| 00:09.32 | *** join/#asterisk n188s (~rbundock@212.183.128.36) |
| 00:10.54 | saxa | hi, has the CLI welcome message with version and copyright been removed in 1.8.x ? |
| 00:11.13 | saxa | Im just getting asterisk CLI> |
| 00:11.31 | saxa | directly when I start asterisk -rvvvvvv |
| 00:11.42 | saxa | is that a normal behavior _ |
| 00:11.45 | saxa | ? |
| 01:25.55 | *** join/#asterisk dharper (~dharper@ppp118-209-102-217.lns20.mel4.internode.on.net) |
| 01:31.04 | *** join/#asterisk shapr (~shapr@nat/digium/x-czpuzebdobajxclz) |
| 01:34.38 | pabelanger | saxa: Already fixed, 1.8.4 should have it |
| 01:41.08 | *** join/#asterisk eMBee (~eMBee@foresight/developer/pike/programmer) |
| 01:41.13 | eMBee | good morning |
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| 02:00.53 | eMBee | how can i use the cli to have it call my phone, and then when i pick up, call a specific extension and connect me to that? |
| 02:01.57 | eMBee | it looks like i could make such a call with the originate command, but i can't see how to make that it calls me first and only calls the extension after i am connected |
| 02:04.20 | pabelanger | eMBee: you'll need to create a script to invoke Originate(), either via AMI, CLI or .call files. |
| 02:04.38 | pabelanger | Or, you call Asterisk, hangup, then asterisk calls you back |
| 02:08.46 | eMBee | well, i can use originate directly on the cli, but how do i make a call that waits before continuing? can i do it in seperate steps? |
| 02:09.12 | eMBee | dial me first, then dial the extension? but how to connect the two? |
| 02:13.21 | *** join/#asterisk foo_ (~matach@135-106.77-83.cust.bluewin.ch) |
| 02:16.48 | foo_ | Hi, I am running asterisk on ESXi over NAT. After hanging up it does not send hangup msg. What is wrong? I have to wait around 5s to disconnect. |
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| 02:26.50 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
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| 02:38.45 | shortcircuit | ~newbook |
| 02:38.46 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 02:42.14 | *** join/#asterisk khlaed (~khlaed@ppp-119-10.pppoe.canl.vu) |
| 02:42.31 | khlaed | hi me brethren! |
| 02:54.27 | pabelanger | eMBee: *CLI> core show application originate |
| 02:54.33 | pabelanger | it will explain how to use it. |
| 02:54.53 | pabelanger | You could also originate a local channel to gain more control over the Dial() command |
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| 03:10.40 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 03:21.56 | eMBee | pabelanger: pabelanger: thanks, i have already found that, it says it will block until the outgoing call gets answered, what it doesn't explain though is how it deals with the extension to connect to. what i want to avoid is that the extension rings before the outgoing call is answered |
| 03:26.45 | *** part/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 03:27.03 | pabelanger | eMBee: Yes, once the number you dial from Originate is answered, the extension specified will be invoked |
| 03:35.16 | eMBee | ah, thanks |
| 03:37.51 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 03:44.38 | pabelanger | Can Set(CDR(foo)=bar) be inherited by other channels? |
| 03:49.52 | *** join/#asterisk NetAtWorld (~support@ool-45794838.dyn.optonline.net) |
| 03:50.01 | NetAtWorld | Hi... |
| 03:51.04 | pabelanger | guess not |
| 03:55.18 | NetAtWorld | i can tell |
| 03:55.28 | NetAtWorld | pabel |
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| 05:49.41 | DrGeek | g'day all |
| 05:50.18 | DrGeek | Does anyone know how to compile modules with the "same compile-time options" that I have on my current install? |
| 05:54.29 | shapr | What's the problem? |
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| 06:07.49 | kuku | I have 2 nics on a server. 1 has public ip and routing, the other has a private subnet where the phones are. I can register from an external ip to the external address, but internall ( when I use the internal Ip address of the sip server ) I get a forbidden upon auth. |
| 06:09.27 | shapr | kuku: Do you correctly get auth required before that? |
| 06:10.21 | kuku | On the external(public) interface yes using x-lite accross the internet. |
| 06:11.07 | shapr | kuku: internally as well? |
| 06:11.10 | kuku | I use the same login/pass, but the private ip, from a pc on the private side of the pbx, and I get bad auth ( I get the error on x-lite AND on the pbx when I debug sip ) |
| 06:11.19 | shapr | are you sniffing the sip traffic with wireshark? |
| 06:11.23 | kuku | public Ip - registers no problem - private, issue. |
| 06:12.09 | kuku | shapr: I'm not, I'm using sip set debug ip 192.168.10.188 (that is the ip of the desktop with x-lite on the internal network ) |
| 06:12.18 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 06:12.37 | DrGeek | shapr, I'm getting this when I build my own codec_ilbc: Module 'codec_ilbc.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 06:13.23 | kuku | http://pastebin.com/455Lf9Fh |
| 06:13.33 | shapr | kuku: I'd suggest using wireshark to capture the sip traffic, that'll likely turn up the problem. |
| 06:16.05 | DrGeek | shapr, this is the whole error message: http://pastebin.com/vGHjiLSz . Any idea how to match the compile-time opts? |
| 06:16.09 | shapr | DrGeek: I don't know, are you able to rebuild the same version? |
| 06:17.36 | kuku | shapr: I needed to specify permit= |
| 06:17.55 | *** part/#asterisk janeNarak (~jane@2001:3c8:c103:a001::19) |
| 06:18.02 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 06:18.29 | DrGeek | shapr, this is an AsteriskNow 1.7 install and I've tried rebuilding the asterisk SRPM, with my tweaks... and still, same problem. |
| 06:18.45 | DrGeek | shapr, Do you know if asterisk has details somewhere to show what libc, gcc, etc. that it was built with? |
| 06:22.25 | sawgood | everything with AN is installed via the RPM process |
| 06:22.49 | sawgood | generally, after a AN install you should still use the yum groupinstall process to add those lib RPM packages |
| 06:23.15 | sawgood | anyways, that is what I've found ... IMHO |
| 06:25.02 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 06:30.55 | DrGeek | sawgood, yep. I always do that when possible. Unfortunately, codec_ilbc isn't available in AN due to (silly) licensing issues |
| 06:32.03 | DrGeek | omg, I fixed it! |
| 06:32.19 | DrGeek | ok, this is ugly... and its probably the *wrong* way to go about it... |
| 06:32.42 | DrGeek | so theres this value called buildopt_sum, calculated by ./configure and embedded in every module that is built. |
| 06:32.53 | DrGeek | Its an md5sum (or some hash). |
| 06:33.12 | DrGeek | If it doesn't match, it yells like the pastie from above. |
| 06:33.39 | DrGeek | so... I loaded the same source tree of asterisk, patched codec_ilbc, manually hacked the buildopt_sum to be the one installed by AN, and Viola! |
| 06:34.15 | DrGeek | If you're curious, that nefarious sum lives here as AST_BUILDOPT_SUM: ./include/asterisk/buildopts.h |
| 06:35.33 | DrGeek | it appears to be a hash of all the MENUSELECT_CFLAGS piped through md5sum. |
| 06:37.38 | DrGeek | and would you believe... I can even dial out! |
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| 07:28.10 | schmidts | good morning |
| 07:28.14 | whiteroom | morning! |
| 07:31.04 | whiteroom | Has anyone ever had an issue with the "database show" command from the CLI not showing all entries of the database? I know other entries exist because I can retrieve their values with a "database get DEVICE/5030 user", for instance. |
| 07:34.36 | schmidts | whiteroom what version do you use? |
| 07:35.48 | whiteroom | I'm currently having this problem on 1.8.2.3, but I believe I have experienced it on the 1.4 branch as well (roughly 1.4.18-1.4.39) |
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| 07:46.37 | schmidts | whiteroom ok thats strange then, 1.8 shouldnt have the problem i thought of ;) |
| 07:47.11 | whiteroom | schmidts so then you've heard of this issue before? |
| 07:48.15 | schmidts | whiteroom no but i had an idea what could cause this |
| 07:50.09 | whiteroom | schmidts i've noticed a possible pattern between the records that do show with a "database show" and the ones that don't. this current system is a trixbox build and data from a previous trixbox install was imported into it via FreePBX. the imported records *may* be the ones that aren't showing for "database show" but are still retrievable directly |
| 07:50.44 | whiteroom | schmidts I don't know if that rings any bells. I'm thinking my next stop will have to be a possible bug report.... or digging into how I can repair or reinitialize the astdb |
| 07:58.49 | schmidts | whiteroom ah ok i understand what you mean, its possible that your astdb file is corrupted. Do you have the same problems after a core shutdown gracefull and starting asterisk new? |
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| 08:01.59 | whiteroom | schmidts I just gave that a try and it didn't seem to make a difference. I ran a "core stop gracefully" from the CLI and then "amportal start" from the command line |
| 08:03.35 | schmidts | whiteroom do you need the data which are in there? cause if not you can try a database deltree to clean it up and restart asterisk |
| 08:03.42 | kaldemar | whiteroom: might be that the FreePBX import feature has screwed up your DB. confirm that scenario somehow. |
| 08:05.16 | whiteroom | schmidts I do need the data. the situation is that I've created a web API that utilizes information that it grabs from the "database show" command, so I need that command to show the entire database contents. I suppose if there was a way of dumping the information via another means I could then re-add it and possibly solve the issue that way |
| 08:06.35 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 08:06.41 | whiteroom | kaldemar it seems as though, if anything, it has just made the particular entries it added a little strange. It would be pretty difficult for me to confirm this behavior without trashing the rest of my config (the import was done a while ago and changed have happened since). |
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| 08:08.05 | kaldemar | whiteroom: you can take a backup of the database, it's in /var/lib/asterisk/astdb unless otherwise configured in asterisk.conf. |
| 08:09.15 | kaldemar | then start with a clean DB, putting similar keys and values into it. then try the show commands before and after an import. |
| 08:10.09 | whiteroom | kaldemar: my concern is more with the changes to the mysql database, system recordings, and agi scripts that a freepbx import will also cause. I see what you're saying though. It seems like there should be an easy way to dump the database and examine the record differences or something. |
| 08:11.32 | *** join/#asterisk joachim_- (~joachim@post.comvie.no) |
| 08:12.00 | kaldemar | the folks at #freepbx might have some better tips on verifying the import functionality. |
| 08:12.50 | joachim_- | Is there any way to define dynamic queue members in the sql database? Cant find much info about it |
| 08:13.00 | whiteroom | kaldemar that's a good idea, I'll check with them too |
| 08:13.42 | whiteroom | joachim_-: I believe you want to look into asterisk realtime functionality |
| 08:14.44 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
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| 08:16.24 | kaldemar | joachim_-: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue |
| 08:17.04 | joachim_- | whiteroom and kaldemar: thanks. =) |
| 08:18.39 | whiteroom | kaldemar: this isn't a very fun answer, but may be a solution: http://fonality.com/trixbox/node/36005 |
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| 08:23.49 | kaldemar | whiteroom: or make a small app that goes through the entries in the freepbx db and puts them into astdb. probably a bit faster and sustainable than clicking buttons. :) |
| 08:24.39 | whiteroom | kaldemar my thoughts exactly. I'm thinking that freepbx must flag entries as needing to be updated somehow in the mysql db. I'll just go through and create those flags and call it a day (hopefully) |
| 08:33.25 | *** join/#asterisk daxt (~daxt@112.135.64.220) |
| 08:33.34 | daxt | anyone using mydivert ? |
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| 09:12.06 | whiteroom | kaldemar: that worked like a charm. thanks for working through that issue with me |
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| 09:35.55 | daxt | guys anybody integrated mydivert with asterisk ? |
| 09:39.22 | kaldemar | daxt: i guess someone has when they say that is works with asterisk on their front page. are you having a problem with it? |
| 09:40.23 | daxt | kaldemar - i am running asterisk on EC2 , yes i am having issues because when i try to test mydivert with my sip softphone , i don't get the feedback |
| 09:41.37 | kaldemar | how are you testing it and how does the test fail? |
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| 09:47.39 | *** join/#asterisk daxt (~daxt@112.135.94.118) |
| 09:48.44 | daxt | kaldemar can you refer me to the person who integrated it ? |
| 09:51.53 | *** join/#asterisk matach (~matach@135-106.77-83.cust.bluewin.ch) |
| 09:57.29 | *** join/#asterisk Vec (~Vec@84.12.253.146) |
| 09:58.40 | kaldemar | daxt: no, i don't know any. but you're probably suffering from a general type of problem that can be solved without any further knowledge of mydivert. are you willing to show something to debug? |
| 09:59.19 | daxt | kaldemar - yes i will , first i have to locate the problem |
| 10:01.53 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:04.14 | Vec | Hi, I am having a problem with Skype for asterisk, runnning 1.6.2.16.1 and the latest S4A, when I call in from Skype > Asterisk, as soon as I send a Ringing() or Answer(), the channel is hangup immediately. |
| 10:05.22 | shapr | Vec: turn on debug/verose and check the logs? |
| 10:05.50 | Vec | shapr : done all that, there is nothing to see, it just shows a hangup |
| 10:08.05 | shapr | Have you tried it with a totally basic dialplan that just answers and plays back a sound? |
| 10:08.24 | *** join/#asterisk lost_soul (~noymfb@cpe-74-78-191-114.twcny.res.rr.com) |
| 10:08.34 | Vec | [2011-02-16 10:07:01] NOTICE[8485]: core.cpp:2146 sfa_call_hangup: ending call |
| 10:09.29 | Vec | shapr : yeh my dial plan is Wait(3) Answer() Wait(3), it hangs up immediately on the answer. I know by the timing. |
| 10:10.35 | shapr | Have you tried the skype client for Linux on that system? |
| 10:11.13 | shapr | If you have similar problems with the skype client, it might be network settings. |
| 10:11.30 | Vec | shapr : no, but I have tried it on 2 diffirent linux servers |
| 10:11.46 | Vec | shapr : and the server I am currently trying it on, has a public ip |
| 10:11.55 | shapr | Same problem on both? |
| 10:12.25 | Vec | shapr : yeh exactly the same |
| 10:12.46 | Vec | shapr : I have iptables running, but I stopped it to test, and same issue |
| 10:13.19 | Vec | shapr : it used to work, and then one day it just stopped, but we never made any config changes |
| 10:13.29 | shapr | When did it stop working? |
| 10:14.09 | shapr | I'd still suggest you install the graphical Skype client for Linux. You can use the client without X if you use ssh -X |
| 10:22.30 | *** join/#asterisk proute (~AnthonyCB@mail.sysun-technologies.com) |
| 10:22.47 | proute | Hello |
| 10:23.37 | proute | I upgrade my ipbx from misdn to dahdi module. before, with misdn my fax works fine. Now with dahdi fax are corrupted. Do you have any idea. |
| 10:24.05 | proute | Fax are transmitted via g711 Pass through |
| 10:24.28 | proute | I use Digium B410P |
| 10:27.24 | shapr | Hi Anthony :-) |
| 10:27.52 | proute | Hi shapr |
| 10:30.43 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 10:30.46 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 10:31.46 | matach | Hi everyone, few hours back I wrote about my situation and unfortunately I lost my internet connectivity. I am running Asterisk on ESXi over NAT. After hanging up it does not send hangup msg. What is wrong? I have to wait around 5s-7s to disconnect from Asterisk. I did debug and it keeps session. Any help will be appreciated. |
| 10:32.45 | teloniusz | hi. I'd like to catch all incoming SIP requests from a given net, let's say, 192.168.100.0/24, and allow them to do INVITEs without user or password. How to do this? |
| 10:32.53 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 10:33.30 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 10:34.42 | ironm | hello - I need a short hint - thank you in advance ;) - my question: does asterisk 1.6 or 1.8 (CenOS 5.5 x86_64) run without troubles on HP DL 385 G7 Servers (AMD 12 core CPUs)? |
| 10:35.14 | teloniusz | Do I need to create a trunk for this, or an user? I've only had experience with Asterisk GUI till now and it seems that it's not sufficient... |
| 10:35.53 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 10:37.10 | *** join/#asterisk philfine (~philfine@56.4.166.178.rev.vodafone.pt) |
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| 10:38.02 | shapr | ironm: Asterisk by itself should work really well on 12 cores. |
| 10:38.03 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:39.38 | ironm | thank you very much shapr -- I guess (and hope) asterisk can scale well (in case there are two 12 core cpus inside the box) .. |
| 10:40.08 | shapr | 24 cores?! That sounds like fun. |
| 10:40.20 | shapr | ironm: What are you going to DO with that system? |
| 10:41.42 | shapr | ~nat |
| 10:41.42 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 10:41.55 | no1peanut | ironm, Are you gonna have like 3d holographic video conference ? |
| 10:42.56 | shapr | matach: If the BYE is never received, it's probably something to do with your NAT config. |
| 10:43.08 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 10:43.45 | shapr | matach: If you have nat=yes for your peers and canreinvite=no in the [general] section of sip.conf, that may fix it.. but afaiu that runs all RTP traffic through your server. |
| 10:46.01 | matach | shapr: yes I set nat=yes and canreinvite=no |
| 10:46.23 | matach | shapr: I also redirect my RTP ports |
| 10:46.25 | BlackBishop | what does an asterisk box needs more ? (ram/cpu) if all it's supposed to do is have a couple of datacards to transfer voice from/to and some sip accounts ( all < 100 ) ! .. how much ram/cpu would be needed ? |
| 10:47.24 | philfine | Hi everyone, I am looking into setup a minimum yet powerful asterisk box. Although I find it very hard to start with all the default sample files. |
| 10:47.47 | philfine | What is the minimum setup to make sip and dahci work |
| 10:48.20 | philfine | I want to be able to understand everything, but with all the default configuration seems impossible |
| 10:53.02 | ironm | re shapr / no1peanut ad. 1 to server 10k+ calls .. ad. 2 is an video conference possible at all with asterisk 1.6 or 1.8 ? |
| 10:53.32 | shapr | Asterisk can do pass-through video conferencing. |
| 10:53.36 | ironm | s/ an / a / |
| 10:54.19 | ironm | thank you shapr :) |
| 10:56.19 | matach | shapr: Afaiu each extension needs 2 ports. I need it for 4 extensions so in my rtp.conf I have rtpstart=12000 rtpend=12007, but in my opinion it is not a RTP issue. :( |
| 11:01.19 | philfine | Anyone ? Should I keep all the config files as they initially are or can I just remove everything and copy essentials ? |
| 11:03.13 | kaldemar | philfine: if by initially you mean the sample files, don't use them as is. if you remove something that is needed, you'll get an error/warning/notice when you start asterisk. |
| 11:04.05 | philfine | kaldemar: Are the needed files posted somewhere ? |
| 11:04.39 | philfine | Apart from extensions.conf and sip.conf ... what is needed ? |
| 11:06.40 | kaldemar | philfine: needed files depends on what modules you want to use. keep at least asterisk.conf, features.conf, indications.conf, modules.conf, rtp.conf and start narrowing down what you need. |
| 11:07.06 | philfine | Ok |
| 11:07.09 | philfine | Thanks |
| 11:10.46 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 11:21.02 | philfine | kaldemar: I just copied those files. when loading the asterisk it complains of missing files. Should all the other files be there but rather empty ? |
| 11:22.23 | philfine | Does autoload=yes in modules.conf means load everything ? :) |
| 11:23.34 | kaldemar | yes, it loads all modules that it finds in the module directory. |
| 11:23.36 | *** join/#asterisk daxt (~daxt@112.135.72.238) |
| 11:23.55 | *** join/#asterisk digitalirony (www-data@my.ass.looks.just.like.your-face.info) |
| 11:24.23 | philfine | So it is not what I want :D |
| 11:27.07 | philfine | manager.conf is also needed ? |
| 11:29.20 | kaldemar | if you want to use the manager interface. |
| 11:29.49 | kaldemar | you can use autoload=yes and then make a noload statement for the modules you don't want. |
| 11:30.55 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 11:31.59 | *** join/#asterisk daxt (~daxt@112.135.72.238) |
| 11:37.25 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:05.52 | philfine | For now I do not seem to be able to connect to my natted asterisk machine from outside the network |
| 12:05.55 | philfine | Can someone help ? |
| 12:06.15 | philfine | Simply says it doesn't match any peer |
| 12:07.06 | philfine | Should I define somewhere my domain, apart from localnet and externhost ? |
| 12:07.39 | philfine | I mean define the domain in externhost and my local network in localnet |
| 12:07.48 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:08.38 | Dovid | is there any way from the dial plan to see if a channel is up ? |
| 12:09.14 | *** join/#asterisk QN (~mIRC@175.156.208.182) |
| 12:11.31 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 12:11.33 | QN | hi. would like to check if anyone has experiences with setting up asterisk with inphonex (including caller id) |
| 12:13.47 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:15.35 | kaldemar | philfine: define a peer that matches. but yes, do define externaddr/externhost and localnet if you are behind a NAT. |
| 12:45.32 | Dovid | tzafrir: Do you mind looking at this and giving me your opnion ? https://issues.asterisk.org/view.php?id=18722 |
| 12:46.06 | philfine | Should the peer include the domain ? |
| 12:46.12 | philfine | kaldemar: ? |
| 12:46.26 | philfine | should it be user@domain ? |
| 12:48.33 | kaldemar | philfine: no. see "Naming devices" in http://svn.digium.com/svn/asterisk/tags/1.8.2.3/configs/sip.conf.sample |
| 12:48.33 | *** join/#asterisk hlep (~hlep@h-163-251.A212.priv.bahnhof.se) |
| 12:48.51 | philfine | thanks |
| 12:48.52 | QN | can i send special invite base on trunk? |
| 12:48.57 | QN | thanks |
| 12:50.11 | eMBee | good evening |
| 12:50.25 | philfine | type friend == user, right ? |
| 12:50.47 | hlep | is there a slick way to play playtones only if the channel is answered and if not play ringing instead? |
| 12:51.09 | eMBee | i have two extensions where asterisk and the phone (cisco 7960) tell me, they are unavailable. but from either phone i can make calls. i just restarted asterisk and the status doesn't change |
| 12:51.24 | eMBee | what could make these phones be unavailable? |
| 12:51.46 | eMBee | there is a little x next to the line symbol on the phone |
| 12:52.11 | WIMPy | The one that's unavailable? |
| 12:52.19 | eMBee | yes |
| 12:52.48 | eMBee | the phone is registered and i can make calls on that line |
| 12:52.49 | WIMPy | It probably means not registered. |
| 12:53.28 | WIMPy | Thet's two different things. Look at sip show peers. |
| 12:53.32 | eMBee | ok, i am not actually sure if it is registered, but i can make calls |
| 12:54.23 | eMBee | ok, it does not list any ip address for those extensions |
| 12:54.45 | WIMPy | Then it's not registered. |
| 12:54.59 | WIMPy | So indeed unavailable. |
| 12:55.20 | eMBee | that makes sense, but what would cause it to fail to register? hmm |
| 12:55.23 | hlep | enable sip debugging on the ip of the phone and try to register again and look in asterisk |
| 12:55.54 | WIMPy | I'd start with just verbose and debug turned up. |
| 12:56.14 | WIMPy | But the phone might be able to tell you more as well. |
| 12:56.33 | hlep | well with sip debug you will see exactly what the phone and asterisk are saying to eachother |
| 12:57.34 | WIMPy | And that's more than I'd bother to read unless neccessary. |
| 12:57.55 | eMBee | :-) |
| 12:58.26 | eMBee | starts with just debug turned higher |
| 12:58.54 | WIMPy | Ask the phone. It must have some error message. |
| 13:03.05 | QN | anyone has experience with inphonex? |
| 13:03.59 | eMBee | i can not see any errors on the phone. in the settings in status, the messages i can see in 'status messages' are the same that are on a working phone (no time server, unprovisioned proxy_emergency ... |
| 13:04.28 | WIMPy | The phone tells you it's not registered, but nothing else? |
| 13:04.55 | Dovid | any dev's here ? |
| 13:05.55 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 13:09.40 | eMBee | WIMPy: that is what it looks like |
| 13:09.58 | eMBee | it is not a wrong password, because asterisk logs complain about that loudly |
| 13:10.10 | WIMPy | Unfriendly phone. |
| 13:10.37 | eMBee | yes |
| 13:10.49 | hlep | NAT involved? |
| 13:10.59 | eMBee | no |
| 13:11.29 | eMBee | the phone is configured via tftp, that part works |
| 13:12.05 | hlep | i would still suggest looking at the actual sip conversation between the phone and asterisk |
| 13:13.21 | eMBee | ok, how do i do that? |
| 13:13.36 | WIMPy | sip set debug peer <name> |
| 13:13.48 | hlep | or ip <ip> |
| 13:14.04 | WIMPy | Err, no, ip would be neccessay here, as it's not registered. |
| 13:14.14 | kaldemar | QN: you can add headers to the invite in your dialplan with app SIPAddHeader but that's about it. |
| 13:14.48 | hlep | well you will see the registration request from the phone (ip) even if it fails with registration |
| 13:15.02 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:15.34 | eMBee | ok, i set the ip, now watching |
| 13:15.35 | WIMPy | adds a menatal note saying not even basic bebugging to Cisco phones. |
| 13:16.02 | eMBee | ? |
| 13:21.44 | eMBee | still nothing in the logs, do i need to raise the debugging value? |
| 13:23.04 | hlep | what did it say when you tired to register the phone? |
| 13:23.07 | hlep | tried* |
| 13:23.24 | hlep | the sip message |
| 13:23.33 | WIMPy | Maybe the phone isn't trying for some reason? |
| 13:23.52 | WIMPy | tcpdump would be the next stage. |
| 13:24.50 | *** join/#asterisk Guest60326 (~Geek@unaffiliated/cain) |
| 13:25.31 | hlep | if asterisk said nothing when you had sip debug enabled and tried to register, problem most likely lies in between |
| 13:26.07 | eMBee | how would i make the phone want to register, i wonder? |
| 13:26.25 | eMBee | i am just resetting the phone every time |
| 13:26.36 | eMBee | reboot ing it |
| 13:26.42 | hlep | I just use softphones so not sure, but most likely it should try to do it when you start it |
| 13:26.45 | *** join/#asterisk DrkShadow (~andrew@host-72-175-240-62.static.bresnan.net) |
| 13:26.50 | eMBee | nods |
| 13:27.09 | hlep | so did the asterisk say anything when you started the phone? |
| 13:27.43 | DrkShadow | I'm having problems with _one_ of my Asstra phones redialing indefinitely when a 503 Service Unavailable is sent to it. The log of the one that fails and one that works is here -- http://pastebin.com/BvPqmzdq . Even the firmware version is the same, but for some reason the failing one sends a CANCEL and the working one sends an ACK.. wtf? |
| 13:28.00 | eMBee | hlep: it didn't say anything |
| 13:28.10 | hlep | you had working phones right? |
| 13:28.33 | hlep | basic troubleshooting then, switch the one that isn't working with one who is and vice versa |
| 13:29.08 | *** join/#asterisk andygraybeal (~andy.gray@static-50-42-0-34.dsl.athn.oh.frontiernet.net) |
| 13:31.54 | eMBee | yes, ihave a working phone, i just rebooted that to see what would happen |
| 13:33.23 | *** join/#asterisk iulhk (~iulhk@119.152.226.130) |
| 13:33.25 | iulhk | hi all |
| 13:34.56 | leifmadsen | ohai! |
| 13:34.57 | leifmadsen | o/ |
| 13:36.49 | iulhk | where to paste questions ..anybody there ? |
| 13:37.37 | QN | i need some help. in order for me to do a cid passthrough for outgoing call, it requires me to send a 'custom' invite. so i would like to know if there's anyway i could have a 'custom' invite just for a specific trunk, so it'll not affect other trunks in the box. |
| 13:38.29 | eMBee | switching the phones also has no effect |
| 13:40.43 | eMBee | now i am trying to switch config files |
| 13:40.54 | hlep | did the "broken" phone work ok when you moved it to a working location? or was the "working" phone fine on both and the "broken" not on both locations? |
| 13:40.59 | leifmadsen | iulhk: you can use pastebin for large amounts of data |
| 13:41.00 | leifmadsen | ~pb |
| 13:41.00 | infobot | [~pb] 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. |
| 13:41.12 | leifmadsen | iulhk: otherwise if it's just 1-2 lines, just ask your question here |
| 13:41.15 | leifmadsen | ~ask |
| 13:41.15 | 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. |
| 13:41.29 | eMBee | maybe a pointer to the list would help? |
| 13:41.34 | wdoekes2 | QN: what do you mean "custom" invite? |
| 13:41.35 | eMBee | ooops |
| 13:41.44 | eMBee | wrong channel |
| 13:42.03 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:43.31 | eMBee | switching configs switches the error, or rather the error sticks with the hardware, not with the extension |
| 13:44.20 | WIMPy | Something important missing in the configuration that's set in one phone but not the other? |
| 13:44.25 | QN | wdoekes2: if i'm not wrong, the usual invite is like "INVITE sip:the-number-i-am-calling@sip.mysip.com" however, my provider needs me to pass "INVITE sip:my-account-info*the-number-i-am-calling@sip.mysip.com" in order for me to achieve the cid passthrough. so for that, i'll need to make my invite as mentioned. |
| 13:45.14 | QN | wdoekes2: any idea? thanks. |
| 13:45.28 | WIMPy | eMBee: Maybe you try to do a factory reset on both and see if they then behave the same. |
| 13:45.35 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 13:45.58 | eMBee | hmm, yes, since config is over tftp a factory reset is not disruptive, will try |
| 13:49.42 | *** join/#asterisk razu_ (~razu@razu.data.ee) |
| 13:51.38 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 13:53.06 | *** join/#asterisk Olivier_54 (~Olivier_5@maxicours.pck.nerim.net) |
| 13:53.46 | DrkShadow | changing hangupcause=34 to hangupcause=27 resolves my Asstra phone issue. |
| 13:54.05 | eMBee | WIMPy: factory reset doesn't help |
| 13:55.30 | WIMPy | Use diff to compare the configs. |
| 13:55.40 | wdoekes2 | QN: make a new dialplan context and call that instead of SIP/trunk |
| 13:55.43 | *** join/#asterisk Guest60326 (~Geek@unaffiliated/cain) |
| 13:56.18 | wdoekes2 | in that context you do _+X!,1,Dial(SIP/accountinfo*${EXTEN}@trunk) |
| 13:56.48 | wdoekes2 | you call the context using the Local channel: Dial(Local/+12345@mycontext) |
| 13:57.20 | eMBee | WIMPy the only difference is the extension number, also i swapped the configs between the phones, the configs didn't affect the status |
| 13:57.24 | *** join/#asterisk wonderworld (~ww@port-92-201-81-40.dynamic.qsc.de) |
| 13:57.43 | eMBee | oh wait, the working phone just reset and now it is not working anymore |
| 13:57.50 | eMBee | you are on to something |
| 13:57.52 | WIMPy | eMBee: Do they have different software? |
| 13:58.01 | eMBee | nope, same firmware version |
| 13:58.01 | WIMPy | Ah. |
| 13:58.21 | WIMPy | So there is something missing in the config file then. |
| 13:58.29 | eMBee | yes, looks like it |
| 13:59.26 | eMBee | now, i wonder if i can extract the full config of cisco 7960 |
| 14:01.11 | *** join/#asterisk matach (~matach@135-106.77-83.cust.bluewin.ch) |
| 14:10.51 | eMBee | i think i found it: the most obvious too, but hard to find: register with proxy is NO by default |
| 14:11.42 | *** join/#asterisk matach (~matach@135-106.77-83.cust.bluewin.ch) |
| 14:12.46 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
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| 14:24.25 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 14:25.31 | QN | wdoekes2: now i got the invite working, however, i'm stucked on the "From" portion. |
| 14:25.41 | *** join/#asterisk daxt (~daxt@112.135.71.212) |
| 14:26.19 | eMBee | WIMPy, hlep: register with proxy was the culprit, found the setting i needed to add to the default config and now all phones work |
| 14:26.26 | eMBee | thanks for your help |
| 14:26.42 | QN | wdoekes2: my provider requires me to send "From: "First LastName" <sip:5551112222@sip.mysip.com>" but no matter what i do, i'm still sending "From: "5551112222" <sip:12345@sip.myprovider.com>" instead. |
| 14:26.51 | eMBee | cost me some time, but at least now all phones received a facrtory reset so such wierdness should not happen again |
| 14:27.59 | wdoekes2 | QN: as priority 1 in your context, do Set(CALLERID(all)="Firstname LastName" <5551112222>") and make the Dial the next prio |
| 14:28.14 | wdoekes2 | s/")// |
| 14:28.24 | wdoekes2 | s/")/)/ |
| 14:37.26 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 14:37.31 | QN | wdoekes2: sorry to ask, may i know where is the contaxt file? |
| 14:37.54 | *** join/#asterisk daxt (~daxt@112.135.71.212) |
| 14:39.38 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 14:39.38 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 14:39.49 | skrusty | is there a way to fire a custom manager event from an AGI? |
| 14:40.09 | skrusty | is there such a thing as a custom manager event? :/ |
| 14:43.23 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 14:43.31 | wdoekes2 | QN: I don't follow.. you were editing extensions.conf, right? |
| 14:43.53 | QN | wdoekes2: hmm. i've set it, but when i dial, i still sees "From: "Firstname LastName" <sip:12345@sip.myprovider.com>" instead of "From: "Firstname LastName" <sip:5551112222@sip.myprovider.com>" |
| 14:44.35 | wdoekes2 | do you have fromuser set? and sendrpid=no? in sip.conf |
| 14:51.53 | asterisk-learner | Hi, i installed dahdi-tools, dahdi-linux and asterisk 1.4.36, but when doing make menuselect I cannot choose dahdi from the channels drivers menu ... |
| 14:52.20 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-twmxgeajzanewizv) |
| 14:52.55 | asterisk-learner | any ideas ? |
| 14:54.19 | iprouteth0 | might be an option passed during your ./configure |
| 14:54.54 | asterisk-learner | such as ? I didnt modify ./configure |
| 14:55.07 | iprouteth0 | no idea. I'm taking a shot in the dark |
| 14:55.14 | asterisk-learner | should i biuld and install dadhi before ./configure ? |
| 14:55.34 | iprouteth0 | I recently compiled UCSniff, but wanted the gui as well so needed to pass ./configure --enable-gui |
| 14:55.58 | iprouteth0 | yes, I imagine ./configure detects presence of dahdi libraries |
| 14:56.06 | asterisk-learner | mm ok |
| 14:56.12 | asterisk-learner | so now i have to make make clean of asterisk |
| 14:56.19 | asterisk-learner | or should i do make dist clean ? |
| 14:56.26 | iprouteth0 | most ./configure scripts check for lib dependencies and report whats missing |
| 14:56.59 | iprouteth0 | not sure. Never removed anything I manually compiled. And I use gentoo, so asterisk is in portage and is easily compiled |
| 14:57.18 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 14:57.28 | asterisk-learner | mmm ok i will try this route |
| 14:57.28 | iprouteth0 | but if building asterisk on gentoo and you want dahdi, I believe you'd have to do something like |
| 14:57.38 | iprouteth0 | USE="dahdi" emerge asterisk -av |
| 14:57.39 | asterisk-learner | ? |
| 14:58.23 | iprouteth0 | the use flag with gentoo's portage acts sort of like ./configure --enable-dahdi |
| 14:58.46 | iprouteth0 | emerge is gentoo's package management system if you weren't aware |
| 14:59.02 | iprouteth0 | or a utility in it I should say |
| 14:59.15 | iprouteth0 | I'm no expert though. Still learning myself |
| 14:59.27 | JerJer | hence why someone should learn how ot build asterisk themselves |
| 14:59.59 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:59.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:00.16 | iprouteth0 | still learning. I'm actually reading "The Future of Telephony" right now |
| 15:00.21 | leifmadsen | ~newbook |
| 15:00.21 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 15:00.30 | leifmadsen | iprouteth0: I suggest The Definitive Guide :) |
| 15:00.51 | iprouteth0 | thankyou kindly :) always appreciate as many info sources as possible |
| 15:01.04 | leifmadsen | iprouteth0: same book as TFOT, but updated and better :) |
| 15:01.10 | leifmadsen | (based on 1.8 instead of 1.4) |
| 15:01.37 | eMBee | does anyone know if i can get rid of these "HTTP Manager add header action: ping" lines in the log when i have verbose set higher than 3? |
| 15:01.52 | leifmadsen | modify the code? |
| 15:01.53 | iprouteth0 | excellent. I'm at 1.6.2.14 |
| 15:01.59 | leifmadsen | iprouteth0: much of it will apply still |
| 15:02.02 | eMBee | :-) |
| 15:02.09 | iprouteth0 | I had imagined so |
| 15:02.14 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:02.35 | eMBee | i thought at somepoint i saw a way to change a config setting that could turn that off |
| 15:03.03 | leifmadsen | eMBee: what does the logged line look like exactly? |
| 15:03.13 | leifmadsen | (what level of logging for example) |
| 15:03.14 | iprouteth0 | I figure asterisk is a good thing for me to learn right now, as I recently took position monitoring telco network |
| 15:03.27 | leifmadsen | could potentially turn off some AMI logging in the manager.conf file |
| 15:03.30 | iprouteth0 | get to play with 5 different switch platforms |
| 15:06.03 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 15:07.21 | ssureshot | does ubuntu lucid apt install of asterisk work well? |
| 15:09.19 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:11.26 | QN | wdoekes2: yes, i do have fromuser set and not defined sendrpid. |
| 15:12.30 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:13.09 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-33.dsl.net.pk) |
| 15:13.30 | mechbangirc | any comments for fail-safe recording server... |
| 15:13.54 | Dovid | mechbangirc: define fail safe |
| 15:14.14 | mechbangirc | good suggestion |
| 15:14.40 | mechbangirc | one that doesn't piss you off |
| 15:16.30 | *** part/#asterisk mechbangirc (~mechbangi@mbl-82-58-33.dsl.net.pk) |
| 15:16.34 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-33.dsl.net.pk) |
| 15:16.37 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 15:18.17 | eMBee | leifmadsen: at level 4 i get half a dozen lines starting with HTTP Manager add header action: ping |
| 15:18.34 | wdoekes2 | QN: remove the fromuser, it overwrites the CALLERID(num) |
| 15:18.42 | eMBee | they all have "HTTP Manager add header" as their first part |
| 15:18.50 | eMBee | and they come every few seconds |
| 15:19.12 | leifmadsen | eMBee: and there is nothign like [WARNING], [DEBUG] etc on the line with it? |
| 15:19.17 | eMBee | they are caused by the AsterixGUI login |
| 15:19.26 | eMBee | nothing else, that was the full line |
| 15:19.56 | leifmadsen | then you either need to change something in manager.conf or in the code itself as it might just be echoing a line out from the code directly without any configurable method to modify that. |
| 15:20.06 | leifmadsen | I'm guessing the latter |
| 15:24.09 | eMBee | probably |
| 15:24.15 | eMBee | thanks |
| 15:25.27 | JerJer | yawns |
| 15:28.18 | QN | wdoekes2: ah. thanks. |
| 15:28.22 | QN | its working |
| 15:28.24 | QN | =D |
| 15:29.56 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:32.55 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 15:35.17 | Faustov | it's not really a voip question, but perhaps someone here will have a good idea: say you have 10 LANs behind 10 Internet GWs you control, how would you provide internal communication so any host from any lan can reach another? |
| 15:36.30 | Tozz_ | vpn's |
| 15:37.00 | WIMPy | Faustov: Maybe tinc is somethig for you? |
| 15:37.01 | Faustov | that's what I got, just looking for something simplier |
| 15:37.11 | Faustov | sry that was re vpns |
| 15:37.17 | Faustov | WIMPy: googling |
| 15:38.33 | Tozz_ | i dont think there really is anything simpeler then vpn's/tunneling |
| 15:38.44 | Faustov | mhm, this tinc indeed seems earier for a full-mesh solution than my mesh of openvpn servers-clients |
| 15:38.54 | Faustov | easier* |
| 15:39.09 | *** join/#asterisk domedan (~domedan@c83-253-183-67.bredband.comhem.se) |
| 15:39.20 | Tozz_ | tinc is just vpn ;) |
| 15:39.39 | WIMPy | Yes, but mesh instead of ptp. |
| 15:40.31 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:40.33 | Faustov | and that is what I was looking for ;) |
| 15:42.06 | Faustov | WIMPy: have you used it before, btw? |
| 15:42.29 | WIMPy | No. Just read and herad about it. |
| 15:43.31 | WIMPy | I'm on openvpn, I don't need or actually dont want everyone to communicate with everyone. |
| 15:43.52 | Faustov | right |
| 15:44.06 | Faustov | well in my situation this seems exactly what I need |
| 15:44.10 | Faustov | so thanks ;) |
| 15:44.32 | exothermc | anyone know the website for O one or O1 ? |
| 15:45.19 | WIMPy | only knows O2 or 3. |
| 15:45.55 | exothermc | doh www.o1.com |
| 15:46.59 | *** join/#asterisk crakrjak (~merc@rrcs-70-62-156-154.central.biz.rr.com) |
| 15:47.29 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 15:47.53 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:50.20 | asterisk-learner | ok, i managed to compile asterisk and dahdi modules (i now have a chan_dahdi.so in my module folder) |
| 15:50.31 | asterisk-learner | however, I still cant load this module |
| 15:50.56 | asterisk-learner | i get : Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref |
| 15:51.00 | asterisk-learner | any ideas ? |
| 15:51.18 | JerJer | the smdi stuff did not get compiled |
| 15:51.29 | asterisk-learner | mmm how chould i compile it ? |
| 15:51.35 | asterisk-learner | from the menuselect ? |
| 15:51.43 | WIMPy | You dont; have the verion running you built? |
| 15:51.57 | JerJer | pehaps res_smdi |
| 15:53.39 | asterisk-learner | JerJer: res_smdi is available and selected |
| 15:53.43 | asterisk-learner | from make menuselect |
| 15:54.38 | asterisk-learner | even if i load the module manualy form the cli once asterisk starts i cant, so it is not an order issue like i read on the net |
| 15:56.22 | iulhk | will anybody take a look at the issue call disconnecting after every 15-20 seconds "http://asterisk.pastey.net/146247" ?? |
| 15:56.37 | *** join/#asterisk philfine (~philfine@56.4.166.178.rev.vodafone.pt) |
| 15:57.12 | WIMPy | iulhk: Looks like a networking issue. |
| 15:58.22 | asterisk-learner | when running dahd_cfg -vv, is it ok to get 0 channels to configure, if i dont have a digium hardware |
| 15:58.24 | asterisk-learner | ? |
| 15:58.59 | bbryant | yes |
| 15:59.51 | *** part/#asterisk mechbangirc (~mechbangi@mbl-82-58-33.dsl.net.pk) |
| 16:00.05 | asterisk-learner | ok thx |
| 16:01.05 | kaldemar | iulhk: are SIP/9000 and SIP/9001 in the same LAN? |
| 16:01.24 | iulhk | yes |
| 16:01.41 | iulhk | <kaldemar>: yes |
| 16:02.28 | *** join/#asterisk vaewyn (root@jabber.andrews.edu) |
| 16:02.29 | kaldemar | SIP/9001@9001 looks strange. what do you see in sip debug? |
| 16:02.51 | iulhk | <kaldemar>: wait let me paste it |
| 16:02.54 | i_heart_asterisk | friedMonkey |
| 16:02.57 | asterisk-learner | when running module show like dahdi, i only get 2 modules |
| 16:03.06 | asterisk-learner | codec_dahdi.so |
| 16:03.08 | asterisk-learner | and |
| 16:03.12 | asterisk-learner | app_dahdibarge.so |
| 16:03.24 | asterisk-learner | why can't I load chan_dahdi.so ??? |
| 16:03.28 | kaldemar | asterisk-learner: that shows loaded modules. |
| 16:03.43 | *** join/#asterisk friedMonkey (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 16:04.10 | kaldemar | asterisk-learner: as WIMPy asked, are you running the version you just compiled or did you just copy chan_dahdi.so to the module dir? |
| 16:04.30 | asterisk-learner | kaldemar: yes the version i just compiled |
| 16:04.41 | i_heart_asterisk | what up friedMonkey |
| 16:04.56 | friedMonkey | i heart * too |
| 16:05.26 | asterisk-learner | should i delete/remove zaptel from my system or is a service zaptel stop enough ? |
| 16:05.27 | vaewyn | Howdy all! Just upgraded from 1.6 to 1.8... and any time I make a call that leaves the server (DAHDI T1) I get laggy call progress and warnings "Retransmission timeout..." on the SIP phone I am calling from... I can call voicemail on the box though and interact all day and it works great... any ideas? |
| 16:05.32 | asterisk-learner | before using / installing dahdi ? |
| 16:07.15 | QN | wdoekes2: thanks for the help~ |
| 16:07.17 | QN | =D |
| 16:09.51 | frigidzephyr | asterisk-learner: i recommend removing all zaptel related files, some are installed in different directories than the DAHDI files |
| 16:10.20 | *** join/#asterisk lep (~lep@ppp-118-60.21-151.libero.it) |
| 16:10.36 | frigidzephyr | also remove zaptel init scripts, etc |
| 16:10.46 | asterisk-learner | ok i just went into zaptel directory made : make clean and make uninstall |
| 16:11.10 | asterisk-learner | can someone tell me what is chan_features used for ? |
| 16:11.19 | russellb | nothing, ignore it |
| 16:11.23 | asterisk-learner | and also codec_ilbc |
| 16:11.24 | russellb | it was a project that was never finished |
| 16:11.26 | asterisk-learner | ok |
| 16:11.39 | russellb | codec_ilbc is a module that provides the ilbc codec, as surprising as that sounds |
| 16:11.40 | asterisk-learner | i have a .so file from a previous instalation i just delete it ...just in case |
| 16:11.52 | asterisk-learner | russelb: :-P |
| 16:12.03 | asterisk-learner | i meant is dahdi dependant on it |
| 16:12.05 | asterisk-learner | in some way ? |
| 16:13.24 | frigidzephyr | no |
| 16:15.28 | iulhk | <kaldemar>: sip debug are "http://asterisk.pastey.net/146249"? |
| 16:15.41 | asterisk-learner | ok, but what did u mena by zaptel init scripts ? |
| 16:15.46 | asterisk-learner | etc, |
| 16:18.04 | kaldemar | iulhk: SIP/9000 is not in the same LAN with SIP/9001. disable re-invites for SIP/9000. |
| 16:18.47 | iulhk | in sip.conf i hv already mentioned canreinvite=no |
| 16:19.24 | *** join/#asterisk kmcconno1 (~kmcconnon@76.9.102.130) |
| 16:19.29 | iulhk | anyway let me check it again |
| 16:21.14 | vaewyn | So anyone have an idea? 1.6 -> 1.8 upgrade... any call to my servers voicemail from SIP or DAHDI sources works great... calling DAHDI -> server -> SIP or SIP -> server -> DAHDI I get delayed call progress logging... no audio... and SIP retransmit warnings |
| 16:21.52 | kaldemar | iulhk: what version of asterisk are you using and where did you put the parameter? |
| 16:23.24 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 16:23.40 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 16:27.15 | asterisk-learner | As u can see here, i tried using ldd on chan_dahdi.so to see on what it depends, but it doesnt seem to depend on smdi... |
| 16:27.17 | asterisk-learner | http://pastebin.com/rKYnPvd6 |
| 16:29.12 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 16:29.48 | ruben23 | hi my dahdi is broken, can i recomplie it without recompiling asterisk and libpri..? |
| 16:30.53 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 16:31.43 | ruben23 | is it ok guys..? |
| 16:32.07 | WIMPy | You should recompile Asterisk if you recompile dahdi. |
| 16:32.07 | pabelanger | ruben23: how did it break? Kernal update? |
| 16:32.17 | pabelanger | s/Kernal/Kernel |
| 16:32.47 | ruben23 | <PROTECTED> |
| 16:33.31 | WIMPy | How did that break it? Shredded FS? |
| 16:33.35 | rubberneck | Im using Asterisk 1.6.2.15 all peers are SIP. Can I upgrade to 1.8 using my current dialplan and SIP peer configuration? Are all the same applications and modules available in 1.8 as there were in 1.6? |
| 16:33.39 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:34.51 | drmessano-lt | guesses power down = kernel update |
| 16:35.21 | pabelanger | rubberneck: read UPGRADE.txt and CHANGES in the 1.8 source directory |
| 16:36.21 | rubberneck | pabelanger: Guess it makes sense that it would be there. thanks. |
| 16:37.20 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 16:37.40 | *** join/#asterisk Olivier_54 (~Olivier_5@maxicours.pck.nerim.net) |
| 16:45.40 | philfine | What are the modules I should forcefully load to be able to perform sip calls |
| 16:45.50 | philfine | Currently I can make the calls but got no sound |
| 16:45.58 | philfine | using softphones |
| 16:46.05 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:47.31 | malcolmd | if you can make a sip call, then chan_sip is already loaded. if you're not getting audio, there are any number of things that could be at play. try setting directmedia=no for the sip peers in question as first step |
| 16:49.00 | drmessano-lt | Ummm |
| 16:50.39 | friedMonkey | i've discovered an issue where AGI is not notified after a xfer w/ polycom's softkey |
| 16:51.12 | i_heart_asterisk | friedMonkey, you are not the first to discover that issue |
| 16:51.30 | philfine | malcolmd: directmedia is already to no |
| 16:51.36 | friedMonkey | my constituent discovered it |
| 16:51.54 | philfine | Since I am not autoloading modules I assume I might miss some module |
| 16:52.09 | WIMPy | res_rtp? |
| 16:52.20 | i_heart_asterisk | friedMonkey, I've tried to get a fix for it but it is not considered a "BUG" to digium |
| 16:52.30 | friedMonkey | a bug? |
| 16:52.42 | i_heart_asterisk | they siad you have to use asterisk's AMI instead |
| 16:52.42 | friedMonkey | is that like an annoying pest? |
| 16:53.10 | philfine | Well it is not the modules since with autoload it doesn't work as well |
| 16:53.21 | malcolmd | philfine: a phone can individually call into a background or playback of a file? rtp is going to the right address? nat's not hosing things up? |
| 16:53.40 | friedMonkey | shouldn't a bug that has been found be fixed? that is generally what occurs, no? |
| 16:54.18 | i_heart_asterisk | as i said before digium does not consider it a bug |
| 16:54.31 | i_heart_asterisk | "just the way things work" |
| 16:55.23 | friedMonkey | apologies, i did misread. so if I were to discover a problem with my vehicle that could be fixed, a mechanic can say "it's just how it works?" |
| 16:55.55 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 16:56.01 | thehar | i_heart_asterisk: i'm sure if you wrote a patch to the "bug" then it would be fixed. |
| 16:56.15 | chazzam | if you discovered something about your vehicle that didn't work the way you want it to, you can buy an aftermarket piece to fix it to work the way you want it to... |
| 16:56.30 | thehar | it is open source after all |
| 16:56.38 | i_heart_asterisk | thehar, i don't know where to start and apparently no one else does either |
| 16:57.02 | i_heart_asterisk | chazzam, are you suggesting you pay someone to fix this problem w/ asterisk? |
| 16:57.12 | *** part/#asterisk lep (~lep@ppp-118-60.21-151.libero.it) |
| 16:57.27 | chazzam | if its been decided that it isn't a bug, and would thus be a feature request, that would definitely be an option |
| 16:57.45 | chazzam | they have a bounty system... |
| 16:57.49 | drmessano-lt | i_heart_asterisk, you act as if someone said "it's not a bug, that's how it works, too bad"... the issue was the behavior is as expected or not implemented and would be a feature request. |
| 16:58.13 | philfine | BTW, it shows Function MASTER_CHANNEL not registered |
| 16:58.16 | i_heart_asterisk | drmessano-lt, that is exactly what leif told me |
| 16:58.17 | philfine | malcolmd: ? |
| 16:58.18 | drmessano-lt | i_heart_asterisk, asterisk not doing something you would like it to do doesn't just make it a "bug" |
| 16:58.48 | drmessano-lt | i_heart_asterisk, I was here for that convo, and that's NOT what was said |
| 16:58.50 | i_heart_asterisk | drmessano-lt, how can software using AGI function properly in this case? |
| 16:59.13 | chazzam | parse the AMI? |
| 16:59.22 | i_heart_asterisk | chazzam, i said AGI, not AMI |
| 16:59.50 | chazzam | i_heart_asterisk | they siad you have to use asterisk's AMI instead |
| 17:00.14 | i_heart_asterisk | chazzam, why though? because AGI /dialplan is NOT functioning properly in this case |
| 17:00.27 | *** part/#asterisk Montys (~dmartinez@nat/digium/x-dnzuxyrthqxgpgjz) |
| 17:00.57 | i_heart_asterisk | it may be "behavior as expected", but that is BAD behavior |
| 17:01.13 | chazzam | i_heart_asterisk: then put in a bounty on a feature request |
| 17:01.44 | *** join/#asterisk WWGD (~WWGD@c-24-22-54-155.hsd1.wa.comcast.net) |
| 17:01.45 | i_heart_asterisk | chazzam, I would rather spend countless hours learning all of the asterisk code and fixing it myself |
| 17:01.47 | *** part/#asterisk WWGD (~WWGD@c-24-22-54-155.hsd1.wa.comcast.net) |
| 17:01.55 | thehar | then do it and quit spamming us with your complaints |
| 17:02.10 | drmessano-lt | or complaining... err.. jinx |
| 17:02.18 | *** join/#asterisk intrin (~Jana@host-174-44-172-96.hln-mt.client.bresnan.net) |
| 17:02.19 | intrin | hi |
| 17:02.41 | friedMonkey | seems as though, if there's an expected problem, when something should run properly, it should be remedied. |
| 17:02.52 | intrin | asterisk have ways that i can track all my incoming and outgoing calls, names, and info about that call? |
| 17:02.53 | chazzam | it wasn't an expected problem |
| 17:02.57 | intrin | like a CRM |
| 17:03.06 | i_heart_asterisk | it's an "expected behaviour" |
| 17:03.15 | i_heart_asterisk | which is a problem for anyone using AGI |
| 17:03.37 | intrin | i just need to be able to know who i need to follow up with after talking to them, etc too |
| 17:03.48 | thehar | if it was more of a "problem" then i'm sure it would be attended to more or someone else would have "fixed" this "problem" |
| 17:03.48 | chazzam | fairly positive you can access the AMI from within an AGI script |
| 17:04.44 | drmessano-lt | Not implemented = feature request .... implemented but broken = bug |
| 17:05.03 | intrin | is everyone just ignoring me? :x |
| 17:05.04 | i_heart_asterisk | i consider this broken |
| 17:05.11 | drmessano-lt | We know |
| 17:05.28 | intrin | ;/ |
| 17:05.50 | drmessano-lt | intrin, yes it does.. AMI |
| 17:05.56 | thehar | intrin: like CDRs? |
| 17:05.56 | intrin | ? |
| 17:05.59 | intrin | ami stands for? |
| 17:06.01 | i_heart_asterisk | you consider it a "feature request", i'm only asking that a feature work properly |
| 17:06.15 | intrin | not sure what CDRs is |
| 17:06.16 | i_heart_asterisk | i'm not asking for a "new" feature |
| 17:06.35 | intrin | googles |
| 17:06.46 | thehar | intrin: call detail record |
| 17:06.52 | *** mode/#asterisk [+b *!*@*buf.ny.i-evolve.net] by Qwell |
| 17:06.55 | *** kick/#asterisk [i_heart_asterisk!~north@pdpc/sponsor/digium/Qwell] by Qwell (go away) |
| 17:06.56 | intrin | :) |
| 17:06.58 | *** kick/#asterisk [friedMonkey!~north@pdpc/sponsor/digium/Qwell] by Qwell ( go away) |
| 17:06.59 | thehar | thank jesus |
| 17:07.15 | leifmadsen | we've gone over this several times |
| 17:07.20 | leifmadsen | the answer is the same |
| 17:07.21 | Qwell | GEE, YOU MEAN THEY WEREN'T THE SAME PERSON?! |
| 17:07.28 | Qwell | does he think we're stupid? |
| 17:07.50 | intrin | thanks |
| 17:07.52 | intrin | ill google |
| 17:08.01 | thehar | intrin: cdrs should be able to give you all of that |
| 17:08.13 | jaytee | yeah, wasn't i_heart_asterisk moaning about the same thing last week? |
| 17:08.32 | thehar | jaytee: yes |
| 17:08.38 | drmessano-lt | Yes he was |
| 17:08.43 | russellb | Qwell: +q! |
| 17:08.53 | russellb | kick has more flair though |
| 17:09.00 | Qwell | hmm, I forget how +q syntax works |
| 17:09.05 | Qwell | it's a ban with something at the front... |
| 17:09.11 | russellb | i usually just /mode +q <nick> |
| 17:09.18 | thehar | it silences them |
| 17:09.20 | jaytee | russellb, exactly how many pieces of flair are sufficient? :-) |
| 17:09.25 | russellb | 17 |
| 17:09.29 | jaytee | hehe |
| 17:09.29 | thehar | lol |
| 17:09.32 | Qwell | russellb: yeah, but you can +q a hostmask with ban syntax |
| 17:09.34 | WIMPy | Or a ban without kick? Should do the same. |
| 17:09.43 | vaewyn | now give him back his stapler |
| 17:09.49 | drmessano-lt | Would lack of flair be a bug or feature request for more flair? |
| 17:09.50 | russellb | Qwell: yeah |
| 17:09.53 | jaytee | hey, I was reading the new book and I found what I think is a typo |
| 17:10.03 | intrin | do i need a special card to hook an asterisk box to my bresnan modem? or can i just use a nic? |
| 17:10.14 | russellb | jaytee: cool, drop us an email and we'll make sure it's fixed. we have an amazing copy editor that probably found it. :-) |
| 17:10.22 | leifmadsen | :) |
| 17:10.25 | jaytee | when's the book out? |
| 17:10.28 | thehar | yay new book |
| 17:10.32 | WIMPy | intrin: What kind of thing is that? |
| 17:10.33 | russellb | jaytee: russell at russellbryant dot net and leif.madsen at gmail dot com |
| 17:10.34 | leifmadsen | March-April time frame |
| 17:10.50 | leifmadsen | we're just finishing up production and editing |
| 17:11.01 | jaytee | cool, I want to buy a print edition and a Kindle edition |
| 17:11.06 | russellb | jaytee: <3 |
| 17:11.09 | leifmadsen | we like you :) |
| 17:11.11 | intrin | well i got my bresnan modem, and i can login and it has a call manager, but not good enough. i wanna route all the calls through the asterisk box, and have it keep track of everyhting |
| 17:11.26 | russellb | but from oreilly.com and get your ebook drm-free! |
| 17:11.27 | intrin | incoming calls, outgonig calls, when i need to call to follow up with a call. etc |
| 17:11.44 | russellb | s/but/buy/ |
| 17:11.47 | jaytee | heck, I have two copies of *TFOT 2nd edition in print and a kindle version. |
| 17:11.58 | jaytee | one of the print editions is signed by Jared |
| 17:12.25 | thehar | i have the first signed |
| 17:12.30 | thehar | i require the second print signed |
| 17:12.37 | thehar | AND kindle as well! |
| 17:12.43 | jaytee | I love my Kindle |
| 17:12.46 | thehar | <3 |
| 17:13.04 | carrar | did he sign your kindle? |
| 17:13.12 | thehar | hrm no |
| 17:13.13 | jaytee | I got the new 3G+WiFi one as a Christmas gift. |
| 17:13.15 | *** mode/#asterisk [+q *!*@*buf.ny.i-evolve.net] by Qwell |
| 17:13.16 | thehar | i should have them do that this astricon |
| 17:13.21 | *** mode/#asterisk [-b *!*@*buf.ny.i-evolve.net] by Qwell |
| 17:13.27 | thehar | gasp |
| 17:13.30 | thehar | Asterisk Kindle! |
| 17:13.36 | Qwell | +q isn't a ban anymore |
| 17:13.43 | drmessano-lt | I can't wait to put the new book on the shelf between the TFOT 2nd edition and Lawn Darts for Dummies |
| 17:13.48 | WIMPy | Asterisk on the Kindle? |
| 17:13.54 | jaytee | drmessano, lol |
| 17:13.59 | thehar | drmessano-lt: lol indeed |
| 17:14.12 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 17:14.21 | jaytee | ~newbook |
| 17:14.21 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 17:15.30 | anonymouz666 | what happened to tk-dfender? |
| 17:16.11 | intrin | well i guess ill dive into this head first and see where i get :) |
| 17:16.30 | *** join/#asterisk friedMonkey (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 17:19.44 | *** join/#asterisk daxt (~daxt@112.135.92.214) |
| 17:24.33 | jaytee | russell and leif, you've got mail |
| 17:25.17 | carrar | I'd love to download and review the 3rd book |
| 17:25.19 | jaytee | anonymouz666, try in freepbx for TK. He doesn't come in here anymore. |
| 17:25.45 | WIMPy | Wow. He does Freepbx? |
| 17:25.50 | carrar | how does one do that? |
| 17:25.51 | leifmadsen | carrar: sorry, review is closed now -- review was at http://ofps.oreilly.com/ |
| 17:25.55 | carrar | hha |
| 17:26.01 | *** join/#asterisk ke4zvu3 (~jnalley@fedora/ke4zvu3) |
| 17:26.10 | leifmadsen | the only download of the book will be via Kindle or purchased PDF |
| 17:26.18 | leifmadsen | (it'll still be available via HTML) |
| 17:26.35 | carrar | What was wrong with status quo? |
| 17:26.36 | Qwell | leifmadsen: do I get a free signed copy of this book too? |
| 17:26.50 | Qwell | and by that, I mean pdf |
| 17:26.57 | carrar | heh |
| 17:27.10 | carrar | php sign? |
| 17:27.12 | Qwell | -rw-r--r-- 1 qwell qwell 72 2010-04-16 10:57 9780596510480.pdf.sig |
| 17:27.13 | carrar | pgp |
| 17:27.16 | Qwell | heck yeah |
| 17:27.33 | n3hxs | pgp? pretty good piracy? |
| 17:28.15 | drmessano-lt | still has his boxed copy of McAfee PGP |
| 17:28.46 | leifmadsen | Qwell: sure! come visit with russell! |
| 17:28.54 | leifmadsen | carrar: what do you mean? |
| 17:28.57 | Qwell | wat |
| 17:29.03 | leifmadsen | "status quo?" |
| 17:29.07 | leifmadsen | the license hasn't changed |
| 17:29.08 | chazzam | visit canada! |
| 17:29.31 | carrar | By status quo I mean free |
| 17:29.43 | Qwell | carrar: the html will be free |
| 17:29.45 | carrar | didn't make enough money on the resale side? |
| 17:30.04 | carrar | was just curious |
| 17:30.40 | Qwell | leifmadsen: did I ever tell you about how my old boss printed the entire v1 pdf? |
| 17:30.42 | leifmadsen | carrar: it's still freely available via HTML like I said -- so nothing has changed |
| 17:30.49 | leifmadsen | Qwell: no, but I'm sure he wasn't the only one :) |
| 17:30.50 | Qwell | it was huge |
| 17:30.54 | *** part/#asterisk friedMonkey (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 17:30.55 | *** join/#asterisk friedMonkey (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 17:31.11 | carrar | thats nuts to print that out |
| 17:31.12 | carrar | heh |
| 17:31.16 | Qwell | yes |
| 17:31.20 | leifmadsen | Qwell: probably cost more than just buying it |
| 17:31.22 | _Corey_ | Geez, if you're going to print it out just buy a nice copy |
| 17:31.29 | Qwell | leifmadsen: company paid for it |
| 17:31.52 | Qwell | and you know how banks are.. |
| 17:31.56 | leifmadsen | Qwell: I'm sure the company would have just preferred him to get them to buy a book :) |
| 17:31.58 | leifmadsen | heh |
| 17:31.59 | chazzam | lol |
| 17:32.00 | leifmadsen | cheap! |
| 17:32.14 | chazzam | did he print it single sided even? |
| 17:32.16 | leifmadsen | friend of mine printed out an entire 1000 page cisco manual at school |
| 17:32.20 | Qwell | leifmadsen: if it wasn't for personal use, maybe :) |
| 17:32.37 | drmessano-lt | I like the PDF so I can print mine on 3' x 2' sheets |
| 17:32.40 | _Corey_ | I worked at this manufacturing company a long time ago and we used to have a guy who bound all those linux PDF HOWTOS from the Linux Documentation Project |
| 17:32.49 | _Corey_ | I think it took like 2 days |
| 17:32.49 | Qwell | oh lord |
| 17:32.55 | chazzam | :o |
| 17:32.59 | Qwell | tldp is huge |
| 17:33.01 | _Corey_ | I still have my set somewhere ;) |
| 17:33.19 | drmessano-lt | What do you think the market would be for an Asterisk themed childrens book? |
| 17:33.25 | drmessano-lt | My First Asterisk, maybe? |
| 17:34.46 | drmessano-lt | *crickets* |
| 17:34.54 | *** part/#asterisk ke4zvu3 (~jnalley@fedora/ke4zvu3) |
| 17:35.00 | *** join/#asterisk ke4zvu3 (~jnalley@fedora/ke4zvu3) |
| 17:35.37 | _Corey_ | Explaining what a phone system does to my 29 year old fiance is difficult enough... |
| 17:36.22 | ke4zvu3 | hi, i'm planning an asteriskNOW deployment and wanted to know if anyone knew anything about if and how well the newer cisco SPA series of phones work with asterisk? I've used the cisco 7900 series of phones with asterisk before with good success. However, sadly, i can't seem to find any good docs on asterisk/asteriskNOW's support or success stories for the Cisco SPA phones. does anyone know if the firmware is free ? |
| 17:37.08 | _Corey_ | ke4zvu3: It's on the Cisco website |
| 17:37.39 | ke4zvu3 | _Corey_: i presume you mean the firmware, thanks. any thoughts/experiences on its usage in asterisk ? |
| 17:38.08 | *** join/#asterisk mrfaison (~mrfaison@200.13.124.36) |
| 17:38.16 | _Corey_ | Yeah, we had to support some on a deployment recently where we cleaned up after another shop |
| 17:38.22 | drmessano-lt | If the SPA phones truly are descended from the SPA-9xx series, SIP should work fine |
| 17:38.28 | _Corey_ | They're OK, not as nice as the old SPA |
| 17:38.48 | _Corey_ | but they're similar enough that the provisioning process was the same as the Linksys and Sipura stuff |
| 17:39.26 | mrfaison | hi folks! maybe a silly question but: i have installed asterisk 1.6.2 and i configured chan_sccp RC3.1 |
| 17:39.27 | _Corey_ | I won't be buying any myself though, for whatever that's worth |
| 17:39.43 | mrfaison | i just want to configure in sccp.conf a 7941 extension |
| 17:41.15 | wdoekes2 | QN: glad to be of help |
| 17:48.21 | *** join/#asterisk teddy_salad (~bclark@eniccpsc01-v410.mid.sta.suddenlink.net) |
| 17:49.11 | mrfaison | sccp help? |
| 17:51.31 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 17:58.09 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:00.46 | tzafrir | mrfaison, that command should work if you have chan_sccp.so loaded |
| 18:02.24 | RypPn | mrfaison http://chan-sccp-b.sourceforge.net/doc/index.html |
| 18:02.33 | *** join/#asterisk asilva (~andre@2801:88:1000:2:240:a7ff:fe10:28f) |
| 18:03.53 | asilva | Hello i have a gosub under a macro, after the gosub i have some other routines but when the gosub get executed after the hangup i doesnt returns to where is supposed to |
| 18:07.20 | asilva | just so you know the gosub redirects so a context which have switches => DUNDi.... |
| 18:08.31 | mrfaison | tzafrir: i mean i want sccp extensions setup help |
| 18:08.38 | mrfaison | tzafrir: :P |
| 18:09.10 | mrfaison | RypPn: i want to setup noe extension, i dont find clear how to config one extension |
| 18:10.03 | *** join/#asterisk cVsup (~cVsup@189.83.131.149) |
| 18:10.14 | *** part/#asterisk cVsup (~cVsup@189.83.131.149) |
| 18:10.31 | leifmadsen | asilva: GoSub() doesn't return unless you use Return() |
| 18:10.52 | asilva | leifmadsen, how can i use a return under context .... {} |
| 18:11.18 | asilva | leifmadsen, under this context there is only switches {} |
| 18:11.48 | leifmadsen | I don't understand your question |
| 18:12.05 | asilva | Okay let me try to explain again |
| 18:12.06 | leifmadsen | why are you using switches at all? |
| 18:12.22 | asilva | I have a macro which gosubs to a context that has switches {} |
| 18:12.26 | leifmadsen | using the dialplan functions and applications for DUNDi lookups is the preferred method |
| 18:12.39 | leifmadsen | also why would you use a macro to get to a gosub? |
| 18:12.50 | leifmadsen | that whole thing you're doing is the wrong approach |
| 18:12.59 | asilva | using DUNDI funcitions is better than the using under siwtches{} |
| 18:13.30 | leifmadsen | of course |
| 18:13.32 | asilva | i mean get the dundi route manual is better? |
| 18:13.37 | leifmadsen | switches are a very old way of doing things |
| 18:13.38 | asilva | good to know |
| 18:13.47 | leifmadsen | you should never *need* to use switches |
| 18:13.56 | leifmadsen | a Macro() calling GoSub() is just odd |
| 18:14.10 | asilva | just using because of the switch |
| 18:14.17 | asilva | i thought it was a good thing |
| 18:14.21 | asilva | using switches |
| 18:14.32 | asilva | let me get back to DUNDI functions it works properly!!! |
| 18:14.34 | WIMPy | Why are dialpan functions preferred? They have limited functionality. |
| 18:15.30 | asilva | i only use context dialplans to set permission to my users, and all goes to a macro the does all the work including the dundi search and getting the route! |
| 18:15.33 | asilva | Thanks for the help |
| 18:16.23 | leifmadsen | WIMPy: I don't understand what you mean by limited functionality -- I consider switches to be more limited in terms of what you can do and what you can control |
| 18:16.54 | leifmadsen | people are welcome to use whatever works, but a Macro() calling a GoSub() calling a switch is just the weirdest thing I've ever seen |
| 18:17.13 | WIMPy | As usual, dialplan functions only work for complete numbers. |
| 18:17.20 | asilva | have any papers about dundi best practices of use? |
| 18:17.38 | leifmadsen | asilva: read the book |
| 18:17.41 | leifmadsen | ~newbook |
| 18:17.41 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 18:17.48 | leifmadsen | there is a whole chapter on DUNDi |
| 18:17.55 | asilva | nice!! thanks for the hint |
| 18:19.50 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:20.18 | *** join/#asterisk deadpigeon (~deadpigeo@office.xpressamerica.net) |
| 18:23.10 | drmessano-lt | Wondering if I should publish this Asterisk source code on Wikileaks |
| 18:26.45 | drmessano-lt | hmm |
| 18:26.58 | WIMPy | Someone already did it? |
| 18:27.43 | drmessano-lt | No, but after I typed that, my wifi kicked me off for a moment |
| 18:27.52 | drmessano-lt | Anonymous is everywhere, apparently |
| 18:28.15 | WIMPy | Does it have a badword filter? |
| 18:28.31 | drmessano-lt | Nope |
| 18:36.26 | *** join/#asterisk rmcmahon (~rmcmahon@76.9.102.130) |
| 18:42.14 | carrar | asterisk-bad-word-filter.so ? |
| 18:43.15 | Qwell | res_profanity.so |
| 18:43.21 | Qwell | it has both a filter function as well as an add function |
| 18:48.24 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 18:51.24 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:52.12 | tuxxie | what it the best way to show sip extension that are inuse? |
| 18:52.21 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 18:52.49 | tuxxie | i have tried sip show inuse but i get more responces than I have extensions. |
| 18:53.49 | tuxxie | I am looking for away to poll extensions to find users statuses. |
| 18:54.58 | carrar | What about monitoring activity via AMI |
| 18:55.27 | tuxxie | i'm running asterisk 1.6.2 |
| 18:55.51 | *** join/#asterisk fross (~fross@68-112-140-187.dhcp.stcd.mn.charter.com) |
| 18:56.16 | carrar | could also parse out core show channels concise |
| 18:56.48 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v013-068.mobile.uci.edu) |
| 18:59.11 | tuxxie | carrar: could you point me in a good direction to find how to monitor activity via AMI? |
| 19:00.05 | carrar | http://tinyurl.com/4jke5rv |
| 19:01.16 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 19:04.21 | *** join/#asterisk philfine (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 19:05.36 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 19:06.28 | titter | Has any one used ConfBridge to host conferences with over 200 participants? |
| 19:08.18 | frigidzephyr | why not set up an audio stream to MOH and just play MOH to those that call in. Unless they actually need to interact |
| 19:08.56 | titter | All participants will need to be able to interact |
| 19:09.00 | frigidzephyr | wow |
| 19:09.05 | frigidzephyr | yeah never tried that |
| 19:09.14 | frigidzephyr | sounds like a noisy conference |
| 19:09.16 | titter | Me either |
| 19:09.31 | frigidzephyr | 200 people talking over each other |
| 19:10.05 | mrfaison | hi *; how can i configure an extension in sccp.conf? what i need, a mac address, line, ip address? |
| 19:10.35 | titter | It will mostly be a muted call, but interaction is required |
| 19:10.42 | drmessano-lt | titter, i've had 10 in a ConfBridge. I would imagine 200 can't be that much different. Well, maybe. |
| 19:10.53 | drmessano-lt | :) |
| 19:12.07 | titter | Ya I just tested with 20 callers, but I am not to keen, or motivated enough to build the dialplan they require. I am thinking I will test app_konference ... just didn't read if it requires dahdi timing or not. |
| 19:14.59 | *** join/#asterisk odin917 (~Gavin_Sil@64.61.105.226) |
| 19:15.32 | odin917 | hi all, i am going to use an asterisk setup with straight SIP trunk connectivity in and out |
| 19:15.53 | odin917 | what is the significance of how many"lines" my voip phones (hardware) can handle |
| 19:17.29 | Mhaddog | odin917: can you be more specific? |
| 19:17.35 | odin917 | yea |
| 19:17.44 | odin917 | im trying to pick some voip phones to order |
| 19:17.49 | odin917 | for our new phone system |
| 19:17.52 | Mhaddog | ok |
| 19:18.16 | Mhaddog | each phone can hold certain amount of lines, usual is a minimum of 2 |
| 19:18.24 | Mhaddog | or one if you go really cheap |
| 19:18.29 | odin917 | like open connections back to my serveR? |
| 19:18.32 | Mhaddog | and then you can take it from there.... |
| 19:18.57 | Mhaddog | each call will requires 2 channels.... |
| 19:19.01 | WIMPy | No, it's the number of accounts it can connect to. |
| 19:19.17 | odin917 | so if i have one extension per person |
| 19:19.22 | Mhaddog | ok... |
| 19:19.23 | WIMPy | On some cheap phones all accounts will have to be on the same server. |
| 19:19.26 | odin917 | a 2 line phone is fine |
| 19:19.40 | Mhaddog | if you want to have 1 line per phone you can |
| 19:19.44 | WIMPy | You'd only need one. |
| 19:19.47 | Mhaddog | you can have more than one too |
| 19:19.52 | odin917 | ok great |
| 19:19.56 | Mhaddog | but with only 1 you will be ok |
| 19:19.57 | Mhaddog | so |
| 19:20.03 | p3nguin | mrfaison: Extensions are configured in extensions.conf, not in sccp.conf. |
| 19:20.03 | odin917 | so it doesnt matter as far as putting ppl on hold etc |
| 19:20.07 | Mhaddog | question is, you going to use voip? |
| 19:20.23 | WIMPy | no, that's another matter. |
| 19:20.27 | Mhaddog | that will depend on the voip provider and plan you get |
| 19:20.28 | p3nguin | mrfaison: sccp.conf is the file for configuring your sccp devices and the lines on those devices. |
| 19:20.37 | odin917 | im doing pbx in a flash |
| 19:20.41 | odin917 | in a co lo |
| 19:20.46 | odin917 | with a mix of sip trunk providers |
| 19:20.48 | mrfaison | p3nguin: hi! do you know how to configure an sccp extension in asterisk? |
| 19:20.56 | Mhaddog | you can get voip plans with only one call per telephone number (did) |
| 19:21.03 | Mhaddog | or multiple incoming or outgoing calls.... |
| 19:21.07 | p3nguin | mrfaison: Did you bother reading what I just said to you? If not, I would advice you to read it now. |
| 19:21.20 | p3nguin | mrfaison: If you don't understand what I said, tell me which part you have a problem with. |
| 19:21.23 | Mhaddog | you do not need to mix, one provider could offer all you need |
| 19:21.43 | Mhaddog | you mix if you want redundancy or some other thing, flexibility or so |
| 19:21.49 | odin917 | yea |
| 19:21.53 | odin917 | just in case one goes down |
| 19:21.57 | odin917 | i need to be able to call them |
| 19:22.00 | odin917 | :) |
| 19:23.04 | mrfaison | p3nguin: actually i've been reading all day long docs |
| 19:23.21 | mrfaison | p3nguin: but i can't configure an sccp extension |
| 19:23.31 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 19:24.37 | mrfaison | p3nguin: can yo help me? my error now is: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SCCP' (cause 44 - Requested channel not available) |
| 19:26.05 | p3nguin | mrfaison: Did you compile, install, and load chan_sccp.so? Did you correctly configure your device(s) and line(s) for your phone(s) in sccp.conf? |
| 19:26.40 | p3nguin | mrfaison: If you don't configure at least a device and a line on it, calling to an extension that dials an sccp device isn't going to succeed. |
| 19:27.21 | mrfaison | p3nguin: chan_sccp.so is loaded |
| 19:27.52 | mrfaison | p3nguin: my devices and lines may be are wrong, here is where i need help |
| 19:28.30 | mrfaison | p3nguin: i have one 7941 device, and i have its mac address |
| 19:28.40 | mrfaison | p3nguin: what else do i need? |
| 19:29.21 | p3nguin | mrfaison: Which sccp version are you using? |
| 19:29.50 | mrfaison | p3nguin: bur80lmt51*CLI> sccp show version |
| 19:29.51 | mrfaison | Skinny Client Control Protocol (SCCP). Release: 3.0 RC3.1 - (built by '' on '2011-02-10 20:43:19 UTC') |
| 19:30.00 | mrfaison | p3nguin: Skinny Client Control Protocol (SCCP). Release: 3.0 RC3.1 - (built by '' on '2011-02-10 20:43:19 UTC') |
| 19:30.42 | p3nguin | mrfaison: Okay, that's the right branch at least. Did you try to use a sample config file and add in your phone's info? |
| 19:32.06 | mrfaison | p3nguin: that is the part where i am getting lost, tell me if i am wrong: [macaddress] section followed with a [line] section, i am not pretty sure what to put there. |
| 19:32.50 | mrfaison | p3nguin: the phone do not need to register to asterisk right? |
| 19:33.55 | p3nguin | mrfaison: Once you have a working sccp.conf, the phone will register with Asterisk and Asterisk will give the phone its information that you configured in sccp.conf. |
| 19:37.20 | p3nguin | mrfaison: I took a sample config from the chan-sccp-b distribution and added my devices and lines to it, using the templates in the sample. When you add your device, it's going to look something like this: |
| 19:37.39 | p3nguin | [SEP001512345678](defaultdevice) |
| 19:37.42 | p3nguin | devicetype = 7941 |
| 19:37.50 | p3nguin | ...add more config lines here... |
| 19:38.39 | p3nguin | mrfaison: That configures the device. Then you have to add at least one line for that device, after any other device sections. |
| 19:39.40 | p3nguin | It might look something like this: |
| 19:39.58 | p3nguin | [001512345678-a](defaultline) |
| 19:40.05 | p3nguin | ...more config stuff here... |
| 19:41.08 | p3nguin | mrfaison: If you are having problems, at least try to do it by yourself first, then you can pastebin the entire sccp.conf for me to look at. |
| 19:44.32 | p3nguin | mrfaison: In the device section, don't forget to assign the line to the device: button = line, 001512345678-a |
| 19:44.44 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
| 19:45.42 | intrin | can someone point me to the ultimate noobs guide for asterisk ? :) |
| 19:46.06 | p3nguin | ~newbook |
| 19:46.06 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 19:46.06 | WIMPy | ~book |
| 19:46.06 | 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/ |
| 19:46.11 | WIMPy | or |
| 19:46.39 | intrin | tyvm |
| 19:47.04 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 19:51.53 | justdave | is there a reliable way to tell from the logs in asterisk which party terminated a call? I'm not seeing anything obvious (other than sometimes I see a "-- Hungup" and sometimes I don't |
| 19:52.02 | leifmadsen | there is not |
| 19:53.02 | justdave | meh. remote user says "my call got dropped" and I can't tell if it was his connection to us or our connection to the phone company that dropped it. oh well |
| 19:53.06 | leifmadsen | see https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010#AstriDevCon2010-%28P3%29 |
| 19:53.22 | *** part/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 19:53.27 | leifmadsen | ~whohungup |
| 19:54.12 | leifmadsen | infobot: whohungup is <reply> The ability to determine reliably who hung up is a feature for a future version of Asterisk, as discussed at AstriCon 2010. Some additional information is available at https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010#AstriDevCon2010-%28P3%29 |
| 19:54.13 | infobot | okay, leifmadsen |
| 19:54.52 | leifmadsen | I would love to update that page with a link to the branch if I knew where it was :) |
| 19:55.47 | russellb | i don't think anyone has done anything. |
| 19:56.30 | WIMPy | What's that Pre-Dial thing? |
| 19:59.21 | *** join/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 19:59.31 | mrfaison | p3nguin: thanks, i am doing that |
| 19:59.34 | *** join/#asterisk sshock (~sshock@sshock.net) |
| 19:59.42 | mrfaison | p3nguin: let me try... |
| 20:00.09 | jimi_ | We are sending Action Originate to the Asterisk AMI. How can I parse back the response from my command only, without getting the responses from other calls? |
| 20:00.11 | sshock | any voip providers out there offer free calls between users? |
| 20:00.51 | jimi_ | sshock, Like mobile to mobile, but for SIP? |
| 20:00.51 | sshock | I know Gizmo did; |
| 20:00.56 | sshock | yeah, exactly |
| 20:00.59 | WIMPy | sshock: I think most do. |
| 20:01.07 | sshock | do you know if flowroute does? |
| 20:01.13 | jimi_ | Flowroute does |
| 20:01.14 | jimi_ | :P |
| 20:01.18 | WIMPy | jimi_: Set an action ID. |
| 20:01.22 | sshock | ok, good; that simplifies things for me |
| 20:01.45 | jimi_ | WIMPy, I still get back all the responses. Is there a method for parsing back just my data? I hate to use a regex for this |
| 20:01.52 | sshock | I want to set up my neighbor with voip, but they don't really need features of a pbx (yet) |
| 20:01.54 | p3nguin | VoIP.ms allows free usage between customers. |
| 20:02.12 | sshock | so I was thinking of just buying an ATA device and calling it good for now |
| 20:02.27 | p3nguin | Set him up with a sub account and an ATA. |
| 20:02.28 | WIMPy | jimi_: Ugh, there was something. events=no? |
| 20:02.49 | *** part/#asterisk sshock (~sshock@sshock.net) |
| 20:02.52 | jimi_ | WIMPy, Check this out. http://pastebin.com/xAPqr74v I did events off |
| 20:03.04 | WIMPy | jimi_: Depends on how much info you want. |
| 20:03.43 | jimi_ | WIMPy, Trying to determine if the local party answered the call or not. basically, we have a forced click to call. The local party has to answer before it rings the remote party. We want to be able to tell if it was answered locally, or time dout. |
| 20:03.58 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:04.16 | WIMPy | jimi_: Better wait for responses. Not doing so can cause the whole thing to fail. |
| 20:04.40 | mrfaison | p3nguin: by the way, one important question, can asterisk behave as a sccp extension? is there any docs? |
| 20:05.38 | jimi_ | WIMPy, Thats what im trying to do. Just trying to grab the data relevant to my action id only |
| 20:05.49 | WIMPy | jimi_: And you're sending events:off with the originate action. That does not look right. |
| 20:06.37 | WIMPy | You don't wait for the login to succeed. That can cause the following action to fail. |
| 20:06.54 | p3nguin | mrfaison: There is no such thing as an "sccp extension" in Asterisk. sccp is a channel technology, and extensions are dialing rules. |
| 20:07.23 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 20:08.13 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 20:08.47 | raden | someone recomend a multi line phone under $100 |
| 20:09.08 | p3nguin | Cisco 7960 |
| 20:09.19 | p3nguin | ebay, $75 |
| 20:09.35 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:10.42 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 20:16.53 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 20:17.22 | *** join/#asterisk NetAtWorld (~support@ool-45794838.dyn.optonline.net) |
| 20:18.25 | mrfaison | p3nguin: can asterisk be configured in order to place calls in a sccp enabled network? |
| 20:18.28 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 20:20.18 | Kobaz | REF:- INSTRUCTION TO CREDIT YOUR ACCOUNT WITH THE SUM OF (US$10.000,000.00) |
| 20:20.20 | Kobaz | oh boy |
| 20:20.34 | drmessano-lt | Count me in |
| 20:20.41 | p3nguin | mrfaison: What did you think the purpose of configuring sccp.conf was? |
| 20:20.41 | Kobaz | Ministry Of finance Of the Federal Republic of Nigeria Forwarded and contracted us to see to your immediate payment of the sum of US$10,000,000.00 |
| 20:20.48 | Kobaz | i'm totally going to be rich now |
| 20:20.57 | drmessano-lt | Oh |
| 20:20.59 | drmessano-lt | Sorry |
| 20:21.14 | jimi_ | WIMPy, Is there a way to tell if a remote party is busy, using originate |
| 20:21.22 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f396812d.cpe.net.cable.rogers.com) |
| 20:21.36 | drmessano-lt | Kobaz, I misread.. I can't accept that large of a sum... anything over $9,999,999 puts me in a different tax bracket |
| 20:21.42 | drmessano-lt | :( |
| 20:21.43 | Kobaz | too bad |
| 20:21.50 | Kobaz | more for me |
| 20:21.58 | carrar | You mean $9.000,000.00 |
| 20:22.06 | WIMPy | Kobaz: What is that Pre-Dial thing you're working on? Any pointer? |
| 20:22.07 | drmessano-lt | I will split it with you, 80/30? |
| 20:22.20 | Kobaz | yeah sure |
| 20:22.30 | WIMPy | jimi_: You get the information in the hangup event. |
| 20:22.38 | Kobaz | if i send you a check for the full amount plus a handling fee |
| 20:22.49 | Kobaz | all you have to do is deposit the amount and then forward me the rest |
| 20:22.57 | Kobaz | WIMPy: interested in using it? |
| 20:23.20 | drmessano-lt | My buddy Hosni has this account in Switzerland I can use |
| 20:23.22 | drmessano-lt | No probs |
| 20:23.24 | jimi_ | WIMPy, Do you know a parser for this, for dealing w/ 1 action id? This is nuts |
| 20:23.59 | WIMPy | Kobaz: It sounds like it might be useful, but I can't find any information except for the name "Pre-Dial". |
| 20:24.13 | WIMPy | jimi_: DIY |
| 20:24.19 | Kobaz | WIMPy: say SIP/100 is Dial()'ing SIP/123 |
| 20:24.32 | WIMPy | But I think I heard about a perl module once. |
| 20:24.42 | Kobaz | you do noop ${CHANNEL} on SIP/100. what do you get SIP/100-000324 something like that right |
| 20:24.53 | Kobaz | how do you get the channel that SIP/123 is using? *before* SIP/123 picks up |
| 20:25.32 | WIMPy | Hmm. That not pre-dial, thet's in-dial. |
| 20:25.54 | Kobaz | well |
| 20:25.54 | Kobaz | i call it predial |
| 20:26.10 | Kobaz | it's like, before the technology is called, but after running Dial |
| 20:26.27 | Kobaz | it gets inbetween so you can run dialplan on SIP/123-000908 before it gets used |
| 20:26.33 | Kobaz | so you can know what channels it goes to |
| 20:27.03 | WIMPy | Will that be like a macro in Dial? Or where do you get that information? |
| 20:27.27 | Kobaz | yeah |
| 20:27.34 | Kobaz | like you do U or M for a pickup macro |
| 20:27.41 | Kobaz | you do I for pred(I)al |
| 20:28.13 | WIMPy | Ok, Thanks for the explanation. |
| 20:50.01 | mrfaison | hi guys!, can asterisk make sccp calls directly to a CiscoCallManager:extension? |
| 20:55.10 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.7.93) |
| 20:55.12 | DelphiWorld | hi all |
| 20:55.17 | DelphiWorld | how to show siptraces? |
| 20:55.22 | DelphiWorld | asterisk 1.8 |
| 20:55.47 | WIMPy | sip set debug |
| 20:56.42 | DelphiWorld | WIMPy: thanks |
| 20:56.48 | DelphiWorld | WIMPy: and to debug for a specific host? |
| 20:57.02 | WIMPy | sip set debug ip ... |
| 20:57.13 | WIMPy | or peer |
| 21:02.46 | DelphiWorld | WIMPy: thank you. |
| 21:03.26 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:11.40 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 21:13.59 | *** join/#asterisk wonderworld (~ww@port-92-201-81-40.dynamic.qsc.de) |
| 21:14.37 | *** join/#asterisk Alric (~nbowyer@64.6.54.218) |
| 21:18.06 | mrfaison | p3nguin: i think, sccp.conf is minded to have asterisk as a pbx wtih several sccp phones. |
| 21:18.25 | mrfaison | p3nguin: connected to it. |
| 21:19.28 | mrfaison | p3nguin: but now, i want that asterisk can be connected to callmanager, i mean callmanager as a pbx and asterisk as a extension |
| 21:19.54 | WIMPy | Use SIP |
| 21:22.03 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 21:23.06 | citywok | Has anybody built a reverse-proxy load balancer with SSL Support & persistent connections, such that all requests from one client go to the same backend webserver? |
| 21:23.13 | citywok | nginx does pretty much all of that minus the persistent part |
| 21:23.35 | *** part/#asterisk Alric (~nbowyer@64.6.54.218) |
| 21:42.20 | *** join/#asterisk matach (~matach@170-117.76-83.cust.bluewin.ch) |
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| 21:52.01 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v013-068.mobile.uci.edu) |
| 21:53.21 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 21:53.52 | LemensTS | Hello. Do non poe IP Phones use 1 pair or 2 pair of wires? |
| 21:54.39 | WIMPy | 2 |
| 21:54.43 | *** join/#asterisk manji (~manjiki@adsl-142-168.adsl.ntua.gr) |
| 21:55.15 | LemensTS | 1,2,3 and 6 wire for T568B then |
| 21:56.55 | *** join/#asterisk xbp (~mod@siphack.com) |
| 21:57.01 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:01.09 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 22:02.34 | nestAr | man, it's been too long since i've messed with polycom phones, they changed a lot of the configs |
| 22:03.23 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 22:03.34 | Katty | hello my asterisk does not work at all how to fix plz????? |
| 22:04.36 | _Corey_ | lol |
| 22:04.37 | DelphiWorld | Katty: kill it |
| 22:04.38 | thehar | lol Katty |
| 22:04.45 | thehar | Katty: keel it with a steeeeeek |
| 22:04.58 | WIMPy | Use violent force. |
| 22:05.10 | Katty | FORCE PUSH |
| 22:05.32 | thehar | git push asterisk > sun |
| 22:05.44 | *** join/#asterisk coppice (~chatzilla@12.69.71.130) |
| 22:06.27 | WIMPy | Use a bigger hammer. |
| 22:07.01 | _Corey_ | cd /etc/asterisk ; rm -rf * |
| 22:07.19 | p3nguin | Why is asterisk taking ownership of /var/run/ and putting its pid and ctl files there rather than using /var/run/asterisk/ like it should? |
| 22:07.52 | digitalirony | any one with basic linux knowledge, and DNS knowledge looking for a job? In the US, we pay relocation if you qualify |
| 22:08.36 | _Corey_ | sounds challenging |
| 22:08.59 | *** join/#asterisk philfine (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 22:09.27 | Katty | i need a new trollline. |
| 22:09.32 | Katty | no one takes me seriously anymore :< |
| 22:09.53 | philfine | Is it possible to create something as a hierarchy of contexts in the dialplan ? |
| 22:09.59 | thehar | Katty: lol |
| 22:10.07 | thehar | hugs Katty |
| 22:10.26 | p3nguin | philfine: That's what you're supposed to do. |
| 22:10.30 | WIMPy | philfine: Use includes |
| 22:10.32 | DelphiWorld | hug both thehar and Katty |
| 22:11.00 | matach | Hi all, In which file in trixbox 2.8 can I set the "bindport" ? in trixbox 2.6 it was just in sip.conf in [general] section ? |
| 22:11.18 | Katty | hugs thehar |
| 22:11.19 | philfine | So there is no way to define it withouth includes |
| 22:11.21 | Katty | hugs DelphiWorld |
| 22:11.22 | Katty | hugs p3nguin |
| 22:11.23 | thehar | aw |
| 22:11.24 | philfine | I mean several files |
| 22:11.25 | thehar | hugs DelphiWorld |
| 22:11.31 | Katty | MOAR HUGS |
| 22:11.36 | Katty | the world will be a better place |
| 22:11.39 | thehar | moaaaarrr |
| 22:11.44 | thehar | i want a shirt that says that |
| 22:11.54 | WIMPy | philfine: You include contexts into others. That's not about files. |
| 22:11.58 | _Corey_ | philfine: You need not necessarily have separate files, but included contexts are the best way |
| 22:12.10 | WIMPy | But you can include files as well. |
| 22:12.15 | philfine | Ok |
| 22:12.18 | Katty | thehar: iwant a vegan zombie shirt |
| 22:12.22 | Katty | thehar: grrrrrraiiiiiiiiinsssssss |
| 22:12.32 | Katty | walks around like zombie |
| 22:12.32 | thehar | lol |
| 22:14.09 | Katty | leif |
| 22:14.11 | Katty | sweety |
| 22:14.14 | Katty | did you start drinking soda |
| 22:14.51 | leifmadsen | pop! |
| 22:15.02 | leifmadsen | I had a Canada Dry today |
| 22:15.11 | Qwell | that's neither canadian, nor dry |
| 22:15.14 | *** join/#asterisk sshock (~sshock@sshock.net) |
| 22:15.16 | Qwell | it's also not soda (or pop) |
| 22:15.18 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 22:15.23 | leifmadsen | ginger ale! |
| 22:15.25 | leifmadsen | good stuff |
| 22:18.26 | *** join/#asterisk Guest60326 (~Geek@unaffiliated/cain) |
| 22:20.57 | *** part/#asterisk sshock (~sshock@sshock.net) |
| 22:21.58 | p3nguin | Disregard, I fixed it. |
| 22:29.24 | saxa | pabelanger: oh ok, thats nice to hear. thx for the info |
| 22:29.33 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 22:30.55 | *** join/#asterisk rlankfo (rlankford@hahainyourface.com) |
| 22:31.22 | rlankfo | i see vitelity now offers sms but it can't be used with web applications. does anyone know of a service that i can use to send sms messages via an application? |
| 22:31.36 | rlankfo | not sure where else to ask... |
| 22:32.02 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.7.93) |
| 22:33.09 | rlankfo | not interested in teleku or twilio.. their rates are ridiculous. |
| 22:36.47 | _Corey_ | I've seen a few wholesale providers of SMS stuff that have APIs... how much is ridiculous? |
| 22:38.04 | rlankfo | _Corey_: do you have links? |
| 22:38.08 | rlankfo | sounds like exactly what i need |
| 22:38.14 | rlankfo | i'd like to pay a flat rate instead of per sms |
| 22:39.21 | rlankfo | the other option i suppose is to get a server colocated with a GSM gateway but that is still extremely limited and expensive |
| 22:39.49 | WIMPy | google sms bulk |
| 22:41.17 | nestAr | any ideas on this one? |
| 22:41.19 | nestAr | [Feb 16 17:36:11] WARNING[3490]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0xba7b8a8 (len 789) to 0.0.0.100:5060 returned -1: Invalid argument |
| 22:42.14 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 22:42.33 | rlankfo | 0.0.0.100 ? |
| 22:42.37 | nestAr | oops.. i didn't have that exten in sip.conf |
| 22:42.38 | nestAr | 100 |
| 22:43.02 | _Corey_ | rlankfo: I'm looking for my bookmark, but I found them by doing what WIMPy suggested... there are a bunmch |
| 22:43.04 | nestAr | added it to sip.conf and reloaded.. error went away |
| 22:43.56 | rlankfo | thanks guys |
| 22:48.28 | *** join/#asterisk sshock (~sshock@sshock.net) |
| 22:55.12 | sshock | I wrote to flowroute support, and they said they don't support free customer-to-customer calling because they don't proxy media. |
| 22:55.43 | sshock | Of course, using * I can simply map a given phone # to a SIP URI. |
| 22:55.51 | WIMPy | One of the negations must be wrong. |
| 22:56.04 | sshock | however, I want to set up my neighbor with just a SIP ATA device, nothing more |
| 22:57.15 | sshock | I may have a solution there though, if the SIP device supports a speed dial feature to dial a sip uri |
| 23:06.27 | nestAr | anyone doing SLA on polycom? |
| 23:11.49 | p3nguin | sshock: (1401.51) <p3nguin> VoIP.ms allows free usage between customers. |
| 23:12.00 | *** join/#asterisk zooz (~zooz@host86-164-219-51.range86-164.btcentralplus.com) |
| 23:17.56 | *** join/#asterisk Tozz_ (Tozz@hardwire.duocast.net) |
| 23:24.31 | sshock | p3nguin: ok, maybe I will have to try out something like VoIP.ms |
| 23:25.04 | p3nguin | sshock: You can set up a sub account for him and give him an ATA, and you'll be set. |
| 23:26.56 | sshock | what if I stay with flowroute and set up my neighbor with VoIP.ms ? |
| 23:27.10 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:28.03 | p3nguin | sshock: Then you'd be on different ITSPs and will not be able to take advantage of free calling between customers. |
| 23:28.55 | sshock | ok |
| 23:30.50 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:32.10 | sshock | so if I have an ATA device like the Linksys PAP2T, can I call a SIP user on that device directly? |
| 23:32.56 | sshock | I'm guessing no since they only work the opposite direction of what I just said |
| 23:34.34 | p3nguin | The ATA is the SIP "user." I don't see any reason you can't call the device directly. |
| 23:35.24 | sshock | hmm, I'm trying to test it now but my mind has left me because I can't figure out what I should dial |
| 23:35.51 | sshock | I'm using an RTP300 right now and it's internal IP address is 192.168.8.242 |
| 23:36.28 | sshock | I want to try calling it from a soft phone by calling something@192.168.8.242, but I'm not sure what to put for something |
| 23:36.40 | WIMPy | nothing? |
| 23:37.49 | sshock | hmm, that never crossed my mind |
| 23:38.18 | sshock | linphone says "not found" |
| 23:39.35 | WIMPy | That might come from the ATA. If sou you will have to find out what it wants. Probably the configured account. |
| 23:40.21 | justdave | does asterisk support PS/911 service on a PRI line? Not having much luck getting such information out of Google |
| 23:40.38 | sshock | ok; currently I have it configured to register two accounts with my * box |
| 23:41.14 | sshock | but for my neighbor's setup, it would just be configured to register directly with the voip provider |
| 23:42.50 | WIMPy | Why not set up a 2nd account with you? |
| 23:43.25 | sshock | hey, I think this is working; I forgot I changed my name from "phillip" to "phillip_home" |
| 23:43.40 | sshock | also, I forgot on my 2nd line it is port 5061 not 5060 |
| 23:43.45 | *** join/#asterisk navaismo (~navaismo@evdomip-203-96-202.iusacell.net) |
| 23:43.53 | sshock | WIMPy: what 2nd account? |
| 23:44.42 | WIMPy | Sorry, I read the two accounts as being on one device. |
| 23:44.42 | sshock | you mean for my neighbor? |
| 23:45.02 | sshock | oh; well, yeah I do have 2 accounts on one device for me |
| 23:45.42 | WIMPy | Ah, ok. So then... |
| 23:46.22 | sshock | My setup is this: 2 analog phone lines => 2 phone jacks on RTP300 ATA device => 2 SIP accounts on * box => 1 SIP account on flowroute |
| 23:46.52 | sshock | for my neighbor I want to do things simpler (he doesn't need a PBX; just cheaper phone calls) |
| 23:47.42 | sshock | so I'll set up my neighbor with: 1 analog phone line => 1 phone jack on an ATA device => 1 account on "voip provider" |
| 23:48.10 | nestAr | already frustrated with these snom m3 units. |
| 23:48.51 | sshock | so just now I was trying to verify how I would call my neighbor's ATA directly, bypassing all VOIP providers, and I think I just verified that I can no problem |
| 23:55.31 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |