| 00:06.04 | *** join/#asterisk woleium (~woleium@email.tecglobal.net) |
| 00:08.15 | *** part/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 00:26.44 | SeRi | quit |
| 00:26.47 | SeRi | lol |
| 00:28.51 | phix | SeRi: try that again with a / infront :) |
| 00:29.01 | SeRi | lol |
| 00:29.16 | SeRi | phix: I was trying to quit asterisk cli. wrong window :P |
| 00:30.30 | phix | ah hehe |
| 00:30.39 | WIMPy | "You can check out any time you like, but you can never leave." |
| 00:30.48 | phix | I find CTRL+d to be less buttons I need to press |
| 00:31.11 | phix | I mean CTRL+c even |
| 00:31.39 | SeRi | lol |
| 00:31.41 | phix | either that or press CTRL+a d to detach the screen I have the console running on :P |
| 00:36.19 | *** part/#asterisk mmaki (~mmaki@pool-71-109-218-186.lsanca.dsl-w.verizon.net) |
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| 00:47.36 | *** part/#asterisk gigawatts121 (~gigawatts@c-24-13-241-203.hsd1.il.comcast.net) |
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| 01:04.31 | Twitchnln | Evening, anyone in here got any experience reseting spa942 when unit has pw? |
| 01:08.09 | Twitchnln | m, found it |
| 01:09.52 | lauris | how? |
| 01:20.07 | *** join/#asterisk splices0 (~rogerclin@96.27.249.63) |
| 01:20.51 | splices0 | Greetings. I'm setting up an asterisk box that is converting dialogic TDM to SIP and going out on some T1s |
| 01:21.06 | splices0 | its being used to outbound dial a lot of numbers for a provider |
| 01:21.26 | splices0 | some people are posing that asterisk isnt capable of keeping up with tdm-to-sip conversions and dial the calls |
| 01:21.31 | splices0 | anyone have any thoughts? |
| 01:23.27 | p3nguin | phix: or Ctrl+\ to dtach it. |
| 01:24.00 | carrar | You mean SIP/RTP to TDM? |
| 01:24.06 | carrar | PRI |
| 01:24.31 | carrar | Work great with the Digium T1 cards |
| 01:24.33 | carrar | Works |
| 01:24.47 | splices0 | yes |
| 01:24.53 | carrar | g.711u/g.711a |
| 01:25.03 | splices0 | with about 12 T1s? |
| 01:25.08 | splices0 | of conversion |
| 01:25.19 | carrar | You really want to put all your eggs in one basket? |
| 01:26.22 | carrar | I'd put those 12 T1's on a Cisco 5400 and let it do at it's DSP level |
| 01:26.40 | carrar | or something other then a Asterisk box |
| 01:26.58 | splices0 | we have a failover...Its just we are getting a lot of issues and some people in the industry are saying the dialogic tdm to sip conversion by asterisk wouldnt keep up |
| 01:27.52 | carrar | move those 12 over to a DS3 |
| 01:28.17 | pabelanger | well, asterisk does not have a Dialogic channel driver to start. 3rd party might |
| 01:28.18 | splices0 | the dialer software is kind of old and configured for use only on tdm |
| 01:28.20 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 01:28.40 | splices0 | we are converting that to sip |
| 01:28.54 | carrar | I don't use Dialogic |
| 01:29.10 | carrar | 12 very well may be too much |
| 01:29.18 | splices0 | we are using chan_dahdi |
| 01:29.35 | splices0 | with digium cards |
| 01:30.02 | *** join/#asterisk jpsharp (~jsharp@74-95-145-82-Naples.hfc.comcastbusiness.net) |
| 01:30.13 | carrar | What does Digium support say? |
| 01:30.51 | carrar | So you have 3 quad cards? |
| 01:30.59 | *** join/#asterisk KNERD (~KNERD@99.65.2.188) |
| 01:31.00 | splices0 | Not sure...Im not sure if digium support is where to ask -- yes 3 quad cards |
| 01:31.32 | splices0 | We are trying to use it for predictive dialing |
| 01:31.34 | carrar | got the interrupts you should be fine |
| 01:31.43 | carrar | and you install them correctly |
| 01:31.48 | carrar | sync canles |
| 01:31.50 | carrar | cables |
| 01:32.19 | carrar | but if you are not sure, ask digium |
| 01:32.21 | carrar | it's their cards |
| 01:32.25 | carrar | their software |
| 01:32.53 | splices0 | I see -- so a good direction for googling this wouldnt be tdm to sip conversion but for predictive dialing with digium or chan_dahdi? |
| 01:33.15 | carrar | predictive dialing is not the issue |
| 01:33.28 | carrar | SIP to T1 conversion is |
| 01:33.35 | carrar | vs capacity |
| 01:33.54 | KNERD | hey Qwell. my asterisk version 1.8.7.1 memory usage keeps going up on a daily basis and I have to keep restarting it. Why is this? |
| 01:33.57 | splices0 | I see. thank you |
| 01:47.50 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 01:53.21 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
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| 01:59.34 | *** join/#asterisk MrPockets (~JimmyCrac@unaffiliated/mrpockets) |
| 02:00.20 | MrPockets | Can someone recommend a cheap (sub $100) hard phone to use as I screw with PIAF as a side hobby (no real production use) |
| 02:00.48 | lauris | SPA502G ? |
| 02:01.06 | lauris | or any older SPA model |
| 02:01.08 | p3nguin | Ebay can help you find several Cisco/Linksys and Polycom phones for that price. |
| 02:01.48 | MrPockets | Will any voip phone work, or are there different kinds for various applications? |
| 02:02.02 | lauris | yes. |
| 02:02.08 | p3nguin | You'll want to make sure they speak SIP. |
| 02:02.13 | lauris | there are which are only compatible with H323 or SCCP |
| 02:02.16 | p3nguin | That will be the easiest to deal with. |
| 02:02.35 | lauris | MrPockets, look for any SPA model |
| 02:02.46 | MrPockets | Cool. Thanks friends. |
| 02:02.48 | lauris | those will work for sure |
| 02:02.54 | p3nguin | You can use other channels, as well, but SIP is where asterisk seems to have the most attention. |
| 02:03.00 | lauris | have hundreds of them in my network |
| 02:04.08 | p3nguin | An older Cisco 7940 or 7960 would be okay, too, and you can use SIP, SCCP, or MGCP on the phone. Asterisk has channel drivers for all three of those techs. |
| 02:04.30 | lauris | wouldn't say so |
| 02:04.34 | lauris | they have some compatibility issues |
| 02:04.41 | p3nguin | Not really. |
| 02:04.46 | lauris | Yes, they do |
| 02:04.58 | p3nguin | I use only Cisco phones with my Asterisk. |
| 02:05.15 | p3nguin | I use chan_sccp-b, though. |
| 02:05.16 | lauris | so probably you were lucky not to step on any of them |
| 02:05.30 | lauris | i mean SIP compat. problems with 7940 |
| 02:05.43 | lauris | in particular when calls are forwarded from one phone to another |
| 02:05.54 | p3nguin | I use chan_sccp-b on 7960/7940, and SIP on a 7912. |
| 02:06.27 | lauris | and you do kind of "bridging" between sccp and sip ? |
| 02:06.46 | p3nguin | Yes. Asterisk bridges calls. |
| 02:07.01 | p3nguin | I have used SIP on 7960/7940 and they worked fine, but the SIP firmware lacked features on those phones. |
| 02:07.21 | p3nguin | Since they were built for SCCP, it wasn't a surprise. |
| 02:08.51 | carrar | Make call, Receive call, what more could you possibly want!!! :) |
| 02:09.08 | p3nguin | Yeah, right. :/ |
| 02:09.08 | WIMPy | A lot. |
| 02:09.15 | WIMPy | But that's all history. |
| 02:09.16 | p3nguin | It certainly does that part. |
| 02:10.45 | carrar | I would like the spare processing on the phone to partake in the SETI program |
| 02:10.54 | p3nguin | nice |
| 02:11.20 | p3nguin | Might as well do some protein folding or even deal in bitcoin. |
| 02:11.41 | carrar | bitcoin is going no where |
| 02:11.47 | carrar | It's all about aliens |
| 02:12.30 | carrar | Aliens are the only thing thats gonna save the US Gov |
| 02:24.19 | *** join/#asterisk master_of_master (~master_of@p57B5408C.dip.t-dialin.net) |
| 02:33.32 | *** join/#asterisk gravin (~gravin@57.70.50.60.brf01-home.tm.net.my) |
| 02:41.18 | *** join/#asterisk spotter (~spotter@user-12ld063.cable.mindspring.com) |
| 02:41.36 | spotter | any idea why I'd be getting an error like this |
| 02:41.37 | spotter | [Dec 21 21:37:22] WARNING[2791]: file.c:644 ast_openstream_full: File /tmp/opening.wav does not exist in any format |
| 02:41.47 | p3nguin | That's not an error. |
| 02:41.51 | spotter | ok, wanring |
| 02:42.04 | spotter | its not playing anything |
| 02:42.09 | p3nguin | /tmp/opening.wav does not exist in any format. |
| 02:42.14 | spotter | it does |
| 02:42.18 | spotter | and I sox converted it to ulaw |
| 02:42.21 | sunfone | remove .wav from the filename |
| 02:42.23 | p3nguin | What is the app data that you supplied? |
| 02:44.08 | spotter | asterisk doesn't like extension I guess |
| 02:44.16 | spotter | removing .wav in the dialplan helped |
| 02:44.17 | p3nguin | Never has. |
| 02:44.19 | spotter | now getting noise |
| 02:44.26 | spotter | I guess make sense |
| 02:44.35 | spotter | can store sounds in multiple formats |
| 02:44.42 | sunfone | allows asterisk to open the file that matches the channels codec |
| 02:44.51 | p3nguin | It will play the one that is best suited for the call. |
| 02:45.43 | spotter | for some reason my sox converted 16 bit pcm -> ulaw results in just noise |
| 02:45.54 | sunfone | I wonder, if in the case it doesn't exist in the channel's codec, if it will take the time to open the one with the least transcoding effort? |
| 02:46.09 | p3nguin | It will transcode, yes. |
| 02:46.16 | sunfone | or do they all go to slin no matter what? |
| 02:46.43 | spotter | there we go |
| 02:46.48 | sunfone | I get that it will transcode, but some transcoding will be more expensive than others, so does it search in least effort order? |
| 02:47.39 | sunfone | That would be a great certification question :) |
| 03:02.08 | *** join/#asterisk incendiaries (~incendiar@c-76-98-133-208.hsd1.pa.comcast.net) |
| 03:14.52 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:15.03 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:29.08 | F2Knight | <PROTECTED> |
| 03:29.08 | F2Knight | FATAL: Error inserting dahdi_dynamic_ethmf (/lib/modules/2.6.32-37-generic-pae/dahdi/dahdi_dynamic_ethmf.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 03:29.14 | F2Knight | any ideas? |
| 03:29.39 | F2Knight | Dmesg doesn't provide any more help |
| 03:29.41 | F2Knight | [ 8028.517903] dahdi_dynamic_ethmf: Unknown symbol dahdi_dynamic_unregister_driver |
| 03:29.41 | F2Knight | [ 8028.518659] dahdi_dynamic_ethmf: disagrees about version of symbol dahdi_dynamic_receive |
| 03:29.41 | F2Knight | [ 8028.518669] dahdi_dynamic_ethmf: Unknown symbol dahdi_dynamic_receive |
| 03:29.41 | F2Knight | [ 8028.525617] dahdi_dynamic_ethmf: Unknown symbol dahdi_dynamic_register_driver |
| 03:34.37 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:52.00 | *** join/#asterisk gravin (~gravin@175.136.216.153) |
| 03:54.33 | *** join/#asterisk TJNII (~TJNII@tjnii.com) |
| 03:56.32 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 04:06.28 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
| 04:10.02 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 04:14.07 | ChannelZ | F2Knight: old build that doesn't match your current kernel? dunno |
| 04:16.04 | F2Knight | ChannelZ, hoping .. just started a rebuild of the drivers. |
| 04:25.19 | F2Knight | ChannelZ, fail... |
| 04:25.47 | F2Knight | but new message.. |
| 04:25.58 | F2Knight | FATAL: Error inserting dahdi_dynamic_ethmf (/lib/modules/2.6.32-37-generic-pae/dahdi/dahdi_dynamic_ethmf.ko): Invalid module format |
| 04:29.30 | *** join/#asterisk cyford (~allen@c-24-98-175-41.hsd1.ga.comcast.net) |
| 04:29.46 | SeRi | waz up guys |
| 04:30.38 | *** part/#asterisk lauris (~la@unaffiliated/lauris) |
| 04:30.40 | ChannelZ | hmmm |
| 04:31.31 | p3nguin | Yes. |
| 04:31.52 | SeRi | waz up p3nguin |
| 04:32.16 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 04:33.28 | p3nguin | Drinkin' beer, watching "Weaponology: Fire Weapons" on TV. |
| 04:34.34 | SeRi | cool. sounds like fune |
| 04:35.16 | WIMPy | Jo SeRi. How's the brew? |
| 04:35.29 | F2Knight | dahdi_cfg -d2 tells me : DAHDI dynamic span creation failed: Invalid argument |
| 04:35.56 | SeRi | waz up WIMPy |
| 04:36.35 | WIMPy | "random features" |
| 04:36.48 | SeRi | no whiskey for me tonight... feals like a friday for some reason.... been buzzy with the house.... fam comes in on Friday from PR. |
| 04:36.57 | SeRi | WIMPy: lol |
| 04:37.05 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 04:37.11 | scalex000 | hello |
| 04:37.57 | scalex000 | I have a question when i dial from asterisk to Nortel, I want to specific the phone-context in the sip message how to do it? |
| 04:37.59 | WIMPy | Last hour before the solstice. |
| 04:38.52 | SeRi | I might go off line I am working on cleaning up the cable closet.... |
| 04:39.09 | SeRi | wtf... *cable closet* lol |
| 04:39.15 | SeRi | *network closet* |
| 04:39.33 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 04:43.08 | Neptu | hej |
| 04:43.17 | p3nguin | hedge |
| 04:45.13 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 04:46.44 | Neptu | p3nguin: http://pastebin.com/9BEgLg3g any ideas?? |
| 04:48.23 | Neptu | seems the chatroom is hangin out |
| 04:53.18 | Neptu | this is dam anoting |
| 04:53.28 | *** join/#asterisk rdancer (~jm@unaffiliated/rdancer) |
| 04:54.19 | *** join/#asterisk AndyMLi7 (~Adium@unaffiliated/andymli7) |
| 04:54.20 | rdancer | how does one find out which mobile network a telephone number is on? is it even possible? |
| 04:55.17 | AndyMLi7 | I have an asterisk 1.8.6 machine that doesn't like the sip registration its getting from a Cisco 7940. The problem looks a lot like the 1.8.3.3 Cisco regression but I can't make sense of it. |
| 04:55.56 | AndyMLi7 | the sip debug looks like this |
| 04:55.57 | AndyMLi7 | http://pastie.org/private/pazogcx8jn9h9nyoosl47w |
| 04:56.03 | AndyMLi7 | Invalid SIP message - rejected , no callid, len 357 |
| 04:56.17 | AndyMLi7 | chan_sip.c: REGISTER request has no from tag, dropping callid: - etc. |
| 04:57.08 | Neptu | dam dam dam |
| 04:57.10 | Neptu | :P |
| 05:05.04 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 05:07.35 | Neptu | what is the diference of writing meetme(9000@rooms) and meetme(9000)?? |
| 05:13.40 | jpsharp | Is that even a legal syntax? I know you can use the "v" option to play a mailbox@context message, but I dont think meetme(9000@rooms) is valid. |
| 05:14.14 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 05:14.49 | kuku | My RxFax application stopped making sounds when answering the fax. It just sits there with no noise coming out. Any ideas? |
| 05:15.37 | ChannelZ | Answering first? maybe the media stream is broke |
| 05:22.57 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
| 05:25.20 | kuku | It answers, and its quiet |
| 05:28.46 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-pktdayoztjvfqnmc) |
| 05:28.55 | WIMPy | Looks like he found the wrong cable. |
| 05:29.54 | ChannelZ | eh? |
| 05:30.55 | WIMPy | <SeRi> I might go off line I am working on cleaning up the cable closet.... |
| 05:37.09 | WIMPy | Happy new year, BTW. |
| 05:37.52 | *** part/#asterisk AndyMLi7 (~Adium@unaffiliated/andymli7) |
| 05:38.02 | Yourname` | Guys, thanks a ton for your help with DAHDI. Now, I just have call choppiness every now and then. eyeBeam running on ULAW to the router, then to the local PBX, then out the dahdi card. |
| 05:44.06 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
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| 05:53.02 | ChannelZ | oh. |
| 05:54.19 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 05:54.32 | *** part/#asterisk joshaidan (~brianj@24.109.210.41) |
| 06:00.57 | *** join/#asterisk speedtapir (~kunal@ppp-58-11-251-251.revip2.asianet.co.th) |
| 06:01.33 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 06:02.29 | speedtapir | hello. my apologies for disturbing. I have been having serious difficulties in trying to get the register string to register my sip provider but i have always failed(even VPS) |
| 06:03.00 | speedtapir | I tried to get support from the asterisk forum at asterisk.org but got little support/help. |
| 06:08.51 | Neptu | ok now i have conference rooms working... |
| 06:08.55 | Neptu | ;) |
| 06:09.07 | Neptu | dahdi is evil |
| 06:09.15 | Neptu | and gentoo is more evil |
| 06:09.59 | Neptu | I think i will take a shower now :D |
| 06:15.45 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:20.24 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:32.40 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 06:35.13 | ChannelZ | speedtapir: ok well we can't read your mind |
| 06:36.11 | *** join/#asterisk gravin (~gravin@175.136.252.105) |
| 06:38.24 | speedtapir | @ChannelZ in my sip.conf file i have entered the following details :- |
| 06:38.26 | speedtapir | [general] |
| 06:38.26 | speedtapir | allow=ulaw |
| 06:38.26 | speedtapir | allow=g729 |
| 06:38.26 | speedtapir | bindaddr=0.0.0.0 |
| 06:38.26 | speedtapir | port=5060 |
| 06:38.27 | speedtapir | register => 6xxxxx:xxxx@sip.provider.com |
| 06:38.38 | speedtapir | but for some reason it just wont register. |
| 06:39.35 | ChannelZ | please use pastebin in the future |
| 06:39.43 | speedtapir | i see it as a peer on asterisk and not as registry. |
| 06:39.43 | ChannelZ | Do you get an error? What does 'sip show registry' say? |
| 06:40.05 | speedtapir | 0 SIP registrations |
| 06:40.21 | speedtapir | even tried to reload, restart etc. just wont register it... |
| 06:41.14 | speedtapir | tried on my computer, at home, even i tried it on VPS via SSH... |
| 06:41.51 | ChannelZ | you might have some funky characters or something else in your sip.conf causing it not to be parsed I suppose |
| 06:42.18 | speedtapir | could be... |
| 06:43.13 | speedtapir | should I just remove all the characters and just put 1 registration string? |
| 06:43.24 | speedtapir | to see if it registers? |
| 06:43.32 | ChannelZ | is the file readable by asterisk? |
| 06:43.51 | speedtapir | yes. I have managed to connect phones on my local lan |
| 06:44.32 | ChannelZ | so that wasn't your whole sip.conf? Is the register line actually in the [general] section? |
| 06:44.44 | speedtapir | yes. |
| 06:48.15 | ChannelZ | what version of asterisk, and do you have any odd characters in your name/password? |
| 06:48.51 | speedtapir | <PROTECTED> |
| 06:49.22 | speedtapir | username is just numbers |
| 06:49.30 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 06:49.41 | speedtapir | password is mix of text,capital leter and numbers |
| 06:49.46 | speedtapir | letters* |
| 06:53.04 | F2Knight | Still having some issues with Dahdi configuration and loading dahdi_dynamic_ethmf if anyone has some insight to either ? |
| 07:00.45 | *** join/#asterisk gravin (~gravin@175.136.252.105) |
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| 07:01.25 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:07.19 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 07:12.35 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 07:12.44 | SeRi | back |
| 07:12.57 | SeRi | dog is acting wierd... :/ |
| 07:13.14 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 07:18.50 | speedtapir | @ChannelZ : any idea? |
| 07:19.27 | speedtapir | i tried running tcpdump but it just doesnt seem to work |
| 07:23.12 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:25.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:26.57 | speedtapir | hello. Under [general] i have only the string |
| 07:27.02 | speedtapir | is this suffice for it to register |
| 07:27.58 | speedtapir | doesn't seem like |
| 07:28.38 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:34.05 | kaldemar | speedtapir: pastebin your whole sip.conf and the CLI output of "module unload chan_sip.so" and "module load chan_sip.so" |
| 07:34.54 | kaldemar | speedtapir: in sip.conf, don't modify or mask _anything_ but the credentials. |
| 07:35.20 | speedtapir | ok. |
| 07:35.35 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:38.49 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 07:41.59 | *** join/#asterisk XnOSX (~XnOSX__@62.83.235.60.dyn.user.ono.com) |
| 07:43.02 | speedtapir | @kaldemar: http://pastebin.com/d7uARQTA |
| 07:43.11 | speedtapir | #kaldemar |
| 07:43.20 | kaldemar | speedtapir: is that the whole file? |
| 07:43.25 | KNERD | what's a good way to troubleshoot why a AGI script is failing? |
| 07:43.33 | speedtapir | yes |
| 07:44.02 | kaldemar | speedtapir: it lacks [general] and a bind address at least. |
| 07:44.06 | speedtapir | i just want to get my SIP provider registered. |
| 07:45.16 | speedtapir | is bind addr required? |
| 07:45.45 | kaldemar | speedtapir: yes. without it the channel driver is not able to use the network and is completely useless. |
| 07:46.39 | kaldemar | speedtapir: add [general] and udpbindaddr=0.0.0.0 to the top of the file |
| 07:46.58 | speedtapir | ok |
| 07:47.08 | kaldemar | speedtapir: is your asterisk box behind a NAT? |
| 07:47.34 | speedtapir | its running on a VPS so it shouldn't be. |
| 07:47.38 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:47.39 | schmidts | good morning |
| 07:47.46 | KNERD | bad morning |
| 07:48.09 | speedtapir | <PROTECTED> |
| 07:48.49 | kaldemar | speedtapir: running on a VPS doesn't really say anything about the network environment. whether it has a public ip address or not is relevant. |
| 07:49.03 | schmidts | does anyone in here use asterisk above 1.8.7.1? |
| 07:51.58 | speedtapir | @kaldemer: i was unaware of that. |
| 07:52.21 | speedtapir | my sip provider suggested me to try running it from a VPS and I just followed |
| 07:53.41 | kaldemar | speedtapir: before worrying about that some more, let's try to get it to attempt the registration. |
| 07:54.03 | speedtapir | yes. please. i already made the changes |
| 07:56.22 | speedtapir | and reloaded the system. it still does not register. |
| 07:58.13 | speedtapir | @kaldemar: whats a good way testing the network enviornment? |
| 07:59.50 | *** join/#asterisk gravin (~gravin@175.136.216.153) |
| 08:00.08 | schmidts | speedtapir start with ping and traceroute |
| 08:01.33 | speedtapir | schimidts: seems to be pinging fine PING 216.115.69.144 (216.115.69.144) 56(84) bytes of data. |
| 08:01.33 | speedtapir | 64 bytes from 216.115.69.144: icmp_req=1 ttl=53 time=163 ms |
| 08:02.39 | schmidts | speedtapir what OS do you use? |
| 08:02.57 | speedtapir | running ubuntu |
| 08:03.26 | speedtapir | and it does traceroute too. |
| 08:03.35 | schmidts | ok then you could use sipsak (should be available as ubuntu package) to check if they answer to sip request |
| 08:03.46 | speedtapir | ok |
| 08:04.51 | *** join/#asterisk d00gster (~dt@2.90.71.39) |
| 08:05.10 | speedtapir | schmidts: just installed sipsak. going to run it |
| 08:05.31 | schmidts | speedtapi ok give me a second to find the right parameters ;) |
| 08:05.55 | speedtapir | thanks |
| 08:06.42 | schmidts | you can easily start with something like this: sipsak -s sip:YOURUSER@SERVER -vv |
| 08:08.16 | speedtapir | yeah. i was thinking of that. |
| 08:08.40 | schmidts | do you get something back? |
| 08:09.49 | speedtapir | 1 min.. |
| 08:09.53 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 08:10.34 | speedtapir | yes i did. |
| 08:10.38 | speedtapir | putting it on pastebin. |
| 08:11.22 | *** join/#asterisk gravin (~gravin@175.136.216.153) |
| 08:12.02 | speedtapir | schmidts: http://pastebin.com/tWGJeKqn |
| 08:13.10 | schmidts | speedtapir then it doesnt look like a network problem, cause they atleast answer to an option message |
| 08:13.11 | *** join/#asterisk irroot (~gregory@41.52.154.230) |
| 08:14.05 | speedtapir | schmidts: so this means something in the asterisk configuration is wrong? |
| 08:17.04 | kaldemar | speedtapir: enable sip debug in asterisk with "sip set debug on" and do a sip reload. you should see a register message going out if asterisk tries to register. if you don't, check that nothing else in the OS is binding to UDP port 5060. |
| 08:18.42 | speedtapir | kaldemar: i did sip reload and there is no message. could this be because of the verbosity? |
| 08:19.12 | speedtapir | prior to running 'sip reload' I ran 'sip set debug on' |
| 08:19.29 | speedtapir | but there is no register message |
| 08:19.33 | speedtapir | going out. |
| 08:21.43 | speedtapir | kaldemar: I ran 'netstat -vaun' |
| 08:21.58 | speedtapir | and i see : udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 08:23.04 | schmidts | speedtapir i dont know -vaun please try netstat -tulpen |
| 08:23.11 | speedtapir | ok |
| 08:24.33 | speedtapir | this is the result I get when i run 'netstat -tulpen' http://pastebin.com/XJggaVZ8 |
| 08:24.35 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 08:25.25 | schmidts | looks ok |
| 08:25.52 | *** join/#asterisk ollii (~risker@vpn.nbank.de) |
| 08:26.08 | speedtapir | could it be the asterisk version I am using? |
| 08:26.15 | schmidts | what version do you use? |
| 08:26.41 | speedtapir | 1.8.4.4 |
| 08:26.51 | IsUp | hello |
| 08:26.55 | kaldemar | speedtapir: verbosity does not affect sip debug. so no. |
| 08:27.09 | schmidts | could be but i dont think 1.8.4.4 had this problems |
| 08:27.10 | kaldemar | speedtapir: pastebin your current sip.conf |
| 08:27.39 | IsUp | is that pıossible to change session name ('s=') in SIP packets? i did it with editing chan_sip.c |
| 08:27.49 | IsUp | but is there any config to change this value? |
| 08:28.49 | schmidts | IsUp AFAIK no |
| 08:28.55 | speedtapir | kaldemar: |
| 08:28.57 | speedtapir | http://pastebin.com/tU2wJAB6 |
| 08:29.02 | kaldemar | IsUp: sdpsession |
| 08:29.27 | IsUp | kaldemar: is that a variable? |
| 08:29.53 | kaldemar | IsUp: it is a sip.conf parameter. there's also sdpowner. |
| 08:30.20 | IsUp | kaldemar: aah, thank you |
| 08:31.10 | IsUp | i did with it "snprintf(global_sdpsession, sizeof(global_sdpsession), "SBX-1-1-1");" in chan_sip.c and it works so far. i'll try sdpsession in config |
| 08:33.44 | *** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net) |
| 08:34.51 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:35.56 | schmidts | kaldemar cool ;) didnt know this |
| 08:36.19 | *** join/#asterisk Nasga (~Nasga@112.4.118.78.rev.sfr.net) |
| 08:36.20 | kaldemar | speedtapir: there should not be anything wrong with your configs. is DNS working on the box? |
| 08:37.52 | kaldemar | see if "core set debug 10", "module unload chan_sip.so" and "module load chan_sip.so" gives anything useful. |
| 08:39.04 | speedtapir | kaldemar: ok. trying |
| 08:42.44 | speedtapir | kaldemar: [Dec 22 09:42:14] NOTICE[30927]: chan_sip.c:26294 build_peer: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser' |
| 08:42.53 | speedtapir | thats all I get. |
| 08:43.39 | kaldemar | interesting. the exact same config works for be, although i tested with 10.0.0. |
| 08:44.45 | speedtapir | you mentioned DNS |
| 08:45.00 | speedtapir | should i try replacing the sip provider with IP ? |
| 08:45.07 | kaldemar | worth a try |
| 08:46.20 | kaldemar | asterisk 10 prints a load of warnings about not being able to resolve an address, whether there is verbosity/core debug or not. |
| 08:46.30 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 08:47.07 | KNERD | speedtapir: fromdomain=sip.flowroute.com is incorrect |
| 08:47.24 | speedtapir | KNERD ? |
| 08:47.31 | KNERD | that is from YOU, not the provider |
| 08:47.32 | speedtapir | KNERD: i got this from |
| 08:47.41 | speedtapir | the sip provider. |
| 08:47.44 | KNERD | okay |
| 08:47.48 | KNERD | if they say so |
| 08:48.01 | speedtapir | or should I remove that line? |
| 08:48.17 | KNERD | I have added that in my own conf because you could not register so I have to put my IP address in that spot |
| 08:48.29 | kaldemar | fromdomain should have your domain as its value, not the provider's. |
| 08:50.17 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 08:50.57 | KNERD | it can be a domain name or IP address |
| 08:51.08 | speedtapir | i see. can i removied the fromdomain ? |
| 08:51.14 | speedtapir | remove* |
| 08:51.22 | speedtapir | and try |
| 08:51.54 | speedtapir | or add my ip address in the from domain? |
| 08:51.57 | KNERD | yes.. just comment it out with ; |
| 08:52.06 | speedtapir | ok |
| 08:52.09 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:52.26 | KNERD | but you should not need it because you are are acutally registering |
| 08:52.31 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 08:53.16 | speedtapir | yes. by the way. when i run 'sip reload' i do get a bunch of notices..but i see 1 error |
| 08:53.18 | speedtapir | [Dec 22 09:52:26] ERROR[31096]: res_config_ldap.c:1657 parse_config: No directory URL or host found. |
| 08:55.02 | speedtapir | kaldemar: tried replacing the sip.providers with ip and no it doesnt work. |
| 08:56.38 | ChannelZ | are you using realtime config? |
| 08:56.40 | kaldemar | ahem.. res_config_ldap... add "noload => res_config_ldap.so" to /etc/asterisk/modules.conf and restart asterisk. |
| 08:56.41 | speedtapir | this is what happens when I reload : http://pastebin.com/B4JQX8Nu |
| 08:57.18 | IsUp | speedtapir: are you having problems with registration? |
| 08:57.35 | speedtapir | IsUp: Yes. A serious problem |
| 08:58.15 | IsUp | speedtapir: ok wheres your register string in sip.conf? |
| 08:58.42 | kaldemar | IsUp: http://pastebin.com/tU2wJAB6 |
| 08:58.56 | kaldemar | IsUp: that's been dealt with already, scroll back. |
| 08:59.32 | speedtapir | kaldemar: added it |
| 08:59.57 | IsUp | kaldemar: ok i thought first that register string is not under [default] i had same problem before |
| 08:59.58 | speedtapir | adeeded it under [global] |
| 09:00.12 | speedtapir | added* |
| 09:01.02 | IsUp | speedtapir: do you have any special chars in your username/password? also, enable sip debug. check if your asterisk sending REGISTER or not. |
| 09:02.33 | speedtapir | IsUp: already done all those. asterisk just wont send REGISTER |
| 09:03.01 | speedtapir | i think it is cause of the error message i am getting when i reload |
| 09:03.29 | IsUp | speedtapir: if you are talking about LDAP error, thats normal. |
| 09:03.55 | kaldemar | [default] and [global] are both mumbo jumbo in sip.conf. you mean [general]. |
| 09:04.38 | IsUp | speedtapir: send me a register message. i'll pm you my PBX ip. im enabled sip debug here. so i'll see if ur sending anything or not |
| 09:04.52 | speedtapir | ok |
| 09:05.17 | *** join/#asterisk s[X]_ (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 09:05.27 | kaldemar | if "sip show registry" does not even list it, asterisk has a problem with reading the registration statement. |
| 09:06.04 | IsUp | kaldemar: did he post full sip.conf? i think theres something wrong with it |
| 09:06.16 | kaldemar | IsUp: twice. scroll back. |
| 09:06.28 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:06.47 | IsUp | kaldemar: it's only 19 lines :p is that normal? |
| 09:08.02 | kaldemar | IsUp: depends on the definition of normal. :P it works here anyway. |
| 09:08.33 | IsUp | speedtapir: did you add register string for my pbx? still waiting |
| 09:10.03 | speedtapir | yes i dod. |
| 09:10.12 | speedtapir | yes i did |
| 09:10.22 | IsUp | ok, sip reload again |
| 09:10.27 | speedtapir | just now. i just ran reload |
| 09:11.01 | speedtapir | kaldemar: which version do u use? |
| 09:11.14 | kaldemar | speedtapir: 10.0.0 |
| 09:11.22 | IsUp | ok got nothing, btw i have Asterisk 1.8.8.0 |
| 09:11.36 | IsUp | i'll test your config speedtapir, but i am pretty sure. your config is totally wrong and something missing |
| 09:12.04 | speedtapir | IsUp: I have a feeling |
| 09:12.17 | speedtapir | but my ISP gave me these configs so I just followed it. |
| 09:12.54 | speedtapir | IsUp: but kaldemar managed to get it. |
| 09:13.21 | IsUp | yes it works :/ |
| 09:13.38 | IsUp | speedtapir: do you have any special chars in your username/password? |
| 09:13.56 | kaldemar | only gray area left is the username and secret in the regiter statement. |
| 09:13.57 | speedtapir | username is numbers, passworkd is number,caps,and letters |
| 09:14.40 | speedtapir | indeed. should we try to replace username and password and see if we get registration failed. |
| 09:15.19 | IsUp | speedtapir: yes. |
| 09:15.29 | speedtapir | ok |
| 09:17.04 | speedtapir | username and password is speedtapir |
| 09:17.44 | speedtapir | still 0 SIP registration :/ |
| 09:18.03 | speedtapir | i have a feeling this version of asterisk isn't right. |
| 09:18.47 | kaldemar | there are no differences between the sip config parser portions of 1.8.4.4 and 10.0.0. |
| 09:19.30 | speedtapir | this is my sip.conf file --http://pastebin.com/Xq7fdVfK |
| 09:20.36 | speedtapir | what other configuration in asterisk determines how asterisks connects to the network besides sip.conf |
| 09:20.40 | kaldemar | and "sip show registry" still does not list any registrations? |
| 09:20.48 | speedtapir | none. |
| 09:20.50 | kaldemar | speedtapir: regarding sip, none. |
| 09:21.07 | kaldemar | then asterisk isn't reading the file. i just tested 1.8.4.4 and it works too. |
| 09:21.18 | IsUp | but its reading peer, thats strange |
| 09:21.20 | kaldemar | you're modifying the wrong file or something spooky. |
| 09:21.39 | speedtapir | it does show it as peers |
| 09:21.44 | speedtapir | but now as registry. |
| 09:21.47 | speedtapir | not* |
| 09:21.58 | kaldemar | the peer and the registration are not really connected to each other. |
| 09:22.48 | speedtapir | both commands(peer and registration) are read from sip.conf |
| 09:22.51 | speedtapir | right? |
| 09:22.55 | IsUp | yes |
| 09:23.05 | IsUp | speedtapir: ok i have a stupid idea, hold a sec |
| 09:23.13 | speedtapir | ok. |
| 09:23.19 | schmidts | speedtapir maybe you should try "core set debug 10" and then do a sip reload if you see anything else |
| 09:23.29 | IsUp | speedtapir: replace your register line with #include "sip_register.conf" |
| 09:23.39 | IsUp | create a file named sip_register.conf, put your register string there |
| 09:23.49 | speedtapir | ok. |
| 09:27.14 | kaldemar | speedtapir: pastebin the output of "ps axw | grep asterisk ; grep etcdir /etc/asterisk/asterisk.conf" |
| 09:28.39 | petern_ | is it possible to control whether asterisk says "and" when saying numbers? |
| 09:29.15 | speedtapir | kaldemar : here is the output: http://pastebin.com/9bwY8RPs |
| 09:29.34 | IsUp | ouch |
| 09:29.36 | speedtapir | IsUp: tried your idea but seems like no effect |
| 09:29.51 | speedtapir | whats the issue? |
| 09:30.28 | schmidts | speedtapir could you try the debug and sip reload thing |
| 09:30.32 | kaldemar | petern_: be more specific please |
| 09:31.20 | schmidts | petern_ i guess you mean sayunixtime when asterisk says its 7 hours 15 minutes "AND" 23 seconds, right? |
| 09:31.24 | kaldemar | speedtapir: delete the whole registration line from sip.conf and rewrite it by hand. maybe you have some icky non-printing character there that screws it up. |
| 09:31.30 | petern_ | right |
| 09:31.39 | speedtapir | schmdits: nothing when I run sip reload |
| 09:31.53 | petern_ | well, SAY NUMBER 103 is saying "one hundred three", not "one hundred and three" |
| 09:31.54 | schmidts | and what do you see with module reload chan_sip.so |
| 09:32.14 | speedtapir | kaldemar: i have a feeling yes. |
| 09:32.16 | speedtapir | let me do that |
| 09:32.18 | IsUp | speedtapir: 'core stop now' and start asterisk with 'asterisk -vvvvvvvvvvvgfc' and see whats going on. also as kaldermar said, delete WHOLE file as my opinion. write it from stratch |
| 09:33.09 | speedtapir | i think i'll type the whole thing |
| 09:33.20 | speedtapir | and see if things happen. |
| 09:33.38 | IsUp | ok |
| 09:34.25 | IsUp | kaldemar: i see many 'rasterisk r' in his process list. is that normal? |
| 09:34.47 | wdoekes2 | IsUp: those are your asterisk -r instances |
| 09:35.06 | IsUp | wdoekes2: yes i know but why he has 15x these instances? |
| 09:35.16 | schmidts | shouldnt be a problem at all |
| 09:35.26 | IsUp | okay |
| 09:35.44 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:36.11 | KNERD | why is asterisk printing "Huh? Child handler, but nobody there?" in my CLI console? |
| 09:37.15 | kaldemar | IsUp: no. |
| 09:38.02 | schmidts | KNERD strange message, comes from asterisk core itself maybe something went wrong with threads |
| 09:38.23 | petern_ | so... is it controllable? :S |
| 09:38.47 | kaldemar | petern_: very much doubt it, without modifying the source. |
| 09:39.24 | IsUp | KNERD: are you running Monitor, MixMonitor? when you get this output? |
| 09:39.35 | petern_ | and the wiki on voip-info.org says it includes 'and' :S |
| 09:39.47 | KNERD | not I am aware of..I am not even in the asterisk console |
| 09:40.11 | KNERD | it's going directly into BASH |
| 09:40.22 | kaldemar | petern_: voip-info.org is not that good a reference for anything. |
| 09:40.32 | IsUp | KNERD: k see logs |
| 09:41.06 | schmidts | KNERD this could happens when you start asterisk directly from your bash |
| 09:41.41 | schmidts | KNERD btw this message is just printed with: printf("Huh? Child handler, but nobody there?\n"); |
| 09:41.42 | schmidts | so it goes to STDOUT which is your bash |
| 09:42.10 | KNERD | ouch...okay let me look in logs |
| 09:42.16 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 09:42.20 | petern_ | apparently not :) |
| 09:42.23 | IsUp | schmidts: yes, you are right |
| 09:45.36 | speedtapir | kaldemar IsUp : I did write up the entire thing again but not much success as the problem still persists. now running asterisk -vvvvvvvvvvvgfc |
| 09:46.06 | petern_ | ok, it says 'and' with only with en_gb, i was doing other stuff with language to support different voice sets |
| 09:47.08 | schmidts | speedtapir please try module unload chan_sip.so and module_load chan_sip.so |
| 09:47.22 | speedtapir | http://pastebin.com/GBDd8mkY ---> |
| 09:47.31 | speedtapir | schmidts: did that...see above |
| 09:47.48 | speedtapir | for some reason it stops at Reloading SIP... |
| 09:48.30 | schmidts | speedtapir please try sipsak against sip-nv1.flowroute.com |
| 09:48.40 | schmidts | thats the server asterisk tries to talk too |
| 09:49.39 | speedtapir | ok |
| 09:51.58 | speedtapir | ah! |
| 09:51.58 | *** join/#asterisk gravin (~gravin@17.34.49.60.brf01-home.tm.net.my) |
| 09:52.49 | speedtapir | schmidts: http://pastebin.com/e1Myn4h9 |
| 09:52.54 | speedtapir | doesnt seem to work |
| 09:53.02 | speedtapir | 404 not found is what i get. |
| 09:54.18 | IsUp | speedtapir: upgrade to 1.8.8.0 - its my final decision :p |
| 09:54.28 | IsUp | speedtapir: and good luck |
| 09:55.17 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 09:55.17 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 09:55.47 | speedtapir | IsUp: I guess so. |
| 09:55.48 | speedtapir | thanks |
| 09:56.00 | speedtapir | do you mean downgrade or upgrade? |
| 09:56.41 | speedtapir | oh sorry. its upgrade !!1 |
| 09:56.45 | speedtapir | :/ |
| 09:56.50 | schmidts | speedtapir maybe its just your provider ;) |
| 09:57.13 | IsUp | schmidts: no i am %100 sure its not about with provider. |
| 09:59.49 | schmidts | speedtapir it would be interesting to do ngrep while you reload chan_sip to see if anything goes out |
| 10:02.05 | speedtapir | schmidts: let me try |
| 10:03.37 | speedtapir | schmidts |
| 10:03.43 | speedtapir | ngrep seems really strange :S |
| 10:03.47 | speedtapir | how do u use it? |
| 10:04.09 | schmidts | ngrep -t -W byline port 5060 |
| 10:04.52 | speedtapir | ngrep running |
| 10:05.00 | speedtapir | going to run asterisk |
| 10:05.30 | *** join/#asterisk gravin (~gravin@17.34.49.60.brf01-home.tm.net.my) |
| 10:06.41 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:12.18 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 10:13.25 | *** join/#asterisk irroot (~gregory@197.172.13.34) |
| 10:17.37 | Yourname` | Can anyone recommend a trick on a regular PSTN line where it's only one line coming into the PBX so I can squeeze out more channels? |
| 10:17.53 | Yourname` | I'm new to the regular PSTN concept, and used to the SIP concept with unlimited channels.. :( |
| 10:18.18 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 10:18.30 | WIMPy | Like a PRI? |
| 10:18.50 | Yourname` | WIMPy: No, just a regular line. Like a landline. |
| 10:19.01 | Yourname` | It comes into the ATCOM card. |
| 10:19.42 | WIMPy | It is what it is. |
| 10:22.13 | KNERD | I am looking for the jabber module..I no longer wanrt it to load. I have 1.8.7.1 |
| 10:22.16 | Yourname` | Ah, lol, so no tricks eh |
| 10:22.43 | Yourname` | WIMPy: Any suggestions renaming DAHDI/1-1 that shows up in the CLI/CDR to the actual number? |
| 10:23.42 | WIMPy | On current versions, you get the number in the channel name if it's available. |
| 10:23.55 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 10:24.30 | WIMPy | Where current must be like less than a year or so. |
| 10:25.17 | Yourname` | WIMPy: I guess we need to set it. I wish I asked that on FreePBX, heh since that's where I'll need to do it. |
| 10:38.20 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 10:44.03 | IsUp | okay schmidts kaldemar: i found FAIL on his sip.conf |
| 10:44.16 | IsUp | he is added [general] and [flowroute] under original sip.conf |
| 10:44.26 | IsUp | so it was double [general] |
| 10:50.34 | kaldemar | that sort of fuckups do happen when people don't provide what they are asked for. when i asked for the whole sip.conf, it got translated into something else... |
| 10:50.53 | IsUp | haha, anyways, its registered now but |
| 10:51.01 | IsUp | its a miracle to this asterisk works fine. |
| 10:51.04 | IsUp | loader.c:393 load_dynamic_module: Error loading module 'chan_iax2.so': /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_aes_set_encrypt_key |
| 10:51.05 | kaldemar | that's not funny. |
| 10:51.11 | IsUp | too many errors like this |
| 10:51.31 | schmidts | Isup you need ssl installed IMHO |
| 10:52.06 | IsUp | schmidts: its speedtapir's |
| 10:52.43 | KNERD | AGGG!!! How can i stop jabber from loading??? |
| 10:53.10 | IsUp | noload => res_jabber.so mod_jabber.so or whatever, in modules.conf |
| 10:53.17 | schmidts | isup ah ok :D |
| 10:53.29 | KNERD | IsUp: did that , been there |
| 10:54.12 | kaldemar | KNERD: pastebin... |
| 10:54.43 | KNERD | Want to permanently stop jabber and gtalk? http://www.voip-info.org/wiki/view/Asterisk+Google+Talk |
| 10:55.37 | kaldemar | KNERD: then noload chan_gtalk.so as well. |
| 10:56.03 | KNERD | kaldemar: it includes thast |
| 10:56.22 | KNERD | yet jabbers continues to spew out 1 million errors a second |
| 10:57.24 | kaldemar | KNERD: did you restart asterisk? if not, unload the modules by hand. |
| 10:57.50 | KNERD | i only reloaded |
| 10:58.13 | kaldemar | a reload is not enough for modules.conf changes. |
| 10:58.58 | KNERD | yes I now know..thanks///finally |
| 10:59.12 | KNERD | So much for no longer functioning google voice |
| 10:59.47 | KNERD | though I understand it still is fine with Free Switch |
| 11:08.25 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 11:15.18 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
| 11:20.36 | IsUp | ~book |
| 11:20.37 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 11:32.52 | KNERD | no longer in pdf..that sucks |
| 11:33.00 | KNERD | ~burn book |
| 11:33.01 | infobot | ACTION pours gasoline all over book, ignites the fire, and then enjoys some toasty marshmallows with the glorious blaze |
| 11:33.23 | IsUp | haha |
| 11:33.56 | WIMPy | likes that one |
| 11:34.10 | *** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 11:34.11 | krotos | hi all |
| 11:34.25 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 11:34.30 | IsUp | hi |
| 11:35.13 | *** join/#asterisk irroot (~gregory@197.172.123.230) |
| 11:36.45 | schmidts | KNERD you will get pdf if you buy it ;) |
| 11:37.51 | KNERD | it used to be free |
| 11:38.56 | schmidts | KNERD i guess it becomes free again when the next version comes out |
| 11:39.33 | KNERD | it used to be when the same version was out |
| 11:41.18 | schmidts | KNERD maybe wait for an hour or two when leifmadsen will join and ask him directly ;) |
| 11:41.42 | KNERD | yeah! he can pass out copies! |
| 11:41.55 | schmidts | sure if you pay him :D |
| 11:42.03 | KNERD | yes |
| 11:42.05 | krotos | i'm having a strange problemi wiht my asterisk box. I'm using 1.8.8.0 ( but the problem was present also in 1.8.7.X). The problem is that every week ( in a random day) my asterisk box become unaccessible. No pin, no ssh, no screen on VGA. I'm using a debian 6 with compiled asterisk ( no package). |
| 11:42.11 | KNERD | pay him compliments |
| 11:42.23 | krotos | the only way is to hard-reset the machine |
| 11:42.33 | IsUp | krotos: check /var/log/messages /var/log/syslog |
| 11:42.42 | WIMPy | krotos: Get a new one. |
| 11:42.47 | IsUp | krotos: this is a hardware/server issue |
| 11:42.53 | KNERD | sounds like a hardware issue |
| 11:42.59 | IsUp | hey WIMPy |
| 11:43.10 | krotos | IsUp: in log there aren't any trace of problem |
| 11:43.17 | krotos | i think so an hw issue |
| 11:43.33 | krotos | the asterisk box is on debian 6 running on VmWare ESXi |
| 11:44.02 | IsUp | krotos: did you catch any 'segfault' in logs? i am not a linux geek but maybe you can catch something in logs |
| 11:44.03 | krotos | before i had a debian 5, with ast 1.4 and the uptime was 1 year and 7 week :( |
| 11:44.29 | krotos | IsUp: no, there arent segfault in log.. |
| 11:44.35 | WIMPy | So does the host or the guest stall? |
| 11:44.42 | krotos | the guest stall.. |
| 11:44.53 | KNERD | machines do start to go bad over time |
| 11:45.11 | WIMPy | It does happen, yes. |
| 11:46.08 | krotos | KNERD: i dont' understand |
| 11:47.51 | krotos | oh..now i understand :( |
| 11:48.16 | krotos | someone of you use ast 1.8.8 on a debian 6? |
| 11:48.20 | *** join/#asterisk jacco_7564 (~Jacco@195-241-134-176.ip.telfort.nl) |
| 11:51.30 | schmidts | krotos yes |
| 11:54.32 | krotos | schmidts: have you got some problem, as i described previously? |
| 11:54.41 | schmidts | krotos nope ;) |
| 11:54.52 | krotos | on this box i have 5000 call per day |
| 11:54.52 | schmidts | it really sounds like a hardware issue |
| 11:55.28 | jacco_7564 | Hello everyone |
| 11:55.37 | krotos | i've thinked that the load of call/minute is too high for asterisk.. |
| 11:56.05 | krotos | but is not right.some people have more user registered and no problem |
| 11:58.07 | IsUp | 5000 calls on VmWare? |
| 11:58.57 | jacco_7564 | Anyone around who could help me with a problem? |
| 11:59.05 | schmidts | krotos how many users do you have? |
| 11:59.11 | WIMPy | ~ask |
| 11:59.11 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:00.22 | krotos | i'have 1000 Registered user |
| 12:00.46 | krotos | but we have also 2 asterisk box configured in type=peer |
| 12:00.55 | krotos | for our reseller |
| 12:01.36 | schmidts | krotos i have two servers with 4500 peers on each with around 20k calls per day running for nearly 4 weeks without a problem |
| 12:01.53 | schmidts | IsUp and these are running in openvz :P |
| 12:02.02 | jacco_7564 | I am using visual dialplan and i created a IVR menu with it. however, with a certain option callers are sent to a queue, and they hear the MOH but even though the assigned static agent is reachable, the phone wont ring and the call stay in the queue in an infinite loop, CPU useage goes up to 100% and the server crashes |
| 12:02.47 | krotos | schmidts: the call are terminated on pri (with a pri card) o directly in SIP with other provider? |
| 12:04.31 | krotos | i've also got a part of my dialplan ( migrated from 1.4 ) that use app_mysql |
| 12:04.38 | schmidts | one this two server they only do sip to the peers and to two other servers, on these other ones i have 14 PRIs connected but still running 1.2 there :D |
| 12:04.58 | krotos | lol, 1.2 :) |
| 12:05.16 | krotos | "untill it stop works, pleas dont touch!" |
| 12:05.18 | schmidts | still the most performing asterisk version out there |
| 12:05.40 | WIMPy | Did 1.2 support PRI? |
| 12:06.36 | schmidts | sure ;) with good old zaptel :D |
| 12:07.09 | schmidts | but krotos i also have another server running 1.8 and two Pri connected to it which does sip to SS7 |
| 12:07.32 | schmidts | 77332 calls processed |
| 12:07.32 | schmidts | System uptime: 1 week, 1 day, 3 hours, 47 minutes, 44 seconds |
| 12:07.34 | krotos | schmidts: you run your boxes directly on hardware machine or virtualized on (vmware / esxi) |
| 12:08.07 | schmidts | my hosted pbx are in openvz virtual container, the one with Pri are only hardware |
| 12:08.32 | schmidts | i dont trust virtual access to hardware resources |
| 12:08.33 | krotos | ok :( today i make a new fresh machine on a new server |
| 12:08.47 | *** join/#asterisk Tim_Toady (~fuzzy@195.74.247.28.dsl.dyn.forthnet.gr) |
| 12:08.51 | krotos | a dell 2950, and for backup i clone the old one here |
| 12:08.59 | krotos | i can think a solutions with carp / vrrp |
| 12:09.04 | krotos | if no ping, change address |
| 12:09.42 | krotos | schmidts: me too, i use PRI only for faxing. all of the other call are routed to sip provider |
| 12:11.40 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.2) |
| 12:12.49 | krotos | there is a new app in 1.8 that substitute the old app_mysql used on 1.4? |
| 12:12.55 | krotos | or i can continue using this? |
| 12:13.15 | schmidts | with 1.8 you can still use it but you should think about moving to odbc |
| 12:13.25 | schmidts | like i should do this too :D |
| 12:13.27 | *** join/#asterisk irroot (~gregory@197.172.70.173) |
| 12:14.38 | krotos | i query the mysql database only for having some info on the callerid( if has secretary up, captive status, callerid presentation..etc) |
| 12:14.52 | krotos | with moving to odbc you say using astdb? |
| 12:17.15 | schmidts | no astdb is the internal db of asterisk |
| 12:17.40 | schmidts | odbc is a database connector, you use this to connect to a database and do queries... |
| 12:18.14 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 12:18.18 | schmidts | but you dont need mysql behind a odbc connection you could also have postgres, oracle, even access if you are masochistic enough for it |
| 12:18.26 | bulkorok | func_odbc is my best friend! |
| 12:18.52 | schmidts | krotos how many mysql connections and queries do you do on a regular call? |
| 12:19.00 | krotos | on a regular call |
| 12:19.52 | krotos | for example, on ougoing call i made 1 connection and 1 query. For incoming call, is the same situation 1 connection and 1 query. |
| 12:20.27 | krotos | but with connection with our reseller type=peer i made 1 connection and 4 query |
| 12:20.32 | schmidts | cute :D |
| 12:20.33 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
| 12:20.46 | krotos | too much query? |
| 12:20.56 | schmidts | i dont have count how many connections and queries and fetch and ... i do but overall i have 50 mysql operations on a single incoming call |
| 12:21.43 | krotos | ok...so cant' be a problem on mysql |
| 12:21.44 | scalex000 | good morning, I have a question how can i specify the phone-context in sip |
| 12:22.02 | schmidts | ok 7 connects, 9 queries, 13 fetch and the rest is clear and disconnect, |
| 12:22.08 | schmidts | shouldnt be a problem at all no |
| 12:22.12 | scalex000 | when I make a debug a sip peer the packet not add this |
| 12:25.53 | jacco_7564 | ~ask I am using visual dialplan and i created a IVR menu with it. however, with a certain option callers are sent to a queue, and they hear the MOH but even though the assigned static agent is reachable, the phone wont ring and the call stay in the queue in an infinite loop, CPU useage goes up to 100% and the server crashes |
| 12:25.53 | infobot | jacco_7564: what are you talking about? |
| 12:26.37 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 12:29.32 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:29.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:29.38 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:29.38 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:33.11 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 12:34.22 | creativx | <3 FOP2 |
| 12:35.02 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
| 12:41.22 | jacco_7564 | Anyone around who could help me with a server crashing loop? |
| 12:41.48 | anonymouz666 | just get rid of safe_asterisk ;))) |
| 12:42.44 | kaldemar | jacco_7564: a caller being in a queue is not likely to crash the server. look at the CLI during a call with verbosity enabled and see what happens. |
| 12:49.07 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 12:49.17 | jacco_7564 | Can I paste the piece of the log that shows the loop here? |
| 12:50.43 | schmidts | jacco_7564 ~pb |
| 12:50.53 | schmidts | sorry i meant pastebin ;) |
| 12:50.55 | schmidts | ~pb |
| 12:50.55 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 12:52.41 | jacco_7564 | http://pastebin.com/5gmij2R5 |
| 12:54.14 | schmidts | jacco_7564 your dialplan is just wrong, you have a goto(1) in there which goes on top of this extension and start from the beginning |
| 12:54.27 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 12:54.34 | IsUp | jacco_7564: loop is normal, write a proper dialplan |
| 12:54.46 | scalex000 | hello everybody, I need help |
| 12:54.52 | IsUp | ~ask |
| 12:54.52 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:54.59 | jacco_7564 | what should be changed then? |
| 12:55.14 | schmidts | jacc0_7564 remove this goto(1) there |
| 12:55.22 | scalex000 | I need to know how to add in sip packet phone-context |
| 12:55.24 | jacco_7564 | I use visual dialplan to make my dialplan |
| 12:55.56 | schmidts | jacco_7564 then you should something else which generate a proper dialplan |
| 12:55.58 | kaldemar | jacco_7564: then don't use it if it produces unusable dialplans. |
| 12:56.08 | IsUp | scalex000: be more specific |
| 12:56.57 | jacco_7564 | is the goto(1) specified in extensions of in queues? |
| 12:57.03 | scalex000 | when i dial to another system I need to put a phone-context tag to identify private network or public |
| 12:57.17 | kaldemar | jacco_7564: extensions, it is a priority in an extension in your dialplan. |
| 12:57.38 | scalex000 | so, I was debugging the sip peer but I not see asterisk send phone-context tag |
| 12:57.41 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 12:58.09 | schmidts | scalex000 phone-context is normally a static config parameter, i guess you are looking for SipAddHeader |
| 12:58.15 | kaldemar | scalex000: you can't select the context in the originating system. what you can do is deliver some information in an X-header and use that in the context selection in the receiving end. |
| 12:58.44 | scalex000 | ok |
| 12:58.55 | scalex000 | sipaddhearder |
| 12:58.59 | scalex000 | let me try |
| 12:59.18 | IsUp | scalex000: SIPAddHeader(phone-context: blabla) |
| 12:59.42 | scalex000 | before dial the number right? |
| 12:59.52 | IsUp | scalex000: yes, before Dial |
| 12:59.57 | scalex000 | ok |
| 13:00.00 | scalex000 | thank you |
| 13:00.01 | jacco_7564 | I have quite a few extension files, custom, additional, vdp(this one is created by visual dialplan), where can i find the looping problem? |
| 13:00.02 | scalex000 | let me try |
| 13:00.39 | IsUp | jacco_7564: dont use visual dialplan if you want a working pbx. |
| 13:00.49 | schmidts | jacco_7564 grep from-queue -ri * |
| 13:01.02 | schmidts | and follow what IsUp said ;) |
| 13:01.43 | jacco_7564 | I am too unfamiliar with asterisk to write my own dialplan, everything is working as it should with vdp except for the queue |
| 13:02.48 | IsUp | jacco_7564: if you are unfamiliar with dialplan, what else? yuo should read book. asterisk = dialplan |
| 13:03.13 | kaldemar | scalex000: non-standard headers should be prefixed with X-, so use X-phone-context: context. you'll get the context value in the receiving end with ${SIP_HEADER(X-phone-context)} |
| 13:03.15 | IsUp | jacco_7564: just follow book and there are plenty docs on voip-info.org |
| 13:03.39 | scalex000 | kaldemar, thank you |
| 13:04.14 | kaldemar | jacco_7564: the file where you can find "[from-queue]" |
| 13:04.53 | jacco_7564 | well i got everything up and running. gotoiftimes, invallid options, menu and authentication. I can make calls to and from the system |
| 13:05.17 | jacco_7564 | But i am ofcourse still learning |
| 13:06.47 | IsUp | folks i have a real dumb question. is that possible to send/receive fax over SIP? ive never tried it |
| 13:09.29 | IsUp | we have zyxel com56k and other old equipments. i want to get rid of them |
| 13:10.03 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:10.51 | jacco_7564 | Here is my extensions_additional http://pastebin.com/PzkZ22rv |
| 13:17.00 | kaldemar | jacco_7564: and a huge pile of crap it is. you better ask in #freepbx how to modify it, people there know better how to not interfere with freepbx. |
| 13:18.48 | jacco_7564 | freepbx/elastix is bad? |
| 13:21.03 | kaldemar | jacco_7564: many people here prefer to use the config files directly. GUI's make some tasks easy, but only some. |
| 13:22.29 | jacco_7564 | Yeah...I can understand that point of view for people who are experienced asterisk users, but for newcomers like me |
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| 13:40.06 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 13:40.27 | scalex000 | kaldemar, how asterisk handle the URI MAP |
| 13:50.58 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
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| 13:52.51 | puzzled | hi |
| 13:57.32 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 13:59.08 | tompaw | Hello. |
| 14:00.25 | tompaw | I have this uber-simple dialplan: http://pastecode.com/jI - and my question is: how do I tweak it so the "h" extension is NOT called if an agent in the queue picked up the call? |
| 14:00.52 | tompaw | Basically, I have 2 curl calls - I want the top one to called if the caller disconnected before being picked up by an agent. |
| 14:00.57 | *** join/#asterisk PsuedoICE (~PsuedoICE@flanderscorp.com) |
| 14:01.02 | tompaw | And the bottom one to be called if an agent picked up. |
| 14:01.22 | tompaw | It works fine, the only problem is - even if an agent picks up the call, "h" exten is still executed. |
| 14:02.03 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 14:02.29 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:03.11 | kaldemar | tompaw: it doesn't matter if the call is picked up or not, h will get executed if it exists in the current context. |
| 14:04.28 | kaldemar | the problem is not how to prevent h from being executed, but how to let h known what happened before it. |
| 14:05.19 | *** join/#asterisk serafie (~erin@nat/digium/x-wcvaagkdkqmgyfln) |
| 14:05.34 | tompaw | well, the queue macro will always be executed before 'h', so I guess I can simply check in my db if that ${EXTEN} call was answered and only mark it as dropped if it wasn't... |
| 14:05.41 | tompaw | but it's not a "clean" solution |
| 14:06.26 | tompaw | kaldemar: is it possible to "let h know what happened" in the dialplan? |
| 14:06.51 | tompaw | hey, how about setting a global variable in the Queue Macro? |
| 14:07.03 | tompaw | and then checking it in "h"? |
| 14:08.48 | kaldemar | CDR(disposition) should be "NO ANSWER" unless the call was answered. |
| 14:09.21 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 14:09.36 | tompaw | kaldemar: http://pastecode.com/jJ |
| 14:09.39 | tompaw | how about that approach? |
| 14:09.49 | kaldemar | so h,1,ExecIf($["${CDR(disposition)}" = "ANSWERED"]?Set(...)) |
| 14:10.04 | tompaw | right... yours is way simpler :P |
| 14:11.00 | tompaw | wonder if his concept would work anyway. |
| 14:11.05 | *** part/#asterisk beaver_rrr (beaver@crabhost.org) |
| 14:11.17 | kaldemar | tompaw: your should work too. |
| 14:11.26 | tompaw | thanks mate! :-) |
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| 14:13.11 | PsuedoICE | We own a switchvox but it does not support the QSIG protocol. When connecting it to our phone system via T1 crossover, internal extensions caller ID is not passing through to the switchvox. It was explained that I could setup an asterisk gateway between the phone system and switchvox allowing the asterisk gateway to conenct physcailly via T1 crossover to the phone system with QSIG, then |
| 14:13.12 | PsuedoICE | SIP=>SIP from the gateway to the switchvox. Honestly, this is problaby the way it needs to be setup anyway if we want to expand to other facilities. However, I am having a hard time understanding exactly what I need to do to configure the gateway itself. Can someone point me to an article or how to for something this specific. I have not been able to find exactly what I needed from Google |
| 14:13.12 | PsuedoICE | searches, or do not fully understand that what I found was the actual answer. |
| 14:13.56 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 14:14.02 | scalex000 | kaldemar, how asterisk handle the URI MAP |
| 14:14.23 | kaldemar | scalex000: what "URI MAP"? |
| 14:14.44 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:15.10 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 14:15.30 | kaldemar | PsuedoICE: http://ofps.oreilly.com/titles/9780596517342/asterisk-OutsideConn.html |
| 14:15.39 | scalex000 | kaldemar, SIP URI MAP |
| 14:16.37 | scalex000 | kaldemar, I think nortel need to set the same SIP URI MAP to identify is private or public network |
| 14:16.53 | scalex000 | the only difference I saw is phone-context in invite |
| 14:17.28 | kaldemar | scalex000: DIY |
| 14:17.43 | PsuedoICE | kaldemar: I actually have that book on my desk. I'll re-read that section. |
| 14:18.17 | scalex000 | lol |
| 14:18.26 | kaldemar | scalex000: there is no built-in way to use. |
| 14:20.23 | kaldemar | scalex000: in other words, you need to grab what ever headers you want in dialplan and do decisions based on them. |
| 14:21.19 | scalex000 | let me paste something so you understand |
| 14:23.10 | kaldemar | i know what nortel means by URI map. |
| 14:25.43 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 14:25.52 | scalex000 | kaldemar, look this http://pastebin.com/nHh2Z2qA |
| 14:26.20 | scalex000 | kaldemar how to make the invite add phone-context |
| 14:30.28 | *** join/#asterisk jhirley (~chatzilla@adsl-108-130-32-197.mia.bellsouth.net) |
| 14:35.23 | *** join/#asterisk scalex000 (~chatzilla@186.6.0.239) |
| 14:38.15 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:39.15 | *** join/#asterisk corretico (~luis@190.211.94.6) |
| 14:42.59 | *** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk) |
| 14:44.32 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:47.37 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:49.37 | scalex000 | kaldemar, any idea |
| 14:50.04 | kaldemar | scalex000: you already added the header in the invite. did it in a ugly manner though, not using X-phone-context like you were told to. |
| 14:50.25 | scalex000 | :P |
| 14:50.38 | kaldemar | scalex000: now what do you want next? |
| 14:50.55 | kaldemar | scalex000: to handle that information in the other end? |
| 14:51.33 | scalex000 | no, In nortel they say in the tutorial to set a private netwoek the SIP URI MAP need to be the same |
| 14:52.18 | kaldemar | "SIP URI MAP" means very little outside nortel devices. |
| 14:52.43 | scalex000 | the problem is when I dial from asterisk to nortel, nortel think is from public network and not from private network even voip trunk |
| 14:54.19 | kaldemar | find out what the mapping really does to an invite. |
| 14:56.09 | scalex000 | well, to identify the network |
| 14:56.15 | scalex000 | I check a diagram |
| 14:56.35 | scalex000 | only, cdp and udp are consider privete network |
| 14:56.47 | scalex000 | unknow, private.unknow |
| 14:57.06 | scalex000 | they take like public network |
| 14:57.36 | scalex000 | when asterisk send the invite not send the phone-context |
| 14:57.37 | *** join/#asterisk Twitchnln (~Adium@c-98-242-79-16.hsd1.ga.comcast.net) |
| 14:58.06 | scalex000 | so nortel response phone-context=unknow@ipaddress |
| 14:58.18 | kaldemar | scalex000: you're speaking in terms that mean nothing in asterisk nor SIP. what does the feature do to the INVITE message? |
| 14:59.12 | scalex000 | :P |
| 14:59.22 | scalex000 | Im speaking about sip connection |
| 14:59.27 | scalex000 | INVITE message |
| 14:59.54 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 14:59.54 | kaldemar | does it add a header by the name "phone-context" and set its value as unknown, unknown.private, special.private, CDP, UDP, unknown.e164 |
| 14:59.59 | kaldemar | etc.? |
| 15:00.13 | carrar | Katty, install HELLO KITTY PBX PLEASE |
| 15:00.19 | Katty | yes plz. |
| 15:00.22 | Katty | with pink |
| 15:00.23 | Katty | kthx |
| 15:00.28 | kaldemar | Katty: sledgehammer |
| 15:00.48 | Katty | that might damage the prety case tho!! |
| 15:00.50 | Katty | unacceptabuhls |
| 15:01.27 | Katty | so i have a new mac |
| 15:01.30 | Katty | my first /ever/ |
| 15:01.41 | carrar | w00t |
| 15:01.43 | scalex000 | this sip uri map have nortel |
| 15:01.44 | kaldemar | in that case, two sledgehammers! |
| 15:01.48 | carrar | laptop or desktop? |
| 15:02.01 | Katty | it's a laptop. macbook pro |
| 15:02.06 | schmidts | Hey katty you should learn do not broke your pbx everyday |
| 15:02.06 | carrar | Nice |
| 15:02.08 | Katty | it vera shiny |
| 15:02.10 | carrar | got the SSD drive in it? |
| 15:02.15 | Katty | no |
| 15:02.20 | Katty | but i was thinking about getting one for it |
| 15:02.24 | carrar | yeah do |
| 15:02.25 | Katty | i got vmware fusion for it |
| 15:02.32 | carrar | it will be faster then a ferret! |
| 15:02.32 | Katty | so i have my winders and debian too |
| 15:02.38 | Katty | ferrets are not fast |
| 15:02.42 | Katty | they zzzz for 18hrs a day sir |
| 15:02.44 | carrar | hence faster |
| 15:02.52 | Katty | and bouncey bouncey the rest of teh day |
| 15:02.57 | carrar | I gots fusion on mine too |
| 15:03.08 | Katty | it's akward. |
| 15:03.08 | Katty | i feel like a user. |
| 15:03.18 | Katty | first i was all like CTRL A WHY U NO WORK |
| 15:03.22 | carrar | just hang out at coffee shops |
| 15:03.31 | Katty | and then i was like mac, y u no have eject cdrom button?! |
| 15:03.42 | Katty | amongst other issues |
| 15:03.58 | carrar | Install adium yet? |
| 15:03.59 | Katty | nah. i learn quickly |
| 15:04.02 | Katty | what is adium |
| 15:04.09 | carrar | You all in one chat client |
| 15:04.11 | carrar | your |
| 15:04.13 | Katty | oh |
| 15:04.16 | Katty | no i was using trillian |
| 15:04.17 | carrar | it's sexy |
| 15:04.22 | Katty | cause that's what i used on my winders. |
| 15:04.31 | Katty | but i don't mind trying it out |
| 15:04.42 | carrar | free yourself from windows thinkinfg |
| 15:04.56 | carrar | and be sure to install VLC |
| 15:04.58 | Katty | it's been nice |
| 15:05.00 | Katty | yes, i did that |
| 15:05.07 | Katty | i had to watch my lost girl movies |
| 15:05.17 | carrar | <PROTECTED> |
| 15:05.27 | Katty | and chrome, and spotify, and skype |
| 15:05.40 | Katty | but other than that, i've not done a whole lot other than just tinker with it and such |
| 15:06.02 | carrar | Libre Office |
| 15:06.14 | Katty | i have ms office |
| 15:06.27 | Katty | does libre office have an exchange connector thingy |
| 15:06.54 | carrar | not sure |
| 15:07.01 | carrar | I don't use exchange |
| 15:07.11 | Katty | mkay |
| 15:07.22 | carrar | I use PINE |
| 15:07.42 | carrar | Cause it's worked great for the last 20 years |
| 15:07.56 | carrar | and I've never gotten a virus from it :) |
| 15:11.00 | Katty | fancy |
| 15:12.20 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:13.24 | Katty | wondderrr boyyy |
| 15:13.29 | Katty | what is the secret of your powwerrr |
| 15:14.39 | The_Boy_Wonder | Katty: tree bark |
| 15:15.02 | joesuffceren | carrar: the last time I used PINE I was 13 years old, and it made me cry. Was trying to use it in conjunction with kermit to download an attachment or some such. :-) Ended up making a 45 minute drive with a floppy drive instead. |
| 15:15.17 | Katty | *hee* |
| 15:15.19 | Katty | hugs The_Boy_Wonder |
| 15:17.02 | The_Boy_Wonder | tree hugger! |
| 15:21.24 | *** join/#asterisk irroot (~gregory@197.106.127.41) |
| 15:21.51 | Katty | i'm having font problems this morning |
| 15:21.58 | Katty | the mac terminal just doesn't look right )= |
| 15:25.01 | *** join/#asterisk Twitchnln (~Adium@50-73-75-61-static.hfc.comcastbusiness.net) |
| 15:30.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:30.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:34.56 | *** join/#asterisk PsuedoICE (~PsuedoICE@flanderscorp.com) |
| 15:35.47 | PsuedoICE | I'm coming behind another administrator, and when I try to edit the chan_dahdi.conf and save the changes, after a reboot those changes get erased . . . . any ideas? |
| 15:35.51 | carrar | pine with attachements is hella simple |
| 15:38.49 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:44.31 | joesuffceren | carrar: it's been 14 years, and I was a clueless kid, then. I'm sure it wasn't pine's fault. That's just my only memory of it. |
| 15:46.18 | Katty | i'm allergic to pine. |
| 15:46.23 | Katty | or was it cedar..hmm |
| 15:46.57 | Qwell | notes to refrain from giving Katty a pine car air freshener |
| 15:47.25 | Katty | much appreciated, sir |
| 15:48.05 | elred_ | Hi |
| 15:48.10 | Katty | hi elred |
| 15:48.58 | elred_ | I am achieving a call and i receive 180+SDP, 2 seconds after i receive 183+SDP, then asterisk CANCEL the calls without user's manipulation |
| 15:49.03 | elred_ | hi Katty |
| 15:49.08 | elred_ | is this normal/known bug ? |
| 15:50.36 | *** join/#asterisk Twitchnln (~Adium@50-73-75-61-static.hfc.comcastbusiness.net) |
| 15:50.47 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.151.65) |
| 15:52.10 | Katty | i can't say i've seen it before. |
| 15:52.19 | Katty | but you might try asking laterish, when more people are awake |
| 15:52.35 | Qwell | Katty: Halp to be fixing it please. |
| 15:52.40 | schmidts | lazy americans sleep so long, day is nearly over here ;) |
| 15:52.49 | Katty | Qwell: i'll fix you in a minute. |
| 15:52.54 | Qwell | hawt |
| 15:53.20 | Katty | schmidts: it's true. i sleep forevers. |
| 15:53.26 | Katty | schmidts: if left alone, i can nap a good 10hrs. |
| 15:53.54 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:54.17 | schmidts | Katty same for me but kids are not often away from home, so even on sundays at 6 in the morning i have to get up ;) |
| 15:55.38 | Katty | while i have no children, i do have a doggy who insists i get up at 7 to let him out |
| 15:55.45 | Katty | so i feel your pain. kind of. |
| 15:56.02 | schmidts | hehe ;) |
| 15:59.11 | jacco_7564 | almost 5 oclock over here, workday over! |
| 15:59.36 | schmidts | same here |
| 15:59.50 | schmidts | cya all tomorrow, or later if i find another bug to work on :D |
| 16:00.34 | Katty | jacco_7564: sooo jealous. |
| 16:00.49 | jacco_7564 | Hehehe ive been in for 8 hours though! |
| 16:01.58 | Katty | you've done your time then |
| 16:02.00 | Katty | get outta here |
| 16:02.05 | Katty | leaf! |
| 16:02.21 | jacco_7564 | Ye waiting for my ride to come pick me up |
| 16:02.54 | Katty | ride, y u no come quicker?! |
| 16:03.01 | Katty | ...that's what she said? |
| 16:03.08 | jacco_7564 | hahaha omg |
| 16:03.08 | Katty | (that's NOT what she said) |
| 16:03.35 | jacco_7564 | lol |
| 16:04.05 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:04.23 | jacco_7564 | where are u from katty? |
| 16:04.26 | Katty | hi kavan |
| 16:04.36 | Katty | from the midwest |
| 16:04.46 | Katty | missouri, if you'd like something more specific. |
| 16:05.54 | Katty | also! |
| 16:06.06 | jacco_7564 | Id rather like something more pacific! |
| 16:06.06 | Katty | my mac does not seem to have an alt. so how do i alt 1,2,3 etc in irssi :< |
| 16:06.29 | Katty | orange you so funny. |
| 16:06.33 | jacco_7564 | You right click |
| 16:07.00 | jacco_7564 | Now im even more funierest |
| 16:07.10 | chuckf | Katty: see what happens when you move to a mac, nothing works right |
| 16:07.14 | chuckf | :) |
| 16:07.20 | leifmadsen | Katty: alt == option |
| 16:07.21 | Katty | hehehe |
| 16:07.26 | jacco_7564 | Haha my dad's a big mac fan |
| 16:07.30 | Katty | ™ <- what on earth is that |
| 16:07.38 | Katty | that is not alt 2 |
| 16:07.41 | leifmadsen | Katty: Trademark symbol |
| 16:07.44 | carrar | TM |
| 16:07.49 | carrar | Capital T |
| 16:07.50 | Katty | <PROTECTED> |
| 16:07.51 | carrar | Capital M |
| 16:07.56 | leifmadsen | shift+option == alt maybe then |
| 16:08.03 | Katty | puts mac on head, hopes for osmosis |
| 16:08.13 | Katty | no :< |
| 16:08.13 | leifmadsen | I have a macbook pro -- it runs windows xp |
| 16:08.18 | Katty | € |
| 16:08.18 | leifmadsen | native |
| 16:08.23 | carrar | haha |
| 16:08.24 | carrar | sad sad |
| 16:08.27 | Katty | i swear... |
| 16:08.32 | Katty | i am just going to open my debian vm |
| 16:08.34 | leifmadsen | I have osx tiger... it's useless |
| 16:08.38 | Katty | that will fix everything |
| 16:08.40 | leifmadsen | and I refused to pay for upgrades |
| 16:08.43 | carrar | I have ferrari, I put in VW Interior |
| 16:08.52 | jacco_7564 | debian, i wish i was that cool |
| 16:09.14 | Katty | brb |
| 16:09.41 | jacco_7564 | anyway, im outa here, cya! |
| 16:09.59 | Katty | ahhh |
| 16:10.03 | Katty | that's better |
| 16:10.33 | Katty | except the font is messed up |
| 16:10.34 | Katty | grrr |
| 16:11.26 | Katty | yay! |
| 16:11.47 | Katty | my alt works!!! *happydance* |
| 16:13.47 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 16:18.13 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 16:18.14 | tompaw | How can I access 'meetme list' data via Manager? |
| 16:19.27 | Katty | very carefully |
| 16:19.45 | Katty | but i've never done it before. you should probably ask someone besides me. |
| 16:22.12 | tompaw | I guess I'll have to use "Command" :/ |
| 16:22.35 | carrar | Meeeska |
| 16:22.36 | carrar | Mooska |
| 16:22.37 | carrar | Mouseketeer |
| 16:23.09 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 16:23.11 | carrar | tompaw, use the Mouskatools |
| 16:23.27 | carrar | Say, "Oh Tootles" |
| 16:23.47 | Katty | sayyy fuzzypickles!!! |
| 16:23.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:26.24 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 16:30.10 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 16:30.48 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 16:31.32 | *** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk) |
| 16:37.12 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 16:41.19 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:45.26 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 16:52.46 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 16:57.36 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 16:58.35 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 16:59.42 | ChannelZ | blinks |
| 16:59.59 | *** join/#asterisk gravin (~gravin@17.34.49.60.brf01-home.tm.net.my) |
| 17:00.24 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:02.42 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 17:05.31 | *** join/#asterisk woleium (~woleium@email.tecglobal.net) |
| 17:06.55 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-202.mobile.uci.edu) |
| 17:09.57 | *** join/#asterisk CyfordTechnologi (~allen@c-24-98-175-41.hsd1.ga.comcast.net) |
| 17:10.38 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 17:11.08 | p3nguin | Why do people dial and then abandon the call? I hear voices from the surrounding cubes, but the person who called never says a word. |
| 17:11.45 | Katty | maybe they're skeered of the accent? |
| 17:12.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:12.15 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:13.24 | anonymouz666 | anyone know if would be possible to a child channel export a var to parent channel? |
| 17:13.36 | anonymouz666 | or the parent channel import the var from child channel? |
| 17:20.13 | *** join/#asterisk irroot (~gregory@197.171.12.1) |
| 17:21.00 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 17:21.54 | p3nguin | Scared of Allison? No waiz! |
| 17:22.22 | p3nguin | Maybe they don't have a keypad, so they can't enter any digits to continue the call. |
| 17:27.29 | *** join/#asterisk bshipman (~bshipman@fw1.safedataisp.net) |
| 17:27.53 | bshipman | good afternoon all, looking for a spot of advice on dahdi and wanpipe |
| 17:28.07 | Katty | i advise..chocolate |
| 17:28.08 | Katty | and caffeine |
| 17:29.45 | bshipman | check on caffeine, thinking about a mexican lunch though perhaps chocolate thereafter |
| 17:33.10 | p3nguin | That may or may not help you solve the issue you're having. |
| 17:34.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:34.40 | bshipman | i can't create any more telephony related problems at least |
| 17:35.15 | bshipman | i have installed dahdi 2.5.x and it doesn't seem to be playing nice w/ the current versions of wanpipe 3.5.x |
| 17:36.20 | bshipman | there is no clear uninstall script for dahdi to downgrade to the 2.2 series and i would like to avoid hosing the system just by willy nilly rm -rf |
| 17:39.33 | p3nguin | Maybe next time you'll remember to use your package manager. |
| 17:42.30 | Katty | mmm mexican lunch |
| 17:42.46 | Katty | i had pizza tho |
| 17:42.50 | Katty | was nom. |
| 17:42.56 | tzafrir | bshipman, just install on top of the new one |
| 17:43.07 | tzafrir | Though why would you go as far back as 2.2? |
| 17:43.49 | tzafrir | wonders if Sangoma builds with 2.6.0(-rc) |
| 17:44.23 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 17:44.36 | IsUp | hi |
| 17:44.53 | RypPn | tzafrir yes it does :) |
| 17:47.20 | tzafrir | Any simple way to add/remove voicemail on a mailbox from the command-line? |
| 17:47.35 | tzafrir | http://tzafrir.org.il/~tzafrir/test_voicemail does not work for some reason I still fail to understand |
| 17:47.52 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-202.mobile.uci.edu) |
| 17:48.17 | tzafrir | (I just need Asterisk to signal a voicemail) |
| 17:49.11 | bshipman | p3nguin, it was a source build - i didn't see a yum-based route |
| 17:50.38 | p3nguin | You should have built your own package from the source. It is an extremely simple procedure. |
| 17:50.43 | p3nguin | ~pacman |
| 17:50.43 | infobot | Using your operating system's package manager is the best way to manage software on your system. The package manager allows you to install/remove/change software safely and completely (in most cases). Using the package manager prevents software from being installed with potentially no way to control it, making it very difficult to remove software in many instances. |
| 17:50.51 | bshipman | tzafrir, awesome thx ... I was looking to go to 2.2 because of a forum post from one of the sangoma techs to a person with roughly my same problem indicating an incompatability with the 3.4.x series of wanpipe |
| 17:52.33 | *** join/#asterisk kikohnl (~kotis@72.253.138.39) |
| 17:52.57 | Katty | pulls her hair out |
| 17:53.13 | Katty | pouts |
| 17:54.48 | Katty | what program for vnc are you guys using on mac |
| 17:55.06 | bshipman | JollyFastVNC |
| 17:55.12 | RypPn | realvnc Katty |
| 17:55.30 | bshipman | caveat ... its got a weird fisheye feature you have to turn all the way off |
| 17:55.32 | RypPn | just download the enterprise version and select viewer only at the customise bit |
| 17:55.39 | Katty | k |
| 17:56.45 | corretico | hola, hi |
| 17:58.49 | p3nguin | When I see a warning saying decodeMP3: Junk at the beginning of frame, what do I need to do to my mp3 to get the junk out of it so I can stop seeing the warnings all the time? |
| 17:59.21 | corretico | me again... I need assitance with my asterisk and avaya... using sip trunk. |
| 18:01.08 | p3nguin | ~ask |
| 18:01.09 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:01.30 | p3nguin | also... |
| 18:01.32 | p3nguin | ~book |
| 18:01.32 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:01.52 | *** part/#asterisk resno (~resno@unaffiliated/resno) |
| 18:06.53 | Katty | realvnc is doin the job |
| 18:08.38 | Katty | RypPn: ty |
| 18:09.08 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 18:12.18 | RypPn | Katty nps :) |
| 18:17.53 | *** join/#asterisk CrossWired (~chatzilla@65.210.186.34) |
| 18:18.11 | *** join/#asterisk l2trace99 (~jr@rrcs-71-43-104-238.se.biz.rr.com) |
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| 18:26.28 | corretico | when I try to use a sip trunk with the avaya, i get the following error: -- SIP/to_promovil-0c552d28 is circuit-busy |
| 18:29.06 | [TK]D-Fender | corretico, Meaningless error. |
| 18:29.23 | [TK]D-Fender | corretico, Like yesterday you should be showing us the complete call with SIP debug enabled |
| 18:38.55 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
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| 18:48.36 | bshipman | running dahdi_cfg -vvv |
| 18:48.53 | bshipman | generates: a notice about the config file |
| 18:49.10 | bshipman | and then: line 0: Unable to open master devce '/dev/dahdi/ctl' |
| 18:49.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:49.24 | bshipman | lspci shows 2 sangoma cards present |
| 18:49.28 | bshipman | any thoughts? |
| 18:51.22 | corretico | <[TK]D-Fender> http://pastebin.com/DgLh07Q2 |
| 18:57.13 | *** join/#asterisk coreyf1513 (~cfarrell@75-130-93-234.dhcp.wlmn.ct.charter.com) |
| 18:57.23 | Katty | server commencing liftover proceedure!!!! whrrrrrrrrrrrrrrrrrrrrrrrrrrr |
| 18:57.44 | Qwell | lifts Katty |
| 18:57.46 | Qwell | moves 3 ft |
| 18:57.49 | Qwell | sets Katty down |
| 18:58.36 | [TK]D-Fender | corretico, SIP/2.0 488 Not Acceptable Here |
| 18:58.53 | [TK]D-Fender | corretico, typcailly a codec or IP refusal. You are offering ALAW & GSM |
| 18:59.10 | [TK]D-Fender | corretico, go verify what the BCM allows |
| 18:59.55 | Katty | :< |
| 19:00.21 | Katty | way to make fun of my typo |
| 19:03.43 | coreyf1513 | does any version of asterisk support authenticated sip notify? Trying to configure NOTIFY Event: resync for Cisco SPA525G's, but they require this to be authenticated. BE C.3.2.1 isn't replying to the 401 challenge for sip notify. |
| 19:08.13 | *** join/#asterisk cyford (~allen@c-24-98-175-41.hsd1.ga.comcast.net) |
| 19:08.44 | corretico | <[TK]D-Fender>let me check |
| 19:17.39 | p3nguin | Why doesn't queue strategy linear work like it is stated to work? It doesn't seem to matter what order I put a pair of members, the one I want to be second is always picked first, leaving the one I want to be first not selected. |
| 19:20.32 | Katty | dear server, shhhhh |
| 19:20.38 | Katty | tries to comfort whiny server |
| 19:21.01 | Katty | 1u server, y u no be quiet?! |
| 19:21.26 | Katty | tries applying blanket >.< |
| 19:22.13 | [TK]D-Fender | Katty, Small runs hot, hot runs loud. |
| 19:24.34 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 19:24.45 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 19:24.49 | [TK]D-Fender | eek |
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| 19:34.50 | _Corey_ | p3nguin: I don't think linear ever worked right for what it's worth |
| 19:35.45 | p3nguin | I've reloaded it a few times and now the order is the way I wanted it, but I don't know if it will remain consistent. |
| 19:36.03 | p3nguin | The next problem I have is with timeout and timeoutpriority. |
| 19:36.16 | _Corey_ | I had a customer try to use it back in the early 1.4 days and I remember testing it a lot and finding the results totally unpredictable |
| 19:36.42 | _Corey_ | There was a patch out there that basically created a linear mode that worked but I don't know if it's still out there |
| 19:37.10 | _Corey_ | That said, I don't know whether linear got re-written at some point in between, so what I'm saying could be past its expiration date |
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| 19:38.52 | _Corey_ | My firefox is having fits today... |
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| 19:41.01 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:44.39 | p3nguin | Why is there no description/usage of the "retry" parameter in queues.conf? |
| 19:47.15 | pdtpatrick1 | Question .. i know you need to have dahdi and libpri installed for timing purposes. But you shouldn't have to have chan_dahdi.so loaded right? |
| 19:49.45 | pabelanger | pdtpatrick1: right, you just need to load res_timing_dahdi |
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| 19:59.02 | krotos | hi all |
| 20:01.00 | krotos | i usually had cdr over mysql using cdr_mysql module. Now i want to migrate and use cdr_adaptive_odbc |
| 20:01.17 | krotos | i've got to configure cdr_odbc and cdr_adaptive_odbc too? |
| 20:01.20 | krotos | (.conf file) |
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| 20:07.40 | p3nguin | Since there is no description or usage information for 'retry' in queues.conf comments... is the value a number of retries or a timeout before retrying? |
| 20:08.08 | [TK]D-Fender | inter-dial timeout IIRC |
| 20:09.06 | pabelanger | krotos: no, one or the other. |
| 20:09.23 | p3nguin | I'll set it to a ridiculous number and see how it behaves. |
| 20:10.03 | krotos | pabelanger: and define the main dsn in odbc.ini, right? |
| 20:14.15 | pabelanger | krotos: cannot remember. Been a while since I've done ODBC and CDRs |
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| 20:57.30 | *** join/#asterisk descention (~scott@c-71-58-188-66.hsd1.pa.comcast.net) |
| 20:57.35 | descention | hey all |
| 20:58.07 | descention | I'm just going to idle in here while I contemplate whether or not to install asterisk for fun. |
| 21:02.21 | _Corey_ | descention: fun and profit to be had by all... |
| 21:03.59 | descention | fun I hope for, profit I doubt. Just considering using asterisk as a personal PBX instead of pbxes.org for sipdroid. I have to figure out how to set it all up before I start installing everything I find though. |
| 21:10.28 | descention | yay for long projects |
| 21:10.53 | leifmadsen | infobot: tell descention about thebook |
| 21:13.08 | descention | leifmadsen: thank you |
| 21:13.26 | *** part/#asterisk mnicholson (~mnicholso@nat/digium/x-qqfbqebjdvsiblrl) |
| 21:16.05 | *** join/#asterisk cusco_ (~trilili@88.157.128.26) |
| 21:16.06 | cusco_ | hi |
| 21:16.34 | cusco_ | using chan gtalk, can I some how read chat text and set it as variable in dialplan? even after the call is established? |
| 21:18.32 | akrohn | is gtalk an xmmp protocol? |
| 21:18.42 | akrohn | xmpp* |
| 21:18.49 | jpsharp | Yes it is. |
| 21:19.10 | akrohn | seems that you could use the Jabber support built into asterisk then, cusco_ |
| 21:19.11 | cusco_ | yes |
| 21:19.22 | cusco_ | that is what chan_gtalk already uses |
| 21:19.43 | cusco_ | what function or app should I use to read chat text? |
| 21:20.26 | *** part/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 21:21.01 | cusco_ | with jabber set debug on |
| 21:21.02 | akrohn | I have no experience in xmpp with asterisk, but .... "JABBER_RECEIVE waits (up to X seconds) for a XMPP message and returns its content. Used along with JabberSend" |
| 21:21.05 | cusco_ | I can see the chat text in cli |
| 21:21.24 | cusco_ | ok let me read on that, thanks |
| 21:21.38 | akrohn | np =) |
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| 21:40.40 | cusco_ | akrohn: I only have JabberJoin JabberLeave JabberSend JabberSendGroup JabberStatus |
| 21:40.44 | cusco_ | no jabber recieve |
| 21:41.14 | akrohn | =/ |
| 21:50.34 | Qwell | cusco_: It's new. Upgrade! |
| 21:50.55 | cusco_ | hu? in what version? |
| 21:51.12 | cusco_ | im using 1.8.7.1 |
| 21:53.37 | cusco_ | I was using debian packages.. got tired of compiling |
| 21:53.37 | cusco_ | lol |
| 21:53.52 | cusco_ | anyway its just a test box |
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| 21:54.07 | cusco_ | I will see that later then.. thanks Qwell |
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| 22:44.55 | pdtpatrick1 | Question - im trying to understand sip show channelstats |
| 22:45.01 | pdtpatrick1 | http://paste.pocoo.org/show/NEAGVzOM1fk4kMpBFqAm/ |
| 22:45.39 | pdtpatrick1 | the packet lost is pretty high. HOw can one further troubleshoot? |
| 22:46.58 | phix | hi gang! |
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| 22:48.27 | krotos | exit |
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| 23:03.44 | jpsharp | Does DAHDI still support the old X100P cards? |
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