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01:05.39 | dijib | anybody experiences with asterisk11 & dahdi? |
01:05.41 | dijib | also fax. |
01:06.06 | WIMPy | ~ask |
01:06.06 | 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. |
01:06.11 | dijib | oi |
01:06.16 | dijib | infobot how are you tonight? |
01:06.17 | infobot | dijib: I think you lost me on that one |
01:06.17 | dijib | :) |
01:07.34 | dijib | i guess what im trying to say is same => n,MeetMe(${EXTEN},dsFMcorx);; is crashing. |
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01:08.18 | WIMPy | Why do you still use MeetMe? |
01:08.42 | dijib | well then what is dahdi's application? |
01:08.56 | dijib | Bridge? or something man i guess thats root cause. wrong app. |
01:09.15 | WIMPy | But a crash is never good. You should open an issue on jira for that. |
01:09.30 | WIMPy | dahdi is for some telephony hardware. |
01:09.49 | WIMPy | Just as tha name says. |
01:09.54 | WIMPy | ~dahdi |
01:09.54 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
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01:10.34 | dijib | then what was the bridge dependent on the dahdi drivers? |
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01:10.51 | WIMPy | That was MeetMe. |
01:11.21 | dijib | no it didnt crash sorry call failed... im so melodramatic... THIS IS NOT A PRODUCTION ENVIRONMENT. lower severity. |
01:11.38 | WIMPy | But since Asterisk 10 ConfBridge might be a better solution. |
01:11.39 | Orii | hey has asterisk been working better on freebsd? last time i heard it was still a little iffy |
01:13.02 | dijib | thanks wimpy you saved the day. just a little extensions-fu here and i will br fixed right up |
01:13.15 | dijib | but then i have a fax issue. but let me fix this first. |
01:14.15 | Orii | also is there a new version of the O'reilly Asterisk book? |
01:14.26 | WIMPy | ~book |
01:14.26 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
01:15.30 | Orii | thank you |
01:15.51 | Orii | anyone able to answer my question about asterisk support on freebsd? |
01:16.46 | WIMPy | can't, but unless you want to use hardware interfaces, it shouldn't make a difference. |
01:18.28 | Orii | cool |
01:18.35 | Orii | what if i do? |
01:19.57 | WIMPy | There is a dahdi-freebsd. That's all I can say. |
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01:24.12 | dijib | core show modules |
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01:28.59 | dijib | ok core show applications does not show Confbridge |
01:29.46 | WIMPy | module load app_confbridge |
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01:30.06 | WIMPy | It might just lack a valid config file. |
01:30.11 | dijib | unable |
01:31.36 | dijib | ls -asl /usr/lib/asterisk/modules/ | grep app_confbridge.so |
01:31.37 | dijib | 1752 -rwxr-xr-x 1 root root 1792071 Oct 16 22:13 app_confbridge.so |
01:32.27 | WIMPy | Turn up verbose and debug. It might tell you why it doesn;t load. |
01:33.27 | dijib | newp |
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01:42.44 | dijib | WIMPy: for some reason i cant unload it or load it, core show applications shows ConfBridge and im pussled |
01:42.47 | dijib | puzzled |
01:43.41 | dijib | app_confbridge.so Conference Bridge Application 0 |
01:43.41 | dijib | 1 modules loaded |
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01:45.28 | parasitodelsur | dijib!!!!!!! |
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01:46.29 | WIMPy | Hmm. Try to noload it in modules.conf, restart, turn up verbose and debug and then try to load it manually again. |
01:46.39 | parasitodelsur | dijib: how you been? |
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01:47.52 | dijib | ive been alright thanks parasitodelsur is that a new handle? |
01:50.32 | dijib | after noload |
01:50.32 | dijib | Unable to load module app_confbridge.so |
01:50.33 | dijib | Command 'module load app_confbridge.so' failed. |
01:50.51 | dijib | same without .so |
01:50.56 | dijib | srry mistake |
01:51.41 | parasitodelsur | dijib: nope is a secondari |
01:51.44 | parasitodelsur | is me SeRi |
01:51.52 | parasitodelsur | I have a ghost |
01:51.54 | parasitodelsur | fixing now |
01:51.55 | dijib | yooooo hommie |
01:52.02 | dijib | crazy it has been forever |
01:52.19 | dijib | i need to get my asterisk 11 fax working. |
01:52.37 | parasitodelsur | hahaha nice seeing you again! |
01:53.03 | dijib | definitly... i was just looking through my dailplan and thought of the conferences we had... ive got a confbridge issue |
01:53.31 | parasitodelsur | what is going on? |
01:55.00 | SeRi | dijib: you still have your bridge upo? |
01:56.02 | WIMPy | No output with debug and verbose? Do you have debung enabled in logger.conf? |
01:56.39 | dijib | no thats the problem... i need a bridge for conerence calls and MeetMe and ConfBridge have both died in my asterisk 11-rci upgrade |
01:57.39 | dijib | the problem is app_confbridge.so is not working. the call dies when initiated. i also cannot load reload or unload it, yet it does get loaded on a core restart |
01:57.52 | dijib | the file exists |
01:57.54 | WIMPy | There's an rc2. |
01:57.59 | dijib | securities are correct |
01:58.03 | dijib | there is?!!? |
01:58.06 | dijib | oml |
01:58.36 | dijib | wait no i must be running rc2, installed it last week |
01:58.51 | dijib | crap i have rc1 |
01:59.01 | dijib | thats core show version |
01:59.25 | SeRi | dijib: upgrade |
02:00.31 | WIMPy | Maybe I shoudl do that as well... |
02:01.32 | dijib | im upgrading now... :S |
02:01.38 | WIMPy | Ugh. Tons of changes. |
02:01.43 | dijib | are their? |
02:01.54 | dijib | lots to break |
02:02.22 | WIMPy | Since my last update, yes. |
02:02.35 | WIMPy | That was a fourtnight ago. |
02:02.47 | dijib | see that |
02:03.29 | SeRi | I am still put with 1.8.14.1 :) |
02:04.52 | WIMPy | wonders why there's a substantial pause after Installing modules from addons... |
02:05.30 | dijib | i need FAXing |
02:05.57 | SeRi | faxing is also not working? |
02:07.00 | dijib | yes |
02:07.09 | dijib | has not worked for a long while |
02:07.10 | dijib | lol |
02:08.40 | SeRi | lol |
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02:09.42 | dijib | now i have a pjproject error on install |
02:10.18 | dijib | this: |
02:10.20 | dijib | http://forums.asterisk.org/viewtopic.php?p=179485 |
02:10.26 | newtonr | WIMPy: you at Astricon? |
02:10.44 | WIMPy | Nope. Several thousands of kilometes too far away. |
02:10.45 | SeRi | man CC is undergoing a massive DDoS |
02:10.52 | SeRi | WIMPy: lol |
02:11.26 | newtonr | WIMPy: ahhh |
02:23.38 | SeRi | uggghhh been working on rwiring my network |
02:24.08 | SeRi | I hate making cat6 cables |
02:26.51 | dijib | why? |
02:28.45 | dijib | make[5]: ccar: Command not found |
02:28.46 | dijib | make[5]: *** [../lib/libpj-x86_64-unknown-linux-gnu.a] Error 127 |
02:30.55 | SeRi | dijib: beacuse they are a bitch to make unlike cat5 cat6 uses this stupid inserts for RJ-45 |
02:31.42 | WIMPy | Don't use them if you don't like them. |
02:35.14 | dijib | GIG-E is worth it |
02:36.31 | WIMPy | loves distcc |
02:38.45 | dijib | im not so much in love with it |
02:39.13 | dijib | there we go... past that |
02:39.33 | dijib | had to do a make distclean in res/pjproject |
02:39.36 | dijib | then reconfigure |
02:40.26 | WIMPy | rc2 doesn't look good on my netbook so far. |
02:40.36 | dijib | then reconfigure |
02:40.48 | dijib | netbook why? |
02:40.57 | WIMPy | play |
02:41.06 | dijib | Asterisk 11.0.0-rc2 built by root @ swissarms on a x86_64 running Linux on 2012-10-22 02:32:46 UTC |
02:42.06 | WIMPy | Looks like the pjlib stuff didn't like distcc :-( |
02:42.29 | SeRi | WIMPy: mehhhh I am just bitching |
02:42.32 | SeRi | :) |
02:43.10 | SeRi | GigE POE are expensive |
02:44.26 | dijib | ok i still have the same app_confbridge.so issue |
02:44.51 | WIMPy | Hmm. sip still needs websockets? |
02:45.15 | dijib | ? |
02:45.27 | WIMPy | I thought that was changed. |
02:45.36 | dijib | dependency of asterisk? |
02:45.57 | WIMPy | inside Asterisk |
02:46.02 | dijib | ya |
02:47.14 | WIMPy | So technically sip needs to be in extended support now since it depends websockes which are in extended support or what? |
02:51.24 | dijib | your telling me. |
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02:54.42 | dijib | i cant for the life of me fix this. |
03:02.58 | dijib | installing 10.9.0 |
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03:39.23 | dijib | anybody a 10.9.0 fax expert? |
03:44.07 | josefig | is there an API to handle the regular features of asterisk ? something like |
03:44.19 | josefig | all the features in an abstraction layer ? |
03:48.55 | dijib | no clue |
03:49.00 | dijib | what are you trying to do |
03:49.56 | josefig | dijib: an api to handle some users in the cloud |
03:50.10 | dijib | no idea |
03:56.32 | DuPie | theres AMI |
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03:58.26 | josefig | DuPie: oh Manager Interface |
03:59.19 | DuPie | thats really what you're looking for |
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04:02.41 | josefig | DuPie: yes, maybe I can extend the AMI in order to support such a several things from my customers. |
04:03.54 | josefig | thanks |
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04:57.54 | oyugik | Hey guys what could cause an incoming call rule not to be obeyed |
04:58.04 | oyugik | in the case of a ring group |
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05:05.23 | dijib | Cannot reserve FAX session - session limit exceeded |
05:06.50 | kaldemar | oyugik: #freepbx |
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05:26.37 | oyugik | @kaldemar I am not using freepbx |
05:27.59 | Wiretap | oyugik: what are you using then |
05:28.07 | Wiretap | asterisk doesn't have 'ring groups' or 'incoming call rules' |
05:28.35 | oyugik | I am asterisk |
05:28.41 | oyugik | and dahdi |
05:28.51 | oyugik | with principles from asterisk gui |
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05:29.15 | oyugik | I have installed asterisk-gui to manage |
05:30.35 | oyugik | sorry for the confusion I am using asterisk gui |
05:32.28 | oyugik | this is how my extension.conf looks like |
05:32.47 | oyugik | and this is the dump from the debugger |
05:33.10 | oyugik | http://pastebin.com/zrGaRpsp --> extensions.conf |
05:33.45 | oyugik | http://pastebin.com/P6kqB5XC --> asterisk debug dump |
05:41.47 | ChannelZ | Doesn't look like any of your devices are actually working |
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05:44.24 | oyugik | eh, really |
05:44.33 | [TK]D-Fender | None registered |
05:45.01 | [TK]D-Fender | helps to actually look at the status of what you think you should be calling... |
05:45.01 | oyugik | what could be causing thus problem? |
05:45.10 | [TK]D-Fender | NOT REGISTERED. |
05:45.42 | oyugik | what the extensions or the device? |
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05:45.54 | oyugik | are not registered |
05:46.42 | ChannelZ | SIP/6006, SIP/6004, SIP/6010... |
05:46.47 | [TK]D-Fender | ^^^ |
05:47.08 | oyugik | okay let me have them up |
05:48.52 | ChannelZ | wanders off to watch Homeland |
05:59.14 | oyugik | @ChannelZ now there is no activity on the debug mode |
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06:07.33 | kaldemar | oyugik: what debug mode? |
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06:08.58 | oyugik | when I check the logs at the terminal from #asterisk -r |
06:09.34 | [TK]D-Fender | "sip set debug on" |
06:09.47 | oyugik | ok |
06:09.49 | [TK]D-Fender | If you aren't looking with SIP DEBUG enabled then you aren't really looking... |
06:09.57 | [TK]D-Fender | Same goes for any in-depth call debugging |
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06:10.26 | oyugik | done |
06:10.33 | kaldemar | if "asterisk -r" is all you're doing, you're not even seeing verbosity. "core set verbose 10" |
06:10.45 | oyugik | ok |
06:11.06 | oyugik | Verbosity was 3 and is now 10 |
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06:14.04 | oyugik | the thing that amazes me is that sometimes the calls come in and then they stop mysteriously |
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06:46.57 | mirela666 | oyugik: bast thing is to sniff traffic and try to catch that situation, and when it happens you can see who sends hangup and why |
06:49.32 | ChannelZ | I think he meant the calls stop coming |
06:50.01 | ChannelZ | But who knows really, with as little information as we're getting |
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07:02.29 | dijib | soup ChannelZ? |
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07:08.34 | mirela666 | yep lol |
07:10.40 | ChannelZ | mmmmm soup |
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08:40.33 | oquidave | Hello, i've properly installed configured odbc with asterisk and now am now am at the point of configuring res_odbc.conf to allow asterisk to connect to mysql through odbc. i've configure the res_odbc.conf file but when i odbc show from the asterisk CLI, nothing shows...here's my config file and other info. thanks |
08:40.38 | oquidave | http://pastebin.com/C93SKtX8 |
08:41.50 | oquidave | however,my odbc.ini is not in /etc/odbc.ini but in /etc/unixODBC/odbc.ini could this be the problem? |
08:42.00 | wdoekes | oquidave: first make sure isql works |
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08:44.18 | oquidave | wdoekes: it does check out http://pastebin.com/PG37B5eP |
08:45.46 | wdoekes | core set verbose 20 |
08:45.49 | wdoekes | module reload res_odbc.so |
08:48.18 | oquidave | wdoekes: it says res_odbc.c:820 load_odbc_config: Limit should be a number, not a boolean: '0'. Disabling ODBC class 'asterisk' ...i've checked the limit and it's 0 ...limit => 0 ...so to what figure should i change it? |
08:49.03 | wdoekes | see configs/res_odbc.conf for an explanation what limit does |
08:49.19 | oquidave | wdoekes: seen it...am changing it to 5 |
08:50.27 | oquidave | wdoekes: thanks it's now up and running!!!! |
08:50.54 | oquidave | wdoekes: " core set verbose 20" why does it have to be 20? |
08:51.27 | wdoekes | because then you get *all* verbose messages |
08:51.36 | wdoekes | but you don't need it unless you're debugging something |
08:51.50 | oquidave | wdoekes: so i can change it back to 3 |
08:52.01 | wdoekes | (I think 11 is the highest number used, so 20 is safe) |
08:52.03 | wdoekes | sure |
08:52.21 | oquidave | wdoekes: okay thanks |
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09:02.06 | ChannelZ | I want to say last I looked 5 or 6 was the max ever used |
09:04.38 | oquidave | ChannelZ: okay, i'lll go with 5. thanks |
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09:05.50 | ChannelZ | It doesn't hurt or anything, just that it's not limitless. I run normally at 3 which pretty much tells me everything I've ever needed to know |
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09:48.01 | awk | Hi guys, currently my system writes cdr's to flat file and mysql db. and the only option I can see under cdr.conf is enable = yes / no... Now how can I have it so it doesn't write the flat file but keeps writing to the DB? |
09:48.03 | awk | Many thanks! |
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09:51.41 | wdoekes | awk: ; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv |
09:51.44 | wdoekes | ; file, define the [csv] category in this file. No database necessary. The example |
09:51.47 | wdoekes | ; config files are set up to provide this kind of output by default. |
09:52.03 | wdoekes | the inverse is most likely true too |
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10:03.46 | awk | I found what I had to disable just cdr_custom module :) |
10:06.19 | oyugik | hey guys I found a solution for my problems earler |
10:07.05 | oyugik | i was using a context "from-pstn" that wasnt explicitly defined in my extensions.conf |
10:07.19 | oyugik | hence causing difficulties in the hang up event |
10:08.11 | oyugik | hence channels got confused |
10:08.22 | oyugik | in the execution of dialplan |
10:08.43 | oyugik | I probably should blog about it. But what the heck |
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10:23.29 | oquidave | hello guys am just realling on asterisk realtime now and i've discovered there's a db schema/structure that you've to follow which i've read. However, to make my development faster, are there some .sql files with this schema that i can just import into my mysql db and simply fill in the values? thanks |
10:25.58 | oquidave | something similar to the sample files for instance |
10:28.42 | wdoekes | oquidave: contrib/realtime |
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10:32.05 | oquidave | wdoekes: exactly what i need! thanks again...now i can just import the sql file |
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10:55.06 | nappy | hello everyone... |
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10:56.52 | nappy | would like to ask if there is a solution or somebody has implement a web conference in asterisk? |
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10:59.08 | oquidave | asterisk realtime is really cool. now i can build a web app where users can register into the system so easily without tinkering with the sip.conf file. |
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11:07.22 | nappy | im using trixbox... is there any application/module for that?? |
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11:07.50 | nappy | hi oquidave,,.. is that for web conferencing? |
11:08.44 | nunne | On a "powerfull" system (2x6core xenon, 8-16gb ram) how may registrations (pref. with qualify) can asterisk 10 handle? or is it still recommended that openser handles the regs? |
11:11.37 | kaldemar | nunne: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
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11:35.33 | gavimobile | how can I view the exact caller id that the destination caller sees? |
11:36.21 | WIMPy | Place a web cam above hos phone. |
11:36.25 | WIMPy | his |
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11:37.54 | WIMPy | What exactely do you want to know? |
11:38.34 | nappy | WIMPy: lol |
11:38.35 | gavimobile | WIMPy: my diagnosis is unclear as of now, however "some" landline destination parties see my telephone number as an international number |
11:39.02 | gavimobile | I used the word some cause only some complain, but I believe that other landlines using the same telco see the number as it should |
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11:39.15 | WIMPy | What do you send and what kind of connection? |
11:39.49 | gavimobile | I don't understand the question |
11:39.53 | gavimobile | I am using sip |
11:40.19 | gavimobile | and it only happens with this dialplan exten => _0NNXXXXXX,1,Set(CALLERID(num)=${OUTCID2}) |
11:40.20 | gavimobile | <PROTECTED> |
11:40.35 | gavimobile | "sometimes" |
11:40.36 | fredericve | On ubuntu lucid, while building asterisk 11 I get this error: |
11:40.37 | fredericve | i486-linux-gnu-ar: Command not found |
11:40.44 | oquidave | nappy: no, still just playing with asterisk.... |
11:40.57 | fredericve | anyone knows how to fix this? |
11:41.06 | WIMPy | Well, it might be them using different routes then. |
11:41.09 | nappy | ohhh okay.. i thought its used with conferencing |
11:41.28 | gavimobile | WIMPy: I have a log with sip debug when calling a destination party which sees an international number |
11:41.39 | BorjaGVO | Hi everyone...I've got some weird thing happening here. Asterisk seems not finding a file (vm-youhave) although it's there, under /var/lib/asterisk/sounds/es/vm-youhave.gsm, Asterisk is not able to open it. I even did chmod 777 to the file and no sucess in case it could be permission issue. Anyone can help me with this? |
11:41.47 | WIMPy | fredericve: Does it have some pseudo package like build-essentials? |
11:41.49 | gavimobile | WIMPy: if it uses different routes, does this mean its not in my control? it never happenes when I call cellphones |
11:42.30 | fredericve | WIMPy: yes build-essential is installed |
11:42.46 | gavimobile | BorjaGVO: in your dialplan, try to leave the extention out |
11:42.54 | oquidave | BorjaGVO: make sure the file extension is not included |
11:43.01 | WIMPy | gavimobile: Possibly. Maybe you can use another format that won't show the issue. But you should check with your ITSP what they expect. |
11:43.25 | gavimobile | WIMPy: what do you mean by other format? im unclear |
11:43.36 | gavimobile | but thanks, I will speak with my telco provider. |
11:43.39 | BorjaGVO | gavimobile, oquidave: I'm using FreePBX so I don't know if modifying the dialplan is a good idea |
11:43.46 | BorjaGVO | I mean....it worked before |
11:43.48 | oquidave | BorjaGVO: plus the file path should be correct e.g Playback(en/vm-youhave) |
11:43.54 | WIMPy | gavimobile: Like you sending your number in national or international format. |
11:44.08 | gavimobile | BorjaGVO: this is a channel for asterisk, not freepbx |
11:44.32 | BorjaGVO | Well, it's directly connected to asterisk... |
11:44.39 | gavimobile | WIMPy: well the weird thing is sometimes it works and sometimes it doesn't |
11:44.54 | WIMPy | BorjaGVO: Try #freepbx |
11:44.58 | gavimobile | BorjaGVO: you just said you don't know how to modify the dialplan |
11:44.59 | BorjaGVO | alright |
11:45.04 | BorjaGVO | yes |
11:45.13 | gavimobile | if you don't know how to modify the dialplan how are the folks in the channel suppose to help you? |
11:45.21 | BorjaGVO | I know how to..but I don't suppose to have to... |
11:45.30 | WIMPy | gavimobile: Unless you do it differently, it must happen somewhere else. |
11:45.35 | BorjaGVO | anyway, I'll try at #freepbx |
11:45.37 | BorjaGVO | thank you |
11:45.50 | gavimobile | BorjaGVO: for freepbx support you should speak to them in their channel |
11:45.56 | gavimobile | sorry :-p |
11:46.21 | gavimobile | WIMPy: this is why I want to monitor it.. in my cdr it doesn't show the caller id the party sees |
11:46.22 | BorjaGVO | gavimobile: np |
11:46.28 | gavimobile | this is why I asked my initial question |
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11:47.15 | gavimobile | if I can monitor what they see (without a webcam in front of their phone :-p) than I can troubleshoot when and where it happenes |
11:47.37 | WIMPy | gavimobile: If you explicetely set the callerID before dialing out, try to put a verbose in between. If you're not 100% sure it might do something. |
11:47.38 | gavimobile | WIMPy: unless you have another idea to approach the issue |
11:48.13 | WIMPy | gavimobile: Try to find the pattern when it works as expected and when not. |
11:48.22 | gavimobile | hrm... |
11:48.31 | gavimobile | im just thinking what I would put inside verbose() |
11:48.38 | gavimobile | callerid? |
11:48.43 | gavimobile | ${callerid} |
11:48.51 | WIMPy | CALLERID(num) |
11:48.53 | gavimobile | something like that |
11:48.57 | gavimobile | got it |
11:49.14 | gavimobile | WIMPy: ill go ahead and do that now |
11:51.31 | gavimobile | WIMPy: is this kosher? http://pastebin.com/ziUfFHkk |
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11:56.19 | fredericve | WIMPy: I fixed the error by symlinking /usr/bin/ar to /usr/bin/i486-linux-gnu-ar |
11:57.47 | WIMPy | gavimobile: yes |
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11:59.52 | gavimobile | WIMPy: still no go |
12:00.32 | WIMPy | Meaning? |
12:00.36 | gavimobile | the verbose works, and it shows which number is beind set by callerid |
12:00.53 | gavimobile | so I tried changing it to local so it wouldn't be international, but the call is still showing up as international |
12:01.06 | gavimobile | ill send a pastebin to help you understand more what im refering to |
12:02.06 | gavimobile | http://pastebin.com/xxxz7ZHK |
12:03.12 | gavimobile | the caller id which was set in example 2 is the caller id I want them to see. my trunk provider is from the states which is why I set the caller id the way it was in example 1. |
12:03.18 | WIMPy | So you have an idea, who to blame then. |
12:03.28 | gavimobile | my trunk provider? |
12:03.54 | WIMPy | But the two calls look the same. |
12:04.20 | WIMPy | Oh, you changed tit after the verbose. |
12:04.25 | WIMPy | -t |
12:04.56 | WIMPy | Yes, your provider or some other place that's outside of your control. |
12:05.00 | gavimobile | WIMPy: I lost you.. I thought the point to add verbose was to see in the output what number was being used in caller id |
12:05.21 | gavimobile | WIMPy: if its not in my providers control, than who else can I speak to |
12:05.32 | WIMPy | I see the difference in the voerbose output of the set. |
12:06.01 | WIMPy | Noone. Talk to your provider. If it's not them, they will have to find out who it is. |
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12:13.23 | gavimobile | WIMPy: thanks so much! |
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14:03.50 | devdvd | Hi all, I'm looking at setting up an * box to do 200 users, voicemail, web interface, 50 concurrent calls and g729 (assume 50 simultanious conversions). I realize this isn't a high amount for * but I was wondering if anyone here has experience with that many calls (using g729) and what kind of hardware you use. |
14:04.07 | devdvd | probably do conferencing as well |
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14:05.03 | oquidave | hello,am trying query a mysql db from within asterisk dialplan using odbc...http://pastebin.com/5XE62SbK ...however am finding it hard to debug where the problem could be. Could someone give me some leads. ive pastebin has enough info of what i've already done. thanks |
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14:07.13 | [TK]D-Fender | oquidave, Looks null... seems fine to me... |
14:08.08 | cusco | hello |
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14:08.27 | cusco | I was reading up on http://forums.asterisk.org/viewtopic.php?p=170058 but im not sure that my issue is the timming source |
14:08.38 | cusco | basically we have 2 PRI cards 4 span each |
14:08.40 | cresl1n | devdvd: Just buy a modern, moderately spec'd server |
14:08.55 | cusco | in one of them we have 2 PRI from telco1 and 1 PRI from telco2 |
14:09.20 | [TK]D-Fender | devdvd, Your users are expected to be remote from your server? |
14:09.22 | cusco | and what happens is that in telco1 some calls there is no audio from client calling in... |
14:09.38 | devdvd | TK, yes |
14:10.13 | kaldemar | oquidave: ${EXTEN} has 751 in it. your DB only has a row with "50" as exten. |
14:10.17 | [TK]D-Fender | devAnd basic box from today should be more than enough. I'd personally aim for an i5+ |
14:10.27 | [TK]D-Fender | oquidave, As kal said.... |
14:10.37 | [TK]D-Fender | kaldemar* |
14:10.48 | cusco | looking at dmesg I have lots of lines like: http://paste.debian.net/202642/ |
14:11.02 | cusco | can I not use telco's timing source and use a internal timing source instead? |
14:11.25 | cusco | I'm using: dahdi-linux-complete-2.5.0+2.5.0 |
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14:12.13 | oquidave | kaldemar: sorry, i think i get it now...i put 50 because am calling from extension 50...so the idea was that it takes the user extension as an argument!! thanks |
14:12.44 | oquidave | kaldemar: so i'll have to change it to caller id instead |
14:12.54 | oquidave | kaldemar: thanks again |
14:12.59 | kaldemar | oquidave: sounds like you need to give a thought or two on what an extension is in asterisk. |
14:13.59 | jeffspeff | ok, i'm missing something here. I'm trying to setup a secondary voicemail system. I'm using realtime configuration. when in asterisk console and I do a 'voicemail show users for PW' it says that the command failed. What could i be doing wrong? |
14:14.24 | jeffspeff | i'm using mysql for realtime driver. mysql is installed on the same localhost |
14:14.30 | oquidave | kaldemar: extensions are like code-lines in dialplan though they can also be used "phone numbers" for the user. I keep missing it up!! |
14:14.47 | devdvd | cuts jeffspeff |
14:14.59 | jeffspeff | ? |
14:15.06 | devdvd | <twister |
14:15.20 | jeffspeff | oh |
14:15.22 | devdvd | lol |
14:15.31 | jeffspeff | any ideas on this? |
14:16.24 | devdvd | when you do voicemail show users for ? |
14:16.28 | devdvd | does it show up in that list |
14:16.31 | jeffspeff | i've double checked the config files against my production system which also uses realtime for voicemail; the only difference is on the production system the voicemail was originally setup through flat file, then converted to realtime |
14:16.59 | jeffspeff | no |
14:17.21 | jeffspeff | i've got a sneaky suspicion that it's not actually parsing the database |
14:17.36 | devdvd | right, do you have anything else realtime on that box? |
14:17.38 | devdvd | or just voice mail |
14:18.00 | jeffspeff | voicemail is the only thing setup |
14:18.42 | devdvd | what does realtime mysql status tell ya |
14:19.07 | devdvd | ive never tried to setup voicemail in realtime, so im just stabbin at it here |
14:19.22 | jeffspeff | it shows that the db is setup |
14:19.55 | devdvd | whats your extconfig entry look like for that? |
14:20.02 | devdvd | the voicemail i mean |
14:20.21 | jeffspeff | voicemail => mysql,cisco,voicemail |
14:21.42 | jeffspeff | then in res_config_mysql.conf i specify: dbhost = 127.0.0.1 dbname = cisco dbuser = root dbpass = password dbport = 3306 dbsock = /va/rlib/mysql/mysql.sock requirements=warn |
14:21.43 | devdvd | oh |
14:21.49 | jeffspeff | obviously all on seperate lines |
14:21.55 | devdvd | try this |
14:22.17 | devdvd | move voicemail.conf to some other directory (like /home/asterisk or whatever) |
14:22.22 | devdvd | then just touch voicemail.conf |
14:22.30 | devdvd | then change ownership to the asterisk user |
14:22.37 | devdvd | make the voicemail.conf blank |
14:22.48 | devdvd | then restart * |
14:23.24 | jeffspeff | ok, done and same results |
14:23.31 | devdvd | ok |
14:23.44 | devdvd | hold let me try to set realtime up on my side and we'll work through it |
14:23.51 | jeffspeff | ok |
14:24.02 | kaldemar | jeffspeff: is "command failed" all it says? |
14:24.24 | cusco | can I dial a sip peer, but on dialplan before dialing set the fromuser SIP option ?? |
14:24.38 | cusco | overriding whatever is set in sip.conf ? |
14:24.43 | jeffspeff | localhost*CLI> voicemail show users for PW |
14:24.44 | jeffspeff | Command 'voicemail show users for PW' failed. |
14:25.38 | jeffspeff | cusco, no users being set in sip.conf. only inbound sip user from other system |
14:27.28 | cusco | ?= |
14:27.58 | cusco | I mean.. in sip.conf I have a peer from a remote sip gateway. I dial say SIP/peer/xxxxxxxxx ... |
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14:28.23 | devdvd | jeff, he wasn't responding to you |
14:28.30 | jeffspeff | oh |
14:28.35 | jeffspeff | :p |
14:28.38 | cusco | but I would like to be able to set the sip header "From: <sip:FROMUSER@...>" dinamically |
14:28.50 | cusco | in dialplan |
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14:29.31 | cusco | jeffspeff: reading on your issue do you have any vm module loaded in asterisk? also vm in extconfig set to use a database? |
14:29.52 | jeffspeff | yes, vm module is loaded, and yes set to use correct db |
14:30.27 | cusco | module show like vm; module show like voicemail |
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14:39.25 | devdvd | yea not working for me either |
14:40.28 | cusco | how do I increment a variable on ael? Set(a=$[${a} + 1]) ? |
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14:45.30 | jeffspeff | devdvd, i fixed it |
14:45.36 | devdvd | ah ok cool |
14:45.39 | devdvd | what'd ya do |
14:46.09 | jeffspeff | well in res_config_mysql.conf dpbass != dbpass |
14:46.12 | jeffspeff | simple type |
14:46.14 | jeffspeff | *typo |
14:46.23 | devdvd | interesting |
14:46.30 | jeffspeff | i knew i was missing something rediculously simple |
14:46.45 | devdvd | but it showed up as connecting fine with realtime mysql status |
14:46.54 | jeffspeff | yep |
14:47.36 | devdvd | interesting |
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15:03.07 | wtfitsme | Hello, I have an issue where a carrier is sending me media (RTP) on a port that is not in the range I specified in rtp.conf - This is happening on a 1.4.23 box. Shouldn't Asterisk negotiate this in the SDP body? |
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15:14.03 | BorjaGVO | New to Asterisk source code...I'm going over file.c, and I don't get this function: struct ast_filestream *ast_openstream(struct ast_channel *chan, char *filename, char *preflang) |
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15:14.57 | BorjaGVO | filename2 is set t0 null: char filename2[256]=""; right? where is the filename assigned? |
15:15.07 | danfromuk | Has anyone had problems integrating the new version of XLite into Asterisk? Specifically, the way that XLite forces a specific dialplan? |
15:15.12 | BorjaGVO | I'm not a programmer so I might be missing something...sorry. |
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15:16.39 | jpsharp | That's not being set to null, that's being set to an empty string. Big difference. |
15:16.59 | BorjaGVO | well...sorry |
15:17.10 | BorjaGVO | it is a big difference...right. Empty string... |
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15:18.14 | BorjaGVO | source code: http://doxygen.asterisk.org/asterisk1.0/file_8c-source.html |
15:18.25 | Qwell | so what are you asking? |
15:18.46 | [TK]D-Fender | danfromuk, Thre is no such "integration". It is a dumb softphone. |
15:20.14 | danfromuk | [TK]D-Fender: yes however the new version doesnt seem to be compatible with my dial plans, and even forces incorrect dialling. They've tried to make the dialling plan simpler, but in fact broken it. |
15:20.52 | [TK]D-Fender | show us where you even set it... |
15:21.11 | danfromuk | [TK]D-Fender: for example, the user dials 9 & local dial code & number, and xlite sends +countrycode & 9 & local dial code + number. |
15:21.31 | danfromuk | The 9 is messing everything up and I can't seem to remove it. I was wondering if anyones seen that with the latest version |
15:22.42 | danfromuk | http://imagebin.org/232838 |
15:24.22 | [TK]D-Fender | danfromuk, That looks like trash. I'd just strip it out.... |
15:24.36 | danfromuk | Can't. It insists on having numbers. |
15:24.50 | danfromuk | I think its a bug on counterpath's part |
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15:32.46 | danfromuk | Ok, got round it by adding x before each number dialled. That makes xlite think that i'm dialling a local extension and doesnt add any digits automatically |
15:34.58 | jpsharp | That's awfully presumptious of xlite. |
15:35.53 | danfromuk | Bit annoying that xlite now comes with an advert/integration with softphone.com. Are there any alternatives that I can offer my clients? How easy is it to make one? |
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15:39.15 | fullstop | itsp? |
15:39.47 | [TK]D-Fender | danfromuk, Yes there is an alternative... their COMMERCIAL products |
15:39.58 | [TK]D-Fender | X-Lite is for chumps |
15:40.27 | fullstop | I don't mean to intrude, but I'm considering an itsp again and I was wondering if there was an updated list. |
15:40.27 | danfromuk | Agreed however, I'm a pretty good programmer and therefore hate buying products that I know I could make. |
15:40.31 | fullstop | voip-info is kind of strange now |
15:42.19 | fullstop | infobot: itsp? |
15:42.19 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
15:43.25 | BorjaGVO | no one can help me with the code? Come one! ;) |
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15:49.47 | navaismo | what code BorjaGVO and what are you trying to do? |
15:50.25 | BorjaGVO | this code: http://doxygen.asterisk.org/asterisk1.0/file_8c-source.html |
15:50.31 | devdvd | BorjaGOV, try asking in #asterisk-dev |
15:50.45 | BorjaGVO | I'm trying to see where filename2 is assigned.. |
15:51.04 | Qwell | BorjaGVO: in the snprintf() calls |
15:51.15 | BorjaGVO | I mean, I just see an empty string (in function struct ast_filestream *ast_openstream(struct ast_channel *chan, char *filename, char *preflang) |
15:52.10 | jmls1 | ConfBridge([confno][,bridge_profile[,user_profile[,menu]]]) |
15:52.23 | jmls1 | doesn't this mean that all options are optional ? |
15:53.00 | BorjaGVO | @Qwell: alright,..didn't know that function...though it was obvious... |
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16:58.52 | edve | Hi people, i've got a question for all of you that could help me with Fail2ban with asteriskNow. Actually i've install the software and it seems it does'nt work correctly .. |
16:59.03 | d_preston215 | Is there a priority difference between a call generated via AGI and a call generated from a call file? |
16:59.37 | d_preston215 | I have a weird issue where it seems calls generated from a call file have a lower priority over other calls. |
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17:00.32 | JerJer | d_preston215: define priority |
17:05.37 | [TK]D-Fender | d_preston215, AGi does NOT generate "calls" |
17:06.01 | jpsharp | d_preston215: Well, you have to remember that call files are generated when asterisk gets around to scanning the spool directory, whereas a call generated through something like AMI or the originate command is done as soon as the command is entered. |
17:06.32 | jpsharp | s/call files/calls originated by call files/ |
17:06.54 | jpsharp | infobot: Thank you |
17:06.54 | infobot | jpsharp: de nada |
17:09.43 | jeffspeff | has anybody tried to do dial-up modem connection through a sip provder using a sip to pstn adapater? |
17:09.47 | jeffspeff | *adapter |
17:09.50 | jpsharp | And there is a 1 second pause at the beginning of every loop of the spool scan. |
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17:10.34 | jpsharp | jeffspeff: masochistic much? I've heard of people doing it and failing miserably at it. |
17:11.04 | jpsharp | Or rather, trying to do it. |
17:11.21 | jeffspeff | somebody was asking about changing their phones over to asterisk but, their business requires them to ocassionaly do a modem connection to medicare systems. |
17:11.39 | WIMPy | jeffspeff: Kind of. I tried a X.75 connection on my NGN line. It actually worked without issue apart from a horrible low speed. |
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17:11.59 | jeffspeff | it didn't sound like a good or stable idea to me, but didn't know how badly it would fail |
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17:12.14 | jpsharp | jeffspeff: Save yourself the heartburn and heartache and tell them to keep a POTS line for the modem. |
17:12.22 | JerJer | 9600 baud, if you are lucky |
17:12.34 | JerJer | but its not going to be reliable |
17:12.47 | jeffspeff | ok, thanks for confirming |
17:13.34 | WIMPy | Better tell them to exchange whatever they use for something that uses IP itself. |
17:13.56 | jpsharp | WIMPy: That would require the US government to come out of the technological dark ages. |
17:14.32 | JerJer | we've gotten the stamp machines to work by disabling all data compression and error correction (you will have to find the correct AT command string for your modem) |
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17:17.14 | saxa | hi everybody, its a long time since i lastly was here :) |
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18:04.45 | Micc_ | So are there any good pre astricon things to do today? |
18:05.39 | WIMPy | Get loaded? |
18:06.04 | edve | Anyone here knows about fail2ban ? |
18:08.36 | WIMPy | It's said it can be used to automatically shoot yourself in the foot. |
18:08.49 | jmetro | has anyone heard of transferring somehow breaking so that the transferred call goes to an outside number? somehow transferring to a 2 digit extension is hitting outside. |
18:09.28 | WIMPy | Looks like you have the wrong context somewhere. |
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18:17.44 | jmetro | Hm. Anything else you can think of besides the context? |
18:18.35 | WIMPy | no |
18:18.35 | dijib | anybody got a fax machine they can send me a test fax from? |
18:20.39 | WIMPy | If you want it via PSTN you should tell us, what country. |
18:20.50 | drzed | hi there, i'm connected to a PSTN via IO |
18:21.03 | drzed | hi there, i'm connected to a PSTN via ISDN, but foreign calls do not work |
18:21.08 | [TK]D-Fender | jmetro, Show us it actually happening... |
18:21.14 | dijib | doesnt matter. canada |
18:21.30 | drzed | the error msg is: Channel 0/1, span 1 got hangup request, cause 3 |
18:21.34 | WIMPy | drzed: define "doesn't work" |
18:21.55 | dijib | toll-free canada |
18:22.17 | drzed | i can dial national numbers, but internationals have this problem |
18:22.22 | WIMPy | drzed: Are you allowed to call abroad? |
18:22.30 | drzed | yes, i am |
18:22.42 | WIMPy | dijib: Probably not toll free from abroad. |
18:22.49 | drzed | at least it was working perfectly fine with my old telephone system |
18:23.14 | drzed | i thoght it could have to do something with the foreign double zero perfix |
18:23.16 | dijib | then i have another number |
18:23.25 | drzed | but removeing them does not help |
18:23.35 | dijib | sorry im mixing my issue ith drzed's |
18:24.05 | WIMPy | drzed: "No route to destination" is not a common one. Have you tried different destinations? |
18:24.15 | saxa | hey WIMPy :) |
18:25.22 | WIMPy | drzed: Does the cause come from your telco or farther away? |
18:26.03 | drzed | WIMPy: how can i find outß |
18:27.05 | WIMPy | drzed: You will see the location if you enable pri debug. |
18:29.34 | WIMPy | drzed: Have you tried different destinations? |
18:29.46 | drzed | unfortunately my version does not support libpri |
18:29.50 | Micc_ | Anyone have any experience playing with asterisk SCF? Is it getting good yet? |
18:29.56 | drzed | yes i tries three diffrent countries |
18:30.21 | [TK]D-Fender | Micc_, It's dead. |
18:30.58 | WIMPy | drzed: You said you're connected via ISDN |
18:31.20 | drzed | yess, indeed i am |
18:31.29 | [TK]D-Fender | Micc_, http://blogs.digium.com/2012/09/14/asterisk-scf-pause/ |
18:31.39 | WIMPy | drzed: So how are you connected? |
18:32.00 | drzed | using an isdn card |
18:32.12 | WIMPy | misdn? lcr? |
18:32.13 | drzed | on the pci slot |
18:32.16 | drzed | zaptel |
18:32.36 | WIMPy | The you are using libpri. |
18:33.14 | drzed | ups, you are right |
18:33.17 | WIMPy | And zaptel has been replaced by dahdi several years ago, BTW. |
18:33.49 | drzed | WIMPy: http://nopaste.info/338996f93b.html <-- debug log |
18:34.48 | cusco | Im reading on again about dahdi timing sources... telco provides us the service, so we should use its timing source, right? |
18:34.51 | WIMPy | drzed: Yes, you're setting the type of number to national. That doesn't work for international, obviousely. |
18:34.55 | WIMPy | And not for local, either. |
18:35.05 | cusco | could I use a internal timing source? such as timerfd ? |
18:35.16 | coppice | [TK]D-Fender: was there ever any significant work on SCF? |
18:35.34 | WIMPy | drzed: See the last one on http://voice.yeti.dk/Asterisk_vs_ISDN/5 |
18:36.16 | [TK]D-Fender | coppice, Never really looked at it myself. I figure it's all vaporware until a real beta test or RC anyway, and in this case proven to be a worthwhile metric for how much I should care... |
18:37.08 | coppice | [TK]D-Fender: I looked a couple of times over the last couple of years and it mostly seemed to be hot air |
18:38.24 | [TK]D-Fender | coppice, You already know that I hold your opinions in high regard. I suspect that their methodologies proved to be more stop-gap than anything revolutionary and the code just swam upstream to Asterisk itself |
18:38.54 | [TK]D-Fender | coppice, Diminishing returns plus probable challenges in marketability |
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18:40.27 | drzed | WIMPy: great, thx very much! |
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18:44.19 | WIMPy | Added a litte explanation. |
18:45.44 | fr0ggie | Hello all! I've got asterisk setup with a Android SIPUA (remote, behind NAT) over UDP connecting. Asterisk make calls via google talk (gv) and i dont hear echo on my end.. However, the other end hears themself echoing pretty bad. Anything I can do? |
18:45.55 | drzed | WIMPy: perfect, thx again |
18:46.05 | WIMPy | Is that correct terminology? Do you call the prefix to place national calls (if required) a "trunk prefix" or is there a better term? |
18:46.18 | drmessano | fr0ggie: What is the other end using? |
18:46.25 | fr0ggie | drmessano: PSTN |
18:46.39 | cusco | how would I tell dahdi to use res_timing_timerfd.so as a timing source? |
18:47.19 | fr0ggie | Android phone (builtin UA) -[SIP]-> asterisk -[gtalk/jabber]-> google voice -[PSTN]-> Other end |
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18:49.20 | drmessano | fr0ggie: I suppose its possible your endpoint is looping the speaker audio back down the line.. Are you on speakerphone? |
18:50.01 | fr0ggie | nope |
18:50.27 | fr0ggie | speakerphone annoyingly enough doesnt work in android's builtin sipua |
18:51.22 | drmessano | Tried another client? |
18:51.54 | fr0ggie | Not yet, wasnt sure if could be my client since the other end is only one that hears any echo at all |
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18:52.57 | drmessano | It's not Asterisk.. so it's either your client or something on the PSTN end of the google voice connection |
18:53.32 | fr0ggie | drmessano: Not what i was hoping to hear :) I was hoping for a config edit and it work, ill try different client, back soon |
18:53.39 | drmessano | ok |
18:53.51 | fr0ggie | drmessano: any recommendations for on android? |
18:54.01 | drmessano | No, I hate Android :) |
18:54.55 | fr0ggie | drmessano: me too, but there isnt so much choice until i cobble a working linux onto my phone and free it from the clutches of google >:) |
18:54.57 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
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18:57.01 | drmessano | I like iOS.. it just works. No need to tinker or guess or wonder if your particular android implementation is borked because your provider decided to tweak it |
18:57.50 | drmessano | I spend all day dealing with crappy android devices that stop sending email, stop receiving email, start authenticating using old credentials, and other mysterious crap. Cant stand it |
18:57.53 | fr0ggie | drmessano: nah, i run cyanogen, my provider sucks |
18:58.23 | drmessano | Besides that, it's Java based. Nothing that I hate more than Java |
18:58.27 | fr0ggie | Problem i have with iOS devices is the fact that im stuck running whatever OS they decide i should run, its like i dont really own the hardware merely paid a bunch to lease it |
18:58.36 | fr0ggie | drmessano: Amen. java is like sewage |
18:58.44 | fr0ggie | When i see it or smell it, i vomit all over |
18:58.53 | cusco | drmessano: whenever I try to explain that to people I tend to get bashed, so I just stay quiet |
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18:59.44 | fr0ggie | i only use android because $50 for android phone beats $500 for ipod phone (besides, i already have ipod) |
18:59.49 | cusco | and... jailbreak your ios, and you can controll your hardware |
18:59.56 | drmessano | I like the fact that my IOS devices work. Like, when I want to make a call, or send an email, or need to VNC/RDP/SSH into a machine while I am heading down the highway 4 hours from home, I know it works. I wouldn't last a week praying over an Android device to comply |
19:00.37 | fr0ggie | cusco: can i port my kernel to it? sadly not, apple has cryptographic controls that force you into iOS |
19:00.53 | drmessano | "It's fine.. if it starts acting up, I just reboot it" "Oh, and what happens if it fucks up 4 times in one day?" "I just reboot it 4 times, duh" |
19:00.55 | drmessano | Yeah.... |
19:00.56 | fr0ggie | my ipod, it is jailbroken 4.3.? |
19:01.11 | cusco | fr0ggie: I'm sure you can, there is even a project on running android on ios.. once the jailbreak hacks the boot firmware |
19:01.12 | fr0ggie | i definitely dont like android. Not one bit |
19:01.45 | fr0ggie | until obama gives me my change back, i cant afford to buy more shiney things :( |
19:01.58 | cusco | I agree on that part |
19:02.16 | saxa | http://pastebin.com/j35p5NUr |
19:02.21 | drmessano | I agree completely there.. if you want hardware freedom, and IOS device won't give you that. I prefer IOS.. The hardware is pretty and all, and works.. but the OS is what sells me on Apple devices. |
19:02.29 | saxa | any idea why i'm getting that error ? |
19:03.33 | drzed | fr0ggie: ++ |
19:03.35 | fr0ggie | drmessano: i like iOS, i even sorta like the shiney hardware it runs on.. but i feel like i am indefinitely renting instead of buying.. like.. if you piss apple off, no more SHSH signing == no more restore/updates (unless of course you save them, and you risk getting stuck without apps then, since no app store) |
19:04.50 | navaismo | saxa, routing issues |
19:05.04 | drmessano | I save all my hardware freedom for my PCs and Laptops. My mobile devices are too important to not think about feature set and stability |
19:05.48 | fr0ggie | saxa: I may be wrong here, but i had that problem when i was using tcp mode sip (trying to avoid a braindamaged NAT firewall), i switched to udp, enabled nat=comedia and updated firmware on router == fixed. Probably simpler fix tho that these guys will know :) |
19:05.54 | drmessano | If Ubuntu pitches a fit and I need to hammer out a fix, I can tolerate that. I don't need 100% uptime for my desktop or laptop |
19:05.59 | fr0ggie | drmessano: I like to play with my hobby OS when i get bored |
19:06.36 | fr0ggie | even a pure linux on phone, i would find that fantastic |
19:06.50 | drmessano | I want an Ubuntu phone one of these days |
19:07.07 | fr0ggie | <PROTECTED> |
19:07.17 | drmessano | YES I LOVE UNITY DESKTOP NOW STOP YELLING AT ME AND RINGING MY DOORBELL AND RUNNING OFF YOU DAMN KIDS |
19:07.53 | fr0ggie | i dont think i'd be very happy with ubuntu on a phone, mostly because phones tend to be ram starved and more CPU usage == more heat and less battery life. Battery life is why i want to piss from a helicopter onto google hq >:) |
19:08.54 | fr0ggie | if i boot my phone into my nice happy little linux install, i can use finch and irssi all day long over 3g, minus the whole having to carry a keyboard with me |
19:09.07 | fr0ggie | If i run android, after 6-8 hours, i have to swap batteries |
19:09.35 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
19:10.01 | drmessano | They've done a lot of work with power management on Ubuntu in preparation for mobile devices, specifically phones. |
19:10.08 | drmessano | I would love to see some benchmarks |
19:10.27 | puzzled | evening all |
19:10.38 | *** join/#asterisk devCloud (~devand@209.5.190.9) |
19:11.26 | drmessano | Shuttleworth can see the writing on the wall. I am confident they will be able to do with Ubuntu what M$ wont be able to do with Windows 8 |
19:12.12 | chuckf | if they don't piss off all the unpaid contributiors before that happens |
19:12.21 | drmessano | Want to talk about scary? BSOD OS on my mobile device |
19:12.31 | drmessano | chuckf +1 |
19:12.53 | fr0ggie | i have no problem with trying to be cluebie friendly, hell its great getting more people to come to the shiney, tasty penguin meat... but i dont so much care for ubuntu's lets make it shiney n blingy til it needs 1gb video ram, 16gb of ram, and a 16384 core CPU to even move the mouse :( |
19:13.31 | fr0ggie | taking over 15 seconds to boot to a login manager? Unacceptable |
19:14.19 | drmessano | Hardly.. I have Ubuntu running on some pretty crappy machines and they hum right along |
19:14.38 | saxa | navaismo: ok, thx, but i do not remember that i changed anything |
19:14.44 | fr0ggie | my netbook, i hit power button and within 15 seconds, im sitting at the login screen, 3 seconds after i login, i've got a desktop waiting for me. |
19:15.07 | fr0ggie | Its more acceptable for 30-45 seconds on a desktop/laptop, but a phone? I should turn it on and within 5 seconds be able to use it |
19:15.17 | saxa | fr0ggie: thx |
19:15.32 | navaismo | saxa, is that a remote extension? |
19:15.39 | WIMPy | Same is true for desktop VOIP phones |
19:15.43 | saxa | yes the one behind 2 nats |
19:15.48 | drmessano | Um.. |
19:16.07 | saxa | i mean one nat at home and the other side has an * server behind a nat |
19:16.33 | drmessano | While unity may be the basis for the UI, the boot and login processes will be very different on a production Ubuntu device. I dont expect you'll have to boot and login to make a call :) |
19:17.23 | fr0ggie | drmessano: correct-- it should boot immediately on receiving power to a fully booted state, with your lock screen sitting active (background init basically) |
19:17.50 | navaismo | saxa, did you have configured the nat settings on your asterisk box? externhost or externip, localnet, and nat=yes in the peer |
19:17.52 | drmessano | It takes as long for Ubuntu to boot on my laptop as it does to cold boot my ipad 2. 15 seconds or so |
19:18.35 | saxa | navaismo: yes, all is like it was |
19:18.55 | fr0ggie | how fast laptop? my netbook is 1.6ghz single core, 2gb ram, 16gb ssd |
19:19.02 | saxa | navaismo: I just need to recheck my * side modem firewall |
19:19.24 | saxa | since on the home side i know I have not touched anything at all |
19:19.42 | saxa | thx for now |
19:19.54 | saxa | tomorrow in my office i check that thing |
19:20.33 | drmessano | Dual core 2.53GHZ, 3GB RAM, 200GB non-SSD |
19:20.37 | drmessano | I think |
19:23.13 | fr0ggie | dual core helps IMMENSELY with startup times |
19:23.49 | drmessano | Dual 2.16 actually |
19:24.28 | drmessano | With an encrypted home and swap it takes a little closer to 45 seconds |
19:24.40 | drmessano | But I am paranoid |
19:25.17 | fr0ggie | understandable |
19:25.22 | fr0ggie | people steal shit too often |
19:25.32 | fr0ggie | also, feds like to steal shit |
19:25.44 | jmetro | 2.5gh 6 core, 8gb ram, 2x500gb sata |
19:25.52 | fr0ggie | laptop? |
19:25.57 | jmetro | pc =D |
19:26.03 | fr0ggie | what is pc/ |
19:26.21 | jmetro | Tower. Big tower. |
19:26.30 | fr0ggie | Ohh, space heater |
19:26.42 | fr0ggie | i like my lap warmer |
19:26.58 | fr0ggie | i hear it temporarily reduces fertility, i hope so. |
19:27.01 | jmetro | only a space heater cause its AMD 6 core XD |
19:27.05 | drmessano | I run an Atom box at home. All that heat is a waste |
19:27.52 | drmessano | I stopped worrying about being the fastest kid on the block when I became an adult lol |
19:33.57 | fr0ggie | if it can make world overnight, im happy |
19:36.29 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
19:36.35 | jmls1 | damn, 11 beta2 is pretty stable |
19:36.52 | Qwell | jmls1: You're still on the beta? |
19:36.58 | jmls1 | ugh. |
19:37.00 | jmls1 | rc |
19:37.03 | Qwell | ahh |
19:37.08 | jmls1 | latest svn, actually :) |
19:37.23 | jmls1 | latest 11 svn branch to be pedantic |
19:37.52 | drmessano | You're supposed to call it TRUNK |
19:37.55 | drmessano | As in |
19:37.58 | drmessano | I R ON 11 TRUNK |
19:38.07 | drmessano | SVN == TRUNK |
19:38.20 | jmls1 | yikes. Shouting ! |
19:38.24 | drmessano | lol |
19:38.40 | drmessano | Now, repeat after me... |
19:38.50 | jmls1 | Asterisk SVN-branch-11-r374932 |
19:39.08 | jmls1 | doesn't say TRUNK in there :-D |
19:39.10 | drmessano | I R USE SIP TRUNK ON ASTERISK 11 TRUNK |
19:39.24 | drmessano | You may need to practice it a bit |
19:39.56 | jmls1 | System uptime: 5 days, 12 hours, 9 minutes, 2 seconds (up since we installed and fired it up) |
19:40.06 | Qwell | days behind |
19:40.17 | jmls1 | 57663 calls processed |
19:40.20 | drmessano | and when someone says "Thats not trunk, that Asterisk 11 branch from SVN".. You tell them "I R WGET IT FROM SVN TRUNK! U NO TELL ME!" |
19:40.55 | drmessano | I should give a talk at Astricon next year about this. Geesh |
19:41.12 | jmls1 | jeez. All I wanted to say was "thanks devs for making this so stable and helping me run my business!" :P |
19:41.24 | drmessano | lol |
19:42.05 | drmessano | I'm not a dev. I am here for the clown show and the chicken nuggets. |
19:42.25 | drmessano | OOOPS.. I mean, NOT CHICKEN NUGGETS. BACK OFF |
19:42.26 | jmls1 | yeah. I got slapped in -dev for mentioning the figures last time :'( |
19:42.53 | jmls1 | nm. I'm pleased to be using * |
19:42.58 | Nugget | eyes drmessano |
19:42.58 | drmessano | Some of those guys in -dev are pretty psycho. I appreciate their work, but glad they are kept in cages and fed with a long spork |
19:43.40 | drmessano | telnet? |
19:43.41 | Nugget | telnet is eeeeeeevil! |
19:44.25 | drmessano | Y R TALNET EVAL? I R TALNET TO PREEPBX FOR SSH WITHOUT WINDOWS |
19:44.36 | drmessano | ^_^ |
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20:03.08 | chuckf | SVN? Why are they not using GIT like everyone else? |
20:03.52 | saxa | because there is not much difference between one and the other :) |
20:04.32 | dijib | fr0ggie: its a lie! i believe im now super fetile. |
20:04.42 | dijib | thanks R200 |
20:08.14 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
20:36.49 | *** join/#asterisk infobot (~infobot@rikers.org) |
20:36.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.9.0 (2012/10/08), 1.8.17.0 (2012/10/08), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Coming to AstriCon? /join #astricon |