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| 00:25.04 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:25.04 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:53.46 | dunderproto | <PROTECTED> |
| 01:07.31 | *** join/#asterisk matrix1233 (~matrix123@197.0.36.136) |
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| 01:47.07 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 01:47.08 | xochilpili | hi all |
| 01:47.09 | *** join/#asterisk Nate15329 (~Nate15329@99-72-98-70.lightspeed.oshkwi.sbcglobal.net) |
| 01:47.33 | xochilpili | i have many of this "attacks" or "i dont know what means" like this: https://pastebin.com/Siyx05RJ |
| 01:47.41 | xochilpili | how can i stop them? |
| 01:50.59 | xochilpili | when i type : sip show channels i got many ip address and user/ANR i dont recognize with Peer <guest> |
| 01:58.17 | tuxd00d | xochilpili: In sip.conf, what is allowguest set to? |
| 02:00.09 | xochilpili | tuxd00d, thanks for answer i have allowguest=no |
| 02:00.29 | xochilpili | i also have : alwaysauthreject=yes |
| 02:00.53 | tuxd00d | I have to run, but check your PCAP or use something like sngrep to see whatâs going on. |
| 02:01.08 | xochilpili | sngrep ? |
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| 02:15.21 | xochilpili | tuxd00d, i have installed sngrep and im capturing, now, i see alot of "attackers" making several "INVITES" and "REGISTER" but i dont realize how to stop this |
| 02:17.24 | agent_white | xochilpili: You need to limit your traffic to be between your trunk<->pbx<->UAs exclusively. They're not using your trunk to send those INVITES. |
| 02:19.31 | xochilpili | agent_white, thanks for answer, i dont fully understand what u mean, i think that u are talking about deny= and permit= options in sip.conf? |
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| 02:30.21 | agent_white | xochilpili: Yep! Those are well worth looking into. Though, the traffic is passing through your network, down to your PBX. Maybe look into separating voip traffic in a VLAN? |
| 02:33.41 | agent_white | Lots of things you can do to avoid. https://github.com/EnableSecurity/sipvicious - you can use this to test how far the traffic is able to egress into your network. |
| 02:34.46 | agent_white | (far from uncommon to see some traffic with a sipvicious tag on it) |
| 02:53.32 | xochilpili | agent_white, but im connecting from outside, i mean, my personal extension is at home, and the server is in the cloud, i have no static ip address, so how can i use permit= if i have no a static ip address? |
| 02:58.25 | igcewieling | packet captures would show packets before iptables filters them |
| 03:00.12 | igcewieling | xochilpili: use a vpn, change the permit line when your IP changes, use port knocking, use fail2ban, get a static IP, use iptables to filter out some of the common bots. |
| 03:01.15 | igcewieling | What *I* do, is to allow guest and autocreate peers, send them to a jail context and play tt-monkeys. The bots / scripts seems to stop trying when it "succeeds". |
| 03:01.21 | xochilpili | igcewieling, i have installed fail2ban and sometimes it blocks my server's ip address |
| 03:01.40 | igcewieling | xochilpili: there is no good answer to a non-static IP address. |
| 03:02.12 | igcewieling | now you have a handful of other things to look into as well. |
| 03:02.39 | xochilpili | igcewieling, i mean in the server's fail2ban config, sometimes fail2ban blocks server's ip address and i cant login |
| 03:03.02 | igcewieling | You should already be using a VPN to get access to your internal network, you could run voice over that |
| 03:03.16 | igcewieling | xochilpili: yes, that is one of the drawbacks to using fail2ban. |
| 03:07.25 | xochilpili | igcewieling, i also have downloading sipvicious then i have typed : ./svcrack.py myserver's ipaddr -u 100 and then in asterisk-cli i have a lot of REGISTER <guest> |
| 03:07.40 | xochilpili | i dont get it, how to protect from this? |
| 03:07.56 | xochilpili | i have done almost everything in this: http://blogs.digium.com/2009/03/28/sip-security/ |
| 03:08.21 | xochilpili | with "almost everything" im not using users.conf because i only have my extension |
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| 03:10.02 | igcewieling | there is no one perfect way. There are only layers of not perfect ways. I'm not going to help with iptables, but here is a part of my iptables rules: https://pastebin.com/zQRhHa37 |
| 03:10.13 | igcewieling | ~users.conf |
| 03:10.13 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 03:10.30 | igcewieling | Anything which tells you to use users.conf should be considered wrong. |
| 03:13.06 | igcewieling | sorry, that is a bad paste. standby |
| 03:13.58 | igcewieling | Here is the correct one: https://pastebin.com/V5Uks0Vp |
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| 03:16.26 | xochilpili | igcewieling, as i can see you have every rule in iptables to block all those, but if there's a new one then u add it into your iptables? |
| 03:16.33 | xochilpili | are u using fail2ban also? |
| 03:17.18 | xochilpili | what u mean with this: "Anything which tells you to use users.conf should be considered wrong". English issues: i think you mean, do not use users.conf ? |
| 03:17.22 | igcewieling | I don't use fail2ban. When I see a new one, I add it. If you don't want to use iptables or fail2ban then set up a vpn and stop fighting with the problem. |
| 03:18.19 | xochilpili | igcewieling, may i share my iptables to you, in order to have some feedback from u? |
| 03:19.00 | igcewieling | You missed the part where I said " I'm not going to help with iptables." There is a world wide web of information about iptables. |
| 03:19.53 | xochilpili | igcewieling, yes, sorry, i omitted that part :D |
| 03:20.38 | xochilpili | igcewieling, can i ask, in asterisk do you have tcp or udp also are u using 5060 or 5061 ?? |
| 03:21.07 | igcewieling | Almost all peers configured on my main call servers have static IP addresses. The only ones which don't are support staff. |
| 03:21.57 | igcewieling | xochilpili: I don't block the requests. I accept them and put them in a context which plays annoying sounds. most stop trying soon after they get that. |
| 03:22.32 | igcewieling | *HOWEVER* I've been using Asterisk since early 2006 and generally know what I'm doing with the configurations I deal with. |
| 03:23.01 | igcewieling | A novice should *NEVER* enable allowguest or autocreatepeer |
| 03:23.46 | xochilpili | igcewieling, i have allowguest=no |
| 03:23.50 | xochilpili | in sip.conf |
| 03:24.11 | igcewieling | you should have that and autocreatepeer=no |
| 03:24.28 | igcewieling | allowguest=no is the most important to have. |
| 03:27.18 | xochilpili | igcewieling, oks added: autocreatepeer=no |
| 03:28.10 | xochilpili | igcewieling, but i need to ask, in iptables (sorry about it), when do you declare 'SIP-SCAN' ? |
| 03:31.29 | agent_white | xochilpili: "friend" may be something to look at if you use the trunk for both inbound and outbound. |
| 03:31.55 | agent_white | Also... "insecure=" and "context=". |
| 03:32.34 | agent_white | dtfmdode=rfc2833 to save all of us the hassle of inbound dtmf shit |
| 03:32.35 | agent_white | :P |
| 03:32.42 | agent_white | s/inbound/inband/ |
| 03:34.04 | agent_white | "host=", "fromdomain=", "bindport=", "bindaddr=". |
| 03:34.09 | agent_white | shrugs |
| 03:34.13 | xochilpili | agent_white, i have insecure=invite and context=from-local then i have a [default] in extensions.conf with only _X.,1,Hangup(21) and s,1,Hangup(21) |
| 03:34.20 | agent_white | Should be enough if you're looking to solve your issue with asterisk config. |
| 03:34.58 | xochilpili | agent_white, wait please, baby steps :D |
| 03:35.12 | xochilpili | can i share a part of my sip.conf in order to have some feedback? |
| 03:35.28 | agent_white | :P Basically my point is that you're still allowing traffic to traverse all the way to your pbx, leaving your pbx to authenticate the traffic. |
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| 03:35.51 | agent_white | It shouldn't reach that far in the first place. These other options generally are just good safey measure on top of it all. |
| 03:36.29 | agent_white | xochilpili: Go for it! I'm Just mosying in though to say my 2cents before I wander off again. |
| 03:36.37 | agent_white | But that definitely is the best start :) |
| 03:39.21 | xochilpili | agent_white, https://pastebin.com/6ecZsT5F << a part of my sip.conf |
| 03:44.24 | agent_white | xochilpili: Your externip is set to (I'm assuming since you blanked it out) your WAN IP, and your bindip is wildcard while your bindport is 5060; that means, it would be fair to say that any and all SIP traffic bound for port 5060 of your WAN IP is guaranteed to reach your PBX) |
| 03:44.43 | agent_white | _almost_ guaranteed. Just saying as this config is setup to allow this exact thing to happen. |
| 03:46.06 | agent_white | Stop the traffic before it reaches the PBX. $10 says if you had no PBX but SIP phones on your network, you would be getting lots of ghost calls from places 'appearing to be a local extension'. |
| 03:46.13 | agent_white | :P |
| 03:47.29 | xochilpili | yes i have receive a lot of ghost calls, but i dont realize exactly what are u suggesting to do in order to stop this |
| 03:47.46 | xochilpili | >>Stop the traffic before it reaches the PBX << iptables ?? |
| 03:49.36 | agent_white | YAP |
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| 03:53.24 | agent_white | xochilpili: Also, if you kinda want super-duper definitive answers to the quesion "is my trunk registered", set qualify to yes. |
| 03:55.01 | agent_white | At least, if you maybe hook responses to the keepalive (OPTIONS) somewhere in your monitoring or whatnot. |
| 03:57.19 | agent_white | Good for folks who like to see 'heartbeats' to ensure things are alive. |
| 03:58.09 | xochilpili | agent_white, im lost :D what you mean with : " if you maybe hook responses to the keepalive (OPTIONS)" somewhere? |
| 04:02.32 | xochilpili | agent_white, im trying with this iptables rules: https://pastebin.com/a8rAAey9 but i cant login from my zoiper at my celphone |
| 04:03.00 | xochilpili | <PROTECTED> |
| 04:12.44 | xochilpili | igcewieling, still here= |
| 04:12.45 | xochilpili | ? |
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| 05:15.37 | xochilpili | igcewieling, i have use a part of your iptables, but sipvicious is not working |
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| 07:47.11 | phrearch | morning |
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| 07:59.43 | Alblasco1702 | phrearch, morning |
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| 08:10.11 | Yorch | morning |
| 08:10.44 | Yorch | hope today someone could give me a hand with monitoring on zabbix.... i'm still working on it... |
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| 08:40.03 | sotoz | Hello, can I use ARI to access the AMD application? I can't find an endpoint in the ARI for this. Am I missing something? |
| 08:54.54 | Samot | Don't think you can. |
| 09:04.54 | sotoz | Is there a workaround for that? |
| 09:05.07 | Samot | I don't know. |
| 09:05.22 | Samot | What do you want to use ARI for with AMD? |
| 09:05.44 | sotoz | Can I create an outgoing channel with the ARI, do a continue in dialplan that will execute the AMD application, set some channel vars and then read them with ARI |
| 09:05.50 | sotoz | or something like that |
| 09:06.05 | Samot | Well.. |
| 09:06.08 | Samot | I use AMI |
| 09:06.12 | Samot | So yes. |
| 09:06.21 | Samot | You can originate a call.. |
| 09:06.27 | Samot | Send it to the context that has AMD in it. |
| 09:06.46 | sotoz | hm |
| 09:07.53 | Samot | So I'm guessing the /channels POST for originate will do it for you |
| 09:08.40 | sotoz | yes, I'm already using that to originate a channel |
| 09:08.52 | Samot | OK.. |
| 09:09.12 | sotoz | so after the answer, I should a ContinueInDialPlan |
| 09:09.21 | Samot | No. |
| 09:09.24 | Samot | It's Originate |
| 09:09.38 | Samot | The command doesn't change if you use AMI or ARI |
| 09:09.43 | Samot | Or straight from the dialplan. |
| 09:09.45 | sotoz | ok |
| 09:09.56 | sotoz | so how do I use the AMD application then? |
| 09:10.19 | Samot | What do you mea? |
| 09:10.24 | Samot | What do you mean? |
| 09:10.29 | Samot | How to use it in general? |
| 09:10.33 | Samot | Or how to call on it? |
| 09:10.45 | sotoz | I want to originate the channel and depending of if it is a machine or not do different things |
| 09:10.50 | Samot | OK |
| 09:11.05 | Samot | You understand how Originate works, right? |
| 09:11.17 | sotoz | I think so.. |
| 09:11.38 | sotoz | it creates the channel, and dials it |
| 09:11.48 | Samot | You specify the extension, context and priority you send the call to once the endpoint answers. |
| 09:12.28 | sotoz | so what extension should I give in case that I'm originating from a Stasis application |
| 09:12.35 | Samot | So you Originate that opens a channel to dial to 1NXXNXXXXXX via your trunk. |
| 09:12.48 | Samot | That dial is ANSWERED |
| 09:12.52 | sotoz | aha |
| 09:13.07 | Samot | It sends it to the exten,context,priorty |
| 09:13.30 | Samot | Now that call is considered INBOUND |
| 09:13.45 | sotoz | well, I'm using the "app" parameter to pass the channel in my stasis application |
| 09:13.50 | Samot | So the callee now can send back DTMF, you can check for talking..etc |
| 09:14.02 | Samot | I don't use ARI |
| 09:14.03 | sotoz | I see that this is mutally exclusive with extension/context/priority |
| 09:14.07 | Samot | Haven't overly messed with it |
| 09:14.13 | Samot | But it's kinda the same concept. |
| 09:14.17 | sotoz | ok |
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| 09:14.20 | Samot | No. |
| 09:14.28 | Samot | You can send the call to an app as well |
| 09:14.38 | Samot | You can either send it to an app or the context, etc |
| 09:14.55 | sotoz | "extension: string - The extension to dial after the endpoint answers. Mutually exclusive with 'app'." |
| 09:15.00 | Samot | You should read up on both Originate and AMD so you understand how they can work together. |
| 09:15.09 | Samot | Yes |
| 09:15.16 | Samot | You choose an APP or an EXTENSION |
| 09:15.20 | Samot | One or the other. |
| 09:15.28 | sotoz | ok, so in my case I should use app |
| 09:15.40 | Samot | That's up to you. |
| 09:15.54 | sotoz | as I have a stasis app that has the business logic of what should happen if the answer is from a machine or human |
| 09:15.58 | Samot | Again, I do this with AMI and just send calls to an extension, context, priority |
| 09:16.10 | sotoz | ok |
| 09:16.12 | Samot | Then why do you need AMD? |
| 09:16.38 | sotoz | I need asterisk to detect the machine so I decide then what to do |
| 09:17.07 | Samot | So then sending answered call straight to the stasis app immediately will be pointless. |
| 09:18.02 | sotoz | why? in case it has a status as a channel var if it is a machine or not then I can choose what to do with that call (hangup/transfer/play etc) |
| 09:18.18 | Samot | If it goes straight to the stasis app.. |
| 09:18.25 | Samot | How do you know if it's a machine or human? |
| 09:18.33 | Samot | How will the stasis app know what to do? |
| 09:18.39 | sotoz | I was thinking of setting somehow a channel var |
| 09:18.48 | sotoz | in the dialplan |
| 09:18.52 | Samot | When are you going to call on AMD? |
| 09:19.05 | sotoz | after the answer |
| 09:19.25 | Samot | 5:15:32 AMÂ <sotoz>Â ok, so in my case I should use app <<-- WHAT APP? |
| 09:19.42 | sotoz | "app: string - The application that is subscribed to the originated channel. When the channel is answered, it will be passed to this Stasis application. Mutually exclusive with 'context', 'extension', 'priority', and 'label'." |
| 09:19.51 | sotoz | I was referring to this app ^^ |
| 09:19.56 | Samot | I know. |
| 09:20.11 | Samot | How do you know if the call is answered by machine or human? |
| 09:20.21 | Samot | How will the stasis app know what to do with the call once it's answered?! |
| 09:20.30 | sotoz | I don't. That's why I wanted to use the AMD application with the ARI somehow |
| 09:20.33 | sotoz | but it's not supported |
| 09:20.41 | Samot | Right |
| 09:20.49 | Samot | So instead of sending the call to the stasis app. |
| 09:20.54 | Samot | You need to send to AMD |
| 09:21.02 | Samot | THEN the stasis app. |
| 09:21.12 | sotoz | aha |
| 09:21.22 | Samot | You have the start and the end of the processs |
| 09:21.25 | Samot | You don't have a middle |
| 09:21.31 | Samot | That's what AMD is for. |
| 09:21.59 | sotoz | ok :) thanks for your valuable info. I'll try that. |
| 09:22.35 | Samot | 5:15:03 AMÂ <Samot>Â You should read up on both Originate and AMD so you understand how they can work together. |
| 09:22.49 | sotoz | :) |
| 09:23.06 | sotoz | oh it's 5:15 there??? go to sleep already :P |
| 09:23.16 | Samot | I just woke up. |
| 09:23.19 | sotoz | heh |
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| 10:20.28 | andycol | Hi Guys |
| 10:20.48 | andycol | i upgraded from asterisk 1.8 to asterisk 13 and now the queue log is no longer showing callerid |
| 10:20.52 | andycol | for inbound calls |
| 10:20.54 | andycol | any ideas why? |
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| 12:18.27 | mdee | hello, when i try to open some ari via swagger or curl asterisk goes to down, this behavior related to v.13.14.0 and higher, 14.3 and higher, in 14.2 everything is ok |
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| 13:30.26 | sotoz | mdee: what do you mean : asterisk goes to down? |
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| 13:36.17 | CyberJacob | am I right in thinking that PJSIP can't do IP-based authentication? |
| 13:37.59 | Samot | No. It does. |
| 13:38.13 | [TK]D-Fender | absolutely |
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| 13:52.28 | mdee | sotoz: i mean - it crashed |
| 13:52.53 | mdee | usr/sbin/asterisk: symbol lookup error: /usr/lib64/asterisk/modules/res_ari.so: undefined symbol: stasis_app_get_debug_by_name with this error |
| 13:53.15 | mdee | autoload all modules solved my problem |
| 13:53.55 | mdee | then i search modules from wich ari's depends |
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| 15:35.01 | net | anyway to disable dtmf tone on incoming calls to be PCI complaince? |
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| 16:01.45 | Kobaz | what's asterisk's preferred payload type for RFC2833 DTMF? |
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| 16:27.30 | imcdona | I've been working with the Corosync devs to address the res_corosync.so module crashing Asterisk when used with Corosync 2.x. See here: https://github.com/corosync/corosync/issues/57 They've written a patch for Asterisk. When might we see this bug fix released for certified Asterisk 13? |
| 16:29.00 | file | if included recently it would go into the next major certified release months in the future |
| 16:29.57 | Kobaz | mmm corosync |
| 16:29.59 | Kobaz | i need to get into that |
| 16:30.03 | imcdona | Currently the dev is waiting for the "Contributor licence" agreement in order to post commit the change |
| 16:30.04 | Kobaz | i'm sure i'll be crashing it too |
| 16:30.13 | Kobaz | that's what happens |
| 16:30.21 | Kobaz | i try new things in asterisk, and it crashes |
| 16:30.24 | Kobaz | and then if fix it :) |
| 16:30.35 | Kobaz | s/if/i/ |
| 16:31.10 | Kobaz | my current side project is setting up ceph |
| 16:33.39 | imcdona | If someone can move the process along to get the corosync dev's aggreement reviewed that would help. ;) |
| 16:33.49 | file | it has to be reviewed by legal |
| 16:34.09 | file | noone can move that along, but it's generally done quickly |
| 16:34.09 | imcdona | ahh...what are we looking at in terms of time-frame? |
| 16:34.19 | file | a few hours or a day |
| 16:34.32 | imcdona | Okay. Perfect. I'll pass that along. Thanks |
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| 16:57.50 | DanQuinney | How does one add a PJSIP header that contains params? If I do the following; http://paste.codebasehq.com/pastes/m5tf98yrpytwsetl5d, I see the following; http://paste.codebasehq.com/pastes/w8y5soy1ow86j250p9 |
| 16:57.55 | DanQuinney | Without the params it's fine |
| 17:00.42 | file | escape them by putting \ in front |
| 17:00.49 | file | otherwise the configuration parser treats it as a comment |
| 17:02.16 | DanQuinney | infront of the ; |
| 17:02.23 | file | yes |
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| 17:05.07 | DanQuinney | that worked a treat, thanks file :) |
| 17:05.14 | file | you're welcome |
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| 18:39.45 | iulhk | hi |
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| 18:43.03 | iulhk | trying to use set_callerid (if i am using this way "$AGI->set_callerid(123456789); it works fine" if i use this way $AGI->set_callerid(123456789PR1234); it works but in sip client call logs underneath of call record 123456789PR1234 and asterisk printed? how to remove asterisk word? |
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| 18:48.36 | [TK]D-Fender | iulhk, Use the normal function for this or set the name |
| 18:48.36 | Download-Fritz | Hey, good day! Would someone happen to know whether it is intended that variables in exten dialplan 'instructions' always resolve to an empty char? |
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| 18:49.06 | [TK]D-Fender | Download-Fritz, please rephrase that |
| 18:49.17 | Download-Fritz | exten => 123${TEST}456,... it always matches 123456, whatever the value of TEST is |
| 18:49.30 | [TK]D-Fender | that is not a variable |
| 18:49.43 | [TK]D-Fender | that is a CONSTANT and must be hard set under [globals] |
| 18:49.59 | [TK]D-Fender | It only loks like a variable |
| 18:49.59 | Download-Fritz | I set it via 'setvar' in the peer definition |
| 18:50.06 | [TK]D-Fender | Doesn't work that way |
| 18:50.10 | Download-Fritz | I know that :) |
| 18:50.14 | Download-Fritz | I asked if that was intended |
| 18:50.30 | [TK]D-Fender | exten => 123${TEST}45 <---- that is a CONSTANT reference and only applies to those set under [globals] |
| 18:51.37 | Download-Fritz | Yes, thanks, but my question is whether that is intended / by-design. I was thinking about filing a bug before I found this channel |
| 18:53.50 | [TK]D-Fender | yes, it was always intended and documented as such |
| 18:53.56 | [TK]D-Fender | it was never a variable |
| 18:54.22 | rmudgett | Otherwise it would make finding an exten much more difficult. |
| 18:54.41 | Download-Fritz | I see, thanks |
| 18:54.48 | rmudgett | i.e., Slow as molasis in January |
| 18:54.52 | [TK]D-Fender | https://www.voip-info.org/wiki/view/Asterisk+Dialplan+Globals |
| 18:55.00 | Download-Fritz | I hoped I could get around using the phone number multiple times across the config files |
| 18:55.24 | [TK]D-Fender | There is a suggestion that you might be able to change the value in runtime.... |
| 18:55.40 | iulhk | this is for set callerid "$AGI->set_callerid(123456789), what is for set callerid name ? |
| 18:55.40 | [TK]D-Fender | not sure how you could do that while still trying to succeed at a pattern match though |
| 18:55.56 | [TK]D-Fender | iulhk, Use the normal function for this |
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| 18:57.01 | Download-Fritz | globals would be 'bad design' for my purpose, guess I need to just mention the number once more. Thank you very much"! |
| 18:58.31 | iulhk | <[TK]D-Fender>: if i will use normal, then i will not be able to send what exactly i want to SET. so that's why i am trying to append and then sending to callee modified callerid or caller name, ? |
| 18:59.27 | [TK]D-Fender | Why not? |
| 19:00.32 | iulhk | like my actual caller is 123456789 and i want when it calls to callee, callee should get 123456789PR or etc ? |
| 19:01.01 | [TK]D-Fender | USE THE REGULAR DIAPLAN FUNCTION TO SET THIS |
| 19:01.52 | iulhk | ok |
| 19:06.08 | sawgood | AstriCon 2017 (Orlando, Fl)! |
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| 19:56.50 | DanQuinney | good excuse for a holiday sawgood |
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| 19:59.25 | qakhan | hi all, i am using sip realtime, sometime extens shows in sip show peers and sometime not |
| 19:59.42 | qakhan | what could be the problem? |
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| 20:14.58 | klow | Hello, I've been making the switch over to PJSIP driver on some systems, and I'm wondering if theres a way to get more logging? the chan_sip logging would say things like "crypto suite not support" or "no matching codecs found" and other stuff like that in the /var/log/asterisk/full , but even if "pjsip set logger on" and "logger set level debug" im not |
| 20:14.58 | klow | really getting helpful hints on why a call might be failing .. ? |
| 20:15.34 | klow | the pjsip logger gets me the SIP Dialog, but it seems the chan_sip driver would give more hints in the asterisk logfile |
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| 20:20.42 | DanQuinney | hey guys, I've got an odd issue, I've got an incoming number that's dialling out to a mobile, if the caller hangs up the phone then the mobile still thinks it's the call still; http://paste.codebasehq.com/pastes/dpmhavemi9xgniamdu |
| 20:21.14 | DanQuinney | i'm bloody confused |
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