| 00:00.07 | angryuser | MikeJ ivr in mysql ? torturing yourself ? |
| 00:00.10 | bkw_ | _khan: map an extension to it |
| 00:00.20 | _khan | angryuser: i know the about dialplans, actually i want to call from each analogue line to check their oustanding bills!!! :( |
| 00:00.35 | joobie | fair enuf |
| 00:00.40 | MatBoy | angryuser: you mean me ? |
| 00:00.47 | joobie | but if i get asterisk working this weekend |
| 00:00.51 | joobie | without freepbx |
| 00:00.58 | joobie | then frepbx is a cop out. |
| 00:01.04 | joobie | the challenge is set.. |
| 00:01.08 | angryuser | oh yes MatBoy |
| 00:01.26 | joobie | joobie to get * working in one weekend without freepbx = freepbx is a copy out |
| 00:01.27 | bkw_ | who did IVR in mysql? |
| 00:01.39 | bkw_ | joobie: not really. |
| 00:01.51 | bkw_ | joobie: most people just want a phone system and could care less how it works. |
| 00:02.05 | MatBoy | angryuser: yes it seems so, no what I want to do is make a IVR that ask a person to enter some digits, that must query a DB if it exists, if it exists I need to transfer the call |
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| 00:02.38 | Qwell | MatBoy: func_odbc :D |
| 00:03.14 | angryuser | _khan so set up a dial as i explained upper |
| 00:03.18 | Qwell | it was written for exactly that type of thing |
| 00:03.31 | MatBoy | Qwell: yes, I have read that, but cmd MySQL seem to be easier... even than not much info about howto do this. |
| 00:03.48 | _khan | angryuser: ok, thanks. m trying will get back to u. |
| 00:05.06 | angryuser | _khan whateversuitsyou,1,Dial(Zap/25/${EXTEN}) |
| 00:06.21 | MatBoy | Qwell: but still than, I can't find good examples about it |
| 00:08.42 | _khan | angryuser: now i understand, thanks. |
| 00:09.14 | angryuser | MatBoy try to write some basic with NoOp everywhere, it's what i do usually ;) |
| 00:09.43 | joobie | hmm guys.. zaptel make menuconfig is complaining that i dont have ncurses installed - but i do (ncurses-5.5-24.20060715) |
| 00:10.03 | angryuser | joobie and dev package also ? |
| 00:10.10 | joobie | nope not the devel |
| 00:10.10 | MatBoy | angryuser: yes I have read about that, that works OK ?> |
| 00:10.21 | joobie | ill install it now.. thanks angry |
| 00:11.55 | angryuser | noop here noop there, sql is standart, eventually you will find what missing.... |
| 00:12.12 | joobie | cheers angry, that worked |
| 00:13.33 | joobie | brb rebooting |
| 00:14.19 | MatBoy | angryuser: mhh, let me look into that |
| 00:14.35 | MatBoy | it must be doable, but I have the idea when I search that I want to do stuff no-one uses :S |
| 00:17.09 | angryuser | MatBoy if there in no examples at all, cli 'core show function/application' should help you and it is big for mysql and odbc |
| 00:17.30 | angryuser | there is* |
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| 00:18.19 | _khan | angryuser: cheers, it works... |
| 00:18.46 | joobie | hmm is this fixed - http://lists.digium.com/pipermail/asterisk-bugs/2008-June/019377.html ? |
| 00:18.53 | joobie | i have the same error with zaptel make install |
| 00:19.08 | MatBoy | angryuser: ok, that is nice to know :) |
| 00:19.11 | MatBoy | thanks |
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| 00:20.10 | angryuser | joobie 'make config' in zaptel directory , uncheck xpp or download from svn |
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| 00:20.40 | angryuser | ~centos |
| 00:20.41 | jbot | centos is probably an Enterprise-class Linux Distribution derived from sources freely provided to the public by a prominent North American Enterprise Linux vendor. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 00:20.52 | joobie | ta angry |
| 00:20.56 | joobie | was just looking into the src |
| 00:21.08 | joobie | if i comment out the typedef it's ok |
| 00:21.21 | joobie | but then it's using the original.. |
| 00:21.26 | joobie | might do ur above suggestion - cheers |
| 00:22.28 | angryuser | going to sleep |
| 00:23.22 | joobie | night man |
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| 00:31.35 | MatBoy | lol |
| 00:32.03 | MatBoy | I was testing a faxline, couldn't figure out why the fax was rining and not coming in... it was transfered :P |
| 00:32.13 | MatBoy | I know for sure that someone is awake now :D |
| 00:32.21 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 00:32.59 | _khan | Incoming call receive after 2 rings on zap channels (analogue line)..... How to directly receive the call without any ring |
| 00:37.13 | _khan | MatBoy: feels that, every has slept now :) |
| 00:38.26 | MatBoy | _khan: hehe |
| 00:38.57 | MatBoy | is there a reason why a asterisk box doesn't pick up calls directly when you rebooted it ? sip show peers shows everything OK |
| 00:39.12 | MatBoy | busy tone |
| 00:39.48 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 00:40.18 | _khan | MatBoy: till channels not configured completely....... |
| 00:41.51 | MatBoy | _khan: no I had CID configured :P |
| 00:42.19 | MatBoy | hehe, I need to empty that fax because I let it ring for a couple of minutes sometimes :P |
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| 01:13.15 | joobie | guys does * work as a sip client behind nat, to a sip proxy on the net? |
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| 01:13.20 | joobie | im reading that there's a few problems |
| 01:15.03 | MatBoy | mhh, faxing seems to be slow |
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| 01:16.04 | MatBoy | ow, it was a photo :P |
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| 01:18.23 | MatBoy | yeah ! this is nice |
| 01:19.31 | joobie | hmmmmmmmm |
| 01:19.38 | joobie | i have setup sip.conf to connect to my sip provider |
| 01:19.47 | joobie | how can i test if the connectino is working? |
| 01:20.05 | joobie | i restarted asterix and cat'ed the logs in /var/log/asterisk.. nothing coming up in there really that says it's connected |
| 01:20.12 | jjshoe_ | sip show registry ? |
| 01:21.09 | joobie | where do i execute that command jj? |
| 01:21.12 | joobie | sip binary doesnt exist on cli |
| 01:21.24 | jjshoe_ | it's an asterisk command. |
| 01:21.32 | joobie | how do i execute asterisk commands? |
| 01:21.46 | jjshoe_ | joobie you should read up on asterisk a bit. |
| 01:22.04 | joobie | kk |
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| 01:22.56 | joobie | mad :) |
| 01:23.06 | joobie | the cli is like cisco with ? query commands |
| 01:23.07 | joobie | very slick |
| 01:23.39 | joobie | jj, what about connecting two phones to the asterisk box? |
| 01:23.46 | joobie | would this be done via sip in the lan? |
| 01:23.50 | joobie | i have polycom 320's |
| 01:23.53 | joobie | and poe |
| 01:24.50 | warewolf | any asterisk software/hardware hackers around? I've got an old creative/innomedia "voipblaster" I'd like to use w/ asterisk. |
| 01:25.22 | warewolf | I'm aware of the g.723 codec "issues" but noticed that there was a 723 plugin for asterisk ... anyone ever manage to get a voipblaster working with asterisk? |
| 01:26.09 | jjshoe_ | joobie there's a free asterisk book somewhere to download, I woould recommend getting it |
| 01:26.10 | jjshoe_ | ~book |
| 01:26.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:27.49 | joobie | ya im just not sure |
| 01:27.59 | joobie | do they use sip in a lan environemnt to conncet to asterisk? |
| 01:28.01 | joobie | is that the best way?? |
| 01:28.27 | joobie | i just have poe rj45 to the polycom phones.. and my asterisk is a sip client to a 3rd party provider on the net |
| 01:28.47 | joobie | so i want to connect the polycom's via poe to my network talking to asterisk.. |
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| 01:47.09 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 01:54.56 | _khan | how to ignore the incoming call on Zap (analougue) ??? |
| 01:55.40 | [TK]D-Fender | _khan: don't answer on it. |
| 01:55.53 | [TK]D-Fender | _khan: do "exten => s,1,Hangup |
| 01:55.55 | _khan | how?? |
| 01:56.16 | _khan | but id does'nt hangup the call, call is still in ringing |
| 01:56.36 | [TK]D-Fender | yes... hence * is not answering it. Exactly like you requested |
| 01:56.44 | _khan | and hanup again & again is executing |
| 01:57.13 | _khan | then, what's the solution?? |
| 01:59.58 | _khan | http://pastebin.com/pastebin.php?dl=d2008b993 |
| 02:01.28 | _khan | [TK]D-Fender: where m i on mistake?? |
| 02:06.35 | [TK]D-Fender | _khan: bad link |
| 02:08.26 | [TK]D-Fender | _khan: == Starting Zap/32-1 at welcomeivr,s,1 failed so falling back to exten 's' == Starting Zap/32-1 at welcomeivr,s,1 still failed so falling back to context 'default' |
| 02:08.42 | [TK]D-Fender | _khan: You apparently haven't learned to look at where you are even sending your calls to. |
| 02:09.08 | [TK]D-Fender | _khan: and why is your pastebin output split in half? |
| 02:17.35 | *** join/#asterisk chandi (n=chandi@76-10-149-224.dsl.teksavvy.com) |
| 02:20.48 | chandi | hi guys, I've got a little question. I'm now using IAX with my provider but I'd like to do something similar to canreinvite and I know it's possible from what I've read (a post from Mark Spencer) |
| 02:21.21 | chandi | is it done automatically or a config has to be changed like with SIP ? |
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| 02:34.09 | [TK]D-Fender | chandi: http://www.voip-info.org/wiki-Asterisk+config+iax.conf |
| 02:37.41 | chandi | thanks a lot :) |
| 02:38.46 | *** join/#asterisk Fiapo-CE (i=Fiapo-CE@201.70.137.40) |
| 02:39.36 | WilliamK | wow, suddenly quiet... |
| 02:40.02 | WilliamK | hello TK |
| 02:40.18 | Fiapo-CE | how do i set id caller on zaptel, it is possible? I'd to remove the area code from my number. |
| 02:40.32 | joobie | guys anyone got current sip software for the polycom 320's ? |
| 02:40.40 | joobie | or know where to download it.. |
| 02:40.53 | [TK]D-Fender | Fiapo-CE: on PRi yes, analog no. |
| 02:41.02 | [TK]D-Fender | joobie: You get it from our reseller |
| 02:41.05 | [TK]D-Fender | your* |
| 02:41.08 | WilliamK | joobie, who did you buy the polycom through? usually the VAR provides an FTP site |
| 02:41.14 | Fiapo-CE | [TK]D-Fender, it is pri. |
| 02:41.44 | [TK]D-Fender | Fiapo-CE: so go set the CID and test it |
| 02:42.18 | Fiapo-CE | [TK]D-Fender: where? zapata.conf? |
| 02:42.48 | [TK]D-Fender | Fiapo-CE: no, in your DIALPLAN, before you DIAL |
| 02:43.20 | chandi | Another little question. I have an app making call files and then it transfers to an extension which call sendDTMF. It used to work fine with SIP but now it keeps on sending the first digit |
| 02:43.28 | chandi | and it never stops |
| 02:43.42 | chandi | it starts the sound and it never ends :( |
| 02:43.43 | joobie | i bought the phones through voip-warehouse.com.au |
| 02:43.44 | chandi | any clue ? |
| 02:43.56 | joobie | but their support really sucks - don't think ill be able to get the software through them.. |
| 02:44.06 | joobie | ive rung them about 20 times in the last year and never got through to anyone |
| 02:45.04 | [TK]D-Fender | chandi: You'd have to pastebin the cli output of a failed call along with all related dialplan, call files, etc |
| 02:46.22 | joobie | are thre any other ways to get the software? anyone got it? |
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| 02:48.25 | *** join/#asterisk Fiapo-CE (i=Fiapo-CE@201.70.137.40) |
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| 02:48.46 | Fiapo-CE | [TK]D-Fender: sorry, could you repeat? |
| 02:49.02 | joobie | that's odd - http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html is that the software ? |
| 02:49.09 | joobie | sip software 2.2.2 |
| 02:49.14 | joobie | it's a 42MB download |
| 02:50.04 | chandi | d-fender : http://pastebin.com/m35f4e6ed thanks :) |
| 02:50.32 | kash | [TK]D-Fender: don't you ever sleep |
| 02:50.39 | [TK]D-Fender | Fiapo-CE: no, in your DIALPLAN, before you DIAL |
| 02:50.50 | chandi | wait, forgot cli output |
| 02:51.42 | chandi | new pastebin :http://pastebin.com/m5e8df2ca |
| 02:51.46 | WilliamK | joobie, my suggestion would be to purchase a phone through a more cooperative VAR or call Polycom and complain that the var isn't being supportive in providing you updated firmware (IF they are an authorized VAR) |
| 02:52.01 | WilliamK | that in itself may be the issue if they're not auth'd by polycom |
| 02:52.27 | chandi | d-fender : it does the same thing even when I call my DID from the PSTN . Not only with a call file |
| 02:55.25 | Fiapo-CE | [TK]D-Fender: my dial is exten => s,2,Dial(Zap/g0/${ARG1},45,Ttr) where can i change to remove the area code? |
| 02:55.43 | *** part/#asterisk korihor (n=korihor@190.39.163.45) |
| 02:57.56 | [TK]D-Fender | Fiapo-CE: I told you to set it BEFORE you dial. |
| 02:58.02 | [TK]D-Fender | Fiapo-CE: "core show function CALLERID" |
| 02:59.11 | [TK]D-Fender | chandi: So you get a continuous "1" tone? |
| 03:00.45 | chandi | [tk]d-fender well.. My ears are not good enough to make the difference between the different numbers but I do get a continuous tone every time but the last time I've tried which sounded like a short one followed by a continuous one |
| 03:00.59 | chandi | but that could be a signal cut though |
| 03:01.26 | [TK]D-Fender | chandi: Ok, it looks like its doing fine, so I'm not sure what to tell you. |
| 03:01.37 | [TK]D-Fender | chandi: I'd be worrying about that as well. |
| 03:01.59 | joobie | WilliamK, what about that download though? |
| 03:02.09 | joobie | is that not the updated firmware? |
| 03:02.43 | WilliamK | there is newer firmware |
| 03:02.48 | chandi | tk-d-fender I'll try to upgrade my * to a a newer version. The strange thing is that it was working perfectly under SIP. Are there different ways to send DTMF through IAX as with SIP (in-band, info, etc) ? |
| 03:03.09 | WilliamK | 2.2 works well don't get me wrong, but there is a newer release out |
| 03:03.15 | joobie | ahh |
| 03:03.17 | joobie | which is the newer? |
| 03:03.24 | [TK]D-Fender | chandi: What are you on now? |
| 03:03.34 | joobie | or WilliamK do you have the newer one? |
| 03:03.36 | chandi | tk-d-fender : 1.4.0 |
| 03:03.38 | joobie | then i can d/l from u. |
| 03:03.41 | [TK]D-Fender | 3.X is apprently unstable ATM and trouble. |
| 03:03.52 | WilliamK | 3.0.3_revB |
| 03:03.53 | [TK]D-Fender | chandi: 1.4.0? ANCIENT |
| 03:03.57 | kash | i use 3.0.3 |
| 03:04.05 | chandi | tk-d- ok ok.. I'll upgrade then |
| 03:04.09 | WilliamK | trouble how so? |
| 03:04.21 | WilliamK | would rather know sooner than later |
| 03:04.33 | [TK]D-Fender | WilliamK: have heard of crashing, config parms moved around, etc. Got an opinion to share? |
| 03:05.36 | joobie | interesting |
| 03:05.45 | joobie | given 3.X is buggy, is the 2.2 the latest in the 2.X branch? |
| 03:05.54 | [TK]D-Fender | joobie: 2.2.2 |
| 03:05.58 | joobie | great :) |
| 03:06.01 | joobie | now i dont feel so bad :P |
| 03:06.05 | joobie | heh |
| 03:06.09 | joobie | thanks |
| 03:06.38 | WilliamK | haven't rolled 3.0.3b yet... looking at the changelog now |
| 03:06.50 | WilliamK | I see where they moved some values though and added some |
| 03:07.44 | WilliamK | boatload of changes |
| 03:08.40 | [TK]D-Fender | WilliamK: Yes, very promising, but not great reports of stability yet |
| 03:09.25 | WilliamK | more changes than I care to digest tonight... my mind is stuck on wireless tonight |
| 03:09.35 | joobie | sup with your wireless will? |
| 03:09.48 | WilliamK | trying to find the best AP controller and WLAN probes |
| 03:10.04 | chandi | bbl after compilation |
| 03:10.07 | WilliamK | was thinking about Demarc Tech's APs (reasonable), yet I want a few more features |
| 03:10.45 | joobie | is this for indoor use? |
| 03:10.52 | WilliamK | I really want MIMO and AD integration abilities along with capabilities for doing digital certs and association of VLAN based on userID |
| 03:10.55 | WilliamK | yeah |
| 03:11.19 | WilliamK | really don't want to have to do a radius between an AD server as a translator |
| 03:11.24 | WilliamK | kinda bites having to do that |
| 03:11.44 | joobie | i use the cisco 800 series in the office |
| 03:11.46 | joobie | very stable |
| 03:12.39 | WilliamK | meshed or just single ? |
| 03:14.50 | joobie | single |
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| 03:22.09 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 03:24.26 | cesar_CR | hello guys :) |
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| 03:29.23 | Fiapo-CE | help me, my full number is XXX-YYYY-ZZZZ, when I call to somebody, my number appears like XXX-YYYY-ZZ. would be YYYY-ZZZZ. how do i fix it? please. |
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| 03:35.23 | [TK]D-Fender | Fiapo-CE: DET YOUR CALLERID BEFORE YOU DIAL |
| 03:36.54 | joobie | guys im getting chan_sip.c: Registration from '<sip:1234@192.168.72.245>' failed for '192.168.72.50' - No matching peer found in the log |
| 03:37.01 | chandi | d-fender : it was indeed my Ancient version of *. Must have been made before 2nd world war ;) |
| 03:37.11 | joobie | i just updated the bootrom and sip software.. configed it all up.. the phone is plugged in and that comes in the logs |
| 03:38.08 | joobie | i defined in sip.conf '[extension]; type=user; secret=aaaa; context=default; mailbox=1234' .. is that not enough? |
| 03:38.44 | [TK]D-Fender | joobie: you didn't set a host |
| 03:39.15 | joobie | does the host have to be static fender? |
| 03:39.27 | Kyoshi | could be dynamic |
| 03:39.32 | Kyoshi | but state 'dynamic' |
| 03:39.34 | Kyoshi | as the host |
| 03:39.45 | joobie | like i have defined a password in the sip software for the phone.. if i use that to specify different passwords for different phones, is that enough? |
| 03:39.47 | joobie | ahh |
| 03:39.56 | [TK]D-Fender | host=dynamic <- |
| 03:40.10 | joobie | if i go down that path, will the phones dynamically assign to a extenion in the sip.conf based on the passwd? |
| 03:40.21 | Kyoshi | nopes |
| 03:40.25 | Kyoshi | not based on the password |
| 03:40.40 | Kyoshi | you have to assign extensions to the phones based on the [name here] in the sip.conf |
| 03:40.46 | Kyoshi | such as |
| 03:40.50 | Kyoshi | [215] |
| 03:40.53 | Kyoshi | or |
| 03:40.56 | joobie | ahh |
| 03:40.58 | Kyoshi | [298] |
| 03:41.00 | [TK]D-Fender | joobie: that tells * that that DEVICE (FFS stop calling it an 'extension') that it is allowed to REGISTER to it. |
| 03:41.12 | Kyoshi | hehehe |
| 03:41.13 | Kyoshi | FFS |
| 03:41.15 | joobie | :) |
| 03:41.17 | joobie | sry ehhe |
| 03:41.19 | joobie | thanks tho |
| 03:41.25 | joobie | damn exciting - my first * build |
| 03:41.26 | Kyoshi | i love that |
| 03:41.28 | Kyoshi | ffs |
| 03:41.39 | joobie | almost there i think.. i can see i have setup the sip client to my sip provider ok |
| 03:41.42 | joobie | just the phones :P |
| 03:41.46 | joobie | k gona try that.. thanks! |
| 03:41.50 | joobie | btw 4 number extensions are ok ya? |
| 03:41.52 | joobie | like 1234? |
| 03:41.54 | Kyoshi | yea |
| 03:41.57 | joobie | cool |
| 03:41.59 | joobie | brb testing:P |
| 03:44.19 | joobie | NOTICE[4657] chan_sip.c: Registration from '<sip:1234@192.168.72.245>' failed for '192.168.72.50' - No matching peer found |
| 03:44.22 | joobie | still getting that |
| 03:44.30 | Kyoshi | pastebin.ca |
| 03:44.33 | joobie | sec ill paste |
| 03:44.33 | joobie | ya |
| 03:44.38 | joobie | k one sec |
| 03:44.39 | Kyoshi | show the sip.conf |
| 03:47.08 | joobie | http://pastebin.ca/1076287 |
| 03:47.16 | joobie | i pasted some of the configs that are going to the handset too |
| 03:47.44 | joobie | you can probably tell by that config - but im setting up asterisk to be a sip client to my sip provider.. then the phones connect to asterisk on my LAN |
| 03:47.47 | [TK]D-Fender | joobie: that should have been "type=peer" <- |
| 03:48.24 | Kyoshi | macaddr.cfg? |
| 03:48.27 | joobie | for 1234? |
| 03:48.44 | joobie | Kyoshi, macaddr is substituted for the handset's mac addr |
| 03:48.53 | joobie | polycom sucks that in when it loads up |
| 03:48.57 | Kyoshi | ahh |
| 03:49.01 | joobie | seperate cfg per handset |
| 03:49.01 | Kyoshi | yea |
| 03:49.04 | Kyoshi | i know |
| 03:49.05 | Kyoshi | k |
| 03:49.06 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 03:49.08 | joobie | cool |
| 03:49.09 | Kyoshi | so lets see |
| 03:49.32 | joobie | sry man.. im new to all this, duno if im stating the obvious.. the whole thing foreign to me:P |
| 03:50.25 | Kyoshi | so you're trying to make the phone config marry to an ext? |
| 03:50.54 | Kyoshi | lemme go read on this |
| 03:51.15 | joobie | yea.. marry to an internal ext |
| 03:51.27 | joobie | so when i plug in the polycom it takes ext 1234 |
| 03:51.49 | joobie | that's the first goal.. second goal is to make it so when i dial the ext voip number it goes through to ext 1234 |
| 03:51.59 | joobie | and also 1234 to be able to dial other extensions and also external |
| 03:52.04 | joobie | small steps tho;P |
| 03:52.25 | joobie | i did 'sip show registry' and it looks like the asterisk has established a connection to the sip provider |
| 03:58.18 | joobie | type=peer works for the phone |
| 03:58.30 | joobie | but doesnt allow me to dial external numbers through the sip provider |
| 03:58.36 | joobie | do i need to setup a dialplan for that or sumthen? |
| 03:59.48 | Fiapo-CE | [TK]D-Fender not working :( |
| 04:03.24 | [TK]D-Fender | Fiapo-CE: and you're not SHOWING |
| 04:04.30 | *** join/#asterisk freezey (n=freezey@c-68-83-177-254.hsd1.nj.comcast.net) |
| 04:14.38 | jblack | [TK]D-Fender: You're such a voyer |
| 04:18.10 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 04:22.35 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 04:30.16 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 04:32.43 | joobie | freak |
| 04:32.45 | joobie | this is killin me |
| 04:32.55 | joobie | anyone able to lend a hand? getting another error now.. |
| 04:33.14 | joobie | will paste bin my conf |
| 04:33.18 | kamanashisroy | hi, is there any manager command to send text to a channel ? |
| 04:35.25 | joobie | http://www.pastebin.ca/1076322 that is my config for sip.conf and extensions.conf .. when i go to dial a local number on my handset i see this coming up .. '[Jul 20 00:25:41] NOTICE[5163] chan_sip.c: Call from '1234' to extension '95310341' rejected because extension not found.' |
| 04:35.54 | joobie | i setup an exten in extensions.conf with pattern matching to match that 9 digit number and pipe it out to comvergence, but doesnt work |
| 04:37.08 | kamanashisroy | joobie: where is your default context ? |
| 04:38.02 | freezey | anybody see that IAX workaround for countries that are blocking the IAX and SIP ports? |
| 04:38.07 | kamanashisroy | joobie: 1234 is bind in the default context !! |
| 04:39.01 | joobie | kamanashisroy, i removed the default context |
| 04:39.15 | joobie | i basically wiped the sameple extensions.conf and did my own. because there's so much crap in there that i thought it was conflicting |
| 04:39.45 | _khan | freezey: use ipfilter to change or bypass the ports for IAX |
| 04:39.49 | kamanashisroy | joobie: no problem .. but the asterisk is searching the extension in the default context .. |
| 04:40.06 | joobie | ahh yea i forgot that line in sip.conf |
| 04:40.15 | freezey | _khan, was at the HOPE conference today and 2 guy designed this perl script that will assist in doing this |
| 04:40.34 | *** join/#asterisk l2cache (n=l2cache@m7a5e36d0.tmodns.net) |
| 04:41.16 | joobie | kamanashisroy, i updated that context to users.. is this all you can see that is wrong? |
| 04:41.39 | kamanashisroy | what did you set ? |
| 04:41.54 | _khan | freezey: when protocol is blocked then the only way to use tunneling & if the ports are blocked then change that ports accordingly |
| 04:41.56 | joobie | under [1234] i set context to 'users' |
| 04:42.01 | joobie | .. in sip.conf |
| 04:42.18 | joobie | restarting asterisk.. ill try to dial again |
| 04:43.16 | freezey | _khan, i understand i am just saying that a few guys at this conference came up with a cool perl script that handles alot of this stuff as well as injection dos etc |
| 04:43.31 | joobie | kamanashisroy, one step closer.. now getting this error 'chan_sip.c: Failed to authenticate on INVITE to '"1234 Rich"' |
| 04:44.09 | _khan | freezey: who one??? |
| 04:44.31 | freezey | _khan, who one? |
| 04:45.26 | kamanashisroy | joobie: is the 1234 registered there ? try use insecure=very .. |
| 04:45.26 | _khan | freezey: i also need those script :) |
| 04:46.53 | freezey | _khan, yeah i am waiting for this guy to post it he said he was gong to do it tonite but he did not... it was cool during the injection part of it he basically brings your asterisk process up to 90% of system usage because he is sending tons and tons of IAX connections to the box but instantly dropping them and each time asterisk creates a session ID for them |
| 04:46.55 | freezey | so its pretty neat |
| 04:47.17 | joobie | kamanashisroy, where did you get 1234 from? i thought that was just the extension number for my handset |
| 04:48.06 | kamanashisroy | joobie: I thought you dialed 1234 .. as it is specified in the log ..! |
| 04:48.11 | joobie | the TISP has provided me a username / password.. i specified that though both in the comvergence section and in the register under general section of sip.conf |
| 04:48.22 | kamanashisroy | joobie: s/dialed/sent call/ |
| 04:48.26 | joobie | my handset is 1234 |
| 04:48.34 | joobie | but i am dialling a 9 digit number |
| 04:48.41 | kamanashisroy | joobie: handset ? is it sip client ? |
| 04:48.54 | joobie | kamanashisroy, the handset is a sip client to asterisk |
| 04:48.59 | _khan | joobie: check your sip whether it is registered or not "sip show registry" |
| 04:49.03 | joobie | and asterisk is a sip client to the sip provider |
| 04:49.36 | kamanashisroy | joobie: as _khan said .. check your sip registry |
| 04:49.36 | joobie | it says sip.comvergence.com.au:5060 is state 'Registered' |
| 04:50.00 | kamanashisroy | joobie: is it for 1234 ? |
| 04:50.09 | joobie | no |
| 04:50.30 | joobie | but kamanashisroy i wanted to set this up so my asterisk box registers to the sip provider |
| 04:50.40 | _khan | joobie: what codec ur provider is using?? |
| 04:50.42 | joobie | so my asterisk box is sip client and sip provider is the sip server |
| 04:50.55 | kamanashisroy | joobie: your dial statement is wrong .. |
| 04:51.08 | joobie | then when my handsets in the lan want to go via the sip provider.. they connect to the asterisk box via sip.. and then route through that connection the asterisk box already has registered to the sip provider |
| 04:51.10 | kamanashisroy | joobie: you are doing it in IAX style .. ... please recheck |
| 04:51.28 | joobie | SIP Codecs: G729, G711A 20ms |
| 04:51.34 | joobie | that is from the sip provider |
| 04:51.35 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:52.06 | kamanashisroy | joobie: I think you wanted to write SIP/${EXTEN}@comvergence not SIP/comvergence/${EXTEN} .. |
| 04:52.26 | joobie | i see |
| 04:52.30 | joobie | let me try |
| 04:52.31 | joobie | sec |
| 04:52.43 | _khan | joobie: first register it on other client to check whether you can dial a number from the sip phone or not |
| 04:54.26 | joobie | kamanashisroy, same error |
| 04:55.18 | joobie | _khan, what do you mean? setup another handset and dial that handset via asterisk? |
| 04:55.26 | joobie | to avoid the sip provider? |
| 04:55.27 | kamanashisroy | joobie: are you trying from 1234 ? |
| 04:55.31 | joobie | yes |
| 04:56.02 | joobie | i pick up handset 1234, then i dial the 8 digit number and press dial |
| 04:56.04 | joobie | then i get that invite error |
| 04:57.53 | _khan | joobie: register this account on any softphone like xlite, eyebeam & dial number from softphone, it is working or not?? |
| 04:58.12 | joobie | ahh i see |
| 04:58.22 | joobie | i havent tried that yet _khan |
| 04:58.29 | joobie | is there a mac softphone? |
| 04:58.32 | joobie | decent one |
| 04:58.32 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 04:58.33 | joobie | ill try |
| 04:59.22 | _khan | use xlite |
| 04:59.25 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:59.27 | joobie | kk |
| 04:59.29 | joobie | downloading |
| 05:00.55 | kamanashisroy | joobie: use "insecure=very" in 1234 sip account .. |
| 05:01.24 | kamanashisroy | joobie: ^^ just for testing .. |
| 05:01.46 | joobie | kk |
| 05:01.47 | joobie | will try |
| 05:01.52 | joobie | downloading xlite meanwhile |
| 05:04.10 | joobie | same error kaman |
| 05:04.56 | joobie | gona try x-lite.. sec |
| 05:06.59 | joobie | hmm |
| 05:07.09 | joobie | x-lite requires a domain, the sip provider didnt provide this |
| 05:07.25 | jaytee | try insecure=port,invite instead if you're running 1.4 |
| 05:07.42 | _khan | joobie: domain is same as ip address |
| 05:07.51 | joobie | okie |
| 05:07.56 | joobie | can i use hostname there too khan? |
| 05:08.02 | _khan | yes |
| 05:08.08 | jaytee | FQDN |
| 05:08.44 | joobie | sweet |
| 05:08.45 | joobie | that works :) |
| 05:08.49 | joobie | x-lite that is |
| 05:09.05 | joobie | jaytee, where is that set? |
| 05:09.15 | joobie | under [1234] ? |
| 05:09.17 | jaytee | sip.conf for that phone |
| 05:09.20 | jaytee | or device |
| 05:09.25 | joobie | k |
| 05:09.37 | joobie | so you guys think it's a problem with [1234] rather than [comvergence] right? |
| 05:10.12 | _khan | what u have define under [1234] |
| 05:10.35 | _khan | joobie: is 1234 your ID?? |
| 05:10.46 | joobie | sec ill pastebin a fresh one khan |
| 05:10.46 | joobie | ya |
| 05:10.49 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 05:10.50 | joobie | 1234 is my handset id |
| 05:11.16 | _khan | joobie: sip.conf & extension.conf both |
| 05:11.58 | joobie | http://www.pastebin.ca/1076343 that is sip.conf |
| 05:12.39 | joobie | http://www.pastebin.ca/1076344 that is extensions.conf |
| 05:13.18 | joobie | .. The error I get is - [Jul 20 01:12:51] NOTICE[6043] chan_sip.c: Failed to authenticate on INVITE to '"1234 Rich" <sip:sipusername@192.168.72.245>;tag=as46df2190' |
| 05:14.19 | joobie | btw the handset i have set the sip provider as * .. asterisk has comvergence set as the sip provider |
| 05:15.05 | joobie | jaytee, i tried that option you suggested and got the same error |
| 05:15.25 | jaytee | did you do a sip reload after you made the change? |
| 05:17.02 | joobie | nod |
| 05:17.03 | joobie | well |
| 05:17.16 | joobie | i did /etc/init.d/asterisk restart |
| 05:17.27 | joobie | presuming this does a sip reload? |
| 05:17.37 | jaytee | what are you running on? Debian? |
| 05:17.43 | joobie | centos |
| 05:18.54 | joobie | jaytee, after i do the restart, if i do 'sip show registry', the reg.time has updated. |
| 05:19.06 | jaytee | you should be able to type service asterisk restart in centos, it's what I do if you're running the right startup script. And then you run asterisk -vvvvvr to remote connect to the CLI. |
| 05:19.28 | *** join/#asterisk _khan (n=shariq@124.29.194.207) |
| 05:19.52 | joobie | ahhh |
| 05:20.03 | joobie | yea service command is a shortcut to /etc/init.d scripts |
| 05:20.24 | joobie | just force of habit.. i think redhat introduced service command only in the last few years |
| 05:21.20 | jaytee | in Debian or Ubuntu it's still the old explicit command |
| 05:22.00 | jaytee | so the sip.conf you posted has all the real personal data masked I hope because those account names are too generic. |
| 05:22.11 | joobie | ahh |
| 05:22.16 | joobie | maybe redhat ripped it from debian :P |
| 05:22.37 | joobie | yea, sipusername and sippassword are masked |
| 05:22.42 | joobie | the rest are as-is |
| 05:22.49 | jaytee | no, I think RedHat came out with service. Ubuntu still doesn't have it. |
| 05:22.57 | joobie | ahh |
| 05:23.34 | jaytee | ok, because that right there would kill registration with an outside sip provider. did you follow the wiki for sipnat? |
| 05:24.02 | _khan | asd |
| 05:24.03 | joobie | my work paid for me to do the rhce.. that was back in RHEL3. i think they just introduced service in that version .. but my memory sux |
| 05:24.33 | joobie | i didnt jaytee |
| 05:25.14 | jaytee | if you're behind a nat'd firewall you might need to refer to it. |
| 05:25.19 | jaytee | ~sipnat |
| 05:25.19 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:26.26 | joobie | * is behind nat firewall |
| 05:26.29 | _khan | joobie: your sip should be like this [sipusername] user=sipusername secret=sippass host=ip or domain fromuser=sipuser allow=g711a |
| 05:27.08 | joobie | interesting khan |
| 05:27.39 | joobie | ill try reformat to that |
| 05:27.41 | joobie | sec |
| 05:27.55 | joobie | is sipuser the same as sipusername ? |
| 05:27.59 | _khan | yes |
| 05:28.21 | joobie | kk |
| 05:28.44 | joobie | [sipusername] im leaving as [comvergence] though - is that ok? |
| 05:28.46 | joobie | just so the dialplan is clearer |
| 05:30.16 | _khan | create a dialplan as _001X.,1,Dial(SIP/sipusername:sippass@host/${EXTEN:2},,tTo) |
| 05:31.55 | joobie | ok same error as before.. though havent put in the dialplan |
| 05:31.58 | joobie | ill wack that in now |
| 05:31.59 | _khan | and remove or comments all of other for time being |
| 05:32.18 | _khan | reload extensions and sip in * |
| 05:32.18 | joobie | kk |
| 05:32.38 | joobie | remove nat=yes, canreinvite=no also ? |
| 05:32.53 | _khan | leave as it is |
| 05:32.56 | joobie | kk |
| 05:33.22 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 05:35.59 | joobie | [Jul 20 01:35:49] WARNING[6388] frame.c: Cannot allow unknown format 'g711a' |
| 05:37.35 | joobie | _khan, http://www.pastebin.ca/1076360 that's the latest configs |
| 05:39.16 | joobie | ok that's interesting |
| 05:39.22 | joobie | totally different error with that config _khan |
| 05:39.29 | _khan | just comment disallow=all & allow=g711a |
| 05:39.31 | joobie | <PROTECTED> |
| 05:39.43 | joobie | kk |
| 05:40.33 | joobie | WARNING[6517] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. |
| 05:40.36 | joobie | not sure what that means.. |
| 05:41.39 | joobie | hmm |
| 05:41.44 | joobie | [Jul 20 01:41:31] WARNING[6591] chan_sip.c: No such host: sip.comvergence.com.au/195317272 |
| 05:41.50 | joobie | WARNING[6591] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 05:42.02 | joobie | it resolves fine on the * box |
| 05:42.14 | joobie | sip.comvergence.com.au that is |
| 05:42.57 | _khan | sip is registered on * ? |
| 05:43.24 | joobie | sip.comvergence.com.au:5060 sipusername 105 Registered Sun, 20 Jul 2008 01:41:55 |
| 05:43.32 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:43.55 | joobie | long shot - is it possible it's trying to resolve 'sip.comvergence.com.au/195317272' rather than 'sip.comvergence.com.au' ? |
| 05:44.20 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 05:44.37 | joobie | hmm |
| 05:44.39 | joobie | sec khan |
| 05:44.42 | joobie | im going to try the ip |
| 05:44.47 | joobie | get rid of dns all togehter |
| 05:45.23 | *** join/#asterisk mattchis (n=mattchis@c-98-195-121-100.hsd1.tx.comcast.net) |
| 05:46.36 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 05:50.20 | mandh | Hi all |
| 05:55.11 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 05:55.17 | tessier | Hello all |
| 05:55.32 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 05:56.55 | mandh | i enable "recordformat=gsm |
| 05:56.55 | mandh | " at agents.conf that record files fine ,but i wann that in recored file name , that date of call is record also |
| 05:57.12 | tessier | I have a phone in context "hoang". I have another phone in context "copilot". Each of these context has a few 3 digits extensions. Context hoang includes exten => _XXX,1,Dial(SIP/${EXTEN}) in the dialplan. If a phone in context hoang dials the 3 digit extension number of a phone in context copilot it successfully rings that phone. But the copilot phone is in a different context. So how are they able to call each other? |
| 05:57.47 | mandh | coz the files now record like :agent-1010-1216431671-111176.gsm |
| 06:05.22 | xiando | tessier: interesting! I'm gonna experiment with that on my boxen right now, the security implications are .. interesting. |
| 06:07.02 | xiando | unless it has include context somewhere (or something like that) then it's not supposed to be able to do that |
| 06:09.27 | tessier | xiando: Yeah, that's why I'm really confused |
| 06:12.16 | lanning | tessier, the "exten => _XXX,1,Dial(SIP/${EXTEN})" does that. you have sip channels defined as 3 digit number, so, it is a direct one to one match, no translation. |
| 06:12.42 | tessier | lanning: That's what I was afraid of. So how do fix that? Have to specify a context somehow? |
| 06:13.07 | lanning | you don't do wildcarding like that. |
| 06:13.23 | tessier | The real problem that turned me onto this is that I have two different teliax accounts |
| 06:13.23 | lanning | you have to spell out the extensions for each context. |
| 06:13.52 | tessier | Yes, I just figured that out and made that change and it works. |
| 06:14.58 | tessier | However I have a phone in hoang context with extensions like exten => _91NXXNXXXXXX,2,Dial(SIP/hoang-teliax/${EXTEN:1}) but when the phone in hoang context calls the call is going out through SIP/teliax instead and getting charged to the wrong teliax account. |
| 06:15.13 | tessier | I thought they would be related problems. Now I'm not so sure... |
| 06:15.58 | tessier | Actually, it says it is dialing hoang-teliax on the console: -- Called hoang-teliax/18583490123 |
| 06:16.17 | tessier | But it went out my SIP device which is just called "teliax". |
| 06:17.16 | lanning | make sure the two sip.conf channels ID themselves differently (username and secret) |
| 06:19.22 | tessier | Yes, they do. They use two different teliax accounts with different usernames and secrets. |
| 06:20.45 | *** join/#asterisk Pons (n=pons@unaffiliated/pons) |
| 06:21.01 | lanning | what do you mean "went out my SIP device", what device? |
| 06:22.08 | tessier | In sip.conf I have two sip accounts/devices. One called "teliax" the other "huang-teliax" |
| 06:22.14 | tessier | hoang |
| 06:22.24 | tessier | They each have a unique username/secret |
| 06:22.37 | lanning | ah, you are calling devices, I am calling channels |
| 06:22.45 | Pons | hello guys, anyone with chan_mobile knowledge? I'm on 1.4.20 (debian version), chan_mobile rev 454 with 1.4.x patch, and it keeps getting the damn "reason (104)" error and disconnecting. |
| 06:22.45 | tessier | Ok, I will call them channels. |
| 06:23.05 | Pons | What would be the best, 1.6 beta and 1.6 addons, or anything else? |
| 06:23.09 | tessier | In sip.conf I have channels defined for all of my phones and a pair of channels defined for each of my accounts with teliax |
| 06:23.28 | tessier | I have phone extensions 397, 398, 399 all register in context copilot |
| 06:24.08 | tessier | and context copilot includes context "outgoing" which has lines like exten => _91NXXNXXXXXX,2,Dial(SIP/teliax/${EXTEN:1}) |
| 06:24.38 | lanning | have you run a "sip debug" to see if you are sending the right info in the INVITE packet? What is telling you that you are using the "teliax" channel? Is it just the bill? If so, it might be an issue with the teliax service. |
| 06:25.20 | tessier | I also have extensions 400 and 401 which register in context hoang. And context hoang includes hoang-outgoing which has lines like exten => _91NXXNXXXXXX,2,Dial(SIP/hoang-teliax/${EXTEN:1}) |
| 06:25.26 | tessier | No, I haven't. Let me try that... |
| 06:26.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:29.27 | tessier | lanning: I did a sip debug peer hoang-teliax and when I made a call it went out through that channel. |
| 06:29.34 | tessier | And successfully rang my cell phone. |
| 06:29.37 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 06:30.40 | lanning | ok, if that call shows up in the "teliax" channel's CDR on the teliax service, then it is their problem. |
| 06:31.08 | tessier | SIP/hoang-teliax-082 (None) Up Bridged Call(SIP/401-081da418) |
| 06:31.08 | tessier | SIP/401-081da418 918583490123@hoang-o Up Dial(SIP/hoang-teliax/18583490 |
| 06:31.14 | tessier | That's from show channels. |
| 06:31.21 | tessier | It's definitely going out the hoang-teliax channel. |
| 06:31.35 | tessier | And showing up in the teliax channel's cdr. |
| 06:31.36 | tessier | Hrm |
| 06:31.54 | tessier | Is it possible that teliax has never had anyone try to register two accounts from the same IP before? |
| 06:31.55 | lanning | they might be grouping by IP address, and using which ever registration happens first/last... |
| 06:32.00 | tessier | Could be... |
| 06:40.14 | joobie | guys anyone know how to resolve the "chan_sip.c: Failed to authenticate on INVITE" error? |
| 06:44.17 | florz | is there any way I could make asterisk not recognize the pickup extension from features.conf as anything special, but rather process it through the dialplan as any other number? |
| 06:44.50 | florz | s/as any/like any/ |
| 06:45.27 | florz | not quite ... =:-) |
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| 07:30.17 | mandh | what "Calls Abandoned" means |
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| 09:12.48 | joobie | guys what dialplan allows phones to dial eachother in the same lan? |
| 09:13.44 | mvanbaak | eh ? |
| 09:15.00 | joobie | mvanbaak, i've defined two handsets in sip.conf |
| 09:15.13 | joobie | both have a sip connection to asterisk |
| 09:15.22 | joobie | one is ext 1233 and the other 1234 |
| 09:15.34 | joobie | what dialplan do i need so i can pick up 1233 and dial in 1234 and get the other phone ringing? |
| 09:16.16 | mvanbaak | exten => _123X,1,Dial(SIP/${EXTEN}) |
| 09:17.46 | joobie | thanks mvan |
| 09:22.12 | joobie | mvanbaak, how's your sip knowledge? i've had a problem all day im about to get stuck into.. not sure where to start with this one though |
| 09:22.54 | mvanbaak | just ask the question |
| 09:23.41 | joobie | my * box is hooked up via sip to my sip provider.. then i have these two phones hanging off the * box, connected up via sip to the * box. |
| 09:24.04 | joobie | when i try to dial an external number via the connection to the sip provider, i get a busy tone and an error like '[Jul 20 05:22:59] NOTICE[32712] chan_sip.c: Failed to authenticate on INVITE to '"1233" <sip:sipusername@192.168.72.245>;tag=as3a945c09'' |
| 09:24.08 | *** part/#asterisk arekm (i=arekm@pld-linux/arekm) |
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| 09:26.28 | joobie | http://www.pastebin.ca/1076465 that is my sip.conf and extensions.conf |
| 09:26.56 | joobie | also when i do 'sip show registry', it says the connection to the sip provider is 'Registered' |
| 09:27.12 | joobie | have also tried connecting directly from my laptop to the sip provider using x-lite, it works fine.. |
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| 09:48.17 | bobface | morning all - quick stupid question - how can i download festival on AsteriskNow |
| 09:48.36 | bobface | i can't find any gcc or dpkg or rpm tools |
| 09:50.38 | mvanbaak | asteriskNow has conary as package manager |
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| 09:51.42 | bobface | ok cool |
| 09:51.44 | bobface | thanks |
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| 09:53.22 | joobie | woops |
| 09:53.28 | joobie | mvanbaak, still around? |
| 09:53.37 | mvanbaak | yup |
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| 09:53.47 | mvanbaak | can you put that paste on pastebin.com ? |
| 09:53.56 | mvanbaak | I cannot reach pastebin.ca for some weird reason |
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| 10:00.03 | joobie | sure thing |
| 10:00.19 | joobie | ahh bugger |
| 10:00.24 | joobie | do u have the url mvanbaak ? |
| 10:00.28 | joobie | i lost it when my client closed |
| 10:01.09 | mvanbaak | http://www.pastebin.ca/1076465 |
| 10:03.37 | joobie | cheers |
| 10:03.59 | joobie | http://pastebin.com/m59faa600 |
| 10:04.02 | joobie | is that readable? |
| 10:04.09 | joobie | i can repost without the numbers if you want |
| 10:04.26 | joobie | the line numbers from pastebin.ca came through.. :P |
| 10:04.45 | mvanbaak | pretty unreadable yeah |
| 10:04.54 | mvanbaak | hhmm, I know why I cannot reach pastebin.ca |
| 10:05.03 | joobie | sec ill repaste |
| 10:05.22 | mvanbaak | they do have an ipv6 address in the DNS, but their webserver does not listen on ipv6 |
| 10:06.45 | joobie | mvanbaak, try this http://pastebin.com/m13c6fc78 |
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| 10:06.53 | joobie | ahh |
| 10:07.02 | joobie | isnt there ipv6 -> ipv4 proxies? |
| 10:07.10 | mvanbaak | yeah |
| 10:07.12 | mvanbaak | ah well |
| 10:07.17 | mvanbaak | ah, that paste is readable |
| 10:07.32 | joobie | cool :) |
| 10:09.01 | mvanbaak | so, what's not working |
| 10:09.14 | mvanbaak | calls from 1233 to 1234 are working ? |
| 10:10.02 | joobie | yup |
| 10:10.44 | joobie | just when i try to dial an external number,w hich should go via comvergence, it says '[Jul 20 06:10:17] NOTICE[32712] chan_sip.c: Failed to authenticate on INVITE to '"1234" <sip:sipuser@192.168.72.245>;tag=as1193723f'' |
| 10:10.53 | joobie | that happens if i try to dial out from either handset |
| 10:11.44 | mvanbaak | can you pastebin the CLI output after you have done 'core set verbose 3' |
| 10:12.00 | joobie | sure |
| 10:12.55 | joobie | well that's good to know :P |
| 10:12.57 | joobie | sec ill pastebin |
| 10:13.53 | joobie | http://pastebin.com/m1738fa65 |
| 10:15.37 | mvanbaak | hhmm, I really think we need the output of a sip debug |
| 10:15.54 | mvanbaak | do: sip set debug on |
| 10:15.58 | joobie | okie |
| 10:16.01 | mvanbaak | try to call outbound |
| 10:16.04 | mvanbaak | and pastebin that |
| 10:16.04 | joobie | should i increase my buffer for this?:P |
| 10:16.08 | joobie | sure |
| 10:16.13 | mvanbaak | erm, probably |
| 10:16.16 | joobie | heeh |
| 10:16.17 | joobie | k |
| 10:16.44 | joobie | it didnt like that syntax |
| 10:16.57 | joobie | i can specify an ip tho |
| 10:17.16 | joobie | should i put *'s ip or comvergence? |
| 10:17.31 | joobie | ahh my bad |
| 10:17.33 | joobie | i can do it by peer |
| 10:17.34 | joobie | sec |
| 10:19.38 | angryuser | ~enter |
| 10:19.38 | jbot | the enter key is not a substitute for punctuation. Use a period '.', exclamation mark '!', question mark '?', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 10:21.42 | joobie | mvanbaak, http://pastebin.com/m1b657216 |
| 10:22.11 | joobie | angryuser, i'm learning - just used a ',' ;) |
| 10:22.33 | joobie | mvanbaak, huge dump from that debug.. i set debug on the peer being '1234' (the handset dialing external) |
| 10:23.58 | mvanbaak | joobie: weird. It thinks comvergence is on your local network |
| 10:24.19 | joobie | hmm |
| 10:24.24 | joobie | im using polycom phones |
| 10:24.39 | joobie | the phones are configured to use 192.168.72.245 as the sip proxy |
| 10:24.58 | joobie | duno if they are related |
| 10:25.08 | joobie | but calls to comvergence sip would still go to the lan first |
| 10:25.38 | joobie | btw mvanbaak, how did you determine from that log the call was going to the local netwokr? |
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| 10:26.56 | mvanbaak | oh wait, this is only the debug for the phone ? |
| 10:30.24 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 10:31.53 | Mike8861 | hello all, i have been running a asterisk for 6 months, the user and call duration has been increase for the past 2 weeks. asterisk beging to freeze and stop working |
| 10:32.33 | angryuser | Mike8861 have you rebooted for 6 months ? |
| 10:32.50 | Mike8861 | how to troubleshoot the problem? SPEC: Celeron/512MB ram, connected user: 100 concurrent calls: 10, no PSTN trunk, pure SIP |
| 10:33.07 | Mike8861 | yes. i have rebooted a few times |
| 10:33.18 | Mike8861 | but this 2 weeks, likely, i have reboot a few hours |
| 10:33.19 | joobie | ya |
| 10:33.20 | joobie | i couldnt turn debug on in genreal.. had to be specific to a peer or ip |
| 10:33.32 | mvanbaak | huh ? |
| 10:33.36 | mvanbaak | sip set debug on |
| 10:33.57 | angryuser | Mike8861 any log traces? What do you meen by freeze ? |
| 10:34.30 | joobie | ahh mvan |
| 10:34.50 | Mike8861 | the server itself, including linux, apache, asterisk....i dunno how to say, stop responsding, it cannot be ping, no HTTP reply, SIP client cannot connect |
| 10:34.54 | joobie | sip set debug worked |
| 10:34.58 | joobie | without 'on' |
| 10:35.04 | joobie | im running 1.4 btw |
| 10:35.07 | mvanbaak | joobie: ah. |
| 10:35.12 | mvanbaak | yeah, thought so |
| 10:35.16 | Mike8861 | any hope ? |
| 10:35.46 | mvanbaak | there's always hope |
| 10:35.48 | angryuser | Mike8861 could it be hardware problem ? run memtest check you hdd, check temperature ect.. |
| 10:36.44 | joobie | mvanbaak, http://pastebin.com/m7d618365 |
| 10:36.48 | joobie | heh :) |
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| 10:37.14 | Mike8861 | angryuser: thank you. I will post the log when i get to the server]\ |
| 10:37.31 | angryuser | Mike8861 clean you pc from dust, sometimes it helps |
| 10:37.42 | Mike8861 | angryuser: does this related to the increase of user ? |
| 10:38.24 | angryuser | Mike8861 it depends , if you do transcoding or not , look in 'top' |
| 10:38.33 | joobie | Mike8861, have you looked at dmesg if anything odd is being reported? |
| 10:38.41 | joobie | or other log files? |
| 10:39.03 | Mike8861 | angryuser: no transcoding are being proceed, the only one u7sed a ulaw |
| 10:39.16 | joobie | also when teh box stops responding to pings, is it still alive from the local console? |
| 10:39.58 | Mike8861 | angryuser: TOP shows lack of RAM, this become more a problem after i installed openfire |
| 10:40.31 | Mike8861 | joobie: how to check the message ? i need to confirm about localconsole, never try that yet |
| 10:41.07 | joobie | what OS are you using? |
| 10:41.10 | Mike8861 | joobie: however, when i press the reboot button on the machine, sometime it cannot bootup succesfully |
| 10:41.36 | joobie | if apache and ping responses stop, etc - it sounds like a general resource issue |
| 10:41.43 | Mike8861 | joobie: as i remember, it halt at DEV <<--- something like that |
| 10:41.52 | Mike8861 | joobie: i am using centOS |
| 10:42.04 | joobie | ahh - my fave OS :P |
| 10:42.07 | angryuser | Mike8861 do what i suggested, and come back with results |
| 10:42.09 | mvanbaak | joobie: it's not sending the correct username etc to comvergence. |
| 10:42.09 | Mike8861 | joobie: me 2 |
| 10:42.10 | joobie | type 'dmesg' |
| 10:42.19 | Mike8861 | joobie: centos people is good people |
| 10:42.19 | joobie | also check /var/log/messages |
| 10:42.33 | joobie | heh :) |
| 10:42.46 | mvanbaak | joobie: From: "1234" <sip:sipuser@192.168.72.245>;tag=as5725fff6 |
| 10:42.59 | Mike8861 | joobie: when to type dmesg ? before it halt or after reboot ? or just type at any time |
| 10:43.09 | mvanbaak | joobie: try to set a valid outbound callerid |
| 10:43.11 | angryuser | im off |
| 10:43.14 | joobie | type it now.. see if there's anything odd coming up.. |
| 10:43.30 | mvanbaak | joobie: something like: Set(CALLERID(num)=some_number_here) |
| 10:43.40 | joobie | for example, my hdd has a few bad sectors.. if i type it now, ill get a few drive read / write errors in dmesg coming up.. it's a good indicator |
| 10:43.46 | Mike8861 | joobie: i cannot get to the machine now, we do not have network console or pdu, no hope until monday |
| 10:43.52 | mvanbaak | use the callerid you have at comvergence |
| 10:44.23 | *** part/#asterisk hads (n=hads@120.138.17.30) |
| 10:44.31 | Mike8861 | joobie: i will type memtest, top, and dmesg, also check /var/log/messages |
| 10:44.33 | joobie | mvanbaak, where do i set that caller id? |
| 10:45.34 | joobie | mike, ya.. good start. Because the whole system is grinding to a halt - presumably it is asterisk, though you should confirm this - will be easy to do if you have access to a live console when it's slowing down |
| 10:46.11 | mvanbaak | joobie: in your dialplan |
| 10:46.53 | joobie | top will show you which application is causing it, if it's an app issue.. dmesg and messages are a start if the problem is lower level than the problem application |
| 10:47.35 | joobie | mvanbaak, k sec.. ill have a quick read and implement |
| 10:47.35 | joobie | nice find btw ;P |
| 10:47.43 | Mike8861 | joobie: what grade of hardware do i need, if i am planning to have 400 concurrent user and 20 concurrent calls with SIP, no transcoding. |
| 10:48.29 | Mike8861 | celeron is just a transitional |
| 10:51.44 | joobie | mvanbaak, 'exten => _X.,1,Dial(SIP/comvergence/${EXTEN}) exten => _X.,n,Set(CALLERID(num)=61390010641)' - is that right? that 61 number is the username the SIP provider gave |
| 10:52.38 | joobie | Mike, im not the best person to ask RE asterisk - i'm pretty new to it.. grab the asterisk o'rielly book online (free) - it's got a section in one of the earlier chapters that talks about what hardware you should use for how many users |
| 10:52.59 | Mike8861 | joobie: thanks |
| 10:53.01 | Mike8861 | ~book |
| 10:53.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 10:53.16 | joobie | i'm a *nix sysadmin, that's how i sorta have some *nix knowledge.. but asterisk specific im limited |
| 10:54.06 | Mike8861 | joobie: u should get a asterisk admin as girlfriend |
| 10:54.11 | joobie | mvanbaak, i put that in the dialplan above, it's still using 1234. |
| 10:54.19 | joobie | haha :) |
| 10:54.27 | joobie | why is that? |
| 10:58.15 | mvanbaak | no idea |
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| 11:01.31 | Radi0ShacK | hello |
| 11:02.20 | joobie | mvanbaak, it's like the phone is overriding the caller id |
| 11:02.36 | joobie | i did 'localhost*CLI> sip show peer 1234' .. and it says 'Callerid : "" <61390010641>' |
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| 11:10.51 | joobie | mvanbaak, still alive? I think i got caller ID setup ok - but still no good |
| 11:12.08 | joobie | mvanbaak, http://pastebin.com/m6cbae374 that's the latest dump |
| 11:22.02 | joobie | hey mvanbaak .. i have to head off for a few hours |
| 11:22.05 | joobie | thanks for all your help |
| 11:22.16 | joobie | may have to pick this up later.. cheers mate |
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| 12:11.35 | deever | re |
| 12:13.16 | deever | connecting with openwengo directly from within my lan to my asterisk of my lan, it works perfectly! |
| 12:13.51 | deever | but now, over openvpn with tap and the same ip i have in my lan, i get: |
| 12:14.17 | deever | chan_sip.c:15055 handle_request_register: Registration from '<sip:nobody@astbox>' failed for '192.168.0.42' - No matching peer found |
| 12:14.50 | oej | Do you have a peer named "nobody"? |
| 12:14.57 | deever | as it seems, wengo is supplying a wrong username, isn't it? |
| 12:15.03 | deever | no |
| 12:15.14 | deever | so i better ask in the wengo channel? |
| 12:36.47 | *** join/#asterisk zydoon (n=zydoon@41.225.155.169) |
| 12:37.02 | *** part/#asterisk zydoon (n=zydoon@41.225.155.169) |
| 12:37.53 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:41.05 | MatBoy | it can be me, but the digium site is always so slow |
| 12:46.37 | *** join/#asterisk bmg505 (n=leon@196-209-8-9-ndn-esr-2.dynamic.isadsl.co.za) |
| 12:47.19 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-182-220-rndf-esr-5.dynamic.isadsl.co.za) |
| 12:47.34 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 12:53.24 | *** join/#asterisk okizo (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 12:53.25 | okizo | hi |
| 12:53.52 | okizo | i got a small on my asterisk system |
| 12:56.14 | okizo | i configure my sipura 3000 with asterisk server |
| 12:56.31 | okizo | ich there any one here ? |
| 12:56.38 | angryuser | yes |
| 12:56.49 | okizo | well |
| 12:56.51 | okizo | hi an |
| 12:57.05 | okizo | i configure my sipura 3000 |
| 12:57.11 | okizo | with my asterisk server |
| 12:57.48 | okizo | when i dial the number to access to my pstn lline |
| 12:57.49 | okizo | when i dial the number to access to my pstn lline |
| 12:58.01 | okizo | after to ring |
| 12:58.10 | okizo | after two ringtone |
| 12:58.14 | angryuser | take your time |
| 12:58.16 | okizo | it connect to my pstn line |
| 12:59.12 | okizo | but the problem is when i dial the local dial number it disconnect from my asterisk server |
| 13:00.11 | okizo | and also if u on the pstn line |
| 13:00.14 | angryuser | let's resume, so you dial from sipura to fxo and you cant dial out ? |
| 13:00.22 | okizo | it respond bussy |
| 13:00.38 | okizo | no |
| 13:01.04 | okizo | i create 3 sip extension from my asterisk server |
| 13:01.48 | okizo | i use one sip extension with my sipura 3000 |
| 13:01.55 | okizo | from my work |
| 13:02.17 | okizo | which is working actually is : |
| 13:02.47 | okizo | from home i can dial to my work over my asterisk server's sip extension |
| 13:03.06 | okizo | now what i have done is : |
| 13:03.35 | okizo | i connect my work's pstn line on my sipura3000 |
| 13:03.41 | okizo | and i also configure it |
| 13:04.23 | angryuser | okizo you wrote a half page without really sayng what is going wrong, resumer |
| 13:05.00 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 13:05.38 | okizo | from home if i dial 123 to join my work's pstn line |
| 13:06.13 | okizo | from home if i dial my sip extension : "123" to join my work's pstn line |
| 13:06.47 | okizo | after 10 sec it connect automatically to my work's pstn line |
| 13:07.27 | okizo | so heard my work's dialtone |
| 13:07.57 | okizo | if i dial outside number from my work's line |
| 13:08.09 | okizo | it disconnect from my asterisk server |
| 13:08.31 | angryuser | pastebin your extensions.conf the part when you dial out |
| 13:08.49 | okizo | and using from home pstn line if i dial to my works pstn line i heard saying it bussy |
| 13:12.59 | okizo | sorry u don't understand my queston |
| 13:15.17 | angryuser | okizo pastebin your extensions.conf |
| 13:15.22 | angryuser | ~pastebin |
| 13:15.23 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:17.41 | okizo | ok |
| 13:17.46 | okizo | plz wait |
| 13:20.57 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 13:21.23 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 13:22.55 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:23.47 | troubled | hey guyes, is there a standalone asterisk front end? if so, whats the name? |
| 13:26.05 | okizo | no leave it |
| 13:26.10 | okizo | nothing working |
| 13:26.21 | okizo | thx angryuser |
| 13:26.22 | okizo | bye |
| 13:27.11 | [TK]D-Fender | troubled: clarify "standalone" |
| 13:27.48 | troubled | [TK]D-Fender: I am trying to think of a web front end i seen once that was recommended. iirc, it was actually shipped with a centos based install |
| 13:28.17 | troubled | so I guess it wouldn't be standalone per se. but I am trying to find the name and see if perhaps it was available from svn or something |
| 13:28.32 | MikeJ | trixbox |
| 13:28.58 | [TK]D-Fender | troubled: most of them are "standalone, and jsut get used in distro installs. |
| 13:29.03 | MikeJ | its a full iso install but not distributed by digium |
| 13:29.29 | [TK]D-Fender | troubled: FreePBX , ScopServ (not free), Thirdlane, etc are all "standalone". |
| 13:29.45 | troubled | ya, we seen that trixbox come up. i will have to look and see if its the same one I seen before |
| 13:30.08 | [TK]D-Fender | troubled: You seem reversed from what you are asking. |
| 13:30.25 | [TK]D-Fender | troubled: Are you loking for a GUI that DOESN'T jsut come bundled with a whole ISO? |
| 13:31.07 | angryuser | troubled elastix also |
| 13:32.26 | troubled | [TK]D-Fender: me and another guy are trying to figure out the name of a front end I seen once. iirc, it was mentioned in some install howto. I cant remember if it was a centos based full install or not though |
| 13:32.36 | angryuser | [TK]D-Fender i wonder why [TK] ? |
| 13:32.51 | [TK]D-Fender | troubled: And now you aren't being specific at all about what you're looking for. bundled or not. |
| 13:32.54 | troubled | freepbx interface looks sorta familiar though. still looking |
| 13:33.15 | angryuser | centos based instal standalone script is elastix alos |
| 13:33.15 | troubled | [TK]D-Fender: I am looking for the screenshots of an app I seen a year or two ago |
| 13:33.38 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 13:33.38 | [TK]D-Fender | troubled: Yet another unnamed search. |
| 13:33.49 | [TK]D-Fender | troubled: do you have ANY clue what you're looking for? |
| 13:34.12 | [TK]D-Fender | troubled: "app" can be ANYTHING. |
| 13:34.52 | troubled | [TK]D-Fender: I am not in the market to install anything myself. I am just trying to track down an old app for someone else that I seen |
| 13:35.08 | troubled | angryuser: ya, I think I tried elastix once along time ago, definetly not that one |
| 13:35.46 | angryuser | old asterisk@home ? |
| 13:35.47 | [TK]D-Fender | troubled: Well I guess you may as well jsut google "asterisk" and "scrrenshot", cross your fingers and say a prayer and also hope that the picture is still up even if the "app" is still being used |
| 13:36.15 | troubled | angryuser: hmmm, rings a bell, let me google |
| 13:36.38 | troubled | it might have been freepbx though. the screenshots I remember had a similar look/feel. |
| 13:36.39 | [TK]D-Fender | troubled: for GUI-like bits : Flash Operator Panel (FOP), Hud (and Hudlite) |
| 13:37.00 | angryuser | it is 'asterisknow' now |
| 13:37.06 | troubled | ok, sec |
| 13:37.09 | [TK]D-Fender | troubled: For reporting : Areski CDY |
| 13:37.30 | [TK]D-Fender | angryuser: what is 'asterisknow' now? |
| 13:37.34 | [TK]D-Fender | CDR* |
| 13:37.43 | MikeJ | troubled: freepbx is front end in trixbox |
| 13:37.45 | troubled | angryuser: awesome, asterisknow was it |
| 13:37.53 | troubled | I remember the install screenshots and everything |
| 13:38.50 | troubled | angryuser: _much_ apreciated :) |
| 13:39.26 | angryuser | so why [TK] ? |
| 13:39.28 | angryuser | ;) |
| 13:40.11 | angryuser | [TK]D-Fender asterisk@home before |
| 13:40.23 | [TK]D-Fender | angryuser: No, it wasn't |
| 13:40.35 | [TK]D-Fender | Andry Asterisk@Home became trixbox |
| 13:40.52 | [TK]D-Fender | angryuser: rather |
| 13:41.08 | angryuser | [TK]D-Fender heh ok i was thinking @home was build by digium&community |
| 13:41.35 | [TK]D-Fender | angryuser: A@H used FreePBX... you're not looking too deep here.. |
| 13:41.43 | MikeJ | no.. they forced them to change the name |
| 13:41.49 | angryuser | never used @home anyway |
| 13:51.28 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:59.02 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 13:59.02 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 14:08.26 | MatBoy | is * addicted |
| 14:18.21 | *** join/#asterisk _sc0tty_ (n=_sc0tty_@217.144.147.41) |
| 14:23.52 | tompaw | Hello |
| 14:24.03 | tompaw | What average load is safe considering Asterisk operation? |
| 14:24.30 | tompaw | Right now I have 40 sip channels with non-digium/g729 giving me around 0.50 cpu load average. |
| 14:24.38 | tompaw | Should I start worrying? |
| 14:25.17 | [TK]D-Fender | tompaw: How is it doing? |
| 14:26.33 | tompaw | Well, the CPU(s) idle is over 95% |
| 14:26.36 | [TK]D-Fender | tompaw: what are the machine's specs? |
| 14:26.45 | [TK]D-Fender | tompaw: and thats BAD? |
| 14:26.52 | tompaw | quadro 3Ghz Xeons. |
| 14:26.57 | tompaw | 4 GB ram |
| 14:27.05 | tompaw | running @ 64-bit *Now build. |
| 14:27.23 | tompaw | [TK]D-Fender: I don't think so, I'm just confirming, it's my first time :P |
| 14:27.51 | codefreeze-lap | tompaw: really, use 'sipp', and find out your limits. First, do really short phone calls, and pump up the speed until asterisk can't handle it. Then, do really long calls, and build up the number until asterisk can't keep up. You should know your limits. |
| 14:27.57 | [TK]D-Fender | tompaw: Worried about * NOT posing a load issue |
| 14:28.14 | tompaw | [TK]D-Fender: excuse me? |
| 14:28.38 | tompaw | codefreeze-lap: didn't know the tool, thanks! |
| 14:28.49 | [TK]D-Fender | tompaw: tompaw>Well, the CPU(s) idle is over 95% <- if * is a low load, why are you worried? |
| 14:31.02 | tompaw | [TK]D-Fender: I'm not worried, I'm just asking if there are no hidden things I should be worrying about. Like the other factors to check (something I can't even think of). |
| 14:31.55 | [TK]D-Fender | tompaw: Transcoding, echo cancellation, & recording are the things that put a real load on a system |
| 14:34.09 | tompaw | [TK]D-Fender: now as you mention it, my CLI (which now looks like those wooden thing in railway tracks watched from a train running at 100mph) produces sth like this from time to time: |
| 14:34.13 | tompaw | [Jul 19 16:33:01] NOTICE[9889]: rtp.c:787 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: |
| 14:34.50 | [TK]D-Fender | tompaw: indeed * does not support *. tell your clients to stop using it, and if its your ITSP, you probably don't have a choice |
| 14:35.03 | [TK]D-Fender | CNG* |
| 14:36.42 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 14:37.22 | tompaw | [TK]D-Fender: actually, it's Arbinet. If it's impossible to re-configure, I guess I'll have to live with it. |
| 14:42.24 | MatBoy | it is simply possible to collect DTMF tones isn't it ? |
| 14:42.38 | MatBoy | I mean collect so that it becomes one string |
| 14:42.53 | [TK]D-Fender | MatBoy: Impossible no, easiy & practical with * in the middle of a call... no. |
| 14:43.05 | [TK]D-Fender | MatBoy: "when" is the question. |
| 14:43.16 | [TK]D-Fender | MatBoy: If you're talking about prompting for input, then yes. |
| 14:43.35 | [TK]D-Fender | MatBoy: You can build an IVR to collect by digit, or "cor show application READ" |
| 14:43.41 | [TK]D-Fender | MatBoy: You can build an IVR to collect by digit, or "core show application READ" |
| 14:43.42 | MatBoy | [TK]D-Fender: yes, prompting, it's needed for my query project where I'm working on |
| 14:44.40 | MatBoy | [TK]D-Fender: an IVR would be over the top for it I guess or do you recommend it ? |
| 14:45.49 | [TK]D-Fender | MatBoy: look ar "read' and see if it does it for you. otherwise, collect the digits yourself in an IVR |
| 14:46.18 | MatBoy | [TK]D-Fender: yep I will |
| 14:46.22 | MatBoy | thanks again ! |
| 14:55.12 | *** join/#asterisk Radi0ShacK (n=puts@41.232.113.183) |
| 14:57.02 | *** join/#asterisk academy (n=adam@unaffilated/academy) |
| 14:58.14 | academy | I have a SIP trunk coming into Asterisk for PSTN Termination and SIP phone registered to Asterisk. If I make a call to a PSTN number, does the RTP stream go through Asterisk or direct to the SIP Trunk provider's media gateway? |
| 15:04.09 | De_Mon | academy draw that scenario with pictures and arrows and stuff and the answer should become clear |
| 15:07.31 | academy | De_Mon: That won't help me. The RTP stream should be seperate to the SIP stream and it should go direct. I wondered whether it actually does. |
| 15:09.03 | tristanbob | academy, by default all asterisk calls go through asterisk (SIP + RTP) |
| 15:10.04 | academy | tristanbob: ok, thanks |
| 15:10.26 | tristanbob | the only caveat is if you turn on reinvite |
| 15:10.43 | *** join/#asterisk ManxPower (n=manxpowe@94.sub-70-222-194.myvzw.com) |
| 15:10.55 | tristanbob | although I don't think that will work over SIP trunk |
| 15:11.04 | tristanbob | only SIP to SIP agents |
| 15:11.31 | tristanbob | so you can be sure that RTP goes through asterisk before SIP trunk |
| 15:11.48 | tristanbob | academy, why do you have a media gateway if you have asterisk? |
| 15:13.02 | *** join/#asterisk DSpair (n=D-Spare@74-130-9-203.dhcp.insightbb.com) |
| 15:13.16 | DSpair | G'day all. |
| 15:13.47 | DSpair | I need a little assistance in configuring a ring group for my fax server lines. Can someone assist me? |
| 15:16.49 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 15:23.35 | _sc0tty_ | hi All. could do with a bit of a pointer. I have asterisk attempting an INVITE to an outbound proxy. the proxy returns a 407, but rather than authenticating, asterisk gives the notice "handle_response_invite: failed to authenticat on INVITE to"[user]", passing a 503 back to the client extension. Wireshark shows that asterisk did not attempt to authenticate after the 407, it just gave up. what setting do I use to turn on authentication? |
| 15:29.40 | *** join/#asterisk [netman] (n=netman@68.Red-88-8-164.dynamicIP.rima-tde.net) |
| 15:33.48 | xiando | _sc0tty_: are you using SIP/nameofthe[infosection]ofthatproxy? it won't care about the [thatproxyinfo] if you just go SIP/sip.domain.tld |
| 15:37.39 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 15:41.17 | _sc0tty_ | xiando: so you are saying if i change extensions.conf so that my Dial() clause is Dial(SIP/[numbertodial]@sip.provider.tld,,r) instead of the current Dial(SIP/[numbertodial]@nameofsip.conf[]section, it wil work? i will try now.,,r) |
| 15:42.48 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 15:42.50 | _sc0tty_ | I still get the same. wireshark shows asterisk responding to the provider's 407 with an ACK, but it does not then resend the INVITE with the digest authentication. |
| 15:43.02 | *** join/#asterisk tloges (n=tiagolog@201-14-220-32.nhoce701.dsl.brasiltelecom.net.br) |
| 15:46.08 | *** join/#asterisk cestchiant (n=none@ARouen-153-1-18-118.w90-17.abo.wanadoo.fr) |
| 15:46.19 | cestchiant | hello :-) |
| 15:46.36 | cestchiant | I use Asterisk 1.4.21 on etch. |
| 15:46.56 | cestchiant | When I call after a random time, my call is "closed" |
| 15:47.27 | cestchiant | I recieve incomin call via T2 and Patton and I send my outbound call via Internet |
| 15:47.40 | cestchiant | I have the same porblem in outbound and incoming call. |
| 15:48.02 | _sc0tty_ | I do not see "Auth Attempt" in the console when this happens, yet looking at the source of chan_sip, do_proxy_auth() should emit this. the only reason this would not happen is of the conditional "if(!p->options && !(p->options = ast_calloc(1, sizeof(*p->options)))) failed, or if asterisk is ignoring the incoming response. |
| 15:48.03 | cestchiant | My calls are stopped without reason.... |
| 15:48.15 | cestchiant | do you have any ideas about my problem? |
| 15:48.17 | cestchiant | thanks |
| 15:52.57 | *** join/#asterisk xenonex (n=xenonex@88.204.197.158) |
| 15:53.15 | kash | cestchiant: probably timeout value is set too low |
| 15:53.19 | kash | also, check verbosity level |
| 15:53.25 | kash | core set verbose 10 |
| 15:53.27 | kash | or something |
| 15:54.14 | cestchiant | kash: where can I found this timeout? |
| 15:55.27 | kash | read the asterisk manual |
| 15:55.36 | cestchiant | for example When I call someone, my call is "stopped" after 30 seconds. I redial the same someone and I speak during 10 minutes... |
| 15:55.49 | kash | also, it would help if you would paste your dialplan |
| 15:55.59 | kash | into pastebin |
| 15:56.13 | cestchiant | ok |
| 15:57.35 | cestchiant | but, it's strange, because this problem don't appear before... |
| 16:00.20 | _sc0tty_ | I am adding |
| 16:01.29 | _sc0tty_ | i have added some log calls to chan_sip.c and recompiled, it seems that do_proxy_auth() never gets called from handle_response_invite(), when the 407 is received. any ideas why? |
| 16:02.25 | *** join/#asterisk tloges (n=tiagolog@201-14-220-32.nhoce701.dsl.brasiltelecom.net.br) |
| 16:03.15 | cestchiant | kash: http://pastebin.com/d13ba6411 (I put my log about asterisk....) at 18:46 o'clock, my call is "stopped" without reason.... |
| 16:04.09 | kash | this isn't the support channel for AMP |
| 16:04.13 | *** part/#asterisk deever (n=deever@static.172.68.46.78.clients.your-server.de) |
| 16:04.33 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 16:05.05 | cestchiant | :( |
| 16:05.18 | kash | see topic |
| 16:05.21 | kash | thanks |
| 16:05.57 | xiando | I had to look it up. "AMP provides a web-based, user-friendly administrative interface to Asterisk and is a standardized implementation of Asterisk (i.e. dialplan) that is maintainable, flexible and extensible." http://www.voip-info.org/wiki/view/AMP+Resellers |
| 16:06.21 | xiando | If you bought the AMP then ask the seller for support. |
| 16:07.52 | cestchiant | ok, but if my channel is closed, I don't think that problem is in relation of amp |
| 16:11.31 | kash | yes |
| 16:11.32 | kash | yes it is |
| 16:11.41 | kash | they modify so much shit in asterisk, it's hardly recognisable. |
| 16:14.15 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 16:15.15 | cestchiant | ok, :) |
| 16:17.40 | kash | cestchiant: #freePBX |
| 16:18.37 | *** join/#asterisk ManxPower (n=manxpowe@167.sub-75-248-204.myvzw.com) |
| 16:19.13 | *** join/#asterisk FunkyGMT (n=PeterHay@modemcable245.49-57-74.mc.videotron.ca) |
| 16:19.17 | FunkyGMT | Hi all |
| 16:19.33 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:20.09 | FunkyGMT | I have a question. What's the function to save, and to retrive, a defined variable in Asterisk. (user-customisable phone number, by example) |
| 16:20.38 | cestchiant | kash: thanks (but nobody seems available ) |
| 16:24.16 | ManxPower | astdb |
| 16:24.28 | ManxPower | see DBGET DBSET functions |
| 16:24.29 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 16:26.55 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 16:30.07 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 16:36.10 | *** join/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 16:36.21 | *** part/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 17:00.36 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 17:00.40 | drfreeze | Hello |
| 17:01.08 | drfreeze | Anyone have a minute to help me get a pri card up and running: wcte12 |
| 17:01.40 | drfreeze | it appears to be recognized and zttool gives it a green status |
| 17:01.59 | drfreeze | but, I'm not sure how to see it inside asterisk. There are no zap channels |
| 17:06.18 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:09.00 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 17:09.33 | *** join/#asterisk pons (n=pons@unaffiliated/pons) |
| 17:10.03 | *** join/#asterisk tomcruize (n=elgroper@69.77.169.14) |
| 17:10.12 | tomcruize | hi |
| 17:11.28 | tomcruize | I have a fast busy, sangoma on 1.4.21.1 asterisk. Seems the D channel is down. Anyone care to lend a hand? |
| 17:11.44 | drfreeze | Getting the following error on the T1 card: |
| 17:11.45 | drfreeze | http://pastie.textmate.org/private/lcwlbbayihnmrvywczjwtq |
| 17:13.25 | pons | guys, anyone here has had any problem with dtmf and chan_mobile? I'm using 1.4.20 from debian, and chan_mobile rev 454 with 1.4.x patch. DTMF are detected when calling any IAX extension, I hear them and they are also shown in debug, but when I spect it with a WaitExten, no DTMF is detected in debug, any suggestions? |
| 17:13.54 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 17:14.08 | [TK]D-Fender | drfreeze: pastebin your configs. |
| 17:14.20 | drfreeze | [TK]D-Fender: ok. just a sec |
| 17:16.54 | drfreeze | [TK]D-Fender: http://pastie.textmate.org/private/nzxmcxivl0b6mpvwblmoq |
| 17:18.01 | [TK]D-Fender | drfreeze: "ztcfg -vvvv" |
| 17:19.09 | drfreeze | [TK]D-Fender: |
| 17:19.10 | drfreeze | http://pastie.textmate.org/private/6qhxwhotzrb2jqkxkfcxuw |
| 17:20.09 | [TK]D-Fender | drfreeze: ok, stop * and restart |
| 17:20.39 | [TK]D-Fender | hey. |
| 17:20.44 | [TK]D-Fender | Your signalling is broken |
| 17:20.54 | [TK]D-Fender | ;signalling=pri_cpe |
| 17:20.56 | [TK]D-Fender | signalling=fxo_ls |
| 17:21.05 | drfreeze | [TK]D-Fender: switch that back to pri_cpe? |
| 17:21.08 | [TK]D-Fender | drfreeze: you commented out PRI and switched ot ls. |
| 17:21.15 | [TK]D-Fender | drfreeze: what do you have plugged into that? |
| 17:21.38 | drfreeze | the t1 that goes to the existing adtran |
| 17:22.15 | [TK]D-Fender | drfreeze: And what does IT put out? |
| 17:22.27 | drfreeze | IT? |
| 17:22.33 | [TK]D-Fender | the Adtran. |
| 17:23.09 | drfreeze | the adtran converts the t1 to analog - we are circumventing it |
| 17:24.45 | drfreeze | [TK]D-Fender: changed the signalling back to pri_cpe, now the module loads |
| 17:24.58 | drfreeze | app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 17:25.20 | [TK]D-Fender | drfreeze: And your answer does not denote your having any knowledge of what you're working with. |
| 17:25.34 | [TK]D-Fender | drfreeze: And never paste useless error messages like that. |
| 17:25.47 | [TK]D-Fender | drfreeze: that doesn't tell me what yuo TRIED to dial |
| 17:26.06 | drfreeze | [TK]D-Fender: sorry. dialed my cell phone |
| 17:26.15 | drfreeze | iow, trying to see if I can dial out |
| 17:26.21 | [TK]D-Fender | drfreeze: Another completely meaningless answer... |
| 17:26.36 | tomcruize | pastie: hi! |
| 17:27.50 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 17:27.51 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 17:31.31 | *** join/#asterisk Segnale007 (n=Segnale0@host141-4-dynamic.18-79-r.retail.telecomitalia.it) |
| 17:37.23 | DSpair | Hey gang, having a problem using iaxmodem. Anyone will to assist? |
| 17:37.26 | drfreeze | [TK]D-Fender: This is the first time I have played with a TE card. Are these cards supposed to be assigned IP addresses? |
| 17:37.38 | tomcruize | hi, I am newbie. Can anyone help with a fast busy on pri, Sangoma card/Asterisk 1.4.21.1 |
| 17:37.50 | [TK]D-Fender | drfreeze: No, they are not NIC's |
| 17:40.35 | jeev | fenderino |
| 17:40.52 | jeev | i got my asterisk/squid/multiwan box |
| 17:40.52 | *** join/#asterisk AMUG (n=junky@96.20.137.156) |
| 17:40.56 | jeev | now i gotta prep it.. freebsd 7? :D |
| 17:44.15 | *** join/#asterisk RoyK (n=roy@212.17.150.132) |
| 17:46.57 | [TK]D-Fender | AMUG: y0, long time no hear. How goes? |
| 17:50.08 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 18:04.38 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 18:17.57 | pons | guys, what would be the best way to call one extension first and when it answers, call another one? |
| 18:18.52 | [TK]D-Fender | pons: to do what exactly? |
| 18:19.41 | pons | i'm planning some internal webpage, you put your phone number and the one you want to call, asterisk calls you and then the one you call |
| 18:22.49 | pons | ideas? |
| 18:24.47 | _sc0tty_ | Hi all. when asterisk sends an INVITE to a SIP peer, i can use the sip.conf fromuser=blah clause to manipulate the From: tag in the outgoing INVITE. however, this only seems to change the <sip:blah@domain> part of the From: tag. how do I change the "alias" part at the start of the From tag, in the quotation marks? |
| 18:28.00 | DSpair | Hey guys, I fixed my IAXmodem problem, but when I try to call in to the asterisk box on one of the fax lines, I get ring-no-answer... Any suggestions? |
| 18:31.15 | DSpair | In the asterisk console, I see the call come in on a Zap channel, the server rings the iaxmodem channel, but there is never an answer. |
| 18:32.03 | DSpair | Oh, wait, nevermind. It appears to have come on-line now. |
| 18:33.17 | DSpair | Woohoo!!! It's all working!!! YAY!!! |
| 18:39.18 | *** join/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 18:45.01 | DSpair | Thanks again for all of the help I have gotten here (no sarcasm intended). I now have a VERY good solution in place thanks to all of your help!!! |
| 18:46.36 | DSpair | Have a good weekend all!!! |
| 18:55.00 | *** join/#asterisk dieno (i=771e6d38@gateway/web/ajax/mibbit.com/x-53eb2f802df308fc) |
| 18:55.37 | dieno | hmm can any one tell me what type of hardware i need to get local DIDs |
| 18:56.31 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 18:56.57 | WhiteWolf | dieno: you can get local dids from a handful of voip providers |
| 18:57.30 | dieno | WhiteWolf hmm rite but in egypt i cant find any :) |
| 18:58.49 | WhiteWolf | dieno: in that case: a pri card... perhaps for ISDN, and the necessary supporting pc -- then just get a line from the local telco |
| 18:59.38 | dieno | WhiteWolf ok this is what i was thinkin but how many i can get in one ISDN |
| 18:59.46 | dieno | i mean how many DIDs |
| 19:01.15 | *** part/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 19:01.18 | WhiteWolf | dieno: well, dids are seperate from the "lines" |
| 19:01.25 | WhiteWolf | you could have 10 dids but only one incomming channel |
| 19:01.27 | dieno | hmm ok thnx |
| 19:01.38 | WhiteWolf | obviously still only able to handle 1 call |
| 19:01.53 | dieno | what abt E1 line |
| 19:02.49 | WhiteWolf | a full E1... 1.5ish mbit is 24 call channels |
| 19:03.13 | WhiteWolf | there's a couple reserved in that |
| 19:03.18 | dieno | hmm ok |
| 19:03.24 | WhiteWolf | 1 for digital data, and 1 for call setup, so it gives you about 22 channels |
| 19:03.25 | dieno | i though its around 30 - 32 |
| 19:03.58 | WhiteWolf | oops, my fault |
| 19:03.59 | *** join/#asterisk Itiliti (n=Itiliti@75.150.198.1) |
| 19:04.03 | WhiteWolf | i just quoted for a T1 |
| 19:04.13 | dieno | ohh ok i was also not sure abt that |
| 19:04.14 | dieno | :) |
| 19:04.14 | WhiteWolf | E1 are slightly faster at 2mbit and have 32 time slots, 2 reserved for 30 usable |
| 19:04.24 | Itiliti | I am using a PRI, and would like to be able toi use *67 to block my caller ID on some calls. How can I do that? |
| 19:05.05 | WhiteWolf | Itiliti: make an extention that matches *67..... then just blank the callerid before sending i suppose |
| 19:05.15 | Strom_M | no. |
| 19:05.27 | Strom_M | you have to set the caller ID presentation restriction flag |
| 19:05.33 | WhiteWolf | there you go |
| 19:05.46 | dieno | but how can these provider get alots of DIDs do they have any Hardware |
| 19:06.23 | WhiteWolf | dieno: the telco gets blocks of dids which they resell to you... and of course they've a lot of hardware |
| 19:06.33 | WhiteWolf | my pri work is more on the digital data rather than the voice side |
| 19:06.43 | WhiteWolf | so, Strom_C is right of course |
| 19:07.20 | dieno | hmm kool thnx for all this knowledge |
| 19:07.22 | Itiliti | where do you set the caller id restriction flag? |
| 19:07.35 | Itiliti | can you do it per call to enable/disable it? |
| 19:07.40 | Strom_M | Itiliti: look at the setcallingpres application or somesuch |
| 19:07.41 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-141-153.neo.res.rr.com) |
| 19:07.52 | WhiteWolf | Itiliti: yes |
| 19:08.03 | *** join/#asterisk hardhatpat (n=pat@c-24-21-226-34.hsd1.mn.comcast.net) |
| 19:08.03 | Itiliti | will do thx.. |
| 19:08.55 | Strom_M | WhiteWolf: also, you're a bit muddled on how T1 works compared to E1 |
| 19:09.10 | Strom_M | T1 doesnt reserve a full channel for framing like E1 does |
| 19:09.11 | WhiteWolf | Strom_C: at no surprise to myself |
| 19:09.37 | Strom_M | T1 uses a single bit once every 24 frames |
| 19:09.45 | WhiteWolf | Strom_M: ahh ha |
| 19:10.31 | hardhatpat | if i send a missed call to a separate voicemail system, is there any way to maintain the original callerid? |
| 19:10.35 | Qwell | bit muddled every 24 frames? |
| 19:13.04 | WhiteWolf | Strom_M: thank you for clarifying, still in my learning stage about the technical workings of PRI interfaces |
| 19:14.07 | Strom_M | WhiteWolf: it's really not that difficult |
| 19:14.19 | Strom_M | just go read up on T1, E1, and ITU-T Q.931 |
| 19:14.34 | WhiteWolf | Strom_M: indeed not, but i obviously misread/misunderstood so correcting me helps me learn |
| 19:20.22 | _sc0tty_ | can I use ${} style variables in sip.conf as well as extensions.conf? |
| 19:28.42 | gramulhaozin | anyone wants to sell 4 Cisco 7940 power supplies ? |
| 19:31.00 | Qwell | gramulhaozin: ebay |
| 19:33.13 | gramulhaozin | need it overnight |
| 19:34.51 | WhiteWolf | cisco direct would always work |
| 19:35.01 | WhiteWolf | if you need-it-now-and-when-i-say-now-i-mean-now support |
| 19:37.34 | ManxPower | _sc0tty_: NO! |
| 19:41.13 | *** join/#asterisk Jumpie (n=jumpie@pool-96-231-155-62.washdc.fios.verizon.net) |
| 19:41.17 | Jumpie | hey guys :) been awhile |
| 19:41.37 | Jumpie | greetings fender |
| 19:42.08 | Jumpie | kinda silly issue, havent actualy rebooted my system in awhile, was kinda lyin dormant last few months, well now somehow my zaptel card doesnt work, and i can't remember what i had to do to renitialize it |
| 19:44.19 | jaytee | service restart zaptel? |
| 19:44.25 | jaytee | then restart * |
| 19:47.48 | Jumpie | within asterisk? or from command line |
| 19:48.39 | Jumpie | btw hey jaytee :D |
| 19:50.24 | jaytee | hey Jumpie |
| 19:50.43 | jaytee | from command line, exit the * console |
| 19:51.00 | jaytee | what distro are you running? |
| 19:54.10 | Jumpie | centos 5 |
| 19:54.17 | Jumpie | service doesnt work |
| 19:54.42 | Jumpie | Linux ippbx.jumpieism 2.6.18-53.1.19.el5xen |
| 19:59.24 | *** join/#asterisk angryuser (n=sldf@78.115.233.228) |
| 20:01.06 | Jumpie | i cant remembe the command i used last time lol |
| 20:03.22 | angryuser | stop asterisk reload mod's |
| 20:03.27 | angryuser | rmmod |
| 20:04.34 | angryuser | and then modprobe |
| 20:05.47 | Jumpie | rmmod doesnt work |
| 20:06.23 | angryuser | what have you typed ? |
| 20:06.51 | Jumpie | i stopped all the asterisk mods |
| 20:06.52 | Jumpie | stop now |
| 20:06.56 | Jumpie | which took me back to shell |
| 20:07.01 | Jumpie | i typed rmmod and it doesnt recognize |
| 20:07.05 | Jumpie | sorry ffs..im rusty lol |
| 20:07.14 | angryuser | rmmod zaptel |
| 20:07.16 | Jumpie | this was just running smooth for months lol |
| 20:07.31 | Jumpie | thats what im saying rmmod zaptel doesnt work, it doesnt recognize the rmmod command |
| 20:07.48 | Jumpie | maybe path issue? |
| 20:08.13 | angryuser | nope, you dont have tha command ? under 'root' also ? |
| 20:08.31 | *** join/#asterisk __yy (n=misha@blk-224-201-7.eastlink.ca) |
| 20:08.33 | Jumpie | im as root |
| 20:08.33 | *** part/#asterisk __yy (n=misha@blk-224-201-7.eastlink.ca) |
| 20:08.35 | Jumpie | and it still doesnt like it |
| 20:08.56 | Jumpie | centos :D does that make a diff? |
| 20:08.58 | jaytee | if you type lsmod what do you get? |
| 20:09.08 | angryuser | what car do you need to initialize ? fxo fxs ? |
| 20:10.08 | jaytee | did you compile zaptel from source or use a package? |
| 20:10.13 | angryuser | type zaptelhardware or something you will see what driver is used, rmmod it and modprobe it |
| 20:10.16 | Jumpie | fxo |
| 20:10.37 | Jumpie | lsmod, rmmod not valid commands |
| 20:10.44 | Jumpie | i compi;led it no problem |
| 20:10.49 | Jumpie | bujt like i said i havent had to mess with it in ages lol |
| 20:11.11 | jaytee | jumpie are you running as root? |
| 20:11.14 | Jumpie | i dont see what would suddenly make it not work anymore |
| 20:11.17 | Jumpie | jay yeah |
| 20:11.31 | jaytee | then something's hosed with your path |
| 20:11.48 | jaytee | if it's not recognizing the lsmod command or the rmmod command |
| 20:11.55 | angryuser | yep rmmod must be everywhere |
| 20:12.02 | Jumpie | hmmm |
| 20:12.08 | Jumpie | i dont recall using those when i instralled it |
| 20:12.17 | Jumpie | im gonna go back to m y notes i got buried somewhere |
| 20:12.32 | Jumpie | where should it be, pathwise? |
| 20:12.53 | jaytee | Jumpie when you finished running make during compile did you run make install and then chkconfig zaptel on ? |
| 20:13.13 | Jumpie | yea...cause this has been workin fine for months |
| 20:13.32 | Jumpie | oh crap hold on..possibly i did some kernel updates today |
| 20:13.34 | Jumpie | havent rebooted yet |
| 20:13.38 | Jumpie | would that have anything to do with it? |
| 20:13.45 | angryuser | this command are part of any linux, and i edited patch's a long time ago so, ask linux guys |
| 20:13.47 | Jumpie | i still dont see the command suddenly not being available |
| 20:13.48 | *** join/#asterisk Yoshinoya (n=yan@netblock-208-127-50-44.dslextreme.com) |
| 20:13.49 | angryuser | or reboot |
| 20:13.53 | jaytee | type echo $PATH at the command line and make sure you see /sbin |
| 20:13.57 | jaytee | or reboot |
| 20:14.20 | jaytee | because /sbin is where lsmod and rmmod live |
| 20:14.23 | Jumpie | lol hmm yeah |
| 20:14.39 | Jumpie | lemme add it there...thats right i remember this...i had a messed up path |
| 20:14.44 | Jumpie | i gotta make it stick so id ont have to manually keep appending |
| 20:15.06 | Yoshinoya | hey I am using AsteriskNow with vitelity |
| 20:15.14 | Jumpie | now it works angryuser :D |
| 20:15.22 | Yoshinoya | I want to set inbound calls to work, where is the setting in vitelity |
| 20:15.26 | Yoshinoya | to point to my server? |
| 20:15.32 | Jumpie | module zaptel does not exist...wtf |
| 20:15.34 | Jumpie | it just disappeared |
| 20:16.05 | Jumpie | not found..hmm not good |
| 20:16.07 | Jumpie | on modprobe |
| 20:16.10 | angryuser | jumpie reboot |
| 20:16.18 | Jumpie | k |
| 20:16.19 | Jumpie | brb..thx |
| 20:18.48 | Yoshinoya | hrm, does anyone here use Vitelity? if not, what service do you use and recommend for a novice? |
| 20:22.27 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:23.03 | angryuser | Yoshinoya service for what ? |
| 20:23.11 | Yoshinoya | DID |
| 20:23.35 | Yoshinoya | I use vitelity currently, and I am having a hard time configuring it to work, even tho I think the only files I edit is sip.conf and extensions.conf |
| 20:23.40 | *** join/#asterisk Jumpie (n=jumpie@pool-96-231-155-62.washdc.fios.verizon.net) |
| 20:23.43 | Yoshinoya | (using ASteriskNOW) |
| 20:23.50 | Jumpie | man somethin is messed up, when i booted up it said /dev/zaptel was not found |
| 20:23.56 | Jumpie | and modprobe doesnt find it...wtf did my card crap out |
| 20:24.07 | Jumpie | it was a cheap 1 port fxo wildcard lol |
| 20:24.33 | angryuser | Yoshinoya so you configured some sip trunk's i suppose ? |
| 20:24.59 | angryuser | Jumpie ask fender i think he loves questions like that |
| 20:25.24 | Yoshinoya | I believe so (I am not too clear what a "trunk" is), but I am trying to configure it so when I get inbound calls |
| 20:25.27 | Jumpie | lol |
| 20:25.32 | Yoshinoya | it is not just the busy/bad configuration sound |
| 20:25.35 | Jumpie | i just dont get why it suddenly doesnt work |
| 20:25.55 | Jumpie | well supposedly ' SIP TRUNK' is a taboo term and admins here flip out over the usage :D |
| 20:26.31 | angryuser | i know, whatever |
| 20:27.20 | Yoshinoya | I am not sure what I am doing wrong because Vitelity gives me the lines of code to put in |
| 20:27.21 | angryuser | Yoshinoya so you got sip friend configured with provider params and another sip/zap/whatever with that did associated ? |
| 20:27.53 | Yoshinoya | right now, i just have the provider added (but I think this is for outbounds) |
| 20:28.01 | Yoshinoya | and also I have the sip.conf and extensions.conf configured |
| 20:28.09 | angryuser | Yoshinoya ok so you want inbound ? |
| 20:28.20 | Yoshinoya | that is correct |
| 20:28.43 | angryuser | Yoshinoya and where inbound call would arrive ? (technology) |
| 20:28.47 | Jumpie | im recompiling zaptel |
| 20:28.52 | Jumpie | cross fingers lol |
| 20:28.56 | angryuser | Jumpie have fun |
| 20:29.14 | Yoshinoya | I have a DID number with vitelity, so I want the inbound call to come in and they hear the default voicemail |
| 20:29.17 | Yoshinoya | in AsteriskNow |
| 20:29.20 | Yoshinoya | the Welcome greeting |
| 20:29.29 | Jumpie | lol |
| 20:29.43 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 20:29.43 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 20:30.03 | Jumpie | my brain spins when i watch all th is install script verbage |
| 20:30.44 | angryuser | Yoshinoya ok add 'incoming route' with that did and action for it |
| 20:31.20 | Yoshinoya | is that through vitelity or asterisknow? |
| 20:32.13 | angryuser | Yoshinoya your provier will send you your DID number you have when incoming call arrives, you need to tell asterisk what to do |
| 20:34.31 | angryuser | so if your did is 333333 then 33333,1,Playback(MessageAudioFile) in context where is you sip frien from provider is |
| 20:37.10 | Jumpie | hey angryuser lol.well recompiling it worked |
| 20:37.24 | Jumpie | but when i do a ztcfg instead of saying 1 channels configured, it says 1 channels to configure |
| 20:37.39 | Jumpie | i already checked the zaptel.conf and zapata.conf and they are correct...is that just a verbage difference or am i missing somethin? |
| 20:38.29 | angryuser | same thing |
| 20:38.58 | Jumpie | k |
| 20:57.42 | *** join/#asterisk mandd (n=moo@bas1-toronto61-1279395161.dsl.bell.ca) |
| 20:58.42 | mandd | any idea why asterisk would do this: http://pastebin.com/m5b453c3c |
| 20:58.42 | mandd | once in while |
| 20:58.44 | mandd | but after reload jsut works fine |
| 20:58.47 | mandd | just* |
| 21:00.56 | jaytee | pastebin your sip.conf |
| 21:01.01 | mandd | ok |
| 21:03.53 | jaytee | have you ever run sip show peers after those warning messages have appeared? |
| 21:03.54 | mandd | http://pastebin.com/m76ac3c2a |
| 21:04.03 | mandd | yes I did, all connected |
| 21:05.04 | jaytee | pastebin extensions.conf too |
| 21:05.22 | mandd | can I spot it from repeating |
| 21:05.23 | mandd | REGISTER attempt 1 to 4168491170@tor3.voipportal.ca |
| 21:05.23 | mandd | Really destroying SIP dialog '2e4b280b1f948fb914812ec650721075@192.168.2.30' Met |
| 21:05.25 | mandd | all the time? |
| 21:05.56 | mandd | it is in there jaytee |
| 21:06.07 | mandd | a bit lower. |
| 21:06.12 | jaytee | ah, I see |
| 21:06.15 | jaytee | thnx |
| 21:06.31 | mandd | it has been fine for a few days now |
| 21:06.36 | mandd | but after that reload |
| 21:06.58 | mandd | is there a way to force asterisk to reload once in a while, like every few hours? |
| 21:07.12 | Qwell | mandd: Why would you? |
| 21:07.43 | mandd | reload fixes things like 'SIP' (cause 3 - No route to destination) |
| 21:07.48 | mandd | sometimes I am not here |
| 21:07.53 | mandd | and phone go down |
| 21:07.56 | mandd | phones* |
| 21:08.04 | mandd | i know itdoesnt "fix" it |
| 21:08.15 | mandd | but at least it gets the line back up |
| 21:13.02 | jaytee | in your extensions.conf file none of the lines in your outbound-local and outbound-fax that run the Congestion app have a timeout set. Try using a minimum of 1 or 2 secs |
| 21:13.12 | Jumpie | woot my zap works now:D |
| 21:13.45 | mandd | ok |
| 21:15.17 | jaytee | manda, is there a nat'd firewall between you and your sip provider for the netout sip account? |
| 21:15.59 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 21:18.26 | drwelby | Is there any kind of a program that simplifies editing Polycom .cfg files? Like something that looks like the web front end and spits out the xml? |
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| 22:07.52 | ManxPower | drwelby: If Polycoms are too hard to manage the way they are then VoIP is not the industry for you. |
| 22:07.58 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 22:08.17 | ManxPower | I don't think I've edited my main polycom config files in at least 2 years. No need. |
| 22:09.07 | drwelby | ManxPower: It's Saturday and I reserve the right to maximize laziness on weekends ;) |
| 22:21.40 | *** join/#asterisk schmooze (n=schmooze@rrcs-67-52-214-19.west.biz.rr.com) |
| 22:21.41 | ManxPower | drwelby: I think there is some polycom gui config file builder, but I think it causes cancer. Do a google search of the mailing list archives |
| 22:21.43 | ManxPower | ~mailinglist |
| 22:21.44 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 22:24.43 | drwelby | manx: thanx |
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