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| 02:50.41 | SeRi | p3nguin: I was testing :) |
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| 04:29.33 | SeRi | p3nguin: you avail? |
| 04:32.16 | p3nguin | Sort of. |
| 04:32.50 | SeRi | one sec |
| 04:34.42 | SeRi | I have g722 I wanted to see if you are avail to test how good it is |
| 04:34.55 | p3nguin | It will sound exactly the same to me. |
| 04:35.05 | p3nguin | I don't have any phones with wideband. |
| 04:35.24 | SeRi | ah I see |
| 04:35.31 | SeRi | ok cool. |
| 04:35.35 | SeRi | how you been? |
| 04:36.04 | p3nguin | Rather typical, it seems. |
| 04:36.15 | SeRi | how was christmas? |
| 04:36.51 | p3nguin | It was okay. I got a couple useful things. |
| 04:37.06 | SeRi | cool what you got? |
| 04:38.49 | p3nguin | I got a new polo shirt and a gift card for one of my favorite stores. |
| 04:42.49 | SeRi | nice!!!! |
| 04:44.50 | SeRi | I got a couple of things. I am happy :D |
| 04:44.59 | SeRi | My phone was my gift to my self |
| 05:04.24 | SeRi | :/ |
| 05:06.27 | SeRi | well here I am I just flashed my brothers xoom and I think it went ok...... not sure :/ |
| 05:06.54 | ChannelZ | I have a Zoiper beta that does wideband if you want it |
| 05:07.19 | k-man | i've been trying to upgrade a snom 300 from v7 to v8 of the firmware. it seems to go through the motions but when it reboots its still on v7 |
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| 05:12.15 | SeRi | ChannelZ: That would be awesome! |
| 05:12.25 | SeRi | that would help me test |
| 05:15.33 | SeRi | ChannelZ: Thanks |
| 05:15.46 | ChannelZ | Apparently you don't do DCC |
| 05:16.02 | SeRi | Mhhhh I thought I had it for auto send |
| 05:16.19 | SeRi | can you email it to me? |
| 05:16.26 | SeRi | I mean auto accept |
| 05:17.23 | ChannelZ | hmm yeah doesn't seem to be doing anything. I can email it |
| 05:18.46 | SeRi | ChannelZ: I sent you a pm. Thanks! |
| 05:21.33 | ChannelZ | np |
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| 05:41.59 | WIMPy | He, life is back :-) |
| 05:42.13 | WIMPy | k-man: Are you sure the file is correct? |
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| 07:56.59 | IsUp | morning |
| 07:57.09 | WIMPy | Hi IsUp |
| 07:57.23 | IsUp | how are you WIMPy? |
| 07:57.41 | WIMPy | Within usual limits :-) |
| 07:58.59 | IsUp | cool :) |
| 08:00.35 | WIMPy | So what are we going to break today? |
| 08:02.49 | IsUp | well, i'll run 'dd if=/dev/urandom of=/dev/sda' and i'll ask you why my Asterisk is not working. |
| 08:03.35 | WIMPy | Ok. Go ahead. |
| 08:03.46 | IsUp | :) |
| 08:04.03 | WIMPy | Id' use /dev/zero however. It's a lot faster. |
| 08:04.30 | IsUp | yeah. btw, u got SSD? |
| 08:05.08 | WIMPy | No, but I run my netbook from a SD card. |
| 08:05.30 | WIMPy | And I used to start my 2nd RPC from a CF card. |
| 08:05.39 | IsUp | seriously? |
| 08:05.51 | WIMPy | yes |
| 08:06.13 | SeRi | well |
| 08:06.56 | WIMPy | The bad thing it that the internal card reader, although it's PCI-e, is slower tan an external one connected via USB. |
| 08:07.12 | SeRi | waz up WIMPy |
| 08:07.25 | WIMPy | But it doesn't matter that much. The access times are the important bit. |
| 08:07.42 | WIMPy | I need a house. |
| 08:07.56 | SeRi | I need a few mill dollars. |
| 08:08.08 | WIMPy | That would help as well. |
| 08:08.09 | IsUp | :) |
| 08:08.47 | SeRi | :P |
| 08:08.49 | WIMPy | I got some drill bits. 600 and 1000mm. But I'm missing the walls to use them on. |
| 08:09.01 | SeRi | lol |
| 08:09.42 | WIMPy | But at least I can be reasonable sure now that I won't get stuck in a wall any more. |
| 08:10.05 | SeRi | lol |
| 08:10.31 | SeRi | I been hacking my brother motorola xoom. looks like wifi versions is very simple to hack |
| 08:10.48 | SeRi | I also got a cruz micro tablet that was very simple to hack as well |
| 08:11.10 | SeRi | both will run csip |
| 08:11.29 | IsUp | what can i run on busybox? |
| 08:11.54 | WIMPy | "on"? |
| 08:11.54 | SeRi | what do you mean? |
| 08:12.18 | WIMPy | busybox can contain a lot of stuff. |
| 08:12.25 | SeRi | indeed |
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| 08:14.28 | WIMPy | is generating md5s over his whole storage raid to find possible duplicates. |
| 08:19.40 | IsUp | i have a rouıter, its running on "memdisk" as far i understand. so it has limited space. |
| 08:19.50 | IsUp | is that possible to run sip client/server on it? |
| 08:20.28 | WIMPy | If you have enough memory. |
| 08:21.12 | IsUp | okay what can i run? |
| 08:21.53 | IsUp | ill test it |
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| 08:50.08 | IsUp | nah, i cant run anything |
| 08:50.53 | WIMPy | that is very little. |
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| 09:24.58 | IsUp | strange. we have no problems today. |
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| 09:34.45 | WIMPy | What kind of problem were you expecting? |
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| 10:00.15 | fish9370 | hello what for res_timing_timerfd.c ? can i unload it ? |
| 10:00.43 | WIMPy | Do you have dahdi running? |
| 10:00.50 | fish9370 | yes |
| 10:00.57 | WIMPy | Then you don;t need it. |
| 10:01.20 | fish9370 | and res_timing_pthread.c ? |
| 10:01.28 | WIMPy | Same. |
| 10:01.45 | fish9370 | can you say what for it? |
| 10:01.54 | WIMPy | Both are alternative timing sources in case you don;t have dahdi running. |
| 10:02.54 | fish9370 | can it bring to crash? |
| 10:03.18 | WIMPy | Ony one timing source can be active. |
| 10:03.40 | WIMPy | So they shouldn;t do anything. |
| 10:04.03 | fish9370 | now i think this one active res_timing_timerfd.c |
| 10:04.25 | WIMPy | Try 'timing test'. |
| 10:04.37 | fish9370 | Attempting to test a timer with 100 ticks per second. |
| 10:04.38 | fish9370 | Using the 'timerfd' timing module for this test. |
| 10:04.38 | fish9370 | It has been 1000 milliseconds, and we got 100 timer ticks |
| 10:05.23 | WIMPy | That's unfortunate. You should noload the timer modules. |
| 10:05.40 | WIMPy | Or deselect them in menuconfig if you built yourself. |
| 10:06.02 | fish9370 | i try to unload |
| 10:06.47 | fish9370 | dahdi, you mean driver ? |
| 10:06.54 | WIMPy | yes |
| 10:06.57 | fish9370 | or hardware ? |
| 10:07.18 | WIMPy | The dahdi module is enough. |
| 10:07.54 | WIMPy | But if you don't have any hardware on it, you might not need dahdi. |
| 10:08.01 | fish9370 | i have trouble when it's not was loaded.. |
| 10:09.10 | fish9370 | can i describe my problem? |
| 10:09.22 | WIMPy | sure |
| 10:10.58 | fish9370 | i have agi scripts.. they route calls.. when i have incoming call, and transfer it to anather peer (SIP), sometimes asterisk frize.. |
| 10:12.09 | fish9370 | this agi scripts is hang as procesess.. |
| 10:12.42 | fish9370 | and only kill -9 is help.. |
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| 10:16.08 | WIMPy | There have been reports that non-dahdi timing can cause some issues. But I don't remember the details nor do I know if they still exist. |
| 10:16.30 | WIMPy | I've been running on timerfd for over a year now without any issues. |
| 10:19.29 | fish9370 | have you agi? |
| 10:19.48 | WIMPy | Yes, but only for caller ID name lookup. |
| 10:20.13 | fish9370 | and testing transfer? and have you BLF? with notifications? |
| 10:20.44 | WIMPy | I hardly use transfer, but BLF is in use. |
| 10:20.56 | WIMPy | What kind of notifications? |
| 10:21.06 | fish9370 | for BLF.. |
| 10:21.27 | fish9370 | SIP NOTIFY PACKET |
| 10:21.33 | WIMPy | What use would BLF be without notifications? |
| 10:22.21 | fish9370 | do you think it resolve my problem? onloadin this timer? |
| 10:23.15 | WIMPy | As I didn't have any issues with timerfd, I can't say. |
| 10:23.52 | fish9370 | what version have you? |
| 10:24.24 | WIMPy | I'm upgrading very often here. |
| 10:24.37 | fish9370 | and now? |
| 10:24.51 | fish9370 | i have 1.6.2.21 |
| 10:25.13 | WIMPy | SVN-trunk-r348794M |
| 10:25.24 | WIMPy | Huh? |
| 10:25.47 | WIMPy | Did that already include alternative timing sources? I thought that was a 1.8 feature. |
| 10:26.20 | fish9370 | no, it is old thing.. |
| 10:26.43 | WIMPy | Yes, you version IS old. |
| 10:28.18 | fish9370 | and 1.8 is stable? |
| 10:28.37 | WIMPy | Yes |
| 10:28.51 | fish9370 | and 10? |
| 10:29.24 | WIMPy | And so was the 10-rc1. But I personally wouldn't recommend the official 10.0.0. |
| 10:29.31 | fish9370 | i need just fix this bug.. (( |
| 10:30.11 | WIMPy | Go for 1.8. That's the long term support version. |
| 10:31.08 | fish9370 | if it not help, i try.. |
| 10:31.51 | fish9370 | thanks.. |
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| 11:11.16 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 11:11.27 | elliot98 | gives a friendly wave |
| 11:12.20 | elliot98 | calls end with a 481 call leg/transaction does not exists |
| 11:12.27 | elliot98 | why is this happening? |
| 11:12.57 | elliot98 | as if * is making the calls just disappear without any BYE packet |
| 11:13.27 | IsUp | hello |
| 11:13.31 | WIMPy | I've got that issue when receiving calls from Linksys phones when the session-timer hits. |
| 11:13.43 | IsUp | elliot98: its probably about session timer |
| 11:13.51 | IsUp | elliot98: whats your asterisk version? |
| 11:17.09 | *** join/#asterisk d00gster_ (~dt@176.45.138.161) |
| 11:21.18 | elliot98 | still on 1.4 |
| 11:21.25 | elliot98 | what's session timer? |
| 11:22.22 | WIMPy | can't even remember if they existed in 1.4. |
| 11:22.38 | WIMPy | You need to see the history channel. |
| 11:23.00 | IsUp | 1.4 doesnt support session-timer |
| 11:23.16 | elliot98 | also seems calls are dropping mid-call |
| 11:23.49 | IsUp | elliot98: i had same issue. did you check your CDRs? |
| 11:24.14 | IsUp | elliot98: http://www.vocal.com/sip/session_timers.html -> explained here |
| 11:24.27 | elliot98 | CDRs so far only state ANSWERED for the calls |
| 11:24.40 | elliot98 | so Hangupcause doesn't indicate anything |
| 11:24.45 | WIMPy | Sounds exactely lika a session-timer issue, but if they didn;t exist way back then, I have no idea. |
| 11:24.59 | elliot98 | calls are also dropping |
| 11:25.17 | IsUp | elliot98: basicly, your provider (peer) sends you UPDATE packet after 1800 secs (default value), for refresh the current call |
| 11:25.30 | *** join/#asterisk irroot (~gregory@197.169.75.224) |
| 11:25.53 | IsUp | elliot98: but Asterisk is unable to send anything back. 1.4 doesnt supports session-timer. so after 1800 secs, call is dropping but channel is alive in Asterisk |
| 11:25.53 | *** part/#asterisk irroot (~gregory@197.169.75.224) |
| 11:25.56 | elliot98 | wouldn't the RTP packets be keeping thing up? |
| 11:26.31 | WIMPy | That's the rtptimeout thing. |
| 11:26.38 | IsUp | elliot98: you can use rtptimeout, i didnt it test it yet. but you can use this. |
| 11:27.35 | elliot98 | I'll look into that...need to check if RTP are lagging for some reason |
| 11:27.48 | IsUp | elliot98: just to be sure, enable sip debugging and place a test call (incoming), then pastebin your output |
| 11:28.00 | IsUp | elliot98: lets see if your provider sets session-timer or not |
| 11:28.19 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 11:29.37 | elliot98 | sip debugging didn't show any UPDATE packets |
| 11:29.57 | elliot98 | if a call times out for some reason, should * send a BYE packets? |
| 11:29.59 | elliot98 | *packet' |
| 11:31.42 | IsUp | elliot98: take a look on INVITE first. check if you have session-timer in headers or not |
| 11:32.29 | k-man | WIMPy, its the file from the snom site. in fact, you don't even upload the file, you just provide a link to the firmware on the snom site |
| 11:33.39 | elliot98 | Asterisk was the one that sent the INVITE |
| 11:34.49 | elliot98 | so it wouldn't put it in the headers if it's not supporting it |
| 11:36.32 | WIMPy | k-man: Usually you provide the URL of a file containing Information about the firmware and the location. |
| 11:36.42 | k-man | WIMPy, hmm... |
| 11:37.28 | k-man | WIMPy, this is the instructions I am following |
| 11:37.30 | k-man | http://wiki.snom.com/Firmware/V8/3x0 |
| 11:38.47 | WIMPy | Ok, and what happens if you do that? |
| 11:39.19 | k-man | it goes through the motions, reboots but reboots to 7 again |
| 11:40.02 | WIMPy | So does it download the file? |
| 11:40.16 | k-man | um... not sure |
| 11:40.28 | WIMPy | I guess it doesn't start the flash process? |
| 11:41.07 | WIMPy | Has the phone been provisioned? |
| 11:41.18 | k-man | WIMPy, no, its not provisioned |
| 11:41.32 | k-man | err... as in its not connected to my asterisk server yet |
| 11:41.35 | k-man | if thats what you mean |
| 11:42.09 | WIMPy | No. the question was if it has been configured automatically. |
| 11:43.16 | k-man | WIMPy, ok, now I'm showing my asterisk ignorance. not sure what you mean by that? or is that the thing where the phone downloads its settings automatically from a server of some sort? |
| 11:43.32 | WIMPy | That's the one. |
| 11:44.11 | k-man | WIMPy, no, it hasn't as I don't have that setup here. this is just my home asterisk setup |
| 11:44.26 | WIMPy | Is it a new phone? |
| 11:44.38 | WIMPy | If not, did you reset it? |
| 11:44.38 | k-man | no, bought it second hand |
| 11:44.46 | k-man | no, I didn't - good point |
| 11:45.04 | WIMPy | Or if yu have a tftp server handy, you can always flash it from th bootloader. |
| 11:45.41 | k-man | WIMPy, ok, I might try resetting it first, if that fails i;ll try the tftp |
| 11:45.43 | *** join/#asterisk TimeRider (~steve@5ad5126d.bb.sky.com) |
| 11:45.44 | k-man | thanks WIMPy |
| 11:51.43 | IsUp | elliot98: are you having problems with incoming or outgoing calls? |
| 12:01.23 | elliot98 | looks like it's in both directions |
| 12:02.32 | elliot98 | IsUp: need to check the RTP packets, maybe something is going quircky with them |
| 12:02.54 | elliot98 | also, is there anyway for wireshark to recognize Digium cards for packet analysis? |
| 12:03.22 | IsUp | elliot98: wireshark recognizes SIP / RTP well. i dont know about DAHDI. |
| 12:03.40 | IsUp | elliot98: debug an "incoming" call. not outgoing. check session timer headers. |
| 12:03.52 | WIMPy | elliot98: There is something on the issue tracker, but your dahdi hardware is not related to SIP. |
| 12:04.34 | elliot98 | yeah, was just asking in passing if Wireshark can somehow grab data from Digium cards |
| 12:05.08 | WIMPy | Reminds me of my to-do-list. |
| 12:05.32 | WIMPy | Try to find out if you can get a copy of the signalling from dahdi directly. |
| 12:11.16 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 12:11.49 | hrolf | Hi, when I execute "core stop gracefully" from the Manager API, do I get some sort of event? |
| 12:18.30 | IsUp | elliot98: did you check INVITE packets from your provider? |
| 12:22.09 | zamba | anyone using a linksys sipura spa2102 here? |
| 12:22.20 | zamba | i need to provision a cfg that will unlock the device |
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| 13:12.32 | iulhk | trying to install zaptel 1.4.12 & asterisk 1.6 when installing asterisk-1.6 and giving command make menuselect every time getting " XXX app_meetme" what dependency is missing can anybody assist? using debian lenny ? |
| 13:13.48 | WIMPy | That's a thing for the history channel. |
| 13:14.09 | WIMPy | Try to use dahdi and a still supported Asterisk 1.8. |
| 13:17.06 | iulhk | <WIMPy>: i never tried dahdi before, but do we have option for ztdummy in dahdi ? |
| 13:17.27 | WIMPy | That is included in the main module. |
| 13:17.41 | WIMPy | But you don't need dahdi for timing any more. |
| 13:18.00 | WIMPy | Except for MeetMe which can often be replaced bu ConfBridge. |
| 13:18.21 | iulhk | yes, i want to use meetme so that's y asking |
| 13:18.31 | iulhk | so means i have to install dahdi too ? |
| 13:18.53 | WIMPy | yes |
| 13:19.30 | iulhk | thanx buddy :) |
| 13:27.44 | iulhk | one more other thing, using realtime asterisk1.6 , if peer 100 is connected with peer 200 third peer 300 should not make call to peer 100 or 200, peer 300 should get busy instead of ring ? |
| 13:29.20 | WIMPy | In 1.6 you had 'call-limit's. |
| 13:29.24 | elliot98 | IsUp: no UPDATES show up is sip debugging |
| 13:30.48 | iulhk | unable to see call-limit , there is busylevel what is for ? |
| 13:31.23 | IsUp | elliot98: debug INVITE, not UPDATE. place an incoming call to your asterisk. and debug INVITE. |
| 13:31.52 | IsUp | elliot98: asterisk returns 'method not implemented' to an UPDATE message. it's normal. |
| 13:33.25 | *** join/#asterisk iulhk (~iulhk@116.71.186.37) |
| 13:33.38 | iulhk | unable to see call-limit |
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| 13:35.36 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:35.47 | WIMPy | You might have to make extra configuration to get it to work with realtime, but I'm not in to that. |
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| 16:55.45 | Cupidon | hey |
| 16:56.23 | Cupidon | can someone help me ? |
| 16:56.41 | Cupidon | i just install asterisk with yum on centos 5.7 |
| 16:57.30 | Cupidon | I can not connect |
| 16:57.37 | Cupidon | eny softphone |
| 16:58.05 | Cupidon | my firewall is disabled |
| 16:58.40 | socomm | eny softphone? |
| 16:58.47 | Cupidon | yes |
| 16:59.18 | socomm | Can you link me to this eny softphone, i've never heard of it. |
| 17:00.02 | Cupidon | I tried three up now ex : bria ... |
| 17:00.46 | Cupidon | xlite ... |
| 17:01.24 | socomm | Oh, you meant ANY softphone |
| 17:02.16 | Cupidon | yes |
| 17:02.16 | socomm | my guess is you've something misconfigured - possibly asterisk itself or the values you've put into your softphone |
| 17:04.15 | Cupidon | you have something indices where i shoud could look ? |
| 17:05.51 | [TK]D-Fender | Cupidon: "sip set debug on" <- go look if the phone is even reaching * |
| 17:08.19 | Cupidon | thank you |
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| 17:34.37 | p3nguin | Perhaps you did not configure Asterisk to do anything. |
| 17:37.24 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 17:38.58 | nny | having a one way audio issue. The pbx has 2 interface, wan and local. Any phones connecting to the wan interface and using the outbound sip trunk are fine, any local phones connecting through the pbx to the sip trunk get the one way issue. Any thoughts? |
| 17:39.04 | nny | here is my iptables http://pastebin.com/vpYpngTY |
| 17:39.19 | nny | i have canreinvite=no in sip.conf too |
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| 17:42.13 | nny | nm |
| 17:42.20 | nny | figured it out i think.. sigh |
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| 18:32.20 | Cupidon | nny I rechecked all still the same result i have |
| 18:32.52 | Cupidon | i cant conect et all |
| 18:33.56 | Cupidon | I'll try reinstalling |
| 18:34.29 | p3nguin | Did you install it already? |
| 18:35.12 | Cupidon | yes |
| 18:35.14 | p3nguin | Then you do not need to install it again. |
| 18:35.24 | Cupidon | <Cupidon> can someone help me ? |
| 18:35.26 | Cupidon | <Cupidon> i just install asterisk with yum on centos 5.7 |
| 18:35.26 | Cupidon | <Cupidon> I can not connect |
| 18:35.26 | Cupidon | <Cupidon> eny softphone |
| 18:35.26 | Cupidon | <Cupidon> my firewall is disabled |
| 18:35.38 | p3nguin | But you do have to configure it. Asterisk isn't magic. Go configure it to do something. |
| 18:35.56 | Cupidon | its conigured |
| 18:36.00 | p3nguin | Show me. |
| 18:36.02 | p3nguin | ~pb |
| 18:36.02 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 18:36.10 | p3nguin | Pastebin your configs. |
| 18:36.26 | Cupidon | 2 sec |
| 18:39.00 | [TK]D-Fender | Forget configs. Show us the SIP debug of your softphone even reaching your server |
| 18:39.08 | p3nguin | I'm mostly interested in sip.conf and extensions.conf. |
| 18:39.25 | [TK]D-Fender | Doesn't matter what you configured on your server if it hasn't reached it yet |
| 18:39.31 | [TK]D-Fender | prove that, then move forward |
| 18:39.53 | Cupidon | http://pastebin.com/NCP6dYjW |
| 18:41.03 | [TK]D-Fender | Cupidon: FreePBX isn't supported in this channel. |
| 18:41.04 | p3nguin | That's a pretty useless sip.conf. |
| 18:41.06 | [TK]D-Fender | ~freepbx |
| 18:41.06 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:41.25 | [TK]D-Fender | And as I said go prove that the softphone is even reaching your server first |
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| 18:43.30 | Cupidon | ok [TK]D-Fender thanks eny way |
| 18:44.19 | [TK]D-Fender | Cupidon: Where is the SIP debug? I gave you the command to go look for this an hour and a half ago... |
| 18:52.07 | Cupidon | [TK]D-Fender hey soory i am new in asterisk .. |
| 18:52.08 | Cupidon | . this is my 1 install with yum |
| 18:52.23 | Cupidon | i was using asterisk now |
| 18:52.38 | Cupidon | thet was working |
| 18:52.39 | [TK]D-Fender | Cupidon: Where is this SIP debug? |
| 18:52.59 | Cupidon | wer i shoud look ? |
| 18:53.03 | [TK]D-Fender | Cupidon: You keep giving us a history of what youv'e done and not looking at what is happening |
| 18:53.05 | [TK]D-Fender | SIP DEBUG |
| 18:53.15 | [TK]D-Fender | Told 5 times now and gave you the command even |
| 18:53.24 | Cupidon | yes |
| 18:53.40 | Cupidon | i dont have in x-lite |
| 18:53.58 | [TK]D-Fender | Who said anything about X-Lite? |
| 18:54.15 | [TK]D-Fender | You said you were using soft-phones and they weren't working. It doesn't matter which one you're using |
| 18:54.21 | [TK]D-Fender | SIP debug is an * CLI function |
| 18:54.25 | [TK]D-Fender | Look at what * is seeing |
| 18:54.30 | Cupidon | ok |
| 18:54.42 | Cupidon | soory now i understed |
| 18:56.22 | Cupidon | localhost*CLI> sip set debug on |
| 18:56.23 | Cupidon | SIP Debugging enabled |
| 18:56.23 | Cupidon | localhost*CLI> |
| 18:59.21 | Cupidon | http://pastebin.com/DCPYVxm8 |
| 18:59.26 | bbourdage | I am having issues dialing out VIA a SIP trunk to Bandwidth.com, I have the phones and the SIP trunk registerd, I am geeting a 401,403 errors. I apologize in advance, I know this is an easy one, I a day or so into learning asterisk. I have used pastebin for the details http://pastebin.com/u/bbourdage |
| 19:01.14 | [TK]D-Fender | Cupidon: bad auth, like it says |
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| 19:02.26 | [TK]D-Fender | bbourdage: Try again, with verbose 10, and do not mask anything execpt passwords |
| 19:02.47 | bbourdage | ok, 2 min |
| 19:04.46 | bbourdage | I have repasted into the same pastebin for debug @ level 10 |
| 19:05.43 | [TK]D-Fender | SIP/2.0 403 Your IP is not authorized to use this service. |
| 19:06.02 | [TK]D-Fender | And I do not see standard verbose in that debug |
| 19:06.19 | Cupidon | [TK]D-Fender what i need to change ? |
| 19:06.23 | Cupidon | to work |
| 19:06.36 | bbourdage | I had called Bandwidth and they confirmed that that IP is valid, I will call and escalate |
| 19:06.48 | Cupidon | i chek te password user is ok |
| 19:07.58 | [TK]D-Fender | Cupidon: Show us your SIP.CONF entry for that device masking only the password, and a new attempt... a COMPLETE one. You only included the error response in that last PB |
| 19:08.44 | [TK]D-Fender | bbourdage: "sip show registry". See if you have registered successfully with them. They may refuse out-calls untill you have registered |
| 19:10.29 | bbourdage | I am not registered with them, they authenticate via IP, I am escalating on the phone with them now. |
| 19:10.53 | Cupidon | [TK]D-Fender i have : /etc/asterisk/sip.conf |
| 19:11.20 | Cupidon | wic which ones |
| 19:11.28 | Cupidon | i have 3 |
| 19:11.34 | Cupidon | sip.conf |
| 19:11.43 | [TK]D-Fender | Cupidon: [100] <------ |
| 19:11.48 | [TK]D-Fender | go find it |
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| 19:26.33 | bbourdage | Fender, I was able to register, sip show registry shows correct, same error, waiting for respond from engineer @ bandwidth. Thanks for your help |
| 19:26.53 | bbourdage | sip reload |
| 19:50.40 | Cupidon | [TK]D-Fender Thank you very much for the help and soory to disterb you |
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| 20:30.57 | bbourdage | Fender, Bandwdith was able to add IP to approved servers, and outgoing calls are working now. Thank you for your help |
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| 21:12.20 | bbourdage | I am looking for help on using a database for SIP peers and friends. I seem to be getting some conflicts on field names and empty values, for example, if the host is set to dynamic, and you have a statis peer in the database, when you issue a sip reload, you get a failure warning on the function getaddrinfo ?. Has anyone run into this before ? |
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| 21:17.54 | p3nguin | I think you're going to use either realtime (from the database, and never ever run "sip reload") or you're going to use static conf. I've never seen anyone happily use both together on the same system. |
| 21:19.57 | bbourdage | I plan to use the realtime, but I am seeing errors, when the system loads the database, it it expecting a value for ipaddr, when the host is filled out for the peer ? |
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| 22:06.57 | jarrod | :q! |
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| 22:50.46 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:50.46 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0 (2011/12/15), 1.8.8.0 (2011/12/15), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 23:03.25 | mroe | I am having a problem with a pri span, incoming calls work but outgoing dies. Here is pri debug: http://pastebin.com/a9hyHaW7 |
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