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| 00:34.22 | tacloban | any ideas on why my greeting.wav file is not being played when voicemail is started for inbound calls? |
| 00:34.42 | tacloban | all i get is the default lady's voice |
| 00:36.07 | tacloban | i've tried recording messags using the voicemail menu |
| 00:36.14 | tacloban | using x-lite softphone |
| 00:38.43 | tacloban | my extension.conf has the following line: |
| 00:38.46 | tacloban | exten => s,n,VoiceMail(100@VoicemailContext) |
| 00:39.06 | tacloban | the caller can leave messages just fine, btw |
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| 00:44.50 | carrar | try exten => s,n,VoiceMail(100@VoicemailContext,u) |
| 00:45.02 | tacloban | let me give it a try |
| 00:48.03 | tacloban | wow, that worked, thanks |
| 00:55.27 | carrar | Amazing what can be done by reading |
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| 01:25.11 | xSmurf | is it me or if I want to externally delete/move a message from the voicemail spool I need to rename all the other files in that spool |
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| 01:29.55 | carrar | just you |
| 01:30.09 | carrar | and yes |
| 01:30.21 | carrar | you need to do that |
| 01:30.52 | carrar | might look at the included scripts with the source for help on that |
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| 02:23.26 | luke-jr | [Jan 3 14:51:48] ERROR[23450]: chan_sip.c:3917 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 02:23.34 | luke-jr | what does this error actually mean? |
| 02:23.45 | jimi_ | serious network trouble |
| 02:28.11 | luke-jr | ⦠|
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| 02:41.53 | shapr | luke-jr: I'd guess it means lots of packet loss? |
| 02:42.14 | luke-jr | nope, tcpdump doesn't even show it going on the wire |
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| 04:09.21 | b14ck | Hi all. Anyone know of a guide to getting shoutcast streaming working for music on hold with asterisk 1.6? |
| 04:10.02 | b14ck | I've only been able to find a really old article that doesn't look accurate. |
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| 07:12.42 | nicola_pav | hello. upon recording calls on demand |
| 07:13.03 | nicola_pav | files are saved as: auto-xxxxxxx-xxx-xxxxxxx.wav |
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| 07:13.20 | nicola_pav | any hint what file can i edit ti change this kind of format? |
| 07:15.36 | kaldemar | nicola_pav: dialplan variable TOUCH_MONITOR_FORMAT |
| 07:16.59 | nicola_pav | kaldemar: i only find TOUCH_MONITOR variable |
| 07:18.30 | nicola_pav | kaldemar: when i set the parameters to record always, i get another format: OUTxxxx-xxx-xxx.WAV |
| 07:19.12 | kaldemar | what version are you on? |
| 07:19.57 | nicola_pav | 1.4.36 |
| 07:21.47 | kaldemar | there's TOUCH_MONITOR_FORMAT in your version too. |
| 07:22.07 | kaldemar | what do you mean by "when i set the parameters to record always"? |
| 07:24.25 | nicola_pav | in hte freepbx on the extensions configuration. i set the record to always |
| 07:25.03 | nicola_pav | when i set it to on demand and press *1 during the call the format saved is auto-xxxxxxx-xxx-xxxxxxx.wav |
| 07:25.13 | nicola_pav | i kind need to change it |
| 07:25.17 | nicola_pav | i don't know how |
| 07:26.12 | kaldemar | i have no idea about what freepbx does. about that, you should ask in #freepbx. |
| 07:26.16 | drmessano | nicola_pav, go to GENERAL SETTINGS |
| 07:26.23 | drmessano | Call Recording Format: |
| 07:26.29 | drmessano | It's right there |
| 07:27.30 | nicola_pav | drmessano: its WAV |
| 07:27.51 | nicola_pav | but it saves it as wav when set on demand |
| 07:28.03 | nicola_pav | when set to always, it saves it as WAV as supposed to |
| 07:28.27 | drmessano | Which version of FreePBX? |
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| 07:29.34 | nicola_pav | 2.5.2.2 |
| 07:30.21 | drmessano | It may be a bug.. but no one is going to support 2.5 |
| 07:30.27 | drmessano | You need to upgrade to 2.8 and test |
| 07:31.44 | nicola_pav | i understand |
| 07:31.56 | nicola_pav | drmessano: thanks anyway |
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| 07:32.09 | kaldemar | or find where and how the format variable is set and override what freepbx does. |
| 07:32.47 | kaldemar | my guess is that it sets it in a macro or [globals] or both. |
| 07:35.25 | drmessano | Upgrading is the easier answer and takes about 10 mins |
| 07:35.39 | drmessano | Then if it's still an issue, it can be reported properly |
| 07:35.40 | nicola_pav | i will see :) |
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| 08:49.43 | zplinux | how can I see diff between versions of libpri to understand how a bug was fixed? |
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| 08:50.09 | WIMPy | You already mentioned it: diff |
| 08:50.25 | shapr | zplinux: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.2-current ? |
| 08:50.33 | shapr | oh, that sort of diff |
| 08:50.48 | zplinux | shapr: I want to see the actual code |
| 08:51.20 | zplinux | gitweb offers this service, should i use my local svn to get this info? |
| 08:52.01 | tamiel | zplinux: yes do an svn diff between two revisions |
| 08:52.07 | WIMPy | You can view the diff between your local copy and the current one with 'svn diff'. |
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| 09:06.12 | zplinux | WIMPy: how can I see what fix was made toward this: |
| 09:06.14 | zplinux | https://issues.asterisk.org/view.php?id=15892 |
| 09:06.26 | zplinux | I want to see how this issue was fixed? |
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| 09:06.52 | zplinux | so I guess I need to get that version and rollback 1 step , right? |
| 09:07.54 | WIMPy | yes |
| 09:08.24 | zplinux | when I checked out now I got Checked out revision 2177 |
| 09:08.42 | zplinux | how can I get the version that had the fix? |
| 09:09.05 | zplinux | is it 15892 according to the URL I gave? |
| 09:10.16 | zplinux | I am still new to version control and mostly use mercurial |
| 09:10.59 | WIMPy | No, that's the issue number. I don't see a version number there. |
| 09:11.17 | zplinux | right |
| 09:11.27 | zplinux | this is why I am asking here |
| 09:12.30 | zplinux | I am not familiar with how the bugzilla for asterisk works, so how can I find the reference to that bug? |
| 09:13.23 | kaldemar | issue found in "1.4.10.1", "appears to be resolved in libPRI 1.4.10.2". either compare those two releases or try to find a note in http://svn.digium.com/svn/libpri/tags/1.4.10.2/ChangeLog to get revisions to compare. |
| 09:14.13 | kaldemar | but, looks like there's only one note between those versions. |
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| 09:26.02 | kaldemar | zplinux: with "svn log" you'll find when the 1.4.10.2 release was made and will be able to do a "svn diff -r1120:1151". |
| 09:26.16 | zplinux | hmm , ok |
| 09:27.44 | zplinux | kaldemar: you are very kind , thnaks |
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| 09:37.12 | zplinux | seems a small change |
| 09:38.23 | zplinux | the card I got in this case is |
| 09:38.26 | zplinux | Wildcard TE121 single-span T1/E1/J1 card |
| 09:39.15 | zplinux | and the fix is mainly for t200 and t203 - so If I see the channel schedualr issue, upgrading wont solve it - right? |
| 09:39.32 | zplinux | I am guessing here, but want to understand asterisk deeper |
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| 09:49.54 | E-bola | Is there a smart way to make asterisk loadbalance/failover between multiple sip accounts(providers)? |
| 09:50.23 | E-bola | as in a sort of roundrobin method where it tries a random sip account in a group of 3 to dial out via |
| 09:51.05 | E-bola | I basically have 3 different sip accounts which i would like to use as a pool, and when users dialout it picks a random/available sip account and dials out via that account |
| 09:54.07 | tzafrir_laptop | E-bola, dial to all of them (using &) in a random order? |
| 09:55.07 | E-bola | tzafrir_laptop: Hmm isnt that sort of abusive? |
| 09:55.22 | tzafrir_laptop | no idea |
| 09:55.23 | E-bola | I mean if i understand you correctly, that would dial out via all of them and use whichever one picks up first? |
| 09:55.31 | tzafrir_laptop | yes |
| 09:55.55 | E-bola | Ya, i dont think thats nice behavior, not to mention i've no idea if some providers would charge for the attempt to dialout |
| 09:56.06 | tzafrir_laptop | alternatively, you try them serially |
| 09:56.20 | E-bola | Ya, but whats the smartest way to do that? |
| 09:56.21 | tzafrir_laptop | If one is broken, how long will it take you to notice that? |
| 09:56.27 | E-bola | Isnt there a better way than checking dialstatus? |
| 09:56.51 | tzafrir_laptop | If it takes you two seconds or so, I guess it's a problem |
| 09:57.03 | tzafrir_laptop | If the timeout is 200ms, it's something you can live with |
| 09:57.44 | E-bola | Either the status of the peer is unreachable (its all iax peers so im referring to the status in iax2 show peers) or the peer will send me back some sort of error |
| 09:58.24 | E-bola | Would it be posssible to have asterisk utilize that its already known that the peer is unreachable? |
| 09:59.58 | WIMPy | It always does. |
| 10:03.09 | E-bola | So there isnt a roundrobin type function builtin? To have asterisk serially try multiple dialout accounts you need to script it yourself? |
| 10:03.41 | WIMPy | Either count or use the random function. |
| 10:04.47 | E-bola | its not so much making soemthing that picks an account thats pretty easy, its more the part where you have to check for the dialstatus top determine if you need to try another account |
| 10:05.21 | E-bola | I'd imagined it could have been built into the dial command, by allowing a peer to be reachable on multiple IP's |
| 10:06.03 | WIMPy | You have to do it yourself. And checking the ststus can be tricky. |
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| 10:09.52 | zplinux | ok, I understand that the t200 is a chip inside pri cards - right? |
| 10:09.58 | zplinux | it is not the card name |
| 10:11.28 | WIMPy | zplinux: It's the name of a timer. |
| 10:11.36 | WIMPy | i.e. software |
| 10:11.46 | zplinux | haha , ok, so this fix may solve the issue |
| 10:18.41 | E-bola | WIMPy: :( thats the least desireable answer for me hehe |
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| 11:50.54 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 11:52.13 | shamelessn00b | Hi, I just installed g729 on my asterisk box, now when the soft phones I'm using dont support g729 |
| 11:52.20 | shamelessn00b | and they are not getting registered |
| 11:52.31 | shamelessn00b | g711 is also available on the server |
| 11:54.06 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 11:58.40 | shamelessn00b | http://pastebin.com/gyEfYX1a |
| 11:58.48 | shamelessn00b | thats the output of core show translation |
| 11:59.32 | shapr | shamelessn00b: I'd suggest you check sip.conf to see if you have disallowed all codecs other than g729a |
| 12:00.01 | shamelessn00b | shapr why? |
| 12:00.14 | shamelessn00b | I want to use other codecs too |
| 12:00.23 | shapr | oh, not *registered* ... I misread |
| 12:00.45 | shapr | shamelessn00b: I think it has nothing to do with g729 being installed if your soft phones cannot register to the server. |
| 12:01.01 | shamelessn00b | heres what I want to do, register a softphone with asterisk that does not support g729 |
| 12:01.07 | shapr | but you can check to be sure by unloading the g729 module and try to register the softphones again |
| 12:01.13 | shapr | wait what? |
| 12:01.27 | shamelessn00b | dial to an extension which dials to a sip trunk |
| 12:01.35 | shamelessn00b | which is configured to use g729 |
| 12:01.42 | shamelessn00b | and let asterisk do transcoding |
| 12:01.42 | shapr | oh, you want to try transcoding |
| 12:01.47 | shamelessn00b | yes |
| 12:02.02 | shapr | the easy way to do that is to disallow g729 in your sip.conf for the softphone registration |
| 12:02.22 | shamelessn00b | I have allow=alaw and allow=ulaw |
| 12:02.26 | shamelessn00b | disallow=all |
| 12:02.28 | shamelessn00b | .. |
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| 12:02.52 | shapr | Did you reload sip or restart asterisk after making those changes? |
| 12:03.02 | shapr | Did you make those changes to only the user defined for your softphone? |
| 12:03.03 | *** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 12:03.31 | kaldemar | shamelessn00b: in what order do you have the allow and disallow lines? |
| 12:03.58 | shamelessn00b | http://pastebin.com/FaE3Yg4A |
| 12:04.54 | kaldemar | shamelessn00b: putting the disallow last will disallow all codecs, no matter what allow lines you have before it. put the disallow first and then the allow lines. |
| 12:05.02 | shamelessn00b | ok |
| 12:05.06 | shamelessn00b | didnt know that |
| 12:05.11 | shamelessn00b | thanks |
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| 12:21.51 | ironm | good afternoon - hopefully you can give a short hint for a soundconverter (from .mp3 to .wav) for linux as my asterisk 1.2 crashes quite often with .mp3 moh files - thank you in advance for any ideas. |
| 12:22.14 | WIMPy | sox |
| 12:22.47 | ironm | thank you very much WIMPy :) |
| 12:27.07 | *** part/#asterisk zplinux (~zplinux@213.8.57.217) |
| 12:36.03 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.23.48) |
| 12:36.06 | DelphiWorld | hey guys |
| 12:36.15 | DelphiWorld | please how can i let my asterisk box accept early media? |
| 12:36.31 | DelphiWorld | i have dids |
| 12:36.37 | DelphiWorld | but is sending 180 no 183 |
| 12:37.10 | ironm | WIMPy, hmm .. : sox FAIL formats: no handler for file extension `mp3' |
| 12:37.39 | DelphiWorld | lol 1.8.1.1? |
| 12:37.42 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:37.42 | DelphiWorld | isn't 1.8.0.1? |
| 12:37.43 | DelphiWorld | :P |
| 12:43.25 | ironm | DelphiWorld, 1.2 |
| 12:43.45 | DelphiWorld | ironm: i need early media |
| 12:48.09 | *** join/#asterisk zplinux (~zplinux@213.8.57.217) |
| 12:48.28 | zplinux | hi all, is there a program to send messeges using SIP, like a chat? |
| 12:48.38 | zplinux | command line |
| 12:48.39 | zplinux | for scripting |
| 12:49.39 | shapr | sipp? |
| 12:50.12 | shapr | zplinux: http://sipp.sf.net/ |
| 12:51.11 | zplinux | thanks |
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| 13:01.21 | *** join/#asterisk anita_voip (7aa3a445@gateway/web/freenode/ip.122.163.164.69) |
| 13:01.48 | anita_voip | Hi ! Can someone help with libss7 asterisk error I am getting when making outbound call ? |
| 13:02.15 | anita_voip | The error is ERROR[19658]: chan_dahdi.c:11917 dahdi_ss7_error: !! Unable to handle message of type 0x3 on CIC 1 |
| 13:03.59 | shamelessn00b | anita_voip: I use chan_ss7 by netfors |
| 13:04.36 | shamelessn00b | and probably I'm the only one using ss7 on asterisk here |
| 13:05.07 | *** join/#asterisk Lord_Rahl (~Lord_Rahl@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 13:05.54 | sivang | morning all |
| 13:06.19 | sivang | I fixed the issue I had uesterday |
| 13:06.21 | shamelessn00b | morning |
| 13:06.23 | Lord_Rahl | stupid ? about asterisknow. Do I need to have a centos install cd as well? I am trying to install it and it states it can not find a centos cd |
| 13:06.49 | sivang | sizzers: It was not NAT or anything else, just need to add 'username' to the provider setup |
| 13:06.52 | sivang | sizzers: in sip.conf |
| 13:07.18 | sivang | so no user id matching, no issue with dialpan no problem with insecure=invite etc |
| 13:07.20 | kaldemar | Lord_Rahl: afaik, it's supposed to be a complete distro. |
| 13:08.27 | Lord_Rahl | kaldemar, Thank that what I thought to. Hmmm this is my 2 download and burn to |
| 13:08.44 | anita_voip | is chan_ss7 better than libss7 ? |
| 13:08.46 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.23.48) |
| 13:10.00 | anita_voip | ok let me know if anyone here has used libss7 on asterisk |
| 13:16.05 | anita_voip | does chan_ss7 work with asterisk 1.6 ? |
| 13:16.47 | *** part/#asterisk zplinux (~zplinux@213.8.57.217) |
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| 13:36.20 | aiksa[LV] | exit |
| 13:39.17 | *** join/#asterisk nzw (~rychu@85.11.67.125) |
| 13:39.19 | nzw | hi all |
| 13:39.44 | nzw | anyone using t38modem with hylafax ? |
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| 13:44.58 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:45.09 | jaytee | good morning |
| 13:46.56 | Lord_Rahl | has anyone download and use the asterisknow 32 bit lately? I had downloaded it on two different computer Windows & Linux and the fail when burning. |
| 13:47.12 | Lord_Rahl | they* |
| 13:55.01 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 13:55.49 | jaytee | Lord_Rahl, are you burning at the slowest speed possible? |
| 13:56.37 | jaytee | Lord_Rahl, also if both computers can't burn a useable image then the download is probably corrupted. |
| 13:56.49 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:56.59 | Lord_Rahl | that my next step. I redownload check the md5 sum and it good. so it must be the burn or my cdr drive |
| 13:57.09 | jaytee | I downloaded the latest build of AsteriskNow 1.7 32 bit but haven't had the time to try a test install. |
| 14:00.05 | Lord_Rahl | i am burn at 4x I will test it in a hour lol |
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| 14:11.29 | anita_voip | i changed the nai to dynamic but still the same error |
| 14:11.30 | anita_voip | ERROR[19658]: chan_dahdi.c:11917 dahdi_ss7_error: !! Unable to handle message of type 0x3 on CIC 1 |
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| 14:15.01 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:15.22 | oelewapperke | how do you submit a patch to asterisk ? I've done the SVN thing and created a patch, sent a mail with patch attachment to asterisk-dev, but my mail isn't showing up in the archives |
| 14:15.30 | oelewapperke | so I think the mail is being ignored |
| 14:15.30 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 14:17.50 | kaldemar | oelewapperke: http://www.asterisk.org/developers/bug-guidelines http://issues.asterisk.org |
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| 14:49.28 | *** join/#asterisk sivang (~sivang@unaffiliated/sivang) |
| 14:49.30 | sivang | hi all |
| 14:49.48 | sivang | I am using the default sample dialplan with an outgoing config |
| 14:49.56 | sivang | which works well for outgoing calls, and incoming |
| 14:50.07 | sivang | however with incoming calls it just plays the tt-wasels and disconnects |
| 14:50.15 | sivang | I want it to call an extension |
| 14:50.58 | sivang | http://paste.pocoo.org/show/314677/ |
| 14:51.13 | sivang | this is the dialplain, can somebody tell me why the incoming call is going into default and not to internal ? |
| 14:52.00 | Katty | good morning |
| 14:52.09 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 14:52.31 | jaytee | sivang, you should not use the sample except as a reference, not for real use. |
| 14:52.37 | jaytee | I'd advise reading the book |
| 14:52.39 | jaytee | ~book |
| 14:52.39 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 14:54.15 | sivang | jaytee: that's after reading the book, mind you :) |
| 14:55.31 | sivang | jaytee: the setup is after arriving at page 101 |
| 14:58.30 | sivang | jaytee: so my bad, sorry, it is the dialplan the book tells you to create in extensions.conf |
| 14:58.38 | sivang | jaytee: I'm sure this is something very simple that I am missing |
| 14:59.02 | sivang | as was with using 'username' in addition to 'fromuser' to make auth work with the external sip provider |
| 15:03.46 | sivang | consults voip-info.org |
| 15:05.54 | *** join/#asterisk evangelion (~manzy_zet@212.183.172.126) |
| 15:06.20 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 15:07.31 | *** join/#asterisk WonTu (~WonTu@p57B56C49.dip.t-dialin.net) |
| 15:07.40 | evangelion | can i make dial() to jump to $WHATEVER after called party reject the call? |
| 15:07.45 | *** part/#asterisk WonTu (~WonTu@p57B56C49.dip.t-dialin.net) |
| 15:10.14 | evangelion | reject -> CANCEL |
| 15:12.47 | *** join/#asterisk sourcode (~code@ppp-58-8-92-192.revip2.asianet.co.th) |
| 15:15.07 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:15.36 | jaytee | sivang, what is the context= for your sip provider in your sip.conf? it should be set to context=incoming_calls |
| 15:15.53 | jaytee | if you haven't set it calls will enter the default context |
| 15:16.27 | sivang | jaytee: yes, so that's what happening and the settings is there, I will pb the conffile |
| 15:16.35 | jaytee | ok |
| 15:20.06 | sivang | there is an 's' context there thought |
| 15:20.08 | sivang | though |
| 15:20.18 | sivang | what does NoOp() mean? and ,1, ? |
| 15:20.21 | *** join/#asterisk luke-jr (~luke-jr@2001:470:5:265:20e:a6ff:fec4:4e5d) |
| 15:21.13 | sivang | jaytee: http://paste.pocoo.org/show/314698/ |
| 15:21.40 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:21.41 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:26.05 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 15:28.05 | kaldemar | sivang: NoOp is an application that does nothing. it can be used for debug output to print whatever is between (). ,1, as in exten => s,1,Noop() means priority 1, i.e. where an extension will start upon a match. |
| 15:28.58 | kaldemar | read up on dialplan syntax and structure. from the book for example. |
| 15:29.53 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:30.43 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 15:30.48 | IsUp | hello |
| 15:31.19 | IsUp | I am always using Asterisk 1.4 on many servers, never tried 1.6 or 1.8 |
| 15:31.28 | IsUp | should i switch to 1.8? |
| 15:31.52 | IsUp | i am mostly using phpagi on Asterisk, also SIP |
| 15:32.03 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 15:32.13 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:32.19 | IsUp | yeah i already visited that page |
| 15:32.32 | IsUp | but it doesnt give an idea, technically |
| 15:32.52 | IsUp | can you tell me major changes between 1.4 and 1.8 |
| 15:33.14 | kaldemar | read UPGRADE* in a source package. |
| 15:34.03 | IsUp | sure, thank you |
| 15:34.39 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 15:36.14 | sivang | kaldemar: will do , thanks |
| 15:36.34 | sivang | jaytee: any idea why it's not entering the incoming_calls context? |
| 15:37.11 | IsUp | sivang: whats wrong? |
| 15:40.26 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 15:41.05 | jaytee | sivang, did you change the context in sip.conf before you pasted it or was it set to that already? |
| 15:41.19 | *** join/#asterisk skrusty (~skrusty@83.166.176.39) |
| 15:41.42 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:41.44 | IsUp | and make sure you reloading your configuration after you make changes |
| 15:41.56 | jaytee | was just going to ask him that :-) |
| 15:42.05 | IsUp | :) |
| 15:45.07 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 15:45.48 | IsUp | anybody knows whats the maximum length of 'useragent=' in sip.conf? |
| 15:54.43 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.ngn.onecommunications.net) |
| 15:55.40 | fullstop | Hi all. Is it possible for a channel variable to be available to the Queue() "macro" parameter? |
| 15:56.31 | fullstop | That is, are the channel variables discarded before the macro executed when a queued call is connected to an agent? |
| 15:57.04 | IsUp | you can check by using 'core show channel <channel>' on CLI |
| 15:57.23 | IsUp | so you can see if variable is still here or not |
| 15:59.04 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:59.32 | fullstop | I'll give it a shot.. |
| 16:00.00 | evangelion | does CHANNEL() function return called channel information when called from Dial() cmd macro ( ie: Dial(SIP/100,M(returnCalledChannelInfo)) )? |
| 16:01.14 | IsUp | evangelion: you can test :p |
| 16:01.49 | evangelion | IsUp: ack! |
| 16:02.07 | evangelion | IsUp: I think it does not =( |
| 16:02.09 | jaytee | sometimes it's more fun to just try something to see if it works |
| 16:02.17 | IsUp | jaytee: exactly. |
| 16:03.16 | evangelion | err! IT DOES =) |
| 16:03.26 | jaytee | yay! |
| 16:04.11 | IsUp | hehe |
| 16:05.10 | fullstop | boo |
| 16:07.03 | sivang | IsUp. jaytee : I alerady tested with that in there, yes, let me restest can't hurt |
| 16:08.24 | IsUp | yeah sure |
| 16:08.35 | *** join/#asterisk Defraz (~Defraz@63-226-95-147.dia.static.qwest.net) |
| 16:09.17 | fullstop | It looks like my channel variables are discarded, perhaps because I am using a local channel.. =/ |
| 16:10.04 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 16:10.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:10.58 | Katty | hi russell |
| 16:11.14 | russellb | hi2u2 |
| 16:11.41 | Katty | hugs russellb |
| 16:11.44 | sivang | jaytee: so the context was there all the time. |
| 16:11.48 | IsUp | hello russellb |
| 16:11.53 | sivang | I've noticed that reloads sometime don't work |
| 16:11.54 | Katty | hugs jaytee |
| 16:11.55 | sivang | and you have to restart |
| 16:12.11 | sivang | I'm reloading configs through asterisk -r |
| 16:12.17 | sivang | and then dialplan reload/ sip reload |
| 16:12.29 | sivang | is this okay? |
| 16:12.36 | drmessano | asterisk -rx "dialplan reload" |
| 16:12.38 | sivang | maybe -r does not suppoe to work that way? |
| 16:12.51 | sivang | drmessano: I don't mind entering the CLI |
| 16:13.28 | drmessano | It's really not a matter of mind.. |
| 16:13.32 | sivang | the weasels are still eating the sounds system... :/ |
| 16:13.43 | drmessano | Why the extra step if you're just exiting the CLI to make changes, then back in |
| 16:14.00 | fullstop | too many hugs |
| 16:14.20 | drmessano | hugs Katty |
| 16:14.28 | fullstop | Thankfully everyone uses Dial(). har har |
| 16:14.30 | *** join/#asterisk acxty (c86bef37@gateway/web/freenode/ip.200.107.239.55) |
| 16:14.48 | Katty | hugs drmessano |
| 16:15.12 | IsUp | damn weasels.h264 |
| 16:15.12 | drmessano | Katty, I have some bad news |
| 16:15.18 | Katty | drmessano: oh? |
| 16:15.46 | acxty | Hi guys, I am trying to make a call using ami. I make the conection but asterisk break it instantly It doesn't show any error http://dpaste.com/294100/ |
| 16:15.46 | drmessano | Mila broke up with Macaulay |
| 16:16.01 | drmessano | Can you believe it? |
| 16:16.02 | Katty | eh? |
| 16:16.06 | drmessano | I thought they would make it |
| 16:16.13 | sivang | Mila is Russian? |
| 16:16.19 | Katty | who wha? |
| 16:16.23 | fullstop | From reading the documentation, it looks like the AGI, macro and gosub parameters are executed on the calling party's channel... but it doesn't seem to be doing that. |
| 16:16.27 | fullstop | The home alone kid |
| 16:16.29 | drmessano | Mila Kunis broke up with Macaulay Culkin |
| 16:17.08 | sivang | drmessano: it gives me auto completion |
| 16:17.08 | Katty | who is that |
| 16:17.08 | drmessano | sivang, I use the up arrow |
| 16:17.08 | fullstop | and the chick from "That 70's Show" |
| 16:17.08 | drmessano | Mila was awesome in Black Swan |
| 16:17.18 | Katty | oh you mean natalie portman |
| 16:17.19 | fullstop | I've not seen Black Swan yet, but I've heard that it was very good. |
| 16:17.26 | Katty | and whats his face ballet guy |
| 16:17.33 | drmessano | Black Swan was incredible |
| 16:17.53 | fullstop | I'm sure that it can't hold a candle to "Shark Attack 3: Megalodon" |
| 16:17.56 | drmessano | Edge-of-your-seat-mindf**k |
| 16:18.26 | drmessano | Shark Attack 3 was pretty hot |
| 16:19.32 | drmessano | I'm VERY much looking forward to Fast5 |
| 16:19.32 | acxty | any idea? |
| 16:19.32 | fullstop | I especially liked how they reminded you that it was taking place in Mexico by making sure that a Mexican flag is visible in every scene. |
| 16:19.32 | fullstop | Otherwise, the Bulgarian accents might confuse the viewer. |
| 16:19.43 | sivang | fullsto++ |
| 16:19.46 | drmessano | If Fast5 doesn't get best picture.. |
| 16:20.10 | sivang | What should i do with those weasles? :) |
| 16:20.20 | fullstop | pop them? |
| 16:20.45 | fullstop | You might need a monkey and a mulberry bush to do that, though. |
| 17:23.32 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:23.32 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.1.1 (2010/12/15), 1.6.2.15 (2010/12/08), 1.4.38 (2010/12/08), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 17:23.42 | jaytee | ~botsnack |
| 17:23.42 | infobot | aw, gee, jaytee |
| 17:25.23 | sivang | jaytee: now dealing with something else more urgent :) |
| 17:25.37 | IsUp | i told to u :p |
| 17:25.52 | *** join/#asterisk Tim_Toady (~moi@188.4.42.12.dsl.dyn.forthnet.gr) |
| 17:27.38 | Naikrovek | holy crap: http://www.youtube.com/watch?v=uTysXITBCmk |
| 17:27.43 | Naikrovek | ^^^ off topic |
| 17:28.52 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:31.46 | *** join/#asterisk slackytude (~slacky@drms-590def5b.pool.mediaWays.net) |
| 17:39.51 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 17:39.55 | *** join/#asterisk Tim_Toady (~moi@188.4.42.12.dsl.dyn.forthnet.gr) |
| 17:42.21 | *** join/#asterisk diemos (~diemos@173-8-148-213-SFBA.hfc.comcastbusiness.net) |
| 17:50.55 | *** join/#asterisk Besticles (~larry@209-58-227-178.static-ip.telepacific.net) |
| 17:51.22 | *** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 17:52.53 | Besticles | Question, I have a AMI/FastAGI application I wrote. Sometimes I need to kill my program, and start it up again with bug fixes. Problem is that sometimes my 1800 inbounds stop working. I know what causes it, the only way I've been able to fix it is to do a service asterisk stop, and a service dahdi stop and start it up again. Is there a easier way of fixing this problem? |
| 17:58.36 | *** join/#asterisk iscario (~quassel@31.248.101-84.rev.gaoland.net) |
| 17:59.19 | Katty | hello my asterisk does not work at all how to fix plz |
| 17:59.57 | IsUp | hahahaha Katty |
| 18:00.10 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 18:00.19 | carrar | reboot |
| 18:01.13 | carrar | Might have dirty power, clean the contacts on the power cord with a metal brush |
| 18:02.30 | iscario | hello, i would need help to compile asterisk-addons on openBSD... I always get an error with a missing file asterisk.h. Could you tell me what i need to install ? I installed asterisk-1.6 from the packages, and mysql-server from the packages too. thx |
| 18:02.59 | pabelanger | iscario: asterisk-1.6.-dev ? |
| 18:03.10 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 18:03.17 | *** join/#asterisk doolph (doolph@unaffiliated/doolph) |
| 18:03.33 | pabelanger | you'll need to source code for asterisk, from the openBSD port |
| 18:03.37 | doolph | anyone here know if a2billing works with asterisk 1.8.1 |
| 18:04.02 | SuPrSluG | having an issue with sip subscriptions. most phones ok, but a few aren't showing their subscriptions to an extension, when they should be. phone side or asterisk issue? |
| 18:04.44 | *** join/#asterisk dajhorn (~dajhorn@adsl-71-158-166-44.dsl.rcsntx.sbcglobal.net) |
| 18:05.11 | iscario | pabelanger: so i need to install asterisk-1.6 from ports first (binary packages isn't useful right ?) |
| 18:06.52 | pabelanger | iscario: You need to install the Asterisk headers on your system. So, depending how you installed Asterisk, from source or ports, you'll need the same headers |
| 18:08.40 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:10.06 | iscario | ok pabelanger. How could i know where are the headers ? while doing ./configure --with-asterisk=/usr/local/share/asterisk/ (where is the asterisk.h in fact) but it keeps telling me that my asterisk installation should be broken... And i receive a asterisk.h not found when building... |
| 18:10.51 | pabelanger | iscario: how did you install Asterisk? |
| 18:11.05 | IsUp | probably yum or apt :p |
| 18:11.09 | iscario | pkg_add asterisk-1.6.xxxx pabelanger |
| 18:11.16 | IsUp | tooo baaad |
| 18:11.42 | pabelanger | iscario: then you are missing the headers, you'll need to see if you can install asterisk-1.6.xxxx-dev |
| 18:11.51 | pabelanger | cannot remember how it is done on OpenBSD |
| 18:12.43 | *** join/#asterisk twanny796 (~twanny@a171.201.adsl.nextweb.net.mt) |
| 18:13.18 | twanny796 | just installed asterisk-gui, but the trunks that I had configured do no show up! |
| 18:14.44 | twanny796 | s/no/not |
| 18:15.49 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 18:16.34 | IsUp | twanny796: take a look to your /etc/asterisk/sip.conf file |
| 18:18.52 | IsUp | i am leaving, thanks for help everyone |
| 18:19.02 | IsUp | have a great day |
| 18:20.52 | *** part/#asterisk doolph (doolph@unaffiliated/doolph) |
| 18:25.09 | *** join/#asterisk Guifort (~Guifort@79.80.79.191) |
| 18:27.06 | Guifort | Hello all |
| 18:27.40 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v011-081.mobile.uci.edu) |
| 18:28.55 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 18:28.55 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 18:31.06 | Guifort | I have a question about the incoming call on asterisk, I use this configuration : http://pastebin.com/sMY0wn29 and it work fine. |
| 18:31.52 | Guifort | But, it is possible to add an another incoming line why the same host ? |
| 18:32.09 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:32.30 | Naikrovek | yes |
| 18:32.37 | Naikrovek | not only possible, but common |
| 18:32.46 | Naikrovek | these are called trunks |
| 18:32.52 | Naikrovek | usually |
| 18:33.59 | Guifort | hm, for exemple I would like use 2 account from ippi , and an account = 1 SIP Account or MGCP , but how ? |
| 18:36.10 | Naikrovek | set up one in sip.conf, the other in mgcp.conf? |
| 18:36.53 | Naikrovek | doesn't know really |
| 18:38.01 | *** join/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
| 18:38.30 | Guifort | aa no the 2 line from ippi are in sip.conf |
| 18:38.44 | kalimc | someone just dropped 4 E1/T1 TDM to Ethernet bridges (redphone) on my desk, what on earth are they for? They thought I could use them lol |
| 18:53.39 | *** join/#asterisk petaflot (~dave@85-218-21-37.dclient.lsne.ch) |
| 18:54.16 | *** join/#asterisk anita_voip (7aa3a445@gateway/web/freenode/ip.122.163.164.69) |
| 18:54.24 | anita_voip | Hi all |
| 18:54.34 | anita_voip | update on libss7, |
| 18:55.06 | anita_voip | upgrading to wanpipe version 3.5.18 solved the unknown 0x3 error, thanks to Mitul of enterux |
| 18:55.30 | anita_voip | set-up time has reduced, but still no outbound calls |
| 18:56.28 | anita_voip | debian*CLI> originate Dahdi/g1/9415000417 extension -- Hungup 'DAHDI/1-1' [Jan 5 00:26:13] WARNING[28119]: func_strings.c:778 csv_quote: No argument specified! |
| 18:56.36 | anita_voip | any pointers ? |
| 19:00.03 | petaflot | hello! I am trying to set up an asterisk server with a short config, but I cannot call one phone from another (Call from '' to extension '10' rejected because extension not found in context 'default'). config can be found here: http://pastebin.com/cYcSzDK0 I am definitely missing something obvious but I can't see what |
| 19:04.59 | pabelanger | petaflot: *CLI> dialplan show 10@default |
| 19:05.33 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:05.49 | petaflot | pabelanger: oops: There is no existence of 'default' context |
| 19:07.39 | petaflot | in case you wonder where I got my config from, I followed this howto: http://jeremy-mcnamara.com/asterisk/how-to-configure-asterisk/ |
| 19:09.29 | TimeRider | HOLD button on phone no longer holds with Asterisk 1.8 (worked in 1.6) Nortel 1535 phone - can hold call via feature codes (freepbx + latest 1.8 Asterisk compiled from source) |
| 19:09.37 | TimeRider | Softphone allows HOLD button no problem |
| 19:10.20 | kalimc | Will this allow me to hook my home phones to a SIP provideR? http://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=64 |
| 19:10.40 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 19:11.31 | *** join/#asterisk oktay (~oktay@217.131.112.52) |
| 19:11.45 | petaflot | pabelanger: I changed the "users" to "default" and that fixed it |
| 19:11.47 | petaflot | thanks for the clue |
| 19:12.19 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:16.56 | Qwell | ~book |
| 19:16.57 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 19:17.16 | Qwell | petaflot: there are a lot of places that are out of date. the book is the best place, really |
| 19:17.17 | carrar | a book of clues? |
| 19:17.39 | Qwell | out of date, or just wrong, really |
| 19:19.29 | oktay | i keep reading that SIP ALG implementations are almost always broken. That true? |
| 19:19.34 | petaflot | Qwell: so it looks.. |
| 19:20.15 | Qwell | oktay: SIP ALG by its very nature is broken |
| 19:20.20 | oktay | Qwell: is there an epub version of the book? |
| 19:20.50 | oktay | turns out my router has SIP ALG enabled by default. I had to call the ISP to get my admin level increased to be able to disable it |
| 19:21.04 | oktay | but then I learned that the SIP port is blocked at their side anyway |
| 19:21.29 | Qwell | it isn't up to a router to modify packets. that entire idea of that is fundamentally flawed. |
| 19:21.30 | oktay | They provide voip also. |
| 19:21.51 | oktay | Yeah. Good thing I saw that in the log. I had no idea SIP ALG existed. |
| 19:22.15 | oktay | And the ISP will open the port. I think I will be able to use their router which has two FXS ports on it. |
| 19:22.37 | Qwell | unlikely |
| 19:22.50 | oktay | how come? |
| 19:23.17 | Qwell | because its a telco router, with some unknown "voip" protocol (is it cable? it isn't voip at all), to which you have no access |
| 19:23.26 | oktay | it uses SIP |
| 19:23.27 | petaflot | maybe one of you knows why I can call my grandstream HW phone from linphone, but not the opposite? I get a 403 on the phone, asterisk says failed to authenticate device |
| 19:23.32 | oktay | and I have access to the whole device now |
| 19:23.58 | oktay | it's not cable. it's fiber |
| 19:24.22 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 19:29.07 | *** join/#asterisk misterrrr-t (~mrt@dslb-084-056-119-207.pools.arcor-ip.net) |
| 19:32.46 | anita_voip | anyone using libss7 ? |
| 19:33.47 | leifmadsen | ~ask |
| 19:33.47 | 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. |
| 19:34.57 | oktay | I am here against my better judgement |
| 19:38.35 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 19:38.40 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 19:50.27 | *** join/#asterisk nunners (~chatzilla@host86-134-212-68.range86-134.btcentralplus.com) |
| 19:51.13 | dlynes | Has anyone else experienced hangups with blind transfers on asterisk 1.8.1.1? |
| 19:51.42 | petaflot | I've been playing a little now, and I got something strange: extension 10 was a grandstream, ext 11 was linphonec. I swapped the two (in config of linphone and grandstream), and now the grandstream answers calls both to ext 10 and 11, and I cannnot place any calls with the grandstream (chan_sip.c:12800 check_auth: username mismatch, have <10>, digest has <11>) |
| 19:53.19 | *** join/#asterisk misterrrr-t (~mrt@dslb-084-056-119-207.pools.arcor-ip.net) |
| 19:53.55 | nunners | Could someone take a look at the following sip debug and give me some suggestions as to why it always failed with registration failed? the sip device is a gigaset c450ip http://pastebin.com/6yRFuppR thanks in advance |
| 19:54.42 | misterrrr-t | hi! i have problems compiling mISDN (latest version) on opensuse kernel 2.6.31.14-0.6. can anyone help please? |
| 19:55.45 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 20:04.24 | *** part/#asterisk petaflot (~dave@85-218-21-37.dclient.lsne.ch) |
| 20:09.06 | *** join/#asterisk OneFix_Work (~onefix@205.133.146.124) |
| 20:09.35 | OneFix_Work | I'm looking for a cheap SIP provider to test with our setup at work |
| 20:18.08 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 20:19.36 | *** join/#asterisk jblack (~jblack@pool-71-181-222-126.sctnpa.east.verizon.net) |
| 20:20.09 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 20:20.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:21.24 | *** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 20:26.56 | *** join/#asterisk hamakohako (~chatzilla@202.138.135.220) |
| 20:35.50 | OneFix_Work | <PROTECTED> |
| 20:36.12 | OneFix_Work | IF you can hack the hardware to get the SIP credentials... |
| 20:38.55 | Qwell | Doing so violates the ToS, and will get your account closed. |
| 20:41.24 | Naikrovek | magicjack does not use sip |
| 20:41.26 | Naikrovek | i don't think |
| 20:41.33 | Naikrovek | and yeah what qwell said - against the TOS |
| 20:45.58 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v011-081.mobile.uci.edu) |
| 20:46.34 | OneFix_Work | Qwell: Do they chack that? |
| 20:46.39 | OneFix_Work | check even |
| 20:46.41 | Qwell | they can, sure |
| 20:47.00 | OneFix_Work | Qwell: Can do are 2 different things :) |
| 20:47.21 | OneFix_Work | that should be "can and do" instead of "can do" |
| 20:47.32 | carrar | just go do it |
| 20:47.37 | carrar | let us know |
| 20:47.42 | carrar | report back |
| 20:48.03 | carrar | write about it on your blog |
| 20:48.38 | carrar | but just don't do anything ILLEGAL |
| 20:48.50 | OneFix_Work | carrar: Well, the fact that there is even an app to pull the information (Fiddler) and forums devoted to it makes me belive that they aren't too persistent |
| 20:49.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:49.18 | OneFix_Work | carrar: Well, that's not what this is for, it's to test voicemail integration |
| 20:49.36 | carrar | everything on the internet is for testing |
| 20:50.40 | OneFix_Work | carrar: Then again, what about NetTalk...is it against the TOS to pull the NetTalk SIP credentials? |
| 20:51.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:57.41 | OneFix_Work | carrar: Apparently MagicJack has a new system that changes passwords regularly, which makes it pretty useless as a SIP provider now |
| 20:59.49 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:00.24 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 21:00.37 | raden | why is there not a addons tarball for 1.8 ? |
| 21:01.40 | Qwell | raden: because as the UPGRADE.txt says, it's now included in Asterisk itself. |
| 21:01.49 | Qwell | (you did read it, right?) |
| 21:02.26 | russellb | Qwell: be nice. |
| 21:02.44 | carrar | be mean!! |
| 21:02.56 | Qwell | who's being mean? |
| 21:03.03 | carrar | russellb is!! |
| 21:03.38 | dlynes | Qwell, I didn't read it either, but when I ran make menuselect, I noticed it didn't need it |
| 21:04.08 | Qwell | well, we always put that kind of thing in there. it should be read on major upgrades, every time. makes things a lot easier. |
| 21:07.49 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-233-75-73.mia.bellsouth.net) |
| 21:08.33 | OneFix_Work | Actually, to do testing, it looks like SIPGate is a good service...they provide a free DID and 60 minutes of calling. |
| 21:14.07 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 21:15.13 | *** join/#asterisk xarmiex (~asulter@64.121.4.75) |
| 21:17.05 | xarmiex | anyone do realtime queues/agents ? |
| 21:18.13 | xarmiex | seems like only the oldest queue call that is up shows that the agent is 'in use' for the rest of the agent that are the phone it says 'not in use' even though they are on a call |
| 21:18.16 | xarmiex | im stumped |
| 21:19.11 | Katty | peeks in |
| 21:19.22 | Katty | OHAITHAR |
| 21:28.02 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:29.56 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:31.16 | raden | heya Katty |
| 21:31.22 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:31.53 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:32.53 | diemos | Anyone have a recommendation for a SER server? I'm attempting to install OpenSER(Kamailio) on a small appliance just for testing, but was wondering if there's any other options. |
| 21:33.58 | Katty | hugs raden |
| 21:34.02 | Katty | is anyone around STL? |
| 21:34.30 | carrar | WHAT |
| 21:34.47 | carrar | OpenSIPS |
| 21:34.57 | russellb | Kamailio! |
| 21:35.18 | *** join/#asterisk Godfather_ (~godfather@42.Red-88-9-178.dynamicIP.rima-tde.net) |
| 21:35.23 | carrar | FreeBSD! |
| 21:35.24 | diemos | Ah, so it's worth the wait to install Ruby on Rails I assume? lol |
| 21:35.44 | Katty | is a debian girl. |
| 21:35.47 | diemos | 133MHz processor is killing me -.- |
| 21:36.00 | Katty | i acknowledge there are other distros. |
| 21:36.02 | Katty | that are also good. |
| 21:36.11 | Katty | no distro wars please. >.< |
| 21:36.24 | Katty | OR I"M GOING TO START THROWING COOKIES |
| 21:36.30 | diemos | WINDOWS |
| 21:36.32 | diemos | 98SE |
| 21:36.34 | diemos | sup. |
| 21:36.42 | Katty | that's just dirty |
| 21:36.47 | russellb | Katty: your last comment seems to encourage distro wars. |
| 21:36.53 | diemos | ^ |
| 21:36.57 | Katty | oh. we can throw cookies regardless. |
| 21:37.01 | russellb | oh ok |
| 21:37.08 | carrar | Atari TOS |
| 21:37.10 | diemos | i'd rather throw cookie dough |
| 21:37.11 | Katty | it can be like the food fight from the roller dirby movie |
| 21:37.13 | Katty | whip it |
| 21:37.15 | diemos | much messier |
| 21:38.05 | Katty | installing debian from 1 cd, and then downloading all the packages is nice. |
| 21:38.08 | Katty | it just takes /forever/ |
| 21:38.15 | Katty | do do do. |
| 21:38.18 | raden | throws a cake at Katty :) |
| 21:38.38 | raden | debian rocks for server |
| 21:38.43 | raden | opensuse for desktop |
| 21:38.46 | diemos | installing gentoo from source, then downloading the packages and compiling from source is awesome. |
| 21:38.55 | raden | LMAO |
| 21:38.58 | diemos | especially when you get to X |
| 21:39.00 | carrar | LFS |
| 21:39.22 | diemos | tried it before, gave up in an hour |
| 21:45.58 | *** join/#asterisk Alric (~nbowyer@64.6.54.218) |
| 21:57.15 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 22:03.11 | *** join/#asterisk bragon (~Alexandre@81.93.247.165) |
| 22:03.16 | bragon | Hi |
| 22:03.30 | bragon | In asterisk 1.6 Playback application have been remplaced ? |
| 22:04.49 | russellb | no. |
| 22:08.21 | bragon | what are common issue a Playback() don't want to work ? |
| 22:09.47 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:12.48 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 22:15.19 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:15.59 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 22:16.03 | carrar | bragon, your file is wrong format or in wrong location |
| 22:21.06 | Corydon76-home | Or the channel is not answered |
| 22:21.24 | Corydon76-home | Or in the case of early media, progress has not been noted yet |
| 22:21.39 | Corydon76-home | Try using the Progress() app first |
| 22:23.16 | carrar | Or perhaps you're not even running Asterisk! |
| 22:26.58 | bragon | Ok i'll verify that |
| 22:30.42 | *** join/#asterisk nsgn (~nsgn@cpe-24-27-57-106.austin.res.rr.com) |
| 22:31.07 | Naikrovek | is there a good home audio channel on this server |
| 22:31.26 | nsgn | so i swapped network cards and had to re-assign my static IP in an asterisk box. i can still access the web browser based services via the network as usual...but none of the phones hit the asterisk box anymore. ideas? i've already shut off iptables and make sure i can ping both the phones and the box |
| 22:31.34 | nsgn | (also rebooted the phones and the box a few times) |
| 22:31.50 | nsgn | Naikrovek, ask in #electronics |
| 22:31.54 | nsgn | they're decent over there |
| 22:32.00 | Naikrovek | mkay thanks |
| 22:32.07 | Naikrovek | eeh invite only |
| 22:32.23 | Naikrovek | n/m i'll google it |
| 22:32.29 | bragon | include => euroweb-opened|09:00-18:00|mon-fri|*|* |
| 22:32.30 | bragon | this |
| 22:32.48 | bragon | GotoIfTime() in 1.6 right |
| 22:33.05 | *** part/#asterisk Alric (~nbowyer@64.6.54.218) |
| 22:33.05 | bragon | (difficult to migrate an asterisk 1.2 to 1.6 ... ) |
| 22:35.49 | Naikrovek | yeah |
| 22:35.51 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 22:36.27 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 22:39.13 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
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| 22:48.27 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
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| 23:16.34 | *** part/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 23:20.09 | *** join/#asterisk DarkRift (~dark@modemcable233.53-81-70.mc.videotron.ca) |
| 23:21.31 | DarkRift | is there a way to tell the "get data" agi command to wait for x seconds when some pressed a key while the sound was playing, and specify a sound to wait Y time if now key was pressed in that time? |
| 23:21.44 | DarkRift | if no * |
| 23:25.40 | DarkRift | guess not ... |
| 23:27.19 | pabelanger | DarkRift: rephrase your question, I don't understand |
| 23:27.40 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 23:27.44 | *** join/#asterisk jetlag (jetlag@pool-173-61-245-217.cmdnnj.east.verizon.net) |
| 23:34.08 | DarkRift | in fact what I want to achieve is to have multiple get data commands one after the other, when someone start to enter data, the agi command should wait until the person press #. The behavior I'm having now, once the timeout is reached the agi command returns with whatever the buffer has in it so far, instead of waiting for the user to press # |
| 23:35.22 | DarkRift | I'm trying to find a way, in AGI, to have multiple get data instead of using sox to mix them and play one get data |
| 23:35.45 | DarkRift | but when the timeout occurs, if the person didn't press # yet and entered digits, the command return with half the data |
| 23:37.03 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 23:37.59 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 23:39.57 | DarkRift | I'll bb later |
| 23:46.09 | nsgn | man i still cant solve this so any help is appreciated |
| 23:46.35 | nsgn | i changed some hardware out in my asterisk box and now i can't get any phones to register against it |
| 23:46.51 | nsgn | however i can get to the web panel in a browser and ping it at the IP all the phones are expecting it to be at |
| 23:46.54 | *** part/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 23:47.02 | nsgn | i can open up asterisk cli and things appear normal (though my knowledge is limited) |
| 23:47.07 | nsgn | what may i have gotten mixed up? |
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| 23:47.13 | *** join/#asterisk zgor (~zgor@34.224.220.87.dynamic.jazztel.es) |
| 23:47.28 | nsgn | the registration simply times out. nothing shows in the log of the phone trying to register |
| 23:47.42 | nsgn | log of a softphone just shows a pure timeout, no response from asterisk |
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| 23:55.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 23:58.05 | brainiac | Does anyone know why calls that come in to a phone that's already on a call cause one-way audio on a call already in progress? |