| 00:21.01 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 00:21.20 | nny | looking for some polycom help/advice. I have phones registering and working with a pretty normal config. I want to change them so they register via A Name (DynamicDNS set to IP(s) and reregister the the same domain on failure or rereg. |
| 00:21.46 | nny | I have them registering via the domain but when I reboot them on a different network they still try to access the old IP |
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| 02:05.00 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 02:16.30 | lex2 | I give up, I can't figure this out |
| 02:16.40 | lex2 | All settings seem to be correct |
| 02:16.52 | lex2 | Still getting no registration |
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| 02:47.07 | Penguin | lex2: Pastebin your sip config including the register statement. Don't mask anything but your passwords. |
| 02:48.52 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:51.59 | lex2 | Penguin: I'm using the web UI |
| 02:52.14 | Penguin | Then you're asking/commenting in the wrong channel. |
| 02:52.23 | lex2 | Just a sec |
| 02:53.56 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:54.28 | lex2 | Penguin: the sip config seems to be OK, it's just a routes issue |
| 02:54.48 | Penguin | No it isn't. If you can't register, it's a sip config problem. |
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| 02:55.55 | lex2 | sip show peers shows as online |
| 02:56.08 | lex2 | sip show registry shows 0 |
| 02:56.17 | Penguin | Does sip show registry show that it is registered? |
| 02:56.31 | Penguin | Then it isn't registered. |
| 02:56.32 | lex2 | 0 sip registrations |
| 02:56.56 | lex2 | What's not registered, my softphone to the server or the trunk to the server? |
| 02:57.21 | Penguin | There's no trunk. There's only asterisk and peers. |
| 02:57.45 | Penguin | sip show registry shows registrations from your asterisk TO other peers. |
| 02:58.13 | lex2 | wiretap*CLI> sip show peers |
| 02:58.14 | lex2 | Name/username Host Dyn Forcerpor |
| 02:58.14 | lex2 | t ACL Port Status |
| 02:58.14 | lex2 | vitel-inbound/shaf_wireta 66.241.99.208 N 5 |
| 02:58.14 | lex2 | 060 Unmonitored |
| 02:58.14 | lex2 | vitel-outbound/shaf_wiret 64.2.142.215 N 5 |
| 02:58.14 | lex2 | 060 Unmonitored |
| 02:58.15 | lex2 | 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] |
| 02:58.24 | Penguin | ~pb |
| 02:58.24 | 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. |
| 02:58.29 | Penguin | pastebin |
| 02:58.37 | lex2 | Whoops |
| 02:58.42 | lex2 | sorry :?( |
| 02:58.46 | Penguin | If your asterisk is supposed to register to another place, sip show registry is what shows those. |
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| 02:59.40 | lex2 | Any clue what's misconfigured? |
| 02:59.49 | Penguin | I'm still waiting to see something useful. |
| 03:00.00 | Penguin | sip debug would be useful. |
| 03:00.11 | lex2 | no such command |
| 03:00.23 | Penguin | Does vitelity even show up at all in sip show registry? |
| 03:00.37 | lex2 | no |
| 03:01.05 | Penguin | If asterisk is configured to register, it would show it there and show that it is not registered or registered. You did say it wasn't registered, but it would still show up as not registered. |
| 03:01.43 | lex2 | Can I screenshot? |
| 03:02.02 | Penguin | That's not really necessary. |
| 03:02.44 | Penguin | If you ask your question in an asterisk channel, you'll get an asterisk answer. If you want help with configuration by GUI, you've come to the wrong place. |
| 03:03.01 | lex2 | OK |
| 03:03.12 | lex2 | Thanks anyway, I'll wait in the other chan |
| 03:03.16 | Penguin | Check your sip.conf to see if there is a register statement present at all. |
| 03:04.41 | Penguin | It would look like: register => username:password@inbound16.vitelity.net |
| 03:10.06 | lex2 | Penguin: http://pastebin.com/wCMtXVyK |
| 03:12.45 | Penguin | #include sip_registrations.conf |
| 03:12.46 | Penguin | the GUI must have put your registration in that file. Check it for the register statement. |
| 03:12.46 | Penguin | If it isn't there, you probably didn't configure the registration for inbound calls. |
| 03:12.46 | Penguin | I don't know anything about FreePBX, so I can't tell you how to do that. I can only tell you from the asterisk standpoint. |
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| 03:15.08 | lex2 | Penguin: http://pastebin.com/UWKTwwsd |
| 03:17.48 | Penguin | That seems to indicate the problem is that you didn't configure registration for inbound calls. |
| 03:18.12 | lex2 | I only want to make outbound calls |
| 03:18.26 | lex2 | I'll do inbound calls later when I get familiar with this |
| 03:18.30 | Penguin | They probably require registration before they'll allow outbound. |
| 03:18.56 | lex2 | ok |
| 03:18.59 | Penguin | It's common to require registration first. It's a sort of pre-authentication mechanism. |
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| 03:20.19 | lex2 | weird. I entered the details in the web ui but it didn't update the config |
| 03:20.46 | Penguin | I think there's an 'apply' button. You pressed it? |
| 03:20.54 | lex2 | ubmit changes, yes |
| 03:20.57 | lex2 | *submit |
| 03:21.05 | Penguin | I see. |
| 03:23.10 | Penguin | It's hard to believe there is no one around the freepbx channel to help you with configuration. |
| 03:23.28 | lex2 | Yeah, nobody's responding |
| 03:23.43 | Penguin | They must be sleeping. |
| 03:24.05 | lex2 | maybe. Although I've been in there for at last 7 hours |
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| 03:25.20 | Penguin | It looks like you've been there for six. And seven hours ago was the last time there was activity. |
| 03:25.50 | lex2 | fair enough, will have to try tomorrow |
| 03:26.03 | lex2 | But I can't understand why it's not updating |
| 03:27.13 | Penguin | Unfortunately, I wouldn't have any idea about that. |
| 03:27.20 | lex2 | Sure |
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| 05:04.04 | fling | Hello. Are you guys doing video conferencing? |
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| 05:14.46 | fling | Is it possible to mix videostreams? |
| 05:16.19 | WIMPy | no |
| 05:17.02 | WIMPy | They can only be switched, depending on who's taling. |
| 05:17.30 | fling | Is it possilble to use something standalone for that? like ffmpeg? |
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| 05:22.03 | ChannelZ | hmm.. anyone have experience with tc on linux and ingress filters? |
| 05:26.56 | fling | WIMPy: I want something similar to this -> http://mirror.dno.so/incoming/2014.10.21-12%3a25%3a22.991988040.png |
| 05:27.10 | fling | WIMPy: when multiple incoming video streams are mixed together. |
| 05:27.29 | fling | WIMPy: and then all the people on the bridge see the mixed result. |
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| 05:40.09 | ChannelZ | nm.. seemed to be a burst problem |
| 05:45.56 | fling | Is it possible to send all the video streams to some app and then receive a resulting video from it? |
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| 07:52.32 | hrolf_ | Hi #asterisk |
| 07:52.41 | hrolf_ | Is there any command to empty all members from a queue? |
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| 08:37.10 | jepperl | Hello there. If i have two asterisk servers exchanging states and whatnot, with trunks registrered on both servers, are there any ways i can control which server the incoming calls will reach? Right now it seems random |
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| 08:40.30 | Chainsaw | jepperl: Those "incoming calls", are they SIP as defined in an SRV record? |
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| 08:51.00 | jepperl | Chainsaw: No. We use a normal DNS A record to point to our Asterisk servers which is connected directly to the internet. Each asterisk server runs an equal configuration in sip.conf where the same trunk is defined. This causes incoming calls from the sip trunk providers to be taken by either Asterisk 1 or Asterisk 2 |
| 08:53.25 | Chainsaw | jepperl: If you wish to have more sophisticated control you need SRV records. |
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| 09:02.24 | *** join/#asterisk woopstar (~woopstar@unaffiliated/woopstar) |
| 09:03.38 | woopstar | Hi guys. If you have two Asterisk servers running with a SIP Trunk connecting them. Both servers connect to the same SIP trunk provider. How would asterisk / or the sip trunk provider decide which asterisk server will receive an incoming call? |
| 09:09.09 | OrNix | maybe last registered? |
| 09:10.46 | linuxgeek | The last registered one will get the call |
| 09:13.08 | OrNix | I'm doing Queue(support,,,,60) with timeout=15, retry=5. Tell me please, how can I play once to caller sorry.alaw before hangup after timeout? |
| 09:13.35 | woopstar | Oh. So there is no way to distribute the incoming calls between the servers? |
| 09:13.56 | woopstar | I have a sip trunk from a provider, and I want both servers to handle the incoming calls? |
| 09:14.23 | OrNix | you can balance calls between servers with dialplan |
| 09:14.54 | woopstar | Yeah, but it will always be the last one registred that gets the call, and the it will balance the calls through the dialplan ? |
| 09:23.02 | Zogot | woopstar: could you not look into connecting the 2 asterisk machines together? |
| 09:23.26 | Zogot | woopstar: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html might be old docs |
| 09:23.27 | woopstar | Zogot: Why not? |
| 09:23.29 | Zogot | but gives the idea |
| 09:23.45 | Zogot | we have it here to connect the phone systems between our 2 locations |
| 09:24.05 | woopstar | And both servers connect to the same sip trunk provider? |
| 09:24.23 | Zogot | well only 1 has to, then you can use the normal dial plan stuff |
| 09:24.32 | Zogot | and it can call the other connected asterisk machine extensions |
| 09:25.13 | Zogot | machine a: extension 100 has the trunk, machine b: extension 200 no trunk. machine a has a dialplan to also call 200, will work :p |
| 09:25.17 | Zogot | stupid example |
| 09:25.50 | woopstar | but what if machine a goes down then? |
| 09:27.10 | Zogot | well then you have a bigger issue than just machine b not being able to be reached |
| 09:27.16 | hrolf_ | Hi #asterisk |
| 09:27.23 | hrolf_ | Is there any command to remove all members from a queue? |
| 09:27.45 | Zogot | woopstar: if its a common issue (machine a going down) they can always register the trunk on machine b when it does |
| 09:27.58 | woopstar | Zogot: thing is.. in my setup. If machine a goes down, machine just takes over and handles everything |
| 09:29.42 | Zogot | woopstar: so how does that work for phone provisioning? |
| 09:29.55 | woopstar | im trying to create a scalabe high available system, where two or more asterisk servers uses the same mysql database for sip friends, trunks etc... and uses the same sip trunks.. |
| 09:31.17 | woopstar | Zogot: Phones connect to either asterisk 1 or 2 using round robin dns with sticky address. So they can register on either 1 or 2. But problems is, that incoming calls is always served by one asterisk servers. And this explicit causes issues when a queue is hosted on another server than the incoming call is taken by |
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| 09:58.00 | Pernat | hi. how to anwser call from voip operator? |
| 10:01.48 | WIMPy | Same as any other call. |
| 10:03.13 | Pernat | yeah. but i want to configure asterisk to redirect calls from VoIP operator to local sip, but it even don't want to register in voip server |
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| 10:17.46 | WIMPy | ~book |
| 10:17.46 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:18.00 | WIMPy | Pernat: Did you try that ^ ? |
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| 10:25.41 | Pernat | i tried configuration from voip provider but it didn't work. asterisk cli don't say nothing, when i call voip number it says that network is busy |
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| 10:30.43 | Pernat | when i configure softphone i can anwser calls. so it is asterisk missconfiguration |
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| 10:59.19 | casdude | hi |
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| 10:59.57 | casdude | I have an issue where when dialing a number the first couple of times fail and then it eventually goes through |
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| 11:00.31 | casdude | I have an output of the log file does anyone have any ideas what could be causing it http://pastebin.com/71D9HaD5 |
| 11:01.34 | casdude | it seems like the "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17" may indicate that there was an issue |
| 11:02.12 | casdude | but what it was is fairly unclear |
| 11:02.25 | casdude | any advice would be greatly appreciated |
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| 11:15.14 | casdude | hi |
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| 11:27.18 | net | if i use GotoIf($["${EXTENID}" < "10000"]?valid:end) - this is not working whereas if i use GotoIf($["${EXTENID}" < "9000"]?valid:end) it works |
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| 11:33.49 | kaldemar | net: drop the quotes when comparing numbers |
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| 11:35.32 | WIMPy | casdude: It said what the issue was: BUSY |
| 22:24.01 | *** join/#asterisk infobot (ibot@rikers.org) |
| 22:24.01 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.13.1 (2014/10/20), 1.8.31.1 (2014/10/20); Standard: Asterisk 12.6.1 (2014/10/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 22:46.15 | MarkS- | Hello, is it possible to transfer someone to a queue without using the blindtransfer keys (idea is to use the transfer buttons on the phone and reply with nothing or with something like "queued" when queueing someone when they have called the wrong number) |
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| 23:43.22 | [TK]D-Fender | MarkS-: huh? |
| 23:50.34 | MarkS- | [TK]D-Fender: it is possible to use the blind transfer option in features.conf to transfer someone to a queue (dialplan extension), however you have to wait for someone else to answer before you can drop the connection if you don't use the blind transfer option |
| 23:51.14 | MarkS- | I want to do it without using the blind transfer option from features.conf and disconnect at the moment someone is in the dialplan/extension again (in the queue waiting for someone to answer) |
| 23:51.22 | [TK]D-Fender | You are combining too much into that sentence |
| 23:52.22 | [TK]D-Fender | You asked if you COULD do a blind transfer.. and then said you DON'T want to. |
| 23:52.43 | [TK]D-Fender | Attended has its rules, blind has its rules. |
| 23:52.46 | [TK]D-Fender | Take your pick |
