| 00:28.48 | *** join/#asterisk binary110 (50002375@gateway/web/freenode/ip.80.0.35.117) |
| 00:29.42 | binary110 | hey guys hoping someone could help me help myself.. i have a simple Linksys SPA921 phone behind a NAT (no port forwarding or dmz) and today it said "Calls forwarded".. i checked the phone's web-admin and it's forwarding my calls to an Israli mobile number (what!!!) |
| 00:37.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:47.25 | binary110 | i think it's related to me configuring a STUN server on my phone |
| 00:47.32 | binary110 | certainly since then |
| 01:08.06 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 01:10.44 | Juggie | sounds hacked |
| 01:11.46 | binary110 | yeah for sure.. my port 80 on public ip loaded my phone's web-admin without any password |
| 01:12.02 | binary110 | the only thing i've changed lately was add a STUN server to the phone settings |
| 01:12.19 | Juggie | id reset it to defaults, flash a good firmware |
| 01:12.29 | Juggie | and start over with the phone firewalled |
| 01:12.34 | Juggie | not on a public ip |
| 01:12.47 | binary110 | i will do, thanks |
| 01:13.01 | binary110 | but what confuses me.. is phone is behind NAT.. no DMZ.. no port forwarding |
| 01:13.32 | binary110 | i've unplugged my phone now but port 80 loaded the phone's webadmin from the internet side |
| 01:13.32 | Juggie | then it makes no sense why port 80 on your public ip would load your phone admin interface |
| 01:13.38 | binary110 | exactly! |
| 01:13.44 | binary110 | i'm so stumped |
| 01:13.59 | binary110 | my only suspicion in STUN |
| 01:14.14 | binary110 | STUN is new to me but i didnt think it does it would do that |
| 01:14.30 | binary110 | or would* |
| 01:14.36 | Juggie | im not famaliar with stun really but that would be a suprise |
| 01:15.27 | binary110 | i know upnp can open ports..but not stun |
| 01:16.37 | Juggie | well not sure if this is an accident on your part (or not) |
| 01:16.47 | Juggie | but i would investigate the router as well |
| 01:18.01 | binary110 | will do, thanks |
| 01:18.06 | binary110 | it's an ISP-supplied one |
| 01:18.15 | binary110 | but i have to use it, annoyingly, as it's a cable router |
| 01:19.40 | Juggie | most isp supplied stuff support bridge mode |
| 01:19.45 | Juggie | and then you can use your own |
| 01:38.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:43.27 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 02:06.37 | michoelc | binary110: I would check that your router is not vulnerable to the rom-0 attach - http://rootatnasro.wordpress.com/2014/01/11/how-i-saved-your-a-from-the-zynos-rom-0-attack-full-disclosure/ |
| 02:07.46 | michoelc | binary110: if so someone could easily get your password and set up port forwarding |
| 02:21.59 | *** join/#asterisk Naikrovek (3252a0b4@gateway/web/cgi-irc/kiwiirc.com/ip.50.82.160.180) |
| 02:45.48 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 02:50.29 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 02:55.04 | *** join/#asterisk joesmoe (~joesmoe@c-73-181-72-237.hsd1.co.comcast.net) |
| 03:03.40 | *** join/#asterisk D30 (~deo@203.177.9.115) |
| 03:38.27 | *** join/#asterisk mushroomed (~mushroome@li173-111.members.linode.com) |
| 03:48.51 | *** join/#asterisk D30 (~deo@203.177.9.115) |
| 03:57.46 | *** join/#asterisk joesmoe (~joesmoe@2601:1:b200:da9:211:32ff:fe18:ef25) |
| 04:07.27 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 04:31.05 | *** join/#asterisk Omnipotent (~Josh@unaffiliated/the-intel/x-1526920) |
| 04:31.26 | Omnipotent | Is there anyway to configure asterisk for SIP client to client calls |
| 04:31.53 | ChannelZ | Asterisk isn't a proxy |
| 04:32.06 | Omnipotent | I am not going to register VOIP i.e. I am not going to call physical phones |
| 04:32.20 | Omnipotent | The way I am achieving client to client skype like calls is, giving extentions to each number |
| 04:32.24 | Omnipotent | ChannelZ, proxy? |
| 04:32.38 | Omnipotent | I have no idea, where you got that idea. |
| 04:32.48 | ChannelZ | well using Asterisk isn't predicated on having an ITSP |
| 04:33.09 | ChannelZ | It's also PBX. The P being Private |
| 04:33.17 | Omnipotent | Yep |
| 04:33.53 | Omnipotent | What I am trying to ask is, Currently, I gave my friend an extention and me one, so we have to remember those extensions to call each other (or have to add in client as contact of course) |
| 04:34.01 | Omnipotent | but, what I wanted is, "account" to account calls. |
| 04:34.04 | ChannelZ | as opposed to what |
| 04:34.22 | ChannelZ | An extension doesn't have to be numbers. |
| 04:34.36 | Omnipotent | You register accounts in SIP.conf so you should be able to say, Call ChannelZ |
| 04:34.44 | Omnipotent | similar to what skypename is |
| 04:34.57 | ChannelZ | First, separate SIP peer names from extensions, they are unrelated |
| 04:35.26 | Omnipotent | Yes? They go in SIP.conf and the latter in extensions.conf |
| 04:35.33 | ChannelZ | If you are using SIP softphones, you can use "barf" as an extension to dial if you want. |
| 04:38.14 | ChannelZ | I'm not really sure how you want it to work differently if you don't want to dial numbers or add a contact in your phone that dials something. |
| 04:38.34 | ChannelZ | (Is it supposed to read minds?) |
| 04:39.03 | Omnipotent | It is supposed to know Account names. |
| 04:39.10 | Omnipotent | That is what I mean. |
| 04:39.35 | ChannelZ | exten => joe,1,Dial(SIP/JoesDevice) |
| 04:39.36 | Omnipotent | Reading mind could be a possibility, sure, why not... |
| 04:39.56 | Omnipotent | ChannelZ, I gotcha, when you said "<ChannelZ> An extension doesn't have to be numbers." |
| 04:40.01 | ChannelZ | Hey everone, call me at sip:joe@asterisk-server.com |
| 04:40.01 | Omnipotent | Thanks a lot :) |
| 04:40.09 | ChannelZ | etc |
| 04:40.41 | Omnipotent | ChannelZ, except there is a small problem in there, in most of softphone client softwares (esp. made for smart phones) there are only dial pads |
| 04:40.49 | Omnipotent | No characters :P |
| 04:40.51 | ChannelZ | Yes, I can't help that |
| 04:40.55 | Omnipotent | Aye |
| 04:41.00 | Omnipotent | Just saying |
| 04:41.11 | Omnipotent | but thanks a lot :) |
| 04:42.18 | ChannelZ | CSipSimple for Android will let you dial by text |
| 04:43.46 | Omnipotent | aye, even Zoiper does it seems, except that you have to CLICK where it shows you the dialed numbers |
| 04:44.11 | Omnipotent | I tried using Ekiga for my desktop, it seems to have that "registering" timeout very short |
| 04:44.23 | Omnipotent | It registers, unregisters, registers, unregisters and then shows unregistered |
| 04:44.25 | Omnipotent | >_> |
| 04:44.43 | Omnipotent | If I increase that time out, the cycle repeats and if it gets on registered before the timeout, then yay |
| 04:45.22 | Omnipotent | I chose Zoiper instead for the desktop as well, but it seems to be that, I can talk from my phone zoiper to a friend, but can't from desktop zoiper |
| 04:45.25 | Omnipotent | Foo |
| 04:46.15 | ChannelZ | that's some other problem |
| 04:48.24 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 04:49.09 | Omnipotent | mm yeah |
| 04:49.13 | Omnipotent | Linphone looks promising. |
| 04:49.25 | Omnipotent | It works in current configurations at least |
| 04:49.27 | Omnipotent | neat. |
| 05:17.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 05:27.56 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 05:30.22 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 05:30.28 | *** join/#asterisk CeBe (~CeBe@port-92-206-30-149.dynamic.qsc.de) |
| 05:39.39 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 06:27.05 | *** join/#asterisk jhlavacek (~jirka@80.215.174.8) |
| 06:36.32 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 06:47.46 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:56.16 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:00.19 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:02.37 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 07:05.46 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:05.53 | Zogot | ahoyhoy |
| 07:09.59 | *** join/#asterisk D30 (~deo@203.177.9.115) |
| 07:31.41 | ChannelZ | You seem to be having trouble operating your telephone machine. |
| 07:39.17 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:2514:28b4:b98c:fc8f) |
| 07:41.04 | *** join/#asterisk joesmoe (~joesmoe@2601:1:b200:da9:211:32ff:fe18:ef25) |
| 08:04.33 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 08:07.32 | robscow | I think someone replied but I didn't have enough scrollback :( so here goes again... I'm trying to play a wav file while waiting between 100 Trying, and 180 Ringing. I can do this with Playtones, but not with Playback or Background. is this possible? I can play the sound file, but it waits until it's finished playing before trying the call (even with Background), I'm trying to fill the 2 second delay when calling out, |
| 08:07.33 | robscow | <PROTECTED> |
| 08:18.06 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:21.38 | ChannelZ | I think the closest you can come without rigging up something completely custom is to use the MOH argument of Dial |
| 08:21.57 | ChannelZ | but that plays through until the channel is answered |
| 08:26.29 | robscow | ChannelZ, thanks, i'll try that one too just to see it in action |
| 08:26.34 | kaldemar | or r, but that screws up real progress. |
| 08:28.09 | robscow | kaldemar, yeah, was using r originally but my new provider doesn't like it anyway |
| 08:35.53 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 08:43.41 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 08:44.19 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 08:45.39 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:18.57 | *** join/#asterisk sekil (~sekil@217.24.20.4) |
| 09:20.30 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:21.06 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 09:32.29 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:34.44 | *** join/#asterisk qakhan (~qakhan@39.59.97.8) |
| 09:37.31 | *** join/#asterisk Manu18 (56ddeaa8@gateway/web/freenode/ip.86.221.234.168) |
| 09:37.36 | Manu18 | Bonjour |
| 09:37.49 | Manu18 | je viens d'installer asterisk |
| 09:37.58 | Manu18 | Mon telephonneest bien enregistré |
| 09:38.13 | Manu18 | par contre j'ai un message d'erreur quand je lance un appel exterieur |
| 09:38.47 | Manu18 | chan_sip.c:20366 handle_response_invite: Received response: "Forbidden" from '"09538902xx " <sip:0953890201@mon ip>;tag=as554587eb' |
| 09:39.08 | Manu18 | Quelqu'un pourrait m'aider? |
| 09:47.59 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 09:48.36 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 09:49.52 | *** join/#asterisk hardwire (~hardwire@201-93-237-24.gci.net) |
| 09:51.30 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:07.53 | *** join/#asterisk qakhan (~qakhan@39.59.97.8) |
| 10:25.10 | *** join/#asterisk tparcina (~tomo@212.92.192.54) |
| 10:38.14 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 10:38.17 | Milos | How am I still getting these? |
| 10:38.30 | Milos | [Apr 14 22:34:37] NOTICE[484] chan_sip.c: Registration from '"14034" <sip:14034@IP:5060>' failed for '94.23.216.191:5092' - Wrong password |
| 10:38.38 | Milos | I don't have inbound 5060 open... |
| 10:42.00 | davlefou | hi, i have seen serveral syntaxe for realtime extension, like that switch => Realtime/@ or switch => Realtime/@extensions, what is the good? |
| 10:42.53 | davlefou | and that : switch => Realtime/mycontext@realtime_ext |
| 10:49.31 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:55.59 | kaldemar | davlefou: switch => Realtime/[context]@[family][/options] <-- if context is not given, the current context is used. if family is not given, "extensions" is used. |
| 10:59.56 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 11:01.23 | davlefou | and family, what is it? |
| 11:02.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:11.34 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 11:20.54 | *** part/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 11:31.49 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 11:33.59 | qakhan | when i try to make outbound call through SIP trunk i hear music and when callee accept the call we both cannot hear each other. |
| 11:34.06 | qakhan | here is my sip debug http://pastebin.com/pu9u8VPG |
| 11:42.11 | kaldemar | qakhan: what did your service provider have to say about the contact data header in SDP? |
| 11:46.44 | qakhan | kaldemar plz paste that line here i could not find it |
| 11:52.45 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 11:57.38 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 12:02.14 | kaldemar | qakhan: the same o= line i told you to ask about 4 days ago. |
| 12:02.15 | davlefou | kaldemar, is it better to user serveral switch or include? |
| 12:02.52 | kaldemar | davlefou: you be the judge. i don't use realtime extensions at all. |
| 12:03.37 | davlefou | this sort of information seems secret on internet |
| 12:05.21 | qakhan | kaldemar here is the problem. if i use same sip trunk on ther elastix it works fine with same config |
| 12:05.40 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 12:06.04 | kaldemar | qakhan: show proof of that. |
| 12:06.53 | qakhan | here http://pastebin.com/BjHr42ZS |
| 12:08.19 | *** join/#asterisk valeech (~valeech@pool-71-171-123-210.clppva.fios.verizon.net) |
| 12:08.45 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 12:10.42 | davlefou | qakhan, you use alaw/ulaw codec, is your internet accés good? |
| 12:11.37 | qakhan | yes |
| 12:11.57 | davlefou | you asterisk was local? |
| 12:12.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:12.27 | kaldemar | qakhan: in that laste pastebin which you said is a working example, your service provider answers with c=IN IP4 10.200.7.157. |
| 12:12.43 | kaldemar | qakhan: in the one that does not work, they have 0.0.0.0. |
| 12:13.18 | kaldemar | owners (o=) do not match either. |
| 12:14.20 | kaldemar | and the port in the m header is 0 also. |
| 12:14.33 | kaldemar | go blame them and their HuaweiSoftX3000. |
| 12:18.21 | *** join/#asterisk qakhan (~qakhan@39.59.51.220) |
| 12:18.37 | qakhan | kaldemar i know that and i am worried out it why it is happening with same config |
| 12:20.23 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 12:20.25 | [TK]D-Fender | kaldemar: Could you relink those 2 PB's? |
| 12:23.27 | kaldemar | [TK]D-Fender: sure, http://pastebin.com/pu9u8VPG http://pastebin.com/BjHr42ZS |
| 12:24.06 | kaldemar | qakhan: don't be so sure there is something wrong with your configs. |
| 12:24.51 | qakhan | ok |
| 12:26.00 | [TK]D-Fender | kaldemar: Well it's not the same peer being used for one... |
| 12:26.26 | [TK]D-Fender | kaldemar: So 2 peers to the same device |
| 12:26.42 | kaldemar | [TK]D-Fender: not even the same box. |
| 12:27.04 | [TK]D-Fender | kaldemar: ? |
| 12:27.14 | [TK]D-Fender | kaldemar: Different server's too? |
| 12:27.15 | kaldemar | [TK]D-Fender: the latter is an elastix provided by "them", the first that does not work is an asterisk box configured by "him". |
| 12:27.25 | kaldemar | that's how i understood it. |
| 12:27.47 | [TK]D-Fender | From: "device" <sip:2847801@172.29.44.242>;tag=as0865f8e1 |
| 12:27.49 | [TK]D-Fender | From: "MMA" <sip:2847801@172.29.44.242>;tag=as233f5b17 |
| 12:27.56 | [TK]D-Fender | rthen why do they have the same IP ADDRESS? |
| 12:29.21 | kaldemar | redundant boxes, i guess. |
| 12:29.26 | qakhan | [TK]D-Fender i use same sip trunk at a time in both boxes |
| 12:29.28 | qakhan | no |
| 12:29.56 | kaldemar | and both boxes are using the same ip address? really? |
| 12:30.09 | qakhan | first i tried in elastix it worked then unplugged cable and plugged in asterisk |
| 12:30.15 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 12:30.18 | [TK]D-Fender | qakhan: no, your SERVER IP is the same on both |
| 12:30.39 | qakhan | yes i use at a time |
| 12:32.59 | qakhan | did you guys noticed this line o=HuaweiSoftX3000 12608159 12608159 IN IP4 10.200.0.7 |
| 12:33.00 | kaldemar | one at a time, that should be. |
| 12:33.02 | [TK]D-Fender | Yeah, that "owner" deal looks off |
| 12:33.09 | [TK]D-Fender | like one is a prxy'd call targeting it |
| 12:33.13 | [TK]D-Fender | proxy* |
| 12:33.40 | [TK]D-Fender | o=HuaweiSoftX3000 25703702 25703703 IN IP4 10.200.0.7 |
| 12:33.46 | kaldemar | c header has 0.0.0.0 & m header has 0 as port => "Peer doesn't provide audio" |
| 12:33.52 | [TK]D-Fender | o=- 14922548 14922549 IN IP4 10.200.7.157 s=SBC call |
| 12:34.35 | qakhan | sip trunk ip is 10.200.7.157 but why it is going to 10.200.0.7 |
| 12:35.13 | [TK]D-Fender | Codecs being offered is not the same, etc |
| 12:35.43 | [TK]D-Fender | kaldemar: Was this a "no-audio" issue? |
| 12:35.55 | kaldemar | [TK]D-Fender: yes. |
| 12:36.38 | qakhan | [TK]D-Fender so it is codec issue? |
| 12:37.05 | [TK]D-Fender | Reliably Transmitting (NAT) to 10.200.7.157:5060: INVITE sip:0537707501@10.200.7.157 SIP/2.0 <--- THEY ARE NOT BEHIND NAT. |
| 12:37.42 | [TK]D-Fender | gUESS WHAT... THEY ip IT TELLS YOU TO USE IS not THE ONE YOU DIALED. aND YOU ARE overriding it. |
| 12:42.46 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 12:44.56 | qakhan | what its mean |
| 12:46.09 | [TK]D-Fender | What part is unclear? |
| 12:46.29 | qakhan | <PROTECTED> |
| 12:46.33 | [TK]D-Fender | One of your servers is treating that device as though it were behind NAT. It. Is. NOT. |
| 12:47.11 | [TK]D-Fender | You set your peer to believe that it isn NAT'd. So when they send a DIFFERNT IP for audio, asterisk IGNORES IT |
| 12:48.18 | dwell | qakhan: asterisk -rx 'sip show settings' | grep Externaddr |
| 12:49.39 | [TK]D-Fender | dwell: No... |
| 12:50.01 | [TK]D-Fender | dwell: this isn't an ASTERISK IP issue, it's a PEER IP issue |
| 12:50.16 | dwell | why not?) he will see the difference between natted and lan adddr |
| 12:50.54 | [TK]D-Fender | dwell: You didn't read the debug. And the IP the PEER is reporting back is not the same. |
| 12:51.01 | dwell | plus - yep - peer option |
| 12:51.03 | [TK]D-Fender | it's the REMOTE AUDIO that * is ignoring. |
| 12:51.11 | qakhan | http://pastebin.com/4GySEMEY |
| 12:51.22 | [TK]D-Fender | dwell: externaddr is for ASTERISK's IP which is not the problem. |
| 12:51.34 | [TK]D-Fender | qakhan: FIX YOUR PEER |
| 12:53.36 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 12:53.42 | qakhan | [TK]D-Fender but in sip.conf i did nat = no |
| 12:53.58 | [TK]D-Fender | [08:46]qakhanReliably Transmitting (NAT) to 10.200.7.157:5060: INVITE sip:0537707501@10.200.7.157 SIP/2.0 <--- THEY ARE NOT BEHIND NAT. |
| 12:54.01 | [TK]D-Fender | ^^ See this? |
| 12:54.10 | [TK]D-Fender | it says NAT. Fix your peer |
| 12:54.57 | kaldemar | [TK]D-Fender: it is the (NAT) paste that works. |
| 12:54.59 | [TK]D-Fender | your peers are different |
| 12:55.08 | [TK]D-Fender | So make them match |
| 12:55.28 | kaldemar | that does not affect what the service provider sends in SDP. |
| 12:55.48 | [TK]D-Fender | kaldemar: It affects if * cares about the IP in SDP or not... |
| 13:00.19 | qakhan | i made nat=yes in my ext but i am still get Reliably Transmitting (no NAT) to 10.200.7.157:5060: on asterisk box |
| 13:00.36 | [TK]D-Fender | qakhan: Make your peers match. |
| 13:00.52 | [TK]D-Fender | qakhan: kaldemar says it's the other box that works, so make it match |
| 13:01.28 | [TK]D-Fender | qakhan: Which is to say that by that thought you CAN'T trust what it is telling you to use. |
| 13:01.37 | [TK]D-Fender | qakhan: So set it to "yes" |
| 13:01.41 | [TK]D-Fender | qakhan: And test |
| 13:02.36 | qakhan | [TK]D-Fender i copied 1 ext config from elastix and pasted in asterisk but it didnt work |
| 13:02.48 | [TK]D-Fender | qakhan: One says NAT, the other doesn.t |
| 13:02.59 | [TK]D-Fender | if the peers are the same then a GLOBAL setting is not. |
| 13:03.13 | [TK]D-Fender | Sa make the end result match |
| 13:05.31 | qakhan | is there any setting in sip.conf related to nat? |
| 13:07.17 | [TK]D-Fender | qakhan: Apparently your WORKING one is saying "nat=yes". Make your NON-WORKING PEER say it as well. |
| 13:07.50 | dwell | qakhan: yes externip and localnet |
| 13:11.41 | *** join/#asterisk Manu18 (56ddeaa8@gateway/web/freenode/ip.86.221.234.168) |
| 13:11.46 | Manu18 | Re bonjour |
| 13:11.51 | [TK]D-Fender | dwell: You have not been reading. This is all on a local LAN with local subnets, not with a server with a public IP. |
| 13:12.01 | Manu18 | J'aurais besoin d'aide svp pour un soucis sur Asterisk |
| 13:12.09 | [TK]D-Fender | qakhan: "nat=yes" <- put it in your non-working peer and test. |
| 13:12.20 | Manu18 | French ? |
| 13:12.30 | Milos | I have a queue configured for Asterisk. If the caller calls up and then hangs up while waiting in the queue, CDR logs this as "BUSY" instead of "NO ANSWER". The queue_log does have ABANDON, so why is the CDR showing BUSY? |
| 13:12.38 | [TK]D-Fender | Manu18: vas-y |
| 13:12.46 | qakhan | [TK]D-Fender i already tested that nat=yes |
| 13:12.59 | [TK]D-Fender | qakhan: Show us the new peer and the new call |
| 13:13.07 | qakhan | ok |
| 13:13.34 | [TK]D-Fender | Milos: You'd have to look at the full call-flow of the dialplan. |
| 13:14.00 | Milos | [TK]D-Fender, as in, -vvv etc? |
| 13:14.24 | Manu18 | chan_sip.c:20366 handle_response_invite: Received response: "Forbidden" from |
| 13:14.32 | Manu18 | Cela a chaque fois que je fais un appel externe |
| 13:15.19 | [TK]D-Fender | Milos: yes |
| 13:15.30 | Milos | [TK]D-Fender, I've got it here. Doesn't show anything relating to busy. |
| 13:15.36 | Milos | [TK]D-Fender, would you be able to take a look? |
| 13:15.47 | Katty | FENDER |
| 13:15.48 | Manu18 | [TK]D-Fender: veux tu que je paste mon sip.conf |
| 13:15.53 | Katty | YOU ARE IN TROUBLE. |
| 13:15.54 | Manu18 | et extension.conf |
| 13:15.58 | [TK]D-Fender | Manu18: votre peer n'est pas bien configurer et l'authorisation inclus est refuse |
| 13:16.28 | Manu18 | je paste mon peer alors |
| 13:16.31 | Milos | on parle francais? lol |
| 13:16.33 | Manu18 | car je ne comprends rien |
| 13:16.37 | [TK]D-Fender | Milos: PASTEBIN is your friend. make a complete call in to test, show the call and the queue log & CDR |
| 13:16.41 | davlefou | [TK]D-Fender, vous parlez français? |
| 13:16.46 | Milos | d'accord |
| 13:17.00 | Milos | toute le monde parle francais |
| 13:17.00 | [TK]D-Fender | Manu18: demontre tout sauf masquer le secret. |
| 13:17.04 | [TK]D-Fender | ~pb |
| 13:17.04 | 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. |
| 13:17.06 | [TK]D-Fender | ^^^^ |
| 13:17.31 | davlefou | Super! |
| 13:17.44 | Manu18 | http://pastebin.com/hn3zhBbL |
| 13:17.45 | davlefou | C'est plus facile! |
| 13:17.48 | [TK]D-Fender | davlefou: Pas de tout... |
| 13:18.12 | Milos | [TK]D-Fender, http://bpaste.net/show/O3K6WSVfGr10Fun8VOJ9/ ca suffit? |
| 13:19.16 | Milos | [TK]D-Fender, in that scenario I hung the call up prematurely before anyone on the other end answered. So that should mean 'NO ANSWER' but the CDR shows 'BUSY'. |
| 13:19.21 | [TK]D-Fender | Manu18: j'ai dit de tout montrer sauf le secret. Veuiller le refaire et inclure le debug du CLI ave "sip set debug on". |
| 13:19.53 | [TK]D-Fender | Milos: Again with the log bits I asked for.... |
| 13:20.03 | Milos | [TK]D-Fender, sorry I missed that part. |
| 13:21.54 | Milos | [TK]D-Fender, voila http://bpaste.net/show/mhM6zatFoXfpEXZCtC2Q/ |
| 13:22.02 | Milos | merde I left the number in the second paste |
| 13:22.04 | Milos | oh well ;p |
| 13:22.38 | Manu18 | ok |
| 13:22.43 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:23.21 | [TK]D-Fender | Milos: -- Executing [numero@incoming:4] Queue("SIP/vibe-0000003c", "default,TtKk") in new stack |
| 13:23.29 | qakhan | [TK]D-Fender here is http://pastebin.com/YmiBPzDX |
| 13:23.34 | [TK]D-Fender | Milos: "SIP/vibe-00000034","","Queue","default,TtKk", |
| 13:23.43 | [TK]D-Fender | Milos: Clearly not the same channel number |
| 13:24.06 | [TK]D-Fender | qakhan: -- Executing [0537707501@internal:2] Dial("SIP/801-00000016", "SIP/STC-Outbound/0537707501") in new stack |
| 13:24.14 | [TK]D-Fender | qalDoes that say [801] to you? |
| 13:24.21 | [TK]D-Fender | qakhan: Does that say [801] to you? |
| 13:24.36 | [TK]D-Fender | qakhan: Why are you showing me [801]? I told you to fix the peerk your are dialing from |
| 13:24.40 | [TK]D-Fender | peer* |
| 13:25.22 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-qeuqksvlcqphiuiq) |
| 13:25.23 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:26.58 | Manu18 | voila =====> http://pastebin.com/X9XgUGWi |
| 13:27.42 | Milos | [TK]D-Fender, http://bpaste.net/show/CBf1hxxmi15INpXDfsfw/ |
| 13:27.56 | [TK]D-Fender | SIP/2.0 403 Wrong login or password |
| 13:28.06 | qakhan | who you are calling peer? to ext? |
| 13:28.29 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:28.36 | kaldemar | qakhan: [peer] in sip.conf |
| 13:28.36 | [TK]D-Fender | Tel que avertit... mauvais usager ou mot de passe |
| 13:29.08 | kaldemar | qakhan: in this case, [STC-Outbound]. |
| 13:30.17 | qakhan | kaldemar do i put nat=yes under [STC-Outbound]? |
| 13:30.55 | Manu18 | [TK]D-Fender: tu as regardé? |
| 13:31.00 | kaldemar | qakhan: yes. and "sip reload" in CLI. |
| 13:31.41 | qakhan | ok |
| 13:33.06 | [TK]D-Fender | Manu18: Je regarde.... |
| 13:33.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:33.17 | Manu18 | ok Merci |
| 13:33.49 | [TK]D-Fender | Manu18: Je t'ai repondu deja... SIP/2.0 403 Wrong login or password |
| 13:34.00 | [TK]D-Fender | [09:28][TK]D-FenderTel que avertit... mauvais usager ou mot de passe |
| 13:34.30 | Manu18 | je ne vois pas ou par contre |
| 13:36.57 | Milos | [TK]D-Fender, any idea about the BUSY? I appreciate the help regardless. |
| 13:37.21 | [TK]D-Fender | Manu18: nat=yes <- devrait etre "no", et utilise "defaultuser=USERNAME" en place de "username=" |
| 13:37.34 | Manu18 | ok je teste |
| 13:37.39 | [TK]D-Fender | Milos: Not sure on this one....they all do it consistently? |
| 13:37.49 | Milos | [TK]D-Fender, yeah, consistent. |
| 13:38.35 | Manu18 | pareil |
| 13:38.42 | Qwell | Manu18: omelette du fromage |
| 13:38.44 | Manu18 | Cela ne change rien |
| 13:39.04 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 13:39.20 | Manu18 | Qwell: ??? |
| 13:39.32 | [TK]D-Fender | Qwell: http://bpaste.net/show/CBf1hxxmi15INpXDfsfw/ <--- looks like a CDR bug. the call is hard-answered 3 different ways, never calls "busy" or any kind of override yet still results in "busy" in CDR |
| 13:39.42 | [TK]D-Fender | Manu18: PB <-------- |
| 13:39.55 | Qwell | [TK]D-Fender: I haven't been paying attention. It's all French, to me. |
| 13:40.03 | Manu18 | Jene comprends pas [TK]D-Fender . |
| 13:40.06 | Milos | Qwell, I've been speaking in English. |
| 13:40.07 | Manu18 | PB??? |
| 13:40.10 | [TK]D-Fender | Qwell: just check the PB. |
| 13:40.16 | [TK]D-Fender | Manu18: PASTEBIN <- |
| 13:40.20 | Manu18 | ok |
| 13:40.23 | Milos | le bin de paste |
| 13:40.38 | Milos | -> BP |
| 13:40.39 | [TK]D-Fender | Qwell: He's pretty bilingual. |
| 13:40.52 | [TK]D-Fender | (or more) |
| 13:41.14 | Manu18 | http://pastebin.com/Yg2h0jaM |
| 13:41.21 | Katty | hugs Qwell |
| 13:41.28 | [TK]D-Fender | Katty: Mew. |
| 13:41.45 | Milos | quel horreur |
| 13:41.46 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 13:41.46 | Katty | [TK]D-Fender: oh i see how it is. NOW you say hi. |
| 13:42.26 | [TK]D-Fender | Manu18: Je ne voit pas votre nouveau configuration, ni le "sip debug" qui s'appartient a l'appel |
| 13:42.38 | Manu18 | :( je le paste |
| 13:42.55 | [TK]D-Fender | Katty: Multi-tasking a half doezen plus conversations.... you know how it is... |
| 13:43.04 | Qwell | Queue updates the CDR when it hits a queue member. |
| 13:43.05 | Milos | all this french is making me hungry |
| 13:43.06 | [TK]D-Fender | Katty: Across multiple languages no less... |
| 13:43.22 | Qwell | BUSY is the correct state. |
| 13:43.28 | Milos | What |
| 13:43.43 | [TK]D-Fender | Qwell: How so... they memeber didn't report it back, and the call was answered... this isn't like dialing a device direct... |
| 13:44.10 | [TK]D-Fender | Qwell: they were ringing..... |
| 13:44.12 | Qwell | There are no timestamps here, but I suspect that the ring time is very low. |
| 13:44.36 | Milos | What does the ring time have to do with anything? |
| 13:44.40 | Qwell | That, or they're hanging up before the queue member has a chance to answer. |
| 13:44.44 | Milos | Yes. |
| 13:44.47 | Milos | That doesn't mean it's busy. |
| 13:44.47 | Qwell | they == caller, who abandoned the queue |
| 13:44.50 | Qwell | Sure it does. |
| 13:44.53 | Milos | That means it's NO ANSWER. |
| 13:44.55 | *** join/#asterisk qakhan (~qakhan@182.185.146.29) |
| 13:44.56 | Milos | So why does it say BUSY? |
| 13:44.59 | Manu18 | http://pastebin.com/5J2TQWSC |
| 13:45.07 | qakhan | kaldemar did you check my PB |
| 13:45.19 | Katty | [TK]D-Fender: i certainly do! |
| 13:45.24 | Katty | hugs [TK]D-Fender |
| 13:45.26 | [TK]D-Fender | Qwell: and the call has an explicit answer at the start plus a forced-answer playback... |
| 13:45.34 | Milos | ^ |
| 13:45.36 | Qwell | doesn't matter - the queue is changing the CDR |
| 13:45.42 | Katty | Qwell: my father is in the hospital )= |
| 13:45.45 | Katty | Qwell: heart problems. |
| 13:45.48 | Qwell | Katty: eep! |
| 13:45.50 | [TK]D-Fender | Qwell: that is messed up.... |
| 13:46.55 | Qwell | [TK]D-Fender: search app_queue for ast_cdr_busy - there are a few ways it can happen |
| 13:47.04 | Milos | Qwell, that is messed up. This should not be expected behaviour. |
| 13:47.05 | [TK]D-Fender | Qwell: Still messed up.... |
| 13:47.26 | Qwell | There were no free queue members. That's different from no answer. |
| 13:47.28 | Milos | Why are you arguing your way into making that expected behaviour? |
| 13:47.30 | [TK]D-Fender | Qwell: like all weather reports saying "currently raining" with clear skys..... |
| 13:47.36 | qakhan | [TK]D-Fender here my sip debug after net=yes under peer http://pastebin.com/R1Auk7TZ |
| 13:47.41 | Qwell | I'm telling you what the code does, and has done for many many years. |
| 13:47.47 | Manu18 | Je vois bien qu'il y a wrong password mais je ne vois ou ce trouve ce mauvais pass |
| 13:47.55 | Milos | Heartbleed was unnoticed for many years. |
| 13:48.11 | Qwell | rolls his eyes |
| 13:48.37 | Katty | if i hear one more thing about heartbleed. |
| 13:48.39 | Milos | Ah well, good to know it's normal at least. |
| 13:48.40 | Qwell | ^^^^ |
| 13:48.41 | Katty | i might start gigglign uncontrollably |
| 13:48.50 | [TK]D-Fender | Qwell: Well I've never noticed it personally, so I can't comment there. However... it was calling multiple members. They were free, AND ringing. |
| 13:49.01 | kaldemar | qakhan: let me guess, still no audio? |
| 13:49.42 | qakhan | i hear music on my end but when calle accept call then no voice on both end |
| 13:50.10 | Qwell | [TK]D-Fender: The caller hung up. At the time of the hangup, the queue was busy. |
| 13:50.12 | Manu18 | [TK]D-Fender: l'erreur est mauvais pass au niveau de mon client freephonie ou asterisk ? |
| 13:50.25 | kaldemar | qakhan: so, what did your provider say about the SDP? |
| 13:50.54 | Milos | Qwell, saying something has been 'happening for years' doesn't mean it's been happening most appropriately for years. So I may submit a bug, because reading the logs should allow the ability to distinguish between BUSY and NO ANSWER even if it was in a queue where the original caller hung up prematurely. |
| 13:50.59 | [TK]D-Fender | Manu18: Freephonie vous dit qu'une des deux n'est pas pareil. |
| 13:50.59 | qakhan | where is that SDP? |
| 13:51.54 | kaldemar | qakhan: in the SIP messages. c=IN IP4 0.0.0.0 and m=audio 0 RTP/AVP 0 8. |
| 13:52.07 | Qwell | If you want better information, use CEL. |
| 13:52.12 | *** join/#asterisk mjordan (~matt@nat/digium/x-hnrtuqbtqdobuwcb) |
| 13:52.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:52.13 | Milos | CEL is? |
| 13:52.29 | Qwell | able to track more than a single line for a complex call. |
| 13:53.07 | Milos | So... cdr is deprecated? |
| 13:53.34 | Qwell | No, it just isn't useful for a call that is more complex than A calls B. |
| 13:54.00 | Milos | It's actually pretty simple. |
| 13:54.06 | Milos | A called B, and A hung up. |
| 13:54.09 | Milos | Before B answered, that is. |
| 13:54.15 | Qwell | B is not a single entity. |
| 13:54.15 | Milos | I don't see the complexity. |
| 13:54.26 | Milos | Regardless, nobody at B answered. |
| 13:54.31 | Milos | Therefore the call was unanswered. |
| 13:54.33 | Qwell | A called B, B on behalf of A called C and D |
| 13:54.45 | Qwell | leaves |
| 13:54.53 | Milos | o/ |
| 13:55.14 | Katty | roots |
| 13:56.27 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 13:57.59 | Manu18 | D'ou peut venir cela : SIP/2.0 403 wrong password |
| 13:58.11 | qakhan | kaldemar they didnot say any thing on it |
| 13:58.17 | [TK]D-Fender | Manu18: Freephponie <- |
| 13:58.22 | *** join/#asterisk Naikrovek (cc3624f5@gateway/web/cgi-irc/kiwiirc.com/ip.204.54.36.245) |
| 13:58.26 | Naikrovek | ahoy |
| 13:58.40 | Manu18 | [TK]D-Fender: le mot de pass est bon j'ai verifié. |
| 13:58.56 | [TK]D-Fender | Manu18: Il ne sont pas d'accord |
| 13:59.23 | Milos | verifié encore 3.000 fois |
| 13:59.28 | Milos | je vais dormir |
| 13:59.31 | Milos | au revoir |
| 13:59.38 | *** join/#asterisk qakhan (~qakhan@182.185.146.29) |
| 13:59.42 | Milos | + merci |
| 13:59.45 | Manu18 | [TK]D-Fender: il faut que je le change d'apres toi ? |
| 14:01.05 | [TK]D-Fender | Manu18: From: "0953890201 " <sip:0953890201@62.210.195.144>;tag=as758e3116 <- pas certain pourquoi, mais je vois une espace d'extra ici.... |
| 14:01.26 | [TK]D-Fender | Manu18: Vrevalider toute... et essay de le changer apsres |
| 14:01.37 | Manu18 | ok |
| 14:03.28 | qakhan | yes kaldemar |
| 14:03.56 | Manu18 | [TK]D-Fender: "<- pas certain pourquoi, mais je vois une espace d'extra ici.... [16:01] <[TK]D-Fender> Manu18: Vrevalider toute... et essay de le changer apsres" je n'ai pas compris |
| 14:04.23 | [TK]D-Fender | Manu18: "0953890201 " <- esace apres le "1" |
| 14:04.46 | Manu18 | ok |
| 14:04.47 | *** join/#asterisk happy-dude (~Adium@darwin-mbp2012-sxc.wireless.rit.edu) |
| 14:05.54 | [TK]D-Fender | Manu18: "sip show registry" <- |
| 14:06.20 | Manu18 | freephonie.net:5060 N 0953890201 1785 Registered Mon, 14 Apr 2014 16:05:42 |
| 14:06.50 | *** part/#asterisk happy-dude (~Adium@darwin-mbp2012-sxc.wireless.rit.edu) |
| 14:09.02 | [TK]D-Fender | Manu18: faire une nouveau PB avec votre config courant et une autre appel |
| 14:09.19 | Manu18 | je refais une nouvelle config? |
| 14:09.55 | davlefou | Manu18, tu es chez freephonie? |
| 14:10.03 | Manu18 | oui |
| 14:10.16 | davlefou | Tu veux utilise la ligne sip free? |
| 14:10.26 | Manu18 | oui* |
| 14:10.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:10.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:10.55 | davlefou | As tu été dans ton web gestionnaire de box pour activer l'option sip? |
| 14:11.06 | Manu18 | oui davlefou |
| 14:11.13 | Manu18 | ca a deja fonctionné |
| 14:11.26 | Manu18 | Cela peut pas venir de iptables? |
| 14:11.59 | davlefou | Manu18, as tu ouvert les ports su r ta freebox et tu l'a redémmarré? |
| 14:12.12 | Manu18 | je suis suir un dedié |
| 14:12.49 | [TK]D-Fender | [10:11]Manu18Cela peut pas venir de iptables? <- non. il dit que t'es pau authorisee.... |
| 14:13.15 | Manu18 | je te repaste une nouvelle fois cela a chnagé dans les logsd |
| 14:14.23 | *** join/#asterisk qakhan (~qakhan@182.185.146.29) |
| 14:14.27 | Manu18 | http://pastebin.com/FnuV8WNL |
| 14:15.43 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ivwmfzndpknbxdnd) |
| 14:15.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:15.55 | SuperNull | guys i have an ATA that is to dtmfmode of RFC2833, the sip.conf explicitly says RFC2833 on asterisk, yet other servers in our network seem to only recognize inband dtmf and ignore the out of band stuff.. |
| 14:18.30 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 14:18.59 | qakhan | [TK]D-Fender here is my sip debug http://pastebin.com/R1Auk7TZ |
| 14:19.16 | *** part/#asterisk volga629_ (63e06bde@gateway/web/freenode/ip.99.224.107.222) |
| 14:19.55 | davlefou | Manu18, es que tu es autorisé à appeler hors du réseau free? |
| 14:20.08 | Manu18 | ouibien sur |
| 14:21.35 | *** join/#asterisk labmis (81cec8de@gateway/web/freenode/ip.129.206.200.222) |
| 14:21.46 | labmis | Hello! |
| 14:23.31 | labmis | got a question concerning the presence state support since version 11. Does this work with the PUBLISHed presence state that snom hardphones support? |
| 14:23.47 | davlefou | Pourrais tu me dire quel est le probléme? |
| 14:23.54 | davlefou | Pourrais tu me dire quel est le probléme Manu18 ? |
| 14:24.18 | Manu18 | Je ne peux pas faire d'appels vers l'exterieur |
| 14:25.07 | mjordan | labmis: No. Asterisk does not initiate PUBLISH requests when presence or other state changes occur. |
| 14:26.34 | davlefou | Et tu peux en recevoir? |
| 14:26.48 | Manu18 | je ne l'ai pas configurer pour en recevoir |
| 14:26.52 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-vghrbklbvllhmdkk) |
| 14:26.53 | labmis | @mjordan: but can I read the state from asterisk? The state info from the phone in the PUBLISH is encoded in xml inside a "im"-tag. Don't really know which standard this is. XMPP? SIMPLE? |
| 14:27.52 | mjordan | labmis: PUBLISH is simply a SIP method. Asterisk has limited support for inbound PUBLISH requests, and generally won't process most event packages it receives from other devices. Generally, the only PUBLISH requests we'll process are those related to call completion |
| 14:27.57 | davlefou | Manu18, pourquoi? |
| 14:28.24 | Manu18 | je dois avoir un soucis dans mon extension.conf |
| 14:28.29 | Manu18 | pour cela que je ne recois pas. |
| 14:28.30 | labmis | @mjordan: ahhh! thanks a lot! :) |
| 14:29.20 | davlefou | Manu18, pourrais tu me pastebin tes comptes sip/iax et ton diaplan? |
| 14:29.28 | Manu18 | ok |
| 14:29.37 | Manu18 | avec l'erreur |
| 14:31.34 | Manu18 | http://pastebin.com/wPYUt8NM davlefou |
| 14:34.18 | Manu18 | davlefou: les appels ext fonctionnent |
| 14:34.43 | Manu18 | les appels interne aussi |
| 14:34.54 | Manu18 | tout fonctionne en fait |
| 14:34.59 | Manu18 | Merci à tous pour viotre aide |
| 14:35.13 | davlefou | c'est bizzare d'avoir deux comptes un peers et un friend pour ta freephonie |
| 14:35.52 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 14:36.16 | [TK]D-Fender | [Apr 14 16:28:57] WARNING[9969]: chan_sip.c:5498 create_addr: No such host: freephonie-out |
| 14:36.21 | [TK]D-Fender | T'as change le NOM |
| 14:36.29 | [TK]D-Fender | [free_out] |
| 14:36.32 | Manu18 | oui |
| 14:36.42 | [TK]D-Fender | [freephonie-out] N'existe plus. |
| 14:36.49 | [TK]D-Fender | <PROTECTED> |
| 14:36.57 | [TK]D-Fender | pis t'as pas adapter ton extension |
| 14:37.12 | Manu18 | Par contre des que je decroche ca raccroche... |
| 14:39.01 | Manu18 | La, je comprends pas trop |
| 14:43.20 | Manu18 | davlefou: ??? |
| 14:43.28 | davlefou | Tu es parti de quel tutorial? |
| 14:44.00 | Manu18 | plusieurs lol |
| 14:44.08 | davlefou | Quel asterisk utilise tu? |
| 14:44.10 | Manu18 | davlefou: Pv c possible? |
| 14:44.24 | Manu18 | <PROTECTED> |
| 14:44.44 | Manu18 | quand je f un appel ext ca sonne je decriche |
| 14:44.49 | Manu18 | et asterisk raccroche |
| 14:45.29 | Manu18 | par contre quand j'appele de l'ext vers asterosk ca ne raccroche pas |
| 14:46.05 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-wzhdymlyohpbewga) |
| 14:49.16 | *** join/#asterisk valeech (~valeech@50.242.62.166) |
| 14:51.27 | *** join/#asterisk bitvilag (~HUNbitvi@62-165-202-131.pool.digikabel.hu) |
| 14:51.30 | bitvilag | hey everyone |
| 14:51.45 | bitvilag | I was wondering if anyone did play around with spa3102? |
| 14:52.03 | bitvilag | I would need some help with it. Everything works fine except the caller id. |
| 14:52.16 | [TK]D-Fender | bitvilag: show us screenshots of your config for it. |
| 14:52.23 | Manu18 | galere galere l'histoire |
| 14:52.41 | bitvilag | which part? |
| 14:52.50 | [TK]D-Fender | http://tinypic.com/ <- |
| 14:52.55 | [TK]D-Fender | All of the PSTN side of it |
| 14:53.00 | bitvilag | oks |
| 14:53.17 | davlefou | Manu18, déjà rationnaliser tes comptes, |
| 14:53.46 | davlefou | un pour free en friends et un pour toi, |
| 14:54.24 | Manu18 | ok |
| 14:54.39 | davlefou | Manu18, Tu es sur un serveur dédier? |
| 14:54.48 | Manu18 | oui |
| 14:54.51 | Manu18 | online |
| 14:54.59 | davlefou | genre ovh? |
| 14:55.07 | Manu18 | pour moi je me met en peer? |
| 14:55.26 | davlefou | friend tout le monde, |
| 14:55.34 | Manu18 | ok c'estfait |
| 14:55.38 | davlefou | pour les appels sortant, c'est user |
| 14:55.48 | Manu18 | ok |
| 14:55.59 | Manu18 | appels entrant et sortant ca fonctionne |
| 14:56.14 | Manu18 | par contre des que j'appele et que ca decroche asterisk raccroche |
| 14:58.18 | davlefou | Ta box, elle est en ip fixe? |
| 14:58.42 | davlefou | Quel linux utilise tu sur ton dédier? |
| 14:58.49 | Manu18 | debian 7 ip fixe |
| 14:59.14 | davlefou | Option sip reg désactivé? |
| 14:59.52 | Manu18 | oui |
| 14:59.57 | bitvilag | http://i62.tinypic.com/2jaixhs.jpg |
| 15:00.25 | bitvilag | http://i59.tinypic.com/2r58dpv.jpg |
| 15:00.32 | davlefou | linksys? |
| 15:00.44 | bitvilag | http://i57.tinypic.com/r7u0es.jpg |
| 15:00.46 | bitvilag | yep |
| 15:01.11 | Manu18 | linksys? |
| 15:01.49 | davlefou | ta dernier conf sur le server avec les mises à jours? |
| 15:02.48 | bitvilag | so D-Fender? Any ideas? |
| 15:02.48 | davlefou | Si c'est pour une seul ligne chez free, que tu as téléphone voip, pourquoi utilise tu un pbx asterisk? |
| 15:04.05 | Manu18 | c'est pour ma societe |
| 15:04.23 | Manu18 | on va acheter 3 ou 4 tel |
| 15:04.27 | Manu18 | je dois faire la mise enplace |
| 15:04.38 | davlefou | et une seul ligne? |
| 15:04.49 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-dfgmzbisoisolhzb) |
| 15:05.07 | Manu18 | non on va prendre un abonement |
| 15:05.30 | davlefou | un abonement? |
| 15:05.48 | Manu18 | oui chez un fournisseur |
| 15:07.05 | Manu18 | <PROTECTED> |
| 15:07.28 | [TK]D-Fender | bitvilag: Change your PSTN Answer Delay to 5 seconds and retest. Also make sure your mode is set right for your country as the standard varies |
| 15:08.29 | bitvilag | I tried 5s did not help. About the mode. actually not sure if it right or not. No idea where to look for |
| 15:09.26 | davlefou | Autre que free? |
| 15:10.35 | Manu18 | oui bien sur free nepropose pas plusieur lignes |
| 15:10.42 | Manu18 | surement OVH je pense |
| 15:11.08 | [TK]D-Fender | bitvilag: https://supportforums.cisco.com/discussion/10636241/spa3102-cid |
| 15:11.57 | davlefou | Manu18, ovh, je maitrise la conf, |
| 15:12.19 | Manu18 | Vu que sa raccroche a chaque fois que mon interlocuteur decroche |
| 15:12.20 | davlefou | Manu18, prend tes lignes et aprés tu configure, |
| 15:12.23 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:12.27 | Manu18 | cela peut venir des codecs? |
| 15:12.40 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 15:12.44 | [TK]D-Fender | non |
| 15:13.03 | [TK]D-Fender | C'est une problem d'authorisation. |
| 15:13.09 | Manu18 | encore? |
| 15:13.33 | Manu18 | quand j'appele sur la ligne asterisk ca fonctionne |
| 15:13.41 | Manu18 | je peux te paste le cli ? |
| 15:13.42 | davlefou | Manu18, Tu perd ton temps, car chez ovh les configuration seront différente |
| 15:14.16 | Manu18 | par contre la sa peut venir d'iptables? |
| 15:14.43 | [TK]D-Fender | Manu18: NON |
| 15:14.54 | [TK]D-Fender | Manu18: T'ecoute pas. Leur serveur te REFUSE |
| 15:15.24 | [TK]D-Fender | Manu18: Soit votre peer n'est pas correct ou il y a une problem avec votre compte chez freephonie |
| 15:15.36 | Manu18 | la, il ne me refuse plus vu que je peux emmettre des appels |
| 15:16.03 | davlefou | Manu18, tout ce que va faire se a jeter quand tu changera de founisseur! |
| 15:16.57 | Manu18 | mais d'un point de vu personnel je veux que ca fonctionne |
| 15:17.13 | Manu18 | car j'utiliserais toujours maligne free pour emettre des appels |
| 15:18.32 | davlefou | Alors du dois tester du côté des ip et du fichier configuration. |
| 15:18.56 | *** join/#asterisk ageis (kevin@207.12.89.97) |
| 15:19.03 | ageis | whats preferred method of forwarding a call to another number within the dialplan? |
| 15:19.27 | file | Goto? |
| 15:20.21 | [TK]D-Fender | ageis: What is this "forward" you're referring to? |
| 15:20.37 | [TK]D-Fender | ageis: So far this sounds like you just want to Dial out. |
| 15:20.53 | [TK]D-Fender | ageis: elaborate on the scenario |
| 15:20.57 | ageis | incoming call to a verizon GSM number |
| 15:21.40 | [TK]D-Fender | ageis: How is that "GSM number" arriving at *? |
| 15:21.50 | ageis | ? |
| 15:21.55 | davlefou | Manu18, déjà commencer par utilise la dernier version, 11.8.1 |
| 15:22.00 | [TK]D-Fender | that was not a complete sentence |
| 15:22.20 | [TK]D-Fender | give a proper linear description of the call flow you want to have happen |
| 15:22.39 | davlefou | Aprés tu fais des essayes, je te dis amicalement, cela peut te prendre plusieurs semaine, |
| 15:23.20 | Manu18 | je m'en doute |
| 15:23.23 | Manu18 | merci de l'aide |
| 15:23.29 | Manu18 | je vais faire la MAJ |
| 15:23.36 | ageis | [TK]D-Fender: incoming call to DID --> Asterisk box --> SIP user --> User's personal cell phone not running a SIP client |
| 15:24.27 | [TK]D-Fender | ageis: So go dial out to your cell via a peer you've set up to dial out with |
| 15:24.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:24.50 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-ckzqgslhymklhpay) |
| 15:25.06 | ageis | [TK]D-Fender: Dial() alone will connect the channel? |
| 15:25.47 | [TK]D-Fender | ageis: that is what Dial() does... have you not been using it all this time? |
| 15:26.05 | ageis | [TK]D-Fender :P |
| 15:26.08 | ageis | thanks |
| 15:26.15 | [TK]D-Fender | ageis: You already Dial() this "SIP user". |
| 15:27.00 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-lavmeqghfnvsvduj) |
| 15:30.53 | *** join/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 15:31.04 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginm.net) |
| 15:31.23 | catphish | is it common to be able to configure sip handsets to only allow calls from servers they're registered to? |
| 15:32.02 | bitvilag | i am talking with my provider. what questins should i ask? |
| 15:32.35 | bitvilag | i guess dtmf mode |
| 15:32.36 | [TK]D-Fender | bitvilag: "What is our CallerID standard?" |
| 15:32.50 | [TK]D-Fender | bitvilag: No. There is no such thing as "dtmf mode" for FXO |
| 15:32.59 | bitvilag | i c |
| 15:33.01 | [TK]D-Fender | bitvilag: it's CID signalling |
| 15:33.15 | [TK]D-Fender | bitvilag: Some use DTMF, others use FSK, etc |
| 15:33.30 | [TK]D-Fender | Timing involved (between rings, etc) |
| 15:34.23 | bitvilag | i c |
| 15:34.39 | *** join/#asterisk marceloamorim (bd5ac048@gateway/web/freenode/ip.189.90.192.72) |
| 15:37.07 | ageis | [TK]D-Fender: do outside lines give the same ${DIALSTATUS} as SIP users? |
| 15:37.15 | mjt | is there some summary of the default/sample configuration, what is enabled in there and what should be disabled for a simple sip gateway? |
| 15:37.29 | [TK]D-Fender | ageis: "outside lines" doesn't tell us what you're talking to.... |
| 15:37.35 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 15:38.03 | [TK]D-Fender | ageis: SIP has status codes, so does ISDN. |
| 15:38.22 | ageis | yes thats what i mean |
| 15:43.02 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-gjzuassvyazzenst) |
| 15:46.39 | bitvilag | D-Fender: I went through like 4 people and noone knew the answer lol |
| 15:56.51 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 16:01.57 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-ofjjtidburfdighh) |
| 16:07.57 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 16:08.40 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 16:16.38 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:31.42 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:38.07 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 16:53.02 | *** join/#asterisk spartan7 (~centos@wsip-68-107-210-139.tu.ph.cox.net) |
| 17:00.17 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 17:00.56 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 17:19.29 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-cptkgizmgrnlsekk) |
| 17:24.31 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 17:27.22 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
| 17:30.05 | *** join/#asterisk imcdona (~Thunderbi@75.151.101.97) |
| 17:32.26 | davlefou | Did there are several g729? I have g729 under my soft and my asterisk and ovh. if i don't use g729 with ovh son is good |
| 17:32.33 | davlefou | why? |
| 17:33.49 | [TK]D-Fender | G729 is compressed and obviously sounds worse than G.711 |
| 17:35.55 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 17:36.26 | coppice | G.711 is compressed, just not as much |
| 17:38.31 | [TK]D-Fender | Well technically "companded" as I've read |
| 17:39.08 | davlefou | [TK]D-Fender, yes but : Softphone ---G729/gsm--> Asterisk server dedier ovh ----alaw--> ovh sip, this is better than : softphne ---g729---->asterisk server dedier ovh ---g729---> ovh sip |
| 17:39.33 | coppice | compand means compress + expand |
| 17:40.08 | [TK]D-Fender | I should read up on the fine print of that... |
| 17:40.49 | davlefou | it is non sens for me! |
| 17:40.57 | coppice | the key thing is its far lossier than many people realise. |
| 17:41.36 | davlefou | i have by from digium my g729 liscence |
| 17:42.47 | rrittgarn | having a brain fart, there an easy way to invert a time range when using GotoIfTime, eg, looking for all hours that are !(05:00-17:00) for example |
| 17:44.30 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 17:44.45 | *** join/#asterisk calum_ (~calum_@host86-140-199-167.range86-140.btcentralplus.com) |
| 17:44.55 | [TK]D-Fender | rrittgarn: if you are looking for specific "not" hours for other specifc days, then wrap the order |
| 17:45.30 | [TK]D-Fender | rrittgarn: But there is no logic inside of that timespec |
| 17:50.58 | rrittgarn | so do (17:00-5:00,mon-fri,*,*) essentially? |
| 17:52.39 | [TK]D-Fender | pretty much |
| 17:52.50 | rrittgarn | k thanks |
| 17:53.06 | mjt | why almost every asterisk cli command given on voip-info.org pages does not work on asterisk? |
| 17:54.25 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:54.34 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:54.35 | Qwell | mjt: voip-info is awful and should not be used. |
| 17:55.21 | mjt | oh well :( |
| 18:03.10 | *** join/#asterisk CeBe (~CeBe@port-92-206-30-149.dynamic.qsc.de) |
| 18:11.05 | [TK]D-Fender | mjt: Because it is poorly maintained and syntax is version specific. |
| 18:19.25 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginm.net) |
| 18:28.43 | *** join/#asterisk josefig (~josef@unaffiliated/josefig) |
| 18:29.02 | josefig | Hi, is there a way to forward the traffic and outbound by a certain IP ? I mean I have 2 ips and I want some traffic going out by 1 ip and the other by the other ip, is it possible ? |
| 18:29.31 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-141-181.user.veloxzone.com.br) |
| 18:29.49 | anonymouz666 | I am back on |
| 18:30.44 | [TK]D-Fender | josefig: "man iptables" |
| 18:30.59 | anonymouz666 | iptables -F |
| 18:31.06 | [TK]D-Fender | josefig: * does not multi-home well |
| 18:31.14 | josefig | [TK]D-Fender: mmm I see |
| 18:31.34 | anonymouz666 | kamailio does a good job on this |
| 18:32.30 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:34.12 | *** join/#asterisk Defraz (~Defraz@24-117-69-71.cpe.cableone.net) |
| 18:34.25 | *** part/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:35.08 | *** join/#asterisk retentiveboy_ (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:40.13 | mjt | is autoload=no in modules.conf not generally recommended? |
| 18:40.30 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 18:47.55 | [TK]D-Fender | mjt: Depends on your concept of security & responsibility |
| 18:48.55 | mjt | actually i thin my current probs has nothing to do with modules, because with autoloading it still doesn't work :) |
| 18:48.59 | mjt | think* |
| 18:50.59 | [TK]D-Fender | mjt: You could try the reverse approach.... |
| 18:53.23 | mjt | Answer(500),Playback(hello-world),Hangup() -- that's a tutorial, debug console says it is executing them correctly, but the phone is 100% silent. |
| 18:54.13 | mjt | inter-phone call is sort of working, but again, the callee phone is completely silent until the other end picks up the phone |
| 18:54.25 | *** join/#asterisk navaismo (~navaismo@201.124.133.34) |
| 18:54.35 | mjt | is a complete newbies... |
| 19:00.26 | josefig | anonymouz666, you said kamailio ? |
| 19:06.44 | [TK]D-Fender | mjt: SIp has several networking considerations, especially where NAT is concerned |
| 19:07.01 | mjt | there's no NAT, eveerything is on a GigE LAN |
| 19:07.09 | [TK]D-Fender | mjt: You should probably better describe What you are calling in from, and how it connectes to * exactly. |
| 19:07.33 | [TK]D-Fender | If you have no audio local you should check your firewalls |
| 19:07.44 | [TK]D-Fender | 5060 + rtp.conf's ports, all UDP |
| 19:07.54 | mjt | i'm using 2 softphones -- csipsimple from android and ekiga from linux. No firewall is set up anywhere. |
| 19:08.54 | [TK]D-Fender | I'm sensing more than just "eveerything is on a GigE LAN" |
| 19:09.15 | mjt | well yes. android is on wifi which is bridged to the same net |
| 19:09.51 | *** join/#asterisk karlh626 (~karlh626@addr-199.21.193.173.nptpop-cmts-cable-sub.rdns-bnin.net) |
| 19:12.52 | [TK]D-Fender | first prove each to * direct |
| 19:12.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:14.11 | mjt | should local sip client work on the same machine as the server? |
| 19:15.25 | lvlinux | mjt: yes but you'll have to set the SIP port on the client to something other than 5060 if * is using that port |
| 19:16.40 | mjt | suspected that much. and ekiga does not allow to set sip port ;) |
| 19:17.14 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:17.37 | lvlinux | you could change your * port if you really wanted to go that far (but you'd have to change the destination port on all the clients too) |
| 19:18.02 | lvlinux | why not use linphone instead of ekiga? |
| 19:18.57 | lvlinux | (it allows you to change the local port) |
| 19:19.13 | mjt | heh. i wasn't sure why i ditched linphone. now i remember. |
| 19:19.25 | mjt | it crashes on startup here finding a tv-tuner card as /dev/video0 |
| 19:19.50 | lvlinux | unload the tv card module then |
| 19:20.00 | mjt | yeah, already did |
| 19:20.21 | lvlinux | hmm --- then delete the linphone config and try it again |
| 19:20.46 | mjt | i mean, it works after rmmod |
| 19:20.53 | lvlinux | oh ok |
| 19:21.12 | lvlinux | then can probably go into the prefs and disable video |
| 19:21.44 | lvlinux | and set the input device as static picture or something and I wouldn't think it would look for the tv card then? |
| 19:22.18 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:29.52 | mjt | quite good expirience with linphone. When it's configure wizard - which meets you at startup - does not let you to actually _save_ the config ("Apply" button is inactive) :) |
| 19:30.31 | davlefou | mjordan, sflphone is good to |
| 19:30.40 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 19:30.49 | lvlinux | or get the yate client---it's nice and works well |
| 19:32.56 | mjt | nah. linphone is also silent |
| 19:32.59 | mjt | on the local machine |
| 19:33.24 | mjt | ..and it just crashed when i tried to hit "Hangup" button |
| 19:38.47 | mjt | demo ext. echotest (600) is working from all clients (but linphone which crashes) |
| 19:39.13 | lvlinux | lol ditch linphone... |
| 19:40.06 | mjt | i just tried the same echotest when going from gsm network to external IP using ipsec |
| 19:40.08 | mjt | it also works |
| 19:40.17 | mjt | from android + csipsimple |
| 19:48.37 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 19:52.40 | *** join/#asterisk lovesroot2014 (47a3342a@gateway/web/freenode/ip.71.163.52.42) |
| 19:53.34 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
| 19:53.37 | [sr] | hi |
| 19:53.37 | *** join/#asterisk ledoktre (~ledoktre@216.51.224.229) |
| 19:53.49 | [sr] | does anyone has a linksys 3102, 1x fxs 1x fxo ? |
| 19:54.47 | ledoktre | greetings. a quick question related to SipAddHeader --- any way to change one thats already set for an existing call? I set the ring style for calls coming in. Someone gets parked. I want it to ring a different ring if they sit on hold for 45 seconds and get routed back in. |
| 19:56.17 | anonymouz666 | [sr]: me |
| 19:56.40 | anonymouz666 | ledoktre: I am afraid to say...that this is not possible. |
| 19:57.02 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 19:57.07 | anonymouz666 | at least I can't think a way to do that |
| 19:57.19 | anonymouz666 | (easy way) |
| 19:58.12 | ledoktre | anonymouz666 - I use the SIPAddHeader to set Alert-Info and pick a ring tone. That parts fine. But when they come back from the parked queue, I can't set it again is the problem. I am just wondering if you can modify the variable :) Im sure you understood that from my previous post but (hoping) clarifying ;) |
| 19:58.32 | [TK]D-Fender | When they get routed back in... is that NOT via the dialplan? |
| 19:59.03 | [sr] | anonymouz666: having some trouble connection the pstn line and make it act like a trunk |
| 19:59.12 | [sr] | but still reading :) |
| 19:59.48 | anonymouz666 | [sr]: everybody has problems with spa3102. you should be lucky. I got some echo problems that is not fixable |
| 20:00.33 | anonymouz666 | [TK]D-Fender: it looks like atxfer behaviour, lots of things are done only in features code. |
| 20:00.34 | [TK]D-Fender | [sr]: Thre are dozens of easily google-able guides for this... |
| 20:00.35 | [sr] | echo thats fine for me, just want the trunk working |
| 20:01.10 | ledoktre | [TK]D-Fender - yes via dial plan. But looking for a distinctive ring so you answer it accordingly |
| 20:01.34 | [TK]D-Fender | Well when your dialplan calls a device... that is a NEW call to that device.. |
| 20:01.38 | coppice | most people would say an echoy trunk is not working |
| 20:01.39 | [TK]D-Fender | not an "existing" one |
| 20:02.05 | [TK]D-Fender | coppice: It's working TOO well... I can hear them.... TWICE! |
| 20:02.33 | anonymouz666 | ledoktre: well if hits dialplan than add another sipaddheader |
| 20:02.39 | anonymouz666 | calls again with different value |
| 20:03.33 | anonymouz666 | sipaddheader only works for INVITEs if you wanna modify other headers, you should be using a sip proxy |
| 20:03.58 | ledoktre | [TK]D-Fender: I understand. But copied the original dial plan for the incoming calls and just changed the Alert-Info but not working. You clarified what I needed -- I'll verify the ring itself is working on the phones. |
| 20:04.27 | anonymouz666 | paste the CLI logs |
| 20:04.31 | ledoktre | anonymouz666: Thats what I was doing. in the [parkedcallstimeout] section, I just set another SIPAddHeader line, but it is defaulting to the regular ring. |
| 20:05.37 | ledoktre | okay will do |
| 20:06.04 | ledoktre | http://pastebin.com/a9fwghdq |
| 20:06.07 | lovesroot2014 | Hi -> Can anyone recommend a professional asterisk support/consulting service? I've got an existing legacy setup but need help locking it down. |
| 20:07.25 | [TK]D-Fender | ledoktre: And is it going out? |
| 20:07.36 | anonymouz666 | ledoktre: the alert info value is valid? did you see the header in sip invite? |
| 20:07.39 | [TK]D-Fender | ledoktre: And is the the specific way those devices require it to be? |
| 20:07.54 | [TK]D-Fender | [16:07]anonymouz666ledoktre: the alert info value is valid? did you see the header in sip invite? <- Of course he didn't ... neither did we |
| 20:08.11 | ledoktre | [TK]D-Fender - yes works fine. Rings back just fine -- just uses Low Trill instead of Low Double Trill. Originally tried using Ringback Style, but it didn't work either |
| 20:08.38 | [TK]D-Fender | ledoktre: Perhaps you should verify the legal values |
| 20:09.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:09.13 | ledoktre | [TK]D-Fender - I am using the names from my polycom sip-interop.cfg file. I use these same variables in other areas, like internal calls to give a distinctive ring. |
| 20:09.23 | ledoktre | And they work fine there. |
| 20:09.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:09.59 | ledoktre | Phone just requires Alert-Info to be set before dial - nothing too special there. |
| 20:11.26 | ledoktre | I'll run some debug - thats fine. I also just noticed the channel name is the same -- perhaps the Alert-Info is already set (and sticks) from when the call originally came in |
| 20:11.52 | lovesroot2014 | Can anyone tell me why the from is null?: chan_sip.c: Call from '' to extension 'REMOVEDNUMB' rejected because extension not found. |
| 20:14.05 | [sr] | anonymouz666: one info i cant find is, for the trunk, is there a default user+pwd, or i can set it? been searching and dont see any conf for this |
| 20:14.43 | [sr] | i see alot examples with different usernames |
| 20:14.53 | [sr] | and others with any user/pass |
| 20:16.04 | mjt | lovely. So, re-installing asterisk-core-sounds-en package (debian) re-registering some sound alternatives and *finally* my asterisk produces some sounds. |
| 20:16.44 | anonymouz666 | I don't remember my friend. You have to configure FXO port to register the account in Asterisk. then you dial out to that ID and send the DTMFs (D option) forward |
| 20:16.57 | anonymouz666 | I used that way |
| 20:17.09 | anonymouz666 | the other ID is the FXS port |
| 20:17.28 | mjt | [TK]D-Fender: i think that's enough to prove that network is not a problem... ;) |
| 20:17.49 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 20:19.45 | mjt | gosh. those gsm-format sounds are of very low quality, i don't understand about 80% of what she's saying... |
| 20:19.56 | anonymouz666 | lol |
| 20:20.07 | anonymouz666 | isn't that bad |
| 20:20.14 | mjt | heh |
| 20:21.24 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:21.25 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:21.34 | Qwell | woah, it's a malcolmd |
| 20:21.37 | ledoktre | [TK]D-Fender, anonymouz666: Figured out it WAS the same channel. Had to run SIPRemoveHeader(Alert-Info:) before trying to set it again. Workin' like a peach now |
| 20:21.44 | malcolmd | Qwell: weird, eh? |
| 20:21.46 | anonymouz666 | mjt: opus is the way |
| 20:21.48 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 20:22.16 | mjt | opus can have bad quality too |
| 20:22.18 | davlefou | some one knows the soluce? chan_iax2.c:4459 realtime_peer: Failed to parse sockaddr '(null)' for ipaddr of realtime peer 'ipaddr'? |
| 20:23.34 | anonymouz666 | mjt: opus can go from narrowband to fullband |
| 20:24.00 | mjt | yeah, and can use different encodings/algorithms too |
| 20:24.37 | anonymouz666 | ledoktre: congratulations |
| 20:24.53 | mjt | not a lot of clients supports opus however, and my packaging of asterisk doesn't support it either |
| 20:25.27 | anonymouz666 | is asterisk working fine with webrtc clients? |
| 20:26.26 | anonymouz666 | I am very interested in webrtc stuff and also IMS networks |
| 20:27.16 | ledoktre | anonymouz666: thanks! |
| 20:27.33 | *** part/#asterisk ledoktre (~ledoktre@216.51.224.229) |
| 20:27.35 | davlefou | anonymouz666, did you have install opus under your asterisk? |
| 20:27.47 | davlefou | anonymouz666, sflphone can use opus. |
| 20:29.59 | anonymouz666 | nice to know |
| 20:30.06 | anonymouz666 | jitsi too? |
| 20:30.08 | mjt | does asterisk 1.6 support opus? Somehow I don't think so.. |
| 20:30.27 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 20:30.41 | anonymouz666 | the patch i saw is for asterisk 11 |
| 20:30.46 | anonymouz666 | didn't try yet |
| 20:31.04 | mjt | it went with asterisk12 |
| 20:31.11 | mjt | i've 1.6 ;) |
| 20:32.46 | mjt | errm |
| 20:32.55 | mjt | i've 11.6 not 1.6 ;) |
| 20:36.33 | *** join/#asterisk aness (~aness@2a02:fe0:c310:581:75c0:135a:74f3:ba06) |
| 20:46.16 | *** join/#asterisk valeech (~valeech@pool-71-171-123-210.clppva.fios.verizon.net) |
| 20:53.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:54.43 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 21:00.16 | *** join/#asterisk joesmoe (~joesmoe@c-73-181-72-237.hsd1.co.comcast.net) |
| 21:01.52 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 21:03.29 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 21:04.08 | *** join/#asterisk c|oneman (cloneman@1337.montrealdark.com) |
| 21:04.33 | Weezey | I've disabled "tk" in the dial string but when I am receiving inbound calls and press #, the transfer menu still comes up, how do I shut that off? |
| 21:04.36 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 21:05.06 | c|oneman | might be offtopic, does anyone know the difference between provisioning a polycom phone via Switchvox, or manually entering the SIP information on the phone? |
| 21:05.07 | [TK]D-Fender | Show us |
| 21:05.44 | [TK]D-Fender | c|oneman: There is a lot more that just SIP information in provisioning.... |
| 21:06.09 | c|oneman | assuming you didn't care about features and wanted most of them to be default |
| 21:06.47 | [TK]D-Fender | c|oneman: "default" may not be what you want it to be |
| 21:07.10 | [TK]D-Fender | c|oneman: Provisioning lets you configure all of that without messing with Polycom's web interface |
| 21:07.29 | c|oneman | not to mention some of the features are outright not changeable in that web interface |
| 21:07.45 | Weezey | c|oneman: also, you can do it remotely, in bulk. |
| 21:07.59 | c|oneman | I mean, what if I self provisionned using my own FTP, but then used switchvox SIP information, what would I lose? |
| 21:08.14 | c|oneman | basically my switchvox is pushing old bootroms and I want a new one |
| 21:08.15 | [TK]D-Fender | c|oneman: More like "most" of the features |
| 21:08.38 | [TK]D-Fender | c|oneman: Then I'd say you should go provision them yourself |
| 21:08.59 | [TK]D-Fender | c|oneman: Are you sure it's the BOOTROM you're after? |
| 21:09.14 | c|oneman | nope, I'm very confused by all these layers of software :) |
| 21:09.22 | c|oneman | 4.0.6 is the latest of whatever. |
| 21:09.43 | [TK]D-Fender | "whatever" is VERY important. |
| 21:09.59 | [TK]D-Fender | Typically what users really care about is the SIP APPLICATION |
| 21:10.03 | Weezey | Gonzo is a whatever. |
| 21:10.23 | c|oneman | 4.0.6 I think is the bootrom |
| 21:10.28 | [TK]D-Fender | No, Gonzo is a Muppet :) |
| 21:10.45 | [TK]D-Fender | c|oneman: I think you should verify what it is you want exactly... |
| 21:10.54 | [TK]D-Fender | Or think you want... |
| 21:14.11 | *** join/#asterisk aness (~aness@cm-84.215.76.18.getinternet.no) |
| 21:39.58 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 21:58.17 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 21:59.20 | *** join/#asterisk digiv (~digiv@as1.si.umich.edu) |
| 22:00.29 | *** join/#asterisk aness (~aness@cm-84.215.76.18.getinternet.no) |
| 22:36.09 | *** join/#asterisk regdar (~regdar@72.186.222.82) |
| 22:48.59 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 22:58.46 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 23:10.23 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-108-132-18-211.mia.bellsouth.net) |
| 23:16.05 | *** join/#asterisk jlnt (~jlnt@adsl-99-119-165-116.dsl.rcsntx.sbcglobal.net) |
| 23:16.55 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-108-132-18-211.mia.bellsouth.net) |
| 23:17.16 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-108-132-18-211.mia.bellsouth.net) |
| 23:21.52 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 23:26.02 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 23:30.56 | jlnt | Qwell, I see you are still roaming the Asterisk world |
| 23:41.30 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 23:43.29 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:50.31 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 23:56.45 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 23:59.36 | *** part/#asterisk jlnt (~jlnt@adsl-99-119-165-116.dsl.rcsntx.sbcglobal.net) |
