| 00:05.10 | ChrisInSydney | hey all |
| 00:06.10 | ChrisInSydney | here is a good one. I have a client with Vocaltone. The hosts they have asked us to point at are on the domain at vocaltone.net.au |
| 00:06.11 | slav3_kitten | what up ChrisInSydney |
| 00:06.31 | wonderworld | hey chris |
| 00:06.37 | ChrisInSydney | the domain doesn't resolve. Hasn't done since yesterday middau |
| 00:06.43 | ChrisInSydney | midday |
| 00:07.09 | ChrisInSydney | they wont gove me IP addresses and I have tried multiple other things. Nothing |
| 00:07.39 | ChrisInSydney | I have also set up alternative SIP addresses to forward to and numbers. Nothing |
| 00:08.12 | ChrisInSydney | Haven't heard boo since 5:30pm where I was told it was "Telstra issue" |
| 00:08.30 | ChrisInSydney | I think I am going to drive to their office across town |
| 00:08.45 | ChrisInSydney | no inbund for a group of 6 companies |
| 00:08.53 | ChrisInSydney | not small either |
| 00:09.07 | ChrisInSydney | thats my rant |
| 00:09.21 | ChrisInSydney | F()# |
| 00:09.40 | wonderworld | sounds like an ugly chain of problems |
| 00:09.45 | ChrisInSydney | hey slav3_kitten wonderworld |
| 00:09.52 | slav3_kitten | ChrisInSydney, have you tried a bigger hamer? |
| 00:10.08 | ChrisInSydney | I'm about to |
| 00:10.10 | ChrisInSydney | :-) |
| 00:11.07 | ChrisInSydney | its called drive to the carrier's office. Then drive to the chairman of the board of the customers with porting forms |
| 00:11.20 | ChrisInSydney | if I dont get what I want |
| 00:11.48 | ChannelZ | FWIW vocaltone.net.au resolves for me |
| 00:12.05 | dijib | ChrisInSydney: vocaltone.net.au = 210.80.188.202 resolves http just doesnt respond to pings |
| 00:12.09 | ChannelZ | vocaltone.net.au has address 210.80.188.202 |
| 00:12.26 | paulc | slav3_kitten: Nope, that wasn't me.. (but it rings a bell - think I followed along on that conversation) |
| 00:12.38 | ChrisInSydney | need aus7.vocaltone.net.au |
| 00:13.04 | ChannelZ | 203.166.11.116 |
| 00:13.08 | dijib | same. |
| 00:13.12 | ChrisInSydney | [Nov 22 11:10:59] WARNING[4687]: chan_sip.c:2938 create_addr: No such host: aus7.vocaltone.net.au |
| 00:13.16 | ChrisInSydney | thanks |
| 00:13.19 | ChrisInSydney | I'll give it a go |
| 00:13.28 | ChannelZ | infobot: dns for aus7.vocaltone.net.au |
| 00:13.33 | paulc | slav3_kitten: I was hoping it was a quick'n'easy "Yes, a2billing will help us bill for a click to call type service" but it's a pig to get working, so we might end up rolling our own.. or bailing on the project.. |
| 00:13.36 | ChannelZ | Interesting |
| 00:13.55 | ChannelZ | Perhaps they are having routing problems. |
| 00:16.31 | slav3_kitten | ChrisInSydney, sorry your stuff is busted though |
| 00:17.26 | ChrisInSydney | nahh. Looks like it has just come good. Was f&^%ed 7 mins ago |
| 00:18.10 | dijib | if you look at the flash banner on the vocaltone.com.au site... what does untime mean? |
| 00:19.10 | slav3_kitten | untime is typo for funtime :D |
| 00:19.26 | ChrisInSydney | working |
| 00:19.39 | ChrisInSydney | faaaaaarrrrrrrkkkkkkk |
| 00:19.50 | dijib | that would be enough for me to want to switch from that itsp |
| 00:20.19 | ChrisInSydney | they went off for two days. Absolutle nothing. Their own numbers didn't even work. |
| 00:20.45 | dijib | itsp fail |
| 00:20.46 | ChrisInSydney | I was told that it was a DDoS on their services, but they didn't seem to have any backups at all |
| 00:20.52 | ChrisInSydney | yup. |
| 00:20.54 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 00:21.07 | ChrisInSydney | Now where are those porting forms ? |
| 00:21.19 | ChrisInSydney | Thanks all sooo much for that info |
| 00:21.19 | Micc | Thats why we have multiple data centers, just in case. |
| 00:21.27 | ChrisInSydney | Hey Micc |
| 00:21.29 | ChrisInSydney | same here |
| 00:21.39 | ChrisInSydney | multiple centres, multiple carriers |
| 00:21.45 | Micc | hey Chris. |
| 00:22.04 | Micc | It doesn't help much if your carrier doesn't have redundancies though. |
| 00:22.27 | ChrisInSydney | not my carrier |
| 00:22.40 | ChrisInSydney | I inherrited them with the client |
| 00:22.49 | Micc | oh, yeah thats not good. |
| 00:23.04 | Micc | We've had to deal with inherrited customers on really bad setups too. |
| 00:23.32 | Micc | We ended up porting them all over to our main carrier after the first problem. |
| 00:23.58 | ChrisInSydney | time to port them. I'm signing a couple of agreements with some wholesalers of my own and the guys I use can give me good wholesale too |
| 00:24.36 | ChrisInSydney | this is strike #2, but given the timeframe and what happened last time, time to move |
| 00:24.47 | Micc | There's nothing more frustrating that customers down and nothing you can do but wait on someone else. |
| 00:24.53 | ChrisInSydney | We have had some inter-carrier number routing issues |
| 00:25.13 | ChrisInSydney | srage ones, whole groups of numbers uncontactable between carriers |
| 00:25.58 | Micc | we've had some numbers stay in the loosing carrier's switch for a while that screws things up, but nothing major like that. |
| 00:26.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:26.36 | ChrisInSydney | wierd shit. We have 8 digits + 1 digit area code. We would have a 6 digit mask dissapear of the grid |
| 00:27.06 | ChrisInSydney | you could call out and you could call any number within the group of numbers |
| 00:27.26 | ChrisInSydney | but fro outside that group, disconnected or not available / switched off |
| 00:28.07 | ChrisInSydney | I heard someone cut a fibre trunk as a "F.. You" to their odl employer |
| 00:28.11 | ChrisInSydney | just pub talk |
| 00:28.15 | ChrisInSydney | nothing confirmed |
| 00:28.22 | *** join/#asterisk wonderworld (~w@dsdf-4db554c4.pool.mediaWays.net) |
| 00:29.08 | ChrisInSydney | but e haad one day where we had around 35% of our customers with some sort of inbound routing issue. ISDNs were down too. Even in the City |
| 00:29.59 | ChrisInSydney | all the carriers went to back up and the routing all got tangled up. IP and PSTN |
| 00:30.41 | ChrisInSydney | We just called people via direct SIP where we could and told them to sit sight |
| 00:30.54 | ChrisInSydney | took us a couple of hours, then we went to the pub for a long lunch |
| 00:31.13 | ChrisInSydney | sit tight |
| 00:31.35 | Micc | damn, sounds like a rough day. |
| 00:32.06 | ChrisInSydney | Micc. same with us. I always leave the old registrations / routes active until I'm 1000% sure |
| 00:32.39 | ChrisInSydney | it was. Had three of those in the last two years. usually 2-4 hours of grief. This one was all day |
| 00:34.00 | Micc | I don't think we would ever survive an all day outage. |
| 00:34.02 | ChrisInSydney | one company Dodo, had a router that crashed the Telstra network. Almost all of our customers were affected in some way, but most had redundancy |
| 00:34.17 | ChrisInSydney | it want just us, it was the big boys too |
| 00:34.55 | Micc | If its affecting the big boys too, customers usually understand. |
| 00:35.07 | ChrisInSydney | I guess the equivilent of AT&T and Verizon going off line |
| 00:35.11 | ChrisInSydney | you |
| 00:35.14 | ChrisInSydney | yup |
| 00:35.19 | ChrisInSydney | they did |
| 00:35.48 | ChrisInSydney | thats why we called them all in advance and took the afternoon off |
| 00:40.45 | ChrisInSydney | I phoned the obmudman's office, just to load the gun, not to fire it, just to have it ready |
| 00:40.55 | ChrisInSydney | for this issue |
| 00:41.22 | ChrisInSydney | They get 2000 complaints a week. Thats in a country of 20 million people |
| 00:41.40 | slav3_kitten | that's not too terrible |
| 00:42.24 | slav3_kitten | that's only .01% |
| 00:42.39 | ChrisInSydney | nahh. probably not. I recon 30% would be people who just like to complain |
| 00:43.00 | slav3_kitten | i may have fucked my math up, i'm not good at math |
| 00:43.36 | ChrisInSydney | thats 99.99% of customers who aren't pissed off enugh to call a free call number |
| 00:44.19 | slav3_kitten | which is damn good |
| 00:44.40 | slav3_kitten | you need to move to america an run the isp i use. then i can be happy with them |
| 00:45.30 | ChrisInSydney | no I dont, and you wouldn't be happy, cause I'd just be at the pub ;-) |
| 00:46.08 | ChrisInSydney | Alcoholics go to meetings. I go to the pub ;-) |
| 00:46.15 | slav3_kitten | lol |
| 00:48.13 | ChrisInSydney | actually, I dont get out that much, and I almost never get the opportunity to take a long lunch. |
| 00:48.36 | ChrisInSydney | anyway, I have service calls to do. Thanks all |
| 00:50.01 | ChrisInSydney | BTW what are you people using to monitor / alert when trunks go down. We're just using dial plans and alternative routes with the carrier to trigger alerts when a call needs a reroute |
| 00:51.03 | ChrisInSydney | I've got the SNMP stuff loaded, on the new systems, but I haven't seen what sort of info / alerts it gives me |
| 00:51.35 | slav3_kitten | snmp with asterisk? |
| 00:59.19 | Maliuta | slav3_kitten: it can be done |
| 00:59.42 | Maliuta | unlike snmp on a billion 5102[s] |
| 01:00.30 | slav3_kitten | stupid question... 5102? |
| 01:01.03 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 01:08.42 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:08.46 | *** part/#asterisk deo (~deo@222.127.13.226) |
| 01:28.45 | dijib | hey anybody conscious of how long back i can do in voipms call logs? |
| 01:33.01 | SeRi | dijib: like cdr? |
| 01:33.12 | dijib | yeah but voip.ms's |
| 01:33.59 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.128) |
| 01:38.44 | dijib | p3nguin: wasn't you that informed me of a way you could look back until account creation? |
| 01:40.08 | SeRi | dijib: yes you can. |
| 01:40.14 | SeRi | but it has to be month by month |
| 01:40.30 | SeRi | not first day day of accountr creation till present |
| 01:40.42 | SeRi | you can go month by month. |
| 01:43.20 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 01:43.54 | SeRi | dijib: you there? |
| 01:44.03 | dijib | ya im here did you just come into conf? |
| 02:05.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:06.30 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 02:08.21 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 02:08.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:26.19 | Maliuta | slav3_kitten: it's a model of ADSL2+ modem |
| 02:27.23 | Maliuta | slav3_kitten: the one sitting on top of my rack, and that I have had for 7 years now. It doesn't give me the info I want (i.e. ADSL sync rates) via snmp |
| 02:34.29 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.155) |
| 02:34.44 | slav3_kitten | ah |
| 02:51.55 | *** join/#asterisk dfgas (~dfgas@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 02:53.42 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.83.48) |
| 03:00.48 | *** part/#asterisk ircNewbz_ (~ircnewbz@unaffiliated/ircnewbz) |
| 03:02.37 | *** join/#asterisk coppice (~chatzilla@m121-202-66-47.smartone.com) |
| 03:05.53 | SeRi | dijib: you around? |
| 03:15.02 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 03:15.09 | dijib | yup |
| 03:15.22 | dijib | your still logged in |
| 03:22.14 | p3nguin | you're |
| 03:22.43 | p3nguin | "your" means something else. |
| 03:27.47 | p3nguin | slav3_kitten: I don't want you to be mislead, the i extension does not magically match a call when there isn't something else to match. 'i' isn't a catch all; 'i' is the invalid extension, which is run from things such as WaitExten or BackGround where a caller enters an extension. |
| 03:34.38 | *** join/#asterisk ideaman55 (~ideaman55@199.30.186.240) |
| 03:35.30 | *** join/#asterisk ziz212 (~ziz212@203.115.2.202) |
| 03:37.57 | slav3_kitten | p3nguin, thanks |
| 03:38.49 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 03:38.59 | dfgas-cr48 | dijib, hmmm |
| 03:39.10 | p3nguin | And extensions are matched by the most specific to the most general, so _XXX would match before _XX. would. |
| 03:39.38 | p3nguin | s/by/from/ |
| 03:40.12 | p3nguin | _12345. would match before _123. would |
| 03:40.16 | p3nguin | etc. |
| 03:41.19 | ziz212 | Hi, I need some help form you all guys |
| 03:42.03 | Nivex | ALL of us? |
| 03:42.08 | Nivex | that's going to be expensive |
| 03:42.37 | p3nguin | If someone didn't already tell you, the best catch-all pattern is going to be something like _X and _X. to match one digit and to match two or more digits, which have not already been matched in a more specific pattern. |
| 03:42.48 | ziz212 | yes |
| 03:43.01 | ziz212 | I will pay for each and every one |
| 03:43.06 | *** join/#asterisk nix8n82 (~AndChat21@24.143.10.93) |
| 03:43.07 | ziz212 | :) |
| 03:43.28 | ziz212 | what do you all say? |
| 03:44.17 | *** join/#asterisk cyborg-one (~cyborg-on@188.115.130.215) |
| 03:44.20 | p3nguin | I think I read where it was explained that _. will match ALL extensions, including the ones you do not want to match. As tk put it, "o shit a fax" will also be matched by _. (which is bad). |
| 03:47.15 | *** join/#asterisk joobie (~joobz@unaffiliated/moo0o0ooo00o0o0o) |
| 03:47.48 | joobie | hey guys.. when an invite request is sent that specifies the audio RTP and port, is the port that is specified in the INVITE request coming from the port range defined on the phone or from the port range defined on asterisk? |
| 03:47.50 | ziz212 | friends I need some help in understanding the way of taking outbound concurrent calls from astesisk? |
| 03:47.59 | ziz212 | How autodialer software are used to take outbound calls, like x number of concurrent calls ? How it is impemented in there? |
| 03:50.25 | FireAndIce | Hi everyone!! |
| 03:51.11 | FireAndIce | I'm trying to configure my asterisk server for Dynamic Realtime. But facing issues. |
| 03:52.00 | ziz212 | friends any information for me if some one have time? |
| 03:52.52 | FireAndIce | Here is the metadata of my users database. http://paste.ubuntu.com/1376376/ |
| 03:54.01 | FireAndIce | my extconfig.conf, http://paste.ubuntu.com/1376380/ |
| 03:54.23 | FireAndIce | Please help.. |
| 03:57.57 | FireAndIce | executing 'sip show peers' after reloading chan_sip.so does not show the users from the ast_sipfriends. |
| 03:59.11 | ziz212 | is it use call files to take concurrent x no of calls? or any methology? |
| 04:01.55 | dijib | hey im back now, but have old whirllly drying my clothes 4 ft from phone |
| 04:02.12 | SeRi | is cool |
| 04:02.15 | SeRi | look at your pm |
| 04:02.22 | *** join/#asterisk pcAngel (~yoink@S0106602ad07d2b78.vc.shawcable.net) |
| 04:03.29 | pcAngel | Hi guys When I originate a call to an external phone number (ie my cell phone) and then when answered, connect it to a queue, the caller ID of the dialed phone number shows up as Unknown/Unknown when answered in the queue - I have no idea where to look, because I'm not setting it to blocked in my dial plan |
| 04:04.17 | pcAngel | In the asterisk manager interface I see a tonne of masquerades/renames, and a <zombie> channel-could those be part of the problem? |
| 04:06.18 | ziz212 | all have questions.. gurus are busy... |
| 04:09.58 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-pmiqggdkwqwebgxr) |
| 04:15.47 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 04:17.43 | gusto | so... |
| 04:17.52 | gusto | today i am early ;-) |
| 04:33.46 | dfgas-cr48 | dijib, yo |
| 04:34.46 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
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| 04:51.01 | joobie | guys for RTP |
| 04:51.23 | joobie | does the phone advertise the port that it wants to listen for audio (speaker) or the output audio port (mic) ? |
| 04:52.21 | ziz212 | guy How can we config more concurrent calls in astersik? is it relaed to call files? |
| 04:52.37 | ziz212 | how to specify the numebr in dial plan? |
| 04:53.16 | *** join/#asterisk mokmeister (~mokmeiste@109.76.71.159) |
| 04:56.50 | *** join/#asterisk whtsup (~whtsup@180.92.141.172) |
| 04:57.02 | whtsup | [Nov 22 09:55:16] WARNING[21575]: db.c:135 init_stmt: Couldn't prepare statement 'CREATE TABLE IF NOT EXISTS astdb(key VARCHAR(256), value VARCHAR(256), PRIMARY KEY(key))': disk I/O error |
| 04:57.02 | whtsup | [Nov 22 09:55:16] WARNING[21575]: db.c:180 db_create_astdb: Couldn't create astdb table: disk I/O error |
| 04:57.07 | whtsup | asterisk not starting |
| 04:57.16 | whtsup | im newbie wht to do |
| 04:57.17 | whtsup | plz help |
| 05:01.50 | dfgas-cr48 | dijib, hey are you around |
| 05:01.52 | dfgas-cr48 | ? |
| 05:03.29 | dfgas-cr48 | SeRi, ? |
| 05:03.44 | gusto | joobie: over SIP |
| 05:04.48 | gusto | joobie: and it depends on if you have NAT set to =yes or to =comedia or to =rport or to =off |
| 05:06.45 | gusto | joobie: on NAT=yes it's like =rport + =comedia, =rport tells him to use rport and =comedia waits until the connection comes from the other side and uses that port instead |
| 05:07.51 | gusto | joobie: with NAT=no they just try to connect to each other's advertised ports over SIP |
| 05:07.56 | gusto | directly |
| 05:09.35 | gusto | so when i am right the difference between =rport and =no is just that one side does not pick a port freely but uses that one which is proposed by the other side |
| 05:11.46 | gusto | so =rport would be the choice when you are sure that he can connect to that port you opened, like when you have a home router that does respect that and =comedia you need when you are double natted, like when you come over mobile broadband, because there you can not be sure if the port you open will be the same that it comes out on WAN |
| 05:20.50 | gusto | whtsup: hey |
| 05:21.34 | gusto | whtsup: when you are a newbie you should not try to put the astdb into a database |
| 05:22.01 | gusto | whtsup: rather create the directory where astdb should be in and after he creates the file, he will start |
| 05:22.56 | tonikasch | bye |
| 05:23.13 | gusto | whtsup: look into your <prefix>/etc/asterisk/asterisk.conf to find out where he expects the astdb to be and unload any ODBC shit |
| 05:24.12 | ChannelZ | actually depending on your version, I thought astdb is a DB now... sqlite or whatever |
| 05:25.19 | ChannelZ | Yeah.. in Asterisk 11 |
| 05:25.21 | gusto | yes |
| 05:25.46 | gusto | but that log looked like he would try to put that astdb in a database ... like postgres or whatever |
| 05:26.35 | gusto | as soon as asterisk can create that astdb file i do not care what he uses as a db backend if sqlite or db4 or whatever ... astdb is useless though |
| 05:27.41 | ChannelZ | in any event I'm guessing if it is Asterisk 11, he does not have write access to /var/lib/asterisk |
| 05:28.02 | gusto | to me astdb is put in ram and of course it is created everytime as new what may not be suggested, but i never had any problems with it |
| 05:28.43 | gusto | yes ... to me /var/lib/asterisk did not even exist, because on openwrt for example you have /var -> /tmp -> ramfs |
| 05:28.46 | ChannelZ | It's always been a persistent database. SIP peers are cached there even. |
| 05:29.10 | ChannelZ | oh.. well yeah wrt is a totally different story. |
| 05:29.15 | gusto | so i have edited the /etc/init.d startup script to add mkdir /var/lib/asterisk in there and that works |
| 05:39.15 | *** join/#asterisk fritz09 (~Adium@2002:542e:50e1:e472:2468:5ac6:94c6:3c6) |
| 06:19.38 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 06:35.03 | slav3_kitten | p3nguin, you about. got some questions on brute force attack detection |
| 06:37.50 | slav3_kitten | also [Nov 21 22:55:32] ERROR[15799][C-000000ae]: pbx.c:3832 ast_func_read: Function CHANNE not registered |
| 06:38.04 | *** join/#asterisk Tabrenus (~Tabrenus@213.211.132.86.static.edpnet.net) |
| 06:49.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 06:51.44 | kaldemar | slav3_kitten: that error is a typo |
| 06:52.23 | kaldemar | or due to a typo i mean |
| 06:53.22 | slav3_kitten | face palms |
| 06:53.39 | slav3_kitten | yes, yes it is. i had read that section 3 times looking for it |
| 06:54.32 | slav3_kitten | i also have about two hundred variations of Inbound unauthenticated call to 221011972592735467 : "as123456" <as123456> : 37.8.34.106 |
| 06:54.48 | slav3_kitten | IP is the same though |
| 06:57.35 | slav3_kitten | so guess i need to make a firewall rule |
| 06:58.05 | WIMPy | gm |
| 06:58.54 | slav3_kitten | is this kinda think common? |
| 06:58.57 | slav3_kitten | thing* |
| 06:59.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 06:59.29 | WIMPy | What? People trying to relay calls via anything? Sure. |
| 06:59.55 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 06:59.58 | slav3_kitten | that's a relay attempt? |
| 07:00.01 | WIMPy | Doing it for e-mail was yesterday. Now it's SIP. |
| 07:00.15 | WIMPy | Unless it was you. |
| 07:01.17 | WIMPy | I always find it interesting to see what kind of numbers they try to call. |
| 07:01.47 | slav3_kitten | nah it wasn't me |
| 07:03.17 | WIMPy | Most calls I see are to UK numbers. Often with rather bizarre prefixes. |
| 07:04.50 | slav3_kitten | is there any good way to minimize the attempts while allowing guests |
| 07:05.10 | WIMPy | No |
| 07:05.41 | WIMPy | Make sure you don;t allow guests to do anything you don't want and ignore them. |
| 07:08.16 | *** join/#asterisk svnNB (~svn@host103-70-dynamic.60-82-r.retail.telecomitalia.it) |
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| 07:10.22 | slav3_kitten | WIMPy, can i get you to call my bridge for a test |
| 07:12.03 | WIMPy | yes |
| 07:18.51 | *** join/#asterisk chris-NB (~chris@fw.commpany.at) |
| 07:21.21 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 07:21.26 | hrolf | Hi asterisk |
| 07:21.55 | hrolf | I have a problem, where I get the error SIP 503 Service Unavailable |
| 07:22.15 | slav3_kitten | hi hrolf |
| 07:22.36 | hrolf | I'm using a SIP peer and it is connected to another Asterisk machine which does the actual dialing through the SIP peer |
| 07:23.17 | hrolf | http://pastebin.com/3h0JuaKm this is the console log that I get |
| 07:23.39 | hrolf | <PROTECTED> |
| 07:23.41 | hrolf | <PROTECTED> |
| 07:23.53 | hrolf | Is it something to worry about? |
| 07:24.06 | hrolf | we are doing outbound dialing through that SIP peer |
| 07:24.17 | hrolf | Is it an error? |
| 07:24.51 | WIMPy | Whatever you called either didn't like what yu called or didn't like you. |
| 07:25.23 | hrolf | WIMPy: what does that mean? |
| 07:25.49 | hrolf | WIMPy: Do you mean that the number I dialed didn't answer that's why I'm getting this? |
| 07:26.26 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:26.32 | WIMPy | No. Your call wasn't accepted in the first place. |
| 07:27.45 | hrolf | WIMPy: So how do I go about figuring out the cause? |
| 07:34.39 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
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| 07:37.20 | Sheeplet | lo all |
| 07:37.31 | v0lZy | love the nick :D |
| 07:37.35 | v0lZy | hi. |
| 07:45.50 | slav3_kitten | i think sleep should find me |
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| 09:07.42 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 09:09.44 | nunne | Using asterisk 1.4 setting the "setqueueentryvar=yes" in my queue. I should be able to use the QEHOLDTIME and QORIGINALPOSTION variables? Im having Local/-channel members. And I'm trying to put these variables inside these channels (setting posotion in CALLERID(name).. But I can't use them it seems.. Tried to use them after the call has been bridged my a M()-macro in Dial. But they still show up as empty. Any ideas anyone? |
| 09:11.18 | Chainsaw | People still use 1.4? |
| 09:11.55 | nunne | Chainsaw, yeah. It's an embedded system. Why wouldn't people use it? :p |
| 09:12.42 | WIMPy | You should find tons of reasons in the changelogs. |
| 09:13.21 | Chainsaw | nunne: Because Asterisk 11 is where the action is, and people who want minimal churn are on 1.8? |
| 09:13.34 | Chainsaw | nunne: History has left you behind. You get to fend for yourself. |
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| 09:21.13 | nunne | WIMPy, I know WHY i should use it. And I do use version 10 on our server based systems. But for this particular customer I'm stuck with 1.4.. And I was under the assumption that setqueueentryvar vas avaible for this and functioned in this way :/ |
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| 10:21.20 | BorjaGVO | Hi everyone...I'm looking for a program that helps simulate high loads of SIP calls (with RTP). I saw soemthing about pjsip but not sure if there is a program as in their website offer a library. Any suggestions? Thank you... |
| 10:22.24 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:25.11 | unicron | hi borja, try going here and scrolling down to Traffic generators |
| 10:25.14 | unicron | http://www.voip-info.org/wiki/view/How+To+Debug+and+Troubleshoot+VOIP |
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| 10:43.09 | BorjaGVO | unicron: thanks |
| 10:47.02 | WIMPy | Wow. Cool feature. I can place and receive calls to other phones but not to applications. |
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| 11:41.25 | pietro | hello |
| 11:41.54 | pietro | I need to know "Our Tag" and "Their Tag" in my dialplan |
| 11:42.40 | pietro | is there a way ? (without a system(asterisk -rx "sip show chan $CALLID| grep …." ? |
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| 11:58.54 | funky1 | hi guys, still having some audio problems with my asterisk, incoming calls works fine, with outgoing calls the incoming audio works fine, but outgoing audio from asterisk sip client comes and goes during call, a few minutes it is working fine then it is gone and then it comes back again what could the reason be, any ideas? |
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| 12:01.49 | *** join/#asterisk vetal_ (~chatzilla@213.154.200.243) |
| 12:03.09 | vetal_ | Dear All, I got problem on Asterisk 10.3.0 - with SIP messages, auth_message_requests = yes in sip.conf, but there is still possibility to send messages from unauth sips |
| 12:10.51 | *** part/#asterisk beaver_rrr (beaver@crabhost.org) |
| 12:15.24 | fling | Hello! Are you using Voxeo? |
| 12:18.47 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
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| 12:22.27 | vetal_ | no |
| 12:23.56 | fling | I need a voxeo guru to help me getting +99. DID |
| 12:24.07 | fling | I can't figure it out how to do so |
| 12:24.28 | fling | http://markusgoebel.blogspot.ru/2008/03/free-bridge-from-skype-to-phone.html |
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| 12:34.03 | fling | funky1: a lag? |
| 12:34.45 | slav3_kitten | fling, http://evolution.voxeo.com/ click the create an account |
| 12:34.58 | fling | slav3_kitten: clicked! |
| 12:35.19 | fling | slav3_kitten: I already have three accounts :P |
| 12:35.33 | slav3_kitten | are they evolution developer accounts? |
| 12:36.05 | fling | slav3_kitten: umm probably, I may create some apps with free phone numbers |
| 12:36.31 | slav3_kitten | the article specificially states evolution developer acount |
| 12:36.34 | slav3_kitten | yawns |
| 12:41.10 | fling | slav3_kitten: yes, I log it at http://evolution.voxeo.com/ |
| 12:41.34 | slav3_kitten | email em then |
| 12:41.50 | slav3_kitten | you american fling ? |
| 12:42.18 | fling | no slav3_kitten I'm russian |
| 12:42.42 | slav3_kitten | well hell, now i can't wish you a happy holiday |
| 12:42.55 | fling | omg why |
| 12:43.09 | slav3_kitten | P.S. i should call you to verify russianess (i love russian accents ^.^) |
| 12:43.19 | slav3_kitten | fling, because you guys don't have a thanksgiving today |
| 12:43.25 | fling | lol, ok :p |
| 12:43.34 | fling | slav3_kitten: and thank you! :] |
| 12:44.13 | slav3_kitten | wait what did i d |
| 12:50.11 | vetal_ | Is any ideas: I got problem on Asterisk 10.3.0 - with SIP messages, auth_message_requests = yes in sip.conf, but there is still possibility to send messages from unauth sips |
| 12:55.51 | funky1 | fling: how could it investigate that? could also be that my router has sip alg as slav3_kitten suggested but would in that case audio return during the call? would a sip alg not mess up all audio so i would not hear anything all the time? |
| 12:58.19 | slav3_kitten | funky1, what's up man? |
| 12:59.06 | fling | funky1: arsterisk -rvvvv |
| 12:59.39 | fling | funky1: make shure you have qualify option set for your peer |
| 12:59.55 | fling | funky1: so you will see the lag in the console in this case |
| 12:59.57 | slav3_kitten | funky1, first off... ALGs will do screwy stuff sometimes. i'd call them unpredictable |
| 13:09.07 | funky1 | hm i'm afraid it is the ALG, just called my provider, nothing they can do about it, will try the qualify suggestion and see if i can see something in cli |
| 13:09.16 | funky1 | thanks for the suggestions so far! :) |
| 13:09.36 | vetal_ | I found when it happend |
| 13:10.33 | vetal_ | while auth_message_requests = yes, if send message from 100, and there is peer 100 - it is declined, but if you send from 101 (it is not in peers) message accepted |
| 13:11.12 | WIMPy | Do you have guests allowed? |
| 13:13.15 | vetal_ | yes |
| 13:14.36 | nunne | SIP-ALG is a nightmare.. I usually never get it working with it. Usually you can get more success actually puttin nat=no when dealing with it - but i'm not sure qualify will help since it will get picked up my the ALG anyway. |
| 13:14.49 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:15.02 | nunne | you dont have access to the equipment yourself? |
| 13:16.22 | vetal_ | WIMPy, is there is way to allow quests for calls, but do not allow for messages? |
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| 13:17.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:20.21 | WIMPy | vetal_: I have no idea. |
| 13:21.36 | fprior | backups and redundancy, what's better or what are you using: rsync or raid ? |
| 13:22.03 | WIMPy | both |
| 13:22.17 | WIMPy | One is not a substitute for the other. |
| 13:22.34 | slav3_kitten | WIMPy, you know anything about cisco 7911's |
| 13:23.48 | WIMPy | nope |
| 13:24.02 | slav3_kitten | damn |
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| 13:26.13 | fprior | WIMPy, so RAID1 to redundancy and rSync to backup files to another machine. is correct ? |
| 13:26.48 | WIMPy | yes |
| 13:33.57 | nunne | fprior, think of raid as only a hardware guarantee, not software. if you loose a harddrive for example. I had a customer last week that has raid and got a faulty drive. inserted new one, it got rebuilt.. but the whole partition table was lost forever.... they had NO BACKUP.. was able to recover the files, but now all files has cool "hard disk sector" names like f02304230432.doc etc ;) |
| 13:34.25 | nunne | I think they will buy an backup-option now though :P |
| 13:35.02 | WIMPy | So the raid didn't work, either. |
| 13:35.21 | slav3_kitten | nunne, hardware raid is fairly good. i've seen many problems like that with software raid |
| 13:36.29 | WIMPy | Hardware raid has the big disadvantage thet you easily get screwed if the controller dies and you can't find a compatible rplacement. |
| 13:36.36 | fprior | nunne, that was a problem of raid. |
| 13:36.38 | nunne | slav3_kitten, was de-facto software raid.. nvraid.. i'm fairly confident dmraid.. but nvidias software raid is really broken |
| 13:37.00 | WIMPy | And unfortunatly the hardware versions are often slower. However they managed to do that. |
| 13:37.17 | slav3_kitten | that is a really good point |
| 13:37.50 | nunne | WIMPy, I think you should go with hardware raid sollution that is known to be supported for quite some time. like if you buy HPs smart-array cards etc. |
| 13:38.07 | fprior | Asterisk peformance could be affected from raid ? |
| 13:38.42 | nunne | but the nvidia raid was terrible.. it crashed the windows server after a few minutes until i was able to update the driver. and it was hard to do that in the little timeframe you had before it crashed ;) |
| 13:38.55 | WIMPy | You know what th eI in RAID stands for? So that's usually not an option. |
| 13:38.58 | nunne | fprior, depends on what your doing with your asterisk I should say |
| 13:39.46 | *** join/#asterisk salz212 (~chatzilla@203.215.176.22) |
| 13:39.52 | fprior | nunne, nothing special, minimum number of simultaneous calls |
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| 13:40.20 | salz212 | Are there any success casing of using Asterisk' Channel variable RTPQOS Audio all for calculating MoS ? |
| 13:40.21 | nunne | fprior, don't think you will notice anything really. |
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| 13:42.19 | *** join/#asterisk MatthewJava89 (59ef7f9e@gateway/web/freenode/ip.89.239.127.158) |
| 13:42.27 | MatthewJava89 | hi i have a problem |
| 13:42.41 | *** join/#asterisk iulhk (~iulhk@182.185.163.186) |
| 13:42.47 | MatthewJava89 | how overwrite dst in cdr |
| 13:43.57 | MatthewJava89 | when i call to my asterisk is dst is number "9412421" i want write to dst "11" |
| 13:43.59 | [TK]D-Fender | You don't. |
| 13:44.18 | [TK]D-Fender | It's read-only. |
| 13:44.19 | WIMPy | Goto |
| 13:44.28 | iulhk | using realtim asterisk, is there any way to use same sip name by using realm in same sippeers table ? |
| 13:44.58 | MatthewJava89 | realtim?? i use agi |
| 13:45.41 | [TK]D-Fender | MatthewJava89, He isn't talking to you. |
| 13:46.17 | MatthewJava89 | ok:) |
| 13:46.52 | MatthewJava89 | i want replace number call with "93142" to "11" |
| 13:47.08 | MatthewJava89 | how replace in dst ? |
| 13:47.31 | nunne | MatthewJava89, maybe you can use the userfield collumn? |
| 13:47.36 | WIMPy | MatthewJava89: Goto 11 and put your stuff there. |
| 13:47.55 | WIMPy | Goto is the only way to change destination. |
| 13:48.53 | MatthewJava89 | hmm i dont change destination real i want replace save number to database cdr column destination |
| 13:49.14 | WIMPy | No go |
| 13:49.42 | nunne | MatthewJava89, if you can control the application you use for outputting cdr info you can use the userfield for "custom information". |
| 13:49.54 | nunne | so just put 11 in the userfield and use that for data collection |
| 13:50.15 | MatthewJava89 | ok thx |
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| 14:19.22 | iulhk | can we use different realm with one asterisk server ? |
| 14:20.40 | bulkorok | iulhk: you mean one server with different IPs? |
| 14:24.41 | iulhk | bulkorok: if i am using one asterisk server, i hv two different clients, (clientA, clientB) both client will be using default sippeers table for realtime registration, i want clientA end-users' can register name john as sip peer as well clientB end-users' can also register name john as sip peer, is it possible? |
| 14:25.11 | *** join/#asterisk deo (~deo@112.198.90.89) |
| 14:25.43 | bulkorok | I suppose yes... but in theory both clients would ring at the same time |
| 14:25.49 | *** part/#asterisk deo (~deo@112.198.90.89) |
| 14:26.56 | nunne | iulhk, is it necessary to have same username on the sip peer? |
| 14:27.23 | nunne | isn't that something you can solve within the dialplan instead? |
| 14:28.02 | iulhk | <nunne>: this is requirement, i hv heard that it can possible by using different realms, so that's y asking :) |
| 14:29.40 | freckle | iulhk: you can have multi domain support in kamailio... not sure Asterisk can do it |
| 14:30.12 | iulhk | in sippeers table if i will change unique key against name then it can be handle by using agi, but don't what happened if i will change sippeers.name uniqueness |
| 14:31.06 | iulhk | <freckle>: yes, exactly, the person who told me, that company using kamailio, is there any posibility in asterisk ? |
| 14:31.36 | [TK]D-Fender | <iulhk> bulkorok: if i am using one asterisk server, i hv two different clients, (clientA, clientB) both client will be using default sippeers table for realtime registration, i want clientA end-users' can register name john as sip peer as well clientB end-users' can also register name john as sip peer, is it possible? <- NO |
| 14:31.51 | [TK]D-Fender | realm is not a class of separation in *. |
| 14:32.19 | nunne | iulhk, I think realm is usually used for seperating passwords between proxys. but i dont think asterisk can handle many peers with the same actual name |
| 14:32.34 | [TK]D-Fender | It can't. Absolute dead-end. |
| 14:32.44 | iulhk | <[TK]D-Fender>:Got it :) |
| 14:32.52 | nunne | ie only the passwords will be different between realms. but i dont know if its possible to have different realms in asterisk either? :P |
| 14:33.25 | [TK]D-Fender | nunne, It will apply to comms to the peer entry, but not allow 2 to share the same name, etc. |
| 14:33.56 | nunne | [TK]D-Fender, I see. yeah. I don't really think asterisk was designed at all with this in mind? |
| 14:34.20 | [TK]D-Fender | nunne, Not at all. |
| 14:34.36 | [TK]D-Fender | nunne, * is not a SIP proxy, switch, gateway, etc. |
| 14:34.41 | [TK]D-Fender | ~b2bua |
| 14:34.42 | infobot | b2bua is, like, a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 14:34.47 | *** join/#asterisk barkabus (~dario@2.176.43.195.dsl.static.ip.kpnqwest.it) |
| 14:34.49 | [TK]D-Fender | D-U-M-B. |
| 14:35.01 | [TK]D-Fender | Don't get too creative :) |
| 14:35.26 | [TK]D-Fender | So go shove kamalio, SER, etc in front and use * for back-end services |
| 14:37.53 | nunne | [TK]D-Fender, yeah.. And thats the way I would like it to stay :D |
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| 14:46.31 | navaismo | the r-series works fine with centos 6.3 any OS its ok? |
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| 16:24.49 | *** join/#asterisk con3x (~pkinnaird@kobol.geeksoc.org) |
| 16:24.55 | con3x | Hello |
| 16:35.40 | *** join/#asterisk freckle (~jon@firebox.enta.net) |
| 16:41.12 | con3x | Hello |
| 16:45.43 | navaismo | ~ask |
| 16:45.43 | 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. |
| 16:50.54 | con3x | Well then, does anybody have any experience setting up ISDN in the UK on BT using DAHDI and a HFS-C BRI card, we've got the card connected but we get an error that it cannot get a TEI. |
| 16:51.59 | WIMPy | Do you have a ptp or a ptmp line? |
| 16:52.35 | con3x | As far as we're aware ptmp, BT haven't been very helpful |
| 16:53.35 | WIMPy | Because that's the difference. ptp doesn't have TEI management. |
| 16:54.13 | WIMPy | But unless you plan to connect other devices in parallel, cofiguring ptp should always work. |
| 16:55.52 | con3x | Nope just the one switch connected up to our primary line, is there any setting that would cause dahdi to force TEI? |
| 16:59.40 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
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| 17:01.59 | WIMPy | What do you mean by that? |
| 17:02.16 | WIMPy | You can switch to ptp o skip that part. |
| 17:02.25 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 17:03.23 | WIMPy | Or are you thinking about a static TEI? I'm not sure any of the drivers can do that. |
| 17:04.51 | paulc | Are they any a2billing users around? I'm trying to get web callback to work and having troubles.. the call makes it to cc_callback_spool, and a2b-callback-daemon picks it up, but the result ends up being ERROR every time and I'm not sure why. Is there a way to debug what's being sent in through AMI and see its full response back? |
| 17:08.44 | con3x | WIMPy: Two seconds and I'll post the error after I get this box back up. Will it still connect if I swap modes? |
| 17:10.08 | con3x | Here's how its set up at the minute :) |
| 17:10.42 | con3x | http://pastebin.com/JnhLXYF3 |
| 17:13.59 | WIMPy | ptp should work, even if it's wrong. |
| 17:15.21 | WIMPy | But if you get a message that it can't get a TEI that suggests that your line doesn't have TEI management. |
| 17:17.18 | con3x | I'll give it a try and see if it works :) Thanks for your help |
| 17:23.01 | SeRi | done with my online shopping for christmas |
| 17:30.29 | *** join/#asterisk cusco (~tralala@ovh.tretas.eu) |
| 17:30.32 | cusco | hi |
| 17:30.46 | cusco | I'm having a issue that I am not sure if it is networking or configuration |
| 17:30.55 | cusco | let me try and describe it |
| 17:32.01 | cusco | we have a machine in our partners office with pptp-client dialing up to our router, its ip is 10.10.20.1, and our router's is 10.10.20.2 ... then we have our asterisk here behind our router 10.100.100.5 that can comunicate with 10.10.20.1 |
| 17:32.21 | cusco | thing is the remote asterisk tries to send sip to 10.10.20.2 instead of 10.100.100.5 |
| 17:48.26 | *** join/#asterisk italorossi (~textual@189.124.196.68) |
| 17:50.18 | navaismo | ~book |
| 17:50.19 | 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 |
| 17:55.15 | *** join/#asterisk serafie (~erin@76.73.167.231) |
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| 18:17.14 | sereal-work | I was looking for something on google and I saw this call log, what does JaK mean? |
| 18:17.16 | sereal-work | <--- SIP read from UDP:209.213.178.252:1041 ---> |
| 18:17.16 | sereal-work | jaK |
| 18:20.27 | sereal-work | oh this is the full log... http://www.freepbx.org/forum/freepbx/users/forward-call-from-sip-trunk-to-dahdi-zap-trunk not that it matters |
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| 18:31.32 | AviMarcus | Hey. Anyone know how to call UK tollfree for free? my past carriers don't seem to be working. And I rarely have traffic there. |
| 18:32.47 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 18:38.37 | [TK]D-Fender | sereal-work, Just a Keep-alive |
| 18:39.07 | [TK]D-Fender | sereal-work, junk SIP packet just to keep a NAT whole open. |
| 18:39.21 | [TK]D-Fender | sereal-work, Same purpose as SIP Options functionally |
| 18:40.22 | sereal-work | ah thanks. |
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| 19:01.56 | dfgas | ughhhh |
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| 19:25.03 | AndChat311001 | hi! im Javier, from Argentina. I wanna make a question, about odbc configuration for asterisk running on debian |
| 19:30.56 | *** join/#asterisk kontinuity (~Adium@122.167.73.248) |
| 19:30.59 | kontinuity | hi all |
| 19:31.12 | kontinuity | is there an ubuntu repo for Asterisk 11? |
| 19:31.22 | kontinuity | ubuntu precise to be precise :) |
| 19:32.42 | [TK]D-Fender | kontinuity, Nope |
| 19:33.16 | kontinuity | [TK]D-Fender: will there be one? or is everyone expected to build one from now onwards? |
| 19:34.17 | [TK]D-Fender | kontinuity, There is no "expectation" except that which you invent, and the isn't any announced yet that I'm aware of. |
| 19:35.01 | *** join/#asterisk dereksky (~derek@unaffiliated/dereksky) |
| 19:40.03 | derek | hi guys! quick question here |
| 19:40.31 | derek | would you recomment/discourage having asterisk running into a VM ? on an ESX host |
| 19:42.44 | *** join/#asterisk ujjain2 (~ujjain@unaffiliated/ujjain) |
| 19:44.06 | kilgorex | derek: This came up in the asterisk mailing list a few months ago, basically the responses were " you can do it, but don't share the machine with other VM's, especially something like Exchange". |
| 19:44.50 | kilgorex | derek: I'll see if I can find the post... Might help... |
| 19:45.53 | derek | i'm using asterisk since a couple of years… got a couple of setup there and there… but one of them is currently running in a VM (on a host with two other vms that are not really resources hungry) |
| 19:46.29 | derek | everything was fine but since a week, almost all of the sip peers are becoming unreachable and then reachable each 2 minutes or so |
| 19:46.47 | derek | and i'm really knocking my head off on this issue |
| 19:49.42 | kilgorex | derek: I think that's where this particular mailing list post was going, having problems but not sure if the VM was at fault. |
| 19:50.24 | derek | i'm not sure too… we allocate a dedicated nic and reserved some cpu for that particular vm |
| 19:50.26 | derek | no luck |
| 19:50.58 | kilgorex | derek: sorry, not sure if the other VM services causing prob... |
| 19:51.43 | derek | we shutted them down as a test and nothing |
| 19:55.34 | kilgorex | derek: do you get anything from debugging the SIP, are you sure it's down to the VM? |
| 19:56.23 | derek | kilgorex: actually i don't blame the vm or anything… i'm trying to figure out what is going on |
| 19:56.39 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:56.58 | derek | kilgorex: i don't see anything special in the sip debug… neither in a pcap on the server and the client |
| 19:57.20 | derek | kilgorex: we experience the issue using both physical phones and softphones |
| 20:12.13 | *** part/#asterisk kontinuity (~Adium@122.167.73.248) |
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| 20:43.38 | *** join/#asterisk anskywalker (~anskywalk@181.29.135.122) |
| 20:43.55 | anskywalker | hello, im javier from argentina |
| 20:44.01 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 20:44.53 | anskywalker | i have a trouble configuring sip users through |
| 20:44.57 | anskywalker | odbc |
| 20:45.55 | anskywalker | i hope someone can help me |
| 20:45.59 | anskywalker | tanks |
| 20:46.18 | [TK]D-Fender | ~ask |
| 20:46.18 | 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. |
| 20:51.13 | anskywalker | i followed all steps on asteriskdocs.org to configure sip users in a mysql database using extconfig realtime. now, if i write "odbc s |
| 20:51.22 | anskywalker | "odbc sh |
| 20:51.50 | anskywalker | "odbc show" on asterisk console it say "yes" |
| 20:52.12 | anskywalker | but sip users are not loaded from mysql table |
| 20:53.09 | anskywalker | i think i have some mistake in the sip configuration, to tell asterisk to search sip users in the database |
| 20:53.44 | anskywalker | i followed the same steps from asteriskdocs website |
| 20:54.25 | anskywalker | chapter 16 |
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| 21:01.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:02.21 | *** join/#asterisk anskywalker (~jazanon@200.69.203.181) |
| 21:03.54 | anskywalker | i configured asterisk to read sip data from a mysql table using odbc. if i write 'odbc show' over asterisk console it say 'conected = yes' so im connected to my database. but asterisk doesnt read the sip data from the table i set up on extconfig.conf |
| 21:04.14 | anskywalker | still search the sip user data on the users.conf |
| 21:06.10 | [TK]D-Fender | You also need res_odbc.conf setup.... |
| 21:06.36 | anskywalker | i have it |
| 21:08.48 | anskywalker | how can i share my config files to you? |
| 21:08.58 | anskywalker | i don't wanna paste all over irc |
| 21:09.12 | navaismo | ~pb |
| 21:09.12 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 21:11.56 | anskywalker | this is my res_odbc.conf placed in /etc/asterisk http://pastebin.com/BVfSdAQ7 |
| 21:13.45 | anskywalker | this is my extconfig.conf placed in /etc/asterisk http://pastebin.com/BnD6vbrZ |
| 21:15.01 | navaismo | this values "mysq_schema_name,table_name" are the real values?? |
| 21:15.19 | anskywalker | no... i just replaced the original values with that in order to protect the database |
| 21:15.52 | anskywalker | instead of that i write the real mysql schema name and the real table name |
| 21:16.18 | navaismo | ok |
| 21:19.17 | navaismo | and you have the asterisk connector in the odbc.ini? |
| 21:21.23 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:21.24 | anskywalker | yes... this is the file http://pastebin.com/93zdeLrP |
| 21:24.40 | anskywalker | and when y type "show odbc" on asteisk console, the console answers this http://pastebin.com/UkAPm6R3 |
| 21:24.55 | anskywalker | the "last connection attempt" value its really wrong |
| 21:25.56 | anskywalker | i don't know if i can trust on this command to check if asterisk connects to the mysql schema |
| 21:26.15 | navaismo | the connector works? ---> echo "select 1" | isql -v asterisk-connector |
| 21:27.34 | anskywalker | yes, this is the answer http://pastebin.com/tUC3k2Gg |
| 21:31.04 | navaismo | so everything looks like in the manual, when you run "sip show peer <yourpeer> load" from the cli what show asterisk? |
| 21:36.00 | anskywalker | it says "Peer not found." |
| 21:37.22 | navaismo | did you reload asterisk after setup the ODBC & extconfig settings? |
| 21:39.46 | anskywalker | yes I did it |
| 21:40.53 | anskywalker | i tried something... i changed the name of the mysql table in the extconfig.conf file |
| 21:41.01 | anskywalker | i write the name of an inexistent table |
| 21:41.14 | anskywalker | i reloaded asterisk |
| 21:41.22 | anskywalker | and it doesn't show any error |
| 21:41.54 | anskywalker | so i realice now i cannot see (or asterisk is not showing) database related errors |
| 21:43.21 | *** join/#asterisk blitzrage (~blitzrage@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:43.21 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 21:44.27 | ghost75 | is it enough to copy /var/lib/asterisk/astdb for db backup purposes ? |
| 21:46.51 | navaismo | anskywalker, you have the correct modules in the odbcinst.ini like Driver = /usr/lib/libmyodbc3.so |
| 21:46.52 | navaismo | Setup = /usr/lib/libodbcmyS.so |
| 21:47.34 | navaismo | anskywalker, this show your profile driver from linux shell: odbcinst -q -d |
| 21:48.34 | anskywalker | this is the odbcinst.ini http://pastebin.com/TBP5vbsE |
| 21:48.52 | anskywalker | i check and the libmyodbc files are in the rigth folders |
| 21:49.36 | anskywalker | this is the answer to the odbcinst -q -d |
| 21:49.36 | anskywalker | http://pastebin.com/FrJhVMfq |
| 21:51.26 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 21:52.59 | navaismo | anskywalker, can you login in mysql with that user and password and run: SELECT * from <your_sip_buddies_table>; |
| 21:54.44 | navaismo | the asterisk reload was a reload from cli or you shutdown the service with service asterisk stop then service asterisk start?? |
| 21:55.31 | anskywalker | yes i can login the mysql and select from the table data |
| 21:55.57 | anskywalker | i made the two things, a core restart gracefully and a service stop and start |
| 21:56.45 | navaismo | weird check the hints here to debug the issues http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Database_id247345.html |
| 21:56.56 | navaismo | i cant help you more since you seems to have all in order |
| 22:02.02 | anskywalker | ok, i will check that hints |
| 22:02.09 | anskywalker | thanks you navaismo |
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| 22:03.56 | navaismo | np |
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| 23:01.24 | diijiib | soup y'all |
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| 23:29.36 | dfgas | diijiib, did you get my pm? i also found a build of dd-wrt for my router |
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| 23:52.27 | SeRi | diijiib: jump in |
| 23:52.45 | SeRi | happy thanksgiving all |
| 23:54.48 | dfgas-cr48 | if it would be better i can flash that, otherwise centos is installed, ssh is installed and i am not sure what else. I found ipcop. little over 60mb and is gear to run atleast 486 |
| 23:55.36 | *** join/#asterisk FunkyGMT (~Adium@groupemtaconseil-139-2.cust.b2b2c.ca) |
| 23:56.15 | FunkyGMT | Good evening everyone.. |
| 23:56.54 | FunkyGMT | I have a quick question.. how much maximum should I "tolerate" on a linux server for Asterisk ? |
| 23:57.28 | navaismo | maximum of what?? |
| 23:57.36 | SeRi | dfgas-cr48: you there? |
| 23:58.22 | FunkyGMT | load |
| 23:58.27 | dfgas-cr48 | yup |
| 23:58.28 | FunkyGMT | forgot a word.. |
| 23:58.32 | FunkyGMT | how much maximum load* |
| 23:58.32 | carrar | depends on the number of CPU's |
| 23:58.37 | FunkyGMT | Quad Xeon |
| 23:58.43 | carrar | 4 |
| 23:59.21 | FunkyGMT | <PROTECTED> |
| 23:59.22 | FunkyGMT | Ok. |
| 23:59.29 | dfgas | in here now |
| 23:59.46 | WIMPy | load can mean anything. |