| 00:19.28 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-lhsbhyfmznxpwiqk) |
| 00:34.42 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 01:03.19 | *** join/#asterisk celer (~celer@pc1.robertwaltersjp-unet.ocn.ne.jp) |
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| 02:04.12 | snadge | im gonna get flamed for this.. but you guys are pretty cool, and hopefully wont lart me for something woefully offtopic |
| 02:04.27 | snadge | freeswitch doesn't seem to have a concept of fromuser ? |
| 02:09.32 | drmessano | https://wiki.freeswitch.org/wiki/Variable_sip_from_user |
| 02:09.41 | drmessano | 20 seconds of Googling |
| 02:10.55 | snadge | hyeah i found that too |
| 02:10.59 | snadge | see also todo |
| 02:11.43 | snadge | and it doesn't really suggest what you should do with that variable.. or where it goes or anything.. i know nothing about freeswitch, but thats what $parentcompany uses.. and they're like, yeah, just connect a pbx up to us.. and we wont help you with anything.. just figure it out ;) |
| 02:12.10 | drmessano | Freeswitch has an IRC channel. I've heard they are completely responsive in there |
| 02:12.29 | drmessano | Seems like that would be time better spent |
| 02:17.01 | Samot | Naw, asking questions in an unrelated channel is better. |
| 02:23.00 | snadge | absolutely.. i fear that if i go in there.. i'll run into people from $parentcompany ;) |
| 02:23.41 | snadge | im just working around it.. its interesting that asterisk 14 has added proper support for SRV lookups |
| 02:23.58 | snadge | so clearly this is becoming a problem for people integrating asterisk with freeswitch/kamailio |
| 02:24.57 | snadge | you can only auth by a single ip.. so you have to duplicate the context for each ip address.. done some googling on that.. yep, sure enough, that's how you do it |
| 02:25.31 | Samot | What are you trying to do? |
| 02:25.37 | drmessano | Proper support for SRV lookups? |
| 02:25.50 | Samot | Yeah, I let that slide. |
| 02:26.24 | drmessano | I thought it had it 10 years ago |
| 02:26.37 | Samot | Yeah, I use it quite a lot. |
| 02:27.25 | drmessano | Maybe they added the oft-unrequested feature of following weights and priorities |
| 02:28.34 | drmessano | But since most providers use RRDNS.. |
| 02:29.04 | drmessano | IDK.. DNS SRV seems more like a client problem, and one that's implemented well in many places |
| 02:29.13 | Samot | <param name="from-user" value="809XXXXXXX"/> |
| 02:29.38 | Samot | Wouldn't that be equivalent of "fromuser"? |
| 02:29.58 | Samot | Or even what drmessano said: sip_from_user |
| 02:31.01 | snadge | what i meant by proper support for srv lookups.. if a hostname resolves to more than one ip.. asterisk doesn't care |
| 02:31.18 | snadge | so you cant say.. host=sip.myprovider.com .. and have that actually mean, more than 1 ip address |
| 02:31.32 | drmessano | That has nothing to do with DNS SRV |
| 02:31.45 | drmessano | Thats standard DNS A Lookups |
| 02:31.46 | snadge | kind of |
| 02:31.49 | drmessano | No |
| 02:32.10 | drmessano | DNS SRV Lookups are entire different |
| 02:32.19 | *** join/#asterisk mrhelpmann (~mrhelpman@i.am.mrhelpmann.xyz) |
| 02:32.43 | Samot | Completely. |
| 02:33.02 | Samot | Because with a DNS SRV I can put sip.provider.com in there.. |
| 02:33.04 | snadge | i get that an SRV record is not related to an A record |
| 02:33.16 | Samot | And the SRV record will tell me which destination to use. |
| 02:33.21 | Samot | Base on the priority and weight. |
| 02:33.37 | snadge | what i dont get.. is why the two are confused in this way |
| 02:33.46 | snadge | right so its a terminology fail |
| 02:34.12 | Samot | Well one, unless they are doing load balancing people don't assign more that one IP to a FQDN. |
| 02:34.21 | snadge | either way.. if you say.. host=sip.myhost.com.au .. and that resolves to 1.1.1.1, 2.2.2.2, 3.3.3.3 as an example |
| 02:34.28 | Samot | It can't |
| 02:34.31 | drmessano | How are they confused? |
| 02:34.35 | snadge | asterisk will pick one of the above ips.. and if the call comes from another.. it will be rejected |
| 02:34.39 | drmessano | It's very clear that they are different |
| 02:35.08 | drmessano | chan_sip doesn't support multiple hosts |
| 02:35.12 | drmessano | chan_pjsip does |
| 02:35.29 | drmessano | and that has nothing to do with DNS SRV |
| 02:35.55 | drmessano | chan_sip is DEAD |
| 02:35.58 | snadge | right.. its just inconvenient thats all.. so what i've had to do is create a duplicate peer for each ip address |
| 02:36.16 | drmessano | so use chan_pjsip |
| 02:36.40 | snadge | yeah that would be nice.. except theres a whole bunch of custom stuff that depends on chan_sip that would need to be rewritten |
| 02:36.48 | drmessano | chan_sip is DEAD |
| 02:36.49 | snadge | including agi scripts, billing system etc |
| 02:37.03 | drmessano | It's in maintenance mode |
| 02:37.26 | snadge | it doesn't change the fact that we can't wave a magic wand and switch to pjsip |
| 02:37.33 | snadge | thats going to take considerable effort |
| 02:38.03 | Samot | How many hosts do you have to make a peer for? |
| 02:38.03 | lorsungcu | just imagine its a very heavy wand you need to work hard to wave |
| 02:38.04 | snadge | if it was just asterisk.. sure.. why not, follow one of the guides.. change over to pjsip.. done |
| 02:38.35 | snadge | its all the other crap.. scripts written by people years ago, who have left etc.. the usual story |
| 02:38.41 | drmessano | This is like a constant moving goalpost |
| 02:38.51 | drmessano | 1. Nothing to do with DNS SRV |
| 02:38.58 | drmessano | 2. Everything to do with chan_sip |
| 02:39.09 | drmessano | 3. Resolved by using chan_PJSIP in 13, not 14 |
| 02:39.37 | drmessano | or making a bunch of peers |
| 02:39.38 | snadge | https://wiki.asterisk.org/wiki/display/AST/New+in+14 |
| 02:39.50 | snadge | bullet point one |
| 02:40.26 | drmessano | What about it? |
| 02:40.35 | snadge | which sounds like its not related to what i was talking about before.. but thats okay, i get that now |
| 02:40.43 | Samot | You mean for Unbound? |
| 02:40.46 | drmessano | Its completely unrelated |
| 02:40.48 | Samot | A DNS service? |
| 02:40.52 | Samot | Like BIND? |
| 02:41.45 | Samot | So they fixed SRV and NAPTR lookups for Unbound. |
| 02:41.50 | drmessano | Your issue is host= can only contain one IP address |
| 02:41.56 | Samot | ^^ That. |
| 02:42.16 | drmessano | Nothing to do with DNS SRV, nothing to do with DNS changes in 14, everything to do with chan_sip |
| 02:42.37 | snadge | okay sure.. im reading about naptr and srv and sip now |
| 02:42.40 | drmessano | I dont think you understand what DNS SRV records are |
| 02:42.46 | snadge | just because im curious as to what that actually means |
| 02:43.37 | Samot | Wait, you're making comments on how DNS SRV has been "properly fixed" in 14 and you don't know what it means/does?! |
| 02:44.08 | lorsungcu | <PROTECTED> |
| 02:44.25 | snadge | someone confused me the other day by saying "asterisk doesn't support SRV lookups" .. and then i confused that with a single ip per host= .. which is a chan_sip limitation |
| 02:44.26 | drmessano | Samot: Indeed |
| 02:44.47 | drmessano | Asterisk has supported DNS SRV lookups since 1.4 |
| 02:45.12 | Samot | Yup. |
| 02:45.36 | snadge | right.. so i think i get it now.. you can do a NAPTR lookup, which can return an SRV record |
| 02:45.37 | drmessano | I believe whats been missing is it doesn't follow weights and priorities.. I just returns the first host. But I dont know any providers that implement that anyway |
| 02:45.41 | snadge | which you then use to look up an A record |
| 02:45.45 | snadge | right |
| 02:46.04 | snadge | that doesn't seem like a big deal to me at all |
| 02:46.15 | snadge | i guess someone is excited about it though |
| 02:46.34 | drmessano | It has applications to improve things over RRDNS |
| 02:46.46 | drmessano | But its not fixing anything thats BROKEN |
| 02:46.53 | Samot | And ties the lookups to a specific protocol and service. |
| 02:46.59 | drmessano | Which is why its taken 10 years to get here |
| 02:47.05 | Samot | So my A record for my sip domain can go one place.. |
| 02:47.21 | Samot | and my UPD/SIP requests go someplace else. |
| 02:47.43 | snadge | right.. it also seems to imply that it will do basic redundancy as well.. eg.. try the first priority |
| 02:47.48 | drmessano | Where I see DNS SRV implemented is basically just having a SIP resource record for any given domain. Period. Not having to look up the actual host name. Period |
| 02:47.49 | snadge | if that doesnt work.. try the second one |
| 02:47.59 | Samot | No. |
| 02:48.02 | Samot | That's not how it works. |
| 02:48.14 | Samot | I can set up SRV records to have priority... |
| 02:48.34 | Samot | And I can set up SRV records to have the same priority with different weights. |
| 02:48.58 | Samot | So I can send all my primary requests to a SRV record with 4 destinations.. |
| 02:49.02 | Samot | Weight them as 25% each |
| 02:49.20 | Samot | All requests will be split.. |
| 02:49.32 | Samot | I make an A record with 4 destinations... |
| 02:49.47 | Samot | Whichever one responds, wins. |
| 02:49.48 | Samot | DIFFERENT |
| 02:49.49 | drmessano | But again |
| 02:49.58 | drmessano | If I have RRDNS.. |
| 02:50.14 | drmessano | Its Pseudo random, and there's load balancing in place anyway |
| 02:50.25 | snadge | before when i said try the first priority.. what i meant was.. try the lowest weight, within that priority |
| 02:50.26 | drmessano | So sure.. this is another way of doing it |
| 02:50.48 | drmessano | But this DOESNT address the whole "I have one host and the call is coming from another" |
| 02:50.48 | snadge | it still seems like its up to the client though whether it does that or not |
| 02:51.21 | drmessano | 99% of SIP clients have DNS SRV implemented, with weights and priorities |
| 02:53.09 | Samot | That's generally what those Use DNS SRV and Add DNS SRV Prefix settings devices have. |
| 02:53.19 | snadge | in any case.. you've confirmed that pjsip handles multiple ips, and chan sip does not.. and likely never will |
| 02:53.27 | Samot | No, it won't. |
| 02:53.34 | Samot | PJSIP is the Chan_SIP replacement. |
| 02:53.59 | snadge | i was actually surprised to see that chan_sip is still in asterisk 14 |
| 02:54.10 | snadge | ive been expecting it to go away for some time now |
| 02:54.12 | Samot | It's not going away anytime soon. |
| 02:54.24 | Samot | Because like you said "Just can't hit a button and convert to PJSIP" |
| 02:55.05 | snadge | if we didn't have complicated custom agi scripts.. and a billing system / management portal that utilised the mysql backend for the realtime database |
| 02:55.12 | drmessano | It will sit around in Zombie Jesus mode for years |
| 02:55.15 | snadge | it would make the process a lot more straight forward |
| 02:55.27 | drmessano | What does that have to do with chan_SIP? |
| 02:55.40 | snadge | the format of the realtime database has significantly changed between the two |
| 02:56.08 | snadge | should you be poking things in and out of that outside of asterisk? probably not.. but apparently some people do |
| 02:56.08 | drmessano | *sigh* |
| 02:56.27 | drmessano | Lots of people do |
| 02:56.51 | WIMPy | That's usually the point, isn't it? |
| 02:56.58 | Samot | Are you referring to adding things to the database? |
| 02:57.17 | snadge | right so as just one example.. adding a DID to someones billing account |
| 02:57.23 | drmessano | Apparently few people interface externally with Asterisk |
| 02:57.31 | drmessano | Someone better tell Digium |
| 02:57.52 | snadge | the code which does that, also pokes the number into the asterisk database.. so that it actually does something when you call it etc.. redirecting it.. diverting it.. all that kinda fun stuff |
| 02:58.21 | drmessano | sed -i s/SIP/PJSIP * |
| 02:58.27 | Samot | NO! |
| 02:58.31 | drmessano | Feel free to paypal me my consulting feeds |
| 02:58.31 | Samot | Not sed! |
| 02:58.34 | drmessano | Feel free to paypal me my consulting fees |
| 02:58.51 | snadge | i haven't looked into the AGI side of it much.. but im going to presume that some of the parameters have changed |
| 02:58.59 | drmessano | Yep |
| 02:59.01 | drmessano | sed -i s/SIP/PJSIP * |
| 02:59.57 | snadge | so when chan_sip is deprecated.. i start seeking employment elsewhere :P .. because we know its just going to sit on life support until it fails, spectacularly |
| 03:00.07 | snadge | hehe |
| 03:00.39 | snadge | i was pleased to see that a cdr handling bug has been fixed in ast 14 though.. just annoyed that 14 isnt an lts |
| 03:00.44 | snadge | and that fix is unlikely to be backported |
| 03:01.03 | drmessano | lol |
| 03:01.11 | drmessano | Why does it matter that it's not an LTS? |
| 03:01.26 | snadge | lts you can neglect for a much longer period of time |
| 03:01.33 | snadge | very important :P |
| 03:01.41 | Samot | It's not a question of when Chan_SIP is deprecated... |
| 03:01.48 | Samot | It *is* deprecated. |
| 03:01.50 | drmessano | Maybe you are in the wrong business |
| 03:02.00 | Samot | There's no further development on it outside of the community. |
| 03:02.23 | drmessano | chan_sip was deprected when 13.0.0 was released |
| 03:02.27 | Samot | LTS releases do not mean you can ignore them longer. |
| 03:02.28 | drmessano | deprecated |
| 03:02.35 | Samot | Yeah, that too. |
| 03:02.54 | drmessano | Samot: When you don't give a shit, whatever time you can buy to do nothing else is a good thing |
| 03:03.09 | drmessano | So +1 for LTS and no-progress initiatives |
| 03:03.25 | drmessano | Zombie Jesus Admin |
| 03:03.28 | snadge | when things are in production, you dont want to be changing them every 6 months.. you could call it laziness i guess |
| 03:03.35 | drmessano | 6 months? |
| 03:03.44 | drmessano | Non-LTS releases are supported for like 2 years? |
| 03:04.22 | drmessano | If you're not upgrading every couple of years, you're not progressing anyway |
| 03:05.02 | Samot | Upgrading is for whippersnappers. |
| 03:05.18 | snadge | i completely agree.. you should be aiming to upgrade every few years to stay current.. i think the idea behind an lts is it buys you some additional time.. you're not forced to |
| 03:05.37 | snadge | the alternative is to go out of support.. which im sure plenty of businesses do |
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| 03:07.24 | snadge | im starting to think i'd be better off going back to development, rather than systems admin.. the irony being, i became a sysadmin because im lazy |
| 03:07.38 | Samot | No, that sounds right. |
| 03:07.48 | snadge | at least as a dev.. i can just point at the out of date production systems.. and laugh.. and whinge about how everything isn't the latest xyz, and its not my problem |
| 03:08.03 | Samot | Expect it is. |
| 03:08.26 | Samot | Development means you should be developing the next stage/platform. |
| 03:08.53 | Samot | So things are out of date because development hasn't developed the new upgrade path for the systems. |
| 03:10.21 | Samot | Lazy developers are worse than lazy sysadmins. |
| 03:10.46 | snadge | the sysadmins want to use code from 1972.. and the devs want to use bleeding edge nightly git |
| 03:11.01 | snadge | thats the way it usually goes |
| 03:11.01 | Samot | What is with the extremes? |
| 03:12.13 | snadge | thats right.. it was illustrated with an extreme, but the idea is that a compromise between them needs to be reached |
| 03:12.50 | snadge | devs sometimes end up rewriting stuff to support older versions.. and sysadmins sometimes end up installing "unstable" stuff out of repo etc.. to get stuff devs have written, to work |
| 03:14.05 | snadge | in telco.. and with a dev background.. i was shocked at the level of neglect and just how far out of support, actual functioning businesses, seem to go |
| 03:14.28 | snadge | and i hope that doesn't apply to the medical industry, or aerospace etc.. or god help us |
| 03:44.53 | Samot | Well, 5 minutes into watching the debate it's clear Hillary needs to learn not to sound so scripted. |
| 03:48.18 | Samot | Does Trump just want to go to war with Mexico? |
| 03:49.15 | Samot | He must want a new supply source for Trump Water. |
| 03:55.23 | snadge | he's going to win isnt he.. i dont want to encourage political debate.. i think southparks giant douche vs turd sandwich sums it up nicely... once again for many people, it comes down to choosing the least worst of two candidates |
| 03:56.24 | snadge | as a neutral observer from a foreign country.. i cant help but feel he has the upper hand.. even if hes just insulting her.. for some reason, that is funny |
| 03:57.17 | snadge | the guy is a racist asshat, and she's a career criminal.. good lord |
| 04:02.22 | Samot | Trump needs to stop using Michigan as an example of how things failed. |
| 04:03.15 | Samot | Or they need to use it against him. |
| 04:04.26 | Samot | This is Romenycare for the Dems and they fail to use it. |
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| 08:07.08 | guest609_ | How does one change the variables passed with the API OriginatingCall function? Before the call(INVITE) is sent. |
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| 08:44.56 | mkn__ | <PROTECTED> |
| 08:48.07 | Lope | hey guys I've been out of VoIP for a while. I'm trying to connect to a South African VoIP provider that only accepts Connections from SA. I've routed my internet connection using openVPN through a SA IP. Now I can connect to my SIP server. But I can't hear any audio when I call a time-checking service. I've set my listening port number on my SIP client to 1234, and I've forwarded incoming TCP/UDP 1234 from the router there to my machine on that lan, and from m |
| 08:48.08 | Lope | y machine on that LAN to 1234 on my machine. (I'm going through NAT twice) |
| 08:49.00 | Lope | I've also setup the iptables rule to allow incoming data on 1234 on my machine, and my SIP client is listening on port 1234 as well. |
| 08:49.21 | Lope | Am I missing something? I've done this successfully before, but it's been a while. |
| 08:50.42 | *** join/#asterisk jay8232 (596920bd@gateway/web/freenode/ip.89.105.32.189) |
| 08:51.18 | jay8232 | Does anyone know why a contact is shown with status unknown and RTT nan when executing pjsip show contacts ? |
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| 09:17.15 | dan_j | jay8232: I seem to recall a bug about that. What version are you running? |
| 09:26.16 | jay8232 | dan_j: pjsip 2.5.5 and asterisk 13.11.0 |
| 09:27.09 | jay8232 | fixed in 14.0 perhaps? |
| 09:29.52 | dan_j | No, I think it was fixed in 13. I'm just checking what i'm running |
| 09:31.50 | dan_j | Hmm. It seems it was fixed in 13.11.0 |
| 09:31.54 | dan_j | See changelog |
| 09:32.08 | dan_j | 2016-08-22 17:08 +0000 [6da8511a6a] res_pjsip: Default endpoints to the "offline" status |
| 09:32.11 | dan_j | http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13-current |
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| 10:00.51 | jay8232 | hm.. thats interesting since I run 13.11.0 :) |
| 10:01.25 | jay8232 | Ill try 14.0.0 then |
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| 11:06.49 | jay8232 | dan_j: FYI: I still get nan/unknown with 14.0.0 |
| 11:07.41 | dan_j | hmm. I'm sure last time I checked, it was fixed. |
| 11:08.08 | dan_j | I'm just busy so I can't check. I've not got pjsip in production yet. Just testing |
| 11:08.56 | jay8232 | k |
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| 12:37.15 | toshywoshy | I am getting 'core dumped' once I try to call out, any suggestions |
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| 13:11.33 | Lope | I was having trouble connecting to my asterisk server from behind NAT on a dodgy connection |
| 13:12.07 | Lope | So now I've connected to my server with openvpn. the openvpn server and my PC can ping each other |
| 13:12.21 | Lope | iptables is allowing everything from the other IP on both sides. |
| 13:13.03 | Lope | I can register but I can't make a call and keep getting messages on the asterisk server saying my client is lagged, unreachable, not replying to crtical packets etc. |
| 13:13.16 | Lope | I've even made the VPN TCP so that no packets get dropped. |
| 13:14.12 | Lope | It's not an MTU problem. I've tested sending 1500 byte pings. |
| 13:14.31 | toshywoshy | Lope: did you check the firewall and vpn for rtp ports/packets |
| 13:15.20 | Lope | yes, as I said, iptables is allowing everything from the opposing IP on both sides. |
| 13:16.09 | Lope | when I did a packet dump I saw a whole load of unauthorized packets. |
| 13:16.15 | Lope | which was strange. |
| 13:16.51 | Lope | I get lots of sip poke noanswer messages on the server. |
| 13:17.28 | toshywoshy | what is the ping time? |
| 13:18.36 | Lope | consistently around 445ms |
| 13:18.54 | Lope | with the occasional (1/20) 700 or 1000ms |
| 13:21.30 | toshywoshy | I think that is why you are getting the lagged / unreachable, I have always had problems if the ping time is larger, even my android wifi phone gives me problems and it has a ping time of about 200ms |
| 13:22.39 | Lope | is there some asterisk setting to adjust for the high ping? |
| 13:24.57 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 13:25.22 | Samot | Unreachable means the endpoint isn't responding. |
| 13:26.19 | Lope | I've just found a way to test UDP connectivity with netcat |
| 13:26.20 | Samot | This is most likely a NAT issue. |
| 13:26.25 | toshywoshy | You can try to enable QoS on the network, but that didn't solve my problem |
| 13:26.35 | Lope | Samot: but there's no NAT involved here |
| 13:27.24 | Samot | Lope: Those errors and statuses mean that the endpoint isn't responding to Asterisk's SIP messages. |
| 13:27.39 | Samot | So you need to look a what's happening on the network with the endpoint. |
| 13:27.56 | toshywoshy | Lope: how is OpenVPN configured, as a server or server-bridge |
| 13:28.28 | Lope | I can send UDP in both directions |
| 13:29.43 | Lope | On my server I've got openVPN server running (TCP). on my client (with the sip client) I've got openvpn client. They connect and can ping each other up to a mtu of 1500. |
| 13:29.52 | Samot | So when an endpoint is unreachable, you can see Asterisk sending messages to the endpoint and the endpoint replying to them? |
| 13:30.26 | toshywoshy | Lope: if openvpn is set as a server it is doing natting, and there is your problem |
| 13:30.32 | Lope | oooh shit |
| 13:30.38 | Lope | i just realized the problem! |
| 13:30.52 | Lope | Samot: you're right, it is a NAT problem. |
| 13:31.26 | Lope | I originally configured asterisk to run behind NAT on my server. So it's got that setting which locks it's IP to the internet IP. |
| 13:31.57 | Lope | But now my client is connecting via openVPN and not via the internet IP. |
| 13:32.02 | Lope | so that's probably the issue. |
| 13:33.35 | Samot | The IP your client connects to and the external IP that Asterisk uses do not have to be the same. |
| 13:33.40 | Lope | I'm a little rusty on asterisk/voip stuff. it's been a while. |
| 13:33.59 | Lope | oh really? |
| 13:34.18 | Lope | because I have externip=1.2.3.4 where 1.2.3.4 is my server's internet IP. |
| 13:34.28 | Samot | OK. |
| 13:34.32 | Lope | However my SIP client can't reach that ip. |
| 13:34.45 | Lope | because it's connecting via openvpn. |
| 13:34.55 | Samot | I'm guessing the VPN is using a LAN. |
| 13:35.10 | Lope | Samot: can you clarify? |
| 13:35.26 | Lope | the VPN is effectively a LAN on a different subnet to the LAN that the asterisk server LAN is on. |
| 13:35.34 | Lope | so I just forwarded the one IP to the other. |
| 13:35.40 | Lope | and vice versa. |
| 13:35.52 | Lope | there's no NAT. they can each talk to the real LAN IP. |
| 13:36.52 | Samot | You've added both subnets as part of the local network for Asterisk? |
| 13:37.13 | Lope | Samot: I've not configured the openvpn LAN in asterisk, no. Is there a way to do that? |
| 13:37.26 | Lope | ohhh, I see. there's a localnet option. |
| 13:37.53 | Lope | so how can I specify multiple localnets? I currently have localnet=192.168.4.0/24 |
| 13:38.45 | Samot | Multiple lines. |
| 13:38.51 | Lope | okay google says, yep. |
| 13:40.36 | Lope | then I suppose my asterisk server should have an IP on the localnet? |
| 13:40.41 | Lope | Because it's still not working. |
| 13:41.30 | Samot | Uhm, yes. |
| 13:42.11 | toshywoshy | how can you ping the server if it does not have an ip on that subnet? |
| 13:44.39 | toshywoshy | Samot: do you have experience with debugging asterisk? |
| 13:45.02 | Samot | In what manner? |
| 13:46.33 | toshywoshy | I am getting 'core dumped' when I try to phone out, in verbose mode I see 'killed by SIGSEGV' and gdb in backtrace talks about a '.clone ()' problem |
| 13:46.45 | toshywoshy | But I do not know what to do now |
| 13:47.08 | Samot | You submit the backtrack to the issue tracker. |
| 13:49.13 | toshywoshy | You mean the JIRA tracker of Asterisk? |
| 13:49.17 | Samot | Yes. |
| 13:49.39 | Samot | Or you detail your issue here and post a pastebin link with the backtrace. |
| 13:49.50 | Samot | Maybe someone will see it and be able to help. |
| 13:50.00 | Samot | I won't be able to. |
| 13:50.08 | Samot | Or you can do both. |
| 13:50.32 | *** join/#asterisk Lope (~Lope@59.89.179.154) |
| 13:51.01 | Lope | Samot: this is a problem. |
| 13:51.13 | Samot | What's that? |
| 13:51.34 | Lope | I can't add an IP in the openvpn network's IP range to my asterisk container. |
| 13:51.53 | Lope | they're separate networks. |
| 13:52.13 | Samot | Lope: This is a network issue. |
| 13:52.22 | Lope | I think it's an asterisk issue |
| 13:52.27 | Samot | Why? |
| 13:52.36 | Lope | netcat has no trouble talking between the 2 nodes |
| 13:52.36 | Samot | What makes you think it's an Asterisk issue? |
| 13:52.40 | toshywoshy | Lope: use openvpn server-bridge and bridge-utils |
| 13:52.43 | Lope | so asterisk shoudln't have a problem |
| 13:53.24 | Samot | Lope: I will ask again. When an endpoint goes unreachable, you can see Asterisk sending messages to the endpoint and the endpoint replying? |
| 13:53.41 | Samot | Or do you see Asterisk re-transmitting the same request over and over again? |
| 13:54.00 | Lope | Samot: I've not done a tcpdump on the asterisk container |
| 13:54.11 | Lope | hmm |
| 13:54.32 | Lope | I think the asterisk bridge idea is a very good one. |
| 13:54.34 | Samot | Because again, unreachable means that ASTERISK *is not* getting replies from the endpoint. |
| 13:54.57 | Samot | I'm going to guess that when it's unreachable, the endpoint can still make calls? |
| 13:54.57 | Lope | Samot: well, sometimes SIP does weird stuff with IP addresses |
| 13:55.35 | Lope | my SIP client sort of thinks it's making calls but they don't seem to start ticking and there's no audio. |
| 13:55.54 | Lope | But that's hard to draw any conclusions from. |
| 13:56.00 | Samot | Because that's a NAT issue. |
| 13:56.09 | Lope | The registration process takes a long time. |
| 13:56.13 | Lope | Which is unusual. |
| 13:56.15 | Samot | Nework. |
| 13:56.17 | Samot | Network. |
| 13:56.30 | Lope | Yeah I'll look into openvpn bridge. |
| 13:56.42 | Samot | Again, all the issues you are having are caused by network issues. |
| 13:56.57 | Samot | NAT/network... |
| 13:57.21 | Lope | so why does ping and netcat not have any trouble? |
| 13:57.33 | Samot | ping is imcp. |
| 13:57.49 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 13:57.59 | Samot | You're also the one requesting the traffic. |
| 13:58.04 | Lope | maybe it's because asterisk wants to be on the same localnet as the endpoint? |
| 13:58.15 | Samot | Just like the endpoint can make an outbound call... |
| 13:58.19 | Samot | No. |
| 13:59.04 | Lope | I'll try openvpn bridge |
| 14:00.01 | Lope | oh but if I use a bridge, then I need to use tap, which is going to be slower? |
| 14:00.16 | Lope | (slower than tun) or is irrelevant for VoIP? |
| 14:00.43 | Lope | This is just for testing and occasional calls, not heavy duty. |
| 14:03.53 | *** join/#asterisk kharwell (kharwell@nat/digium/x-muvkkxncrczqwovp) |
| 14:04.48 | toshywoshy | tun is layer 3 vs tap layer 2, so you have an extra ethernet packet overhead |
| 14:06.22 | Lope | toshywoshy: yeah |
| 14:06.24 | toshywoshy | if you want to use a tun with a openvpn server, you will need to setup iptables correctly |
| 14:06.43 | Lope | I see that server-bridge can't be used with static keys. Extra hassle :/ |
| 14:06.53 | toshywoshy | so you need 5060/udp and 10000:20000/udp |
| 14:07.06 | toshywoshy | static keys? |
| 14:07.27 | Lope | openVPN keys can be static or SSL |
| 14:07.42 | Lope | private key or public key encryption. |
| 14:08.09 | toshywoshy | oh, yes, I never used the static one, forgot they even exist |
| 14:08.26 | Lope | hmm, I find it less hassle |
| 14:08.27 | toshywoshy | as for the hassle, use easy-rsa |
| 14:08.57 | Lope | hmm. i'm not convinced openvpn-bridge is the way to go. |
| 14:09.04 | Lope | this SHOULD work. |
| 14:09.08 | toshywoshy | then adjust iptables |
| 14:09.12 | Lope | I'm going to give it some thought. |
| 14:17.22 | *** join/#asterisk pjm (~pjm_freen@uhfsatcom.plus.com) |
| 14:18.02 | *** join/#asterisk onixx (1000@bas1-stetherese38-2925260772.dsl.bell.ca) |
| 14:19.43 | igcewieling | bridging networks is rarely a good idea. |
| 14:22.47 | Lope | wow. looks like my connection sucks |
| 14:22.57 | Lope | I just did a ping and got many multi-second pings. |
| 14:23.04 | Lope | one was over 10sec |
| 14:24.04 | toshywoshy | igcewieling: why? |
| 14:25.32 | igcewieling | toshywoshy: excessive traffic. you do not want broadcast traffic to go across high latency links. |
| 14:25.55 | igcewieling | I am, of course, assuming the term bridging means bridging and not routing. |
| 14:26.31 | Kobaz | i cant connect to the conference wireless from the 6th floor anymore |
| 14:26.34 | Kobaz | :( |
| 14:26.51 | Lope | I'm doing a tcpdump now. |
| 14:27.10 | igcewieling | Lope: seeing lots of arps? |
| 14:27.13 | Lope | I see the asterisk server is sending loads of 401 Unauthorized to the client when it tries to make a call. |
| 14:27.26 | igcewieling | Lope: that is normal. |
| 14:27.52 | igcewieling | All calls start with an attempt without auth info, gets rejected, call is retried with the auth info. That's how it works. |
| 14:28.13 | Lope | igcewieling: no arbs regarding these 2 machines. only 2 arps for the gateway. |
| 14:28.28 | igcewieling | Lope: no bridging then |
| 14:28.46 | Lope | igcewieling: ah I see. but it doesn't seem to authorize. Or maybe the connection is just very lagged. |
| 14:28.51 | Lope | I'm going to try ping again. |
| 14:29.28 | Lope | yeah, it's pretty horrible. Around 500-3052. |
| 14:29.49 | Lope | The call actually seemed to work without openvpn. But the jitter was horrendous. |
| 14:29.58 | Lope | So I thought maybe openvpn with TCP would sort it out. |
| 14:30.04 | Lope | however the ping is just too bad. |
| 14:30.22 | Lope | Maybe I need to rather try some jitter correction stuff. |
| 14:31.00 | igcewieling | Lope: are you on satellite or a 3rd world country like the USA? |
| 14:31.18 | Lope | lol. Unfortunately internet-wise. I'm in india :/ |
| 14:31.31 | *** join/#asterisk jennie (~chatzilla@unaffiliated/jennie) |
| 14:31.54 | jennie | hello, how to start troubleshooting if I am getting ' circuit busy' while calling |
| 14:32.11 | jennie | I have opened putty and webadmin panel too |
| 14:32.16 | igcewieling | Lope: I'm sorry to hear that. One of my customers has staff in India. The internet sucks. |
| 14:32.21 | Lope | gtg bbl |
| 14:34.46 | jennie | anyone? |
| 14:35.15 | [TK]D-Fender | Show us the call. |
| 14:35.16 | [TK]D-Fender | ~pb |
| 14:35.16 | infobot | somebody said pastebin was 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. |
| 14:35.18 | [TK]D-Fender | ^^^ |
| 14:35.31 | [TK]D-Fender | ' circuit busy' <- means nothing |
| 14:35.45 | [TK]D-Fender | You need actual channel level debug on to see what's really happening |
| 14:35.52 | [TK]D-Fender | "sip set debug on" <- chan_sip |
| 14:36.01 | [TK]D-Fender | "pjsip set logger on" <- pjsip |
| 14:36.20 | [TK]D-Fender | Others depending what you're calling over |
| 14:37.38 | toshywoshy | igcewieling: yes there is an overhead, but with tuns you have much more configuration and that makes it more complex as in this case |
| 14:39.48 | igcewieling | toshywoshy: I assume he meant routing. Only a fool would bridge LANs |
| 14:43.40 | onixx | exit |
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| 18:27.59 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.0 (2016/09/26); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 18:41.30 | jrun_ | what is the diff between (!) and (+) for templates? |
| 18:43.35 | hexanol | jrun_: (+) is not specific to template: https://wiki.asterisk.org/wiki/display/AST/Adding+to+an+existing+section |
| 18:43.47 | hexanol | see also https://wiki.asterisk.org/wiki/display/AST/Template+Syntax |
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| 19:07.59 | Eze_Arg | Hi to all!, im having problems with dropped calls in the middle of the conversation. random durations (some call last 5 minutes, some a little bit more but they eventually get dropped.). What I see on the logs is the following (no errors are found before these lines) |
| 19:08.02 | Eze_Arg | <PROTECTED> |
| 19:12.07 | Eze_Arg | any clue? |
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| 19:15.12 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 19:16.15 | [TK]D-Fender | Eze_Arg, that doesn't show SIP debug to prove what side caused it to end or why |
| 19:16.22 | [TK]D-Fender | enable SIP debug and show another complete call |
| 19:16.36 | [TK]D-Fender | ~pb |
| 19:16.36 | infobot | from memory, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:16.37 | [TK]D-Fender | ^^^ |
| 19:16.47 | [TK]D-Fender | PASTEBIN it, do not flood. |
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| 19:19.04 | Eze_Arg | Ok D-Fender, will do that and come back, thanks ! |
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| 19:27.35 | tangel | sorry to bother you guys. anyone have a url about getting the message waiting indicator to work on cisco 7940 series phones? |
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| 19:34.35 | [TK]D-Fender | There is no url. If you specify the mailbox on your peer it should jsut work. |
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| 19:37.10 | artisangoose | Via ARI, is it possible to record a channel that is in a mixing/dtmf_events bridge? |
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| 19:48.35 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.1 (2016/09/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 20:12.40 | igcewieling | I'd tell a UDP joke, but you might not get it. |
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| 20:24.34 | DivideBy0 | doesnt acknowledge igcewieling |
| 20:24.59 | igcewieling | I'd tell a UDP joke, but you might not get it. |
| 20:25.03 | igcewieling | 8-) |
| 20:25.12 | DivideBy0 | artisangoose: yes, you can use snoop on the channel or you can record the whole bridge, also there's #asterisk-ari |
| 20:25.45 | artisangoose | Good to know about #asterisk-ari, thanks! We found some information about snooping, but the docs are confusing. :P We'll give it a try. For now, recording the bridge is sufficient. |
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| 20:27.08 | DivideBy0 | artisangoose: do you want just audio coming from the channel? This will do it: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-record |
| 20:27.51 | artisangoose | Yes, that's what I want, however I can't record the channel via ARI whilst the channel is in a bridge. And we need the channels to be bridged. |
| 20:28.18 | DivideBy0 | I don't believe that to be true, but I've never tried |
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| 20:28.42 | DivideBy0 | if it's true, create a snoop channel, then record that one |
| 20:29.19 | file | a snoop is best described as splitting the audio stream of a channel and duplicating it elsewhere... |
| 20:29.44 | DivideBy0 | I thought he was best described as a rapper that will do anything for $ |
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| 20:34.26 | igcewieling | or a beloved comicstrip beagle |
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| 20:43.16 | artisangoose | @DivideBy0 The ARI call to record a channel currently in a bridge does not fail, however no audio file is created for the channel. Further, if a recording has been started on a channel, it is not possible to join the channel into a bridge. Just get an error saying "channel XXX is currently being recorded" |
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| 20:44.51 | file | yeah, record is like calling Record() in the dialplan |
| 20:45.16 | file | a Snoop channel is what you have to use if you want to record that specific channel |
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| 20:50.34 | *** topic/#asterisk by wilhelm.freenode.net -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.0 (2016/09/26); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 20:50.47 | file | eh |
| 20:50.54 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.1 (2016/09/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 20:58.09 | DivideBy0 | I'm blame jetlag for the date loss |
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| 23:38.15 | scv | here's an interesting one |
| 23:39.16 | scv | just had somebody report to me that their voicemail has excessive gain after 7 seconds in each message |
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| 23:39.25 | scv | <PROTECTED> |
| 23:39.43 | scv | afaik vmgain/other gain options are DAHDI only right? |