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| 00:05.56 | WIMPy | mentok: No |
| 00:06.06 | mentok | no what |
| 00:06.08 | mentok | oh sorry. |
| 00:06.23 | WIMPy | grep your log or write somethign to display it in realtime. |
| 00:06.23 | mentok | Heh sorry, forgot I asked in here. |
| 00:36.13 | FuriousGeorge | anyone using asterisk with centos? |
| 00:36.36 | FuriousGeorge | the sysvinit script that comes with asterisk 13 does not work in centos 7. |
| 00:37.36 | FuriousGeorge | PID 707 read from file /var/run/asterisk/asterisk.pid does not exist or is a zombie. |
| 00:38.04 | FuriousGeorge | asterisk.service never wrote its PID file. Failing. |
| 00:38.37 | FuriousGeorge | rather than doing something hackish i thought someone in here may have a good init script for the rhel family |
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| 00:43.26 | RadicalDev | haven't enjoyed centos 7 much... this may help http://community.freepbx.org/t/start-asterisk-and-systemd/23115 |
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| 04:47.35 | moe` | uh does this work? |
| 04:47.47 | moe` | I just adjusted my ipfw |
| 04:47.54 | moe` | is there anybody out there? |
| 04:48.17 | moe` | should be now via my squid proxy |
| 04:49.20 | moe` | ok all that aside |
| 04:50.01 | moe` | if I can ask the greybeards here, does anyone know of a simple device to connect a analog line to an IP enabled SIP client? |
| 04:50.32 | moe` | does digium have such a thing that does not require a complete asterisk instance? |
| 04:51.00 | moe` | basically connect analog, hook it up to a SIP client on asterisk |
| 04:51.24 | moe` | the box has to do the analog to digital, and do the SIP uplink |
| 04:51.30 | moe` | this must exist |
| 04:51.40 | moe` | comments anyone? |
| 04:51.55 | moe` | [TK]D-Fender ? |
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| 05:07.13 | Samot | moe`: Are you talking about ATAs and FXS gateways? |
| 05:07.47 | Samot | Or are you saying I want to use X-Lite to connect to my phone line? |
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| 05:10.58 | linux4life | is there an address I can hook an aggregator to in order to display an rss feed of blogs.digium.com? |
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| 06:05.56 | moe` | sorry guys, I was looking for analog to SIP "adapters" |
| 06:06.18 | moe` | http://www.ebay.ca/itm/Unlocked-Linksys-PAP2-NA-2ports-Voip-gateway-ATA-1E-NEW-VOIP-PHONE-Adapter-SIP-G-/170840669684?hash=item27c6e5bdf4:g:mvAAAMXQO21Ruua8 |
| 06:06.24 | moe` | that sorta thing |
| 06:06.40 | moe` | so an analog line becomes a SIP client |
| 06:06.47 | moe` | so easy, so good. |
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| 06:24.27 | Lope | I'm trying to use my Nexus VoIP account from behind NAT using an outbound proxy, Siproxd. I'm getting an error "407 Proxy Authentication Required". I googled and saw something saying that if the phone authenticates with the proxy, the outbound proxy will/should pass the auth data on to the registrar. Apparently I can put user accounts in siproxd_passwd.cfg but I've not been able to find an example or the format that data must appear in this file? There is n |
| 06:24.27 | Lope | o sample file in my debian install. |
| 06:27.27 | Lope | scratch that. I just found a sample in the source. |
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| 06:49.45 | Lope | I've added the username and pass to my passwd file for siproxd. The Siproxd log acknowledges that my client authenticates itself successfully. But I'm still having the same issue with my SIP Registrar (provider). |
| 06:50.14 | Lope | I've got no problem with almost the exact same config (except host, user, pass) connecting to my Asterisk box from behind NAT through Siproxd. |
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| 09:10.50 | Lope | What is a realm, when specified on my softphone? is it the same as a context in sip.conf in Asterisk? |
| 09:12.05 | MaliutaLap | an auth realm ... user@mydomain.com |
| 09:12.29 | Lope | ah, thanks |
| 09:13.01 | MaliutaLap | normally you split the username from the realm, except in registration |
| 09:13.05 | Lope | I tried *. I'm just getting "404 not found" when trying to dial my test extension. |
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| 09:15.41 | MaliutaLap | what is the test extension? can you pastebin your sip.conf? |
| 09:16.02 | MaliutaLap | you may not even need to set the realm |
| 09:17.12 | Lope | Well, I've got 2 sip phones in the same context. I can call my test extension from my PC. It works, but maybe half the time I get no audio. |
| 09:19.48 | Lope | That's my sip.conf at the moment http://codepad.org/i41QhI8G |
| 09:22.21 | Lope | If I use the 'mobile' account from my PC it works. So must be something wrong in CSipSimple on Android. |
| 09:22.52 | Lope | I'm connecting via Siproxd a outbound SIP proxy |
| 09:25.41 | Lope | Strangely now when I try dial from my phone I get 408 Request Timeout. My tcpdump on ppp0 catches 0 packets. Which means it's not getting past Siproxd. |
| 09:26.34 | Lope | hmm: this doesn't look good: sip_utils.c:1179 sip_find_direction: unable to determine direction of SIP packet |
| 09:26.56 | Lope | (siproxd msg) |
| 09:32.22 | MaliutaLap | from the cli "sip show peers" ? |
| 09:32.57 | Lope | Nah the problem is actually between CSipSimple (Android) and Siproxd. The call is never making it out of the network. |
| 09:32.58 | MaliutaLap | I wouldn't be using an android softphone for testing |
| 09:33.43 | MaliutaLap | should it be making it out of the network? You should be able to dial both of those hosts from within * |
| 09:33.49 | Lope | I'm getting all of this traffic that looks like this: Android>Siproxd Invite, Siproxd>Android Status: 408 Request Timeout. |
| 09:33.57 | MaliutaLap | if they are both registered |
| 09:34.11 | Lope | Repeated over and over continuously. Perhaps I've got a time mismatch between my phone and the router. |
| 09:34.24 | MaliutaLap | and if you register directly to the * host? |
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| 09:34.56 | Lope | Well I've connected CSipSimple (Android) to my * Host before, when I had a different router that did black magic to make VoIP over NAT work. |
| 09:35.05 | MaliutaLap | I have had only bad experiences with sip proxies ... NAT was ok, but I got a /29 from my ISP specifically so I could give my * host a separate IP |
| 09:35.30 | Lope | But now I'm trying to control the black magic myself (Siproxd), and so far I still need to get CSipSimpleAndroid to play nice with Siproxd. |
| 09:35.44 | MaliutaLap | Lope: the iptables nat conntrack module works pretty well |
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| 09:36.05 | Lope | My * host has it's own static IP. My * is a VPS on the public internet. My PC softphone and my android softphone are behind NAT. |
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| 09:36.13 | MaliutaLap | you _should_ be able to use that, and then just set the NAT stuff in * properly |
| 09:36.44 | MaliutaLap | the only time it becomes an issue is if you have a dynamic IP, then you need to change it in * when your IP changes |
| 09:36.52 | Lope | MaliutaLap: Oh, can you please show me some example configs? PLEEEEEASE :) |
| 09:37.22 | Lope | Yeah I have a dynamic IP but I don't care about that. this is just for testing my * works properly. |
| 09:37.38 | MaliutaLap | so the * host is fine, on a static ... if the clients are behind a NAT then it's just your NAT rules that need doing |
| 09:37.42 | Lope | So I don't mind changing my IP on * every time it changes at the test-office. |
| 09:38.17 | Lope | What do I need to do to get multiple softphones to connect to Asterisk? |
| 09:38.28 | Lope | Should each softphone connect from a different local port? |
| 09:38.38 | Lope | Or does it not matter? |
| 09:39.23 | MaliutaLap | first you need to load nf_conntrack_sip and nf_nat_sip |
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| 09:39.51 | MaliutaLap | the clients behind a NAT doesn't matter, * shouldn't need to know about the NAT |
| 09:40.41 | Lope | My nat is straight up "IPTABLES -A POSTROUTING -o ppp0 -j MASQUERADE" with some TCPMSS hax. |
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| 09:41.40 | Lope | Well when I last tried to connect my clients from behind NAT they did stupid stuff like sending their IP on the LAN to the * server. Like foo@192.168.x.x |
| 09:41.48 | Lope | And obviously * can't reply to that. |
| 09:42.14 | MaliutaLap | you will need to be running contrackd, and the have some lines above that |
| 09:42.53 | Lope | MaliutaLap: do you have an example config you can share, with IPTABLES, a SIP definition and whatever other asterisk settings necessary? |
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| 09:43.42 | Lope | I want to test QUEUES. So my one softphone behind my NAT will be a member of the queue. Then another softphone behind my NAT will call up and pretend to be a customer, so I can test the system. |
| 09:43.43 | jkroon | MaliutaLap, do you have CONFIG_NF_CONNTRACK_SIP set? |
| 09:43.57 | jkroon | and CONFIG_NF_NAT_SIP? in the kernel. |
| 09:44.26 | MaliutaLap | jkroon: I do |
| 09:44.36 | jkroon | your experience and mine are seemingly very, very different. my advice. disable it and see if that doesn't solve your problems. |
| 09:44.36 | WIMPy | nf_nat_sip is evil. |
| 09:44.37 | MaliutaLap | Lope: look at https://home.regit.org/netfilter-en/secure-use-of-helpers/ for a start |
| 09:45.15 | MaliutaLap | WIMPy: I agree. But it's a required evil for some |
| 09:45.37 | jkroon | MaliutaLap, yes, if you expect SIP packets to be dropped randomly, and blocked pointlessly. |
| 09:45.46 | jkroon | and if you're OK with that. |
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| 09:46.02 | WIMPy | With Asterisk it will most probably cause more harm than good. |
| 09:46.16 | jkroon | for everybody else - disable it. even just having it LOADED causes problems. |
| 09:46.50 | WIMPy | What else could you do with it, more than having it loaded? |
| 09:47.08 | Lope | so would I need to add this "iptables -A FORWARD -m conntrack --ctstate RELATED -m helper --helper sip -d $ISP_RTP_SERVER -p udp -j ACCEPT" in addition to my normal NAT stuff? (-A POSTROUTING -o ppp0 -j MASQUERADE) |
| 09:47.15 | jkroon | it's supposed to mark related traffic for RTP purposes. so that conntrack RELATED will allow the RTP traffic. |
| 09:47.58 | WIMPy | That's the other module. |
| 09:48.02 | jkroon | I believe it also modifies packets with respect to contact ports etc for rtp purposes and then sets up expects for those, but since it doesn't work properly and caused significant issues for us we just nuked it. |
| 09:48.14 | WIMPy | And that can indeed be usefull. |
| 09:48.36 | jkroon | agreed. if it worked. |
| 09:48.45 | Lope | So would you all advise that I do not use Siproxd? |
| 09:48.45 | *** join/#asterisk d00gster_ (~d00gster@unaffiliated/d00gster) |
| 09:48.53 | MaliutaLap | I wouldn't |
| 09:49.00 | WIMPy | nf_nat_sip mangles the sip packets. nf_conntrack_sip provides the RELATED option. |
| 09:49.34 | Lope | So far siproxd is incompatible with Nexus.co.za and CallCentric.com. it works with SFLPhone on my PC, but so far not CSipSimple (Android) though maybe I've not found the right setting for CSipSimple yet. |
| 09:49.37 | jkroon | we had conntrack drop SIP packets at a point. perhaps that has been fixed. |
| 09:49.46 | MaliutaLap | someone is here uses the contrack and nat modules exclusively to allow SIP/RTP into their network |
| 09:49.54 | Lope | It seems like maybe I've got a time mismatch between my android phone and Siproxd. |
| 09:50.12 | WIMPy | So it's safe to have nf_conntrack_sip loaded and can be usefull. having nf_nat_sip loaded will mess with wour sip traffic and should be avoided. |
| 09:50.12 | Lope | It's instantly rejecting the phone's attempts at calling with request timeout, continuously. |
| 09:50.54 | MaliutaLap | sip proxies are bad mmm'kay |
| 09:50.56 | MaliutaLap | :) |
| 09:51.20 | jkroon | now if someone knows of a decent sip load balancer ... capable of helping me to deal with 100k+ SIP clients ... |
| 09:51.24 | MaliutaLap | jkroon: how long ago was that? |
| 09:51.31 | WIMPy | Or maybe it's rather SIP that should be avoided :-) |
| 09:51.39 | snadge | jkroon, you mean like kamailio? :p |
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| 09:51.44 | MaliutaLap | WIMPy: there is that too |
| 09:52.01 | jkroon | snadge, sigh ... in that direction once more :(. |
| 09:52.06 | jkroon | will check it out agai. |
| 09:52.18 | MaliutaLap | WIMPy: I'd probably set up another * instance and trunk them together with IAX |
| 09:52.27 | snadge | seriously.. we're an asterisk shop.. and i gave up on trying to load balance asterisk, given then limitations of our existing setup |
| 09:52.34 | snadge | if you were building it from scratch.. you might be able to |
| 09:52.45 | Lope | Okay, so Sip Proxies are bad. (seems like it to me). So I'll look at implementing some Conntrack magic on my NAT. |
| 09:52.48 | snadge | we got bought out.. and $upstream uses freeswitch.. with kamailio in front of it |
| 09:53.02 | jkroon | snadge, my benchmarks currently, with qualify=60 I can get to around 50k users on a single server. |
| 09:53.02 | WIMPy | MaliutaLap: Yes. That works. |
| 09:53.07 | jkroon | i've got a client that wants 2m. |
| 09:53.28 | snadge | if you're just connecting a and b parties.. then you dont really need asterisk do you |
| 09:53.32 | jkroon | fortunately I don't need single-server, but still, 40 servers is not going to fly. |
| 09:53.34 | WIMPy | Lope: What are you trying to talk to each other? |
| 09:53.41 | Lope | Can someone please answer my conntrack question: |
| 09:53.42 | Lope | would I need to add this "iptables -A FORWARD -m conntrack --ctstate RELATED -m helper --helper sip -d $ISP_RTP_SERVER -p udp -j ACCEPT" in addition to my normal NAT stuff? (-A POSTROUTING -o ppp0 -j MASQUERADE) |
| 09:54.04 | WIMPy | snadge: no |
| 09:54.18 | snadge | asterisk is barely adequate as a pbx.. at the provider level.. lolno |
| 09:54.38 | jkroon | Lope, normally I just have MASQUERADE, but auto-helper assignment is deprecated. I've not idea how to send traffic to the required helpers. |
| 09:54.54 | Lope | Wimpy: I've got a queue on my *, I need to connect 1 or 2 softphones to * as members of the queue. Then I need another softphone to call * as a customer so I can test it all. All my softphones are behind the same NAT. I control the router, it's debian. |
| 09:55.09 | MaliutaLap | Lope: iptables -A PREROUTING -t raw -p tcp --dport 5060 -d <asterisk_ip> -j CT --helper sip |
| 09:55.18 | MaliutaLap | and for udp |
| 09:55.39 | jkroon | Lope, with the right * options that should work without the conntrack helper assuming that the rtp ports are open. |
| 09:55.41 | WIMPy | Lope: Then don't try to fix the sip packets on the firewall and enable nat support for the peers in Asterisk. |
| 09:55.59 | MaliutaLap | jkroon: the * isn't on the NAT, it's the clients |
| 09:56.15 | jkroon | MaliutaLap, yes, and asterisk has options to assist with dealing with that. |
| 09:56.24 | WIMPy | [22:49] mlhess has joined #asterisk (~mlhess@drupal.org/user/102818/view) |
| 09:56.25 | jkroon | in sip.conf, check for nat= |
| 09:56.26 | WIMPy | [22:50] Echo6 has left IRC (Quit: Leaving) |
| 09:56.34 | WIMPy | [22:50] membiblio has joined #asterisk |
| 09:56.36 | Lope | jkroon: where do I specify the RTP (udp only?) ports for *? |
| 09:56.36 | WIMPy | (~membiblio@pool-71-112-149-68.pitbpa.fios.verizon.net) |
| 09:56.41 | WIMPy | Argh. |
| 09:57.07 | WIMPy | Yes, and Asterisks support for peers behind NAT works wuite well. |
| 09:57.19 | WIMPy | rtp.conf |
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| 09:57.26 | Lope | jkroon: are the RTP ports meant to be open on the * box only? And then I just let the clients punch holes through my NAT as necessary? |
| 09:57.27 | jkroon | Lope, the phones I normally use sip over tcp going to asterisk (recommended) if there is NAT (this stops the majority of "helpful" firewalls from breaking things). Then in rtp.conf you can find the port numbers. |
| 09:57.39 | jkroon | and setting the right nat= values in sip.conf should sort you out. |
| 09:57.59 | Lope | jkroon: isn't TCP shitty for calls? |
| 09:58.09 | jkroon | Lope, tcp for *control*, not voice. |
| 09:58.14 | jkroon | voice (rtp) remains on udp. |
| 09:58.40 | Lope | jkroon: oh, yeah i figured the SIP (signalling) would always use TCP. |
| 09:59.04 | jkroon | so then on your firewall you need to allow all udp "connections" to the server on the RTP ports (udp), and port 5060 either udp or tcp depending on which you choose to use. |
| 09:59.10 | WIMPy | Nope. Usually UDP as well. |
| 09:59.28 | jkroon | normally UDP. not too many ITSPs supporting SIP/TCP that I'm aware of. |
| 09:59.41 | Lope | jkroon: on my NAT firewall I allow anything outgoing, so that shouldn't be a problem. |
| 10:00.03 | jkroon | Lope, then I suggest you remove the SIP modules completely and stick with the normal masquerade. |
| 10:00.06 | MaliutaLap | jkroon: true, if the clients are always behind nat, then having 'nat=yes' in the peer definition |
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| 10:00.42 | jkroon | and set nat=force_rport,comedia |
| 10:01.15 | jkroon | or nat=auto seems to be a newer option but I haven't had the guts to test that at a large scale yet. |
| 10:02.32 | jkroon | can anybody explain why jumping from 30k SIP peers to 40k SIP peers (al with host=dynamic, all of them registered) with qualify=60 causes CPU usage to double? from 10k=>20k=>30k the CPU increased with a nice steady 5% on each jump. |
| 10:04.03 | Lope | okay so from beginning to end: 1. (plain) MASQUERADE on the NAT router. 2. sip.conf: [general]nat=force_rport,comedia 3. set RTP ports in rtp.conf 4. allow 5060 (tcp/udp) in and RTP port range (udp) in the * box's firewall? |
| 10:05.26 | WIMPy | That's the usual setup, yes. |
| 10:05.50 | Lope | jkroon: how can I force the use of TCP only for port 5060? Is that something I do on my softphones? or should I block UDP 5060 on my * box? |
| 10:06.17 | WIMPy | You have to configure your phones to use tcp. |
| 10:07.05 | jkroon | not all clients support it, but udp works equally well as long as your firewall doesn't try and fudge with it. |
| 10:08.07 | Lope | Can anyone recommend a decent linux softphone? I find Linphone to be terrible to configure. So clunky and pretty much inoperable. SFLphone lacks a contact list but has the most control in other ways. Ekiga has a contact list, but lacks control over RTP ports etc. |
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| 10:22.11 | cusco | hi.. we have a DDI that when a call comes in goes to a queue, and this queue has local members. pretty straight forward. Now is there a way, based on the callerid that I can try to deliver this call fromq ueue to a member first? |
| 10:22.28 | cusco | like preffered agent for caller? |
| 10:22.57 | cusco | I was thinking in the local dialplan that queue dials, to change the destination... |
| 10:23.08 | cusco | but perhaps you have other ideas of how this is doen? |
| 10:26.12 | jkroon | cusco, if you figure out a way ... please let me know. almost like a type of affinity (ie, how much does this specific caller and agent "like" each other, could be implemented as a type of penalty adjustment in the backend perhaps) |
| 10:26.19 | jkroon | but that's code modifications. |
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| 10:27.28 | jkroon | perhaps one could modify app_queue to read HASH(queue_affinity,${agent}) when calculating the next agent to be called, then one just need to populate the hash before going into Queue() ?? |
| 10:27.54 | jkroon | or even a specialized QUEUE_AFINITY(${agent}) function provided by app_queue. |
| 10:28.38 | WIMPy | Sounds like big plans. |
| 10:31.08 | Lope | I put tcpenable=yes under my softphone's SIP definition. It complained: 'TCP' is not a valid transport for 'testfoo'. we only use 'UDP'! ending call. |
| 10:31.21 | Lope | What's up with that? |
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| 10:33.00 | WIMPy | transport= |
| 10:33.15 | Lope | I've also got tcpenable=yes in [general]. Ok will try that. |
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| 10:34.16 | Lope | Thanks :) |
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| 10:37.00 | Lope | chan_sip.c:25660 handle_request_invite: Call from '' (1.2.3.4:1234) to extension '1234' rejected |
| 10:37.38 | Lope | Any ideas why it's saying call from (empty) ? The context of my softphone has been set to a ascii string with no other characters. |
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| 11:04.42 | cusco | also when a queue dials local exten as a member, if I want it to skip to the next one is there a difference if I return Busy() or Hangup() ? |
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| 12:33.01 | mlhess | WIMPy Did you need something? |
| 12:33.55 | utrack | Hi! my Asterisk has "Opus Codec" in the `core show codecs`, but I feel that it's disabled. Any way to check? |
| 12:34.19 | utrack | and do I need any patches or not |
| 12:35.08 | [TK]D-Fender | utrack, Use it |
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| 12:43.22 | Lope | WIMPy: jkroon MaliutaLap: thanks for your help earlier. It's such a relief to have a simple, reliable setup! |
| 12:44.12 | jkroon | Lope, glad you came right. just remember: less is more. Simpler is ALWAYS better. |
| 12:44.55 | Lope | jkroon: regarding your CPU usage jump you're probably running into a hardware limitation caused by some limiting configuration. |
| 12:45.25 | Lope | If I were you I would look into the logging, caching, paging etc intervals. Are you saving the calls to disk or something? What about virtual memory? |
| 12:46.01 | Lope | You're probably hitting some kind of IO or buffer saturation point and the CPU is juggling like crazy to keep things afloat. |
| 12:46.04 | jkroon | Lope, there is no dialplan on that even yet, however, this should give you an idea of what I'm up to: |
| 12:46.20 | jkroon | 50000 sip peers [Monitored: 50000 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 12:46.31 | Lope | jkroon: I agree. apt-get remove --purge siproxy. |
| 12:47.06 | jkroon | no disk involved. I had to a long time ago already move astdb off disk into RAM as well. |
| 12:47.51 | jkroon | my biggest issue at this point in time is chan_sip is single threaded and I can't upgrade to 13 trivially but it's definitely going to need to happen. |
| 12:48.27 | jkroon | especially since this client wants to aim at some rather insane number of SIP clients. |
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| 12:48.45 | Lope | jkroon: hmm, what about running multiple asterisk installs on the same server in LXC etc? |
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| 12:49.19 | Lope | why not just upgrade to 13? |
| 12:49.48 | Lope | It's probably easier in the long term than managing a crazy setup. |
| 12:50.30 | jkroon | Lope, that's part of the solution we need to look at. |
| 12:50.45 | jkroon | Lope, it's a non-trivial migration unfortunately. |
| 12:51.00 | Lope | What about logging, maybe asterisk is logging something to disk? have you checked your syslog and dmesg are not logging crazily to disk? |
| 12:51.01 | jkroon | and even asterisk 13 won't go to the scale we need on a single server. |
| 12:51.39 | jkroon | Lope, that would show up under iostat and i'm not seeing anything strange there. the CPU I'm measuring is purely userspace and system ticks as found in /proc/$(pidof asterisk)/stat. |
| 12:51.52 | Lope | If you can do multiple servers easily then I'd consider running them in LXC, better performance than virtualization. |
| 12:52.30 | jkroon | Lope, there are easier ways. on gentoo we've already built support for running any number of asterisk instances on the same host. |
| 12:53.13 | jkroon | ie, same linux instance. so just ln -s asterisk /etc/init.d/asterisk.name ... setup the config correctly and off you go. need to just bind on different IPs. |
| 12:54.28 | Lope | yeah, you're on your own now :p |
| 12:54.52 | Lope | What CPU are you running on? |
| 12:55.01 | Lope | Maybe it's a CPU limitation? |
| 12:55.03 | jkroon | Intel(R) Xeon(R) CPU E31220 @ 3.10GHz |
| 12:55.17 | jkroon | chan_sip is single-threaded for all practical purposes (rx at least). |
| 12:55.22 | Lope | What arch is that? sandy bridge, ivy, haswell etc? |
| 12:55.46 | jkroon | good question. |
| 12:55.57 | jkroon | but it shouldn't make those jumps. |
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| 12:56.15 | jkroon | oddly enough, 50k users is using the exact same CPU as 40k users ... |
| 12:56.22 | Lope | Well, it's worth checking if you run out of questions, find more questions until you find the answer you need. |
| 12:56.53 | jkroon | that's something I happen to be quite good at. so no need to worry about me - i'll manage this one :) |
| 12:57.04 | Lope | Yeah well, if you put x amount of water down a pipe y wide, it can handle, but then try putting x*1.2 and you might find you've got a bottleneck problem. |
| 12:57.26 | Lope | It might be that your CPU usage was 40% or whatever, but some component in it was frying it's tits off. |
| 12:58.29 | Lope | The other thing I thought of is the kernel limits on number of open files. That can affect open connections etc |
| 12:58.40 | jkroon | LOL, well, it's still below 50% which is where we think we want to push to on average (keep sufficient buffer space for spikes etc ...) |
| 12:59.03 | jkroon | Lope, ran into that particular problem about 4 years ago already :). ulimit is your friend. |
| 12:59.27 | Lope | hehe yeah, I know. I ran into that issue load testing a webserver :P |
| 12:59.35 | jkroon | in this case we've got one socket, udp on port 5060 so no fd limits (current benchmarks is running with ulimit 1024) |
| 13:00.01 | jkroon | actual open fd's is about 25 though. |
| 13:00.23 | Lope | You should definitely try different CPUs. At very least it will be interesting. |
| 13:00.41 | jkroon | mostly the number of available clock cycles on a single core is the important bit. |
| 13:00.55 | jkroon | http://jkroon.blogs.uls.co.za/it/voip/asterisk-massively-speeding-up-those-register-requests - that is pretty much the only asterisk disk IO issue I ever bumped into. |
| 13:01.15 | Lope | You've probably considered using separate hosted VPS's etc as well? Do you get much more bang for the buck renting your own dedi? |
| 13:01.34 | jkroon | recordings can get an issue, but they don't actively use sync() of fsync() as does sqlite. |
| 13:02.28 | jkroon | Lope, yes, our "per instance" cost then is something like R180/month for entry level requirements. With the current setup with multiple asterisk instances on one server our cost is approximately R15/instance. |
| 13:02.48 | Lope | lol, are you in South Africa? |
| 13:02.54 | jkroon | indeed I am :) |
| 13:03.10 | Lope | hahaha. Cool. Almost no nerds left there in Zuma can't count land. |
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| 13:59.12 | Lope | Can anyone recommend a decent Linux Softphone? |
| 13:59.56 | Lope | SFLPhone: Has annoying audio bug requiring me to type a number for the sound to work (dbl click history is hit and miss with ALSA) |
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| 14:09.57 | Lope | Linphone's basically unusable. Twinkle's UI is broken on ubuntu. Yate's contact list is broken. |
| 14:10.04 | Lope | Ekiga actually seems to work... |
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| 14:28.45 | kippi | hey |
| 14:29.00 | kippi | can anyone think of any reason why I am only able to start 100 mettle rooms? |
| 14:29.04 | kippi | meetme |
| 14:50.01 | rmudgett | The meetme mixing is single threaded and handled by DAHDI |
| 15:12.45 | avb | lorsungcu: im using jitsi. its written in java, but not that bad |
| 15:12.54 | avb | ups |
| 15:12.59 | avb | wrong nick |
| 15:13.00 | avb | sorry |
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| 17:10.42 | Kevin` | is there a way with normal dialplan funtions to execute a dialplan and/or shell comman when a user presses certain dtmf keys? |
| 17:10.48 | Kevin` | during a call |
| 17:11.00 | WIMPy | "features" |
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| 17:21.15 | ice9 | just installed Asterisk for the first time, created a sip account in sip.conf and trying to login via softphone and getting "wrong password" in asterisk log, however the username and password matches the conf |
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| 17:22.57 | ice9 | also sip show peers, doesn't show the account I created |
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| 17:25.33 | ice9 | sip show users doesn't retrieve any users either |
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| 17:27.07 | WIMPy | Smells like a broken config. |
| 17:27.39 | ice9 | oh seems I should create account in users.conf not sip.conf! |
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| 17:29.35 | WIMPy | ~users.conf |
| 17:29.35 | 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. |
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| 17:30.44 | ice9 | ~sip.conf |
| 17:35.02 | ice9 | by default call encryption is enabled in asterisk? |
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| 17:35.42 | WIMPy | No |
| 17:36.04 | WIMPy | And there are different types of encryption. |
| 17:36.55 | file | it also requires client support |
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| 17:42.38 | ice9 | so whine type do you recommend, srtp, zrtp or tls? |
| 17:42.57 | ice9 | client supports all! |
| 17:45.14 | ice9 | /whine/which |
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| 18:02.30 | qakhan | can we send calls to agents in a queue on time base, e.g. 8AM to 4PM calls go to Agent1 - Agent 4 and 4PM to 12AM calls go to Agent5 - Agent8 |
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| 18:03.34 | WIMPy | Use different queues and some *IfTime thing. |
| 18:05.21 | ice9 | astgenkey generates .key and .pub files, how to convert them to PEM? |
| 18:05.57 | qakhan | i cannot use different queue cuz we have integration with an app which use queue name to show caller id when call comes in |
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| 18:15.34 | RadicalDev | Any idea when, with asterisk 13, I originate a call from an agi script, the call is answered, things happen, and then the caller hangs up, the original channel drops back into the dialplan at the same context,exten,priority that call the agi script that originated the call? |
| 18:15.58 | RadicalDev | things happen = call is bridged, etc |
| 18:16.56 | RadicalDev | with asterisk 1.8, the caller channel drops to context,exten,priority+1 |
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| 20:05.01 | linetrace | got a timing question: which is better, a timing interface for which `timing test` takes 1003-1009 milliseconds or a timing interface for which `timing test` loses 1-3 ticks, but consistently takes 1000 milliseconds? |
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| 20:15.00 | linetrace | i've been working to get asterisk more stable on os x and am currently helping test res_timing_kqueue... naturally, kqueue should be more performant on os x & BSDs and is the one that takes a consistent amount of time, but currently misses ticks (res_timing_pthread is only other available and takes an inconsistent amount of time) |
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| 20:49.47 | afallison | I'm using Asterisk 13.6 with Twilio SIP Trunk. I let Twilio manage all call recordings. When the call ends, Twilio sends a header back called `X-Twilio-CallSid`. On my outbound trunk I use the Dial Option Ttb(outbound-hangup,s,1) which currently just Noop with misc headers and values for testing, including `X-Twilio-CallSid`. When the phone is hung up and outbound-hangup runs it looks like it is returning SIP_HEADER headers from m |
| 20:50.07 | afallison | So when I hangup a call, a BYE is sent from the SIP phone to Asterisk followed by another BYE sent from Asterisk to Twilio. My question is, how can I capture that final response from Twilio's trunk? Or can I? |
| 20:50.17 | afallison | Thanks! |
| 20:58.28 | jeff | afallison: your first message truncated at "it looks like it is returning SIP_HEADER headers from m" |
| 21:02.41 | afallison | from my SIP phone and not from the trunk. |
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| 22:22.18 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.1 (2016/04/14), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 23:11.26 | avb | ; same=n,GotoIf($["x${SIP_HEADER(P-Asserted-Identity)}" != "x"]?doDial) |
| 23:11.26 | avb | ; same=n,SIPAddHeader(P-Asserted-Identity: <tel:${CLID6}>) |
| 23:11.57 | avb | guys, how its possible that SIP_HEADER(P-Asserted-Identity) returns a variable, but dial() is sending a call without it? |
| 23:12.12 | avb | is confused |
| 23:12.26 | avb | that happends with forwarded call |
| 23:13.16 | avb | i assume it already have a pai header, but dial() is not reusing it :-/ |
| 23:13.40 | avb | if i noop() it, there is a value |
| 23:15.59 | avb | http://pastebin.com/yhqSvLEV |
| 23:16.06 | avb | thats a whole dialplan |
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