| 00:00.05 | mrjazzman | ChannelZ: if that q was to me, Basically i think i need a queue that is in round robbin but rather than stations, i need the calls to go to external SIP calls (as in make a call through my provider to an external number). |
| 00:00.42 | *** join/#asterisk Wiktor (wiktor@2001:470:9a06::1) |
| 00:01.05 | Wiktor | hi, anybody online and have some time to help me out ? :) |
| 00:01.22 | beanie | TazzNZ - it says to change the date format but does that mean replicating it then by way of adding the new format in the custom file? |
| 00:01.38 | beanie | sorry was just getting the oven pride out as you do at 1am ha |
| 00:04.51 | Wiktor | what are the possible reasons of situation when my [incoming] call from outside isn't handled by the dialplan instructions ? in sip.conf i have context=incoming in [general], and in extensions.conf under incoming i have exten => s,1,Answer()? |
| 00:08.04 | TazzNZ | Wiktor: try exten => _X.,1,Answer() |
| 00:08.46 | TazzNZ | beanie: if it goes under [general], you can only have 1 of them |
| 00:09.23 | Wiktor | i have very basic configurations, written from zero. Something is down with them, because changing them to default ones works like a charm. here are them http://pastebin.com/LVWRMtfX |
| 00:09.28 | Wiktor | TazzNZ: thx, ill try it now |
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| 00:12.13 | Wiktor | TazzNZ: still nothing, i'm not getting through. SIP debugs shows me that i'm getting to asterisk, but no stuff in console, and telefon goes "network busy" |
| 00:13.45 | Wiktor | and also when i'm trying to call my cellphone from sip client it searches that number in internals |
| 00:14.49 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 00:25.14 | beanie | TazzNZ will it cause any problems if i edit that file - i have been told it would |
| 00:25.26 | beanie | i.e. the logger.conf |
| 00:26.05 | *** part/#asterisk Defraz (~Defraz@209.141.122.71) |
| 00:32.04 | mrjazzman | quick one - i think what i want is queues and i've setup a test one but i can't make the Sip member dial. Any suggestions? |
| 00:35.43 | beanie | is anybody around who can help me get my fail2ban up and working - im scared of leaving the server insecure when i return to my mums down south later today |
| 00:36.57 | mrjazzman | beanie: for asterisk or ssh? |
| 00:39.37 | Penguin | Did you read the fail2ban guide on voip-info.org? |
| 00:39.41 | Penguin | beanie: ^ |
| 00:39.50 | WIMPy | mrjazzman: Do you want RR or rather just call the whole lost for the fastest one to pick the call? |
| 00:40.29 | mrjazzman | WIMPy: RR is preference as it's a list of mobiles which may go t o voicemail if one of them is on the phone |
| 00:40.49 | Penguin | beanie: The guide should be pretty reasonable. I remember a time when I followed it and got fail2ban working without too much trouble. |
| 00:40.50 | WIMPy | VM is evil. |
| 00:40.59 | mrjazzman | yes yes it is |
| 00:41.20 | mrjazzman | WIMPy: i have a queue, i can see the 2 sip members , i call the number and get the on hold music, but it doesn't seem to dial the members |
| 00:41.42 | WIMPy | Look at your console. |
| 00:41.51 | mrjazzman | members show up as "Not in use" but not sure if that is they don't have any calls or they aren't logged in |
| 00:41.54 | mrjazzman | WIMPy: that's from the console |
| 00:42.08 | beanie | mrjazzman - i'm currently trying to secure my server as im very scared - im going back down to care for my mother later today for another couple of months |
| 00:42.15 | beanie | i want to leave the server secure but accessible to me |
| 00:42.48 | beanie | penguin - thats the guide im really struggling with :) |
| 00:42.49 | mrjazzman | beanie: Penguin has linked you to the asterisk side but fail2ban comes pre-configured for ssh on ubuntu/debian |
| 00:42.51 | beanie | i am trying though! |
| 00:42.52 | WIMPy | And what do you see while you have a call? |
| 00:42.58 | mrjazzman | beanie: happy to throw you the config if you want |
| 00:43.30 | beanie | well the question i have os as follows: http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 00:43.37 | Penguin | The necessary configs should be on the voip-info.org fail2ban guide. |
| 00:43.45 | WIMPy | And (Not in use) just means idle. |
| 00:43.45 | beanie | on there, it says "Likewise, you willl also need to ensure the date format has been changed in logger.conf to "dateformat=%F %T"." |
| 00:44.00 | Penguin | I've tried to keep it updated when possible. |
| 00:44.02 | beanie | now ive been told time and time again not to edit the original files |
| 00:44.02 | mrjazzman | WIMPy: call enters the queue, (sorry trying to deal with this box being live for DR also so getting a lot of other calls coming through that i need to filter through) |
| 00:44.07 | beanie | but i dont see how it is otherwise possible |
| 00:44.27 | mrjazzman | WIMPy: ok get the ring, answer and queue |
| 00:44.37 | mrjazzman | and then a bunch of SIP RTP COS updates but that's it |
| 00:44.44 | beanie | mrjazzman - thanks for the offer but im not quite sure what id do with it |
| 00:44.45 | mrjazzman | doesn't seem to dial the other sip members |
| 00:45.16 | WIMPy | That SIP COS stuff comes up when aSIP channel is created, so that looks like something IS dialled. |
| 00:45.27 | Penguin | beanie: My dateformat is commented out, therefore default. |
| 00:45.39 | mrjazzman | WIMPy: hmmmmm ok thanks i'll chase a little more |
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| 00:58.28 | Penguin | beanie: If you are following the guide and run into a specific problem, I'll try to help you through it if you'll just tell me what the specific issue is. |
| 01:02.58 | mrjazzman | Ok this is annoying me.... I can get a call into a queue, but can't get it to ring a SIP agent... Any pointers? |
| 01:03.21 | WIMPy | Turn up verbose if you don't see anything. |
| 01:04.03 | mrjazzman | WIMPy: k trying |
| 01:07.45 | TazzNZ | beanie: if you edit logger.conf, the next time you apply changes it will be overwritten |
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| 01:39.13 | mrjazzman | WIMPy: ok verbose is up to 99 - i see it go into hold |
| 01:39.15 | mrjazzman | but nothing else |
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| 01:51.32 | Penguin | mrjazzman: How many members are in the queue? What is your member type for the queue member(s)? |
| 01:53.35 | mrjazzman | Penguin: 2 members, trying to set them to be external numbers called through my sip provider |
| 01:53.44 | mrjazzman | Penguin: stratergy is now ringall (was rrmemory) |
| 01:53.54 | mrjazzman | re member type, whatever is default |
| 01:54.23 | Penguin | mrjazzman: How are you defining the members? There is no default. The members use whatever channel you set up. |
| 01:54.34 | mrjazzman | Penguin: http://pastebin.com/6SNjVDth |
| 01:54.40 | mrjazzman | that's my queues.conf |
| 01:54.59 | mrjazzman | Penguin: thanks for the assist. Still getting my head around all of this |
| 01:55.22 | Penguin | <PROTECTED> |
| 01:55.33 | Penguin | SIP/1234567890@exetel is an invalid channel. |
| 01:55.51 | mrjazzman | ok |
| 01:55.59 | Penguin | If you are trying to dial an external number, I'd recommend using a Local channel. |
| 01:56.00 | mrjazzman | so i have a sip partner called "exetel" |
| 01:56.13 | mrjazzman | so LOCAL/123456780@exetel? |
| 01:56.21 | WIMPy | Why is that invalid? |
| 01:56.40 | Penguin | For example, if you have a context used for outbound calls that is named 'outbound', you could use Local/1234567890@outbound. |
| 01:57.23 | Penguin | If you must use a DEFINED SIP PEER, try using the correct syntax: SIP/exetel/1234567890 |
| 01:57.33 | WIMPy | doesn't see why you can't use sip/itsp/number as member. |
| 01:57.57 | WIMPy | The other syntax should do the same, shouldn't it? |
| 01:59.27 | mrjazzman | ok |
| 01:59.35 | mrjazzman | if i define SIP/exetel/123567890 |
| 01:59.38 | Penguin | I seem to remember having problems using SIP channels as queue members, so I switched to using local channels a long time ago. |
| 01:59.41 | mrjazzman | i get (unknown) when doing a queue show |
| 01:59.53 | mrjazzman | when i do SIP/36248783244@exetel it shows "not in use" |
| 02:00.05 | mrjazzman | also tried Local/12783791@exetel and it came up as "invalid" |
| 02:00.29 | Penguin | If exetel isn't a valid dialplan context, it should have shown that. |
| 02:01.07 | mrjazzman | exetel is a sip.conf definition |
| 02:01.22 | Penguin | That doesn't make it a dial plan context. |
| 02:01.53 | Penguin | If you can't "dialplan show exetel" and get a good response, it's not a dial plan context. |
| 02:03.14 | Penguin | If you have configured a dial plan context where you do outbound dialing through that peer, use that context in a local channel. |
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| 02:05.50 | mrjazzman | Penguin: i get extenion information |
| 02:06.01 | mrjazzman | Guessing i need one for the queue? |
| 02:06.33 | Penguin | You can create a new context for putting queue extensions if you want, but it's not necessary. |
| 02:06.41 | Penguin | <Penguin> If you have configured a dial plan context where you do outbound dialing through that peer, use that context in a local channel. |
| 02:06.45 | Penguin | THIS ^ |
| 02:08.34 | mrjazzman | Penguin: ok i'll go RTFM on trhat part |
| 02:08.35 | mrjazzman | thanks |
| 02:09.05 | Penguin | Did you not already write some dialplan? |
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| 02:10.17 | mrjazzman | Penguin: i have stuff in extensions.conf that handles incoming |
| 02:10.25 | mrjazzman | Penguin: part of that is using SIP/<Number>@exetel |
| 02:10.26 | mrjazzman | and it works |
| 02:10.39 | mrjazzman | but i'm guessing i'm missing a default outbo8und dialplan |
| 02:10.55 | Penguin | I advise you to read the book on how to form a proper dial string. |
| 02:10.57 | Penguin | ~book |
| 02:10.57 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:10.57 | mrjazzman | is new |
| 02:12.08 | mrjazzman | Penguin: thanks I do mean to get myself a book on this. TBH we use asterisk only in a limited fashon and have been able to fudge our way through so far. I would love to get it to do more but time & effort is being pushed on other things. It's only that we're using it to deal with ISDN outages that htis requirement has hit the fan |
| 02:12.17 | mrjazzman | Penguin: thanks for your help |
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| 02:13.44 | Penguin | I'll give you a tip. The format you should use is SIP/peername/number2call. |
| 02:13.53 | mrjazzman | ta |
| 02:14.08 | Penguin | perhaps SIP/exetel/${EXTEN} |
| 02:18.43 | zopsi | pjsip usernames can be alphanumeric right? |
| 02:20.17 | [TK]D-Fender | [20:44]beanienow ive been told time and time again not to edit the original files <- beent hrough that 50 times as well.. you can change the CUSTOM ones. |
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| 02:52.19 | mrjazzman | Penguin: thanks for your help |
| 02:52.22 | mrjazzman | managed to get it working |
| 02:52.31 | mrjazzman | WIMPy: thank you also for your help |
| 02:52.33 | mrjazzman | appreciate it |
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| 03:31.19 | *** join/#asterisk ledoktre (~ledoktre@216.51.224.229) |
| 03:32.25 | ledoktre | quick question fellas -- is there a way to store a devstate (Custom:Test123 for example) to astdb? Saving it there I suppose is not hard, save it when you set it. But what about when the system reboots and reloading the devstate back? |
| 03:41.57 | [TK]D-Fender | On start-up run a refresh based off of the AstDB to reset them |
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| 03:46.15 | ledoktre | [TK]D-Fender - Greetings. That sounds like just what I need. Is there a link handy for that? You've fed me the help bot thing on here a few times in the past ;- ) |
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| 03:47.09 | [TK]D-Fender | Read the sample extensions.conf. You'll see something highly triggerable there... |
| 03:49.45 | ledoktre | thanks :) No matter what anyone else says. You're okay ;-D |
| 03:58.10 | [TK]D-Fender | unappreciated by the lazy and deaf.... such is my lot in life. |
| 03:58.39 | ledoktre | Well, make sure no one is looking, and go ahead and pat yourself on the back. |
| 04:03.28 | stevePearPear | hi i noticed in the SIP Invite SDP sent by Asterisk, when it was sent from Asterisk to the SIP provider, there was no a=rtcp:port IN IP4 xxx.xxxx.xxx.xxx where xxx should be Asterisk public IP address |
| 04:04.02 | stevePearPear | is that normal? cuz I saw this field when my SIPML5 sent an INVITE to Asterisk |
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| 04:11.26 | ledoktre | [TK]D-Fender: I saw `#exec /path/to/file` which I could probably use to reload that one thing. But from what you wrote, I got the impression you could have it refresh everything, not explicitly set things. ? |
| 04:12.16 | [TK]D-Fender | because it COULD be hit with a "reload" and not just a "first load" ... you'd have to reset EVERYTHING in each pass |
| 04:14.54 | ledoktre | right now I'm only storing maybe 5 things in the astdb. reset everything isn't a big deal - I'm just trying to do this the "asterisk" way. I/m just not sure exactly what that is. I'm googling on it and Ive read that sample file a couple of times. I'm trying :) |
| 04:17.29 | [TK]D-Fender | in your script.... set everything "off" then back on if AstDB is set for it |
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| 04:29.04 | [TK]D-Fender | bedtime, checking out... |
| 04:36.00 | ledoktre | good night. I think i did it just by setting a global variable with its value loading from the DB. I *think*. |
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| 04:38.44 | stevePearPear | hey is sip_nat.conf still used in asterisk 12? |
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| 04:55.16 | x86 | ok... so is there no longer an option to run chan_motif to Google, without the requirement of ICE support? |
| 04:55.51 | x86 | I rebuilt my home general purpose server completely, but I made a backup of the entire /etc/asterisk first |
| 04:58.06 | x86 | the server had been running 11.3, and now I'm trying to get 11.10 running, but every time I try to make a call it gives me reorder tone right after bitching about lack of ICE support (which, is enabled in rtp.conf as well as sip.conf... not sure why it wont actually enable, even though explicitly configured) -- how can I disable ICE support completely, short of reverting back to an older release? |
| 04:58.51 | x86 | because "icesupport=no" in rtp.conf and sip.conf appeared to have no effect at all after asterisk was restarted, still instant reorder tone |
| 04:59.30 | x86 | where is the "FOAD ICE" option? ;) |
| 05:02.05 | x86 | alternatively, if I could get asterisk to actually enable ICE somehow (~@$T!#%Y!<#%YL#%Y<), that would probably work also |
| 05:03.34 | x86 | Jun 10 01:03:48] WARNING[29205][C-00000001]: chan_sip.c:10103 process_sdp: Declining non-primary audio stream: audio 2240 RTP/AVP 0 8 18 101 |
| 05:03.38 | x86 | [Jun 10 01:03:48] ERROR[29198][C-00000001]: chan_motif.c:821 jingle_add_google_candidates_to_transport: Unable to add Google ICE candidates as ICE support not available or no candidates available |
| 05:04.09 | x86 | (icesupport=true in rtp.conf) |
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| 05:04.56 | x86 | meh I'm going to bed... hopefully no one tries to on-call call me ;) |
| 05:06.44 | stevePearPear | i was having ICE support not enabled too few days back and seems like Asterisk requires pjproject for STUN/ICE |
| 05:06.48 | stevePearPear | not sure if it helps |
| 05:07.35 | stevePearPear | hey can I check for those with 2 ways audio, when you check rtp debug, do you always have from and to for both ways? |
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| 05:12.10 | stevePearPear | right now all i have is only Got RTP packet from xxx.xxx.xxx.xxx and Sent packet to yyy.yyy.yyy.yyy |
| 05:12.25 | stevePearPear | there isn’t GOT RTP packet from yyy.yyyy.yyy.yyy SENT packet to xxx.xxx.xxx.xxx |
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| 05:29.39 | maani | Hi, with "dahdi show channels" I can see just incomming calls on pri not outgoing calls is there a way to see all calls on pri link ? |
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| 06:10.38 | ChannelZ | maani: actually I don't think 'dahdi show channels' is showing you anything except all the configured channels. To see actual channels in use, try 'core show channels' |
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| 07:58.09 | ruben23 | anyone familiar with this error after install ---> Illegal instruction (core dumped) <--- asterisk 1.8 |
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| 08:33.39 | KNERD | ruben23: you on a virtual macine? |
| 08:34.59 | KNERD | and what exact vesion you using? |
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| 08:45.27 | stevePearPear | Hi I’m just curious if this was an issue with anyone else: https://code.google.com/p/sipml5/issues/detail?id=156#makechanges |
| 08:45.39 | stevePearPear | using Asterisk hosted on AWS had that problem where its one way audio |
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| 09:22.25 | michael_work | stevePearPear, did you tried to run wireshark on server and on computer? |
| 09:23.15 | michael_work | do you use stun? |
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| 09:34.14 | stevePearPear | yeah i used wireshark |
| 09:34.22 | stevePearPear | and there’s stun, I could see it too |
| 09:35.15 | stevePearPear | is it weird eg. if there are 2 set of ip, xxx.xxx.xxx.xxx and yyy.yyy.yyy.yyy, when i do rtp set debug on, i only see SENT rtp packet to xxx.xxx.xxx.xxx and RECIEVE from yyy.yyy.yyy.yyy and there’s no SENT rtp packet to yyy.yyy.yyy.yyy |
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| 09:49.23 | michael_work | did you try to run wireshark on the end that needs to send? |
| 09:49.37 | michael_work | maybe it's sending but to the wrong ip? |
| 09:49.49 | michael_work | you need to run wireshark on 3 points |
| 09:49.55 | michael_work | server/leg1/leg2 |
| 09:49.59 | michael_work | and check what's wrong |
| 09:50.08 | michael_work | check in SDP what ip they use |
| 09:50.13 | michael_work | check NAT settings and so on |
| 09:50.20 | michael_work | firewalls |
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| 13:26.08 | defswork | Intermittent one way audio on internal only calls ? any suggestions for what it might be ? we've ran out of ideas |
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| 13:49.06 | jameswf | defswork: are your phones and users on the same LAN |
| 13:51.09 | defswork | jameswf, yes |
| 13:51.18 | defswork | jameswf, all on the same switch also |
| 13:51.58 | defswork | every now and then I see high ms in the sip peers list (1sec+) |
| 13:52.11 | defswork | not sure if thats related though |
| 13:52.46 | jameswf | defswork: try splitting your networks. You probably have user(s) streaming media or pulling down torrents or doing some other intensive tasks |
| 13:54.08 | defswork | jameswf, we've put the majority of the handsets on their own switch |
| 13:55.06 | jameswf | move em all to their own switch and subnet |
| 13:57.21 | Katty | oh where is my hairbrush |
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| 13:58.30 | Matt | frowns |
| 13:58.42 | Katty | you turn that frown upside down. |
| 13:59.18 | Matt | if you can assist with the magic incantation to make * behave with my upstream provider's Nortel box I will :) |
| 13:59.40 | WIMPy | Katty: Use more Asterisk and you won't need a hairbrush any more. |
| 13:59.52 | Katty | i love you, WIMPy |
| 14:00.03 | Matt | I've got it registering, and incoming calls seem fine, but the upstream box is rejecting outbound calls with a 404 |
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| 14:00.47 | WIMPy | thinks he's not as bad as Asterisk. |
| 14:00.54 | WIMPy | .. most of the times. |
| 14:00.59 | WIMPy | ;-) |
| 14:01.08 | Matt | comparing packet dumps between * talking to $provider (which isn't working), and a GXP2000 talking to $provider (which does work), it looks like it's the domain on the destination address in the INVITE |
| 14:02.11 | Matt | the GXP2000 sends an INVITE From: sip:me@ourdomain.com To: sip:number@ourdomain.com to proxy.provider.com |
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| 14:02.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:02.28 | Katty | WIMPy: i'm not entirely sure what you're trying to tell me lol |
| 14:02.31 | Katty | hugs sruffell |
| 14:02.36 | Matt | whereas * sends an INVITE From: sip:me@ourdomain.com To: sip:number@proxy.provider.com to proxy.provider.com |
| 14:02.38 | sruffell | waves |
| 14:03.02 | Matt | and I'm not sure how to change that behaviour |
| 14:03.02 | Katty | how're you dear |
| 14:04.14 | WIMPy | Matt: Set the host to what you want in the to and outboundproxy to the proxy. |
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| 14:07.05 | Matt | ok, lemmie have a poke at that |
| 14:09.28 | Matt | so when I set host=ourdomain.com, it starts sending packets to ourdomain.com, not proxy.provider.com |
| 14:09.50 | Matt | and as ourdomain.com is out webserver, and doesn't run any SIP services, that doesn't work :) |
| 14:11.26 | [TK]D-Fender | mattPB your peer. |
| 14:11.30 | [TK]D-Fender | ~pb |
| 14:11.31 | 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. |
| 14:11.34 | [TK]D-Fender | ^^^^^^^^^^ |
| 14:11.54 | Katty | hi fender bender. |
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| 14:13.14 | [TK]D-Fender | Katty: Mew. |
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| 14:19.53 | Matt | one mo |
| 14:20.04 | Matt | (I'm kinda tinkering with this in between real work) |
| 14:22.07 | Matt | you want the one that was working for incoming, but fails with outgoing, or the one I just changed that's not registering (cause it's talking to the wrong endpoint)? |
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| 14:24.45 | [TK]D-Fender | your peer to your provider has nothing to do with your regsitration to it |
| 14:25.19 | [TK]D-Fender | You were discussing calls to them not being formatted properly, so show us our peer entry |
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| 14:27.28 | Matt | http://pastebin.com/Fexfy9QP |
| 14:27.40 | Matt | that what you're after? |
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| 14:31.48 | [TK]D-Fender | mattWhy are the outboundproxy & host the SAME? You jsut said they were supposed to be different. |
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| 14:32.14 | [TK]D-Fender | host ad fromdomain should be the same. |
| 14:32.20 | [TK]D-Fender | Not the outboundproxy |
| 14:32.34 | [TK]D-Fender | "say I'm from X, send to Y" |
| 14:33.28 | Matt | if I change the host entry to read 'host=myworkdomain.com', then the peer goes unreachable, and tcpdump shows it trying to talk to the IP that myworkdomain.com resolves to |
| 14:34.22 | Matt | the box I'm talking to here is odd |
| 14:34.37 | [TK]D-Fender | disable qualify for it and test |
| 14:35.20 | Matt | so host=myworkdomain.com and qualify=no ? |
| 14:37.12 | [TK]D-Fender | yes |
| 14:37.21 | Matt | ahha, that might have done the trick, thanks |
| 14:38.45 | Matt | now I just need to deal with codecs |
| 14:39.17 | [TK]D-Fender | disallow=all <- before your allow |
| 14:39.23 | Matt | nods |
| 14:39.30 | Matt | I have no idea what the other end accepts |
| 14:39.40 | [TK]D-Fender | because if you don't everything else may be allowed from [general], which could mean "everything" |
| 14:39.50 | [TK]D-Fender | look at the call.... you'll see. |
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| 14:42.36 | Matt | excellent, looks like I may have this working |
| 14:43.08 | Matt | thanks muchly |
| 14:43.21 | Matt | I really need to read more into this stuff |
| 14:45.32 | [TK]D-Fender | You're welcome |
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| 16:09.15 | SpaceInvaders | Hey, in sip.conf when it says "With the current situation, you can do one of four things:" and one of the four things is "a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1" what is it "listening" for? |
| 16:09.54 | pabelanger | SpaceInvaders, a SIP packet |
| 16:10.47 | SpaceInvaders | thanks! |
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| 16:22.25 | Katty | safe travels, leifmadsen |
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| 16:52.39 | zopsi | mjordan: so I was able to get everything working except the digium phones. For some reason the extensions work fine if I use a softphone, but anytime I dial on the digium phones I get failed immediately with little information in the cli. Any ideas? |
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| 16:54.01 | mjordan | zopsi: not enough information :-) |
| 16:54.16 | zopsi | mjordan: I'll dig into the logs. |
| 16:54.28 | mjordan | zopsi: pb your res_digium_phone.conf |
| 16:54.36 | mjordan | zopsi: also, check the CLI to see if you have any sessions |
| 16:54.43 | mjordan | digium_phones show sessions, IIRC |
| 16:55.05 | zopsi | I have one active session for the one phone I have configured for now. |
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| 16:56.57 | zopsi | http://pastebin.com/mTSgYP7D |
| 16:57.04 | zopsi | mjordan: see above |
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| 17:18.33 | zopsi | mjordan: I'm heading off to work, but let me know if you have any ideas as I really love the new chan_pjsip and DPMA module |
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| 18:35.43 | qakhan | hi all, i am getting following message when i try to run agi |
| 18:35.48 | qakhan | <PROTECTED> |
| 18:36.30 | [TK]D-Fender | clearly thte module offering the app is not being loaded |
| 18:37.24 | qakhan | WARNING[11488]: loader.c:908 load_resource: Module 'res_agi.so' already exists. |
| 18:38.15 | [TK]D-Fender | Show us your modules list, an attempt to reload the module, an actual full call attempt, etc |
| 18:41.14 | qakhan | [TK]D-Fender this module show like http://pastebin.com/n28iCQMM |
| 18:41.43 | Qwell | qakhan: just 'module show' |
| 18:42.29 | andi | Hi |
| 18:42.44 | qakhan | here http://pastebin.com/nVHc9nFq |
| 18:42.51 | andi | Is there an easy way to get VoiceMail() to not say the digits of the exten which was called? |
| 18:46.09 | qakhan | ok i got it. there was , after AGI |
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| 18:51.34 | [TK]D-Fender | andi: record an actual message |
| 18:52.32 | Qwell | lunch |
| 18:52.36 | Qwell | wrong channel |
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| 19:00.57 | Vendigroth | Hi everyone, Is it normal for asterisk to take up to 5 seconds to reply to an OPTIONS message? Is there a reason why it could be happening? (it does seem to happen under load but still.. 5 is a lot) |
| 19:01.53 | Chainsaw | Vendigroth: DNS delays come to mind. |
| 19:02.09 | Chainsaw | Vendigroth: Asterisk 11 and below will not entertain such delays. Provide a local resolver if you have to. |
| 19:06.13 | zopsi | mjordan: sorry I'm back. Did you get a chance to look at my res_digium_phone.conf? |
| 19:06.28 | Vendigroth | Chainsaw: We're on 11.6.0, and have a local dns, but you still might be right. i'll see if that's the cause. thanks. |
| 19:06.59 | Chainsaw | Vendigroth: It will deal with DNS problems worse than words can describe. One peer can in fact hold up the other. |
| 19:07.27 | Chainsaw | Vendigroth: Should that not be it, if you have STUN (or ICE) configured, ensure that the STUN server in question is responsive. |
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| 19:18.44 | qakhan | when we need asterisk licenses? is it not free? |
| 19:21.33 | [TK]D-Fender | What licenses? |
| 19:22.12 | Chainsaw | I can only think of G729 transcoding licenses. |
| 19:22.33 | Chainsaw | just bought a DSP card |
| 19:22.36 | [TK]D-Fender | I can think of all sorts of things... |
| 19:22.41 | Chainsaw | If I have to pay money I want hardware offload. |
| 19:22.59 | [TK]D-Fender | But second-guessing a vague & unusable question like that is a waste of time. |
| 19:23.11 | [TK]D-Fender | qakhan: What licenses? |
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| 19:27.45 | qakhan | http://www.digium.com/en/products/asterisk/licensing |
| 19:27.54 | qakhan | [TK]D-Fender look at this |
| 19:30.29 | [TK]D-Fender | qakhan: That page explains it all by itself |
| 19:31.15 | qakhan | if you setup an asterisk for a company then do i have to take license? |
| 19:32.09 | [TK]D-Fender | qakhan: You clearly have not read that page |
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| 19:34.13 | qakhan | i read it but i could not understand it. i sent mail to digium and asked the same question. and they said yes i have to buy license |
| 19:35.38 | [TK]D-Fender | qakhan: Are you looking to MODIFY the Asterisk source code and REDISTRIBUTE it? |
| 19:35.58 | [TK]D-Fender | qakhan: "An Asterisk commercial license provides customers the legal means both to modify Asterisk and to incorporate it into a product, without the obligation of providing the resulting code under the GPLv2 license." ,_ because this is VERY clear |
| 19:36.14 | malcolmd | they may not have understood your question, or perhaps they did. asterisk can be used freely under the terms of the GPLv2. if your usage of Asterisk runs counter to the GPLv2, then it can be licensed, for a fee, from Digium as an alternative. |
| 19:37.24 | qakhan | ok if i integrad asterisk with my app then do i need license? |
| 19:38.24 | malcolmd | if your usage of Asterisk runs counter to the GPLv2, then you cannot use Asterisk under GPLv2. you'd have to instead license it from Digium. |
| 19:39.16 | [TK]D-Fender | [15:35][TK]D-Fenderqakhan: Are you looking to MODIFY the Asterisk source code and REDISTRIBUTE it? |
| 19:40.17 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 19:40.47 | qakhan | no modification in asterisk |
| 19:41.10 | [TK]D-Fender | Then you obviously don't need to buy a license for it |
| 19:41.41 | Chainsaw | The Fender has decided. This decision is binding. |
| 19:42.18 | [TK]D-Fender | gavels |
| 19:42.20 | malcolmd | unless you're not distributing the source code of Asterisk along with your end-application, or if you create code that links to Asterisk that's also not distributed in source-format and also under GPLv2 or less restrictive licenses |
| 19:43.30 | *** join/#asterisk u0m3_ (~u0m3@92.80.75.201) |
| 19:45.48 | qakhan | ok |
| 19:46.45 | Katty | i blame Qwell. |
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| 20:21.30 | DigiDaz | Hi all, I have a problem I'm trying to understand. We have an asterisk PBX that is doing direct media with the carrier via a freeswitch billing box. To compensate for any missing BYEs, etc we have enabled session timers on the freeswitch box. We have what just appears to be a single extension on the asterisk box, a speedtouch 2030 that is causing problems. Whenever the reinvite occurs between the asterisk and freeswitch the call gets hung up. Anyone any |
| 20:21.30 | DigiDaz | ideas? |
| 20:24.09 | *** join/#asterisk navaismo (~navaismo@187-178-254-98.dynamic.axtel.net) |
| 20:25.56 | newtonr | DigiDaz, do you know who hangs up the call? freeswitch, asterisk , the phone? |
| 20:26.33 | newtonr | you say "the call gets hung up". you should be able to determine who is hanging up by looking at a packet capture from the Asterisk and freeswitch systems and then running it down |
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| 20:27.05 | DigiDaz | newtonr: Thanks for replying, this is an old issue I'm just kicking back to life so I cannot remember exactly. |
| 20:27.55 | newtonr | sounds like your first step would be getting packet captures at each point to see what is hanging up first |
| 20:28.29 | DigiDaz | I've replaced the extension with a Yealink already and all is well but it has been haunting me for a while why the session timer between asterisk and freeswitch would be affected by a single extension |
| 20:28.56 | DigiDaz | I did have captures at the time, I wonder if I can dig any out |
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| 20:49.45 | Katty | so. what'd i miss? |
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| 21:20.30 | jormun | I’m having some trouble with getting an extension working on Asterisk 11.10 |
| 21:22.20 | [TK]D-Fender | Show us |
| 21:22.21 | [TK]D-Fender | ~pb |
| 21:22.22 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 21:22.24 | [TK]D-Fender | ^^^ |
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| 21:31.53 | jormun | well ultimately it’s a problem with the contact address being sent to my extension from Asterisk |
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| 21:32.24 | jormun | the extension is receiving a private IP in the contact, when it should be getting a public IP |
| 21:32.44 | [TK]D-Fender | Show us |
| 21:33.05 | jormun | I have the following lines in the [general] secion of sip.conf to set external and local info |
| 21:33.05 | [TK]D-Fender | Next, that is a device comm issue, not an "extension" issue |
| 21:33.07 | jormun | externaddr:54.86.168.171 |
| 21:33.08 | jormun | localnet=172.31.45.98/255.255.255.0 |
| 21:33.09 | [TK]D-Fender | extension = extensions.conf |
| 21:33.27 | [TK]D-Fender | PASTEBIN the actual SIP debug from your call or other comm attempt |
| 21:33.53 | [TK]D-Fender | [17:33]jormunexternaddr:54.86.168.171 <- ":" != "=" |
| 21:33.59 | jormun | welp, I just saw the problem upon pasting it here. |
| 21:34.09 | jormun | and you are right it’s that damned colon! |
| 21:34.53 | jormun | turns red and tries to sneak out quietly |
| 21:46.51 | leifmadsen | i prefer to sneak out loudly |
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| 23:43.50 | chris349 | When I send a call to a SIP peer Asterisk replaces the callerid with the name of the SIP peer, is there any way to instead pass the recieved callerid? |
| 23:46.53 | [TK]D-Fender | Only reason it'd override is if you set "fromuser" |
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