| 00:06.11 | ruben23 | lvlinux: but its already registered... ---> 887643/887643 xxx.xxx.xxx.yyyy D N A 65476 OK (35 ms) |
| 00:07.54 | ruben23 | but the weird part when i check sip show registry no one is registered but the logs displays on sip show peers.. |
| 00:08.27 | ruben23 | display this on sip show peers ---> 887643/887643 xxx.xxx.xxx.yyyy D N A 65476 OK (35 ms) |
| 00:16.53 | [TK]D-Fender | ruben23, Meaningless. |
| 00:17.26 | [TK]D-Fender | ruben23, The fact you have registered has no impact on the call failing its auth. |
| 00:17.55 | [TK]D-Fender | ruben23, And what you have shown is a PEER QUALIFY RESPONSE.... not prrof of "registration" |
| 00:18.35 | [TK]D-Fender | ruben23, That "OK" It gives you just says the other side is RESPONDING to *'s SIP OPTIONS requests. This proves absolutely nothing about your having set any kind of auth up right at all. |
| 00:19.09 | ruben23 | [TK]D-Fender: ok so the server itself is not registered yet.then causing this inbound issue..coz my dialout is working so far |
| 00:20.49 | [TK]D-Fender | <ruben23> [TK]D-Fender: ok so the server itself is not registered yet. <- I have NOT said this. |
| 00:21.15 | [TK]D-Fender | ruben23, I have said what you showed us was not what you claimed it to be and that you have provided no proof. |
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| 03:46.28 | hebber | Anyone know why asterisk would do a SIP CANCEL after the other replies with 183 Session Progress? |
| 04:00.11 | *** join/#asterisk Vutral (WoV54MVnZM@mirbsd/special/Vutral) |
| 04:07.46 | Penguin | The caller hung up. |
| 04:09.51 | hebber | Penguin: no the caller is still active on the event |
| 04:11.41 | WIMPy | Show us |
| 04:12.09 | Penguin | Is it reproducible? |
| 04:12.09 | hebber | a moment |
| 04:12.35 | hebber | happens every single time |
| 04:13.50 | hebber | Phone -> asterisk1 -> asterisk2 -> provider (reason for Asterisk2 is network not accessible for asterisk1) |
| 04:14.56 | WIMPy | And who starts to cancel? |
| 04:15.20 | hebber | phone |
| 04:15.23 | hebber | http://pastebin.com/7UaSDzJs |
| 04:15.30 | hebber | no |
| 04:15.40 | hebber | For me it seems to be asterisk2 |
| 04:17.32 | WIMPy | Looks like no licence to me. |
| 04:18.06 | hebber | really? what kind of licence? g729? |
| 04:18.22 | WIMPy | yes |
| 04:18.27 | hebber | and how did you spot that if thats the case? |
| 04:18.46 | WIMPy | One call is G.711 and the othr G.729. |
| 04:19.18 | hebber | and server2 doesn't have licence - doh - only server one have a Digium card |
| 04:23.22 | hebber | YES!, got a step further - provider are now accepting calls - thank you for helping me out |
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| 08:10.21 | andycol | hi guys |
| 08:11.02 | andycol | has anyone got a solution for if i have multiple sip trunks all coming from the same ip, what happends is all incoming calls just go over the first trunk |
| 08:11.26 | Penguin | Define "sip trunk." |
| 08:14.43 | andycol | so i have server A which is my core server which is connecting to server b which has multiple trunks all registering to Server A, incoming calls are coming from server A to B but they seem to all go via the first registered trunk |
| 08:19.14 | andycol | any ideas? |
| 08:19.47 | Penguin | I was still waiting on the definition. |
| 08:20.02 | andycol | sip trunk meaning connection between 2 sip server |
| 08:20.02 | Penguin | Anyway... |
| 08:20.03 | andycol | sip trunk meaning connection between 2 sip servers |
| 08:21.34 | Penguin | It seems like what you really mean is that you have a couple SIP registrations from one client (server B) to a server (server A) and a couple peers defined on each for each other... |
| 08:21.45 | Penguin | How are you matching calls? Are you using type=peer for the peer entries? |
| 08:22.10 | andycol | yes type=peer is correct |
| 08:22.24 | andycol | do u want to see an example of a trunk |
| 08:22.25 | Penguin | That's why it matches the first one every time. |
| 08:22.36 | Penguin | I know what trunks are, and SIP doesn't have them. |
| 08:22.44 | Penguin | Common misconception. |
| 08:23.00 | Penguin | But the reason is because of how type=peer does matching. |
| 08:23.05 | Penguin | It matches IP/port. |
| 08:23.17 | andycol | ok so what should i change type= to |
| 08:23.28 | Penguin | If you want to match a different peer definition, use type friend and specify the defaultuser. |
| 08:23.41 | andycol | where do i specify defaultuser? |
| 08:23.56 | Penguin | Within the peer definition, of course. |
| 08:24.10 | andycol | can u show me an example |
| 08:24.35 | Penguin | [serverA] |
| 08:24.40 | Penguin | defaultuser=serverB |
| 08:24.49 | Penguin | host=serverA.domain.com |
| 08:24.55 | Penguin | type=friend |
| 08:24.59 | Penguin | et cetera |
| 08:25.18 | andycol | on server A host is set to dynamic as server B registers to A |
| 08:25.19 | Penguin | The peer entry. |
| 08:25.23 | andycol | will that work |
| 08:25.37 | Penguin | type=friend tries to match usernames. |
| 08:25.52 | Penguin | type=peer only matches IP/port and the first one will always match first. |
| 08:26.42 | andycol | is default user important, what if it is left blank? |
| 08:27.00 | Penguin | You can't match the username if you don't specify it. |
| 08:27.17 | andycol | ok and im guessing the defaultuser will be the username correct? |
| 08:27.22 | Penguin | That is correct. |
| 08:27.35 | andycol | ok let me try this :) |
| 08:27.42 | andycol | thanks will let you know shortly |
| 08:28.11 | Penguin | You have some user names specified already because you send registrations from one to the other. The register statement requires the username. |
| 08:28.20 | Penguin | So you're part of the way there already. |
| 08:28.34 | Penguin | Just add defaultuser and change type to friend. |
| 08:28.48 | andycol | im changing it to friend on both servers? |
| 08:28.56 | andycol | and defaultuser on both servers as well? |
| 08:30.34 | Penguin | If all calls are going from serverA to B, and B matches the first peer entry every time... |
| 08:30.44 | Penguin | You can just set type=friend on B only. |
| 08:31.00 | Penguin | But set the defaultuser on A. |
| 08:31.15 | andycol | ok let me test that |
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| 08:35.12 | andycol | will it only work once i have done it on all the trunks? |
| 08:35.31 | andycol | u set it up on one of the trunks but still comes in on the first trunk |
| 08:36.13 | andycol | can i show u one of my trunk definitions? |
| 08:36.22 | Penguin | ~pb |
| 08:36.22 | 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. |
| 08:36.38 | andycol | http://pastebin.com/sSnth58L |
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| 08:37.24 | Penguin | Remove the insecure specification. |
| 08:37.37 | andycol | ok |
| 08:38.15 | andycol | k done let me test |
| 08:38.25 | andycol | now i get |
| 08:38.34 | andycol | username mismatch, have <123>, digest has <234> |
| 08:39.05 | Penguin | Yeah. You've got the usernames in the wrong place. |
| 08:39.17 | Penguin | The defaultuser has to match the OTHER SIDE. |
| 08:39.33 | andycol | sorry im confused where should username be? |
| 08:39.41 | Penguin | So if server A has a [serverB] section in it... |
| 08:39.57 | Penguin | Server A's peer entry has to say defaultuser=serverB |
| 08:40.22 | Penguin | Wait, I think I said that wrong. |
| 08:40.29 | andycol | ok |
| 08:40.40 | Penguin | In server A |
| 08:40.52 | Penguin | You have [serverB] section |
| 08:40.58 | Penguin | yes? |
| 08:41.25 | Penguin | In server B, you have [serverA] section. |
| 08:41.28 | andycol | where would i have a server B section? |
| 08:41.40 | andycol | under host? |
| 08:41.47 | Penguin | Server A has to send defaultuser=serverA to the other side. |
| 08:42.09 | Penguin | The defaultuser defined on one side has to be the user name configured in the other server. |
| 08:42.22 | andycol | yes that is right |
| 08:42.43 | andycol | username is the same on both sides as thats how they authenticate |
| 08:42.53 | Penguin | http://pastebin.com/Ag7tknm2 |
| 08:43.20 | Penguin | See how defaultuser is matching the name on the other side? |
| 08:43.28 | Penguin | You have the same username on both servers? |
| 08:43.37 | andycol | yes, i see you dont have username= |
| 08:43.50 | Penguin | username is no more |
| 08:43.55 | Penguin | It is defaultuser now. |
| 08:44.06 | andycol | ok so should i remove username on both sides and use defaultuser |
| 08:44.09 | Penguin | Yes. |
| 08:44.28 | andycol | does it matter that on both sides my [23] is the same? |
| 08:44.34 | Penguin | If the name is the same on both sides, then the defaultuser will also bt the same on both sides. |
| 08:44.41 | Penguin | It should be fine. |
| 08:44.43 | andycol | ok makes sense |
| 08:44.47 | andycol | let me test it quickly |
| 08:44.49 | andycol | :) |
| 08:46.39 | Penguin | If you have any still set to type=peer, they will match the IP of the host= line first. You have to change them all to type=friend if you want to match by name. |
| 08:46.53 | andycol | ok |
| 08:48.15 | Penguin | And insecure=invite means don't bother checking for user name. |
| 08:48.23 | Penguin | So you have to remove those too. |
| 08:49.17 | andycol | will this stop T38 working? |
| 08:49.36 | Penguin | I wouldn't think so. Why would it? |
| 08:49.55 | andycol | just was wondering because we thought canreinvite is what T38 needs |
| 08:50.13 | Penguin | directmedia |
| 08:50.18 | Penguin | not canreinvite |
| 08:50.36 | Penguin | canreinvite is like username -- changed. |
| 08:50.36 | andycol | oh ok |
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| 08:50.57 | Penguin | And directmedia has nothing to do with matching by name instead of IP. |
| 08:51.17 | andycol | ok |
| 08:51.41 | Penguin | directmedia just keeps asterisk in the media path after the call is set up. |
| 08:58.26 | andycol | still get username mismatch |
| 08:58.54 | *** join/#asterisk s7r_m (~s7r@openvpn/user/s7r) |
| 09:01.31 | andycol | this happens as soon as i remove insecure=invite |
| 09:01.34 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 09:02.35 | Penguin | Yeah, because you're now trying to match names. insecure=invite means "I don't care about user names." |
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| 09:03.03 | andycol | but then why does it give the username mismatch error |
| 09:03.19 | Mohsen_Hassani | Hello guys. I'm using asterisk on Debian. How can I check if asterisk is working properly? There is not asterisk in /etc/init.d/ for me. |
| 09:03.37 | Penguin | The wrong username is being sent. Why? That part I'm not sure of without seeing the call. Even then I don't know if I'll know. |
| 09:03.59 | *** part/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 09:04.57 | andycol | ok i will have to figure that one out |
| 09:05.07 | andycol | must have to do with Server A which is running a2billing |
| 09:05.55 | Penguin | I've seen that username mismatch thing before, but I don't know how to fix it. Did you restart both asterisks? |
| 09:11.46 | andycol | ok i will play with it |
| 09:11.49 | andycol | thanks for helping :) |
| 09:12.25 | andycol | cant restart the asterisk on ServerA during the day has 10000 concurrent calls |
| 09:12.36 | Penguin | Yeah, that would be a problem. |
| 09:13.05 | andycol | yea :) |
| 09:13.35 | Penguin | There's always core restart when convenient or core restart gracefully. Be careful, though. |
| 09:14.26 | andycol | yea unfortunately i have to do it after 11 at night :S |
| 09:15.13 | Penguin | The time is 0315 here! Perfect for rebooting without any screaming. |
| 09:16.53 | andycol | ur lucky here its 11:16 am :S |
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| 10:30.32 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.1 (2014/11/20), 11.14.1 (2014/11/20), 1.8.32.1 (2014/11/20); Standard: 12.7.1 (2014/11/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 12:10.36 | tom486 | hello im currently receiving the following with some sip registrations: |
| 12:10.57 | tom486 | Contact: <sip:username@180.215.214.19:42744;transport=UDP;ob> |
| 12:10.57 | tom486 | Contact: <sip:username@01261334I4:42744;transport=UDP;ob>;expires=0 |
| 12:10.57 | tom486 | Expires: 900 |
| 12:10.57 | tom486 | Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS |
| 12:11.18 | tom486 | the second contact string has a corrupt address causing the asterisk box to jam sip registrations |
| 12:11.38 | tom486 | is there any setting causing this or is there an easy way to get fail2ban to ban these types of requests |
| 12:19.55 | wdoekes | tom486: install a local dns cache, e.g. dnsmasq |
| 12:20.37 | wdoekes | (put interface=lo in the config, so you won't be a ddos amplifier) |
| 12:24.28 | tom486 | ok will give it a go |
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| 12:32.25 | tom486 | @wdoekes so i just apt-get install dnsmasq then update /etc/dnsmasq.conf to have interface=lo ??? |
| 12:33.59 | wdoekes | yea, or put a file custom.conf (or interface.conf) in /etc/dnsmasq.d/ |
| 12:34.06 | wdoekes | (which allows for easier upgrades) |
| 12:34.56 | tom486 | so this basically doesnt forward these bad packets? |
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| 12:37.19 | tom486 | @wdoekes n e ways thanks it seems to be working. |
| 12:38.23 | wdoekes | no, dnsmasq will speed up the resolving |
| 12:38.37 | wdoekes | so you won't be *as* affected |
| 12:40.05 | wdoekes | if this is a register flood, you can consider using fail2ban or iptables hashlimit |
| 12:40.45 | tom486 | it seems to be that a particular make of wifi key malforms the sip register packet |
| 12:41.21 | tom486 | causing the invalid host <sip:username@01261334I4:42744;transport=UDP;ob>; |
| 12:42.21 | tom486 | the ipaddress is in the line prior to the malformed packet, can fail2ban still pick this up? |
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| 12:48.48 | wdoekes | not sure what you want, nor what you mean with "asterisk .. jam[s] sip registrations" |
| 12:49.35 | wdoekes | supplying two contacts is valid per RFC, in this case the 2nd one is an "unregistration" |
| 12:49.54 | wdoekes | not sure what asterisk chan_sip does with that though |
| 12:50.03 | tom486 | its the second one that is hang |
| 12:50.26 | tom486 | as cannot resolve 1261334I4:42744 |
| 12:53.04 | wdoekes | yes, but that's a "temporary" hang, assuming your resolver times out within a short enough time |
| 12:53.21 | wdoekes | see: man 5 resolv.conf # for options for timeout |
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| 12:58.59 | tom486 | thanks will give that a go |
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| 14:28.55 | Katty | drmessano: GUESS WHO"S BIRTHDAY IT IS TODAY |
| 14:34.41 | tomodachi | zapatas? |
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| 15:04.53 | Penguin | Who is birthday? |
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| 16:13.46 | WIMPy | Hmm. Is Queue() so clever that it doesn't count calls that ended because of network issues as abandoned? |
| 16:14.15 | WIMPy | The statistics at a site that had massive networking issues today don't add up. |
| 16:16.03 | skrusty | is there an event for those calls leaving the queue? |
| 16:16.31 | WIMPy | Good question. Let me check... |
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| 16:18.03 | WIMPy | No. That doesn't seem to be the case. |
| 16:19.01 | WIMPy | I use QueueLog in the h extensions and those logs come in bigger numbers than COMPLETE* or ABANDON entries. |
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| 16:22.25 | pabelanger | WIMPy, Ya, Queue() has an odd definition of abandoned. |
| 16:22.48 | WIMPy | Odd? That actually seems pretty clever. |
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| 17:39.44 | stefan27 | when a sip peer has wss as transport, is there any way for configure asterisk so that it detects whenever such a sip peer loses its connection and when it does disconnect any calls it has? (otherwise when the sip peer loses power (cant send BYE) i have to wait rtptimeout seconds for call to die) |
| 17:40.46 | WIMPy | What's wrong with rtptimeout? |
| 17:43.14 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:43.37 | mjordan | That's actually what is appropriate about rtptimeout. Just because a websocket has broken doesn't mean that the 'call' itself is dead. |
| 17:43.55 | mjordan | media does not flow over websockets :-) |
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| 17:48.03 | stefan27 | right, but i never witnessed websocket breaking without media breaking too, and the same web application that has the webrtc client also queries asterisk for active calls and if user reloads page it gets wrong info for rtptimeout=15 seconds |
| 17:48.15 | stefan27 | im scared to lower it further |
| 17:48.31 | mjordan | well, then, the answer is "no". |
| 17:48.55 | WIMPy | Why? I have it set to 9. That should be enough. |
| 17:49.26 | WIMPy | (note that it's off by one) |
| 17:49.54 | stefan27 | i dont know how input devices or networks can behave |
| 17:50.04 | stefan27 | default value was 60 i think |
| 17:50.38 | WIMPy | Who is going to wait for 61 seconds for audio to re-appear? |
| 17:52.11 | stefan27 | what happens if you mute microphone? |
| 17:52.39 | WIMPy | That doesn't stop RTP. |
| 17:53.01 | WIMPy | Putting a call on old could. But you still should get at least keep-alives. |
| 17:53.48 | stefan27 | I'll try setting it to 5 seconds and see if something funny ever happens |
| 17:54.02 | stefan27 | chees |
| 17:54.03 | stefan27 | cheers |
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| 18:48.54 | Milenco | lvlinux, I wasnt interested in Videocalling, I was helping 'okdamn' :) |
| 18:56.15 | Penguin | OK. Damn! |
| 19:10.06 | Milenco | :P |
| 19:11.34 | file | I'd like to know how prevalent video calling actually is... |
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| 19:20.48 | Milenco | me too, little experience with it |
| 19:21.08 | Milenco | got it setup once but only got 320x240 videostreams with no working option to change it |
| 19:21.20 | Milenco | but didnt look into it much |
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| 19:27.04 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@217.147.53.97) |
| 19:28.18 | [ProB]CrazyMan | Hi I have an problem, my asterisk box had tonigt an powerloss, and now I cannot get it to run back, I use misdn with LCR, LCR receive the calls, but they get not transferded to asterisk |
| 19:30.11 | Milenco | what do your logs say? |
| 19:31.12 | WIMPy | Is the channel connected? Do you see Asterisk in lcradmin state? |
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| 19:33.23 | [ProB]CrazyMan | http://pastebin.com/5P6ikqbF |
| 19:34.02 | [ProB]CrazyMan | where do I see asterisk in lcradmin? |
| 19:34.05 | WIMPy | Looks like it isn't. |
| 19:34.26 | WIMPy | Last in the interface section. |
| 19:34.46 | WIMPy | Either chan_lcr isn't loaded or can't connect. |
| 19:35.00 | [ProB]CrazyMan | I#m using LCR 1.12 |
| 19:35.39 | [ProB]CrazyMan | what do I have to start that lcr is wworking ? |
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| 19:36.05 | WIMPy | Is chan_lcr loaded? |
| 19:36.14 | [ProB]CrazyMan | and what kernel moduls must be loaded |
| 19:36.29 | WIMPy | In Asterisk. |
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| 19:39.10 | [ProB]CrazyMan | chan_lcr is loaded us count =0 |
| 19:39.24 | [ProB]CrazyMan | use count == null |
| 19:39.36 | WIMPy | What about permissions? |
| 19:41.34 | [ProB]CrazyMan | permissions from what? |
| 19:41.46 | WIMPy | LCRs socket. |
| 19:42.27 | [ProB]CrazyMan | ?? |
| 19:42.53 | WIMPy | Have you ever looked in to the config? |
| 19:43.12 | WIMPy | Try 'core set debug 2 chan_lcr' |
| 19:44.55 | [ProB]CrazyMan | doesnt get any output |
| 19:45.38 | WIMPy | And it is not connected? |
| 19:46.18 | [ProB]CrazyMan | how di I see its connected? |
| 19:46.56 | [ProB]CrazyMan | in lcradmin i dont see the state to asterisk |
| 19:48.58 | WIMPy | If you don't see "Remote: asterisk" it isn't connected. |
| 19:49.17 | WIMPy | If it isn't connected you should see it trying to connect in the *CLI. |
| 19:50.22 | WIMPy | So if neither is the case, maybe you should try to unload and load the channel. |
| 19:53.00 | WIMPy | It's about finishing time for me. So if you have more questions, ask quick. |
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| 19:54.07 | [ProB]CrazyMan | i dont know what to do |
| 19:55.40 | [ProB]CrazyMan | I do not get any error when loading the module |
| 19:56.10 | WIMPy | No output, even with debug on? |
| 19:57.31 | [ProB]CrazyMan | no |
| 19:59.38 | WIMPy | No idea then. I can only assume it's not really running or not starting its reconnect timer. |
| 20:00.26 | [ProB]CrazyMan | do I need to start sth, on shell |
| 20:00.31 | [ProB]CrazyMan | misdn ? |
| 20:00.43 | WIMPy | no |
| 20:01.08 | [ProB]CrazyMan | funny |
| 20:01.12 | WIMPy | That seems to be running for what I see in your log snippet. |
| 20:01.22 | [ProB]CrazyMan | I restarted asterisk agan |
| 20:01.26 | [ProB]CrazyMan | now it wors |
| 20:01.33 | [ProB]CrazyMan | currios |
| 20:02.41 | [ProB]CrazyMan | thx WIMpy for your time |
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| 20:03.10 | WIMPy | Good night then. |
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| 21:32.28 | PHunter | I have an odd Question.. while I know that I can get Channel Vars through AGI, is there a way to pass channel variables from my dialplan to an AMI Event instead? |
| 21:33.33 | PHunter | Say, when the Dial Event is triggered, pass a variable 'outsideNum'? |
| 21:34.49 | PHunter | (Asterisk 11.14.2 btw) |
| 21:35.29 | [TK]D-Fender | SendEvent() |
| 21:35.42 | PHunter | :O! |
| 21:35.53 | PHunter | googles SendEvent constantly for next 10 minutes. |
| 21:35.54 | [TK]D-Fender | You really should read *'s application and function lists.... |
| 21:36.17 | PHunter | I have read a lot of AMI events, and done google searches.. I think Ill make that my next stop. Thank you. |
| 21:36.23 | Penguin | core show applications like event ? |
| 21:37.02 | PHunter | I honestly didn't think there was but I am clearly wrong.. so thanks again! |
| 21:37.03 | [TK]D-Fender | HMMM |
| 21:37.18 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_UserEvent |
| 21:37.23 | [TK]D-Fender | UserEvent rather |
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| 21:46.12 | [TK]D-Fender | Checkout time here at the office. Heading home. BBL |
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| 22:40.51 | jpsharp | I'm having a brainfart moment. How do I prevent Asterisk from sending out 183/early media no matter what I get back from the termination provider? |
| 22:41.29 | *** join/#asterisk [1]Kiwi (~Kiwi@ip-118-90-67-126.xdsl.xnet.co.nz) |
| 22:42.24 | [1]Kiwi | Hi |
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| 22:46.03 | [1]Kiwi | can anyone tell me an easy way to have call forwarding enabled that will take the call and forward it out to a mobile with out ringing the internal phones it would normally ring. I was getting the user to just use the call foward feature on the hardphone its self but this still rings other phones as well as mobile |
| 22:47.00 | jpsharp | You have a dial string with multiple devices? |
| 22:47.01 | [TK]D-Fender | jpsharp, progressinband=no <- IIRC |
| 22:47.25 | [TK]D-Fender | jpsharp, Otherwise * forwards early media I suppose if the calling end has it to begin with. You should be able to prevent this by forcibly answering the initial channel before issuing the dial. This would have CDR and billing consequences however |
| 22:47.55 | [TK]D-Fender | [1]Kiwi, It's your dialplan... you dial whatever you want whenever you want. |
| 22:49.03 | [1]Kiwi | yeah i need to figure out how I can make a simple solution |
| 22:50.18 | [1]Kiwi | jpsharp yes I do have a dial string with multi devices |
| 22:50.25 | [TK]D-Fender | [1]Kiwi, lookup some flag you've set to indicate that you shouldn't call the device you're about to. If it is set then dial the other place instead. |
| 22:50.37 | jpsharp | Then you'll need to forward the call before you hit that dialstring. |
| 22:50.41 | [TK]D-Fender | <[1]Kiwi> jpsharp yes I do have a dial string with multi devices <- clearly not what you should be doing |
| 22:51.13 | [TK]D-Fender | [1]Kiwi, Check a flag to see if you should skip your "Plan A", and act on it |
| 22:54.37 | [1]Kiwi | could I use the hadrphones call forward feature ( which the user likes) but have this phone at the top of the prioritys |
| 22:55.11 | [TK]D-Fender | Once you hit a phone-based forward your dialplan gets 100% redirected |
| 22:55.40 | [1]Kiwi | do you thinnk thats an ok solution ? |
| 22:55.51 | [1]Kiwi | or a bit rubbish ? |
| 22:55.59 | jpsharp | I have progressinband=no and Asterisk still sends out 183 + early media |
| 22:59.06 | [TK]D-Fender | jpsharp, You've seem my "Plan B" |
| 22:59.09 | [TK]D-Fender | seen* |
| 22:59.25 | jpsharp | I have? |
| 22:59.33 | [TK]D-Fender | [1]Kiwi, Phone-based can't be reversed by an admin without direct access to the phone. This is not ideal. |
| 22:59.37 | [TK]D-Fender | (often) |
| 23:03.16 | [1]Kiwi | ok i wont do it this way then are there any examples of dialplan you can send me a link to so I can do some learning about how to do this |
| 23:04.39 | [TK]D-Fender | [1]Kiwi, "core show function DB", "core show application gotoit" |
| 23:04.52 | [1]Kiwi | thank s |
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| 23:15.07 | [1]Kiwi | [TK]D-Fender so my end result might look like this to the user - dial 49 to enable call fwd and 50 to disbale it ? |
| 23:16.09 | [TK]D-Fender | Something like that |
| 23:16.15 | [1]Kiwi | cool |
| 23:34.46 | [TK]D-Fender | heads out for the evening... |
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