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| 00:34.05 | newmember | good day |
| 00:34.21 | newmember | I have been trouble shooting *1.4 connection to callcentric |
| 00:34.41 | newmember | I keep seeing a 404 error going back to CC |
| 00:35.00 | newmember | my asterisk error is this: NOTICE[3370]: chan_sip.c:15503 handle_request_invite: Call from '1777XXXXXXX' to extension '1777XXXXXXX' rejected because extension not found. |
| 00:35.27 | newmember | so I changed my extn to this: exten => _X!,1,NoOp(New call from New call from Callcentric PDX 1503XXXXXXX) |
| 00:35.51 | newmember | thinking that this would catch everything coming in but still I get the 404 error going back to CC |
| 00:37.52 | newmember | sip.conh has: context=from-trunk and externsions.conf has: [from-trunk] |
| 00:40.56 | newmember | yeah got it |
| 00:41.00 | newmember | gees |
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| 00:43.38 | newmember | well it works with the exten => _X!,1,NoOp filter I need to restrict it a bit |
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| 01:01.17 | ruben23 | guys anyone can help, i got a DID number and upon dialing it.. it get this error and no calls or calls incoming -----> http://pastebin.com/LMQdKZDk |
| 01:02.08 | ruben23 | on my trunkinbound context i got this ----> http://pastebin.com/XbbQNzM5 |
| 01:02.31 | ruben23 | but why its not matching the extensions at all..? |
| 01:05.06 | newtonr | ruben23, I don't think _X. will match 's', but I'm not sure |
| 01:06.08 | newtonr | That is, the call is coming in to extension s and you are trying to match a string starting with 0 to 9 and then one or more characters. |
| 01:06.43 | ruben23 | newtonr: what should i do..? |
| 01:07.46 | newtonr | ruben23, change your pattern match to be an 's', so that it can find the s extension... or make sure the far end actually sends a DID |
| 01:08.27 | newtonr | for now, you might just change _X.,1,AGI to s,1,AGI |
| 01:09.44 | newtonr | I'd help more but I've got to go! |
| 01:09.53 | newtonr | Send a mail to the asterisk-users list if no one is online to help you out. Later! |
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| 01:44.03 | alexw | In voicemail 2 email - the attachment is there but it cannot be played |
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| 06:59.05 | Zogot | ahoyhoy |
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| 08:06.15 | JeroenBo | I am looking for somone with experience with connecting a Panasonic PBX to asterisk (dialplan) |
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| 08:27.59 | Zogot | Ahoyhoy |
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| 08:48.39 | Stefan27 | lsof | grep asterisk | wc -l outputs 50962 for me. is that normal for a 32-bit fedora installation of ast 12.5? with 50 peers or so |
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| 08:50.37 | Stefan27 | i just noticed a webpage for which the user had 357 for the same query, from 2010 though. |
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| 09:28.16 | al_nz1 | evening all |
| 09:28.30 | al_nz1 | I have a piaf server locally which works great |
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| 09:29.22 | al_nz1 | I put a instance in EC2, and modified the config of my SEP<mac>.cnf.xml replacing the LAN IP with the WAN IP of my EC2 (which has ports forwared) but the phone wont register |
| 09:29.35 | al_nz1 | is there a sip debug command to watch registration attempts? |
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| 09:44.13 | JeroenBo | <PROTECTED> |
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| 09:50.06 | al_nz1 | JeroenBo: yeah its UNAUTH |
| 09:50.18 | al_nz1 | just trying to work out why...the pass is right |
| 09:50.29 | Zogot | can you use Asterisk Sorcery for other things than pjsip? like extensions(contexts)? |
| 09:51.09 | Zogot | and if so, is there some kind of reference i could find in the asterisk sources to determine the keys that are needed ? |
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| 10:06.23 | al_nz1 | I do now! |
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| 10:50.50 | dan_j | Hi. I've got a macro that makes the callee press 1 in order to accept the call. However when the callee answers (prior to pressing 1), the ringing sound that the caller hears stops. I believe thats because the early audio is no longer being transmitted. |
| 10:50.58 | dan_j | Any idea how I can keep the ringing sound going? |
| 10:52.19 | file | Zogot, no - only PJSIP really uses it right now |
| 10:52.35 | al_nz1 | anyone here know nigel from the uk? |
| 10:52.47 | al_nz1 | cant remember his nic |
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| 11:16.42 | Zogot | file: ah fair enough, thanks for answering |
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| 11:44.35 | Zogot | there are many more tables defined with alembic for pjsip. mainly ps_contacts, ps_endpoint_id_ips. i imagine they require configuration with sorcery |
| 11:45.15 | Zogot | is there a means to find that out? i register my peer (as defined in the Setting up PJSIP realtime page) but no values are updated in ps_aors for example, is that correct? |
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| 12:08.57 | file | yes, that is correct |
| 12:09.14 | file | the act of registering creates a contact in ps_contacts, it doesn't modify the AOR |
| 12:09.44 | Zogot | ah yep its ok now, not sure if its something i missed, but if i dont supply a timers_min_se above 90 it breaks :p |
| 12:10.26 | file | define breaks |
| 12:10.50 | Zogot | https://gist.github.com/zogot/f11b62fd24313cac1f4f |
| 12:11.06 | Zogot | thats when timers_min_se is 89, when i change it to 91, all works fine |
| 12:11.13 | Zogot | thats when i attempt to make a call, from 102 to 101 |
| 12:11.17 | Zogot | (or vis versa) |
| 12:11.42 | file | ah |
| 12:11.50 | file | file an issue |
| 12:12.53 | file | need to add min/max to that option |
| 12:13.07 | Zogot | its not a problem from flat files? |
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| 12:13.21 | file | it would be a problem there as well |
| 12:13.33 | file | same code is used for it all |
| 12:13.58 | Zogot | aye but none of the examples of pjsip endpoints reference any timers_min_se values |
| 12:14.08 | Zogot | perhaps there was some sort of default value used in flat files or so |
| 12:14.22 | file | there's a default value if it's not specified or present |
| 12:15.44 | Zogot | is it perhaps then just an issue for the realtime database structure (just trying to determine what best to say in the issue ticket) or something that should have a more global fix, that in flat files it also should have a minimum or so :p apologies for stupid questions, rather new to asterisk/sip things |
| 12:15.59 | file | it's an issue everywhere |
| 12:16.05 | file | the code which handles the various fields is common to everything |
| 12:16.11 | Zogot | ah ok |
| 12:16.44 | file | just say that if you put in a timers_min_se below a certain number it crashes like above |
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| 12:38.24 | Tuju | say that i've multiple companies in same phone with different lines, each line is separated in proxy with own context. How do i'm supposed separate them in registration phase? |
| 12:39.00 | Tuju | if i keep the default sip-contaxt/domain, those accounts are not found. |
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| 12:40.41 | Tuju | i cannot use different dns names in proxy dns name, because Cisco dropped support for multiple proxies from c7975 firmware. Previous versions had the support, but now it's gone. |
| 12:41.04 | *** join/#asterisk dipanjan (671b082b@gateway/web/freenode/ip.103.27.8.43) |
| 12:42.32 | dipanjan | Asterisk is unable to detect BYE when it is waiting for a DTMF input. It just stays there for hours. I am using version 1.6 because that is what came with my distro. Can anybody please suggest a solution? |
| 12:45.06 | Zogot | hey file, if you perhaps have a moment to check over the ticket, ensure I supplied the information as best as possible it would be appreciated. (i reviewed the submitting issues page also) https://issues.asterisk.org/jira/browse/ASTERISK-24336 |
| 12:48.32 | file | 'tis fine |
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| 12:50.06 | Tuju | http://pycall.readthedocs.org/en/latest/usage.html |
| 12:50.16 | Tuju | file: saw my question? |
| 12:53.26 | file | yes. |
| 12:53.37 | [TK]D-Fender | dipanjan: Show us. |
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| 12:58.21 | Tuju | file: sip show domains lists DNS-domains and context mappings, when that dns-domain is used? in registration? From proxy dns name (that maps in ip to that particular host?) |
| 12:58.35 | *** join/#asterisk dict (~dict@ul.dolphinict.co.uk) |
| 12:59.08 | file | I would suggest telling the channel what you are trying to do instead of getting into specifics and getting people confused |
| 13:00.41 | dict | Hi there, I'm getting a seg fault in asterisk on a call hangup. The backtrace goes from ast_cdr_detach to post_cdr then cdr_handler. For some reason cdr_handler calls sqlite libs which then cause a coredump. I didn't think sqlite was used unless you took steps to switch to it from MySQL. |
| 13:01.02 | dict | Can anyone suggest what might be calling sqlite to give me an idea of where to look to find the cause? |
| 13:01.31 | Tuju | file: I told above. I've a phone with different lines that maps into different sip context in same asterisk proxy. |
| 13:01.44 | file | Asterisk isn't a proxy |
| 13:01.44 | Tuju | each line is a different company. |
| 13:01.56 | Tuju | okay, i call it then asterisk. |
| 13:02.15 | file | but you shouldn't have to mess with SIP domains and such... each one should just be a different user |
| 13:02.47 | Tuju | in that case asterisk says that it cannot find such line/user name |
| 13:03.31 | file | show the configuration and console output |
| 13:03.52 | Tuju | i assume that incoming registrations get mapped into context, way or another. I i don't create those DNS-name....context mappings, those lines doesn't exist in its viewpoint. |
| 13:04.20 | Tuju | problem is that that damn cisco phone cannot have different proxy for each line button. |
| 13:04.29 | file | I don't understand why you'd need different proxies ... |
| 13:04.32 | Tuju | it has to be the same ip. |
| 13:04.39 | file | unless you want them to all go to different servers |
| 13:04.53 | Tuju | in contrary, there is only one asterisk. |
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| 13:05.33 | Tuju | i could do more dns names that map into same ip. like a.example1.com and a.example2.com |
| 13:06.01 | file | if you are trying to use SIP domains in chan_sip then I don't know that, so I can't help |
| 13:06.11 | Tuju | and those would be same ip address and come into same server, but sip REGISTER packets would have different From whatever strings and each line would get mapped into different context. |
| 13:06.15 | file | people rarely end up using that support |
| 13:06.32 | Tuju | i don't know what chan_sip is. |
| 13:06.58 | Tuju | but someone must have written something about use of domain= in sip.conf, right? |
| 13:07.21 | Tuju | this can't all be folklore around bondfire and "show me a config" right? |
| 13:07.45 | file | I personally have never used the support. There is some documentation in sip.conf.sample, there may be more elsewhere. |
| 13:07.51 | file | The number of people who use the support is small. |
| 13:08.03 | Tuju | yep, i've got the same feeling. |
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| 13:08.43 | Tuju | reminds me times when mysql had their documentation in single web page that was supposed to be for all versions. |
| 13:09.06 | *** join/#asterisk jepperl (3e2c87e7@gateway/web/cgi-irc/kiwiirc.com/ip.62.44.135.231) |
| 13:09.34 | file | Most people who do multi-tenant don't use SIP domains, they just have different SIP usernames and use dialplan contexts to separate things for routing purposes. |
| 13:09.47 | Tuju | i'd like to see somethingl ike http://www.postgresql.org/docs/9.3/interactive/index.html you can click versions and it's always up-to-date. |
| 13:10.18 | Tuju | file: i tried that but those lines don't register because those lines are, reasonably - in different context. |
| 13:10.47 | Tuju | if i don't map context somehow, incoming REGISTER is in 'default' and hence that line doesn't exist for that packet. |
| 13:10.55 | file | er, no |
| 13:10.59 | file | it doesn't work like that |
| 13:11.05 | Tuju | well, how does it? |
| 13:11.24 | jepperl | Hey guys. I'm currently having an issue where sometimes, usually during SIP REGISTER packets are being sent, the asterisk CLI will become unresponsive. By that i mean; i can type anything into the CLI and press enter, and nothing happens. 'jklahsdkjhaskhdsa' or 'reload' or 'sip reload' etc.. nothing.. i am running the cli with 'sudo asterisk -vvvvv |
| 13:11.24 | jepperl | r' btw. Have anyone experienced this? the asterisk version i am running is 11.12.0 |
| 13:11.26 | file | if you aren't using SIP domains then the user portion of the From header is used to find a SIP entry in sip.conf |
| 13:11.48 | file | if authenticate passes then it is registered |
| 13:11.53 | Tuju | file: and all lines are on same 'flat domain' ? |
| 13:11.58 | file | yes. |
| 13:12.08 | Tuju | and then the context gets mapped. right. |
| 13:12.24 | file | I don't understand what you mean by "gets mapped" |
| 13:12.24 | Tuju | then it makes sense to hose whole domain=plaaplaa |
| 13:12.39 | Tuju | file: becomes effective for that line. |
| 13:12.50 | file | sure |
| 13:13.12 | Tuju | then i need to figure out why my lines whine 'not a local domain' whatever. |
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| 13:22.30 | dipanjan | [TK]D-Fender: you want a screenshot? I will need to wait for the next time someone hangs up when waiting for DTMF. |
| 13:22.52 | dipanjan | [TK]D-Fender: I am controlling it through AGI (perl) |
| 13:22.57 | [TK]D-Fender | dipanjan: A screenshot of what? |
| 13:23.40 | dipanjan | [TK]D-Fender: I didn't get what you wanted me to show when you said 'show us' |
| 13:24.04 | [TK]D-Fender | dipanjan: the CALL |
| 13:24.12 | [TK]D-Fender | dipanjan: Prrof of what is happening. |
| 13:26.15 | dipanjan | [TK]D-Fender: this is the line in the perl AGI where it is stuck: $dtmf = $AGI->get_option($promptstring[$j], $components[4], $timeout); |
| 13:26.29 | [TK]D-Fender | dipanjan: 2 more things : AGI's can continue after a hangup. a handler is called and if you don't do things properly your script will just keep running. This is the most likely. |
| 13:27.05 | [TK]D-Fender | dipanjan: Next 1.6 isn't a specific branch, and 1.6.0, 1.6.1, and 1.6.2 are all no longer supported at all. |
| 13:27.38 | [TK]D-Fender | dipanjan: So if there is some sort of issue that is actually Asterisk's fault, your branch is not going to get fixed |
| 13:27.51 | [TK]D-Fender | dipanjan: Time to upgrade |
| 13:27.55 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 13:29.01 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-wefqlgjsiqmyxvim) |
| 13:29.12 | dipanjan | [TK]D-Fender: yes, I know 1.6.2 is ancient. But this is what my distro installed by default (less hassles). I have a callback handler to handle hangups. It works fine except when asterisk is waiting for an input |
| 13:30.44 | dipanjan | [TK]D-Fender: I am quite sure this has been fixed in later versions as there are several bugs reported on this issue. Was just wondering if doing someting different would save me going down the upgrade path, and probably breaking a few things. |
| 13:30.49 | [TK]D-Fender | dipanjan: Your distro is either equally ancient, or is just bad. You should not be running that branch. Even showing us anything won't change anything in the end if it's an Asterisk issue |
| 13:30.55 | [TK]D-Fender | dipanjan: You'll have to upgrade |
| 13:31.33 | dipanjan | [TK]D-Fender: yes, the distro is ancient too. Did not touch it for fear of breaking things. |
| 13:31.55 | dipanjan | Thanks. I will look into how to upgrade safely, without breaking many things. |
| 13:32.01 | [TK]D-Fender | dipanjan: Well things are broken now. Time to move forward |
| 13:36.40 | dipanjan | [TK]D-Fender: good point :D |
| 13:38.03 | *** part/#asterisk dipanjan (671b082b@gateway/web/freenode/ip.103.27.8.43) |
| 13:38.59 | dict | What's the best way to deal with segfaults? I'm reading backtraces but getting the feeling that I am in over my head. I have 7 segfaults, 4 refer to utils.h whilst 3 refer to cdr_handler just before segfaulting. |
| 13:44.27 | Stefan27 | did you add any code yourself, which may have caused the segfault? |
| 13:44.49 | hurdman | hi, i can pass sip video call, but is-it possible to havec video during the ring tone before answer ? |
| 13:45.10 | dict | We've not touched the asterisk code. The box is running Asternic but it is only asterisk that segfaults. |
| 13:45.28 | *** join/#asterisk BeachBall (~eXcAliBuR@206.162.174.6) |
| 13:46.34 | dict | I'm just trying to match the backtraces with what happened immeditately prior. One of the cdr_handler ones happened when somebody logged out of a queue. |
| 13:47.37 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-qtgzrnopufksltov) |
| 13:47.37 | BeachBall | how do i get debugging on? |
| 13:48.00 | [TK]D-Fender | beatthere are over 10 different KINDS of debugging.. care to be more specific? |
| 13:48.08 | [TK]D-Fender | BeachBall: there are over 10 different KINDS of debugging.. care to be more specific? |
| 13:48.13 | BeachBall | sip |
| 13:49.12 | [TK]D-Fender | depends on version |
| 13:49.30 | BeachBall | sip debug on doesn't work |
| 13:49.50 | [TK]D-Fender | sip <tab> will expand CLI's command list... |
| 13:50.01 | BeachBall | oh nice |
| 13:50.04 | BeachBall | i didn't know that trick |
| 13:50.05 | BeachBall | ;D |
| 13:50.13 | BeachBall | ok got it |
| 13:50.15 | BeachBall | set |
| 13:50.17 | [TK]D-Fender | Only been there since... forever-ish |
| 13:50.18 | BeachBall | now |
| 13:50.24 | [TK]D-Fender | Next? |
| 13:50.27 | BeachBall | it's on but i don't see anything |
| 13:50.36 | BeachBall | verbose... |
| 13:50.38 | BeachBall | i can do that |
| 13:51.33 | BeachBall | that tab thing is great |
| 13:53.03 | BeachBall | i still don't see debugging info |
| 13:53.21 | BeachBall | i have core set debug 10 |
| 13:53.25 | BeachBall | sip set debug on |
| 13:53.29 | [TK]D-Fender | You seem to be operating on the assumption that something is even making it to your server in the first place. |
| 13:53.30 | BeachBall | core set verbose 10 |
| 13:53.32 | [TK]D-Fender | ^ |
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| 13:58.02 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 14:03.46 | hurdman | can someone help me on early video media with sip ? |
| 14:05.20 | Katty | hello my asterisk does not work at all how to fix pls |
| 14:05.57 | newtonr | hurdman, Maybe! ask a specific question, and then pastebin some debug to demonstrate the issue. |
| 14:07.11 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 14:09.22 | rrittgarn | Katty: Please try turning it off and then back on again. If that doesn't work perhaps an early lunch break will be helpful |
| 14:11.04 | Katty | lunch break is always helpful! |
| 14:11.35 | Katty | pokes drmessano |
| 14:11.47 | hurdman | newtonr: i have got video during my sip call, but i do not have any video before the answer , i can't see any error anywhere |
| 14:12.59 | jepperl | katty: maybe the power coupling has been polarized? |
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| 14:15.53 | newtonr | hurdman, I've not messed with early media as video, but perhaps you need to call the progress app somewhere. https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application |
| 14:16.28 | newtonr | hurdman, you might also look at a pcap to verify the video is being sent and arriving where you want it to go. |
| 14:17.24 | Katty | jepperl: i'll polarize your power coupling in a minute. |
| 14:17.24 | hurdman | newtonr: i have tried progess :/ and my video stream is visible into wireshark |
| 14:18.29 | newtonr | Is the receiving client capable of early media? |
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| 14:20.01 | hurdman | linphone should, yes |
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| 14:21.16 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:23.15 | newtonr | hurdman, post on the asterisk-video list with a link to your pcap, asterisk debug and the dialplan you are using. Someone will likely be able to help you figure it out. Maybe you are not calling Progress at the right part of your dialplan. |
| 14:25.37 | dict | Right, I've gone through all my logs and asterisk is core dumping each time on a call hangup. It is core dumping right at the end of the hangup process. In all but one case it happens before app_mixmonitor.c kicks in as there are no logs relating to it. |
| 14:26.05 | dict | One of the logs looks like app_mixmonitor.c did all of its work before asterisk bailed out. |
| 14:26.10 | dict | annoyingly incosistent. |
| 14:26.18 | dict | *inconsistent |
| 14:26.41 | hurdman | newtonr: ok thx for your advice. i'll do that |
| 14:27.31 | newtonr | dict, file a bug! https://issues.asterisk.org/jira |
| 14:27.43 | *** part/#asterisk hurdman (~ygcheny@c3p0.r0b0t.fr) |
| 14:28.13 | newtonr | with a backtrace! https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 14:28.30 | dict | I have 7 backtraces that are all slightly different :/ |
| 14:29.32 | newtonr | compiled with DONT_OPTIMIZE and BETTER_BACKTRACES ? |
| 14:29.58 | newtonr | attach'em all |
| 14:30.07 | newtonr | (to the issue) |
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| 14:31.56 | dict | I believe DONT_OMPTIMIZE is enabled on the distro. Not sure about BETTER_BACKTRACES |
| 14:33.01 | dict | Hmm, just found this which sounds like half of my backtraces |
| 14:33.02 | dict | http://pbxinaflash.com/community/index.php?threads/sql-lite-causing-asterisk-to-crash.14134/ |
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| 14:36.40 | dict | Found the .ini for sqlite and I'll turn those off. If it turns out to solve the problem then it might be a distro issue rather than asterisk itself. |
| 14:38.14 | *** join/#asterisk michael_work (~michael-w@bzq-82-168-31-134.red.bezeqint.net) |
| 14:40.03 | michael_work | there is this function SHARED that can be usefull, but the question is how do i clean the data after i finsihed to use it |
| 14:40.50 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:41.39 | michael_work | i mean no to set variable to nothing as set(shared(foo,channel)=) but to actually delete it as there would be no need after i completed certain operation |
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| 14:43.22 | mjordan | michael_work: if nothing needs the shared channel variable - that is, all channels that cared about it have gone away - Asterisk will tidy it up. |
| 14:43.24 | marceloamorim | guys, morning, I`m trying to build my own table on mysql, I used func_odbc, but now, I`m geting information about the cel ( sql ) on the cli |
| 14:43.37 | marceloamorim | like > [INSERT INTO cel (eventtype,eventtime,userdeftype,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,peeraccount,uniqueid,linkedid,userfield,peer) VALUES ('CHAN_START',{ts '2014-09-17 11:42:18'},'','','','','','','s','ramais','SIP/3000-00000004','','',3,'','','1410964938.4','1410964938.3','','')] |
| 14:44.40 | michael_work | mjordan, thanks |
| 14:45.56 | Tuju | i'm wondering where cisco phones get their From: domain |
| 14:46.20 | michael_work | so to get sip cause from originated channel thru AMI i had to use b() in dial to set hanguphandler to set sip cause from there to shared variables so i can get it after from h in original originated channel |
| 14:46.23 | michael_work | \o/ |
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| 14:48.22 | michael_work | it might be related to https://issues.asterisk.org/jira/browse/ASTERISK-22042 |
| 14:48.25 | michael_work | but not sure |
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| 15:10.50 | marceloamorim | there is any way to enable the func_odbc and disable all messages(mysql) from the other features? |
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| 15:21.06 | newtonr | marceloamorim, I don't believe so. You can just turn up or down verbose or debug. I can't remember which the SQL messages are on. |
| 15:22.08 | marceloamorim | yeah, I`m try to use app_mysql now |
| 15:24.11 | marceloamorim | maybe is better, looks deprecated, but I want to use cel at the way it is and use another table for other things. |
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| 15:52.36 | marceloamorim | weird, after I removed the func_odbc.conf from the /etc/asterisk and remove the module, the cel mysql log keeping appear at the CLI, and I restart the asterisk already |
| 15:52.54 | marceloamorim | before I used for the first time, I didn`t have those verboses |
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| 16:11.41 | Tuju | i can't get my trunk match into line settings between two asteriskskskssks. |
| 16:16.05 | lvlinux | Tuju: what exactly do u mean? |
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| 16:18.12 | Tuju | i can see that asterisk A sends OPTIONS packet that is for From: a-trunk@172.16.1.1 and i've domain=172.16.1.1 listed, but it still says "404 not found" |
| 16:18.33 | Tuju | that domain setting is in asterisk B |
| 16:18.52 | Tuju | and asterisk B has line set with: [a-trunk] |
| 16:19.13 | Tuju | and i've another box where it works just fine. |
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| 16:22.07 | lvlinux | do u have a pb of ur sip.conf? |
| 16:22.19 | lvlinux | err both of them rather |
| 16:23.21 | [TK]D-Fender | Tuju: Nothing tells me the reason for the 404 there. You seem to assume it's auth-based. |
| 16:23.28 | [TK]D-Fender | Tuju: But I'm not seeing proof |
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| 16:26.20 | Tuju | i'm now trying to compare to working one and actually there are those 404 not founds too, apparently for that OPTIONS packet |
| 16:26.28 | Tuju | but it still registers. |
| 16:26.54 | [TK]D-Fender | 404 is not necessarily an AUTH failure |
| 16:27.02 | [TK]D-Fender | and you aren't showing anything useful |
| 16:37.40 | cunningpike | [TK]D-Fender: Still fighting the good fight, I see... :-) |
| 16:38.39 | [TK]D-Fender | cunningpike: Another likely exercise in futility |
| 16:40.34 | cunningpike | Indubitably |
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| 16:54.46 | Tuju | yeah, now i found the problem. those packets gets dropped somewhere on the way. |
| 16:55.09 | Tuju | i can see them leaving from my dmz firewall interface, but don't see them coming in another end. |
| 16:55.13 | [TK]D-Fender | 404 = answer, not a drop |
| 16:55.32 | Tuju | ack |
| 16:56.32 | BeachBall | ok, so I have no audio coming from my asterisk... where do i start |
| 16:56.40 | BeachBall | calls are going through ok |
| 16:56.44 | BeachBall | and it use to work |
| 16:56.55 | BeachBall | nothing was touched, did have a power outage |
| 16:57.01 | [TK]D-Fender | Look at your calls |
| 16:57.23 | BeachBall | i have debugging on |
| 16:57.24 | BeachBall | :} |
| 16:58.13 | BeachBall | [TK]D-Fender: i need more details |
| 16:58.25 | [TK]D-Fender | Go get them |
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| 17:00.11 | BeachBall | gotta dumb it down a bit more |
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| 17:07.28 | BeachBall | i did a playback of hello-world, and couldn't hear it |
| 17:07.57 | BeachBall | could something have fried when the power went out? |
| 17:10.16 | [TK]D-Fender | Yes, "something" could have fried. What could have fried that actually has anything to do with this is another matter. |
| 17:10.34 | BeachBall | so where do I start? |
| 17:10.45 | BeachBall | pastebin my sip.conf and extensions.conf? |
| 17:11.14 | BeachBall | 11.0.0 |
| 17:11.17 | BeachBall | 11.9.0 |
| 17:11.19 | BeachBall | is the version |
| 17:11.27 | [TK]D-Fender | Configs are like giving us a showrrom photo of your car when it was new and asking why it crashed |
| 17:11.38 | BeachBall | right |
| 17:11.47 | BeachBall | so i update my asterisk to 11.12 ? |
| 17:11.53 | BeachBall | and hope for the best |
| 17:11.57 | [TK]D-Fender | [12:56][TK]D-FenderLook at your calls |
| 17:12.08 | BeachBall | that doesn't mean anything to me |
| 17:12.12 | [TK]D-Fender | cunningpike: The adventure continues... |
| 17:13.05 | BeachBall | <PROTECTED> |
| 17:13.08 | BeachBall | i don't hear it |
| 17:13.13 | BeachBall | the system is playing it |
| 17:13.21 | BeachBall | i have the volume up |
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| 17:21.30 | BeachBall | i'm an idiot, it's not my fault |
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| 17:42.10 | BeachBall | right |
| 17:42.16 | BeachBall | installed 11.12 and still no sound |
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| 18:04.07 | anonymouz666 | mjordan: I just confirmed that this message is due (s)RTCP being dropped - Got SRTP from 192.168.200.17:54613 - failed to decrypt information. |
| 18:04.49 | anonymouz666 | it parses rtcp attribute, increases one port, and send the rtcp there. |
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| 18:13.40 | mjordan | Hm. Is the SDP that is being offered from the phone using the 'rtcp' attribute and incrementing the RTCP port by something more than 1? |
| 18:17.15 | anonymouz666 | in the example the rtp was sent to 192.168.200.17:54612 |
| 18:17.24 | anonymouz666 | rtcp 192.168.200.17:54613 |
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| 18:17.39 | mjordan | not quite what I asked :-) |
| 18:17.46 | mjordan | Generally, Asterisk will sent RTCP to the negotiated RTP port + 1 |
| 18:18.02 | mjordan | there, is however, an 'rtcp' attribute that can be sent in an SDP that will specify a port other than the 'default' |
| 18:18.15 | mjordan | did the phone use that attribute, and did it specify something other than the negotiated RTP port + 1? |
| 18:19.27 | anonymouz666 | no, it was RTP port + 1. |
| 18:19.37 | anonymouz666 | rtcp= in SDP |
| 18:19.45 | mjordan | k. That would be expected then. |
| 18:20.00 | mjordan | Sounds like something didn't apply the crypto policy to the RTCP stream. |
| 18:20.27 | anonymouz666 | any suggestions? |
| 18:20.48 | anonymouz666 | on where to inspect |
| 18:24.01 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 18:25.12 | mjordan | IIRC, you're running some of Olle's branches with this, right? |
| 18:25.41 | anonymouz666 | tried, teapot-1.8, rtcp-pinefrog-11, asterisk-11.12 with lifetime patch |
| 18:25.46 | anonymouz666 | I am open to test anything |
| 18:26.35 | mjordan | well, you're running modified Asterisk. I think at this point, you may want to see if Olle is running into anything similar with his work. |
| 18:26.51 | mjordan | If he isn't seeing these same issues, it may be worthwhile working with him to find out what the difference is. |
| 18:27.03 | mjordan | Otherwise, we really need to have the issue reproduced in something that is unmodified |
| 18:29.19 | anonymouz666 | I unnderstand, gonna try to get in touch if him you he appears online |
| 18:29.33 | anonymouz666 | with wim when |
| 18:31.46 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 18:33.49 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:36.21 | *** join/#asterisk areski (~areski@80.174.128.93.dyn.user.ono.com) |
| 18:40.13 | BeachBall | still has no audio |
| 18:40.17 | BeachBall | and no one is helpingme |
| 18:40.18 | BeachBall | >:( |
| 18:40.46 | [TK]D-Fender | You've provided no useful detail and aren't looking at the call. |
| 18:41.07 | BeachBall | your not telling me what "looking at the call" means |
| 18:41.09 | Chainsaw | Listen to the Fender. It speaks truth. Not politely, but truth all the same. |
| 18:41.45 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-kmcbxmtmcfxfothz) |
| 18:43.09 | BeachBall | http://pastebin.com/aiSqP58s |
| 18:43.48 | BeachBall | does that help? |
| 18:45.34 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 18:48.16 | [TK]D-Fender | That isn't from the start of the call. |
| 18:48.43 | [TK]D-Fender | nor is that SIP DEBUG which is what you asked how to get earlier ... but don't seem to think is important. |
| 18:48.57 | [TK]D-Fender | You still haven't given us details of what's in play here |
| 18:49.52 | BeachBall | http://pastebin.com/U1uebESV |
| 18:49.54 | BeachBall | hows that |
| 18:58.21 | [TK]D-Fender | I'm seeing multiple IP's involved in this call and you are not providing DETAILS about what is in play with your server |
| 18:58.33 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 19:02.36 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 19:06.39 | *** join/#asterisk theron (~theron@173.252.71.2) |
| 19:11.36 | *** join/#asterisk aross42 (~aross@h46-190.reznet.ucalgary.ca) |
| 19:12.56 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:15.27 | BeachBall | phone connected to server all internal. dialing an extension that plays hello world |
| 19:17.13 | [TK]D-Fender | <--- SIP read from TCP:10.75.2.78:36469 ---> <- Polycom coming from |
| 19:17.15 | [TK]D-Fender | INVITE sip:303@10.75.2.190:5060;user=phone;transport=tcp SIP/2.0 <--- calling your server |
| 19:18.26 | [TK]D-Fender | o=root 1243169667 1243169667 IN IP4 206.162.174.18 <-- handing your phone a PUBLIC IP for audio |
| 19:18.34 | [TK]D-Fender | Because you didn't set up your LOCALNETS right |
| 19:18.43 | [TK]D-Fender | Fix your basic NAT and network settings |
| 19:21.10 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 19:23.55 | BeachBall | i c |
| 19:24.13 | [TK]D-Fender | <--- Reliably Transmitting (NAT) to 10.75.2.78:36469 ---> <-- * was also told to believe that your phone was behind NAT |
| 19:28.17 | BeachBall | all better |
| 19:28.19 | BeachBall | :) |
| 19:28.22 | BeachBall | thanks [TK]D-Fender |
| 19:28.27 | BeachBall | you the best |
| 19:28.40 | BeachBall | localnet was set to server's ip |
| 19:28.41 | BeachBall | :( |
| 19:28.47 | BeachBall | instead of subnet |
| 19:28.54 | BeachBall | bounces out |
| 19:28.56 | *** part/#asterisk BeachBall (~eXcAliBuR@206.162.174.6) |
| 19:29.49 | marceloamorim | guys, could you explain a little better how should I get the MYSQL(Fetch)? |
| 19:30.27 | marceloamorim | I used the mysql connect and I query the conection, but when I use the fetchid I don`t know how to get a specific result from there |
| 19:32.03 | *** join/#asterisk marlinc (~marlinc@2001:981:f1c6:1:44da:5ab0:4a6c:a67) |
| 19:36.00 | *** join/#asterisk dhardison (uid20219@gateway/web/irccloud.com/x-uxrhbrjhcoahfqxn) |
| 19:41.51 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 19:52.25 | marceloamorim | like, MYSQL(Connect connid 127.0.0.1 dbuser dbpass dbname) , MYSQL(Query resultid ${connid} SELECT 'eventtype' from cel_custom), MYSQL(Fetch fetchid ${resultid} eventtype) should I get the word on this eventtype? |
| 19:52.53 | marceloamorim | word I mean the result |
| 19:54.49 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 20:05.44 | marceloamorim | the result from there query is 1 |
| 20:06.19 | marceloamorim | this is my first time using my own table using the asterisk, I didn`t find anything so far |
| 20:13.39 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:22.53 | marceloamorim | I got it, the Fetch works putting the result on that variables |
| 20:24.00 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:32.07 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 20:34.45 | marceloamorim | nice |
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| 20:41.16 | *** join/#asterisk Magus` (cgrady@ip70-173-121-137.lv.lv.cox.net) |
| 20:43.18 | Magus` | hi, having an odd issue with asterisk here in my office, and I'm not finding anything about an issue like this on Google... every few days/weeks, with no rhyme or reason, every extension will suddenly ring but show the call coming as from itself |
| 20:43.40 | Magus` | has anyone seen an issue like this, or perhaps know what to even search for? |
| 20:48.23 | drmessano | Magus`, sounds like a SIP attack |
| 20:48.33 | drmessano | Do you have 5060 open to the public internet? |
| 20:48.41 | Magus` | would that cause an extension to ring itself? |
| 20:48.48 | Magus` | *could, I guess |
| 20:48.57 | Magus` | not sure, let me ask |
| 20:49.13 | drmessano | Simple...The calls hit Asterisk as to <ext> from <same ext> |
| 20:49.22 | drmessano | So the probes look like they're coming from the same device |
| 20:49.41 | Magus` | I see |
| 20:51.36 | Magus` | I am not able to connect to that port from outside, so it doesn't appear to be open |
| 20:52.40 | Magus` | still waiting for a response for final word; I'm not directly in charge of anything, I'm just tired of this happening so much and wanted to see if I could get ideas to pass along :) |
| 20:53.28 | Magus` | hm, unless we just use a non-standard port.. I'm getting a possible yes from the network guys |
| 21:00.53 | Magus` | and of course, the one directly in charge of the box is out to lunch... guess I'll have to resume investigating later :) thanks for the info so far :) |
| 21:13.34 | drmessano | Connect how? |
| 21:17.50 | Magus` | I had tried telnet since I couldn't get ahold of the right person |
| 21:17.51 | Nugget | telnet is eeeeeeevil! |
| 21:17.57 | Magus` | heh |
| 21:18.05 | eirirs_ | telnets Nugget |
| 21:18.10 | Magus` | waiting for him to return and then I'll find out for sure |
| 21:20.56 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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