| 00:01.34 | navaismo | hmm 380 isnt a redirect/transfer |
| 00:02.08 | jeffspeff | but it tells the originating caller to try to complete the call at a different location though right? |
| 00:02.28 | jeffspeff | i don't want to transfer from 1 number to another |
| 00:03.10 | navaismo | https://lists.cs.columbia.edu/pipermail/sip-implementors/2009-March/022169.html |
| 00:04.48 | jeffspeff | sorry, long day, 305 not 380 |
| 00:06.44 | navaismo | ok i googled the definition of that response but dont know how it will help you: |
| 00:07.05 | navaismo | "305 Use Proxy. The requested resource must be accessed through the proxy given by the Contact field. The Contact field gives the URI of the proxy. The recipient is expected to repeat this single request via the proxy. 305 responses must only be generated by the user agent servers (UASs)." |
| 00:09.45 | jeffspeff | the call comes in from provider (system-A) to my main box (system-B) system-B then sends a 305 with the ip of another box (system-C). system-A then repeats the same call attempt to system-C |
| 00:12.36 | jeffspeff | or am i not understanding 305 correctly? |
| 00:15.39 | navaismo | well, dont know just googled that today, but as far i undertand in the last 5mins the 305 must work like that. IP1 sent an invite to IP2, IP2 response with 305 and the Proxy addr(IP3) at the contact header, IP1 ack IP2 then IP1 invites IP3 |
| 00:16.22 | jeffspeff | i just started looking at this today myself. |
| 00:16.33 | jeffspeff | now to find out how to do that in the dialplan |
| 00:22.52 | [TK]D-Fender | Transfer() is a 302 redirect and WILL throw the call completely off your server |
| 00:23.17 | jeffspeff | but 302 must go to a different number |
| 00:23.21 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 00:23.25 | jeffspeff | i want to use the same numbers |
| 00:23.59 | jeffspeff | [TK]D-Fender, i think 305 is more like what i'm trying to do |
| 00:24.57 | navaismo | i have no idea how handle sip response via dialplan |
| 00:25.24 | [TK]D-Fender | what "same numbers"? |
| 00:25.31 | [TK]D-Fender | You specify the taret URI |
| 00:25.57 | jeffspeff | [TK]D-Fender, are you referring to Transfer() ? |
| 00:26.02 | [TK]D-Fender | yes |
| 00:27.01 | jeffspeff | so, it's supposed to be Transfer(SIP/${IP-ADDRESS}) ? not necessarily with the variable, but the IP goes in palce |
| 00:27.04 | jeffspeff | *place |
| 00:27.48 | jeffspeff | i understood it to be more like exten=1234,1,Transfer(SIP/5678) |
| 00:28.25 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 00:28.44 | navaismo | jeffspeff, i think you need a sip proxy to handle that kind of request before reach an asterisk |
| 00:29.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:29.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:30.00 | jeffspeff | navaismo, are you referring to my example exten=1234,1,Transfer(SIP/5678) or the 305 response? |
| 00:30.14 | [TK]D-Fender | SIP/5000@1.2.3.4 |
| 00:30.21 | [TK]D-Fender | Full URI |
| 00:30.44 | jeffspeff | [TK]D-Fender, let me give that a shot real quick. thanks |
| 00:31.57 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 00:31.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:32.26 | navaismo | jeffspeff, I referring to 305 response |
| 00:46.15 | jeffspeff | [TK]D-Fender, that works like a charm. thanks |
| 00:47.02 | *** join/#asterisk serafie (~erin@24.214.158.242) |
| 00:58.54 | Katty | hello cupcakes! |
| 01:00.06 | [TK]D-Fender | jeffspeff: You're welceom |
| 01:00.09 | [TK]D-Fender | welcome* |
| 01:00.11 | [TK]D-Fender | asjkdhadgjoeuiftywqneocrtwer |
| 01:00.28 | Katty | hi dear |
| 01:09.41 | *** join/#asterisk hectorini_mx (~hector@189.177.42.156) |
| 01:11.07 | carrar | someone has cupcakes? |
| 01:11.43 | Katty | you do! |
| 01:11.57 | Katty | sneaks cupcakes over |
| 01:12.39 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:15.44 | lorsungcu | not real sneaky when youre like |
| 01:15.44 | lorsungcu | hey guys |
| 01:15.49 | lorsungcu | im over here, sneaking |
| 01:15.53 | lorsungcu | cupcakes.. |
| 01:18.20 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 01:26.57 | carrar | puts the secret cupcakes in the secret safe using the secret code of 1234 |
| 01:27.15 | lvlinux | jerks up a phone and dials 1234 |
| 01:27.43 | lvlinux | darn all circuits are busy... |
| 01:27.56 | carrar | Allison likes to say that |
| 01:28.04 | lvlinux | lol |
| 01:28.25 | carrar | need some Allison paradoy messages |
| 01:28.38 | carrar | Her doing spoofs on her own messages |
| 01:28.40 | *** join/#asterisk droemel (~droemel@p4FCAD835.dip.t-dialin.net) |
| 01:29.14 | carrar | "All circuits are busy, hahah I'm just kidding, you just can't dial the correct number of digits" |
| 01:40.48 | [TK]D-Fender | "No we're not busy, we're just IGNORING you." </DirectoryInassistance> |
| 01:42.27 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 01:45.23 | lvlinux | yeah that would be funny |
| 01:53.47 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 01:55.30 | *** join/#asterisk picard276 (~chatzilla@nat-128-84-124-0-572.cit.cornell.edu) |
| 01:55.45 | picard276 | hey guys i had a sip kind of conceptual question |
| 01:56.00 | picard276 | i havec. noticed when using sip in some public cafes (etc) ports are blocked |
| 01:56.02 | picard276 | by default |
| 01:56.16 | picard276 | aka i can't reach a stun server |
| 01:56.32 | picard276 | or i can't open an RTP connection on 10,000 |
| 01:57.07 | picard276 | is there anyway i can tunnel this over port 80 |
| 01:57.15 | picard276 | or an automatic open port.. |
| 01:57.38 | picard276 | i mean signaling is UDP so it does not have an incoming port (hence whatever port its send out over should not matter) |
| 01:58.12 | picard276 | but RTP does have an incoming port (which could potentially be blocked if the firewall filters out all incoming requests that are not on certain ports?) |
| 01:59.04 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 02:02.15 | lorsungcu | picard276: use VPN |
| 02:02.28 | picard276 | right |
| 02:02.33 | picard276 | that was going to be my next question |
| 02:02.35 | picard276 | use a vpn |
| 02:02.39 | picard276 | that will tunnel all traffic |
| 02:02.41 | picard276 | right ? |
| 02:02.47 | lorsungcu | if you tell it to |
| 02:02.56 | [TK]D-Fender | Whatever you choose to send over it |
| 02:03.20 | picard276 | but i feel lik ea VPN |
| 02:03.22 | picard276 | is really |
| 02:03.24 | picard276 | "slow" |
| 02:03.28 | picard276 | so much overhead |
| 02:03.38 | lorsungcu | so much? |
| 02:03.42 | picard276 | well |
| 02:03.44 | lorsungcu | please explain |
| 02:03.45 | picard276 | VPN's drop |
| 02:03.46 | picard276 | a lot |
| 02:03.49 | lorsungcu | really. |
| 02:03.57 | picard276 | PPTP or openVPN are not really stable enough i think for a phone call |
| 02:04.03 | picard276 | especially when u are on public wifi |
| 02:04.08 | lorsungcu | k |
| 02:04.28 | picard276 | if u are on an MPLS or something very high end... sure .. (but in that case hopefully your enviornment is setup to handle voip correctly) |
| 02:04.34 | lorsungcu | MPLS != VPN |
| 02:04.48 | lanning | I think it is more the "public WiFi" than the VPN software. |
| 02:04.55 | lorsungcu | i've been on a pptp VPN for 5+ hours now |
| 02:04.59 | lorsungcu | without a single drop |
| 02:05.11 | lorsungcu | all my voice traffic goes over L2TP/ipsec |
| 02:05.22 | picard276 | lorsungcu |
| 02:05.24 | picard276 | where are you? |
| 02:05.24 | lorsungcu | have customers on ipads using them exclusively over pptp with softphones |
| 02:05.28 | lanning | I have a openvpn from my desk at work to my home, it goes down maybe once a month or so. |
| 02:05.34 | lorsungcu | they are as stable as the connection you're on |
| 02:05.40 | picard276 | right |
| 02:05.51 | picard276 | lorsungcu i know that |
| 02:06.00 | picard276 | im just saying... on a connection like that.. a VPN is pretty stable |
| 02:06.01 | lorsungcu | <picard276> |
| 02:06.02 | lorsungcu | PPTP or openVPN are not really stable enough i think for a phone call |
| 02:06.09 | picard276 | the public wifi is obviously hte problem |
| 02:06.22 | picard276 | but if a VPN drops it has to reconnect.. if a few phone packets drop.. the call continues |
| 02:06.44 | picard276 | L2TP/ipSec interesting.. why do you do all your voice traffic like that (the reason i said above or something else?) |
| 02:06.50 | lorsungcu | go lab something with 1% packet loss |
| 02:06.55 | lorsungcu | let me know how great the call sounds |
| 02:07.06 | lanning | the issue is that usually the VPN interface goes down when it disconnects, then it gets a new IP when it reconnects. |
| 02:07.08 | lorsungcu | if the wifi is that bad, you won't be fixing it |
| 02:07.19 | picard276 | right .. that is the point im making |
| 02:07.22 | lorsungcu | lanning: that is entirely up to how you configure it |
| 02:07.28 | lanning | true |
| 02:07.41 | picard276 | its not just the IP |
| 02:07.43 | picard276 | its more |
| 02:07.44 | [TK]D-Fender | If your connection is bad, stop blaming other things |
| 02:07.44 | picard276 | the time |
| 02:07.46 | picard276 | for hte VPN to reconect |
| 02:07.50 | [TK]D-Fender | Use IAX2 on an odd port |
| 02:07.55 | [TK]D-Fender | Use a VPN. |
| 02:07.57 | [TK]D-Fender | Do whatever you want |
| 02:07.59 | picard276 | why IAX2 over SIP? |
| 02:08.06 | lorsungcu | none of that matters if your connection is that bad |
| 02:08.08 | [TK]D-Fender | You already know the piece, just make up your mind and d it |
| 02:08.09 | ChannelZ | In Soviet Russia, VPN tunnels YOU! |
| 02:08.22 | lanning | heh |
| 02:08.24 | [TK]D-Fender | picard276: Because it isn't SIP |
| 02:08.26 | lorsungcu | that sounds somewhat dirty, ChannelZ |
| 02:08.29 | picard276 | ? |
| 02:08.38 | picard276 | what is your beef with sip? |
| 02:08.44 | lorsungcu | rofl |
| 02:08.49 | lorsungcu | TK stop talking shit about SIp |
| 02:08.51 | lorsungcu | i've had it |
| 02:08.53 | [TK]D-Fender | Who said *I* had a beef with SIP? |
| 02:09.00 | [TK]D-Fender | YOU are the one with "issues" |
| 02:09.06 | [TK]D-Fender | and we're proposing alternatives |
| 02:09.08 | [TK]D-Fender | IAX2 is one |
| 02:09.10 | lanning | mmmm.... beeeefff |
| 02:09.52 | lorsungcu | if you can;t maintain a VPN connection |
| 02:09.55 | [TK]D-Fender | picard276: If your connectivity is flakey then just put a bullet in your head. You'll take what you can get. If they block direct SIP, then VPN or alternate protocol it is... |
| 02:09.58 | lorsungcu | your calls will essentially be worthless |
| 02:10.10 | picard276 | kk |
| 02:10.12 | picard276 | yea |
| 02:10.14 | picard276 | i knw |
| 02:10.24 | lanning | actually with OpenVPN via UDP, you can have really long timeouts |
| 02:10.27 | [TK]D-Fender | So get your hands off your nuts and SEIZE THE DAY! |
| 02:10.59 | lanning | you can basically use a phone line. hang up, then redial and the VPN will still be active... :) |
| 02:11.04 | picard276 | so Ipsec |
| 02:11.12 | lorsungcu | no. |
| 02:11.14 | [TK]D-Fender | or whatever |
| 02:11.28 | picard276 | well i will probably build in someting that if it blocks direct sip |
| 02:11.31 | picard276 | to just fire up a VPN |
| 02:11.35 | picard276 | but try direct SIP first |
| 02:12.14 | [TK]D-Fender | heads out for a while. |
| 02:12.15 | lanning | direct will always have lower latency, which is nice. |
| 02:12.21 | picard276 | right |
| 02:12.32 | picard276 | its just a litlte problem i have ran into before |
| 02:12.37 | [TK]D-Fender | IAX2 is a SINGLE port and you can set your server to whatever alternate port you want. |
| 02:12.52 | [TK]D-Fender | there's an esay plan |
| 02:12.56 | [TK]D-Fender | easy plan* |
| 02:13.10 | lanning | ya, it is way easier than figuring out RTP failures... |
| 02:13.18 | lorsungcu | latency will not change appreciatively with a VPn |
| 02:13.20 | picard276 | hmmm. interesting |
| 02:13.36 | picard276 | can you . |
| 02:13.39 | lanning | depends on the VPN |
| 02:13.44 | lorsungcu | jesus christ |
| 02:14.03 | picard276 | im not 100% familiar with IAX2... but do u have to register.. because im not doing sip registering... im just calling a sip extenion |
| 02:14.16 | picard276 | or can i send URI 's with IAX2 |
| 02:14.31 | [TK]D-Fender | IAX2 is NOT SIP |
| 02:14.51 | lorsungcu | you can user certificates, picard |
| 02:15.06 | [TK]D-Fender | registration is optional for the same reasons as SIP |
| 02:15.15 | [TK]D-Fender | Go read |
| 02:15.16 | [TK]D-Fender | go use it |
| 02:15.38 | picard276 | kkk |
| 02:15.45 | picard276 | thanks guys |
| 02:15.47 | picard276 | appreciated |
| 02:16.01 | ChannelZ | Registering should be renamed to something less authoritative like "informing" |
| 02:16.50 | lorsungcu | hmm |
| 02:17.11 | lorsungcu | i dont get what people's issue with PPTP is |
| 02:18.26 | picard276 | it can be hacked |
| 02:18.32 | picard276 | easier |
| 02:18.36 | picard276 | than openVpn |
| 02:18.39 | lorsungcu | show me an exploit |
| 02:18.40 | picard276 | idk about ipsec |
| 02:18.53 | lorsungcu | and your issue is NAT/firewalls |
| 02:18.55 | picard276 | there is something called |
| 02:19.05 | picard276 | i have to google it |
| 02:19.17 | lorsungcu | go for it, ill be here |
| 02:19.57 | picard276 | http://www.computerworld.com/s/article/9229757/Tools_released_at_Defcon_can_crack_widely_used_PPTP_encryption_in_under_a_day |
| 02:20.00 | picard276 | this is not the article i originally was looking for |
| 02:20.02 | picard276 | but that is one |
| 02:20.10 | lorsungcu | using MS-CHAP |
| 02:20.15 | lorsungcu | find me something recent |
| 02:20.57 | picard276 | http://linuxhouse.eu/networking/hack-pptp-vpn |
| 02:21.14 | lorsungcu | same thing |
| 02:21.17 | lorsungcu | keep going |
| 02:21.23 | picard276 | there is a non chap article |
| 02:21.26 | picard276 | but i dont remember wher eit is |
| 02:21.38 | lorsungcu | k |
| 02:21.43 | lorsungcu | but it solves your issue |
| 02:21.49 | lorsungcu | is perfect for what you want |
| 02:21.54 | picard276 | you say to use ipsec though? |
| 02:21.58 | lorsungcu | and you come up with 1000 reasons not to use it |
| 02:22.05 | lorsungcu | I think that's somewhat funny |
| 02:22.16 | picard276 | i dont know anything about ipsec |
| 02:22.35 | lorsungcu | no, i say to use PPTp |
| 02:36.34 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 02:37.46 | *** join/#asterisk s_enya (~enya@fw.technoface.co.jp) |
| 02:37.55 | s_enya | hello |
| 02:41.30 | carrar | IPSEC FTW |
| 02:42.55 | lorsungcu | ipsec is unnecessary for getting aorund NAT |
| 02:43.29 | carrar | s/ipsec/vpn/ |
| 02:45.12 | carrar | however |
| 02:45.15 | carrar | IPSEC FTW :) |
| 02:45.27 | carrar | Just cause I'm a fav of IPSEC |
| 02:45.30 | carrar | fan |
| 02:45.58 | carrar | it |
| 02:46.08 | carrar | it's the first thing I turn on after picking up my cell phone |
| 02:49.06 | deo | guys how do we turn off this notifications when accessing the CLI > http://pastie.org/7132309 |
| 02:52.26 | carrar | iax2 set debug off |
| 02:53.03 | carrar | core set debug off |
| 02:53.14 | carrar | core set verbose 3 |
| 02:53.48 | deo | hi carrar thanks this works iax2 set debug off .. i thought this is a verbose message from sip |
| 02:55.52 | carrar | yeah if you don't know how to turn off debug messages they can get pretty annouying |
| 02:56.04 | deo | yeah they are... |
| 03:05.31 | s_enya | I use Xeon E3-1220 hardware +16GB RAM, asterisk 11.1.0 and sipp-3.2. |
| 03:06.45 | s_enya | I put sipp command is "sipp -sn uac 200.100.99.98 -s 200 -r 200 -m 15000 -rtt_freq 1000 -trace_rtt -stat_delimiter" |
| 03:07.45 | s_enya | it result has many errors. |
| 03:08.15 | carrar | It hurts when I bite myself! |
| 03:12.13 | carrar | or maybe start with -r to 5 |
| 03:45.18 | *** join/#asterisk bipul (73fd9070@gateway/web/freenode/ip.115.253.144.112) |
| 03:48.12 | s_enya | carrar: I bite myself. I understand error cause. |
| 03:48.20 | *** join/#asterisk bipul_ (73fd9f29@gateway/web/freenode/ip.115.253.159.41) |
| 03:50.23 | bipul_ | Hellow |
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| 04:01.04 | bipul | Hi |
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| 04:54.05 | mnathani | It the PSTN generally oversubscribed? As in if everyone were to pick up the phone and make a call would some people not be able to get through? |
| 04:54.40 | mnathani | If so, then what percentage of people would have to try to call in order to get a consistent busy signal? |
| 05:16.52 | TriJetScud | mnathani: it generally depends |
| 05:17.11 | TriJetScud | in the old days if you were calling from one CO to another CO then yes they'll over subscribe the trunks |
| 05:18.01 | TriJetScud | of course if people happened to call each other within that same CO builiding there shouldn't be any busy tones as the switching matrix coicidentially is a non-blocking design ;) |
| 05:22.50 | igcewieling | gads, I hate expect |
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| 05:37.08 | Addisk | Does anybody know on the top of their head if its possible to configure the IVR(free pbx) to timeout to an extension(like operator)? |
| 05:39.42 | [TK]D-Fender | Addisk: yes |
| 05:42.37 | Addisk | where do you see that option? |
| 05:42.55 | Addisk | in the IVR, i see timeout message to play, but i can't find a timeout extension to foward to |
| 05:43.23 | [TK]D-Fender | you make it in the options. |
| 05:43.26 | [TK]D-Fender | "t" <- |
| 05:44.07 | Addisk | thats the option? |
| 05:44.15 | Addisk | instead of putting a number? |
| 05:44.31 | [TK]D-Fender | Depending on your version |
| 05:44.45 | [TK]D-Fender | In 2.10 there is "Timeout Destination" as a drop down |
| 05:44.48 | Addisk | i'm on 2.7 |
| 05:44.53 | [TK]D-Fender | then yes, "t" |
| 05:44.59 | [TK]D-Fender | and that is quite old |
| 05:45.10 | Addisk | planning to upgrade soon |
| 05:45.16 | Addisk | but need something quick and dirty for now |
| 05:45.24 | Addisk | the newest one is 2.11 ? |
| 05:46.50 | [TK]D-Fender | 2.10 is current, 2.11 is beta due for release ... later |
| 05:47.46 | Addisk | i've actually been out of this since forever |
| 05:48.01 | Addisk | i set this system up like 3-5 years ago and never touched it since |
| 05:49.41 | Addisk | hold crap lol |
| 05:49.43 | Addisk | i don't believe this |
| 05:49.45 | Addisk | System Uptime:83 weeks, 4 days, 4 hours, 7 minutes |
| 05:50.08 | Addisk | lol damn this system bound to die if it reboots |
| 05:50.57 | Addisk | the t works, thanks |
| 05:51.34 | [TK]D-Fender | You're welcome |
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| 07:30.45 | linocisco | is there any asterisk jobs channel? |
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| 09:37.21 | linocisco | is there any asterisk jobs channel? |
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| 10:12.51 | jermey_g | does any one have benchmarks asterisk conference capabiliies? version 1.4 to latest, we can use whichever perform best? |
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| 11:21.00 | k01 | hi all, i'm new in asterisk and i'm trying to configure it from scratch. i've a patton sn4112 to handle external calls, everything works fine except incoming calls. this is my sip.conf http://pastebin.com/MSswyTZY and here it is my extensions.conf http://pastebin.com/70GNdCDv. The problem in the asterisk log is simply "Call from '10001' to extension '10001' rejected because extension not found in context |
| 11:21.02 | k01 | 'default'." |
| 11:21.45 | k01 | any help is appreciated, thanks :) |
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| 11:26.19 | WIMPy | Did you read the message? |
| 11:29.55 | kaldemar | k01: the call does not start in the s extension. your dialplan does not have priority 1 for the 10001 extension, and that's why it says the extension is not found. "s" is not a "match all". |
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| 11:35.36 | k01 | kaldemar: thank you very much, i've changed to exten => 10001,1,Dial(SIP/300,120,r) and remove exten => s,1,Answer() and now works, thank you again ;) |
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| 11:36.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:36.19 | WIMPy | wonders how that message could be made more obvious. |
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| 11:38.35 | kaldemar | by saying something about priority 1 maybe. |
| 11:39.22 | kaldemar | "...because not matching extension with priority 1 found in context..." |
| 11:46.24 | WIMPy | Could help. But that's more often not the issue, I think. |
| 11:47.44 | WIMPy | k01: What do you think could make it more obvious? |
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| 12:52.21 | vlad_starkov | Question: How can I verify .tar.gz archive with .asc gpg key? |
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| 13:14.00 | setkeh | <PROTECTED> |
| 13:14.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:14.53 | [TK]D-Fender | setkeh, ... and the pastebin I asked for? |
| 13:15.23 | setkeh | [TK]D-Fender: [Mar 27 09:14:30] WARNING[22671][C-00000005]: pbx.c:4621 pbx_extension_helper: No application 'VoiceMailMain,s101@default' for extension (house, 601, 1) is the only error |
| 13:15.29 | setkeh | pastebinning configs now |
| 13:15.35 | Greenlight | I'm putting together some reporting on call stats based on CDR's. I'm seeing that if a call enters a queue and is never answered, the CDR for that call shows a disposition of "BUSY". Is this reliable; or are there situations where it'll show "ANSWERED" ? |
| 13:15.54 | [TK]D-Fender | setkeh, right there is a problem.. you don't use a comma after the application name. That is soooo * 1.0 <-0 |
| 13:16.32 | [TK]D-Fender | setkeh, And the "s" doesn't belong there either on anything resembling a server from the past 5 years |
| 13:16.43 | [TK]D-Fender | setkeh, Your syntax is almost a decade out... |
| 13:17.21 | setkeh | wiki's FTW XD |
| 13:17.35 | setkeh | http://sprunge.us/KIXa extentions.conf |
| 13:17.39 | [TK]D-Fender | voip-info.org = decrepit |
| 13:17.47 | setkeh | http://sprunge.us/MDMS |
| 13:17.53 | setkeh | voicemail.conf |
| 13:18.15 | [TK]D-Fender | Tons of sample crap you should permanently remove in there. |
| 13:18.31 | [TK]D-Fender | Waiting for the CALL TRACE with SIP DEBUG........ |
| 13:18.42 | [TK]D-Fender | And a confirmation of what version of * you're now running... |
| 13:19.10 | setkeh | [TK]D-Fender: yeah i intend too :D just trying to make it work first :P |
| 13:20.29 | setkeh | Asterisk 11.2.1 built by root @ ibmxxxxxxx.digium.internal on a x86_64 running Linux on 2013-03-06 18:55:26 UTC |
| 13:22.06 | [TK]D-Fender | Awaiting call dump..... |
| 13:22.23 | setkeh | oh sorry how do i do that ?? |
| 13:22.39 | [TK]D-Fender | * CLI -> "core set verbose 10" , "sip set debug on" |
| 13:23.10 | setkeh | ty :D |
| 13:27.42 | setkeh | https://gist.github.com/setkeh/5254136 |
| 13:27.47 | setkeh | there wego :D |
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| 13:29.53 | [TK]D-Fender | setkeh, exten =>101,2,Voicemail(101@default) |
| 13:29.53 | [TK]D-Fender | exten => 601,1,VoiceMailMain,s101@default |
| 13:30.01 | [TK]D-Fender | setkeh, the first format is valid, the 2nd isn't |
| 13:30.22 | [TK]D-Fender | setkeh, Which I already confirmed |
| 13:30.33 | setkeh | awesome thanks :D |
| 13:30.33 | [TK]D-Fender | <[TK]D-Fender> setkeh, right there is a problem.. you don't use a comma after the application name. That is soooo * 1.0 <-0 |
| 13:30.50 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 13:30.54 | [TK]D-Fender | ~book |
| 13:30.55 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:30.56 | [TK]D-Fender | ^^^ |
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| 13:36.16 | setkeh | [TK]D-Fender: in that formatif i dial 101 from my tab (101) it just calls itself ?? |
| 13:39.13 | fling | is not it working fine with selinux? |
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| 13:45.11 | [TK]D-Fender | setkeh, I don't understand your question.... |
| 13:46.14 | setkeh | [TK]D-Fender: you said exten =>101,2,Voicemail(101@default) is the correct one to add to extentions but when i dial 101 it just calls my tablet |
| 13:46.33 | [TK]D-Fender | setkeh, that is priority #2, not 1 |
| 13:46.48 | [TK]D-Fender | exten => 101,1,Dial(SIP/setkehtab,20) |
| 13:46.48 | [TK]D-Fender | exten =>101,2,Voicemail(101@default) |
| 13:46.59 | [TK]D-Fender | you made your first step dial. |
| 13:49.21 | setkeh | [TK]D-Fender: ooh i seethe issue |
| 13:49.45 | setkeh | [TK]D-Fender: the original issue was being able to access voicemails XD |
| 13:49.48 | setkeh | thats my bad |
| 13:50.10 | setkeh | which is why i had exten => 600,1,VoiceMailMain,s100@default |
| 13:51.06 | setkeh | the "Step dial" and recording seems tobe working great ijustneed to be able to access it now :D |
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| 14:03.55 | setkeh | how wouldi goabout that ?? |
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| 14:07.16 | igcewieling | setkeh: since exten => 600,1,VoiceMailMain,s100@default is not valid I'm not sure what you are talking about. It isn't valid. |
| 14:07.29 | igcewieling | make it valid and it might work |
| 14:07.51 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-137-199.user.veloxzone.com.br) |
| 14:08.24 | setkeh | igcewieling: i understand that :D i just needto know what i needto do so i can dial say 12 or *12 or somthing like that to listen to the voicemails :D |
| 14:09.07 | *** join/#asterisk jetlag (jetlag@pool-71-168-200-61.cmdnnj.east.verizon.net) |
| 14:09.35 | [TK]D-Fender | setkeh, "core show application voicemailmain" <-\ |
| 14:11.25 | *** join/#asterisk theridge (~kvirc@adsl-108-82-201-194.dsl.chi2ca.sbcglobal.net) |
| 14:12.46 | igcewieling | exten => 12,1,VoicemailMail(100@default,s) exten => *121,VoicemailMail(100@default,s) |
| 14:12.49 | igcewieling | there you go |
| 14:13.06 | igcewieling | sorry, too early in the morning |
| 14:13.18 | igcewieling | <PROTECTED> |
| 14:13.31 | [TK]D-Fender | Still too early... |
| 14:13.55 | [TK]D-Fender | 3rd time's the charm ... but 47th time is JOB SECURITY |
| 14:14.03 | igcewieling | [TK]D-Fender: I'll go back to my home dimension and try again tomorrow |
| 14:14.27 | setkeh | ooh i see |
| 14:14.29 | igcewieling | All the info he needs is in The Book anyway |
| 14:14.50 | setkeh | igcewieling: yeah ill be buying iton payday :D |
| 14:14.51 | igcewieling | Gads, I hate expect. |
| 14:15.10 | igcewieling | setkeh: until then, why not read the online edition for free? |
| 14:18.47 | Katty | hello my asterisk does not work at all how to fix?? plz answer is urgent thx |
| 14:19.24 | setkeh | igcewieling: ahh found it thnx :D |
| 14:20.03 | igcewieling | Katty: I recommend more fiber. |
| 14:20.25 | Katty | igcewieling: i've got 4 needles full of fiber. |
| 14:20.34 | Katty | igcewieling: making Wilfred's brother this morning ^_^ |
| 14:20.46 | Katty | igcewieling: http://42ndknitstreet.blogspot.com/2013/03/wilfred-giraffe-for-henry.html <- wilfred. |
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| 14:25.50 | carrar | maybe a wife and child too! |
| 14:25.58 | setkeh | ok now im getting "No application 'VoicemailMailMain' for extension (house, 12, 1)" and i have read the Voicemail partof the book minue the conclusion and couldent find anything to suggest this |
| 14:28.01 | kaldemar | read VoicemailMailMain a few times |
| 14:28.46 | kaldemar | if no light bulb, compare it to 'VoiceMailMain' |
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| 14:30.59 | setkeh | kaldemar: derp XD ty |
| 14:31.18 | [TK]D-Fender | <[TK]D-Fender> setkeh, "core show application voicemailmain" <- |
| 14:31.45 | slav3_kitten | pounces Katty |
| 14:31.51 | anonymouz666 | anyone using asterisk 11 in production? |
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| 14:37.11 | Greenlight | anonymouz666: Yes |
| 14:37.48 | anonymouz666 | Greenlight: how many calls/day? |
| 14:38.28 | Greenlight | Circa 250-500k |
| 14:39.15 | Greenlight | Other systems will less, but busiest is around that iirc |
| 14:39.29 | anonymouz666 | stable so far? |
| 14:39.46 | Greenlight | Yup, very |
| 14:40.20 | Greenlight | Only ever needed to restart to apply system updates. Never crashed. |
| 14:40.30 | anonymouz666 | 11.2.1? |
| 14:40.58 | Greenlight | Ranges from 11.0.1 to 11.3rc1 |
| 14:41.01 | Katty | hugs slav3_kitten |
| 14:41.19 | slav3_kitten | how you been cat? |
| 14:42.00 | Katty | not too shabby, just go over a cold and that's wonderful!!! how're you? |
| 14:43.24 | Greenlight | Had some niche case isses on one system, where it deadlocked, but that was probably myself to blame for briding a call the moment it joined a queue. Fixed around that. Aside from that all good and stable on 11 |
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| 14:46.08 | din3sh | hello all |
| 14:46.20 | Greenlight | hello |
| 14:48.08 | igcewieling | Asterisk 11 has been far more stable than I'd expect from such a young release. |
| 14:48.33 | *** part/#asterisk theridge (~kvirc@adsl-108-82-201-194.dsl.chi2ca.sbcglobal.net) |
| 14:49.06 | din3sh | anyone here running cisco 6945 (SIP) with asterisk ? |
| 14:53.31 | slav3_kitten | i'm doing decently enough Katty, family not so much but we will get through it |
| 14:53.46 | slav3_kitten | din3sh, i'm rocking cisco 7900 series with sip images |
| 14:54.52 | [TK]D-Fender | din3sh, I don't see any standard resellers carrying it, and it doesn't lok too exciting... |
| 14:55.14 | [TK]D-Fender | look* |
| 14:56.54 | Free99 | just wondering: what does RTPproxy do in terms of STUN? so far as I understand it, it keeps a sip session alive transparently between a NAT'd client and the server...isn't that what STUN is? |
| 14:58.40 | [TK]D-Fender | STUN is not a proxy |
| 14:58.56 | din3sh | i also run cisco 7900 series , like 400 of them |
| 14:59.00 | din3sh | no real issues |
| 14:59.14 | din3sh | with firmware version 8.x |
| 14:59.26 | din3sh | the cisco 6945 is a pain in the ass |
| 15:00.18 | din3sh | also tried chan_sccp with 7900 series, pretty cool to run cisco with its native protocol |
| 15:02.17 | din3sh | running into issues with the cisco 6945, supposed to the successor of the 7911 line |
| 15:02.40 | din3sh | anyway |
| 15:03.21 | *** join/#asterisk blizzow (~jburns@75-171-154-172.hlrn.qwest.net) |
| 15:04.38 | din3sh | <PROTECTED> |
| 15:04.57 | din3sh | [TK]D-Fender: any idea why? |
| 15:06.46 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:09.53 | [TK]D-Fender | nope |
| 15:10.36 | din3sh | lol nevermind, thnks |
| 15:11.22 | WIMPy | Maybe you have some no longer supported option in your sip conf? |
| 15:12.00 | WIMPy | I have a feeling thay can cause random effects. |
| 15:12.15 | din3sh | hey WIMPy |
| 15:13.12 | din3sh | my sip.conf http://pastebin.com/JvjmFYxU |
| 15:14.41 | din3sh | WIMPy: the card i swapped some time back seems to have resolved some of my issues, (HDLC error etc) |
| 15:16.49 | WIMPy | Only some? |
| 15:17.49 | din3sh | stll having occasional HDLC |
| 15:18.08 | WIMPy | How often? |
| 15:18.22 | din3sh | once in 2 weeks |
| 15:18.24 | din3sh | or so |
| 15:18.54 | WIMPy | Ah, ok. Doesn't sound bad. |
| 15:27.36 | din3sh | yeah I no longer have major issues with calls coming through the E1 (at least no issues has been reported) |
| 15:28.41 | din3sh | but when i checked the logs, i saw occasional HDLC errorr with alarm on span 1 |
| 15:28.54 | din3sh | but also timing slips between asterisk and nortel on span 2 |
| 15:30.06 | igcewieling | din3sh: not uncommon. make sure your have the sync source (timing) correct on each span. |
| 15:31.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.81) |
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| 15:32.53 | din3sh | span 1 (asterisk) has pstn as timing source, span 2 (nortel) has asterisk as source |
| 15:33.28 | din3sh | my question is if there is some kind of fault in line on span 1, will it transfer to span 2? |
| 15:34.07 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 15:35.03 | Free99 | if I have asterisk running on sip.blah.com, is the domain blah.com or sip.blah.com? |
| 15:35.27 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:36.25 | [TK]D-Fender | Free99, this is a basic DNS question. has nothing to do with SIP |
| 15:36.37 | [TK]D-Fender | Free99, FQDN points to IP. that's all there is |
| 15:38.16 | Free99 | so domain is just blah.com |
| 15:39.19 | drmessano | If you have DNS SRV records, and other hosts are resolving your DNS SRV records, you can have a SRV record pointing your TLD to that sip.domain.com. That's how SIP URI's are supposed to work. But regardless, if your box is at sip.domain.com, it's at sip.domain.com. Host records are host records. |
| 15:39.22 | [TK]D-Fender | <Free99> if I have asterisk running on sip.blah.com, <- what does this mean then? |
| 15:39.45 | Free99 | iunno man, this is really complex software for me, I'm used to working with LDAP and such, which seemed to require a bit less guru-ness |
| 15:40.02 | [TK]D-Fender | Free99, this has nothing to do with "software" |
| 15:40.10 | [TK]D-Fender | Free99, this is your understanding of DNS |
| 15:40.38 | [TK]D-Fender | Free99, SIP has no role in the FQDN you choose. |
| 15:40.53 | Free99 | [TK] ordinarily, you're right, but when I'm having issues I try to nail every possibility down. Sometimes software isn't as intuitive/rule-following as you'd like to think |
| 15:41.17 | [TK]D-Fender | Free99, this is DNS. there is no mystery. |
| 15:41.29 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 15:41.43 | [TK]D-Fender | Free99, "bob.myserver.com" and "myserver.com" could resolve to completely different IP's |
| 15:42.05 | Free99 | look, I'm trying to register a user in the sipusers database, there's a field for "fromdomain", and I'm trying to understand what that field is for |
| 15:42.07 | Free99 | that's all |
| 15:42.45 | [TK]D-Fender | Free99, your phones won't be using that field, and only certain providers require it |
| 15:43.15 | [TK]D-Fender | And it changes what goes into the FROM header when you send out an invite |
| 15:43.44 | Free99 | I love when I'm midway through typing a reply and I get my question answered |
| 15:43.48 | [TK]D-Fender | That part IS purely SIP |
| 15:44.05 | *** join/#asterisk kpettit (~kpettit@99-116-144-119.lightspeed.hstntx.sbcglobal.net) |
| 15:44.09 | [TK]D-Fender | because it isn't used for any kind of internal routing or matching |
| 15:46.35 | Free99 | so the fqdn simply points to a machine, but all machines below the domain are seen as parts of the domain. Duh, I guess I'd never thought about it so poignantly |
| 15:48.10 | [TK]D-Fender | That's the difference between SIP routing (layer 7) vs TCP/IP (layer 4) |
| 15:49.16 | Free99 | ok. So just so I'm clear here: my sip server is sip.bleh.com, but the sip domain is bleh.com, right? |
| 15:51.02 | [TK]D-Fender | Free99, Again, this isn't for YOUR server |
| 15:51.53 | [TK]D-Fender | Free99, You won't be using this for any of your devices. |
| 15:52.34 | [TK]D-Fender | Free99, PROVIDER peer entires may require this so that you flag as being part of their netwrk and so they can separate SIP domains even though a proxy IP may be shared across multiple SIP domains |
| 15:56.12 | Free99 | ok then. I could just leave this as null then, since the devices don't need it. Thanks |
| 15:56.16 | WIMPy | din3sh: As far as timing goes, yes. |
| 16:00.16 | *** join/#asterisk jetlag (jetlag@pool-71-168-200-61.cmdnnj.east.verizon.net) |
| 16:00.31 | *** join/#asterisk RyanTG (~Thunderbi@mailer.americassuppliers.com) |
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| 16:06.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 16:51.44 | skrusty | hi, can anyone help me understand what "Rejecting offer with image stream due to UDPTL initialization failure" means when trying to use a fax machine to send a fax out via asterisk to PSTN? |
| 16:51.53 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:54.07 | *** join/#asterisk navaismo (~navaismo@189.241.122.125) |
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| 17:06.52 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 17:10.56 | *** join/#asterisk sputnik_too (~sputnik@funtoo/user/sputnik-too) |
| 17:11.38 | igcewieling | skrusty: Fax -> FXS port on Asterisk -> FXO port on asterisk ? |
| 17:11.55 | igcewieling | if not, then properly describe the setup |
| 17:12.18 | skrusty | fax -> spa122 -> asteris -> sip provider |
| 17:12.37 | skrusty | i've got rid of the error now |
| 17:12.47 | skrusty | getting an unknown rtp codec notice now |
| 17:12.53 | skrusty | which i suspect is the ata using t30? |
| 17:13.03 | skrusty | (according to asterisk guru) |
| 17:13.24 | skrusty | just connecting to customers vpn now to see if there's a setup issue |
| 17:14.33 | Free99 | I seem to be getting a lot of these in my asterisk console when I try making a call between two phones: [Mar 27 13:13:08] WARNING[5267]: chan_sip.c:20335 handle_response_invite: just did sched_add waitid(28461) for sip_reinvite_retry for dialog 7a2cbbfc58e0b0fb531fd7fa58eaed91@darkstart.com in handle_response_invite |
| 17:15.50 | igcewieling | Free99: according to a digium developer, that message is harmless |
| 17:16.15 | igcewieling | The only "bug" is that a harmless message is a WARNING instead of a NOTICE |
| 17:16.45 | Free99 | igcewieling: the problem is that I cannot make a phone call between the phones... might that warning/notice be related? |
| 17:17.00 | igcewieling | Free99: it is not. |
| 17:17.04 | Free99 | I'm talking about the console spitting out at least 60 of those messages in a second |
| 17:17.08 | Free99 | (shrug) ok |
| 17:17.15 | igcewieling | then report the bug on jira |
| 17:17.31 | igcewieling | more than likely you have some OTHER issue like NAT issue. |
| 17:19.56 | Free99 | igcewieling: if it were NAT, wouldn't calling the asterisk server result in one-way audio? |
| 17:20.04 | Free99 | because I can make the echo test ok |
| 17:20.48 | Greenlight | What is it you are calling ? |
| 17:21.08 | igcewieling | Free99: NAT issues can manifest in many ways. 1) Is ASTERISK behind NAT 2) are the phones behind NAT 3) are the phones behind different NATs |
| 17:21.59 | Free99 | greenlight & igcewieling: I have an asterisk server on public internet, firewall only opened on 5060 and ssh, the phones are behind the same NAT though |
| 17:22.08 | *** join/#asterisk serafie1 (~erin@static-host-66-18-61-84.epbinternet.com) |
| 17:22.24 | igcewieling | Free99: so Asterisk and both phones are on the same network? |
| 17:24.06 | Greenlight | Sounds like directmedia is enabled for the peers |
| 17:24.34 | Greenlight | If both can echo test okay |
| 17:25.55 | igcewieling | Free99: pastebin the cli output of a failed call with sip debug enabled |
| 17:25.59 | igcewieling | someone here might find something in the result |
| 17:27.17 | *** join/#asterisk bipolar (~bipolar@204.186.46.94) |
| 17:34.36 | Free99 | hmm |
| 17:35.27 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 17:35.36 | Free99 | asterisk is outside the NAT, the phones are on the same network, but inside their NAT |
| 17:35.52 | Free99 | when I say same, I mean on my own private network |
| 17:36.08 | Free99 | chowing down, just a sec :) |
| 17:36.15 | igcewieling | then you need directmedia=no |
| 17:39.52 | *** join/#asterisk serafie1 (~erin@static-host-66-18-61-84.epbinternet.com) |
| 17:40.53 | Free99 | igcewieling: where would I set that? I have asterisk running in realtime |
| 17:41.28 | igcewieling | Free99: You need to set it, how to do it with realtime is left as an exercise for the reader. |
| 17:41.50 | Free99 | ;) thanks igcewieling |
| 17:42.17 | igcewieling | on OUR system we would either put it in sip.conf [general] or in the database for the peers, the field is....directmedia |
| 17:45.44 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:45.44 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:46.12 | Free99 | yeah, I tried sticking it into sip.conf, reloading, rebooting the phones.. no dice |
| 17:47.24 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 17:48.09 | [TK]D-Fender | Free99, Show us the DB dump & call with SIP DEBUG enabled |
| 17:48.10 | [TK]D-Fender | ~pb |
| 17:48.10 | 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. |
| 17:48.11 | Free99 | igcewieling: do you find having asterisk exposed to the internet to be dangerous? I keep finding articles saying I should use a proxy like kamailio in front of asterisk, but I have a feeling this is where a lot of my problems seem to be coming from |
| 17:48.12 | [TK]D-Fender | ^^^ |
| 17:48.33 | [TK]D-Fender | Free99, filter ONLY the secrets |
| 17:48.40 | Free99 | can do |
| 17:48.50 | igcewieling | we use iptables to limit access |
| 17:50.46 | Free99 | db dump= "database show" right/ |
| 17:51.17 | [TK]D-Fender | Free99, Not from * |
| 17:51.30 | [TK]D-Fender | Free99, dump your DB table from it's source |
| 17:51.33 | [TK]D-Fender | its* |
| 17:52.00 | Free99 | from the mysql console you mean? |
| 17:52.13 | Free99 | or how I set it up? |
| 17:52.52 | [TK]D-Fender | sure, console |
| 17:56.35 | Free99 | looks kind of silly, but: http://pastebin.com/N0grYh8Y |
| 17:56.59 | Free99 | that's the DB dump of "select * from sipusers;" |
| 18:00.57 | navaismo | canreinvite=yes |
| 18:02.29 | Free99 | a portion of me making a call, because it's at least 9000 lines (literally) in less than 2 seconds: http://pastebin.com/D9KFSdAA |
| 18:03.43 | navaismo | incomplete sip debug |
| 18:04.14 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 18:06.37 | Free99 | ok, so if I add debug to the logger.conf line about messages, then I can just copy to disk right? b/c I've been copy pasting (and it sucks) |
| 18:07.07 | [TK]D-Fender | Free99, You are missing setting directmedia=no , and the canreinvite=yes is BAD |
| 18:08.00 | Free99 | I didn't change canreinvite, also because I actually need the media stream to go through my server continuously (which canreinvite changes right?) |
| 18:08.29 | Free99 | directmedia=no I set in sip.conf, let me make sure its under the [general] part |
| 18:09.57 | *** join/#asterisk ujjain2 (~ujjain@unaffiliated/ujjain) |
| 18:12.57 | [TK]D-Fender | Free99, set in your database |
| 18:13.33 | Free99 | so /var/log/asterisk/messages isn't giving me what the console spits out for "sip set debug on" |
| 18:13.42 | Free99 | despite having changed logger.conf |
| 18:14.09 | Free99 | I set it as "messages => notice,warning,error,debug" |
| 18:14.29 | Free99 | I'll try that [tk] |
| 18:14.46 | navaismo | did you executed the logger reload? |
| 18:15.01 | Free99 | navaismo yes, I ran "logger rotate" |
| 18:15.19 | navaismo | hmmm reload != rotate |
| 18:15.44 | Free99 | weird, they both respond with "logger restarted" but ok |
| 18:15.47 | Free99 | let me try again |
| 18:17.50 | Free99 | http://pastebin.com/0uNC9csC <-- all it gives me |
| 18:18.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.141) |
| 18:19.55 | *** join/#asterisk bchia (~Adium@nat/digium/x-gmqyhzmyhdocofwh) |
| 18:20.32 | Free99 | ah ok I got it |
| 18:22.49 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 18:22.55 | Free99 | http://pastebin.com/1wvCr7mp |
| 18:23.59 | Free99 | it's weird b/c I can call 600 from both phones simultaneously and everything works. I'll try the canreinvite and directmedia [tk] |
| 18:37.27 | Free99 | so I set directmedia=no and canreinvite=no for both users in my database, still no luck. at least now asterisk shows the two as being online though |
| 18:39.03 | Free99 | [tk]d-fender or navaismo, would you suggest I just try this without using kamailio? |
| 18:40.38 | *** join/#asterisk din3sh (2988577a@gateway/web/freenode/ip.41.136.87.122) |
| 18:40.58 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 18:41.11 | navaismo | so you are using kamalio because....? |
| 18:41.56 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 18:43.30 | *** join/#asterisk jsjc (~Adium@227.Red-2-136-107.dynamicIP.rima-tde.net) |
| 18:44.55 | Free99 | I can't find the article, but they pretty convincingly laid out why your services would be hardened and more efficient by having kamailio handle UAs, while asterisk just did logic |
| 18:44.58 | Free99 | problem so far is, I cant get it to work lol |
| 18:45.22 | Free99 | oh here: http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch |
| 18:46.09 | [TK]D-Fender | Free99, What is your end deployment expected to look like? |
| 18:47.43 | Free99 | I'm trying to just make a basic system where people use phones to call each other, routed through a2billing (which I like b/c its open source |
| 18:48.05 | Free99 | but um... not too sure how large the deployment will get |
| 18:48.13 | Free99 | so ideally I could scale it up as need be |
| 18:48.19 | [TK]D-Fender | Free99, Give us a guesstimate |
| 18:48.29 | Free99 | lets say 400 users initially |
| 18:48.44 | Free99 | total |
| 18:49.05 | [TK]D-Fender | A single * box could handle that easily |
| 18:49.06 | Free99 | simultaneously, it's supposed to be international, so i'd say like 40 calls per hour or something |
| 18:50.18 | Free99 | I took what opentelecoms said with a grain of salt, but I figured I'd go for the hard route first just to make the system resilient and expandable. But ok, whatever kamailio is out |
| 18:51.05 | Free99 | because of some networking requirements, all the media streams, like rtp and such, have to go through the * box though. That's possible, right? |
| 18:51.16 | [TK]D-Fender | Correct |
| 18:51.53 | Free99 | can asterisk overcome the NAT stuff as rtpproxy was supposed to, or do I keep that too? |
| 18:52.06 | [TK]D-Fender | Not needed |
| 18:54.18 | Free99 | ok. So I changed my asterisk port to 5060, and restarted the server after shutting down kamailio and rtpproxy |
| 18:54.31 | Free99 | now when I call 600, I get nothing |
| 18:56.39 | navaismo | your phones point to asterisk server ip? |
| 18:57.17 | Free99 | yep |
| 18:57.27 | Free99 | correct port, and the phones register |
| 18:58.00 | Free99 | "sip show peers" says they have a hostname that is for within my home router network though.. guessing NAT isn't working yet? |
| 19:00.07 | *** join/#asterisk kresp0 (~kresp0@81.61.25.130.dyn.user.ono.com) |
| 19:04.26 | navaismo | sip debug again for failed call |
| 19:04.44 | Free99 | ok |
| 19:07.02 | *** join/#asterisk chatran (~chatran_@187.112.197.56) |
| 19:07.27 | chatran | hello guys |
| 19:07.42 | chatran | i have a big trouble !!! with pap2t |
| 19:08.05 | drmessano | !!! |
| 19:08.19 | chatran | can anyone give me the basic config or paste in anywhere to get work fast ? |
| 19:08.29 | drmessano | Factory reset it.. |
| 19:08.31 | chatran | sip.conf / extension.conf |
| 19:08.33 | drmessano | Go to Line 1 |
| 19:08.40 | chatran | well this i didnt try |
| 19:08.43 | drmessano | Put in the user, password, and proxy |
| 19:08.44 | drmessano | Done |
| 19:08.47 | [TK]D-Fender | chatran, There is no "config". requireing magic. Fill in the server IP, user & pass. THE END |
| 19:08.57 | drmessano | Jinx |
| 19:09.25 | chatran | well its work but half work, becose on pap2t they cant hear me when theyr call me |
| 19:09.35 | chatran | and line goes down |
| 19:09.37 | drmessano | ~sipnat |
| 19:09.37 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 19:09.40 | chatran | with an error |
| 19:09.47 | chatran | drmessano is not nat im sure |
| 19:09.56 | chatran | i did reset modem and put new modem |
| 19:10.02 | Free99 | http://pastebin.com/B4FGGMww |
| 19:10.09 | chatran | i think i am missing someting on config |
| 19:10.19 | chatran | i will paste the error |
| 19:10.20 | chatran | sec |
| 19:10.32 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 19:11.49 | Free99 | something's wrong with that aocomputing link, it doesn't seem to come up |
| 19:12.06 | chatran | http://pastebin.com/iKM3nShw |
| 19:12.21 | *** join/#asterisk citywok (~kvirc@208.186.203.146) |
| 19:14.14 | [TK]D-Fender | Free99, It's offline |
| 19:14.20 | [TK]D-Fender | Maybe back someday.... |
| 19:14.52 | chatran | [TK]D-Fender can you take a look on my error? |
| 19:15.04 | Free99 | pity. I added "nat=yes" and "qualify=yes" to sip.conf, going to try it |
| 19:16.59 | Free99 | ok, so I got 600 working again |
| 19:17.24 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:17.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:17.29 | Free99 | but I cannot call between the two phones, they are both "unreachable" by asterisk |
| 19:24.56 | *** join/#asterisk sputnik_too (~sputnik@funtoo/user/sputnik-too) |
| 19:25.06 | Free99 | how odd. I setup the two phones to use stunserver.org, but only one of them is listed as reachable by asterisk |
| 19:25.54 | Free99 | one of them is at port 5060, but the other is linked up to asterisk with port 64099 |
| 19:26.06 | Free99 | 64099 being the unreachable one |
| 19:30.15 | navaismo | free you need to configure externip/externhost, localnet, and the nat set properly in your peer conf, then be sure the ports are forwarded if you *box isnt public etc etc |
| 19:30.44 | Free99 | * is public, phones are not |
| 19:31.00 | Free99 | thought externip and such was for if * was behind nat |
| 19:32.22 | *** part/#asterisk voxter (~hardcore@70.36.63.61) |
| 19:32.39 | *** join/#asterisk lvlinux (~n1gg@adsl-98-86-95-97.tys.bellsouth.net) |
| 19:33.40 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 19:37.17 | Free99 | if I have nat=yes in sip.conf, but nat=no in the user database, the DB overrides, right? |
| 19:37.31 | Free99 | actually, in general, the DB overrides? |
| 19:38.28 | navaismo | maybe its time to read the: |
| 19:38.30 | navaismo | ~book |
| 19:38.30 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:39.14 | Free99 | :-/ my bad, |
| 19:39.39 | Free99 | they have a 4th version for asterisk 11 you know |
| 19:39.48 | Free99 | ofps.oreilly.com/titles/9781449332426/asterisk-DB.html#dynamic_realtime |
| 19:40.29 | navaismo | well I only point to the book i cant handle the bot urls |
| 19:41.03 | Free99 | oh. well yeah, the fourth version is free and online for the time being |
| 19:41.14 | Free99 | (shrug) just fyi |
| 19:45.53 | navaismo | thanks |
| 19:47.53 | *** join/#asterisk RyanTG (~Thunderbi@mailer.americassuppliers.com) |
| 19:49.13 | igcewieling | Free99: stop using a stun server. it is not needed |
| 19:49.32 | Free99 | ?? but it didn't work at all until I dod |
| 19:49.34 | Free99 | *did |
| 19:50.06 | igcewieling | Free99: Asterisk has NAT traversal technology. When you try using a client NAT traversal technology they often interfere with each other. |
| 19:50.23 | igcewieling | Free99: It didn't work because something is set up wrong, not because you are not using STUN |
| 19:50.26 | Free99 | hmm. ok |
| 19:50.56 | Free99 | so for the nat field, would you suggest I use "force_rport" or "comedia" or..? |
| 19:51.06 | igcewieling | Asterisk didn't even correctly support STUN until Asterisk 11, do you think for all those years people with clentts behind NAT were not working |
| 19:51.23 | igcewieling | Free99: I recommend setting it to nat=yes because of a bug in Asterisk |
| 19:51.28 | Free99 | ok |
| 19:51.44 | igcewieling | make SURE you are not forearding any ports on the client nat router, make sure you have directmedia=no |
| 19:52.05 | navaismo | or canreinite in this case |
| 19:52.14 | igcewieling | navaismo: how old of a version is he running? |
| 19:52.40 | Free99 | I'm on 1.8.10.1 |
| 19:52.41 | navaismo | dont know i just see that in his table |
| 19:52.56 | Free99 | but I can move up to 11 if I have to (kind of want to actually) |
| 19:53.02 | igcewieling | he must have used an old version of the database schema |
| 19:53.42 | Free99 | igcewieling: I'd been following this tutorial for a while http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb |
| 19:54.03 | igcewieling | Free99: get your db scheme from the asterisk source code contrib/ directory |
| 19:54.54 | igcewieling | Free99: *nod* You are "that guy who decided, for his first install, to use the most complicated way to set up asterisk." takes balls. |
| 19:55.10 | navaismo | :D |
| 19:56.12 | Free99 | lol iunno, I didn't like having to reload asterisk for each new user and stuff I added |
| 19:56.32 | Free99 | besides, I come from the LDAP side of the fence, I like keeping stuff in databases |
| 19:56.44 | Free99 | balls=stupid sometimes tho |
| 19:57.00 | igcewieling | not sometimes, usually. |
| 19:57.14 | Free99 | (shrug) trial by fire I suppose |
| 19:57.52 | igcewieling | so is dumping a 12 yr old into the middle of the wilderness, but that doesn't mean it is a good idea. |
| 19:58.04 | igcewieling | start simple, build on it. |
| 20:01.54 | Free99 | so I just noticed my router has something called "SIP ALG" on it.. currently disabled. Keep it that way right? |
| 20:02.25 | Free99 | I mean igcewieling, you're trying to help me make my system work without any weird NAT setup right? |
| 20:03.02 | Free99 | also igcewieling, 12 year old in the wilderness is a rite of passage in the dominican republic lol |
| 20:03.10 | Free99 | or, it was anyway |
| 20:03.29 | igcewieling | SIP ALG means "screw up other nat traversal" |
| 20:04.21 | Free99 | brb |
| 20:07.56 | *** join/#asterisk iprouteth0_ (ccf60469@gateway/web/freenode/ip.204.246.4.105) |
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| 20:09.32 | Free99 | weird that having any sort of sip port forwarding in the router screwed NAT up |
| 20:09.50 | Free99 | but that part works, thanks a million igcewieling |
| 20:10.10 | iprouteth0_ | some routers do not handle sip very well |
| 20:10.25 | iprouteth0_ | may also need to forward your RTP ports |
| 20:10.32 | Free99 | apparently even recent cisco routers :-/ |
| 20:10.45 | iprouteth0_ | actual cisco router or SOHO cisco router? |
| 20:10.55 | Free99 | SOHO.. |
| 20:11.01 | lorsungcu | yes them especially |
| 20:11.08 | iprouteth0_ | hard to say. probably running vxworks. |
| 20:11.23 | iprouteth0_ | I tend to prefer OpenWRT for firmware when I can and highly recommend it |
| 20:11.34 | lorsungcu | just buy mikrotik |
| 20:11.42 | lorsungcu | and never touch cisco anything again |
| 20:11.53 | lorsungcu | or sketchy custom firmware :) |
| 20:12.03 | iprouteth0_ | cisco is all well and good for telco/enterprise, but the SOHO leaves much to be desired |
| 20:12.08 | Free99 | tomato is pretty good too, I stopped messing with ddwrt when I tried that ad-program for dd-wrt and it gave my friend a virus via one of the ads lol |
| 20:12.18 | Free99 | not too surprising |
| 20:12.26 | iprouteth0_ | lorsungcu: yeah custom stuff can definitely be touchy, I'll give you that |
| 20:12.27 | Free99 | tbh, even their enterprise stuff sometimes sucks |
| 20:12.47 | *** join/#asterisk jblack (~jblack@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 20:12.55 | iprouteth0_ | well, either you can go church of brainslayer for DD*WRT and have a very limited developer base, or OpenWRT which has a greater community of developers |
| 20:13.20 | iprouteth0_ | community support is better with openwrt IMHO, though it's still pretty good with dd-wrt |
| 20:13.26 | Free99 | we had a cisco VPN system that was bought before I got here, cost like 2 grand, regular linux box worked way faster and better with openvpn in terms of integrating with the LDAP |
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| 20:13.56 | iprouteth0_ | Free99: Doesnt matter the size of cisco equipment. Some of our CRS1s have had some surprising difficulties for how much they cost |
| 20:13.57 | igcewieling | iprouteth0_: the outgoing audio generally opens the proper NAT port malling |
| 20:14.10 | iprouteth0_ | I work for a telco and we use a mix of cisco and alcatel equipment |
| 20:14.42 | Free99 | we had a real problem where we bought some 10G layer 3 switches from cisco and their throughput was actually like 7.55 based on my tests with iptraf |
| 20:15.04 | Free99 | swapped it to an HP that was 3/4 the price and it was way closer to 10 |
| 20:15.07 | iprouteth0_ | igceweiling: true, but I have encountered instances where it wouldnt work without forwarding it. Usually this happened when using remote end-points |
| 20:15.16 | iprouteth0_ | and was on ISP supplied CPE |
| 20:16.01 | iprouteth0_ | how many people here using google voice integration? |
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| 20:29.42 | igcewieling | iprouteth0_: there would be more of Google didn't keep changing their protocols |
| 20:30.21 | iprouteth0_ | I think there needs to be more explanation regarding XMPP priority settings in the documentation |
| 20:30.34 | iprouteth0_ | that one detail saved me headaches once I figured it out |
| 20:30.41 | leifmadsen | iprouteth0_: I look forward to your patches |
| 20:31.13 | igcewieling | iprouteth0_: for some reason I seem to remember the sample .conf file getting better comments recently, but I could be wrong -- I don't follow the google support closely. |
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| 21:03.04 | leifmadsen | in Asterisk 11 the Google support is significantly better |
| 21:03.08 | leifmadsen | prior to that, it's a crap shoot |
| 21:03.14 | Qwell | your face is a crap shoot |
| 21:03.22 | leifmadsen | Qwell: <3? |
| 21:03.27 | Qwell | sure!@ |
| 21:03.48 | leifmadsen | ya well fuck you |
| 21:03.53 | Free99 | are those testicles or a heart? Guess we'll never know. |
| 21:04.24 | leifmadsen | heart shaped tea bag |
| 21:04.40 | Free99 | yesah, bout that. |
| 21:04.50 | Free99 | *yeah |
| 21:06.52 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.2 (2013/03/27), 10.12.2 (2013/03/27), 1.8.20.2 (2013/03/27), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:06.53 | Free99 | having an issue where my context is defined, my user has a default context too, but none of the commands in the context are run |
| 21:07.30 | Free99 | how can I be sure asterisk loaded the context? |
| 21:07.55 | navaismo | dialplan show context |
| 21:09.15 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.2 (2013/03/27), 10.12.2 (2013/03/27), 1.8.20.2 (2013/03/27), DAHDI-linux 2.6.3 (2013/03/14), DAHDI-tools 2.6.3 (2013/03/14), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:09.48 | Qwell | mjordan: DAHDI is still 2.6.2, no? |
| 21:10.06 | mjordan | Well, there's a 2.6.3 on the downloads |
| 21:10.09 | Qwell | -rc1 |
| 21:10.26 | mjordan | doh |
| 21:10.58 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.2 (2013/03/27), 10.12.2 (2013/03/27), 1.8.20.2 (2013/03/27), 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 |
| 21:11.10 | Free99 | lol |
| 21:11.10 | mjordan | it was showing 2.6.1, which I knew wasn't right :-P |
| 21:14.51 | [TK]D-Fender | We support only the most current mistakes here! |
| 21:15.12 | [TK]D-Fender | is a dedicated follower of fashion... |
| 21:22.49 | iprouteth0_ | The biggest trouble I had with asterisk and google voice integration was most certainly related to the priority setting |
| 21:26.02 | Free99 | I keep getting an error about write() returning improperly, why is that? |
| 21:28.15 | pabelanger | Free99: agi script? |
| 21:28.19 | pabelanger | or ami? |
| 21:28.19 | Free99 | yep |
| 21:28.22 | Free99 | agi |
| 21:28.26 | pabelanger | then fix your script |
| 21:28.46 | pabelanger | you are likley closing your connect before asterisk is finishing writing |
| 21:30.01 | pabelanger | connection* |
| 21:30.08 | Free99 | thanks pabelanger |
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| 21:34.14 | iprouteth0_ | leifmadsen: what is the best channel to submit suggested documentation improvements?: |
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| 21:54.20 | mjordan | iprouteth0_: by documentation improvements, do you mean to the wiki, or to the application/function/etc. documentation in Asterisk? |
| 21:58.57 | leifmadsen | if it's changes to a configuration file, then you make the change in the file, and do "svn diff" then submit the resulting diff upstream to the jira system |
| 22:00.10 | iprouteth0_ | thanks |
| 22:02.28 | iprouteth0_ | mjordan: might be a bit of both the wiki and comments in the conf files |
| 22:03.55 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 22:07.40 | mjordan | if it's specific wiki articles, contact mdavenport@digium.com. If it's the auto-generated documentation (which is a fair amount of it - you can tell because it will have an "Import Version" section on the page), then it has to be updated in the source first, a diff made, and the diff attached to a Jira issue |
| 22:08.00 | mjordan | er, malcolmd@digium.com. Not the other one. |
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| 23:29.02 | Free99 | hey there, I'm having trouble with this application called a2billing, does anyone know about it? |
| 23:29.46 | *** join/#asterisk gusto (~gusto@adsl-dyn101.78-98-21.t-com.sk) |
| 23:31.08 | Katty | HELLO CUPCAKES. |
| 23:33.56 | paulc | Mmm cupcakes.. are we talking normal ones? or ridiculous ones with an over abundance/copious amount of frosting? |
| 23:34.34 | Free99 | why not just frosting in a cupcake paper? put a maraschino cherry on top, just for nutrition reasons |
| 23:34.47 | Katty | why not both?! |
| 23:34.55 | igcewieling | paulc: I think the bundance/copious amount of frosting people know better than to come 'round these parts. |
| 23:35.18 | Katty | cupcakes with foo-fur-a |
| 23:36.43 | Katty | wow, there's an actual thing called Cupcake Porn. |
| 23:36.50 | Katty | flickr and pinterest |
| 23:36.57 | Katty | and tumblr |
| 23:37.47 | paulc | I dunno - I just remember as a kid getting some kind of lemon sponge cupcake from the town baker.. and it was awesome.. and it had like 2mm of icing on the top. Flat. Plain. Simple. And it was AWESOME. |
| 23:38.01 | paulc | today's crazyness with 3 pounds of frosting on top? just.... unnecesssary! |
| 23:38.14 | igcewieling | paulc: exactly |
| 23:38.25 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 23:40.52 | paulc | Yes! Exactly! Just like this: http://www.cupcakesrecipe.com/wp-content/uploads/2013/02/lemon-cream-cupcakes-4.jpg |
| 23:40.59 | igcewieling | Katty: Rule 34 |
| 23:41.00 | paulc | (now I feel my afternoon work is complete!) |
| 23:41.12 | Katty | we have rules now?! |
| 23:41.23 | Katty | no one told me we had rules! |
| 23:41.24 | igcewieling | decides to not get paulc started on people who put frosting on cookies |
| 23:41.36 | paulc | That's just plain wrong |
| 23:41.38 | Katty | like those delicious cookies from sam's club |
| 23:41.54 | paulc | I'll admit to putting cookies in the fridge though. And KitKats.. best way to have them.. cold chocolate.. Mmm... |
| 23:42.06 | igcewieling | a good cookie can stand on its own without the crutch of frosting. |
| 23:42.15 | *** part/#asterisk jdoe (jdoe@falseprophet.ca) |
| 23:42.32 | Katty | i agree. |
| 23:43.35 | Katty | pb cookies! |
| 23:43.40 | Katty | red velvet white choc chip cookies! |
| 23:43.50 | Katty | snickerdoodles! |
| 23:47.53 | *** part/#asterisk iprouteth0_ (ccf60469@gateway/web/freenode/ip.204.246.4.105) |
| 23:48.12 | carrar | STOP the madness |
| 23:49.45 | Katty | ICE the madness! |
| 23:49.54 | Katty | applies cookies to carrar |
| 23:52.23 | [TK]D-Fender | ma-ma-ma-ma-ma-ma-ma-ma-mad-mad-mad. ma-ma-ma-ma-ma-ma-ma-ma-madness! |
| 23:52.43 | [TK]D-Fender | counted that out entirely too many times for accuracy |
| 23:53.02 | Katty | applauds [TK]D-Fender's effort. |
| 23:53.25 | Katty | that's about 99% more efforts than i can currently muster |
