| 00:09.32 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 00:25.50 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 00:26.01 | mattwj2002 | good day all |
| 00:26.02 | mattwj2002 | :) |
| 00:26.10 | WIMPy | Good morning. |
| 00:26.24 | mattwj2002 | hi WIMPy |
| 00:28.55 | mattwj2002 | WIMPy: what are you up to |
| 00:28.57 | mattwj2002 | :) |
| 00:28.59 | *** join/#asterisk KNERD (~KNERD@24.175.253.226) |
| 00:29.22 | WIMPy | Lots of broken stuff. |
| 00:29.31 | mattwj2002 | WIMPy: that sucks |
| 00:30.27 | WIMPy | About two hours ago the GPU driver exploded. So I built the latest linux. Now the NFS server is broken. |
| 00:30.50 | mattwj2002 | oh man! |
| 00:30.57 | mattwj2002 | sounds like a bad day |
| 00:30.58 | mattwj2002 | :( |
| 00:31.00 | WIMPy | And there was Asterisk as well, but I don't think I have to mention that part, do I? |
| 00:31.06 | KNERD | I wonder if I should file a bug report...with latest update CentOS 6.5 versions of asterisk 1.8.x, 11 and 12 fail to run after compiling without errors... ...tried on 4 different machines with different CPUs...upon starting asterisk I get "Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk" |
| 00:31.12 | mattwj2002 | nope |
| 00:32.20 | WIMPy | Ah, the good old broken compiler again? |
| 00:33.41 | KNERD | seems so |
| 00:34.12 | KNERD | oh...also tried with the "build native" unchekced also |
| 00:34.44 | WIMPy | Did you try to build other stuff? |
| 00:34.50 | KNERD | no |
| 00:34.53 | KNERD | only asterisk |
| 00:35.08 | WIMPy | Looks like I'm not the only one having a bad night. |
| 00:35.18 | mattwj2002 | hey me too! |
| 00:35.33 | mattwj2002 | I couldn't fix a freaking phone issue at work |
| 00:35.41 | mattwj2002 | I am a bit upset about it |
| 00:35.51 | WIMPy | But I think I don't care any further for today. Writes from Omni work. Reading isn't essential at the moment. |
| 00:36.39 | WIMPy | (neiter are Linux clients) |
| 00:37.46 | mattwj2002 | does anyone have any recommendations for some really inexpensive SIP phones? |
| 00:37.57 | mattwj2002 | the physical kind not a softphone |
| 00:37.59 | mattwj2002 | :) |
| 00:38.22 | WIMPy | Not me |
| 00:38.35 | mattwj2002 | okay cool |
| 00:38.56 | mattwj2002 | not a big deal.... |
| 00:39.00 | *** join/#asterisk mmikeym (~mikeym@184.70.65.118) |
| 00:39.13 | mattwj2002 | I probably shouldn't waste my money on that right now anyways |
| 00:39.44 | WIMPy | I still like the old Snoms most. |
| 00:40.07 | mattwj2002 | WIMPy: any particular model? |
| 00:40.20 | WIMPy | 360 |
| 00:41.08 | WIMPy | Or 320 if you don't want much display. Otherwiese it's the same. 370 has more beef under the lid as well. |
| 00:41.39 | KNERD | ?bugs |
| 00:42.03 | WIMPy | Although I genereally find the user Interface rather strange, the Digiums phones aren't bad if you need only one account. |
| 00:42.15 | KNERD | what is that bot prefix? |
| 00:42.23 | KNERD | &bugs |
| 00:42.24 | WIMPy | ~ |
| 00:42.28 | KNERD | thanks |
| 00:42.30 | KNERD | ~bugs |
| 00:42.36 | WIMPy | issues.asterisk.org |
| 00:42.44 | KNERD | oh..thanks |
| 00:43.03 | mattwj2002 | D40? |
| 00:43.07 | mattwj2002 | not bad :) |
| 00:43.17 | WIMPy | Oh. The Linux client came back to life. |
| 00:43.48 | WIMPy | mattwj2002: BTW: Both of them take AGES to boot :-( |
| 00:45.19 | mattwj2002 | your servers? |
| 00:45.37 | mattwj2002 | or were you talking about the phones |
| 00:45.38 | mattwj2002 | :) |
| 00:45.53 | WIMPy | No the Snom and even more the Digium phones. |
| 00:46.11 | mattwj2002 | got ya |
| 00:46.29 | mattwj2002 | I think I'll stick to my softphones at the moment |
| 00:46.38 | mattwj2002 | but I was just curious you know |
| 00:46.48 | WIMPy | They surely start faster. |
| 00:46.58 | mattwj2002 | :) |
| 00:47.30 | mattwj2002 | I have a very barebones asterisk box |
| 00:47.45 | mattwj2002 | it is running on a raspberry pi |
| 00:48.20 | WIMPy | I thought about abusing a Pi as a phone adapter. |
| 00:48.35 | mattwj2002 | a phone adapter? |
| 00:48.35 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:48.46 | mattwj2002 | what do you mean a phone adapter? |
| 00:48.58 | WIMPy | To connect a legacy phone. |
| 00:49.17 | mattwj2002 | they don't have an fxs or fxo on them though |
| 00:49.24 | mattwj2002 | how would you connect to it? |
| 00:49.41 | mattwj2002 | *fxs |
| 00:49.53 | WIMPy | No, but they have USB. |
| 00:50.05 | mattwj2002 | oh hehe |
| 00:50.08 | WIMPy | And I'm going for BRI. |
| 00:50.27 | mattwj2002 | sounds interesting |
| 00:50.28 | mattwj2002 | ;) |
| 00:51.10 | WIMPy | Although... I still have that Horstbox... |
| 00:54.09 | mattwj2002 | that is like me.... |
| 00:54.48 | WIMPy | You also got one of them? |
| 01:00.57 | mattwj2002 | no |
| 01:01.17 | mattwj2002 | I have a OBI100 I could use instead of buying a phone |
| 01:01.18 | mattwj2002 | :) |
| 01:01.57 | mattwj2002 | it is just an ATA |
| 01:01.58 | mattwj2002 | :) |
| 01:02.15 | WIMPy | just takes apart an Samsung IAD. |
| 01:03.41 | WIMPy | Only infineon stuff. Doesn't look like a good candidate for alternative firmware. |
| 01:03.58 | mattwj2002 | hehe |
| 01:04.29 | WIMPy | But it seems pretty usable the way it is. |
| 01:04.31 | mattwj2002 | I am not familiar with a Samsung IAD |
| 01:04.42 | mattwj2002 | is that similar to a Cisco IAD? |
| 01:04.54 | mattwj2002 | multiple analog lines? |
| 01:04.58 | WIMPy | 3 FXS and 1 BRI. |
| 01:05.03 | mattwj2002 | nice dude! |
| 01:05.04 | mattwj2002 | :D |
| 01:05.12 | WIMPy | Integrade PBX. |
| 01:05.24 | mattwj2002 | do you have ISDN in your area? |
| 01:05.29 | WIMPy | Integrated PBX. |
| 01:05.36 | mattwj2002 | sweet |
| 01:05.46 | WIMPy | If you still have it, yes. |
| 01:06.14 | mattwj2002 | ok no new installs? |
| 01:06.44 | WIMPy | Only on very explicit request. |
| 01:07.40 | mattwj2002 | got ya |
| 01:07.45 | mattwj2002 | kind of like dialup here |
| 01:07.46 | mattwj2002 | :) |
| 01:08.03 | WIMPy | 2 years ago we still had some 30% BRIs, but since last year they doen't sell phone lines any more. |
| 01:10.39 | mattwj2002 | got ya |
| 01:10.53 | WIMPy | Well, there are two telcos the still sell lines and one that will give you one if you are begging hard. |
| 01:11.17 | mattwj2002 | yeah it is like the PSTN here in the US |
| 01:11.27 | mattwj2002 | lots of alternatives I am sure |
| 01:11.53 | WIMPy | SIP, SIP or SIP. |
| 01:11.57 | mattwj2002 | hehe |
| 01:12.00 | mattwj2002 | I love sip! |
| 01:12.03 | WIMPy | So no replacement available. |
| 01:12.07 | mattwj2002 | I have sip on my cell phone |
| 01:12.15 | mattwj2002 | it uses my data connection |
| 01:12.17 | mattwj2002 | :) |
| 01:12.59 | mattwj2002 | I am terminating with a company called callwithus |
| 01:13.01 | WIMPy | Data connections are extremely expensive here. |
| 01:13.36 | mattwj2002 | I have a 100 minute plan with 5 GB per month for $30 usd per month prepaid |
| 01:13.51 | mattwj2002 | how expensive are we talking? |
| 01:13.58 | WIMPy | Sounds ok. |
| 01:14.15 | WIMPy | 3G for 15 EUR (data only) |
| 01:14.15 | mattwj2002 | it is fairly inexpensive by US standards |
| 01:14.24 | mattwj2002 | not bad |
| 01:14.48 | WIMPy | Well... |
| 01:15.03 | WIMPy | Not an alternative to DSL or cable. |
| 01:15.30 | mattwj2002 | no definitely not |
| 01:15.43 | mattwj2002 | but for a on the go solution doesn't sound bad :) |
| 01:15.43 | WIMPy | Same goes for LTE. |
| 01:16.01 | mattwj2002 | mobile solution |
| 01:16.27 | WIMPy | If I use it, I go for 1GB / 24h for 1.99/ |
| 01:17.13 | mattwj2002 | cool |
| 01:17.46 | WIMPy | That's the best solution of you only need it sometimes. |
| 01:18.06 | mattwj2002 | definitely |
| 01:18.14 | mattwj2002 | the 15 EUR plan is for a month? |
| 01:18.32 | WIMPy | 30 days to be exact. |
| 01:18.52 | mattwj2002 | okay got ya |
| 01:19.15 | WIMPy | But that's all on the cheaper networks, off course. |
| 01:19.43 | mattwj2002 | still real good |
| 01:20.13 | mattwj2002 | that is one thing I like about this callwithus sip service |
| 01:20.23 | mattwj2002 | you can pick your route type |
| 01:20.46 | WIMPy | If you want the ones with better coverage, prices are a lot higer. |
| 01:20.54 | mattwj2002 | they have standard, premium, or pstn |
| 01:21.15 | mattwj2002 | I suppose it depends where you live and work |
| 01:21.17 | mattwj2002 | :) |
| 01:21.28 | mattwj2002 | oh what type of coverage will get you by |
| 01:22.00 | WIMPy | It's ok in the city. If you get out the differece is ratehr obvious. |
| 01:22.17 | WIMPy | Probably most obvious when you're using trains. |
| 01:22.27 | mattwj2002 | make sense |
| 01:41.59 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
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| 04:34.45 | KNERD | anyone used the intel compuler instead of gcc? |
| 04:37.01 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:58.46 | ectospasm | would that be advisable, given that Asterisk has some GNU dependencies? |
| 04:59.47 | KNERD | well asterisk no longer executing probler on centos anymore |
| 05:00.13 | KNERD | gotta try soemthing |
| 05:38.18 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
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| 05:38.53 | ectospasm | KNERD: are you compiling from source? That should be unnecessary |
| 05:39.07 | ectospasm | you can even add e.g. the AsteriskNOW repos if you want |
| 05:40.02 | *** join/#asterisk Entelin (~Paruza@c-71-63-229-101.hsd1.mn.comcast.net) |
| 05:41.45 | *** join/#asterisk boratynskikamil (5bef9959@gateway/web/freenode/ip.91.239.153.89) |
| 05:42.11 | Entelin | I am making a simple "voicemail" prompt by using Record , however I don't see an option with this command to run a system command after? I tried doing a call to System directly after, however if the user hangs up during the recording, I dont think it continues down the priority list? |
| 05:44.12 | KNERD | ectospasm: yes I am compiling from source, but asterisk is now borked under centos |
| 05:45.10 | ectospasm | Entelin: you may want to set up the h extension |
| 05:45.26 | ectospasm | ...and run System($processing_command) in the h exten |
| 05:45.46 | Entelin | <PROTECTED> |
| 05:49.16 | Entelin | I assume this is context specific right? |
| 06:01.16 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 07:10.55 | Entelin | hey so, the record app, if the user hangs up the recording is just discarded apparently... Whats the point of the k option then? Is there something I'm missing here or doesn't this make this app kinda useless? |
| 07:11.44 | Entelin | I'm using the k option and it still doesnt create the file on a hangup |
| 07:14.43 | KNERD | ~book |
| 07:14.43 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:19.27 | Entelin | ive read that whole book |
| 07:19.44 | Entelin | the section on that function does not address or even mention this issue |
| 07:21.18 | KNERD | that was for me |
| 07:21.27 | Entelin | ah ok |
| 07:21.46 | KNERD | i am looking for a command to show all loaded modules, but I see none |
| 07:22.01 | Entelin | module show |
| 07:22.05 | Entelin | from the cli? |
| 07:23.49 | KNERD | yes |
| 07:23.55 | Entelin | yeah thats how |
| 07:24.05 | KNERD | oh..thanks |
| 07:24.29 | ChannelZ | The k option of Record works for me |
| 07:24.49 | KNERD | wow....asterisk on CentOS is not seriously borked |
| 07:24.59 | ChannelZ | Record(/tmp/fart.ulaw,,,k) |
| 07:25.18 | Entelin | what version of asterisk are you running? |
| 07:25.23 | KNERD | *CLI> module show Module Description Use Count chan_sip.so Session Initiation Protocol (SIP) 0 1 modules loaded |
| 07:25.31 | ChannelZ | 12 |
| 07:25.37 | KNERD | CLI> sip show peers No such command 'sip show peers' (type 'core show help sip show' for other possible commands |
| 07:25.42 | Entelin | that might be the difference, ive been using 1.8 |
| 07:25.53 | Entelin | just because it was in package management on my server |
| 07:26.08 | ChannelZ | Quite old |
| 07:26.27 | Entelin | yeah ill upgrade this at some point |
| 07:27.08 | ChannelZ | well make sure your parameters are even right, I don't know if Record has changed that far back |
| 07:27.24 | Entelin | same => n,Record(/var/spool/asterisk/monitor/${FILE}.wav,4,300,k) |
| 07:27.30 | Entelin | its the same, and it does work |
| 07:27.36 | Entelin | the params all work etc |
| 07:27.43 | Entelin | and the k option is in the docs for that version... |
| 07:27.50 | Entelin | but yeah on hangup its an issue |
| 07:36.34 | boratynskikamil | ChannelZ: Question to you. Could you advice me, what should I check if calling SIP->SIP noone can hear each other. |
| 07:36.39 | boratynskikamil | I mean, both of them. |
| 07:39.29 | Entelin | your firewall for sure |
| 07:39.38 | Entelin | as the first thing i mean |
| 07:39.55 | Entelin | if either end are behind nat then you also need to look into the sip nat settings |
| 07:46.55 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 07:47.05 | mattwj2002 | hi guys |
| 07:47.19 | mattwj2002 | I am using a service called callwithus |
| 07:47.29 | mattwj2002 | they have three levels of service |
| 07:47.36 | mattwj2002 | standard, premium and pstn |
| 07:47.48 | mattwj2002 | anyone know what is the highest quality one? |
| 07:50.52 | *** join/#asterisk chare (~chare@50-47-81-172.evrt.wa.frontiernet.net) |
| 07:51.25 | mattwj2002 | I originally assumed the highest quality was pstn, but now I am working if it isn't premium |
| 07:51.32 | mattwj2002 | *wondering |
| 07:51.50 | *** part/#asterisk chare (~chare@50-47-81-172.evrt.wa.frontiernet.net) |
| 08:06.19 | KNERD | the one with the highest price |
| 08:08.14 | mattwj2002 | well in the US they are the same |
| 08:08.30 | KNERD | then look at their FAQ |
| 08:08.48 | KNERD | and look at prices out of USA |
| 08:10.37 | mattwj2002 | KNERD: do the Bahamas count? |
| 08:10.57 | KNERD | that is not in US |
| 08:12.27 | mattwj2002 | premium is higher for UK calls |
| 08:13.18 | *** join/#asterisk evilman_home (~evilman_h@89-179-77-66.broadband.corbina.ru) |
| 08:13.51 | mattwj2002 | same for norway |
| 08:13.52 | mattwj2002 | :) |
| 08:14.49 | mattwj2002 | now PSTN is more expensive for China |
| 08:14.50 | mattwj2002 | :P |
| 08:15.11 | mattwj2002 | it doesn't really matter..... |
| 08:15.24 | mattwj2002 | I am just messing around anyways :) |
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| 08:46.40 | boratynskikamil | Damn Entelin is gone. |
| 08:46.46 | boratynskikamil | I disabled my firewall at all for tests. |
| 09:06.10 | *** join/#asterisk g_r_eek (~g_r_eek@46-236-35.adsl.cyta.gr) |
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| 09:39.26 | ChannelZ | boratynskikamil: firewall or config. |
| 09:39.31 | ChannelZ | or both |
| 09:40.01 | boratynskikamil | ChannelZ: As I told. Firewalls are off. :-) So, where should I look for mistake in config? |
| 09:40.54 | ChannelZ | Is the asterisk box its self the firewall, or is it behind it? (IE is it behind NAT?) |
| 09:41.24 | boratynskikamil | ChannelZ: Asterisk isn't behind NAT. SIPs and Asterisk are in the same LAN. |
| 09:42.34 | ChannelZ | And you have no audio? Strange. |
| 09:42.41 | boratynskikamil | See next one bug. When I call SIP2 from SIP1 I hear MusicOnHold. |
| 09:42.53 | ChannelZ | Turn on sip debug, make a test call, then pastebin the result |
| 09:43.01 | boratynskikamil | ok. |
| 09:43.05 | boratynskikamil | From the begining, right? |
| 09:43.08 | ChannelZ | yes |
| 09:43.15 | boratynskikamil | So I do something like that. |
| 09:43.26 | boratynskikamil | 1. SIP1 calls SIP2. |
| 09:43.34 | boratynskikamil | 2. SIP2 answers. |
| 09:43.38 | boratynskikamil | 3. SIP2 hangups. |
| 09:43.44 | boratynskikamil | 4. SIP1 calls SIP2 one more time. |
| 09:44.19 | ChannelZ | just one call, let's not make a mess of this |
| 09:47.33 | boratynskikamil | http://www.wklej.org/id/1287359 |
| 09:47.40 | boratynskikamil | ChannelZ: ^ here you go. |
| 09:56.09 | ChannelZ | ok.. so HD01 calls HD04. Does HD01 hear the music on hold from asterisk while HD04 is ringing? |
| 09:56.21 | boratynskikamil | HD01 calls HD04. |
| 09:56.30 | boratynskikamil | HD01 hears MoH during the call right. |
| 09:56.43 | boratynskikamil | When HD04 asnwers call, they do not hear each other. |
| 09:56.59 | boratynskikamil | HD01 tells something HD04 does not hear that. HD04 tell something and HD01 does nto hear that. |
| 09:57.15 | boratynskikamil | When HD01 hangups phone, or HD04, does not matter. |
| 09:57.26 | boratynskikamil | And HD01 calls one more time, HD01 does not hear MoH. |
| 09:58.13 | ChannelZ | What phones are these? I don't recognize the user-agent |
| 09:58.25 | boratynskikamil | QuteComs. |
| 09:58.39 | ChannelZ | And have you done an echo test on the individual endpoints to make sure they even work themselves between asterisk? |
| 10:00.05 | boratynskikamil | ChannelZ: I am able to asnwer phone from ISDN card and GSM card. |
| 10:00.21 | ChannelZ | huh? |
| 10:00.38 | boratynskikamil | HD01 and HD04 are able to answer DAHDI calls, in example. |
| 10:00.53 | boratynskikamil | They are members of a Queue. |
| 10:01.00 | ChannelZ | What I mean is, create a test extension, say 1234.. in it, do a Playback() of something, and then Echo() |
| 10:01.10 | ChannelZ | Call extension 1234 from HD0 |
| 10:01.10 | boratynskikamil | Ok. |
| 10:01.12 | boratynskikamil | I will do it. |
| 10:01.16 | boratynskikamil | Yes, yeah, I know. |
| 10:01.19 | boratynskikamil | Moment. |
| 10:01.26 | ChannelZ | You should hear whatever sound file, and then be able to talk and hear yourself echo'd back. |
| 10:02.42 | boratynskikamil | ChannelZ: Ok. |
| 10:02.45 | boratynskikamil | I did somethig like that. |
| 10:02.52 | boratynskikamil | I quited QuteCom, right? |
| 10:03.15 | boratynskikamil | Runned it one more time and did echo-test. |
| 10:03.18 | boratynskikamil | I worked. |
| 10:03.31 | boratynskikamil | I heared myself with an echo. :-) |
| 10:03.33 | ChannelZ | and HD04? |
| 10:03.40 | ChannelZ | does it work as well? |
| 10:04.45 | boratynskikamil | ChannelZ: When I restart QuteCom, yep. |
| 10:04.49 | boratynskikamil | But... what is strange. |
| 10:05.21 | *** join/#asterisk michael_work (~michael@212.199.182.172) |
| 10:05.52 | boratynskikamil | When I call HD04, and do an echo-test one more time, I do not hear this nice girl and myself, too. :-) |
| 10:06.43 | boratynskikamil | When I relog, it works well one more time. |
| 10:07.15 | ChannelZ | Not sure what to say, maybe it's broken. |
| 10:08.17 | boratynskikamil | QuteCom? |
| 10:08.25 | ChannelZ | I don't see anything particularly wrong in your sip debug, although it does look like you have canreinvite turned on, so it's trying to make both endpoints connect directly to each other and maybe they don't like that either. |
| 10:08.38 | ChannelZ | yes |
| 10:08.57 | boratynskikamil | ChannelZ: Wold you like to see my sip.conf? |
| 10:09.02 | boratynskikamil | Would* |
| 10:09.27 | ChannelZ | wouldn't hurt, just XXX out your secret= |
| 10:09.40 | boratynskikamil | ChannelZ: Ok (: |
| 10:09.48 | boratynskikamil | Do not make me an idiot, please. :-P |
| 10:10.41 | ChannelZ | Are you running these Qute clients on windows, mac, .. ? |
| 10:11.04 | boratynskikamil | WIndows. :-( |
| 10:16.24 | ChannelZ | this client is weird |
| 10:16.42 | boratynskikamil | ChannelZ: http://wklej.org/id/1287393 |
| 10:18.00 | boratynskikamil | ChannelZ: Weird you mean, easy to understand or strange? :-) |
| 10:19.49 | ChannelZ | strange like it doesn't seem to work so well. |
| 10:22.52 | boratynskikamil | ChannelZ: Did you mean, I screwed that up? :-) |
| 10:23.13 | ChannelZ | I can't get audio to work on this thing at all. |
| 10:23.46 | boratynskikamil | ChannelZ: Hm? |
| 10:25.42 | boratynskikamil | ChannelZ: You mean I didn't define audio condecs in SIP.conf? |
| 10:25.59 | ChannelZ | It transmits audio, although it's all screwed up, but it doesn't seem to receive audio. This QuteCom client is not working. |
| 10:27.27 | boratynskikamil | ChannelZ: It is all screwed up? Nice. :-) So why I receives audio, when I got a call, for example? |
| 10:27.35 | boratynskikamil | Before I try to call SIP, it works properly. |
| 10:27.48 | boratynskikamil | Forwarding too... |
| 10:27.49 | ChannelZ | More indication this client is a piece of crap |
| 10:28.05 | ChannelZ | Try something else, like Zoiper or Blink. They work. |
| 10:29.05 | boratynskikamil | ChannelZ: Ok, i will try to, give me a moment, pelase. |
| 10:45.23 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 10:45.32 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 10:52.22 | ChannelZ | Well things have either gone incredibly well or incredibly badly. |
| 11:02.13 | boratynskikamil | ChannelZ: DAMN! |
| 11:02.15 | boratynskikamil | It works. |
| 11:02.33 | boratynskikamil | Damn. I tried to repair it from yesterday and I thought I fucked up configs. |
| 11:03.46 | ChannelZ | No just seems like a shitty softphone. Doesn't look to be in development anymore |
| 11:04.01 | boratynskikamil | ChannelZ: Right. |
| 11:04.18 | boratynskikamil | Ok, could you tell me what can I repair in my configs? If you told I screw them up? :-) |
| 11:04.19 | ChannelZ | When the bugtracker for a softphone has a bug entered that says "calls don't work", it kind of says a lot. |
| 11:06.02 | ChannelZ | Anyway have fun, I have to go to bed |
| 11:06.26 | boratynskikamil | ChannelZ: Cya. Thank you a lot for help. |
| 11:08.04 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
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| 12:28.52 | boratynskikamil | Does Zoiper support Contact groups? |
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| 15:22.41 | boratynskikamil | Guys. Does Zoiper supports call forwarding? |
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| 17:36.01 | maruen | Hi there, anyone avaiable on this channel now? |
| 17:36.52 | ectospasm | I'm barely paying attention |
| 17:36.54 | ectospasm | !ask |
| 17:37.01 | ectospasm | ~ask |
| 17:37.01 | 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. |
| 17:37.29 | *** join/#asterisk lnb (~lnb@CPE000347b24a71-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 17:38.23 | maruen | Im getting some error when trying to load the module chan_dongle.so in my Mac OS X Asterisk |
| 17:38.49 | maruen | I could install asterisk 12.0.0.0 on Mac OS X Mavericks, and its working fine if I did not load this module |
| 17:39.18 | ectospasm | is there a reason you're using Asterisk 12? |
| 17:39.30 | ectospasm | I admit I know nothing about chan_dongle |
| 17:39.37 | maruen | <PROTECTED> |
| 17:39.37 | maruen | Referenced from: /Library/Application Support/Asterisk/Modules/chan_dongle.so |
| 17:39.42 | maruen | the error is this one above |
| 17:39.59 | ectospasm | maruen: what do you need chan_dongle for? |
| 17:40.02 | maruen | no, could be other version |
| 17:40.25 | maruen | I want it for send and receive sms messages with huaway modems |
| 17:40.33 | ectospasm | do you expect this Asterisk instance to see production? |
| 17:40.58 | maruen | Yes, I expect |
| 17:41.06 | maruen | it can be on production? |
| 17:41.12 | ectospasm | do you know if the chan_dongle module exists in Asterisk 11? |
| 17:41.31 | maruen | in does not exists, I donwloades apart from it |
| 17:41.40 | maruen | Downloaded* |
| 17:41.43 | af_ | it seems a cool, channel, it would be great test it |
| 17:41.59 | ectospasm | maruen: so it's a third party module |
| 17:42.08 | maruen | yeap |
| 17:42.27 | af_ | oh, it's not in the official Digium source tree? |
| 17:42.32 | maruen | no |
| 17:42.33 | ectospasm | maruen: you might want to check to see what version of Asterisk chan_dongle is developed against, and install that one. My guess it isn't built for 12 |
| 17:42.58 | maruen | it isn't probably |
| 17:43.46 | maruen | is there an official module built within asterisk official modules to send and receive SMS ? |
| 17:43.51 | ectospasm | 12 has a lot of experimental stuff in it |
| 17:43.59 | maruen | understood |
| 17:44.13 | maruen | what version do you recomend for test it ? |
| 17:44.17 | ectospasm | maruen: I don't know about transmitting SMS |
| 17:44.30 | maruen | ectospasm: no problem then... |
| 17:45.15 | ectospasm | it takes a bit of work, but many mobile providers give email addresses for SMS, like 1234567890@txt.att.net |
| 17:45.22 | maruen | but I think in this particulary error that I'm getting, it's relatively the third part module instead something of asterisk |
| 17:45.23 | ectospasm | ...but receiving is a whole nother beast |
| 17:46.14 | maruen | Nice to hear about email solutions, I didn't know about that, but as I will need also receive, it wond fit for my needs.... |
| 17:46.36 | ectospasm | no, it won't |
| 17:47.08 | maruen | so, should it worth test it in another version of asterisk? |
| 17:49.01 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 17:49.02 | *** join/#asterisk lorsungcu (~anonymous@74-95-143-116-Minnesota.hfc.comcastbusiness.net) |
| 17:49.20 | ectospasm | maruen: find the documentation for chan_dongle, and determine what version of Asterisk it's built for. Better hope it's 1.8 or 11 |
| 17:50.38 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 17:51.02 | af_ | what a boring day.... pfffff |
| 17:51.26 | af_ | I am stuck configuring a sip phone... what deal |
| 17:52.31 | maruen | in my linux, I make it work with version 1.8 |
| 17:52.53 | maruen | but isn't to much deprecated the version 1.8 comparing with 11 ? |
| 17:53.03 | ectospasm | af_: that differs greatly by the phone you're configuring. Polycoms suck ass when configuring. Digium phones are a breeze in comparison |
| 17:53.20 | ectospasm | maruen: well, 1.8 will EOL before 11 |
| 17:53.42 | ectospasm | maruen: but 1.8 is a lot more mature |
| 17:53.50 | ectospasm | (read == less buggy) |
| 17:54.29 | maruen | whan mean EOL ? |
| 17:56.48 | af_ | oh, thanks ectospasm now I am using unexpensive GS |
| 17:57.06 | af_ | but may be a problem in my network, that is poorly managed |
| 17:57.32 | ectospasm | last I heard (which was a while ago), Grandstream phones were substandard quality |
| 17:57.42 | af_ | or an old serial modem, server it's stuck! |
| 17:57.55 | ectospasm | but then again, I used to work at Digium |
| 17:57.57 | af_ | pfff. it's gone mad on the serial pci |
| 17:58.31 | af_ | is it good job there ectospasm ? |
| 17:58.39 | af_ | fun? |
| 17:58.56 | ectospasm | good, fun, but not really high paying (in my position, at least) |
| 17:59.22 | af_ | well, there is money problems all around the world..... bad times.... |
| 18:00.03 | af_ | fun may be a plus. have to reset my internet conn, sucky server down... have nice time |
| 18:18.58 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 18:22.25 | [TK]D-Fender | [12:53]ectospasmmaruen: but 1.8 is a lot more mature [12:53]ectospasm(read == less buggy) <- not sure I'd stand on that. 11 has been around a while and didn't any real show-stoppers on release. Both are being maintained so it's a question of apps being version-dependent. |
| 18:22.35 | [TK]D-Fender | There is no reason to touch 1.8 for a enw install otherwise |
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| 19:04.15 | *** join/#asterisk jsjc (~Adium@179.Red-79-150-65.dynamicIP.rima-tde.net) |
| 19:11.22 | lnb | wondering if anyone here knows a fix for calls with follow me not going to voice mail if not answered? |
| 19:11.59 | [TK]D-Fender | Not an Asterisk problem. |
| 19:12.22 | lnb | please prove that statement |
| 19:13.25 | [TK]D-Fender | That isn't using Asterisk's followme command. |
| 19:15.14 | [TK]D-Fender | Which also has no implicit concept of "voicemail" |
| 19:19.42 | *** join/#asterisk cyford (~cyford33@2601:0:9300:2fc:2cf3:d4f6:2f6f:bdf7) |
| 19:20.12 | cyford | <PROTECTED> |
| 19:31.51 | lnb | clearly, the ring time is ignored in the follow me setting, only the Initial ring time is working |
| 19:32.12 | lnb | so something is very wrong |
| 19:32.26 | [TK]D-Fender | With FreePBX maybe. Nothing to do with Asterisk's functioning. |
| 19:33.05 | lnb | what file might this be in.. i have looked at several extension_ files but the syntax isnt clear to me |
| 19:34.43 | [TK]D-Fender | Not a topic for here.... |
| 19:35.41 | lnb | is exten => *320,n,Goto(vmret,1) mean go to voice mail? |
| 19:36.14 | Penguin | We can't know that; you haven't shown us the REST of the dial plan. |
| 19:36.16 | [TK]D-Fender | Not a topic for here.... |
| 19:36.23 | [TK]D-Fender | Penguin: FreePBX <- |
| 19:36.40 | lnb | or maybe exten => 320,n,Macro(exten-vm,320,320,0,0,0) |
| 19:36.45 | Penguin | And if it's FreePBX, I don't even care about the rest of the dial plan. |
| 19:37.18 | [TK]D-Fender | lnb: There is no support here for debugging FreePBX processing issues here. |
| 19:47.15 | *** join/#asterisk paule32 (paule32@dslb-178-000-064-233.pools.arcor-ip.net) |
| 19:47.18 | paule32 | hello |
| 19:47.36 | paule32 | how can i add a user and password to pbx ? |
| 19:47.50 | [TK]D-Fender | What "user" are you talking about? |
| 19:47.59 | paule32 | sip account |
| 19:48.07 | Penguin | I think users are people. |
| 19:49.18 | paule32 | <PROTECTED> |
| 19:49.46 | *** join/#asterisk Ta^3 (~tacvbo@fixed-203-252-93.iusacell.net) |
| 19:50.47 | [TK]D-Fender | paule32: "vi sip.conf" |
| 19:51.11 | [TK]D-Fender | paule32: that is AMI... that has nothing to do with a "user", or "SIP" |
| 19:52.31 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 19:54.53 | paule32 | i have following code: |
| 19:54.55 | paule32 | http://codepad.org/9O7dU9nz |
| 19:55.12 | paule32 | and i am would like to initate a session |
| 19:55.38 | [TK]D-Fender | That is AMI, not "SIP". |
| 19:55.58 | [TK]D-Fender | What are you trying to actually do? |
| 19:57.31 | paule32 | i would like to call a sip account, that get dialtone, check it, e.g. dialtone 1 and 2 -> do response with dial tone 2 1 |
| 19:57.52 | [TK]D-Fender | paule32: that code has nothing to do with what you just said |
| 19:58.05 | [TK]D-Fender | ~book |
| 19:58.05 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:58.12 | [TK]D-Fender | Go read the book. ---^ |
| 19:58.29 | [TK]D-Fender | sip.conf , extensions.conf |
| 19:58.37 | WIMPy | Ah, paule32 is back |
| 19:58.50 | paule32 | hehe |
| 19:58.56 | paule32 | hi WIMPy |
| 19:59.34 | paule32 | as basis of: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 19:59.49 | paule32 | i would login into an account |
| 20:00.04 | [TK]D-Fender | paule32: Told you twice now.. lets try a third... AMI has NOTHING to do with the goal you stated |
| 20:00.21 | [TK]D-Fender | paule32: You do not seem to have any comprehension of what AMI is. |
| 20:00.42 | WIMPy | What's the goal today? |
| 20:00.51 | [TK]D-Fender | [14:57]paule32i would like to call a sip account, that get dialtone, check it, e.g. dialtone 1 and 2 -> do response with dial tone 2 1 |
| 20:01.17 | [TK]D-Fender | And his "code: [14:54]paule32i have following code: [14:54]paule32http://codepad.org/9O7dU9nz |
| 20:01.25 | WIMPy | Dialtone modem text transmission? |
| 20:02.04 | [TK]D-Fender | WIMPy: We'll see if he tries turning it into that pipe-dream again... |
| 20:04.41 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 20:07.08 | paule32 | what role plays "deny" in example user "mark" in /etc/asterisk/manager.conf ? |
| 20:07.15 | paule32 | not allowed? |
| 20:07.28 | [TK]D-Fender | Correct |
| 20:08.34 | paule32 | when i try to connect from lan - the user is only visible in my net? |
| 20:08.38 | paule32 | 0.0.0.0 |
| 20:09.11 | [TK]D-Fender | Depends what you ALLOW |
| 20:14.54 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 20:55.32 | *** join/#asterisk u0m3 (~u0m3@109.96.148.19) |
| 20:59.17 | *** join/#asterisk m0sphere (~m0sphere@S01060018e78c9cff.cg.shawcable.net) |
| 21:00.01 | m0sphere | anyone ever use google tts and accidently get themselves blocked by google temporarily? |
| 21:07.27 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 21:12.57 | paule32 | so |
| 21:13.34 | ChannelZ | us |
| 21:14.10 | paule32 | i login in into pbx, it seems successfull, because CLI response with: == Manager 'user' logged in from x.x.x.x |
| 21:14.50 | WIMPy | This can't end well. |
| 21:14.51 | *** join/#asterisk iulhk (~iulhk@bba589495.alshamil.net.ae) |
| 21:14.52 | paule32 | when i close the programm, the message ".. logged if from ..." will apear |
| 21:15.49 | paule32 | now, i have a loop, in it, i try to recieve some bytes from pbx server |
| 21:16.04 | paule32 | but it will be fail |
| 21:16.25 | *** join/#asterisk spditner (~simon@206-248-134-67.dsl.teksavvy.com) |
| 21:16.31 | [TK]D-Fender | paule32: Why are you messing with AMI at all right now? |
| 21:16.45 | [TK]D-Fender | paule32: there is no point if you haven't set everything else up first |
| 21:17.16 | spditner | If I have two SIP peers, one with disallow=all;allow=ulaw, and the other with disallow=all;allow=g722, asterisk should step in and transcode, correct? |
| 21:17.50 | WIMPy | yes |
| 21:19.19 | paule32 | ah, the loop was fail, get message with 27 bytes, and then by next read: "Asterisk Call Manager/1.1" |
| 21:20.05 | paule32 | i see, it would be useful to read the new documentation |
| 21:22.02 | paule32 | cool |
| 21:22.50 | paule32 | in the 3 read, i get "Success" |
| 21:23.12 | paule32 | which means, the login was correct |
| 21:24.10 | ChannelZ | praisings to the lord! |
| 21:34.27 | *** join/#asterisk mmikeym (~mikeym@184.70.65.118) |
| 21:34.57 | paule32 | i will not be a lord, i will that computers do what i want - in constructive way |
| 21:37.27 | paule32 | i must say: excelent api documentation |
| 21:43.32 | *** join/#asterisk lorsungcu (~anonymous@74-95-143-116-Minnesota.hfc.comcastbusiness.net) |
| 21:50.05 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 21:51.29 | paule32 | PlayDTMF - how can i open a channel? or is it opeb by login? |
| 21:51.54 | paule32 | opeb = open |
| 21:53.25 | [TK]D-Fender | paulI cannot stress this enough .... AMI HAS NOTHING TO DO WITH THIS. |
| 21:53.34 | [TK]D-Fender | paule32: You are using the wrong tool for the job |
| 21:53.44 | [TK]D-Fender | paule32: AMI is not for PROCESSING CALL |
| 21:53.46 | [TK]D-Fender | +s |
| 21:58.32 | paule32 | ok, must left, tomorrow new woork week .. till then thanks for listen |
| 22:24.34 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 22:35.12 | *** join/#asterisk g_r_eek (~g_r_eek@46-236-35.adsl.cyta.gr) |
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| 23:13.43 | m0sphere | is anyone at all using googletts.agi? I need to confirm if its broken for everyone, or if theres something wrong with mine |
| 23:23.09 | [TK]D-Fender | you last asked about a temporary lockout... this sounds like longer term now... |
| 23:24.48 | m0sphere | me? |
| 23:25.12 | [TK]D-Fender | [15:59]m0sphereanyone ever use google tts and accidently get themselves blocked by google temporarily? |
| 23:25.19 | m0sphere | yea |
| 23:25.23 | [TK]D-Fender | Apparently |
| 23:26.10 | m0sphere | i went to the 'sorry' page for google thinking that would work, but it didnt, so i'm not sure if perhaps google changed something on the back end that prevents the agi script from working, or if somethinge else is wrong |
| 23:26.34 | *** join/#asterisk traph (~traph@unaffiliated/traph) |
| 23:27.44 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 23:27.56 | mattwj2002 | hi everyone :) |
| 23:28.17 | WIMPy | lo |
| 23:28.28 | mattwj2002 | hey WIMPy |
| 23:29.17 | mattwj2002 | on g729 if you have licenses on the softphones and your provider is g729.....when do you need an asterisk g729 license? |
| 23:29.22 | mattwj2002 | thanks in advance |
| 23:30.53 | WIMPy | Whenever Asterisk has to do something to/with your audio. |
| 23:31.14 | mattwj2002 | ah |
| 23:31.17 | WIMPy | And in that case you'd even need two. |
| 23:31.28 | [TK]D-Fender | Where "anything" requires transcoding that is |
| 23:31.51 | mattwj2002 | so definitely for a conference bridge |
| 23:31.58 | WIMPy | Anythign that doesn't? |
| 23:32.03 | WIMPy | yes |
| 23:32.18 | mattwj2002 | I think I am going to avoid g729 |
| 23:32.19 | mattwj2002 | :) |
| 23:32.33 | mattwj2002 | there are a lot of really good low bandwidth codecs |
| 23:32.34 | [TK]D-Fender | [18:31]WIMPyAnythign that doesn't? <- straight recording doesn't have to necessarily |
| 23:32.52 | [TK]D-Fender | WIMPy: playback only does if the source file isn't g729 |
| 23:33.01 | *** join/#asterisk lorsungcu (~anonymous@74-95-143-116-Minnesota.hfc.comcastbusiness.net) |
| 23:33.12 | WIMPy | Well, but you'd need a licence at another time then. |
| 23:33.35 | [TK]D-Fender | There are a number of clear places... |
| 23:33.45 | [TK]D-Fender | sneakiest of which being inband indications.... |
| 23:34.18 | [TK]D-Fender | Which should really be a goal on the tracker..... |
| 23:34.23 | [TK]D-Fender | to eliminate |
| 23:40.14 | mattwj2002 | so is anyone working on any interesting asterisk projects? |
| 23:41.00 | [TK]D-Fender | Depends on your idea of interesting I guess |
| 23:41.20 | mattwj2002 | true :) |
| 23:41.29 | WIMPy | So true. |
| 23:46.34 | [TK]D-Fender | Almost a decade back was already using it as a jukebox and to make me coffee... |
| 23:48.08 | mattwj2002 | hehe [TK]D-Fender sounds interesting |
| 23:48.09 | mattwj2002 | :) |
| 23:48.45 | mattwj2002 | [TK]D-Fender: I have the local weather radio stream on mine |
| 23:49.04 | *** join/#asterisk cyborg-one (~cyborg-on@31.31.108.80) |
| 23:50.22 | *** join/#asterisk Tim_Toady (~fuzzy@83.212.108.130) |
| 23:53.29 | mattwj2002 | speaking of weather.... |
| 23:53.52 | mattwj2002 | if you guys want to test your astereisk box on a nice test number try this link |
| 23:53.57 | mattwj2002 | http://www.nws.noaa.gov/pa/recordedforecasts.php |
| 23:54.07 | mattwj2002 | weather from around the US |
| 23:54.08 | mattwj2002 | :) |
| 23:54.17 | WIMPy | To test what? |
| 23:54.37 | mattwj2002 | like a sip trunk or pstn connection |
| 23:54.44 | mattwj2002 | it just play the weather |
| 23:55.06 | mattwj2002 | *plays |
| 23:55.13 | WIMPy | Don't you have test numbers for that? |
| 23:56.13 | mattwj2002 | good point |
| 23:58.45 | [TK]D-Fender | His wheel will be rounder, I'm sure of it |