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| 01:40.14 | BeachBall | i'm getting an error when running make with asterisk 13.10 - at make chan_iax2.o error 4 |
| 01:46.05 | BeachBall | i disabled iax in menuselect = all good |
| 01:46.09 | BeachBall | :/ |
| 01:46.17 | BeachBall | just being here makes me smarter |
| 01:46.21 | BeachBall | scary |
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| 03:51.02 | drmessano | No it didnt |
| 04:54.03 | freebs | calm down |
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| 10:07.48 | Haris | is there a tool for sip to check ws/wss based connectivity ? |
| 10:28.36 | Samot | sip show peers |
| 10:28.50 | Haris | not in that sense |
| 10:29.00 | Samot | Then in what sense? |
| 10:29.09 | Samot | Clarity is your friend when asking questions. |
| 10:29.23 | Haris | like I can manually simulate the register, options process of sip call as we do with troubleshooting smtp connections by manually telnet to the smtp server and going through a smtp transaction |
| 10:29.43 | Samot | Clarity is your friend when asking questions. |
| 10:29.53 | Samot | I have no idea what you just said. |
| 10:29.58 | Haris | I need a tool or way to manually register and exchange headers for ws connection with asterisk |
| 10:30.01 | Haris | at server end |
| 10:30.05 | Samot | Yes. |
| 10:30.14 | Samot | sip set debug on |
| 10:30.44 | Haris | it shows output. it doesn't actually register or exchange options headers to make and keep the sip connection |
| 10:31.00 | Haris | I need a manual way to register with a sip server like asterisk |
| 10:31.13 | Haris | and manually exchange headers to go through the register process |
| 10:31.22 | Samot | What do you mean by "manual way"? |
| 10:31.37 | Haris | like telnet to a smtp server and then going through the smtp transaction |
| 10:31.56 | Haris | ehlo domain |
| 10:31.57 | Haris | mail from: this-and-that |
| 10:32.01 | Haris | rcpt to: this-and-that |
| 10:32.03 | Haris | data |
| 10:32.14 | Haris | type data in multi-line fashion and exit with \n\n and . |
| 10:32.23 | Samot | Uhm. |
| 10:32.33 | Haris | Is there a manual way to telnet a sip server, and manully exchange headers |
| 10:32.42 | Haris | or a tool for it |
| 10:32.45 | Haris | like wscat |
| 10:32.48 | Haris | npm install ws |
| 10:32.49 | Samot | You realize how many headers are in a SIP messege? |
| 10:33.07 | Haris | 12-13 in one time sending |
| 10:33.26 | Haris | just like a http client sending its request to server end |
| 10:33.33 | Samot | Why do you even need to do this? |
| 10:33.50 | Haris | to troubleshoot if I can actually succeed in connecting to the server via ws/wss |
| 10:34.03 | Haris | register with it, and/or stay connected |
| 10:34.08 | Samot | Wouldn't a WSS support client do that for you? |
| 10:34.13 | Samot | Since it would send the information |
| 10:34.29 | Samot | Acknowledge the auth-challenge. |
| 10:34.36 | Samot | Send back the proper response in www-digest format. |
| 10:34.47 | Haris | I'v been trying. perhaps I'm doing it the wrong way. I was not getting anywhere with a sip client that does ws. I mean get no output on errors or where the problem cams up |
| 10:34.50 | Haris | cams = came |
| 10:35.34 | Haris | ws clients don't mention errors or where problem came up .. just like a browser stays mute no matter what error comes up |
| 10:36.15 | Haris | I tried barebones, eyebeam |
| 10:38.41 | Haris | with ws/wscat I was able to connect to ws port. but I couldn't do anything further, like register or exchange headers, so I couldn't actually check if the server end was setup correctly |
| 10:46.04 | Samot | Did you actually go and install a pure Asterisk server? |
| 10:46.16 | Samot | Or are you still trying to do this on a FreePBX system? |
| 10:51.42 | Samot | Well? |
| 10:52.39 | Haris | hmm |
| 10:52.42 | Haris | no |
| 10:52.55 | Haris | this is the freepbx distro install I'm moving on with at present |
| 10:53.19 | Haris | I'm hoping most stuff is pre-configured/ready on server end |
| 10:53.53 | Haris | with a manual asterisk install, I'm not sure everything works correctly on centos |
| 10:54.12 | Haris | I have the other VM turned on, where I installed stuff manually. but not using it at present |
| 10:54.37 | Haris | if server end is ok, then I only have to worry about client end |
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| 11:08.53 | Samot | Haris: What part of "It's not going to work on FreePBX how you want." was hard to understand yesterday? |
| 11:09.23 | Haris | I'm not using freepbx on that VM |
| 11:09.30 | Samot | Flat out told that everything you are doing at this point is pointless because of how FreePBX handles WebRTC/WS calls. |
| 11:09.30 | Haris | I'm using the asterisk installed on it |
| 11:09.39 | Samot | That's what I just asked! |
| 11:10.26 | Haris | asterisk installed on VM is manually configurable ? |
| 11:10.28 | Samot | So you have an Asterisk server that you installed but haven't really configured and aren't using it. |
| 11:10.48 | Haris | Correction: asterisk installed on that VM is manually configurable ? |
| 11:10.54 | Samot | Instead you're still jerking around with the FreePBX install that's not going to do what you want. AND you're in here asking about it. |
| 11:11.25 | Haris | ok. so you want me to manually install everything all over again ? rather than just flush the prev config and do the config manually again ? |
| 11:11.45 | Samot | On the standalone Asterisk server? Yes. |
| 11:12.01 | Samot | Get rid of the FreePBX server. It's not going to do what you need/want. |
| 11:12.07 | Samot | Stop messing with it. |
| 11:12.43 | Haris | I was not asking about the asterisk server. I was asking for a tool that one can use to troubleshoot sip connectivity |
| 11:12.51 | Samot | There isn't one. |
| 11:12.53 | Samot | Move on. |
| 11:12.55 | Haris | via its various transports |
| 11:13.34 | Haris | that's a big ouch for the whole community or anyone who's trying to setup a voip box from the ground up |
| 11:14.12 | Haris | I have the other box. let me check if ws connectivity works on it |
| 11:14.19 | Samot | What you need to be asking now is "I've installed Asterisk X.X and I have followed instructions X, I've done A, B and C and now I'm getting Result Y [verbose debugs follow]. |
| 11:15.02 | Haris | Nope. don't need to ask that |
| 11:15.18 | Haris | I'm not doing text book exam here |
| 11:15.50 | Haris | I need to enable functionality that it comes with and test if that functionality works |
| 11:18.10 | Haris | its like making a virtualhost in apache and testing if that virtualhost is configured right. its like not follow exactly that manual page and that's the only thing that this piece of software can do |
| 11:18.44 | Haris | there'a a million scenarios' for which a virtualhost can be configured |
| 11:19.28 | Samot | virtualhosts have nothing to do with getting Asterisk configured to support WebRTC or TLS calls. |
| 11:19.29 | Haris | I built asterisk from the ground up with ws/wss support. I'm trying to test if it was built correctly. there's no manual page that helps me do that |
| 11:19.36 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support |
| 11:19.38 | Samot | No? |
| 11:19.54 | Haris | it tells me vaguely how to develop it. but not how to test it |
| 11:19.58 | Haris | if it doesn't work |
| 11:20.08 | Haris | Correction: or what to do if its not working as intended |
| 11:20.34 | Samot | How you test it is using the debug tools that are provided within Asterisk. |
| 11:20.40 | Haris | hmm |
| 11:21.02 | Haris | asterisk debug tools report output. they can't be used to test or troubleshoot connectivity or to simulate a basic sip session |
| 11:21.18 | Haris | which is .. what I need at this point in time |
| 11:21.23 | Samot | What connectivity? |
| 11:21.39 | Samot | You mean if your WebRTC client is sending SIP messages to Asterisk? |
| 11:21.44 | Samot | Yes, it will tell you that. |
| 11:22.06 | Samot | If you don't see the messages then you need to figure out why they aren't making it there. |
| 11:22.18 | Haris | hmm |
| 11:22.37 | Samot | tcpdump will monitor any traffic on your interfaces to see what is happening. |
| 11:22.58 | Samot | So there are various ways to see if your "connection" is making it to the Asterisk server. |
| 11:24.57 | Haris | that's ok. that part doesn't take much to make it work. simulating a sip registration or header exchange is what I need |
| 11:25.11 | Samot | Then use a client or a phone. |
| 11:25.16 | Samot | Like everyone else. |
| 11:25.40 | Samot | Somehow everyone else has managed to troubleshoot their issues in that manner. |
| 11:25.45 | Haris | is there a client which can show output of headers it sent or received for doing what it does ? |
| 11:26.08 | Samot | No. You can use wireshark on your computer the client is on. |
| 11:26.09 | Haris | debug output exchange in verbose mode at client end |
| 11:26.11 | Samot | Watch the traffic. |
| 11:26.13 | Haris | hmm |
| 11:26.29 | Haris | need to learn how wireshark does that |
| 11:27.17 | Samot | Then I guess you start learning. |
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| 11:38.33 | Haris | SIP Debugging re-enabled |
| 11:38.34 | Haris | <PROTECTED> |
| 11:38.34 | Haris | <PROTECTED> |
| 11:38.50 | Haris | ws connection is starting up. but don't know how to proceed |
| 11:38.51 | Haris | hmm |
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| 11:48.50 | Samot | So now you're showing results without explaining the steps you took to get those results. |
| 11:49.17 | Samot | What have you done to configure Asterisk for web socket support and calls? |
| 11:53.13 | Haris | umm.. |
| 11:53.24 | Haris | exactly as that page said. config http.conf, sip.conf |
| 11:53.35 | Samot | "That page"? |
| 11:53.39 | Samot | Which page is that? |
| 11:53.56 | Haris | https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support |
| 11:54.07 | Haris | res_http_websocket is loaded |
| 11:54.21 | Haris | nat=no |
| 11:56.44 | Samot | That's two things. What about the rest? |
| 11:59.04 | Samot | Prove what you've actually configured and done. |
| 11:59.07 | Haris | it says enable http server - done. it says set bindport - done |
| 11:59.08 | Samot | ~pb |
| 11:59.08 | 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. |
| 11:59.17 | Samot | Show it. |
| 11:59.20 | Haris | Ok |
| 12:02.25 | Haris | this is manually installed server asterisk/freepbx 13 |
| 12:02.33 | Samot | NO FREEPBX! |
| 12:02.37 | Haris | but this asterisk is also configured through freepbx |
| 12:02.46 | Haris | it has no ssl cert |
| 12:02.59 | Samot | FFS. |
| 12:03.12 | Haris | ok. well, it'll take me some time to get all things re-done. |
| 12:03.18 | Samot | Then do them. |
| 12:03.20 | Haris | *sigh*. bbl |
| 12:03.25 | Samot | Don't come in here with things half-ass. |
| 12:03.42 | Samot | And no FreePBX. |
| 12:03.46 | Samot | Straight Asterisk install. |
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| 12:03.58 | Haris | centos 6 ? |
| 12:04.06 | Samot | Whatever OS you'd like. |
| 12:04.10 | Samot | It's Asterisk. |
| 12:04.28 | Samot | It plays nice on CentOS, Ubuntu, Debian, etc. |
| 12:05.04 | Haris | ok. going with c7 |
| 12:06.00 | Samot | And when you come back for help it's "I followed X, did A, B, C and got result Y" |
| 12:06.17 | Samot | A,B and C better include settings up TLS properly. |
| 12:06.42 | Samot | No "How do I do?" type questions. |
| 12:06.58 | Haris | I have the cert handy. it'll be done. |
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| 14:15.04 | ntz | hello |
| 14:15.07 | *** join/#asterisk WIMPy (~wimpy@85.183.95.26) |
| 14:15.19 | ntz | i need to ask for one rather generic thing: |
| 14:16.00 | ntz | I'm designing some software solution for fax based on hylafax, t38modem and asterisk doing full routing based on fax numbers |
| 14:16.35 | ntz | I am completely not an expert on telephonia but my solution works (somehow) fine but probably not following a RFCs |
| 14:17.30 | ntz | I can send/receive faxes with users from ldap determined by subaddress when the fax.no. is in format $base_number#$subaddress (note # character) |
| 14:18.25 | ntz | does "xxx" in typical number like +32-2-12345-xxx represent subaddress ? |
| 14:18.35 | ntz | i'm little bit confused by this |
| 14:19.56 | ntz | what I'd like to point out is, that in this case the subaddress is somehow enwrapped in the fax number (part before #) so on the target device I see: |
| 14:20.31 | ntz | Aug 3 14:39:21 srv0 FaxGetty[18617]: RECV FAX (000000134): recvq/fax000000045.tif from 1111212, subaddress 07, 2 pages in 0:00:22 |
| 14:20.53 | WIMPy | Has anyone ever used subaddresses? |
| 14:21.10 | ntz | ^^ for testing purposes I just created a dialplan with sample numbers (not connected to outside world) like you see 1111212 |
| 14:22.17 | ntz | so from the hylafax POV the EXT is `1111212' and the subadress is somehow hidden inside t38 protocol |
| 14:28.07 | [NC] | ntz: subaddress is a FAX thing, it is actually part of T.30, so yes hidden inside T.38. Asterisk has no knowledge about this FAX subaddress Sender and receiver need to support and use this. |
| 14:36.05 | ntz | [NC]: it does work for me, I have no problem with that .. I'm only asking, for that our customer wants to have fax numbers in ldap and they also want to be compatible with outside world so how it works with assigning fax numbers (or tel numbers - I guess it's same) |
| 14:36.42 | ntz | they shall have a national prefix (eg +32) and the rest nin-digit-number from some provider, but who's that provider ? |
| 14:37.10 | ntz | we're now testing that in the bubble of own network that is not connected to outside world and it works as we set it in the bubble |
| 14:37.46 | ntz | if we fax to four-digits number it works .... if we create fifteen-digits numbers in dialplan it will work also |
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| 14:50.25 | [NC] | ntz: You need a VoIP provider that has numbers in the country you need them. (you could use a traditional telco provider with an ATA too..) |
| 14:55.26 | ntz | [NC]: and does it work like that the provider gives me say first six numbers eg 123456 and let me freely assign the rest of 3 ? so the format is (with nat prefix) +$nat_prefix-123456-$extension ? |
| 14:56.36 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 14:56.48 | ntz | eg for belgium +32-123456-789 where 123456 is the number I pay for to a provider and the rest 789 is my exclusive range I can do whatever I want with that ? |
| 14:57.20 | Samot | What do you mean by that? |
| 14:57.29 | Samot | Do what with it? |
| 14:57.37 | Samot | It's the last three digits of the number. |
| 14:58.56 | ntz | Samot: example from our company - we have some prefix for telephone numbers eg +420-123456-xxx and this rest part is somehow managed within our company (IP phones) and our admins can freely assign/re-assign among the employees |
| 14:59.28 | Samot | Are you sure it's not because they got a range of numbers from the provider? |
| 14:59.45 | Samot | Because in US/Canada it's NPA-NXX-XXXX |
| 14:59.51 | ntz | o.O |
| 14:59.57 | Samot | The XXXX is the same as your XXX |
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| 15:00.18 | ntz | Samot: okay, so it works like that we'll get some available range (or pool) ? |
| 15:00.19 | Samot | Companies get a range of number in XXXX and generally do assign extensions based on that XXXX part. |
| 15:00.26 | Samot | Yes. |
| 15:00.36 | ntz | Samot: good, thanks |
| 15:00.47 | Samot | You just don't randomly pick that last part of the DID out of the air. |
| 15:00.58 | Samot | They need to know how to route that number and it's calls. |
| 15:02.31 | ntz | Samot: just to explain you - I'm responisble for making it running but my knowledge about dialplan-related and other phoney-related RFCs is rather none, so actually for me everything works when we're in the PoC bubble and we can create a dialplan as we want so I am asking what I shall expect when it comes to prod |
| 15:03.06 | ntz | Samot: so in other words, I shall expect the range of numbers that are for us |
| 15:03.35 | ntz | and it will be basically 9-digits long (without NAT prefix) |
| 15:04.39 | ntz | NAT == National |
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| 15:09.49 | amessina | gtjoseph: In trying to compile the Asterick 13 branch, including the UUID generation commit from https://gerrit.asterisk.org/#/c/3404/, I get ...asterisk.c:4435: undefined reference to `ast_pbx_uuid_get' |
| 15:10.24 | gtjoseph | hmmm. you may have to do a distclean |
| 15:16.15 | amessina | gtjoseph: This is in a Koji/mock chroot for Fedora 24 x86_64 which starts off with a clean slate |
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| 15:16.49 | gtjoseph | something's inconsistent... how are you pulling the source? |
| 15:19.40 | amessina | gtjoseph: I start with the released asterisk-13.10.0 tarball, then apply the diff between 13.10.0 to branches/13 on top |
| 15:19.52 | gtjoseph | ok, let me check |
| 15:26.01 | gtjoseph | amessina: I think it's something in the diff. How was it generated. Can you try just checking out branches/13? |
| 15:27.43 | amessina | While on the '13' branch... git diff --patch --stat -M 13.10.0 > asterisk-13-branch-update.patch |
| 15:30.13 | gtjoseph | weird. the 13 branch by itself compiles fine of course. Can you try something else? Do a git format-patch instead of git diff? It'll create a patch file for each commit but you can apply them all with git am |
| 15:30.21 | amessina | gtjoseph: I'll try building the checkout of branches/13 manually this afternoon. I've got to brush up on building from the git checkout. |
| 15:30.33 | gtjoseph | no worries |
| 15:32.36 | WIMPy | ntz: You seem to be talking baout DDi, not subaddresses. |
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| 16:26.46 | amessina | gtjoseph: After removing LOW_MEMORY from MENUSELECT_CFLAGS, which I had enabled previously, I'm able to compile with https://gerrit.asterisk.org/#/c/3404/ |
| 16:27.23 | gtjoseph | is LOW_MEMORY soemthing you need? |
| 16:28.03 | gtjoseph | not that it's OK if it fails |
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| 16:30.33 | amessina | Not really. It's ok for me to disable it, but using less memory seemed like something reasonable to put in ;) |
| 16:31.34 | scv | LOW_MEMORY can cause a lot of odd issues |
| 16:31.48 | *** part/#asterisk frawd (~francois@167.red-80-33-231.staticip.rima-tde.net) |
| 16:31.48 | scv | also, dialplan strings get a hard limit of 255 characters and there is *no* warning if it is truncated |
| 16:32.14 | gtjoseph | yeah it can. it're really meant for extremely limited environments like embedded devices. |
| 16:32.35 | scv | even in those cases its pretty un-necessary these days |
| 16:33.16 | scv | average usage is like 80MB or so peak in my environment |
| 16:36.44 | gtjoseph | amessina: would you file a issue for this? |
| 16:37.31 | amessina | gtjoseph: yes, I can. |
| 16:37.39 | gtjoseph | thanks. |
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| 16:40.28 | amessina | gtjoseph: https://issues.asterisk.org/jira/browse/ASTERISK-26273 |
| 16:40.35 | amessina | Thanks for your help. |
| 16:40.46 | gtjoseph | no prob |
| 16:42.37 | davlefou | Bonjour, sauriez vous où sont stocké les messages d'accueil de voicemail? |
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| 16:53.57 | hdon | hi all :) does a 100% vanilla asterisk really need to persist astdb? we're seeing a lot of syncs to this file and we're thinking of just moving someplace ephemeral and in RAM only |
| 16:54.09 | hdon | (or is there a way to tell asterisk to sync writes for this file to disk less often?) |
| 16:55.19 | file | if you are configuration based it's used to persist inbound SIP registrations and in the case of PJSIP also subscriptions so they survive across Asterisk restarts |
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| 17:22.32 | scv | am i correct in understanding that under chan_pjsip if a qualify OPTIONS is sent and the response is received it should refresh the contact expiry timer |
| 17:24.14 | [TK]D-Fender | no |
| 17:24.22 | [TK]D-Fender | that isn't bumping a registration timer |
| 17:24.44 | [TK]D-Fender | that's just a keep alive to see if it should give up period. The device is still required to re-reg on interval |
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| 17:28.19 | scv | thanks |
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| 17:28.42 | scv | i think there might be an ALG in the path here overwriting the registration expiry :/ |
| 17:30.31 | scv | yep. wtf |
| 17:31.26 | SpaceInvaders | if you load a variable with a file (e.g. myvar=FILE(/test.txt,u) can you treat that var like a file and read it line-by-line (e.g. FILE(myvar,line#,#lines,l) ? I'm guessing no but that'd be handy :) |
| 17:31.37 | SpaceInvaders | er... |
| 17:31.48 | SpaceInvaders | if you load a variable with a file (e.g. myvar=FILE(/test.txt,u) can you treat that var like a file and read it line-by-line (e.g. FILE(${myvar},line#,#lines,l) ? I'm guessing no but that'd be handy :) |
| 17:33.04 | file | scv, you aren't Antonis Psaras are you? |
| 17:33.22 | scv | no |
| 17:33.26 | scv | why do you ask |
| 17:33.38 | file | an issue was filed against chan_sip yesterday re: registration expiry |
| 17:33.47 | scv | ah |
| 17:33.50 | scv | this is under pjsip |
| 17:33.58 | file | ack |
| 17:34.37 | scv | i took a whack at the compact headers issue btw |
| 17:34.44 | scv | the problem is in pjsip itself |
| 17:34.57 | [TK]D-Fender | SpaceInvaders, Good reason to start looking at AGI, etc |
| 17:35.13 | scv | the generic header function assumes that there will be a compact header alternative always |
| 17:35.19 | file | scv, yeahhhhhh |
| 17:35.32 | scv | i suppose it could be worked around by passing the full header as the compact version |
| 17:35.34 | scv | but that's icky |
| 17:35.38 | file | https://issues.asterisk.org/jira/browse/ASTERISK-26241 |
| 17:35.48 | file | tharrrr be the issue for that |
| 17:36.00 | SpaceInvaders | [TK]D-Fender I'm still working on understanding AMI 3: |
| 17:36.12 | scv | is pjproject responsive to bug reports? |
| 17:36.23 | SpaceInvaders | but I'll keep that in mind as I plan on reading up on that, as well. Thank you! |
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| 17:36.32 | file | depends |
| 17:36.38 | file | it's faster if we do it usually |
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| 17:47.55 | scv | there's no issue patching pjsip sources from asterisk? |
| 17:48.14 | scv | wouldn't that be inconsistent if the user opted to build with the system copy of pjsip |
| 17:48.23 | file | we submit any changes upstream |
| 17:48.28 | scv | oh, i see what you mean |
| 17:48.52 | file | the only patches we have in the bundled stuff are changes which are not yet in a released version of PJSIP |
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| 17:51.07 | scv | in the interim that should be fine then, anything in patches/ is applied from what i see so i'll just drop a fix in there |
| 17:51.13 | scv | works for my internal builds i guess |
| 17:51.38 | scv | if the patch works properly i'll put it up on that ticket |
| 17:52.48 | file | kk |
| 17:53.58 | scv | of course this modem/router has no ALG toggle |
| 17:53.59 | scv | -_- |
| 18:01.05 | hdon | file, thanks. that doesn't seem terribly critical to our operation. i'm tempted to try this.. |
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| 18:01.18 | file | hmm? |
| 18:01.25 | hdon | file, putting astdb in a tempfs |
| 18:01.26 | hdon | tmpfs |
| 18:01.28 | file | ah |
| 18:02.07 | hdon | or maybe asking asterisk to sync the sqlite db to disk less aggressively |
| 18:03.13 | scv | we're seeing a lot of syncs to this file and we're thinking of just moving someplace ephemeral and in RAM only |
| 18:03.14 | scv | er |
| 18:03.30 | scv | hdon: we just recently switched to that technique |
| 18:03.45 | scv | astdb is in a tmpfs and we sync it to persistent storage every 10 minutes |
| 18:04.22 | scv | the reason behind that is to improve registration speed however, since there's a single lock taken on sqlite and REGISTERs will block until that lock is free |
| 18:06.03 | scv | i dont think its an issue under pjsip though |
| 18:06.16 | scv | wont be a concern once i get rid of the last of chan_sip |
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| 18:17.53 | hdon | scv, yeah i'm a little worried about locking during the copy blocking asterisk, too. but we'll play around with it and see what's acceptable. so far, it seems like if we lost the astdb during a power failure or something, it wouldn't be a big deal. |
| 18:26.57 | scv | hdon: the copy wouldn't lock the file |
| 18:27.14 | scv | the biggest issue you'd need to worry about would be copying while its being synced |
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| 18:29.58 | Echo6 | Hey guys! Can I receive calls for multiple DIDs using IP based registration instead of account based? |
| 18:30.59 | SpaceInvaders | as the new guy here, I think you may need to add some additional detail >:) |
| 18:31.40 | Echo6 | I have a provider that wants to route my calls via IP instead of account registration. |
| 18:32.19 | Echo6 | My concern is that if I have multiple PBX's behind the same IP the calls will only go to one of the systems |
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| 18:58.16 | DivideBy0 | Echo6: you can usually specify a port with the ip, so pbx0 is forwareded on port 5060, pbx1 is 5061, etc |
| 18:58.41 | DivideBy0 | depends on provider, but that's how I've gotten around it |
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| 19:00.03 | scv | ah grandstream |
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| 19:01.18 | hdon | scv, a copy that doens't lock the database is pretty useless |
| 19:01.26 | hdon | scv, unless you have atomic file copy |
| 19:02.01 | scv | "sip re-register before expiration (in seconds)" actually means send register at interval+before time |
| 19:02.53 | HRH_H_Crab | hi all, im very new to voip / sip / asterisk but ive been making some good progress. ive got a slightly odd network configuration, i think pretty similar to the user in this bug report: https://issues.asterisk.org/jira/browse/ASTERISK-22564 |
| 19:03.04 | HRH_H_Crab | i suspect it may be possible that i am seeing the same issue that he reported. |
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| 19:03.47 | HRH_H_Crab | i have the symptoms that he reports: incoming calls to my asterisk server work perfectly (with directmedia=no) |
| 19:04.20 | HRH_H_Crab | outgoing calls do not have any inbound audio! the person i call can hear me fine, i cannot hear them. |
| 19:04.37 | scv | hdon: you could use ami to trigger an exclusive lock by sending the query directly |
| 19:04.42 | HRH_H_Crab | what im wondering is how to determine from his debug where the problem is happening - it doesnt make sense to me. |
| 19:05.35 | HRH_H_Crab | (im trying to work out whether my issue *really* is the same as his) |
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| 19:07.27 | HRH_H_Crab | as an aside i did a packet trace, and it looks like all the audio does get from my sip provider (ipv6) to my asterisk server. |
| 19:07.42 | HRH_H_Crab | its just that it is not then properly bridged to my ipv4 handset |
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| 19:36.38 | shido6 | DivideBy0 - you could use ports - or you can use a Session Border Controller or and EdgeMarc or a kamailio instance |
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| 19:53.18 | hdon | scv, well, the .backup command isn't a query. i don't think you can send it over the query API. i think it's only an sqlite3 shell command. there are C APIs for sqlite3 to do similar things that are probably used in the .backup command impl |
| 19:53.30 | hdon | scv, if i'm understanding you right |
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| 20:47.24 | rcp | hi guys, how can I ban a number |
| 20:47.32 | rcp | block it from calling me. |
| 20:48.11 | rcp | I tried this database put blockcaller number-here 1 |
| 20:48.17 | rcp | but it didnt seem to block them |
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| 20:54.29 | rcp | do I need to restart asterisk once I add a number to it? |
| 20:54.41 | robmal | dialplan reload |
| 20:54.45 | shido6 | no |
| 20:54.52 | shido6 | what robmal said. |
| 20:55.05 | rcp | if I restart it, does that do the same? reload dialplan? |
| 20:55.17 | shido6 | yes but it tears down existing calls, too |
| 20:55.36 | rcp | dialplan reload gives me loads of options |
| 20:56.51 | rcp | ive just restarted |
| 20:56.52 | rcp | thansk |
| 20:56.57 | rcp | sorted I hope. |
| 21:00.40 | hdon | rcp, what options did it give you? |
| 21:01.06 | hdon | rcp, "dialplan reload" command will usually spit out a lot of information as it loads the dialplan, but it's straightforward. you don't have to make choices or enter new information to reload the dialplan. |
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| 22:13.30 | SpaceInvaders | How do I make this work? testvar="this is line 1\nThis is line 2" ? |
| 22:14.16 | SpaceInvaders | The only thing I can find in the docs on esc seq is on pattern matching and doesn't seem to apply |
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| 22:25.02 | SpaceInvaders | I don't understand why dialplan functions is missing function_strings |
| 22:29.09 | [TK]D-Fender | ? |
| 22:29.36 | [TK]D-Fender | * doesn't do line breaks and fancy things |
| 22:31.36 | SpaceInvaders | will \n do a line break? |
| 22:31.59 | SpaceInvaders | oh |
| 22:32.00 | SpaceInvaders | got it |
| 22:32.29 | SpaceInvaders | Thank you :-) |
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