| 00:00.14 | Thiago_Lima | run make clean before? |
| 00:10.43 | *** join/#asterisk hi365 (n=hi365@94.159.148.161) |
| 00:18.40 | dzup | hi, i recieve "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" when asterisk -r, i have read a lot in google, dis a bunch of stuff but still no look, i change permits, on /var/run/asterisk, create asterisk:asterisk: chmod everything to him, asterisk -&; asterisk -r; reboot, no luck at all, my asterisk -V said: Asterisk 1.4.21.2~dfsg-1ubuntu3 ...any idea? |
| 00:19.50 | dzup | ...the same apply when i did the make clean;make;make install; from sources. |
| 00:35.22 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
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| 00:40.34 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 00:43.59 | eric256 | hey, if i want to put callers into a que, hold for 30 seconds, then play an announcment, then hold for 7 minutes, then play another announcment (even the same one)... is that possible? the announcement timer looks hardcoded |
| 00:54.42 | x86 | 7 minute hold time? that's ludacris |
| 01:04.59 | *** join/#asterisk thansen (n=thansen@wsip-24-120-249-99.lv.lv.cox.net) |
| 01:07.12 | f0ner00t | Hey is there a way to force reg on a trunk? |
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| 01:39.58 | Ralcala | hi |
| 01:40.27 | Ralcala | someone here was brent couple days ago? |
| 01:40.51 | Ralcala | cause he help me about * server on fc9... |
| 01:42.01 | *** join/#asterisk hi365 (n=hi365@94.159.148.161) |
| 01:42.08 | Ralcala | nope |
| 01:42.09 | Ralcala | k |
| 01:42.17 | Ralcala | someoen here is awake? |
| 01:42.50 | Ralcala | online? |
| 01:43.00 | Ralcala | live? |
| 01:43.41 | slaney | yes |
| 01:43.46 | Ralcala | finally |
| 01:43.49 | slaney | I don't know who you are talkng about |
| 01:44.00 | Ralcala | it was an op |
| 01:44.11 | slaney | why are you running a production server on fedora |
| 01:44.31 | Ralcala | why not? |
| 01:44.42 | slaney | because it's extremely unstable? |
| 01:44.42 | x86 | fedora is crap |
| 01:44.47 | slaney | and not for such things? |
| 01:44.57 | Ralcala | what dist you recommend me... |
| 01:45.20 | slaney | I don't know, I have no idea what your needs are |
| 01:45.31 | Ralcala | poduction needs... |
| 01:45.37 | x86 | Ubuntu Server, Debian, rPath, Gentoo.... all good ones for Asterisk |
| 01:45.47 | slaney | I run Ubuntu for everything I possibly can |
| 01:45.55 | slaney | unless I need RH compatibility |
| 01:46.00 | x86 | I personally run Ubuntu Server 8.10 on my production Asterisk boxes |
| 01:46.00 | slaney | in which case I use CentOS |
| 01:46.09 | slaney | for asterisk? |
| 01:46.11 | slaney | we use Ubuntu |
| 01:46.20 | x86 | slaney: CentOS is almost as bad as Fedora ;) |
| 01:46.25 | slaney | heh |
| 01:46.27 | slaney | not quite |
| 01:46.35 | x86 | slaney: rPath is starting to look kinda cool though |
| 01:46.40 | slaney | it's at least somewhat stable |
| 01:46.42 | slaney | yeah |
| 01:46.47 | slaney | rpath does look interesting |
| 01:46.53 | slaney | I have only messed a little with it |
| 01:47.13 | x86 | i like rPath appliances for ESX servers |
| 01:47.22 | slaney | sure |
| 01:47.37 | *** part/#asterisk doug (i=doug@breakout.telerama.com) |
| 01:47.58 | slaney | Ralcala: if you don;t know what distro to use, I suggest you hire an admin |
| 01:48.17 | Ralcala | iam an admin |
| 01:48.27 | slaney | ok |
| 01:48.27 | Ralcala | i use RH since 1990 |
| 01:48.44 | Ralcala | no problems qith FC |
| 01:48.48 | slaney | ok |
| 01:48.51 | slaney | great |
| 01:49.26 | Ralcala | i got an production server working with FC3 |
| 01:49.42 | Ralcala | but now i update and got problems... |
| 01:49.45 | slaney | how are the updates on that? |
| 01:49.58 | slaney | still getting security patches? |
| 01:50.10 | Ralcala | nop |
| 01:50.13 | slaney | yeah |
| 01:50.14 | slaney | I know |
| 01:50.17 | slaney | fale |
| 01:50.47 | Ralcala | if i show you the sip headers of the client with XLite can you help? |
| 01:50.55 | slaney | maybe |
| 01:50.57 | Ralcala | k |
| 01:51.00 | slaney | I am not an asterisk guru |
| 01:51.05 | Ralcala | who is? |
| 01:51.08 | x86 | dude |
| 01:51.15 | slaney | I am here lurking to learn |
| 01:51.16 | x86 | RH wasn't even around in 1990 |
| 01:51.20 | slaney | heh |
| 01:51.32 | Ralcala | not |
| 01:51.34 | x86 | Linux was not even thought of until ~1991, and not even released until early 1993 |
| 01:52.00 | dzup | hi, i recieve "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" when asterisk -r, i have read a lot in google, dis a bunch of stuff but still no look, i change permits, on /var/run/asterisk, create asterisk:asterisk: chmod everything to him, asterisk -&; asterisk -r; reboot, no luck at all, my asterisk -V said: Asterisk 1.4.21.2~dfsg-1ubuntu3 ...the same apply when i did the src compile(make clean;make;make install; from |
| 01:52.14 | slaney | x86: I was letting that one go |
| 01:52.26 | slaney | I figured it was a rounding error |
| 01:52.34 | x86 | http://en.wikipedia.org/wiki/History_of_Linux |
| 01:52.39 | slaney | :) |
| 01:53.08 | x86 | "Since the initial release of its source code in 1991, it has grown from a small number of C files under a license prohibiting commercial distribution to its state in 2009 of over 370 megabytes of source under the GNU General Public License" |
| 01:53.23 | x86 | where was i getting 1993? |
| 01:53.40 | Ralcala | red hat 6.0 |
| 01:53.59 | Ralcala | i dont remember the year exactly... |
| 01:54.06 | slaney | Ralcala: regardless. Running a prod system on Fedora is foolish |
| 01:54.13 | slaney | if you want help, show us the problem |
| 01:54.38 | Ralcala | its your opinion not mine... |
| 01:54.43 | slaney | no |
| 01:54.52 | slaney | Fedora project is of the same opinion |
| 01:55.09 | slaney | I am not going to debate that with you |
| 01:55.12 | Ralcala | k |
| 01:55.20 | Ralcala | it work for me |
| 01:55.25 | slaney | great |
| 01:55.30 | slaney | I think that is fantastic |
| 01:55.36 | slaney | I wouldn't do it |
| 01:55.39 | slaney | but go on |
| 01:55.47 | Ralcala | k |
| 01:56.12 | Ralcala | i just registering client with Xlite on Windows with no problems... |
| 01:56.13 | X-Rob | dzup, have you STARTED asterisk before you try to connect to it? try 'asterisk -vvvg' |
| 01:56.24 | Ralcala | but know i upgrade i got troubles... |
| 01:56.26 | Kobaz | http://pastebin.com/m3222719d i'm having problems with originate via the ami with asterisk 1.6... the call kinda goes through, the remote phone rings for about 10ms... looks like a bug |
| 01:56.37 | *** join/#asterisk jeff_phillips (n=fircuser@m3f0436d0.tmodns.net) |
| 01:59.43 | Ralcala | and well? |
| 01:59.55 | Ralcala | can i paste you the headers? |
| 02:00.00 | slaney | Ralcala: that's not a question |
| 02:00.07 | Ralcala | k |
| 02:00.31 | slaney | honestly, I can tell that english is not your first language, but I have no idea what the question is |
| 02:00.32 | Ralcala | i cant register with XLite client on windows with * 1.6 on FC9 behind FW |
| 02:00.51 | slaney | ok |
| 02:01.15 | Ralcala | you right its not my first language... |
| 02:01.44 | slaney | that's fine |
| 02:01.54 | slaney | so |
| 02:02.01 | slaney | does it work with other clients? |
| 02:02.03 | dzup | X-Rob: http://pastebin.ca/1380531 |
| 02:02.11 | slaney | can you connect from anything? |
| 02:02.30 | Ralcala | i got 2 servers one with FC3 working and other with FC9 not working.... |
| 02:02.42 | Ralcala | and i know its * server issue |
| 02:02.42 | slaney | FC3 is running which version of * |
| 02:02.46 | Ralcala | 1.0 |
| 02:02.53 | jaytee | lol |
| 02:02.54 | Ralcala | old i know |
| 02:03.00 | slaney | ... |
| 02:04.24 | slaney | ok |
| 02:04.25 | slaney | so |
| 02:04.36 | slaney | let's pretend that doesn't matter |
| 02:04.58 | slaney | and I am getting beyond my area of understanding |
| 02:05.01 | slaney | but |
| 02:05.03 | X-Rob | dzup, last line. Your modules are bad. Delete everything in /var/lib/asterisk/modules and recompile asterisk |
| 02:05.09 | X-Rob | and reinstall |
| 02:05.22 | slaney | I am guessing you have tried to connect, and gotten an error of some sort |
| 02:05.25 | slaney | ? |
| 02:05.33 | Ralcala | yep |
| 02:05.42 | slaney | can we see that error |
| 02:05.50 | slaney | use pastie |
| 02:05.51 | Ralcala | sip headers? |
| 02:06.03 | Ralcala | or client error |
| 02:06.04 | slaney | I don't know what that means |
| 02:06.05 | *** join/#asterisk blkry (n=blkry@24-241-112-018.dhcp.gnvl.sc.charter.com) |
| 02:06.07 | slaney | both |
| 02:06.26 | Ralcala | XLite error 408 request timeout... |
| 02:07.04 | slaney | no |
| 02:07.06 | slaney | use pastie |
| 02:07.08 | slaney | or other |
| 02:07.14 | slaney | paste the EXACT error |
| 02:07.51 | slaney | I am probably not the person that is going to help you ultimately, and these guys will want to see this stuff |
| 02:08.38 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 02:08.56 | Ralcala | ok i now on pastebin now what |
| 02:09.03 | dzup | X-Rob: ok, doing so |
| 02:09.17 | dzup | X-Rob: thanks |
| 02:09.19 | slaney | Ralcala: paste the problem |
| 02:09.26 | slaney | and then give us the link |
| 02:09.30 | Ralcala | k |
| 02:09.41 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 02:09.41 | Ralcala | http://pastebin.com/m751a40c |
| 02:10.10 | Ralcala | these are the sip headers with server FC9... |
| 02:10.27 | Ralcala | now i send you the sip headers with FC3 |
| 02:10.34 | slaney | I don't care about that |
| 02:10.50 | slaney | fc3 is just confusing the issue |
| 02:11.11 | slaney | keeping in mind I have no idea what this is about |
| 02:11.12 | Ralcala | http://pastebin.com/m12c50405 |
| 02:11.23 | slaney | where did you get that stuff you pasted |
| 02:11.24 | Ralcala | yes i see now |
| 02:11.39 | Ralcala | i am admin |
| 02:11.43 | slaney | heh |
| 02:11.45 | slaney | ok |
| 02:11.47 | slaney | I am done |
| 02:12.52 | Ralcala | ok make me an favor and pass these links to asterisk support fellas and i´ll back tomorrow, ok.... |
| 02:13.09 | slaney | yeah |
| 02:13.18 | slaney | I will tell them when I get in in the morning |
| 02:13.24 | slaney | hey asterisk devs |
| 02:13.27 | slaney | here are some links |
| 02:14.18 | Ralcala | mmmmm |
| 02:14.20 | Ralcala | cya |
| 02:14.25 | Ralcala | and thanks |
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| 02:47.19 | *** mode/#asterisk [+o mog] by ChanServ |
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| 03:17.09 | fluvvell | I have some issues connecting a remote thomson 2030, it works then it stops working. I have all the ports forwarded, and am told the phone is setup right. I have had two conversations with a user on the phone, now its not passing audio through. |
| 03:23.29 | *** join/#asterisk mandd (n=moo@67.204.24.118) |
| 03:40.15 | Thiago_Lima | tzafrir_laptop, machine installed again from scratch and put the correct sources ... so far no problem, we see the end ... thanks ... |
| 03:43.54 | *** join/#asterisk xpot (n=james@67.222.236.132) |
| 03:45.06 | xpot | Help: anyone know what would cause the following WARNINGs? |
| 03:45.06 | xpot | [Apr 2 15:34:18] WARNING[10740]: channel.c:753 ast_best_codec: Don't know any of 0x0 formats |
| 03:45.06 | xpot | [Apr 2 15:34:18] WARNING[10740]: chan_iax2.c:10252 iax2_request: Unable to create translator path for unknown to ulaw on IAX2 |
| 03:46.09 | xpot | using asterisk-1.6.0 and asterisk-gui-2.0 both are svn latest branches |
| 03:53.25 | JunK-Y | xpot: just add allow=ulaw after ur disallow=all |
| 03:54.38 | pdmmm | codec |
| 03:54.47 | pdmmm | you got none set |
| 03:54.53 | pdmmm | wut junky said |
| 04:02.56 | xpot | Junk-Y: lol, I just checked. sure enough "disallow=all" and no allows! |
| 04:03.48 | xpot | not sure how that happened, I attempted to test the gui and I clicked ulaw, gsm, etc. I wonder if there is a bug in the gui? |
| 04:05.12 | JunK-Y | dunno, i dont use gui |
| 04:05.26 | JunK-Y | but if u add allow=ulaw u should be able to call, right? |
| 04:05.40 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
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| 04:06.07 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3bb65dde841f3c84) |
| 04:06.41 | xpot | JunK-Y: right. I Have used the cli now for about 2 years... trying to implement a GUI so I don't have to be the one managing it all the time. |
| 04:06.53 | xpot | this is a new setup btw |
| 04:09.25 | xpot | All: there is now bug, tried it again changing the priority of codecs in the gui and it worked that time. For some reason the first setup wrote "disallow=all" but didn't finish writing my allow piece. ?? |
| 04:09.31 | xpot | now = no -=0) |
| 04:15.40 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 04:39.57 | leifmadsen | hey :) |
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| 05:05.09 | jblack | leifmadsen: what are the odds of the * book ever making it to the kindle? |
| 05:05.21 | JunK-Y | http://www.clickatell.com/developers.php |
| 05:05.29 | JunK-Y | wow, that warning is really professional. |
| 05:05.41 | jblack | It's a nasty, evil drm format, but it would be so convienent.. |
| 05:05.53 | stabler | ~book |
| 05:05.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 05:21.19 | JunK-Y | do you guys know a good lib to use with gsm/gprs? (to abstract AT commands) |
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| 06:56.03 | Micc | I have a customer with an spa8000 and when they pickup their lines it stutters but thers no message waiting. It gives a strange message waiting type dial tone before the regular tone. I've tried everything to fix it, but nothing seems to work. |
| 06:56.17 | Micc | Anyone have any experience like this? I have other customers with spa8000's and they work great. |
| 06:56.25 | Micc | all of them running the same firmware. |
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| 07:41.20 | Ryan09 | Hello, I'm looking for a copy of the back port of func_devstate.c for 1.4, I have the link to it in svn but it seems to be down atm |
| 07:53.41 | zaafouri | hi all |
| 07:55.02 | *** join/#asterisk RichiH (i=richih@freenode/staff/richih) |
| 07:55.46 | RichiH | i need a cheap & reliable box that i can plug a POTS line into which will feed calls into our asterisk |
| 07:59.03 | zaafouri | I have this error for inbound call : Unable to create channel pf type 'SIP' (cause 20 - Unknown) |
| 07:59.18 | zaafouri | how can I fix it pls ! |
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| 10:59.30 | miloux | Anyone know which codecs works natively on windows (Sending voicemail to e-mail)...and which of those produce the smallest files? |
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| 11:34.41 | lesouvage | What operating panel would you advice for a callcenter with around 15 queues in operation? |
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| 11:48.46 | axscode | hi, anyone can tell me how can i use SMS function of asterisk? |
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| 12:02.48 | elred | that's silent today |
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| 12:34.18 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 12:34.55 | brunner | Can anyone point me to an npa nxx termination rate list for *any* voip provider? |
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| 12:36.25 | JunK-Y | brunner: they change for each provider. |
| 12:36.32 | brunner | of course |
| 12:36.49 | JunK-Y | so why asking for any provider? |
| 12:37.15 | brunner | because it gives me an idea of the relative cost of terminating calls to each exchange |
| 12:37.26 | JunK-Y | miloux: use wav |
| 12:37.28 | brunner | JunK-Y: do you know of such a list? |
| 12:37.44 | miloux | JunK-Y: which one.. |
| 12:37.49 | JunK-Y | no, cause it doesnt exists, it changes for every provider. |
| 12:38.57 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 12:38.59 | brunner | what do you mean it doesn't exist? I've seen several lists that have prices for each NPA/NXX |
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| 12:39.36 | JunK-Y | so its gonna be one list per provider, not one general list |
| 12:40.25 | brunner | no shit |
| 12:40.38 | brunner | didn't you just point out that it changes per provider? |
| 12:40.45 | brunner | how could it be a universal list? |
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| 12:41.32 | JunK-Y | im trying to say that there's no universal list for 5 minutes |
| 12:42.02 | brunner | JunK-Y: go back and look for what I said, and then look up "any" and "all" in an english dictionary |
| 12:43.53 | JunK-Y | i think you have to look for every provider one by one. |
| 12:44.32 | brunner | JunK-Y: why not just say, "I don't have a link to any lists like that right now, sorry." |
| 12:45.16 | JunK-Y | cause i wanted to talk to you :) |
| 12:45.22 | brunner | apparently |
| 12:46.15 | brunner | http://www.termination.com/assets/iCall_Termination_BlendRates_20061101.txt |
| 12:46.19 | brunner | see, there we go |
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| 12:47.59 | JunK-Y | http://www.flowroute.com/services/rates/ |
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| 12:51.00 | brunner | Okay, JunK-Y, could your list be a list of rates for every *country* or a list of rates for every *npa and nxx*? |
| 12:51.08 | brunner | I'll give you a few minutes to figure it out |
| 12:51.57 | JunK-Y | thats not my list, that's your job :) |
| 12:52.34 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:52.36 | brunner | JunK-Y: dude, in the future if you don't know what someone is asking, just don't answer |
| 12:54.01 | JunK-Y | i will remember that next time im trying to help you. |
| 12:54.14 | brunner | I appreciate it |
| 12:54.35 | JunK-Y | anyway, [TK]D-Fender is in da house! |
| 12:54.38 | [TK]D-Fender | resumes watching "Biting The Hand that Feeds" |
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| 12:57.46 | [TK]D-Fender | brunner: So... how's that workin' out for you? |
| 12:57.50 | [TK]D-Fender | </drphil> |
| 12:58.04 | brunner | [TK]D-Fender: How is what working out for me? |
| 12:58.09 | brunner | the list? |
| 12:58.14 | brunner | fine, thanks |
| 12:58.20 | [TK]D-Fender | brunner: Whatever it is... I just got in. |
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| 12:58.57 | [TK]D-Fender | brunner: Good to hear. |
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| 13:19.06 | Zeeek | Jello! |
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| 13:42.54 | rue_mohr | the incomming line has call waiting, to use it, they need to be able to wink the pots line from the voip set.... ideas how? |
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| 13:48.18 | [TK]D-Fender | rue_mohr: Plenty of docs show how to call Flash() from features.conf dynamic feature |
| 13:48.39 | [TK]D-Fender | rue_mohr: And your system has finally, and completely, jumped the shark. |
| 13:49.08 | rue_mohr | jumped the shark? |
| 13:49.24 | rue_mohr | gone ass backwards by using call waiting on an analog line? |
| 13:50.24 | [TK]D-Fender | rue_mohr: Yes |
| 13:51.20 | [TK]D-Fender | rue_mohr: Guess now they'll have to have telco vm as well, etc, and check for it manually because * doesn't |
| 13:51.42 | aiksa[LV] | hi everyone. |
| 13:51.59 | aiksa[LV] | just managed to compile t38modem as static binary. woohoo. |
| 13:52.15 | aiksa[LV] | as _working_ static binary :P |
| 13:52.36 | aiksa[LV] | no more fiddling around with opal and ptlib sources for some time |
| 13:53.17 | aiksa[LV] | that is - untill they will introduce some major changes to their code that is |
| 13:53.50 | rue_mohr | [TK]D-Fender, they do |
| 13:54.01 | rue_mohr | maybe I didn't mention there is only 1 line for the main business? |
| 13:54.09 | rue_mohr | and they insist on keeping it that way |
| 13:54.43 | rue_mohr | and I been wondering about having * retrieve it |
| 13:54.51 | [TK]D-Fender | rue_mohr: I bet the insisted on buying a Ferrari to do their grocery shopping with too. |
| 13:55.16 | rue_mohr | its making a good demo system |
| 13:55.43 | coppice | Ferraris suck for shopping. have you seen the luggage space? |
| 13:56.00 | [TK]D-Fender | coppice: I see the point is NOT lost on you... good :) |
| 13:56.02 | rue_mohr | I think he meant a 60' semi trailer |
| 13:56.06 | rue_mohr | not the point |
| 13:57.26 | rue_mohr | they really like th system though |
| 13:57.28 | [TK]D-Fender | rue_mohr: they need call waiting insist on 1 line, will have to manually monitor and maintain telco vM. They should get a bunch of stupid phones and just run it like you would a normal residence |
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| 13:58.15 | rue_mohr | they got a bit tangled up as they were trying to hand over a call by putting it on hold and picking it up on antoher phone, I successfully got them to understand they have to explicitly deliver the call |
| 13:58.28 | [TK]D-Fender | rue_mohr: these people are the kind of dumb that only ends with an encounter with a baseball bat in a dark alley |
| 13:58.50 | coppice | [TK]D-Fender: Someone I was at school with took his driving test in his mum's Rolls-Royce, just to show off, and was then pissed off that he was only licenced to drive automatics, and had to take his test again to drive a sports car :-\ |
| 13:59.31 | [TK]D-Fender | coppice: Interesting that transmissions are independently licensed.... |
| 14:00.03 | coppice | it so much easier to drive an automatic that many countries distinguish |
| 14:01.01 | *** join/#asterisk stmaher (n=stephen@80.68.89.200) |
| 14:01.04 | stmaher | Hi guys.. |
| 14:01.32 | coppice | Doesn't the US? are there just too many automatics there for people to bother with different classes of licence? |
| 14:01.57 | stmaher | Having a problem with Music on hold.. It works between two sip phones attached to the same asterisk.. but not from a Dahdi call. |
| 14:01.58 | [TK]D-Fender | coppice: North America doesn't care. |
| 14:01.59 | stmaher | any ideas? |
| 14:03.01 | coppice | UK driving school say people get through the automatic test in about 40% of the lessons a manual licence takes |
| 14:03.54 | *** join/#asterisk [gnubie] (i=patintin@119.56.59.14) |
| 14:03.55 | rue_mohr | dahdi local set or co call |
| 14:04.29 | [gnubie] | waves |
| 14:04.59 | [gnubie] | how do i remove the tick/static sounds when in call session? |
| 14:05.00 | stmaher | rue_mohr was that for me? |
| 14:05.06 | rue_mohr | yea |
| 14:05.13 | stmaher | its a local set.. |
| 14:05.30 | rue_mohr | does it operate in a different context? |
| 14:05.43 | stmaher | to which the sip phones? |
| 14:05.44 | rue_mohr | or so you mean transfer dosn't work |
| 14:05.45 | stmaher | yes |
| 14:05.52 | stmaher | that too doesnt work |
| 14:05.56 | rue_mohr | aha |
| 14:05.59 | stmaher | LOL |
| 14:06.04 | stmaher | Sherlock holmes :-) |
| 14:06.11 | rue_mohr | the only way I got mine to work was to use the hook flash |
| 14:06.19 | stmaher | hook flash? |
| 14:06.50 | rue_mohr | i flash the hook on the pots phone, it gives me my dail tone, I dial the destination, when I hang up its transfered |
| 14:07.08 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.24.187) |
| 14:07.16 | rue_mohr | you tap the phone hangup button |
| 14:07.47 | stmaher | herm ok? is there not a better solution? |
| 14:07.59 | stmaher | the hard sip phones have a hold / transfer button |
| 14:08.06 | rue_mohr | http://lists.digium.com/pipermail/asterisk-users/2006-January/134190.html <- [TK]D-Fender that wouldn't work, I'm using *X for the speed dial system |
| 14:08.23 | rue_mohr | stabler, on the pots phones I couldn't get anything else to work |
| 14:08.29 | rue_mohr | er stmaher |
| 14:08.41 | rue_mohr | I couldn't get anything that used inband audio to work |
| 14:09.15 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:09.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:09.19 | leifmadsen | I don't want to know your name! |
| 14:09.24 | leifmadsen | I don't want to meet your mom! |
| 14:09.38 | leifmadsen | [TK]D-Fender: I just want! |
| 14:12.43 | [TK]D-Fender | leifmadsen: ! ! ! |
| 14:13.11 | [TK]D-Fender | rue_mohr: Yes, it would work. |
| 14:13.31 | rue_mohr | different meaning with call established? |
| 14:13.55 | rue_mohr | cause its a feature not part of the dialplan... |
| 14:14.31 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
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| 14:57.13 | *** join/#asterisk _pepo_ (n=pepo@200.55.224.2) |
| 14:57.15 | _pepo_ | hi friends |
| 14:57.25 | *** join/#asterisk freckle (n=chatzill@195.74.96.118) |
| 14:57.29 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 14:59.14 | _pepo_ | please, How do I test my voicemail? I want to record my message (sound) automatically in many mailboxes, so then test the stress in the server |
| 14:59.51 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 15:08.22 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 15:09.48 | [TK]D-Fender | _pepo_: You test it by USING it |
| 15:10.06 | [TK]D-Fender | _pepo_: So go leave a ton of VM's in boxes if you fell like it. |
| 15:10.44 | *** join/#asterisk [netman] (n=netman@142.Red-83-50-80.dynamicIP.rima-tde.net) |
| 15:10.53 | _pepo_ | ok, How do I leave a ton of vm? Is there some app to do that? |
| 15:11.20 | Dovid | _pepo_ i dont think so. |
| 15:11.23 | [TK]D-Fender | _pepo_: No, you'll have to script something yourself |
| 15:11.24 | Dovid | u would need to call in |
| 15:11.32 | Dovid | u can write script |
| 15:11.36 | Dovid | oh TK answered for me ;) |
| 15:11.49 | Dovid | nm. i see now TK typed first. i need sleep |
| 15:12.11 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:12.42 | _pepo_ | sorry I dont know how do that? Is there some way to make script inside of Asterisk? How do I use a pre-recorded file? |
| 15:12.55 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 15:14.54 | [TK]D-Fender | _pepo_: Set up a bunch of .call files that will play a sound file on one side and dumping them into Voicemail on the other. |
| 15:15.31 | Kobaz | http://pastebin.com/m3222719d i'm having problems with originate via the ami with asterisk 1.6... the call kinda goes through, the remote phone rings for about 10ms... looks like a bug |
| 15:15.45 | _pepo_ | give me 5 min.... I will see what is .call files |
| 15:18.35 | _pepo_ | wow.... I am learning so much :) |
| 15:20.31 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
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| 15:30.59 | Zeeek | In 30 minutes, Michael Robertson of Gizmo5 and OpenSky is with us on the #voip-users-conference - join that channel anytime |
| 15:32.05 | Zeeek | http://tr.im/joinvoip for how to call via SIP, PSTN, string and cans... |
| 15:35.15 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 15:38.51 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.95) |
| 15:39.00 | mazpe | how about group voicemail? |
| 15:39.07 | mazpe | _pepo_: http://www.shift8.biz/blog/?p=6 |
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| 15:43.42 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.95) |
| 15:45.18 | cesar_CR | hello guys, when using asterisk as a SIP client is there any way to make * send a SUBSCRIBE request to te provider ? |
| 15:46.15 | UQlev | cesar_CR: isn't it the same as register? |
| 15:46.53 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:47.10 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:47.19 | *** join/#asterisk genin (n=phrame@ANice-252-1-28-192.w82-122.abo.wanadoo.fr) |
| 15:47.22 | genin | allo |
| 15:47.31 | genin | anyone use an audiocodes mp202 |
| 15:48.02 | *** join/#asterisk [netman] (n=netman@142.Red-83-50-80.dynamicIP.rima-tde.net) |
| 15:49.10 | cesar_CR | UQlev, dont really know because with an softphone the capture shows a SUBCRIBE request to the provider an with * I doesnt happend. I don really know if there is any diff ... |
| 15:49.43 | genin | no mp202 users |
| 15:49.49 | genin | im trying to get a dialplan working |
| 15:50.04 | genin | but the mp202 only has one line for the dialmap stuff |
| 15:50.28 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 15:50.32 | genin | and i need to put ten lines in like in the mp11x model |
| 15:50.33 | freckle | why do you want * SUBSCRIBE to your provideR? |
| 15:51.26 | freckle | a softphone is a UA . whereas * is a B"BUA |
| 15:51.31 | freckle | B2BUA even :-) |
| 15:51.41 | cesar_CR | the thing is that I can make calls, but I cant recibe calls from my provider |
| 15:52.32 | freckle | receiving calls is nothing to do with SUBSCRIBE |
| 15:52.40 | cesar_CR | I've been traying to find out what is the problem, I thing is the REGITER procedure, because with a softphone is diferent that with * |
| 15:53.21 | freckle | from asterisk CLI do a SIP SHOW REGISTRY |
| 15:53.27 | cesar_CR | humm, ok now the thing is that * is sending to the provider an 401 error |
| 15:53.36 | freckle | does i show your provider registered? |
| 15:53.41 | *** join/#asterisk [netman] (n=netman@142.Red-83-50-80.dynamicIP.rima-tde.net) |
| 15:53.45 | cesar_CR | freckle, yes is registered |
| 15:53.58 | freckle | ok so what happens if you try to call in ? |
| 15:54.57 | cesar_CR | in the CLI i get only " Using rtp .... " and in the other side, a busy tone |
| 15:55.30 | freckle | cesar_CR: that doesn't give me much info |
| 15:55.51 | UQlev | cesar_CR: do you want to share your SIP account via *? |
| 15:55.57 | freckle | if you get busy then your provider can't reach your * box or your asterisk box is rejecting the call |
| 15:56.27 | cesar_CR | freckle, UQlev I am doing a capture to share it, one second |
| 15:57.57 | *** join/#asterisk jstoker (n=jstoker@80-41-160-106.dynamic.dsl.as9105.com) |
| 15:58.31 | *** part/#asterisk jstoker (n=jstoker@80-41-160-106.dynamic.dsl.as9105.com) |
| 15:58.42 | UQlev | cesar_CR: do you have in your sip.conf "register => username:password@sip.myisp.com/username"? |
| 15:59.00 | cesar_CR | freckle, UQlev without debug I only get this: Using SIP RTP CoS mark 5 |
| 15:59.52 | cesar_CR | freckle, I hace username@domain:secret:authuser@host/username |
| 15:59.53 | freckle | cesar_CR: thats not enough info ... set verbose 99 and post full debug on pastebin or CLI output |
| 15:59.55 | cesar_CR | have |
| 16:00.14 | bkw_ | Who's IP is this 66.96.218.5 |
| 16:00.18 | bkw_ | anyone in here? |
| 16:00.24 | *** join/#asterisk MikeJ_ (n=MikeJ@freeswitch/developer/mikej) |
| 16:00.31 | leifmadsen | bkw_: I'm here! |
| 16:00.50 | bkw_ | does that ip belong to anyone here? |
| 16:00.54 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 16:01.04 | leifmadsen | seems like a long shot you'll get a response... |
| 16:01.12 | leifmadsen | I wouldn't respond if I didn't know why you wanted to know ;) |
| 16:01.22 | bkw_ | the bastards are sip spamming |
| 16:01.28 | leifmadsen | fun |
| 16:01.32 | bkw_ | they sent invites to every damn phone on my network |
| 16:01.35 | bkw_ | at the same time |
| 16:01.57 | bkw_ | test phones no less but anoying |
| 16:02.11 | UQlev | bkw_: use firewall |
| 16:02.35 | *** join/#asterisk [netman] (n=netman@142.Red-83-50-80.dynamicIP.rima-tde.net) |
| 16:02.39 | cesar_CR | freckle, UQlev * sends this to the provider: Sending to 201.194.212.242 : 5060 (no NAT) |
| 16:02.39 | cesar_CR | Using INVITE request as basis request - LU-1238774270629221@provider |
| 16:02.39 | cesar_CR | No user '89119997' in SIP users list |
| 16:02.39 | cesar_CR | Found peer 'icesip' for '89119997' from provider:5060 |
| 16:03.07 | freckle | cesar_CR: dont flood use pastebin |
| 16:03.17 | cesar_CR | sorry!!! |
| 16:03.28 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 16:03.28 | freckle | no worries |
| 16:03.47 | bkw_ | UQlev: its my test bed... and I just wanted to strangle the guy |
| 16:04.06 | cesar_CR | freckle, I will paste the * debug in pastebin, one second |
| 16:04.23 | *** join/#asterisk RMod (n=rmod@unaffiliated/rmod) |
| 16:04.41 | leifmadsen | bkw_: I just got it too I think |
| 16:04.50 | leifmadsen | [Apr 3 11:53:26] NOTICE[18910]: chan_sip.c:14585 handle_request_invite: Call from '' to extension '010380445945248' rejected because extension not found. |
| 16:04.50 | leifmadsen | [Apr 3 11:58:32] NOTICE[18910]: chan_sip.c:14585 handle_request_invite: Call from '' to extension '00380445945257' rejected because extension not found. |
| 16:05.40 | jameswf | LeddyHM, does 1.4.24.1 still do both dahdi and zaptel? |
| 16:05.50 | leifmadsen | jameswf: yes |
| 16:05.58 | leifmadsen | I think there is a config option somewhere |
| 16:06.10 | jameswf | wasnt zaptel going to be depricated @ 1.4.22 |
| 16:06.10 | bkw_ | leifmadsen: yes thats it |
| 16:06.14 | UQlev | leifmadsen: may be your SIP provider doesn't like leading 0 |
| 16:06.15 | leifmadsen | bkw_: nice nice... |
| 16:06.23 | bkw_ | they just hit me again too |
| 16:06.26 | leifmadsen | UQlev: no, that was an incoming attempt |
| 16:06.45 | bkw_ | Contact: <sip:MeucciSolutions@66.96.218.5> |
| 16:06.49 | leifmadsen | jameswf: yes, deprecation != removal |
| 16:06.56 | leifmadsen | fun... |
| 16:07.05 | bkw_ | INVITE sip:010380445945248@99.185.85.3 SIP/2.0 |
| 16:07.16 | bkw_ | leifmadsen: yep you got hit too |
| 16:07.37 | leifmadsen | just enabled authrejectalways=yes because I hadn't previously :) |
| 16:07.43 | jameswf | leifmadsen, guess that is why I still get away with not using <core> show |
| 16:07.52 | leifmadsen | jameswf: aye |
| 16:08.00 | cesar_CR | freckle, UQlev here it is http://pastebin.com/m15572a34 |
| 16:08.12 | jameswf | stupid core |
| 16:08.17 | leifmadsen | bkw_: luckily it didn't ring anything locally :) |
| 16:08.25 | bkw_ | leifmadsen: yah its just anoying |
| 16:08.33 | leifmadsen | jameswf: you can use cli_aliases.conf to get around it in 1.6.2.x |
| 16:08.55 | leifmadsen | jameswf: so you don't have to change anything if you don't want to |
| 16:09.03 | leifmadsen | just load the 1.4 template |
| 16:09.03 | coppice | bkw_: unless you have the vulnerability they are hunting for :-) |
| 16:09.06 | freckle | cesar_CR: here is your problem No user '89119997' in SIP users list |
| 16:09.10 | jameswf | leifmadsen, not motivated enough to type core deffinately less motivated to do extra confs not to type core |
| 16:09.29 | leifmadsen | jameswf: you're not motivated enough to uncomment a pre-made template? |
| 16:09.29 | bkw_ | coppice: haha doubt it |
| 16:09.36 | leifmadsen | jameswf: I already made the template for you |
| 16:09.38 | jameswf | pretty much :) |
| 16:09.51 | leifmadsen | ok, I'll be sure to be unmotivated to answer additional questions :) |
| 16:10.30 | jameswf | will wait for the book... better yet the Asterisk movie (that would be exciting) |
| 16:10.41 | *** join/#asterisk [netman] (n=netman@83.50.80.142) |
| 16:10.49 | UQlev | cesar_CR: paste your "sip show peers" and "sip show registers" |
| 16:11.11 | cesar_CR | freckle, I am confused about 89119997 is my cell, but I am calling to my SIP account in my provider tha is 2601####, why is he searching my cellphone in my * box ? |
| 16:11.22 | cesar_CR | UQlev, one second |
| 16:12.15 | UQlev | cesar_CR: you have to have in your sip.conf register> and account for your SIP provider |
| 16:12.48 | cesar_CR | UQlev, I have like this: username@domain:secret:authuser@host/username |
| 16:12.51 | SuPrSluG | how many legs on a call before you lose sound? |
| 16:13.03 | SuPrSluG | is there a limit? |
| 16:13.17 | UQlev | cesar_CR: can you show all your sip.conf w/o secrets |
| 16:13.34 | cesar_CR | ok UQlev |
| 16:14.07 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) |
| 16:15.47 | UQlev | cesar_CR: and try to change register as follows: username:secret@host/username |
| 16:16.40 | flohack | Hi! I'm just setting up a snom phone with two identities. Both register fine and show up with 'sip show peers'. As soon as I change the identity and try to place a call using the phone I get "check_auth: username mismatch, have <7707>, digest has <7705>". Any ideas what the proble could be? |
| 16:17.59 | cesar_CR | UQlev, http://pastebin.com/m6b3f9e51 |
| 16:21.29 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:21.46 | UQlev | cesar_CR: I see it is duly registered |
| 16:23.48 | UQlev | cesar_CR: and you can not dial 89119997? |
| 16:24.10 | cesar_CR | I get a busy tone, but I can make calls |
| 16:27.47 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 16:30.12 | cesar_CR | UQlev, its very extrange right :) but I'll keep traying |
| 16:30.32 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 16:31.54 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 16:32.07 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 16:32.25 | flujan | hello guys, I am having problems with the blind xfer at asterisk 1.4.24. |
| 16:32.44 | flujan | I am trying to use the xfer feature of the softphone and the # key but both are not working. |
| 16:33.05 | flujan | Attendee transfer is working using * |
| 16:33.07 | flujan | any ideas? |
| 16:33.52 | *** join/#asterisk asaleem_ (n=amjad@modemcable210.78-70-69.static.videotron.ca) |
| 16:34.54 | asaleem_ | any recommendations for 6000 channels using asterisk? |
| 16:35.43 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:38.05 | flohack | Noone here who can help with multiple registrations using snom phones? |
| 16:46.11 | flujan | here goes my features.conf file |
| 16:46.11 | flujan | http://pastie.org/436064 |
| 16:48.32 | *** join/#asterisk infinity1 (n=brendon@web2.artsopolis.com) |
| 16:48.49 | jameswf | Anyone with PIAF installed? |
| 16:49.35 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 16:49.46 | CrashSys | Anyone have a document on how to get Zaptel install in Centos5? |
| 16:49.50 | CrashSys | modprobe zaptel fails |
| 16:49.52 | CrashSys | cant find module |
| 16:50.24 | jameswf | make sure you have the appropriate kernel-devel |
| 16:52.40 | *** part/#asterisk BrianR___ (i=brianr@osiris.978.org) |
| 16:53.13 | CrashSys | maybe i'll just "yum install any-other-linux-distro" to fix it... |
| 16:53.15 | CrashSys | :( |
| 16:53.22 | jameswf | damnit sound server just crashed... thanks ubuntu |
| 16:55.25 | *** join/#asterisk rhc (n=matt@qnan.org) |
| 16:55.33 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:55.49 | rhc | does anyone know how to parse the .tlv files on cisco devices? e.g. CTLFile.tlv or a CTLSEP<mac>.tlv |
| 16:58.32 | Qwell | rhc: You don't.. |
| 16:59.00 | rhc | Qwell: ? what do you mean? |
| 16:59.24 | rhc | i know its not necessary to parse them if i'm just setting up a network, but i actually want to know what's inside the files |
| 16:59.40 | Qwell | Nobody but Cisco knows. |
| 17:00.09 | Qwell | It's just encryption stuff |
| 17:04.05 | flohack | Is there a way to make asterisk authenticate peers via username, instead of ip address? |
| 17:08.41 | [TK]D-Fender | flujan: pastebin your failed call attempt and transfer from beginning to end |
| 17:13.58 | *** join/#asterisk Davedan (n=me@CBL217-132-90-11.bb.netvision.net.il) |
| 17:14.04 | flohack | Can someone please tell me what to do? I can't use type=friend because of call limit issues, but I need to use several lines from one ip address. Is there a way to do that with asterisk 1.4? |
| 17:14.47 | Qwell | flohack: instead of trying to work around how things work, how about you explain what you want to do? |
| 17:14.54 | Qwell | that would be easiest, I suspect |
| 17:16.14 | flohack | Qwell: Sure! I need to have multiple queues a user can be logged in, therefore I need to limit calls to phones. I also need to support multiple identities on one phone, which does not seem to work when using peers for the phone accounts. |
| 17:16.58 | flohack | Because of the requirement for call limits I cannot use friends (I can't remember the exact problem I ran into, but I think it is quite well known) |
| 17:18.56 | flohack | Therefore I need be able to make specific peers authenticate on the From user string instead of ip address. I could of course write a patch which adds an option to specify that specific peers should be authenticated against username, but I can't believe that this is actually required. |
| 17:20.16 | flohack | Qwell: Any ideas? |
| 17:23.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:24.59 | flujan | [TK]D-Fender: http://pastie.org/436096 here is the pastie |
| 17:25.08 | *** join/#asterisk fusa (n=fusa@dpc6744201027.direcpc.com) |
| 17:26.56 | [TK]D-Fender | flujan: Ok, I see it prompting for the transfer, what happens next? |
| 17:27.31 | flujan | the incoming call drops and the transfered extension do not ring |
| 17:27.52 | flujan | looks like the extension are not being executed at asterisk |
| 17:28.16 | [TK]D-Fender | :| |
| 17:29.48 | flujan | [TK]D-Fender: It should exec the _XXXXX,1,Dial(SIP/{EXTEN},1,60,rt) extension on the dialplan |
| 17:29.53 | flujan | but it is not executing it |
| 17:30.29 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 17:31.31 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:33.25 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 17:37.43 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 17:38.54 | cesar_CR | guys anybody know why could be the reason of a messange 423 Interval Too Brief from my SIP provider ? |
| 17:39.16 | flohack | Is still nobody able to help with the peer vs. multiple indentities issue? |
| 17:40.35 | *** join/#asterisk tecnico (n=tecnico@75.76.169.148) |
| 17:48.30 | leifmadsen | flohack: just ask the question |
| 17:51.59 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 17:52.56 | [intra]lanman | cesar_CR: they require a minimum registration expires... and yours is smaller than that? what's the provider? |
| 17:53.20 | flohack | leifmadsen: Its a few lines up in the history |
| 17:53.31 | cesar_CR | is from Costa Rica, the use a Lucent equipement ? |
| 17:54.20 | cesar_CR | [intra]lanman, wasn't a question |
| 17:54.25 | cesar_CR | :) |
| 17:54.51 | cesar_CR | [intra]lanman, can I set that on sip.conf ? |
| 17:55.17 | [intra]lanman | probably... i haven't used asterisk in years... but i think there's a setting for it |
| 17:55.33 | [intra]lanman | look for expire or timeout maybe? |
| 17:56.18 | [intra]lanman | oh, or look for 120... i think that's the default |
| 17:56.28 | [intra]lanman | was back in 1.2 when i used it ;-) |
| 17:56.37 | cesar_CR | ok :) thanks |
| 17:57.35 | flohack | leifmadsen: Any ideas? |
| 18:06.40 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-f2d5aa9a3c8f2176) |
| 18:12.07 | KavanS | if I'm using "&" dialing to dial multiple extensions, can I evaluate if a certain session (I have 3 concurrent dials) comes back with congestion/busy? |
| 18:12.42 | KavanS | I've done a DIALSTATUS evaluation before, but with multiple dials at the same time....how would I evaluate the 1st, 2nd or 3rd dial? |
| 18:12.56 | [TK]D-Fender | KavanS: When would you do something based on this result? |
| 18:13.46 | KavanS | the goal is to attempt to dial out an SIP provider if the PSTN (middle dial command) is congested |
| 18:14.35 | [TK]D-Fender | KavanS: What are the other 2 devices in your dial doing in the mean-time? |
| 18:15.02 | KavanS | ringing 2 additional local SIP extensions |
| 18:16.14 | *** join/#asterisk MmixX (n=mmixx@61.14.191.137) |
| 18:20.12 | nkohh | has anyone dealt with "netvoice"? netvoice.ca |
| 18:21.47 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 18:21.53 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 18:26.53 | [TK]D-Fender | KavanS: Then make your 2nd item in that dial a Local Channel instead of whatever you're using now, and it can have whatever failover logic you want. |
| 18:31.22 | *** join/#asterisk nicoAMG (i=asgplat@asgplat.expomaderas.com) |
| 18:36.13 | *** part/#asterisk JunK-Y (n=junky@modemcable206.69-20-96.mc.videotron.ca) |
| 18:36.42 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 18:37.36 | DavidR2008 | has anyone used the switch statement in extensions.conf and if so could explain how it works to me? I can't find much about it in the documentation |
| 18:39.00 | *** join/#asterisk twanny796 (n=jmx@85.232.204.152) |
| 18:39.30 | twanny796 | is there a linux based free skype solution ? |
| 18:41.09 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 18:41.51 | *** join/#asterisk algorithmn (n=blake@s15263727.onlinehome-server.com) |
| 18:43.01 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 18:43.41 | Qwell | twanny796: how about...Skype? |
| 18:44.17 | coppice | too simple |
| 18:45.10 | KavanS | [TK]D-Fender, do you have any examples of this? |
| 18:45.39 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 18:46.10 | [TK]D-Fender | KavanS: Lookup "local channels" on the WIKI |
| 18:46.26 | KavanS | I have failover macro in another section of my config, but I'm not sure how to dial multiple extensions and utilize this macro |
| 18:47.12 | KavanS | ahh nice, thank you |
| 18:47.24 | KavanS | this is what I needed |
| 18:48.28 | KavanS | ok, lunch time....cheers! |
| 18:53.55 | *** join/#asterisk ruben23 (n=ITadmin@124.107.3.178) |
| 18:55.59 | tzafrir_laptop | Qwell, that's the problem. Not the solution |
| 18:56.34 | *** join/#asterisk fusa (n=fusa@dpc6744201027.direcpc.com) |
| 19:07.45 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 19:09.37 | WHYS | I'm trying to #include a file in extensions.conf, a reload tells me it does not exist. It does exist and is spelled correctly. ???? |
| 19:11.42 | hardwire | if DTMF tones were to be easily confused.. would 8 and 6 US be the most mistaken? |
| 19:12.53 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:12.54 | WHYS | no, they have different Hor&vert frequencies. |
| 19:15.05 | twanny796 | @Qwell: something like a linux based skype to sip gateway? |
| 19:15.11 | [TK]D-Fender | WHYS: And you're not showing us anything. |
| 19:15.38 | WHYS | I was hoping for a quick "oh that" |
| 19:15.44 | [TK]D-Fender | WHYS: And unless you're on a bleeding-edge SVN I highly doubt its anything but user error otherwise everyone would be in here freaking out about it already :) |
| 19:15.56 | WHYS | :) |
| 19:16.03 | WHYS | I'll clooect and pastbin |
| 19:18.26 | Qwell | hey tzafrir_laptop, do you know much about building dahdi for use on xen? |
| 19:18.37 | Qwell | specifically wrt dahdi_dummy |
| 19:18.46 | tzafrir_laptop | Qwell, no |
| 19:18.47 | *** join/#asterisk zippytech (n=ron@75.149.24.162) |
| 19:18.56 | tzafrir_laptop | I know that there's a pending patch... |
| 19:19.48 | Qwell | I'm running into a rather interesting problem.. for xen, I'm going to need to disable USE_RTC or whatever, but...here's the catch. |
| 19:19.59 | tzafrir_laptop | the problem is that normally nither RTC nor hires timers are available there |
| 19:20.07 | Qwell | all the builds are being done at once against the various flavors (standard, xen, pae, etc) |
| 19:20.11 | tzafrir_laptop | and often the kernel is not built with HZ=1000 |
| 19:20.30 | Qwell | presumably USE_RTC wouldn't even be set on xen builds, would it? |
| 19:20.41 | zippytech | hi have have polycom phone that when a get a call coming in i pick up then i try to transfer the call to another user on the system the user see's my phone not the call that came in |
| 19:20.56 | WHYS | #include error: http://pastebin.com/d67a3f485 |
| 19:21.26 | tzafrir_laptop | Qwell, http://bugs.digium.com/view.php?id=13930 |
| 19:21.28 | zippytech | i have a few grandstreams and when they tranfer it show the caller that called the phone system |
| 19:21.51 | tzafrir_laptop | Though I'm not sure what this does to a non-xen version |
| 19:21.52 | zippytech | is there a setting some where in the polycom |
| 19:22.15 | Qwell | this isn't going to be fun |
| 19:22.53 | Qwell | damn you xen - implement an RTC |
| 19:22.58 | [TK]D-Fender | WHYS: no quotes |
| 19:23.23 | WHYS | Line 58 - single quotes. |
| 19:24.29 | WHYS | arg. Should be double quotes. |
| 19:24.37 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:25.38 | WHYS | Yep, Thanks [TK]D-Fender! It needs to be double quotes. |
| 19:26.58 | Katty | haiguys |
| 19:27.03 | Katty | i've come to HUG YOU ALL |
| 19:27.09 | Katty | well. a few of you. |
| 19:27.12 | Katty | PREPARE YOURSELVES. |
| 19:27.37 | carrar | I would like hug pls |
| 19:27.46 | Katty | carrar: who are you. |
| 19:27.47 | tzafrir_laptop | Qwell, can you test that patch? Any problem applying it to older versions? |
| 19:27.49 | carrar | haha |
| 19:27.56 | carrar | I'm ME!! |
| 19:28.06 | Qwell | tzafrir_laptop: problem is, I can't apply a patch to just one flavor |
| 19:28.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:28.12 | Katty | hugs anthm |
| 19:28.15 | Katty | hugs Qwell |
| 19:28.25 | [TK]D-Fender | carrar: You can't be "Me", *I'm* ME!!! Fraudster! |
| 19:28.26 | carrar | surely I have a pic someplace on the innerneb |
| 19:28.27 | anthm | hi |
| 19:28.29 | Katty | hugs [TK]D-Fender |
| 19:28.36 | carrar | WHAT#@#$@ |
| 19:29.06 | Katty | pls to produce identification for hugging porpoises. |
| 19:29.24 | WHYS | I'm someone else |
| 19:29.30 | Corydon76-dig | hugs Katty |
| 19:29.34 | Katty | hugs Corydon76-dig |
| 19:30.03 | tzafrir_laptop | carrar, here's one picture of me: http://upload.wikimedia.org/wikipedia/en/thumb/6/69/WindowsME.png/290px-WindowsME.png |
| 19:30.23 | Qwell | Not enough blue in that picture |
| 19:30.31 | Qwell | should be a brighter blue too |
| 19:30.40 | carrar | ok Katty |
| 19:30.40 | carrar | pub 1024D/BB29A64E 2002-08-09 |
| 19:30.40 | carrar | <PROTECTED> |
| 19:31.06 | Katty | Information provided could not be parsed. |
| 19:31.08 | Katty | Please try again. |
| 19:31.09 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:31.11 | carrar | haha |
| 19:31.18 | Qwell | carrar: unless anybody here who she trusts has said they trust you... |
| 19:31.29 | carrar | yeah |
| 19:31.31 | carrar | possible |
| 19:31.40 | carrar | I've been to a few keyring parties at nanog |
| 19:32.20 | Katty | where did jbot go? |
| 19:32.26 | Qwell | ~jbot |
| 19:32.27 | infobot | rumour has it, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 19:32.40 | Katty | infobot: are you jobt? |
| 19:32.49 | Katty | infobot: are you jbot? |
| 19:32.57 | Katty | infobot: danny boy? |
| 19:32.57 | infobot | methinks danny boy is http://www.youtube.com/watch?v=OCbuRA_D3KU -- *sniffle* Oh Danny Boy!!! *sniffle* |
| 19:32.58 | tzafrir_laptop | carrar, I can't fund such a record on pgp.mit.edu |
| 19:33.03 | Katty | infobot: YOU ARE JBOT! |
| 19:33.03 | infobot | I think you lost me on that one, Katty |
| 19:33.09 | carrar | it's there |
| 19:33.41 | Katty | infobot: <3 |
| 19:33.41 | infobot | <3 is, like, not >4, or not 3, or the emulation of a love symbol |
| 19:33.57 | carrar | http://pgp.mit.edu:11371/pks/lookup?search=tim%40osburn.com&op=index&fingerprint=on |
| 19:34.14 | Katty | file: did you put that one in infobot? it sounds like you. |
| 19:34.56 | tzafrir_laptop | Katty, TimRiker did |
| 19:34.56 | file | no! |
| 19:35.21 | Katty | ohiseethen |
| 19:35.24 | Katty | hugs file |
| 19:35.29 | file | tackle hugs Katty |
| 19:35.36 | Katty | file: howarecu |
| 19:35.43 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 19:35.48 | file | Katty: well!u? |
| 19:35.58 | Katty | file: peachy. |
| 19:37.00 | Katty | file: i got a new blackberry |
| 19:37.07 | file | Katty: is it black? |
| 19:37.08 | Katty | file: not the storm tho |
| 19:37.12 | Katty | file: mostly |
| 19:37.23 | file | Katty: oic |
| 19:38.05 | Katty | file: 8330 |
| 19:38.14 | Katty | file: has camera. |
| 19:38.22 | file | Katty: eep! do you use it to spy on people? |
| 19:38.28 | Katty | file: of course. |
| 19:38.34 | file | Katty: excellent |
| 19:38.38 | Katty | file: got to keep an eye on missouri law enforcers. |
| 19:39.39 | *** join/#asterisk AndyML (n=AndyML@pool-96-245-116-32.phlapa.fios.verizon.net) |
| 19:39.40 | Katty | sad. |
| 19:39.51 | Katty | this hard drive refuses to partition |
| 19:39.55 | AndyML | <PROTECTED> |
| 19:39.56 | file | hammer. |
| 19:40.08 | Katty | deposits hd in toilet, flushes |
| 19:40.14 | Katty | fixed! |
| 19:41.12 | Katty | AndyML: is that a... digium card? |
| 19:41.18 | AndyML | sure is |
| 19:41.22 | AndyML | transcoder |
| 19:41.23 | Katty | hrm. |
| 19:41.31 | Katty | i have a wiki on dahdi for sangoma |
| 19:41.33 | AndyML | sorry, "voice processor" |
| 19:41.39 | Katty | i could link it, if you think it might possibly be useful |
| 19:41.45 | Katty | but it's a sangoma t1 card. |
| 19:41.59 | AndyML | thanks Katty, i think its pretty different. |
| 19:42.08 | Katty | k |
| 19:42.12 | AndyML | it offloads g729 / 723 to a PCI card. |
| 19:43.14 | AndyML | looks like it "just works"... |
| 19:43.19 | AndyML | that is weird to me. |
| 19:43.35 | Qwell | tzafrir_laptop: am I reading this patch wrong? Does it just remove RTC for all cases? |
| 19:44.34 | tzafrir_laptop | Seems to. Maybe this is too harsh? |
| 19:44.55 | Qwell | lots of kernels still have only RTC |
| 19:45.09 | Qwell | now, if the "normal" timer is better than RTC, maybe it's a good patch |
| 19:46.40 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 19:48.47 | prakriti | are there any sip clients that can connnect to and use multiple sip accounts simultaneusly |
| 19:48.47 | prakriti | ? |
| 19:49.18 | bmoraca | i imagine that virtually all of them support multiple "lines" |
| 19:49.21 | jplank | its crazy how finicky zaptel could be - was having an issue with a my tdm lines, rebuilt the box on the same exact server, using a backup of the original server, using the same ISO I used to roll out the original box, same card same everything, yet this box doesn't have the issue, go figure |
| 19:49.33 | prakriti | well, most of them only connct to a single sip server at once |
| 19:49.44 | prakriti | and the server itsefl supports multiple lines/extensions |
| 19:50.00 | Katty | wow |
| 19:50.01 | jplank | what do you mean by client? |
| 19:50.04 | Katty | 2 bad hard drives in a row )= |
| 19:50.23 | prakriti | We are setting up an asterisk install per "city" for our company |
| 19:50.51 | prakriti | I want a SIP client for my desktop that I can connect to all the asterisk servers at once, to take calls etc |
| 19:51.02 | jplank | eyebeam |
| 19:51.11 | jplank | the paid version of xlite |
| 19:51.33 | prakriti | thanks, i'll look into it |
| 19:51.49 | Katty | i think i'd just set the extension, to forward to an iax extension registered on another server |
| 19:52.29 | Katty | but that's just me |
| 19:52.30 | Deeewayne | offers Katty a lemon cookie |
| 19:52.40 | Katty | oooh! |
| 19:52.42 | Katty | hugs Defraz |
| 19:52.43 | Katty | oh |
| 19:52.46 | Katty | hugs Deeewayne |
| 19:53.57 | rob0 | What kind of hugfest is this?!? |
| 19:54.57 | carrar | It's a discriminatory one!!! :) |
| 19:55.48 | jaytee | obviously some people just aren't "huggable" |
| 19:56.21 | carrar | I'm not from Japan |
| 19:56.52 | carrar | hugging isn't in their culture |
| 19:57.26 | Katty | hugs rob0 |
| 19:57.30 | Katty | hugs jaytee, twice. |
| 19:57.40 | jaytee | ooooh, I got two :-) |
| 19:57.40 | carrar | Though every time I go there I bring the hugging culture with me |
| 19:58.07 | jameswf | creeps quietly to the corner |
| 19:58.57 | prakriti | jplank : do you know how many accounts I can set up on a single eyebeam client? |
| 20:14.51 | *** join/#asterisk jstoker (n=jstoker@80-41-158-78.dynamic.dsl.as9105.com) |
| 20:15.13 | *** part/#asterisk jstoker (n=jstoker@80-41-158-78.dynamic.dsl.as9105.com) |
| 20:15.23 | *** join/#asterisk freh (n=freh@83.101.31.107) |
| 20:15.58 | jplank | eyebeam, I think can take quite a few. |
| 20:16.02 | jplank | xlite only takes one |
| 20:16.06 | freh | Can anyone help me out with asterisk 1.6, Dahdi and ISDN BRI? |
| 20:16.47 | freh | I have 3 ISDN BRI lines with a digium B410p card |
| 20:17.02 | freh | I have 2 phone numbers with each line |
| 20:17.25 | *** part/#asterisk rhc (n=matt@qnan.org) |
| 20:17.42 | *** join/#asterisk j_kroon (n=jkroon@41.240.190.12) |
| 20:17.45 | freh | On dialout, I can't get it to dial out with the second number of each line |
| 20:18.28 | Katty | yeah, i have a light that keeps coming on in my car--then going off |
| 20:18.31 | Katty | what's wrong with it? |
| 20:19.08 | jplank | Katty: have you tapped it? |
| 20:19.33 | jplank | :) |
| 20:19.52 | *** join/#asterisk emoboyforever (n=kvirc@89.241.220.74) |
| 20:20.05 | *** part/#asterisk emoboyforever (n=kvirc@89.241.220.74) |
| 20:21.32 | Katty | freh: there you go! have you tapped it? |
| 20:23.04 | Katty | i'm starting to think there's something wrong with this motherboard |
| 20:23.23 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 20:23.33 | freh | Katty, Have you tapped it? |
| 20:23.40 | Katty | 3 cdrom drives, 4 hard drives, two ide cables, a sata cable, and another cd later... |
| 20:23.44 | Katty | it still refuses to cooperate |
| 20:24.01 | jplank | did you tap it? |
| 20:24.17 | Katty | why yes, i did. and i hugged it too! |
| 20:24.29 | jplank | well then I'm out of ideas :P |
| 20:24.52 | Katty | i have one! |
| 20:24.53 | Katty | ram |
| 20:25.03 | jplank | right, ram it into the wall |
| 20:25.06 | jplank | that could work too |
| 20:25.16 | *** join/#asterisk f0ner00t (i=f0ner00t@rubik.cluenet.org) |
| 20:25.35 | Katty | i'll try that next. |
| 20:25.37 | *** part/#asterisk f0ner00t (i=f0ner00t@rubik.cluenet.org) |
| 20:25.37 | jplank | just make sure the wall is dry, you don't want to hurt the board |
| 20:26.04 | Katty | k |
| 20:26.20 | jplank | has anyone else gotten so tired of customers requests that you say no to things you know could be done, just cause? |
| 20:26.49 | Katty | oh i've gone past that |
| 20:27.00 | Katty | but yeah |
| 20:27.35 | jplank | I had a customer that has had me change things so many times, when he asked me if I could set it up so his users don't have to dial 9 to get out, I told him it was impossible |
| 20:28.40 | jplank | even though this is the type of customer whos going to look it up and call me out on it anyway, it felt good saying no for a change |
| 20:28.49 | tzafrir_laptop | hmm... when you see something iwth too many buzzword TLAs, check if it is RFC5513-compliant |
| 20:29.23 | Katty | setup an extension to run smbclient |
| 20:29.32 | Katty | and send lots of popups to his windows workstations in the building |
| 20:30.08 | jplank | that brings up a good idea |
| 20:30.27 | jplank | next client that asks me for win pops of calls, is getting netsend messages :) |
| 20:30.57 | Katty | did you know windows eject can be ran remotely |
| 20:31.06 | jplank | hahah |
| 20:31.08 | jplank | even better |
| 20:31.14 | Katty | i'm thinking extension 210 should definately be eject |
| 20:31.33 | jplank | everytime the phone rings, the CDROM ejects |
| 20:31.38 | jplank | have the customer explain that one |
| 20:31.42 | Katty | hot |
| 20:31.52 | Katty | oh the things i've done... |
| 20:31.53 | Katty | shame on me |
| 20:32.05 | Katty | my workstation is running smoothly now. |
| 20:32.06 | Katty | go figure |
| 20:32.12 | Katty | chunks ram stick at wall |
| 20:32.15 | *** join/#asterisk DanyWalker (n=elmo@201.230.46.241) |
| 20:32.18 | jplank | that'd be worth the hassle just to hear the client explain the issue |
| 20:32.28 | DanyWalker | <PROTECTED> |
| 20:32.32 | Katty | it does WHAT?! |
| 20:32.39 | jplank | the only PC that runs smoothly is like the only safe PC, its the one locked up in the closet |
| 20:32.41 | DanyWalker | how i cant make a restart asterisk with phpagi ' |
| 20:32.41 | Katty | sounds like a short... |
| 20:32.45 | Katty | we'll need to replace the motherboard |
| 20:32.48 | Katty | that'll be 300 bucks please. |
| 20:32.58 | jplank | maybe its the flux capacitor |
| 20:33.00 | DanyWalker | how i cat do a restart asterisk with phpagi ' |
| 20:33.36 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 20:34.16 | jplank | exec( "asterisk -r -x restart now"); |
| 20:34.39 | jplank | think that syntax is correct |
| 20:36.04 | DavidR2008 | just an update, I had this weird problem where softphones on laptops connected via VPN were sending the wrong address to *, the solution was to add the nat=yes |
| 20:38.21 | Katty | i haz tummy ache )= |
| 20:40.20 | *** join/#asterisk CrazyTux (n=brandon@ip72-211-214-95.oc.oc.cox.net) |
| 20:42.00 | jplank | thanks for the info david! |
| 20:44.09 | Katty | hmm. |
| 20:48.31 | *** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 20:50.42 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 20:51.59 | BCS-Satori | Anyone having an issue with asterisk-addons-1.6.1.0-rc3 and 1.6.2.0-beta1. I can not make them, I keep getting chan_ooh323 function errors and strange characters |
| 20:52.19 | Qwell | BCS-Satori: Are you compiling them against the correct versions of Asterisk? |
| 20:52.34 | BCS-Satori | Yes. all fresh downloads from several minutes ago on a new box |
| 20:52.50 | BCS-Satori | the configure works fine but the make fails i will post a paste 1 sec |
| 20:53.14 | Qwell | What versions of Asterisk are you compiling them against? |
| 20:53.50 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:54.11 | tzafrir_laptop | Qwell, I wonder what it would take for the confiugre script of -addons to check that version |
| 20:54.25 | BCS-Satori | Qwell: http://rafb.net/p/vJ43QW89.html and Asterisk 1.6.0.8 |
| 20:54.48 | tzafrir_laptop | you need either newer asterisk or older -addons |
| 20:54.59 | Qwell | so, like I was asking... are you compiling against the right version of Asterisk? You are not. |
| 20:55.09 | BCS-Satori | Qwell as I just noticed... lol |
| 20:55.47 | BCS-Satori | Qwell: ignore me lol my bad |
| 20:56.35 | Katty | pokes asterisk.org |
| 20:56.43 | Katty | pokepoke |
| 20:56.51 | Katty | where did asterisk go :< |
| 20:57.48 | Katty | nnoooooes |
| 20:57.52 | Katty | must have my tarballs |
| 20:58.20 | Katty | no ping )_= |
| 20:58.51 | Qwell | Katty: downloads.digium.com |
| 20:58.56 | Katty | yeah i just went there |
| 20:59.14 | carrar | it's back |
| 21:03.05 | _pepo_ | ... I am using call files and I can call to other users, but How do I redirect to Voicemail? |
| 21:03.16 | Katty | wonders where dhadi is |
| 21:03.32 | slykens | hi all... just built 1.6.0.8 but it's failing on start with: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data ... any ideas? |
| 21:03.47 | slykens | fwiw, 1.6.0.7 fails the same way, but 1.6.0.6 is fine |
| 21:04.14 | Qwell | http://bugs.digium.com/view.php?id=14819 |
| 21:05.47 | slykens | thx... guess it's back to 1.6.0.6 for now. :) |
| 21:06.36 | Katty | Qwell: where are you hiding dadhi drivers |
| 21:06.42 | Katty | infobot: dahdi |
| 21:06.42 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 21:07.08 | Katty | ooh |
| 21:07.11 | Katty | asterisk.org is backup |
| 21:07.34 | DavidR2008 | _pepo_: if you know how to use a call file, then create a special context and/or extension for voicemail and set the context and extension fields of the call file to this special area |
| 21:09.19 | DavidR2008 | are there any commands/functions/etc. that allow you to ascertain the ip registered in sip for the current call and/or for a specific sip user while in the dialplan? |
| 21:09.57 | timeshell | infobot: sip |
| 21:09.57 | infobot | sip is probably http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 21:10.04 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 21:13.30 | DavidR2008 | later |
| 21:16.53 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:17.40 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 21:18.00 | *** join/#asterisk ruben23 (n=ITadmin@124.107.3.178) |
| 21:28.38 | Qwell | slykens: putnopvut just fixed that in svn. It'll be in the next release we make (which may be shortly - we'll see) |
| 21:33.20 | tzafrir_laptop | Is there any code that relies on 'show file version' to show $Revision$ and will break if it will be replace with $Id ? |
| 21:33.58 | tzafrir_laptop | err... wrong channel |
| 21:39.45 | *** join/#asterisk AndyCrawford (n=AndyCraw@dynamic-65-161-142-80.tvscable.com) |
| 21:40.04 | *** join/#asterisk ThxGiving (n=quassel@f048070004.adsl.alicedsl.de) |
| 21:40.16 | ThxGiving | hi |
| 21:41.03 | ThxGiving | has anyone tried mISDN2 with chan_lcr yet? |
| 21:48.36 | tzafrir_laptop | I heard about a few such |
| 21:55.12 | ThxGiving | well... with hfc cards in nt mode.. mISDN v1 is very unstable |
| 21:55.27 | ThxGiving | sometimes it even segfaults the kernel |
| 21:55.45 | ThxGiving | plus you cant compile v1 with the newest kernel |
| 21:55.57 | ThxGiving | so im trying mISDN2 now ^^ |
| 21:56.08 | ThxGiving | but theres no native chan for it yet |
| 21:56.18 | ThxGiving | so you have to use linux call router inbetween |
| 21:57.05 | ThxGiving | chan_lcr works so far but i only can call from asterisk to isdn devices.. but not the other way around |
| 22:08.01 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 22:10.58 | *** join/#asterisk CrazyTux (n=brandon@ip72-211-214-95.oc.oc.cox.net) |
| 22:11.23 | jameswf | SO the dev list has been dead for 27+ hours kinda creepy did someone blow something up? |
| 22:11.39 | [TK]D-Fender | goes to hide the last of the bodies |
| 22:20.26 | tzafrir_laptop | ThxGiving, IIRC latest misdn1 does build with latest kernels. But I don't really follow it |
| 22:21.00 | ThxGiving | yes somehow it builds.. but i have problems getting it to work with chan_misdn |
| 22:21.14 | ThxGiving | misdn2 is much easier |
| 22:21.57 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:33.06 | jplank | I am completely stumped |
| 22:34.34 | jplank | I've been having this problem with no caller id and after hanging up a call over a TDM channel, * picks back up the line as if another call was coming in |
| 22:34.43 | jplank | swapped the card and server no change |
| 22:35.03 | jplank | rebuilt another server in my office with a new card, tested, everything worked, installed it at the customers prem |
| 22:35.05 | jplank | same issue |
| 22:35.52 | jplank | but when I test the POTS line, everything seems good, get caller ID on my test set and the voltage looks perfect |
| 22:40.54 | *** join/#asterisk Pan3D (n=Pan3D@node2.sensoryresearch.net) |
| 22:55.36 | *** join/#asterisk cr4z3d (n=cr4z3d@unaffiliated/cr4z3d) |
| 22:55.50 | cr4z3d | hey can asterisk use SCCP over TLS for cisco ip phones? |
| 23:07.09 | *** join/#asterisk jeff_phillips (n=fircuser@m460436d0.tmodns.net) |
| 23:08.06 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 23:10.59 | *** join/#asterisk fusa (n=fusa@dpc6744201027.direcpc.com) |
| 23:16.22 | jeff_phillips | hi |
| 23:29.54 | *** join/#asterisk xirdal (n=xirdal@203-mo7-3.acn.waw.pl) |
| 23:35.54 | *** part/#asterisk xirdal (n=xirdal@203-mo7-3.acn.waw.pl) |
| 23:39.43 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 23:41.10 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 23:41.29 | KavanS | ok...I tried local channels and dialing but I did not have success |
| 23:41.39 | KavanS | any suggestions on evaluation if a zap channel is in use with an if statement? |
| 23:43.08 | *** join/#asterisk harryv (n=harry@vangberg.name) |
| 23:43.26 | harryv | hey, there's only sporadic news on skype for asterisk. does anybody know a bit more about the status? |
| 23:48.30 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 23:50.21 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
