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| 07:10.04 | Parham | Hi all. Is there an invalid extension handler, like we have hangup handlers? |
| 07:10.18 | Zogot | i |
| 07:10.50 | Parham | Zogot: I'm looking for something global that doesn't have to be defined in every context. Like this: https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
| 07:11.36 | Zogot | Parham: I would then think about planning better context structure, as if you do it right you shouldn't have to define it so often |
| 07:11.50 | Zogot | Parham: use of #include or other solutions |
| 07:11.54 | Zogot | Gosub and so on |
| 07:12.00 | Zogot | sorry, wrong include |
| 07:12.06 | Zogot | [context]include => other_context |
| 07:12.34 | Parham | Zogot: Ah. I didn'tk now that. So I can define "i" and such event handlers in a context and include it in other contexts? |
| 07:12.52 | Zogot | so you could have something like [invalid_handler]exten => i,1,Playback('invalid') |
| 07:12.58 | Zogot | and in others include => invalid_halder |
| 07:13.12 | Zogot | got it? |
| 07:13.22 | Parham | Zogot: Thank you. You are great! |
| 07:13.27 | Zogot | Parham: no prob |
| 07:15.21 | Zogot | Parham: a thing to remember about includes, asterisk will find the reference in the own context first before include lines |
| 07:15.23 | Zogot | iirc |
| 07:16.01 | Zogot | so if you have _[0-9*]+,1,NoOp(First Context} and then include => other_context that defines the same but NoOp(Other Context) you will only see First Context |
| 07:16.37 | Zogot | Parham: https://wiki.asterisk.org/wiki/display/AST/Include+Statements+Basics the green text at the bottom :) |
| 07:18.15 | Parham | Zogot: That makes sense. Thanks. |
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| 08:59.15 | Parham | Does anyone know how I can enable a module at compile time without going through the menuselect interface? |
| 08:59.31 | Parham | I tried menuselect/menuselect --enable chan_sip, but it doesn't seem to be reliably working. |
| 09:05.10 | Parham | Apparently, if I run 'sudo make menuselect' just once, the 'menuselect/menuselect --enable chan_sip' command works. But I'm trying to write a shell script, and can't really run 'sudo make menuselect' because it creates a user interface. Is there a way to create the menuselect files before I run the menuselect/menuselect command? |
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| 10:30.12 | snadge | weird question.. but where is the default location for notices? is it /var/log/asterisk/notices or notice |
| 10:30.27 | snadge | without the s.. a few of our servers are inconsistent which makes writing a script that parses the notices log.. annoying :D |
| 10:34.49 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 10:34.56 | ModFather | Hi All |
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| 10:50.51 | mzbotr | Is it generally easier to send outbound calls over a SIP proxy "device" or a URL? |
| 10:51.25 | mzbotr | I have a registered device @ didlogic.net, but I've had huge difficulties in setting up outbound calling from my asterisk PBX |
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| 10:57.16 | Parham | Zogot: Sorry to bother you, but do you know how I can find more information on using menuselect from the command line? I keep googling, but I don't find how I can generate the menuselect.makeopts and menuselect-tree and such. |
| 10:59.38 | Zogot | Parham: you saw the page on the wiki? |
| 10:59.44 | Zogot | Parham: past this: https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options |
| 10:59.49 | Zogot | im not sure what else i could offer you |
| 11:00.44 | Parham | Zogot: Ah, so that is the right page. I had seen it. So it's probably something to do with my system. Thanks a lot, I'll take a deeper look. Sorry to bother you again. |
| 11:01.09 | Zogot | Parham: no worries man, you'll just have to excuse me if it takes a while to respond |
| 11:01.15 | Zogot | its sprint plan/retro day today |
| 11:01.27 | Zogot | so you caught me on the day im least behind the comp |
| 11:01.33 | Parham | Uh oh! |
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| 12:21.58 | andycol_500 | has anyone seen this error before |
| 12:21.59 | andycol_500 | Unable to create request with auth.No auth credent als for any realms in challenge |
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| 12:27.40 | mzbotr | Any thoughts on didlogic as a SIP provider? I've been having huge difficulties placing outbound calls on asterisk through them. |
| 12:28.17 | ModFather | mzbotr am using Twilio SIP trunking |
| 12:28.25 | ModFather | seems fine so far |
| 12:29.02 | mzbotr | Are you able to place a .call file in asterisk without authentication trouble w/ them? |
| 12:29.29 | [TK]D-Fender | call files have no impact on this |
| 12:30.28 | mzbotr | didlogic is very picky. Registry string will only work when I supply the extension, making outbound SIP very difficult. |
| 12:31.03 | ModFather | hi [TK]D-Fender ! |
| 12:31.30 | [TK]D-Fender | What "extension", and what makes this difficult? |
| 12:32.14 | mzbotr | register => user:pass@didlogic.net /exten |
| 12:33.12 | mzbotr | if information is in [didlogic], /didlogic is an instant failure. also, supplying any proxy addresses will mess up the authentication horribly. |
| 12:33.17 | [TK]D-Fender | that "exten" has nothing to do with OUTBOUND calls |
| 12:33.40 | [TK]D-Fender | and has nothing to do with a perr definition |
| 12:33.49 | [TK]D-Fender | peer* |
| 12:35.20 | [TK]D-Fender | And from the config sample I'm seeing on their site they don't even care about the /exten part of the register particularly |
| 12:35.29 | mzbotr | You're right, but they do care about CLI passthrough |
| 12:35.42 | mzbotr | They say a lot about it on the custom caller ID page they run. |
| 12:36.24 | mzbotr | And the .call files have been hitting the same errors as I had trying to place calls through the asterisk CLI, so I think this is probably the problem. |
| 12:37.15 | mzbotr | They would want some kind of billing stub w/ personal information that should be interesting to spokeo if you want that. |
| 12:38.04 | [TK]D-Fender | Then properly set the callerid before the call-out |
| 12:38.08 | andycol_500 | does anyone know if it is possible to modify the contact header in your sip invite with chan_sip or even pjsip |
| 12:49.17 | ModFather | [TK]D-Fender how are you today? |
| 12:50.02 | [TK]D-Fender | still breathing... |
| 12:51.01 | ModFather | i will pray to the god for you to continue breath for ever mate! |
| 12:51.27 | *** join/#asterisk hedie (50efa501@gateway/web/freenode/ip.80.239.165.1) |
| 12:51.32 | hedie | hi @all |
| 12:52.00 | hedie | i have some difficulties with the compiling of sip_chan.c |
| 12:52.02 | ModFather | hi hedie |
| 12:52.10 | hedie | it would be great if anyone could help me with this |
| 12:52.24 | hedie | i menat chant_sip.c ^^ |
| 12:52.31 | hedie | chan_sip.c |
| 12:52.54 | ModFather | paste the error output to pastebin.ca and someone who will see it could help |
| 12:53.46 | hedie | is done: http://pastebin.ca/3206445 |
| 12:53.48 | mzbotr | Is it correct to use "host" or "hosts" in a sip.conf entry for a provision URL? |
| 12:54.47 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 12:54.47 | [TK]D-Fender | what is a "provision" URL? |
| 12:55.04 | [TK]D-Fender | and "hosts" is not a valid parm name in any section in sip.conf. "host" is |
| 12:55.35 | mzbotr | fromdomain. |
| 12:56.11 | ModFather | hedie gcc: internal compiler error: Killed |
| 12:56.26 | hedie | yep |
| 12:56.34 | hedie | but its only with chan_sip.c |
| 12:56.46 | hedie | if i exclude this module, then it compile completly |
| 12:56.54 | hedie | and tells me to rund make install |
| 12:56.55 | [TK]D-Fender | "fromdomain" changes the @ portion of your invite on calls going out |
| 12:57.05 | hedie | but i need the sip module since i want to use my SIP Account |
| 12:57.23 | ModFather | hedie it seems to me an error with memory.. how much memory do you have? |
| 12:57.37 | hedie | huh... |
| 12:57.49 | hedie | 32mB vSWAP and 128MB RAM |
| 12:57.59 | ModFather | ^^ thats kinda low |
| 12:58.08 | hedie | yep its an VPS Service |
| 12:58.15 | hedie | for 4 USD / YEAR |
| 12:58.24 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 12:58.29 | [TK]D-Fender | Should have spent 5$USD |
| 12:58.46 | WIMPy | So kill everything you can and try again. |
| 12:58.54 | ModFather | hedie You appearantly ran out of memory and the kernel decided to kill gcc process. This |
| 12:58.55 | ModFather | is likely not a GCC bug. |
| 12:58.58 | ModFather | not even an asterisk bug buddy |
| 12:59.15 | ModFather | if it was asterik my lord [TK]D-Fender could help you |
| 12:59.53 | hedie | in idle, my server uses 2.5MB/128MB and 4MB of the 32MB SWAP |
| 13:00.04 | hedie | so im not sure how efficient it would be to kill something else |
| 13:00.09 | ModFather | hedie why you dont use Amazon Ec2 free tiers? |
| 13:00.24 | hedie | is that also such cost effective? |
| 13:00.36 | WIMPy | Well, you need more RAM. Or you need to compile on another machine. |
| 13:01.11 | ModFather | hedie : https://aws.amazon.com/ec2/pricing/ read for the Free Tier |
| 13:01.26 | ModFather | my daugther using a free tier for her box from amazon |
| 13:01.30 | ModFather | she is happy with it |
| 13:01.46 | hedie | sounds good |
| 13:01.58 | hedie | if i want to compile it outside of my VPS |
| 13:02.28 | hedie | do i only need to run make in the directory and after that, copy and paste this directory back to my VPS? |
| 13:02.36 | ModFather | i would only use VPS for a psybnc or an eggdrop ;P |
| 13:03.26 | hedie | ^^ lol |
| 13:03.48 | ModFather | hedie no please dont do that :) |
| 13:04.09 | ModFather | just move on a better vps box or just open an account to amazon, and use Free Tier |
| 13:04.51 | hedie | freetier is time limited |
| 13:04.58 | ModFather | its for 1 year |
| 13:05.15 | ModFather | collect more 5$ on that year and pay $10 for a double ram vps box man |
| 13:05.35 | ModFather | or make new Amazon S3 account after 1 year |
| 13:06.06 | hedie | how much ram would be enough for an working asterisk server? |
| 13:06.43 | WIMPy | Depends on what you want to do with it. |
| 13:07.07 | WIMPy | People run it on plastic routers. So it can get very little. |
| 13:07.17 | ModFather | yes as WIMPy said, depends on the "load" you are going to use |
| 13:07.34 | hedie | i just want to route one SIP Account (VirtualNumber) to multiple devices |
| 13:07.47 | hedie | but usually there will be only one call at a time |
| 13:07.57 | WIMPy | It starts with what modules you load. |
| 13:10.03 | hedie | im not very familliar with asterisk actually. so im not sure what a minimum system needs |
| 13:10.27 | hedie | after installing asterisk i planned to install freePBX as GUI |
| 13:10.59 | ModFather | for personal use: 400 MHz x86, 256 MB RAM |
| 13:11.33 | ModFather | for a small office/home less than three lines and five sets: 1 GHz x86, 512 MB RAM |
| 13:12.11 | ModFather | so you are going for the minimum requirement : 400 MHz x86, 256 MB RAM |
| 13:12.34 | hedie | ok so 512MB / 256MB SWAP would be ok |
| 13:12.40 | hedie | with 1vCPU Core |
| 13:12.52 | WIMPy | It will definitely work with less, but it may be some work to tweak things. |
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| 13:26.12 | hedie | i have ordered an VPS with 512MB RAM and 256SWAP |
| 13:26.15 | hedie | i hope it will work now |
| 13:26.38 | ModFather | it could work |
| 13:27.14 | hedie | thank you for your help |
| 13:27.23 | ModFather | you are welcome |
| 13:28.58 | ModFather | [TK]D-Fender to change music and replace the current wait music its a big headache? |
| 13:29.46 | [TK]D-Fender | Depends on your definition. |
| 13:30.05 | ModFather | i want to change the current wait music to a song of bob marley |
| 13:32.05 | [TK]D-Fender | WHEN? |
| 13:32.33 | [TK]D-Fender | you used "change" and "replace" in the same sentence there... |
| 13:32.35 | ModFather | exten => 5555,1,Playback(silence/1) |
| 13:32.35 | ModFather | exten => 5555,2,Queue(general-queue) |
| 13:32.50 | ModFather | i want to replace the current music to something else |
| 13:32.52 | [TK]D-Fender | Your queue can specify its own MOH |
| 13:33.10 | [TK]D-Fender | Set the class and then make sure it's configured in musiconhold.conf |
| 13:34.51 | ModFather | can you explain me what the class should be? |
| 13:34.53 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:34.59 | ModFather | i just checked the musiconhold.conf |
| 13:36.01 | [TK]D-Fender | it should be whatever you define |
| 13:36.27 | ModFather | can you give me an example if you had it handy? |
| 13:36.42 | [TK]D-Fender | Asterisk provides you sample configs |
| 13:36.44 | [TK]D-Fender | read them |
| 13:37.14 | ModFather | ^^ |
| 13:40.55 | ModFather | [TK]D-Fender do you believe is a good choice to use Bob Marley for MOH ? |
| 13:41.22 | [TK]D-Fender | Are you paying the LICENSING FEES for rebroadcast rights? |
| 13:41.47 | ModFather | no but i would pay if needed |
| 13:43.18 | [TK]D-Fender | it is |
| 13:43.36 | [TK]D-Fender | And since you're loking to SPECIFICALLY select this I have no idea where you go for it |
| 13:44.00 | ModFather | i will search about the rebroadcast rights |
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| 14:17.51 | d-skull | Hello, how might i turn off verbose style logging that has VERBOSE[1871] logger.c NOTICE blah blah and WARNING blah blah (it's taking up a good 12gb of space every week) |
| 14:18.40 | WIMPy | logger.conf |
| 14:18.54 | d-skull | logger.conf says do not edit this file, modified by freepbx |
| 14:19.20 | d-skull | also, there are no actual lines that funciton in that file, all commented |
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| 14:19.21 | WIMPy | Then you're asking in the wrong place. Try #freepbx. |
| 14:22.08 | inpain | Has anyone setup blf on Cisco SPA5xx phones with asterisk 13.6.0? I have yet to make line keys show anything but the current handsetâs status. Monitoring other lines is not working at all. |
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| 14:47.05 | vjfromgt | i am looking for a developer for chan_mobile - add android sms - am i in the right place? my email is vjfromgt@gmail.com |
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| 15:00.15 | emdk | I'm having issues with the ooh323 channel driver. Remote end sending TCSRelease after RTP is transmitting bidirectionally and I see an "Asn1Error: -2 at ooh323c/src/decode.c:67 and "Error: unbalanced structure" in the H323_log file with a traceleve=6 |
| 15:03.02 | glNito | So if PJSIP supports multiple endpoints registered to the same extension, does that mean I can easily replicate the shared line functionality of your traditional analog phones? |
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| 15:34.06 | [TK]D-Fender | glNito: No. That's only part of the picture |
| 15:34.57 | [TK]D-Fender | glNito: You still won't see calls in progress by another device or that one is held, or steal a held call of theirs, etc |
| 15:35.16 | [TK]D-Fender | Basically it is LESS than BLF, and you just get the same call offer as the other. |
| 15:35.20 | [TK]D-Fender | You don't see their state |
| 15:38.16 | loko- | I am running into an issue where my Cisco 7960 phone cannot receive SIP calls - debug logs shows the line is busy/congested when it calls the SIP/X. However, using a soft phone (same extension / authentication) can receive calls just fine. Any ideas on what may block the Cisco phone from I assume registering correctly? |
| 15:39.23 | glNito | I see. |
| 15:39.28 | glNito | Thanks for the clarification. |
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| 16:14.26 | [TK]D-Fender | "busy/congested" as presented is meaningless |
| 16:14.52 | [TK]D-Fender | We'd need to see what was asctually attempted |
| 16:14.55 | [TK]D-Fender | ? pb |
| 16:15.01 | [TK]D-Fender | ~pb |
| 16:15.05 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:15.07 | [TK]D-Fender | ^^^ |
| 16:15.47 | craigify | Hello fender |
| 16:15.59 | craigify | you seem to answer a lot of questions on here |
| 16:16.22 | WIMPy | He sure does. |
| 16:17.23 | loko- | [TK]D-Fender, http://pastebin.com/dPNPLXiP |
| 16:18.17 | [TK]D-Fender | loko-: SIP/2223334444 <- if this is your actual SIP device that the cisco should have been registered to we'll need to drill further |
| 16:18.25 | [TK]D-Fender | "sip show peer 2223334444" |
| 16:18.29 | [TK]D-Fender | and "sip set debug on" |
| 16:18.33 | [TK]D-Fender | Then show us a new call with both |
| 16:20.59 | loko- | [TK]D-Fender, http://pastebin.com/zw3dG7Ki |
| 16:21.25 | [TK]D-Fender | ok, and the call with SIP debug... |
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| 16:24.32 | loko- | [TK]D-Fender, HEre is the call. I stopped filtering the number, so the extension listed is correct - http://pastebin.com/cwHyVRwX |
| 16:26.10 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 16:26.15 | [TK]D-Fender | Your phone doesn't seem to ack that name |
| 16:26.25 | [TK]D-Fender | basically improper setup of the phone itself |
| 16:26.54 | loko- | thanks - I will try reconfiguring the phone |
| 16:27.05 | [TK]D-Fender | but before we do, you no longer have verbose on in there |
| 16:27.09 | [TK]D-Fender | let's see it to be clear |
| 16:27.19 | [TK]D-Fender | "core set verbose 10" |
| 16:27.25 | [TK]D-Fender | We should not be seeing LESS than the first PB |
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| 16:39.43 | loko- | I reconfigured the phone with the same settings as my softphone, but still seeing the same thing. |
| 16:40.42 | [TK]D-Fender | Show the new proper call debug |
| 16:40.45 | [TK]D-Fender | Verbose AND sip |
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| 16:43.08 | loko- | http://pastebin.com/ykBhZqgx |
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| 17:02.45 | [TK]D-Fender | loko-: Clearly we ar NOT seeing SIP debug for the call to that phone |
| 17:03.07 | [TK]D-Fender | loko-: You seem to have restricted it to your provider end..... |
| 17:06.15 | loko- | [TK]D-Fender, I am not sure? Not sure how it could be restricted to provider. This is a bare minimum asterisk setup. I am open to ideas if you have any ideas? Thanks again for your help. |
| 17:06.26 | [TK]D-Fender | "sip set debug on" |
| 17:06.36 | [TK]D-Fender | We are NOT seeing the invite going out to your phone |
| 17:07.04 | [TK]D-Fender | eally destroying SIP dialog '5de8c346359927147c95fc5e0a7f093b@204.16.241.9:5060' Method: INVITE |
| 17:07.05 | [TK]D-Fender | <PROTECTED> |
| 17:07.14 | [TK]D-Fender | Seems comms went out, but no debug for packets this time |
| 17:08.56 | loko- | the only thing I can think of is NAT - but I did not have any port forwards setup for my soft phone and it receives just fine. |
| 17:09.17 | [TK]D-Fender | No.... |
| 17:09.25 | [TK]D-Fender | In earlier debug we SAW the cisco respond |
| 17:09.30 | [TK]D-Fender | And we are NOT seeing proper debug now |
| 17:09.50 | [TK]D-Fender | reissue the debug command, "sip show peer XXXX" for it and then call again |
| 17:10.07 | [TK]D-Fender | Because we are getting less now which is not sensible. |
| 17:11.34 | loko- | [TK]D-Fender, http://pastebin.com/kVbuQY0u |
| 17:13.14 | [TK]D-Fender | <--- SIP read from UDP:73.52.254.109:5060 ---> |
| 17:13.15 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 17:13.19 | [TK]D-Fender | It IS getting the request |
| 17:13.21 | [TK]D-Fender | and rejecting it |
| 17:13.39 | [TK]D-Fender | It does not like the username from the loks of it |
| 17:13.45 | [TK]D-Fender | To: <sip:4123450602@73.52.254.109:5060;user=phone;transport=udp> |
| 17:15.42 | loko- | hmm - I have name, shortname, and authenication name all set to: 4123450602 on the Cisco phone |
| 17:18.50 | [TK]D-Fender | Not sure I can comment further on this.... |
| 17:18.59 | [TK]D-Fender | but the phone is getting the request and doesn't like it |
| 17:20.17 | loko- | no problem, I will keep playing with the phone. Thanks for your help! |
| 17:20.50 | [TK]D-Fender | You're welcome |
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| 17:24.17 | nny | Hi. Dealing with some carrier issues, want to ensure I am doing all I can. The issues I had were 1.) had to add ww to the number in Dial due to the telco not getting the first digit every time (or most times). 2.) 11 rx gain and 7 txgain due to very quiet lines (tested using milliwatt to achieve the nominal 14500). Now I am getting reports of crosstalk and if I drop the volume I get echo. Can't win, gonna call the telco but wondering if |
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| 17:33.58 | lvlinux | nny: what line hardware are you using? |
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| 17:34.11 | lvlinux | nny: line interface I mean |
| 17:34.38 | nny | lvlinux: Rhino RCB8FXX, set to hardware EC |
| 17:35.06 | lvlinux | k |
| 17:36.07 | lvlinux | well the reason you get echo if you lower the levels is the hardware EC doesn't have enough to work with so doesn't kick in. But crosstalk i don't know... |
| 17:36.15 | lvlinux | sounds like the line does need work. |
| 17:36.31 | lvlinux | and the dtmf issue is probably related to the line volume too I would think. |
| 17:36.37 | nny | yeah, and I figure the crosstalk is due to bad lines and raising the gain just gives it more opportunities to happen |
| 17:37.01 | nny | lvlinux: even after the gain adjustment I had to add the 1 second pause in, it's like the teclo isn't ready when the line is first opened (but slow enough for humans) |
| 17:37.30 | lvlinux | yeah you need to give them a call sounds like. |
| 17:37.38 | nny | k thanks! |
| 17:38.58 | lvlinux | np |
| 17:39.05 | lvlinux | good luck :-) |
| 17:40.47 | nny | haha yeah, they're switching over to a telco that uses VoIP to the CPE so this is something I have spent more time on than is worth already. Was hoping to just smooth it over until then but the crosstalk is just another problem. Should have known |
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| 19:07.25 | pecanha | Hello everyone. How can I set a variable for a unique.id/call which has a number to be playback to the channel? Something like: 1) if new call, set a variable for this session with number=20151023303 as example. Then, when the caller select a queue, I check if the variable number is set and I execute let me say: SayDigits(number). Would this be a good approach? |
| 19:08.44 | [TK]D-Fender | Could work |
| 19:13.25 | pecanha | nice |
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| 19:18.49 | vinrock | Any other active voip channels on freenode? |
| 19:20.06 | [TK]D-Fender | Nothing "generic" |
| 19:20.14 | [TK]D-Fender | there are some for other platforms |
| 19:20.27 | [TK]D-Fender | If your question is generic it can still belong here probably |
| 19:20.29 | [TK]D-Fender | So just ask |
| 19:20.33 | vinrock | I just need to learn the technology in general, guess this looks like a good place to start |
| 19:20.57 | vinrock | I inherited an att ip flex system in disarray, guess its time to expand upon the resume |
| 19:25.32 | [TK]D-Fender | What protocols does it use? What are you expectting to configure it to communicate with? |
| 19:26.31 | [TK]D-Fender | So far that looks liek a SIP trunking SERVICE, not a PBX. |
| 19:26.56 | [TK]D-Fender | Or is there a sub-product for that as well? |
| 19:27.06 | vinrock | sip trunk over bonded t1's, we're just experiencing a ton of odd behavior and i don't know enough about tele-anything to really begin diagnosis. Figure i'd start learning how all this works from scratch, build a lab and then have the ability to tell which vendor is bullshitting me. |
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| 19:27.30 | vinrock | well yeah but like i said, i need to learn it -all- |
| 19:27.43 | vinrock | pbx is a samsung officeserv |
| 19:27.49 | [TK]D-Fender | I'd start by elobarating on "odd behaviour" |
| 19:28.17 | vinrock | random inbound calls hitting 'all circuits are busy', sudednly outbound toll-free numbers are not going through with 'co hung up' message on handset |
| 19:28.18 | vinrock | etc |
| 19:28.52 | vinrock | idk if this is a hardware failure at the iad, the cisco managed router taking a dump, the implenetation of it all, the pbx |
| 19:29.56 | [TK]D-Fender | Ok, sounds like you need to read up on the general SIP RFC |
| 19:30.08 | [TK]D-Fender | And whatever debug mechanisms you have on your PBX. |
| 19:30.08 | vinrock | Yes I do |
| 19:30.16 | [TK]D-Fender | The rest is also packet dumps from your router |
| 19:30.38 | vinrock | I'm waiting for samsung to open the thing back up, previous service company locked it down and refuses to give access |
| 19:30.39 | [TK]D-Fender | (or your PBX if you can) |
| 19:31.07 | [TK]D-Fender | Well that confirms 1 vendor is already bullshitting you ... |
| 19:31.14 | [TK]D-Fender | It's now a question of how much more. |
| 19:31.18 | vinrock | oh I know, and he alwready got the boot |
| 19:31.24 | vinrock | which got me suspicious |
| 19:31.36 | vinrock | I wouldnt be surprised if this guy was half-assing it the whole time |
| 19:32.25 | [TK]D-Fender | Do not preclude the possibility of his being a COMPLETE ass. |
| 19:32.33 | [TK]D-Fender | And just marginally lucky |
| 19:32.37 | vinrock | hah oh yeah |
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| 20:47.47 | mub | You guys da real MVP |
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| 20:57.04 | WIMPy | What? |
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| 22:57.27 | lvlinux | if i have an * box with multiple IPs, how should I set the realm option in sip.conf? |
| 23:00.44 | lvlinux | clients register to the server via the IPs --- i.e. they do not use the hostname of the box. |
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| 23:34.49 | ModFather | lvlinux do you mean something like that: |
| 23:34.57 | ModFather | domain= lvlinux.com |
| 23:35.08 | ModFather | domain = lvlinux2.com |
| 23:35.15 | ModFather | domainsasrealms=yes |
| 23:35.34 | lvlinux | yes but i need to use IPs, since everything is on a LAN, and there isn't any DNS. |
| 23:35.52 | lvlinux | can I put IPs for the domain? |
| 23:36.20 | lvlinux | or will it think it's an fqdn with a wierd tld lol? |
| 23:36.54 | ModFather | nah i dont think sou |
| 23:37.00 | ModFather | you can try it, i bet its gonna work |
| 23:37.37 | lvlinux | k thanks. I was looking at the examples and saw the domainsasrealms item but didn't know if I should try it w IPs or not. |
| 23:40.30 | ModFather | you are welcome, its gonna work |
| 23:41.33 | WIMPy | Do you have any idea how the realm is linked to hostnames or IPs? |
| 23:41.54 | lvlinux | WIMPy: me? |
| 23:42.02 | WIMPy | yes |
| 23:42.16 | lvlinux | what do you mean? |
| 23:42.50 | WIMPy | Do you know what one has to do with the other? |
| 23:43.42 | lvlinux | nope to be honest I don't. I thought realm was sortof an auth mechanism that got hashed with the other auth stuff. |
| 23:43.59 | WIMPy | yes |
| 23:44.49 | lvlinux | so does that actually mean it's just an arbitrary text string, regardless of whether it contains an IP or hostname? |
| 23:45.02 | WIMPy | yes |
| 23:45.40 | WIMPy | It shouldn't matter. |
| 23:46.12 | lvlinux | ok good---so why do we have both "realm" and "domain" settings? |
| 23:47.29 | WIMPy | Because you can use multiple domains (as in name spaces) with identical user names. |
| 23:47.40 | WIMPy | But I never looked in to that. |
| 23:48.25 | lvlinux | ok |
| 23:48.43 | lvlinux | Thanks WIMPy for the good info, as always :-) |
| 23:49.09 | WIMPy | Well, you know I'm not that in to SIP. |
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