00:11.57 | *** join/#asterisk SpenglerMBP (~spenglerm@pool-98-117-215-165.bltmmd.fios.verizon.net) |
00:15.04 | SpenglerMBP | having some problems connecting an external sip client to my asterisk box which is behind a nat |
00:15.14 | SpenglerMBP | can anyone give me some pointers? |
00:15.21 | SpenglerMBP | I have been researching for a few days |
00:15.43 | SpenglerMBP | I have forwarded ports 5060udp/tcp and 10000-20000udp/tcp |
00:15.46 | SpenglerMBP | still no dice |
00:24.49 | [TK]D-Fender | "sip set debug on" |
00:25.00 | [TK]D-Fender | And go look at the actual comm attempts |
00:40.41 | *** join/#asterisk suneye (~atcmmi@116.25.194.224) |
00:41.31 | *** join/#asterisk atcmmi (~atcmmi@116.25.194.224) |
00:42.40 | Spengler1 | when i enable sip debug on it scrolls so fast |
00:42.46 | Spengler1 | anyway to output to a file? |
00:50.31 | [TK]D-Fender | just cut&paste... |
00:53.13 | *** join/#asterisk suneye (~atcmmi@116.25.194.224) |
01:09.05 | Spengler1 | i'm looking now ; anything specific i should look for? |
01:09.36 | [TK]D-Fender | All of is. If you can't tell what you're looking for ... show us. That's the point. |
01:09.43 | [TK]D-Fender | ~pb |
01:09.43 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
01:09.46 | [TK]D-Fender | ^^^ your friend |
01:11.09 | Spengler1 | http://pastebin.com/6JGZjSHN |
01:11.16 | Spengler1 | 2005 is the external sip extension |
01:11.22 | Spengler1 | 2003 is a hard phone |
01:12.01 | [TK]D-Fender | Go get a complete registration attempt... or complete call.. or something |
01:12.05 | [TK]D-Fender | a llt more than that... |
01:12.09 | [TK]D-Fender | you shoudl pages worth.... |
01:12.16 | [TK]D-Fender | lot* |
01:12.19 | [TK]D-Fender | gah.... |
01:15.25 | Spengler1 | http://pastebin.com/GjttSQ4Y |
01:17.46 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
01:19.59 | carrar | <PROTECTED> |
01:20.02 | carrar | woops |
01:20.15 | [TK]D-Fender | Spengler1: Reliably Transmitting (no NAT) to 66.241.99.27:5060: Contact: <sip:s@192.168.4.204:5060> |
01:20.42 | [TK]D-Fender | Spengler1: You have given Vitelity a PRIVATE IP as the means of sending you calls. This shows your sip.conf is not set up right to work behind NAT |
01:21.12 | Spengler1 | in my configuration? |
01:21.36 | Spengler1 | how should i change it? |
01:21.43 | [TK]D-Fender | Spengler1: add : nat=yes , directmedia=no , localnet=x.x.x.x/y.y.y.y (for your subnet) , externaddr=(your WAN IP or dyndnsname, etc) |
01:22.05 | [TK]D-Fender | Spengler1: and in their peer, "nat=no", because THEY are not behind NAT |
01:22.32 | [TK]D-Fender | Apply all this and show a new trest |
01:22.36 | [TK]D-Fender | test* |
01:23.05 | Spengler1 | the nat=yes , etc goes under the global config right? |
01:23.49 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
01:24.29 | [TK]D-Fender | yes |
01:24.31 | [TK]D-Fender | all of that |
01:24.49 | [TK]D-Fender | * needs to know what is loca nad what isn't and what to report to others when you call out |
01:24.51 | Spengler1 | when you say their peer do you mean their config []? |
01:25.16 | [TK]D-Fender | [vitelity] or whatever that peer is for your setup for them |
01:25.22 | Spengler1 | gotcha |
01:25.24 | Spengler1 | brb |
01:25.37 | [TK]D-Fender | there were two parts being fixed here, your setup behind NAT and them NOT being behind it. |
01:25.51 | [TK]D-Fender | So once you get that down then we can see how everything else looks |
01:26.21 | [TK]D-Fender | this is just a test of my new keyboard. |
01:26.24 | [TK]D-Fender | oops... |
01:27.55 | *** join/#asterisk ani216 (8ec5e97e@gateway/web/freenode/ip.142.197.233.126) |
01:27.59 | ani216 | Hello |
01:28.55 | Spengler1 | I made the changes , reloaded and here is the result |
01:28.57 | Spengler1 | http://pastebin.com/A1FTUFks |
01:29.04 | Spengler1 | thanks for your help |
01:30.16 | *** join/#asterisk ani216 (~ani216@142.197.233.126) |
01:30.20 | [TK]D-Fender | znat=yes is deprecated, use nat=force_rport,comedia instead <- do follow that |
01:30.44 | ani216 | if someone has a chance and has video experience with asterisk 10 i have a small problem |
01:30.50 | [TK]D-Fender | I didn[t accomodate the latest changes. It should still work.. but no sense in doing the job half-right |
01:31.17 | [TK]D-Fender | Reliably Transmitting (NAT) to 192.168.4.131:5060: <--- [2003] is NOT behind NAT it would seem .. you should set them accordingly |
01:31.18 | Spengler1 | also should i set that on my external sip peer? |
01:31.37 | Spengler1 | 2003 is behind nat |
01:31.47 | [TK]D-Fender | Retransmitting #8 (NAT) to 192.168.4.1:41174: <- and this is a LOT of retransmits.... |
01:31.55 | [TK]D-Fender | This phone LOOKS local .... is it? |
01:32.13 | Spengler1 | the local phone is 2003 ( 192.168.4.131) |
01:32.20 | Spengler1 | the external is 2005 |
01:33.12 | [TK]D-Fender | Show an actual call now. |
01:35.05 | Spengler1 | http://pastebin.com/ANRBVSJr |
01:37.46 | [TK]D-Fender | Reliably Transmitting (no NAT) to 66.241.99.27:5060: REGISTER sip:inbound28.vitelity.net SIP/2.0 Contact: <sip:s@192.168.4.204:5060> |
01:37.52 | [TK]D-Fender | Still not right |
01:38.02 | [TK]D-Fender | pastebin your general section. mask only passwords |
01:39.19 | Spengler1 | http://pastebin.com/YWgHQaKZ |
01:40.05 | [TK]D-Fender | ftp.crsmd.com <- ping it from CLI |
01:40.19 | [TK]D-Fender | did you create a HOST entry for it that resolves as LOCAL?> |
01:40.50 | Spengler1 | no ; its my dyndns ; it resolves to my external DNS |
01:40.58 | Spengler1 | to my external WAN i mean |
01:43.17 | [TK]D-Fender | Show me |
01:43.54 | Spengler1 | http://pastebin.com/Fpt519YW |
01:44.36 | [TK]D-Fender | Ok, something is a little out here.. restart * to be sure |
01:44.44 | [TK]D-Fender | and pastebin "sip show settings |
01:45.05 | [TK]D-Fender | OH, and add externrefresh=180 |
01:45.13 | ani216 | [TK]D-Fender: do you happen to know about video support on * 10? |
01:45.14 | [TK]D-Fender | just because it could change over time |
01:45.25 | [TK]D-Fender | ani216: It does. |
01:45.45 | ani216 | do you know who i could talk to about getting help with it? i have a cisco phone and linphone and its not passing video |
01:45.51 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
01:45.51 | Spengler1 | http://pastebin.com/NW3vJ7Uz |
01:45.59 | [TK]D-Fender | needs to finish rewriting his NAT guide... |
01:46.31 | [TK]D-Fender | <PROTECTED> |
01:46.33 | [TK]D-Fender | <PROTECTED> |
01:46.38 | [TK]D-Fender | Something wrong there.. pastebin it again... |
01:46.44 | [TK]D-Fender | [general] that is |
01:46.49 | [TK]D-Fender | make sure you've got all of it |
01:46.53 | [TK]D-Fender | (masking passwords) |
01:46.59 | Spengler1 | i will again |
01:47.27 | [TK]D-Fender | ani216: I'd start by showing us a call with an actual error in it. |
01:47.39 | ani216 | ok let me get the debug..thank you |
01:47.44 | Spengler1 | http://pastebin.com/zSe8MF9E |
01:48.11 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
01:49.42 | [TK]D-Fender | externaddr=ftp.crsmd.com <- change for externhost ... |
01:50.03 | [TK]D-Fender | it SHOULDN't require that one.. but lets give it a whirl. the restart * and "sip show settings" |
01:51.06 | Spengler1 | change ftp.crsmd.com |
01:51.16 | Spengler1 | or change externaddr to externhost |
01:53.57 | [TK]D-Fender | latter |
01:54.03 | ani216 | [TK]D-Fender: here is the log.. http://pastebin.com/kde2aEmJ |
01:55.14 | [TK]D-Fender | ani216: that is not a compete call, doesn't show any video codecs being offered and doesn't actually show any packets from either other side |
01:55.45 | ani216 | sorry first time having to debug..let me find out how to cap it all |
01:58.12 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
01:58.53 | Spengler1 | http://pastebin.com/qv8JgK2t |
01:58.59 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
02:00.15 | [TK]D-Fender | <PROTECTED> |
02:00.16 | [TK]D-Fender | <PROTECTED> |
02:00.19 | [TK]D-Fender | <PROTECTED> |
02:00.19 | [TK]D-Fender | BETTER |
02:00.24 | [TK]D-Fender | now try another call. |
02:05.05 | Spengler1 | still no audio |
02:05.40 | Spengler1 | [Jun 3 22:05:25] WARNING[17237]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 73774AD7C87FB2C3CE714C6696122468FBF7755F for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
02:05.43 | Spengler1 | seeing this as well |
02:06.31 | [TK]D-Fender | show the full call |
02:09.18 | Spengler1 | http://pastebin.com/expXzAYm |
02:09.44 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
02:10.01 | ani216 | [TK]D-Fender: sorry about there..here ya go i think this is right.. http://pastebin.com/xDx36YF4 |
02:12.22 | [TK]D-Fender | Spengler1: Retransmitting #5 (NAT) to 192.168.4.1:7080: <-- seem you still didn't set nat=no for that local phone |
02:12.52 | [TK]D-Fender | hold on a sec.. |
02:12.58 | [TK]D-Fender | Something looks off.. |
02:13.09 | [TK]D-Fender | <--- SIP read from UDP:192.168.4.1:7080 ---> INVITE sip:2003@ftp.crsmd.com SIP/2.0 |
02:13.22 | [TK]D-Fender | That looks like a aLOCAL IP is targeting your WAN IP... |
02:13.36 | [TK]D-Fender | which is NTO GOOD |
02:13.38 | [TK]D-Fender | NOT |
02:13.47 | [TK]D-Fender | This runs into a hairpin NAT issue |
02:14.16 | [TK]D-Fender | o=- 65627 25434 IN IP4 10.192.148.118 <- these other IP's are looking off as well |
02:14.37 | [TK]D-Fender | I think you'd better give a much better description of your networking here.... |
02:15.43 | Spengler1 | i just disabled nat on phones |
02:15.47 | Spengler1 | on locals i mean |
02:16.02 | [TK]D-Fender | ani216: use a proper SSH client for your debugging. this one is skipping every other line and littering your output with ANSI crap |
02:16.26 | [TK]D-Fender | those other IP's look unaccounted for and I'm not sure what to trust right now. |
02:16.42 | [TK]D-Fender | So be VERY cler of exactly what is where, of what routing your behind, etc |
02:17.14 | Spengler1 | its a cisco rv220w |
02:17.33 | Spengler1 | all phones / server behind in the 192.168.4.0/24 network |
02:17.45 | Spengler1 | the 2005 extension is on my iphone |
02:17.50 | Spengler1 | connected to the cell network |
02:17.53 | [TK]D-Fender | change your phones so they point to your server's IP, not the DYNDNS name |
02:17.58 | [TK]D-Fender | this has to stay local |
02:18.07 | Spengler1 | yes they r pointing local |
02:18.17 | Spengler1 | they were configured through dpma |
02:18.31 | [TK]D-Fender | --- SIP read from UDP:192.168.4.1:7080 ---> INVITE sip:2003@ftp.crsmd.com SIP/2.0 <- oh no they aren't |
02:18.37 | [TK]D-Fender | look at the target |
02:18.47 | [TK]D-Fender | not @localIP |
02:18.54 | [TK]D-Fender | and I can't help you with DPMA.... |
02:19.14 | [TK]D-Fender | User-Agent: Acrobits Softphone/5.3.3 <- which THIS isn't |
02:19.24 | [TK]D-Fender | this softphone is not setup to point to the IP direct.. |
02:19.42 | Spengler1 | I would point it to my external ip? |
02:21.59 | [TK]D-Fender | it clearly has the DYNDNS name right now |
02:22.01 | [TK]D-Fender | this is BAD |
02:22.10 | [TK]D-Fender | it should only have been told about the LOCAL IP. |
02:23.17 | Spengler1 | well if i disable my port forwarding rules then the phone wouldn't be able to connect to the asterisk server right? |
02:25.21 | [TK]D-Fender | you said everything was LOCAL |
02:25.36 | Spengler1 | yes except for the softphone |
02:25.46 | Spengler1 | i'm saying to prove that it is local |
02:25.53 | Spengler1 | disable the port forward on the firewall |
02:26.19 | [TK]D-Fender | facepalms |
02:26.25 | [TK]D-Fender | Lets try this again... |
02:26.35 | [TK]D-Fender | that SOFTPHONE says it is coming for YOUR network |
02:26.44 | [TK]D-Fender | YouWhere precisely is it? |
02:26.54 | Spengler1 | it is on the cellular network |
02:27.06 | [TK]D-Fender | How the hell is it sourceing as LOCAL then? |
02:27.18 | [TK]D-Fender | You leave **WIFI** enabled on it? |
02:27.28 | Spengler1 | i disabled wifi |
02:27.39 | [TK]D-Fender | [22:18][TK]D-Fender--- SIP read from UDP:192.168.4.1:7080 ---> INVITE sip:2003@ftp.crsmd.com SIP/2.0 <- oh no they aren't |
02:28.13 | Spengler1 | the 192.168.4.1 would indicate it is coming from the gateway? |
02:28.50 | [TK]D-Fender | If you have any kind of SIP helper / ALG on that router, BURN IT |
02:29.05 | Spengler1 | it is disabled |
02:29.10 | [TK]D-Fender | Then what is that IP? |
02:29.26 | Spengler1 | 192.168.4.1 is default gateway |
02:29.38 | [TK]D-Fender | Your networking is screwed up |
02:29.50 | [TK]D-Fender | your router should be be listed as the originator of that outside call. |
02:30.05 | [TK]D-Fender | It is either just completely broken, or it's setup is |
02:30.41 | Spengler1 | what would it look like if it were proper? |
02:31.25 | [TK]D-Fender | It's not be coming from your router |
02:31.45 | Spengler1 | it would be coming from the ip of the phone correct? |
02:31.53 | [TK]D-Fender | yes |
02:32.03 | Spengler1 | its like it is translating wrong? |
02:32.06 | [TK]D-Fender | your router IS proxying the comms |
02:32.14 | [TK]D-Fender | it isn't translating.. it is PROXYING |
02:32.59 | [TK]D-Fender | I've heard nothing but trouble from thr RV series... |
02:33.08 | Spengler1 | what do you recommend? |
02:33.17 | [TK]D-Fender | Anything else.. |
02:33.22 | [TK]D-Fender | but first beat it up as best you can |
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02:38.21 | Spengler1 | what router do you typically use? |
02:41.40 | [TK]D-Fender | I jsut recently switched to Microtik. Boring Linksys works fine, aso do most cheap routers.... DD-WRT has had issues, as has OpenWRT at times. |
02:41.55 | ani216 | [TK]D-Fender: what ssh program do you recommend to log it? im on windows |
02:42.51 | [TK]D-Fender | zPutty |
02:42.54 | [TK]D-Fender | -z |
02:43.15 | ani216 | thats what im using :( |
02:45.16 | [TK]D-Fender | In a VM by any chance? |
02:46.17 | Spengler1 | hey [TK]D-Fender ; could you look at my configuration for the [2005] extension and see if it is correct? |
02:46.40 | [TK]D-Fender | Spengler1: No point until you fix your router. |
02:46.57 | [TK]D-Fender | Spengler1: This is a DOA issue |
02:47.47 | Spengler1 | any hints as to how to fix it |
02:49.57 | Spengler1 | http://pastebin.com/NY7a0n1X |
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02:53.07 | [TK]D-Fender | The problem isn't your peer. It's your router. |
02:53.10 | [TK]D-Fender | Go read the manual |
02:53.19 | [TK]D-Fender | Go double-check your settings. |
02:53.28 | [TK]D-Fender | Then make it quintuple or more. |
02:55.48 | ani216 | [TK]D-Fender: is this better? http://pastebin.com/UQPggUk7 |
02:57.17 | Spengler1 | I had a look at a lot of info, and it starts to look to me that the RV220W not only hides the LAN form the WAN, but also the other way around. So, an SMTP-client from the outside will be translated to the inside IP address of the router (e.g. 192.168.1.1) and a free port will be found for the mapping. |
02:57.22 | Spengler1 | i found that lol |
03:00.15 | Spengler1 | damn cisco |
03:00.38 | ani216 | Spengler1: thats what im saying i got a 8945 video phone and its not working right inside :( |
03:00.50 | Spengler1 | I do have an apple time capsule ; maybe i should hook that up and convert my rv220w into a straight router with no nat |
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03:07.15 | [TK]D-Fender | ani216: I see you jsut grep'd out any spaces anywhere |
03:07.37 | ani216 | best i could do :) |
03:07.41 | ani216 | :( oops |
03:07.51 | ani216 | it was a lot worse |
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03:09.39 | [TK]D-Fender | ani216: Capabilities:us-(gsm|ulaw|alaw|h263),peer-audio=(ulaw|alaw|g729|ilbc|g722|slin16)/video=(h264)/text=(nothing),combined-(ulaw|alaw) |
03:09.51 | [TK]D-Fender | ani216: they aren't starting offering video |
03:12.56 | [TK]D-Fender | ani216: And you only debugged one HALF of the call |
03:15.53 | ani216 | ok ill try again tomorrow |
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05:11.25 | linocisco | hia all |
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05:53.49 | p7ank5te7 | Anyone familiar with using MPG123 with MusicOnHold? |
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06:27.02 | ChannelZ | I think I used it once a couple of years ago for fun |
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06:31.00 | p7ank5te7 | ChannelZ: Is there a restriction on the file names? I've got it set and if I have no spaces or anything, the file will play, but if I put it back in the directory I want it to, it causes asterisk to blow out errors like hot cakes and nothing for hold music. |
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06:33.36 | p7ank5te7 | blows out: res_musiconhold.c:643 monmp3thread: poll() failed: Interrupted system call |
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06:43.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
06:43.59 | p7ank5te7 | ChannelZ: I just found an old IRC log of you helping someone with this same exact error.. LOL |
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07:00.58 | linocisco | hi all |
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08:49.04 | alexann | Hallo, I'd like to hear someone's opinion about the difficulty of this project we are about to do. Consider that I don't know anything of asterisk, so one of the goals now is to understand who we should hire in order to do this. But, I need to have some numbers for our client by this evening, that is the cost of the various service providers. Basically we want to create a bot (well, asterisk) that is able to call the users phones, have a s |
08:49.05 | alexann | conversation with them where each line pronounced by the system (they'll be audio file) depends on the previous answer of the user (speech recognition, in fact we need to intercept the audio stream and pass it to a 3rd party speech recognition engine) and some logic that can be handled by an external module. Saying that the requirements are up to 200 concurrent conversations, and that the conversations will take place in the USA only, wha |
08:49.05 | alexann | services should we buy? One VOIP provider, one hosting solution for asterisk. How difficult is it to write the asterisk configuration? Can you advice some provider for these services? Is anybody up for a short term contract position? Thank you |
08:51.37 | ChannelZ | I say, hire some illegals to make the calls. |
08:54.14 | alexann | heh, why? |
08:54.55 | alexann | the ASR is not an issue, as long as we can intercept the audio stream |
08:56.34 | alexann | can you help me to separate the actors in such a project: professionals, software, facilities? |
08:59.13 | ChannelZ | sorry it's WAY past my bed time. You'll probably have better luck with a response in a few hours when some people start waking up |
09:00.01 | alexann | hehe, ok thanks :) |
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09:07.38 | alsax | I'll stay… on CET |
09:21.32 | jacekowski | alsax: is the conversion predefined? |
09:22.01 | jacekowski | or it's using some kind of ai? |
09:22.43 | jacekowski | google has voice recognition that can be linked into asterisk |
09:23.17 | jacekowski | but probably all you need is pretty much standard ivr type system |
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09:25.29 | alsax | hey there jacekowski : it's using some kind of ai, plus we have a custom speech recognition engine |
09:25.40 | alsax | se basically all we need to do is to get the audio stream |
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09:29.41 | alexann | do you know what is needed? I have seen voip providers |
09:29.50 | MrQuist | Freeaqingme, hi there |
09:30.18 | Freeaqingme | ohi |
09:30.42 | alexann | for what I get, they take care of the telephony part, then I need a server where to install asterisk |
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09:48.27 | the_5th_wheel | howdy, Im having a strange issue on a asterisk box. its running asterisk 11, and my snom 370, which has BLFs setup is not working, every time an event is sent it sends back sip 400. Anyone else experience that? |
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10:33.29 | MrQuist | have you checekd the callroutes? |
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11:40.03 | Freeaqingme | Hey guys, can anyone explain how I should interpret this regex when used as an extension 'number'? _[0-9+]! |
11:40.09 | Freeaqingme | more specifically, the ! at the end? |
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11:41.43 | kaldemar | Freeaqingme: it's not a regex, but a pattern: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
11:47.54 | msaraiva | So, the problems with Motif and Hangouts have started... |
11:48.18 | msaraiva | Google "ftl" |
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12:00.02 | Freeaqingme | kaldemar, ntx |
12:00.03 | Freeaqingme | * tnx |
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12:53.59 | danfromuk | Hi, when a call comes in, it can go through a number of Local/ channels before being answered ending in multiple cdr lines. I need to be able to group calls together so I can see which cdr lines are connected to the same call. I thought that uniqueid would help but that increments a few times during the call. |
12:54.35 | danfromuk | I thought I could set CDR(userfield) but it only affects one cdr line and doesnt get inherited to new channels. |
12:54.59 | danfromuk | I also set CDR(accountcode) which does get inherited when dialing other Local/ channels |
12:55.55 | Greenlight | I guess you could send something in the SIP header, and than set that to a custom CDR field |
12:56.12 | Greenlight | Hmm but not with Local actually, so scratch that |
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12:56.29 | kaldemar | danfromuk: what's your issue? |
13:00.05 | danfromuk | kaldemar: A call comes in, rings an extension for a few minutes, then diverts to an external number. Due to the dialplan, this results in a few cdr lines. I want to set a value at the start of the incoming call which is inheritted and written in each line of the cdr. |
13:00.19 | danfromuk | In the same way that cdr(accountcode) is inherited for each line. |
13:01.19 | kaldemar | why are you not using accountcode? |
13:01.33 | Greenlight | You can prefix a variable with __ to ensure it's inherited |
13:01.54 | danfromuk | kaldemar: I am, but that contains the client's account number. |
13:01.57 | Greenlight | And then pick that up in the 2nd call and use it to set somethng in CDR |
13:02.14 | kaldemar | one _ for inherited once, two for indefinitely. |
13:02.41 | danfromuk | Does __CDR(userfield)="test" work at all? |
13:03.04 | Greenlight | No, I don't think you can inherit those |
13:03.34 | Greenlight | But Set(__CustomCDRField=Test) would then allow Set(CDR(userfield)=${CustomCDRField}) |
13:03.44 | Greenlight | So you can do it that way |
13:03.52 | danfromuk | Greenlight: Ok, good idea. I'll give that a try. |
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13:31.25 | saint_ | Is there a special way to prirotize g729 over g711 in Asterisk / |
13:31.54 | Greenlight | The order of the defenition of allowed codecs decides the preferred one |
13:32.17 | Greenlight | For example allow=g729,alaw would prefer g729 |
13:32.26 | Greenlight | BUt there's no guarentee the other side will play ball |
13:32.31 | saint_ | so ig I have g729,ulaw and my call always do g711 , then is the issue on the asterisk , or at my provider ? |
13:32.40 | danfromuk | Greenlight: your suggestion works. thanks. |
13:32.57 | Greenlight | danfromuk: Excellent, glad to help! |
13:32.58 | saint_ | I mean.. is this something that you guys are using and is working ? or is ti known to have issues ? |
13:33.19 | Greenlight | saint_: A negiotiation occurs |
13:33.21 | saint_ | cause I found this as an issue, but it does not resolve my problem.. https://issues.asterisk.org/jira/browse/ASTERISK-6037 |
13:33.21 | LieutPants | [ASTERISK-6037] [Status: Closed] codecpriority=caller does not seem to work - https://issues.asterisk.org/jira/browse/ASTERISK-6037 |
13:33.32 | Greenlight | There is no guarentee |
13:33.42 | Greenlight | You say "I support g729 and ulaw, but prefer g729" |
13:33.49 | igcewieling | saint_: the solution is to only allow g729 or ulaw, not both. |
13:33.52 | Greenlight | The other side says "Ok, lets use ulaw" |
13:34.10 | Greenlight | Yea, as igcewieling said, that's you're best bet |
13:35.10 | Greenlight | It sounds like you always want to use g729, so why not define the peer as such |
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13:41.35 | saint_ | igcewieling: i guess.. I'll buy some more licenses .. |
13:46.23 | carrar | Do it in hardware |
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13:46.50 | saint_ | carrar: is it sold by digium ? you have an url ? |
13:47.05 | carrar | well assuming you have to transcode, yeah |
13:48.33 | carrar | http://www.digium.com/en/products/telephony-cards/voice-compression |
13:49.02 | Katty | performs drive by hugging on carrar |
13:49.08 | carrar | woah |
13:49.29 | Katty | woahs. |
13:49.31 | Katty | yes? |
13:49.32 | carrar | falls over from the sudden act of kindness!! |
13:49.57 | Katty | :> |
13:50.03 | Katty | props carrar back up |
13:50.25 | Katty | carrar: my ear is better today! infection subsiding! :> |
13:50.31 | carrar | returns to his usual mannequin pose |
13:50.36 | Katty | carrar: AND the scale reports i am now down 24lbs total :> |
13:50.54 | carrar | 24 lbs!! |
13:51.02 | carrar | Where is it all going |
13:51.02 | Katty | yesh. |
13:51.10 | Katty | into energies when i'm running, hopefully. |
13:51.20 | carrar | gonna turn into a skinny toothpick! |
13:51.24 | Katty | can't run for a bit tho )= not while i have an ear infection |
13:51.51 | carrar | You need to stop putting dirty objects in your ear |
13:52.06 | carrar | hard habit to break |
13:52.34 | Greenlight | saint_: One option if you're limited on licences, is to catch the hangupcase from a failed dial and then dial again over a different peer (to the same carrier but requsting ulaw) |
13:52.53 | Greenlight | I *think* that would work.. |
13:54.34 | carrar | saint_, curious why you want to use g729? |
13:54.47 | Greenlight | Limited bandwdith surely? |
13:54.55 | carrar | but is that really the case |
13:56.08 | Greenlight | If not then he's mad to use g729 if he has available bandwidth for other codecs |
13:56.25 | carrar | Simply MAD |
13:56.34 | Greenlight | mmhmm |
13:56.51 | saint_ | carrar: limited bandwidth. |
13:56.59 | Greenlight | Phew, he's sane :) |
13:57.09 | carrar | How much BW do you have? |
13:57.19 | igcewieling | Greenlight: no, he isnt, but not because of that |
13:57.45 | Greenlight | :) |
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13:58.43 | saint_ | carrar: I have a T1, but it's for a non profit organization. they did not spend money into bqndwidth management / qos , so when they are all on the internet downloading crap, it kills the g711 calls. it's all choppy. until they stop streaming stuff. |
13:59.07 | carrar | it will also kill g729 then too |
13:59.20 | carrar | unless you have qos on that T1 on both sides |
13:59.46 | Greenlight | g729 will actually suffer more |
13:59.58 | Greenlight | It's less tolerant of jitter and loss |
14:00.30 | Greenlight | T1 isn't a lot of bandwidth to play with at all - can't you even get ADSL or something for "normal" net access ? |
14:00.33 | carrar | I have a T1 here to my house and without qos on it, it screws up my single call when I am doing stuff on the internet |
14:00.44 | Greenlight | I can well beleive it |
14:01.08 | carrar | You are better off with a two DSL's |
14:01.10 | saint_ | carrar: that's the issue here.. we have a meeting tonight. i'm will recommend that they switch to comecast. |
14:01.12 | carrar | one for voice |
14:01.13 | Greenlight | TCP was never designed to be friendly to other stuff using the pipe |
14:01.16 | carrar | one for internet |
14:01.20 | Greenlight | +1 |
14:01.23 | Greenlight | Best way to go |
14:01.31 | carrar | if you can't do qos on the T1 |
14:01.32 | saint_ | now.. do you guys have SIP providers, or regular ISDN connected to your Asterisk boxes ? |
14:01.36 | [TK]D-Fender | [09:59]GreenlightIt's less tolerant of jitter and loss <- It's actually more tolerant. As is iLBC |
14:01.48 | [TK]D-Fender | saint_: Yes. |
14:01.52 | saint_ | they are using a sip provider here, but it does not support SRTP / SIPTLS .. |
14:02.01 | Greenlight | [TK]D-Fender: Oh, well that's news. I |
14:02.05 | carrar | saint, All SIP |
14:02.09 | saint_ | [TK]D-Fender: does your provider support encryption ? |
14:02.18 | [TK]D-Fender | saint_: Dunno... maybe |
14:02.24 | saint_ | carrar: what about you ? do they support encryption ? |
14:02.29 | [TK]D-Fender | saint_: Does my provider matter to you? |
14:02.34 | Katty | IT MATTERS TO ME. |
14:02.37 | carrar | I don't know any provider that supports encryption in the RTP stream |
14:02.47 | saint_ | [TK]D-Fender: if your provider is better than MINE, then yes :D |
14:02.59 | carrar | You can get SIP over a private MPLS connection if you need privacy |
14:03.02 | [TK]D-Fender | saint_: Then maybe you should ask a better question. |
14:03.04 | [TK]D-Fender | ~polls |
14:03.05 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
14:03.10 | [TK]D-Fender | ^ |
14:03.18 | [TK]D-Fender | RABID. WEASELS. |
14:03.36 | [TK]D-Fender | They only snack on phone systems, what they really yearn for is the blood of phone-admins. |
14:03.49 | Greenlight | hides |
14:04.02 | saint_ | ok, so let me ask a better question.. who has a sip provider that support srtp / siptls ? |
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14:04.16 | Katty | i'm totally grinning right now. |
14:04.21 | carrar | totally |
14:04.34 | Katty | totally is still a cool word, right? |
14:04.43 | Greenlight | totally |
14:04.45 | Katty | excellent. |
14:05.37 | [TK]D-Fender | WOAH |
14:05.44 | [TK]D-Fender | </keanu> |
14:05.48 | carrar | saint_, you probably have a better chance at just setting up your own IPSEC tunnel for your audio |
14:06.18 | Greenlight | Or, if you're that worried go with ISDN |
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14:06.57 | carrar | WHAT ARE YOU TRYING TO HIDE, YOU MUST BE DOING SOMETHING ILLEGAL!! :) |
14:07.02 | Greenlight | Or colocate a server which is connected to ISDN or directly to carrier, then do your encryption from there to your box |
14:07.32 | carrar | Just talk in code |
14:07.38 | Greenlight | A little paranoia never hurt anyone |
14:07.41 | carrar | THe green monkey rides the yellow cow |
14:07.45 | Greenlight | adjusts his tinfoil hat |
14:10.28 | Katty | breaks out the tin cans and string |
14:10.45 | Katty | krrr Katty to home base, come in home base, over. krrrr |
14:10.47 | carrar | You need to encrypt that string |
14:11.05 | Katty | krrr 10-4 krrrr |
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14:14.55 | saint_ | I guess you guys are not working around NYC ..? |
14:15.31 | [TK]D-Fender | Correct. That is indeed a guess. |
14:15.45 | [TK]D-Fender | You may now claim a plushie from the bottom rack |
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14:50.16 | igcewieling | saint_: neither are you as far as I can tell. |
14:50.32 | igcewieling | aren't you the one with the Wireless ISP? |
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14:52.33 | SpenglerMBP | can anyone make a recommendation for a good small business router that works well with asterisk? |
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14:54.46 | [TK]D-Fender | SpenglerMBP: http://routerboard.com/RB2011UAS-2HnD-IN |
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15:26.48 | Freeaqingme | Does asterisk have any functionality to 'reformat' a phone number to international notation? So that 0881234567 is changed into +33881234567 ? (I could google it as well, but have no idea how such thing is called) |
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15:29.16 | WIMPy | +33${EXTEN:1} |
15:30.31 | Freeaqingme | WIMPy, but if someone already dialed +33881234567 it'll now become +33+33881234567. I'm looking for a somewhat generic way to normalize that |
15:30.54 | WIMPy | Use patterns. |
15:31.08 | Freeaqingme | k. tnx |
15:31.11 | navaismo | or a execif/gotoif |
15:32.04 | WIMPy | Or a loopback switch if you need it realtime. |
15:32.19 | Freeaqingme | at least that sounds funky ;) |
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15:37.49 | drmessano | Anyone know how to use a blank space as a delimiter with CUT? |
15:38.15 | drmessano | Not "a", but single blank spaceS |
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15:42.17 | Greenlight | Hmm getting a really strange issue at one customer. It's like we're getting cross lines. They're making outbound calls and randomly getting crossed call of letting agencies, estate agents other random things, and they can hear the conversaton but not speak to them. I thought this sort of thing didn't really happen anymore, but I've just heard it myself on the call recordings. We're fully SIP |
15:42.18 | Greenlight | to our carrier. |
15:43.41 | Greenlight | I can't see how anything at my side could create this behaviour - is this likely to be an issue on the PSTN network itself? |
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15:57.10 | Free99 | hey, is mjordan in here? |
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16:01.27 | Free99 | So I have an issue where my call terminator changed their DNS entry to a different IP, but it seems my asterisk server was caching the old resolved IP. I kept getting a "Registration Timeout message". A simple "sip reload" fixed everything, so.. how can I prevent this from happening again? |
16:02.01 | Greenlight | What version are you running ? |
16:03.14 | Free99 | Greenlight, 1.8.10.1 |
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16:04.11 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::3) |
16:04.13 | cusco | hi folks |
16:04.28 | Greenlight | There i a conf file that allows you to specify your own TTL if I remember correctly |
16:04.37 | Greenlight | *is a |
16:04.38 | Greenlight | Umm |
16:04.41 | cusco | I'm trying to determine if current call audio glitches, come from high IO |
16:04.59 | Greenlight | dnsmgr.conf |
16:05.21 | cusco | we had this problem not long ago, got a separate disk for mysql (realtime for asterisk) and a disc for the remaining SO |
16:05.21 | WIMPy | Free99@ You need to use the dnsmgr and configure it to do refreshes. |
16:05.43 | cusco | load average is pretty high, altough cpu is low |
16:05.45 | Free99 | WIMPy, wish I'd seen this in the asterisk book :-/ |
16:06.00 | Free99 | Greenlight, WIMPy thanks |
16:06.17 | Free99 | what are managed dns lookups btw? |
16:06.50 | WIMPy | 1. a cache and 2. th possibility to refresh it. |
16:07.40 | Greenlight | cusco: Are you doing any call recording? |
16:07.50 | Free99 | Hmm.. this seems like a pretty necessary feature. Why does asterisk not have this enabled by default? |
16:08.02 | cusco | Greenlight: some, really few compared to call volume |
16:08.05 | cusco | but am... |
16:08.14 | cusco | I can state however |
16:08.22 | cusco | calls are recorded to tmpfs |
16:08.29 | Greenlight | Ok, good |
16:08.30 | cusco | and then in the end of the call, moved to the disk |
16:08.33 | Greenlight | Perfect |
16:08.40 | Greenlight | That's the best way |
16:08.46 | Free99 | well w/e, thanks anyway fellas |
16:08.46 | cusco | now that you mentioned it |
16:08.54 | cusco | I could move them to another disk.. less busy |
16:09.03 | Greenlight | Naa, don't worry |
16:09.09 | cusco | I'm not even sure I have a IO problem :/ |
16:09.19 | Greenlight | Doesn't sound like an IO issue |
16:09.24 | cusco | but load average is comming high |
16:09.26 | cusco | anc cpu is low |
16:09.33 | Greenlight | Yea, asterisk can be deceptive |
16:09.40 | Greenlight | When looking at load average |
16:09.50 | Greenlight | Lots of short lived threads |
16:09.52 | cusco | and load average comes high .. at random peaks |
16:10.02 | cusco | hmm ... |
16:10.06 | Greenlight | I've a system that peads l/a at 50.00 |
16:10.16 | cusco | o.O |
16:10.16 | Greenlight | CPU usage shows asterisk at 300% max |
16:10.18 | Greenlight | 8 cores |
16:10.30 | talntid | how many users? |
16:10.31 | cusco | right |
16:10.37 | Greenlight | 600~ channels |
16:10.48 | Greenlight | No call quality issues |
16:10.51 | talntid | hmm, i suppose that's more than mine |
16:10.55 | cusco | 60 active channels |
16:10.55 | cusco | 30 active calls |
16:11.06 | cusco | really little here |
16:11.11 | Greenlight | WE're even doing COnfBrdige and other timing sensitive stuff |
16:11.16 | cusco | there is plenty of BW (net) available |
16:11.27 | talntid | ~75 active calls here, on a dual core, 2gb ram virtual machine |
16:11.28 | Greenlight | So, I wouldnm't immediately see high l/a as an issue |
16:11.40 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
16:12.09 | Greenlight | Are you using FreePBX ? |
16:12.13 | Greenlight | Or lots of queues |
16:12.41 | Free99 | actually just discovered that Linode had my resolv.conf setup to point to DNS servers with the wrong entry for my terminator |
16:12.57 | Greenlight | Free99: "wrong" entry --- how is that? |
16:13.17 | cusco | me? |
16:13.20 | Greenlight | Free99: Did they jump serverrs ahead of their TTL's |
16:13.24 | cusco | asterisk, lots of queues |
16:13.24 | Greenlight | cusco: Yes |
16:13.30 | cusco | lots |
16:13.32 | cusco | relative |
16:14.03 | cusco | root@ami /mnt/root# asterisk -vrx "queue show"|grep strategy|wc -l |
16:14.05 | cusco | 61 |
16:14.06 | Greenlight | We used to have a very odd issue on boxes with both FreePBX and a great many queues in heavy use |
16:14.24 | Greenlight | Basically had to reboot the box nightly or else we'd get call quality issues |
16:14.25 | cusco | its plain asterisk, no freepbx |
16:14.31 | cusco | o.O |
16:14.36 | Greenlight | Very very very odd. |
16:14.55 | cusco | also... |
16:14.56 | *** join/#asterisk ylamhang (~chatzilla@UNVLON55-1176057127.sdsl.bell.ca) |
16:14.58 | cusco | well.. |
16:14.59 | cusco | dunno |
16:15.10 | cusco | need to get more info on this call quality issue |
16:15.14 | Greenlight | Moved away from both lots of asterisk Queues and from FreePBX. So we're raw asterisk and manager our own queues, and all is fine no need to reboot to avoid call quality issues |
16:15.22 | Greenlight | When was your box last rebooted? |
16:15.26 | Free99 | Greenlight, I changed my resolv.conf so the first entry is (bleh) google's 8.8.8.8 and reloaded... where before I was getting some weird IP that had no corresponding reverse lookup, it suddenly started working after restarting resolveconf |
16:16.03 | Greenlight | Free99: Sounds like you have some DNS issues |
16:16.17 | Greenlight | Free99: Also, I'd always recommand using a local DNS server |
16:16.19 | talntid | why bleh google's dns? |
16:16.27 | Greenlight | DNS goes down, so does asterisk |
16:16.38 | Greenlight | It's all very nasty |
16:16.40 | talntid | i know local dns is better/faster, but google's dns is very good regardless |
16:17.01 | Greenlight | Google's DNS may be good, but relies in his internet connectionk |
16:17.06 | drmessano | I use DNSMASQ as a local resolver, and it uses Google DNS. |
16:17.10 | Greenlight | His connection dies, and wham |
16:17.21 | drmessano | I basically set it up as a caching proxy |
16:17.25 | Free99 | talntid, I just don't like putting all my eggs in one basket |
16:17.32 | Greenlight | drmessano that's exactly what you want |
16:17.39 | drmessano | I know it is |
16:17.40 | drmessano | lol |
16:17.46 | Greenlight | :) |
16:17.52 | Free99 | its weird b/c linode's always been good |
16:17.54 | talntid | I see. I thought you were inferring it was a bad service :) |
16:18.13 | drmessano | I've done that since nearly day 1. DNS + Asterisk = recipe for disaster |
16:18.19 | Free99 | talntid, nah. I just also don't trust google very much. |
16:18.36 | Greenlight | drmessano: mmmhmm learned from bitter experiance here too! |
16:18.42 | talntid | k :) |
16:18.45 | drmessano | Trust is an interesting thing |
16:19.02 | talntid | I don't trust you, but I'll use you for 80% of my internet needs.. :P |
16:19.40 | drmessano | lol |
16:19.42 | talntid | trust IS an interesting thing. |
16:20.29 | drmessano | I was remarking to someone earlier about how my "better judgement" on some of these larger, more popular entities and the necessity of moving away from them because they're all evil has only provided me with a poor experience all around |
16:21.04 | talntid | i love when people tell me they don't trust google - google is too powerful, but they use google as a search engine because "it's better", they use gmail as their email because "it's better" =D |
16:21.05 | drmessano | Seems like evil is a requirement of reliability |
16:21.19 | Free99 | talntid, I personally don't. I have an android phone which I've rooted and configured iptables to block google services. |
16:21.25 | Free99 | so (shrug) |
16:21.36 | talntid | the fact you use an android phone is funny. google product. |
16:21.50 | drmessano | Like when I was hot shit and moved some of my domains from my Godaddy default DNS to Cloudflare because they are/were so awesome and not evil |
16:21.59 | drmessano | Except when they break, horribly |
16:22.02 | Free99 | talntid, the rom is open source and I've looked at it |
16:22.20 | Free99 | I can't make an assurance about the hardware tho |
16:22.23 | talntid | go md5 the firmware on the phones ;) |
16:22.25 | talntid | right. |
16:22.34 | talntid | the acual silicon chips. |
16:22.45 | drmessano | is learning to leave shit well enough alone |
16:22.46 | Free99 | talntid, I have replacement firmware too lol |
16:23.41 | Free99 | talntid, I also don't use google for search, I use duck-duck-go so :P |
16:23.57 | drmessano | I always Ask Jeeves |
16:24.03 | drmessano | That guy, always right |
16:24.12 | Free99 | drmessano, screw it, altavista |
16:24.32 | talntid | ;) |
16:24.49 | Free99 | you remember their translate function? it was so cool! |
16:24.56 | Free99 | *ahem* anyway |
16:25.10 | drmessano | Ask.com MUST be correct and awesome because Oracle bundles their toolbar with Java, and I trust BOTH of them |
16:25.20 | Free99 | hahahaha |
16:25.46 | cusco | am still looking at iostat |
16:25.53 | cusco | Greenlight: iostat -dx 5 -p |
16:25.57 | cusco | what do you get ? :p |
16:26.02 | cusco | after 3 times |
16:26.02 | Free99 | hey drmessano if I wanted to emulate your dnsmasq setup, are there any good tutorials you followed? |
16:26.12 | *** join/#asterisk suporte85 (~guardadig@187.56.25.195) |
16:26.43 | drmessano | Sorta.. I have a Asterisk/FreePBX install guide that I think was last updated for Ubuntu 12.10. The DNSMASQ bits are at the beginning. Hang on |
16:27.17 | drmessano | http://www.2l2o.com/how-to/asterisk-without-tricks |
16:28.18 | drmessano | It's a handful of lines dumped into an additional conf in /etc/dnsmasq.d/ .. so we're not overwriting anything existing |
16:32.24 | thehar | ohai |
16:37.09 | Free99 | so what's weird is, I've set my resolv.conf back to being a symlink but after restarting the service it is empty |
16:37.12 | Free99 | wtf? |
16:37.19 | Free99 | wrong room for this so w/e |
16:41.28 | Greenlight | cusco: http://pastebin.com/34NmKjgd |
16:43.15 | Greenlight | I don't think your issue is IO related tbh |
16:43.54 | Greenlight | Are you on server hardware ? |
16:44.28 | Free99 | drmessano, is it a good idea to run asterisk with rtprio 10? |
16:44.43 | joesmoe_ | okay back to the drawing board |
16:44.44 | joesmoe_ | time to redo pbx |
16:44.55 | Greenlight | Free99: It's always advisable to priortise it |
16:45.01 | joesmoe_ | this time going with the freepbx distro instead of the asterisk now |
16:45.22 | Greenlight | joesmoe_: Compile from source if you can |
16:46.11 | Free99 | Greenlight, but by how much? I never liked the possibility of locking myself out due to process jamups |
16:46.41 | Greenlight | The canary prevents that doesn't it |
16:46.57 | Greenlight | At least that was my understanding |
16:47.13 | Free99 | ah, that's also what its for? I thought it was for if the service just died mysteriously it'd get restarted |
16:47.31 | Free99 | http://asteriskfaqs.org/2010/11/24/asterisk-tips/astcanary.html |
16:47.35 | Free99 | you're right Greenlight |
16:47.57 | Greenlight | So that should keep you safe |
16:49.02 | Greenlight | Just be sure to feed it every now and again |
16:49.18 | Greenlight | Mine likes dried nuts and bread |
16:49.50 | Free99 | Greenlight, I feed mine Lafaber pellets :P I actually have two small parrots, so.. |
16:49.55 | Free99 | lol |
16:50.07 | Greenlight | :) |
16:50.25 | Free99 | um after updating rtprio, how do I enforce this config change without a reboot? |
16:50.41 | Greenlight | I think the process would need restarted |
16:51.31 | Free99 | so I go into htop and the pri is -11? |
16:51.46 | Greenlight | That's what mine shows yes |
16:52.01 | Free99 | ok |
16:52.06 | Free99 | cool, that was easy :) |
16:52.40 | Greenlight | And, if you do anything in your dialplan that calls external apps on the server, I always "nice" them |
16:53.06 | Free99 | oh crud, I have an AGI written in PHP |
16:53.17 | Free99 | is that going to get swamped? |
16:53.27 | Greenlight | I wouldn't be overly worried |
16:56.20 | Free99 | thanks Greenlight |
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17:00.29 | ylamhang | hi |
17:00.39 | ylamhang | is there a way to make asterisk sent rtcp report? |
17:01.09 | ylamhang | can anyone point to some documentation or examples? Thanks |
17:01.37 | Greenlight | https://wiki.asterisk.org/wiki/display/TOP/RTCP |
17:02.14 | ylamhang | thank you Greenlight |
17:02.24 | ylamhang | I will take a look at it |
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17:09.03 | cusco | Greenlight: your io writes are above mine, but await are below |
17:09.04 | cusco | o.O |
17:10.19 | Greenlight | Hardware RAID controller with BBU and cache |
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17:12.14 | cusco | ah cool |
17:12.15 | cusco | ok ok |
17:12.19 | cusco | adaptec ? :) |
17:12.27 | Greenlight | Dell |
17:12.40 | Greenlight | H700 1gb if memory serves |
17:13.20 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-lxeskdelwomirlig) |
17:14.15 | Greenlight | If your concerned about disk IO then leave your calls in memory (just while testing) and disble all logging |
17:14.28 | Greenlight | That should almost eliminate all disk IO and prove one way or another |
17:14.58 | Greenlight | Just be sure to re-enable your tmpfs -> disk script once you're done testing :) |
17:16.38 | Greenlight | Right - leaving office now - laters! |
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17:43.00 | Freeaqingme | Most of the Dutch telco's do an anwer() before they forward you to voicemail. Is there any way to ignore that forward and proceed as if the call was unanswered? |
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17:48.37 | [TK]D-Fender | Freeaqingme: "core show application amd" |
17:48.38 | *** join/#asterisk kresp0 (~kresp0@35.Red-83-39-25.dynamicIP.rima-tde.net) |
17:48.57 | Freeaqingme | [TK]D-Fender, I thought that one wasn't ready for production yet? |
17:49.12 | Freeaqingme | or 'yet', at least not reliable yet? |
17:50.03 | [TK]D-Fender | Nothing is perfect here |
17:50.10 | Freeaqingme | heh |
17:50.12 | [TK]D-Fender | It's OK, and it's all there is. |
17:50.15 | Freeaqingme | I blame nobody for that ;) |
17:50.19 | Freeaqingme | cool. tnx |
17:55.41 | ani216 | Hello |
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17:57.39 | ani216 | [TK]D-Fender: are you still around? |
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17:58.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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17:59.43 | jmetro | dat idle |
17:59.46 | [TK]D-Fender | ani216: yes |
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18:00.51 | ani216 | ok so for you to see the log of a 2 way call i have to set debug on both peer 1 and peer 2 correct and just log all that |
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18:17.32 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
18:17.46 | *** join/#asterisk grongi (~gringo@unaffiliated/gringo) |
18:19.32 | *** join/#asterisk infobot (~infobot@rikers.org) |
18:19.32 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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18:23.23 | PhenZen | its driving me crazy :( everything looks like it is set right too |
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18:30.32 | navaismo | PhenZen, and what is the issue? |
18:30.45 | PhenZen | when i call the other video phone i only get audio, no video |
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18:31.38 | igcewieling | are you sure the other phone support h264? |
18:31.44 | PhenZen | yes |
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18:32.46 | navaismo | h264? |
18:32.48 | navaismo | --->combined - (gsm|ulaw|alaw|h263) |
19:09.43 | *** join/#asterisk infobot (~infobot@rikers.org) |
19:09.43 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
19:10.26 | SpenglerMBP | is anyone in here running multiple statics IPs out of their house? |
19:10.46 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
19:12.37 | WIMPy | SpenglerMBP: Asterisk doesn't care. If your routing gets it right, fine, if not, you're screwed. |
19:13.06 | WIMPy | Or in short: Multiple default routes=bad. |
19:13.12 | PhenZen | navaismo: disabled all firewalls still a nogo :( |
19:14.30 | TriJetScud | disconnect |
19:14.32 | TriJetScud | oops |
19:14.40 | drmessano | I'm trying to take a var that |
19:14.47 | drmessano | GAH ENTER KEY FAIL |
19:15.08 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
19:15.10 | WIMPy | And the shift key as well. |
19:16.38 | *** join/#asterisk TriJetScud (znc@van-app-svr.ad.v10networks.ca) |
19:17.25 | drmessano | I'm trying to take a var that has mutiple words separated by spaces, parse the last word, and set it as a new var. I think using CUT along with FIELDQTY (to give me the number of the last field to feed to cut) will do the trick.. I'm just lost on how to address the blank spaces as the delimiter |
19:17.52 | drmessano | Any thoughts? |
19:18.32 | PaybackTonyB | Can you use an AGI to do it? |
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19:19.17 | WIMPy | Bad for performance, but certainly easier. |
19:19.51 | PhenZen | damnit, firewalls are ALL down and still no video :( |
19:19.57 | navaismo | PhenZen, its weird i see the marks in the cli output hmmm |
19:20.00 | WIMPy | Unless you have an AGI running anyway. |
19:20.11 | drmessano | So I am not crazy that the blank spaces are going to be an issue? Because it's basically one line of dialplan, which works except for no way to address the blank spaces. |
19:21.06 | drmessano | Gah |
19:21.14 | navaismo | PhenZen, not sure if a tcpdump can show you if the phones are passing video frames but you may try it just to be sure |
19:21.23 | PhenZen | ok |
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19:23.15 | PaybackTonyB | We use PHP AGI's and are able to get fairly good performance (11.4) |
19:23.22 | PaybackTonyB | well, compared to our expectations at least |
19:23.33 | igcewieling | drmessano: paste your function needing to cut on space which don't work |
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19:23.55 | WIMPy | Well, startig an AGI is always a fork and that's really bad. |
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19:24.22 | jmetro | i just execute php scripts using shell |
19:24.24 | igcewieling | WIMPy unless it is a FastAGI |
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19:24.35 | PaybackTonyB | Our hope was that (with our very custom application) we could get 1k concurrent calls per box |
19:24.40 | jmetro | i see calls come in and type out the script that should run for that call in realtime |
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19:24.56 | PaybackTonyB | Which we can |
19:26.16 | WIMPy | Yes, I'm sure that's the reason 1. FastAGI exists and 2. has that name. |
19:26.26 | drmessano | igcewieling, Set(command=${CUT(utterance,,${FIELDQTY(utterance," ")})}) |
19:26.35 | drmessano | The " " was my last attempt |
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19:37.42 | PhenZen | navaismo: could it being set to udp only affect it? |
19:38.03 | igcewieling | drmessano: try Set(command=${CUT(utterance,,${FIELDQTY(utterance, )})}) |
19:39.13 | navaismo | PhenZen, dont think so, did you the frames in the tcpdump? |
19:39.32 | igcewieling | you people and your attachment to " |
19:39.34 | PhenZen | not yet |
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19:40.28 | navaismo | igcewieling, we love the " |
19:40.53 | igcewieling | navaismo: the " is like fire, it must be used carefully |
19:40.55 | drmessano | igcewieling, I tried the , ) first. It seemed sensible and more likely to work. The ' ', " ", and some other silly things like (utterance,\ ) (ha, yep, escaping a space) were guesses |
19:42.01 | igcewieling | drmessano: nested functions can be hell to troubleshoot, break it into two statements just to make things easier |
19:42.06 | drmessano | More to the heart of it, I have a NoOp(${FIELDQTY(utterance,<SOMETHING>)}) in there I have been beating up by itself to find something thats at least valid for FIELDQTY, and nothing seems to work |
19:42.17 | drmessano | Yep ^ |
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19:46.19 | igcewieling | drmessano: cowboy up and look at the code 8-) |
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20:07.02 | PhenZen | navaismo: i ran tcpdump and i see G711 traffic and also h234 (not decoded yet) |
20:07.09 | PhenZen | oops h264 |
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20:39.22 | ani216 | navaismo: i ran the tcpdump and see all the traffic for G.711 but only a few for 264 |
20:40.15 | ani216 | im just going to download asterisknow and install that and try |
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20:52.48 | navaismo | what about plain asterisk? |
20:55.57 | jmetro | navaismo: +1 |
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20:57.07 | igcewieling | because people seem to prefer to spend a week learning a complicated GUI instead of a week learning config files? |
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20:58.42 | WIMPy | Someone once said "The only legitimate use for using the keyboard is wordprocessing". |
20:59.44 | jmetro | steve jobs |
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21:01.19 | WIMPy | Then someone once thought that more tan one button on a mouse would bee too complicated to use. |
21:01.57 | jmetro | ^ again |
21:02.36 | WIMPy | And someone thought that telephony is only about being able to talk to people in another location. |
21:04.47 | mountainm2k | If I call 1<mycell> I do not get caller-ID through, but <mycell> its coming through. I have seperate contexts for local and LD, but they are exactly the same... What else could I be missing? |
21:05.25 | mountainm2k | Carrier confirms that they do NOT see the data when I call it as a LD number |
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21:06.23 | [TK]D-Fender | go look at the calls |
21:06.26 | drmessano | Lets not forget "Why do you need an Epyx FastLoad Cartridge?? Your games already load REALLY FAST" - My Dad |
21:07.04 | WIMPy | You don't need CallerID to be able to talk. |
21:08.13 | mountainm2k | [TK]D-Fender: Let me see if I can do a DumpChan() right before the Dial() on the outbound call... |
21:08.44 | [TK]D-Fender | mountainm2k: No, look at th4e ACTUAL CALL. |
21:08.54 | [TK]D-Fender | LIke what you pass as you dial |
21:09.05 | mountainm2k | [TK]D-Fender: Not sure where to do that then... |
21:09.20 | [TK]D-Fender | Well what are you dialing? |
21:09.30 | WIMPy | 'core set verbose 3' and compare the two calls. |
21:10.08 | [TK]D-Fender | not verbose.. |
21:10.14 | [TK]D-Fender | prove what the CAHNNEL is given |
21:10.35 | mountainm2k | Not verbose, and I'm on really old ABE, so it isn't "core set" anyway :-P |
21:10.54 | mountainm2k | I really need to fix that, but at this point it'll behuge task to get it updated |
21:11.36 | WIMPy | remembers the 1.4 days when gettig callerID out through dahdi was random. |
21:11.45 | mountainm2k | How can I see what the channel, in this case a ZAP pri span |
21:12.07 | WIMPy | What version? |
21:12.34 | mountainm2k | WIMPy: you're going to scream at me |
21:12.36 | mountainm2k | Asterisk B.2.2.1:75261 built by root on 2007-07-16 00:36:40 UTC build |
21:12.52 | mountainm2k | I told you it was old |
21:13.16 | WIMPy | I can see it's old, but I can't see what version that's from. |
21:13.48 | igcewieling | mountainm2k: I doubt we can help with ABE |
21:13.59 | WIMPy | But I just said that callerID on PRI was random on 1.4. I guess that might be what you're seeing. |
21:14.03 | mountainm2k | I actually have a whole new box, with new cards, and everything -- just need to get everything built on it and port over all my existing configs, including fax server, etc... It'll be a huge pia. |
21:14.27 | mountainm2k | Yeah, I wouldn't swear to it, but I think this is probably 1.2-ish... |
21:14.32 | WIMPy | But it shouldn't be related to what you dial. |
21:15.02 | igcewieling | putting a Wait(.25) or Wait(.5) is usually sufficient to get your inbound CallerID Name on PRI |
21:15.09 | WIMPy | What I found way back then was that it depended on the last incomming call on the channel being used. |
21:15.28 | WIMPy | Huh? |
21:15.32 | mountainm2k | igcewieling: What's weird is, as I said, *local* calls are getting the Caller-ID |
21:15.45 | WIMPy | (we're on SENDING CallerID, BTW) |
21:17.05 | igcewieling | ah. We never ever had problems with outgoing calls and CallerID Number. |
21:17.37 | igcewieling | mountainm2k: define "local". "local" as in "on the same telco" or "local as in different telco, same city" ? |
21:18.28 | WIMPy | Are you sure it's related to how you dial? Do you use different groups? |
21:20.21 | mountainm2k | I have only one carrier, a PRI, so in a sense, its all the same |
21:20.44 | mountainm2k | the contexts were written such that certain employee phones could be limited to only local, or only LD, or international |
21:20.51 | mountainm2k | but in practice, they basically look the same |
21:21.16 | WIMPy | Do they look basically the same or are they exactely the same? |
21:22.08 | mountainm2k | http://pastebin.ca/2389293 |
21:22.31 | mountainm2k | hah -- I muted out the company name, but not the numbers... <smack> |
21:22.38 | mountainm2k | oh well, now you could figure us out, lol |
21:22.53 | igcewieling | NEVER EVER DO THAT SetCallerID("foo" <(303) 345-${CALLERIDNUM}>),a) |
21:23.01 | mountainm2k | note that 303 and 720 are overlayed |
21:23.08 | mountainm2k | OK, whatchew want me to do instead |
21:23.09 | igcewieling | CallerID number has no dashes, no spaces, no parands |
21:23.24 | mountainm2k | note that it works that way for the [outbound-local] context |
21:23.38 | igcewieling | no wonder your ld carrier tosses out the CID |
21:23.53 | mountainm2k | They're not tossing it if I call from [outbound-local] |
21:24.02 | mountainm2k | actually they say there' not getting it at all |
21:24.16 | mountainm2k | from outbound-ld but they ARE getting it from outbound-local |
21:24.27 | WIMPy | I would also expect them to filter that ID. |
21:24.31 | mountainm2k | even if its the same number -- ie, 1303xxxxxxx or 303xxxxxxx |
21:24.43 | Qwell | mountainm2k: You say that as if IRC didn't give you away... |
21:25.10 | *** join/#asterisk blizzow1 (~jburns@67.50.165.58) |
21:25.38 | igcewieling | um 1303xxxxxxx or 303xxxxxxx are not the same number, the second one is valid callerid and the first one is not. |
21:25.38 | mountainm2k | well, they don't -- If I call out of [outbound-local] my DID shows up |
21:25.38 | mountainm2k | its awsome |
21:25.38 | mountainm2k | that's the number I'm *calling* |
21:25.40 | WIMPy | Back to verbose and comparing two calls. |
21:25.53 | [TK]D-Fender | mountainm2k: No extra cars, EVER |
21:25.57 | [TK]D-Fender | chars* |
21:26.21 | WIMPy | And if there's nothing obvious, add pri intense debug. |
21:26.46 | igcewieling | mountainm2k: generally people here want you to fix the broken parts before trying to diagnose the issue because often, once you fix the broken parts the issue no longer happens. |
21:26.57 | mountainm2k | TK, ok, what should it look like in this case? SetCallerID("foo" <303345${CALLERIDNUM}>),a) ??? |
21:27.13 | WIMPy | yes |
21:27.15 | igcewieling | mountainm2k: other than the extra )? |
21:27.27 | mountainm2k | crap... ok, one sec |
21:27.46 | igcewieling | BTW, you should not have quotes there either |
21:28.02 | igcewieling | but that does matter since name is removed before the local telco hands off the call |
21:28.04 | WIMPy | And what's your "dialplan" configuration in zaptel.conf? |
21:28.06 | mountainm2k | igcewieling: Generally I'm on here diagnosing my broken parts because I don't know what parts are broken |
21:28.10 | mountainm2k | lol |
21:28.54 | WIMPy | hopes the name doesn't even get sent. |
21:28.56 | mountainm2k | Level3 is taking the name |
21:29.24 | mountainm2k | its up to the receiving end as to if they accept it or look it up in the ... NAM database |
21:29.31 | mountainm2k | (might have the wrong name there) |
21:29.34 | igcewieling | mountainm2k: I suppose the receiving telco throws it out, just don't expect most telcos to pass the name |
21:30.23 | mountainm2k | that's exactly what Level3 rep told me a couple hours ago. I'm passing it on anyway, but I realize most carriers are getting it from the database, not from me, and that's fine. |
21:30.52 | WIMPy | There's no standard way to transfer names anyway. |
21:31.22 | mountainm2k | Level3 told me they accept it, and hand it off to the next carrier over ss7 |
21:31.34 | igcewieling | we get a lot of weird callerid with inbound calls on our level 3 service. |
21:31.37 | mountainm2k | but in any case -- I can leave that out if its causing the problem, butr i don't think it is |
21:31.49 | WIMPy | Asterisk seems to support sending it as DISPLAY which is in violation of Q.931 but seems to work for some telcos. The safer way is to use the calling party subaddress. |
21:32.27 | mountainm2k | For incoming, i want to take what Level3 sends me, which is usally not much, but if they don't send anything I want to display the ANI instead |
21:32.32 | mountainm2k | and since I'm on PRI, I should get that |
21:32.36 | mountainm2k | but again -- different problem |
21:33.09 | mountainm2k | So per http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerID |
21:33.25 | mountainm2k | I should have the format: SetCallerID("foo" <number>|a) |
21:33.36 | mountainm2k | you guys are saying to remove the quotes and the <> |
21:33.41 | mountainm2k | ? |
21:33.55 | *** join/#asterisk alagar (~helpdesk@vsusg1.vernalissystems.com) |
21:33.58 | WIMPy | No, not the <>/ |
21:34.03 | WIMPy | . |
21:38.30 | *** join/#asterisk netman (~netman@178.121.20.95.dynamic.jazztel.es) |
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21:44.46 | mountainm2k | Sorry, got a unrelated phone call and now back... http://pastebin.ca/2389306 |
21:44.50 | mountainm2k | no quotes, no dashes |
21:45.09 | mountainm2k | now it doesn't send outbound for local OR ld |
21:45.58 | WIMPy | We still haven't seen a call. |
21:46.23 | WIMPy | And the question about "dialplan" parameters is also still unanswered. |
21:46.32 | mountainm2k | OK, dialplan -- checking... |
21:46.39 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
21:47.06 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
21:47.24 | mountainm2k | the word "dialplan" does not exist in my zaptel.conf |
21:48.05 | saliak | I'm having issues with SendFAX. The error i get is "Cannot open source TIFF file". Do you think that implies a permission error, or a "i don't understand this file format" error? i set the file and directory as 777 prior to dropping my .call file in the spool directory |
21:48.43 | mountainm2k | saliak: Is selinux enabled? run: sestatus |
21:49.07 | igcewieling | pridialplan=unknown |
21:49.26 | WIMPy | And prilocaldialplan=unknown. |
21:49.38 | mountainm2k | want me to add those to zaptel.conf? |
21:49.40 | saliak | mountainm2k: installing it now.. |
21:49.47 | WIMPy | But I think it defaulted to national, which should be ok in this case. |
21:50.26 | saliak | mountainm2k: it's disabled |
21:51.07 | mountainm2k | saliak: Next sanity-check items would be, is the disk full, are you out of inodes... df -h and df -i |
21:51.27 | saliak | mountainm2k: nah, all good on that front |
21:51.28 | mountainm2k | Can the user asterisk runs as write the file, if you just su to that user and touch or edit the file |
21:52.42 | mountainm2k | WIMPy: wondering if I have SetCIDName and SetCIDNum instead... When did those show up... |
21:53.08 | WIMPy | I only know they have long gone. |
21:54.03 | saliak | mountainm2k: yeah, i can sudo -u asterisk and touch the tiff file |
21:54.19 | mountainm2k | So I see, now it wants set(callerid(num=nnn)) |
21:54.46 | WIMPy | CALLERID(num)= |
21:59.34 | mountainm2k | saliak: Not sure what else -- the obvious stuff isn't it... |
22:00.06 | saliak | mountainm2k: do you think that it's actually unable to access the file? or something about the file format? |
22:00.12 | WIMPy | Look at the calls. |
22:01.15 | mountainm2k | saliak: Is it giving you an error indicating the file is corrupt or unreadable? Sounded like you were creating a new file, and that was failing, which is why I was thinking permissions, selinux, etc... |
22:01.34 | mountainm2k | WIMPy: I can make test-calls all day, what do I need to look at? |
22:01.46 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
22:01.58 | WIMPy | Let me scroll up... |
22:01.59 | saliak | mountainm2k: it just says "cannot be opened". one could interpret that a few ways, eh? |
22:02.34 | WIMPy | [20:48] puzzled has joined #asterisk |
22:02.36 | WIMPy | (~patrick@2001:980:5e31:1:d8c9:6e71:1037:bfa3) |
22:02.50 | WIMPy | Oops. Not the whole file :-( |
22:02.54 | WIMPy | <WIMPy> Back to verbose and comparing two calls. <WIMPy> And if there's nothing obvious, add pri intense debug. |
22:03.00 | WIMPy | better |
22:04.46 | mountainm2k | WIMPy: OK, nothing obvious... http://pastebin.ca/2389334 |
22:05.16 | mountainm2k | the SPOTXCHANGE|INC is throwing me -- I might have to get rid of that coma |
22:05.33 | WIMPy | That's what just cought my eye. |
22:05.49 | ani216 | [TK]D-Fender: i got my video working :) |
22:05.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.197) |
22:06.02 | ani216 | navaismo: got my video working :) |
22:07.01 | mountainm2k | WIMPy: It would seem that might have been the whole problem |
22:07.14 | mountainm2k | now, 9,13038856111 and 9,3038856111 are both working |
22:07.41 | WIMPy | Quite possible. The dialplan parser is not very clever. |
22:08.01 | mountainm2k | yeah, is it any better now than it used to be in 1.2 / 1.4 ? |
22:08.03 | WIMPy | Now if you had provided the verbose output whe I asked an hour ago..... |
22:08.05 | navaismo | what was the issue ani216 ? |
22:08.15 | WIMPy | Not much, no. |
22:08.18 | ani216 | updated to asterisk 11 and it works flawless |
22:08.43 | mountainm2k | WIMPy: Hah, well, I normally work at verbose 13... I had gone over that several times, and as you said, nothing grabbed me |
22:09.38 | mountainm2k | Not sure what if anything I can do about incoming -- I always asume asterisk will display it if it comes in from level3 |
22:09.50 | mountainm2k | and that, as others have said, I have found to be less than reliable |
22:10.07 | WIMPy | What exactely? |
22:10.36 | igcewieling | mountainm2k: for incoming add a Wait(.5) |
22:10.56 | mountainm2k | Calls come in and display on the SIP handset as "unknown", but the name will show up in voicemail |
22:10.59 | mountainm2k | just as an example |
22:11.07 | igcewieling | mountainm2k: sometimes the telco sends the CID name as a facility message after the call setup, the wait waits for that mesage to arrive |
22:11.07 | mountainm2k | that happens from time to time |
22:11.17 | [TK]D-Fender | Shouldn't have to wait.. this is dialing OUT |
22:11.30 | igcewieling | [TK]D-Fender he switched directions on us |
22:11.32 | mountainm2k | tk -- the dialing OUT problem appears to be fixed |
22:11.38 | mountainm2k | hahahah I did indeed |
22:11.42 | [TK]D-Fender | mountainm2k: enable PRI DEBUG |
22:11.49 | WIMPy | The number should always work. |
22:11.55 | [TK]D-Fender | mountainm2k: Which you should have done from the start |
22:12.13 | igcewieling | mountainm2k: make sure to have (I think) facility=yes or facilityenable=yes (check the sammple config file) |
22:12.17 | mountainm2k | Holy crap, that's a lot of data |
22:12.35 | *** join/#asterisk dorphalsig (~dorphalsi@181.50.255.162) |
22:13.19 | mountainm2k | pri debug span 1 -- might tell me if/why I'm not able to transfer off-net, too... I keep meaning to look into that... |
22:14.51 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
22:15.14 | WIMPy | Might be a better idea to concentrate on upgrading. |
22:15.33 | dorphalsig | Hi! I'm trying to set up an IP Trunk through NAT (port forwarding), but the call is acting up. It rings ok but when I answer the phone the called party hears ringing and the caller hears some strange echo |
22:15.38 | mountainm2k | I don't disagree |
22:15.58 | dorphalsig | when looking @ the console I'm getting a SIP retransmit error |
22:16.08 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
22:16.37 | WIMPy | dorphalsig: No IP, no VOIP. That's the way it is. |
22:17.08 | dorphalsig | WIMPy: a SIP trunk, sorry ate the S |
22:17.31 | WIMPy | Yes, you still need IP to the peer to be able to use it. |
22:18.08 | carrar | T1 to channel bank |
22:18.16 | carrar | plug in analog phones |
22:18.53 | WIMPy | Someone should make (working) IAX phones again. |
22:19.11 | carrar | Apple talk phones |
22:19.17 | carrar | heh |
22:19.38 | mountainm2k | +1 for IAX phones |
22:19.53 | mountainm2k | Is there even still a IAX ATA in production? |
22:20.05 | WIMPy | Don't think so. |
22:20.29 | WIMPy | Has there ever been more tan the IAXy? |
22:21.03 | WIMPy | IAX is so much better than the SIP shit. |
22:21.20 | mountainm2k | +100 |
22:21.30 | dorphalsig | So anyway, I turned debug on and I see the Invite answered by a Trying then Two Ringing and then an OK |
22:21.32 | mountainm2k | the NAT rule is one-line, and it just works, with no crap |
22:22.15 | WIMPy | And you can get things like charging information from one Asterisk to another. |
22:22.37 | mountainm2k | and PTT, if you're like me and using app_rpt for stuff |
22:22.45 | mountainm2k | (completely seperate from this ABE box) |
22:22.55 | dorphalsig | When I hang up I get: WARNING[6665]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission |
22:23.52 | dorphalsig | AFAIK that means there is something fishy with my NAT, right? |
22:23.59 | dorphalsig | However I cant quite put the finger on it |
22:24.19 | dorphalsig | Anybody cares to give me a hand? |
22:24.24 | mountainm2k | dorphalsig: Guessing the RTP packets are not crossing through the NAT... Describe your NAT device / firewall |
22:24.36 | WIMPy | dorphalsig: Yes, you are obviousely not able to communicate with that peer. |
22:24.55 | mountainm2k | the SIP setup packets are getting through, but RTP is clearly not. |
22:25.06 | dorphalsig | but the forwarding is there |
22:25.12 | mountainm2k | RTP is a PIA... Generally you want to avoid NAT'ing that if you possibly can. |
22:25.14 | WIMPy | Looks like not even SIP is getting through all the time. |
22:25.20 | mountainm2k | the forwarding is there for RTP? |
22:25.56 | dorphalsig | yeap my telco set me up to use 20000 to 20019 |
22:26.12 | mountainm2k | bare in mind those are UDP |
22:26.24 | dorphalsig | Yeap, UDP forwarded |
22:26.28 | mountainm2k | what type of firewall/NAT are you using? |
22:26.37 | dorphalsig | My firewall is an IPCOP box |
22:27.03 | WIMPy | Does that have any hind of SIP support? |
22:27.28 | dorphalsig | Dont think so. I just forwarded ports 20000 to 20019 to my * box |
22:27.53 | mountainm2k | does it use iptables? |
22:28.01 | dorphalsig | yes |
22:28.12 | mountainm2k | Can you have it load additional modules? |
22:28.22 | dorphalsig | I guess... |
22:28.23 | WIMPy | lsmod|grep sip |
22:28.32 | mountainm2k | put a -i in that sip |
22:28.36 | mountainm2k | I mean grep |
22:28.45 | mountainm2k | n/m it is lower case |
22:29.21 | mountainm2k | See if you can get it to load nf_conntrack_sip and nf_nat_sip |
22:29.27 | WIMPy | No! |
22:29.33 | mountainm2k | ? |
22:30.02 | dorphalsig | ? |
22:30.11 | WIMPy | nf_conntrack_sip is ok, using that you can avoid forwarding rtp ports, but NEVER EVER load nf_nat_sip or you end up in hell. |
22:30.33 | mountainm2k | because it will do the rewriting for you so you don't have to STUN? |
22:30.51 | WIMPy | Yes. |
22:31.17 | WIMPy | But my experiences with letting that module doing it were bad. |
22:31.31 | WIMPy | Let Asterisk do it. It's better at it. |
22:31.39 | mountainm2k | I must say I have minimal experience with NAT'ing SIP, but what little I have done, I used both modules, and it works well. |
22:32.02 | mountainm2k | But again, I use it for one extension, going from our PBX (behind the firewall) out to our video bridge (outside the firewall) |
22:32.10 | Juggie | i'd argue its pointless to load any module that interfeers with sip or rtp |
22:32.32 | mountainm2k | Juggie: Not pointless if it doesn't work otherwise :-) |
22:32.55 | WIMPy | It definitely works without. |
22:33.06 | mountainm2k | So in any case: dorphalsig: If you can get IPCOP to load nf_conntrack_sip -- do so |
22:33.14 | Juggie | with proper configuration, basic symetrical nat should take care of everything. |
22:33.49 | Juggie | caveats are: when asterisk sends internal ip to external clients (solution externip=) |
22:34.20 | WIMPy | Yes, you need to configure Asterisk appropriately. |
22:34.38 | Juggie | asterisk properly configured there is no need for any modules. |
22:35.02 | WIMPy | But without the conntrack module, you need to open the rtp ports. |
22:35.16 | Juggie | yeah yo need basic nat loaded |
22:35.24 | Juggie | but nothing to interfeer with that |
22:35.39 | dorphalsig | loading the module |
22:35.42 | Juggie | its important to understand how it works before interfeering :) |
22:35.56 | WIMPy | Yes, helps a lot. |
22:36.13 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
22:36.18 | Juggie | and it also depends on your config |
22:36.29 | WIMPy | dorphalsig: The important thing was to make sure the nf_nat_sip module is NOT loaded. |
22:36.54 | Juggie | are you 1) nated sip client, public ip for *. 2) public sip client, nated * or 3) nated sip client, nated *. |
22:37.13 | Juggie | if you plan for scenario #3, then all others will work ;) |
22:37.34 | dorphalsig | WIMPy: No SIP Modules loaded |
22:37.48 | WIMPy | Ok, that's good. |
22:37.56 | dorphalsig | insmod nf_conntrack_sip |
22:37.57 | dorphalsig | right? |
22:38.01 | Juggie | forwarding of rtp ports is also entirely unnecessairy |
22:38.48 | WIMPy | Yu can do so, but that has no effect on your SIP issue. |
22:39.05 | WIMPy | goes looking for food. |
22:39.33 | Juggie | actually.. mm its been a while.. you might need to foward rtp for a double nat scenario... |
22:39.39 | Juggie | unless the far end sip client does upnp |
22:39.43 | Juggie | ya you prolly do ;) |
22:41.36 | ani216 | does anyone know of some URI to test video? |
22:43.25 | *** join/#asterisk dorphalsig (~dorphalsi@181.50.255.162) |
22:44.57 | navaismo | uh? What do youi mean with test? |
22:45.16 | [TK]D-Fender | Double NAT = Please stand on that plastic sheet over there in the corner..... |
22:50.03 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
22:50.07 | *** mode/#asterisk [+o pabelanger] by ChanServ |
22:50.37 | dorphalsig | What options should I use for nf_conntrack_sip? |
22:51.19 | WIMPy | None, but you need to fix the SIP issue first anyway, before going to RTP. |
22:53.15 | dorphalsig | WIMPy so how do I deal with it? AFAIK the sequence is ok, no? |
22:53.38 | WIMPy | Yes, but you have retransmissions. That should not happen. |
22:54.17 | dorphalsig | WIMPy. So Where do I start looking? I have a trace of the call, if I pastebin it could you give it a look?\ |
22:56.18 | dorphalsig | http://pastebin.ca/2389814 |
22:56.26 | WIMPy | Lots of tcpdumping at various places of the communication path is what usually helps finding the bad spot. |
22:57.47 | WIMPy | Looks ok in principle, but that retransmission thing is not good at all. |
22:59.32 | dorphalsig | declares himself an absolute n00b in that |
23:00.10 | WIMPy | just feels lucky if it works. |
23:05.43 | jmetro | same => n,gotoiftime(8:00-5:00,mon-fri,*,*?Business |
23:05.44 | jmetro | <PROTECTED> |
23:06.18 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.197) |
23:08.09 | *** part/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
23:08.28 | jmetro | oh christ |
23:08.33 | jmetro | thats 8am to 5 am |
23:13.49 | dorphalsig | WIMPy: This is the tcpdump of the firewall |
23:13.50 | dorphalsig | http://pastebin.ca/2389822 |
23:16.12 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
23:21.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-uixdodrwsfxifkjv) |
23:22.39 | dorphalsig | Anybody home? |
23:23.21 | WIMPy | There's nothing to read. Look on both sides of it where something goes missing. |
23:33.31 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
23:33.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
23:35.04 | *** join/#asterisk Shariff (~chatzilla@2001:980:167b:1:407b:48f1:3dcd:ed97) |
23:35.06 | Shariff | Hi there |
23:37.15 | Shariff | I have a fritz!box router with 2 telephone (FXO?) ports.. Is it possible to somehow let Asterisk manage those ports in some way? Should the fritz!box become a client to asterisk.. but don't you get a weird setup then? Setup being: Fritz connects to internet. On Lan is the asterisk system, which is connected to a SIP provider on the internet.. So the phone goes from fritz to asterisk to fritz... |
23:37.16 | Shariff | ...to internet... ? |
23:38.44 | WIMPy | You can do it any way yo like. Register Asterisk to the FB or register the FB to Asterisk or you could even connect to it via chan_capi. |
23:39.18 | Shariff | Interesting.. do you have a recommendation which is the 'best' way to go? |
23:39.37 | WIMPy | The one you like. |
23:40.20 | WIMPy | If you use CAPI the ports on the FB will be handled from Asterisk directly. |
23:40.49 | WIMPy | When using SIP you have the routing on the FB as well. |
23:41.41 | Shariff | Does it hurt performance having 1 sip client be a host to another.. so asteriskSIP =>FB -> VOIP PRovider? |
23:42.16 | WIMPy | no |
23:42.31 | Shariff | Thanks a lot for asking these newbie questions :D |
23:42.52 | WIMPy | The "best" way to go on about it also depends on what you want to do. |
23:43.28 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
23:43.45 | Shariff | Simply have a hand full of phones be able to talk with one and other.. some of which are not IP phones.. |
23:43.58 | Shariff | Btw.. do I need any hardware in the asterisk system to work with capi? |
23:44.04 | WIMPy | The FB can do that. |
23:44.11 | WIMPy | No |
23:45.16 | Shariff | WIMPy: Yeah.. but 1 phone messes that up :) I have a cisco phone who uses * codes and FB uses ** as a prefix for an extension.. so every time I try ** on the cisco phone (SPA525G) I get an invallid number error |
23:45.24 | Shariff | That is the problem I'm basically trying to solve |
23:46.04 | WIMPy | You need to configure that on the phone. |
23:47.05 | Shariff | I can do that? |
23:47.19 | Shariff | blinks in amazement :D |
23:47.54 | WIMPy | It's called dialplan or something. |
23:49.34 | Shariff | So I should be able to use **999 if I change the dialplan, as far as you know? |
23:49.45 | WIMPy | yes |
23:50.12 | Shariff | That should make life less complicated :D |
23:50.52 | Shariff | Thanks a lot!! |
23:53.00 | dorphalsig | WIMPy: Ok, I just ran a quick comparison |
23:53.13 | dorphalsig | the number of packets is the same (would that mean no packet loss??) |
23:53.28 | dorphalsig | I'm testing my two endpoints. My firewall and my * box |
23:53.54 | WIMPy | Then maybe some packets don't go to the correct destination. |
23:55.35 | Shariff | WIMPy: You are the best!! Once you told me what to look for.. It was as easy as adding **xxx to the dialplan.. that's all it took |
23:55.40 | Shariff | Thanks a million! |
23:56.21 | WIMPy | Usually the easiest is to just leave it blank. |
23:58.26 | dorphalsig | WIMPy: I ran this basic diff, everything seems to be in place? http://www.diffchecker.com/diff |
23:58.43 | dorphalsig | sorry its http://www.diffchecker.com/muxcxgqr |