK. |
| 15:53.25 | ariel_ | do I need to change the bus or device number? |
| 15:53.29 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 15:53.30 | tzafrir_laptop | If it doesn't, please pastebin: grep ^pri_protocol /etc/dahdi/xpp.conf; dahdi_hardware -v |
| 15:53.42 | ariel_ | I have the first two as E1 last 2 as T1 CAS |
| 15:54.24 | Katty | ARIEL! |
| 15:54.25 | tzafrir_laptop | that would be: pri_protocol/xbus-00/xpd-0[23] T1 |
| 15:54.27 | Katty | squeals! |
| 15:54.27 | tzafrir_laptop | I believe |
| 15:54.29 | Katty | glomps ariel_ |
| 15:54.41 | irroot | cmendes0101 please open a issue and send me the url ill post patch |
| 15:54.58 | ariel_ | hi Katty |
| 15:55.08 | ariel_ | but I need the frist two ports to be E1's |
| 15:55.41 | *** join/#asterisk bmg505 (~leon@196-209-163-87.dynamic.isadsl.co.za) |
| 15:56.54 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:58.26 | tzafrir_laptop | ariel_, E1 is an implicit default |
| 15:58.33 | ariel_ | http://pastebin.com/37i3Lvc6 |
| 15:58.48 | ariel_ | yes I know but it is only seen the first 2 spans |
| 15:59.03 | ariel_ | I need to get the last 2 setup as well |
| 16:02.12 | cmendes0101 | irroot: Should I open with component being chanspy or chan_sip? |
| 16:02.49 | *** join/#asterisk bmoraca (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:02.54 | irroot | chan_sip format error format != nativeformat incomat channel |
| 16:03.13 | irroot | ive isolated the patch |
| 16:04.13 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 16:04.21 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 16:06.42 | Katty | i do believe it is time to stuff my face |
| 16:06.54 | Katty | g'later. |
| 16:07.04 | irroot | TTFN |
| 16:07.15 | *** part/#asterisk jhmichelle352 (~jhmichell@41.69.251.53) |
| 16:09.29 | *** part/#asterisk dhorner_mb (~dhorner_m@184.18.40.39) |
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| 16:17.27 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 16:26.43 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 16:42.06 | *** join/#asterisk cxreg (count@62.f9.1243.static.theplanet.com) |
| 16:44.17 | irroot | to much bugginess today |
| 16:45.52 | cxreg | does anyone work with the git mirror on gitorious? |
| 16:46.08 | cxreg | trying to figure out how to mingle asterisk and asterisk-tools to make it actually compile properly |
| 16:51.29 | tzafrir_laptop | cxreg, I normally work with them |
| 16:51.32 | tzafrir_laptop | any problems? |
| 16:52.19 | tzafrir_laptop | Generaly path/to/asterisk-tools/setup asterisk |
| 16:52.24 | irroot | deewayne wwww sorted the problem out not a clean solution but usable one |
| 16:52.30 | tzafrir_laptop | (in the checkout of asterisk) |
| 16:54.12 | *** join/#asterisk AB3LS (~AB3LS@pool-96-249-151-45.hrbgpa.fios.verizon.net) |
| 16:55.25 | E-bola | tzafrir_laptop: Would it be better to use the dahdi timer over the pthread one in debian, even if I dont have any dahdi hardware? |
| 16:55.37 | cxreg | tzafrir_laptop: oh ok, i couldnt get asterisk by itself to build due to missing menuselect |
| 16:55.58 | *** join/#asterisk davlefou (~david@41.225.128.121) |
| 16:55.59 | tzafrir_laptop | E-bola, what kernel? |
| 16:56.14 | tzafrir_laptop | cxreg, did you run that setup script? |
| 16:56.19 | cxreg | no, doing it now |
| 16:56.50 | E-bola | tzafrir_laptop: 2.6.32-5-xen-686 |
| 16:57.27 | tzafrir_laptop | I see some bug reports about timerfd not working well on some VMs |
| 16:57.32 | E-bola | Reason I'm asking is that i sometimes have issues with MOH being choppy/degraded when we get calls via sip trunks, while iax trunks seem less affected |
| 16:57.34 | tzafrir_laptop | dahdi would be worth a shot |
| 16:57.45 | E-bola | tzafrir_laptop: ya doesnt work well at all as far as i can see, if at all |
| 16:57.46 | cxreg | tzafrir_laptop: right that was it, thanks~! |
| 16:58.13 | E-bola | tzafrir_laptop: Actualy i didnt think timerfd libs/modules was included in default debian kernels? |
| 16:58.21 | E-bola | I couldnt get it to compile my self anyway |
| 16:58.40 | tzafrir_laptop | E-bola, included, and works, but reported to not work well enough |
| 16:58.49 | tzafrir_laptop | but also try pthreads |
| 16:59.00 | tzafrir_laptop | It's probably simpler |
| 16:59.05 | E-bola | Im running pthreads now |
| 16:59.11 | E-bola | works pretty well |
| 16:59.24 | E-bola | I think i simply put too many extensions on the server |
| 16:59.29 | E-bola | got about 80 |
| 16:59.51 | E-bola | its not like it really should be a problem, but it feels like increasing the amount of extensions have put more strain on "something" |
| 17:00.39 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:02.03 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:02.03 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:10.01 | *** join/#asterisk QuantumSchema (~QuantumSc@rrcs-24-227-92-3.se.biz.rr.com) |
| 17:10.13 | QuantumSchema | Good afternoon all! |
| 17:10.40 | irroot | lo good evening |
| 17:10.55 | QuantumSchema | Quick question... |
| 17:11.15 | ariel_ | how do I tell dahdi not to setup any echocancel ? |
| 17:11.37 | QuantumSchema | When using "autopause" or "autologoff" in a queue config and QUEDEVSTATE is set to True... is there any way to have it actually maniuplate a devstate? |
| 17:21.37 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-gsbaielxdyclnekj) |
| 17:24.02 | QuantumSchema | Bueller? :-) |
| 17:24.04 | *** join/#asterisk cerberus_za (~coert@196-215-125-233.dynamic.isadsl.co.za) |
| 17:24.47 | doolittlework | i quick question regarding realtime, is there any dialplan commands to list all connected sip peers? |
| 17:25.04 | Katty | woo |
| 17:25.05 | Katty | it's hot out |
| 17:25.36 | doolittlework | where you from Katty here we are feezing our buts off |
| 17:25.50 | _Corey_ | doolittlework: If you have 'rtcachefriends' enabled, 'sip show peers' will work |
| 17:26.24 | doolittlework | _Corey_: is that in the sip.conf? |
| 17:26.35 | _Corey_ | yeah |
| 17:26.43 | doolittlework | thank you |
| 17:26.48 | _Corey_ | np |
| 17:27.06 | Katty | well if your'e freezing your tails off i would guess i am in the opposite hemisphere |
| 17:27.13 | Katty | it's a good, solid, logical guesstimate |
| 17:27.25 | Katty | cause tilt is like that, yo |
| 17:29.49 | Deeewayne | offers Katty a lemon Italian Ice |
| 17:30.26 | doolittlework | must be hell |
| 17:30.30 | Katty | WOOt |
| 17:30.39 | Katty | accepts, inhales |
| 17:30.50 | Katty | Deeewayne: how's de weather out there todays |
| 17:31.20 | doolittlework | where can i find more info regarding realtimesip? |
| 17:31.55 | Deeewayne | Katty, I work from home and only stepped outside for 2 minutes to walk around my driveway, but weather.com says it feels like 93 |
| 17:32.32 | irroot | is freezing winter is not cool |
| 17:34.02 | Katty | icky :< |
| 17:34.06 | Katty | irroot: i wanna come visit |
| 17:34.12 | Katty | irroot: do you have a guest room? |
| 17:34.23 | Katty | irroot: and wireless internets |
| 17:34.26 | irroot | i doo indeed |
| 17:34.32 | Katty | :>> |
| 17:34.36 | irroot | can do wifi |
| 17:34.36 | Katty | BRT BOOKING PLANE |
| 17:34.56 | irroot | cool |
| 17:35.08 | Katty | i wonder how much the ticket would cost |
| 17:35.11 | irroot | be aware it gets freaking cold |
| 17:35.14 | Katty | irroot: /query me with airport code |
| 17:36.45 | irroot | goes to -4c overnight |
| 17:36.54 | Deeewayne | Boulder CO appears to be a comfy 77 |
| 17:36.55 | Katty | what is that in F |
| 17:37.07 | Katty | dayum |
| 17:37.20 | Katty | cheapest ticket is $2,393.40 |
| 17:37.33 | Katty | 21hr 21min flight |
| 17:38.04 | Katty | Deeewayne: jealous |
| 17:38.29 | irroot | 24.8 |
| 17:38.29 | coppice | no commercial flight is 21:21. that take in flight refueling |
| 17:38.51 | Kobaz | boulder is always a comfy 77 |
| 17:38.54 | irroot | 24.8F |
| 17:39.15 | Katty | 12:39 pm Depart St. Louis (STL) 12:39 pm Depart St. Louis (STL) 12:39 pm Depart St. Louis (STL) |
| 17:39.17 | Kobaz | i lived in golden, co for several months |
| 17:39.19 | Katty | irroot: oh that's nothin |
| 17:39.26 | Kobaz | i was outside in shorts and a tshirt on new years day |
| 17:39.29 | irroot | but most days its 10c 50F in winter |
| 17:39.30 | coppice | its a comfy 23 here, but only because the air con is working hard |
| 17:39.35 | Katty | try like -30 or -40F and you've got missouri weather |
| 17:39.38 | Katty | in winter |
| 17:40.02 | irroot | lol ok |
| 17:40.19 | irroot | but we dont do double glazing or underfloor heating |
| 17:40.27 | coppice | Katty: how come? missouri isn't in the north |
| 17:40.30 | Katty | what's double glazing? |
| 17:40.30 | Deeewayne | Kobaz, I'll be in Boulder and Conifer next week |
| 17:40.37 | Katty | coppice: how come what? |
| 17:40.45 | irroot | heat proofing windows |
| 17:40.54 | coppice | Katty -40F in missouri |
| 17:41.19 | coppice | double glazing is good in hot climates too |
| 17:41.41 | Katty | coppice: i'm not an expert in that dept |
| 17:42.35 | Kobaz | mmm |
| 17:46.10 | QuantumSchema | Does anyone know the C function for setting the device state? I'm trying to set a device state during the Ring No Answer portion.... |
| 17:46.24 | QuantumSchema | portion of app_queue.c... |
| 17:46.30 | irroot | yeah |
| 17:46.38 | irroot | was working with it today :P |
| 17:46.42 | QuantumSchema | LoL |
| 17:46.43 | irroot | see RB 1119 |
| 17:47.21 | irroot | https://reviewboard.asterisk.org/r/1119/diff/#index_header |
| 17:47.53 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-249-008.pools.arcor-ip.net) |
| 17:50.34 | QuantumSchema | hmmmmm |
| 17:51.02 | Katty | hmmer. |
| 17:51.47 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-069.mobile.uci.edu) |
| 17:52.00 | *** join/#asterisk irroot (~irroot@197.171.157.97) |
| 17:53.10 | irroot | QuantumSchema newstate = ast_parse_device_state(tmp->member->interface); |
| 17:53.14 | irroot | update_status(qe->parent, tmp->member, newstate); |
| 17:53.29 | irroot | why ?? |
| 17:53.39 | QuantumSchema | Thanks! |
| 17:53.42 | QuantumSchema | Well.... |
| 17:53.53 | QuantumSchema | app_queue lets you set "autopause". |
| 17:54.31 | irroot | yes PUNT my patch adds a delay after last call before autopause kicks in ... |
| 17:55.00 | QuantumSchema | But... if you're using anything in a dialplan that maniuplates hints or AstDB for logic (e.g. BLF), and when the autopause happens, nothing is modified. |
| 17:55.50 | irroot | mmm |
| 17:56.06 | irroot | im intrested |
| 17:56.20 | irroot | i know my "ignorebusy" has problems |
| 17:56.38 | irroot | and i use the code above to check/set it |
| 17:57.05 | *** join/#asterisk mechbangirc (~mechbangi@115-186-140-80.nayatel.pk) |
| 17:57.57 | mechbangirc | hi can I forward two different external ports to single internal port 5060 in my router which then points to my asterisk behind nat server |
| 17:58.03 | irroot | where does this creep into app_queue |
| 17:58.11 | QuantumSchema | Sorry...got pulled away... |
| 17:58.21 | QuantumSchema | So yeah... |
| 17:58.38 | QuantumSchema | I'm doing alot of stuff with logic based off of hints or AstDB... |
| 17:59.08 | QuantumSchema | So I could use something like #1 to (un)pause for users and based on an AstDB value and have the dialplan set it. |
| 17:59.31 | QuantumSchema | But when autopause happens, it does nothing but pauses the user. |
| 18:00.01 | irroot | i have extended QUEUE_MEMBER to (queue,paused,interface) R/W |
| 18:00.38 | irroot | SET(QUEUE_MEMBER(1234,paused,SIP/100)=0/1) |
| 18:00.39 | QuantumSchema | So what I'm thinking is somwhere in line 3387 to 3406 there's something that sets an AstDB or Hint.... |
| 18:01.39 | QuantumSchema | line 3387 to 3406 in the 1.8.4 app_queue.c |
| 18:02.31 | QuantumSchema | Actually.... not an AstDB but a hint for my case. |
| 18:03.37 | irroot | in the rna function |
| 18:03.53 | QuantumSchema | Yup! |
| 18:04.41 | irroot | could be good |
| 18:05.03 | QuantumSchema | Thank ya! That's what I was thinking. |
| 18:05.09 | irroot | put the delay code there |
| 18:05.20 | QuantumSchema | What lines were those? |
| 18:05.33 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 18:05.58 | irroot | its in pending for trunk |
| 18:08.28 | QuantumSchema | Is there a way to use this (newstate = ast_parse_device_state(tmp->member->interface);) on a custom device state? |
| 18:11.28 | Katty | wonder if the movie Gamer is any good |
| 18:12.15 | Qwell | Katty: 5/10 |
| 18:12.25 | dr0ck | http://i.qkme.me/2fhd.jpg |
| 18:12.44 | Katty | Qwell: why only 5 of 10? |
| 18:12.54 | Katty | Qwell: bad story line? bad acting? |
| 18:13.02 | Qwell | not my rating |
| 18:13.24 | Katty | oh |
| 18:13.26 | Katty | k'then |
| 18:17.53 | *** join/#asterisk sourcode (~code@ppp-115-87-207-206.revip4.asianet.co.th) |
| 18:20.22 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:22.51 | QuantumSchema | Ahh... did irrot disappear? |
| 18:22.58 | QuantumSchema | :-( We were brainstorming! |
| 18:24.01 | pigpen | Question: Is there a way to have asterisk launch an event (email or such) when a caller hangs up without being connected (answered), yet, if they are connected, there is no event? |
| 18:24.52 | pigpen | Currently, we have calls falling into a queue, but since this is a "sales" type operation, they want to have cid info to call the person back. |
| 18:24.54 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 18:25.11 | pigpen | If there is no way, then we will be looking to have it developed. |
| 18:26.08 | QuantumSchema | I'm sure there could be a way to launch an agi script right before the hang up in the dialplan. |
| 18:26.41 | pigpen | but how can it anticipate a hang up? |
| 18:27.02 | pigpen | For that matter, it would have be conditional if the call was bridged. |
| 18:27.09 | pigpen | ie: to a sip device |
| 18:27.27 | pigpen | I know there are hangup events, but that is all hangups. |
| 18:27.37 | pigpen | not just the ones that are not bridged. |
| 18:27.45 | QuantumSchema | ahh...got me there. |
| 18:27.53 | QuantumSchema | why would the call be bridged? |
| 18:28.09 | pigpen | answered, bridged, etc... |
| 18:28.25 | pigpen | ie: pstn bridged to sip |
| 18:28.51 | pigpen | so if call from a sip phone to another sip phone, they are bridged. |
| 18:29.36 | QuantumSchema | I'm terribly confused then. |
| 18:29.43 | QuantumSchema | Would the call become bridged before it's answered then? |
| 18:30.00 | QuantumSchema | And would sip to sip happen outside the organization? |
| 18:31.06 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 18:31.19 | pigpen | if the call is not answered, it is not bridged. |
| 18:31.28 | pigpen | if the call is answered, it is bridged. |
| 18:31.55 | pigpen | Note: I think the term "bridged" is correct. someone correct me if I am using it out of context. |
| 18:32.08 | pigpen | sip outside the org? No. |
| 18:32.27 | QuantumSchema | So you're wanting to capture the CID of inbound calls that are not answered? |
| 18:32.56 | pigpen | exact: pstn (analog) --- Audiocodes-----(SIP)------Asterisk-------(SIP)------SIP Device |
| 18:33.12 | pigpen | QuantumSchema, yes, and have an action for them. |
| 18:33.35 | pigpen | There is a hangup event, but I believe it is for all hangup events. |
| 18:33.46 | pigpen | even the ones that were answered. |
| 18:34.23 | QuantumSchema | Ahh... |
| 18:34.45 | pigpen | yeah, it is a little goofy. |
| 18:34.58 | pigpen | To add icing: I need this for 380 locations. |
| 18:34.59 | QuantumSchema | Well, I just tested via a PSTN to SIP call w/ the PSTN phone hanging up and you do get a hangup to work with. |
| 18:35.26 | QuantumSchema | But you're right... the hangup happens to all calls.... |
| 18:35.40 | pigpen | yeah, we may have to just setup a script that runs on all hangup events, and have it determine if the call was bridged or not. |
| 18:35.45 | pigpen | Dam sales people. |
| 18:37.58 | pigpen | I think it was the "h" extension |
| 18:41.50 | pigpen | I may be able to use the h exten, grab the channel info, launch a script to grep out that channel in the cdr, see if it was bridged, if not, grab the cid info and email it. |
| 18:41.54 | pigpen | mabye. |
| 18:42.03 | pigpen | s/mabye/maybe |
| 18:43.44 | *** join/#asterisk SuperNull (~SuperNull@24-148-101-238.ip.mhcable.com) |
| 18:44.36 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 18:46.53 | SuperNull | Hey guys, what versions of asterisk support dumping of msgXXX.txt into the voicemail dir to generate a VM Notify ? im using 1.2.7 and it doesn't appear to be doing it.. |
| 18:49.30 | _Corey_ | pigpen: Why don't you grab one of the channel variables to tell you what happened with the call? You can use DIALSTATUS or the queue equivalent |
| 18:50.40 | *** join/#asterisk davlefou (~david@41.225.114.186) |
| 18:51.08 | _Corey_ | if you have the 'setinterfacevar=yes' and 'setqueuevar=yes' you will have some good info |
| 18:53.47 | pigpen | yeah, acutally using the channel info, I can grab the call via the cdr. |
| 18:54.07 | pigpen | I should be able to tell if it were connected as the other channel would be something stupid like "s" |
| 18:54.43 | Qwell | SuperNull: 1.2.7? Do you have any idea how many security vulns that version contains? |
| 18:55.02 | pigpen | yeah, that got your attention. |
| 18:55.44 | russellb | there were 890 commits to asterisk 1.2 after that version :-) |
| 18:55.46 | _Corey_ | coming out of the queue or via dial? |
| 18:57.48 | pigpen | well, don't know yet. |
| 18:58.42 | SuperNull | uhh |
| 18:58.47 | SuperNull | yeah. |
| 18:58.48 | SuperNull | well |
| 18:59.07 | SuperNull | we have a slight problem with a customer who wont upgrade his old ass hardware that for what ever reason only works with 1.2 |
| 19:10.06 | _Corey_ | pigpen: I would just cover both and examine the DIALSTATUS first, then if it doesn't exist go through the queue ones... obviously this can be done a lot of different ways |
| 19:10.47 | pigpen | Yeah, you are probably right. |
| 19:11.08 | pigpen | I am trying to see what shows up in the cdr for unanswered calls in the queue right now. |
| 19:11.43 | pigpen | An option like "SalesPeopleAreAnal=yes" would be a nice setting |
| 19:11.58 | *** join/#asterisk [netman] (~netman@246.Red-83-40-0.dynamicIP.rima-tde.net) |
| 19:12.01 | _Corey_ | lol |
| 19:12.23 | _Corey_ | every once in a while people ask me to do the whole "call back the person" like OrderlyQ does |
| 19:12.37 | _Corey_ | that's similarly painful but in a different way |
| 19:12.44 | pigpen | yeah |
| 19:13.07 | pigpen | this one lady -has- to have any calls not answered be sent to her so she can call them back. |
| 19:13.26 | pigpen | I guess it is better to have over zealous employees than lazy ones like at Walmart. |
| 19:13.49 | _Corey_ | i'm usually happy to help those actually trying to do their jobs |
| 19:14.49 | *** join/#asterisk [netman] (~netman@246.Red-83-40-0.dynamicIP.rima-tde.net) |
| 19:15.23 | pigpen | Ok, the queue exits the cdr as "BUSY" |
| 19:15.49 | *** join/#asterisk [netman] (~netman@246.Red-83-40-0.dynamicIP.rima-tde.net) |
| 19:15.55 | pigpen | now to marry that to dialstatus |
| 19:16.11 | pigpen | and a good stiff cup of coffee |
| 19:16.59 | *** join/#asterisk [netman] (~netman@246.Red-83-40-0.dynamicIP.rima-tde.net) |
| 19:20.32 | pdtpatrick__ | Question for u smart folks .. im on one asterisk box but trying to use a context from another box.. do i do switch +> DUNDi/jn and then do exten => _17024445241,n,Dial(Local/context,1) ? |
| 19:20.50 | russellb | you don't need DUNDi for that necessarily |
| 19:21.21 | russellb | you can just Dial() it over SIP or IAX2 a ndthat'll be fine |
| 19:21.36 | russellb | if you want asterisk to do extension lookups before it runs anything, you could use the IAX2 switch in the dialplan |
| 19:21.43 | Katty | hi |
| 19:21.49 | pdtpatrick__ | what would u suggest? currently i have something like this exten +> _8882226665,n,Dial(Local/66700@dundi-jn/n) and this works perfectly |
| 19:22.19 | pdtpatrick__ | russellb can u please give me an example |
| 19:22.25 | russellb | oh, well if you're already using DUNDi, then yeah, a DUNDi switch will do it |
| 19:23.49 | pdtpatrick__ | so would i then be able to get the context on the other box doing ? exten => _8882226665,n,Dial(Local/context@dundi-jn/n) |
| 19:32.29 | doolittlework | what does the qualify=yes do in sip.conf? |
| 19:32.54 | doolittlework | i can only see my realtimesip extensions if i have it set to yes |
| 19:34.34 | *** join/#asterisk radic (~radic@dslb-094-216-228-150.pools.arcor-ip.net) |
| 19:35.22 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-aoljaczothncxreu) |
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| 20:33.26 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:33.26 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4.2 (2011/06/04), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 20:34.41 | Katty | stares at clock |
| 20:34.52 | Katty | i can't make it another 1.5hrs |
| 20:34.58 | Katty | CAN"T DO IT |
| 20:34.59 | Katty | cries |
| 20:47.00 | *** join/#asterisk oliver1 (~oliver@manz-590efdfc.pool.mediaWays.net) |
| 20:48.54 | *** join/#asterisk Guizmo (~Guizmo@adsl-178-38-150-243.adslplus.ch) |
| 20:50.28 | *** join/#asterisk jc318 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 20:50.31 | jc318 | hi |
| 20:50.56 | jc318 | I need some help with setting up my first Asterisk server, anyone available? |
| 20:51.25 | Katty | is available |
| 20:51.42 | Kobaz | Katty: coming to astricon this year? |
| 20:51.49 | Katty | probably not |
| 20:51.55 | Guizmo | Hello.I bought a digium isdn bri card (B410P) for an asterisk server. I wanted to connect a NT Box on it (NT1+2ab), and connect only the S port on one of the port of the card (in TE mode). Is it correct to do it like this ? |
| 20:52.49 | Kobaz | aw |
| 20:57.00 | jc318 | Basically I have a broadband connection, Asterisk 1.8.3.2 on HP t5710 thin client, running CentOS 4.x (will update later), 7940 & 7960 hardphones, a sip account with 3 DiD numbers some initial calling credits in account. I am trying to set up a home office, with 2x deskphones on two different desks, each have their DiD and can call internal/external unlimited. This is a new house, when I did |
| 20:57.00 | jc318 | not have broadband, in the offline period I read free Asterisk book 2 times in the last 4 months and set up several systems to try (trixbox,pifa etc). now I want to go with basic asterisk without any gui to learn better |
| 20:57.38 | WiretapWork | Guizmo, normally you'd connect the SELV interface of the NTU to a TE interface on the card |
| 20:59.57 | jc318 | the only problem is I am confused about the design. should the phones register to the ITSP or the local asterisk server and why? I have read a tutorial and the guy explains there is no reason to register to internal server as voicemail is in ITSP side and talks about using different ports for each phone so that (for example emergency services) can call back directly to the unique device if |
| 20:59.57 | jc318 | needed. is this what I should do or should I set up phones register to internal server and that registers with the ITSP? (also I have a draytek router and I can forward ports but not as flexible as linux based firewall) would this system work with only port tcp/udp 5060 redirected from router to asterisk server? (is there a way to use rtp config to use it in say 'passive mode' or is it |
| 20:59.57 | jc318 | mandatory to open some rtp ports)? |
| 21:00.44 | Guizmo | WiretapWork: on the NT box, there is: ab1, ab2, 2 S ports, U port. If I read the manual correctly it is the S port ? is it the SELV you are talking about ? |
| 21:01.02 | WiretapWork | Guizmo, that is not standardised, you would need to check with your provider |
| 21:01.37 | Guizmo | ah ok. I will check with them, I saw they have a line for pbx. thanks |
| 21:04.10 | p3nguin | jc318: Most people who use Asterisk allow Asterisk to do all the work. There is a single virtual circuit between the ITSP and the Asterisk system where all calls pass. Voicemail would be handled on your Asterisk system and all phones will register to Asterisk. |
| 21:04.15 | Katty | hi p3nguin |
| 21:04.31 | p3nguin | Hello, katty. How's it going today? |
| 21:04.39 | Katty | it's goin. hows things out your way |
| 21:05.00 | p3nguin | Despite the 100 degree temp today, I am still alive. |
| 21:05.08 | Katty | :>>> |
| 21:05.10 | Katty | most excellent |
| 21:05.21 | WiretapWork | jc318, no offense bro, but you read far too much :P |
| 21:06.24 | jc318 | WiretapWork: yeah seems so. |
| 21:06.57 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 21:10.17 | jc318 | I read NAT is not ideal but I do not have resources to provide direct access to Asterisk at the moment. If I 'port redirect' tcp/udp5060 to single NIC internal Asterisk box's (internal) IP address, and do the same for RTP ports, how bad would that be? can I call outbound and receive calls inbound? |
| 21:11.32 | WiretapWork | ?nat |
| 21:11.37 | WiretapWork | err |
| 21:11.37 | WiretapWork | !nat |
| 21:11.41 | WiretapWork | ~nat |
| 21:11.41 | infobot | [nat] Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 21:11.48 | WiretapWork | jesus I'm off my game this morning |
| 21:12.36 | jc318 | yes I am reading at the moment it seems like it can work, I just wanted to ask why is there so many NAT=problem posts. Seems straightforward (so far) |
| 21:13.25 | p3nguin | jc318: You'll need to forward UDP 5060 and the UDP port range as configured in rtp.conf (usually 10000-20000). I personally run multiple Asterisk systems behind NATs and use ITSPs on the public Internet. |
| 21:13.27 | WiretapWork | because there are plenty of idiots using NAT |
| 21:19.20 | *** join/#asterisk timholum1 (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 21:21.51 | jc318 | p3nguin: I have configured localnet and externaddr parameters, redirected tcp/udp5060 and UDP10k-20k to Asterisk's internal address. What should I choose in nat= parameter? (no/force_rport/yes/comedia) |
| 21:25.32 | ChannelZ | yes |
| 21:25.35 | *** join/#asterisk robertj (~Rob@dumbledore.athenshousing.org) |
| 21:30.47 | p3nguin | jc318: If your Asterisk is behind NAT, you need to set nat=yes in the general sip.conf section. |
| 21:31.45 | p3nguin | jc318: You'll also want to set directmedia=no to always keep Asterisk in the media path. |
| 21:33.20 | WiretapWork | p3nguin, if nat is turned on you can generally do without directmedia and let your localnet stuff handle itself |
| 21:34.18 | *** part/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 21:34.26 | p3nguin | jc318: Also, if you have a dynamic public IP address, you should consider using externhost and a dynamic DNS hostname so you don't have to update externaddr all the time. |
| 21:34.46 | p3nguin | If it's static, then externaddr is fine. |
| 21:35.16 | p3nguin | And stop forwarding TCP 5060 unless you have a good reason to foward it. |
| 21:35.47 | jc318 | I have a static IP I was planning to run some services when I ordered this line |
| 21:36.20 | jc318 | thanks for all the help. I am hoping to make my first external call tonight |
| 21:39.50 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 21:45.37 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 21:47.47 | timholum1 | does anyone know of a good tutorial on setting up phoneprov in asterisk 1.8 ( I am using polycom phones and would like them to be configured automaticly by my phone system |
| 22:01.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:07.33 | *** join/#asterisk Syrex (~syrex@dsl-146-17-172.telkomadsl.co.za) |
| 22:12.00 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 22:15.46 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 22:18.40 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 22:49.48 | sunfone | Anyone using LXC on Ubuntu to host asterisk instances? |
| 22:53.22 | *** join/#asterisk cardy (~cardy@host65-16-97-138.birch.net) |
| 22:57.56 | cardy | Hi guys, I'm real new to Asterisk - I am trying to set up a Sip trunk to VOIP Innovations with Debug turned on I can see the call coming in but I don't know how to get any further - could some one point me in the right direction. How do I read the debug info? |
| 22:59.11 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 23:04.11 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 23:10.07 | paulc | cardy: want to pastebin your console output and I'll take a look? |
| 23:13.32 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 23:17.38 | cardy | Thank you paul |
| 23:17.38 | cardy | SIP/2.0 200 OK |
| 23:17.39 | cardy | Via: SIP/2.0/UDP 64.136.174.30:5060;branch=z9hG4bK1fdb41b7;rport=5060;received=192.168.2.220 |
| 23:17.39 | cardy | Contact: <sip:192.168.32.24:60707> |
| 23:17.39 | cardy | To: <sip:1004@192.168.32.24:60707;rinstance=39969020f0fba488>;tag=9e32195e |
| 23:17.39 | cardy | From: "Unknown"<sip:Unknown@64.136.174.30>;tag=as40b4c0cf |
| 23:17.39 | cardy | Call-ID: 469659126900a3bb1dd443ff5dd557c1@64.136.174.30 |
| 23:17.40 | cardy | CSeq: 102 OPTIONS |
| 23:17.40 | cardy | Accept: application/sdp |
| 23:17.41 | cardy | Accept-Language: en |
| 23:17.41 | cardy | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO |
| 23:17.42 | cardy | Supported: replaces |
| 23:17.42 | cardy | Allow-Events: presence, message-summary, tunnel-info |
| 23:17.43 | cardy | Content-Length: 0 |
| 23:18.11 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 23:18.30 | WiretapWork | ~pastebin @ cardy |
| 23:18.37 | WiretapWork | ~pastebin |
| 23:18.37 | infobot | [~pastebin] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 23:18.46 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 23:18.59 | cardy | Oh dear OK sorry i am new to IRC also |
| 23:19.40 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 23:21.41 | cardy | I see pasted now at pastebin.com Cardy |
| 23:23.31 | WiretapWork | cardy, you have ot paste the pastebin link back here |
| 23:24.02 | cardy | http://pastebin.com/ZkYrGjvm |
| 23:24.13 | cardy | Ahh OK |
| 23:25.47 | WiretapWork | cardy, what is your problem though, what you've pasted is a single 200 OK |
| 23:25.57 | WiretapWork | not the entire call, with verbose turned to 7 |
| 23:26.01 | WiretapWork | core set verbose 7 |
| 23:27.06 | cardy | I'm sorry - I am about 3 days into teaching myself Asterisk - let me see if i can figure out how to do that |
| 23:28.56 | WiretapWork | well, you managed to turn debug on |
| 23:29.26 | *** join/#asterisk Dobah (~chatzilla@cpe-98-150-177-51.hawaii.res.rr.com) |
| 23:29.57 | cardy | is it asterisk -vvvvvv -g -dddddd -c |
| 23:30.01 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 23:31.55 | WiretapWork | cardy, are you running asterisk as a daemon? |
| 23:32.23 | WiretapWork | normally it would be 'asterisk -r' |
| 23:32.28 | WiretapWork | followed by 'core set verbose 7' |
| 23:32.33 | WiretapWork | if it were running as daemon |
| 23:35.33 | cardy | http://pastebin.com/5d17rXDR |
| 23:35.37 | cardy | Hows that? |
| 23:36.32 | p3nguin | Just f.y.i.: Unless something has changed in the last few months, verbosity does not increase once you reach level 4, so "core set verbose 4" and "core set verbose 400000" would produce the exact same output. |
| 23:38.13 | p3nguin | That paste shows a sip debug, which is unrelated to core verbosity. |
| 23:41.03 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:41.16 | WiretapWork | cardy, SIP debug is for figuring out what happens when peers talk to each other, not for figuring out what happens once a call has been accepted by * |
| 23:41.34 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 23:41.38 | cardy | http://pastebin.com/3sSvddMJ |
| 23:41.49 | WiretapWork | and all I see is a SIP Ping in that second pastebin too |
| 23:41.59 | cardy | Ok what about that one |
| 23:42.20 | WiretapWork | we need a verbose |
| 23:42.21 | WiretapWork | Looking for s in from-sip-external (domain 192.168.2.220) |
| 23:42.31 | WiretapWork | is the only line of any relevance to the debuggin process |
| 23:42.35 | WiretapWork | and by itself, it is worth nothing |
| 23:43.02 | cardy | Sorry like I said this is all pretty new to me i am a Shoretel guy and Comdail too. |
| 23:43.20 | cardy | So what do I need to do to get more info |
| 23:43.28 | p3nguin | If you started Asterisk using -vvvvvv like you suggested you did, you should be seeing SOMETHING. |
| 23:43.45 | cardy | Let me try again |
| 23:43.48 | cardy | one sec |
| 23:43.53 | p3nguin | "core set verbose 4" if you are already on the CLI. |
| 23:44.12 | p3nguin | If it was 0, it should report that it was 0 and is not at least 4. |
| 23:44.19 | p3nguin | s/not/now/ |
| 23:45.16 | cardy | shows that it was 7 is now 4!! |
| 23:45.42 | p3nguin | I would have expected to see some type of output. Maybe logger.conf needs adjustment? |
| 23:46.03 | p3nguin | I can't remember if the logger settings affect the verbose output or not. |
| 23:46.43 | WiretapWork | cardy, do you actually have anything in from-sip-external to accept calls? |
| 23:47.02 | cardy | Ill paste its contents ?? would that help? |
| 23:47.10 | p3nguin | You can find that out by "dialplan show from-sip-external" |
| 23:47.17 | cardy | maybe not |
| 23:47.38 | p3nguin | If you have nothing, make sure you have run "dialplan reload" after saving any changes to extensions.conf. |
| 23:47.38 | cardy | where do I find that? |
| 23:47.46 | cardy | Ok |
| 23:49.19 | p3nguin | If you forgot to run dialplan reload, then the dialplan show command above will surely clue you in. |
| 23:49.22 | WiretapWork | wonders if cardy should start with a gui-based installation to get a working system and then pick at it from there |
| 23:50.01 | cardy | http://pastebin.com/Y7mK949s |
| 23:50.08 | p3nguin | No, cardy should not. |
| 23:50.08 | p3nguin | No one should, actually. |
| 23:50.19 | jc318 | Q: are there any "ultra basic SIP trunk example configuration" file pack somewhere? there are millions on google but many of them seem to be outdated (several parameters get rejected by Asterisk 1.8) |
| 23:50.30 | cardy | I have FreePBX |
| 23:51.28 | WiretapWork | cardy, hang on, this is a FreePBX setup? |
| 23:51.36 | WiretapWork | oh god, yes it is |
| 23:51.45 | WiretapWork | and you're starting asterisk from the CLI and hand-editing configs? |
| 23:51.56 | WiretapWork | consider your freepbx setup broken |
| 23:52.04 | WiretapWork | and you should be directing it to #freepbx |
| 23:52.05 | WiretapWork | not here |
| 23:52.18 | cardy | O dear |
| 23:52.32 | WiretapWork | freepbx is designed to handle _all_ configuration with no finger fiddling by the user |
| 23:52.40 | cardy | but i have not been doing any modifying of file from the cli |
| 23:53.09 | WiretapWork | also, if your calls are hitting from-sip-external, you have your trunk set up wrong, as theyre being treated as anonymous calls |
| 23:54.21 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:54.25 | cardy | Ok - so what do i need to do? |
| 23:54.31 | WiretapWork | fix your trunk setup |
| 23:54.48 | WiretapWork | and I'd recommend setting asterisk up properly so that it starts on boot |
| 23:55.12 | WiretapWork | you may want to use one of the freepbx distros, however.... this is not the channel for this... time to move to #freepbx |
| 23:56.05 | p3nguin | And you probably didn't break anything by hand-editing configs, because FreePBX usually puts them back the way they were the very next time you hit the bit Apply button from the web interface. |
| 23:56.19 | p3nguin | s/bit/big/ |
| 23:56.28 | drmessano | Unless you edited SIP.conf or Extensions.conf |
| 23:56.36 | WiretapWork | or manager.conf |
| 23:56.36 | p3nguin | They don't go back? |
| 23:56.39 | drmessano | no |
| 23:56.41 | WiretapWork | p3nguin, no |
| 23:56.43 | cardy | no i see not to edit those |
| 23:56.46 | drmessano | the base files do not get overwritten |
| 23:56.51 | drmessano | only the includes |
| 23:56.52 | WiretapWork | many of the config files are not overwritten just added to |
| 23:57.03 | cardy | I didnt change anything -- http://knowledgebase.voipinnovations.com/index.php?/vikb/Knowledgebase/Article/View/160/20/what-settings-do-i-need-for-asterisktrixboxfreepbx-to-use-voip-innovationss-wholesale-service |
| 23:57.13 | p3nguin | That's good to know in the event that I ever care to support FreePBX. |
| 23:57.24 | cardy | that was the set info from the carrier - it's all i had to work wiith |
| 23:57.44 | WiretapWork | cardy, I think you're missing my point... this is the wrong channel for freepbx support |
| 23:57.46 | cardy | I did manage to set up a sip trunk between the freepbx and my shoretel system though |
| 23:57.50 | p3nguin | You do not have the permissions required to view this article, or the article does not exist. |
| 23:58.43 | cardy | No well thanks very much for all your time - I am sorry I didnt know I was in the wrong channel |
| 23:59.08 | cardy | Cheerio then |
| 23:59.19 | p3nguin | It's too bad we don't have enough room in our channel topic to mention that. |
| 23:59.54 | *** part/#asterisk cardy (~cardy@host65-16-97-138.birch.net) |
