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| 02:21.24 | jeffspeff | once a phone registers upon boot, does it register again periodically or does the sip qualify from * take care of that? |
| 02:21.57 | WIMPy | periodically |
| 02:22.48 | jeffspeff | how can i change the register periods? is that done on the device itself or through the sip settings for that peer? |
| 02:22.57 | WIMPy | both |
| 02:23.14 | jeffspeff | does one override the other or do they both apply? |
| 02:23.40 | WIMPy | They hopefully agree on somethign that works. |
| 02:25.02 | jeffspeff | what i'm getting at is, i have a firewall rule that allows 12 registers from a single IP within 60 seconds. what i'm finding is that over the course of about an hour or so, the public ip of the office where those phones are gets denied through the FW because of too many register attempts |
| 02:25.33 | WIMPy | How many clients from that site? |
| 02:25.47 | jeffspeff | i don't want to increase the hitcount too much because that begins to defeat the purpose. there are 50+ currently |
| 02:25.52 | WIMPy | And registrations ony? |
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| 02:26.04 | WIMPy | That's what I said :-) |
| 02:27.50 | jeffspeff | ? |
| 02:27.59 | jeffspeff | WIMPy, here's the iptables i'm currently using http://pastebin.com/xTJfBNKS |
| 02:28.04 | WIMPy | jeffspeff: YOu probably just didn't get the call completed. 6/s is FAR too little. |
| 02:28.09 | WIMPy | Which also means that a rate limit is pretty pointless :-( |
| 02:28.30 | jeffspeff | i saw that earlier, that rtp issue has been resolved. |
| 02:29.23 | jeffspeff | the issue i'm having now is that the phones stop responding after an hour or so. if i turn off iptables, they all come back |
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| 02:30.14 | WIMPy | And I'm not so sure it's a good idea to have all those rules independantly. |
| 02:30.52 | jeffspeff | what do you mean? |
| 02:31.28 | WIMPy | Just because someone get's blocked by one rule doesn;t mean ha can't continue with the others. |
| 02:33.18 | jeffspeff | well, if he's trying to hack me via register attempts then i cover that with the TCP and UDP registry blocks, same for invites. is there a better way to do it? |
| 02:34.15 | WIMPy | And why are you DROPping attempts form one scanner while REJECTing those from another? |
| 02:35.38 | WIMPy | The best solution would be a user space firewall, but I didn't try that, yet. |
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| 02:49.21 | WIMPy | Using --seconds after a --set should always be true. |
| 02:50.49 | jeffspeff | it's still a work in progress. i found the example with those reject rules on voip info site and thought that due to the UA strings they applied to helped block that scanner specifically or something |
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| 02:51.50 | WIMPy | If it is to be effective, you will block legit users as well. |
| 02:52.02 | WIMPy | Just takes someone making a typo. |
| 02:52.22 | jeffspeff | and i'm ok with that since i configure all of the phones. |
| 02:52.28 | WIMPy | Or redialling too fast. |
| 02:52.42 | jeffspeff | that's something i'm needing to tweak too |
| 02:54.45 | WIMPy | And you can still make calls with that 6/s limit? |
| 02:54.52 | jeffspeff | yeah |
| 02:55.03 | jeffspeff | i need to up that some |
| 02:55.11 | WIMPy | wondes what the default bucket size is. |
| 02:55.54 | jeffspeff | i'm thinking i should remove that rate limit rule |
| 02:57.10 | jeffspeff | and if i set the other rules to only apply to --state NEW then that would solve the issue of the phone re-registering and someone redialing wouldn't it? |
| 02:57.34 | [TK]D-Fender | UDP = stateless |
| 02:57.47 | jeffspeff | thanks |
| 02:57.59 | WIMPy | iptables = not stateless |
| 02:58.33 | WIMPy | And no you shouldn't. That would completely disable them. |
| 02:59.16 | jeffspeff | ah, because it wouldn't be considered new anymore after the first attempt |
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| 02:59.50 | WIMPy | Not after Asterisk sent out the first reply. |
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| 03:04.19 | jeffspeff | WIMPy, would it work better if instead of matching just REGISTER AND INVITE if i matched what * replies to bad register and invites requests with? |
| 03:05.23 | WIMPy | That's what things like fail2ban do, just on a longer journey. |
| 03:06.28 | WIMPy | uses AMI to get the information, but it's still not realtime, off course :-( |
| 03:07.53 | jeffspeff | i have alwaysauthreject set to yes so any bad registers should always reply with the same thing right? |
| 03:08.06 | WIMPy | And yes, I do get false positives regularly. So the concept is definitely flawed. |
| 03:08.11 | WIMPy | yes |
| 03:08.47 | jeffspeff | what about the invites? are there any other sip methods other than registers and invites that i should protect against? |
| 03:09.33 | WIMPy | OPTIONS can give away clues about existing extensions, IIRC. |
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| 03:12.55 | WIMPy | SUBSCRIBE would also be a candidate. |
| 03:13.51 | jeffspeff | should i just filter the entire sip rfc? lol |
| 03:14.04 | WIMPy | Maybe. |
| 03:14.45 | WIMPy | doesn't make any differeces. |
| 03:14.49 | jeffspeff | on a separate note, is there a way in sip to limit number of active calls for an individual peer/user/friend? or do you have to do that counting method in the dialplan? |
| 03:16.26 | WIMPy | The COUNT* functions are the preferred way now. |
| 03:16.42 | WIMPy | But I think the limits still work. |
| 03:18.08 | jeffspeff | where are those configured at? |
| 03:18.26 | WIMPy | sip.conf |
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| 06:03.10 | juned | Hi Guys |
| 06:03.31 | juned | I am having some problem with System command |
| 06:04.05 | juned | I am posting data using curl and writing its response in one text file using System command |
| 06:06.52 | juned | https://paste.debian.net/311897/ |
| 06:07.13 | juned | here is the output of Asterisk processing |
| 06:07.43 | juned | and this is my dialplan command |
| 06:07.43 | juned | exten => h,n,System(echo "\nResponse ----\n${RESPONSE}\n=======================================================" >> ${LOG_FILE}) |
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| 06:30.04 | WIMPy | Embedding newline does not look like a good idea. |
| 06:31.00 | juned | Hi WIMPy: if remove \n will it solve my issue ? |
| 06:31.42 | WIMPy | Not sure what your issue is, but that looks like a potential issue to me. |
| 06:32.19 | juned | Actually Response contains lots of html tags so i think its breaking somewhere and its not writing in file |
| 06:33.00 | juned | is there anyway i can pass it as string something like "${RESPONSE}" ? |
| 06:33.30 | WIMPy | Yes. That way. |
| 06:33.47 | WIMPy | But that might be taken very literally again. |
| 06:35.29 | juned | let me check with that ... |
| 06:37.15 | juned | as we already have response in "" this won't work i guess |
| 06:37.28 | juned | WIMPy: what do you suggest ? |
| 06:37.42 | WIMPy | Quotes end up literal as well, usually. |
| 06:38.09 | WIMPy | There's no need to nest quotes. |
| 06:39.05 | juned | Okay but that way its not working :( |
| 06:41.06 | WIMPy | Passing parameter can e a PITA. What exactely do you need to do? |
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| 06:44.57 | drmessano | That looks so fundamentally flawed.. Why don't you use system() to write to specific lines in the file and use something like PHP to interpret the page |
| 06:45.15 | drmessano | Rather than dumping the whole thing with a giant echo |
| 06:47.51 | juned | drmessano: if there something minor i am missing here then i would like to fix it first |
| 06:48.45 | WIMPy | Might be a good idea to find a good solution for the problem instead of fixing one that might not make sense anyway. |
| 06:51.00 | juned | WIMPy: all right, can you suggest me anything to do the same ? |
| 06:51.18 | WIMPy | What exactely do you need to do? |
| 06:51.56 | juned | WIMPy: once call completes i am posting some data using curl |
| 06:52.34 | juned | and I want to write that response in some log file to trace the problem in future |
| 06:53.14 | juned | drmessano: can you give small example to achieve this using php ? |
| 06:53.57 | WIMPy | So you post some data using CURL and only want to log the result, but you are not using any result otherwise? |
| 07:06.29 | juned | <PROTECTED> |
| 07:06.44 | juned | WIMPy: I am setting response like this |
| 07:07.16 | WIMPy | Either I'd call CURL via system and redirect directly to the log from there, |
| 07:08.14 | WIMPy | or if you want to save some resources, you can avoid calling external stuff altogether by using AMI. |
| 07:08.35 | WIMPy | But that requires a little programming. |
| 07:08.51 | juned | :( |
| 07:10.04 | juned | if i call CURL using system command then how do i write result from there itself |
| 07:10.35 | WIMPy | Same as you do with echo. |
| 07:12.51 | juned | exten => h,n,Set(RESPONSE=${System(CURL request) |
| 07:12.56 | juned | something like this |
| 07:12.57 | juned | ? |
| 07:13.28 | WIMPy | System is not a function. And teher's nothing to Set. |
| 07:13.50 | WIMPy | Did you read about dialplan basics? Either in the book or on the wiki? |
| 07:14.29 | juned | Yeah read while ago... |
| 07:15.59 | WIMPy | You only need System, nothing else. |
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| 07:37.05 | juned | WIMPy: exten => h,n,System(curl command 2>&1 logfile.txt ) |
| 07:37.05 | juned | I can do something like this correct ? |
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| 07:37.53 | WIMPy | I#m missing the actual redirection, but otherwise, yes. |
| 07:38.40 | juned | ohh okay i ll try |
| 07:38.44 | juned | Thanks .... |
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| 08:10.36 | wyoung | WIMPY!!!!!!!1 |
| 08:11.39 | WIMPy | Here! |
| 08:13.36 | wyoung | I am still having issues with registering with SIP. If I don't reload asterisk every minute or so I cannot receive incoming calls. |
| 08:14.06 | WIMPy | Are you behind NAT? |
| 08:14.12 | wyoung | Yes |
| 08:14.20 | wyoung | Cisco router handles that |
| 08:14.30 | WIMPy | And the router is extremely shitty? |
| 08:14.37 | wyoung | It is Cisco so yes :) |
| 08:14.45 | wyoung | Overpriced too |
| 08:14.54 | WIMPy | Cisco challenged networking... |
| 08:15.23 | WIMPy | I never needed it, but isn't there a keepalive option in Asterisk? |
| 08:15.45 | MaliutaLap | ~sipnat |
| 08:15.45 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 08:15.47 | wyoung | My raspberry PI is faster at routing than this Cisco one |
| 08:16.11 | WIMPy | RPis are real slow at networking. |
| 08:16.16 | wyoung | WIMPy: yes there is a keepalive option, and I have tried it |
| 08:16.26 | wyoung | WIMPy: Faster than this cisco |
| 08:16.42 | WIMPy | Does the Cisco have some SIP ALG enabled? |
| 08:16.50 | wyoung | It can only route at 10Mbit |
| 08:17.01 | wyoung | I have disabled all ofthe SIP crap on the router |
| 08:17.03 | WIMPy | (although I'd expext it to always fail then) |
| 08:17.08 | wyoung | and SIP VLAN |
| 08:17.36 | wyoung | I can pastebin asterisk and cisco router config if you think that will help |
| 08:17.39 | WIMPy | Get a decent router. |
| 08:17.59 | WIMPy | That's probably faster and cheaper. |
| 08:18.07 | wyoung | Not entirely sure it is the router that is causing the issue |
| 08:18.21 | wyoung | plus customer won't be happy, they spent over $700 on it |
| 08:19.27 | WIMPy | If they can spend 700$ on some challenging stuff, I guess they shold have a spare 30$ for somethign that works. |
| 08:20.01 | WIMPy | tcpdump/wireshark are your friends to prove what's going on. |
| 08:20.09 | wyoung | Customer logic is they have already spent $700 on a "good" (well known brand name) router, why should they spend anything else on it? |
| 08:20.22 | wyoung | what am I looking for on tcpdump / wireshark? |
| 08:20.59 | WIMPy | So they don't have to spend another 700$ for your time to figure how to make it work. |
| 08:21.08 | WIMPy | Just if you receive anything at all. |
| 08:21.25 | wyoung | I cannot see the call come in at all |
| 08:21.57 | wyoung | hmmm i might be able to dump some packets on the router too, just need to look at reference manual again |
| 08:22.13 | WIMPy | So unless the ISP screwes things up, there's the router left. |
| 08:23.22 | wyoung | hmm, possible, ISP is the largest telco here (for PSTN) |
| 08:24.04 | wyoung | I wouldn't put it past them |
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| 09:21.46 | o-zone | hi all, i've got an issue with queues using LocalAgent |
| 09:23.24 | o-zone | in details, LocalAgents simply relay call signal to real extensions but, when someone answer the call, no audio |
| 09:24.12 | o-zone | i have an OpenSIPS router that forward to Asterisk for queue needs |
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| 09:31.01 | yun1989 | hello all ? |
| 09:31.35 | o-zone | hi yun1989 |
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| 14:06.51 | kannan | hello. In asterisk certified 1.8 , is it possible to use filters for custom columns in cdr_adaptive_odbc, the examples show only src and dst ? |
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| 14:12.13 | WIMPy | Show? Where? Asteris doesn't show anything. It just writes there. |
| 14:13.03 | kannan | WIMPy , i meant in examples in asteriskdocs |
| 14:13.44 | WIMPy | doesn't understand. |
| 14:14.56 | kannan | filter <CDR variable> => <content> is syntac in cdr_adaptive_odbc.conf. can this be a custom column also? |
| 14:15.18 | kannan | samples and examples deal with src, dst or accountcode only |
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| 15:03.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 16:50.43 | qakhan | i am trying to connect 2 asterisk throught sip trunk. but it is keep saying wrong password. here is my config http://pastebin.com/sQwVtKqk |
| 16:53.09 | [TK]D-Fender | You shouldn't be registering BOTH ends. |
| 16:53.33 | [TK]D-Fender | in fact both are on FIXED IP's locally.. you shouldn't be registering AT ALL |
| 16:55.07 | qakhan | [TK]D-Fender you mean register => |
| 16:55.09 | qakhan | right? |
| 16:55.20 | [TK]D-Fender | What else would I be talking about? |
| 16:55.31 | qakhan | :) |
| 16:55.32 | [TK]D-Fender | Registing is when one side DOESN'T know where the other will be |
| 17:18.43 | dan_j | Hi. I'm trying out imap voicemail storage for the first time. It seems to really slow down the startup of asterisk when there are a large-ish number of mailboxes as it seems to check them all before continuing with the load. Is there any way to get asterisk to do all that in a background thread so that it doesnt block the startup process? |
| 17:18.57 | *** join/#asterisk cstk421_ (~cstk421@c-68-46-158-24.hsd1.mi.comcast.net) |
| 17:19.44 | cstk421_ | having issues finding support for hylafax for use with elastix. any channel |
| 17:19.49 | cstk421_ | any channels you would suggest ? |
| 17:19.52 | [TK]D-Fender | #elastix |
| 17:20.10 | cstk421_ | [TK]D-Fender: been there no one seems to know much about the fax functionality |
| 17:20.24 | [TK]D-Fender | Then no channel. Hit their forums, etc |
| 17:20.55 | cstk421_ | [TK]D-Fender: k |
| 17:23.55 | qakhan | [TK]D-Fender i removed the register now when i call from phone 3288 registered on server A to phone 000308004 registered on server B, call does not go through here is sip debug from server B |
| 17:23.56 | qakhan | http://pastebin.com/qM0mewV0 |
| 17:24.36 | [TK]D-Fender | because it is using the CALLERID as the username |
| 17:24.44 | [TK]D-Fender | you hsoul be setting FROMUSER in your peers. |
| 17:25.10 | [TK]D-Fender | And allowing the rpid to cover the CID part |
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| 17:37.28 | qakhan | Thanks [TK]D-Fender it is working now. but why there is SIP/2.0 401 Unauthorized |
| 17:37.37 | qakhan | in sip debug |
| 17:37.58 | [TK]D-Fender | because it is a CHALLENGE |
| 17:38.12 | [TK]D-Fender | it clearly is not the END of the conversation |
| 17:40.46 | qakhan | so can we ignore it? |
| 17:40.49 | [TK]D-Fender | yes |
| 17:40.55 | [TK]D-Fender | It clearly isn't stopping you |
| 17:40.55 | qakhan | ok Thanks |
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| 18:13.30 | philfry | I'm having a problem with polycom phones where the do not disturb doesn't do anything. I'm really not sure of the best way to troubleshoot why it isn't working. Any idea what could be causing that not to work? I recently rebuilt/upgraded the phone server and hasn't worked since doing that. I compared the old configs and i don't see any differences. |
| 18:20.56 | robmal | From what firmware did you upgrade the phones? |
| 18:22.07 | philfry | well some of the phones are running the highest available. I have a few 501s that are EOL and they are on 3.1.7. The only polycom phone that works is a VVX500 phone. |
| 18:22.12 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:23.42 | robmal | The config syntax changed a lot after 3.X so you might want to check that. |
| 18:24.06 | robmal | Also, export config from one of the phones that don't work and paste it somewhere. |
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| 18:28.50 | babak | mjordan: Hi I read in dev mailing list you mentioned about David Lee patch for adding Redis to Asterisk, is it available for test ? |
| 18:29.02 | [TK]D-Fender | DND is normally never disabled in any sample config |
| 18:29.13 | mjordan | Er |
| 18:29.17 | [TK]D-Fender | So you see it on the phone as "on", and it is NOT rejecting the calls? |
| 18:29.19 | mjordan | Which post? |
| 18:29.29 | babak | mjordan: http://lists.digium.com/pipermail/asterisk-dev/2015-March/073344.html |
| 18:29.47 | mjordan | "David Lee has a patch up (that I *still* |
| 18:29.47 | mjordan | need to write tests for, I'm way behind) that adds support for |
| 18:29.47 | mjordan | RabbitMQ." |
| 18:29.55 | mjordan | silly line breaks, sorry about that. |
| 18:29.59 | mjordan | RabbitMQ, not Redis. |
| 18:30.12 | mjordan | 6 months later; still behind. |
| 18:30.30 | mjordan | But those patches do exist, and they should be easy to apply. Note that what David wrote was for CEL/CDRs. |
| 18:31.22 | babak | mjordan: I also found this: https://github.com/tic-ull/func_redis |
| 18:32.02 | mjordan | You're more than free to try it; since it wasn't contributed upstream, I have no idea what it will do |
| 18:32.23 | philfry | so yeah i can set DND by using the feature code |
| 18:32.34 | philfry | I got it to kinda work with a soft key |
| 18:33.03 | robmal | Oh, that. |
| 18:33.26 | robmal | One of my favourite issues with polycoms. |
| 18:33.28 | robmal | http://community.polycom.com/t5/VoIP/Server-based-DND-using-FAC-NOTIFY/m-p/8042#M1080 |
| 18:34.38 | philfry | oh yuck |
| 18:36.21 | babak | mjordan: thx will try ,it seems he asked for Code Review request: http://lists.digium.com/pipermail/asterisk-dev/2015-July/075029.html |
| 18:37.28 | *** part/#asterisk nuken (~nuken@187.112.208.221) |
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| 18:38.22 | mjordan | Alas, that's not really how code reviews work in the Asterisk project, which is probably why he didn't get much feedback. |
| 18:39.32 | mjordan | Generally, although everyone who participates in the project can decide what they'd like to do, most of us don't review code in other's repos. |
| 18:39.33 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Code+Review |
| 18:42.17 | babak | mjordan: I will email this conversation for him, he may send a new code review request |
| 18:42.33 | philfry | is there a certain brand of phone that you guys recommend. I've mostly just used polycom and digium phones so far. We actually have issues with digium with the do not disturb as well but different issues. |
| 18:44.54 | igcewieling | philfry: For our customers we use ony Polycom, except when we need cordless, then we go with VTech (I think). |
| 18:45.20 | igcewieling | philfry: your DND issue is lack of sync between phone DND state and PBX DND state? |
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| 18:58.03 | philfry | well from my understanding at some point asterisk changed from using sip to feature codes for DND? I could be wrong in my understanding. I think the polycom software is still trying to send them over sip. The only polycom phone working currently is a newer one. |
| 19:00.46 | igcewieling | philfry: In my experience it is not possible to sync CF or DND status between the phone and the server. |
| 19:01.41 | ks3 | You can use macros in your polycom config to set local DND and also send a feature code to server, iirc |
| 19:02.22 | robmal | ^ The 'uglier' way |
| 19:02.28 | igcewieling | The closest solution we came up with was to remap the polycom DND and CF buttons to *XX codes and make and use the idle screen and a php script on a web server to display the CF and DND state. Horribly convoluted and we decided to live with the annoyance rather than try to push a rube goldberg-ish solution. |
| 19:02.35 | philfry | yeah i was sending the feature code using the soft key but the one thing i couldn't figure out is how to do it so when someone calls it goes to voicemail instead of them getting a busy signal |
| 19:02.49 | igcewieling | ks3: that doesn't make the phone show "DND" on the display. |
| 19:03.07 | ks3 | It does if you toggle local DND status as part of the config |
| 19:04.11 | igcewieling | ks3: interesting, though still does not reflect the actual asterisk DND state, just a best guess. |
| 19:04.13 | ks3 | Did it at my last job, don't have access to configs anymore but 99% sure we had it working both. Unless, of course, the feature code didn't make it to server. Or phone rebooted into a different state than what the server had. |
| 19:04.28 | ks3 | Yep, definitely wasn't perfect. But "mostly" worked. |
| 19:04.49 | [TK]D-Fender | Except there is no such thing as "Asterisk DND state" |
| 19:06.21 | igcewieling | My three biggest complaints about Polycom are 1) no CF / DND sync, 2) limited subset of xhtml on non-VVX phones, embarassingly crude displays on low end non-VVX phones. |
| 19:08.13 | robmal | What do you need on those non-VVX phones that hurts you the most? I've made a n-way conference setup with macros and some php to skip the built-in limit of 3/4 participants and so far the feedback is 'awesome'. |
| 19:09.45 | igcewieling | robmal: *I* don't need more, but our customers get terribly confused between *67 (DND) SVC and the DND button on the phone. Oh, fixed in the most recent release: once the phone loads the master directory once it never checks it again for updates.. |
| 19:10.17 | igcewieling | the fact you cannot edit the contact directory xml and expect the phone to pick up the changes. |
| 19:11.00 | robmal | Ad.1 http://community.polycom.com/t5/VoIP/FAQ-Using-Enhanced-Feature-Keys-EFK-macros-to-change-key/td-p/5705 Yes you can ;-) |
| 19:11.34 | robmal | Ad.2 I've made an ugly hack that emulates admin login to the phone's web panel and pushes the xml in their format :-) It works just fine. |
| 19:14.02 | igcewieling | robmal: Remapping hardkeys is not tough |
| 19:14.38 | igcewieling | Our solution so far has been to deal with "we took off DND but the calls are still going to voicemail!!!!!" on a case-by case basis. |
| 19:16.02 | igcewieling | robmal: define "pushes the xml in their format"? |
| 19:16.07 | robmal | Well, that's the solution we're using right now, but i think its ugly and plan on writing some asterisk module once i figure out how this monstrosity works. |
| 19:16.18 | robmal | One sec. |
| 19:19.05 | robmal | http://pastebin.com/iXujhgjE |
| 19:19.56 | igcewieling | robmal: indeed. How do you get the phone to keep the local copy in sync with what the server has? |
| 19:20.44 | igcewieling | I think that Admin interface might be only available in firmware 4.x and later. |
| 19:21.16 | robmal | The phones send updates to the value set in CONTACTS_DIRECTORY="/Dir/" so each phone sends /Dir/$macaddr.xml, i just push it into our db and it all stays consistent. |
| 19:21.25 | robmal | This works on FV 4.0.8 now |
| 19:21.29 | igcewieling | We have hundreds of phones which are too old to support 4.x :-( |
| 19:21.49 | robmal | I'd love to help but i don't have any ;-) |
| 19:22.06 | robmal | Also, the old admin panel was the worst shit imaginable. |
| 19:22.40 | igcewieling | *nod* I stopped doing any work on them long before 4.x came out. |
| 19:23.53 | robmal | I'll ask tomorrow if anyone has some pre 4.x phones laying around. |
| 19:27.02 | igcewieling | My best guess is 10% - 20% (100 - 200) of our phones cannot upgrade past 3.x. I won't have much time to revisit the issue for a while, but I am curious what could be done on the old phones. |
| 19:30.29 | robmal | I'll check our 'don't-look-in-here' closet tomorrow, but afair there are only some old linksys spa942 there. If you're really interested i'll talk to my boss, i'm sure it's possible but we'll neet to do that boring NDA crap etc. |
| 19:33.28 | robmal | Also: https://i.imgur.com/4N2T6eq.jpg In case you copied the pastebin. |
| 19:34.53 | igcewieling | robmal: don't spent any effort, I ask only out of curiousity. |
| 19:36.57 | robmal | No, i'd love to make this available for pre 4.X phones but i'm pretty sure i don't have access to any so i need to obtain one and have some reason for my boss to buy just one piece ;-) |
| 19:37.28 | robmal | The latter part is much harder than figuring out how to push the config to the phone. |
| 19:43.26 | robmal | Ok, 2 more brain cells engaged in this topic. |
| 19:43.35 | robmal | igcewieling: What firmware are you using? |
| 19:44.10 | philfry | yeah i didn't realize this problem was so convoluted. We have many 3.x phones that won't get 4.x. |
| 19:44.29 | robmal | philfry: 3.1.7, right? |
| 19:45.29 | robmal | Damn, the 2 braincells were wrong, polycom software matrix won't allow me to downgrade my 335 or 650. |
| 19:46.00 | philfry | yeah that is what i have on the 501. I can check the firmware that I have on the 335, and 430 phones. |
| 19:46.20 | robmal | http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 19:47.22 | robmal | Looks like i can't do anything with 3.X with my current hardware, i'll check Narnia tomorrow. |
| 19:48.59 | igcewieling | robmal: most phones are running something close to the latest available for the phone. |
| 19:49.10 | igcewieling | robmal: look at the polycom downgrader download |
| 19:49.49 | igcewieling | oh, wait. are these SOUNDPOINT or VVX phones, robmal? |
| 19:50.19 | robmal | I've got it running on both, the one i pasted is for soundpont. |
| 19:50.42 | *** join/#asterisk reveal (reveal@2401:7000:0:7::e376:fc6) |
| 19:50.57 | igcewieling | I'm pretty sure the soundpoints can be downgraded to 3.x, but the VVXs require firmware 4.x or later. |
| 19:51.10 | philfry | I noticed the SIP 450 does something completely different. You can try to enable DND from within features but it actually doesn't do anything. |
| 19:51.39 | igcewieling | philfry: on polycoms DND's default action is to send back a busy when the call arrive to the phone. |
| 19:52.16 | robmal | Yes. |
| 19:52.17 | philfry | oh okay |
| 19:55.17 | robmal | igcewieling: "The Matrix" (tm) says i can downgrade my 650 to even 3.0.4 but not 3.1.4, 3.1.6, 3.1.7. Please check your firmware and msg me, afair 3.X it's possible but a bit more complicated. |
| 19:58.16 | igcewieling | Key, Microsoft just called me to let me know I have a virus! 8-| |
| 19:59.42 | philfry | so would you recommend upgrading to 4.x or will that not fix anything? I do have some phones that support 4.x. |
| 20:01.00 | robmal | philfry: You'll notice a huge difference in the web panel. |
| 20:01.12 | robmal | Also: your provisioned config will stop working ;-) |
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| 20:05.55 | dan_j | Why would asterisk say "No voicemail provider registered" repeatedly? |
| 20:05.59 | dan_j | I'm using IMAP |
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| 20:56.04 | philfry | okay stupid question...it appears there is an upgrader to upgrade polycom phones to 4.x. Is that correct? Do you need an account on connect.polycom.com to get it? |
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| 21:06.19 | vader- | any of you guys have experience with the Polycom Soundstation IP 7000 phone? Does it always take forever to boot? |
| 21:06.23 | *** join/#asterisk doome_ (~doome@77-234-67-131.pool.digikabel.hu) |
| 21:06.54 | WIMPy | All IP phones take forever to boot. |
| 21:07.52 | vader- | this phone literally takes 5 minutes to boot |
| 21:08.09 | vader- | or longer |
| 21:08.11 | WIMPy | Ok, sounds like a new record. |
| 21:08.40 | WIMPy | Is it really booting ot does it have a timeout reaching some configured server? |
| 21:09.55 | igcewieling1 | vader-: I'll let you know when I finally get one working 8-| |
| 21:12.20 | vader- | it's booting and configuring |
| 21:12.43 | vader- | igcwieling Polycom's configuration files and compatibility matrixes are one of the most confusing |
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| 21:13.52 | philfry | i know a lot of people despise digium phones but for the most part it has made my life easier. |
| 21:14.16 | vader- | i mainly run cisco 7940 phones |
| 21:14.23 | WIMPy | So far they were the slowest to boot. |
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| 23:04.24 | vader- | so i can't figure out why after rebooting this polycom phone, when you make a call you can't hear the person on the other end⦠they can hear you but you can't hear them |
| 23:04.35 | vader- | it's usually the first 1-3 calls have this problem |
| 23:04.45 | vader- | then after that I can make 30 calls and not have any issues |
| 23:05.19 | WIMPy | "Well, it works most of the time" |
| 23:06.13 | *** join/#asterisk doome_ (~doome@77-234-67-131.pool.digikabel.hu) |
| 23:06.18 | vader- | ya that isn't good enough for our president who has been "embarrassed" by this phone during board of trustee meetings |
| 23:06.28 | vader- | so i have been tasked to correct the issue |
| 23:08.54 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:11.21 | vader- | here is the sip debug http://pastebin.com/UNFk1MpP |
| 23:11.54 | vader- | not sure if anyone can maybe see the issue |
| 23:12.17 | vader- | i just tried updating the firmware on the phone to see if it helped but no dice |
| 23:12.56 | vader- | im at 4.0.9 on the polycom firmware |
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| 23:19.39 | talntid | does anyone know where to find actual working psql schema for asterisk 13? for things like realtime sip, voicemail |
| 23:20.08 | talntid | i thought it was included somewhere with the asterisk source |
| 23:20.38 | WIMPy | Should be somewhere in contrib |
| 23:21.36 | rmudgett | See https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic |
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| 23:23.33 | talntid | just found that, rmudgett, thanks!! :) |
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