| 00:00.31 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 00:26.10 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:41.10 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 00:48.01 | *** join/#asterisk jmordica (uid18332@gateway/web/irccloud.com/x-yirpolofdceulraa) |
| 00:48.26 | jmordica | Anyone around? |
| 00:50.08 | jmordica | I use adaptive jitter buffer on 13.4 and I'm wondering why it adapts quickly on the way up (during network congestion) but it is very slow to come back down to normal. |
| 00:50.44 | jmordica | It's not such a big deal but it causes the audio to be garbled while it adjust to different thresholds. |
| 00:51.22 | jmordica | Is there another param I can set to speed it up on the way back down (after network congestion is resolved)? |
| 00:55.08 | jmordica | https://www.irccloud.com/pastebin/QRMz068O/ |
| 01:12.18 | *** join/#asterisk bjhaid (49b001ef@gateway/web/freenode/ip.73.176.1.239) |
| 01:12.21 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-znrltgubpnecccyf) |
| 01:12.40 | bjhaid | hi |
| 01:12.59 | bjhaid | I am playing with asterisk, I can get local calls to work fine |
| 01:13.52 | bjhaid | however when I try to make calls to a mobile network (I am trunking with a sip provider telbo), I can clearly hear what is being said on the other end, but the other end wouldn't here me |
| 01:15.24 | bjhaid | I am running Asterisk 11.13.1~dfsg-2+b1 on Debian Jessie |
| 01:31.33 | *** join/#asterisk mbecroft (~user@ak2.becroft.co.nz) |
| 01:48.23 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-237-81.hhui7.cht.bigpond.net.au) |
| 01:49.03 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-237-81.hhui7.cht.bigpond.net.au) |
| 02:02.10 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 02:05.52 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 02:14.09 | *** join/#asterisk Dovid (~dovid@ool-4356e96f.dyn.optonline.net) |
| 02:25.47 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-98-234.dynamic.qsc.de) |
| 03:09.59 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:10.53 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 03:14.57 | *** join/#asterisk Chotaire (~chotaire@vegetarian.cannibal.club) |
| 03:23.36 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-jfvdftadnpluvwzv) |
| 03:56.17 | *** join/#asterisk jermey_g (jermey_g@c-099c72d5.021-183-73746f3.cust.bredbandsbolaget.se) |
| 04:42.25 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 05:05.04 | *** join/#asterisk no2pencil (~no2pencil@direct2.akroncdnr.com) |
| 05:58.15 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:14.56 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 06:33.50 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 06:48.15 | *** join/#asterisk Milos (~milosivan@pdpc/supporter/student/milos) |
| 06:48.28 | Milos | erm |
| 06:48.36 | Milos | [Jul 12 18:38:46] NOTICE[2847][C-00000151] chan_sip.c: Call from '' (192.187.109.154:5070) to extension '20401148422885410' rejected because extension not found in context 'default'. |
| 06:48.40 | Milos | this looks concerning |
| 06:48.52 | Milos | did someone manage to get into my asterisk server? |
| 06:49.32 | Milos | I don't have 5060 open... |
| 06:49.42 | Milos | nor do I see any strange IPs in sip show peers |
| 06:49.49 | Milos | but somehow that log is making me think that someone was authorised to make a phone call? |
| 06:57.56 | [TK]D-Fender | The call was authorized and only failed because you aren't set up to process the number they passed you |
| 06:58.02 | [TK]D-Fender | Which is not good. |
| 06:58.48 | Milos | yeah |
| 06:58.55 | Milos | that's exactly what I thought |
| 06:59.01 | Milos | so now I need to figure out how this is happening |
| 06:59.14 | Milos | they are actively trying to make calls, every 6 or so minutes for the last 48 hours or so |
| 07:01.56 | [TK]D-Fender | You're allowing un-authed call. |
| 07:01.59 | [TK]D-Fender | So disable that |
| 07:02.29 | Milos | ah |
| 07:02.40 | [TK]D-Fender | allowguest=no under [general] |
| 07:02.50 | Milos | but... how are they initiating the connection? I don't allow inbound connection to 5060 directly from the Internet |
| 07:02.51 | [TK]D-Fender | And then firswall them out |
| 07:03.12 | [TK]D-Fender | You are |
| 07:03.35 | Milos | I really am not. |
| 07:04.20 | [TK]D-Fender | we can see where it's coming from |
| 07:04.23 | [TK]D-Fender | It isn't lying. |
| 07:04.49 | [TK]D-Fender | chatever port chan_sip is listening on it getting the packets |
| 07:04.49 | Milos | I understand your logic and I'm not saying it's wrong, but I'm not allowing inbound connection. I just tried. |
| 07:05.56 | [TK]D-Fender | perhaps you need to adjust your testing and evaluation |
| 07:06.19 | Milos | https://bpaste.net/show/b82667a7c19a |
| 07:06.27 | Milos | one sec |
| 07:08.47 | Milos | where's chan_sip set? I'm 100% sure it's 5060 already, but it's not defined in the config anywhere so I assume it's being defaulted |
| 07:10.06 | [TK]D-Fender | udp 0 0 0.0.0.0:5060 0.0.0.0:* 2798/asterisk |
| 07:10.17 | Milos | cool |
| 07:10.37 | [TK]D-Fender | Now your netstat proves nothing about your claim that nothing is allowed from the internet |
| 07:10.48 | Milos | never said it did |
| 07:11.01 | Milos | this only started yesterday it seems, I do have 3 WAN connections on this box, but it's all firewalled as far as I can tell |
| 07:11.39 | [TK]D-Fender | Look closer |
| 07:11.43 | Milos | I'll check my conntrack table to see if there's anything in there |
| 07:12.24 | Milos | gw1 ~ # conntrack -L | grep 192.187.109.154 |
| 07:12.24 | Milos | udp 17 1817 src=192.187.109.154 dst=103.52.207.55 sport=5071 dport=5060 src=103.52.207.55 dst=192.187.109.154 sport=5060 dport=5071 [ASSURED] mark=4 secctx=null helper=sip use=1 |
| 07:12.24 | Milos | udp 17 3374 src=192.187.109.154 dst=103.52.207.55 sport=5070 dport=5060 src=103.52.207.55 dst=192.187.109.154 sport=5060 dport=5070 [ASSURED] mark=4 secctx=null helper=sip use=1 |
| 07:12.29 | Milos | okay, it's there |
| 07:13.16 | Milos | it's possible that was established while I was messing with the firewall yesterday, I don't remember if I temporarily disabled it |
| 07:13.30 | Milos | so it would be allowed an established,related entry |
| 07:13.44 | [TK]D-Fender | best ditching conntrack entirely |
| 07:13.53 | [TK]D-Fender | UDP is stateless anyway |
| 07:14.00 | Milos | doesn't matter, it still has a state with conntrack |
| 07:14.14 | Milos | I can just kill the conntrack session but that would be no fun |
| 07:14.45 | [TK]D-Fender | trace it back and ICBM = fun |
| 07:14.54 | Milos | :D |
| 07:16.18 | [TK]D-Fender | Alrighty, bed time here. Go get 'em |
| 07:33.16 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:47.27 | *** join/#asterisk D30 (~D30@58.71.19.178) |
| 08:31.47 | *** join/#asterisk Megagolgoth (~Megagolgo@AMontsouris-652-1-192-16.w90-24.abo.wanadoo.fr) |
| 08:34.00 | *** join/#asterisk dadrc (~quassel@unaffiliated/dadrc) |
| 09:00.42 | *** join/#asterisk johnny_|_ (~johnny@unaffiliated/johnny-/x-2623418) |
| 09:42.32 | *** join/#asterisk pchero (~pchero@0x555140b5.adsl.cybercity.dk) |
| 11:50.45 | *** join/#asterisk asterisker (2ed1c983@gateway/web/cgi-irc/kiwiirc.com/ip.46.209.201.131) |
| 11:51.46 | *** join/#asterisk bluenemo (~bluenemo@unaffiliated/bluenemo) |
| 11:52.07 | asterisker | hi fellas |
| 11:58.22 | asterisker | there's a problem with our compony asterisk system |
| 11:58.23 | asterisker | this is the code: |
| 11:58.23 | asterisker | <PROTECTED> |
| 11:58.23 | asterisker | <PROTECTED> |
| 11:58.23 | asterisker | <PROTECTED> |
| 11:58.23 | asterisker | <PROTECTED> |
| 11:58.23 | asterisker | i set it to playback some voice when inbound calls come to this queue |
| 11:58.24 | asterisker | but the voice play to agent who recieve that call. |
| 11:58.24 | asterisker | anyone can help? |
| 12:00.37 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 12:03.39 | asterisker | info |
| 12:03.42 | asterisker | info help |
| 12:04.05 | WIMPy | ~pb |
| 12:04.05 | 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. |
| 12:06.45 | asterisker | http://pastebin.com/rNTTxFh6 |
| 12:08.34 | WIMPy | So you want to play something to the caller? When? |
| 12:09.40 | asterisker | right before answer by agent |
| 12:16.59 | asterisker | no one? |
| 12:33.56 | Cyford33 | your using freepbx and set it up wrong |
| 13:00.02 | *** join/#asterisk Dovid (~dovid@ool-4356e96f.dyn.optonline.net) |
| 13:05.57 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 13:21.19 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 13:38.10 | *** join/#asterisk CeBe (~CeBe@port-92-200-98-234.dynamic.qsc.de) |
| 13:40.45 | *** join/#asterisk karelk (~karel@84-72-164-65.dclient.hispeed.ch) |
| 13:46.42 | *** join/#asterisk Cyford33_y (~cyford@c-73-137-1-6.hsd1.ga.comcast.net) |
| 13:47.01 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 13:47.27 | *** join/#asterisk mlhess (~mlhess@drupal.org/user/102818/view) |
| 14:43.48 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 14:49.52 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5F3F7.dip0.t-ipconnect.de) |
| 15:10.58 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:17.52 | *** join/#asterisk n3ob (~ed@2a00:d880:6:320:82fa:b33e:3d20:4763) |
| 15:32.43 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 15:32.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:37.52 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 15:41.49 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 15:45.27 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 15:46.21 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 16:10.23 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 16:10.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:24.54 | *** join/#asterisk gusto (~gusto@2001:4c50:62f:ad13:8219:34ff:fecf:17f0) |
| 16:28.26 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 16:41.26 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 16:45.19 | *** join/#asterisk Megagolgoth (~Megagolgo@AMontsouris-652-1-39-197.w83-202.abo.wanadoo.fr) |
| 17:26.42 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 17:52.42 | *** join/#asterisk crooksey (~smuxi@host86-169-48-89.range86-169.btcentralplus.com) |
| 17:54.51 | crooksey | Hi guys, at work the powers that be have decided to migrate to VOIP, and have asked us (network admins) to evaluate the possibilty of hosting/running this in house. We have say 1 public phone number that routes to a switchboard, then can be picked up by X amount of users in the office. Then whenever outgoing calls are made from a VOIP handset, they use the same number (so caller ID on the end user phone shows the same |
| 17:54.51 | crooksey | number as the one called in on). |
| 17:56.14 | crooksey | Then we have 20 mobile users, if migrating to VOIP can we also route all mobile calls through the VOIP server? |
| 17:56.28 | crooksey | So I am fairly confident we could setup and configure the local phones and handsets, its more a question of the mobile users where I am struggling to find much documentation |
| 17:56.43 | [TK]D-Fender | What do you define as "mobile calls"? |
| 17:57.10 | nbjoerg | cell phones? |
| 17:57.18 | [TK]D-Fender | that is the DEVICE |
| 17:57.25 | [TK]D-Fender | What NETOWRK are you referring to? |
| 17:58.20 | crooksey | Ok, so re-phrase.. Can I route all calls from say an iphone we have, through the asterisk box, then on to the final destination e.g any landline number or cell number? |
| 17:59.29 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 17:59.50 | [TK]D-Fender | again, not clear abouty how the iphone is talking to anything yet |
| 17:59.52 | nbjoerg | crooksey: I take it you don't want to use a soft-phone app |
| 18:00.30 | [TK]D-Fender | nbjoerg, He isn't clear about what he's talking about yet. Let's wait till we get an actual answer |
| 18:00.36 | nbjoerg | yeah |
| 18:00.44 | crooksey | nbjoerg: yea I don't want to if poss, but I have just realised data connections will be a problem |
| 18:00.57 | crooksey | Let me start from the top.. |
| 18:01.46 | crooksey | So problem, we have a landline number in the UK, we have an office with 10 handsets. |
| 18:02.30 | crooksey | We want all calls that hit that number to route through the asterisk box, certain handsets we speicfy to be on the "group" line |
| 18:02.50 | crooksey | Some phones can be called directly via an extension |
| 18:03.16 | crooksey | If a certain extension is used, it routes through to a cell number |
| 18:03.31 | crooksey | All calls handled through the box must be recoreded |
| 18:03.38 | [TK]D-Fender | that sounds more like TO a cell number and not THROUGH it. |
| 18:04.00 | nbjoerg | (or FROM) |
| 18:04.06 | crooksey | Yea, but we want to record all calls routed through the box too a cell |
| 18:04.29 | nbjoerg | that's more or a less just a normal outgoing call from your PBX |
| 18:04.35 | [TK]D-Fender | so you want your inbound call to dial OUT to a cell phone via the standard PSTN? |
| 18:04.43 | crooksey | yES |
| 18:04.56 | [TK]D-Fender | That is no different than dialing anything else |
| 18:05.01 | crooksey | OK |
| 18:05.17 | [TK]D-Fender | you just have choose HOW you call it |
| 18:05.44 | [TK]D-Fender | What are you planning on calling it via? |
| 18:05.49 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:05.49 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:06.10 | crooksey | [TK]D-Fender: could you explain? |
| 18:06.25 | crooksey | Normal SIP I guess |
| 18:06.27 | [TK]D-Fender | That iPhone isn't going to starting ringing via magicv |
| 18:06.35 | [TK]D-Fender | What SERVICE are you planning on using to call it? |
| 18:06.57 | crooksey | I don't know, I have just started evaluating VOIP |
| 18:07.21 | crooksey | Not quite sure what connects to what etc |
| 18:07.27 | [TK]D-Fender | How are we even talking cvoip yet? |
| 18:07.36 | [TK]D-Fender | We talking about a call going from your SERVER to your PHONE.\ |
| 18:07.42 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 18:08.06 | [TK]D-Fender | Are you planning on calling the iPhone via the REGULAR CELLULAR NETWORK? |
| 18:08.19 | nbjoerg | how is your server getting the call in first place? |
| 18:08.23 | [TK]D-Fender | Or via it's DATA connection? |
| 18:08.29 | crooksey | nbjoerg: SIP? |
| 18:08.34 | nbjoerg | ok |
| 18:08.43 | nbjoerg | (could have been via phone hardware too) |
| 18:09.25 | crooksey | Well my landline number will be registered with my SIP provider, which will then forward all calls from the landline number to my server right? |
| 18:11.14 | [TK]D-Fender | yes |
| 18:11.21 | crooksey | Cool. |
| 18:11.27 | [TK]D-Fender | that part isn't the important part of this scenario |
| 18:11.44 | [TK]D-Fender | this is easy part. |
| 18:11.46 | crooksey | And then outgoing calls from a VOIP phone, send data to the SIP provider which then makes the call? |
| 18:11.50 | crooksey | (for basics) |
| 18:11.58 | [TK]D-Fender | You need to be clear abot how the PBX will call those mobile devices |
| 18:12.07 | crooksey | Yea, thats my concern |
| 18:12.20 | [TK]D-Fender | it shouldn't be a concern |
| 18:12.24 | [TK]D-Fender | It should be a DECISION |
| 18:12.37 | crooksey | Whats the most common way, as the mobile devices must answer the call via a regular cell signal |
| 18:13.00 | [TK]D-Fender | Are you going to have your PBX call them via the PSTN so that it arrives on the VOICE connection it has with the carrier they are on? |
| 18:13.06 | [TK]D-Fender | there is no common |
| 18:13.10 | [TK]D-Fender | This ideaa does not exist |
| 18:13.16 | [TK]D-Fender | YOU have to choose how YOU want to do things |
| 18:13.22 | crooksey | ok |
| 18:13.42 | [TK]D-Fender | Those devices are DUMB PHONES. They have phone numbers directly on them (as delivered via a carrier) |
| 18:13.47 | [TK]D-Fender | That is ONE way |
| 18:13.54 | crooksey | So via the PSTN i guess |
| 18:14.07 | [TK]D-Fender | They also could have DATA plans on the or you might expect them to hop onto WIFI NETWORKS |
| 18:14.14 | [TK]D-Fender | that is a completely DIFFERNT way |
| 18:14.32 | [TK]D-Fender | And you need to have picked how you intend to do this |
| 18:14.58 | [TK]D-Fender | If you are calling them via their associated PSTN number then your server is calling the PSTN via some service. The iPhone doesn't matter in this case |
| 18:15.13 | crooksey | ok cool |
| 18:15.24 | [TK]D-Fender | If you call them using DATA, then that would be voip DIRECTLY between your server and the iPhone and the iPhone would have to be running a VoIP CLIENT |
| 18:15.32 | [TK]D-Fender | So you have to CHOOSE |
| 18:15.37 | crooksey | No, PSTN |
| 18:16.02 | [TK]D-Fender | then the iphone doesn't matter in this question. It's just your server calling out to a number on the PSTN |
| 18:16.09 | [TK]D-Fender | And it could be ANY number for all that matters |
| 18:16.18 | crooksey | OK perfect, thats that sorted |
| 18:16.32 | [TK]D-Fender | So the fact of them being "mobile" or "iPhone" has no impact on this |
| 18:17.03 | [TK]D-Fender | So yes, your server can get calls in for an ITSP and you can call out al you want via services like that as well. |
| 18:17.09 | [TK]D-Fender | ~itsp |
| 18:17.09 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:19.17 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-hsqmlezqeoqcqqdy) |
| 18:26.28 | *** join/#asterisk Cyford33_y (~cyford@c-73-137-1-6.hsd1.ga.comcast.net) |
| 18:29.54 | WangDang | ~itsplist-ca |
| 18:29.55 | infobot | somebody said itsplist-ca was Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 18:54.22 | *** join/#asterisk johnny_|_ (~johnny@unaffiliated/johnny-/x-2623418) |
| 19:07.44 | *** join/#asterisk TazzNZ (~pieter@office-nat2.ytechops.co.nz) |
| 19:42.28 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 19:42.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:48.53 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 19:48.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:57.28 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 19:57.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:01.53 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 20:02.50 | *** join/#asterisk Dovid (~dovid@ool-4356e96f.dyn.optonline.net) |
| 20:05.00 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 20:05.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:12.50 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 20:12.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:20.08 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 20:20.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:24.17 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 20:24.30 | xochilpili | hi all |
| 20:25.26 | WIMPy | lo you |
| 20:25.42 | xochilpili | i have a question: why do i receive 'Receiver incomming SIP connection from unknown peer to 9900972599932957" ? |
| 20:26.02 | xochilpili | i have "allow guest <no>" in freepbx |
| 20:26.07 | WIMPy | Because someone tries to call that number via your system. |
| 20:26.29 | WIMPy | Doesn't seem to work then. But that's for #freepbx. |
| 20:26.38 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 20:26.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:27.45 | xochilpili | WIMPy, but it came from some ip which i dont recognize |
| 20:27.51 | xochilpili | cames |
| 20:28.58 | WIMPy | Looks like you do allow guests. |
| 20:29.05 | WIMPy | Or FreePBX does. |
| 20:29.38 | xochilpili | WIMPy, yes, you're right, i saw the sip.conf |
| 20:29.43 | xochilpili | it was enabled |
| 20:29.46 | xochilpili | thanks :D |
| 20:33.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:34.20 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 20:34.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:34.51 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:40.21 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-irlowbxzjmszbghg) |
| 20:54.06 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 20:54.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:17.02 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 21:22.19 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 21:22.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:24.15 | *** join/#asterisk jdgirard (~quassel@232.90.50.123.dsl.dyn.mana.pf) |
| 21:38.53 | *** join/#asterisk monsterco (~monsterco@69.90.218.42) |
| 21:40.56 | *** part/#asterisk monsterco (~monsterco@69.90.218.42) |
| 21:46.35 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 21:46.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:59.31 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 21:59.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:01.06 | *** join/#asterisk jonathono (~jonathon@2605:6000:101a:8032:b408:b9c9:b891:66dd) |
| 22:02.19 | jonathono | Anyone using Simon Telephonics SIP to Google Voice gateway? I'm getting all circuits are busy on a new setup. |
| 22:02.39 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:03.25 | *** part/#asterisk Simon-- (~sim@2606:6a00:0:28:5604:a6ff:fe02:702b) |
| 22:08.20 | jonathono | Simon Telephonics shows google voice account authorized and device registered |
| 22:08.50 | jonathono | sip show peers trunk status = ok |
| 22:14.02 | *** join/#asterisk pchero (~pchero@0x555140b5.adsl.cybercity.dk) |
| 22:14.40 | [TK]D-Fender | none of those messages means anything. |
| 22:14.50 | [TK]D-Fender | Enable SIP debug and look at what is actually happening |
| 22:17.13 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 22:22.26 | jonathono | Ok I think this is the relevant error: TRUNK: SIP/simonics DISABLED |
| 22:31.23 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 22:36.00 | jonathono | If I set my vitelity trunk as secondary on the outbound route, it is used, and the call connects. |
| 22:57.00 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 22:57.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:12.24 | lvlinux | ~book |
| 23:12.24 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:34.00 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 23:37.00 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 23:38.37 | *** part/#asterisk jonathono (~jonathon@2605:6000:101a:8032:b408:b9c9:b891:66dd) |
| 23:42.02 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-gsyhiuapvvwrbhjz) |