| 00:09.44 | *** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor) |
| 00:12.47 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 00:14.07 | sarthor | Hi, there is a hotel, I want to install asterisk server there, there are 110 rooms, already normal telephone sets installed in each room, " http://www.pinongroup.net/admin/image/product/KX-TS500MX_W.jpg ", what will be the cheap and best solution to make these normal telephone sets work with asterisk server. |
| 00:25.33 | [TK]D-Fender | 4 gateways like a Media5 4124 or an AudioCodes MP-124 |
| 00:25.39 | [TK]D-Fender | 5 actually |
| 00:25.47 | [TK]D-Fender | @ 24 ports each |
| 00:28.13 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 00:28.13 | *** mode/#asterisk [+o ChanServ] by lindbohm.freenode.net |
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| 02:07.36 | nsgn | odd question. i'm in a situation where i may need to pipe some audio very low latency over a high speed data network. the audio is music, however. SIP is well suited for my low latency needs, but i'm curious what codecs/hardware could be used that can handle the full spectrum of music, not just the narrow bands of voice |
| 02:26.50 | PCS | Q: can someone recommend a lowcost VPS for hosting asterisk? |
| 02:32.26 | nsgn | hmm..seems g722 can suit my needs. now the question is this: what IP phones out there can i take a line level input and output to/fron? |
| 02:32.28 | nsgn | *from |
| 02:41.55 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:49.40 | jpsharp | PCS: I've done it on linode as well as Amazon EC2. |
| 02:50.24 | PCS | jpsharp: thanks. |
| 02:50.37 | jpsharp | Did you find a consultant to help you earlier? |
| 02:51.15 | PCS | jpsharp: not yet. |
| 02:51.54 | jpsharp | Then allow me to introduce myself :) |
| 02:52.05 | PCS | u helped me before jpsharp. :) |
| 02:52.45 | jpsharp | oh. Well, then I can probably help you again. |
| 02:55.59 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 02:58.46 | jpsharp | nsgn: I've never seen an IP phone with line in/out, only headset in & out. YOu might end up just using a PeeCee. |
| 02:59.21 | nsgn | jpsharp, yeah..its just for a situation where a PC would be less than reliable. an IP phone with autoanswer configured would be rock solid |
| 03:04.06 | jpsharp | Maybe an embedded PC or fanless via board booted from flash. |
| 03:05.52 | nsgn | jpsharp, hm, ok. wonder if a raspberry pi would be sufficient. i've got a few |
| 03:06.14 | jpsharp | If you can somehow convince the codec to run in the onboard DSP. |
| 03:06.41 | nsgn | jpsharp, would you happen to know why MOH doesn't seem to play in g722 even if i'm connected to asterisk with a pure g722 phone and can confirm the traffic flow is g722. does MOH recompress in something lower? |
| 03:06.54 | nsgn | is there a way to override? calls sound nice but music sounds kindof...not |
| 03:09.37 | jpsharp | Not off the top of my head. I've not used G722. |
| 03:10.54 | nsgn | hmm |
| 03:12.06 | jpsharp | Though MOH is going to sound like crap since it originates as 8-bit 8Khz. |
| 03:14.00 | nsgn | jpsharp, what do you mean? |
| 03:14.23 | nsgn | i'm starting to think asterisk can play nice quality MOH but that when you use some utility like freepbx to prep the file it may be preset to crush it |
| 03:23.00 | jpsharp | Nevermind, I was thinking of something different. |
| 03:42.11 | nsgn | are there any softphones out there that can do siren14? |
| 03:44.40 | jpsharp | Seems that Zoiper can do it. |
| 03:44.49 | nsgn | hm. never heard of it. i'll try it. thanks |
| 03:46.00 | jpsharp | I use the basic version. Doesn't have all the fancy licensed codecs, but it works well. |
| 03:56.51 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:56.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:57.31 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 04:00.48 | *** join/#asterisk hiyo (~Puppy@unaffiliated/hiyo) |
| 04:01.47 | hiyo | what is asterisk all about (in layman terms |
| 04:01.50 | hiyo | ) |
| 04:02.12 | jpsharp | basically a computerized telephone switch. |
| 04:02.39 | jpsharp | That can connect VOIP, analogue, and/or digital telephone circuits together. |
| 04:02.40 | hiyo | what does that mean? |
| 04:02.47 | hiyo | oh |
| 04:03.16 | carrar | Please read this |
| 04:03.18 | carrar | ~book |
| 04:03.18 | 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 |
| 04:04.13 | *** part/#asterisk hiyo (~Puppy@unaffiliated/hiyo) |
| 04:04.50 | jpsharp | Well now you scared him off. |
| 04:04.51 | jpsharp | or her. |
| 04:08.07 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 04:19.36 | nsgn | jpsharp, oh snap, the basic version doesn't have siren14? |
| 04:19.42 | nsgn | i just installed it and can't seem to find it :D |
| 04:24.06 | jpsharp | Nope. Gotta buy it since they paid money to license it. |
| 04:28.24 | nsgn | i was hoping they had a trial. i've never even heard siren codec. i kindof want to see if it works for my needs before licensing |
| 04:34.39 | nsgn | i didn't know my momma was on irc |
| 04:57.27 | *** join/#asterisk nveid (~nveid@184.91.24.135) |
| 04:57.41 | nveid | What siptrunk providers support video? |
| 05:00.28 | jpsharp | I don't think you'll find any, since they're mostly PSTN->SIP and therefore no video. |
| 05:13.15 | nveid | So how are video calls accomplished with VOIP? |
| 05:21.29 | jpsharp | Private servers that everyone registers to or connecting to servers directly with something like an SRV record. |
| 05:22.46 | jpsharp | No need for a "provider" |
| 05:23.36 | jpsharp | klmnp;km"{} |
| 05:23.37 | jpsharp | PKjkmopjm |
| 05:23.50 | jpsharp | my cat says hi. |
| 05:31.28 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:32.39 | SeRi | lol |
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| 06:12.52 | *** join/#asterisk ziz212 (~chatzilla@203.115.9.110) |
| 06:12.59 | ziz212 | hi, |
| 06:16.03 | nveid | jpsharp: whats one of these servers that everyone connects to? |
| 06:16.56 | ziz212 | hi, my pbx giving erros " Got SIP response 400 "Bad Request" back from 109.68.162.233:5060" |
| 06:17.25 | ziz212 | why is that? pls help? |
| 06:34.38 | ziz212 | hello some body out there????????? |
| 06:58.15 | atan | ziz212, there is likely a body out there somewhere. This channel gets quiet at times :) |
| 06:58.36 | ziz212 | ok |
| 06:58.44 | ziz212 | sorry for using that word |
| 06:58.52 | atan | No harm done. |
| 06:59.25 | atan | With regard to your post I would guess you're trying to register something that isn't connecting correctly. Perhaps a bad user, secret, or something like that. |
| 06:59.29 | ziz212 | so I am getting "Got SIP response 400 "Bad Request" in dialing out through sip soft phone |
| 06:59.46 | ziz212 | yes it might be |
| 06:59.56 | atan | Is the SIP phone itself connected to asterisk? `sip show peers` |
| 07:00.04 | ziz212 | yes |
| 07:01.13 | atan | If you put something simple in your dialplan for it to hit, does it get that far? |
| 07:02.24 | ziz212 | yes |
| 07:02.33 | ziz212 | that is there |
| 07:02.47 | ziz212 | pls wait I will copy the error |
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| 07:11.45 | *** mode/#asterisk [+o Qwell] by lindbohm.freenode.net |
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| 07:12.52 | ziz212 | can I give you a teamviewer out to see the error |
| 07:12.59 | ziz212 | still I cant get it |
| 07:13.47 | ziz212 | it says "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 127" |
| 07:19.44 | ziz212 | atan: are you there |
| 07:20.05 | atan | I am here off and on, I just happened to look back here now |
| 07:21.48 | atan | Working on a few database projects which has me a bit distracted :-) |
| 07:22.14 | ziz212 | ok |
| 07:22.20 | ziz212 | so can you pls help me |
| 07:22.34 | ziz212 | can i give you teamviwer to see |
| 07:22.52 | ziz212 | because i am in a bit back to successs |
| 07:23.18 | ziz212 | step behind the winning line |
| 07:23.31 | ziz212 | I ve dont all ... |
| 07:23.40 | ziz212 | but there is an issue |
| 07:24.03 | atan | From what I can tell so far it sounds like your dialplan is trying to send your call but you're not formatting it correctly or something so it's returning the bad request |
| 07:26.35 | ziz212 | yes as a help can you pls see my configs |
| 07:26.47 | ziz212 | i will give you a teamviewer |
| 07:26.51 | ziz212 | session |
| 07:26.52 | *** join/#asterisk Matthias (~Matthias@2001:15c0:670f:ffff::2) |
| 07:41.44 | *** join/#asterisk Marquel (~Marquel@unaffiliated/marquel) |
| 08:16.14 | WIMPy | ziz212: What are you trying to dial? |
| 08:19.43 | ziz212 | oh WIMPy : I am trying to up my sip truk and dial out .. it is not working |
| 08:19.51 | ziz212 | trunk is up |
| 08:20.03 | ziz212 | sip phones are registerd |
| 08:20.06 | ziz212 | but not working |
| 08:22.02 | ziz212 | some help needed |
| 08:22.12 | WIMPy | Oh, well, I guess you have to look at sip debug to find out more. There's a good chance that you find a clear text description of what's going on there. |
| 08:23.06 | ziz212 | yes done .. found that "Got SIP response 400 "Bad Request" back from 109.68.162.233:5060" |
| 08:23.53 | WIMPy | Not that helpfull. |
| 09:16.59 | ziz212 | WIMPy: ok all the details are there http://pastebin.com/jRzziXH9 |
| 09:17.54 | ziz212 | pls help me |
| 09:24.31 | ChannelZ | The call is being rejected by your ITSP (or whatever 'mydivert-out' is) |
| 09:24.55 | ChannelZ | Probably they don't like the way the number is formatted |
| 09:25.44 | ChannelZ | What country are you in? |
| 09:29.24 | ChannelZ | (either that or they just don't like who you are.. they don't ask you for auth at all) |
| 09:30.10 | ChannelZ | your From seems a little odd |
| 09:35.53 | ziz212 | Thanks ChannelZ: is it something related to caller id and dialling no bit patten or no of digits? |
| 10:08.11 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 10:08.11 | *** mode/#asterisk [+o ChanServ] by lindbohm.freenode.net |
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| 10:41.52 | *** join/#asterisk Papierkorb (~papierkor@i577A2B8E.versanet.de) |
| 10:43.47 | Papierkorb | Hi. Recently found out about Asterisk. Sounds amazing. But I'm wondering if it is possible to use a (old) modem to make and answer calls? |
| 10:48.05 | ectospasm | no |
| 10:48.16 | ectospasm | modems do not transmit voice |
| 10:48.55 | ectospasm | you don't necessarily need special hardware to make and answer calls, though |
| 10:49.07 | ectospasm | ...just an appropriate VoIP provider |
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| 11:22.43 | *** part/#asterisk Papierkorb (~papierkor@i577A2B8E.versanet.de) |
| 11:25.39 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
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| 12:37.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:55.40 | atan | hah wouldn't that be sweet if we could just use nice 56k modems for a PSTN :D |
| 12:55.54 | atan | Those must be so cheap these days |
| 13:30.53 | gusto | well, i had one 33,6k modem that did telephone as well |
| 13:31.05 | gusto | but that one is gone |
| 13:31.21 | gusto | and wouldnt matter nowadays though, because it was ISA and not PCI |
| 13:32.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-84-111-20-159.red.bezeqint.net) |
| 13:35.40 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
| 13:36.28 | robl^ | hey everyone. is svn.asterisk.org having problems? I can't seem to do an update and not sure if it's my ISP or if its a server issue |
| 13:37.12 | Deeewayne | robl^, I was just going to ask the same question |
| 13:39.26 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 13:39.35 | robl^ | Deeewayne: thanks for the confirmation. at least I know its not just me. |
| 13:53.01 | *** join/#asterisk aberrios (~Lucia@psiclik.plus.com) |
| 13:54.48 | aberrios | data centre having trouble again? |
| 13:55.13 | robl^ | aberrios: appears so. svn issues? |
| 13:55.24 | aberrios | yup |
| 13:55.40 | robl^ | you are the third person to mention it in the past 20 minutes |
| 13:56.00 | aberrios | was the same last week |
| 13:56.34 | robl^ | ..and that's a lot for a sunday morning in the US (Digium's HQ) |
| 13:57.14 | robl^ | server is running. I can ping it. but it seems that the svn service is having issue |
| 13:57.26 | *** join/#asterisk AviMarcus (~avi@bzq-79-182-222-15.red.bezeqint.net) |
| 13:57.29 | AviMarcus | Hey. Anyone here live in canada? have a quick Q |
| 14:09.00 | aberrios | hmm its b0rked |
| 14:11.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 14:19.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:20.09 | *** join/#asterisk benlangfeld (~Adium@unaffiliated/benlangfeld) |
| 14:20.23 | benlangfeld | Is there anyone around who can kick packages.asterisk.org back into life? |
| 14:20.39 | aberrios | i would if i could |
| 14:20.55 | aberrios | waiting on it myself |
| 14:21.02 | benlangfeld | Who is responsible for it? |
| 14:23.29 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 14:24.08 | [TK]D-Fender | My guess would be "Digium". |
| 14:24.21 | benlangfeld | Yeah, that's a little broad :) |
| 14:25.06 | benlangfeld | Unfortunately our best contact there just departed. |
| 14:32.18 | aberrios | are there any mirrors about for packages and svn>? |
| 14:38.52 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 14:40.09 | [TK]D-Fender | Qwell: PING |
| 14:50.17 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 14:55.45 | aberrios | pokes the DC |
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| 15:11.24 | aberrios | i take it someone over at digium must be aware of the problems... |
| 15:15.38 | benlangfeld | who knows |
| 15:17.16 | benlangfeld | I wouldn't necessarily count on it |
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| 15:28.16 | atan | On a Sunday :D I'm sure they're itching to get to work to fix it :D |
| 15:34.56 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 15:41.52 | robl^ | what?!?! they LEAVE work? I thought they were all chained to their desk 24/7 and were just fed pizza and cola 3x a day. |
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| 16:21.00 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 16:21.01 | [sr] | hi WIMPy |
| 16:21.06 | [sr] | hi all |
| 16:55.40 | Marquel | morning |
| 16:56.08 | Marquel | is there an option to find all database keys below <family>/<key>/ and iterate over them? |
| 16:57.28 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:57.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 17:23.15 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 17:23.48 | gusto | i am wondering, that for all topics there are channels on freenode, but for hacking of women ... there should be a channel ##women :-) |
| 17:24.25 | gusto | and #asterisk would be even more popular, because the only thing women like more than television is telephone |
| 17:28.35 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:30.55 | gusto | tsss. there is such channel |
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| 18:33.35 | SeRi | test |
| 18:38.56 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:46.09 | Unchained | test ok |
| 18:53.28 | *** join/#asterisk apurvtwr (8002d2cb@gateway/web/freenode/ip.128.2.210.203) |
| 18:55.47 | apurvtwr | hey jpsharp: you there? |
| 18:55.51 | *** join/#asterisk k610 (~Instantbi@host-85-201-12-229.brutele.be) |
| 18:58.38 | apurvtwr | Hi everyone, I am trying to connected asterisk with google voice client. I was able to do that and it was working fine until my computer crashed yesterday and now it's giving an error |
| 18:58.52 | apurvtwr | ERROR[3577]: res_jabber.c:3127 aji_initialize: JABBER ERROR: No Connection |
| 18:59.00 | apurvtwr | does anyone know how to fix that? |
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| 19:21.29 | [TK]D-Fender | Marquel: No |
| 19:22.26 | [TK]D-Fender | Marquel: You can do that outside of * with a lib that can access the BDB or SQLite DB that lies underneath or parse it out from a text dump |
| 19:38.09 | apurvtwr | anyone here who has faced this error before : res_jabber.c:3127 aji_initialize: JABBER ERROR: No Connection |
| 19:39.01 | apurvtwr | there are forums where people have asked the same question but to no solution |
| 19:39.02 | apurvtwr | http://forums.digium.com/viewtopic.php?f=1&t=79392 |
| 19:43.33 | jpsharp | You don't see a proper connection in "jabber show connections"? |
| 19:45.43 | apurvtwr | Jabber Users and their status: [gtalk] my_email@gmail.com/Talk - Disconnected |
| 19:46.10 | apurvtwr | i am using the same configurations as you suggested the other day |
| 19:46.23 | apurvtwr | and it was working. |
| 19:46.41 | jpsharp | did your external IP change or anything? |
| 19:47.47 | apurvtwr | no. it is the same as the one specified in gtalk.conf |
| 19:49.28 | apurvtwr | it was working till yesterday. then my computer crashed. stopped working. I reinstalled everything, but the error remains. |
| 19:50.05 | apurvtwr | to uninstall I removed asterisk from /etc/asterisk and /var/log/asterisk |
| 19:51.02 | apurvtwr | is there something else that I should have removed? |
| 19:51.47 | jpsharp | No, the config files are stored in /etc/asterisk |
| 19:52.07 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:52.09 | apurvtwr | ok. |
| 19:54.58 | jpsharp | Have you checked the basic things like does that machine have internet connection? |
| 19:55.40 | apurvtwr | i am talking from the same machine |
| 19:56.08 | jpsharp | Just checking. |
| 19:56.51 | apurvtwr | yeh sure. no problem. :-) |
| 19:56.57 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 19:57.25 | *** join/#asterisk frawd (~francois@79.Red-88-9-167.dynamicIP.rima-tde.net) |
| 20:03.00 | *** join/#asterisk madd (~m@188.40.175.238) |
| 20:03.34 | madd | hello |
| 20:04.30 | madd | I couldn't fetch any packages of packages.asterisk.org nor any http-request if I try that with curl |
| 20:05.22 | madd | [root@mx02 ~]# curl http://packages.asterisk.org |
| 20:05.22 | madd | curl: (7) Failed to connect to 2001:470:e0d4::e5: Das Netzwerk ist nicht erreichbar |
| 20:05.45 | madd | curl seems to try ipv6 but my system has got only ipv4 |
| 20:06.52 | madd | but host packages.asterisk.org offers also an ipv4 address |
| 20:07.08 | jpsharp | A lot of programs have the option to force ipv4 only with the -4 option. |
| 20:07.09 | madd | is packages with ipv4 down? |
| 20:07.41 | [TK]D-Fender | yes |
| 20:07.52 | jpsharp | Oh, yes it is. |
| 20:09.13 | madd | hmmm mirror? |
| 20:10.08 | madd | would be nice to find this info in the IRC-Topic |
| 20:11.01 | [TK]D-Fender | if Ops (mostly Digium people) were around to do that... theyd have notified others and fixed it already |
| 20:12.07 | madd | ok, hope so. Server is down for serveral hours. |
| 20:14.04 | madd | jpsharp: thx for the -4 hint. I forgot this option |
| 20:15.58 | madd | have got someone sample configs for 1.8? |
| 20:16.08 | madd | centos 5 |
| 20:17.53 | *** join/#asterisk k610 (~Instantbi@host-85-201-12-229.brutele.be) |
| 20:24.59 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 20:24.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:25.52 | robl^ | svn is fown too |
| 20:25.58 | robl^ | down even |
| 20:32.01 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 20:45.39 | *** join/#asterisk apurvtwr (8002d2cb@gateway/web/freenode/ip.128.2.210.203) |
| 20:45.52 | apurvtwr | jpsharp: you there? |
| 20:46.17 | *** join/#asterisk Nemus (~Nemus@c-67-161-251-38.hsd1.ut.comcast.net) |
| 20:50.20 | jpsharp | Aye |
| 20:50.39 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:50.39 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:55.30 | apurvtwr | any clue of how to fix it? I have tried pinging from another computer to this one.. and it works so i guess external ip is correct |
| 20:55.51 | apurvtwr | tried with another gmail account .. the problem still remains |
| 21:04.02 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 21:06.46 | *** join/#asterisk alsuren_ (~dlaban@host109-155-8-48.range109-155.btcentralplus.com) |
| 21:27.04 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 21:28.20 | *** join/#asterisk k610 (~Instantbi@cable-78.29.241.186.coditel.net) |
| 21:28.52 | *** join/#asterisk ska (~ska@unaffiliated/ska) |
| 21:33.34 | *** join/#asterisk k610 (~Instantbi@cable-78.29.241.186.coditel.net) |
| 21:38.43 | *** join/#asterisk k610 (~Instantbi@cable-78.29.241.186.coditel.net) |
| 21:41.50 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 21:42.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:43.59 | *** join/#asterisk k610 (~Instantbi@cable-78.29.241.186.coditel.net) |
| 21:46.17 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 21:48.59 | Neptu | hej just wondering if its worth to install apt-get asterisk with version 1.8 or compile 2.0... what you think? |
| 21:49.17 | *** join/#asterisk k610 (~Instantbi@cable-78.29.241.186.coditel.net) |
| 21:50.25 | Neptu | the only difference that i know is that chatrooms can have better quality |
| 21:50.53 | [TK]D-Fender | ...... 2.0?\ |
| 21:51.36 | ska | Has anyone uses Asterisk on pfsense? |
| 21:55.24 | Neptu | [TK]D-Fender: sorry 10 |
| 21:55.30 | Neptu | donno what i was talking about |
| 21:56.03 | Neptu | just checking the documentation about whats new in 10 and having a sqllite db can be handy |
| 21:56.30 | [TK]D-Fender | Except it is implented as SQLite. |
| 21:56.57 | [TK]D-Fender | That is just the backend. The front end still has a BDB layer so there isnt direct access to the SQLite behind the scenes |
| 21:57.40 | Neptu | aha so no tampering with the database |
| 21:57.51 | Neptu | jur |
| 21:58.09 | Neptu | donno gonna compile 10 |
| 21:58.13 | Neptu | and look happy |
| 21:58.17 | [TK]D-Fender | You CAN... but the data written to it is formatted with BDB in mind. |
| 21:58.43 | Neptu | BDB <-- no idea |
| 21:59.23 | [TK]D-Fender | As in? |
| 21:59.30 | Neptu | just gonna setup test thingie for me i played with asterisk before and was kinda of fin |
| 21:59.33 | [TK]D-Fender | BDB is what AstBD was using since the start.... |
| 21:59.41 | Neptu | Binary Data base? |
| 21:59.53 | Neptu | Beautiful Data base |
| 22:00.02 | [TK]D-Fender | Bekeley DB |
| 22:00.13 | Neptu | ah |
| 22:00.13 | Neptu | ok |
| 22:01.06 | Neptu | so no mess it with it... i saw you can do some scripting in python... I will check that up instead... |
| 22:02.12 | [TK]D-Fender | What do you actually want to do? |
| 22:09.48 | Neptu | just to play around make a system for home maybe add a server at my parents or get a spanish phone number and use it directly from my htc |
| 22:10.01 | Neptu | i mean mostly play a bit with it |
| 22:10.33 | [TK]D-Fender | I was asking more about the DB side of that.... |
| 22:13.35 | Neptu | I was thinking to pharse backslash and get all the betamax routes to be used somewhere i have some experience with sql lite i thougth i can trick asterisk to use those routes on database level |
| 22:13.53 | Neptu | bu asterisk probably has another way of thinking... |
| 22:14.19 | [TK]D-Fender | Use routes = dialplan |
| 22:14.59 | [TK]D-Fender | SAsterisk itself doesnt inherently use any DB for that so yu can do whatever LCR analysis you want in making your call flow, but that isn't * itself |
| 22:15.56 | Neptu | strange does not compile the source from the subversion directly... i think it did before |
| 22:16.00 | apurvtwr | is anyone right now able to use google voice with Asterisk? |
| 22:17.03 | apurvtwr | it was working a few days back, but jabber show connections shows me disconnected all the time now |
| 22:17.12 | Neptu | [TK]D-Fender: I saw some python lib for asterisk maybe can help me to deal with the dialplan without having to use nasty telnet commands |
| 22:17.12 | Nugget | telnet is eeeeeeevil! |
| 22:18.50 | apurvtwr | jpsharp: you still there? |
| 22:19.38 | Neptu | donno what lib SS7 is, do I need this? |
| 22:19.39 | apurvtwr | you said something about the problem with external IP address? |
| 22:20.06 | apurvtwr | I found a IRC log where someone faced a similar problem http://ibot.rikers.org/%23asterisk/20120116.html.gz |
| 22:20.15 | [TK]D-Fender | Neptu: What is this about telnet and not having to deal with the dialplan? |
| 22:20.25 | [TK]D-Fender | Neptu: You aren't making much sense.... |
| 22:20.42 | apurvtwr | seems like IvP4 - IvP6 address |
| 22:21.00 | apurvtwr | how can I force asterisk to use IvP4 ? |
| 22:21.08 | Neptu | Well a friend explained to me that there is a port you can connect over telnet and basically run quite a lot of commands to change most of things in asterisk |
| 22:21.38 | Neptu | he worked on a company developing for asterisk for a year or so, I belive he knows what he talks about |
| 22:23.15 | [TK]D-Fender | Neptu: His description is vague and largely incorrect. |
| 22:23.25 | Neptu | ok |
| 22:23.34 | Neptu | ill talk with him tomorrow |
| 22:23.47 | [TK]D-Fender | Neptu: AMI is a socket interface for manipulating calls in progess, but ewhen a call comes in that his dialplan. that is what determins what steps a call goes through. |
| 22:23.51 | [TK]D-Fender | ? book |
| 22:23.56 | [TK]D-Fender | ~book |
| 22:23.56 | 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 |
| 22:23.58 | [TK]D-Fender | ^^^ |
| 22:24.52 | [TK]D-Fender | There is no inherent relationship between something like a SIP entry for a phone and a SIP entry for an ITSP you want to use to terminate calls. |
| 22:25.15 | [TK]D-Fender | things like SIP are 10% of your configuration job. |
| 22:25.50 | [TK]D-Fender | When a call comes in you have to look at what is dialed and take whatever actions are appropriate to your goals. That is the dialplan. |
| 22:27.21 | [TK]D-Fender | AGI Does allow you to do most of your logic outside of *'s command set and call commands for bread & butter things like playing back audio, getting input ete though. |
| 22:27.26 | [TK]D-Fender | etc* |
| 22:27.41 | [TK]D-Fender | But it's best used only when necessary |
| 22:54.40 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 22:54.53 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 23:01.04 | apurvtwr | Has anyone seen this error before: ERROR[14811]: res_jabber.c:3127 aji_initialize: JABBER ERROR: No Connection |
| 23:01.13 | apurvtwr | or can help me resolve it? |
| 23:01.39 | *** join/#asterisk Alex_Bkash (b4ea514a@gateway/web/freenode/ip.180.234.81.74) |
| 23:02.08 | Alex_Bkash | hello, Is there any IAX2 client for symbian? |
| 23:04.04 | [TK]D-Fender | I can't imagine anyone would bother. |
| 23:05.32 | Alex_Bkash | can i make iax callls from symbian? |
| 23:05.58 | [TK]D-Fender | if you have an IAX2 app for it probably |
| 23:06.30 | Alex_Bkash | is thr any opensource one? |
| 23:07.02 | WIMPy | iaxcomm |
| 23:07.22 | WIMPy | Or you could use libiax. |
| 23:19.03 | Alex_Bkash | thanks a lot |
| 23:23.14 | jpsharp | apurvtwr: Can I see your config files, with passwords removed, of course. |
| 23:23.33 | apurvtwr | sure. one sec |
| 23:27.26 | apurvtwr | jabber.conf : http://pastebin.com/PXkfYxD7 |
| 23:29.22 | apurvtwr | gtalk.conf: http://pastebin.com/R8s56D91 |
| 23:30.51 | apurvtwr | extensions.conf: http://pastebin.com/7ZvY8Grg |
| 23:31.07 | apurvtwr | well it's the changes in the extensions.conf that i have added here |
| 23:32.48 | jpsharp | I'm not sure. I can't duplicate your symptoms here. |
| 23:34.05 | apurvtwr | strange. |
| 23:35.01 | apurvtwr | hm. what all directories must i remove to remoe everypossible settings that asterisk has made? |
| 23:36.00 | jpsharp | /etc/asterisk, /var/lib/asterisk, /var/spool/asterisk |
| 23:36.35 | apurvtwr | ok. Let me remove all 3 directories and make fresh installation |
| 23:36.36 | apurvtwr | thanks |
| 23:36.54 | jpsharp | If you're doing a full, fresh installation, also wipe /usr/lib/asterisk |
| 23:37.21 | apurvtwr | ok. I will remove that too. should i reinstall iksemel ? |
| 23:37.30 | jpsharp | Nah. |
| 23:37.36 | apurvtwr | or that is independent |
| 23:37.38 | apurvtwr | hm. ok |
| 23:38.57 | jpsharp | It is independent of Asterisk. |
| 23:39.34 | apurvtwr | i see. |
| 23:39.55 | apurvtwr | give me 5 mins. will install everything and be right back |
| 23:42.59 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 23:42.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:45.54 | *** join/#asterisk jrgill (~jrgill@unaffiliated/jrgill) |
| 23:46.17 | apurvtwr | ok. removed the directories: /etc/asterisk , /var/lib/asterisk, /var/spool/asterisk and /usr/lib/asterisk. Installed Asterisk by executing "./configure, make , make install" |
| 23:46.55 | apurvtwr | modified jabber.conf, gtalk.conf and extensions.conf to the same configuration I sent you |
| 23:47.21 | apurvtwr | and it still shows disconnected |
| 23:47.40 | jpsharp | Can you telnet to talk.google.com on port 5222? |
| 23:48.10 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 23:48.36 | apurvtwr | a sec let me try |
| 23:52.14 | apurvtwr | nope. it seems to be stuck on trying 74.125.142.125... |
| 23:52.48 | jpsharp | Well, that's your problem. |
| 23:53.19 | apurvtwr | hm. i see. |
| 23:53.52 | apurvtwr | thanks. though it's strange.. that suddenly my network provider blocked it. |
| 23:54.30 | jpsharp | I get 74.125.45.125 for an IP for talk.google.com |
| 23:55.32 | apurvtwr | I guess google has multiple region specific servers |
| 23:56.21 | apurvtwr | even ping talk.google.com, resolves it to 74.125.142.125 for me |
| 23:56.50 | jpsharp | Try hard setting the IP to 74.125.45.125 instead of talk.google.com |
| 23:57.40 | jpsharp | Or at least see if you can telnet to that IP on port 5222 |
| 23:57.53 | jpsharp | I can connect to both from here. |
| 23:58.09 | jpsharp | jsharp@asterisk:~$ telnet 74.125.142.125 5222 |
| 23:58.09 | jpsharp | Trying 74.125.142.125... |
| 23:58.09 | jpsharp | Connected to 74.125.142.125. |
| 23:58.13 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:58.13 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 23:58.56 | apurvtwr | tried that 74.125.142.125 |
| 23:59.00 | apurvtwr | Connected to 74.125.45.125 |
| 23:59.15 | apurvtwr | but not to 142.125 |
