| 00:04.09 | navaismo | any hints to avoid that |
| 00:04.20 | igcewieling1 | navaismo: what version of Asterisk? |
| 00:04.40 | igcewieling1 | navaismo: I had a similar issue a few weeks ago. |
| 00:08.04 | ghost75 | hundreds of sip channels oO |
| 00:10.22 | navaismo | 1.8.11 |
| 00:11.14 | igcewieling1 | My solution was to just restart the box |
| 00:11.21 | igcewieling1 | well restart asterisk |
| 00:11.44 | navaismo | igcewieling1, asterisk version 1.8.11. 414 sip channels/dialogs right now many with (nothing) last message ACK |
| 00:12.01 | navaismo | yep that was the solution here, but eventually it happens again |
| 00:12.13 | igcewieling1 | navaismo: sip channels are created for qualify messages, but they should not be hanging around for terribly long |
| 00:12.29 | igcewieling1 | navaismo: try updating |
| 00:13.10 | navaismo | that is my fear... :'( |
| 00:28.27 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 01:02.34 | BludSuckingFiend | heh, conf2ael doesn't recognize "same =>" |
| 01:10.43 | *** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806) |
| 01:14.43 | phix | igcewieling1: So you tried turning it off then on again/ |
| 01:29.56 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 01:52.30 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:52.31 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 02:29.35 | ShoreTel | hi |
| 02:30.21 | WIMPy | lo |
| 02:45.32 | BludSuckingFiend | so wouldn't you think that if encryption=no is set on a device, it wouldn't use encryption if it is offered by the phone? |
| 02:46.08 | BludSuckingFiend | outgoing... nevermind |
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| 03:12.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 03:26.01 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 04:24.22 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 04:26.32 | lorsungcu | anyone know of a way to support this in asterisk? http://wiki.snom.com/Features/XML_Parsing/ParsingRules/sip_eval |
| 04:28.41 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:44.28 | pabelanger | lorsungcu: what? |
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| 07:18.13 | *** join/#asterisk fulcan (brads@2600:3c00::f03c:91ff:fe70:f0a9) |
| 07:19.56 | fulcan | I am trying to migrate a vicidial/asterisk box and after migration and changing this IP, I get this error -> WARNING[17702]: app_meetme.c:3145 admin_exec: Conference number '8600051' not found! |
| 07:26.00 | ectospasm | fulcan: note that's a WARNING and not an ERROR... it means what it says, MeetMe conference number 8600051 doesn't exist |
| 07:26.17 | ectospasm | you're probably passing that to Meetme: Meetme(8600051) |
| 07:26.35 | ectospasm | It has nothing to do with your migration, unless vicidial is doing something funky |
| 07:26.49 | ectospasm | and we don't support vicidial here. |
| 07:28.13 | fulcan | everything is vanillia, I just don't work with the mysql side of the asterisk db so I am a little stumped where the meetme mapping is. |
| 07:29.10 | ectospasm | fulcan: it's in your dialplan |
| 07:29.14 | ectospasm | extensions.conf |
| 07:31.17 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 07:33.37 | fulcan | I see very little of the functionality of this phone system in extensions.conf I see all of the peer configs etc in mysql. |
| 07:35.34 | ChannelZ | extensions can be in a database too |
| 07:36.30 | fulcan | somewhere there is an ip change needed and I can find where. |
| 07:38.28 | fulcan | can't |
| 07:40.26 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 07:42.43 | ChannelZ | A missing MeetMe conference doesn't have much to do with any IP anywhere |
| 07:47.28 | fulcan | this image was coppied yesterday. the original is still running live now. the only difference is ip. |
| 07:48.07 | fulcan | me = monkey w/ lightbulb. |
| 07:58.23 | ChannelZ | shrugs |
| 07:58.33 | ChannelZ | With no more information, we don't stand much of a chance. |
| 07:59.12 | fulcan | there is an endless amout of info to provide, what is best for you first? |
| 07:59.38 | fulcan | not much in extensions. |
| 08:00.05 | ChannelZ | well it has to be somewhere. Presumably someone is dialing something to cause this warning |
| 08:00.45 | ChannelZ | if extensions.conf is set for realtime, they'd be in the database. |
| 08:00.48 | fulcan | http://pastie.org/6364526 |
| 08:01.29 | fulcan | grep'ed the old ip address |
| 08:03.20 | ChannelZ | hmm. This gtalk protocol style auto-detect seems to still be a fail. |
| 08:03.32 | fulcan | http://tinypic.com/r/29c47x4/6 |
| 08:05.08 | fulcan | I would of expected to see the other ip there... |
| 08:05.33 | ChannelZ | What of meetme.conf? Is there a conference called 8600051? Or is it configured for dynamic conferences? |
| 08:09.14 | fulcan | http://pastie.org/6364548 |
| 08:09.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:09.42 | fulcan | cat /etc/asterisk/meetme-vicidial.conf is where that came from |
| 08:10.04 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 08:10.07 | ChannelZ | I missed anything you said after my last |
| 08:10.54 | fulcan | this is the regular meetme.conf http://pastie.org/6364552 |
| 08:11.56 | ChannelZ | and what's in meetme-vicidial.conf |
| 08:11.58 | fulcan | line 4 of the first meetme conf => 8600051 |
| 08:12.14 | fulcan | http://pastie.org/6364548#4,16 |
| 08:14.54 | ChannelZ | does 'meetme list' on the console show all those? |
| 08:17.14 | fulcan | grrr, what does it want http://pastie.org/6364569 vicidial list names? |
| 08:19.05 | ChannelZ | I might be wrong. I don't have any machines still running meetme so I can't even look at the commands |
| 08:22.20 | ChannelZ | guessing it's not parsing your configs correctly, either meetme.conf isn't getting loaded or the included one with all those conferences isn't. Maybe permissions related. Maybe path related. |
| 08:23.43 | fulcan | I think you are correct |
| 08:24.33 | *** join/#asterisk puzzled (~patrick@5ED2AFC0.cm-7-3c.dynamic.ziggo.nl) |
| 08:25.17 | fulcan | my only question would be, does the error mean it can't find 8600051 or meetme-vicidial.conf ? which one exactly is it missing? |
| 08:25.18 | ChannelZ | the module is loaded but it says that conference doesn't exist so there's very few other things it could be |
| 08:26.10 | ChannelZ | Asterisk loads meetme.conf when the meetme module is loaded. Yours #includes another config file so it gets read at that point in meetme.conf as if the contents were there in place of that line. |
| 08:26.54 | fulcan | fyi reload output -> http://pastie.org/6364595 |
| 08:27.02 | ChannelZ | So either it's failing to do that because it can't read/find the meetme-vicidial.conf file for whatever reason, or it is but there's something goofy in it it chokes on and doesn't wind up parsing the thing. |
| 08:27.25 | ChannelZ | do a 'module reload app_meetme' |
| 08:27.53 | fulcan | so you are saying what I am thinking is that it isn't hitting the file? |
| 08:27.55 | fulcan | ok |
| 08:29.14 | fulcan | I was wrong, and thats a neat trick -> http://pastie.org/6364603 |
| 08:29.43 | ChannelZ | ok so it's at leats reading it |
| 08:29.49 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 08:30.23 | fulcan | yep, so it is choking after that.... grrrr I'm not even sure what is next... one sec. |
| 08:31.23 | ChannelZ | do "help meetme" - I just don't remember what all the commands were but I'm pretty sure one of them will list all the conferences it knows about |
| 08:32.46 | fulcan | http://pastie.org/6364610 it wants a conference number? |
| 08:33.57 | ChannelZ | what version of asterisk is this? |
| 08:38.15 | fulcan | 1.4 |
| 08:38.17 | fulcan | :/ |
| 08:40.23 | fulcan | old enough to have grand children |
| 08:41.29 | ChannelZ | well I'm not sure what to say. |
| 08:42.04 | WIMPy | 1.6 = children, 1.8 = grand children? |
| 08:42.58 | fulcan | 1.4=archives? |
| 08:43.39 | fulcan | grand papies long forgotten? |
| 08:44.11 | WIMPy | Luckily. |
| 08:44.31 | fulcan | :) |
| 08:44.37 | WIMPy | I remeber having quite some "interesting features" in 1.4. |
| 08:45.56 | fulcan | believe it or not, vicidial ships it's latest iso with 1.4. It's nasty bro..... |
| 08:46.14 | fulcan | nasty :/ |
| 08:46.16 | WIMPy | It must be dead. |
| 08:46.25 | WIMPy | Or probably already rotten. |
| 08:47.12 | fulcan | nope very much alive and craved by telecomm folk around here. far from dead, just horribly programmed. |
| 08:49.04 | WIMPy | I'd call anything <1.8 constricted usable. |
| 08:49.33 | fulcan | for folks whom fear the command line, vicidial is empowerment *smirk* |
| 08:49.47 | *** join/#asterisk kchehab (~david@62.84.68.161) |
| 08:50.45 | ChannelZ | Do the conferences in meetme.conf work? 8600 or 8601? |
| 08:51.19 | kchehab | hi , i am using asterisk 1.8.20 ,how can i force RTP to use the public ip address as i can see that its sending the RTp to UA private IP address as i already set correctly the localnet= and the externip= |
| 08:51.42 | ChannelZ | So your Asterisk is behind NAT? |
| 08:52.28 | ChannelZ | and the peer neets nat=yes |
| 08:53.11 | ChannelZ | and * will only send RTP to the remote IP if it gets some RTP from them first so chances are something is wrong in the port forwarding or something |
| 08:53.26 | fulcan | I am on an open/routable ip |
| 08:53.48 | ChannelZ | sorry not you, that was for kchehab |
| 08:54.52 | kchehab | ChanServ yes it is |
| 08:54.55 | fulcan | the recording function is somehow routed to 192.168.1.1 ,,, after a breif teleconference with the original tech, he said just create a subif 192.168.1.1 (which is done, errors were worse prior). |
| 08:56.12 | kchehab | ChanServ yes all my peers have qualify=yes and nat=yes |
| 08:56.46 | ChannelZ | then perhaps media isn't even making it from the device out its firewall |
| 08:57.17 | ChannelZ | is Asterisk sending out its correct IP for the remote side to send their media to? |
| 08:57.36 | kchehab | ChanServ i will do a sip debug and chechk |
| 08:58.41 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 08:59.43 | *** join/#asterisk puzzled (~patrick@5ED2AFC0.cm-7-3c.dynamic.ziggo.nl) |
| 09:02.27 | ChannelZ | damn this xmpp/motif is so unreliable |
| 09:02.56 | kchehab | ChannelZ i print my config and debug http://pastebin.com/RVWbnCvZ ,please can you check it |
| 09:04.01 | kchehab | ChanServ and the RTp debug print our when i dial voicemail *98 Sent RTP packet to 192.168.88.248:4002 (type 18, seq 013557, ts 011840, len 000030) |
| 09:04.11 | kchehab | ChanServ my private Ip address |
| 09:06.15 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-145-215.pools.arcor-ip.net) |
| 09:06.32 | ChannelZ | so that device is probably not sending any RTP TO asterisk, either because it isn't, or it can't. |
| 09:07.58 | kchehab | ChanServ but i am dialing the vocie mail the rtp should be oneway from asterisk to UA as it connecting him on its private address not public:port |
| 09:09.49 | ChannelZ | Even with nat=yes for the peer, Asterisk will only send RTP to the IP it got RTP from. If it doesn't get anything, it will send to the IP in the INVITE. |
| 09:10.38 | kchehab | ChanServ is there any thing i can code or fix in the config to let * send traffic to its public Ip address as it registered with its public IP |
| 09:10.53 | kchehab | ChanServ you mean its a UA miss config |
| 09:12.07 | ChannelZ | Yes and no |
| 09:12.21 | kchehab | ChanServ hehehe how is that :) |
| 09:12.44 | kchehab | ChanServ is there a solution for this |
| 09:13.20 | [TK]D-Fender | kchehab: copy it all to another pastebin site, I can't seem to resolve that link. |
| 09:13.27 | ChannelZ | The UA is sending a private IP address because it is unaware of it's public IP address through NAT. Asterisk sort-of knows that the UA is lying because you told it so with nat=yes, but it doesn't arbitrarily send RTP to the IP it thinks it should (even though personally I think there should be a config option to make this happen, but that's another story) |
| 09:14.25 | ChannelZ | One * sees a little RTP come in, that wakes it up and it says "OK, I'll send my own RTP back to THAT IP." |
| 09:14.29 | kchehab | ChanServ i read on asterisk new features media_address= configuration option , can that help ? |
| 09:15.19 | ChannelZ | not really |
| 09:15.45 | kchehab | ChanServ what do you advise ,and do you think that there is a solution for such problem ? |
| 09:16.14 | *** join/#asterisk puzzled (~patrick@5ED2AFC0.cm-7-3c.dynamic.ziggo.nl) |
| 09:16.49 | ChannelZ | well you've got to figure out why your UA isn't sending anything, or if it is, why it might not be making it out of its network and to your * box. |
| 09:17.06 | ChannelZ | Do you have some UDP ports forwarded through the firewall your * server is behind, per rtp.conf? |
| 09:18.13 | kchehab | ChanServ yes i already forward that |
| 09:18.39 | kchehab | ChanServ i mean i already forward udp port from 1000-2000 as i am using SRTP |
| 09:18.53 | kchehab | ChanServ can SRTp be such problem ? |
| 09:20.12 | kchehab | ChanServ no its not i test it with normal RTP same problem |
| 09:22.09 | kchehab | [TK]D-Fender i will copy the debug |
| 09:22.27 | ChannelZ | those port numbers don't seem to be what your * is requesting.. |
| 09:22.44 | ChannelZ | m=audio 13644 RTP/SAVP 18 8 0 101 |
| 09:22.50 | ChannelZ | that's 13644 |
| 09:24.31 | [TK]D-Fender | kchehab: [9092] nat=no |
| 09:24.53 | kchehab | [TK]D-Fender please check it http://pastebin.com/QCC8g3ah |
| 09:25.09 | [TK]D-Fender | kchehab: <--- SIP read from TLS:62.84.68.161:35437 ---> From: "9092" <sip:9092@91.151.233.212>;tag=QSoXkL3BEl6znPTnLsGARKPzhbcRGJw5 c=IN IP4 192.168.88.248 |
| 09:25.22 | ChannelZ | oh, duh |
| 09:25.25 | [TK]D-Fender | kchehab: Their reported RTP IP is WRONG and you left them as nat=NO |
| 09:25.31 | [TK]D-Fender | kchehab: Change it tio YES |
| 09:25.42 | ChannelZ | I'm seeing two different conversations in here, missed that |
| 09:25.45 | kchehab | [TK]D-Fender i am rwally sorry the first debugout put was with nat=no ,as this debug nat=yes |
| 09:26.26 | kchehab | [TK]D-Fender i was doing some test as now nat=yes http://pastebin.com/QCC8g3ah |
| 09:27.01 | ChannelZ | so when you said you were forwarding 1000-2000 was that a type-o? did you mean 10000-20000? |
| 09:27.25 | kchehab | ChanServ yes 10000 to 20000 yes |
| 09:27.44 | kchehab | ChanServ when i open a VPN the call pass normally |
| 09:29.33 | [TK]D-Fender | kchehab: On your 2nd call I see it apss IP's normally... it accepts, answers, plays back -- <SIP/9092-0000001d> Playing 'vm-login.gsm' (language 'en') and then Bria hangs up |
| 09:30.22 | [TK]D-Fender | kchehab: Show us your forwarding |
| 09:30.38 | ghost75 | libopenais3 libcorosync4 libgmime-2.0-2a libresample1 sox libsox-fmt-alsa libsox-fmt-base libsox1b libsybdb5 libvpb0 unixodbc vpb-driver-source odbcinst1debian2 odbcinst |
| 09:30.43 | ghost75 | any of those required by asterisk? |
| 09:31.09 | kchehab | [TK]D-Fender u mean my mikrotik config ? |
| 09:32.38 | ectospasm | ghost75: at least, unixodbc if it's to connect to an ODBC database |
| 09:33.56 | [TK]D-Fender | kchehab: Your router. |
| 09:34.03 | ghost75 | apt lists them when doing autoremove |
| 09:34.10 | kchehab | [TK]D-Fender mikrotik is my router config dst address 91.151.233.212 protocol udp dst-port 6000-65000 action dst-nat 192.168.9.3 (asterisk private ip ) and also i have src nat src address * 192.168.9.3 to public IP |
| 09:34.55 | ghost75 | http://packages.debian.org/de/squeeze/libopenais3 <- cluster framework? |
| 09:36.30 | ectospasm | ghost75: same as libcorosync |
| 09:36.43 | ghost75 | whats that for |
| 09:36.49 | [TK]D-Fender | kchehab: pastebin rtp.conf |
| 09:37.59 | ectospasm | ghost75: http://lmgtfy.com/?q=libcorosync |
| 09:38.39 | ghost75 | yes i wrote already cluster framework, but what this has to do with asterisk |
| 09:39.21 | ectospasm | ghost75: asterisk can be a service provided by a cluster |
| 09:39.34 | ectospasm | ...but Asterisk does not depend on corosync |
| 09:39.40 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.102.114) |
| 09:39.45 | ghost75 | k |
| 09:40.19 | kchehab | [TK]D-Fender [general] rtpstart=10000 rtpend=20000 |
| 09:40.56 | kchehab | [TK]D-Fender as i can see the problem the * is sending rtp to UA private Ip address not its public:higer port |
| 09:41.02 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.102.114) |
| 09:42.31 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 09:43.11 | [TK]D-Fender | kchehab: Looks fine so far... |
| 09:43.28 | [TK]D-Fender | kchehab: * seems to think it's NAT'd so it should send to the same IP |
| 09:44.51 | kchehab | [TK]D-Fender how can fix it ? |
| 09:45.54 | kchehab | [TK]D-Fender is the a config to let it check it media as i can see in line 351. o=root 1724596770 1724596771 IN IP4 91.151.233.212 |
| 09:46.51 | kchehab | [TK]D-Fender * be informed that the clinet been nated i mean |
| 09:47.18 | *** join/#asterisk evilman_home (kvirc@2.92.139.155) |
| 09:48.51 | kchehab | [TK]D-Fender i think this parameter should changeg right c=IN IP4 192.168.88.248 (should be the public UA public IP |
| 09:53.53 | fulcan | Hey list, THANK YOU! My brain is fried. |
| 09:55.11 | ChannelZ | eh? |
| 09:55.25 | kchehab | [TK]D-Fender any ideas or that it :) |
| 09:57.00 | ChannelZ | you need to figure out why the phone isn't sending RTP to Asterisk or why it's otherwise not making it there. |
| 09:59.20 | kchehab | ChannelZ I check if bria (my soft phone) and nothing can be done for that as test blink with the same result |
| 09:59.55 | ChannelZ | It's possibly/probably the firewall on either side |
| 10:00.08 | [TK]D-Fender | c=IN IP4 91.151.233.212 |
| 10:00.19 | [TK]D-Fender | servr offers teh WAN IP for RTP |
| 10:01.04 | kchehab | ChannelZ do you think that firewall can cause this rtp confuse as it already had a sip fix up as i try to stop /start this feature with same result |
| 10:02.27 | [TK]D-Fender | kchehab: your router had better be doing NOTHING extra with this |
| 10:02.30 | [TK]D-Fender | NO ALG |
| 10:02.35 | [TK]D-Fender | this is to be disabled |
| 10:02.45 | ChannelZ | you need to somehow prove 1. the phone can successfully get packets OUT of its network TO your Asterisk server on ports 10000-20000, and 2. that those packets make it IN to your asterisk server though whatever firewall its running. |
| 10:03.22 | kchehab | [TK]D-Fender what do you mean by ALG |
| 10:06.29 | ChannelZ | ALG as in your router trying to sniff/munge SIP packets its self |
| 10:07.11 | kchehab | ChannelZ [TK]D-Fender i will check what i can do to collect more debug thaks in advance |
| 10:07.37 | [TK]D-Fender | <PROTECTED> |
| 10:08.11 | [TK]D-Fender | NO "fix" allowed here. Your router should simply pass the packet, not proxy/rewwrite them |
| 10:08.12 | kchehab | [TK]D-Fender you mean mikrotik yes it has |
| 10:08.17 | [TK]D-Fender | TURN IT OFF |
| 10:08.22 | kchehab | [TK]D-Fender i am now |
| 10:09.28 | eirirs | I just don't understand why someone inventened SIP ALG in router |
| 10:10.18 | ChannelZ | I've never seen one work |
| 10:10.34 | ChannelZ | never got the linux one to anyway |
| 10:10.48 | *** join/#asterisk mwally (~mwally@mwally.opencnam.com) |
| 10:11.16 | fulcan | my GOD computers SUCK! |
| 10:11.17 | kchehab | ChanServ same after disabling sip fix up :( |
| 10:11.57 | ChannelZ | tell me about it. Can't really get gtalk to work reliably. I have about a 25% chance of audio working |
| 10:12.14 | ChannelZ | NAT is a fucking pain the ass. |
| 10:16.37 | [TK]D-Fender | http://forum.mikrotik.com/viewtopic.php?f=7&t=35453 |
| 10:16.50 | kchehab | ChanServ hehehehe 100 % true |
| 10:18.25 | kchehab | [TK]D-Fender reading it |
| 10:18.35 | ChannelZ | I'm going to bed. You can keep talking to ChanServ though, he's tireless. |
| 10:20.12 | kchehab | ChanServ no one in this field is not tire,we have no life brooo |
| 10:20.27 | jzaw | ChannelZ, my isp's stun server .... |
| 10:20.31 | jzaw | stun.aa.net.uk is an alias for natisevil.aasip.co.uk. |
| 10:27.21 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-145-215.pools.arcor-ip.net) |
| 10:27.55 | eirirs | NAT are easy |
| 10:32.40 | WIMPy | Easy to reveil bad protocols. |
| 10:43.24 | *** join/#asterisk speedytel (~speedytel@113.10.165.35) |
| 10:44.39 | speedytel | hi guys |
| 10:46.52 | speedytel | everyone zombie? |
| 10:47.59 | kchehab | WIMPy hehe ,do you have an advice for me |
| 10:48.40 | WIMPy | Don't use SIP? |
| 10:49.08 | kchehab | WIMPy that an advice to do not use asterisk |
| 10:49.13 | kchehab | sip |
| 10:49.47 | WIMPy | Asterisk can do a lot more than SIP. |
| 10:50.07 | kchehab | WIMPy but my UA use sip |
| 10:50.38 | WIMPy | A very common issue. |
| 10:53.30 | ghost75 | sippy |
| 11:04.21 | WIMPy | Hmm. I just had a look at the current version of chan_dahdi.conf and I found the well known parameter 'immediate', but it seems to have a completely different meaning than in previous versions. |
| 11:05.14 | WIMPy | Is that correct? Is it a good idea to use a parameter name that was previouselt used for somethign else? And how do you configure what it used to do now? |
| 11:23.32 | ectospasm | WIMPy: "current version..." which version do you mean? Trunk? The meaning of immediate hasn't changed in 11, so... |
| 11:25.02 | WIMPy | No. it's all ok. I can't read. |
| 11:25.21 | WIMPy | But there's still a lot of stuff I don't understand in there. |
| 11:34.12 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 11:44.07 | eirirs | https://supportforums.cisco.com/thread/303692 |
| 11:44.23 | eirirs | I got exactly the same quesitions, anyone know how to get more verbose info ? |
| 11:45.43 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 11:48.07 | *** join/#asterisk marksaitis (~marksaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 11:48.16 | ghost75 | no files for proxy |
| 11:49.08 | eirirs | ? |
| 11:50.32 | ghost75 | proxy_backup is setting in SIPDefault.cnf |
| 11:51.23 | ghost75 | ah you mean parse error below |
| 11:51.28 | eirirs | yes |
| 11:51.31 | eirirs | 2 errors |
| 11:51.38 | eirirs | would love to know what those 2 errors was |
| 11:56.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 12:00.00 | ghost75 | i have newer model |
| 12:11.49 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 12:20.58 | *** join/#asterisk vahidabdi (~sena@91.98.33.208) |
| 12:22.12 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 12:22.48 | *** part/#asterisk vahidabdi (~sena@91.98.33.208) |
| 12:23.36 | *** join/#asterisk vahidabdi (~sena@91.98.33.208) |
| 12:50.34 | eirirs | ok... narrowed it to be either tos_media: "5" or enable_vad: "0" |
| 12:50.40 | eirirs | rebooting phones to check :P |
| 12:52.40 | eirirs | ok, hello bastard you're busted, tos_media: "5" , uncommented it and the parse error is gone |
| 12:53.06 | eirirs | erm...commented it out |
| 13:05.46 | ghost75 | great no bluetooth in kernel |
| 13:15.54 | eirirs | hmm, no webaccess for 7940G |
| 13:16.49 | ghost75 | old model dont has i think |
| 13:17.36 | eirirs | but I see it can be enabled by CUCM, so gotta be a parameter I can set in SIPmac.cnf but gotta find what the name of parameter are.. |
| 13:18.44 | ghost75 | 7941G has web |
| 13:22.10 | ghost75 | if i dont change kernel is make modules and make modules_install enough for adding new modules? |
| 13:22.46 | jacekowski | sort of |
| 13:23.03 | jacekowski | it works for some modules |
| 13:23.13 | jacekowski | but some modules will need updated kernel |
| 13:24.13 | *** join/#asterisk Demon_ru (~demon@ip253.net222.n37.ru) |
| 13:31.33 | ghost75 | poor atom building kernel |
| 13:42.50 | *** part/#asterisk vahidabdi (~sena@91.98.33.208) |
| 13:44.14 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 13:53.28 | *** join/#asterisk ThoMe (~tm@tm.muc.de) |
| 13:53.29 | ThoMe | hiho |
| 13:53.49 | ThoMe | i would like to logging incomming calls in my cdr database (csv or mysq). |
| 13:54.11 | ThoMe | it works when i have a dialplan like _X. => { Playback(tt-monkeys) } |
| 13:54.28 | ThoMe | but i dont like playback anything or dial().. only logging. |
| 13:54.31 | ThoMe | is it posible? |
| 13:54.55 | ThoMe | only { Verbose(1, Ein Anruf geht ein von ${von} nach ${nach} am ${datum} um ${uhrzeit}); } and write to my csv-file or database. |
| 13:55.05 | ThoMe | can you help me pleasse? |
| 14:03.20 | WIMPy | pong |
| 14:04.01 | WIMPy | Have you taken a look at cdr.conf? |
| 14:04.28 | ThoMe | WIMPy: yes. |
| 14:04.34 | WIMPy | There's a parameter called unanswered. |
| 14:04.37 | ThoMe | but what parameter? :-( |
| 14:04.37 | ThoMe | ah |
| 14:04.41 | ThoMe | unanswered=yes? |
| 14:05.39 | WIMPy | Might help. |
| 14:06.48 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 14:22.04 | *** join/#asterisk xuw (xuw@110.77.229.124) |
| 14:30.52 | ghost75 | compiling 1h now .... |
| 14:31.06 | xuw | i'm trying to figure out how asterisk and a2billing works, but looks like im really stuck now. outbound through SIP provider works fine, but none of the DID i've setup in a2billing, will come through - they only get to the 'please enter complete pin number' ivr. |
| 14:31.49 | xuw | the extensions is probably where the problem is, but im not really sure where i should start... :/ |
| 14:32.03 | WIMPy | xuw: and how are these facts linked to another? |
| 14:32.34 | WIMPy | Are you having an issue with a2billing or with your dialplan? |
| 14:32.46 | xuw | im having an issue with not getting through when dialing the DID numbers |
| 14:33.25 | WIMPy | Coming from your SIP provider? |
| 14:33.38 | xuw | the did's are from the sip provider, yes |
| 14:34.20 | WIMPy | Set verbose to at least 3 and lokk if you can see the call at all. |
| 14:35.21 | xuw | when calling from pstn to did; |
| 14:35.32 | xuw | <PROTECTED> |
| 14:35.32 | xuw | <PROTECTED> |
| 14:35.33 | xuw | <PROTECTED> |
| 14:35.33 | xuw | <PROTECTED> |
| 14:35.36 | xuw | (sorry for flood) :p |
| 14:35.57 | xuw | where 4721984121 is the pstn number |
| 14:36.01 | WIMPy | Use a pastebin next time. |
| 14:36.13 | WIMPy | Is that the end of the call? |
| 14:36.55 | xuw | thats basically all the information that i see in asterisk when dialing with pstn->กรก |
| 14:37.04 | xuw | pstn->did |
| 14:37.44 | WIMPy | So it looks as if the a2billing AGi never returns. |
| 14:38.02 | xuw | didnt wanna paste that much |
| 14:38.06 | xuw | but it does. -- <SIP/siptrunk-00000008>AGI Script a2billing.php completed, returning 4 |
| 14:38.25 | WIMPy | Ok, so the call seeems to work just fine. |
| 14:38.46 | WIMPy | If it doesn't do what you want, change the dialplan. |
| 14:39.11 | xuw | yeah, i was wondering how i would go about to do that.. |
| 14:39.46 | xuw | a2billing has set the DID destinations to SIP/accountnumber@asterisk-ip |
| 14:39.46 | WIMPy | ~book |
| 14:39.47 | 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 |
| 14:40.40 | xuw | ~nomoneyforbooks |
| 14:41.06 | WIMPy | What about reading it in your browser? |
| 14:41.23 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 14:41.36 | xuw | ill try that. thanks. |
| 14:54.29 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 14:55.58 | Greenlight | Does the JITTERBUFFER function act as an audiohook ? |
| 14:59.39 | file | no, it acts as a framehook - 'tis lower level |
| 15:01.52 | ThoMe | WIMPy: merci! :-) |
| 15:02.12 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 15:02.37 | Greenlight | Ahh okay was trying to think of a quick fix for my bug with jitterbuffer stopping audio when bridging, and thought audiohook inheritance might help, guess not :) |
| 15:04.21 | ghost75 | (15:30:52) ghost75: compiling 1h now .... <- still |
| 15:04.47 | WIMPy | distcc |
| 15:05.05 | ghost75 | ? |
| 15:05.27 | WIMPy | Use a faster machine. |
| 15:07.07 | ghost75 | yes |
| 15:09.50 | ghost75 | finish |
| 15:10.10 | ghost75 | 1,5h |
| 15:10.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:11.29 | ghost75 | now make modules :< |
| 15:12.32 | ghost75 | WARNING: modpost: Found 10 section mismatch(es). |
| 15:15.02 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 15:49.52 | ghost75 | now asterisk doesnt detect anymore cli commands -.- |
| 16:04.52 | ghost75 | <PROTECTED> |
| 16:04.53 | ghost75 | <PROTECTED> |
| 16:04.54 | ghost75 | huh? |
| 16:06.28 | ShoreTel | lol |
| 16:06.41 | ghost75 | what this means |
| 16:36.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:42.55 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
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| 16:56.27 | *** join/#asterisk Ice_Strike (Ice_Strike@87.115.83.87) |
| 17:04.24 | igcewieling1 | stop trying to install asterisk on win32? 8-) |
| 17:13.02 | *** join/#asterisk joeb3_ (~Joe@67.143.244.149) |
| 17:15.28 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 17:17.34 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 17:20.11 | *** join/#asterisk beowu1f (~pk@41-133-132-144.dsl.mweb.co.za) |
| 17:22.09 | beowu1f | hi all. im a new too asterisk. i have a setup where i only want to make an internal call from one softphone to another. but i keep getting this message. "'7777' rejected because extension not found in context 'internal'" |
| 17:23.04 | beowu1f | the setup i am following is a very simple one. http://www.roboguys.com/index.php?option=com_content&task=view&id=65&Itemid=47 |
| 17:23.24 | beowu1f | any ideas? |
| 17:23.46 | [TK]D-Fender | beowu1f: You clearly don't have an extension to match 7777 in [internal] |
| 17:24.03 | [TK]D-Fender | beowu1f: So go look at what you think you have in there that was supposed to match ... and doesn't |
| 17:25.43 | [TK]D-Fender | beowu1f: Also that guide is SIX years old and some of the dialplan in there will not work on releases from the past several years without some fixing. |
| 17:26.19 | [TK]D-Fender | beowu1f: They also made some flat-out mistakes.as well... |
| 17:28.24 | beowu1f | [TK]D-Fender: thanks for your feedback. in your oppinion would this be more up to date? http://agix.com.au/blog/?p=2656 |
| 17:28.39 | beowu1f | i am running v1.8 atm |
| 17:29.24 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:29.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:29.44 | [TK]D-Fender | beowu1f: Yes |
| 17:30.19 | [TK]D-Fender | beowu1f: But only by being 1.4, as opposed to the previous guide's 1.0 / 1.2 |
| 17:30.32 | [TK]D-Fender | beowu1f: "canreinvite" was replaced in 1.6 |
| 17:31.03 | [TK]D-Fender | beowu1f: The rest of the basics aren't that different. |
| 17:31.07 | [TK]D-Fender | ? book |
| 17:31.09 | [TK]D-Fender | ^ |
| 17:31.12 | [TK]D-Fender | ~book |
| 17:31.12 | 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 |
| 17:31.16 | [TK]D-Fender | Dangit ^ |
| 17:32.07 | beowu1f | [TK]D-Fender: wonderfull, i'll download now an start reading |
| 17:32.09 | beowu1f | [ |
| 17:32.49 | beowu1f | [TK]D-Fender: they do say on the site that the config is for v1.8.8 |
| 17:35.08 | [TK]D-Fender | beowu1f: The uide has a few mistakes as the other did..... |
| 17:35.35 | [TK]D-Fender | beowu1f: insecure=very <-- this was replaced in *1.4* |
| 17:35.53 | [TK]D-Fender | beowu1f: both guides were written by well-meaning schmucks.... |
| 17:36.12 | beowu1f | [TK]D-Fender: i did see asterisk complaining about =very |
| 17:36.29 | [TK]D-Fender | beowu1f: register =>0812341234:MySIPPass@sip.internode.on.net/s <-- register statement have to come at the END of [general]. Any other settings below it get IGNORED |
| 17:36.54 | [TK]D-Fender | beowu1f: bindaddr=192.168.0.3 <- is unnecessary for a box with 1 IP. |
| 17:37.42 | [TK]D-Fender | beowu1f: [VoIPProvider] nat=yes <-- this has to be NO for almost 100% of providers out there... |
| 17:38.02 | [TK]D-Fender | beowu1f: canreinvite=no <-- this changed to DIRECTMEDIA in 1.6. |
| 17:38.24 | [TK]D-Fender | beowu1f: So basically... another well-meaning blog post written by a fan ... but not an informed one. |
| 17:39.04 | [TK]D-Fender | beowu1f: Actually... it pretty bad all around |
| 17:39.11 | igcewieling1 | all this info is in the UPGRADE-*.txt files included in the asterisk tarball |
| 17:39.15 | [TK]D-Fender | beowu1f: I would slap this guy.... |
| 17:42.04 | beowu1f | [TK]D-Fender: lol.. noob like me i suppose |
| 17:42.43 | beowu1f | [TK]D-Fender: im a first level noob he's a second. |
| 17:43.03 | beowu1f | [TK]D-Fender: in his defense..:) |
| 17:43.25 | [TK]D-Fender | beowu1f: Which makes him dangerous.... giving bad information to people who can't know any better... |
| 17:43.34 | beowu1f | [TK]D-Fender: true |
| 17:44.11 | beowu1f | [TK]D-Fender: can you possibly help me with a simple config? |
| 17:44.30 | beowu1f | [TK]D-Fender: or must i rtfm.. |
| 17:46.37 | beowu1f | [TK]D-Fender: dont want any outbound connections as yet. internal working first |
| 17:48.28 | [TK]D-Fender | beowu1f: Show what you've done. |
| 17:48.31 | [TK]D-Fender | ~pb |
| 17:48.31 | 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. |
| 17:48.34 | [TK]D-Fender | ^ |
| 17:48.52 | [TK]D-Fender | beowu1f: You have a dialplan error so just extensions.conf for now |
| 17:49.07 | [TK]D-Fender | beowu1f: And make sure to mask passwords when you put configs in PB's |
| 17:49.42 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-147.red.bezeqint.net) |
| 17:52.00 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 18:14.23 | beowu1f | [TK]D-Fender: sorry havent responded, i've been playing a bit. managed to get a call working.. my actuall problem before was my file was named extension.conf and not extensions.conf.. noob |
| 18:14.55 | beowu1f | i'm going to take a close look at the manual now though. try understand things from first principles |
| 18:15.04 | beowu1f | [TK]D-Fender |
| 18:15.17 | beowu1f | [TK]D-Fender: thanks for the insight |
| 18:15.55 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 18:15.59 | [TK]D-Fender | np |
| 18:35.31 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-145-215.pools.arcor-ip.net) |
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| 19:01.23 | ghost75 | ectospasm ? |
| 19:05.39 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
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| 19:59.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 20:16.40 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 20:30.19 | danfromuk | Hi. I'm struggling with a firewall/nat issue. Endpoints can connect properly, but whenever they try to Dial, they fail to respond properly to the call setup packets and asterisk decides to end the call. Has anyone seen this before where INVITE sip packets seem to be filtered out but REGISTER and OK seems to be working fine? |
| 20:30.39 | danfromuk | Tried disabling and enabling SIP ALG. It made no difference. |
| 20:32.35 | ChannelZ | Havent heard of anything |
| 20:42.42 | edong23 | danfromuk: nat? well... nat transversal has always worked for me with asterisk, but anyway... perhaps grab a tcpdump on either side of your firewall? |
| 20:42.48 | ChannelZ | Wht does sip debug show? Does the invite reach asterisk but the phones fail to communicate to acks and things afterwards? Your original query was a little unclear |
| 20:46.57 | *** join/#asterisk edwin_quijada (~macaruchi@186.33.81.33) |
| 20:47.01 | edwin_quijada | hi |
| 20:47.56 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.209.86) |
| 20:48.21 | edwin_quijada | I am using Gigaset C610 with asterisk but I dont know how register this base to asterisk. Somebody has used this device |
| 20:50.09 | edong23 | set the username and password and go like damnit! |
| 20:50.20 | danfromuk | ChannelZ: Just doing a pastebin. |
| 20:53.53 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:a12b:399d:f867:7272) |
| 20:55.49 | ChannelZ | edwin_quijada: is that even a SIP phone? |
| 20:57.09 | ChannelZ | it seems like an analog phone in which case you need an ATA of some sort |
| 20:58.55 | edong23 | ... |
| 20:58.56 | edong23 | lol |
| 20:59.40 | danfromuk | ChannelZ: Here is a pastebin showing what i described. |
| 20:59.41 | danfromuk | http://pastebin.com/xAXJp2Ga |
| 20:59.58 | danfromuk | You'll see that the endpoint seems to respond to some SIP packets but not all. |
| 21:00.39 | danfromuk | Its a polycom and we've got loads of the same model connected to our server. |
| 21:00.43 | danfromuk | Without any issues. |
| 21:02.18 | *** join/#asterisk joeb3_ (~Joe@67.143.244.149) |
| 21:03.42 | *** join/#asterisk din3sh (2988521a@gateway/web/freenode/ip.41.136.82.26) |
| 21:05.03 | din3sh | hi all |
| 21:05.33 | WIMPy | lo you |
| 21:06.16 | din3sh | hey WIMPy |
| 21:06.53 | din3sh | 2 weeks i've not heard my client complain after 1 swapped the digium card |
| 21:06.57 | WIMPy | How are things going? |
| 21:07.02 | edwin_quijada | ChannelZ: is a base SIP with dect phone but I dont know how to register to asterisk |
| 21:07.14 | WIMPy | Smae model or another one? |
| 21:07.20 | din3sh | same model |
| 21:07.29 | din3sh | but 5th generation |
| 21:07.31 | leifmadsen | analog or digital card? |
| 21:07.32 | din3sh | instead of 4th |
| 21:08.14 | din3sh | digium told me 4th generation cards are known to have hdlc issues |
| 21:08.26 | WIMPy | wonders how they can fail graueally. |
| 21:08.48 | WIMPy | +d |
| 21:09.23 | din3sh | havent had the chance to look at the logs yet, but client has not complained about previous problems |
| 21:10.12 | din3sh | busy installing another box with a 4 span digium card :D |
| 21:10.52 | ghost75 | cant get bluetooth working durrrr |
| 21:11.18 | ChannelZ | danfromuk: hmm I dunno. It looks like the phone just falls asleep. |
| 21:11.37 | din3sh | is it good practice to use irqbalance to distribute the E1 card load accross different cores? |
| 21:12.03 | din3sh | cat /proc/interrupts shows it hammering a single core out of 16 |
| 21:12.31 | WIMPy | It houldn't really make a difference. |
| 21:13.02 | WIMPy | On a machin that was too slow I made the observations that using only one core worked a tiny bit better. |
| 21:13.50 | din3sh | its a double quad processor box |
| 21:16.27 | ChannelZ | a quad laser? |
| 21:16.55 | ghost75 | a panzer |
| 21:17.04 | ChannelZ | no one can defeat the quad laser! |
| 21:17.28 | din3sh | :D |
| 21:17.56 | ghost75 | i can haz lazers? |
| 21:17.56 | ChannelZ | http://www.youtube.com/watch?v=TNYMxgNKIEU |
| 21:47.42 | edong23 | i immediately thought of nexuiz/quake |
| 21:54.09 | WIMPy | Nice. I get a Session-Timer expired, but I don't see any reinvite attepts. |
| 22:05.20 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 22:37.10 | ^rage^ | hi! |
| 22:37.23 | ^rage^ | anybody using asterisk + zeromq? |
| 22:38.16 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 22:40.14 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:40.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:27.23 | jzaw | hi im trying to get * talking to google voice ... gtalk i suppose |
| 23:27.28 | jzaw | using svn 11 |
| 23:27.44 | jzaw | ive selected chan motif |
| 23:28.06 | jzaw | but its not registering as a channel |
| 23:28.20 | jzaw | [Mar 2 23:23:24] WARNING[27245][C-00000001]: channel.c:5956 ast_request: No channel type registered for 'MOTIF' |
| 23:28.33 | jzaw | can someone help to debug pls? |
| 23:28.49 | jzaw | i had this working with jingle/gtalk.conf |
| 23:29.10 | jzaw | but the switch to motif has caused me to break something |
| 23:29.36 | leifmadsen | jzaw: that means chan_motif is not loaded and/or built |
| 23:29.47 | leifmadsen | e.g. there is no chan_motif loaded into memory |
| 23:30.02 | leifmadsen | in menuselect make sure it is selected and the deps are satisfied |
| 23:30.22 | jzaw | ive checked ... but will check again |
| 23:31.09 | leifmadsen | <PROTECTED> |
| 23:31.13 | leifmadsen | that definitely means chan_motif is not loaded |
| 23:31.18 | jzaw | nods |
| 23:31.37 | jzaw | im just make menuselect ing |
| 23:32.39 | jzaw | channel drivers ... chan_motif selected |
| 23:33.00 | leifmadsen | did it get installed in /usr/lib/asterisk/modules? is it loaded when you do a "module show like motif" ? |
| 23:33.01 | jzaw | chan_gtalk and chan_jingle DEseleted |
| 23:33.11 | leifmadsen | right, you don't care about those though |
| 23:33.22 | leifmadsen | it's chan_motif you need to make sure is loaded and has a configuration file |
| 23:33.27 | jzaw | nods |
| 23:34.54 | jzaw | yep chan_motif.so resent in /usr/lib/asterisk/modues |
| 23:34.56 | jzaw | modules |
| 23:35.03 | jzaw | present |
| 23:35.10 | jzaw | damn keyboard |
| 23:35.25 | jzaw | + has config file |
| 23:36.42 | jzaw | core show channeltypes |
| 23:36.49 | jzaw | does not list motif |
| 23:40.50 | jzaw | leifmadsen, module show like motif |
| 23:40.51 | jzaw | gives |
| 23:40.54 | jzaw | chan_motif.so Motif Jingle Channel Driver 0 |
| 23:40.54 | jzaw | 1 modules loaded |
| 23:41.05 | leifmadsen | shrugs |
| 23:41.06 | leifmadsen | should work then |
| 23:41.17 | leifmadsen | are you literally using MOTIF though? |
| 23:41.21 | leifmadsen | maybe you need to use Motif? |
| 23:41.29 | leifmadsen | seems suspect, but who knows |
| 23:41.34 | jzaw | i was just about to ask about capitalisation |
| 23:41.57 | jzaw | all my other calls are SIP and IAX2 ... no sip nor Sip |
| 23:42.07 | jzaw | will try |
| 23:45.12 | jzaw | nope neither Motif nor motif |
| 23:45.37 | [TK]D-Fender | jzaw: And do you have it configured? |
| 23:45.48 | jzaw | as in ive got a motif.conf ? |
| 23:45.49 | jzaw | yes |
| 23:45.54 | [TK]D-Fender | Show us |
| 23:45.56 | [TK]D-Fender | ~pb |
| 23:45.56 | 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. |
| 23:46.00 | [TK]D-Fender | ^^ |
| 23:46.02 | jzaw | indeed |
| 23:46.12 | [TK]D-Fender | ls -la /etc/asterisk |
| 23:47.43 | pabelanger | leifmadsen: jzaw no, it is not setup properly |
| 23:47.57 | jzaw | http://pastebin.com/DtBGuSk5 |
| 23:47.58 | pabelanger | there was a blog post recently from digium adding new logic about module state |
| 23:48.34 | jzaw | go on pabelanger |
| 23:48.36 | pabelanger | http://blogs.digium.com/2013/02/21/your-asterisk-module-is-loaded-but-is-it-running/ |
| 23:48.45 | jzaw | will read |
| 23:49.01 | pabelanger | but, not sure why mofit is not working |
| 23:49.06 | pabelanger | I have not tried it yet |
| 23:49.41 | [TK]D-Fender | jzaw: Folder dump as requeted please... |
| 23:49.49 | jzaw | ah yes soz |
| 23:49.51 | jzaw | coming up |
| 23:52.29 | jzaw | http://pastebin.com/0j6SHCsk |
| 23:52.37 | jzaw | [TK]D-Fender, ^^^ |
| 23:52.55 | jzaw | pabelanger, is that blog saying its a conf error on my part ? |
| 23:53.02 | jzaw | thats how im reading it |
| 23:53.20 | pabelanger | that is my guess |
| 23:53.28 | jzaw | effectively the module is requested ... loaded ... but wrongly configured |
| 23:53.38 | pabelanger | ya |
| 23:53.44 | [TK]D-Fender | jzaw: Running Asterisk as root? Not great... |
| 23:53.53 | jzaw | yeah i know |
| 23:54.05 | [TK]D-Fender | jzaw: I also see you didn't sepcify an "transport" for those entries you made |
| 23:54.18 | [TK]D-Fender | any* |
| 23:54.21 | jzaw | im listening |
| 23:55.52 | jzaw | ice-udp ? |
| 23:56.23 | jzaw | has read so many google pages now that its all mixing up |
| 23:57.30 | jzaw | i thought i read somewhere that if its not specified ... it reverts to ice-upd |
| 23:57.34 | jzaw | udp |
| 23:57.37 | jzaw | but ill put it in |
| 23:59.42 | *** part/#asterisk tallest_red (~CNZ@ip98-169-207-41.dc.dc.cox.net) |