| 00:12.14 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 00:27.02 | *** join/#asterisk stac (~stac@april-fools/2013/runnerup/stac) |
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| 01:31.53 | file | Hrm |
| 01:32.28 | scv | h |
| 01:33.49 | file | Moo |
| 02:14.54 | *** join/#asterisk boris_t (~boris_t@212.119.243.183) |
| 02:20.52 | qmarchi | Can't get this bloody phone to register... |
| 02:28.43 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-tcmtisjtuvkmxder) |
| 02:40.33 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:45.12 | raspberrypifan | i got some bloddy issues too |
| 02:45.23 | raspberrypifan | asterisk keeps telling me the channel is busy |
| 02:45.25 | raspberrypifan | hangup 17 |
| 02:45.36 | shido6 | change the channel - watch something else |
| 02:45.40 | shido6 | yeesh |
| 02:46.01 | raspberrypifan | lol |
| 02:46.06 | raspberrypifan | hello friendo |
| 02:46.13 | shido6 | hello there |
| 02:50.04 | raspberrypifan | anyone willing to help with my latest troubles http://stackoverflow.com/questions/38862704/asterisk-sip-00000001-is-busy-with-itsp-works-fine-with-other-softphones |
| 03:00.09 | *** join/#asterisk u0m3 (~u0m3@188.27.75.240) |
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| 03:17.35 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 06:19.21 | *** join/#asterisk bhans (ddfd4592@gateway/web/freenode/ip.221.253.69.146) |
| 06:20.14 | bhans | Who tried setting up Asterisk PBX in Japan using KDDI Line? I have question regarding their connection pattern about PBX INS |
| 06:27.31 | wyoung | Never been to Japan but I would love to snowboard there one day |
| 06:33.09 | Haris | ok. asterisk 13 installed on c6 |
| 06:33.12 | Haris | 64bit |
| 06:33.26 | Haris | followed: http://www.voip-info.org/wiki/view/Asterisk+11+Installation+on+CentOS+6 |
| 06:33.30 | Haris | http://blogs.digium.com/2015/02/24/install-asterisk-13-pjsip-centos-6/ |
| 06:33.45 | Haris | http://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+6 |
| 06:33.48 | Haris | http://howto.lintel.in/how-to-install-asterisk-on-centos/ |
| 06:34.32 | Haris | asterisk is running |
| 06:35.50 | Haris | http://pastebin.ca/3681438 |
| 06:37.00 | Haris | what article I can use now to manually create users, extensions for ws/wss / webrtc calls ? |
| 06:41.30 | [TK]D-Fender | WIKI |
| 06:41.37 | [TK]D-Fender | same page you've been pointed to a hundred |
| 06:41.39 | [TK]D-Fender | times |
| 06:42.42 | Haris | ok |
| 06:54.19 | [sID] | As agi send remote-part-id? |
| 06:54.21 | [sID] | I have tried in many ways and I can not :( |
| 07:11.15 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
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| 07:31.55 | [sID] | anybody knows, maybe? |
| 07:32.43 | Haris | haven't known it that much yet |
| 07:36.27 | [sID] | sorry, I do not understand |
| 07:41.20 | Haris | I'm here, but unable to help |
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| 07:48.58 | [sID] | ok |
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| 08:21.29 | ChannelZ | I love linux! My system is currently running on a kernel & modules which have subsequently been purged off the drive, probably a couple of months ago |
| 08:21.44 | ChannelZ | oops! |
| 08:21.53 | ChannelZ | But it runs :D |
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| 09:51.04 | *** join/#asterisk fbnts (~fbnts@s004.spireinns.co.uk) |
| 09:53.20 | fbnts | Hi, I am having an issue with blind transfers using Grandstream phones and asterisk 13.7 - Transferring incoming calls is fine but if I originate a call say to my mobile and then hit transfer then another extension I get Call Transfer Failed on the grandstream and see |
| 09:53.27 | fbnts | "chan_sip.c:23515 handle_response_notify: Got OK on REFER Notify message" on the asterisk console |
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| 12:53.59 | *** join/#asterisk Khronos (~Khronos@72-238-4-152.res.bhn.net) |
| 12:54.03 | Khronos | Hi guys. |
| 12:54.36 | Khronos | What is the suggested maximum number of peers registring to a single host behind a single router? |
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| 12:55.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 12:57.45 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 13:06.50 | newtonr | Khronos, probably depends on how capable the host and the router are. |
| 13:22.37 | ||cw | it absolutely does |
| 13:23.32 | ||cw | a nice cisco IAD is going to handle a lot more than a little $20 netgear |
| 13:23.56 | [TK]D-Fender | Or you could spend a tiny amount of $ and get a VPN router |
| 13:24.48 | ||cw | and it also depends on the usage, registering isn't really a big deal, simultaneous calls is what's going to swamp it |
| 13:25.25 | AviiNL | quick question about GotoIfTime, I have this line "exten => s,1,GotoIfTime(23:00-08:00,*,*,*,?passed:)" does this range 'bleed over' into the next day? eg, monday 11pm to tuesday 8am ? thanks in advance :3 |
| 13:27.52 | *** join/#asterisk crisderock (~Hausmeist@ip9234e635.dynamic.kabel-deutschland.de) |
| 13:27.56 | crisderock | hi |
| 13:28.01 | crisderock | i have a little problem with asterisk 11.13 (and previous versions). i write a macro with ael and the parser generates this macro with extension ~~s~~. now, if i use the application Dial and with M(macroname), asterisk looks for extension s in the macro |
| 13:28.05 | crisderock | is this my fault or maybe a bug? |
| 13:32.59 | [TK]D-Fender | update your system and validate tthat you're still getting it |
| 13:34.32 | crisderock | this is the latest version in debian jessie |
| 13:34.52 | [TK]D-Fender | And we're at 11.23.0 now. |
| 13:34.59 | [TK]D-Fender | that's almost half a branch ago |
| 13:35.29 | crisderock | i know. bit we in a comany and no admin wants self compiled versions in the system |
| 13:35.36 | crisderock | s/comany/company/ |
| 13:35.53 | scv | whats the justification for that |
| 13:36.03 | [TK]D-Fender | Rule #1 here : we don't fix old versions |
| 13:36.54 | [TK]D-Fender | validate against current |
| 13:37.26 | [TK]D-Fender | And if it's a bug then you're going to have to upgrade anyway or find some way around it |
| 13:39.58 | crisderock | are there packages for the current version or only the source? |
| 13:41.39 | scv | debian creates those packages, not digium |
| 13:41.59 | scv | or the package maintainer handling it for debian, w/e |
| 13:42.13 | crisderock | some developer create packages for their software |
| 13:43.27 | scv | stretch is on 13.8 so if you can upgrade the distro there's a router to a newer-but-still-not-the-latest version |
| 13:46.22 | *** join/#asterisk fbnts (~fbnts@s004.spireinns.co.uk) |
| 13:47.27 | *** join/#asterisk Vamp898 (511b7bb0@gateway/web/cgi-irc/kiwiirc.com/ip.81.27.123.176) |
| 13:48.36 | Vamp898 | Hi there, i have an weird problem i just can not fix. When im inside the Netzwork, everything works just fine/perfect. When i am outside the network i can use every service fine (Email, XMPP, RDP; VNC, Whatever) except Asterisk. All services use the same NAT Settings on the same Firewall. The Problem is the following. |
| 13:48.50 | fbnts | Hi, I'm having an issue transferring a call from a Grandstream handset with Asterisk. The transfer fails. On the console I see "chan_sip.c:23515 handle_response_notify: Got OK on REFER Notify message" and if I enable sip debug I see "SIP/2.0 503 Service Unavailable (can't handle one-legged xfers)" |
| 13:49.20 | Vamp898 | I call the Number --> I hear nothing but according to asterisk -rvvv the call is going on well. I hang up and call again, then i hear the other side just fine. I hang up --> the call ends on my phone but according to asterisk the call goes on. If i call again, im in the middle of the conversation. |
| 13:50.33 | Vamp898 | Everytime i ask people tell me my NAT settings are wrong, but nobody is able to tell me what is wrong and why everything else works (even though i have wrong NAT Settings, they only seem to affect asterisk) |
| 13:50.41 | [TK]D-Fender | crisderock, Not us |
| 13:51.23 | [TK]D-Fender | Vamp898, PASTEBIN is your friend. Show us configs and the failed call with SIP DEBUG enabled |
| 13:51.24 | [TK]D-Fender | ~pb |
| 13:51.24 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:51.25 | [TK]D-Fender | ^^^ |
| 13:52.04 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:52.47 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-wdpzhhpespszhobw) |
| 13:55.50 | Vamp898 | Here the log from the call: http://pastebin.com/My2jmsuv i hung up in the end but as you see, nothing happens after i hung up, call is going on. This is the config: http://pastebin.com/yJEBSxui |
| 13:56.30 | [TK]D-Fender | SIP DEBUG <- you missed the big print before |
| 13:56.35 | [TK]D-Fender | "sip set debug on" |
| 13:56.44 | [TK]D-Fender | If you're not looking at SIP debug... you're not looking |
| 13:56.46 | [TK]D-Fender | PB a new call |
| 13:57.05 | [TK]D-Fender | externip=XX.XX.XXX.XXX |
| 13:57.05 | [TK]D-Fender | externhost=sip.hostname.domain |
| 13:57.08 | [TK]D-Fender | yOU SHOULD not HAVE BOTH |
| 13:57.11 | [TK]D-Fender | THIS IS EITHER/OR |
| 13:57.19 | crisderock | nice.. loaded version 13.10.0 from asterisk.org, try to compile and get a undeclared identifier.. ;) |
| 13:57.27 | [TK]D-Fender | oops caps |
| 13:57.47 | [TK]D-Fender | taht should be externaddr, not externhost BTW |
| 13:58.28 | [TK]D-Fender | You're missing the "nat=yes" for your server being behind NAT |
| 13:58.32 | [TK]D-Fender | And sipgate is NOT behind NAT |
| 13:58.50 | [TK]D-Fender | Your settings are all with the wrong keywords, missing or backwards |
| 14:01.32 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 14:02.11 | Vamp898 | [TK]D-Fender: "Asterisk The Definitive Guide" mentions this different. I will try nat=yes and delete externhost |
| 14:03.25 | Vamp898 | Not all devices are behind NAT. The Book mentions i should use auto_force_rport and auto_comedia... is this wrong then? |
| 14:04.41 | [TK]D-Fender | Vamp898, Your provider is NOT behind NAT |
| 14:04.49 | [TK]D-Fender | they ahve real IP's and real routting |
| 14:05.10 | [TK]D-Fender | "Asterisk The Definitive Guide" <- depends which book exactly that this is. |
| 14:05.17 | [TK]D-Fender | what's the DATE on it? |
| 14:05.28 | Vamp898 | [TK]D-Fender: 4th Edition, Covers Asterisk 11 (the one we use) |
| 14:05.42 | [TK]D-Fender | Looks like they left in some legacy stuff... |
| 14:08.08 | Vamp898 | [TK]D-Fender: I removed the NAT completely for the provider and used NAT=yes for the clients. After an restart the behavior seems to be identical. Network-Internal Clients work perfect, smartphones (connection from external networks) have one-way audio and after they hung up, nothing happens. i tried sip set debug on but there are like 1000lines o |
| 14:08.08 | Vamp898 | f logs per second... |
| 14:09.10 | scv | try setting debug for only a single host |
| 14:12.10 | [TK]D-Fender | You don't remove it for your vendor peer |
| 14:12.30 | *** join/#asterisk kharwell (kharwell@nat/digium/x-yzwgijccevzybepp) |
| 14:12.30 | [TK]D-Fender | you should set nat=no explicitly otherwise your [general] one gets applied |
| 14:12.54 | [TK]D-Fender | Finish fixing your configs and show us those first |
| 14:13.03 | [TK]D-Fender | lets validatet that you've done that job properly firstt |
| 14:15.23 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 14:15.49 | Vamp898 | there you go http://pastebin.com/R12ywUN2 |
| 14:16.09 | Vamp898 | i dont know if i do my job properly, obviously "Asterisk The Definitive Guide" was not a good point to start from |
| 14:16.29 | [TK]D-Fender | it was.. |
| 14:16.33 | [TK]D-Fender | butt there's sttill mistakes |
| 14:16.49 | [TK]D-Fender | looks like you're missing section names at line #38 because that's 1 big section |
| 14:16.57 | [TK]D-Fender | and you're nott preventing reinvitets |
| 14:17.05 | [TK]D-Fender | reinvites* |
| 14:17.11 | [TK]D-Fender | "directmedia=no" |
| 14:17.21 | [TK]D-Fender | username=XXXXXX <- should be defaultuser |
| 14:17.38 | [TK]D-Fender | outboundproxy=sipconnect.sipgate.de <- not needed because the host is the same |
| 14:18.05 | [TK]D-Fender | You've also defined a WAN IP butt not what is LOCAL |
| 14:18.07 | Vamp898 | lines 23-48 i got from sipgate, i only changed nat settings on those |
| 14:18.08 | [TK]D-Fender | localnet= <--- |
| 14:18.24 | [TK]D-Fender | you need to set whatt is considered local so it knows what to apply the WAN IP for |
| 14:18.33 | crisderock | oooo kkk.... testet my problem with version 13.2 (cause 13.10 wont compile) and the problem still exists |
| 14:18.55 | [TK]D-Fender | Vamp898, You seem to have a SECOND peer, without a section heading at all starting on line 38 like I said... |
| 14:19.20 | [TK]D-Fender | crisderock, Time to show us code and the CLI dialplan dump |
| 14:19.29 | crisderock | if a macro is called from the dial application, asterisk searches for extension 's' but by parsing the ael, the extension is named '~~s~~' |
| 14:20.47 | Vamp898 | [TK]D-Fender: i know, but this is exactly how sipgate told me to do it... |
| 14:20.58 | crisderock | a second please... have to copy all from the vm.. |
| 14:21.35 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:21.36 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:22.24 | Vamp898 | [TK]D-Fender: i'll fix this up and make one section |
| 14:22.40 | crisderock | http://pastebin.com/xx9cL4nb |
| 14:25.13 | [TK]D-Fender | crisderock, "dialplan show" |
| 14:26.17 | Vamp898 | [TK]D-Fender: this is better? http://pastebin.com/qnbXhW4t |
| 14:26.42 | [TK]D-Fender | Vamp898, Whole thing. Yuo had other fixes to do as well... |
| 14:27.00 | [TK]D-Fender | username=XXXXXX <- didn't fix like I told you to |
| 14:27.07 | crisderock | http://pastebin.com/gT7005eD |
| 14:27.17 | [TK]D-Fender | outboundproxy=sipconnect.sipgate.de <- left this is when I told you it shouldn't be needed |
| 14:27.56 | Vamp898 | [TK]D-Fender: so this is wrong? http://teamhelp.sipgate.de/hc/de/articles/203571302-Wie-konfiguriere-ich-Asterisk-f%C3%BCr-sipgate-trunking- k ill remove it |
| 14:28.17 | crisderock | i had this problem already with the old version and patched this |
| 14:28.30 | [TK]D-Fender | crisderock, Interesting. I'm wondering how this could be as long standing a bug as it appears. go create a ticket on the tracker |
| 14:28.31 | Vamp898 | [TK]D-Fender: username=XXXXX <- should be defaultuser. So you mean i should literaly type "defaultuser" there? |
| 14:28.35 | crisderock | ... but not for the current version |
| 14:29.10 | [TK]D-Fender | Vamp898, Since I'm telling you all these things to fix it should be pretty clear that it's wrong |
| 14:29.30 | [TK]D-Fender | canreinvite = no <- this was changed over 6 YEARS ago |
| 14:29.38 | [TK]D-Fender | Their guide is ancient crap |
| 14:29.49 | [TK]D-Fender | And there is no date on the article |
| 14:30.14 | crisderock | where can i fi9nd the tracker? |
| 14:31.25 | [TK]D-Fender | https://issues.asterisk.org/ |
| 14:33.49 | crisderock | thx |
| 14:34.04 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 14:36.25 | Haris | moving out for a bit |
| 14:36.26 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 14:36.51 | Vamp898 | [TK]D-Fender: is it possible to write multiple networks into localnet? We have multiple networks connected via VPN |
| 14:37.03 | [TK]D-Fender | you can have multtiple |
| 14:37.08 | [TK]D-Fender | 1 per line |
| 14:37.16 | [TK]D-Fender | multiple localnet lines |
| 14:38.42 | crisderock | issue created.. |
| 14:39.17 | [TK]D-Fender | make sure you included the code, and the dialplan show dump |
| 14:39.18 | Vamp898 | [TK]D-Fender: with username=defaultuser i can not connect to sipgate anymore |
| 14:39.22 | [TK]D-Fender | vap |
| 14:39.36 | [TK]D-Fender | Vamp898, ... CHANGE "username" to "defaultuser" |
| 14:39.40 | [TK]D-Fender | not the VALUE |
| 14:39.42 | Vamp898 | ah, now :D |
| 14:39.50 | [TK]D-Fender | the fieldname changed |
| 14:40.12 | [TK]D-Fender | justt like 'canreinvite' has been "directmedia' since 1.6 |
| 14:41.58 | Vamp898 | [TK]D-Fender: better? http://pastebin.com/5MuyQrkF |
| 14:42.48 | [TK]D-Fender | <[TK]D-Fender> and you're nott preventing reinvitets |
| 14:42.48 | [TK]D-Fender | <[TK]D-Fender> reinvites* |
| 14:42.48 | [TK]D-Fender | <[TK]D-Fender> "directmedia=no" |
| 14:42.56 | [TK]D-Fender | add tto [general] and your peers |
| 14:43.14 | [TK]D-Fender | pedantic=false <- should be "yes" really... |
| 14:43.20 | [TK]D-Fender | may * work for parsing... |
| 14:43.22 | [TK]D-Fender | make* |
| 14:43.36 | [TK]D-Fender | oops, you have it in [general] |
| 14:43.40 | [TK]D-Fender | but still add to your peers |
| 14:43.41 | Vamp898 | [TK]D-Fender: with pedantic=yes, we can not transfer calls coming/going from sipgate, thats why we set it false |
| 14:43.47 | [TK]D-Fender | that is very odd.. |
| 14:43.54 | [TK]D-Fender | ok, not an item I'd worry about |
| 14:44.07 | [TK]D-Fender | rest is loking a lot better |
| 14:44.13 | [TK]D-Fender | what are you forwarding tto your server? |
| 14:44.55 | Vamp898 | [TK]D-Fender: you mean in the NAT? |
| 14:45.51 | Vamp898 | [TK]D-Fender: All ports from 5000-20000 UDP |
| 14:46.10 | [TK]D-Fender | I mean that you seem to have confirmed that * is behind that NAT router like the rest of your local devices.... so whatt PORTS have you forwarded from your router to *? |
| 14:47.23 | Vamp898 | [TK]D-Fender: ok, yes, in that case, all ports from 5000 up to 20'000 using UDP protocol |
| 14:47.41 | Vamp898 | iptables btw |
| 14:48.25 | [TK]D-Fender | should just need 5060, and your RTP range from rttp.conf |
| 14:48.59 | [TK]D-Fender | looks a lott better |
| 14:49.01 | [TK]D-Fender | go test itt |
| 14:51.07 | Vamp898 | [TK]D-Fender: i haven't defined anything in rtp.conf and just found out that default is 5000 to 31000....... so the iptables rule is wrong... |
| 14:51.29 | Vamp898 | but still about every 5 calls i have an one-way audio call |
| 14:52.23 | [TK]D-Fender | default should be 10000-20000 based on the sample configs anyway... |
| 14:53.08 | [TK]D-Fender | "sip show settings" and then show the new configs |
| 14:53.16 | [TK]D-Fender | Once that checks out we'll need to see the call itself. |
| 14:54.19 | Vamp898 | [TK]D-Fender: http://pastebin.com/r8smMaSM |
| 14:55.03 | [TK]D-Fender | Externaddr: 81.27.123.176:0 <- not sure on the port not showing... |
| 14:55.26 | [TK]D-Fender | UDP Bindaddress: 0.0.0.0:5060 |
| 14:55.30 | [TK]D-Fender | but the bind shows itt |
| 14:55.33 | [TK]D-Fender | might not be bad |
| 14:55.38 | [TK]D-Fender | show the call |
| 14:55.40 | Vamp898 | [TK]D-Fender: netstat also says 5060 |
| 14:56.05 | [TK]D-Fender | keep in mind I haven't seen your actual phone peers |
| 14:56.15 | [TK]D-Fender | nat=yes |
| 14:56.16 | Vamp898 | how can i enable sip debug only for one call? and is there an easy way to put this into an file? |
| 14:56.22 | [TK]D-Fender | the only template you have shows that... |
| 14:56.32 | Vamp898 | [TK]D-Fender: there is only this template, all clients use this |
| 14:56.34 | [TK]D-Fender | use putty and a big scroll-back buffer |
| 14:56.45 | [TK]D-Fender | right-click on the title-bar and "copy all to buffer" |
| 14:58.00 | Vamp898 | [TK]D-Fender: how can i enable debuging only for one peer. sip set debug intt0011; sip set debug on intt0011 and sip set debug on peer intt0011 are all wrong... |
| 14:58.24 | [TK]D-Fender | learn how to use CLI help |
| 14:58.31 | [TK]D-Fender | "sip set debug <tab> |
| 14:58.49 | [TK]D-Fender | partial line, then hit <tab> to see tthe next keywords available |
| 14:58.56 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-kqettstqqholgxel) |
| 14:58.58 | Vamp898 | [TK]D-Fender: thanks |
| 14:59.07 | *** join/#asterisk ntz (~jdoe@static-84-42-228-122.net.upcbroadband.cz) |
| 14:59.09 | ntz | hello |
| 14:59.21 | ntz | one stupid Q: (I apologize if it hurts too much) |
| 14:59.45 | ntz | if ${EXTEN} in AEL/dialplan represents DID, what represents CID ? |
| 15:03.07 | Vamp898 | [TK]D-Fender: http://pastebin.com/JDVkkksK |
| 15:04.29 | [TK]D-Fender | ntz, It doesn't represent DID, it represents the current extension as is being executed through the dialplan. |
| 15:05.47 | [TK]D-Fender | Contact: <sip:XXXXXX@81.27.123.176:5060> <-- if your WAN IP is right, then you need to look at the calling side. |
| 15:05.54 | ntz | [TK]D-Fender: yeah, sure |
| 15:05.59 | [TK]D-Fender | but do a double-check on your forwarding and firewalls |
| 15:06.25 | [TK]D-Fender | ${EXTEN} is where you are, not how you started. |
| 15:09.56 | ntz | I know the part with EXTEN, JFYI: http://susepaste.org/view/raw/82849134 |
| 15:10.26 | ntz | [TK]D-Fender: it says: From: "root" <sip:T38modem_mgec211@10.0.255.211>;tag=4ce9df21-795d-e611-8dea-005056b01ea6 |
| 15:10.35 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:10.35 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:11.02 | ntz | but it doesn't say the number however the number is somehow encapsulated in t38 protocol so I see on the receiver side a CID (from) number but I am unable to access that from asterisk |
| 15:11.45 | [TK]D-Fender | ntz, Number isn't part of T.38. And the from has nothing to do with EXTEN either |
| 15:11.59 | [TK]D-Fender | ntz, None of those pieces are related to each other. |
| 15:12.12 | ntz | hmmm |
| 15:12.24 | ntz | o.O |
| 15:13.22 | Vamp898 | [TK]D-Fender: i tested it a few times. seems to work now... if this is it, i really thank you very much. Sometimes (every 5 calls) i can see in asterisk that the other side answered, but my app still says "Dialing..." |
| 15:13.59 | ntz | [TK]D-Fender: JFYI, I was trying to resolve problem with asterisk/t38 where I was unable to assign to CALLERID(dnid) anything but both CALLERID(number)= and CALLERID(name)= works |
| 15:14.30 | [TK]D-Fender | You don't assign DNID |
| 15:14.39 | [TK]D-Fender | There is only the number you actually call, and your own callerID |
| 15:15.16 | [TK]D-Fender | Vamp898, like I saidwe'd have to see the calling side... which is the part we haven't seen yet |
| 15:15.28 | ntz | oh my ..... them my apologize, I'm just systems engineer with very limited knowledge about phony/sip related things .... I've made it working at 90%, now I'm trying to get the resting 10% |
| 15:15.37 | ntz | s/them/then/ |
| 15:16.15 | Vamp898 | [TK]D-Fender: but this only happens if i call multiple times the same external number fast, so this could be related to the provider. |
| 15:16.33 | [TK]D-Fender | cisco |
| 15:16.46 | [TK]D-Fender | oops. |
| 15:16.53 | [TK]D-Fender | meaning to ask what router you're using |
| 15:17.32 | ntz | [TK]D-Fender: then my question is - how do I access the CallerID ? I don't see it nowhere in my entire output from sip with enabled debug BUT I see it on the target device passed via the communication of modems (so I guess it's encapsulated in t38 protocol and asterisk don't see it) |
| 15:18.31 | [TK]D-Fender | Go look at the call |
| 15:19.19 | Vamp898 | [TK]D-Fender: Its an Linux Firewall with iptabels (if this question was addressed at me) |
| 15:21.48 | [TK]D-Fender | Vamp898, Ok, so then it shouldn't have anytthing messing with SIP (ALG, etc) in all likelihood |
| 15:22.28 | ntz | I appreciate very very much your will to help me, please tell me if you don't have time or if you're bothered enough and I won't call ya [TK]D-Fender , this is my entire log from doing a fax call between the number 111211 -> 111212 http://susepaste.org/view/raw/a603ab64 |
| 15:22.44 | ntz | I simply don't see there anywhere a string 111211 |
| 15:23.05 | Vamp898 | [TK]D-Fender: i highly think, even when my configs have been borked, it was due to the 5000-20000. In the rtp.conf i used it said 5000-31000 is used per default for my version of asterisk... if that is the case, that would explain why some calls went fine and others not |
| 15:23.53 | ntz | please don't let confuse by numbers prefixed with 1|2|3|4|5, I've for testing purposes put to my dialplan for 111212 target this http://susepaste.org/view/raw/62952916 |
| 15:25.58 | [TK]D-Fender | Vamp898, That is entirely likely |
| 15:26.10 | *** join/#asterisk burnbrighter (~kurt@50-79-211-169-static.hfc.comcastbusiness.net) |
| 15:27.15 | [TK]D-Fender | <PROTECTED> |
| 15:27.16 | [TK]D-Fender | <PROTECTED> |
| 15:27.21 | [TK]D-Fender | those are not things you set. |
| 15:27.29 | [TK]D-Fender | that's are things you see on inbound |
| 15:27.49 | [TK]D-Fender | Your DIAL sets what tthe dialed number is, your callerid = ... the callerid. |
| 15:28.12 | [TK]D-Fender | Dial(SIP/${EXTEN}@T38modem_mgec212/,15,g); <- shouldn't have a trailing / |
| 15:28.25 | [TK]D-Fender | and should use the standard notation of SIP/peer/numbertodial |
| 15:28.27 | [TK]D-Fender | not @ |
| 15:28.49 | [TK]D-Fender | Also that doesn't show what SIP CID presentation settings you're using for that peer... |
| 15:28.50 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:28.56 | [TK]D-Fender | and we're not seeing the call itself |
| 15:30.25 | ntz | ok |
| 15:30.26 | ntz | fair |
| 15:30.31 | ntz | THANKS MUCH !!!!!!!! |
| 15:34.46 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:34.46 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:35.16 | ntz | [TK]D-Fender: now just in theory, I'll change my extensions.ael that way, that I'll be routing that using $some_number_prefix,"XXX" (eg last 3 digits will represent the enduser in range), if I'll be on sending side having the same, what shall be in Dial() that I can access the number from AEL during the call (presumably in block context fax_in_$foo) ? |
| 15:35.55 | Vamp898 | [TK]D-Fender: heading home now, 30 test calls worked fine now. Thank you again very much. |
| 15:36.31 | [TK]D-Fender | Vamp898, You're welcome |
| 15:36.56 | [TK]D-Fender | ntz, I have no idea what you meant there. |
| 15:37.53 | ntz | [TK]D-Fender: that I am capable to see callerid from ael when executing this block http://susepaste.org/view/raw/62952916 |
| 15:38.20 | ntz | because I don't actually see (can access) the caller id (or I don't knbow how) when doing this block |
| 15:38.34 | *** join/#asterisk cbanta (~cbanta@c-50-158-244-235.hsd1.il.comcast.net) |
| 15:38.43 | ntz | even in broken form ${EXTEN}@T38modem_mgec212 |
| 15:38.59 | ntz | wait, ${EXTEN}@T38modem_mgec211 (since sending side is 211) |
| 15:40.05 | [TK]D-Fender | We're still no t seeing what you have in your peer |
| 15:40.20 | ntz | oh, you mean in sip.conf, minute |
| 15:41.04 | ntz | [TK]D-Fender: http://susepaste.org/view/raw/92839808 |
| 15:41.43 | [TK]D-Fender | that seems fine so far |
| 15:42.15 | ntz | [TK]D-Fender: btw: https://www.t38fax.com/support/knowledgebase/asterisk/ |
| 15:42.28 | ntz | [TK]D-Fender: ^ ctrl+f and find on the page "DNID" |
| 15:42.59 | ntz | isn't this solution for me (however in here it doesn't work) |
| 15:43.46 | [TK]D-Fender | there are ***30**** refeerences to DNID including header cut-ups. |
| 15:45.24 | ntz | first one seems good |
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| 16:00.05 | *** join/#asterisk defsdoor (~andy@cpc8-sutt4-2-0-cust254.perr.cable.virginm.net) |
| 16:10.33 | *** join/#asterisk TandyUK2 (5246d3ee@gateway/web/freenode/ip.82.70.211.238) |
| 16:10.49 | TandyUK2 | hey guys, on our system, *2 is for attended transfer |
| 16:11.11 | TandyUK2 | im trying to configure a door entry phone, which require the dtmf string 5*2 to be sent |
| 16:11.27 | TandyUK2 | ofc what im getting is "5, Transfer...." |
| 16:11.33 | shido6 | heheh |
| 16:11.38 | WIMPy | Off course? |
| 16:11.55 | WIMPy | Maybe you shouldn't activate transfers in the Dial() to that door phone? |
| 16:12.00 | TandyUK2 | how do i make asterisk actually send the *2 to the device im 'talking' to, rather than trying to transfer the call? |
| 16:12.13 | WIMPy | (Or from) |
| 16:12.36 | TandyUK2 | what it is tbh is irrelevant |
| 16:12.53 | TandyUK2 | what if im trying to use telehpne banking for example, and i need to send "*2"? |
| 16:13.17 | [TK]D-Fender | Slow down the dial so it doesn't get ack'd as a feature code |
| 16:13.29 | [TK]D-Fender | or stop using them at all |
| 16:13.36 | [TK]D-Fender | DTMF features = ass |
| 16:13.36 | TandyUK2 | i cant do it too slowly, or the door phone times out of programming mode |
| 16:15.02 | TandyUK2 | is there no way to config asterisk to say "**" = send dtmf * |
| 16:15.19 | [TK]D-Fender | change the feature code or stop using them |
| 16:15.20 | [TK]D-Fender | that is all |
| 16:15.21 | TandyUK2 | much like youd use backslash as an escape char in shell |
| 16:15.37 | [TK]D-Fender | <TandyUK2> is there no way to config asterisk to say "**" = send dtmf * <- there isn't |
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| 16:28.54 | shido6 | wow |
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| 18:01.36 | raspberrypifan | does all of asterisk console output get saved to message? |
| 18:02.58 | *** join/#asterisk fbnts (~fbnts@s099.spireinns.co.uk) |
| 18:03.13 | robmal | You set it in logger.conf |
| 18:04.21 | fbnts | Hi, I have upgraded to Asterisk 14 from 13 and have noticed that the console doesn't show verbose logs like incoming call dial plan progression etc. I have connected to the console with -rvvvvvv like I used to. Has something changed between the versions? |
| 18:07.46 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 18:08.23 | raspberrypifan | i see, i just want to get the same logging i get from asterisk to be very verbose in a file so i dont have to watch the console all the time |
| 18:10.24 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 18:10.49 | robmal | Add verbose to messages key. |
| 18:12.57 | *** join/#asterisk ddickenson (uid179041@gateway/web/irccloud.com/x-udwaijibnuxmajsy) |
| 18:13.47 | raspberrypifan | oh cool, |
| 18:14.10 | raspberrypifan | is there anyway i can make the log get automatically feed via network somewhere else |
| 18:15.10 | robmal | Ye, drop it to syslog, let syslog share it over the net and grab it anywhere you want. |
| 18:15.44 | raspberrypifan | cool |
| 18:15.49 | raspberrypifan | thanks |
| 18:16.01 | raspberrypifan | so like mix it in with syslog? |
| 18:16.56 | robmal | There might be a better way but this is the first and easiest thing that comes to my mind. |
| 18:17.14 | raspberrypifan | ill have to do some research |
| 18:21.06 | raspberrypifan | lol turns out its in the manual |
| 18:21.18 | raspberrypifan | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Monitoring_id264504.html |
| 18:22.19 | robmal | This manual was created by the forgotten 13th Apostole, soon after the dinosaurs died. Please use wiki.asterisk.org |
| 18:23.07 | raspberrypifan | whats the difference robmal |
| 18:23.36 | robmal | It's outdated and not the best practice. |
| 18:24.34 | raspberrypifan | oh i see |
| 18:24.45 | raspberrypifan | can i make the log have the specific sip channel i need to text |
| 18:25.53 | robmal | If you enable debug on said peer, ye. |
| 18:29.18 | raspberrypifan | ah i see |
| 18:29.32 | raspberrypifan | its based on the console then |
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| 18:34.40 | raspberrypifan | what i tell the console it will do |
| 18:34.50 | raspberrypifan | kinda wish those cold be indepdent |
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| 20:33.35 | Penguin | On Jitsi, when I haver a voicemail waiting indicator, I press the voice mail call button, and it dials extension 'asterisk'. I'd like to configure it so it dials the appropriate voice mail extension rather than 'asterisk'. Any idea where the setting for this is in Jitsi? |
| 20:34.21 | shido6 | you can have 'asterisk' go to the right vmail extension in the dial plan if 'asterisk' is sent to asterisk. :) |
| 20:34.38 | Penguin | Unless asterisk extension is already configured to do something else. |
| 20:35.07 | Penguin | Which is why I'm asking for someone that knows how to set the correct voice mail extension in jitsi instead of asking if I should reconfigure asterisk. |
| 20:36.02 | shido6 | thats definitely your choice how you want to spend 30 seconds is up to you :) |
| 20:36.47 | Penguin | "Hi, yes, could you please reconfigure your system and all of your users? My jitsi program can't dial *86 like the rest of everyone else." |
| 20:37.09 | shido6 | negative - its a line in the dialplan |
| 20:37.38 | Penguin | I'm not new to asterisk, I'm new to jitsi. |
| 20:38.03 | shido6 | are you using automated provisioning |
| 20:38.34 | Penguin | No, I'm asking if anyone knows how to configure the voice mail calling button in jitsi so I can reprogram it. |
| 20:38.52 | Penguin | I want it to call the correct voice mail extension rather than what it current is dailing. |
| 20:38.55 | Penguin | dialing |
| 20:39.21 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:39.21 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:39.51 | Penguin | On every other hard phone and soft phone I've used, there's a setting somewhere to tell the stupid phone what happens when you hit the Message or VoiceMail button on the phone. |
| 20:40.02 | shido6 | https://jitsi.org/Documentation/FAQ#jitsi-home |
| 20:40.27 | Penguin | I guess that will be a good place to start. |
| 20:43.49 | Penguin | The only reference to 'asterisk' in the entire .jitsi directory appears to be where I have been pushing the voice mail button and it has been sending calls to asterisk@SIPaddr. |
| 20:45.10 | *** join/#asterisk klow (~textual@96.81.150.137) |
| 20:45.17 | shido6 | well - I would add a line in the dial plan and use callerid as auth + pin until I figured out the provisioning process - but we all have holes. |
| 20:45.48 | Penguin | I'll probably make a regular call and dial directly until I can find the setting for the voice mail button. |
| 20:47.50 | *** join/#asterisk Alblasco1702 (~Alblasco1@ip5456b46b.speed.planet.nl) |
| 20:48.15 | shido6 | for s&g's i'll install it and see if i can see what you mean. |
| 20:48.24 | shido6 | what version are you using? |
| 20:48.35 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:48.35 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:48.58 | Penguin | <PROTECTED> |
| 20:49.12 | *** join/#asterisk miralin (~Thunderbi@194.8.128.54) |
| 20:49.53 | Penguin | I found a clue about it. |
| 20:50.23 | Penguin | It's using the contact info for the server as the number to dial for the voice message. |
| 20:51.29 | shido6 | hrmm |
| 20:51.34 | shido6 | i see voicemail subscription uri |
| 20:51.37 | shido6 | and voicemail check uri |
| 20:51.53 | shido6 | under the connection tab |
| 20:52.13 | shido6 | i've read that previous versions may not have honored the info there |
| 20:52.14 | Penguin | I left both of those blank. I actually tried *86@myasteriskaddr at first, but that wasn't any help. |
| 20:53.01 | Penguin | When I start jitsi, it shows there is a voice message, so I click on the indicator. It lists my account name under the voice message "group" on the list. |
| 20:53.17 | Penguin | If I point to the phone symbol, the tip says "call contact." |
| 20:53.37 | Penguin | So I guess I need to figure out how to reprogram the contact for voice mail to be *86 instead of asterisk. |
| 20:54.17 | Penguin | I guess the voice message indicator is actually on top of the call history button. When I click it, it expands the list., |
| 20:55.36 | shido6 | weird - i don't even see the voicemail button |
| 20:55.52 | Penguin | Do you have any new voice mails? |
| 20:56.11 | shido6 | not yet - i'll make one now.. |
| 20:56.24 | Penguin | To the right of the box that says "enter name or number" there is a button that is for call history. |
| 20:56.44 | Penguin | When I start jitsi, there is a red indicator on that call history button with a number 1 on it. |
| 20:57.41 | Penguin | If I press it, the list below gets expanded. It shows Voice messages on the top similar to a group when on the XMPP account. The voice messages group shows my account ID and says "1 new and 18 old messasges" |
| 20:58.32 | shido6 | ok - well mine worked flawlessly - I did put voicemail subscription uri in there and a BAD voicemail check uri - |
| 20:58.50 | Penguin | If I click on the entry on the list, it expands that entry to show the cassette tape icon and a phone icon. |
| 20:59.12 | shido6 | i think I'm starting to like jitsi - thank you for bringing this up - i haven't used jitsi in a very long time |
| 20:59.13 | Penguin | So when you pressed the phone icon to call your voice mail, it called asterisk extension, right? |
| 20:59.39 | shido6 | it FOUND the right extension - the extension i put in the voicemail settings was not used - |
| 20:59.54 | shido6 | I'm using freepbx for this test tho |
| 21:00.05 | Penguin | What extension do you use for voicemail, *98? |
| 21:00.19 | shido6 | *97 was dialed |
| 21:00.27 | shido6 | let me try again |
| 21:00.29 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 21:01.00 | Penguin | Is that the default in fpbx? I thought I remembered it as *98. |
| 21:01.24 | Penguin | *shrug* I don't use fpbx, so I don't know. I'd have to go look it up. |
| 21:02.27 | shido6 | thats what I thought |
| 21:02.30 | Penguin | The weird thing is that even if I put something in the voicemail check URI field, when I restart jitsi, that field has been cleared. |
| 21:02.39 | shido6 | thats what i put in there but IT dialed *97 |
| 21:02.41 | Penguin | It does not retain what I typed in. |
| 21:03.01 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:03.05 | Penguin | You're saying you entered *98 in the box but it dialed *97? |
| 21:03.22 | shido6 | yep - |
| 21:03.31 | Penguin | That's some messed up stuff. |
| 21:03.32 | shido6 | so now I'm looking through the config for that |
| 21:03.57 | Penguin | Yours dials a different numeric extension, mine dials asterisk... neither of which we configured! |
| 21:04.06 | shido6 | right |
| 21:04.13 | shido6 | but I'm thinking there is a config advertising this info |
| 21:04.15 | Penguin | Goofy stuff. |
| 21:08.06 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 21:08.19 | Penguin | I see the value in asterisk, but I can't figure out how to change it. |
| 21:08.33 | Penguin | sip show settings - Voice Mail Extension: asterisk |
| 21:08.58 | Penguin | Got it. |
| 21:09.05 | Penguin | vmexten in sip.conf |
| 21:09.09 | WIMPy | vmextension IIRC |
| 21:09.15 | Penguin | close! |
| 21:09.21 | Penguin | Good guess, though. |
| 21:09.23 | WIMPy | or shorter |
| 21:09.59 | WIMPy | It's been a while since I touched sip.conf. |
| 21:10.03 | shido6 | vmexten= |
| 21:10.06 | shido6 | nice |
| 21:10.14 | Penguin | Now to restart jitsi and see if it's fixed. |
| 21:10.21 | shido6 | in [general] |
| 21:10.24 | shido6 | in sip.conf |
| 21:10.41 | Penguin | Solved. |
| 21:10.44 | shido6 | nice |
| 21:11.11 | shido6 | thanks WIMPy , I learned something new today |
| 21:11.27 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 21:11.27 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 21:11.31 | Penguin | That's funny. I guess because EVERY OTHER phone I have ever used has a configurable voice mail extension IN THE PHONE, I had no reason to ever set the value broadcast by asterisk. |
| 21:12.16 | WIMPy | On most phones there's no reason to configure anything VM if your sip.conf is correct. |
| 21:12.57 | Penguin | My provisioning files for hard phones always had the correct extension written in the messages button setting. |
| 21:13.06 | Penguin | When I use soft phones, I set the value in the setup. |
| 21:13.14 | Penguin | Never used Jitsi before until today. |
| 21:13.30 | Penguin | I'm glad I found that setting in sip.conf. |
| 21:13.34 | shido6 | hehe |
| 21:13.48 | Penguin | I was really blaming it on jitsi! |
| 21:14.06 | shido6 | yeah - the problem was at Layer 8 |
| 21:14.17 | Penguin | That happens occasionally. |
| 21:16.46 | Penguin | Now if I can figure out how to integrate the asterisk contacts and xmpp contacts together, it would be much nicer. |
| 21:17.14 | Penguin | As it is now, I have an XMPP contact and a SIP contact for each person. |
| 21:22.52 | Penguin | And since Asterisk is separate from the XMPP server, trying to "chat" with the SIP contact is useless. |
| 21:31.32 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 21:31.32 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 21:37.46 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 21:37.46 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 22:06.57 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 22:06.57 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 22:10.58 | *** part/#asterisk danielyk (~danielyk@p5B21B974.dip0.t-ipconnect.de) |
| 22:56.13 | *** join/#asterisk piling_ (~piling@unaffiliated/piling) |
| 23:02.53 | *** join/#asterisk piling (~piling@unaffiliated/piling) |
| 23:04.35 | *** part/#asterisk kharwell (kharwell@nat/digium/x-yzwgijccevzybepp) |
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| 23:19.10 | raspberrypifan | is this normal in a sip trace |
| 23:19.12 | raspberrypifan | i=(o=IN IP4 192.168.84.71) |
| 23:19.13 | raspberrypifan | c=IN IP4 209.208.212.140 |
| 23:28.26 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 23:37.24 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 23:54.57 | raspberrypifan | anyone got a sansay vsxi |