| 00:19.57 | *** join/#asterisk tesaf (~root@2602:ffea:1:39d::3257) |
| 00:35.17 | *** join/#asterisk marlinc (~marlinc@ip2.weert.li.nl.cvo-technologies.com) |
| 00:35.28 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 00:57.06 | chandoo | when i add sip entries, does it make any difference if i add at the end |
| 00:57.21 | chandoo | sip show settings doesn't show my entries |
| 00:58.16 | WIMPy | General settings belong under [general]. Otherwise to the peer they're under. |
| 00:59.27 | chandoo | i have /usr/local/etc/asterisk/sip.conf is this location good or do i need to copy it under /etc/asterisk/sip.conf? |
| 00:59.43 | chandoo | WIMPy, i dont see my entries |
| 00:59.50 | WIMPy | Whatever you configured. |
| 01:01.43 | chandoo | yes |
| 01:05.30 | phix | WIMPy: DAHDI :( |
| 01:05.52 | WIMPy | Who's your dahdi? |
| 01:07.32 | phix | TDM400p! |
| 01:07.33 | phix | :) |
| 01:07.41 | phix | but is broken :( |
| 01:08.04 | WIMPy | Analog is evil! |
| 01:08.24 | phix | but cost effective |
| 01:08.38 | WIMPy | Where? |
| 01:09.35 | phix | Analgoue is cost effective than replacing all existing analogue phones in my house with SIP phones and recabling with cat5e/6. |
| 01:10.09 | phix | plus I would need a PoE switch too (as I don't have power sockets in all of the places I have a phone) |
| 01:10.51 | WIMPy | But the interfaces are expensive. |
| 01:11.16 | phix | $320.00, cheaper than replacing all analouge phones + recabling + PoE switch |
| 01:11.47 | WIMPy | You skipped the 20th century. |
| 01:12.00 | phix | Alternatively I could purchase an ATA |
| 01:12.40 | phix | I don't remember SIP being available in the 80's |
| 01:12.41 | WIMPy | DAHDI might be the better choice there. |
| 01:12.52 | WIMPy | No, but ISDN. |
| 01:13.36 | phix | ISDN in the 80's was rediciously expensive, especially in AU |
| 01:14.09 | phix | any way, I need DAHDI workiung |
| 01:14.11 | phix | but it isn't :L( |
| 01:14.12 | WIMPy | Now it's good, cheap and robust technology. |
| 01:14.12 | phix | :( |
| 01:14.23 | phix | WIMPy: in AU it is expensive |
| 01:14.27 | chandoo | i have phone power account i am trying to configure asterisk13 on freenas |
| 01:14.37 | phix | chandoo: get in queue :) |
| 01:14.44 | phix | take a number |
| 01:14.52 | chandoo | :) |
| 01:15.00 | WIMPy | LOL |
| 01:15.13 | chandoo | i am following this link from phonepower |
| 01:15.14 | chandoo | http://www.phonepower.com/wiki/Configuring_Asterisk_To_Use_SIP_Credentials |
| 01:15.16 | phix | I think WIMPy is the only support guy on here :P |
| 01:15.40 | chandoo | i did all the confguration exactly, i am not sure why i dont see it registered |
| 01:15.48 | WIMPy | But I have a tendency to ignore SIP related questions. |
| 01:15.56 | phix | WIMPy: I am also having another issues where I need to reload asterisk every 2 minutes or I cannot phone into my SIP DID |
| 01:16.15 | chandoo | i got proxy as sip2.phonepoer.com, but it is not pinging, so i change it to sip.powerpower.com |
| 01:16.25 | [TK]D-Fender | chandoo, "sip show registry" |
| 01:16.27 | chandoo | which i can ping |
| 01:16.29 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 01:16.31 | phix | yaya [TK]D-Fender is here! |
| 01:16.36 | [TK]D-Fender | chandoo, You seemed to have doubt your configs are even in the right place |
| 01:16.39 | chandoo | [TK]D-Fender, 0 entries |
| 01:16.44 | phix | WIMPy: ok, lets continue fixing my DAHDI :) |
| 01:16.45 | WIMPy | chandoo: Sounds like you need a proxy. |
| 01:16.52 | chandoo | asterisk*CLI> sip show registry |
| 01:16.52 | chandoo | Host dnsmgr Username Refresh State Reg.Time |
| 01:16.52 | chandoo | 0 SIP registrations. |
| 01:16.52 | chandoo | asterisk*CLI> |
| 01:17.01 | [TK]D-Fender | chandoo, Then either you did your configs very wrong, or they are not in the right place or condition |
| 01:17.07 | phix | chandoo: use pastebin to paste in the future |
| 01:17.07 | WIMPy | And phix seems to have an issue with hist natting firewall. |
| 01:17.13 | [TK]D-Fender | chandoo, "core show settings" <- |
| 01:17.20 | phix | WIMPy: Yeah it is probably the CISCO router causing issues again |
| 01:17.44 | phix | WIMPy: I wish I could isolate that but it is in production and I can't gain access to the site after business hours |
| 01:18.06 | phix | WIMPy: so I will just reload asterisk every 2 minutes for now |
| 01:18.28 | WIMPy | phix: Enable keepalive |
| 01:18.44 | chandoo | [TK]D-Fender, http://pastebin.com/e8urMdGB |
| 01:19.02 | phix | WIMPy: What's the config directive called? I will read up on it |
| 01:19.03 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 01:19.27 | [TK]D-Fender | Configuration directory: /usr/local/etc/asterisk |
| 01:19.33 | WIMPy | Take a guess. |
| 01:19.34 | [TK]D-Fender | There is your answer |
| 01:19.57 | chandoo | [TK]D-Fender, yes that is where all the files are |
| 01:20.07 | chandoo | i configured sip.conf from that location |
| 01:20.11 | [TK]D-Fender | "sip reload" |
| 01:20.27 | chandoo | sucessful |
| 01:20.30 | [TK]D-Fender | then "sip show peers" |
| 01:22.39 | chandoo | got http://pastebin.com/VmE2VFip |
| 01:22.42 | chandoo | [TK]D-Fender, |
| 01:22.50 | [TK]D-Fender | So the peer is there. |
| 01:22.58 | [TK]D-Fender | You screwed up your register statment then |
| 01:23.20 | [TK]D-Fender | pastebin your config masking ONLY the secrets |
| 01:30.49 | chandoo | http://pastebin.com/Lp6Hm1Dk |
| 01:31.12 | [TK]D-Fender | You don't even HAVE a register command to your provider |
| 01:31.42 | [TK]D-Fender | You have failed to follow their instructions |
| 01:34.11 | WIMPy | Who still uses register these days? |
| 01:34.33 | janicez | dogecoin |
| 01:34.34 | chandoo | [TK]D-Fender, my mistake |
| 01:34.37 | chandoo | thanks |
| 01:37.01 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 01:42.35 | chandoo | do i have to use register=> or register= |
| 01:43.08 | chandoo | i belive it is after the [phonepower] |
| 01:43.48 | WIMPy | It belongs to general, or you can use callbackextension in the peer instead. |
| 01:44.26 | [TK]D-Fender | You have to do them at the END of [general] and before any other section definition |
| 01:44.29 | [TK]D-Fender | LOCATION MATTERS |
| 01:45.46 | chandoo | [TK]D-Fender, i am able to register now :) |
| 01:45.55 | chandoo | happy to see it registered |
| 01:46.36 | [TK]D-Fender | This sample does NOT include any of the settings required to work from behind NAT. |
| 01:46.40 | [TK]D-Fender | You have a lot of reading to do |
| 01:47.19 | chandoo | sure i will do that |
| 01:47.33 | chandoo | my goal is to configure 4 extenstions |
| 01:53.16 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 01:53.16 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:15.43 | chandoo | what are voip clients that works well with windows and asterisk |
| 02:17.25 | chandoo | is 3cx recommended? |
| 02:17.38 | drmessano | Blink |
| 02:26.12 | *** join/#asterisk coreyfarrell (~coreyfarr@24-177-3-44.dhcp.nwtn.ct.charter.com) |
| 02:36.35 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 02:48.19 | coreyfarrell | has anyone done multiple instances of asterisk on the same system using alternate config paths like the testsuite does? looking for any information about issues with doing this in production. |
| 02:59.44 | *** join/#asterisk infobot (ibot@rikers.org) |
| 02:59.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 03:00.12 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:15.25 | *** join/#asterisk ganbold (~Ganbold@173.244.215.173) |
| 03:18.04 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 03:36.02 | *** join/#asterisk K0HAX (~K0HAX@c-75-72-143-131.hsd1.mn.comcast.net) |
| 03:36.37 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 03:57.34 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 04:07.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 04:37.50 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 05:01.24 | *** join/#asterisk creativx (~ntttt@226.62-97-205.bkkb.no) |
| 05:03.34 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 05:09.45 | *** join/#asterisk KeatonT (~KeatonT@pool-173-64-210-22.dllstx.fios.verizon.net) |
| 05:11.05 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 05:16.15 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 05:18.53 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 05:19.39 | *** join/#asterisk KeatonT (~KeatonT@pool-173-64-210-22.dllstx.fios.verizon.net) |
| 05:37.49 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 05:38.29 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-162-146-39.ph.ph.cox.net) |
| 05:48.06 | *** join/#asterisk kleszcz (tick@178.216.200.29) |
| 05:53.22 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 06:24.58 | *** join/#asterisk bulkorok (~Benjamin_@217.110.197.225) |
| 06:27.28 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 06:43.35 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 07:00.36 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 07:03.14 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:22.12 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:30.44 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 07:31.48 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:38.07 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:44.17 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:45.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 07:52.29 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:00.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:07.45 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 08:08.04 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 08:10.37 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-62-18.om.om.cox.net) |
| 08:11.44 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 08:14.54 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 08:17.01 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 08:17.34 | *** join/#asterisk K1rk (~Kirk@5.135.221.149) |
| 08:17.54 | *** part/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 08:21.33 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 08:24.26 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 08:26.42 | *** join/#asterisk seik0 (b2411142@gateway/web/freenode/ip.178.65.17.66) |
| 08:30.42 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:34.17 | seik0 | Hi. Asterisk is sip-registered to some TelCo-sip-switch to received calls from fixed phones. And everything looks like that TelCo's sip-switch does not send all sip-packets to correctly finish SIP dialog. I know this because asterisk writes on CLI messages "Sheduling" and "Really destruction" of SIP dialogs and it writes a lot of them. |
| 08:35.14 | seik0 | Moreover, looks like after a while some SIP dialogs are not auto-destroyed |
| 08:35.54 | seik0 | And when "sip show channels" shows about 300 active SIP dialogs asterisk starts slowing down |
| 08:37.31 | seik0 | Question is - what information do I really need to collect to send to TelCo so that they will believe in my problem and will not say that it's asterisk issue |
| 08:37.48 | seik0 | because TelCo is hard in such questions |
| 08:37.56 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 08:43.08 | *** join/#asterisk Jayk555 (51896c3d@gateway/web/freenode/ip.81.137.108.61) |
| 08:43.17 | Jayk555 | Hi |
| 08:44.43 | Jayk555 | I have read that I need either dahdi or zaptel to get the meetme.so file on my asterisk build. But i'm being told be colleagues that I don't need either. p |
| 08:44.53 | Jayk555 | please can somebody advise / |
| 08:52.38 | *** join/#asterisk Dodewapwap (~dod@mail.ovive.fr) |
| 08:54.06 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 08:55.26 | Dodewapwap | Hi, I'm trying to set up Asterisk to allow SIP hard phones to communicate with XMPP clients (I use Jitsi and Gajim to test it) |
| 08:56.21 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 08:56.39 | Dodewapwap | So I compiles res_xmpp, chan_motif and configured both, set 'icesupport=true' and added a STUN server in rtp.conf, but I can't manage to make the phones and the XMPP clients communicate |
| 08:56.46 | Dodewapwap | ERROR[25953][C-00000001]: chan_motif.c:759 jingle_add_ice_udp_candidates_to_transport: Unable to add ICE-UDP candidates as ICE support not available or no candidates available |
| 08:57.41 | Jayk555 | I have read that I need either dahdi or zaptel to get the meetme.so file on my asterisk build. But i'm being told be colleagues that I don't need either. please can somebody advise? |
| 08:58.33 | WIMPy | Jayk555: True. Just use the new ConfBridge instead of MeetMe. |
| 08:58.57 | Jayk555 | Does it work the same as meetme? |
| 08:59.14 | WIMPy | No. It has a lot more features. |
| 08:59.45 | WIMPy | The only advantage of MeetMe is that it's faster. |
| 08:59.55 | Jayk555 | but the core functionality is the same right? |
| 09:01.27 | WIMPy | Depends on your definition. It has the same purpose but it works in a different way. |
| 09:02.51 | Jayk555 | oh okay. Well I'm basically going to set one of our numbers up to automatically join a meeting room that is administrated by an extension in my extensions.conf |
| 09:03.59 | Jayk555 | so a person will dial that number, be greeted with our welcome message, be put into the room and told it wont start till I join |
| 09:07.32 | WIMPy | Both have the purpose of connecting more than two callers. |
| 09:07.48 | Jayk555 | Okay that's great. Thanks for your help |
| 09:15.19 | Jayk555 | Just read a document and it says that I can't use a pincode? |
| 09:16.25 | WIMPy | IIRC you can, but I ended up doing it in the dialplan to have more control over what happens. |
| 09:18.02 | WIMPy | There was an old ConfBridge that was not much more than a prrof of concept, really. That one did have VERY little features. Less than MeeyMe. So make sure you're looking at current information for |
| 09:18.09 | WIMPy | Asterisk 11 or newer. |
| 09:22.43 | Jayk555 | ok thanks, I'll look into that |
| 09:31.29 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 09:34.01 | *** join/#asterisk Rico_pb (~rico@unaffiliated/rico29) |
| 09:34.03 | Rico_pb | hello |
| 09:34.23 | Rico_pb | I'm trying to make my asterisk 1.8 realtime mysql tables compatible with asterisk 13 |
| 09:34.48 | Rico_pb | somebody here gave me the name of a script for building asterisk 10+ realtime mysql databases |
| 09:34.50 | Rico_pb | but I forgot it |
| 09:36.29 | Rico_pb | alembic ! |
| 09:39.50 | WIMPy | It should be somewhere in the contib directory. |
| 09:40.29 | *** join/#asterisk cbeyerlein (~cbeyerlei@fra-of-out.net-m.de) |
| 09:44.28 | cbeyerlein | how can I prevent dahdi from spamming my kernel log after every call? "wct4xxp 0000:05:08.0: Unassigning channel 0/2!" |
| 09:46.34 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:59.23 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:02.53 | *** join/#asterisk hehol (~hehol@tdsl-gw.dortmund.loca.net) |
| 10:06.16 | *** join/#asterisk mastahh (~mastahh@89.238.129.100) |
| 10:06.42 | mastahh | Hello, I was wondering if someone would be able to assist with a vanilla build of Asterisk and my SIP client getting error 408? |
| 10:07.26 | WIMPy | ~ask |
| 10:07.27 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:07.57 | *** join/#asterisk mirela666 (~Mirko@D57E13CA.static.ziggozakelijk.nl) |
| 10:09.28 | mastahh | Apologies WIMPy. Let me rephrase my question :) |
| 10:11.03 | mastahh | I am running X-lite 3.0. My client is on IP address 192.168.100.74, Asterisk is on 192.168.100.76. I have entered the following entries in to IPTables iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT && iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT. I have a bog-standard SIP.conf file but I am still receiving the error "Registration Error: 408 - Request Timeout". |
| 10:11.03 | mastahh | I was wondering if there was something I am missing from the initial setup of Asterisk. |
| 10:11.42 | mastahh | I have also ran a packet capture on my Asterisk box and can see SIP traffic on the server from my client also. |
| 10:18.34 | WIMPy | Looks like its a firewall on the client machine then. |
| 10:19.26 | Chainsaw | Is the client wanting to use SIP over TCP? |
| 10:19.39 | mastahh | The client firewall is completely disabled |
| 10:19.47 | mastahh | But in saying that, I will double check! |
| 10:20.42 | mastahh | Yea, it is all turned off - atleast as turned off as Windows allows it in Control Panel. |
| 10:21.08 | Chainsaw | And the anti-virus solution in use is not adding another firewall on top? |
| 10:21.34 | WIMPy | Do you see SIP traffic in both directions? |
| 10:21.41 | WIMPy | Do you see it on the *CLI? |
| 10:21.47 | mastahh | Windows Defender is now disabled. Let me test connectivity again. |
| 10:24.19 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:b803:c1d8:75c9:fd9b) |
| 10:25.33 | mastahh | WIMPy should I load in Asterisk via "asterisk -cvvv" command to see logs whilst in *CLI? |
| 10:25.45 | *** join/#asterisk davlefouAMD (~davlefouA@197.15.195.65) |
| 10:26.24 | WIMPy | You can do that, but you usually attach to a running Asterisk with asterisk -r. |
| 10:27.37 | mastahh | I think one underlying problem I have is that I am using a Cisco UCS with no audio cards because now my X-Lite client is complaining about audio devices. I am going to reinstall Asterisk and try connecting from my laptop directly to a VM withnot any networking infrastructure in between. |
| 10:27.59 | mastahh | I am not 100% sure that our WatchGuard firewall is allowing SIP traffic. |
| 10:28.26 | WIMPy | Didn't you say you saw SIP traffic? |
| 10:29.00 | mastahh | I did, on my Asterisk server. |
| 10:29.05 | WIMPy | And when doing it from the same machine bear in mind that you can't use the same port multiple times on the same IP. |
| 10:29.17 | WIMPy | In both directions? |
| 10:29.51 | mastahh | I am going to re-test this as I am now using my laptop (albeit on a different subnet but still has connectivity to the Asterisk server) and see that I have SIP traffic going bi-directionally. |
| 10:30.29 | WIMPy | So at least your Asterisk seems to be working. |
| 10:30.53 | WIMPy | YOu might just suffer from a NAT issue, off course. |
| 10:30.56 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 10:32.02 | mastahh | My initial plan was to build the Asterisk server on my UCS environment so that if my directors approve of using Asterisk in production, I would not need to rebuild the whole server and could just simply rehost it to our production environment. However, I am in a bit of a pickle whereby I have network infrastructure that I do not have control over inbetween myself and the server |
| 10:33.38 | mastahh | So, my Wireshark packet capture on my laptop is this: I get a SIP packet sent sending a REGISTER packet but then getting ICMP 590 Destination Unreachable |
| 10:33.44 | mastahh | I can ping the server no problems however |
| 10:35.46 | mastahh | Interestingly, I have just ran the command "nmap -v -sV localhost -p 5060" on my Asterisk server and can see the output: PORT 5060/tcp STATE closed -- is this normal? |
| 10:35.53 | WIMPy | Sounds interesting, but instead of looking in to networking issue, I will take a look in to the fridge and see what it has to offer. |
| 10:36.10 | WIMPy | SIP is uaually UDP. |
| 10:36.19 | mastahh | That's what I thought |
| 10:36.25 | mastahh | I think it might just be misleading tbh |
| 10:36.55 | mastahh | [root@localhost ~]# lsof -i:5060 |
| 10:36.55 | mastahh | COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME |
| 10:36.55 | mastahh | asterisk 3343 root 7u IPv4 18385 0t0 UDP *:sip |
| 10:37.16 | mastahh | It all appears OK but just fails to work |
| 10:37.36 | mastahh | [root@localhost ~]# netstat -nlp | grep 5060 |
| 10:37.36 | mastahh | udp 0 0 0.0.0.0:5060 0.0.0.0:* 3343/asterisk |
| 10:40.38 | mastahh | Could this be the issue: |
| 10:40.40 | mastahh | To: "1000"<sip:1000@192.168.100.76> |
| 10:40.40 | mastahh | From: "1000"<sip:1000@192.168.100.76>;tag=dd1ec277 |
| 10:40.56 | mastahh | Contact: <sip:1000@10.6.10.71:53626;rinstance=356b8146a5c7b0eb> |
| 10:41.50 | mastahh | That is from the packet capture I have just taken on the Asterisk server... |
| 10:43.24 | *** join/#asterisk averythomas (~averythom@2607:5300:60:2d42::1) |
| 10:43.40 | *** join/#asterisk tesaf (~root@2602:ffea:1:39d::3257) |
| 10:43.55 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 10:44.55 | WIMPy | 'sip set debug on' |
| 10:45.01 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 10:46.09 | mastahh | WIMPy - nothing happening :( |
| 10:47.41 | mastahh | Judging from the Wireshark captures I have taken from my laptop and the Asterisk server, I have can see a solitary REGISTER packet being sent and received over, and over again |
| 10:48.31 | WIMPy | If you can't see it in Asterisk, it doesn't seem to get there. |
| 10:48.45 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-pllhczxoerpioept) |
| 10:49.41 | mastahh | Right, I think I will just rebuild my Asterisk installation on a VM on my laptop and see if I get any further. My SIP.conf would not be the issue would it? As I said, it is extremely Vanilla, very basic |
| 10:50.43 | WIMPy | It could be, but you already showed it's listening on udp/5060 from 0/0 it's ok so far. |
| 10:51.19 | WIMPy | UNless that's something else, not Asterisk. |
| 10:53.35 | *** join/#asterisk kleszcz (tick@178.216.200.29) |
| 10:53.40 | mastahh | I am wondering if the initial SIP REGISTER packet uses ICMP to establish a connection and then begin to transmit via 5060? |
| 10:54.00 | mastahh | WIMPy please forgive me as I am clutching at straws here and my knowledge of SIP is relatively limited |
| 10:55.07 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 10:55.55 | WIMPy | I think using ICMP first would be an entirely new concept in IP :-) |
| 10:56.06 | mastahh | Haha, fair enough :P |
| 10:56.35 | mastahh | But do you understand my theory? Does the initial REGISTER packet use port 5060 at all? |
| 10:57.51 | Chainsaw | I'm expecting to see ARP, then SIP (and hopefully RTP at some point, when an actual audio stream appears). |
| 10:57.57 | WIMPy | Yes |
| 10:58.10 | Chainsaw | But no ICMP, unless your MTU is wrong. |
| 10:58.38 | *** join/#asterisk CeBe (~CeBe@port-92-200-59-135.dynamic.qsc.de) |
| 10:58.43 | mastahh | I am spoke to my infrastructure manager who believes there is no rule to allow SIP through. Just getting a change control form completed to get the change implemented and re-test |
| 10:59.12 | mastahh | If I can get Asterisk working today, hopefully we can get it implemented internally as our new phone system in the upcoming month! |
| 10:59.53 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:03.16 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-59-135.dynamic.qsc.de) |
| 11:04.52 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 11:05.16 | *** join/#asterisk bulkorok (~Benjamin_@217.110.197.225) |
| 11:07.28 | WIMPy | No SIP allowed. Sounds like a nice place. |
| 11:09.49 | skrusty | ;] |
| 11:21.52 | mastahh | WIMPy until I joined, there was never any sort of interest in telephony. |
| 11:24.17 | WIMPy | I will continue to refure to accept SIP as a form of telephony. |
| 11:30.02 | mastahh | Haha, are you a POTS kind of guy then, WIMPy? |
| 11:32.30 | *** join/#asterisk mirela666 (~Mirko@D57E13CA.static.ziggozakelijk.nl) |
| 11:34.09 | *** part/#asterisk bulkorok (~Benjamin_@217.110.197.225) |
| 11:41.01 | WIMPy | No. Analog is evil! |
| 11:41.58 | WIMPy | There are other VOIP proticolls that are ok, IAX is pretty good, but a little neglected. But so far I see no replacement for ISDN. |
| 11:43.14 | cbeyerlein | WIMPy: but TDM will die, at least in Europe for telephony... |
| 11:43.37 | WIMPy | Yes, We need to destroy everythign that works. |
| 11:43.40 | r00f | the faster the better |
| 11:44.35 | cbeyerlein | oh yeah.. I have done interconnection compatibility test with german DTAG on SIP... horror... |
| 11:44.51 | WIMPy | LOL |
| 11:45.28 | Chainsaw | WIMPy: H323 is quite close to ISDN, closer than SIP is. |
| 11:45.41 | WIMPy | Tehy are pretty famous for not getting their stuff to work. |
| 11:45.42 | Chainsaw | WIMPy: Do you not like that one either? |
| 11:45.56 | WIMPy | H.323 is good, yes. |
| 11:46.01 | cbeyerlein | WIMPy: exactly! "yeah, the spec demands this, but we cannot do it" |
| 11:46.22 | WIMPy | Just that it has the NAT issue as well, off course. |
| 11:52.09 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 11:52.24 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 11:56.42 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:56.57 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 11:59.35 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:02.04 | *** join/#asterisk seik0 (b2413906@gateway/web/freenode/ip.178.65.57.6) |
| 12:03.48 | seik0 | Hi again. When using realtime odbc, does odbc-connection spooling really works? I.e., does each asterisk channel is capable of using free odbc-connection for fetching dialplan if previous odbc-connection is busy? |
| 12:08.46 | seik0 | Now I have situation, when having 45 simultanious calls processing dialplan is very slow |
| 12:09.31 | seik0 | and it is not that specific steps get all the time - time is spread all along the dialplan processing |
| 12:11.10 | seik0 | And odbc-connection used to fetch dialplan is allowed to spool up to 10 connections |
| 12:11.44 | seik0 | And "odbc show" shows 10 max, 5 active and 2-3 in use |
| 12:14.47 | seik0 | the worst scenario I foresee is that when channel is asking for next dialplan step there is only one odbc connection processing all such requests |
| 12:15.25 | seik0 | and spooling works in case different modules fetch data from odbc |
| 12:19.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:19.37 | seik0 | agh, looks like it's not an odbc issue. Dialplan is processing slow even on non-realtime parts |
| 12:20.15 | WIMPy | Anythign special going on on your setup except for using realtime? |
| 12:21.04 | seik0 | going on in asterisk? |
| 12:21.15 | WIMPy | yes |
| 12:21.36 | WIMPy | Switches? CEL? Debugging? |
| 12:22.44 | WIMPy | Build time debugging options? |
| 12:23.06 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 12:23.07 | seik0 | voicemail with odbc storage, call-center, calls recording, IVR |
| 12:23.12 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:23.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:23.28 | seik0 | almost everything interacting with odbc |
| 12:23.36 | WIMPy | recording: Do you perhaps have an I/O issue? |
| 12:23.37 | seik0 | *interact |
| 12:23.59 | seik0 | maybe |
| 12:24.30 | seik0 | btw memory leakage in odbc driver adds pain |
| 12:26.02 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 12:27.44 | seik0 | most important answer, that it is most likely not an odbc issue I got =) |
| 12:32.35 | wdoekes | seik0: s/spool/pool/g |
| 12:54.52 | *** join/#asterisk russellb (~rustlebee@redhat/russellb) |
| 12:54.52 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:56.32 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 12:58.02 | *** join/#asterisk ghoti (~paul@hq.experiencepoint.com) |
| 13:01.40 | seik0 | wdoekes: much better, thanks =) |
| 13:03.05 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 13:07.27 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 13:12.19 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:13.28 | *** join/#asterisk [Outcast] (outcast@nat/cisco/x-rrchkvjmktctmyav) |
| 13:15.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:17.48 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 13:19.23 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 13:24.45 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 13:29.10 | *** join/#asterisk mjordan (mjordan@nat/digium/x-muwespfjtlhrrvow) |
| 13:29.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:32.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:34.13 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:36.41 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-31-201.user.veloxzone.com.br) |
| 13:36.54 | anonymouz666 | hello |
| 13:37.28 | *** join/#asterisk tedstriker (~tedstrike@mig01416328395.demon.co.uk) |
| 13:38.14 | *** join/#asterisk tedstriker (~tedstrike@mig01416328395.demon.co.uk) |
| 13:42.56 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 13:53.35 | cbeyerlein | hm. how comes that GROUP_COUNT and GROUP_MATCH_COUNT take a groupname, but GROUP does not accept a group-name, only a category?! |
| 13:56.35 | cbeyerlein | oh, got it.. Set(GROUP()=groupname) ... |
| 13:57.01 | cbeyerlein | an example would help... https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_GROUP |
| 14:02.51 | *** join/#asterisk mirela666 (~Mirko@D57E13CA.static.ziggozakelijk.nl) |
| 14:06.01 | mastahh | WIMPy I have moved the client and the Asterisk server in to the same subnet and I am still getting the same issue :( |
| 14:06.30 | mastahh | Traceroute on my Windows client only shows there is no external firewall in the way of communication |
| 14:06.42 | mastahh | And I am still getting error 408 |
| 14:10.45 | [TK]D-Fender | What about your server itself? |
| 14:10.48 | [TK]D-Fender | That has a firewall.... |
| 14:10.53 | [TK]D-Fender | prove what's in there |
| 14:11.46 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-lsruandgrazllzqb) |
| 14:11.58 | mastahh | iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT |
| 14:11.58 | mastahh | iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT |
| 14:12.08 | mastahh | I have entered these two commands in to my server |
| 14:13.29 | mastahh | If that is what you was asking for [TK]D-Fender? |
| 14:13.54 | [TK]D-Fender | Just because you have that doesn't mean you don't have OTHER stuff in there |
| 14:14.22 | mastahh | Forgive my ignorance [TK]D-Fender, how can I output all the rules that have been applied to my Asterisk server firewall? |
| 14:14.34 | mastahh | I would still define the configuration of the server to be relatively Vanilla |
| 14:14.57 | [TK]D-Fender | That does not count as "verification" |
| 14:15.01 | [TK]D-Fender | "iptables --list" <---- |
| 14:16.32 | mastahh | Is there a way I can share the output with you? |
| 14:16.38 | [TK]D-Fender | ~pb |
| 14:16.39 | 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. |
| 14:16.40 | [TK]D-Fender | ^^^ |
| 14:16.45 | mastahh | Thanks lol. |
| 14:16.50 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-59-135.dynamic.qsc.de) |
| 14:17.16 | *** join/#asterisk theron (~theron@199.201.64.131) |
| 14:17.36 | mastahh | [TK]D-Fender http://pastebin.com/x0vfdEps |
| 14:18.48 | [TK]D-Fender | Seems OK. |
| 14:18.56 | [TK]D-Fender | Show us the SIP debug from * CLI |
| 14:19.00 | [TK]D-Fender | "sip set debug on" |
| 14:19.42 | mastahh | This is what is confusing me. When I do that, I get no output yet I get confirmation when running a packet capture on my server that I am atleast getting a SIP REGISTER packet from my client. I can try again for you now, though. |
| 14:20.28 | mastahh | "SIP Debugging enabled" and then nothing. |
| 14:20.43 | [TK]D-Fender | Where are you doing this capture? |
| 14:22.17 | mastahh | On the server and on my client. On my client I am using Wireshark and on my server, I am using the following command: /usr/sbin/tcpdump -w sip.pcap -i ens160 |
| 14:24.56 | [TK]D-Fender | Ok, and you see nothing when using "sip set debug on" from * CLI? |
| 14:25.04 | mastahh | Nothing at all. |
| 14:25.28 | mastahh | Could my sip.conf file have anything to do with this also? |
| 14:25.37 | mastahh | Bearing in mind that is extremely basic. |
| 14:26.31 | [TK]D-Fender | No. this is pure networking. I suspect I've misinterpreted your iptables dump. |
| 14:27.26 | mastahh | From what I can make of the IPtables dump, it seems to be allowing pretty much everything through? |
| 14:30.23 | [TK]D-Fender | Since you don't see to have yuo have an allow section with no entries |
| 14:30.37 | [TK]D-Fender | since you have nothing that I can tell has any importance, trash it entirely |
| 14:30.44 | [TK]D-Fender | "iptables --flush" |
| 14:30.49 | [TK]D-Fender | And try again |
| 14:31.38 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:71:4121:9e83:2f2f:972) |
| 14:32.20 | mastahh | Does it automatically rebuild a firewall policy once it has been trashed? |
| 14:32.32 | mastahh | Or will I now need to re-enter those two commands I mentioned previously? |
| 14:35.20 | [TK]D-Fender | one step at a time |
| 14:35.42 | [TK]D-Fender | "Does it" would require my knowing what "it" is. |
| 14:36.07 | mastahh | The firewall residing on the Asterisk server - sorry. |
| 14:36.19 | [TK]D-Fender | that means your entire OS |
| 14:36.32 | [TK]D-Fender | Firewall is not part of Asterisk. |
| 14:36.35 | [TK]D-Fender | That is your OS setup |
| 14:36.53 | mastahh | So does the OS rebuild the firewall policy since we have erased it? |
| 14:37.23 | [TK]D-Fender | Probably |
| 14:37.34 | [TK]D-Fender | I don't what what you are using or how that got build |
| 14:37.38 | [TK]D-Fender | You're just going to have to find out |
| 14:37.44 | [TK]D-Fender | So for now just trash it and test |
| 14:38.01 | mastahh | Trashed and tested and I am still getting the 408 error. |
| 14:38.30 | mastahh | In all fairness, the firewall policy I have shared with you was just there. The only changes I have made were those two commands to explicitly allow SIP and RTP traffic to traverse the server. |
| 14:39.11 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 14:40.28 | [TK]D-Fender | I said trash it |
| 14:40.32 | [TK]D-Fender | not put other stuff in there |
| 14:43.01 | mastahh | Sorry, you're misinterpretting what I have said there. I have not put anything else in the firewall policy. |
| 14:43.37 | mastahh | It is completely empty. |
| 14:44.27 | [TK]D-Fender | Ok, you should be seeing it in SIP debug. "sip show settings" |
| 14:44.37 | [TK]D-Fender | And show us that it's empty as well as your external trace |
| 14:46.20 | mastahh | We have got somewhere [TK]D-Fender |
| 14:46.35 | mastahh | I have done as you said, trashed the firewall and tested again to no avail. |
| 14:46.51 | mastahh | However, I have entered the two lines I mentioned previously and now my SIP softphone has registered |
| 14:47.41 | *** join/#asterisk Qwell (~north@asterisk/developer/Qwell) |
| 14:47.41 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:47.55 | mastahh | Just confirmed that it is showing my host IP in the "sip show peers" output |
| 14:48.21 | mastahh | [TK]D-Fender - you are a gentleman and a scholar. Thank you for your help there! (I hope you're male now I have said that =) ) |
| 14:51.16 | [TK]D-Fender | You're welcome |
| 14:55.25 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-egdlzutufesdtpfb) |
| 14:59.12 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 14:59.42 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 15:00.15 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 15:00.42 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 15:01.12 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 15:01.43 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 15:11.19 | *** join/#asterisk kharwell (kharwell@nat/digium/x-fuyxwxnafifjugek) |
| 15:11.33 | *** join/#asterisk Qwell (north@asterisk/developer/Qwell) |
| 15:11.33 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:23.23 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-pyabpbzzndlgowzr) |
| 15:42.33 | *** join/#asterisk vader- (~Adium@204.183.88.4) |
| 15:55.08 | *** join/#asterisk f0ner00t (~f0ner00t@69-170-21-20.static-ip.telepacific.net) |
| 15:55.09 | f0ner00t | Does anybody know of a good script to ban / jail sip hackers? |
| 15:55.51 | WIMPy | ifconfig eth0 down |
| 15:56.13 | f0ner00t | WIMPy ha ha |
| 15:56.39 | WIMPy | Many like to gamble with fail2ban. |
| 15:57.28 | f0ner00t | Yea |
| 15:57.59 | f0ner00t | But I was talking about a acutal script someone wrote |
| 15:58.45 | [TK]D-Fender | Since fail2ban does do what it says I don't see too much point in reinventing the wheel |
| 15:59.13 | WIMPy | Oh, I do. But I don't know what to make of the word "script" here. |
| 16:09.53 | *** join/#asterisk jab416171 (~jab416171@c-76-27-96-12.hsd1.ut.comcast.net) |
| 16:13.51 | f0ner00t | Thank you Winmpy and [TK]D-Fender |
| 16:15.25 | mastahh | Quick question for my knowledge. "Contexts" in Asterisk are the same as Paritions in the Cisco CallManager world? |
| 16:16.28 | [TK]D-Fender | mastahh: From what I've just read, more or less |
| 16:17.01 | [TK]D-Fender | mastahh: They hold things that can be dialed. Context can't include other contexts in a heirarchy so that you can scale access based on where a call is |
| 16:17.13 | [TK]D-Fender | Phones would point to a context that would include things they are allowed to call. |
| 16:17.25 | mastahh | [TK]D-Fender: is there such a thing as a group of Contexts? |
| 16:17.34 | mastahh | In Asterisk I mean. |
| 16:17.47 | [TK]D-Fender | Incoming calls from the outside world would probably land in other contexts with things they can dial ( including implementing Auto-Attendents, etc) |
| 16:17.58 | [TK]D-Fender | There is no "group" |
| 16:18.15 | mastahh | In the Cisco would, you would have a Calling Search Space which would have a 1:m relationship with Partitions. |
| 16:18.17 | mastahh | *world |
| 16:18.27 | [TK]D-Fender | Every call lands in a SINGLE context. |
| 16:18.34 | [TK]D-Fender | That context can INCLUDE other contexts. |
| 16:18.40 | [TK]D-Fender | The context itself defines this. |
| 16:18.44 | mastahh | And one phone is related to one context. |
| 16:18.48 | [TK]D-Fender | There is no "list for search" |
| 16:18.55 | [TK]D-Fender | the one you're looking in must explicitly INCLUDE others |
| 16:19.43 | mastahh | Thank you for clarification [TK]D-Fender |
| 16:20.23 | [TK]D-Fender | So look at is as a starting point on a branch of a tree. |
| 16:21.27 | WIMPy | More of a net than a tree. |
| 16:22.09 | [TK]D-Fender | A "web" if you will |
| 16:22.31 | [TK]D-Fender | Considering you COULD do stupid stuff like attempting circular includes, etc |
| 16:22.49 | WIMPy | And if you enable international calling, you have a world wide web :-)) |
| 16:22.49 | [TK]D-Fender | * custs you off after a point, though I don't recall the recursion limit |
| 16:23.05 | mastahh | Is there such a thing as a best practice guide for Asterisk roll-outs? |
| 16:23.51 | [TK]D-Fender | Not really. |
| 16:23.51 | mastahh | And do you guys sell Asterisk as VoIP solution in your day-to-day jobs? |
| 16:24.09 | [TK]D-Fender | A few of us do. I don't. |
| 16:25.35 | mastahh | I've come from a "Cisco, Cisco, Cisco" world. I have recently moved companies, and as I said earlier, we have only just started to tap in to telephony since I have come on board and I am looking in to getting Asterisk up and running and demoing to our sales guys here as an alternative to Cisco if they have a customer who is looking in to going down a VoIP route for their telephony. |
| 16:26.38 | [TK]D-Fender | Are you looking to develop your own solution to sell? |
| 16:27.17 | [TK]D-Fender | or for you to maintain on customer's behalf? |
| 16:27.21 | mastahh | I am looking to utilise Asterisk for call processing but look to see what there is out there that can be used alongside Asterisk. |
| 16:27.47 | [TK]D-Fender | What kind of "alongside things" do you have in mind? |
| 16:27.51 | mastahh | Such as a Jabber-esque IM and Presence software client. |
| 16:28.05 | mastahh | Reporting and billing software. |
| 16:28.19 | [TK]D-Fender | These are things you can setup on the side as well. There are solution providers already doing this |
| 16:29.02 | mastahh | The idea is that we can still offer a VoIP telephony service/solution to our customers, but without the six figure cost really. |
| 16:29.14 | *** part/#asterisk f0ner00t (~f0ner00t@69-170-21-20.static-ip.telepacific.net) |
| 16:30.55 | mastahh | Hometime! Speak to you guys later. Thank you again for your help today [TK]D-Fender and WIMPy - pleasure chatting to you guys :) |
| 16:31.54 | [TK]D-Fender | mastahh: You're welcome |
| 16:39.15 | *** join/#asterisk superscrat (asanders@nat/digium/x-ramqljscymrsjvnh) |
| 16:45.22 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 16:56.07 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-229-207-133.lsanca.fios.verizon.net) |
| 17:03.01 | *** join/#asterisk Syco (~syco@217.73.79.34) |
| 17:09.15 | Syco | Hi, I have a problem with the new agent implementation, could someone point me to the right direction please? I have some agents logged in without queues, then I do a ami call to bridge a channel to an agent, using AgentRequest. All this work preferctly, but the agent can't use any dtmf combination defined in featuremap. Is there a way to enable them? Or is there any other way for the agent to do an attended transfer? Thanks. |
| 17:10.15 | *** join/#asterisk Qwell (north@asterisk/developer/Qwell) |
| 17:10.15 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:25.33 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:71:4121:9e83:2f2f:972) |
| 17:29.00 | [TK]D-Fender | Syco: Don't directly bridge them bridge AgentRequest to a local channel that will use dial to dial ANOTHER local channel adding the features you want and will then call Bridge() to snag the targeted channel |
| 17:31.00 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 17:41.49 | *** join/#asterisk wonderworld (~ww@82.113.121.98) |
| 17:46.36 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 17:46.39 | xochilpili | hi all |
| 17:47.34 | xochilpili | i have some problems trying to connect an avaya 1608-l phone to asterisk; in the documentation said that that phones uses h323; i have load ooh323 module in asterisk |
| 17:48.57 | xochilpili | but i can't figure out how to connect with the phone, it discover the asterisk; and i have setting up tftp and http with his config files ; but in some point i just got: "Discover 10.0.2.50" which is the asterisk server |
| 17:54.28 | Syco | Thank you so much. dialing a local channel with the right options solved everything! |
| 17:55.34 | [TK]D-Fender | Syco: Gotta think "sideways". Local channels is the mechanism that makes so many otherwise impossible things in * possible. |
| 18:04.01 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 18:07.37 | *** join/#asterisk danieli (~duniel@unaffiliated/danieli) |
| 18:07.58 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 18:09.07 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 18:31.38 | *** join/#asterisk mastahh (~mastahh@89.238.175.130) |
| 18:36.57 | xochilpili | [TK]D-Fender, do you have any idea how to connect an h323 phones to asterisk? |
| 18:37.43 | [TK]D-Fender | Tell it to register to * if it has top. Set your peer to allow it to register if that's the case. Set your auth right. THE END |
| 18:38.49 | xochilpili | [TK]D-Fender, is the phone, the phone said something like "connecting to callCenter" |
| 18:39.04 | xochilpili | then nothing; just reboot itself |
| 18:39.13 | [TK]D-Fender | Those words don't eman anything to me |
| 18:39.44 | [TK]D-Fender | That does no say what it it is connecting to, what it is connecting for, and why it is restarting. You aren't offering any useful information. |
| 18:40.22 | xochilpili | that's all that the phone said! I know that doesn't mean anything for me either |
| 18:43.59 | [TK]D-Fender | Time to read the phone's manual |
| 18:52.00 | *** part/#asterisk mjordan (mjordan@nat/digium/x-muwespfjtlhrrvow) |
| 18:55.08 | *** join/#asterisk rsdvd (~rsdvd@90.206.88.142) |
| 18:56.24 | rsdvd | evening all.....is there anyone around who could give me some pointers on setting up ISDN2e lines.....I have never used ISDN before and cant get them to work |
| 19:01.04 | *** join/#asterisk hatlessman (~jkitchens@162.243.150.206) |
| 19:01.26 | mastahh | Has anyone used Openfire Spark with Asterisk before? |
| 19:11.53 | *** join/#asterisk leftleg (~JBurke@nsseofirewall.nsseo.org) |
| 19:19.18 | *** join/#asterisk Zogot (~Adium@90-145-117-39.bbserv.nl) |
| 19:22.12 | *** join/#asterisk mjordan (mjordan@nat/digium/x-yczhnvgjzhtcxxek) |
| 19:22.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:27.35 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 19:35.33 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 19:36.59 | hexanol | in Asterisk 13 using chan_sip, if I have a peer with no "allow" and "disallow" option, when I do a "sip show peer <mypeer>", then I see "Codecs : (ulaw|alaw|gsm|h263)" |
| 19:37.29 | hexanol | if I add "disallow = all" and "allow = g729", then "sip show peer <mypeer>" shows "Codecs : (g729)" |
| 19:37.46 | TazzNZ | rsdvd, what hardware you connecting it with ? |
| 19:37.54 | hexanol | and if I then remove the "disallow" and "allow" option, then "sip show peer <mypeer>" shows "Codecs : (g729|ulaw|alaw|gsm|h263)" |
| 19:38.19 | hexanol | i.e. the g729 reference is still there -- I've tested with asterisk 11 and this is a new behaviour |
| 19:38.26 | hexanol | should this be considered a bug ? |
| 19:38.40 | TazzNZ | mastahh, that is just a XMPP client - what are you expecting to happen/do ? |
| 19:39.40 | TazzNZ | hexanol, if you restart asterisk. is the original behaviour restored ? |
| 19:39.46 | hexanol | yep it is |
| 19:39.52 | hatlessman | hexanol: Yeah, I saw something in jira about this, about how you can't disable a peers codecs after they've been enabled. |
| 19:39.54 | hexanol | i.e. g729 disappers from the Codecs list |
| 19:40.08 | hexanol | I've look at the jira but did not found an issue |
| 19:40.18 | hexanol | that said I sometime have a hard time searching some issues in the tracker |
| 19:40.32 | TazzNZ | hexanol, what modules are loaded before and after the reload that sets the codec ? |
| 19:40.55 | hexanol | well no new modules are loaded/unloaded |
| 19:41.00 | hexanol | i.e. I just edit sip.conf |
| 19:41.03 | hexanol | and do "sip reload" |
| 19:42.20 | rmudgett | hexanol: https://issues.asterisk.org/jira/browse/ASTERISK-25182 |
| 19:42.23 | TazzNZ | hexanol, do a "module show like codec" before sip reload |
| 19:42.27 | TazzNZ | and after |
| 19:43.10 | hexanol | rmudgett: thank you, that is indeed the "same" problem as I have |
| 19:52.28 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:10.10 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 20:14.19 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 20:19.07 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:28.03 | cmendes0101 | I've had sipml5 and asterisk 11.12.1 running for a while with transport ws. We are moving the site to https but now need to use wss. I've modified sip.conf and added the transport but not sure how access it. For the websocket url wss://IP:8088/wss but no luck |
| 20:28.46 | cmendes0101 | Before I go upgrading or any large changes was hoping to confirm the correct websocket url format if anyone has got that running |
| 20:29.38 | cmendes0101 | Error from sipml5 is failed to connect to server and I tried ws:.../wss and other way around |
| 20:32.48 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 20:32.54 | aquaguy | Hello! |
| 20:36.35 | aquaguy | Is there a way to play an audio file every 30 seconds? I'm playing some music on hold until an agent picks up the call I would like to tell the client that we are busy and that we'll take his call as soon as possible and play this message every 30 secs. How can this be done? |
| 20:39.19 | TazzNZ | aquaguy - http://www.voip-info.org/wiki/view/Asterisk+cmd+MusicOnHold |
| 20:39.40 | *** join/#asterisk d00gster (~d00gster@ERMLON6511W-LP130-05-1177962827.dsl.bell.ca) |
| 20:39.43 | TazzNZ | so you could do - MusicOnHold (default,30) |
| 20:39.52 | TazzNZ | then play the announcement, and then loop back |
| 20:40.42 | aquaguy | TazzNZ: All right, thanks! |
| 20:44.58 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:47.54 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 20:48.51 | aquaguy | TazzNZ: one last thing, how would you loop back with Goto? |
| 20:52.42 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 20:53.30 | *** join/#asterisk petris (~petris@107.191.103.74) |
| 20:55.56 | TazzNZ | aquaguy, depends on how you are using the queue etc ? |
| 21:02.50 | *** join/#asterisk matkam11 (~mkaminski@173-14-158-249-NewEngland.hfc.comcastbusiness.net) |
| 21:03.17 | *** join/#asterisk petris (~petris@2604:180:0:ad7::73db) |
| 21:04.36 | matkam11 | Hey I was wondering if I could get some help with asterisk13 and corosync. I've been trying to build it and now that I finally have the RPM built, it seems to just be dying when it tries to load res_corosync |
| 21:15.21 | hatlessman | cmendes0101: Everything but the /ws (thas the same for ws and wss) is configured in http.conf |
| 21:16.00 | *** join/#asterisk d00gster (~d00gster@ERMLON6511W-LP130-05-1177962827.dsl.bell.ca) |
| 21:20.15 | *** join/#asterisk petris (~petris@2604:180:0:ad7::73db) |
| 21:24.46 | *** join/#asterisk rsdvd (~rsdvd@90.206.88.142) |
| 21:25.57 | rsdvd | TazzNZ......the card is a Xorcom XPD Astriband2 |
| 21:27.32 | *** join/#asterisk petris (~petris@107.191.103.74) |
| 21:27.38 | *** join/#asterisk Chex (~Chex@swampjax.northnook.ca) |
| 21:30.03 | TazzNZ | rsdvd, have you installed DAHDI ? |
| 21:30.41 | rsdvd | I think so - I am taking over from someone else who set the box up - was supposed to come as a ready to go applicance but seems it wasnt |
| 21:30.58 | TazzNZ | do you have CLI access to the server ? |
| 21:31.27 | rsdvd | I did have - I have lost the VPN - I will see if I can get it re-enabled |
| 21:31.36 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 21:32.12 | aquaguy | Is there a way to enable only certain MAC addresses to register? I'm getting a bunch of registration attempts from 65.182.108.179 and 199.191.59.114. I can't just allow registration from local network because some of my extensions are cellphones but I would like to protect myself from this kind of attacks. Like allowing only to register if they much just a certain MAC address. |
| 21:32.42 | aquaguy | match * |
| 21:33.23 | TazzNZ | aquaguy, I would use fail2ban which will ban IP's that fail for a set amount of time |
| 21:33.35 | TazzNZ | it then un-ban's them |
| 21:34.40 | aquaguy | ok thanks for the advice |
| 21:36.08 | WIMPy | rsdvd: http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 21:36.12 | TazzNZ | aquaguy, also make sure that anonymous calls go where you want them to go |
| 21:36.12 | *** join/#asterisk petris (~petris@107.191.103.74) |
| 21:36.19 | rsdvd | TazzNZ - I am at the CLI now |
| 21:36.45 | TazzNZ | rsdvd, what WIMPy pasted is a good read :) |
| 21:37.09 | aquaguy | TazzNZ: you mean unidentified incoming calls? Like probable attackers or spammers? |
| 21:37.14 | rsdvd | I will read it - but getting punch drunk with all this info |
| 21:37.22 | WIMPy | Well, Not so much if you already have the hardware. |
| 21:37.38 | rsdvd | how can I check dahdi is installed and workling |
| 21:37.52 | TazzNZ | aquaguy, spammers will come from "know sources", if you and I call a spammer the same thing. |
| 21:38.05 | WIMPy | I stumbled across my Astribank last week. Maybe I should try to get that going again? |
| 21:38.12 | TazzNZ | aquaguy, my "warning" is more about people using your PBX to make toll calls |
| 21:39.03 | WIMPy | rsdvd: Did you build it? |
| 21:39.12 | matkam11 | Anyone know what the status is of the corosync module with asterisk 13? |
| 21:39.16 | TazzNZ | I normally configure my PBX with a default "noop" context, and place all my peers/users/etc out of that context |
| 21:40.04 | rsdvd | WIMPy......no it was a readymade server from completePBX.....I have beeb dragged in to try and get it running as the project it now veryt late |
| 21:40.47 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 21:43.26 | rsdvd | TazzNZ / WIMPy - can you give me any more advice ? I am reading thru that doc but any assistance would be greatly appreciated |
| 21:43.57 | TazzNZ | rsdvd, it does help if you ask specific questions :) |
| 21:44.06 | WIMPy | Indeed. |
| 21:44.25 | WIMPy | But it sounds like you got yourself ito trobles water ther. |
| 21:44.41 | rsdvd | TazzNZ - Im sorry Im being vauge......but been thrown in at the deepend |
| 21:45.04 | TazzNZ | rsdvd, we've all been there :) |
| 21:45.19 | TazzNZ | but try it yourself and we will pull you out of the mud :) |
| 21:46.13 | rsdvd | lol - I am trying :-) I have never touched an ISDN line before lol ....... all the installs of Asterisk I have done used AIX/SIP trunks |
| 21:46.53 | WIMPy | Well, it can be massy with Asterisk. |
| 21:47.19 | WIMPy | But at least ist works at all, which doesn't seem to really be the case with other projects. |
| 21:47.24 | rsdvd | :-( so I have noticed |
| 21:48.15 | WIMPy | But I still ave no clue what you're trying to do. |
| 21:49.47 | rsdvd | I am trying to set up an 8 channel ISDN to be the trunk for my outbound routes |
| 21:50.10 | WIMPy | 8 channel or 8 lines? |
| 21:50.17 | rsdvd | 8 lines sorry |
| 21:50.24 | WIMPy | But the question ist more what it is you've got there. |
| 21:51.03 | WIMPy | Wow. Wouldn't an ISDN30 have been cheaper then? |
| 21:51.07 | *** join/#asterisk Chotaire (~chotaire@vegetarian.cannibal.club) |
| 21:51.37 | rsdvd | i didnt spec or buy this - I would not have used ISDN at all cause I have no experience with it |
| 21:52.02 | TazzNZ | isdn30 ? |
| 21:52.10 | WIMPy | Unless you use Asterisk, you don't have to know anythign about it. |
| 21:52.20 | TazzNZ | is isdn8 when they only activate 8 channels ? |
| 21:52.27 | WIMPy | The BT term for a PRI. |
| 21:52.42 | WIMPy | No, he said 8 lines. That would make 16 channels. |
| 21:52.46 | TazzNZ | ffs - why can't they all just call it the same ! |
| 21:52.55 | TazzNZ | so 8* BRI ? |
| 21:53.22 | rsdvd | Its PRI from BT as far as I know |
| 21:53.34 | WIMPy | But as google tells me CompletePBX is a Xorcom thing. So that probably has Astribank support. |
| 21:53.50 | rsdvd | it is Xorcom |
| 21:53.58 | WIMPy | You sid ISDN2e that wouls be a DSS1 BRI |
| 21:54.43 | WIMPy | Im not sure if anyone here has ever seen their configuration interface. I didn't even know they did appliances so far. |
| 21:55.27 | rsdvd | I expect this to be a plug and go system (should be for the price) but its not configured at all |
| 21:55.46 | rsdvd | the insterface loos like like the normal ASterisk web GUI |
| 21:56.03 | WIMPy | Whatever that is. |
| 21:56.05 | WIMPy | FreePBX? |
| 21:56.25 | WIMPy | The screenshot I see on their page does not look familiar in any way to me. |
| 21:56.26 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 21:56.34 | *** join/#asterisk marlinc (~marlinc@ip2.weert.li.nl.cvo-technologies.com) |
| 21:58.00 | *** join/#asterisk Qwell (~north@asterisk/developer/Qwell) |
| 21:58.00 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 21:58.53 | rsdvd | from what I can see the dadhi is installed and workign (I think) |
| 21:58.53 | WIMPy | But first of all, they say it's simple. So why don't you have it working, yet? :-) |
| 21:59.32 | rsdvd | lol - anything but simple so far |
| 22:00.38 | WIMPy | The trouble is that you're rather unlikely to find more than one person who knows how to configure that thing. |
| 22:01.10 | WIMPy | So you should probably go an read their documentation on setting it up. |
| 22:01.23 | rsdvd | there is no docs from them about ISDN |
| 22:02.22 | WIMPy | Sounds strage that they sell a "PBX" and their own ISDN interfaces without telling you how to use it. |
| 22:03.33 | rsdvd | yes - very strange - I wont buy another system from them |
| 22:05.08 | WIMPy | Trouble is, if you're supposed to configure it by clicking buttons, it most likely won't help if we tell you how to write config files. Either vecause they are organized in a different way, |
| 22:05.24 | WIMPy | but mostly because they will probably get overwritten on the next click anyway. |
| 22:05.58 | WIMPy | So yu reallt need to stick to what they provide. |
| 22:08.04 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:12.34 | rsdvd | WIMPy......I give up and gonna read more internet ..... thanks for the help and when i have a proper question I will come back |
| 22:13.07 | WIMPy | Nothing obvious to click? |
| 22:17.20 | rsdvd | wimpy - not really - I think I have it all set up right now.....I have the 'pri show span' shows that 2 of the lines are now up. I will have to go to site to check I can dial out but not doing that at 23.20 :-) |
| 22:17.42 | rsdvd | thanks for the help - no doubt I will be back with questions soon :-) |
| 22:17.53 | WIMPy | Use 'channel originate' to test. |
| 22:19.51 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 22:22.19 | *** join/#asterisk zz_mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 22:22.27 | *** join/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 22:22.28 | rsdvd | thanks WIMPy -that worked :-) I now have the ISDN outbound working :-) |
| 22:22.54 | rsdvd | that is enough for 1 day - inbound tomorrow once we turn off the divert on the phone number |
| 22:23.03 | TazzNZ | rsdvd, now you just need to route your calls that way :) |
| 22:23.53 | WIMPy | Take a phone with you. |
| 22:24.05 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 22:24.29 | WIMPy | None of the current Linux drivers allow you to manipulate forwarding. That only worked with I4L :-( |
| 22:25.35 | *** join/#asterisk ketas (~ketas@226-209-191-90.dyn.estpak.ee) |
| 22:27.43 | WIMPy | Well, it still does, but... |
| 22:28.29 | WIMPy | it's the only use left for it. |
| 22:30.11 | *** join/#asterisk Chotaire (~chotaire@vegetarian.cannibal.club) |
| 22:50.13 | *** join/#asterisk averythomas (~averythom@2607:5300:60:2d42::1) |
| 22:51.09 | *** join/#asterisk Chotaire (~chotaire@vegetarian.cannibal.club) |
| 22:58.03 | *** join/#asterisk startledmarmot (~startledm@cpe-23-242-27-45.socal.res.rr.com) |
| 23:10.43 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fuyxwxnafifjugek) |
| 23:14.23 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 23:22.32 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 23:33.33 | UncleKiwi | hi there, is there an easy way for asterisk to run a program upon being unable to register ie 'ie sip show registry' |
| 23:34.18 | UncleKiwi | im wanting to make a watchdog type thingy |
| 23:35.13 | *** join/#asterisk Chotaire (~chotaire@vegetarian.cannibal.club) |
| 23:37.17 | UncleKiwi | I know I can do with using AMI but it feels like overkill |
| 23:38.54 | UncleKiwi | but maybe its what i have to do |
| 23:44.20 | [TK]D-Fender | Define "being unable to register" |
| 23:45.27 | UncleKiwi | where im registering a number with a SIP provider and they send me calls |
| 23:45.35 | UncleKiwi | but it times out |
| 23:45.38 | UncleKiwi | for example |
| 23:46.33 | [TK]D-Fender | You could try to do this off a log scraper a-la fail2ban |
| 23:46.45 | [TK]D-Fender | but that's kinda worse than just using an AMI watchdog |
| 23:47.01 | UncleKiwi | something odd happens to my router about once per month |
| 23:47.05 | UncleKiwi | or 6 weeks |
| 23:47.12 | UncleKiwi | and it needs a reboot |
| 23:47.19 | UncleKiwi | before asterisk will register |
| 23:47.24 | UncleKiwi | its not the router |
| 23:47.53 | UncleKiwi | but i know if i can make the watchdog reboot the router that will keep everyone happy |
| 23:48.24 | UncleKiwi | hmm fail2ban |
| 23:48.26 | UncleKiwi | maybe |
| 23:50.17 | UncleKiwi | sorry the router needs the reboot |
| 23:50.43 | UncleKiwi | I dont know why and i dont think the ISP wants to help me on the matter |
| 23:50.49 | UncleKiwi | they are a wireless ISP |
| 23:51.00 | UncleKiwi | i think its something in their network causing it |
| 23:59.37 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |