00:00.57 | p3nguin | I prefer to use wiki.asterisk.org for things like this, first. |
00:05.30 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
00:05.42 | Sean-Der | http://pastebin.com/KZRn4nCe |
00:05.47 | Sean-Der | http://pastebin.com/6KR4EzkB |
00:06.04 | Sean-Der | p3nguin: Does CEL default or do I need to uncomment the fields? |
00:07.44 | p3nguin | I'm not sure. Did you consult the page on CEL at wiki.asterisk.org? |
00:08.08 | Sean-Der | https://wiki.asterisk.org/wiki/display/AST/CEL+Configuration+Files |
00:08.14 | Sean-Der | Its blank? |
00:08.22 | *** join/#asterisk wonderworld (wonderworl@gateway/shell/anapnea.net/x-fzeyzrtmzjxyjrqg) |
00:09.56 | p3nguin | Yikes! That's not good. |
00:10.16 | Sean-Der | Could it be vandalism? Is there a revert? |
00:12.01 | p3nguin | I doubt it is vandalism. |
00:12.40 | Sean-Der | [2011-12-10 19:00:44] VERBOSE[3404] cel.c: -- CEL logging enabled. |
00:12.53 | Sean-Der | In my full log it looks like cel is now enabled |
00:13.19 | Sean-Der | [2011-12-10 19:00:44] WARNING[3404] cel_odbc.c: No such connection 'mysql1' in the 'first' section of cel_odbc.conf. Check res_odbc.conf |
00:13.23 | Sean-Der | Any ideas? |
00:21.29 | Sean-Der | hmm in odbc what is the 'dsn'? |
00:22.08 | helen_ | :) |
00:29.17 | *** join/#asterisk TimeRider (~steve@92.40.247.208.threembb.co.uk) |
00:30.51 | Sean-Der | It references your connection in odbc.ini |
00:37.55 | dijib | sup everyone |
00:37.59 | Sean-Der | Okay I have CEL working. Does anyone know of a decent PHP frontend. Or am I going to have to write my own? |
00:39.33 | *** join/#asterisk rotten777 (~quassel@fl-67-233-25-130.dhcp.embarqhsd.net) |
00:41.11 | rotten777 | to anyone awake i have a stupid question... i'm trying to make the default incoming calls for my pbx go to a list of playback commands then to a ring group... I'm using trixbox and have it playing moh by default but i would like to insert custom playback commands before the moh... i can't find where trixbox is putting the commands for the incoming calls |
00:42.04 | WIMPy | ~trixbox |
00:42.05 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
00:42.57 | rotten777 | ahh no wonder.. i should have never moved from my old standard asterisk implementation |
00:42.59 | rotten777 | ugh |
00:43.01 | rotten777 | thanks wimpy |
00:56.13 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
01:08.49 | *** join/#asterisk KojakWolf (~kojak@kojakwolf.com) |
01:53.38 | dijib | anyone wanna hold my p3n15 while i do a pgsql integration. |
02:01.35 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
02:05.19 | carrar | Once you get pgsql installed, maybe then you'll have something for someone to hang onto |
02:10.29 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
02:17.57 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
02:23.03 | *** join/#asterisk master_of_master (~master_of@p57B52C59.dip.t-dialin.net) |
02:58.48 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
03:17.40 | *** join/#asterisk wonderworld (wonderworl@gateway/shell/anapnea.net/x-paawhanfejrxdazx) |
03:41.49 | SeRi | carrar: LOL |
03:42.17 | SeRi | well the bday was fun.... kids had lots of fun |
03:45.58 | *** join/#asterisk radic (~radic@dslb-094-216-229-166.pools.arcor-ip.net) |
04:01.51 | p3nguin | seri: I have a conundrum. |
04:04.22 | SeRi | p3nguin: whats that? :) |
04:04.40 | p3nguin | I'm in a bit of a pickle, Dick. |
04:05.02 | SeRi | lol ok.... whats going on...? |
04:05.22 | p3nguin | I'm drawing a blank trying to route some traffic to another router... |
04:06.13 | p3nguin | One router in two subnets (A and B), serving hosts in subnet A... |
04:06.29 | SeRi | Mhhhh ok..... are you using RIP? |
04:06.31 | p3nguin | Hosts in subnet A need to have a default gateway of router B. |
04:07.00 | SeRi | ok I see. |
04:07.25 | p3nguin | But if I explicitly give them the address of router B, they can't reach it because they do not have an address in subnet B. |
04:07.40 | p3nguin | So I give them a default gw of router A. |
04:07.52 | SeRi | this is all inside vyatta right? |
04:07.55 | p3nguin | But I don't want traffic to go out router A's internet connection. |
04:08.05 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
04:08.24 | p3nguin | Yes, but the concept is going to be plain linux routing and iptables. |
04:08.42 | p3nguin | Should I NAT it? Should I try to route it? |
04:08.46 | SeRi | completly doable. I am doing something similar with pf |
04:09.00 | SeRi | one sec let me pull it up |
04:10.37 | p3nguin | I don't see how I can route it because I need to route everything, i.e. 0.0.0.0/0.0.0.0, from hosts on subnet A (192.168.10.0/255.255.255.0) to router B. |
04:11.01 | p3nguin | I thought maybe snat was the key, but I just don't know. |
04:11.41 | WIMPy | rules are your friend. |
04:11.46 | SeRi | the key is to route and to restrict access to servidces or traffic via iptables |
04:12.14 | p3nguin | I don't care about restricting anything. |
04:12.18 | SeRi | WIMPy: exactly |
04:12.27 | p3nguin | I just need to route all traffic to a router that is not on my subnet. |
04:12.29 | WIMPy | 'ip rule help' |
04:12.41 | SeRi | you can You can route and use ip rules to do what you want |
04:12.54 | SeRi | or even chains |
04:13.13 | p3nguin | Should I do it with nat or firewall? I'll use the vyatta interface to do it. |
04:13.34 | WIMPy | Neither. Do it with routing. |
04:13.42 | p3nguin | nat is going to use the nat table of iptables. firewall is going to use the filter table of iptables. |
04:13.46 | WIMPy | Policy routing. |
04:13.51 | p3nguin | Okay. |
04:13.58 | p3nguin | Let me look at my policies. |
04:15.35 | p3nguin | Perhaps I was over-thinking this. |
04:16.00 | SeRi | I would do routing as well. I dont know much about the policy's but I am sure they would do the job as well |
04:16.51 | p3nguin | It has to be based on the source address, though. That's where I ran into a problem in my mind. |
04:17.18 | WIMPy | Look for the LARTC. It's like ~book. |
04:17.44 | WIMPy | Add a from rule and set up a 2nd routing table. |
04:18.10 | WIMPy | I'm off for a little hardware change. |
04:20.44 | SeRi | p3nguin: I never done it the way WIMPy explained. mine is base on routing and rules |
04:21.13 | p3nguin | I can't do it that way, anyway. I have to do it with something that vyatta interface can handle. |
04:21.22 | p3nguin | It's like using freepbx and then trying to edit your own extensions.conf. |
04:21.32 | SeRi | ahhh I see. :) |
04:21.56 | p3nguin | The vyatta interface will do routing and firewall and nat... |
04:22.16 | SeRi | got cha... I need to find the time to load vyatta. |
04:23.48 | p3nguin | It looks like I might have the tools with "route filtering policy." |
04:24.13 | p3nguin | set policy access-list 100 rule 10 source ... |
04:24.17 | p3nguin | Maybe. |
04:24.25 | SeRi | ah that looks like the spot |
04:25.17 | p3nguin | Wait. This is for RIP. |
04:25.30 | p3nguin | I don't care about RIP. I just need to set some static routing policies. |
04:25.46 | p3nguin | It's one router. I do not need RIP. |
04:25.53 | SeRi | indeed |
04:25.59 | p3nguin | I really think NAT is where I need to look. |
04:26.05 | p3nguin | source nat |
04:26.29 | p3nguin | Anything from the 192.168.10.0/24 network, send it to the other router. |
04:27.05 | p3nguin | I just don't know. Do you see why I said it was a conundrum? |
04:27.14 | p3nguin | s/conundrum/pickle/ |
04:29.17 | SeRi | yes lol |
04:30.56 | SeRi | just use nat and set your rules. |
04:31.26 | dijib | hows the whiskey tonight ? |
04:31.30 | dijib | SeRi: |
04:32.18 | SeRi | no whiskey :) |
04:32.27 | SeRi | kids bday party today :) |
04:32.39 | dijib | i guess you didnt see my last post |
04:33.01 | dijib | i thought that already happend? |
04:33.07 | dijib | was reading sometime... i dont know |
04:33.19 | dijib | anyways none of my business |
04:33.28 | dijib | continue, sup asterisk? |
04:33.44 | SeRi | ????? |
04:33.51 | dijib | i gottah git some pgsql going |
04:34.06 | dijib | and am without a effin clue, ossmosis aint working either |
04:34.11 | dijib | ~osmosis |
04:34.12 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
04:35.18 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
04:35.34 | SeRi | dijib: for what? |
04:36.06 | SeRi | p3nguin: making progress>? |
04:36.22 | p3nguin | I'm looking at the SNAT configuration to see if I can use it for this. |
04:37.50 | SeRi | I see, |
04:39.21 | SeRi | how does vyatta use snat? as source nat? |
04:39.43 | *** join/#asterisk OldMonk (~raju@120.56.163.142) |
04:39.45 | OldMonk | hi |
04:40.18 | SeRi | pqsql is not hard at all. |
04:41.40 | OldMonk | i'm running an agi, which is executing fine. near the end, i call $agi->set_extension(1); extension 1 exists in the context. the agi debug shows <SIP/1001-00000030>AGI Rx << SET EXTENSION 1; however, on completion of the agi, the call ends with: Auto fallthrough, channel 'SIP/1001-00000031' status is 'UNKNOWN' |
04:42.10 | OldMonk | if i set_extension($invalid), the i extension is invoked normally |
04:42.16 | p3nguin | snat is source nat, configured with set service nat rule <rule> type source |
04:42.53 | SeRi | http://www.vyattawiki.net/wiki/NAT#Source_NAT |
04:43.20 | SeRi | p3nguin: many company's have their own defenition of snat thats whay I ask :) |
04:53.10 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
04:58.01 | Naikrovek | my daughter's hamster is going insane. run run run. all night long. run run run run run. |
05:04.31 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net) |
05:05.30 | KingDavidNYC | hello can someone please help me to identify the ip address of the incoming call? |
05:05.40 | KingDavidNYC | via the sip.conf |
05:07.08 | KingDavidNYC | am I the only one at home in front of a computer tonite? |
05:07.32 | *** join/#asterisk nix8n82 (~hmg@75-174-154-76.chyn.qwest.net) |
05:09.31 | p3nguin | seri: http://imagebin.org/index.php?mode=image&id=188014 |
05:10.38 | SeRi | p3nguin: that indeed looks a bit complicated. |
05:11.02 | p3nguin | It actually seems like a very simple scenario, but the solution seems hard to come by. |
05:11.25 | SeRi | KingDavidNYC: what do you mean via sip.conf? If somebody is calling you via sip than do sip show channells in cli and you will see there IP. |
05:11.38 | p3nguin | kingdavidnyc: You can't identify an active call by looking at a conf. You can, however, use "sip show channels" to see addresses of calls. |
05:11.52 | SeRi | s/tehre/their/ |
05:12.03 | SeRi | shit |
05:12.25 | KingDavidNYC | seri: I mean, how can I create a sip.conf entry to route calls from on specific IP addres to an specific context |
05:12.30 | SeRi | sorry for my fucked up spelling... is one of those nigths again |
05:13.29 | KingDavidNYC | seri: the first thing I would do would be to get the ip address from the sip header, when the call comes in, but I was wondering if the other way was the proper way to do it |
05:14.26 | SeRi | KingDavidNYC: you dont have to get it from the sip headers as posted from p3nguin and me you do sip show channels in cli and you can see the ip |
05:15.30 | p3nguin | If you just want to create a peer entry to match based on IP address, create a new peer definition in sip.conf using host=THE-IP-ADDRESS |
05:15.46 | p3nguin | This entry will be for a host which will not register to you, but will send calls to you. |
05:16.07 | p3nguin | It will be based on the IP address when you use type=peer. |
05:16.29 | KingDavidNYC | I did, but I am getting a weird error.. |
05:16.40 | KingDavidNYC | I created a sip.con entry |
05:17.37 | KingDavidNYC | and I am getting a ton of warnings that say "registration from x failed, no matching peer found |
05:18.11 | KingDavidNYC | I created a very simple entry: |
05:18.20 | KingDavidNYC | [Harry] |
05:18.24 | KingDavidNYC | type=peer |
05:18.49 | SeRi | show us the error |
05:18.51 | SeRi | ~pb |
05:18.51 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
05:18.55 | KingDavidNYC | host=the-ip-I want to authorize |
05:20.15 | dijib | holy cow m pissed |
05:20.17 | dijib | im |
05:20.28 | dijib | and eating mmm rice krispy squares |
05:20.34 | dijib | and beer |
05:20.40 | dijib | sleemans |
05:20.42 | p3nguin | kingdavidnyc: Are you trying to create a peer entry for a phone which will register to you? |
05:20.49 | KingDavidNYC | http://pastebin.com/wxhRHRBf |
05:21.16 | KingDavidNYC | p3nguin: no, no registration :) ... I just want to validate the ip |
05:21.19 | p3nguin | Don't get into a habit of trying to hide information when you're asking for help. |
05:21.41 | p3nguin | This error you're using says a phone is trying to register. |
05:21.54 | SeRi | ^^ |
05:22.16 | KingDavidNYC | p3nguin: I just want to receive incoming calls and authorize them by ip |
05:22.45 | KingDavidNYC | p3nguin: that's the weird part... I am not trying to register |
05:22.54 | SeRi | dijib: you ok? |
05:23.08 | p3nguin | To create an entry for any random devie which will not register to you, make it something like this: http://pastebin.com/GEd3YtyH |
05:23.11 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
05:23.59 | SeRi | KingDavidNYC: thats not what the error is saying.... The error defently is showing a peer trying to register |
05:24.05 | SeRi | [Dec 11 05:19:24] NOTICE[18971]: chan_sip.c:22318 handle_request_register: Registration from '"harry" <sip:harry@x.x.x.x>' failed for 'y.y.y.y' - No matching peer found |
05:25.25 | KingDavidNYC | seri/p3nguin: okay guys, sorry, it can then most likely be a sip phone that is trying to register, but I still dont know how to create the entry in sip.conf to authorize incoming calls |
05:25.47 | p3nguin | Already showed you. |
05:25.53 | SeRi | p3nguin: +1 |
05:25.59 | p3nguin | http://pastebin.com/GEd3YtyH <---- |
05:26.38 | p3nguin | Calls from THE-IP-ADDRESS will match this entry, and no authentication should be performed after that. |
05:26.45 | dijib | oh yeah SeRi im good |
05:26.54 | dijib | just doing a case swap on an x86 |
05:27.03 | dijib | just finished the swap. yet to plug it in |
05:27.26 | SeRi | dijib: cool. good luck :) dont burn the mbo :P |
05:27.37 | KingDavidNYC | p3nguin: I am going crazy :)... harry is not a phone, it is a sip.conf entry....why in the world is it trying to register? |
05:27.46 | dijib | burn out the mobo..... this guy new? |
05:27.58 | dijib | its all done, all scres accounted for. |
05:28.04 | KingDavidNYC | p3nguin: I promise you it is not a phone |
05:28.24 | p3nguin | Whatever it is, it is sending registrations. |
05:28.35 | dijib | i need a new i7 |
05:28.36 | dijib | :( |
05:28.48 | p3nguin | You _could_ allow it to register to you if you wanted. |
05:28.57 | SeRi | KingDavidNYC: maybe the source trying to call you is nor calling you instead is trying to register? |
05:29.10 | SeRi | s/nor/not/ |
05:29.28 | p3nguin | Change to host=dynamic and let it register. |
05:29.49 | p3nguin | Be sure to set a context that it can't do any harm. |
05:29.49 | dijib | p3nguin: you are an amaizing asterisk guide, and you must charge people $40 paypal to talk to you. |
05:29.55 | dijib | your that informed |
05:30.06 | KingDavidNYC | seri: yes, I think so... the source is another asterisk box, but there is no entry ther called harry |
05:30.14 | p3nguin | They won't pay. They'll just call me names and continue to ask questions. |
05:30.16 | dijib | awemazing |
05:30.25 | dijib | want $5? |
05:30.55 | p3nguin | I'll accept any cash that anyone wants to offer. |
05:31.03 | dijib | email |
05:31.13 | dijib | ppal? |
05:31.27 | p3nguin | I'll pm it to you. |
05:32.01 | dijib | if i can remember how seri explained multiwindow to me. |
05:32.05 | dijib | alt+ huh? |
05:32.06 | dijib | lol |
05:32.16 | p3nguin | Which window am I on? |
05:32.36 | SeRi | KingDavidNYC: unless you are been the middle box there is no reason why another asterisk box would call you. I think what is happening is that the other asterisk box is not trying to call you via sip is actually trying to register. |
05:32.53 | p3nguin | If I went to window 8, use Alt+8 |
05:32.58 | p3nguin | or |
05:33.01 | p3nguin | /win 8 |
05:33.16 | SeRi | atl+8 = awesomeness |
05:33.24 | SeRi | I love hot keys |
05:33.29 | KingDavidNYC | seri: I think so... I got one more box to check |
05:33.31 | p3nguin | You can also use Alt+ right arrow or left arrow. |
05:33.52 | SeRi | KingDavidNYC: You are confused and looks like you have it misconfigured. |
05:34.02 | p3nguin | This routing thing is really irritating me. |
05:34.14 | SeRi | p3nguin: yeap :) |
05:34.18 | dijib | lol |
05:34.22 | dijib | there we go |
05:34.23 | p3nguin | http://imagebin.org/index.php?mode=image&id=188014 |
05:34.29 | KingDavidNYC | seri: actually, I am beginning to think my box is being attacked |
05:34.32 | p3nguin | The topology is simple. |
05:34.32 | dijib | $5.05 canadian comming your way. |
05:34.39 | p3nguin | The task is difficult. |
05:35.17 | p3nguin | Amount: $5.00 CAD |
05:35.24 | p3nguin | via eCheck |
05:35.38 | SeRi | p3nguin: a vlan would have solve this problem in a simple way.... |
05:36.00 | p3nguin | But then I'll need two vlan capable switches, I think. |
05:36.08 | p3nguin | Actually, I know I will. |
05:36.27 | p3nguin | Because in reality, I have more than just the one PC on subnet A. |
05:36.46 | KingDavidNYC | dijib: are you seriously sending $5? |
05:37.29 | p3nguin | I have a second switch connected to the first switch. The second switch has systems in both subnets A and B, as well. |
05:37.47 | KingDavidNYC | dijib: the man deserves at least $100 |
05:38.06 | SeRi | KingDavidNYC: go for it :) |
05:38.49 | KingDavidNYC | seri: serious, where I come from, $5 is an insult |
05:38.54 | SeRi | I dont think two switches.. one would suffice. |
05:39.17 | SeRi | KingDavidNYC: Is not the money but the thought that counts. dijib did a good thing in return. |
05:40.20 | SeRi | 5 dollars is some nice beer money :) |
05:41.56 | KingDavidNYC | seri: okay, let's just say it is better I dont give my opinion on that one :) |
05:42.04 | SeRi | I do agree p3nguin deserves a lot more but this all part of information sharing so we are all in the same pege here and thats to help on another. |
05:42.07 | ChannelZ | goes for a Slurpee |
05:42.28 | ChannelZ | (and it's not really "sharing" if you pay) |
05:42.49 | SeRi | ^^ |
05:43.15 | SeRi | donation's are welcome |
05:44.09 | p3nguin | I'll have to have another switch to tag traffic from the hosts on my other switch. |
05:44.10 | KingDavidNYC | question: on this one:chan_sip.c:22318 handle_request_register: Registration from '"harry" <sip:harry@x.x.x.x>' failed for 'y.y.y.y' - No matching peer found |
05:44.31 | KingDavidNYC | who is trying to register? x.x.x.x or y.y.y.y?? |
05:44.43 | p3nguin | Yyyy |
05:44.44 | SeRi | "other switch" <----- didnt know that one :) |
05:44.48 | ChannelZ | helen_: you ever get your thing working BTW? I wandered off |
05:44.54 | KingDavidNYC | cause for the life of me I dont know whery the y.y.y.y ip comes from |
05:45.14 | p3nguin | Xxxx is your asterisk. |
05:45.23 | KingDavidNYC | that's right |
05:45.24 | SeRi | KingDavidNYC: what is sad is that you are letting that happen. mis configured * for sure. |
05:45.52 | KingDavidNYC | that means I am being attacked? |
05:46.22 | ChannelZ | If they're not expected IPs.... |
05:46.26 | SeRi | random public IP's should not be trying to register to your box unless you let it happen |
05:46.38 | p3nguin | SeRi: I have a core switch with both routers attached, and another switch daisy chained. |
05:47.13 | KingDavidNYC | great... the only thing that puzzles me is why it says "harry"?... harry was the name of my sip.conf context |
05:47.19 | p3nguin | The second switch has hosts attached which belong in different vlans. |
05:47.35 | ChannelZ | They can use any name they want (to try) |
05:47.54 | ChannelZ | Are you sure this isn't a softphone or something of your own? :P |
05:47.55 | SeRi | KingDavidNYC: look in to iptables, fail2ban and alwaysauthreject=yes |
05:47.55 | p3nguin | So I need to tag at that switch by the port. |
05:48.09 | KingDavidNYC | but harry is the name of one of my contexts.... |
05:48.16 | ChannelZ | so? |
05:48.33 | ChannelZ | Either a lucky guess or someone knows more about your system than they probably should |
05:48.48 | KingDavidNYC | exactly |
05:48.52 | p3nguin | That's your fault for having corresponding words. |
05:49.14 | KingDavidNYC | I only have one more x-lite phone in my office computer |
05:49.27 | KingDavidNYC | dont think so |
05:49.46 | p3nguin | If you want the Harry phone to register, create a peer for Harry. |
05:50.28 | ChannelZ | and then make an exten called 'balls' |
05:50.36 | ChannelZ | mmheh heh hmm heh |
05:51.17 | KingDavidNYC | p3guin: thanks p3nguin |
05:56.15 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
05:56.17 | SeRi | ChannelZ: hahahahah! |
05:56.28 | SeRi | p3nguin: +1 |
05:56.41 | SeRi | I bet fluffy is the password |
05:56.52 | p3nguin | or 'sweaty' |
05:56.57 | SeRi | hahaha |
05:57.27 | SeRi | p3nguin: ok I am starting to see the full pic your top.... |
05:58.27 | p3nguin | http://pastebin.com/GEd3YtyH |
06:01.02 | SeRi | hahahah p3nguin I got one for you! |
06:02.09 | p3nguin | Does it involve routing traffic from 192.168.10.0/24 our another gateway on a different subnet? |
06:02.28 | p3nguin | s/our/out/ |
06:06.31 | SeRi | p3nguin: http://i106.photobucket.com/albums/m260/xcom7/omfg.jpg |
06:06.45 | SeRi | the pic was an ad |
06:06.54 | SeRi | lol |
06:08.28 | p3nguin | The chick pic was an advertisement? |
06:09.58 | SeRi | yes |
06:10.08 | SeRi | hahaha |
06:10.23 | SeRi | something about texas dl |
06:16.39 | dijib | KingDavidNYC: yes i did send him $5 and yes he is worth $1000's |
06:16.42 | dijib | no argument |
06:17.13 | dijib | p3nguin must explain to me how and why he exists. |
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06:17.32 | dijib | fuck texas you guys are gun nutz |
06:17.40 | SeRi | fuck yea |
06:18.08 | SeRi | pussy's... |
06:18.14 | SeRi | lol |
06:18.15 | dijib | lol |
06:18.21 | dijib | dude ive been drinking |
06:18.32 | dijib | and therefor im kinof drunk |
06:18.41 | dijib | and yeh i need to smoke one im thinking |
06:19.35 | nix8n82 | smoke one what? |
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06:22.39 | [TK]D-Fender | dijib: You mean this is you .... BEFORE DRUGS? |
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06:24.33 | p3nguin | That is so irritating. |
06:26.44 | SeRi | [TK]D-Fender: LMAO |
06:26.50 | SeRi | p3nguin: whats going on? |
06:27.45 | p3nguin | It's kind of hard to explain exactly, but the gist of it is that when certain clients disconnect from my session, it locks up screen and kills IRC temporarily. |
06:28.45 | p3nguin | I don't know the cause, but I do know that it happens on more than one FreeBSD system running screen with irssi in it. |
06:37.46 | p3nguin | Maybe I could supernet this and make it work. |
06:37.53 | SeRi | weird... |
06:38.57 | SeRi | maybe... wouldnt that be a bit to much? |
06:39.12 | p3nguin | It's ridiculous that there isn't some way to route based on source address. |
06:39.35 | WIMPy | Get a real Linux. |
06:39.50 | p3nguin | I see you aren't a fan of Debian. |
06:39.54 | WIMPy | You don't use FreePBX either, do you? |
06:40.03 | p3nguin | I don't use FreePBX. |
06:40.23 | WIMPy | Err, if you say, Debinan can;t do it, I may have to retink if it's Linux at all. |
06:40.50 | WIMPy | I thought you were tryin vyatta or however that's spelled. |
06:40.59 | p3nguin | It is Vyatta. |
06:41.26 | p3nguin | It's a regular Linux, but with an interface that combines everything into it rather than using a bunch of separate tools. |
06:42.48 | p3nguin | It could be possible that the interface has the ability to accomplish this task, but without knowing what direction to go, I don't know what commands I need to enter into vyatta. |
06:42.49 | WIMPy | ... which doesn;t allow you to do what you want. |
06:43.00 | p3nguin | I tried natting it, but that didn't work. |
06:43.31 | p3nguin | The firewall does not perform that sort of operation, so that's no good. |
06:43.49 | p3nguin | Routing tables are based on destination, so that's out of the question. |
06:44.23 | WIMPy | Yes, but as I said befor I left, you can have multiple routing tables. |
06:44.44 | WIMPy | And you can have rules (like FROM) to decide which table to use. |
06:44.52 | p3nguin | Is there such a thing as source-based routing? |
06:45.21 | p3nguin | Conventional routing is destination based. |
06:45.35 | WIMPy | That's ehere the rules come in. |
06:45.49 | WIMPy | 'ip rule' ... |
06:48.17 | p3nguin | I'm not sure if I can access that from vyatta. I know it does use 'ip' for some things, but I would have to find out what commands exist for ip rules. |
06:48.37 | WIMPy | 'ip rule help' |
06:49.56 | p3nguin | I can't do it manually. That's exactly the same as hand-editing extensions.conf when you use FreePBX. DO NOT DO IT. |
06:50.35 | WIMPy | Do you see a pattern? |
06:52.09 | SeRi | :/ |
06:54.38 | SeRi | ha! GV will stay free for local US calls in 2012! |
06:54.55 | SeRi | I knew it. there is no way they can become a paid for service been beta |
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07:42.02 | p3nguin | Dammit. I thought I had it solved, then ran into a command MISSING. |
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07:48.03 | SeRi | p3nguin: ? |
07:48.33 | p3nguin | A command was renamed without notification in the documentation. |
07:48.37 | p3nguin | I found it. |
07:48.52 | SeRi | ah |
07:49.03 | p3nguin | They renamed "enable-source-based-routing" to "per-packet-balancing" |
07:49.11 | SeRi | :/ |
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07:51.06 | p3nguin | But it still does not work! |
07:52.24 | SeRi | :( |
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07:53.57 | WIMPy | Why am I not surprised that dahdi won't compile? |
07:55.33 | ChannelZ | I don't know. Why? |
07:56.10 | WIMPy | I upgraded to the latest Kernel before I put the card in. |
07:57.12 | ChannelZ | Is it some strange new kernel from the future? |
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07:57.52 | WIMPy | What linux-stable gave me, 3.2.0-rc1. |
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07:58.04 | WIMPy | 4 rcs behind. |
07:58.59 | WIMPy | But I do wonder why rc1 is considered stable. |
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08:01.38 | ChannelZ | is still in 2.6-land |
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08:03.54 | elliot98 | is there anyway anymore to submit a bug for * 1.4? |
08:04.19 | WIMPy | Yes, but noone wil care. |
08:04.19 | elliot98 | just upgraded to a new version and there seems some regression |
08:04.26 | WIMPy | l |
08:04.35 | elliot98 | version .37 to .42 |
08:04.47 | elliot98 | so versions are doing something |
08:04.55 | WIMPy | 1.4 and 1.6 are EOL |
08:05.21 | elliot98 | true true, but it's a stable system that's been on 1.4 for like forever |
08:06.08 | elliot98 | so what are the newer versions of 1.4 for? |
08:06.12 | WIMPy | Then why did you change it? |
08:06.25 | elliot98 | to fix another issue |
08:07.28 | elliot98 | that issue seems to be fixed in the later version, but another thing comes up |
08:08.35 | p3nguin | You'll be lucky if even a serious security vulnerability in 1.4 will receive attention. |
08:09.52 | elliot98 | but versions of 1.4 are still being released |
08:10.26 | elliot98 | is it possible to run 2 instances of asterisk on the same maching? |
08:11.06 | WIMPy | If you configure them carefully, yes. |
08:13.19 | elliot98 | but probably better to set up virtualbox or some sort of viruatlization...do virutalization and PRIs play well together? |
08:13.36 | p3nguin | If your issue is a security problem, it may get attention. Nothing else really matters. 1.4 has been in security mode for over six months already. |
08:13.37 | WIMPy | Probably not |
08:14.29 | WIMPy | I had serious problems last WE with a Digium PRI card. |
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08:15.24 | WIMPy | Err, vitout any virtualisation that is. |
08:15.57 | p3nguin | seri: I had to give up on it for now. I just changed subnet A to be the same as subnet B so I can assign router B to the dhcp clients. |
08:15.58 | elliot98 | so it could be the card, not the upgrade? |
08:16.40 | SeRi | sorry I was not much of help... :( |
08:16.48 | WIMPy | elliot98: ? |
08:17.11 | elliot98 | WIMPy: the mention of having problems with the PRI card |
08:17.56 | WIMPy | That was a dahdi or card issue. But why do you think tha could have anything to do with your upgrade? |
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08:18.22 | WIMPy | But I wouldn't use PRI with pre 1.8 versions. |
08:18.22 | elliot98 | WIMPy: we've reverted back and forth and it occurs only on the upgrade |
08:18.37 | elliot98 | diffing chan_dahdi.c with the old version yields a lot of changes |
08:18.50 | elliot98 | so someone's still in there tinkering with code |
08:19.03 | SeRi | p3nguin: http://www.ebay.com/itm/Polycom-2201-12560-001-SoundPoint-IP-560-SIP-Phone-/260913553715?pt=LH_DefaultDomain_0&hash=item3cbfa8e133 <---- g722 |
08:19.17 | WIMPy | Are both the old and the new issue PRI related? |
08:20.38 | elliot98 | unfortunately yes, so I can't just pick and choose some modules |
08:20.43 | elliot98 | one is dtmf |
08:20.58 | elliot98 | the other is that the card suddenly stops incoming calls |
08:21.20 | WIMPy | Isn't DTMF done by the card? |
08:21.24 | elliot98 | and need to run a "dahdi restart" to correct things |
08:21.37 | SeRi | p3nguin: http://www.ebay.com/itm/Plantronics-CS70-NC-Headset-System-W-HL10-Lifter-/250951359133?pt=LH_DefaultDomain_0&hash=item3a6dddda9d |
08:21.48 | elliot98 | it is? |
08:22.09 | elliot98 | how do you set that up? |
08:22.48 | WIMPy | Maybe teh Digium cards need the DSP module for that. |
08:23.19 | elliot98 | perhaps it has it, but I have yet find any documentation regarding setting it up |
08:24.14 | WIMPy | Hmm. It could be it's disabled by default. I think I stumbled upon something there. |
08:24.26 | WIMPy | It's a module parameter. |
08:24.46 | elliot98 | a module paramer in chan_dahdi.c? |
08:24.51 | elliot98 | *parameter |
08:25.09 | WIMPy | The kernel module for your card. |
08:25.24 | elliot98 | libpri or dahdi? |
08:25.49 | p3nguin | Domino's to the rescue. |
08:26.07 | elliot98 | but asterisk would still need to be configured somehow to work with it |
08:26.27 | WIMPy | With what? |
08:26.37 | SeRi | p3nguin: lol |
08:26.41 | SeRi | pizza? |
08:27.04 | elliot98 | WIMPy: with the onboard DTMF analysis |
08:27.22 | WIMPy | Why do you think it might not do so? |
08:27.34 | SeRi | p3nguin: should I put a bid on that 560? |
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08:27.45 | SeRi | I can move my 501 to a new home. :) |
08:28.34 | elliot98 | WIMPy: well, for starters, we don't want asterisk to also try to diagnose DTMF, because double DTMF can result |
08:29.14 | WIMPy | Off course I don;t KNOW what Digium does, but I would expect them to support their hardware in Asterisk. |
08:30.16 | elliot98 | seemingly, but how it's done...there must be some doc somewhere |
08:30.17 | p3nguin | Yes, pizza -- I'm pretty friggin' hungry. |
08:30.26 | SeRi | nom nom nom |
08:30.35 | p3nguin | What will you to with the 501? |
08:30.41 | elliot98 | and also, should PRIs be set to inband or rfc2832? |
08:31.00 | SeRi | Ill send it to its "original" home :P |
08:31.11 | WIMPy | elliot98: The doc is called modinfo |
08:31.20 | p3nguin | Back into the closet? |
08:31.23 | elliot98 | where? |
08:31.28 | elliot98 | in dahdi? |
08:31.39 | SeRi | nah ill send it to you. I did offered it to you some time ago. |
08:31.48 | WIMPy | And PRIs usually use inband DTMF. |
08:31.54 | SeRi | I just didnt come threw :( |
08:31.55 | WIMPy | At least if it's peer to peer. |
08:32.11 | SeRi | Ill place my bid at 130.00 |
08:32.13 | p3nguin | Will you benefit from the 560? |
08:32.18 | SeRi | Yes ser. |
08:32.19 | elliot98 | what else is there besides peer to peer? |
08:32.21 | SeRi | 6 lines and HD |
08:32.26 | WIMPy | No, it's the dirver. modinfo that one. |
08:32.36 | SeRi | I been wanting more lines. I actually need them |
08:32.42 | p3nguin | How many lines does the 501 have? |
08:32.46 | SeRi | 3 |
08:33.14 | WIMPy | elliot98: Terminal to switch. That would certainly be keypad. |
08:33.43 | p3nguin | Do you configure an account for each line key, or just use them for speed dials and stuff like that? |
08:34.25 | SeRi | speed dials and stuff like that.... But you could use each one for one individual account |
08:34.30 | elliot98 | keypad? need to bone up on PRI stuff |
08:35.13 | p3nguin | I'm not all that familiar with how Polycom deals with the accounts and line keys. |
08:35.45 | WIMPy | elliot98: Is the DTMF issue the old or the new one? And what's the other? |
08:35.47 | SeRi | Right now #1 Hot #2 Brother Office #3 Brother cell |
08:36.05 | elliot98 | the DTMF is ongoing, the new is issue occurs after the upgrade |
08:36.12 | SeRi | Its very simple actualy.. |
08:36.20 | SeRi | s/its/it's/ |
08:36.41 | SeRi | infobot: you drunk? |
08:38.57 | elliot98 | how does one make any changes in the settings after seeing the params in modinfo? |
08:39.24 | WIMPy | use them as parameter when you load the module, |
08:39.48 | WIMPy | Or put them in the modprobe.d somewhere. |
08:39.53 | elliot98 | in insmod? |
08:40.16 | WIMPy | If you're still using insmod, yes. |
08:40.41 | elliot98 | what is dtmfthreshold? |
08:42.30 | WIMPy | Probably either a level or a duration. No idea. |
08:43.34 | WIMPy | The default seems to be 1000. |
08:47.52 | WIMPy | The lower the value the more hits. |
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08:55.02 | pdtpatrick__ | Question .. which is more powerful... AEL or LUA ? |
09:06.45 | elliot98 | is there a way to restart a specific channel is Dahdi? |
09:07.43 | WIMPy | No |
09:08.00 | WIMPy | Did you only upgrade Asterisk or libpri as well? |
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09:11.43 | elliot98 | WIMPy: upgraded to latest libpri and dahdi |
09:12.13 | WIMPy | Then you should try the new Asterisk with the old libpri. |
09:13.45 | elliot98 | WIMPy: hehe...I only upgraded after the old libpri/dahdi was failing. Thought that was the problem, but aparently not so |
09:13.49 | elliot98 | sorry |
09:14.04 | elliot98 | only upgrade libpri/dahdi after asterisk started to fail |
09:14.18 | elliot98 | so tried all variations already |
09:14.22 | WIMPy | started to fail? What does that mean? |
09:14.33 | elliot98 | WIMPy: that the pri lines started to fail |
09:15.02 | WIMPy | You are aware that you need to rebuild Asterisk after changing libpri? |
09:15.49 | elliot98 | WIMPy: did a full reinstallation after the libpri update |
09:16.21 | elliot98 | the chan_dahdi.c code has significant changes there...don't know who to contact to find out what happened |
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10:24.26 | pdtpatrick__ | WIMPy: if ur PRI is active, once u load chan_dahdi.so .. it should pick up all available channels and resume working (if you disabled prior) right? |
10:25.17 | WIMPy | Disabled? How? |
10:25.57 | pdtpatrick__ | well the PRI was shutdown by provider for a bit so i unloaded the module and switched over to using SIP |
10:26.04 | pdtpatrick__ | but then looks like they will be turning it back on |
10:26.11 | pdtpatrick__ | so i should just have to load the same module |
10:26.27 | pdtpatrick__ | and the channels should (if active) should start working again right? |
10:26.44 | WIMPy | yes |
10:26.59 | pdtpatrick__ | cool deal. Thanks |
10:27.02 | WIMPy | Or you just load it and wait until it works. |
10:27.14 | WIMPy | It shopuld find out by itself. |
10:27.50 | pdtpatrick__ | will try that |
10:27.52 | kaldemar | and there's no need to unload the module. |
10:28.12 | pdtpatrick__ | well i didn't want all those channel unavailable messages to the screen |
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12:50.26 | helen_ | ChannelZ: No. :( |
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18:11.11 | ChannelZ | helen_: so what does the console say when you try to call? I don't see anything blaringly wrong in your config you posted so something else has to be at play |
18:14.55 | helen_ | ChannelZ: http://pastebin.com/MwNdT5C7 |
18:15.32 | ChannelZ | need more verbose output, but the first line pretty much means whatever device you were trying to call is not online or doesn't exist |
18:16.00 | helen_ | It definately exists. |
18:16.02 | helen_ | 1min... |
18:16.11 | ChannelZ | core set verbose 3 |
18:16.19 | helen_ | ChannelZ: It is set to that. |
18:16.32 | WIMPy | Check the device with 'sip show peers'. |
18:16.36 | ChannelZ | then your console is not outputting verbose because there should be more than that |
18:18.36 | helen_ | WIMPy: I told Tom to turn on his client |
18:18.53 | helen_ | and it's coming up unspecified under host on his extension. |
18:18.56 | helen_ | sighs |
18:19.11 | ChannelZ | see |
18:19.15 | ChannelZ | told you so |
18:19.35 | helen_ | ChannelZ: ? |
18:19.52 | helen_ | I don't get it... |
18:20.03 | helen_ | Everyone else has been having problems connecting apart from me. |
18:20.05 | ChannelZ | "the device doesn't exist or is not online" |
18:20.12 | helen_ | it exists |
18:20.19 | helen_ | just not online |
18:20.27 | ChannelZ | yes, like I said, again |
18:20.38 | ChannelZ | You can't call devices you can't reach |
18:20.46 | ChannelZ | so mystery there |
18:20.53 | ChannelZ | s/so/no/ |
18:21.04 | helen_ | It seems Tom dialed the voice maqil extension |
18:21.07 | helen_ | [Dec 11 21:20:10] NOTICE[10126]: chan_sip.c:20163 handle_request_invite: Call from '' to extension '4242' rejected because extension not found. |
18:21.14 | ChannelZ | No he tried to call 4242 |
18:21.26 | helen_ | yes |
18:21.36 | helen_ | which is the voice mail extension |
18:21.50 | helen_ | Call from '' |
18:21.50 | ChannelZ | apparently not, because it is "not found" |
18:21.54 | helen_ | what is that all about? |
18:22.08 | helen_ | ChannelZ: But extension 4242 is there |
18:22.12 | helen_ | because I can call it |
18:22.13 | ChannelZ | No user name/anonymous SIP |
18:22.25 | ChannelZ | Context is everything. |
18:22.42 | helen_ | Ok |
18:22.46 | helen_ | Forget that bit |
18:22.53 | helen_ | How do I make it so I can call Tom |
18:22.56 | helen_ | and he can call me? |
18:23.03 | helen_ | We need this urgently. |
18:23.04 | ChannelZ | Either that was a random person on the net trying that exten, or the other guy's device isn't configured right/your sip.conf isn't configured right and it's not matching a peer and not sending it to the right context. |
18:23.24 | ChannelZ | I don't even know who Tom is, that's new. Your old config referred to 101, 102, 103.. |
18:23.25 | helen_ | ChannelZ: YOu just said my configs are fine earlier. |
18:23.36 | helen_ | ChannelZ: Tom = 101 |
18:23.41 | helen_ | Helen = 102 |
18:23.51 | helen_ | Sorry for being confusing. |
18:23.54 | ChannelZ | sip show peer 101 |
18:24.22 | ChannelZ | or not even that. just do "sip show peers" |
18:24.23 | ChannelZ | Is 101's Host "unspecified"? |
18:25.18 | helen_ | ChannelZ: Doesn't say. |
18:25.26 | ChannelZ | What do you mean it doesn't say |
18:25.45 | helen_ | ChannelZ: THere is no host thingie in there. |
18:25.59 | ChannelZ | sip show peers ? |
18:26.20 | helen_ | Unspecified |
18:26.50 | ChannelZ | So his phone is not registering. You've got to fix that. Whether because his device is not configured right, or it's being blocked by a firewall on his or your side, I couldn't say. |
18:27.21 | helen_ | ChannelZ: Not my side |
18:27.42 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
18:27.46 | WIMPy | If that anonymous call to 4242 was from there, it must be missing the right configuration. |
18:29.11 | ChannelZ | What is 101? A softphone? |
18:29.24 | helen_ | ChannelZ: Yes |
18:29.29 | helen_ | It's my softphone |
18:29.40 | ChannelZ | I thought it was Tom's |
18:29.42 | ChannelZ | :P |
18:29.47 | helen_ | no? |
18:29.52 | helen_ | Tom's is 102 |
18:29.57 | ChannelZ | In event it's not configured right. |
18:30.02 | ChannelZ | then why did you say |
18:30.03 | ChannelZ | <helen_> ChannelZ: Tom = 101 |
18:30.04 | ChannelZ | <helen_> Helen = 102 |
18:30.14 | helen_ | I got it the wrong way around. |
18:31.19 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
18:31.26 | ChannelZ | sighs |
18:31.33 | helen_ | Is it me or is this a hard task? |
18:31.53 | ChannelZ | It's you. No offense meant |
18:32.01 | ChannelZ | It's only hard if you have bizzarre networking setups to work around |
18:32.16 | ChannelZ | So I don't know what device you're dialing, and which from. |
18:32.26 | Neptu | hej I have a headache trying to make work a sip provider like voipgain.com... |
18:32.34 | helen_ | I used to use Axon PBX but it's not free |
18:32.39 | helen_ | and it worked fine |
18:32.47 | helen_ | But asterisk is too complex for me. |
18:32.48 | ChannelZ | But if any of them say 'unspecified' in sip show peers, it's never going to work until you get those devices to register, or hard-code them with a static IP. |
18:32.55 | ChannelZ | AND the phone has to be configured right as well |
18:33.10 | [TK]D-Fender | helen_[Dec 11 21:20:10] NOTICE[10126]: chan_sip.c:20163 handle_request_invite: Call from '' to extension '4242' rejected because extension not found. |
18:33.12 | ChannelZ | What softphone is it? |
18:33.21 | helen_ | ChannelZ: Twinkle |
18:33.25 | [TK]D-Fender | the '' is a tip-off that the calling agent isn't identifying itself properly |
18:33.27 | helen_ | me and Tom use twinkle |
18:33.36 | [TK]D-Fender | For which I'd have to ask precisely how this call was dialed |
18:33.40 | [TK]D-Fender | ^^ |
18:33.41 | helen_ | All OptixLayer staff use twinkle or express talk |
18:33.53 | ChannelZ | And you've added a SIP account to it properly, told it the name (101 or 102) and the password, and the hostname/IP of your Asterisk box? |
18:33.54 | [TK]D-Fender | helen_: So what precisely did they put in the dial box? |
18:34.04 | Neptu | http://pastebin.com/vxRWjCRv |
18:34.12 | helen_ | [TK]D-Fender: ? |
18:34.23 | helen_ | WHy do you want to know that? |
18:34.42 | helen_ | Theres 20 of us |
18:35.16 | Neptu | anyone can give me hand creating my first voip peer... im with thos for 20 min and maybe i need a hand here |
18:35.19 | [TK]D-Fender | helen_: Because that might clearly explain the reson for the call being misidentified |
18:35.24 | [TK]D-Fender | helen_: details matter |
18:35.50 | Neptu | http://pastebin.com/vxRWjCRv -> this is my sip configuration they mention i should use a register command aswell but did not work |
18:35.51 | [TK]D-Fender | helen_: Of course it'd help is we saw the call attempt with SIP debug enabled. |
18:36.07 | helen_ | [TK]D-Fender: They've been calling me |
18:36.11 | helen_ | to test the line |
18:36.17 | helen_ | but haven't gotten through. |
18:36.33 | [TK]D-Fender | mepWell there is no register in there. Also almost no provider anywhere should ever be "nat=yes", and should be "type=peer" as well |
18:36.52 | [TK]D-Fender | helen_: Yes I know. I just asked you precisely what they put in the dial box to try that call |
18:37.02 | helen_ | [TK]D-Fender: 101 |
18:37.42 | *** join/#asterisk brdude (~brdude@c-24-130-20-142.hsd1.ca.comcast.net) |
18:39.05 | [TK]D-Fender | helen_: pastebin your peer configs and the CLI w/ SIP DEBUG enabled for the failed attempt |
18:39.53 | helen_ | [TK]D-Fender: SIP DEBUG enabled? |
18:40.19 | helen_ | peer configs as in extensions.conf and sip.conf? |
18:40.21 | [TK]D-Fender | helen_: "sip set debug on" |
18:40.36 | [TK]D-Fender | helen_: and yes, sip.conf masking only passwords |
18:43.29 | helen_ | [TK]D-Fender: http://pastebin.com/3ZTV40GL |
18:43.36 | helen_ | I removed the passwords |
18:44.35 | [TK]D-Fender | helen_: Ok, lets look at the failed call now |
18:45.52 | helen_ | [TK]D-Fender: Screenshot from a staffer. |
18:45.55 | helen_ | http://magix.megapowers.net/img/wm7YB.png |
18:46.28 | [TK]D-Fender | helen_: Don't care what their client says. Look at * CLI w/ SIP debug as I've requested a few times |
18:47.22 | WIMPy | Smells like a networking issue. |
18:48.00 | helen_ | [TK]D-Fender: SIP/2.0 401 Unauthorized |
18:48.06 | *** join/#asterisk kriegerod (~krieger@79.135.222.22) |
18:49.07 | [TK]D-Fender | helen_: Lets try this again.... |
18:49.11 | [TK]D-Fender | helen_: pastebin the call |
18:51.03 | helen_ | [TK]D-Fender: Theres too much to pastebin. |
18:51.36 | [TK]D-Fender | helen_: I doubt that Pastebin can handle a lot.... |
18:51.52 | *** join/#asterisk chasing`Sol (~cS@41.206.151.28) |
18:51.56 | [TK]D-Fender | helen_: I doubt that. Pastebin can handle a lot.... |
18:52.08 | [TK]D-Fender | Better... period makes all the difference in there. |
18:52.32 | helen_ | [TK]D-Fender: YOu said pastebin the call. |
18:52.46 | [TK]D-Fender | helen_: Yes, and I've seen pastebins well past 3000 lines. |
18:53.04 | [TK]D-Fender | helen_: So somebody telling me "too much" does hold any weight historically |
18:53.13 | helen_ | [TK]D-Fender: Can I just pastebin everything? |
18:53.25 | [TK]D-Fender | helen_: apparently, yes |
18:53.30 | [TK]D-Fender | ~pb |
18:53.30 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
18:53.34 | [TK]D-Fender | www.pastebin.com |
18:53.37 | [TK]D-Fender | slow bot.. |
18:53.39 | [TK]D-Fender | danit |
18:53.41 | [TK]D-Fender | dangit |
18:55.37 | helen_ | STUPID ROUTER |
18:55.39 | helen_ | Ok |
18:55.40 | Neptu | having the same problem to connecto to an external sip provider |
18:55.46 | helen_ | [TK]D-Fender: It goes off my terminal |
18:56.00 | [TK]D-Fender | get a bigger scroll-back buffer |
18:56.08 | helen_ | OMFG |
18:56.16 | [TK]D-Fender | Neptu: You should be looking at SIP debug as well. |
18:56.17 | WIMPy | Use tee |
18:56.29 | [TK]D-Fender | no need for "tee". |
18:56.48 | helen_ | I hope this isn't a password |
18:56.51 | helen_ | Really destroying SIP dialog ' |
18:56.54 | helen_ | after that |
18:56.59 | helen_ | I hope it isn't a password |
18:57.06 | Neptu | [TK]D-Fender: im on it jsut want to be sure my conf file are ok |
18:57.18 | [TK]D-Fender | Neptu: Can't be until you actually look. |
18:57.50 | [TK]D-Fender | Neptu: And aside from the corrections I already gave you |
18:58.30 | Neptu | [TK]D-Fender: Dial(SIP/0046851174812@voipgain) -> this is incorrect? |
18:58.55 | Neptu | i feel this might be one of the problems |
18:59.02 | [TK]D-Fender | Neptu: Not the preferred means, but should be OK. |
18:59.08 | [TK]D-Fender | Neptu: And feeling != loking |
18:59.11 | [TK]D-Fender | looking* |
18:59.18 | helen_ | Forget it. |
18:59.22 | helen_ | This is wasting my time. |
18:59.28 | helen_ | I don't have time to waste. |
18:59.58 | [TK]D-Fender | Pastebinning a call should have taken 30 seconds |
19:00.08 | [TK]D-Fender | Time is being wasted. |
19:00.44 | WIMPy | If you don't have lots of time to waste, Asterisk is not for you. |
19:00.45 | helen_ | [TK]D-Fender: AMd you didn't answer my question. |
19:00.55 | helen_ | WIMPy: So then what is for me? |
19:00.58 | [TK]D-Fender | helen_: What question? |
19:01.04 | helen_ | There are no other PBXs out there. |
19:01.19 | Neptu | [TK]D-Fender: the command is sip debug?? |
19:01.23 | helen_ | 13:56 < helen_> Really destroying SIP dialog ' |
19:01.23 | helen_ | 13:56 < helen_> after that |
19:01.23 | helen_ | 13:56 < helen_> I hope it isn't a password |
19:01.29 | [TK]D-Fender | Neptu: Look up |
19:01.31 | helen_ | ? |
19:01.49 | [TK]D-Fender | helen_: that is a statement, not a question. And now that you're considering it one, no. |
19:01.56 | WIMPy | helen_: I don;t know what your needs are. |
19:02.14 | Neptu | sip set debug on |
19:02.17 | Neptu | ok |
19:02.23 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
19:02.51 | helen_ | Does anyone know of any other PBX software? Don't you dare tell me to google. |
19:02.56 | helen_ | I already did that. |
19:03.40 | helen_ | Ok i'm getting stressed. |
19:03.45 | helen_ | Time out from the computer. |
19:03.56 | WIMPy | helen_: You will get used to that. |
19:04.27 | WIMPy | And wikipedia might have a comparative list of available software. |
19:05.27 | WIMPy | Using telephones is constantly getting more complicated. |
19:05.42 | [TK]D-Fender | So rather that showing a call which should take all of 30 seconds... instead we get paranoia, delays no output, and an immediate attempt to bail for something else. |
19:05.46 | WIMPy | Unfortunatly without giving more features. Rather the opposite. |
19:06.00 | Neptu | [TK]D-Fender: http://pastebin.com/3HSTnYHF -> Seems the dial is not finding the voipgain peer defined in sip.conf... |
19:06.03 | [TK]D-Fender | little speed bump now = full stop |
19:06.47 | Neptu | [TK]D-Fender: I did sip show users and i do not see the voipgain peer there |
19:06.54 | SeRi | [TK]D-Fender: I am now bidding for a 560 :) |
19:07.16 | [TK]D-Fender | Neptu: "sip show peers". |
19:07.39 | [TK]D-Fender | Neptu: And I did not see your corrected condfigs so I have no idea what you have in there right now. |
19:07.55 | Neptu | [TK]D-Fender: I get the same as sip show users 2 sip extension 1001 and 1000 |
19:08.20 | Neptu | [TK]D-Fender: I pastebin my confs w8 |
19:08.20 | [TK]D-Fender | SeRi: 5XX is an odd price-point in retail. If you get an agressive win against it it'd be a very nice phone though |
19:08.38 | [TK]D-Fender | Neptu: Never use "sip show users". Virtually useless. |
19:10.13 | SeRi | [TK]D-Fender: 124.00 is the bid up to. I dont want to go over 130.00. That includes free shipping. I have not been able to fins it less than 200.00. I think I am doing ok so far. the 321 came home for 24.50. |
19:10.26 | SeRi | s/fins/find/ |
19:10.31 | [TK]D-Fender | SeRi: Good up through 160 maybe... |
19:10.36 | SeRi | o wow! |
19:11.13 | SeRi | nice to know. I guess ill hang in there. Thanks [TK]D-Fender |
19:11.14 | [TK]D-Fender | SeRi: It is backlit, HD, etc... check out the retail price range between 335, 450, 560, 650 |
19:11.15 | Neptu | [TK]D-Fender: ai ai never use sip show users... this are my confs... http://pastebin.com/SXv1H4kD |
19:11.20 | [TK]D-Fender | SeRi: www.telephonydepot.com |
19:13.08 | SeRi | [TK]D-Fender: holly shit! I better hang in there! :D |
19:13.16 | [TK]D-Fender | Neptu: From the initial looks of things we should be seeing [voipgain] .... AND [1002]. thats the first and third entries missing. Something is broken there. |
19:13.25 | [TK]D-Fender | Neptu: Make sure you are reloading your config sproperly |
19:13.37 | [TK]D-Fender | Neptu: and that you're actually editing the proper file |
19:13.45 | Neptu | sip reload? |
19:13.49 | Neptu | extension reload? |
19:14.03 | [TK]D-Fender | Neptu: Should do it. "reload" should grab everything and I always do anyway |
19:14.41 | Neptu | doing it |
19:15.27 | Neptu | ok i check what can be broken at 1002 and voipgain... hope i can find it im not familiar with the sintax |
19:18.58 | [TK]D-Fender | Neptu: Syntax looks OK... tht's why I'm wondering if you are working in the actual proper config file, that permissions haven't gotten screwed up, etc |
19:19.15 | [TK]D-Fender | Neptu: "ls -la /etc/asterisk" <- PB |
19:19.29 | Neptu | w8 |
19:19.39 | Neptu | im doing everything as root |
19:19.50 | Neptu | and seems strange extension 1002 is not loaded... |
19:20.45 | [TK]D-Fender | Neptu: How was * installed? |
19:21.02 | Neptu | lrwxrwxrwx 1 asterisk asterisk 45 Dec 7 19:28 sip.conf -> /var/www/html/admin/modules/core/etc/sip.conf |
19:21.15 | Neptu | is the asterisk now version so should be all in place |
19:22.49 | WIMPy | helen_: Looking for new problems instead of solving existing ones? |
19:23.14 | [TK]D-Fender | Neptu: ... you're hand editing a FreePBX system,? |
19:23.44 | Neptu | ? |
19:23.50 | [TK]D-Fender | WIMPy: I'd leave this one alone until ready to concentrate on fixing something... |
19:24.02 | Neptu | im writing sip.conf and extensions.conf yes |
19:24.09 | helen_ | WIMPy: No |
19:24.17 | helen_ | Looking to switch back to Axon |
19:24.18 | [TK]D-Fender | Neptu: that symlink is a dead give-away that you are hand editing the sip.conf on a system that was used with FreePBX to manage it |
19:26.37 | Neptu | [TK]D-Fender: menaing i need to do it someother way?? |
19:27.01 | [TK]D-Fender | Neptu: Meaning you shouldn't be hand-editing file at all and should be using their GUI. |
19:27.17 | Neptu | mmmmmmmmm |
19:27.27 | Neptu | ok how i get to access the GUI? |
19:27.32 | [TK]D-Fender | Neptu: And what you've done already probably breaks it a bit as it is |
19:27.41 | [TK]D-Fender | Neptu: via a web-browser... |
19:28.16 | Neptu | [TK]D-Fender: user and pass by default? |
19:29.05 | [TK]D-Fender | Neptu: "The default username/password is freepbx/fpbx" |
19:29.49 | Neptu | ok im in |
19:29.56 | Neptu | let me familiarize with the enviroment |
19:30.03 | Neptu | and see if i make it work |
19:30.36 | [TK]D-Fender | Neptu: As I said, you need to undo everything you've dones.. including your dialplan.. this is likely to break things.' |
19:30.53 | Neptu | ai ai |
19:40.11 | Neptu | [TK]D-Fender: the web feels dam anoying... |
19:41.24 | [TK]D-Fender | Neptu: Not a common first reaction.... |
19:42.02 | [TK]D-Fender | Neptu: But if you want to run your system yourself it is very rewarding. You should re-install your PBX with a base non-gui setup and start from there |
19:42.30 | [TK]D-Fender | Neptu: Do backup your previous work though as those are a decent start |
19:43.47 | Neptu | i cna not even register now my softphone... |
19:43.56 | Neptu | and I feel i loosed control |
19:44.02 | Neptu | w8 |
19:44.21 | SeRi | p3nguin: you avail? |
19:44.31 | p3nguin | BOOM! |
19:46.39 | WIMPy | What did you break? |
19:46.49 | SeRi | p3nguin: I am having another issue with voip.ms reseller interface |
19:47.00 | p3nguin | wimpy: my Arch Linux |
19:47.32 | SeRi | how can a one account for pay pal fees? if the user pays 25.00 dollars they expect 25 dollars.... |
19:47.50 | SeRi | s/a one/one/ |
19:48.04 | p3nguin | If you pay $25, that's what you get. |
19:49.15 | SeRi | In their accoutn it shows that but in my pal there is 23. and some change |
19:49.55 | p3nguin | If someone adds funds via portal, why does it end up in your paypal account? |
19:50.31 | SeRi | The reseller portal links the pay pal account to your pay pal account |
19:53.59 | *** join/#asterisk PoWeRKiLL (~powerkill@77.127.116.143) |
19:57.14 | *** join/#asterisk nix8n82 (~hmg@71-32-140-32.chyn.qwest.net) |
19:57.54 | *** join/#asterisk beccara (~beccara@mail.ubergroup.co.nz) |
20:09.36 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-075-157-125.pools.arcor-ip.net) |
20:11.25 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
20:12.51 | Neptu | this is a mess |
20:15.54 | SeRi | p3nguin: Is a bit confusing but looks like I got it figured out. |
20:16.06 | SeRi | all deductions happen at the users interface |
20:17.11 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net) |
20:18.15 | KingDavidNYC | hello guys, I am scratching my head here because verbose stops after the first call |
20:19.02 | p3nguin | seri: Does that mean that if a person deposits $25, it will show less than $25 in the interface? |
20:19.26 | SeRi | No. I have an open a ticket for that. |
20:19.46 | SeRi | another bug. voip.ms has to find a way to account for that for resellers |
20:20.16 | p3nguin | It's almost like you are the very first person to ever try to use the reseller interface. |
20:21.11 | SeRi | no shit.... I feal the same way. I cant even belive that they would never thought about that. I guess people just didnt bother for a reason |
20:24.42 | SeRi | they only thing I can come up with is to include a small extra fee under the monthly fee to account for pay pal fees. but that would be bull shit and corporate like. verzion math at is best. |
20:26.50 | p3nguin | Let's say they don't make a deposit every month. You're either going to be undercharging or overcharging. |
20:27.40 | p3nguin | Or if they make a deposit of more or less than you've accounted for, the same could be true. |
20:28.01 | SeRi | Indeed |
20:28.38 | SeRi | well lets see how it goes. I am testing the grounds with my family.... I have a few monkeys I guess.... |
20:30.34 | KingDavidNYC | can you guys help me? this doesn't make sense.. when I call via a DID, I get verbose, but when I call via x-lite, the first call gets verbose and then no more verbose from calls made to that phone |
20:32.17 | WIMPy | KingDavidNYC: The only thing I can tell you is that I have the same issue since about a week. |
20:32.34 | WIMPy | What version are you using? |
20:32.35 | KingDavidNYC | I am using 1.6.2.20 |
20:33.06 | KingDavidNYC | WIMPy: it doesn't make sense |
20:34.15 | [TK]D-Fender | KingDavidNYC: Verbose TO a phone? * can't be exluding based on some destinatin. |
20:34.28 | [TK]D-Fender | KingDavidNYC: You should be showing us both attempts |
20:34.51 | WIMPy | KingDavidNYC: Hmm, that's definitely older than my issue. |
20:34.52 | KingDavidNYC | I mean, when I initiate calls with x-lite this happens |
20:35.10 | WIMPy | But for me it fails as soon as the first call starts. |
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20:35.55 | KingDavidNYC | the first call fails, and then subsequent calls dont seem to arrive to the server |
20:36.10 | [TK]D-Fender | KingDavidNYC: that's the caller, not the callee |
20:36.12 | WIMPy | If it happens I don;t even get any output from things like 'sip show peers' or 'core show application dial' until I restart the remote console. |
20:36.30 | [TK]D-Fender | KingDavidNYC: Details and debug please |
20:36.36 | WIMPy | The calls don;t work or you just don;t see them? |
20:37.05 | WIMPy | And why have I been unable to press ' for quite some months now? |
20:37.32 | p3nguin | Maybe your ' moved. |
20:38.41 | WIMPy | I didn't change keyboard this cenury. |
20:39.29 | Neptu | [TK]D-Fender: reinstaling * from scratch... |
20:40.37 | Neptu | recomend freepxb or asterisk gui?? |
20:41.14 | KingDavidNYC | I got it!! I was sending the calls from the x-lite phone to a different context! |
20:41.39 | Neptu | KingDavidNYC: congratz! |
20:41.45 | [TK]D-Fender | Which should have shown up in CLI like everything else |
20:42.09 | p3nguin | neptu: No. |
20:42.29 | KingDavidNYC | TK: yes TK, that's what has me puzzled |
20:43.13 | KingDavidNYC | TK: but I am not going to bother....got a route to have ready in a couple of hours |
20:43.19 | KingDavidNYC | thanks a lot guys |
20:44.32 | p3nguin | Paper route? |
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20:48.02 | KingDavidNYC | wholesale route |
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20:49.05 | KingDavidNYC | nowadays asterisk people get so low paid that we have to take anything that comes our way :) |
20:50.39 | [TK]D-Fender | "wholesale route" sounds like about 10 minutes work on the outside... |
20:51.45 | KingDavidNYC | WEll... they had a2billing, and since I did not want to spend the time learning it, I am now spending the time writing some basic least cost routing and routing scripts |
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20:52.36 | KingDavidNYC | I figure they may come in handy next time |
20:53.33 | p3nguin | I need to do that, too. |
20:54.03 | p3nguin | I need to compare the cost based on destination and route the call accordingly. |
20:54.48 | KingDavidNYC | I am using mysql+php |
20:57.39 | p3nguin | I don't code, so I'll end up hiring someone to do it for me if I can't do it with basic sql commands and a unix-like shell. |
20:58.38 | KingDavidNYC | I might jsut give you my code |
21:05.18 | p3nguin | seri: Does a chili pepper get more intense spiciness as it gets older and dries out, or does it get milder? |
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21:28.17 | elliot98 | gives a double arm wave |
21:28.46 | elliot98 | ok, so I need to submit a bug for Asterisk 1.4 |
21:28.51 | elliot98 | is that even possible? |
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21:34.48 | patrickod | I'm having trouble debugging two soft clients on the same network. One is registering and functioning correctly the other keeps getting 408 errors and timeouts |
21:35.53 | patrickod | I can switch credentials and the same one still works whil the other continues to fail |
21:43.54 | patrickod | scratch that. For some reason it's working now. weird |
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22:23.11 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-rc2 (2011/11/15), 1.8.7.2 (2011/12/08), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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22:27.09 | Neptu | hej trying freepbx and i think i almost have it ready but still my trunk seems is not connecting properly... http://pastebin.com/FXUPWdpE |
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22:28.05 | Neptu | any ideas? |
22:28.24 | beccara | anyone here got post dial delay logging with asterisk 1.8? |
22:29.29 | WIMPy | beccara: Can you describe what you're after? |
22:29.52 | beccara | I'm wanting to log post dial delays into the CDR's |
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22:30.10 | WIMPy | What delay? |
22:30.37 | WIMPy | From dialling to getting a response? To answer? Or what exactely? |
22:30.50 | beccara | PDD, The time it takes from sending the invite to a trunk carrier and the carrier responding |
22:31.20 | beccara | the only thing floating around is patchs for asterisk 1.2 |
22:31.44 | p3nguin | ~freepbx |
22:31.44 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
22:31.47 | p3nguin | neptu: ^^^^^^^^^^^^^ |
22:31.51 | WIMPy | I don't see any way to find out about that. |
22:32.09 | beccara | it's been done, it's possible |
22:32.24 | beccara | it was under 1.2 and FPBX seems to have those sorts of options aswell |
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22:34.50 | Neptu | p3nguin: hej donno how to solve this I added the registration chain but the trunks are not working... |
22:35.32 | p3nguin | ~freepbx |
22:35.32 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
22:35.34 | p3nguin | neptu: ^^^^^^^^^^^^^ |
22:35.56 | p3nguin | This is a FreePBX concern. And that does not concern us here in #asterisk. |
22:36.40 | Neptu | ok, the problem is the same... need to configure a trunk and have it operational |
22:36.44 | p3nguin | Now did you understand what I'm saying? |
22:36.53 | Neptu | yep |
22:37.15 | p3nguin | So you'll go over there and ask for help configuring your system by FreePBX, yes? |
22:37.34 | Neptu | can u help me configure the trunck over sip.cfg? |
22:37.56 | p3nguin | What part of "FreePBX is unable to be supported here" are you having the most trouble with? |
22:38.06 | p3nguin | It's not an asterisk issue. |
22:38.11 | p3nguin | You aren't using asterisk. |
22:38.15 | Neptu | im not talking about freeçPBX i mean from editing the files |
22:38.28 | Neptu | ;) |
22:38.33 | p3nguin | You're using FreePBX. You don't edit files when you're using FreePBX. |
22:38.56 | Neptu | well they are included as sip_additions.cfg as far as I saw |
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22:39.05 | Neptu | so mix configurations are possible |
22:39.12 | p3nguin | Go ask in the appropriate channel. This is an asterisk channel. |
22:40.21 | p3nguin | You don't go to an automobile repair center and ask for help laying carpet in your bedroom. |
22:41.26 | Neptu | p3nguin: Im asking you to help me to fix sip.cfg file WITHOUT freepbx to have a peer configration... that is all |
22:41.32 | p3nguin | Can't be done. |
22:41.38 | Neptu | ? |
22:41.44 | p3nguin | You use FreePBX. That's all there is to it. |
22:41.54 | WIMPy | If you start editing that, it will the reason why it fails. |
22:41.58 | p3nguin | If you didn't use FreePBX, I'd help you edit sip.conf. |
22:42.07 | Neptu | ok if i install a fresh version of asterisk without freepbx can u help me then? |
22:42.08 | p3nguin | But You don't use Asterisk, so I can't help you. |
22:42.32 | Neptu | ok give me 10 min i get a new fresh compy without freepbx |
22:42.37 | Neptu | ;) |
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22:45.13 | beccara | seems like asterisk just doesn't have PDD, Freeswitch does http://wiki.freeswitch.org/wiki/PDD |
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22:45.54 | p3nguin | You're trying to measure the post-dial delay? |
22:46.18 | beccara | yep |
22:46.21 | beccara | in cdr's |
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23:09.43 | SeRi | damn kids where going crazy dialing numbers. |
23:09.56 | SeRi | I am going to have to restrict outbound calls to those in the db only |
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23:16.29 | SeRi | p3nguin: you avail? |
23:16.50 | p3nguin | sure. |
23:17.12 | SeRi | http://www.ebay.com/itm/250951359133?ssPageName=STRK:MEWAX:IT&_trksid=p3984.m1438.l2649 |
23:17.14 | p3nguin | Do you want them to be able to dial anything at all? |
23:17.17 | SeRi | what do you think? |
23:17.50 | SeRi | p3nguin: no all I want them to dial is a 4 digit ext and 911 |
23:18.07 | p3nguin | Create a new dial plan for their phone. |
23:18.14 | SeRi | got it |
23:18.48 | p3nguin | Very simple. Limit it only to patterns or explicit extensions you want them to dial. Do not include other contexts that have the ability to dial other extensions. |
23:19.03 | SeRi | got it. |
23:19.09 | SeRi | I think I am on the right track |
23:19.12 | SeRi | Thanks |
23:19.16 | SeRi | did you see my link? |
23:19.18 | p3nguin | For example, [kids] context, include "internals" and an extension for 911. |
23:19.29 | SeRi | indeed |
23:19.35 | SeRi | +1 |
23:19.52 | p3nguin | If you like that sort of headset and it isn't too expensive for you, grab it. |
23:20.39 | SeRi | seems resonable... I am just to sure about the in ear thing... |
23:20.56 | SeRi | my ears dont like that |
23:21.07 | SeRi | but for 40 dollars I migth give it a spin |
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23:25.19 | p3nguin | You can always try to resell it if you don't like it. |
23:26.07 | SeRi | true. |
23:26.16 | SeRi | I told my brother to buy it for me :P |
23:26.19 | SeRi | lol |
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23:48.14 | p3nguin | I just keep running into limitations of Vyatta. |
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23:49.14 | p3nguin | First the lack of source-based routing and/or policy-based routing, now it's a problem with hairpin NAT (reflective NAT) not working when there's a dynamic WAN IP address. |