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| 01:32.52 | swimmercol | anyone working on PJSIP project? |
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| 04:55.15 | KNERD | swimmercol: the PJSIP devs are ;-) |
| 04:55.48 | swimmercol | :) |
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| 08:56.17 | jkroon | hi all, i'm reasonably sure this has been asked a million times before. I'm looking for the SIP response code directly after Dial(), according to http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause I should be able to use ${HASH(SIP_CAUSE,${CDR(dstchannel)})} but that just gets me the empty string - alternative suggestions? |
| 09:09.06 | Zogot | jkroon: after a dial, the variable DIALSTATUS is set |
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| 09:09.15 | Zogot | oh, sory misread |
| 09:09.44 | Zogot | jkroon: the question then becomes, why the sip code? |
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| 11:02.25 | louisdk | When you want to use Asterisk do you then need an external SIP service or can Asterisk manage that too? |
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| 11:35.20 | seik0 | hi! I have a problem with chan_extra (for gsm): if there is outgoing call then incoming on same interface, then incoming call replaces outgoing |
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| 11:37.54 | drathir | hi... |
| 11:38.30 | seik0 | in other words, Chan1 -> Gsm(outgoing) --> Chan2..Then Chan3 --> Gsm (incoming) , Gsm(incoming) is congured just to Hangup() all incoming calls. Instead Chan3 becomes connected with Chan1 and Chan2 is executed Hangup() |
| 11:38.38 | drathir | guys wonder of any alternatives for ipkall? |
| 11:39.50 | drathir | its not totally ot but pm is fine too... |
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| 11:42.19 | seik0 | Ok, lets got from other side. What does mean "overlap call" & |
| 11:42.21 | seik0 | *? |
| 11:44.34 | drathir | seik0: conferencial or waitin call ? |
| 11:45.06 | drathir | guessing... |
| 11:45.18 | seik0 | drathir: threewaycalling also here? |
| 11:45.35 | seik0 | It was my guessing too to get rid of a problem |
| 11:45.57 | seik0 | Tried to disable just everything connected with any conference-like options, but no luck |
| 11:46.48 | seik0 | drathir: with chan_extra all incoming calls are loggin to cli as " Accepting overlap call from ... " |
| 11:47.52 | seik0 | In fact, it is possible, that problem is somewhere else |
| 11:59.21 | seik0 | disabling callwating on operator side helped |
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| 11:59.42 | seik0 | looks like chan_extra misunderstand second gsm line |
| 12:02.13 | seik0 | looks like I need further inverstigation for whether it is old chan_extra or else |
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| 12:09.56 | louisdk | does Asterisk include a SIP server or do I need to install an external vSIP server like Kamailio? |
| 12:11.58 | Gugge | both asterisk and kamailio allow registrations and calls through sip |
| 12:12.07 | Gugge | i would call that a sip server |
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| 12:14.40 | louisdk | Gugge, do you have a good howto or step-by-step guide. For some reason I've trouble setting up SIP via Asterisk. |
| 12:14.58 | louisdk | I'm using AsteriskNow. |
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| 12:44.14 | [TK]D-Fender | louisdk, Then you'll have to go to #freepbx for support with it. |
| 12:44.46 | [TK]D-Fender | louisdk, This channel is not meant for 3rd party GUI support. |
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| 13:09.05 | drathir | wonder if and why debian asterisk have in config default user enabled? And if there any guide of secure default configuration? |
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| 13:13.56 | [TK]D-Fender | What is "default user"? |
| 13:19.22 | drathir | [TK]D-Fender: default asterisk user config there are uncommented fields like fullname = New User | userbase = 6000 etc... is that secure? |
| 13:22.05 | drathir | [TK]D-Fender: i wonder if the default asterisk config shipped with debian is fine to open at world is secured to remote access/messup by default? |
| 13:22.07 | jkroon | Zogot, sorry, had to urgently run off, the reason for the SIP code is because 404 and 604 maps to the same HANGUPCAUSE but the provider we're using are using them differently, specifically, 404 indicates that the destination number doesn't exist, whereas 604 indicates that they can't find the callerid we're passing them. |
| 13:22.41 | [TK]D-Fender | drathir, That sounds like you are looking at SAMPLE configs. Those are junk |
| 13:23.07 | [TK]D-Fender | drathir, That is never supposed to be used. They are there to show you some concepts to see them in full. They are not meant to be used |
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| 13:29.12 | drathir | [TK]D-Fender: some sort of debian have files in /etc/asterisk/ and wonder if that configuration is seccure when server is open at world if dont wonder why files not named in scheme .config_sample with inform user to adjust file before use? ofc no offence any kind... only try to understand the way how things are maded workin... |
| 13:29.54 | drathir | [TK]D-Fender: or place files in /etc/asterisk/samples/ ? |
| 13:30.37 | [TK]D-Fender | drathir, If you didn't make the configs.. then they are just SAMPLES. |
| 13:30.55 | [TK]D-Fender | drathir, And you should copy them off and start building your own |
| 13:31.15 | [TK]D-Fender | drathir, basic * shouldn't have anything in sip or extensions that should be considered usable as-is |
| 13:31.26 | [TK]D-Fender | drathir, they are there for learning syntax, etc |
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| 13:32.04 | drathir | [TK]D-Fender: there is somewhere samples of bare minimum files which ones asterisk will workin? |
| 13:33.24 | [TK]D-Fender | drathir, The ones you really need to make yourself are the channel driver configs (sip, pjsip, iax, etc), manager.conf (if needed) and your dialplan (extensions.conf) |
| 13:33.49 | [TK]D-Fender | drathir, the other sample configs are usually a decent starting point for you to get rid of the sample stuff from slowly as you build your own |
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| 13:35.35 | drathir | [TK]D-Fender: oh i see... i will try to implementing them step by step, thanks a lot for really useful starting tips... |
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| 13:35.54 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:39.00 | ccc_ | the different versions of each version (i mean for example 1.8.x) are only bugfixes right? new characteristics are only added only in the yearly releases like 1.8, 10,11,12,13 |
| 13:39.57 | ccc_ | so if i want to upgrade say from 1.4 to 1.8, i should use the latest version of 1.8, which is 1.8.32.3 i think, right? |
| 13:40.12 | [TK]D-Fender | ccc_, 1.8 is EOL. Dead. Forget about it. |
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| 13:41.08 | [TK]D-Fender | ccc_, Only brances you should consider are 11 (I wouldn't), or 13. 13 is LTS and does actually get new functionality in decimal releases. |
| 13:41.59 | ccc_ | i'm afraid i would need to change too many things in my installation to switch to any of those branches but thanks |
| 13:42.39 | ccc_ | regarding my question, i take it that i should use the latest 1.8, that no new changes were made inside that branch? |
| 13:42.45 | ccc_ | pardon my bad english |
| 13:42.53 | [TK]D-Fender | ccc_, like what? |
| 13:43.34 | ccc_ | like support for new codecs, changes in the way functions or applications work, etc |
| 13:43.39 | ccc_ | anything but bugfixes |
| 13:44.07 | [TK]D-Fender | ccc_, What specifically changed that should be difficultt to adapt? |
| 13:44.36 | [TK]D-Fender | ccc_, There are no bug OR security fixes for 1.8 anymore. It is completely dead. |
| 13:44.56 | [TK]D-Fender | ccc_, Seriously get yourself on something supported.... |
| 13:45.30 | ccc_ | thanks for your help anyway |
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| 13:50.20 | drathir | [TK]D-Fender: there is official debian repository of 13. for jessie? |
| 13:50.34 | jkroon | Usage of SIP_CAUSE is deprecated. Please use HANGUPCAUSE instead. ... how do I distinguish between a 404 and a 604 response using HANGUPCAUSE for which both map to 1? |
| 13:51.06 | [TK]D-Fender | There is no official repo for Debian-based at all last I checked |
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| 13:53.49 | drathir | [TK]D-Fender: me wonder if Asterisk 11.13.1~dfsg-2+b1, is "not reccomended" how is the best to automate keep up 2 date with asterisk reccomendations? |
| 13:54.43 | drathir | bc probably that depend how often asterisk is updated and need intervention (rebuild etc.)... |
| 13:56.51 | file | as a project we aim (aim - not guarantee) to release every 6 weeks |
| 13:58.00 | [TK]D-Fender | Not sure on Debian |
| 13:58.11 | file | yeah, that's outside of us |
| 14:00.43 | jkroon | Zogot, so it seems HANGUPCAUSE() has turned into a function which should contain what I need ... if I can get it to work. |
| 14:03.19 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 14:04.44 | drathir | but there is planned or implemented some kind of os independent from source auto build&update system? or normal source like user managed only? |
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| 14:05.15 | [TK]D-Fender | file, Any plans on official debian repo? |
| 14:05.33 | file | it'd be nice, but nothing being worked on currently |
| 14:05.39 | [TK]D-Fender | There you go |
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| 14:08.36 | lgaetz | I have been tasked with finding out how to set the T.38 parameter, T38FaxMaxBuffer. If such a parameter exists, I assume it would be in udptl.conf, but my googling is not revealing any mention of it. Is there a similarly named parmaeter in Asterisk |
| 14:08.55 | drathir | know debian packaging is a nightmare that thinks about some kind of scripted git source fetch + build method in theory probably easier to implement... |
| 14:09.36 | drathir | [NC]: file thanks a lot for confirmation i guess go for now with by hand source build instead of os ones... |
| 14:09.45 | drathir | [NC]: err |
| 14:09.51 | drathir | [TK]D-Fender: ^ |
| 14:10.53 | *** join/#asterisk jonah (~jonah@5751acb3.skybroadband.com) |
| 14:11.47 | drathir | and still in theory would be easier to switch into repository update model when will become available... |
| 14:15.03 | dasjoe | Is there a linter for asterisk configs available? |
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| 14:49.10 | jkroon | https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause - is HANGUPCAUSE() and HANGUPCAUSE_KEYS() only available once the channel on which it has executed has terminated? I need to make a decision after Dial() as to the SIP response code as to whether to retry, and it is turning out to be rather difficult to get that done. |
| 15:06.04 | [TK]D-Fender | dasjoe, a what? |
| 15:08.38 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 15:09.07 | jkroon | [TK]D-Fender, not sure if maybe you could help me? Or at least point me in some direction or possibly advise on what information would help to troubleshoot? I'm busy reading the source code and honestly am struggling to follow what's going on in the internals, what the owner of the hangup etc is ... so SIP adds the cause to p->owner - which I'm guessing is the outbound channel? surely we want to set the cause on the peer channel |
| 15:09.07 | jkroon | (initiating)? |
| 15:09.25 | jkroon | or I'm missing something completely. |
| 15:09.58 | WIMPy | Hangupcauses are usually inherited in both directions. |
| 15:10.08 | WIMPy | Not even sure if there's maybe only one. |
| 15:10.19 | jkroon | ok, let me PB what I've got ... |
| 15:13.39 | jkroon | http://pastie.org/10793630 |
| 15:14.13 | jkroon | note the output at lines 33 and 40. |
| 15:15.01 | WIMPy | IIRC you need to sipply a channel name to HANGUPCAUSE_KEYS. |
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| 15:17.09 | jkroon | you can't, it doesn't take parameters. |
| 15:17.46 | WIMPy | Oh no, that IS the list of channels. |
| 15:18.35 | jkroon | that's how I understood it as well. so basically you split that into channel names and then use HANGUPCAUSE(${other_channel_name}) to get the cause codes. |
| 15:20.50 | *** join/#asterisk hfb (~hfb@96.229.100.56) |
| 15:24.20 | jkroon | ast_channel_hangupcause_hash_set(struct ast_channel *chan, const struct ast_control_pvt_cause_code *cause_code, int datalen) - I'm assuming chan should be the calling channel? |
| 15:24.30 | jkroon | in my case the IAX/ one? |
| 15:24.58 | jkroon | can someone confirm that p->owner is the right channel for that? |
| 15:27.58 | *** join/#asterisk mrmccrac (~Adium@2001-18e8-2-1009-f000-1fb.dhcp6-bl.indiana.edu) |
| 15:28.28 | mrmccrac | pasted this last week but got distracted w/ other things: http://pastebin.com/pjRv03R8 |
| 15:29.02 | mrmccrac | does the first error about it failing have to do with it not finding an appropriate config entry in extensions.conf ? |
| 15:29.16 | mrmccrac | "failed so falling back to exten 's'" |
| 15:30.13 | jkroon | your context for the peer is probably not set the way you think it is. |
| 15:30.33 | jkroon | the call is routing to 5555@default - which is most likely not what you actually want. |
| 15:30.53 | mrmccrac | ill paste an example call file im using |
| 15:31.36 | mrmccrac | http://pastebin.com/raw/fMfMaM2c |
| 15:31.59 | mrmccrac | and my (security sanitized) sip.conf |
| 15:32.29 | [TK]D-Fender | You have no context in your call file |
| 15:33.05 | [TK]D-Fender | and we leftt off where I asked you to prove where and how you were creating the call file in the first place |
| 15:33.15 | [TK]D-Fender | -- Attempting call on SIP/cic/15555555555 for 5555555555@:1 (Retry 1) <--- clearly not ther |
| 15:33.22 | [TK]D-Fender | @ nowhere |
| 15:34.20 | [TK]D-Fender | (or Originate) |
| 15:34.37 | mrmccrac | sip.conf: http://pastebin.com/raw/NvUNSZNL |
| 15:34.47 | mrmccrac | alright ill look at extensions |
| 15:35.10 | jkroon | mrmccrac, on which side is the error? |
| 15:35.34 | jkroon | the error looks like an incoming call coming from your provider. |
| 15:35.45 | [TK]D-Fender | This isn't an "extensions" problem |
| 15:35.49 | jkroon | that said - all the places I can think of to set a context looks like it's set. |
| 15:35.51 | [TK]D-Fender | Your call is wrong from the start |
| 15:35.55 | [TK]D-Fender | You did not specify the context |
| 15:36.12 | [TK]D-Fender | <[TK]D-Fender> and we leftt off where I asked you to prove where and how you were creating the call file in the first place <-- |
| 15:36.20 | mrmccrac | im seeing that error on my side where im trying to make a call |
| 15:36.40 | mrmccrac | im manually crafting the call file myself and placing it in outgoing spool directory |
| 15:36.54 | jkroon | [TK]D-Fender, http://pastebin.com/raw/fMfMaM2c - there is a Context: line ther. |
| 15:37.27 | jkroon | mrmccrac, humour me - is there a newline character on the end of that call file? |
| 15:37.42 | mrmccrac | yes |
| 15:37.50 | mrmccrac | er |
| 15:37.51 | [TK]D-Fender | You are missing a priority as well |
| 15:37.54 | mrmccrac | yes |
| 15:38.02 | [TK]D-Fender | And I asked how itt got to the spool folder |
| 15:38.12 | mrmccrac | # cp /tmp/place_call.call /var/spool/asterisk/outgoing/ |
| 15:38.15 | [TK]D-Fender | NO |
| 15:38.16 | [TK]D-Fender | BAD |
| 15:38.18 | mrmccrac | while im testing |
| 15:38.26 | [TK]D-Fender | You CANNOT just copy the file |
| 15:38.32 | jkroon | mrmccrac, no, no no. it's got got to me a mv from another location in the same filesystem. |
| 15:38.32 | [TK]D-Fender | you have to MOVE it |
| 15:38.42 | jkroon | asterisk is reading the file before the content got properly written. |
| 15:38.48 | mrmccrac | interesting |
| 15:38.54 | [TK]D-Fender | mv from within the same filesystem as the spool folder |
| 15:39.57 | jkroon | mrmccrac, try this: cp /tmp/place_call.call /var/spool/asterisk/tmp/place_call.call && mv /var/spool/asterisk/tmp/place_call.call /var/spool/asterisk/outgoing/ |
| 15:39.57 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-xbyraetauecvtity) |
| 15:39.57 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:40.00 | mrmccrac | alright did a mv but same error |
| 15:40.33 | jkroon | mrmccrac, /tmp and /var/spool/asterisk/ is not on the same filesystem on at least my system. use my command from above. |
| 15:40.41 | [TK]D-Fender | mrmccrac, Show us |
| 15:40.52 | [TK]D-Fender | And prove it |
| 15:41.04 | [TK]D-Fender | make the file in the folder JUST BELOW and mv it yourself |
| 15:41.30 | *** join/#asterisk linuxfool (~james@DHCP-149-248.resnet.ua.edu) |
| 15:41.34 | mrmccrac | # cp /tmp/place_call.call.bak /var/spool/asterisk/tmp/place_call.call && mv /var/spool/asterisk/tmp/place_call.call /var/spool/asterisk/outgoing/ |
| 15:42.07 | jkroon | and the CLI output? |
| 15:42.20 | mrmccrac | ill paste |
| 15:43.10 | mrmccrac | http://pastebin.com/raw/xVVRGNd1 |
| 15:44.24 | [TK]D-Fender | <PROTECTED> |
| 15:44.27 | [TK]D-Fender | Clearly NOT the same |
| 15:44.34 | [TK]D-Fender | it is now getting the context you specified |
| 15:44.47 | [TK]D-Fender | <PROTECTED> |
| 15:44.55 | [TK]D-Fender | So now you are left with a clear dialplan error |
| 15:44.56 | mrmccrac | you think the GRNOC context isn't available on my provider? |
| 15:45.03 | [TK]D-Fender | this has nothing to do with your provider |
| 15:45.06 | [TK]D-Fender | that is a DIALPLAN CONTEXT |
| 15:45.10 | [TK]D-Fender | extensions.conf |
| 15:45.36 | [TK]D-Fender | It doesn't seem you fully understand cwhat a call-file is supposed to do and use yet |
| 15:45.50 | jkroon | GRNOC has to exist locally, ie dialplan show GRNOC must give stuff back. |
| 15:46.10 | [TK]D-Fender | And you must have created it and and exten to match |
| 15:46.12 | mrmccrac | i see, so this: |
| 15:46.43 | mrmccrac | http://pastebin.com/raw/LKERhbA0 |
| 15:46.56 | mrmccrac | is bottom of my extensions.conf |
| 15:47.02 | mrmccrac | i have context outbound and inbound there? |
| 15:47.12 | [TK]D-Fender | [GRNOC[ <--- where do YOU see this context you told your call-file to use? |
| 15:47.26 | [TK]D-Fender | you said "look in [GRNOC]" |
| 15:47.34 | [TK]D-Fender | there is no context with that name |
| 15:47.48 | mrmccrac | it works :) |
| 15:48.14 | mrmccrac | you guys accept paypal? :P |
| 15:48.14 | jkroon | great. now I just need to figure out HANGUPCAUSE() usage ... |
| 15:48.34 | jkroon | mrmccrac, a thank you is adequate for me thanks. |
| 15:48.53 | [TK]D-Fender | yup |
| 15:48.54 | jkroon | there is many people here that have assisted me, so I just see it as "pay it forward" |
| 15:48.56 | mrmccrac | thank you |
| 15:49.00 | mrmccrac | yes i will try to do the same |
| 15:49.13 | mrmccrac | i know this stuff takes time and money from you so |
| 15:49.36 | mrmccrac | lunch time |
| 15:49.38 | *** join/#asterisk tby (zeus@flash.fw42.net) |
| 15:50.01 | jkroon | of which time is really the more valued resouce. |
| 15:53.14 | tby | Hi! one of our Asterisk servers (version 1.8.32.3) started going nuts today, I cant figure out whats causing the issue we have not changed any configs. The logs keep getting these error messages and some agents cant login. WARNING[25252]: channel.c:1544 __ast_queue_frame: Unable to write to alert pipe on Local/8800103@default-00000099;1 (qlen = 96): Resource temporarily unavailable! |
| 15:54.23 | jkroon | file descriptor limits is a likely culprit (ulimit -n) |
| 15:54.39 | tby | oh, good call going to try right away thx |
| 15:55.37 | jkroon | easy to check: f=(/proc/$(pidof asterisk)/fd/*); echo "${#f[@]}"; cat /proc/$(pidof asterisk)/limits |
| 15:55.55 | tby | let me run that |
| 15:55.56 | jkroon | Max open files is the one that's probably too low. |
| 15:56.27 | jkroon | so I typically run with a ulimit of 8192 where the default is typically 1024. |
| 15:57.30 | tby | yea its default 1024 |
| 16:13.05 | jkroon | [TK]D-Fender, ok, so the SIP causes from chan_sip gets added to the created channel but not the requesting channel. |
| 16:13.21 | jkroon | is that a bug or how do I deal with that? |
| 16:13.42 | tby | jkroon: increasing the ulimit solved the problem, thanks! :) |
| 16:13.44 | [TK]D-Fender | no idea |
| 16:14.28 | jkroon | [TK]D-Fender, thanks for a straight forward answer. any idea who might be able to assist? |
| 16:14.43 | [TK]D-Fender | An actual dev probably... |
| 16:15.08 | jkroon | asterisk-dev? |
| 16:15.30 | [TK]D-Fender | that;'s one place to look for them... |
| 16:16.01 | jkroon | Matt Jordan created the documentation at https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause but those examples doesn't work. |
| 16:16.58 | [TK]D-Fender | Show us |
| 16:17.36 | jkroon | http://pastie.org/10793630 |
| 16:18.55 | [TK]D-Fender | jkroon, What version are you running? |
| 16:19.29 | jkroon | 11.21.1 (upgaded from 11.17.1 earlier today to confirm that it's not an already fixed bug) |
| 16:19.41 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 16:20.46 | [TK]D-Fender | ... other stuff I'd prefer to not paste here ... <--- hard to be sure of the impact of this |
| 16:22.32 | jkroon | it basicaly just sets CALLERID(num) stuff. |
| 16:25.27 | jkroon | well, that's really all it does. |
| 16:25.52 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 16:34.59 | *** part/#asterisk ccc_ (50186c15@gateway/web/freenode/ip.80.24.108.21) |
| 16:59.14 | mrmccrac | alright i have it both receiving and dialing calls now .. had to make sure the context was named correctly for the incoming calls based upon what my provider was sending me |
| 17:00.30 | [TK]D-Fender | contextt has nothing to do with what the provider sends you |
| 17:00.45 | [TK]D-Fender | If you're matching a peer it uses whatever context you pointed it to |
| 17:00.52 | mrmccrac | it started working after i fixed the entry in extensions.conf |
| 17:00.54 | [TK]D-Fender | SIP knows nothing about "context" |
| 17:01.05 | [TK]D-Fender | As well it should... that's your auto-callout\ |
| 17:01.10 | [TK]D-Fender | You pointted it to that context... |
| 17:01.16 | [TK]D-Fender | it has nothing to do with your provider |
| 17:01.28 | mrmccrac | this was a different error i was getting for an incoming call |
| 17:01.42 | mrmccrac | [2016-04-11 16:57:02] NOTICE[31653]: chan_sip.c:23613 handle_request_invite: Call from 'GRNOC' (10.234.232.37:8060) to extension '5555555555' rejected because extension not found in context 'cic'. |
| 17:01.51 | mrmccrac | oh |
| 17:02.12 | mrmccrac | i see what you mean |
| 17:02.41 | [TK]D-Fender | Again, that context is what you put in your sip.conf. Not their fault |
| 17:02.45 | mrmccrac | right |
| 17:15.35 | jkroon | [TK]D-Fender, chan_local bug. |
| 17:16.20 | jkroon | just in case you're wondering, my config and code is correct, chan_local is however not handling the control frames. |
| 17:18.04 | [TK]D-Fender | \o/ |
| 17:23.58 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 17:24.55 | qakhan | hi all, i have 1 phone which unregister every day and i have to restart it in order to make it work, i changed the switch port and cable as well. but same issue |
| 17:27.53 | [TK]D-Fender | Why would any of us consdier looking at the cable first? |
| 17:27.58 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 17:29.06 | qakhan | [TK]D-Fender ? |
| 17:29.28 | [TK]D-Fender | What's hard to understand? |
| 17:29.47 | [TK]D-Fender | Nobody would go looking at the CABLE first as being the source of the problem. |
| 17:30.42 | qakhan | ok |
| 17:31.07 | qakhan | i am just telling the steps which i took |
| 17:31.50 | [TK]D-Fender | What else? |
| 17:34.01 | qakhan | checked the phone settings, |
| 17:34.17 | qakhan | all phones have same settings, no change |
| 17:35.06 | *** join/#asterisk n3ob (~ed@2a00:d880:6:320:82fa:b33e:3d20:4763) |
| 17:41.46 | [TK]D-Fender | That tells us nothing |
| 17:44.01 | *** join/#asterisk tmottabr (~tmotta@191.17.222.15) |
| 17:49.10 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 17:56.23 | jkroon | qakhan, phones. makes. models. firmware versions. |
| 17:56.36 | jkroon | i've seen an old SNOM6.X bug causing exactly what you're describing. |
| 18:03.12 | qakhan | jkroon its Polycom VVX 600 |
| 18:03.21 | *** join/#asterisk KaliLinuxGR (~alexandro@adsl-93.109.242.183.tellas.gr) |
| 18:03.29 | jkroon | the person to speak with is chainsaw - he's offline atm. |
| 18:04.15 | jkroon | based on sip debug of the IP, what can you gather? |
| 18:05.21 | qakhan | currently its working, |
| 18:05.45 | qakhan | do you think we can get any usefull infor from sip debug at this time |
| 18:05.52 | jkroon | doesn't matter, get that output. |
| 18:05.55 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 18:06.22 | jkroon | so what we've seen was that it was the initial response to the INVITE that then caused the SNOM to stuff up on the next INVITE. |
| 18:07.31 | jkroon | you can possibly hasten breakage by setting minexpiry, maxexpiry and defaultexpiry rather low. |
| 18:07.53 | jkroon | eg, all to a value of say 120. |
| 18:08.26 | dasjoe | [TK]D-Fender: a linter, a tool which checks asterisk's config files for mistakes and errors |
| 18:08.50 | [TK]D-Fender | No such thing that I've heard for for * |
| 18:09.38 | dasjoe | Boo, so the only way to check one's config for unintended stuff is to read through it carefully or pay somebody else to do so? |
| 18:09.56 | [TK]D-Fender | "This website is best viewed with .... your eyes" |
| 18:10.23 | [TK]D-Fender | "unintended" doesn't mean "wrong systex that can be found with a checker" |
| 18:13.07 | *** join/#asterisk scv (solar@162.217.176.109) |
| 18:13.14 | *** join/#asterisk busymind (~textual@c-24-21-79-235.hsd1.wa.comcast.net) |
| 18:13.28 | scv | is it expected to have no warning when attempting to transcode to a codec that isnt available/loaded |
| 18:13.35 | scv | even with debug/verbose set high |
| 18:13.43 | scv | 13.8.0 |
| 18:14.36 | [TK]D-Fender | not loaded will give you an error. |
| 18:14.37 | *** join/#asterisk pankid (~quassel@mail.nschoonover.com) |
| 18:14.59 | [TK]D-Fender | there should always something... |
| 18:15.03 | [TK]D-Fender | Show us the actual call |
| 18:15.11 | scv | in my case, both legs joined the bridge then left immediately |
| 18:15.25 | scv | cdr record lists normal call clearing, no debug warning error/etc in console or log |
| 18:17.55 | *** join/#asterisk areski (~areski@80.174.128.77.dyn.user.ono.com) |
| 18:18.18 | qakhan | jkroon here |
| 18:18.25 | qakhan | http://pastebin.com/MgTT8kBR |
| 18:19.29 | jkroon | qakhan, there is no REGISTER request in there. |
| 18:20.28 | jkroon | basically what you need to look for: The REGISTER that's successful, then wait until it breaks, check if it sent another REGISTER - if it did - WHY did asterisk reject, if it didn't - figure out WHY it stopped sending REGISTER. |
| 18:21.33 | qakhan | jkroon it is rendom, |
| 18:21.37 | *** join/#asterisk lgaetz_ (~lgaetz@66.185.28.100) |
| 18:21.51 | qakhan | so i have to trun on debug on that peer |
| 18:22.21 | jkroon | at the risk of having a lot of people point out the contrary, but in an IT world I've come to learn that almost nothing is random. it only seems random until you understand the cause. |
| 18:22.27 | jkroon | yes |
| 18:23.12 | qakhan | ok |
| 18:28.47 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 18:47.42 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 18:54.34 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 18:54.55 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 19:06.21 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 19:09.50 | avb | dasjoe: 'dialplan reload' complains about syntax errors. you can wrap this up to a script |
| 19:10.25 | *** join/#asterisk hfb (~hfb@96.247.52.240) |
| 19:15.05 | [TK]D-Fender | very few |
| 19:28.45 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 19:31.21 | *** join/#asterisk hfb (~hfb@96.247.52.240) |
| 19:46.19 | *** join/#asterisk AviiNL (~AviiNL@h178213.upc-h.chello.nl) |
| 20:21.40 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 20:24.43 | *** join/#asterisk elbriga (~elbriga@187.95.109.175) |
| 20:41.11 | *** join/#asterisk Inf0r (uid2810@gateway/web/irccloud.com/x-ryiqwometmcccwep) |
| 20:44.38 | *** join/#asterisk kim_bruning (~kim@2001:980:de98:1:d250:99ff:fe20:5e06) |
| 20:46.50 | kim_bruning | why would exten=>foo,1,GotoIf( [1 = 2]?10:20) always jump to priority 10? |
| 20:47.12 | kim_bruning | I must be misundderstanding the syntax or something |
| 20:49.53 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:50.12 | kim_bruning | wait, do I need to do $[1=2] .... ? |
| 20:50.15 | kim_bruning | says, on rereading |
| 20:50.17 | kim_bruning | dangit |
| 20:50.25 | kim_bruning | thank you channel O:-) |
| 20:52.39 | kim_bruning | I love freenode help channels ;-) |
| 20:53.04 | qakhan | [TK]D-Fender and jkroon there is no sip debug when phone unregister, but here is sip debug when phone get register |
| 20:53.05 | qakhan | http://pastebin.com/Z7PAHZNk |
| 21:12.06 | *** join/#asterisk RawPeacock (~RawPeacoc@206.169.158.138) |
| 21:14.40 | RawPeacock | So I'm following a route and I'm unable to replicate an odd error, but when an extension calls a number it'll get an error stating "this number is unallocated" just from that extension.. has anyone experienced this? |
| 21:18.52 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 21:20.03 | *** join/#asterisk linuxfool (~james@DHCP-149-254.resnet.ua.edu) |
| 21:28.13 | [TK]D-Fender | Show us the call. |
| 22:46.46 | RawPeacock | I feel like a fool I'm going to analyze it a bit more [TK]D-Fender , perhaps tomorrow I can get a full SIP capture and see what might be up |
| 23:08.49 | *** join/#asterisk KaliLinuxGR (~alexandro@adsl-93.109.242.183.tellas.gr) |
| 23:09.32 | *** part/#asterisk kharwell (kharwell@nat/digium/x-aipwduhukwzbiovq) |
| 23:11.43 | *** join/#asterisk linuxfool (~james@DHCP-149-254.resnet.ua.edu) |
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