| 00:05.11 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 00:05.42 | *** join/#asterisk rpifan (~rpi@73.106.75.174) |
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| 00:26.31 | *** join/#asterisk Phrohdoh (18b22005@gateway/web/cgi-irc/kiwiirc.com/ip.24.178.32.5) |
| 00:27.48 | Phrohdoh | So I'm trying to write a custom application with http://www.voip-info.org/wiki/view/Asterisk+.NET but I cannot figure out how to tell asterisk to launch my resulting executable via mono. Does my `.agi` need to be a script file with a shebang, or can I just invoke `mono` directly in said file with the params given? |
| 00:31.00 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
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| 01:00.11 | Samot | No idea. |
| 01:26.35 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 01:32.21 | snadge | scenario.. yealink phone, cisco meraki natted via public internet to asterisk on public internet.. extension is set to DirectMedia: No Force rport: Yes Symmetric RTP: Yes |
| 01:33.43 | snadge | everything seems to be working fine.. except A party calls into B party (the yealink behind the meraki).. audio is fine.. another call comes in to B party, they put 1st call on hold.. speak to 2nd caller.. hang up.. resume 1st call, one way audio |
| 01:53.11 | Samot | Which way? |
| 01:53.13 | Samot | Inbound? |
| 01:55.53 | Samot | If it's inbound, it is a NAT issue. |
| 02:12.47 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
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| 07:16.01 | bittis | goodmorning everyone |
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| 08:57.56 | wyoung | oh hai there bittis ! |
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| 10:38.05 | catphish | is it possible to disable T.38 and use spandsp/receiveFax over SIP in audio mode? |
| 10:49.54 | catphish | i suspect just setting "t38pt_" options to no will do it |
| 10:53.03 | *** join/#asterisk cka_ (~cka@static-5-103-15-252.energifyn.net) |
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| 12:29.53 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 12:31.01 | bounceman | How does one best receive both dtls/SRTP and unencrypted SIP/RTP on the same asterisk? When it originates from the same peer? |
| 12:31.29 | bounceman | GW <--SIP+DTLS+SRTP+RTP-->Asterisk |
| 12:31.33 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 12:32.32 | *** part/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 12:33.02 | bounceman | I mean I know I need to have two seperate contextes, but when the PEER is the same I am not sure how to do it. |
| 12:35.17 | *** join/#asterisk WinterNight (~norbert@h238240.net.pulawy.pl) |
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| 12:47.22 | WinterNight | Hello. :) I am trying to get work Asterisk over OpenVPN on FreeBSD. In tcpdump -i tun0 I see that recives data from client but not responding. Ping from client is working properly. What can I do? |
| 12:48.34 | WinterNight | Without OpenVPN Asterisk is working properly. |
| 12:51.28 | avb | WinterNight: seems you are having a problem with a source address |
| 12:51.37 | cka | WinterNight: Perhaps Asterisk is not listening to the ip of the tun0 interface, and therefore ignores the packets? |
| 12:52.14 | avb | you are register to an ip from vpn subnet while asterisk replies with an ip from another ip |
| 12:52.19 | *** part/#asterisk cka (~cka@static-5-103-15-252.energifyn.net) |
| 12:52.26 | avb | or like cka were saying |
| 12:52.31 | *** join/#asterisk cka (~cka@static-5-103-15-252.energifyn.net) |
| 12:56.04 | WinterNight | Thank You for replys. :) In sip.conf I have bindaddr=192.168.0.1 and localnet=192.168.0.0/255.255.255.0 . OpenVPN are configured "server 192.168.0.0 255.255.255.0". What can I do more? |
| 12:56.37 | avb | sip show settings |
| 12:56.38 | cka | WinterNight: Is 192.168.0.1 the IP of the tun0 interface in the box running asterisk? |
| 12:57.35 | WinterNight | cka : Yes. |
| 12:57.54 | cka | And are your clients all in the network 192.168.0.0/24? Can you get traffic flowing in both directions? (Eg. Can you get replies when using ping from the clients that should connect to asterisk) |
| 12:57.58 | WinterNight | Ok, I will start SSH server. |
| 12:58.41 | avb | what is the point to stop it while developing? :) |
| 13:02.41 | cka | thinks peer resets too many connections... |
| 13:06.54 | *** join/#asterisk WinterNight (~norbert@h238240.net.pulawy.pl) |
| 13:07.32 | WinterNight | This is my "sip show settings": https://paste.ubuntu.com/23633388/ |
| 13:11.56 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 13:20.03 | WinterNight | I have this errors in CLI when I am connecting: https://paste.ubuntu.com/23633442/ |
| 13:20.06 | *** join/#asterisk sl4ck (~slack@85.255.235.96) |
| 13:21.12 | cka | WinterNight: Looks more like a network issue than an asterisk one. Can you ping from 192.168.0.1 to 192.168.0.6? And the other way? |
| 13:26.21 | WinterNight | Ooops. I can't. :( |
| 13:29.18 | cka | WinterNight: Then it seems like you need to check the routing tables on both sides of the tunnel. |
| 13:32.42 | *** join/#asterisk twanny796 (~user@mail.globalcapital.com.mt) |
| 13:33.05 | twanny796 | how can i see all the extensions in a particular pickup group? |
| 13:33.50 | Samot | Not sure there's a way to do that |
| 13:34.00 | bounceman | Anyone know how I can receive both DTLS/SRTP and regular unencrypted SIP/RTP on the same Asterisk from the same peer? |
| 13:34.58 | Samot | transport=tls,udp,tcp |
| 13:47.05 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
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| 14:03.52 | bounceman | Samot: how does that work if you set encryption=yes on the context? |
| 14:04.11 | Samot | What context? |
| 14:04.23 | bounceman | If you use the same context for tls and udp |
| 14:04.45 | bounceman | I believe you have to set encryption=yes, will that not mess with regular udp traffic? |
| 14:06.31 | bounceman | Also setting ICE for that peer. |
| 14:08.11 | *** join/#asterisk sarthor__ (~sarthor@unaffiliated/sarthor) |
| 14:09.20 | *** join/#asterisk seiggy (~seiggy@74.203.105.194) |
| 14:11.19 | bounceman | Samot: do you follow me? |
| 14:11.55 | Samot | I guess you would just have to, I don't know, test it. |
| 14:15.41 | *** join/#asterisk orn (~orn@rtr2.sh23.sip.is) |
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| 14:42.06 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 14:52.02 | bounceman | Samot: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio |
| 14:52.07 | bounceman | For regular udp |
| 14:52.17 | Samot | Show it. |
| 14:54.07 | bounceman | What do you want to see? |
| 14:58.52 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ccebvdoxhcpenlbr) |
| 14:58.52 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:01.48 | *** join/#asterisk Joshua___ (62664be2@gateway/web/freenode/ip.98.102.75.226) |
| 15:03.26 | Joshua___ | anyone know how to dial an extension with click to dial/TAPI driver? I have tried a comma, pipe, pound and semi. |
| 15:03.43 | Joshua___ | using AMI |
| 15:05.20 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_Originate |
| 15:05.23 | [TK]D-Fender | VERY FINE MANUALS |
| 15:05.57 | *** join/#asterisk DiskoTwist (~DiskoTwis@207.115.103.186) |
| 15:06.03 | bounceman | Samot: do you wish to see sip.conf? |
| 15:07.22 | Samot | No, I want to see a call. |
| 15:07.29 | *** join/#asterisk KaliLinuxGR (~alexandro@unaffiliated/kalilinuxgr) |
| 15:08.35 | bounceman | What good will that do in this scenario? |
| 15:09.59 | [TK]D-Fender | Juror : "Why would I want to see the store camera footage of the shooting?" |
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| 16:09.56 | *** mode/#asterisk [+o rmudgett] by ChanServ |
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| 20:19.08 | karelk | I have a SIP trunk provider |
| 20:19.14 | karelk | outgoing calls work OK, |
| 20:19.21 | karelk | but when incoming call comes, I see on Asterisk: |
| 20:19.22 | karelk | SIP/2.0 404 Not Found |
| 20:20.01 | karelk | any idea what might me wrong ? |
| 20:22.17 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 20:23.50 | *** join/#asterisk funxion (324d7d39@gateway/web/cgi-irc/kiwiirc.com/ip.50.77.125.57) |
| 20:24.44 | funxion | does anyone have asterisk working in azure? |
| 20:29.22 | [TK]D-Fender | karelk, Show the actual call with SIP debug enabled. |
| 20:29.25 | [TK]D-Fender | ~pb |
| 20:29.27 | infobot | pastebin is probably 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:29.28 | [TK]D-Fender | ^^^ |
| 20:44.21 | karelk | [TK]D-Fender: here is the log: |
| 20:44.21 | karelk | http://pastebin.com/1y1bnGzC |
| 20:45.03 | [TK]D-Fender | Looking for 41436776766 in incoming (domain 199.144.186.8) |
| 20:45.06 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 20:45.11 | [TK]D-Fender | Means exacttly what it says |
| 20:45.19 | [TK]D-Fender | You don't have a match in your dialplan to process the call |
| 20:45.25 | tomcruise | karelk dutch? |
| 20:46.05 | karelk | tomcruise: no, I have replaced real tell numbers with garbage |
| 20:46.14 | tomcruise | check |
| 20:49.56 | funxion | 192.168.1.100 > 11-27-109-98-CELEBRITY.level3.net: ICMP ip reassembly time exceeded, length 556 Any Clue? |
| 20:52.40 | karelk | [TK]D-Fender: I have following in extensions.conf: |
| 20:52.42 | karelk | exten => 0436776766,1,Answer() |
| 20:52.42 | karelk | same => n,Macro(record) |
| 20:52.42 | karelk | same => n,Set(CHANNEL(hangup_handler_push)=hangup-handler,s,1) |
| 20:52.42 | karelk | same => n,DIAL(SIP/4412&SIP/4414,60) |
| 20:52.42 | karelk | same => n,Hangup() |
| 20:53.18 | tomcruise | change 0436776766 to 41436776766 |
| 20:55.59 | karelk | thanks, it works now. |
| 20:56.00 | karelk | <PROTECTED> |
| 20:56.27 | tomcruise | cool |
| 20:57.04 | [TK]D-Fender | Turns out you need tto match what tthey send you |
| 20:57.06 | [TK]D-Fender | :) |
| 20:58.22 | karelk | yes, thanks a lot for the help. |
| 20:58.26 | karelk | You saved me a lot of gray hairs |
| 20:58.53 | tomcruise | yw :) |
| 20:59.10 | funxion | am I really the only person in here that's tried to put asterisk in azure? |
| 20:59.28 | tomcruise | i guess so funxion |
| 20:59.35 | funxion | that blows |
| 21:00.10 | tomcruise | asterisk AND azure in Google shows you're not the only one tho. |
| 21:00.21 | funxion | I know |
| 21:00.44 | funxion | and apparently toeher people have it working |
| 21:02.00 | funxion | I have internal working, Outbound Working, but inbound is working 5 our of ten calls |
| 21:02.36 | funxion | when I've had nat issues in the past the results are consistant |
| 21:04.59 | tomcruise | 5 out of 10 are working, do you see all of them in debug logs? |
| 21:05.13 | funxion | thats the problem |
| 21:05.21 | funxion | I see the invites in ngrep and tcpdump |
| 21:05.26 | funxion | but nothing in console |
| 21:05.35 | tomcruise | multiple interfaces maybe? |
| 21:05.39 | funxion | some stuff in debug but it doesnt give me much info |
| 21:05.47 | tomcruise | what does debug say? |
| 21:05.49 | funxion | http://pastebin.com/3nwavaVh |
| 21:06.03 | funxion | only one interface |
| 21:06.12 | funxion | azure nsfg is any any |
| 21:06.36 | funxion | using the same exact iptables config from another working asterisk box with same provider |
| 21:07.19 | tomcruise | hm ok, what's the time stamp of a failed call? |
| 21:07.50 | funxion | 2016-12-15 20:11:31 |
| 21:07.54 | funxion | top of the file |
| 21:08.40 | tomcruise | those are just SIP OPTIONS I think |
| 21:09.01 | funxion | that is all of the output received |
| 21:09.07 | tomcruise | qualify=yes |
| 21:09.12 | funxion | yes |
| 21:09.24 | tomcruise | so the calls arrive at the network interface but don't reach the asterisk application |
| 21:09.27 | tomcruise | how strange |
| 21:09.48 | funxion | https://thepasteb.in/p/NxhVD9XNjnqhN |
| 21:09.48 | funxion | I've never seen it before |
| 21:09.54 | tomcruise | i never played around with azure so don't know :( |
| 21:09.58 | funxion | those are the peer options outside of host |
| 21:11.19 | funxion | I'm I wrong but I remember when I had nat issues it usually affected both inbound and outbound |
| 21:13.17 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 21:14.49 | tomcruise | which asterisk version is this? did you try playing around with the nat settings? |
| 21:15.03 | tomcruise | force_rport and/or comedia |
| 21:15.10 | jeffspeff | i have a queue setup with 6 members. each 2 members has a different penalty, which groups the six members into 3 groups. i have queue rules setup to ring the next appropriate group of 2 phones. however, i have found that after both phones ring in the last group the call does not end. I continue to hear MOH even though several seconds ago asterisk informed me that both phones did not answer within the timeout. |
| 21:15.21 | funxion | Asterisk 13.7.1 |
| 21:15.23 | funxion | of course |
| 21:17.09 | tomcruise | there's 1 instance of asterisk running right? |
| 21:17.19 | funxion | yes |
| 21:17.22 | tomcruise | no other sip applications |
| 21:17.27 | funxion | nope |
| 21:17.35 | tomcruise | and in your trace, asterisk does not return any signalling on the failed invites? |
| 21:17.46 | funxion | nothing |
| 21:17.56 | jeffspeff | funxion are you using pjsip or sip? |
| 21:17.56 | funxion | minblown |
| 21:18.01 | funxion | chan_sip |
| 21:18.12 | tomcruise | nothing to be seen in your iptables logs? |
| 21:18.18 | funxion | hmmm |
| 21:18.18 | tomcruise | or the logs iptables fill |
| 21:18.23 | funxion | havent looked there |
| 21:18.35 | tomcruise | somewhere something is dropping traffic before it hits asterisk |
| 21:18.50 | jeffspeff | do you have localnet set for your local network and the public ip set and binding to the local ip? |
| 21:19.00 | tomcruise | good one jeffspeff! |
| 21:20.09 | funxion | yes |
| 21:20.15 | funxion | ips are setup right |
| 21:20.34 | tomcruise | i'd check your logs containing iptables activity |
| 21:20.53 | jeffspeff | just disable iptables while you're testing |
| 21:20.53 | funxion | cant seem to find where its dumping them |
| 21:21.20 | tomcruise | yeah, disabling during testing to isolate would be best |
| 21:21.52 | funxion | done that |
| 21:21.58 | funxion | same results |
| 21:22.13 | jeffspeff | try running sngrep |
| 21:22.38 | jeffspeff | it's a really sweet program that helps diag and show sip stuff |
| 21:22.49 | jeffspeff | it's like tcpdump on steroids |
| 21:24.50 | funxion | geting it now |
| 21:25.00 | jeffspeff | also, from an outside host, you can try telnet'ing to port 5060 (or whatever your sip listening port is) on that host |
| 21:25.13 | jeffspeff | telnet 123.456.789.1 5060 |
| 21:25.17 | [TK]D-Fender | <jeffspeff> i have a queue setup with 6 members. each 2 members has a different penalty, which groups the six members into 3 groups. i have queue rules setup to ring the next appropriate group of 2 phones. however, i have found that after both phones ring in the last group the call does not end. I continue to hear MOH even though several seconds ago asterisk informed me that both phones did not answer within the timeout. <- yes, well queues don't |
| 21:25.17 | [TK]D-Fender | necessarily hang up just because the CURRENTT ring attempt has timed out |
| 21:25.23 | [TK]D-Fender | That's qhat queues normally do |
| 21:25.36 | [TK]D-Fender | Rings some people... then stop and ring someone else |
| 21:26.14 | jeffspeff | [TK]D-Fender how do you get it to hangup or do something else in the dialplan once it's called all members of the queue? |
| 21:26.34 | [TK]D-Fender | "core show application queue" <----- |
| 21:27.19 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 21:27.25 | funxion | I can't telnet to the inbound providers signalling gateway but I'm assuming thats because its setup for inbound only |
| 21:27.31 | jeffspeff | i can see it dial 1, ring 15 sec, dial 2, ring 15 sec, dial 3, ring 15 sec, dial 4, ring 15 sec, dial 5, ring 15 sec, dial 6, ring 15 sec.... now it's done dialing, nobody else left to dial. the timeouts have all been reached. * is just playing the caller MOH |
| 21:28.08 | [TK]D-Fender | "core show application queue" <----- |
| 21:28.18 | [TK]D-Fender | the INTER-AGENT TIMEOUT has been reached |
| 21:28.25 | [TK]D-Fender | that is not the life-span of the QUEUE |
| 21:30.06 | funxion | jeffspeff: just installed sngrep now how to use |
| 21:30.15 | jeffspeff | just type sngrep |
| 21:30.27 | jeffspeff | it will show any sip traffic coming in/out of that box |
| 21:31.17 | [TK]D-Fender | heads out |
| 21:32.26 | funxion | much more insight now |
| 21:32.40 | jeffspeff | :) |
| 21:37.09 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 21:37.49 | *** join/#asterisk Oatmeal (~Suzeanne@2602:306:3676:c60:c1b7:3cd5:8071:e99d) |
| 21:42.02 | funxion | weird part is I see alot of stuff in sngrep but I dojnt see the initial call setup until after 15 rings or so |
| 21:43.09 | jeffspeff | how fast of box are you running * on? sngrep should show realtime network traffic (it does on all of my boxes) |
| 21:43.20 | funxion | very fast |
| 21:43.28 | jeffspeff | i can only think that maybe your box is underpowered or something |
| 21:45.01 | funxion | weird thing I can see is that when * tried sends an options packet I get back 503 |
| 21:50.04 | igcewieling1 | Are you still trying to get Asterisk working with NAT? |
| 21:50.32 | igcewieling1 | always disable dns lookups when using tcpdump or similar. |
| 22:00.10 | *** join/#asterisk rwb (~Thunderbi@65-183-151-239-dhcp.burlingtontelecom.net) |
| 22:00.29 | funxion | freaking weird. So I can see the invites with ngrep but sngrep doesnt see them |
| 22:00.34 | funxion | what the hell |
| 22:18.28 | *** join/#asterisk zerohalo (~zerohalo@uranium.zerohalo.net) |
| 22:19.13 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 22:19.42 | *** join/#asterisk voyonesti (454667ce@gateway/web/freenode/ip.69.70.103.206) |
| 22:20.13 | voyonesti | Hi everyone, any chance someone have experience with remote agent on vicidial ? |
| 22:23.52 | Milos | I had an outage and when my Internet came back, my upstream SIP trunk peer status is UNKNOWN but the registration is successful. That's not very sturdy at all. How can I debug this? |
| 22:23.54 | Milos | I'm running 13.13.1 |
| 22:25.50 | Milos | http://forums.asterisk.org/viewtopic.php?f=1&t=96620 |
| 22:25.53 | Milos | doesn't seem to be the case. |
| 22:34.19 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 22:34.22 | igcewieling1 | do you have qualify enabled? |
| 22:34.47 | Milos | I have it in general, so yes |
| 22:34.55 | *** join/#asterisk lankanmon (~LKNnet@2607:fea8:d20:239:b169:a70a:4bfc:3f1e) |
| 22:34.56 | Milos | I ran sip show peer xxxx and it has expire = -1 |
| 22:34.58 | igcewieling1 | good |
| 22:35.37 | Milos | whereas my SIP clients have a value that counts down. |
| 22:35.41 | Milos | what does expire = -1 mean? |
| 22:36.13 | Milos | is that the qualify countdown? doesn't seem so since qualify interval is 60s |
| 22:36.20 | Milos | and the value was much higher than that |
| 22:36.23 | Milos | for the SIP clients |
| 22:37.17 | *** join/#asterisk voipmonk (~voipmonk@2604:a880:2:d0::191:8001) |
| 22:38.12 | Milos | it does seem to be related to TLS |
| 22:38.27 | Milos | because the upstream SIP client that is not TLS did come back on its own |
| 22:38.30 | Milos | s/client/peer/ |
| 22:48.27 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 22:48.27 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 22:50.39 | Milos | I'll be restarting asterisk unless anyone has suggestions |
| 22:56.04 | voyonesti | Vicidial system using asterisk, When I try to setup a survey campaign with a remote agent this is what asterisk is returning if anyone have time to take a look : http://dumptext.com/lFQaFEZS |
| 22:56.21 | voyonesti | all other campaign are working fine |
| 23:02.58 | WIMPy | Looks like you're calling an invalid destination. |
| 23:04.16 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 23:22.20 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 23:26.17 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 23:35.15 | *** join/#asterisk troyt (~troyt@c-24-11-28-185.hsd1.ut.comcast.net) |
| 23:43.34 | *** join/#asterisk robmal (r@wporzo.pl) |
| 23:43.34 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 23:54.49 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ccebvdoxhcpenlbr) |