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| 00:41.00 | jkroon | hi guys, as of asterisk 1.6.2.19 it seems setvar in chan_dahdi.conf (or users.conf for sections with dahdichan lines) goes missing - can anybody else confirm this? |
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| 00:49.55 | p3nguin | jdoe: Using the same dialing rules, can you send a different caller ID number and complete the call to the same number? |
| 00:52.41 | jkroon | argh, works from chan_dahdi.conf, not from users.conf :( |
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| 00:56.06 | jkroon | ok, i think i've found a bug - anybody around that's more familiar with the code than what I am perhaps that can confirm? it seems setvar gets processed into the confp structure, and then immediately after configuration it's claimed that the function already put it in some pvt - but this only seems to hold for chan_dahdi.conf ... |
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| 01:26.26 | jdoe | p3nguin: yes. |
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| 01:30.52 | jkroon | bug confirmed. and patch created. however, my jira login is still stuffed and a reset doesn't work - is there an admin around that could possibly help me out please? |
| 01:41.54 | *** join/#asterisk box (5006fcbf@gateway/web/freenode/ip.80.6.252.191) |
| 01:42.29 | box | Hello |
| 01:42.56 | box | Is it possible to make Asterisk listen with TCP as well as UDP? |
| 01:43.13 | box | I'm using PBX In A Flash template |
| 01:43.34 | WIMPy | Yes |
| 01:43.48 | WIMPy | If you are talking about SIP. |
| 01:44.03 | box | Yes, for SIP |
| 01:44.55 | WIMPy | tcpenable=yes |
| 01:45.18 | jkroon | for * 1.8 I take it? |
| 01:45.25 | box | It's 1.4 |
| 01:45.44 | box | Sorry, should've mentioned that first |
| 01:45.46 | jkroon | no, time to grab an upgrade :p |
| 01:45.51 | WIMPy | Hmm. I think TCP started with 1.6. |
| 01:46.04 | jkroon | WIMPy, only 1.8 afaik |
| 01:46.39 | WIMPy | 1.8 is the only sensible choice anyway :-) |
| 01:46.42 | jkroon | actually, 1.6.2.19 has tcpenable=yes |
| 01:46.58 | WIMPy | But I'm pretty sure it existed earlier. |
| 01:47.21 | box | WIMPy, jkroon: Ok thank you guys I'll do some research, someone must've released a patch for the older versions:-) |
| 01:47.35 | jkroon | WIMPy, that depends on a LOT of things. <1.6.2.18/19 I would say yes, that's insane. |
| 01:47.56 | jkroon | 1.8 seems to be going very stable very quickly so i'm starting to investigate the upgrade. |
| 01:49.32 | box | I was reading on the Asterisk forums that you shouldn't upgrade if everything is working |
| 01:49.51 | box | Is this true or just an opinion? |
| 01:50.12 | WIMPy | "never change a running system" |
| 01:50.19 | nightrid3r | true as long as you don't need a new feature |
| 01:50.30 | WIMPy | But 1.8 is the only actively supported branch. |
| 01:50.49 | WIMPy | But 1.8 definitely made things better for me. |
| 01:51.07 | WIMPy | Even if there had been IAX issues in the beginnung. |
| 01:57.34 | box | I was just looking at this http://brickybox.com/2007/10/19/sip-tcp-support-in-asterisk |
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| 02:52.12 | tyman | Does asterisk 1.8.4.4 support video using the bria client? If yes, does anyone have a good link for that? |
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| 03:48.54 | afink | I seem to be having a brain lapse, I am positive I wrote custom functions in * 1.4 but I don't remember how. Can anyone point me in the right direction? thanks |
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| 03:52.58 | afink | nm, remembered gosub |
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| 03:56.02 | phatANhappy | anyone on that can help me register a polycom phone? |
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| 04:01.22 | afink | sure |
| 04:01.51 | afink | I have a 550 for reference |
| 04:03.15 | phatANhappy | oh man that would be amazing |
| 04:03.22 | phatANhappy | i have 560's |
| 04:03.29 | phatANhappy | have 3.3.1 loaded |
| 04:03.36 | phatANhappy | have tftp functioning |
| 04:04.18 | phatANhappy | a bit lost on what needs to be in the sip.cfg and phoneEXT.cfg |
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| 04:06.39 | afink | ahh, I see I have only been configuring from the web interface |
| 04:07.18 | titter | You mean sip.conf? and extensions.conf? or are you trying to setup the .cfg files for provisioning? personally I kicked tftp (unreliable over wan) and went with http |
| 04:07.31 | phatANhappy | .cfg files for provisioning |
| 04:08.30 | phatANhappy | As i have learned.... the hard way..... TFTP boogers when trying to transfer files larger then 32meg... |
| 04:08.52 | phatANhappy | safer to use FTP... so i am using FTP - with the DHCP parm to deploy / upgrade local phones |
| 04:09.10 | titter | I haven't gone to 3.3.x yet due to the changes in the files. It's nothing big, the idea is the same, but I still support older 501s, and 500s ... and didn't feel like upgrading all my files (yet). |
| 04:09.32 | phatANhappy | do you think the "conversion" utility will work? |
| 04:09.38 | titter | It should |
| 04:09.47 | titter | I was told to use it by my friend at Polycom |
| 04:10.00 | titter | Just read the new administrators guide |
| 04:10.06 | titter | Their documentation is fairly decent |
| 04:10.11 | phatANhappy | if you are willing to share..... i will plunk it in and give it a try.... |
| 04:10.27 | phatANhappy | AND update you on the functionalty LOL |
| 04:13.38 | titter | Basically, I start with 000000000000.cfg which is the default file the phones look for. This file is application based (application = specific model) In that file, I have the config files section setup as - boot.cfg, phones/provision[MACADDRESS].cfg, phones/sip[MACADDRESS].cfg, phone1.cfg, sip.cfg |
| 04:14.49 | titter | boot.cfg is a custom file I created to setup the boot parameters, https and password info for the phones. it's moslty used for upgrading out of date devices. |
| 04:15.41 | titter | the next two are mac address specific, which is important. this is how I register multiple phones with different credentials, and settings. |
| 04:16.15 | titter | the phones will expand the var [MACADDRESS] to their mac address. so it will search for a file with it's own mac. |
| 04:16.39 | titter | the last two are just the general settings that can be changed across the range of the phones |
| 04:17.08 | atan | http://pastebin.com/5kG1kjYM |
| 04:17.19 | titter | it reads left to right, and anything set in the first files, overwrites information in the later files |
| 04:17.20 | atan | I'm confused why my sip device can't register =\ |
| 04:17.23 | titter | (don |
| 04:18.28 | phatANhappy | i used the 00000.cfg file to ensure all phones have the correct ios / same ios... in this case 3.3.1 |
| 04:19.32 | phatANhappy | i have the mac address files in place.......set to load 2 config_files "x102.cfg, sip.cfg |
| 04:20.00 | phatANhappy | as i believe that is how its supposed to be done... . where x102 -changes per extension - and the sip.cfg stays contant |
| 04:21.06 | titter | phatANhappy: http://pastebin.com/zS1UEKi3 |
| 04:21.19 | titter | that's what is in my provsion[MAC] and sip[MAC] files. |
| 04:21.25 | titter | the sip mac is just for timezone info |
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| 04:22.19 | titter | I personally use the phone's mac address as the username in asterisk with a random generated password |
| 04:22.20 | phatANhappy | WOW - either i am over complicating this --- or i am an idiot.... |
| 04:22.42 | titter | Ya like I said, make your own files with what you need changed on each phone |
| 04:22.53 | titter | 000.cfg will contain all the files you want the phone to read |
| 04:23.02 | titter | let me post that as well |
| 04:23.43 | phatANhappy | i was hoping to find a base config - with most of the standard buttons... will that simple config work? |
| 04:23.47 | titter | http://pastebin.com/tQGXGL7w |
| 04:24.06 | titter | that's the 000.cfg file |
| 04:24.39 | titter | I have to support quite a few different models lol |
| 04:24.42 | phatANhappy | i have that i downloaded with the 3.3.1 code |
| 04:25.05 | titter | correct but notice my changes, the [MACADDRESS] is a big part of making this work |
| 04:25.14 | phatANhappy | i feel for you |
| 04:25.19 | titter | those create the simple per phone files |
| 04:25.48 | titter | then the sip.cfg and phone1.cfg files can be altered, but will change all settings on the remaining phones ... which is fine (excpet for those few custom cases) |
| 04:26.33 | phatANhappy | i got that --- iand i have tried a various phone1.cfg and sip.cfg files ... but have not found a full featured one that works with 3.3.1 |
| 04:26.42 | titter | gotcha |
| 04:27.09 | titter | I haven't gone to the newest firmware yet ... just because I like stability, and what I have is stable lol |
| 04:27.34 | titter | I created a web gui that does most of this work for me |
| 04:28.09 | titter | it spits out my provisioning files, and creates the dialplan/sip.conf additions for me |
| 04:28.24 | phatANhappy | i am not at that level yet... |
| 04:28.28 | phatANhappy | maybe next month... |
| 04:28.38 | phatANhappy | for now.. i need to get 30 phones working... |
| 04:28.57 | titter | Ok they are all 3.3.1? |
| 04:29.01 | phatANhappy | yes |
| 04:29.17 | titter | let me download this new package |
| 04:29.47 | titter | Oh and use the split packages when you can ... makes life much easier. |
| 04:29.59 | titter | Phones download the smaller files much easier lol |
| 04:30.12 | phatANhappy | the bulk of the phones are poly 560's so - can use the same template - and just mod for extension and passwords |
| 04:30.26 | phatANhappy | that must help... process is SLOW |
| 04:31.02 | titter | ya |
| 04:31.57 | titter | Try taking a 500 that is stuck on stuff from 2002 and making it current ... upgrade the bootrom, format, blah blah, reboot, gets new server info, another new bootrom, new app, blah blah wait, then finally it gets the most current stuff. |
| 04:32.09 | titter | Takes like 30 minutes lol |
| 04:34.05 | phatANhappy | i cant wait till that install |
| 04:34.10 | phatANhappy | probably be my next one |
| 04:35.45 | titter | Wow they made this complicated lol |
| 04:36.09 | phatANhappy | then i am only 1/2 crazy |
| 04:36.21 | titter | Looks like phone1 is now features.cfg for the most part |
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| 04:36.30 | titter | sip-interop is the old sip.cfg with all the options |
| 04:36.37 | titter | the others are bit cumbersome |
| 04:37.14 | phatANhappy | so - if i rename sip.interop to sip.cfg ... drop it in place... that should be 70% or better for a full feature set/ config? |
| 04:37.42 | phatANhappy | and if i take features - and mod it - it will work as phone1. and i should be golden? |
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| 04:40.51 | titter | CONFIG_FILES="Config/reg[MACADDRESS].cfg, Config/sip.cfg, Config/features.cfg" |
| 04:40.52 | tainted | anyone know why my callfile defaults to extension:s even though it's specified in the .call ? |
| 04:41.02 | titter | rename sip-interop to sip.cfg |
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| 04:41.32 | titter | take the reg-basic.cfg and full in your info ... and rename it to reg0004F2xxxxxx.cfg |
| 04:41.45 | titter | whatever the mac of the device is |
| 04:42.04 | titter | point the phones via http or ftp to that server, and give it a whirl |
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| 04:43.50 | phatANhappy | i need to mod one of them to define - extension / server data |
| 04:43.59 | phatANhappy | i assume - its reg-basic |
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| 04:45.16 | titter | ya |
| 04:45.42 | titter | http://pastebin.com/zS1UEKi3 -- look at my prov[MAC] for an example |
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| 04:48.22 | phatANhappy | so you are calling 3 files ..... provMAC.cfg reg.cfg and sip.cfg |
| 04:49.17 | titter | I am calling 3 on mine that would equal reg[mac].cfg, sip.cfg, features.cfg for you |
| 04:49.23 | titter | I assume |
| 04:50.27 | phatANhappy | i have now..... copied sip-interop to sip.cfg.... and its loading via the mac access.cfg file |
| 04:50.47 | phatANhappy | also in mac address file = x102.cfg, which = reg.basic |
| 04:51.11 | phatANhappy | next i need to add features.cfg...... |
| 04:51.26 | titter | Should work for 1 phone, but remember with 30 to do, the [MACADDRESS] variable will save you time. |
| 04:51.47 | titter | so x[MACADDRESS].cfg and the file name would be x0004F2xxxxxx.cfg |
| 04:51.59 | titter | whatever the mac is of the phone |
| 04:54.08 | phatANhappy | i am using 0004f....cfg to call ----features.cfg, x102.cfg(reg-basic, Modified), and sip.cfg (sip-interop NOT modified) in that order |
| 04:55.04 | phatANhappy | doing second phone - i was going to copy the x102 to x103 change the login /password and use the macadd2ndphone.cfg |
| 04:55.11 | phatANhappy | lather rinse -repeat |
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| 05:02.53 | phatANhappy | i am not seeing the server config in any of the templates.... |
| 05:02.58 | phatANhappy | i have entension config |
| 05:08.01 | phatANhappy | only site.cfg - has the reg.1.server.1.address information in ti |
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| 05:09.56 | titter | you can add it to any file |
| 05:10.00 | titter | well somewhat |
| 05:10.03 | titter | you can make your own |
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| 05:28.10 | phatANhappy | we ... i made a mess... and its still not working lol |
| 05:28.30 | phatANhappy | i added a the files together - it registers - it dials - but it doent allow you to answer |
| 05:28.58 | phatANhappy | * - you can pick up - but its dead air |
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| 06:25.09 | phatANhappy | anyone with thoughts on why i get dead air? |
| 06:25.45 | irroot | phatanhappy ... nat ?? codec mismatch ?? codec formats not loaded ?? |
| 06:26.03 | phatANhappy | i am at a loss.... |
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| 06:26.26 | phatANhappy | i am struggling to provision polycom 560 phones |
| 06:26.37 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:26.42 | irroot | also try sip set debug |
| 06:26.47 | irroot | get debuging info |
| 06:26.58 | phatANhappy | kk rebooting asterisk server |
| 06:42.42 | phatANhappy | sip phones are registered. |
| 06:42.58 | phatANhappy | internal extension 102 dials - 103 but you cant answer 103 |
| 06:43.00 | phatANhappy | dead air |
| 06:43.10 | phatANhappy | all on the same subnet. |
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| 06:46.52 | phatANhappy | its not allowing to pick up |
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| 06:52.03 | CRCinAU_ | does sip.conf support templates in 1.6.x like in 1.8.x? |
| 06:52.32 | phatANhappy | i dont know.... |
| 06:52.51 | phatANhappy | are you asking? or responding to myquestion |
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| 06:54.48 | irroot | morning jkroon phatanhappy you got a trace ?? debug |
| 06:55.03 | phatANhappy | what do you want me to send? |
| 06:55.12 | CRCinAU_ | I just configured up a 1.8.4.4 and it seems to keep the line in a weird state after missing a call |
| 06:55.15 | gravin | hi |
| 06:55.21 | CRCinAU_ | ie the line light keeps flashing with no call :\ |
| 06:55.34 | gravin | anyone experienced with clustered asterisk setup |
| 06:55.44 | gravin | using drbd & heartbeat |
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| 07:04.55 | gravin | any guide i can follow |
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| 07:09.26 | jkroon | irroot, 1.6.2.19 contained a fix for chan_local thanks. |
| 07:09.50 | jkroon | picked up a few other issues but already cooked up two patches for those. |
| 07:09.58 | irroot | cool |
| 07:10.02 | jkroon | they weren't as involved. |
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| 07:10.07 | jkroon | chan_dahdi stuff. |
| 07:10.36 | irroot | need something to warm up .... go home early whack the zoempoeswyn |
| 07:11.04 | jkroon | hehe |
| 07:11.27 | jkroon | ok, take a wack at back-porting your T.30<->T.38 patches for ast 1.8 for me. |
| 07:11.35 | irroot | lolo |
| 07:11.45 | irroot | lol that been done backport on jira |
| 07:11.48 | jkroon | and if that's too easy, a way to "listen in" on the transmitted fax would be cool. |
| 07:12.01 | irroot | nou vat jy kans |
| 07:12.11 | jkroon | ie, I want to "Monitor()" a T.38 conversation into a .tiff file :p |
| 07:12.15 | jkroon | ek weet :) |
| 07:15.37 | *** join/#asterisk Aven (~user@95.52.229.140) |
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| 07:20.07 | jkroon | irroot, i do have one other issue i just picked up. |
| 07:20.21 | irroot | want my shrinks no ?? |
| 07:20.21 | jacc0 | goede morgen :) |
| 07:20.23 | jkroon | app_queue, using leastrecent strategy doesn't seem to honor the penalty values at all. |
| 07:20.31 | irroot | jacc0 goie more |
| 07:20.50 | jkroon | the dutch/german/afrikaans in this channel is certainly picking up. |
| 07:21.04 | jacc0 | ;) |
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| 07:26.53 | coppice | jkroon: there are legal implications about FAX snooping in many places, so people are reluctant to help you with that |
| 07:27.49 | irroot | jkroon riaan meades from miro says hi he is working with us |
| 07:28.17 | jkroon | coppice, just because it's in "many places" surely shouldn't be a restriction to build the functionality? in our case the client gives explicit permission, so it's not a problem. |
| 07:28.42 | jkroon | irroot, say hi to riaan for me :) |
| 07:30.11 | coppice | you can find a couple of snooping packages for sale, but the ones I have seen are only offered for sale to law enforcement people |
| 07:30.57 | coppice | I have a crude decoder in spandsp, which I use for debug and analysis, but so far I have refused to turn it into a proper monitoring facility |
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| 08:08.45 | hrolf | Is there any limit to the number of SIP accounts we can register from a single IP? |
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| 08:17.12 | irroot | hrolf not that im aware of you will probably need to look at the "insecure" option |
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| 08:40.25 | hrolf | irroot: the 'insecure' option? where is it? |
| 08:40.30 | hrolf | irroot: sip.conf? |
| 08:42.39 | irroot | ;insecure=port ; Allow matching of peer by IP address without |
| 08:42.41 | irroot | <PROTECTED> |
| 08:42.43 | irroot | in sip.conf |
| 08:43.06 | irroot | insecure=port,invite |
| 08:43.23 | irroot | will allow multiple registrations better |
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| 09:26.26 | jacc0 | I'm about to release a patch for an extra param for app_originate :) |
| 09:26.47 | jacc0 | it makes app_originate more usefull |
| 09:27.12 | jacc0 | it will enable you to set a timeout other then the hardcoded 30 seconds |
| 09:27.15 | jacc0 | :) |
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| 09:45.21 | jacc0 | from the overwelming responces I can see everybody has been waiting for this one ;) |
| 09:45.34 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 09:45.48 | mattwj2002 | hi all |
| 09:47.23 | jacc0 | Hi mattwj2002 |
| 09:48.08 | jacc0 | I guess you are comming to hear everthing about the new app_originate patch :P |
| 09:48.40 | mattwj2002 | no but tell me more :) |
| 09:49.00 | jacc0 | I'm about to release a patch for an extra param for app_originate :) |
| 09:49.05 | jacc0 | it makes app_originate more usefull |
| 09:49.09 | jacc0 | it will enable you to set a timeout other then the hardcoded 30 seconds |
| 09:49.14 | jacc0 | :) |
| 09:49.22 | mattwj2002 | I was just coming to praise CallCentric |
| 09:49.23 | mattwj2002 | nice |
| 09:51.41 | mattwj2002 | there prices seem good and I like their features and what not.....the biggest test will be their call quality |
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| 09:58.32 | nightrid3r | mattwj2002 if you like those prices maybe you shoud check store.2600hz.com |
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| 10:00.49 | mattwj2002 | nice |
| 10:00.53 | mattwj2002 | how is the quality though |
| 10:00.55 | mattwj2002 | ? |
| 10:02.00 | jacc0 | https://issues.asterisk.org/jira/browse/ASTERISK-17015?focusedCommentId=179575 |
| 10:02.01 | CaptainPants | [ASTERISK-17015] [Status: Open] [patch] const timeout = 30 makes app_originate pretty useless - https://issues.asterisk.org/jira/browse/ASTERISK-17015 |
| 10:02.04 | jacc0 | there it is |
| 10:03.03 | nightrid3r | mattwj2002 no idea about quality, you'll need to test them |
| 10:03.31 | mattwj2002 | nightrid3r: can these be used for personal use? |
| 10:04.01 | nightrid3r | i guess so, contact them to find out |
| 10:04.44 | mattwj2002 | ok |
| 10:05.04 | mattwj2002 | their prices are dang good though |
| 10:05.05 | mattwj2002 | :) |
| 10:05.53 | jacc0 | please stop sending the tank-you cards; the mailman has been complaining about it to me :P |
| 10:06.30 | nightrid3r | they aim at the big accounts but as a startup they probably don't mind personal accounts, besides most of them are (ex) asterisk contributors |
| 10:10.27 | mattwj2002 | okay :) |
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| 10:46.25 | teloniusz | hi. I have two friend peers and I'm doing call transfer from one to another. I'd like to allow the caller to keep its original CID. Is it done by default or should I put something special into exptensions.conf near Dial()? |
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| 12:02.10 | *** join/#asterisk gravin (~gravin@175.139.237.155) |
| 12:02.16 | gravin | hi |
| 12:02.48 | gravin | I'm trying to setup an inbound DID to all the conference rooms i have |
| 12:02.53 | gravin | asterisk 1.6 |
| 12:03.09 | gravin | exten => s,1,Answer |
| 12:03.09 | gravin | exten => s,n,read(CONFNUM,/var/lib/asterisk/sounds/conf-getconfno) |
| 12:03.09 | gravin | exten => s,n,GotoIf($[ ${CONFNUM} : â776[0-3]â ]?ext-meetme,${CONFNUM},1:) |
| 12:03.10 | gravin | exten => s,n,CBMysql(${CONFNUM}|) |
| 12:03.12 | gravin | exten => s,n,goto(custom-meetme3,s,1) |
| 12:03.31 | gravin | i followed this but if i enter the conf room, it says invalid |
| 12:07.11 | jacc0 | @gravin:what is the exact error? |
| 12:07.30 | gravin | ok hold on |
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| 12:10.47 | gravin | [Jul 6 20:10:02] VERBOSE[4235] file.c: -- <SIP/mobiweb-did-00000034> Playing '/var/lib/asterisk/sounds/conf-getconfno.gsm' (language 'en') |
| 12:10.47 | gravin | [Jul 6 20:10:11] VERBOSE[4235] app_read.c: -- User entered '7171' |
| 12:10.47 | gravin | [Jul 6 20:10:11] WARNING[4235] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '<token>', expecting $end; Input: |
| 12:10.48 | gravin | <PROTECTED> |
| 12:10.50 | gravin | <PROTECTED> |
| 12:10.52 | gravin | [Jul 6 20:10:11] WARNING[4235] ast_expr2.fl: If you have questions, please refer to doc/tex/channelvariables.tex. |
| 12:10.54 | gravin | [Jul 6 20:10:11] VERBOSE[4235] pbx.c: -- Executing [s@custom-meetme3:3] GotoIf("SIP/mobiweb-did-00000034", "0?ext-meetme,7171,1:") in new stack |
| 12:10.56 | gravin | [Jul 6 20:10:11] VERBOSE[4235] pbx.c: -- Executing [s@custom-meetme3:4] CBMysql("SIP/mobiweb-did-00000034", "7171,") in new stack |
| 12:10.58 | gravin | [Jul 6 20:10:11] VERBOSE[4235] file.c: -- <SIP/mobiweb-did-00000034> Playing 'conf-invalid.gsm' (language 'en') |
| 12:11.00 | gravin | [Jul 6 20:10:15] NOTICE[4235] app_cbmysql.c: getConf: 0 |
| 12:11.02 | gravin | [Jul 6 20:10:15] NOTICE[4235] app_cbmysql.c: No User or Admin passwords |
| 12:11.04 | gravin | [Jul 6 20:10:15] VERBOSE[4235] config.c: == Parsing '/etc/asterisk/meetme.conf': [Jul 6 20:10:15] VERBOSE[4235] config.c: == Found |
| 12:11.06 | gravin | [Jul 6 20:10:15] VERBOSE[4235] config.c: == Parsing '/etc/asterisk/meetme_additional.conf': [Jul 6 20:10:15] VERBOSE[4235] config.c: == Found |
| 12:11.08 | gravin | [Jul 6 20:10:15] NOTICE[4235] app_cbmysql.c: Maximum users exeeded, Current users in room: 0 |
| 12:11.10 | gravin | [Jul 6 20:10:15] VERBOSE[4235] file.c: -- <SIP/mobiweb-did-00000034> Playing 'conf-locked.gsm' (language 'en') |
| 12:11.12 | gravin | [Jul 6 20:10:17] NOTICE[4235] app_cbmysql.c: checkMax: -1 |
| 12:11.14 | gravin | [Jul 6 20:10:17] VERBOSE[4235] pbx.c: == Spawn extension (custom-meetme3, s, 4) exited non-zero on 'SIP/mobiweb-did-00000034' |
| 12:11.34 | WIMPy | gravin: Use a pastebin |
| 12:12.37 | gravin | ok sorry |
| 12:12.38 | gravin | http://pastebin.com/9NjGJsCx |
| 12:14.08 | irroot | back at the desk gravin need dialplan snippet too |
| 12:14.30 | gravin | ok |
| 12:15.18 | gravin | context --> http://pastebin.com/vDsLxfUD |
| 12:15.31 | gravin | the conf room is 7171 |
| 12:17.43 | gravin | at first it says invalid conf number, then this conf is loacked |
| 12:17.47 | gravin | locked* |
| 12:18.19 | jacc0 | there is a syntax error in : s,n,GotoIf($[ ${CONFNUM} : â776[0-3]â ]?ext-meetme,${CONFNUM},1:) |
| 12:18.24 | irroot | gravin line 4 of exten broke |
| 12:18.25 | jacc0 | what is it supposed to do? |
| 12:18.35 | irroot | what jacc0 said |
| 12:19.00 | gravin | im not very well verse, i got it from freepbx website |
| 12:19.16 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:19.16 | gravin | http://www.fonality.com/trixbox/forums/trixbox-forums/help/setting-single-inbound-conference-bridge-number-multiple-meetme-conferenc |
| 12:19.17 | nightrid3r | the conf room is 717 but you have 776 |
| 12:19.41 | gravin | at first, but i changed it to 717[0-3] |
| 12:19.43 | gravin | already |
| 12:19.56 | nightrid3r | ok |
| 12:19.58 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 12:20.32 | gravin | fonality website* |
| 12:21.28 | gravin | jacc0 : it is said to check for conference room and then ask for passwd |
| 12:25.52 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 12:26.02 | gravin | any idea... |
| 12:26.26 | *** join/#asterisk SpiderMon (~SpiderMon@68.152.22.33) |
| 12:26.30 | SpiderMon | gm |
| 12:31.33 | jacc0 | s,n,GotoIf($[ "0${CONFNUM}" = "0776" ]?ext-meetme,${CONFNUM},1) |
| 12:31.41 | jacc0 | might be working better |
| 12:32.05 | gravin | it will allow 7171 |
| 12:32.18 | gravin | oh ok |
| 12:32.21 | gravin | let me test |
| 12:33.10 | gravin | but i need several conf in one inbound DID |
| 12:33.23 | gravin | this syntax will only allow one right... |
| 12:33.24 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176141096.dsl.bell.ca) |
| 12:34.41 | gravin | s,n,GotoIf($[ "0${CONFNUM}" = "7171" ]?ext-meetme,${CONFNUM},1) |
| 12:34.54 | gravin | i used this, and entered 7171, it still says invalid conf number |
| 12:35.19 | gravin | from my IP-phone i can dial ext 7171 and enter the conf |
| 12:35.48 | gravin | if i route the incoming DID to conf 7171, it works also |
| 12:38.45 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
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| 12:42.50 | gravin | yes! |
| 12:42.53 | gravin | i solved it |
| 12:43.00 | nightrid3r | nice |
| 12:44.01 | gravin | working dialplan ---> http://pastebin.com/qiqYHCZj |
| 12:44.25 | gravin | just used MeetMe instead of CBMysql |
| 12:46.38 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:47.49 | nightrid3r | gravin put the solution in the asterisk cookbook |
| 12:49.43 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:50.38 | gravin | hi.. ok |
| 12:50.52 | gravin | you mean the forum? |
| 12:52.02 | nightrid3r | oh they already finnished the cookbook project |
| 12:52.16 | nightrid3r | then put it in the forum |
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| 12:53.48 | gravin | ok will also blog it |
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| 12:57.58 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:58.40 | saliak | I'm trying to forward calls from my google voice number into asterisk. for some reason it doesn't seem to be answering correctly. at my asterisk prompt I can see the call come in, main menu play, etc. but on the phone I'm calling from, it just rings. anyone had luck with a setup like this? i've turned off call screening so i don't think GV should be waiting for anything.. |
| 12:59.16 | *** join/#asterisk AVGP (~AVGP@hrmatching1.epfl.ch) |
| 12:59.21 | AVGP | Hey there |
| 12:59.35 | AVGP | I am confused. |
| 13:00.19 | *** join/#asterisk radic (~radic@tmo-096-10.customers.d1-online.com) |
| 13:00.37 | AVGP | I'm running Asterisk 1.8 on a debian |
| 13:00.52 | Chainsaw | AVGP: How's that working out for you? |
| 13:00.57 | AVGP | Pretty good |
| 13:01.08 | AVGP | but: Where the hell is the asterisk-gui storing the configs? |
| 13:01.25 | AVGP | Because I setup a trunk via the gui - and its not in /etc/sip.conf oO |
| 13:01.36 | Chainsaw | AVGP: Ah, I don't use the GUI myself. But perhaps it is using Asterisk in realtime mode? |
| 13:01.49 | Chainsaw | AVGP: In that case all the config is in a database of your choosing (MySQL in most cases). |
| 13:01.49 | AVGP | mh maybe. |
| 13:01.55 | AVGP | Anyway - I've got another question |
| 13:01.57 | nightrid3r | AVGP look in users.conf |
| 13:02.33 | AVGP | We've got a bunch of VoIP-Numbers from a swiss registrar (NetStream) and are signed up for an IP-connect trunk |
| 13:02.41 | AVGP | ah users.conf, okay |
| 13:03.12 | AVGP | the question I have is: Inbound calls are working like a charm from every number we've set up with NetStream |
| 13:03.36 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 13:03.38 | AVGP | What I wanna do now seems simple: I want to set the caller-ID on outbound calls apropriately |
| 13:04.31 | AVGP | Do I have to setup every number seperately as a context to use (so that I would have 1 phone per Context?) or can't I just set the Caller-ID using SetCallerID(...)? |
| 13:04.37 | AVGP | the latter seems not to work :( |
| 13:05.05 | Chainsaw | AVGP: Netstream might be limiting you in what caller IDs you're allowed to set. How are you testing these? |
| 13:05.50 | eduzimrs | im using * with sip realtime peers, * is running in a server different from da peers database. So when the server thats running the DB fall down, all peers get diconnect from *. Anyone knows ? |
| 13:05.52 | Chainsaw | AVGP: (Generally if you can prove ownership of a number block; they are then quite willing to widen the filter for you. But without prior agreement, expect to be limited) |
| 13:06.06 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 13:06.35 | AVGP | Chainsaw: Thats exactly what I encounter |
| 13:06.41 | AVGP | the problem is: The support is pretty stupid |
| 13:07.08 | AVGP | on my request, how to get the Outbound Caller-ID as expected, they linked me to a setup-guide for starface-gui -.- |
| 13:07.17 | AVGP | seriously wtf |
| 13:07.28 | Chainsaw | AVGP: I would start by setting exactly the number you have bought from them. |
| 13:08.06 | Chainsaw | AVGP: To see if that gets through on outbound calls. If it does, you are using the correct method. You can then have a more persuasive argument for the phone monkey and its flowchart. |
| 13:08.36 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mfnfgwnafdsbgvjp) |
| 13:08.53 | AVGP | Yeah we have numbers like xxxxxxxxxxx00, xxxxxxxxxxx01 and so on |
| 13:09.04 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 13:09.16 | AVGP | currently I am connected to their service using 00 only - and thats the CallerID its transmitting |
| 13:09.29 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-246-62.w81-51.abo.wanadoo.fr) |
| 13:09.30 | AVGP | ignoring that in the dialplan, I am specifying 01 etc. |
| 13:09.58 | AVGP | I had hoped, that it would preceed the fromuser-setting used to authenticate with them |
| 13:10.39 | Chainsaw | I just set ${CALLERID(num)} before my call goes out. |
| 13:10.50 | Chainsaw | And name too, but that doesn't carry over ISDN. |
| 13:11.19 | AVGP | as we're going through SIP, that should carry out |
| 13:11.26 | AVGP | Shit. |
| 13:13.10 | Chainsaw | AVGP: And it probably does, until it hits Netstream's filter. If they don't like what you set, it will set the main number. |
| 13:13.21 | WIMPy | Chainsaw: Well, yes and no. |
| 13:13.22 | Chainsaw | AVGP: This is why I have my ISDN hand-off in-house, so I have a bit more control. |
| 13:14.08 | Chainsaw | transfers the call to WIMPy for further elaboration |
| 13:14.21 | WIMPy | sr had the issue last week that he couldn't call out, because the caller names were sent as display IEs. |
| 13:14.53 | WIMPy | And on the other hand you could set the subaddress the way HiPaths do. |
| 13:14.55 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 13:15.00 | WIMPy | If you are able to send them. |
| 13:15.27 | AVGP | Hm, maybe it fails because of the number format. Will play a bit with this first, I think |
| 13:16.26 | WIMPy | The format is definitely important if you send a type (numberplan in Asterisk terms). If not, it still might be importantn. |
| 13:16.59 | AVGP | Silly me... I had id commented out. |
| 13:17.21 | AVGP | Ah and now I know why - it rejects the call immediately, when I set the CallerID differently |
| 13:17.41 | AVGP | even if I set it to ...00, what I am authed with. So its the format, I guess |
| 13:18.12 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:18.26 | WIMPy | You could try "V00". |
| 13:18.36 | *** join/#asterisk sourcode (~code@ppp-58-8-94-161.revip2.asianet.co.th) |
| 13:18.51 | AVGP | Isn't it SetCallerID("xxxxxxxxxx00") etc.? |
| 13:19.02 | WIMPy | See the sample chan_dahdi.conf for those kludges. |
| 13:19.07 | AVGP | okay |
| 13:19.19 | WIMPy | Set(CALLERID(num)=V00) |
| 13:20.23 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 13:21.49 | AVGP | mh. nope, then its just setting the main-number as CallerID |
| 13:22.26 | AVGP | I tried Set(CALLERID(num)=.....01) now - still reaching the destination as ....00 :( |
| 13:23.23 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 13:23.45 | WIMPy | That is a DDI block? |
| 13:23.59 | AVGP | I am pretty new to VoIP and Telephony |
| 13:24.02 | AVGP | whats a DDI-Block? |
| 13:24.21 | AVGP | ah yes |
| 13:24.28 | WIMPy | A block of numbers, not a set of single numbers. |
| 13:24.52 | AVGP | I don't know - the numbers are only varying in the last 2 digits |
| 13:25.01 | AVGP | I guess (and hope) we got them as a block |
| 13:25.13 | AVGP | The contract I have here isn't saying anything about it |
| 13:25.33 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 13:25.46 | AVGP | It - somehow - is possible. I got pointed to a guide for setting up starface via gui with this |
| 13:25.50 | WIMPy | I've been told that you can't configure more than 16 numbrs per line, so that has to be a block. |
| 13:25.53 | AVGP | but we don't have (and don't want) Starface. |
| 13:35.17 | AVGP | this is frustrating |
| 13:42.53 | jacc0 | AVGP: try removing country code |
| 13:43.06 | jacc0 | AVGP: try removing 0s |
| 13:43.16 | jacc0 | AVGP: try removing loca area code |
| 13:43.16 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:43.27 | jacc0 | AVGP: try a mixure of these |
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| 13:47.06 | jacc0 | AVG: and at last try calling your provider :) |
| 13:48.05 | nightrid3r | and YELL at him :) |
| 13:48.40 | AVGP | :D |
| 13:48.42 | *** join/#asterisk Xaviertoor (~Xaviertoo@187.6.117.192) |
| 13:49.00 | AVGP | I'm never yelling at people who show they're incapable of handling things... |
| 13:49.12 | AVGP | I'm just leaving them alone, then. We consider changing the provider :D |
| 13:52.21 | jacc0 | tell me about it; yesterday there where 3 so called "network engeneers" at a customers side. all 3 of them did not understand what I was saying when I told them there was no way of getting to a nother subnet if the provided gateway ip is not available |
| 13:52.29 | WIMPy | In some places you use just the last 4 digits. |
| 13:54.22 | nightrid3r | jacc0 your hiring ? :) |
| 13:55.12 | Faustov | anyone from Lon? which major datacenter had a power failure today? |
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| 13:55.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:55.59 | jacc0 | nope, I'm just an employe |
| 13:57.45 | nightrid3r | damn |
| 13:58.12 | merlin8282 | How about emergency calls via a sip provider ? Do they all route 911/112 ? And How is it when channels are all used, does ChanIsAvail work ? |
| 13:58.16 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:58.33 | Kobaz | you'll have to keep track of the channel limit yourself |
| 13:58.47 | Kobaz | chanisavail won't tell you if the provider will reject new calls |
| 13:59.13 | merlin8282 | Kobaz: ok, do you have a dialplan example ? |
| 14:00.07 | Kobaz | use something like Set(GROUP=pstn); if ${GROUP_COUNT(pstn)} > 10... then playback all circuits busy, and Busy() |
| 14:00.51 | AVGP | Okay - ugly fix: I just have a peer for everynumber called netstreamout<DID> |
| 14:01.32 | AVGP | and an extension going Call(SIP/${EXTEN}@netstreamout${CALLERID(num)}) which works. |
| 14:01.37 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:01.45 | AVGP | doesn't look beautiful in the configs but ... the fuck - it works. |
| 14:09.21 | *** join/#asterisk serafie (~erin@nat/digium/x-fzrbtgcdsnlhdxwc) |
| 14:11.13 | AVGP | okay, thank you all for your patience with a noob like me and your help :) |
| 14:11.46 | merlin8282 | Kobaz: ok, thanks |
| 14:11.58 | d_preston215 | Has anyone got this error dealing with the phone directory: |
| 14:12.01 | d_preston215 | PHP Fatal error: Call to undefined method DB_Error::setFetchMode() in /var/www/html/xmlservices/include/xmlservices_libs.php on line 96 |
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| 14:22.13 | _Corey_ | d_preston215: "core show application directory" for information about the phone directory, #FreePBX for everything else |
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| 14:26.21 | jacc0 | <PROTECTED> |
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| 14:28.19 | d_preston215 | Ok |
| 14:30.25 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 14:30.56 | d_preston215 | Its weird how I can create a php script to connect to the asterisk DB, but that's still messing up. |
| 14:31.40 | kuku | Question: Using XO SIP as a trunk. Everything works fine EXCEPT, when I want to dial from an internal extension, using a different DID than the main DID, to an internal DID. Getting: Failed to authenticate device NOTICE |
| 14:33.00 | d_preston215 | PHP Notice: Undefined index: main in /var/www/html/cisco/services/rss/lib/xtpl.php on line 110 |
| 14:33.06 | d_preston215 | Also got that too. |
| 14:33.20 | d_preston215 | PHP Notice: Undefined index: NextGroup in /var/www/html/xmlservices/PhoneDirectory.php on line 28 |
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| 14:44.33 | *** join/#asterisk takamarou (4b9196ae@gateway/web/freenode/ip.75.145.150.174) |
| 14:44.41 | takamarou | Hi - can anyone help me? I've just loaded up a new trixbox and registered with my VOIP provider (sip). When I try to make outbound calls I am getting the "All circuits are busy now. Please try again later" error. |
| 14:46.33 | irroot | ~trixbox |
| 14:46.34 | infobot | methinks trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 14:46.54 | takamarou | irroot, thanks. That helped me fix my problem. |
| 14:47.29 | tzanger | takamarou: is there a trixbox support channel? |
| 14:47.52 | irroot | #trixbox |
| 14:47.58 | takamarou | tzanger, very few people on it. I would imagine my issue is not actually with trixbox, more likely with PBX |
| 14:48.10 | tzanger | as a general rule closed systems are not supported here for the simple reason that we cannot effect any change, fix any bugs or address any issues. |
| 14:48.58 | takamarou | Would it not be a more useful community if people at least attempted to help? It seems counterproductive to say GTFO. |
| 14:49.19 | jacc0 | does anyone have experiance with handelin SOCA alarms with app_alarmreciever? |
| 14:49.27 | jacc0 | *handeling |
| 14:49.27 | irroot | ~ask |
| 14:49.27 | 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. |
| 14:49.30 | Qwell | takamarou: You aren't using Asterisk though. You might as well come here asking for help with a Microsoft product. |
| 14:49.59 | takamarou | @Qwell, doesn't trix utilize asterisk? Perhaps I'm confused about how these things work |
| 14:50.02 | jacc0 | does it work? |
| 14:50.06 | Qwell | It "uses" "Asterisk". |
| 14:50.21 | tzanger | takamarou: trixbox has *the* shittiest dialplan and logic I've ever seen. I refuse to even try to figure it out on that basis alone |
| 14:50.25 | Qwell | It's some really old fork version they patched to hell. |
| 14:50.35 | irroot | its a branch / fork / frankenasterisk |
| 14:51.08 | takamarou | OK. So perhaps a better question, is what should I use instead? I would prefer not to pay for a solution, but my boss will not be very happy if there isn't some usable frontend on this system. Is there an alternative to trix that doesn't suck? |
| 14:51.24 | tzanger | that's the holy grail |
| 14:51.31 | takamarou | :) |
| 14:51.39 | irroot | thinks he has a funky frontend :P |
| 14:51.40 | tzanger | free, easy to use and free support |
| 14:51.42 | takamarou | OK, perhaps one that only kind of sucks? |
| 14:52.00 | irroot | undoes zip to check |
| 14:52.16 | _Corey_ | takamarou: http://www.asterisk.org/asterisknow/ |
| 14:52.21 | Qwell | takamarou: AsteriskNOW would be vastly superior. Asterisk with your own configs would be best though. |
| 14:52.36 | tzanger | I haven't used asterisknow in forever. I should try it out again |
| 14:52.39 | irroot | takamarou i have a basic web gui we ship customised mostly for south africa there lots of priojects |
| 14:52.43 | *** part/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 14:52.52 | takamarou | Sounds like AsteriskNOW it is, then. Honestly, my boss won't give me the time of day if I show him a CLI |
| 14:53.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 14:53.34 | tzanger | takamarou: that is understandable. you may have to explain to him that he might have to pay money to have a pointy-clicky interface. |
| 14:53.39 | tzanger | or at least one that is supported |
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| 14:54.14 | Qwell | takamarou: If you're able to spend money, just go with Switchvox |
| 14:54.56 | tzanger | I thougth digium bought switchvo a long time ago |
| 14:55.02 | takamarou | @Qwell, that is a very likely option - my preferred option really. But, for it's similar money to go with Switchvox or some hosted solution that we never have to touch. |
| 14:55.33 | Qwell | tzanger: yes |
| 14:55.56 | takamarou | ehh.. that was bad english. We would prefer to go with a hosted solution so we never have to touch it. If the cost is comparable, it'll be hard to justify anything else. |
| 14:56.33 | Qwell | takamarou: hosted stuff is a bit scary. Your entire system goes down (including in-office calls) if your Internet connectionhas any issues. |
| 14:57.03 | Qwell | plus you're stuck with voip, instead of something like PRI |
| 14:57.19 | takamarou | The one we are looking for allows us to redirect to cellphones if our net connection fails. But, if we use an internal Asterisk server we lose our phones if the net dies anyways, right? |
| 14:57.31 | Qwell | You don't lose in-office calling. |
| 14:57.39 | takamarou | right |
| 14:57.42 | Qwell | plus, if you're using something like a PRI, like I mentioned, then...no. |
| 14:58.26 | Qwell | phone companies have much much higher SLAs than internet providers |
| 14:58.31 | Qwell | (there's a reason for that) |
| 14:58.48 | tzanger | Qwell: because the charge more? :-) |
| 14:58.55 | Qwell | tzanger: perhaps |
| 14:59.05 | jaytee | we use Flowroute as an ITSP and we can set a failover route for any DID number we have with them to either a phone number or an IP address. for most of the DID numbers in our office we failover to cell phone numbers so if our net or pbx goes down we still get calls. |
| 15:00.30 | takamarou | jaytee, just signed up for FlowRoute. Nice to know that they offer failover routes |
| 15:01.22 | jaytee | IMHO they have solid service, easy to use management portal and very low rates for metered usage, especially toll free numbers. |
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| 15:07.48 | jacc0 | does app_alarmreciever support SIA-HS as well as SIA ? |
| 15:09.46 | jacc0 | or only SIA? |
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| 15:17.15 | *** join/#asterisk tekzilla (~jon@hmbg-5f767991.pool.mediaWays.net) |
| 15:18.09 | tekzilla | can i run AGI scripts from the CLI for testing purposes ? |
| 15:20.47 | irroot | tekzilla why not run them on a local channel in a new context i do quite a bit of testing that way |
| 15:21.13 | irroot | using originate or dial you can do it from cli |
| 15:22.33 | tekzilla | sorry my asterisk knowledge is very limited, havent set up the server myself |
| 15:23.21 | tekzilla | just found a script to display the caller name and wanted to test it |
| 15:23.51 | tekzilla | running it from the shell with the fake-agi-env.php fails |
| 15:25.09 | tekzilla | could you give me an example how to just call the script from CLI ? |
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| 15:41.49 | *** join/#asterisk jimbo_uk (~IceChat77@84.12.253.146) |
| 15:42.00 | jimbo_uk | Hi guys, |
| 15:42.45 | jimbo_uk | Can anyone help me with a problem i have. I need to take calls into an asterisk system and push them out again to an end user via pstn (SIP/ISDN/ANalogue - its irrelevant) |
| 15:42.59 | jimbo_uk | the challenge is i need to ONLY send 10 calls at a time |
| 15:43.21 | jimbo_uk | the 11th call should be queued and sent when one of the slots is available |
| 15:43.43 | jimbo_uk | the customer can accept way more than 10 concurrent calls, just on this DDI we want to restrict it. |
| 15:44.04 | irroot | use the group/groupcount functions |
| 15:44.10 | jimbo_uk | seems line queue() isn't suitable because it has no idea of the state when a call is made via the local or sip channel |
| 15:44.18 | jimbo_uk | yes i can |
| 15:44.32 | jimbo_uk | but then any advice on how to handle the queueing? |
| 15:45.09 | Qwell | You could do a While loop in the dialplan, but that seems a little inefficient |
| 15:45.27 | Qwell | also the ordering would get screwed quickly |
| 15:45.30 | jimbo_uk | yes, very inefficiend |
| 15:45.32 | jimbo_uk | indeed |
| 15:45.36 | Qwell | hmm |
| 15:45.46 | jimbo_uk | i can write the calls off to a db via an agi |
| 15:45.47 | irroot | device_state function ?? |
| 15:46.08 | jimbo_uk | ..go on? |
| 15:46.14 | Qwell | app_queue does seem like a good way, but I'm not quite sure how you'd do it |
| 15:46.47 | jimbo_uk | app_queue would make life very easy, but the darn thing has no concept of device state for anything other than an agent |
| 15:47.14 | Qwell | sure it does |
| 15:47.16 | jimbo_uk | also the I'm not forwarding the call onto 10 separate numbers - it's just the one |
| 15:47.48 | irroot | local channel perhaps ?? |
| 15:47.55 | jimbo_uk | i tried it earlier with the local channel and the sip channel and it wasnt good |
| 15:48.16 | irroot | mind is numbing |
| 15:48.51 | jimbo_uk | when i had a single queue member of local/123456789@insideout the first call when through fine |
| 15:48.54 | irroot | im outta here on way home |
| 15:49.13 | jimbo_uk | i popped a second call in and then that went through too, regardless that the channel already had a call |
| 15:49.22 | irroot | jimbo there is ringinuse and now in trunk ignorebusy the latter does check state |
| 15:49.22 | jimbo_uk | am I missing something? |
| 15:49.36 | jimbo_uk | ringinuse=no |
| 15:49.55 | jimbo_uk | tell me more about ignorebusy?? |
| 15:50.44 | *** join/#asterisk batfastad (~benbradle@93-97-161-95.zone5.bethere.co.uk) |
| 15:51.17 | irroot | jimbo_uk see https://reviewboard.asterisk.org/r/1119/ |
| 15:51.26 | jimbo_uk | got it :-) |
| 15:52.18 | jimbo_uk | you say it's in trunk? |
| 15:52.23 | *** join/#asterisk avb (~avb@190.122.106.167) |
| 15:52.31 | avb | hey guys |
| 15:52.46 | avb | i need some help with ss7 configuration |
| 15:52.49 | irroot | jimbo_uk see lines 3080+ |
| 15:53.10 | avb | [Jul 6 15:53:04] ERROR[4398]: chan_dahdi.c:13573 dahdi_ss7_error: [1] Received message destined for point code 0x18aa but we're 0x4dd. Dropping |
| 15:53.11 | avb | [Jul 6 15:53:04] NOTICE[4398]: chan_dahdi.c:3312 my_handle_link_exception: SS7 got event: HDLC Abort(6) on span 1/0 |
| 15:53.19 | *** part/#asterisk batfastad (~benbradle@93-97-161-95.zone5.bethere.co.uk) |
| 15:53.32 | avb | any clue whats going on here? |
| 15:53.38 | avb | any hints probably |
| 15:53.50 | avb | its on asterisk 1.8 |
| 15:54.28 | Kobaz | i don't know ss7, but "destined for point code 0x18aa but we're 0x4dd" that looks like some sort of handshake problem, some sort of signalling mismatch? |
| 15:54.50 | irroot | avb looks like telco line problem im with kobaz |
| 15:55.10 | avb | irroot: ok. im thinking the same :) just to make sure |
| 15:55.18 | avb | ok, will ask a guy to contact support |
| 15:55.26 | jimbo_uk | thanks irroot |
| 15:55.31 | avb | thanks guys |
| 15:55.32 | jimbo_uk | going to test now! |
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| 15:55.51 | irroot | jimbo_uk hope it helps a bit |
| 15:57.38 | irroot | jimbo_uk perhaps there needs to be a max active call counter on app_queue this could be useful |
| 15:58.07 | irroot | or rather a maxbusy members |
| 15:58.42 | irroot | buzz me if you dont come right i can work on it perhaps tommorow |
| 15:58.53 | irroot | TTFN |
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| 16:00.52 | jimbo_uk | hey irroot |
| 16:00.56 | jimbo_uk | i'm missing something |
| 16:00.57 | jimbo_uk | look - |
| 16:00.58 | jimbo_uk | Members: |
| 16:00.58 | jimbo_uk | <PROTECTED> |
| 16:01.07 | jimbo_uk | although there is a call going on! |
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| 16:32.39 | avb | wow |
| 16:32.42 | ruben23 | hi guys any one have idea successfull install of dahdi on akeernel-PAe..? |
| 16:32.50 | avb | i took out crc4 option from system.conf |
| 16:32.54 | avb | and it seems working |
| 16:33.04 | avb | will to couple test calls |
| 16:33.11 | avb | Kobaz: |
| 16:36.07 | *** join/#asterisk doug (doug@breakout.telerama.com) |
| 16:36.54 | doug | morning |
| 16:37.14 | doug | what would i use for converting a digit string into spoken numbers? |
| 16:37.29 | Qwell | saynumber? |
| 16:38.38 | leifmadsen | saydigits() or saynumber() |
| 16:40.04 | doug | ah, sayalpha() for me actually, forgot that i was spelling, not numbering... |
| 16:40.30 | doug | i guess my real question is about converting keypad input into alpha |
| 16:40.48 | doug | i've got some file that i seem to have hacked to bits, app_dtmftotext.c |
| 16:40.55 | doug | not part of the standard distro |
| 16:41.05 | doug | what do people use for that sort of thing these days? |
| 16:42.16 | doug | i want the caller to be able to input a name |
| 16:42.25 | doug | DTMF directory style |
| 16:43.04 | Kobaz | avb: |
| 16:44.08 | avb | Kobaz: thanks for the tips :) |
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| 16:57.08 | Kobaz | welcome |
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| 17:00.10 | bongfrog | Good morning all. I am having a tough time with a PRI from paetec and their Account code server. I can dial oiut local , get in bound correctly but LD calls we get silence then a hang up. I have tried progressinband=yes and earlymedia=yes in the sip.conf with no results. THis is a 1.8.4.1 server with a Sangoma a101 t-1 card. Any hints? |
| 17:02.34 | Kobaz | if you can dial out local, and ld is not working, call your tech support, it's on their end |
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| 17:06.48 | irroot | jimbo_uk progress ?? |
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| 17:11.16 | bongfrog | Kobaz - unfortunatly that is not the correct answer, The customer currently terminates this PRI into an Avaya Legend and works correctly. I see a answer with 127 talking about inband progress back from them . |
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| 17:15.54 | Kobaz | is your pridialplan set correctly? |
| 17:16.07 | Kobaz | nice nice |
| 17:16.10 | Kobaz | er wrong window |
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| 17:18.17 | Kobaz | bongfrog: any errors on the pri or t1? |
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| 17:36.30 | jimbo_uk | hey guys |
| 17:36.44 | jimbo_uk | anyone know app_queue well or do i need to go over to the dev forum |
| 17:38.47 | _Corey_ | ~ask |
| 17:38.47 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 18:08.44 | bongfrog | kobaz: none at all even when a ld call is made. i see the connection, just no audio and no responce when digits are pressed (dtmf code followed by # like on the other switch) |
| 18:09.43 | bongfrog | kobaz: I see a coupld of bug reports that seem to reference this type or issue --premature media |
| 18:12.09 | *** join/#asterisk elsimio (~elsimio@lists.debian.org.sv) |
| 18:12.28 | elsimio | hello, somebody here use a2billing? |
| 18:15.22 | *** join/#asterisk rutski (~rutski@96.56.54.186) |
| 18:15.35 | rutski | hey all |
| 18:15.49 | rutski | I've got a new asterisk server here at the office, which we've set up with what we think is an identical configuration to our old one |
| 18:15.59 | rutski | we assigned the old machine's IP to the new machine |
| 18:16.08 | rutski | and then hoped that all of the phones would just connect, and that everything would work |
| 18:16.21 | rutski | but like 4/5th of the phones fail to register with the new machine :( |
| 18:16.31 | rutski | I did some packet sniffing and "sip set debug on" snooping |
| 18:16.46 | rutski | and it does seem that sip REGISTER packets are getting sent by said phones to asterisk |
| 18:16.52 | rutski | but they seem to be silently being dropped |
| 18:16.56 | rutski | no error message, no nothing |
| 18:17.01 | rutski | so I'm a bit stuck here :-/ |
| 18:19.31 | *** join/#asterisk jaminja (~jaminja@unaffiliated/jaminja) |
| 18:20.22 | avb | new problem. Link is working but call connects fast and i hear dead air :) |
| 18:20.49 | avb | seems alignment issue |
| 18:20.57 | avb | as sometimes i hear a person |
| 18:27.03 | joel_oliveira | hi all! quick question: is it possible to plain remove from the voicemail app folders except INBOX and OLD? i.e. the user only gets access to these 2folders. I am using 1.6.2.17 but I can migrate to other version. Thanks :) |
| 18:29.45 | jaytee | rutski, do you have port 5060 open in iptables? |
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| 18:46.11 | rutski | jaytee: definitely, yea; there's no IP tables config on this machine, it's all open |
| 18:46.17 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-rzxctkpceqnngcwd) |
| 18:46.24 | rutski | plus several phones connect properly and work just fine |
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| 18:47.46 | pabelanger | CaptainPants: Where do you get your fancy pants from? |
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| 18:55.17 | russellb | pabelanger: that's a trade secret |
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| 19:04.29 | *** join/#asterisk takamarou (4b9196ae@gateway/web/freenode/ip.75.145.150.174) |
| 19:06.09 | takamarou | Hi all. I'm currently setting up my first Asterisk box. It seems that I've got most things set up right, but am on one last difficulty. I've got calls routing through my VOIP provider (flowroute), but when I call it is saying "Please check the number you have dialed". I've looked at /var/logs/asterisk/full and don't see anything wrong, so it must be something happening between me and flowroute. Any ideas? |
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| 19:07.55 | FLeiXiuS | How do you guys prefer for SIP Trunks with PSTN connectivity? |
| 19:10.41 | avb | guys, how its possible which cic is comming to which channel? |
| 19:10.48 | avb | with ss7 signalling |
| 19:10.50 | nightrid3r | takamarou is the number * sends to flowroute the right format that flowroute expects |
| 19:11.01 | avb | getting constant dead air |
| 19:11.09 | avb | and in very rare cases getting voice |
| 19:11.37 | takamarou | nightrid3r, I believe so. There's not a ton of documentation on flowroute's site... |
| 19:12.00 | irroot | here |
| 19:13.45 | *** join/#asterisk jimbo_uk (~IceChat77@84.12.253.146) |
| 19:14.15 | irroot | jimbo_uk hi |
| 19:14.26 | irroot | hope you coming right |
| 19:14.27 | jimbo_uk | hi |
| 19:14.56 | jimbo_uk | yep, didn't see that we're causing too much of an issue if i'm honest. not exactly awash with other conversation but hey ho! |
| 19:15.07 | jimbo_uk | ok so let me test |
| 19:15.11 | jimbo_uk | i think we're set |
| 19:15.27 | irroot | it is the dev room where it started |
| 19:15.58 | jimbo_uk | yes indeed |
| 19:16.39 | jimbo_uk | ok |
| 19:16.39 | jimbo_uk | lol |
| 19:17.04 | jimbo_uk | i'm using ring all so that screws them all to busy lol |
| 19:17.17 | jimbo_uk | let me change the ring method |
| 19:17.23 | irroot | states showing ok ?? |
| 19:18.09 | jimbo_uk | yes but of course the ALL went to busy lol |
| 19:18.31 | irroot | that is a good thing |
| 19:18.37 | jimbo_uk | yes |
| 19:18.47 | jimbo_uk | didn't hang up and go not in use thoug |
| 19:18.49 | irroot | you can always only leave 1 or 2 in the group |
| 19:18.54 | jimbo_uk | sure |
| 19:19.11 | jimbo_uk | corte show hints shows busy too |
| 19:19.16 | jimbo_uk | let me see what happened |
| 19:19.22 | *** part/#asterisk jaminja (~jaminja@unaffiliated/jaminja) |
| 19:20.16 | jimbo_uk | shite |
| 19:20.27 | jimbo_uk | even after a restart now we're still showing busy.... |
| 19:20.35 | jimbo_uk | how does that work then lol |
| 19:20.45 | irroot | set verbose 3 and see |
| 19:21.03 | jimbo_uk | i did |
| 19:21.18 | jimbo_uk | but now of course it wont ring at all because it sees them all as busy. |
| 19:21.28 | jimbo_uk | so i can't see where it failed. |
| 19:21.39 | jimbo_uk | would have thought a restart now would solve this.... v odd |
| 19:22.00 | irroot | mmm its stored in astdb maybe |
| 19:22.07 | jimbo_uk | yes it is |
| 19:22.12 | jimbo_uk | need to reset that... |
| 19:22.17 | jimbo_uk | hold on |
| 19:22.27 | FLeiXiuS | Any preference for US sip trunk providers? |
| 19:23.04 | nightrid3r | FLeiXiuS store.2600hz.com |
| 19:23.07 | jimbo_uk | nope, can do uk |
| 19:23.14 | jimbo_uk | ok db entries deleted |
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| 19:23.39 | FLeiXiuS | nightrid3r, nice..easy and fast setup |
| 19:23.40 | jimbo_uk | ok, let me check DEVICE_STATE |
| 19:24.45 | jimbo_uk | ok nope |
| 19:24.48 | *** join/#asterisk scalex000 (~chatzilla@186.6.173.9) |
| 19:24.55 | jimbo_uk | the state executes |
| 19:24.55 | *** join/#asterisk ArtemMakhutov (~ArtemMakh@109.46.200.167) |
| 19:25.00 | irroot | can also "dial X@" |
| 19:25.17 | jimbo_uk | huh? |
| 19:25.20 | jimbo_uk | <PROTECTED> |
| 19:25.28 | jimbo_uk | ahhh doh |
| 19:25.36 | ArtemMakhutov | hi, is there a way to escape a semi-colon in asterisk config? |
| 19:26.02 | irroot | Custom:lineh |
| 19:26.09 | irroot | it needs a number here |
| 19:26.11 | scalex000 | good afternoon, I have this issues I connect a nortel and asterisk with sip. when i transfer call to asterisk from nortel its okey but when i try to transfer this call to nortel extension use another sip connection how can i transfer witout use another trunk |
| 19:26.13 | jimbo_uk | yes h is $EXTEN at this point |
| 19:26.43 | jimbo_uk | oddly when the call connects the h context got executed. |
| 19:26.54 | jimbo_uk | we've seen this before.... |
| 19:26.57 | irroot | set a variable with the exten and use it in h |
| 19:27.07 | jimbo_uk | ok |
| 19:27.15 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 19:29.18 | irroot | this should solve your problem without any need for the ignorebusy bits |
| 19:29.31 | jimbo_uk | yes, so i don't need to run trunk for this right? |
| 19:30.01 | jimbo_uk | still 1 problem to solve though - this hangup being executed at wrong time. |
| 19:30.54 | jimbo_uk | irroot, dude - it's 99% |
| 19:31.16 | jimbo_uk | only remaining issue is the fooking thing is hanging up at wrong time, call is still going on. |
| 19:31.22 | jimbo_uk | maybe a trunk issue? |
| 19:31.55 | jimbo_uk | ie trunk asterisk issue? |
| 19:32.23 | bongfrog | Good morning all. I am having a tough time with a PRI from paetec and their Account code server. I can dial oiut local , get in bound correctly but LD calls we get silence then a hang up. I have tried progressinband=yes and earlymedia=yes in the sip.conf with no results. THis is a 1.8.4.1 server with a Sangoma a101 t-1 card. Any hints? |
| 19:32.24 | irroot | could be |
| 19:32.36 | irroot | i need to bail soon |
| 19:32.48 | jimbo_uk | ok dude me too. |
| 19:32.51 | jimbo_uk | bail. |
| 19:33.02 | jimbo_uk | seriously though, let's talk outside of here. we can do business! |
| 19:33.17 | jimbo_uk | do you use any form of im? |
| 19:33.28 | irroot | google is on |
| 19:33.47 | irroot | gregnietsky at g mail dot com |
| 19:34.21 | jimbo_uk | gmail cool. will email u tomorrow. I'm james at 63selwyn dot com |
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| 19:35.11 | irroot | hope it helped |
| 19:35.20 | jimbo_uk | very much so. |
| 19:35.29 | jimbo_uk | speak tomorrow. enjoy evening and thank you. |
| 19:35.42 | irroot | pleasure /me going off |
| 19:36.24 | beek | Interesting turn of a phrase. |
| 19:38.15 | FLeiXiuS | nightrid3r, how difficult is it to connect a 2600hz trunk to asterisk? |
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| 19:47.52 | scalex000 | good afternoon, I have this issues I connect a nortel and asterisk with sip. when i transfer call to asterisk from nortel its okey but when i try to transfer this call to nortel extension use another sip connection how can i transfer witout use another trunk |
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| 19:52.13 | takamarou | Well, I'm getting closer any ways. Now when I try and call from my softphone it says "Call cannot be completed as dialed, please check the number and dial again". Where should I start on debugging this? |
| 19:55.54 | *** mode/#asterisk [+q t_dot_zilla!*@*] by leifmadsen |
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| 20:44.46 | *** join/#asterisk gruvfunk (~chatzilla@cpe-68-172-221-157.hvc.res.rr.com) |
| 20:45.20 | gruvfunk | greetz |
| 20:45.45 | gruvfunk | I don't have any experience so I ask: does the Paging functionality in * work with SIP phones? |
| 20:46.17 | gruvfunk | I've noticed in the sample configs it has been modified for Snom, Polycom, Grandstream and "Other" - what's the consensus for a new deployment? |
| 20:46.38 | gruvfunk | Will any popular modern SIP-based phone be compatible with the Page() ? |
| 20:52.02 | WIMPy | Any phone that can do auto-answer to speakerphone. |
| 20:52.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:53.48 | gruvfunk | WIMPy: so in my testing at home, I have softphones (Xlite) and a traditional analog phone on a SPA-2102 ATA.. I tried and no go, assuming they don't have said feature? |
| 20:54.52 | scalex000 | hello, what its means this WARNING[2504]: chan_sip.c:18374 handle_response_invite: Received response: "Forbidden" from '"3245" <sip:3245@192.168.110.1>;tag=as0644a376' |
| 20:54.57 | WIMPy | Whow could that possibly work with an analog Phone? No idea about Xlite. |
| 20:55.27 | brainiac | I've got a TDM2400P card that shows no open channels, even after restarting Asterisk (1.4.24). Attempts to call out show channel congestion... what can I check? |
| 20:55.39 | Qwell | .24? |
| 20:55.42 | gruvfunk | WIMPy: dunno, assumed since it's plugged into an ATA adapter, it provides the function (I have a 2 line speakerphone analog phone) |
| 20:55.50 | Qwell | You could start by upgrading to something from the last 2 years O.o |
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| 20:56.15 | brainiac | will do |
| 20:57.01 | gruvfunk | Any recommendations for my first desk voip phone? Preferably with at least 2 lines, Speakerphone (that works with * Paging)? Oh and .. wireless? (no Ethernet runs to my desk) |
| 20:57.12 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 20:58.28 | scalex000 | hello, what its means this WARNING[2504]: chan_sip.c:18374 handle_response_invite: Received response: "Forbidden" from '"3245" <sip:3245@192.168.110.1>;tag=as0644a376' |
| 20:59.07 | WIMPy | You don't want wireless. And I like the Snom 3x0. |
| 21:03.57 | *** join/#asterisk Howie69 (~Howie69@host-64-139-254-4.nctv.com) |
| 21:05.13 | gruvfunk | so I need a clever way to hide an ethernet run up 2 floors of the house to my desk, buh |
| 21:05.22 | gruvfunk | Snom looks fancy tho, nice |
| 21:07.31 | jaytee | gruvfunk, do you already have one ethernet run already? if so you could get something like a Polycom 331 that has a built-in switch for a PC. |
| 21:07.50 | WIMPy | BTW: Whe I said I like them, that doesn't mean I really like them, just that all other IP phones I've comte across have benn much worse. |
| 21:08.15 | gruvfunk | no, none to my desk (office) - all servers, routers, infrastructure, etc are in basement - all I have is a normal house phone line running to my office |
| 21:08.45 | WIMPy | 2 or 4 wires? |
| 21:09.02 | gruvfunk | 4, using 2 for PSTN, and 2 for my SPA-2102 VOIP line |
| 21:09.08 | batfastad | gruvfunk: Yeah we use snoms... 821s and 320s and they are decent. wifi phones aren't great. We've got a couple of Snom M9s which are DECT and they work fine. DECT is much better than wifi as it's the technology that's already used with regular cordless phones |
| 21:09.48 | WIMPy | If it's a real phone line, not just bell wire, it has Cat 3, i.e. is good anough for 10MBit ethernet. |
| 21:09.49 | gruvfunk | batfastad: so I can get a DECT, leave the base in the basement and float the M9 around the house, yeah? |
| 21:10.06 | gruvfunk | WIMPy: hey now! you're right |
| 21:10.55 | gruvfunk | i'm definitely looking at cat 3 pairs here |
| 21:11.20 | gruvfunk | would take some rewiring alltogether, since all analog phones are on that same whole-house-run |
| 21:11.35 | gruvfunk | but while I'm at it, throw an FXO in the * and convert to voip phones in the house |
| 21:11.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-njsjtnxifauiwgcf) |
| 21:12.22 | scalex000 | hello, who can help me, I need to send to execute a function on Nortel from asterisk to remove restriction its that possible |
| 21:13.14 | WIMPy | Unable to parse. |
| 21:17.03 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:18.21 | brainiac | I've got a TDM2400P card that shows no open channels, even after restarting and upgrading Asterisk to 1.4.37. Attempts to call out show channel congestion... what can I check? |
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| 21:25.49 | nny | if I want packets going to a specific peer (multiple public ips, multiple routes on box), can I set externip in the peer settings for that peer only? |
| 21:26.40 | WIMPy | Last time I tried Asterisk on a multi homed box, that didn't work out too well. |
| 21:27.08 | batfastad | gruvfunk: That's right. Just like regular cordless phones. The M9 is decent for home use but for business the menus are a bit fiddly for transferring calls and putting on hold |
| 21:27.18 | nny | WIMPy: yeah externip only works for general and nat=yes |
| 21:28.02 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-njsjtnxifauiwgcf) |
| 21:28.39 | nny | WIMPy: they have an mpls circuit connected 10.0.0.20 (NAT) -> public ip -> provider, but the route isonly for that provider and i have my route setup via the route command. Any tricks for mangling those packets to show FROM: public ip of mpls circuit? |
| 21:29.09 | *** join/#asterisk Chinorro (~chino@84.127.234.228.static.user.ono.com) |
| 21:29.33 | WIMPy | I don't think so. |
| 21:29.57 | batfastad | gruvfunk: What I wish really existed is a DECT adaptor for desk phones. So if you want a Snom 821 phone where it's difficult to run an ethernet cable, it could use DECT for communication with a base station instead of wifi |
| 21:30.28 | nny | WIMPy: looking at NAT=route |
| 21:30.57 | WIMPy | batfastad: You can get IP over DECT. |
| 21:31.34 | batfastad | WIMPy: Yeah? How? |
| 21:32.48 | WIMPy | There have been the famous PC-Cards and Serial adapters, Not sure if I've seen anythign USB. |
| 21:33.26 | WIMPy | But I'm not sure it's worth the effort. |
| 21:34.11 | batfastad | WIMPy: Would be great if there was an ethernet->DECT converter/bridge, so a desk phone has no idea it's actually operating over DECT |
| 21:34.19 | WIMPy | Or get a femtocell and a GSM desk phone. |
| 21:34.40 | WIMPy | You would definitely need a router. |
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| 21:37.07 | _Corey_ | Those Panasonic DECT phones aren't bad |
| 21:45.14 | gruvfunk | _Corey_: are you referring to those Pan. DECT add-on phones to the base unit? |
| 21:45.36 | _Corey_ | http://www.infradapt.com/panasonic_sip_phones_asterisk.html |
| 21:46.15 | gruvfunk | exactly what I was thinking |
| 21:46.31 | _Corey_ | You can just use them w/o the base unit if you prefer |
| 21:46.51 | _Corey_ | well, w/o the "corded base station" is what I mean to say |
| 21:46.55 | _Corey_ | you need one base unit |
| 21:48.06 | scalex000 | hello _Corey, how can i send a password from asterisk to another system |
| 21:48.22 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 21:48.24 | scalex000 | I need to execute a function like this function 68 |
| 21:49.14 | _Corey_ | scalex000: not sure I know what you mean... "function 68" ? |
| 21:50.36 | scalex000 | _Corey_,the another system have line restrinction |
| 21:51.49 | gruvfunk | _Corey_: do you know if the corded base is also a sip phone? (duh..?) |
| 21:52.04 | _Corey_ | scalex000: Are you asking how asterisk can dial a DTMF password? |
| 21:52.09 | *** part/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 21:52.10 | _Corey_ | gruvfunk: Yes :) |
| 21:52.24 | gruvfunk | dig it, thx |
| 21:52.25 | scalex000 | _Corey_,just a second |
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| 22:03.38 | nny | WIMPy: this comes down to a network redesign heh. Talking with our team about it, have a proper, normal solution involving a /30 and a propr edge router |
| 22:12.42 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 22:15.55 | paulc | /c/c |
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| 22:25.26 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 22:26.27 | mchou | anyone notice "weird stuff" (TM) going with sip with asterisk 1.8.4.4? |
| 22:26.57 | WIMPy | Weird stuff with SIP is perfectly normal. |
| 22:27.14 | mchou | also the text with "core show applications" is all garbled |
| 22:27.28 | mchou | didn't happen on 1.8.4 |
| 22:27.56 | mchou | like it's a different language encoding or something |
| 22:28.16 | mchou | and no, it's not my terminal |
| 22:28.41 | *** join/#asterisk scalex000 (~chatzilla@186.6.151.91) |
| 22:29.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-003.mobile.uci.edu) |
| 22:29.55 | mchou | even "sip show peers" output was garbled and misaligned |
| 22:30.52 | titter | CHANNEL(rtpqos,audio,all) works but CHANNEL(rtpqos,audio,local_lostpackets) does not ... am I missing someting? |
| 22:38.52 | *** join/#asterisk philippel_mac (~p_lindhei@184.234.121.38) |
| 22:43.35 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 22:44.03 | gruvfunk | mchou: I'm on 1.8.4.4 and I don't have garbled |
| 22:47.28 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 22:48.51 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
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| 23:06.17 | *** join/#asterisk Smirker (Smirker@14-202-69-225.static.tpgi.com.au) |
| 23:06.48 | Smirker | Howdy. being used to app_fax in asterisk 1.6, there's now res_fax in asterisk 1.8. is res_fax free to use? |
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| 23:16.51 | *** part/#asterisk Smirker (Smirker@14-202-69-225.static.tpgi.com.au) |
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| 23:32.44 | redder86 | How do you troubleshoot "SIP/2.0 401 Unauthorized" ? Is there some way to get the CLI to explain *why* the auth failed? |
| 23:34.09 | jc319 | sip set debug on |
| 23:37.25 | redder86 | jc319: thanks, that's where I see the unauth message - looks like the Cisco 7961 phone isn't responding to the challenge |
| 23:37.33 | mchou | gruvfunk_away: what's your arch? |
| 23:46.13 | *** join/#asterisk Kris2k (~kris@modemcable152.16-178-173.mc.videotron.ca) |
| 23:47.46 | Kris2k | Anyone have a clue why ILBC(30ms packet) to ULAW(20ms packet) results in a very bad "robotic" outcome, but the ILBC stream accessing my voicemail works fine ? |
| 23:50.45 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 23:51.06 | *** join/#asterisk [netman] (netman@152.252.22.95.dynamic.jazztel.es) |
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| 23:57.38 | leifmadsen | Kris2k: I'm guessing because something is happening in the conversion between the 30ms packet size and the 20ms packet size |
| 23:57.48 | leifmadsen | I'd suggest first trying to make iLBC work in 20ms packets |
| 23:57.49 | Kris2k | I figured |
| 23:57.52 | leifmadsen | allow=ilbc:20 |
| 23:58.06 | Kris2k | because when I threw both call legs into a meetme bridge, the audio was good |
| 23:58.46 | Kris2k | 1.8 wont allow me to put ilbc to 20ms |
| 23:59.34 | Kris2k | even when defined, a sip show peer will report ilbc:30 |