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| 01:01.52 | dgeary2 | anyone know of a free software softphone that runs on linux and supports iax encryption? |
| 01:03.39 | ledoktre | anyone on here know much about DEVICE_STATE? |
| 01:07.12 | xpot | dgeary2: http://bit.ly/PvNVx4 |
| 01:08.44 | xpot | ledoktre: what do you need to know? |
| 01:16.14 | ledoktre | xpot: I am trying to figure out one of two things. 1) Can I update SIP/some_extension directly, or if I go with Custom:, can I have my notifications (ring, idle, not_idle) sent to the Custom: one for a given extension ? |
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| 01:17.25 | ledoktre | sorry I didnt write clear. One or two. 1) Can I update SIP/some_ext via the dialplan -or- 2) If I use Custom:some_ext, that parts easy but can I have the DEVICE_STATE for SIP/some_ext written to Custom:some_ext instead? |
| 01:17.26 | [TK]D-Fender | ledoktre: No, you cannot modify an actual SIP device state like that |
| 01:17.34 | WIMPy | 1st part: no, 2nd part I don't understand. |
| 01:18.03 | [TK]D-Fender | ledoktre: You can make a COMPOSITE hint that will take the combined state. |
| 01:18.15 | dgeary2 | xpot, thanks for the lmgtfy reference using search terms 'iax', 'linux', and 'softphone'. i am already aware of sflphone (which either cannot do iax encryption or i am using an old version that does not support it) and am playing with simpleopal at the moment. |
| 01:18.23 | WIMPy | Ok, no, but you could watch the changes for the real device state and set the custom state when you see such an event. |
| 01:18.28 | ledoktre | [TK]D-Fender : I suspected that (about not updating the sip device itself. WHat is a composite hint? |
| 01:18.29 | [TK]D-Fender | ledoktre: exten => 123,hint,SIP/abc&CUSTOM........ |
| 01:18.43 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
| 01:18.50 | ledoktre | aahhhhh. So how do you know which takes presedence? |
| 01:19.17 | ledoktre | For example, if I update the CUSTOM to be BUSY, but the device itself is Idle, would BUSY trump? |
| 01:19.47 | WIMPy | the one that's not idle. For other states, you may have to consult the source. |
| 01:20.22 | ledoktre | As you're probably guessing, I am trying to setup a combined state / DND hint :-) |
| 01:20.40 | [TK]D-Fender | Highest state wins |
| 01:20.40 | ledoktre | That way can use my Buddies button to determine if someone is DND or InUse, etc |
| 01:21.14 | ledoktre | [TK]D-Fender, any idea where to find out a list of states priorities then? Where could I find the "pecking order"? |
| 01:21.44 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 01:21.47 | [TK]D-Fender | ledoktre: Should be on the voip-info wiki and I can't recall the source file.. its one of the header or main includes.... |
| 01:22.11 | ledoktre | alright. You (all) have helped a bunch. Answered how to combine states anyway :-) THANK YOU! |
| 01:31.35 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 01:46.25 | ledoktre | [TK]D-Fender : I assume that with a combined state, the hints will show the highest one (useful for Buddies list), but the phone itself will not respect the combined state, it will just use the SIP device itself, correct? |
| 01:46.51 | [TK]D-Fender | "Phone respect"? |
| 01:47.08 | [TK]D-Fender | Don't follow you. A hint is a hint.... |
| 01:47.25 | [TK]D-Fender | Phone A does not look at phone B ever |
| 01:47.41 | [TK]D-Fender | Asterisk tells phones what the hints tell it to say. |
| 01:48.08 | [TK]D-Fender | * is not a SIP presences server or proxy so if Youre thinking of trying to set a state from the phone itself then that is a dead end |
| 01:48.09 | WIMPy | The phones could do it directly as well. |
| 01:48.56 | ledoktre | [TK]D-Fender: what I meant was that if I had like this : SIP/EXT21&Custom:EXT21. The phone will not see the combined state, asterisk will. If I set Custom:EXT21 to BUSY, I expect the phone would still ring if I tried, even though the hint will say it is BUSY. Right? |
| 01:49.02 | WIMPy | Isn't there something now? |
| 01:49.56 | WIMPy | You can check for whatever you want in your dialplan before sending the call to a phone. |
| 01:50.31 | [TK]D-Fender | ledoktre: There is no relationship to whether you can call a device and the state of some hint |
| 01:51.02 | ledoktre | WIMPy: I read about that. A hack for DND basically. |
| 01:51.29 | [TK]D-Fender | presense will not lead to actually preventing you from calling |
| 01:51.51 | WIMPy | DND is usually done on the terminal. But you still can do whatever you want. |
| 01:52.34 | *** part/#asterisk sustav (~vpp@76.73.166.16) |
| 01:55.10 | ledoktre | [TK]D-Fender : I think I'm tracking with you. The SIP notifies that are received are notices only. * receives them, not sets them. Phone does not care about what the hints say. So its a moot point my question of whether the phone would respect the Custom:EXT21 being set to BUSY. It would have to be done in the dialplan as WIMPy mentioned. |
| 01:56.34 | ledoktre | Anyway, this ought to work out quite well either way. It should be a way to have a combined state for the phones to 'watch', so that each user knows the status of the other extensions. |
| 01:59.49 | ledoktre | I am offsite right now, is there anyway to set DEVICE_STATE from cli? I'd love to test the 'core show hints' before I go back to the office tomorrow. |
| 02:00.35 | ledoktre | core set DEVICE_STATE |
| 02:00.51 | WIMPy | devstate change ... |
| 02:04.25 | ledoktre | WIMPy : that did it. It appears like BUSY takes over IDLE, as does RINGING. Will still need to try and find that doc |
| 02:06.01 | WIMPy | As I said >> The one that's not idle. For other states, you may have to consult the source. |
| 02:08.49 | ledoktre | WIMPy : Yup. Working as you said. Thanks for the devstate tip. Worked perfect. |
| 02:09.08 | ledoktre | I've only got one more question, if anyone's up for it. |
| 02:16.33 | *** part/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 02:17.40 | ledoktre | For various reasons, I needed to compile * from source. It installed as root. My other box is running under asterisk user. I wanted to change it to also run as asterisk. I chown'd all the files that asterisk owns. I set AST_USER. We're good there. Are there any other things to watch out for? For example, there is a folder under /proc that asterisk uses on both boxes. The one that... |
| 02:17.42 | ledoktre | ...was already set to asterisk has that folder/files owned by asterisk. This one the folder is still owned by root. I try to chown it, but it doesn't. Forget it? |
| 02:20.02 | WIMPy | Asterisk doesn't use anythin under /proc. And if you're talking about dahdi, that's dahdi. udev might be involved as well. |
| 02:21.26 | ledoktre | then I shall forget about it :) I wasn't sure if a kernel module of asterisk used it or what. |
| 02:22.02 | WIMPy | Read that as Asterisk doesn;t have kernel modules. |
| 02:23.24 | ledoktre | WIMPy : Point well taken. How I even found it was by issuing `cd /; find -uid {asterisk ID}`. Ran that on the box that had asterisk running already as asterisk to find out which files/folders I should chown. |
| 02:23.41 | ledoktre | Both systems were Debian Squeeze, so I figured it should be safe as both versions were very similar |
| 02:24.50 | ledoktre | Difference being one was tarball certified-asterisk-1.8.11-cert7 and the other was deb asterisk-1.8.11.1-1digium1~squeeze |
| 02:30.26 | puchoweb | Please help me understand difference between Asterisk 1.8 and Asterisk 10. I am a hobbist who is trying to install Asterisk using Virtualbox. Any clues appreciated. Thank you. |
| 02:31.36 | WIMPy | See the UPGRADE.txt |
| 02:32.07 | WIMPy | The most obvious difference is the new ConfBridge. |
| 02:32.22 | ledoktre | To WIMPy: and [TK]D-Fender: Thanks for setting me straight, for advice, tips, pointers, and answers. It was all a very big help. So, THANK YOU. Good night. |
| 02:32.35 | puchoweb | Why does YUM install instructions use Asterisk 1.6 at http://www.asterisk.org/downloads/yum webpage? Any clues appreciated? |
| 02:32.41 | WIMPy | Good point. |
| 02:32.48 | WIMPy | goes to bed as well. |
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| 02:47.03 | k1ng_ | hollo people of #asterisk, how are you doing today? |
| 02:47.21 | k1ng_ | i still cant figure out how do i add a user with alphanumeric |
| 02:47.50 | k1ng_ | so i can login to asterisk with sip client using aphanumeric username |
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| 02:59.33 | ledoktre | Hey back for a second, guys. Wondered if anyone in here has messed with Polycom phones and more specifically the ldap/corporate directory. I have mine working, but I really don't know how to use ldap (1st time!). On a regular address book entry in the polycom, I can set bw to 1, and it tells the phone to subscribe to that user's status. Anyone know if this is even possible on corporate... |
| 02:59.35 | ledoktre | ...directories for polycom phones? |
| 02:59.44 | ledoktre | Might be stuck just using local directories |
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| 05:08.10 | nickgaw | hi, what do I do if my Asus netbook hardly turns on after the EFI partition was wiped when pbx in a flash eraced everything? |
| 05:09.56 | nickgaw | I have the restore dvds |
| 05:14.34 | ChannelZ | use them? |
| 05:19.07 | ChannelZ | Hrmmm. I thought SILK was in Asterisk 10 but it doesn't seem to be in my 10.7.. did it get taken back out or something? |
| 05:19.11 | nickgaw | so you are saying the net book might not work with linux? |
| 05:24.50 | nickgaw | can a bios setting be causing the issues? |
| 05:26.49 | ChannelZ | maybe |
| 05:26.57 | ChannelZ | What exactly does it do? |
| 05:27.51 | nickgaw | boots up gets network access then goes into low power mode plugd in |
| 05:30.07 | ChannelZ | boots up as in linux begins to boot or you're just seeing the BIOS go by? |
| 05:30.42 | nickgaw | boots into linux as I can ssh to it briefly |
| 05:32.06 | ChannelZ | hmm I don't really know anything about PBXIAF, what distro it even is |
| 05:32.22 | nickgaw | centos 6.2 |
| 05:35.09 | nickgaw | what version do you use? |
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| 06:18.32 | ChannelZ | personally I'm on Ubuntu |
| 06:18.44 | ChannelZ | oops missed he left |
| 06:22.08 | ChannelZ | Erhm. Silk is a separate DL. |
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| 06:35.57 | morfin | where can i find table schema for asteriskcdrdb |
| 06:36.04 | morfin | for postgresql |
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| 06:43.29 | carrar | in contrib/realtime/postgresql/realtime.sql |
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| 07:54.11 | ennui2342 | morning all |
| 07:55.19 | ennui2342 | I have the same problem you probably see 10 times a day. External sip connection to freepbx setup. Five hours later resorting to irc |
| 07:55.52 | ennui2342 | firewall rules in place. tcpdump shows UDP packets coming in 08:54:06.398944 IP 188.29.197.86.51451 > 192.168.1.81.5060: SIP, length: 611 |
| 07:56.02 | ennui2342 | But I never see anything going out and device wont register |
| 07:58.14 | ennui2342 | Configured freepbx SIP with dyndns hostname and 192.168.1.0/255.255.255.0 |
| 08:09.25 | ectospasm | ennui2342: try #freepbx, we don't support FreePBX in here... |
| 08:09.27 | ennui2342 | Hmm this doesn't look quite right. When I do sip show settings I get Externaddr: x.x.x.x:0 |
| 08:09.45 | ennui2342 | k thx ectospasm will do |
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| 08:44.10 | ChannelZ | normally for Asterisk behind a firewall you must set externaddr and localnet in sip.conf but for FreePBX, no idea where one does that such that it doesn't get stomped on by FPBX |
| 08:46.36 | ennui2342 | I just nailed it - its cause I'm using the incredible pi distrib - it comes with a restrictive iptables! |
| 08:46.47 | ennui2342 | Didn't even think to check that - silly! |
| 08:47.33 | ectospasm | that'll do it |
| 08:47.53 | ennui2342 | on the good side I now have a raspberry pi pbx which I can connect to over my mifi device. Woot |
| 08:48.51 | ennui2342 | Just a bit of configuration and a iPod touch purchase and I can dump my mobile number |
| 08:49.24 | ectospasm | sounds scary |
| 08:50.11 | ennui2342 | Not too bad. I've had astrisk setup before on my mac with a sipura3000, but the pi is a much better option |
| 08:51.03 | ennui2342 | basically can save myself a few hundred on the iPhone 5 purchase by going for a touch then routing all calls via my home PSTn (which I have to have anyway for ADSL) or voip providers |
| 08:51.58 | ennui2342 | SMS is the only difficult bit as my options are crappy web interfaces |
| 08:52.13 | ennui2342 | Wondering if I can do anything by gating through to XMPP |
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| 10:03.12 | redax | hi |
| 10:03.36 | redax | is there anybody running asterisk on linux kernel 3.x ? |
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| 10:08.59 | ectospasm | redax: I'm trivially running it on 3.2.0... |
| 10:09.21 | ectospasm | ...I don't use it much for anything but as a quick way to look up commands |
| 10:25.53 | redax | ah. the main problem I have a HP Proliant ML110 G7 which has the Intel 6 Series chipset C200 |
| 10:26.15 | redax | and seems like the whole system performance is slower than expected. |
| 10:26.47 | redax | my collegue installed elastix on the HP, thus the kernel is locked on 2.6.18 |
| 10:27.23 | redax | now I installed a debian with kernel 3.2, and the performance seems to be reasonable (say fast like h*ll) |
| 10:28.16 | redax | thus I planning to kick out the elastix partition, and use the kernel 3.x with debian, just wanted to know either is it compile fine, and do we have any issues with the new kernels. |
| 10:29.08 | redax | ectospasm: question 1 is answered, it is compiles fine... thanks. |
| 10:30.05 | ectospasm | yeah, compiles fine for me |
| 10:31.16 | redax | the strange thing is, HP telling RHEL5 is supported on ML110/G7. |
| 10:31.25 | redax | :-/ |
| 10:32.30 | ectospasm | that's RHEL5... which is not what Elastix runs... |
| 10:34.13 | redax | isn' Centos5 == RHEL5 ? |
| 10:34.25 | ectospasm | no |
| 10:34.26 | redax | more-less... same kernel at least |
| 10:34.41 | ectospasm | CentOS is based on RHEL5, but it's not (always) a drop-in replacement |
| 10:35.31 | redax | ah. I though they sharing the same kernel, then the performance issues should be the same on rhel5 and centos5 |
| 10:41.13 | redax | btw. |
| 10:41.51 | redax | what is the most prefered distro for asterisk by you? |
| 10:42.28 | redax | debian/ubuntu vs centos6 for example |
| 10:43.57 | ectospasm | no real preference |
| 10:44.12 | ectospasm | I prefer Debian, but it doesn't really matter |
| 10:44.43 | ectospasm | ...it also depends on whether your site policy or personal preference needs to use distro packages |
| 10:44.50 | ectospasm | ...rather than building from source |
| 10:44.57 | ectospasm | I prefer source for staying up to date |
| 10:45.22 | ectospasm | but that makes it harder to maintain from a sysadmin perspective |
| 10:45.27 | redax | me too. just some concrete brained collegues prefer elastix/trixbox and friends |
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| 10:46.12 | redax | I wonder why all the binary asterisk/freepbx distros using Centos5 |
| 10:47.19 | ectospasm | because that has been the most popular with users, from what I can tell |
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| 11:05.05 | *** join/#asterisk OldMonk (~raju@59.177.74.247) |
| 11:05.46 | OldMonk | hi. when using "core show channels" is there any way to force asterisk to show the full channel ID instead of limiting it to 20 characters? |
| 11:06.21 | OldMonk | or some other way to list full channel IDs? |
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| 11:34.08 | OldMonk | is there any way to show confbridge conferences from the cli? |
| 11:34.41 | WIMPy | Did yo try the most obvious? |
| 11:34.47 | OldMonk | which is? |
| 11:34.51 | WIMPy | confbridge <tab> |
| 11:35.19 | OldMonk | sure, that doesn't produce anything |
| 11:35.39 | WIMPy | Do you hace ConfBridge at all then? |
| 11:36.05 | OldMonk | con<tab> completes to config and console |
| 11:36.07 | OldMonk | yes |
| 11:36.29 | WIMPy | Or are you on 1.8? |
| 11:36.33 | OldMonk | yes |
| 11:36.39 | WIMPy | Ah |
| 11:36.51 | WIMPy | Athe ConfBridge in 1.8 was VERY limited. |
| 11:38.00 | OldMonk | as for full channel ids, you can get them with 'core show channels concise', but it's deprecated |
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| 11:38.39 | OldMonk | so no way to get active conf information in 1.8? |
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| 11:59.17 | WIMPy | I can't remember. |
| 11:59.45 | WIMPy | But ConfBridge in 1.8 wasn;t much mor than a proof of concept I think. |
| 12:00.14 | WIMPy | For the real thing use 10 or 11. |
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| 16:01.18 | [ProB]CrazyMan | hi |
| 16:02.34 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 16:03.02 | [ProB]CrazyMan | is there an dirty trick to get translation between g729 and alaw to work? the diginum license is accepted.. but I always get "Dropping incompatible voice frame on IAX2/722323-3303 of format g729 since our native format has changed to 0x8 (alaw)" |
| 16:08.33 | *** join/#asterisk Tim_Toady (~fuzzy@193.92.229.192.dsl.dyn.forthnet.gr) |
| 16:09.04 | *** join/#asterisk DropsOfSerenity (~justin@cpe-76-169-90-202.socal.res.rr.com) |
| 16:09.04 | [TK]D-Fender | You are likely passing your maximum # licenses |
| 16:09.23 | [TK]D-Fender | Show a complete call with a license dump before & after |
| 16:11.42 | [ProB]CrazyMan | 0/0 encoders/decoders of 2 licensed channels are currently in use |
| 16:11.59 | [ProB]CrazyMan | before the call and also during the call its 0/0 |
| 16:17.11 | [TK]D-Fender | is the call dropping? |
| 16:17.51 | [ProB]CrazyMan | no, I just dont here anything and the cli gets flooded by the dropping incompatible message |
| 16:19.03 | [TK]D-Fender | hrm |
| 16:20.12 | [ProB]CrazyMan | Im not shure if this is a problem, so I call from my Iphone via sip G729 --> to an Asterisk in the internet --> IAX G729--> Asterisk in LAN --> SIP alaw |
| 16:20.36 | [ProB]CrazyMan | the astersik in local lan has the g729 license |
| 16:25.13 | [TK]D-Fender | Standard passthrough shouldn't be a problem, but I have seen that when you use a local channel internally even if both ends are G.729 it forces through SLIN.... |
| 16:25.21 | [TK]D-Fender | Yesterdays revelation |
| 16:27.08 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:27.35 | [ProB]CrazyMan | ok and this is a problem? |
| 16:29.08 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:31.19 | [TK]D-Fender | Do your cals use them? |
| 16:31.22 | [TK]D-Fender | it might |
| 16:31.50 | [ProB]CrazyMan | SLIN ? now |
| 16:32.14 | [ProB]CrazyMan | normaly everything here is alaw, but for mobile usage g.729 would be good (maybe) |
| 16:32.48 | [TK]D-Fender | Now ready what I said AGAIN. |
| 16:32.57 | [TK]D-Fender | If you pass through a LOCAL channel |
| 16:47.05 | [ProB]CrazyMan | I dont understand what you want to tell me, but when it forces it to SLIN shouldnt it tell me that it uses SLIN ? |
| 16:52.44 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 16:57.15 | Gugge | [ProB]CrazyMan: if the call passes through a LOCAL channel (for example Dial(Local/something)) it could force SLIN |
| 16:58.13 | Marquel | morning. what would i put into a hint to monitor a single dahdi-channel? |
| 17:03.32 | [TK]D-Fender | Just that channel. The same way youd dial it |
| 17:05.32 | Marquel | [TK]D-Fender: usually i use dahdi groups and then dial a number behind it: DAHDI/G2/${ExTEN} |
| 17:05.59 | [TK]D-Fender | So So specify a channel and not grop |
| 17:06.11 | Marquel | exactly. |
| 17:06.52 | Marquel | you mean like 'exten => *21,hint,DAHDI/1'? |
| 17:07.07 | [TK]D-Fender | yes |
| 17:07.43 | Marquel | [TK]D-Fender: tried that. though the channel reported busy in *> dahdi show channel 1, the hint said "idle". |
| 17:08.42 | [TK]D-Fender | show us |
| 17:09.07 | Marquel | [TK]D-Fender: i'll try to reproduce... |
| 17:11.00 | Marquel | *CLI> core show hint *25 |
| 17:11.02 | Marquel | <PROTECTED> |
| 17:11.39 | [TK]D-Fender | PASTEBIN the whole pile including proof of what channels are in use |
| 17:14.33 | Marquel | that's the pile of show hint... |
| 17:15.51 | *** join/#asterisk jsjc (~Adium@61.Red-79-150-64.dynamicIP.rima-tde.net) |
| 17:17.37 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 17:19.11 | gavimobile | hey folks, I cannot get ipcomms to register.. im using their asterisk trunk sample from their website with my credentials. im also not receiveing anything in the loggs... im assuming I need to change something in the control pannel to forward the did to my ip which I cannot find |
| 17:19.17 | gavimobile | andyone with ideas? |
| 17:20.08 | [TK]D-Fender | idea : Actually look at the registration attempt |
| 17:20.08 | Marquel | well, before i try to pastebin anonymised info: core show channel doesn't know a channel "DAHDI/5", it only knows "DAHDI/i2", which is the group of dahdi channels... |
| 17:20.34 | gavimobile | [TK]D-Fender: I don't see any |
| 17:20.50 | [TK]D-Fender | "sip reload" <- You should |
| 17:21.05 | gavimobile | I did sip reload |
| 17:21.07 | [TK]D-Fender | "sip show registry" ,- And you should see that it tried |
| 17:22.19 | gavimobile | [TK]D-Fender: its not in the list |
| 17:22.25 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 17:22.48 | [TK]D-Fender | Then you didnt even configure your register statement properly |
| 17:23.00 | gavimobile | copy paste from their website |
| 17:23.09 | [TK]D-Fender | Means nothing to me.... |
| 17:23.37 | gavimobile | register=2012344204:xxxxxxxx@sipconnect.ipcomms.net/2012344204 |
| 17:23.50 | gavimobile | maybe it needs an arrow ? |
| 17:23.55 | gavimobile | register => 2012344204:xxxxxxxx@sipconnect.ipcomms.net/2012344204 |
| 17:24.13 | [TK]D-Fender | facepalms |
| 17:24.30 | [TK]D-Fender | When will you people learn not to paste just single lines. The big picture matters |
| 17:24.58 | gavimobile | you want the entire trunk confirg? no prob |
| 17:25.15 | specing | pours a bucket of paint over [TK]D-Fender |
| 17:25.43 | gavimobile | http://pastebin.com/YD1LCKqp |
| 17:25.54 | gavimobile | specing: I wouldn't do that if I were you |
| 17:26.41 | [TK]D-Fender | gavimobile: That is completely invalid |
| 17:27.01 | [TK]D-Fender | REGISTER statements have to occur after the end of [general] and BEFORE any other section. |
| 17:27.04 | [TK]D-Fender | Nowhere else. |
| 17:27.35 | gavimobile | [TK]D-Fender: I also tried moving it in general |
| 17:27.48 | gavimobile | ill go ahead and do that now.. thanks for teaching me something new |
| 17:27.54 | gavimobile | what else is no good there? |
| 17:27.54 | Marquel | [TK]D-Fender: i've did a bit of googling - seems there's no way to give a hint a specific dahdi channel and expect it to actually work. yet. seems i've got to wait until dahdi has been taken a bit further into fitting with sip... |
| 17:28.03 | Marquel | thx. for all your time and answers. |
| 17:28.19 | [TK]D-Fender | Marquel: it DOES works and weve done this for YEARS' |
| 17:28.24 | [TK]D-Fender | And you've shown nothing |
| 17:29.34 | gavimobile | [TK]D-Fender: thanks for that tip, im starting to get some logs! |
| 17:29.52 | gavimobile | ill troubleshoot more later, I gtg for now |
| 17:31.02 | Marquel | [TK]D-Fender: i've asked for a specific way on how to do it. i've got something. i've tried that. it did not work. i know you want more details, and i'd happily copy you an anonymised copy of my entire setup including any information you'd ask for. unfortunately "show the pile" is not exactly a statement i do understand particularly well when the line below is the single line of output i get. |
| 17:31.57 | [TK]D-Fender | All you get? |
| 17:32.00 | [TK]D-Fender | Lokok at a CALL. |
| 17:32.01 | Marquel | and yes, i know about pastebin, and now i know to use it for even two-liners. |
| 17:32.05 | [TK]D-Fender | Prove that line is in use' |
| 17:32.18 | Marquel | [TK]D-Fender: it was ringing. |
| 17:32.30 | [TK]D-Fender | Marquel*25@BLF_Group : DAHDI/5 State:Idle <- I see no watchers |
| 17:32.32 | [TK]D-Fender | I am not seeing it ringing |
| 17:32.36 | [TK]D-Fender | You are showing NOTHING |
| 17:33.42 | *** join/#asterisk wonderworld (~ww@dsdf-4db51104.pool.mediaWays.net) |
| 17:38.11 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 17:38.44 | *** join/#asterisk Carly- (~Wikipedia@unaffiliated/carly-) |
| 17:38.49 | *** part/#asterisk Carly- (~Wikipedia@unaffiliated/carly-) |
| 17:43.04 | Marquel | [TK]D-Fender: that's EXACTLY my point. |
| 17:43.49 | *** join/#asterisk sysnod1359 (~kunal@wf-171-99-190-54.revip9.asianet.co.th) |
| 17:44.15 | Marquel | the channel is ringing, but the hint shows it as idle. |
| 17:44.31 | [TK]D-Fender | It is not ringing... Otherwise youd have a call to show me. |
| 17:45.57 | Marquel | [TK]D-Fender: what exactly do you want to see? 'core show channels' doesn't show DAHDI/5 at all. only DAHDI/i2. 'dahdi show channels' shows it as being in use by the calling CID. |
| 17:48.03 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 17:50.03 | [TK]D-Fender | Ok, I'm done wasting my time waiting for backup... |
| 17:50.31 | Marquel | what EXACTLY do you want? |
| 17:52.25 | Marquel | besides. if you don't trust my statements - why should you trust my pastebin? i've said i'd anonymize it anyway. |
| 17:53.21 | Marquel | and if that is the case: i'm sorry for wasting your time. good bye. |
| 17:55.25 | sysnod1359 | hello. i am having difficulty with asterisk. whenver i try to start asterisk. it wont start. i cannot even access the sip.conf files via console or even through gui folders. it just hangs at the etc/asterisk/ folder. wont let me go further. also have tried to shut down asterisk but it wont. any advise how to deal with it? |
| 17:57.59 | Gugge | fix the filesystem / os |
| 17:58.30 | sysnod1359 | oh ok |
| 18:03.29 | *** join/#asterisk albania-asterisk (~chatzilla@92.60.22.232) |
| 18:03.46 | albania-asterisk | thello there |
| 18:06.37 | albania-asterisk | my server loadaver with asterisk is around 20 and i am scared if it will impact on the calls |
| 18:07.03 | albania-asterisk | i dont have strange message or error in cli of asterisk |
| 18:07.17 | albania-asterisk | but i giving top stats if you can have a look |
| 18:07.18 | albania-asterisk | http://paste2.org/p/2253709 |
| 18:08.09 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 18:08.49 | Gugge | albania-asterisk: crazy load, how many calls? |
| 18:09.19 | albania-asterisk | wimpy or channelz or anybody optinion is much appreciated |
| 18:09.33 | albania-asterisk | gugge: 1500 concurrent calls |
| 18:09.54 | Gugge | well, there is plenty of free cpu power |
| 18:10.29 | albania-asterisk | yes i see , but i am afraid i create any bottleneck somewhere else |
| 18:10.32 | Gugge | how busy are the disk system? |
| 18:10.36 | albania-asterisk | that i am not aware of |
| 18:10.43 | albania-asterisk | how to check them |
| 18:11.01 | Gugge | iostat maybe |
| 18:11.14 | albania-asterisk | let me check |
| 18:11.24 | rbowles | I'd suggest doing a 'taste test' of your server, which can be as simple as just putting a B2B call through it between 2 endpoints your trust and do some echo test or some other typical audio that you'll recognize if you are starting to drop/delay frames |
| 18:11.40 | jkroon | [TK]D-Fender, rofl @ discussion with [ProB]CrazyMan ... *sigh* ... busy trying to figure out WHY that happens. |
| 18:12.23 | albania-asterisk | http://paste2.org/p/2253728 for iostat |
| 18:12.25 | rbowles | when we load test things, we'll fill the box with SIPP calls (as long as SIPP can keep up) and then put through a B2B call listening to music or spoken word |
| 18:12.52 | [TK]D-Fender | jkroon: Caught between people who dont read, and secret squirrels |
| 18:13.03 | rbowles | iostat looks good to my eyes |
| 18:13.16 | jkroon | good luck with those ... my eyes are starting to hurt from looking at chan_local ... |
| 18:13.18 | albania-asterisk | this is production server , i ahve 1500 concurrent calls running perfectly |
| 18:13.36 | rbowles | right, add one more ... your test call |
| 18:13.40 | rbowles | and listen :-) |
| 18:13.48 | albania-asterisk | but things become so delicate on this numbers, and just wanted to get some precausios ideas |
| 18:13.58 | jkroon | albania-asterisk, kernel version? |
| 18:14.21 | albania-asterisk | 2.6 you can see on the paste link http://paste2.org/p/2253728 for iostat |
| 18:14.36 | jkroon | last time i saw load spike that high it was the leap-second bug issue in the kernel. |
| 18:15.25 | albania-asterisk | you think if can affect live traffic on this load somehow ? |
| 18:15.28 | jkroon | resetting the date using the date command should (if this is the case) result in an immediate drop of load. |
| 18:15.41 | jkroon | yes, i know it can, from painful experience. |
| 18:16.03 | albania-asterisk | what do you mean by resetting the date ? |
| 18:16.31 | jkroon | date -s "$(date)" |
| 18:16.38 | jkroon | assuming bash ... |
| 18:17.26 | albania-asterisk | and what does that command with date do ? |
| 18:17.50 | jkroon | takes the current date and sets the system date to it (ie, actually nothing except clear some arb bit in the kernel that causes problems) |
| 18:18.08 | albania-asterisk | i see let me try now then |
| 18:19.42 | albania-asterisk | did it , but dont see any decrease |
| 18:19.46 | jkroon | load should start dropping almost instantaneously if that's your problem. |
| 18:19.53 | jkroon | so that wasn't it, and load is probably legit. |
| 18:19.56 | albania-asterisk | was 19-20 dhe load when i did , and is same |
| 18:20.14 | rbowles | is that 1500 B2BUA calls? |
| 18:20.21 | rbowles | with media? |
| 18:20.35 | jkroon | do you still have idle cpu cycles? do you still have ram? what does your disk io look like? |
| 18:20.43 | albania-asterisk | yes those are p2p media through asterisk of course, transcoding can be only 10% |
| 18:20.46 | jkroon | ie, very simply: can your server deal with the load. |
| 18:21.14 | albania-asterisk | do you see "top" paste i copied here ? |
| 18:21.21 | albania-asterisk | did you... |
| 18:21.46 | jkroon | no, nor do i really care, honestly. it's questions you need to answer for yourself IMHO. |
| 18:22.38 | albania-asterisk | there are plenty of CPU idle cycles |
| 18:22.46 | jkroon | then why are you worried? |
| 18:22.50 | rbowles | i'll replug putting a test call through the box during the most loaded time and either use your ears or some pcap collection on both ends to know if Asterisk is truly not affecting call quality |
| 18:23.05 | albania-asterisk | almost 70% of the cpu is left |
| 18:23.09 | jkroon | has to back rbowles on that one. |
| 18:23.15 | *** join/#asterisk nixofortune (~ishevtsov@w-31.cust-5783.ip.static.uno.uk.net) |
| 18:23.58 | rbowles | having said that, however, seeing your call total tells me I've been paranoid down around 300 B2B calls :-) |
| 18:24.57 | albania-asterisk | because the loadaverag is high thats why i am worried |
| 18:25.08 | albania-asterisk | and i dont understand well the iostat output |
| 18:25.19 | jkroon | try iotop then |
| 18:26.45 | albania-asterisk | just installed and run iotop |
| 18:27.32 | albania-asterisk | its plenty of asterisk instances with read and write speeds, what figure should i look into more |
| 18:27.44 | jkroon | probably network IO |
| 18:30.16 | *** join/#asterisk k610 (~Instantbi@178-118-0-204.access.telenet.be) |
| 18:30.43 | albania-asterisk | please have a look at it, is it fine? http://paste2.org/p/2253774 |
| 18:31.04 | jkroon | albania-asterisk, MAKE A TEST CALL AND LISTEN! |
| 18:31.22 | jkroon | ooh, running asterisk as root are we? |
| 18:31.29 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:31.40 | jkroon | ok, get more ram, you're swapping. |
| 18:32.23 | albania-asterisk | test call i tried everything was clear, without no distortion, but i think it does not represent all live calls.. |
| 18:32.29 | jkroon | free -m should confirm |
| 18:32.32 | albania-asterisk | yes as root |
| 18:32.58 | jkroon | you should probably fix that |
| 18:33.22 | albania-asterisk | <PROTECTED> |
| 18:33.24 | albania-asterisk | Mem: 24097 23994 103 0 432 21364 |
| 18:33.50 | albania-asterisk | there is a lot of cached ram, should it be considered as free as well ? |
| 18:34.12 | albania-asterisk | free 103, cached 21364 |
| 18:35.00 | jkroon | very interesting that iotop is reporting swapin values when you have that much ram around ... |
| 18:35.11 | jkroon | seriously, i don't think you need to worry too much just yet. |
| 18:35.20 | rbowles | <goes to find iotop> |
| 18:37.03 | albania-asterisk | why it has to use swap then, if there is that much of ram cached ? |
| 18:37.14 | jkroon | no clue ... |
| 18:40.46 | albania-asterisk | is there any practical software limitation on the asterisk for concurrent calls ? |
| 18:41.01 | rbowles | probably :-) |
| 18:41.21 | jkroon | not that i'm aware off ... but i guess it depends on how much cpu, ram and network you can throw at the problem :) |
| 18:42.00 | albania-asterisk | thats why i put practically :) something that we experienced as highest |
| 18:42.19 | albania-asterisk | on production preferably |
| 18:42.20 | rbowles | often the walls you hit are configuration related |
| 18:42.25 | jkroon | depends on so many factors it's actually really hard to calculate |
| 18:42.31 | jkroon | eg, how much transcoding? |
| 18:42.42 | jkroon | direct path vs media going via ast |
| 18:42.55 | jkroon | how well does your NICs handle small packets |
| 18:42.57 | albania-asterisk | yes i should have said that |
| 18:43.12 | jkroon | also, how complex is your dialplan? |
| 18:43.16 | albania-asterisk | lets say 10% of transcoding and rest is media carried by asterisk on p2p |
| 18:43.33 | jkroon | jitter buffers? |
| 18:43.38 | albania-asterisk | dialplan does not contain sql calls |
| 18:43.45 | jkroon | there are too many variables to give you a simple answer. |
| 18:43.54 | rbowles | agree with jkroon |
| 18:43.56 | jkroon | sql is not the only complexity ... |
| 18:44.01 | albania-asterisk | yes but i would ask only by experience |
| 18:44.37 | albania-asterisk | lets say max number you encountered on production system preferably |
| 18:44.38 | albania-asterisk | ? |
| 18:44.53 | *** part/#asterisk Marquel (~Marquel@unaffiliated/marquel) |
| 18:45.20 | jkroon | well, on my system I estimate I should be able to deal with around 1500-2000 based on current ram + cpu consumption and extrapolation ... but even that is just a guess as some things scale linearly, others may scale polinomically and yet others logarithmically - so there is no real rules. |
| 18:45.57 | albania-asterisk | that is what i wanted to hear :), thanks |
| 18:46.09 | jkroon | you're 1500 is pretty high from my experience ... think the highest I ever had on my systems was around 300 odd ... no transcoding but no direct media either. |
| 18:46.19 | jkroon | *your |
| 18:46.30 | rbowles | yeah, 1500 on your system is the highest I've seen too :-) |
| 18:46.47 | albania-asterisk | really |
| 18:46.50 | rbowles | i hope you have a second or third box for sharing/failover |
| 18:47.01 | rbowles | that's a lot of calls to drop when your power supply fails |
| 18:47.09 | albania-asterisk | i was hoping you guys were dealing with same numbers |
| 18:47.15 | jkroon | rbowles, based on what i've read above that's a no :) |
| 18:47.19 | *** join/#asterisk vlad_starkov (~vlad_star@wn2nat43.beelinegprs.ru) |
| 18:47.30 | jkroon | albania-asterisk, getting there, but not just yet. |
| 18:48.17 | albania-asterisk | power supply is redundant in the server and also the datacenter has redudant power, this is not a issue |
| 18:48.23 | rbowles | i have about that many calls spread across 10 HP G7s :-) |
| 18:48.51 | rbowles | though I'm doing some other terribly inefficient things like running PERL scripts for each call and reaching out to databases :-) |
| 18:49.00 | albania-asterisk | wow too much servers |
| 18:49.05 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 18:49.12 | albania-asterisk | yes that is the difference |
| 18:49.14 | rbowles | there is never too much ... |
| 18:49.31 | rbowles | when your stuff falls over and you are hauled out to the woodshed |
| 18:49.53 | albania-asterisk | we are trying to keep simple dialplan only basically going through labels via after the hangup cause |
| 18:50.00 | jkroon | with rbowles on that ... i'd rather spread the risk as much as possible... |
| 18:50.16 | albania-asterisk | and all the realtions with the databases we do it out of asterisk application |
| 18:50.20 | albania-asterisk | by using the files |
| 18:50.30 | rbowles | and asterisk does crash |
| 18:50.43 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 18:50.50 | jkroon | although ... i haven't seen one (thankfully) in quite a long (8 months?) time |
| 18:51.08 | albania-asterisk | it never crashed actually |
| 18:51.26 | rbowles | could be my crappy PERL code :-) |
| 18:51.50 | albania-asterisk | but i am working for a carrier wholesale in albania |
| 18:52.34 | albania-asterisk | and basically the calls are coming and going simple handling the cause codes and random and gotoif is much on the dialplan |
| 18:52.38 | albania-asterisk | thats all |
| 18:52.38 | jkroon | and your point is ... ? most people here has something to do with telecoms somewhere, some for carriers, some for smaller companies ... |
| 18:53.12 | jkroon | fender is about the only person that has said he's "just here", and has no affiliation with telecoms ... go figure |
| 18:53.45 | albania-asterisk | my point is that i need some opinions from people who are dealing with 1000+ concurrent calls on one box every day |
| 18:54.00 | jkroon | spread the load. |
| 18:54.16 | albania-asterisk | we have 5 asterisk servers and we are spreading the load by kamailio |
| 18:54.24 | jkroon | let me ask you this: what's the financial impact of running a secondary box? |
| 18:54.31 | rbowles | I could remove my A record shuffle and see ... but ... well ... |
| 18:54.35 | jkroon | what's the financial impact of your single box going down for even 5 minutes? |
| 18:55.03 | albania-asterisk | the calls will go through other left boxes |
| 18:55.32 | rbowles | so you need to know if calls are NOT affected when peak load is on 4 boxes |
| 18:56.30 | albania-asterisk | basically if one server fails |
| 18:56.35 | rbowles | get your test call ready, and drop the A record shuffle in kamailio and see (or do this in a lab) ... learning the lesson while NOT in control is worse |
| 18:56.47 | albania-asterisk | easch 25% will be spread through left 4 boxes |
| 18:57.19 | rbowles | drop to 4 boxes under controlled conditions (is what I meant) |
| 18:57.27 | jkroon | yea, but the strain per server will increase by 25% as well from what it usually runs at :) |
| 18:57.36 | albania-asterisk | and this can increase the load of others too, i needed to hear some people who are working with huge volumes for carriers lets say and see the limits they are considering |
| 18:57.40 | rbowles | because that 5th box will die while you are out at a restaurant and sleeping |
| 18:58.27 | rbowles | or, without any testing, buy a 6th box and put it in and expand when all 6 are at 1500 |
| 18:58.58 | albania-asterisk | yes that is an option to consider |
| 19:00.40 | albania-asterisk | but if asterisk box concurrent calls can hit 3000 concurrent calls and still mantaining same quality , i need to do caluclations better so we are saving on hardware |
| 19:01.16 | jkroon | ... again ... financial impact of dropping 3000 calls compared to running an extra server ... |
| 19:02.35 | albania-asterisk | why there is a financial impact of dropping 3000 calls live but at the same time not causing a downtime period ? |
| 19:03.00 | jkroon | knows of at least one carrier that prefers to not run more than 100-200 calls per SBC simply because a SBC crashing and losing the CDR information costs them money |
| 19:03.02 | albania-asterisk | if there are a lot of dropping in the calls it will impact ACD and therefore have a financial impact |
| 19:03.38 | albania-asterisk | but if all are dropped once, it will not affect statistics |
| 19:03.56 | jkroon | well, if you drop those 3000 calls you're most likely losing that CDR info, preventing you from billing them but still being billed for them (even as a carrier) |
| 19:04.21 | jkroon | anyway, i'm off |
| 19:04.26 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 19:04.34 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:05.14 | albania-asterisk | yes in case of hardware problem , the CDRs of that moment live calls will be lost , you are right |
| 19:05.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 19:06.23 | albania-asterisk | but if we mantain the CDRs in two places at asterisk but also at the sip server (kamailio) we are still safe :) |
| 19:06.26 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 19:07.09 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:10.11 | albania-asterisk | so can you tell me again what kind of other financiall/technical impact can i have but mantaining the asterisk boxes with 1000+ concurrent calls ? |
| 19:10.31 | albania-asterisk | *by |
| 19:12.40 | albania-asterisk | any other idea ? |
| 19:16.12 | rbowles | I won't speak for jkroon, but you don't want your everyday load to be running at levels where you are not sure that if one box goes down hard, will the remaining boxes handle the load |
| 19:16.51 | rbowles | no matter where you ask about Asterisk load support, the informed answer will always be 'it depends' |
| 19:18.18 | rbowles | You've already successfully 'tested' Asterisk load to 1,500 calls ... on your hardware, you don't know how much higher you can go |
| 19:18.28 | albania-asterisk | i would be more curious/interested about current infrastructure supposively of carriers handling 20K concurrent calls with sip server and media through asterisk boxes , in order to see how they scale and precausios activities are they having in the map |
| 19:19.31 | rbowles | I don't know of any, so now you are stuck with the unknown of trying to find a carrier that a) uses Asterisk and b) puts more calls through similar hardware than you |
| 19:19.55 | albania-asterisk | for b) :) i am not asking |
| 19:20.22 | albania-asterisk | they can put less or more , it depends on how they thought about scalibility |
| 19:21.09 | rbowles | and while you are waiting to find that answer, do you know if 4 of your 5 boxes can handle 7,500 calls ? |
| 19:21.38 | albania-asterisk | but sure they are carriers using asterisk |
| 19:21.54 | albania-asterisk | *there |
| 19:22.12 | rbowles | most of the carriers I know are using freeswitch (if they are using open source) ... |
| 19:23.08 | rbowles | and they 'tell me' that freeswitch handles more traffic than asterisk, but I don't necessarily believe everything I hear |
| 19:24.19 | albania-asterisk | i never worked on a freeswitch box, i also heard that it goes more than 5K concurrent calls on box normally |
| 19:24.57 | albania-asterisk | i am very comfortable with asterisk and i am not the one changing this, but who knows |
| 19:25.17 | albania-asterisk | freeswitch has a good thing, that has the nokia sip stack based on RFC |
| 19:25.25 | rbowles | the 'conservative' estimates I've heard are 2,500 calls per core ... but surely it depends on all the other factors that folks mentioned here |
| 19:25.27 | albania-asterisk | but still ... |
| 19:26.27 | rbowles | with a lot of free time, I'd love to put Asterisk/Freeswitch/Yate in a lab and stuff maximum number of B2B calls with media through them and see |
| 19:27.04 | rbowles | are you the primary person responsible for those 5 boxes running 1500 calls? |
| 19:27.08 | albania-asterisk | i would pay to have that time :) |
| 19:27.33 | albania-asterisk | no i am senior engineer of routing and dialplan |
| 19:28.14 | rbowles | cool, so you and your team need to decide how much unknown can you handle |
| 19:28.21 | albania-asterisk | exactly there are two primary engineers taking care of kernel and system/asterisk failovers |
| 19:29.03 | rbowles | either test production load on 4 servers (controlled conditions), then you know that you can lose one box |
| 19:29.06 | albania-asterisk | kernel i mean linux environment |
| 19:29.23 | rbowles | or buy box 6 and wait for all 6 to approach 1500 calls load wise |
| 19:29.53 | rbowles | even if someone tells you they did 3,000 calls on a box similar to your 5, it still 'depends' and you don't know what will happen when you have 1 less box |
| 19:30.18 | albania-asterisk | yes we dont have problem to buy another one or two, we are also keen of reaching some big numbers close to the limits ;) |
| 19:31.09 | rbowles | you won't really know what the limit is until you detect call degradation on your test call under load |
| 19:31.47 | rbowles | whether you do that on your live system or in a lab is your choice :-) |
| 19:42.13 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 19:49.08 | *** join/#asterisk vlad_starkov (~vlad_star@wn2nat43.beelinegprs.ru) |
| 20:03.06 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 20:03.58 | nickgaw | Hi, My Asus netbook is now downloading pbx in a flash but it appears as if the bios has something that it could not unset the power save mode is this a hardware setting in the bios I need to turn off? |
| 20:06.33 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 20:07.13 | pzn | "core sip show peers" is this the right syntax? I upgraded asterisk in debian and this command seems to stop working... is the syntax correct? |
| 20:08.51 | pzn | I know "sip show peers" was in 1.2 and 1.4 versions... now which is the right command? |
| 20:12.13 | specing | sip show peers |
| 20:14.02 | pzn | don't care about my previous question. the bug is related to a debian security update this week that had broken packages... |
| 20:14.08 | pzn | specing, thanks anyway |
| 20:16.20 | *** part/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 20:19.01 | albania-asterisk | hello |
| 20:19.17 | albania-asterisk | is there a way to get sip status code on the dialplan using asterisk 1.4 ? |
| 20:19.38 | kuku | I'm trying to do chanspy with just using the sip extension ( without the channel ) it works 80% of the time, the other 10% it doesn't and the other 10% it connects to an entirely different extension... any advice ? |
| 20:21.57 | ChannelZ | What do you mean you're 'using the sip extension without the channel' ? |
| 20:24.03 | albania-asterisk | is there a way to get sip status code on the dialplan using asterisk 1.4 ? |
| 20:29.21 | *** join/#asterisk AreolaMonster (~AreolaMon@cpe-76-168-44-221.socal.res.rr.com) |
| 20:34.54 | AreolaMonster | Hello, Can someone help me figure out why CID doesn't get changed if it matches a number please? |
| 20:34.57 | AreolaMonster | here's what I have |
| 20:34.58 | AreolaMonster | http://sprunge.us/aOFG |
| 20:43.43 | morfin | hello |
| 20:44.14 | morfin | can anybody help me? cdr_pgsql can use schema? |
| 20:47.00 | morfin | i mean i have 2 schemas in cdr database and i want to specify another schema |
| 20:50.39 | *** join/#asterisk jsjc (~Adium@61.Red-79-150-64.dynamicIP.rima-tde.net) |
| 20:54.46 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
| 20:56.33 | albania-asterisk | is there any difference between the modules cdr-csv and cdr-custom, performance wise and/or guaranteeing no calls are missing ? |
| 21:00.22 | albania-asterisk | i need to know how much safe i am moving to using cdr-custom files for our billing , as they are separate modules? |
| 21:06.33 | *** part/#asterisk albania-asterisk (~chatzilla@92.60.22.232) |
| 21:06.33 | jpsharp | One works just as well as the other. |
| 21:06.35 | jpsharp | But you've already left. |
| 21:06.54 | *** join/#asterisk albania-asterisk (~chatzilla@92.60.22.232) |
| 21:07.26 | albania-asterisk | is there any difference between the modules cdr-csv and cdr-custom, performance wise and/or guaranteeing no calls are missing ? |
| 21:07.29 | albania-asterisk | i need to know how much safe i am moving to using cdr-custom files for our billing , as they are separate modules? |
| 21:09.19 | jpsharp | One works just as well as the other. |
| 21:15.57 | *** join/#asterisk mKn0wt (~mtv@190.181.158.37) |
| 21:20.04 | albania-asterisk | thanks jpsharp , i will proceed to make some testing |
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| 21:51.08 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 22:05.46 | *** join/#asterisk Platinum (~Platinum@72.42.68.245) |
| 22:06.19 | Platinum | hello |
| 22:17.04 | ChannelZ | ahoyhoy |
| 22:17.09 | Platinum | hello |
| 22:18.01 | Platinum | Does anyone have experience with g3si and asterisk? |
| 22:28.22 | ChannelZ | That might be a rare person to find particularly over the weekend |
| 22:29.34 | Platinum | Yes, I guess so. |
| 22:34.31 | volga629 | is nf_conntrack_sip.ko should accept port range ? |
| 22:35.40 | volga629 | [2012-09-22 18:22:22] WARNING[5596]: res_rtp_asterisk.c:2143 ast_rtp_read: RTP Read too short |
| 22:35.40 | volga629 | [2012-09-22 18:22:22] WARNING[30688]: chan_sip.c:9147 process_sdp: ignoring 'video' media offer because port number is zero |
| 22:37.33 | volga629 | I am second try understand what cause tried use tcpdump to check what happened, but can't find any explanation and looked on Google, but this is not a case |
| 22:37.37 | volga629 | day |
| 23:05.12 | ChannelZ | I never had luck getting conntrack_sip to do anything in the past, though I never did video either. It could just be completely kacked |
| 23:07.49 | *** join/#asterisk k610 (~Instantbi@178-118-0-204.access.telenet.be) |
| 23:18.04 | volga629 | I am using shorewall on this vpn box from where come video and voice stream I looked into modinfo and modified helper for sip nat to nf_conntrack_sip ports=5060 sip_direct_signalling=0 sip_direct_media=0 I will try test see if it help just run out of ideas what can not right |
| 23:18.33 | volga629 | be |