| 00:02.17 | *** join/#asterisk etnos (n=etnos@adsl-2-204-26.mia.bellsouth.net) |
| 00:05.11 | *** join/#asterisk Iamnacho (n=Iamnacho@ip174-70-146-115.ks.ks.cox.net) |
| 00:06.11 | carrar | Polycom XML goodness! |
| 00:12.43 | bmoraca | ariel_, you around for a minute? |
| 00:15.36 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 00:35.38 | *** part/#asterisk DarkFibre (n=Administ@64.129.95.226) |
| 00:35.54 | ariel_ | bmoraca, I am now |
| 00:36.31 | bmoraca | i'm having a weird problem with my as5400...was wondering if i might be able to run it past you |
| 00:38.09 | bmoraca | i'm drawing a blank on google and every other forum i know to ask in...but for some reason, if a call comes in off my PRIs and has no calling party number, it fails to match any of my dialpeers |
| 00:38.47 | ariel_ | is it coming in as a unknown? |
| 00:38.58 | bmoraca | no, it's coming in as blank |
| 00:39.08 | ariel_ | I have not seen blank for a pri |
| 00:39.43 | Corydon76-dig | I have |
| 00:39.43 | ariel_ | can you duplicate it? |
| 00:40.03 | bmoraca | from my dialpeer debug, i see "Calling Number=, Called Number=2095389297" and q931 debug also confirms that nothing comes across in calling party info |
| 00:40.04 | Corydon76-dig | If it comes in as blank, it should be routed to the "s" extension |
| 00:40.32 | Corydon76-dig | Oh, calling number, not called number? |
| 00:40.34 | ariel_ | Corydon76-dig, as5400 has no s |
| 00:41.02 | bmoraca | it's really weird |
| 00:41.12 | bmoraca | and i can't find any reference to it at all |
| 00:41.28 | Corydon76-dig | Are these telco PRIs or Asterisk PRIs? |
| 00:41.28 | ariel_ | it should match your called Number |
| 00:41.46 | bmoraca | telco PRIs |
| 00:42.00 | Corydon76-dig | Call your telco then |
| 00:42.39 | bmoraca | Corydon76-dig, it's not a telco problem. it's an as5400 problem. unless calling party should come through as something other than blank if it's blocked |
| 00:43.05 | bmoraca | ariel_, they do match called number...as long as the calling party number is set |
| 00:43.41 | ariel_ | Cisco's OS is expecting something |
| 00:44.05 | Corydon76-dig | wonders why we're discussing non Asterisk equipment on this channel at all |
| 00:46.32 | bmoraca | it was slow and i know that ariel_ uses them so i saw no harm in asking |
| 00:49.28 | ariel_ | I am looking at my admin doc's for the units |
| 00:49.58 | bmoraca | i'm not sure, but i may have solved it...i had assumed that my outbound dialpeer would also match allinbound calls, but it appears that IOS doesn't treat a trunkgroup destination the same as a port destination as far as matching inbound calls goes |
| 00:50.43 | bmoraca | i added some dial-peers which specifically mention my PRI ports as their destination, and i haven't seen the issue again...hopefully that was it |
| 00:51.00 | bmoraca | i don't have a blocked number with which to test, unfortunately |
| 00:51.42 | ariel_ | yes you can setup slots |
| 00:52.43 | bmoraca | i swear...cisco couldn't have made dial-peer more obtuse if they'd tried |
| 00:53.42 | ariel_ | that is why I pair them up to asterisk systems in my locations for routing |
| 00:54.26 | bmoraca | my asterisk system is routing...this is just a media gateway...but for some reason, it's not behaving properly when a call comes in with no ANI |
| 00:59.31 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 00:59.53 | ariel_ | Cisco gives support for this, you can call them and they will assit |
| 01:02.00 | securevoip | is there a default 'torture' script included in asterisk? |
| 01:02.39 | p3nguin | Not that I know of. |
| 01:02.52 | ChannelZ | Where is the default order of columns in Master.csv documented? The sample in cdr_custom is not quite matching up to my actual log |
| 01:03.04 | p3nguin | http://www.voip-info.org/wiki/view/Asterisk+billing |
| 01:03.08 | ariel_ | search for ex girlfriend |
| 01:03.15 | ariel_ | for the torture |
| 01:03.59 | p3nguin | I'm always searching for my next ex-girlfriend. |
| 01:04.13 | p3nguin | to torture. |
| 01:04.24 | p3nguin | ;) |
| 01:04.54 | ChannelZ | thanks |
| 01:05.04 | ChannelZ | * docs are a scattered mess |
| 01:05.08 | ariel_ | a few years ago there was a ivr sample made for sending dialer calls to a round robin hell |
| 01:06.18 | bmoraca | well, thanks for your help, ariel |
| 01:06.21 | bmoraca | it's go-home time |
| 01:08.07 | ariel_ | bmoraca, if you have a sample of what you used, I can show it to my in house Cisco rep, but I need to be in the office and take it to him directly |
| 01:17.15 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
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| 02:46.53 | pentanol | hi guys, anybody use module for skype integration? |
| 02:48.05 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 03:07.24 | *** join/#asterisk jkroon (n=jkroon@dsl-244-25-144.telkomadsl.co.za) |
| 03:08.03 | jkroon | hi all. where can I find a good, concise guide on the SIP protocol? |
| 03:08.41 | jkroon | in particular I'm looking for something that'll help me assist trouble-shooting why calls potentially gets dropped given a SIP trace. |
| 03:21.26 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:21.57 | *** join/#asterisk aidinb (n=Aidin@24-176-209-6.dhcp.lnbh.ca.charter.com) |
| 03:28.06 | jaytee | jkroon, try SIP Demystified by Gonazalo Camarillo |
| 03:28.29 | jkroon | jaytee, thanks. |
| 03:29.07 | *** join/#asterisk MaliutaLap (n=biteme@202.124.73.65) |
| 03:34.33 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:34.52 | jkroon | in the meantime, http://pastebin.co.za/76510 ... does that look sane? I'm trying to figure out the trigger for that 183 Session Progress @ line 150 |
| 03:36.40 | *** part/#asterisk dc3 (n=dc3@tinderbox.complexminds.com) |
| 03:36.52 | jkroon | nm, it seems my script doesn't get all the SIP packets out of the dump just yet, from what I understand 183 Session Progress is essentially a "ringing" indication? |
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| 03:57.06 | sbrath | Do the nf_conntrack_sip and nf_nat_sip moduless for ubuntu work, or do I really have to forward a bunch of ports to get SIP to work where my asterisk is behind a NAT? |
| 03:58.06 | sbrath | I've setup all the extenip, localnet, and nat= stuff, and I can establish a call, but I have no audio. |
| 04:05.36 | *** join/#asterisk freetown2 (n=freetown@ipvpn098241.netvigator.com) |
| 04:05.58 | *** join/#asterisk EyeBaw (n=I@cpc3-sgyl27-2-0-cust52.sgyl.cable.virginmedia.com) |
| 04:06.40 | freetown2 | hello, has anybody tried out the snom 8xx phones? |
| 04:06.47 | ManxPower-work | sbrath: did you forward the ports? |
| 04:06.53 | ManxPower-work | ~phones |
| 04:06.54 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones. Do not consider Cisco phones. Ever. |
| 04:07.11 | ManxPower-work | Well that was not very helpful. |
| 04:07.23 | Mango | chuckles at the last line. |
| 04:07.25 | freetown2 | okay okay, has anybody tried out 802.11n phones then? |
| 04:07.46 | freetown2 | fighting with Datacraft |
| 04:07.53 | sbrath | I haven't forwarded the ports, as I don't want to open a big hole... |
| 04:08.41 | sbrath | I guess I could do a iptables rule to forward for just a VIP and the ports, but then I'm basicly just putting the Asterisk on the live net... Is that really the only way, their's no magical nf_conntrack_sip or something? |
| 04:09.17 | sbrath | ~sipnat |
| 04:09.18 | infobot | [~sipnat] Quick guide on configuring * + 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 |
| 04:09.52 | *** join/#asterisk lordmortis (n=lordmort@203-206-67-161.dyn.iinet.net.au) |
| 04:10.15 | freetown2 | sbrath, there is a sip module for netfilter available |
| 04:10.29 | freetown2 | what are you running? |
| 04:10.48 | sbrath | I see nf_conntrack_sip and nf_nat_sip on ubuntu ( I'm on ubuntu 9.04 LTS) |
| 04:10.54 | *** join/#asterisk lordmortis (n=lordmort@203-206-67-161.dyn.iinet.net.au) |
| 04:11.15 | freetown2 | that should handle things no? |
| 04:11.19 | jkroon | sbrath, nuke those modules. |
| 04:11.38 | jkroon | sbrath, they randomly drop actual SIP packets. |
| 04:11.43 | sbrath | I removed them . |
| 04:11.54 | freetown2 | openbsd for sip natting then? :P |
| 04:12.04 | sbrath | Realize that my ip-tables firewall is a seperate device from the asterisk server.... |
| 04:12.37 | sbrath | I was considering loading freeswitch on a net connected box and using is as a Border Session Controller, and doing it that way. |
| 04:12.45 | freetown2 | just punches a hole via openvpn for sip |
| 04:13.14 | freetown2 | but i guess you don't have that freedom eh sbrath ? |
| 04:13.19 | sbrath | I'm trying to egress SIP to les.net and simplesignal |
| 04:13.22 | jkroon | still, i got scared when I saw this in my logs: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=00:15:17:62:d7:90:00:d0:02:13:e8:0a:08:00 SRC=41.246.214.187 DST=196.35.70.140 LEN=446 TOS=0x00 PREC=0x00 TTL=56 ID=38120 PROTO=UDP SPT=5060 DPT=506 LEN=426 |
| 04:14.05 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:14.12 | jkroon | hmm, for some reason the DPT got screwed, think I changed terminal size but you get the idea. |
| 04:14.21 | sbrath | jkroon: that's probably the rate limit rules in ufw, I hate them and always remove them, and the random dropped message go away. |
| 04:14.32 | freetown2 | are grandstream phones still on the lower end of the quality bracket? I thought their recent models were good? |
| 04:14.33 | jkroon | no, they're not. |
| 04:14.53 | jkroon | that is nf_ct_sip dropping SIP packets that it doesn't know how to cope with. |
| 04:15.04 | sbrath | ahh |
| 04:15.07 | jkroon | and i don't use ufw anyway. |
| 04:15.31 | sbrath | well the netfilter people do say they didn't have time to learn SIP and the it was too complicated for them. |
| 04:15.42 | freetown2 | custom built rules in interfaces jkroon ? |
| 04:15.56 | jkroon | straight from the command line and you know it. |
| 04:16.00 | jkroon | :p |
| 04:16.10 | jkroon | iptables -A INPUT ... etc style stuff. |
| 04:16.24 | freetown2 | hahaha, however, that does not persist.... |
| 04:16.34 | freetown2 | it ain't service iptables save you know :( |
| 04:16.40 | jkroon | it does if you then run iptables-save and use iptables-restore :) |
| 04:16.57 | freetown2 | that's true...and you call from interfaces right? |
| 04:17.08 | jkroon | from interfaces? |
| 04:17.18 | freetown2 | <PROTECTED> |
| 04:17.20 | sbrath | jkroon: are you on CentOS |
| 04:17.26 | jkroon | no i'm not. |
| 04:17.37 | freetown2 | he is not...ubuntu he said...if he refers to ufw... |
| 04:17.52 | freetown2 | or is it debian...ah... |
| 04:18.02 | jkroon | no, sbrath refered to ufw ... debian is closer, but i'm actually using gentoo. |
| 04:18.17 | freetown2 | OH. My apologies. |
| 04:18.22 | sbrath | ohh.. well that makes sense then... |
| 04:18.27 | jkroon | i'd rather use debian if i'm to use a binary distro. and no apology required :) |
| 04:19.03 | jkroon | anyway, back to asterisk. I've got a massive SIP debug (50+MB) out of which I'm trying to rip all SIP packets related to a specific Call-ID |
| 04:19.12 | sbrath | I'm supprised that you use iptables, and arn't writing the assembler code to directly tweak the kernel rules ........ ( Ahh Gentoo build from source linux ) |
| 04:19.31 | freetown2 | sbrath, very funny |
| 04:19.33 | jkroon | is there a tool that can do this or should I just hack away at my regex to figure out why it's missing certain packets? |
| 04:19.35 | sbrath | I used to manage the mirrors servers for gentoo :) |
| 04:19.44 | jkroon | lol @ sbrath |
| 04:20.24 | jkroon | nah, i just enjoy the OS :p |
| 04:20.35 | freetown2 | jkroon, sorry, no idea, i gave up on natting and just went for direct connections or asterisk handling the call |
| 04:21.01 | sbrath | I had one emerge that went south, and I never had time to go back and rebuild.... Had to slap in a ubuntu disk and get back to work .... |
| 04:21.09 | jkroon | i hate nat, and my problem atm has nothing to do with NAT, I am, however, after a tool to extract those SIP traces. |
| 04:21.15 | sbrath | I tried to emerge upgrade to the next release. |
| 04:21.42 | jkroon | except gentoo doesn't really have "releases" per say, but anyway. |
| 04:21.54 | freetown2 | tcpdump? :P |
| 04:21.59 | sbrath | You know their might be a something in wireshark? |
| 04:22.25 | sbrath | I'd bet their is something in there that will glue it all back together. |
| 04:22.55 | sbrath | just make sure that your traces are captured with a big enough window size so you arn't loosing the end of each packet. |
| 04:24.35 | jkroon | hehe, already got the asterisk textual output, so trying to parse that log. |
| 04:25.01 | ManxPower-work | does it have any internal addresses that look like they should be external addresses in it? |
| 04:25.05 | jkroon | and yes, wireshark does have something, but it requires me to have the full trace locally. I deal with about 2GB worth of SIP traffic a day. |
| 04:25.49 | sbrath | Sorry, until I can master this damn natsip issue, I'm toast |
| 04:26.16 | jkroon | which is remote, don't want to capture+copy that every day, would cost me a small fortune. |
| 04:26.35 | jkroon | sbrath, ManxPower-work should be able to sort you out if you can give him enough (accurate) information. |
| 04:26.48 | jkroon | he's very clued up. |
| 04:27.21 | ManxPower-work | jkroon: I bet the problem is with his iptables and I'm very out of practice with iptables and they are not exactly trivial to set up correctly. 8-| |
| 04:27.43 | ManxPower-work | sbrath: Do you have Asterisk -> Linux NAT router -> Internet? |
| 04:27.45 | jkroon | if you can confirm it's firewall i'll assist with the iptables itself. |
| 04:27.53 | freetown2 | sbrath, with nat, i have only ever got one way sound working. You can hear the other side but they cannot hear you. do you have to nat? |
| 04:27.56 | jkroon | my sip is useless again. |
| 04:29.02 | jkroon | ManxPower-work, on second thought, you're probably right already (as per usual). |
| 04:29.03 | jkroon | sbrath, if you can get away from the nat that would be first prize (trust me) |
| 04:29.29 | jkroon | if you can't, then the second best option is to set a smaller range of rtp ports (set it to your concurrency limit + a bit for spare) then only forward those ports. |
| 04:29.45 | jkroon | but whatever you do, stay the hell away from nf_nat_sip please. |
| 04:30.37 | ManxPower-work | you don't want to get fancy with NAT |
| 04:30.38 | jkroon | or alternatively, i found that IAX/2 works very well over NAT. but not everybody has that as an option. |
| 04:31.24 | *** join/#asterisk Znuff (n=ibm86@2001:0:53aa:64c:2c26:2f82:a65a:7c98) |
| 04:31.29 | jkroon | ManxPower-work, frankly, i think someone should just fix the nf_ct_sip module, I'd do it but my SIP knowledge is next to useless. |
| 04:31.38 | Znuff | Hi. |
| 04:31.56 | *** join/#asterisk hluesea (n=hulusika@88.239.32.245) |
| 04:32.10 | jkroon | that would allow one to simply allow outbound port 5060 and have everything else "just work". |
| 04:32.11 | Znuff | Any ideas how could I recover a saved password for a SIP account, saved on a dumb device that can't export its configuration or save it in some readable format? |
| 04:32.42 | jkroon | can you access a web interface on it? |
| 04:32.47 | Znuff | Yup |
| 04:32.52 | jkroon | go to the screen with the password and view source. |
| 04:33.03 | Znuff | <PROTECTED> |
| 04:33.03 | Znuff | :) |
| 04:33.11 | jkroon | let me guess - grandstream? |
| 04:33.14 | Znuff | yup |
| 04:33.20 | jkroon | tcpdump ? |
| 04:33.37 | jkroon | although, that'll probably give you an md5 ... can you force it to try and auth with plaintext? |
| 04:33.52 | Znuff | hmmm, no idea |
| 04:33.57 | jkroon | or just check your asterisk config. |
| 04:34.35 | Znuff | I don't own that asterisk :) |
| 04:35.03 | Znuff | I was thinking of running an asterisk server on my box |
| 04:35.19 | Znuff | And replacing just the SIP server in the web interface |
| 04:38.08 | drmessano | and keep replacing the password with random and running a reload via AGI until it registers? |
| 04:38.16 | drmessano | Fantastic idea.. see ya in 11 years |
| 04:38.38 | *** part/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) |
| 04:39.03 | freetown2 | nobody's hot on wireless eh? |
| 04:39.12 | freetown2 | or not wireless-n anyway... |
| 04:39.15 | *** join/#asterisk chendy_ (n=chatzill@113.91.37.74) |
| 04:39.51 | Znuff | wish this bloody thing would have plaintex auth |
| 04:39.53 | Znuff | text |
| 04:40.52 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 04:41.27 | Znuff | how do I increase verbosity in asterisk? |
| 04:41.36 | Znuff | haven't use it in quite some time now |
| 04:43.51 | dlynes | Znuff, core set verbose 100 |
| 04:44.33 | Znuff | any chance I could see the hash of the password? |
| 04:48.48 | sbrath | ManxPower-work: asterisk -> ( internal routers ) -> iptables firewall doing Nat -> Internet |
| 04:49.43 | ManxPower-work | sbrath: no nat or firewall on the internal routers? |
| 04:49.56 | Znuff | SIP uses SHA for passwords, right? |
| 04:50.01 | sbrath | internal routers are just Cisco's routing around the 10.x space. |
| 04:50.20 | sbrath | only NAT is on the external linux box. |
| 04:50.44 | ManxPower-work | sbrath: on the iptables box doing NAT set up some rules that log the packets you think you should be seeing. See if what is being logged is what you expect. |
| 04:51.04 | ManxPower-work | (I use a Cisco 1720 router for my NAT/Router) |
| 04:51.22 | sbrath | Can you confirm what I should have for sip.conf and such. |
| 04:51.55 | sbrath | I have the following: localnet=10.0.0.0/255.255.255.0 |
| 04:51.57 | ManxPower-work | sbrath: pastebin your sip.conf. Just a warning, I'm here when I'm waiting for a process to complete in another window, then I fire off another process and come back here. |
| 04:52.02 | sbrath | and externalip=1.2.3.4 |
| 04:52.09 | sbrath | externip= |
| 04:52.25 | sbrath | canreinvite=no |
| 04:52.27 | ManxPower-work | you have your real externip of course? |
| 04:52.51 | ManxPower-work | sbrath: ALL your hosts start with 10.0.0.X |
| 04:53.05 | freetown2 | how is asterisk on solaris/opensolaris now? |
| 04:53.55 | sbrath | ManxPower-work: yes |
| 04:54.01 | freetown2 | ~opensolaris |
| 04:54.09 | sbrath | the asterisk box is on the 10.0.0.x net |
| 04:54.12 | jarrod | wow |
| 04:54.16 | freetown2 | ~solaris |
| 04:54.16 | infobot | methinks solaris is like Star Trek movies; only the even-numbered ones are good |
| 04:54.27 | sbrath | I was also going to add qualify=yes to the sip config |
| 04:54.37 | jarrod | any way to protect asterisk from hosing up when cdr mysql server crashes |
| 04:54.38 | ManxPower-work | I would add that just to be safe. |
| 04:55.12 | drmessano | freetown2: Pretty good if you use the Solaris install to download CentOS, burn the ISO to CD, boot the CD, install, and then run Asterisk on top of that |
| 04:55.47 | freetown2 | drmessano, hahaha |
| 04:56.09 | drmessano | </obligatory OS troll> |
| 04:56.37 | freetown2 | heh. shall i start with gcc extensions then :P |
| 04:57.56 | drmessano | I was going to try OpenSolaris once |
| 04:57.59 | drmessano | then a second time |
| 04:58.10 | drmessano | But both times I requested a disc they sent me shit |
| 04:58.24 | freetown2 | ? |
| 04:58.31 | kc8pxy | drmessano: that sounds messy. |
| 04:58.40 | drmessano | lol |
| 04:58.47 | freetown2 | i got the discs...after months |
| 04:59.32 | drmessano | I never got them.. Which means they didnt care if I used their so-obscure-we're-begging-you-to-use-it-by-giving-away-CDs OS |
| 05:00.03 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 05:01.19 | freetown2 | shrugs. i put up with their 'i am a noble' thing on the opensolaris channel when i need help otherwise i don't bother. zfs has done wonders for me for the samba server |
| 05:02.10 | drmessano | I'm sure they're no more elitist than BSD or Debian users |
| 05:02.43 | sbrath | I'd say OpenBSD are the worst I've seen. |
| 05:02.59 | drmessano | "Look at me, I run 300 apps on this box, all in their own jails!" "Yes, but have you ever kissed a girl?" |
| 05:03.04 | drmessano | just sayin |
| 05:03.05 | freetown2 | Debian? hmm, not tried that crowd yet. BSD, no comment. Have not had to go asking them anythng yet too |
| 05:03.13 | freetown2 | hahaha |
| 05:03.49 | sbrath | At least when I ran mirrors.tds.net the Open BSD people were to special to allow anyone to mirror them direct from the main source... You had to "know" somebody to vouch for you before the'd even talk to you ... :) |
| 05:05.33 | drmessano | They're the ones with the workplace safety sign on the website, right? |
| 05:05.52 | drmessano | "Only 1 accident in the last 365 days" |
| 05:05.56 | drmessano | errrr ooops |
| 05:06.01 | drmessano | "Only 2 accidents in the last 365 days" |
| 05:06.12 | sbrath | Only two remote holes in the default install, in a heck of a long time! |
| 05:06.41 | drmessano | It was only 1 remote exploit in the default install, ever.. til they hit number 2 |
| 05:06.48 | sbrath | and cvsup is the mosted f***d up system to manage as a Mirror Admin. |
| 05:06.58 | drmessano | then it was "Only 2 remote exploits, K?" |
| 05:07.02 | jdoe | the debian user groups are full of dangerously helpful mouthbreathers. |
| 05:07.10 | jdoe | people who will happily give you dumb/wrong advice. |
| 05:07.20 | [TK]D-Fender | OpenSolaris "noble"? Bs... more like yuppies trying to think their cool "Yeah we're 'open source' too now! We're cool and won't stab you in the back... like all those other times" |
| 05:07.28 | sbrath | Don't even get me started about the Amiga and OS2 people... Jeesh :) |
| 05:07.35 | jdoe | the debian contributor stuff is full of people bickering over whether something is free or kinda free or free as in freedom to starve on a barren mountainside |
| 05:07.54 | dlynes | laughs at jdoe. |
| 05:08.01 | [TK]D-Fender | OpenSolaris = feigned nobility |
| 05:08.27 | [TK]D-Fender | IceWeasel bitches! |
| 05:08.38 | drmessano | lol |
| 05:08.40 | dlynes | I hate IceWeasel |
| 05:08.43 | drmessano | IceWeasek! |
| 05:08.46 | freetown2 | [TK]D-Fender, oh, they constantly talk about how solaris is used by banks and financial institutes...oh wait, by the 'cream' of society that is responsible for the current mess. |
| 05:08.46 | drmessano | IceWeasel! |
| 05:09.12 | dlynes | IceWeasel, IceApe and IceDove are all majorly lame |
| 05:09.22 | drmessano | My bank runs on Windows NT using the unreleased service pack 7 |
| 05:09.24 | dlynes | I've replaced them all with Firefox and Thunderbird |
| 05:09.29 | dlynes | and X-Chat :) |
| 05:10.01 | dlynes | drmessano, you mean microsoft fixed sp6a? |
| 05:10.09 | dlynes | drmessano, and didn't bother telling anyone? |
| 05:10.10 | Znuff | Can I create a peer in asterisk that accepts *any* password? |
| 05:10.37 | dlynes | Znuff, you mean one that doesn't require a password? |
| 05:10.59 | Znuff | dlynes, something like that, but if password is specified by client, still accept |
| 05:11.16 | dlynes | Znuff, no...but you can create one that doesn't use a password |
| 05:11.22 | drmessano | Theres been a long standing rumor/partially confirmed fact that M$ built an SP7 for certain high paying customers who paid for extended support hotfixes |
| 05:11.42 | Znuff | dlynes, and if the users specifies a password in his configuration? |
| 05:12.11 | dlynes | Znuff, then it tries to use that password, and the other end says sorry, but you're not dynamic |
| 05:12.16 | drmessano | One of those involves the NASA having deployed NT4 SP7 on certain systems internally |
| 05:12.30 | drmessano | But who knows |
| 05:12.42 | Znuff | dlynes, dynamic? |
| 05:12.43 | dlynes | I think they're all doing that, because NT4 is the most stable version of windows ever released |
| 05:12.56 | drmessano | I'm still pissed off they didn't release an SP5 for Windows 2000 |
| 05:13.33 | dlynes | drmessano, it's microsoft's punishment because none of you peons liked their windows hasta la vista |
| 05:14.45 | dlynes | Znuff, register => ... |
| 05:15.02 | drmessano | Vista was punishment for making them add all that security to the codebase for XP SP2 and Server 2003 SP1 |
| 05:15.29 | drmessano | all that damn work that distracted them from releasing Vista with all those features they promised for years and still havent delivered |
| 05:15.35 | dlynes | but xp sp2 still had a shitload of security holes |
| 05:16.23 | dlynes | I think xp is by far their one release with the most of amount of security holes (other than 95/98/ME...but those don't really count, because they all still ran on top of DOS) |
| 05:16.42 | ManxPower-work | ~guis |
| 05:16.43 | infobot | extra, extra, read all about it, guis is "FreePBX/Trixbox is to Asterisk as Windows 95 is to DOS" |
| 05:18.07 | drmessano | XP didnt have a lot of security holes.. the codebase had a lot of security holes that were exposed dramatically in a very short period of time.. Note 90% of those were patched all the way back to NT until the changes were too drastic to implement that far back. |
| 05:18.56 | ManxPower-work | The only time my XP got a virus is when I was in a location without a firewall. |
| 05:19.54 | ManxPower-work | (dialup actually) Pretty cool (NOT!) seeing malware spewing out spam via your 33.6 dialup connection. |
| 05:19.58 | drmessano | lol |
| 05:20.26 | ManxPower-work | Oh! Wait! That was Win2k. Nevermind. |
| 05:20.45 | drmessano | lol |
| 05:21.57 | drmessano | I loved Win2k, but god it was horrible when it came to malware. It would just tear a hole in the thing you couldn't fix without a reinstall. |
| 05:21.58 | jdoe | freetown2: ... right. Solaris is. Not OpenSolaris. Nobody in their right mind uses that. |
| 05:22.03 | drmessano | Worse than a trixbox |
| 05:24.54 | freetown2 | is officially insane |
| 05:28.31 | drmessano | I run OpenDarwin because rather than suffering as an OSX Server admin, or using a mainstream BSD, I CHOOSE to use their heavily modified BSD and suffer unnecessarily |
| 05:29.34 | ManxPower-work | I loved win2k too. Only time it was compromised is when I did something REALLY stupid. |
| 05:29.42 | jdoe | fuck you all |
| 05:29.44 | jdoe | I run A/UX |
| 05:29.52 | OrNix | Hello. * 1.6.1.11, spandsp-0.0.6(snapshot). * receives fax with ReceiveFAX. If sending side requests the conversation after transmission, app_fax gives this error: http://paste.org.ru/?v3byhl. It there a way to repair it? |
| 05:30.28 | ManxPower-work | OrNix: Does app_fax support that? |
| 05:30.42 | freetown2 | jdoe, is your OS available on the net? |
| 05:30.50 | drmessano | jdoe: I use Netware 5.1 as my DESKTOP OS |
| 05:30.54 | freetown2 | first time i have heard of A/UX |
| 05:31.33 | jdoe | I'm sure you can find it somewhere if you try. |
| 05:31.39 | jdoe | finding hardware that can run it is likely harder :P |
| 05:31.46 | sbrath | I've used AIX but not A/UX |
| 05:31.58 | OrNix | ManxPower-work, i don't know. I don't want app_fax to transfer the call somewhere. I want it not to give an error after transmission. |
| 05:32.07 | freetown2 | thinks smething flew over sbrath's head :D |
| 05:32.34 | sbrath | :) |
| 05:32.48 | drmessano | My Kaypro running CP/M is my file server for my Commdore 64 running GEOS |
| 05:35.19 | sbrath | ok,, My Z/80 trs80-model 30 will kick all your buts.... I think you could even fit the readme file for how to install asterisk totaly in RAM! |
| 05:35.26 | sbrath | model 3 I mean : |
| 05:36.31 | drmessano | PRESS PLAY ON TAPE |
| 05:36.34 | drmessano | ^^ I win |
| 05:39.28 | *** join/#asterisk aidinb (n=Aidin@24-176-209-6.dhcp.lnbh.ca.charter.com) |
| 05:50.21 | freetown2 | now I know what to get the asterisk channel going. there should be a ban on OS discussion :D |
| 05:53.36 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 05:53.53 | drmessano | Star Trek sucks, Star Wars was much more realistic |
| 05:54.01 | drmessano | ^^^^^ Take tat |
| 05:54.04 | drmessano | that* |
| 05:54.44 | freetown2 | waits for the Trekkies to rise in defiance... |
| 05:55.28 | voipmonk | sleeps to the opening of Thundercats |
| 05:56.34 | drmessano | Thunder, thunder, thunder, thundercats, HOOOO!! |
| 05:58.41 | freetown2 | OOoohh, I don't feel so old anymore |
| 06:10.19 | voipmonk | :) |
| 06:14.50 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
| 06:21.07 | sbrath | Damn, ManxPower left. |
| 06:21.25 | sbrath | I npw just wiresharked that sip traffic via my NAT fw. |
| 06:22.37 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 06:24.39 | sbrath | According to the wireshark logs my SIP session carrys on a 30 second RTP session, back and forth, and then just aborts. |
| 06:28.12 | ChannelZ | You must be having a very boring conversation. |
| 06:28.22 | sbrath | yep |
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| 06:32.24 | bobsaccamano | hi..i have a query not directly related to asterisk..hope to get some pointers...i want to know if there is a standard error ttreatment in SIP if two vertical service codes are dialed consecutively...like *34*67<DN here> ? |
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| 07:09.49 | ZX81 | is there a way to expand a variable which specifies the application to run? |
| 07:10.27 | ZX81 | kinda like an exec |
| 07:10.53 | ZX81 | I guess you could exec asterisk -rx '${VARIABLE}' but it wouldn't be on the channel |
| 07:10.55 | ZX81 | meh |
| 07:11.27 | ZX81 | need like a dialplan_exec function |
| 07:11.28 | ZX81 | :) |
| 07:11.32 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 07:11.41 | ZX81 | ooh |
| 07:11.49 | ZX81 | maybe I could use execif |
| 07:11.52 | ZX81 | with 1 |
| 07:12.01 | ZX81 | and variable for the app |
| 07:12.02 | ZX81 | :D |
| 07:13.19 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
| 07:13.55 | ZX81 | oh |
| 07:14.00 | ZX81 | slaps head |
| 07:14.11 | ZX81 | Exec(appname(arguments)) |
| 07:14.12 | ZX81 | :D |
| 07:20.40 | Jenna | ZX81, running scripts(external apps) is supported |
| 07:20.54 | ZX81 | yah with system |
| 07:20.59 | ZX81 | I wanted internal |
| 07:20.59 | ZX81 | :) |
| 07:21.05 | ZX81 | for a routing type in MySQL |
| 07:21.14 | ZX81 | so that I could do variable substitution |
| 07:21.19 | ZX81 | i.e. click add ivr |
| 07:21.31 | ZX81 | and the system ask for what you want to fill the variable with |
| 07:21.43 | ZX81 | i.e. what application do you want to execute when someone dials 1 |
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| 07:22.09 | ZX81 | because the routing type is the same for everyone, it needs to set the variables then use the same context |
| 07:22.24 | ZX81 | exec does what I need - I can expand a variable to an application to run |
| 07:22.26 | ZX81 | :) |
| 07:23.00 | jayrod422 | if i have a tdm800p with a echo cancelation module do i need to do anything with enabling echo cancelation in dahdi? |
| 07:24.24 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 07:24.42 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 07:25.08 | ZX81 | just follow the digium installation document |
| 07:25.27 | ZX81 | I'm still running zaptel on most things but really should change everything over |
| 07:25.44 | ZX81 | "Note that unlike zaptel.conf, you have to explicitly set the echo canceller for each channel." |
| 07:25.51 | ZX81 | http://www.voip-info.org/wiki/view/DAHDI |
| 07:26.03 | ZX81 | <PROTECTED> |
| 07:26.16 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net) |
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| 07:44.31 | ChannelZ | hmm.. anyone use asterisk-stat? |
| 07:53.47 | freetown2 | anybody given wireless-n and snom 820 + usb wireless a try? |
| 07:56.35 | ramzi | Hi, i'm using Asterisk 1.2.13, and when i check the time with app-speakingclock the clock skews about 2min every day. the system date and hwclock show right time. If i restart asterisk the time changes to the system time. What can be wrong? |
| 07:59.07 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
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| 08:33.58 | phasip | Hey! I am having trouble with call forwarding, this is what I get: http://pastebin.com/d10d31258 |
| 08:34.19 | phasip | I tried adding the extension in default but it wont work |
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| 08:42.51 | tzafrir | ramzi, I have no idea what app-speakingclock is |
| 08:43.07 | tzafrir | Uses Asterisk timing? |
| 08:44.49 | *** join/#asterisk sulex (n=sulex@88-149-154-95.static.ngi.it) |
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| 08:47.07 | ramzi | tzafrir: yes, its the function that says the time, *60 |
| 08:47.51 | tzafrir | ramzi, why not just use SayUnixTime() ? |
| 08:51.47 | ramzi | tzafrir: i dont know, but the problem is that the clock in asterisk is skewing, it fixes afteri restart asterisk. |
| 08:55.32 | Amorsen | Sounds rather strange that Asterisk would use something different from the system clock... |
| 08:58.11 | tzafrir | ramzi, are you sure SayUnixTime doesn't just use system time? |
| 08:58.48 | tzafrir | Amorsen, I wouldn't be surprised if Asterisk's internal clock is slightly skewed from the system clock |
| 08:58.57 | ramzi | it says the time asterisk got. Asterisk uses system time. but i dont know why the clock in asterisk skew |
| 08:59.00 | tzafrir | (not sure which of them is more accurate) |
| 09:01.15 | ramzi | tzafrir: its not just the saytime functions thats my problem. everything in asterisk thats uses the clock skews. timestamps, time conditions. |
| 09:02.03 | ramzi | tzafrir: but if i restart asterisk the clock gets in sync with system time, but after 24h the time is 2min behind |
| 09:02.29 | Amorsen | tzafrir: Why doesn't Asterisk just stick with the system-provided time? |
| 09:02.43 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:02.56 | tzafrir | Amorsen, as I mentioned above - that surprises me |
| 09:03.21 | tzafrir | I have no idea what this app-speakingclock is |
| 09:04.21 | TommyBotten | Are you running it on a VM rigg? |
| 09:04.46 | TommyBotten | Or more precise - in a virtual machine? |
| 09:04.58 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 09:05.22 | ramzi | no, its not a virtual machine |
| 09:05.22 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 09:05.42 | ramzi | both system clocks are fine, but asterisk skews |
| 09:06.31 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 09:06.59 | TommyBotten | Do you get any notification/error messages? |
| 09:07.09 | *** join/#asterisk Jenna (n=JJ@unaffiliated/jenna) |
| 09:08.24 | doolittlework | Any one know how to get this to work "Junghanns OctoBRI ISDN card" |
| 09:08.58 | doolittlework | this thing is giving me a pain in the backside |
| 09:17.57 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) |
| 09:18.37 | tzafrir | doolittlework, should be supported by wcb4xxp of dahdi |
| 09:22.48 | ramzi | TommyBotten: no messages at all |
| 09:23.23 | doolittlework | tzafrir thx |
| 09:23.25 | doolittlework | exit |
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| 09:39.51 | benngard | sorry if i repeat myself, did put a question here yesterday but did get a disconnect so i am not sure if any had any solution for me, any i ask again |
| 09:41.01 | benngard | i have an extension that dials both my sip phone and my cell phone, working! but when i "dial" my cell i need to strip leading zero from a-number to get caller-id on my cell phone display |
| 09:41.42 | benngard | exten => 0317998975-NOT_INUSE,1,Dial(SIP/0317998975&OOH323/00733025975@Avaya,20,rtT) |
| 09:42.18 | *** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) |
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| 09:47.22 | ChannelZ | benngard: don't understand the question. Based on your Dial, just remove the undesired 0 |
| 09:48.50 | ChannelZ | but I suspect that's not really what you're asking, but what you've asked is unclear |
| 09:49.50 | benngard | yes, but i dont wanna remove it for the sip, just the ooh323 part |
| 09:50.14 | ChannelZ | you have 2 static numbers typed in |
| 09:50.45 | benngard | yes, 1 is my sip phone (0317998975) and 1 is my cell phone (00733025975) |
| 09:51.20 | ChannelZ | and you want to remove the first 0 from one of them.. so just... hit the delete key |
| 09:51.22 | benngard | if for example 0317998976 calls 0317998975, i need to keep the leading 0 for the sip part but not for the ooh323 part |
| 09:52.48 | benngard | like dial(SIP/0317998975& "Strip leading zero before" OOH323/00733025975@Avaya) |
| 09:53.05 | ChannelZ | you are making no sense |
| 09:53.29 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:56.05 | ChannelZ | Dial(12345) is stripping the leading 0 off of Dial(012345) |
| 09:56.38 | *** join/#asterisk stephank (n=traveler@212.178.158.35) |
| 09:56.49 | benngard | i need to strip it a-number, not b-number |
| 09:57.05 | benngard | maybe u dont use the word a and b number? |
| 09:57.24 | fenrus_ | caller and callee |
| 09:57.36 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) |
| 09:57.36 | ChannelZ | no I haven't the faintest idea what you are talking about based on the examples you've given |
| 09:58.26 | ChannelZ | The Dial() examples you've pasted twice now are static |
| 10:00.34 | fenrus_ | ${EXTEN:-3} gives you the last three numbers of the dialed number.. |
| 10:00.35 | fenrus_ | i guess you can do some other magic to remove the first number |
| 10:00.37 | ChannelZ | now if your actual extension is more like "exten => NNNN" then ${EXTEN:1} refers to whatever 'NNNN' is, minus the first number. |
| 10:00.46 | fenrus_ | ah, there it was |
| 10:01.25 | ChannelZ | but what you pasted above was "exten => 0317998975-NOT_INUSE", not a pattern, and you're dialing completely different numbers statically in the Dial() statement, so I have no idea what you mean or want. |
| 10:02.54 | benngard | its not the number i dal, they are working, what i need to do is maniuplate caller-id before sending it to "ooh323" but not changing anything when sending it to "sip" |
| 10:04.25 | benngard | caller = 0317998976 and callee = 0317998975, sip part: caller = 0317998976 but ooh323 part: caller = 317998976 |
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| 10:05.32 | ChannelZ | as in the value of CALLERID(num) -- in which case I'm pretty sure you can't in a multiple channel dial() |
| 10:05.59 | benngard | so sad |
| 10:07.27 | benngard | the i have to talk to the guys who run the call network and se if they can accept the leading zero |
| 10:07.30 | benngard | then* |
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| 10:28.46 | *** join/#asterisk sassyn (n=sassyn@93-173-146-83.bb.netvision.net.il) |
| 10:28.50 | sassyn | hi all |
| 10:28.58 | sassyn | does anyone use asterisk 1.6.20? |
| 10:29.17 | sassyn | I haveing problems with getting it work |
| 10:29.22 | sassyn | with freepbx |
| 10:29.28 | sassyn | \join #freepbx |
| 10:31.46 | Chainsaw | Try /join, with a forward slash. |
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| 10:33.51 | maruz | someone does know which is the max value at right of dot into uniqueid field? 65535, maybe? |
| 10:38.06 | *** join/#asterisk aeon- (n=chatzill@ks302028.kimsufi.com) |
| 10:39.17 | aeon- | Hi |
| 10:40.11 | aeon- | I'm looking for an information : where is stored the voicemail password when a user changes it from the voicemailmain ? does the plugin modify the voicemail.conf file ? |
| 10:41.16 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 10:44.21 | *** join/#asterisk jkroon (n=jkroon@dsl-244-25-144.telkomadsl.co.za) |
| 10:45.57 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-wgyxmcjekjlevipz) |
| 10:48.04 | *** join/#asterisk aeon- (n=cell@ks302028.kimsufi.com) |
| 10:49.18 | *** join/#asterisk smooth_penguin (n=smoove@59.95.21.233) |
| 10:53.14 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 10:53.30 | angryuser_ | do asterisk 1.6 support fully rfc 4028 ? |
| 10:54.44 | *** join/#asterisk garymc (n=garymc@81.138.225.161) |
| 10:54.49 | garymc | hi guys, im having a little bit of trouble with my snom wireless phones. I can call each other but with no voice. Now these phones are connected to my router so i assumed i had to open udp ports 5060 and 10001-20000 I did this but still no voice over the internet network or external. Here is a cli pastebin of my call. http://pastebin.ca/1749457 Pleas help if you can |
| 11:02.02 | *** part/#asterisk vk2dgy (n=rossw@ali-syd-3.albury.net.au) |
| 11:15.13 | sassyn | Hi |
| 11:15.30 | sassyn | does IAX doesn't work in 1.6.20? |
| 11:15.55 | tzafrir | 1.6.2.0 ? |
| 11:16.01 | sassyn | 1.6.2.0 |
| 11:16.04 | tzafrir | What are you trying to talk to? |
| 11:16.04 | sassyn | I mean |
| 11:16.34 | sassyn | I compile and build a new DEB file for the Debian/Ubuntu release |
| 11:16.41 | sassyn | and testing this version |
| 11:16.54 | sassyn | But Having too many problems |
| 11:17.01 | tzafrir | 1.6.2.0 and 1.6.2.1-rc1 build fine here |
| 11:17.05 | tzafrir | (in pkg-voip) |
| 11:17.39 | sassyn | like for example : when dailing a invalid number |
| 11:17.59 | sassyn | I see in the console this: Playing 'cannot-complete-as-dialed.gsm' |
| 11:18.02 | tzafrir | I can't think of any chan_iax2 related patch we needed |
| 11:18.03 | sassyn | but can't hear it |
| 11:18.11 | *** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it) |
| 11:18.25 | tzafrir | can't hear it in? |
| 11:18.34 | sassyn | tzafrir,yep |
| 11:19.55 | sassyn | tzafrir, yes can't hear it |
| 11:20.16 | sassyn | I use zoiper and snom phone |
| 11:20.46 | tzafrir | Any chance that this is a version before the IAX fix? |
| 11:21.28 | tzafrir | http://downloads.asterisk.org/pub/security/IAX2-security.html |
| 11:22.17 | sassyn | No I grub it two days ago |
| 11:22.35 | sassyn | the same thing apper the same with SIP account |
| 11:23.17 | sassyn | u can check out the asterisk log at http://pastebin.com/m5664dcb0 |
| 11:24.51 | sassyn | brb |
| 11:26.00 | pif | hi, is there a digium 4BRI pcie card? |
| 11:26.48 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
| 11:26.52 | tzafrir | sassyn, this is SIP, not IAX |
| 11:27.29 | tzafrir | also, try a more direct test: |
| 11:27.45 | tzafrir | originate SIP/1000 application Playback demo-instruct |
| 11:28.14 | tzafrir | this should call the phone and play a message to it |
| 11:32.13 | *** join/#asterisk Jenna (n=JJ@unaffiliated/jenna) |
| 11:35.43 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 11:39.43 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 11:46.21 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) |
| 11:47.10 | *** join/#asterisk viraptor (n=viraptor@212.11.65.66) |
| 11:48.22 | viraptor | hi, I've got a problem with ${CHANNEL(audioreadformat)} - it's not available at sip hangup, audio was passing through asterisk, I'm running 1.4 series- any ideas? (other variables - write / native also not available) |
| 12:03.28 | *** join/#asterisk nardul (n=kse@212.37.141.188) |
| 12:04.07 | nardul | Afternoon. When i pull calls with *8, on my phone it doesn't display the caller-number. Is there any way to change that? |
| 12:05.37 | sassyn | tzafrir, still around? |
| 12:05.48 | tzafrir | yes |
| 12:06.04 | sassyn | Still not working |
| 12:06.13 | sassyn | I can get into my voice mail |
| 12:06.24 | sassyn | hear the messages |
| 12:06.34 | sassyn | but when dailing error number can't here a thing |
| 12:06.51 | sassyn | r u working with 1.6.2.0? |
| 12:08.12 | tzafrir | partially |
| 12:08.18 | sassyn | what I doing worng? |
| 12:08.25 | sassyn | Maybe u can also help here: |
| 12:08.30 | sassyn | http://pastebin.com/m18993386 |
| 12:08.48 | tzafrir | are the phone and Asterisk on the same LAN? |
| 12:08.53 | sassyn | can't dial to internal extenstion from internal extenstion |
| 12:08.58 | sassyn | yes |
| 12:09.26 | tzafrir | "can't dial from an internal extension to an internal extension" is a dialplan issue |
| 12:09.45 | tzafrir | In your case: a FreePBX issue, its dialplan is horribly complex |
| 12:10.43 | sassyn | so where to look? |
| 12:12.52 | tzafrir | I think that the following is not such a good sign: NoOp("SIP/1000-0000000c", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack |
| 12:12.57 | tzafrir | Try #freepbx |
| 12:13.22 | tzafrir | The target extension does not exist? Was not defined? |
| 12:14.15 | sassyn | I create two extenstions |
| 12:14.18 | sassyn | 1000, 2000 |
| 12:14.21 | sassyn | both connected |
| 12:14.51 | tzafrir | are both SIP phones? |
| 12:16.34 | sassyn | yep |
| 12:16.52 | sassyn | sip show peers show both sip clients |
| 12:22.54 | tzafrir | let's try a low-level call: |
| 12:23.01 | *** join/#asterisk garymc (n=garymc@81.138.225.161) |
| 12:23.08 | tzafrir | originate SIP/1000 application Dial SIP/2000 |
| 12:23.42 | garymc | anyone know why i cant get any voice over my wireless snom phones? ive opened all ports |
| 12:23.45 | garymc | on the router |
| 12:27.40 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) |
| 12:29.46 | sassyn | tzafrir, this is working |
| 12:30.09 | tzafrir | ok. So this is, indeed, a dialplan issue |
| 12:30.42 | sassyn | so it is a problem with the FreePBX? |
| 12:31.33 | tzafrir | In the realm of FreePBX, yes |
| 12:31.42 | sassyn | OK |
| 12:31.50 | sassyn | I will ask #FREEPBX |
| 12:32.34 | sassyn | what is the big diffrent between 1.4 to 1.6? since with 1.4 I had no issue work out of the box |
| 12:34.20 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:35.55 | *** join/#asterisk [netman] (n=netman@193.153.155.169) |
| 12:37.36 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 12:38.27 | garymc | i take it nobody looked at my pastebin as im getting no voice over my wireless extensions with are connected to my router? |
| 12:41.07 | garymc | ive opened the correct ports but im getting no joy :( |
| 12:49.09 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 12:49.24 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 12:52.51 | tzafrir | sassyn, there's no "1.6". 1.6.0, 1.6.1 and 1.6.2 are separate branches on their own, just like 1.2 and 1.4 |
| 12:53.32 | beek | tzafrir: I for one am glad that they've gone to the new naming scheme. |
| 12:54.00 | tzafrir | they're actually back to the original one with 1.8 |
| 12:55.05 | coppice | what the fork are they up to with so many threads? |
| 12:55.47 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:55.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:57.22 | garymc | [TK]D-Fender not about today |
| 13:02.27 | sassyn | tzafrir, What do u think is at the moment the best release? |
| 13:02.58 | tzafrir | sassyn, for what, specifically? |
| 13:03.05 | tzafrir | starting from scratch? |
| 13:03.08 | beek | tzafrir: Yep... and I'm glad to see it. I guess I should have said "new, old naming scheme." |
| 13:05.44 | *** join/#asterisk [netman] (n=netman@193.153.155.169) |
| 13:05.47 | devmod | morning |
| 13:06.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:09.50 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) |
| 13:14.52 | garymc | [TK]D-Fender : Hi could you take a glimpse at my pastebin http://pastebin.ca/1749457 Im not getting voice over extension 300 or 301 to 201. They can call each other but no voice is passed. Im guesing its a port issue on my router but ive opened all relavent ports? |
| 13:15.39 | *** part/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 13:15.48 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 13:15.52 | garymc | ext 300 & 301 are Snom wireless phones with are plugged right into my router away from office. But my office is plugged into that router. So it should work |
| 13:16.02 | [TK]D-Fender | garymc: you haven't described the environment at all and provided no pertinent debug. |
| 13:16.25 | [TK]D-Fender | garymc: No configs either. |
| 13:16.26 | garymc | is that not a debug in my pastebin. been a while since i done any of this |
| 13:16.41 | beek | [TK]D-Fender: Is that a record for you? Eight minutes from signon to first query directed at you... |
| 13:17.03 | [TK]D-Fender | garymc: and how is a "phone" and an "OFFICE" plugged into a router? What sets this phone apart? |
| 13:17.16 | [TK]D-Fender | garymc: SIP dEBUG <_ |
| 13:17.21 | garymc | lol ok i get you |
| 13:17.52 | [TK]D-Fender | beek: No, I've been pounced faster.... |
| 13:17.59 | [TK]D-Fender | beek: mornin' |
| 13:18.14 | beek | Good morning. |
| 13:18.22 | garymc | my office is outside in porta cabins. I have a cable going over to my home where the broadband connection is. My servers all go through this. My SNOM wireless phones are over in the house so i can have office phones and extension on portable phones over there |
| 13:18.53 | garymc | so i need a sip debug. I need someone back in the house now |
| 13:18.55 | garymc | :( |
| 13:19.15 | viraptor | trying again with guys who joined recently - I've got a problem with ${CHANNEL(audioreadformat)} - it's not available at sip hangup, audio was passing through asterisk, I'm running 1.4 series- any ideas? (other variables - write / native also not available) |
| 13:20.34 | devmod | any good/cheap voip phone recommended for asterisk ? |
| 13:20.43 | voipmonk | viraptor: you want the codec for logging purposes? |
| 13:21.48 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 13:22.08 | [TK]D-Fender | garymc: And none of that story tells me anything useful at all. You may as well have just told me what colour the desk the phone was on. |
| 13:22.20 | sassyn | tzafrir, for a stand alone 100 user asterisk machine |
| 13:22.22 | creativx | my desk is a light color |
| 13:22.27 | sassyn | which version? |
| 13:22.30 | [TK]D-Fender | devmod: Polycom or bust <- |
| 13:23.10 | viraptor | voipmonk: yes |
| 13:23.52 | tzafrir | sassyn, that's the version now getting into Elastix and into Debian |
| 13:24.19 | garymc | [TK]D-Fender : Brown |
| 13:26.25 | garymc | [TK]D-Fender : Here is a sip debug. http://pastebin.ca/1749561 not all is in here though :S |
| 13:27.20 | sassyn | tzafrir, which version? |
| 13:27.28 | tzafrir | 1.6.2 |
| 13:27.29 | viraptor | voipmonk: any ideas? |
| 13:28.00 | [TK]D-Fender | garymc: Transmitting (NAT) to 192.168.1.65:5060: Contact: <sip:201@81.134.0.102> |
| 13:28.03 | voipmonk | does SIP_CODEC just force or can it display the existing codec - or is it depricated? |
| 13:28.22 | [TK]D-Fender | garymc: WTF is it calling a Class-C using a public IP as contact? |
| 13:28.28 | [TK]D-Fender | garymc: Fix your NAT settings |
| 13:28.30 | voipmonk | deprecated |
| 13:29.48 | garymc | yeah my server has had to go on a public ip, should i take it off that? |
| 13:30.04 | garymc | we bought 5 static ip address |
| 13:30.29 | garymc | hmm ok |
| 13:30.32 | garymc | ill take a look |
| 13:31.46 | [TK]D-Fender | garymc: its CALLING to a Class-C. It should HAVE an address on a similarly reachable Class-C and not be handing it the public |
| 13:33.10 | garymc | yeah its pissing me off. BT with their static ip rules. Its cos i bought 5 static ip address's you got to give a public address to the server instead of a generic dhcp one |
| 13:33.48 | devmod | [TK]D-Fender, thx |
| 13:34.14 | [TK]D-Fender | garymc: It should be static, thats not the point. |
| 13:34.17 | devmod | any ideas on how to create prompts like these using open source tools? http://www.aupix.com/index.htm?/support/astvideo.htm Any specific requirements for asterisk while encoding to h.263 n h.264? |
| 13:35.00 | [TK]D-Fender | garymc: YOU are the one who configured * to send what it does by not considering that subnet local. |
| 13:35.17 | garymc | ...hmmm ok |
| 13:42.54 | donatas | I installed asterisk-mysql package, but I don't see mysql module |
| 13:42.59 | donatas | only pgsql |
| 13:43.14 | donatas | I need cdr_mysql.conf file |
| 13:44.06 | [TK]D-Fender | donatas: Go create it then |
| 13:44.55 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:45.07 | tzafrir | devmod, ffmpeg and co (mplayer, vlc, whatever) can conver video formats for you |
| 13:45.18 | garymc | [TK}D-Fender : I changed it from public ip to private and still ringing no voice |
| 13:45.45 | [TK]D-Fender | garymc: Your networking description is still piss-poor |
| 13:45.59 | garymc | im thinking their is a problem with my router and its shitty port settings. ie doesnt open ports properly |
| 13:46.04 | [TK]D-Fender | garymc: And you never provide backup without a large hassle |
| 13:47.09 | garymc | Asterisk Server > BROADBAND ROUTER < SnomM3 gateway |
| 13:47.17 | garymc | any better? |
| 13:47.26 | devmod | tzafrir, i did convert them but it wont work. I am wondering if there are any specific options I must use in order for it to work |
| 13:47.50 | [TK]D-Fender | garymc: How is * transmitting from a public host to a PRIVATE SUBNET DIRECTLY? |
| 13:48.13 | garymc | throught the Router |
| 13:48.29 | garymc | its actually set on a private one now to test |
| 13:48.49 | [TK]D-Fender | !?!??!?! |
| 13:48.51 | garymc | as it is new to me this Static ip stuff |
| 13:48.59 | garymc | yeah me too |
| 13:49.06 | [TK]D-Fender | garymc: static is IRRelevANT |
| 13:49.28 | garymc | thats what i thought, but BT do this 5 static IP's through the one router |
| 13:49.39 | [TK]D-Fender | garymc: PUBLIC does not call to PRIVATE |
| 13:50.18 | garymc | so instead of my Asterisk having IP of 192.168.1.73 (as the router would assign automaticly) it would have a Public IP instead regardless |
| 13:50.36 | [TK]D-Fender | garymc: garymc and I have 8 at hoe, and 16 at the office. this is irrelevant. |
| 13:50.42 | garymc | ok |
| 13:50.43 | [TK]D-Fender | garymc: home* |
| 13:50.56 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:51.14 | [TK]D-Fender | garymc: Where is this M3 physically relative to the server? |
| 13:51.44 | garymc | i dont get why i can get them to call each other and they ring but no voice. The M3 is plugged into the router, but 100 yards away in distance from the server |
| 13:52.05 | garymc | the server is also plugged into the router |
| 13:52.20 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:52.35 | [TK]D-Fender | garymc: So that router is also NATing another subnet? |
| 13:52.38 | garymc | when i go into the router software i can see both listed and their relevant ip address |
| 13:52.54 | garymc | yes |
| 13:53.11 | garymc | i guess so |
| 13:53.40 | [TK]D-Fender | drawme ANOTher complete picture of this networking scenario. In the first PB you also didn't bother showing me the START of the call as if the First LEG wasn't important either |
| 13:53.56 | garymc | yeah it disapeared |
| 13:54.15 | garymc | ill have to use putty on my other machine instead of this linux server im on now |
| 13:54.32 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 13:55.01 | garymc | no its ok ill do it now |
| 13:55.17 | *** join/#asterisk ayrjola (n=ayrjola@gw.ccxtech.fi) |
| 13:56.58 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 13:59.22 | garymc | right gotta wait for someone to be in house brb |
| 14:03.30 | *** join/#asterisk etfonhomey (n=etfonhom@ip-64-32-192-35.iad.megapath.net) |
| 14:04.29 | *** join/#asterisk mort_gib (n=mjensen@177.209.244.195.dsl.static.gibconnect.com) |
| 14:07.22 | mort_gib | Hey, does dahdi support bri_cpe_ptmp with a Bri card?? |
| 14:07.37 | donatas | i run on asterisk console: show modules like sql |
| 14:07.42 | donatas | but i got only pgsql.. |
| 14:07.43 | tzafrir | mort_gib, yes |
| 14:07.43 | donatas | no mysql |
| 14:07.57 | sassyn | tzafrir, which linux version r u using? |
| 14:08.14 | tzafrir | (or rather: libpri >= 1.4.4 and asterisk >= 1.6.0 support it) |
| 14:08.19 | sassyn | did u apt-get asterisk from debian ? version 1:1.6.2.0-1 and others) |
| 14:08.21 | mort_gib | tzafrir: process_dahdi: Unknown signalling method 'bri_cpe_ptmp' at line 1231 |
| 14:08.22 | tzafrir | sassyn, various |
| 14:08.29 | tzafrir | elastix, Debian |
| 14:08.39 | [TK]D-Fender | donatas: Do you see a MySQL module in the modules folder? |
| 14:08.54 | tzafrir | mort_gib, which version of asterisk? built with libpri? |
| 14:09.04 | donatas | [TK]D-Fender: /usr/lib/asterisk/modules/app_addon_sql_mysql.so |
| 14:09.42 | tzafrir | I mostly suffer now from limitations of Asterisk 1.4 |
| 14:10.19 | mort_gib | tzafrir: I started without libpri, and I haven recompiled Asterisk AFTER installing libpri -Do I have to do that, or is it enough to recompile Dahdi?? |
| 14:10.42 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:10.55 | [TK]D-Fender | donatas: And when you try to load it manually? |
| 14:10.58 | sassyn | tzafrir, is elastix is better then freepbx? |
| 14:11.06 | [TK]D-Fender | LOL |
| 14:11.13 | donatas | [TK]D-Fender: how can i do it ? :) |
| 14:11.14 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 14:11.22 | [TK]D-Fender | sassyn: Elastix is a DISTRO that INCLUDES FREEPBX |
| 14:11.34 | [TK]D-Fender | Is a tire better than a car? |
| 14:12.04 | *** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca) |
| 14:12.13 | [TK]D-Fender | donatas: "module load app_addon_sql_mysql.so" |
| 14:12.26 | donatas | yea, i run it |
| 14:12.36 | donatas | Loaded app_addon_sql_mysql.so => (Simple Mysql Interface) |
| 14:12.40 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:13.00 | sassyn | [TK]D-Fender, I see |
| 14:13.15 | sassyn | [TK]D-Fender, the screenshot seem diffrent |
| 14:13.38 | [TK]D-Fender | sassyn: because the disto also has MORE stuff. OTher StUFF. |
| 14:13.56 | [TK]D-Fender | sassyn: Guess what... a car has more to see than jsut the tires |
| 14:13.56 | sassyn | [TK]D-Fender, like? |
| 14:14.10 | [TK]D-Fender | sassyn: Novel idea : Go visit their site |
| 14:14.10 | sassyn | [TK]D-Fender, sorry for my stupid questions |
| 14:14.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 14:14.15 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 14:15.07 | jaytee | morning Katty |
| 14:15.17 | beek | morning Katty, jayrod422 |
| 14:15.21 | beek | morning Katty, jaytee |
| 14:15.30 | jaytee | morning beek |
| 14:15.31 | TommyBotten | morning Katty, jaytee, beek |
| 14:15.39 | jaytee | morning Tommy |
| 14:15.49 | beek | morning TommyBotten |
| 14:16.13 | *** join/#asterisk highvoltz (i=rogers@bling.bling.org) |
| 14:16.21 | highvoltz | When you link two asterisk boxes, can you have a call come in one, and ring an extensions on the other in a ringroup? |
| 14:16.40 | [TK]D-Fender | highvoltz: You can do whatever you want |
| 14:17.00 | [TK]D-Fender | highvoltz: Server A gets a call. Does what it wants. Calls out to server 2. It does what it wants |
| 14:17.06 | highvoltz | werd |
| 14:17.26 | Katty | morning |
| 14:17.57 | [TK]D-Fender | Katty: Mew. |
| 14:18.03 | Katty | hugs [TK]D-Fender |
| 14:19.03 | minotaur01 | morning |
| 14:20.25 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 14:20.56 | mbrevda | anyone familar with this issue? WARNING[7004] loader.c: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_state_to_str |
| 14:30.12 | donatas | how to check if configuration of files are good or NOT ? |
| 14:30.23 | jayrod422 | can anyone shed some light on this (asterisk 1.6 dahdi 2.2) i can receive calls just fine but when i place them i get app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 14:30.34 | jayrod422 | tdm800p |
| 14:31.02 | [TK]D-Fender | jayrod422: We aren't seeing what you're doing so we can't tell why it might complain. Where are your configs & debugs? |
| 14:31.29 | jayrod422 | which configs do you want to see and which debugs |
| 14:31.36 | [TK]D-Fender | jayrod422: ......... |
| 14:31.42 | tzafrir | jayrod422, a slightly more complete trace will also help |
| 14:31.43 | *** part/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 14:31.45 | donatas | jayrod422: all configs |
| 14:31.51 | jayrod422 | lol |
| 14:31.52 | [TK]D-Fender | jayrod422: How about DAHDI CONFIGS, and CLI output for your FAILED CALL |
| 14:31.53 | jayrod422 | hold 1 sec |
| 14:32.01 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 14:32.06 | donatas | cdr_mysql.conf,extension.conf,sip.conf |
| 14:32.21 | jayrod422 | ouch |
| 14:33.35 | [TK]D-Fender | [09:30]<donatas>how to check if configuration of files are good or NOT ? <- USE THEM |
| 14:34.52 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:35.07 | Amorsen | Particularly for extensions.conf there are a lot of errors you'll get only at runtime, not at config load time |
| 14:35.38 | *** join/#asterisk Izinucs (n=scunizi@ip72-197-240-157.sd.sd.cox.net) |
| 14:37.20 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.33.202) |
| 14:37.20 | *** join/#asterisk freckle (n=jon@195.74.96.122) |
| 14:38.39 | jayrod422 | http://pastebin.com/m5f3a8e7f |
| 14:39.12 | jayrod422 | please dont hit me with the clue bat |
| 14:39.31 | devmod | is app_conference supposed to compile with asterisk 1.6.2? |
| 14:40.41 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 14:40.57 | [TK]D-Fender | jayrod422: -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/7011-00000046", "DAHDI/1/6574787") in new stack <-- you don't HAVE a "group=1" channel at all |
| 14:41.25 | [TK]D-Fender | OOPS. |
| 14:41.47 | jayrod422 | ??? |
| 14:42.06 | jayrod422 | what file you talking about for group=1? |
| 14:42.27 | [TK]D-Fender | jayNVM.... OK, do "dahdi show status" , "dahdi show channels", and "core show channels" |
| 14:42.34 | jayrod422 | k 1 sec |
| 14:44.12 | jayrod422 | http://pastebin.com/m2d5049d6 |
| 14:44.48 | [TK]D-Fender | jayrod422: Looks like its already in use |
| 14:45.04 | jayrod422 | you mean channel 1 |
| 14:45.59 | [TK]D-Fender | jayrod422: yes |
| 14:46.05 | jayrod422 | should it use the next one then |
| 14:46.15 | jayrod422 | or am i missing where i tell it too |
| 14:46.20 | [TK]D-Fender | jayrod422: no, because you are not dialing a group |
| 14:46.31 | [TK]D-Fender | jayrod422: You are dialing a specific channel |
| 14:47.40 | *** join/#asterisk galeras (n=galeras@186.80.192.102) |
| 14:48.34 | highvoltz | hey guys is there a good resource for SIP QoS configuration? |
| 14:49.10 | jayrod422 | fender i think i know what you are saying |
| 14:49.57 | jayrod422 | TRUNK = DAHDI/G2 is in extensions.conf but everything is group=0 in dhadi-channels |
| 14:50.15 | *** part/#asterisk Izinucs (n=scunizi@ip72-197-240-157.sd.sd.cox.net) |
| 14:51.12 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:52.31 | *** part/#asterisk benngard (n=benngard@213.88.138.230) |
| 14:53.27 | *** join/#asterisk [netman] (n=netman@193.153.155.169) |
| 14:57.10 | jayrod422 | i got it thanks fender |
| 14:58.27 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 14:58.48 | [TK]D-Fender | jayrod422: Good to hear |
| 15:03.31 | jbw | is + a valid extension character ? e.g. exten +1XXXXXXXXXXXXX,... |
| 15:04.06 | [TK]D-Fender | jbw: yes |
| 15:04.08 | *** join/#asterisk Lord-Rahl (n=quassel@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 15:04.21 | jbw | thanks [TK]D-Fender |
| 15:04.39 | EyeBaw | No problem |
| 15:04.50 | jaytee | how do you dial a + from a Polycom's keypad? |
| 15:05.29 | [TK]D-Fender | jaytee: Very carefully |
| 15:05.49 | Chainsaw | jaytee: On a cellphone you hold down the 0, perhaps it's the same there? |
| 15:06.23 | Lord-Rahl | ? for anyone I really like Asterisk Gui vrs Freepbx. Is still been work on? or I they move to freebpx? |
| 15:08.03 | [TK]D-Fender | Lord-Rahl: " or I they move to freebpx?" <- huh? |
| 15:08.12 | [TK]D-Fender | grammar FAIL |
| 15:11.12 | Lord-Rahl | [TK]D-Fender: Is Diagum stoping development of Asterisk Gui Sorry I was unclear |
| 15:11.41 | [TK]D-Fender | Lord-Rahl: No, but their is very little focus on it |
| 15:12.34 | jbw | puzzling. Shouldn't "exten +X." catch "+17027267579" ? I get handle_request_invite: Call from 'jbw' to extension '+17027267579' rejected because extension not found. |
| 15:13.03 | [TK]D-Fender | jbw: It should, but you aren't showing anything pertinent |
| 15:13.26 | jbw | do i need to begin the exten with _? to makes X expand to 0-9? hrm.. let me try |
| 15:13.37 | dlynes | jbw, yes |
| 15:13.41 | jbw | doh |
| 15:13.42 | jbw | thanks |
| 15:13.57 | [TK]D-Fender | jb_ denotes a PATTERN |
| 15:14.12 | dlynes | jbw, '_' means you're stating a pattern...if it doesn't have the '_', it accepts everything verbatim, and must match exactly |
| 15:15.50 | jbw | yep. got it :) asterisk rocks |
| 15:18.24 | Katty | jeebus. |
| 15:18.31 | Katty | what does it mean when your nose randomly starts bleeding? |
| 15:18.36 | Katty | :<<< |
| 15:18.48 | [TK]D-Fender | Katty: Office bone-dry? |
| 15:18.59 | [TK]D-Fender | Katty: probably need a humidifier |
| 15:19.01 | Katty | stares at the space heater running next to her |
| 15:19.15 | Katty | yeah, pretty dry all right |
| 15:19.19 | Katty | my skin is turning scaley |
| 15:19.27 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:19.27 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:19.31 | [TK]D-Fender | Katty: Mystery solved. |
| 15:19.37 | [TK]D-Fender | NEXT!!@!@!@ (c) BKW |
| 15:19.47 | *** join/#asterisk TheDavidFactor (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 15:23.23 | sassyn | [TK]D-Fender, what is the diffrent between asterisk-1.6.1.12.tar.gz to asterisk-1.6.2.0.tar.gz |
| 15:25.10 | beek | sassyn: two separate versions of asterisk. |
| 15:25.39 | beek | 1.6.0.x != 1.6.1.x != 1.6.2.x |
| 15:26.38 | Lord-Rahl | Can someone help me with a dahdi problem. I just setup asterisk 1.4.28 with dahdi 2.2.0+2.2.0 with a Wildcard TDM400P REV I. when I try to dial out I get (cause 66 - Channel not implemented) |
| 15:26.56 | leifmadsen | sassyn: new features and possible ABI changes |
| 15:27.20 | leifmadsen | 1.6.1.x to 1.6.2.x is like 1.2.x to 1.4.x |
| 15:27.54 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 15:29.30 | Lord-Rahl | I have check all over google but unable to find any good info. |
| 15:29.52 | *** join/#asterisk moy (n=moy@189.162.119.207) |
| 15:29.54 | tzafrir | Lord-Rahl, you tried the wrong channel? |
| 15:30.03 | tzafrir | Could you please provide a more complete trace? |
| 15:32.16 | Lord-Rahl | tzafrir: you can see it here http://codepad.org/XwcVIeRz |
| 15:32.21 | *** join/#asterisk medicineman (n=medicine@75.87.82.200) |
| 15:32.22 | Lord-Rahl | wrong one |
| 15:32.32 | medicineman | guten morgen all |
| 15:33.00 | beek | 'ello gov'nor |
| 15:33.32 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 15:35.36 | highvoltz | Will vmware thin-provisioning effect calls within asterisk if virtualized? |
| 15:36.21 | eppigy | <PROTECTED> |
| 15:36.25 | eppigy | y u do dis |
| 15:38.04 | TheDavidFactor | highvoltz? what do you mean? |
| 15:38.35 | TheDavidFactor | are you talking about thin-provisioning the disk device? |
| 15:38.45 | [TK]D-Fender | Lord-Rahl: Go back to #asterisk-gui |
| 15:38.55 | [TK]D-Fender | Lord-Rahl: Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/6000-00000001", "/16163630604") in new stack |
| 15:39.12 | [TK]D-Fender | Lord-Rahl: You didn't specify a trunk in this setup and we do not support the GUI here |
| 15:39.24 | Lord-Rahl | tzafrir:ok |
| 15:39.37 | TheDavidFactor | if so, then not that I know, I've got two asterisk machines I use for development that are both thin provisioned vmware VMs |
| 15:39.45 | *** part/#asterisk Lord-Rahl (n=quassel@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 15:40.11 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:40.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:40.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:40.57 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 15:42.29 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 15:43.13 | Katty | ALLLLLOOOOOOOOO NURSE. |
| 15:44.49 | *** join/#asterisk j_kroon (n=jkroon@dsl-244-25-144.telkomadsl.co.za) |
| 15:45.26 | dlynes | Hello, Nursy Nursy! |
| 15:46.03 | coppice | Katty: are you the Warner sister? |
| 15:46.08 | Katty | sure, why not |
| 15:46.41 | coppice | well, you do get locked up in a tower every night if you are |
| 15:47.26 | Katty | ^_- |
| 15:47.34 | Katty | no thanks then. |
| 15:48.03 | dlynes | wow...amazing |
| 15:48.10 | coppice | I'm not sure actually. do they retreat to the water tower, or get locked in it |
| 15:48.12 | dlynes | 1.6.2.1 hasn't come out yet |
| 15:48.34 | medicineman | how is everyone's morning goin |
| 15:48.38 | dlynes | asterisk must actually be getting more stable, or something |
| 15:49.10 | Naikrovek | of course it is |
| 15:49.58 | Katty | medicineman: kinda meh actually :/ |
| 15:50.10 | dlynes | btw |
| 15:50.17 | medicineman | Katty, why is that? |
| 15:50.24 | dlynes | Has anyone used canaca.com for vps hosting? |
| 15:50.42 | dlynes | Other than voipmonk, that is? |
| 15:51.10 | dlynes | ~seen voipmonk |
| 15:51.11 | infobot | voipmonk is currently on #asterisk (1h 54m 13s), last said: 'deprecated'. |
| 15:51.12 | Katty | medicineman: i don't feel good, but work needed me in this morning to contact one of our vendor software people to sort out some issues. they supposedly have a 4hr turn around time. it's now been almost 2 hours, still no call. |
| 15:51.21 | Katty | medicineman: so i'm sitting here, waiting for a phone call, wanting to go home |
| 15:51.50 | medicineman | Katty: :( that sucks. i've got a feeling they'll be calling any minute |
| 15:52.27 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:53.18 | Carlos_PHX | So our ISP at one facility decided to block all SSH traffic. "For security reasons." |
| 15:53.40 | Katty | medicineman: i sure hope you're right :> |
| 15:53.43 | Chainsaw | Name & shame please, I like to avoid clueless timewasters. |
| 15:53.56 | titter | dlynes: check out hostgator.com |
| 15:54.01 | Katty | Carlos_PHX: ouch )= |
| 15:54.11 | Carlos_PHX | We've never been so secure. |
| 15:54.18 | Katty | i bet. |
| 15:54.21 | Katty | not even YOU can get in |
| 15:54.25 | Carlos_PHX | I can't make changes, and historically, I've broken stuff much more than hackers. |
| 15:54.31 | Chainsaw | Until you set up SSH to run on 443. |
| 15:54.34 | Carlos_PHX | Damn, they are right. |
| 15:54.49 | tzafrir | hmm... set up ssh on port 23? |
| 15:55.12 | Chainsaw | They'd probably close that too. |
| 15:55.12 | tzafrir | It would even be fun |
| 15:55.25 | Chainsaw | But I doubt they'd dare to break HTTPS. |
| 15:55.41 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:55.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:55.45 | Katty | hi Deeewayne |
| 15:56.11 | Deeewayne | Katty, good morning :-) |
| 15:56.37 | Chainsaw | coppice: They get locked in. You see the security guard slamming the door in the intro :) |
| 15:57.27 | Carlos_PHX | It was *only* port 22. |
| 15:57.29 | coppice | I spot a wisely spent youth |
| 15:57.33 | Carlos_PHX | Which made the troubleshooting take forever. |
| 15:57.51 | Carlos_PHX | Assuming it must be me/systems because what kind of enterprise ISP would block 22? |
| 15:58.28 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 15:58.30 | Carlos_PHX | Oh, and it's Time Warner. Who has been absolutely perfect for the last four years. It's a fiber connection into a private facility. |
| 15:58.47 | Chainsaw | Thank you. On the list now. |
| 15:58.51 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:2465:1628:a077:acde) |
| 15:58.52 | cusco | hello |
| 15:58.55 | Carlos_PHX | And only blocked to our BGP-published subnet, not to the native IPs. |
| 15:58.57 | Chainsaw | Hi cusco. |
| 15:58.58 | Katty | ohaider, cusco |
| 15:59.03 | cusco | :-) |
| 15:59.03 | Carlos_PHX | So at least I could SSH to the router. |
| 15:59.07 | Katty | cusco: did you bring me a snack? i'm starving |
| 15:59.18 | cusco | is it OK to have a NoOp(blabla (bla) blah); |
| 15:59.26 | Katty | sure |
| 15:59.28 | cusco | I mean the parentises inside NoOp() |
| 15:59.31 | Chainsaw | Katty: Microwave popcorn okay? |
| 15:59.31 | cusco | cheers |
| 15:59.37 | Katty | Chainsaw: ugah :/ |
| 15:59.42 | Chainsaw | Katty: No? |
| 15:59.52 | cusco | Katty: no, I have candy drops tho :p |
| 16:00.14 | Katty | Chainsaw: i was really hoping for something with cheese |
| 16:00.22 | cusco | hmm :/ |
| 16:00.48 | Chainsaw | Katty: Ham & cheese sandwich? |
| 16:00.56 | Chainsaw | Katty: I think the toaster isn't boxed up yet. |
| 16:02.31 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 16:02.44 | Katty | Chainsaw: :>>>> |
| 16:03.31 | Chainsaw | Katty: Toaster sighting confirmed. It's on the way :) |
| 16:03.42 | Katty | wooooooooot! |
| 16:03.50 | Chainsaw | Katty: (I hate moving, the company kitchen is in shambles. Even the teapot is gone.) |
| 16:03.52 | Ickmund | Is there a way to fire off a command when an agent picks / hangs up a call coming from a queue? |
| 16:04.06 | Katty | Ickmund: call file |
| 16:04.28 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 16:04.33 | Katty | hi CunningPike |
| 16:05.33 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:06.04 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 16:07.43 | Ickmund | Katty, could you elaborate a bit on that? I thought call files were to create outbound calls? |
| 16:08.11 | p3nguin | They are to create calls. |
| 16:09.22 | p3nguin | Creating a call between a channel and an extension... could be an extension that dials outside, dials inside, or some application that doesn't dial at all. |
| 16:09.25 | bmoraca | Ickmund, to my knowledge, the only way to do that is have a 3rd party application monitor AMI. |
| 16:10.33 | Ickmund | bmoraca: ok, thanks |
| 16:11.18 | [TK]D-Fender | Ickmund: Option #2 depends if you use your dialplan to call the member. Dial offers options here, as does the "h" * Standard Extension |
| 16:13.34 | Katty | Ickmund: a call file just starts another leg. |
| 16:13.37 | *** join/#asterisk el_critter (n=voip@200.8.20.159) |
| 16:13.41 | Katty | Ickmund: what you do with that leg is your choice. |
| 16:13.45 | el_critter | Hi |
| 16:13.50 | Katty | Ickmund: if you choose to call out with that leg, super, if not...then do whatever you want. |
| 16:14.14 | Erestar | Anyone have experience using Originate from the manager API? |
| 16:14.17 | Katty | el_critter: ohhaider |
| 16:14.29 | Naikrovek | "ohhaider" lol |
| 16:14.34 | Naikrovek | been a while since i heard that |
| 16:14.35 | el_critter | :) |
| 16:14.54 | Erestar | Am I correct in assuming that I can only originate one call at a time until the previous call is answered? |
| 16:15.21 | [TK]D-Fender | Erestar: No |
| 16:17.28 | Erestar | [TK]D-Fender, I'm piling two originates into the socket in a row, and then reading back... The second call doesn't go through until I answer the first. (I'm testing by calling two separate SIP phones) |
| 16:17.43 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:18.04 | [TK]D-Fender | Erestar: then you're doing it wrong |
| 16:18.08 | Erestar | [TK]D-Fender, Do you have any sense of what I could be doing wrong? Is it hanging on the read or something? |
| 16:18.22 | el_critter | I'm dealing with a T1 seller for the first time and I need help checking compatibility with asterisk, can you help me with that for a minute? |
| 16:18.31 | [TK]D-Fender | Erestar: I also don't see what you're doing.... PASTEBIN is your friend |
| 16:18.33 | [TK]D-Fender | ~pb |
| 16:18.34 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit |
| 16:18.42 | Erestar | Just a sec |
| 16:20.46 | Erestar | [TK]D-Fender, http://pastebin.com/m3296e577 |
| 16:21.19 | Erestar | [TK]D-Fender, The first part is the meat... The call action really just adds lines to a buffer until I send() |
| 16:21.29 | el_critter | Here is the T1 description http://pastebin.ca/1749779 is that compatible with asterisk? |
| 16:21.33 | TheDavidFactor | does anyone know if moh if playing on a channel gets reset or restarted when a dial command is executed with option m? |
| 16:22.07 | p3nguin | Depends on the music. |
| 16:22.42 | [TK]D-Fender | Erestar: you need send them as separate requests. |
| 16:22.48 | [TK]D-Fender | Erestar: Same connection, separate packets |
| 16:23.33 | Erestar | I'm not sure what you mean. I thought \r\n\r\n would indicate a new packet |
| 16:24.19 | bmoraca | el_critter, should work just fine. PRIs are generally easier to deal with, though, and easier to debug than CAS T1s...but it should work fine |
| 16:24.49 | Erestar | Sorry if I'm sounding stupid... This is my first foray into sockets |
| 16:24.55 | [TK]D-Fender | Erestar: I might advise to use the AMI functions in php-agi |
| 16:25.18 | [TK]D-Fender | Erestar: Separate library, plenty of sample code |
| 16:26.50 | Erestar | [TK]D-Fender, I'll take a look, thanks |
| 16:28.22 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 16:28.30 | el_critter | bmoraca: thanks! |
| 16:39.28 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 16:39.32 | _abc_ | hello |
| 16:39.54 | _abc_ | has anyone seen cisco phones with vlan=1 connecting to non-vlan interfaces? |
| 16:40.28 | Erestar | [TK]D-Fender, I just needed to set Async to true |
| 16:40.42 | [TK]D-Fender | Erestar: k |
| 16:41.33 | highvoltz | Anyone reccoemend a cheap SIP trunk thats reliable? |
| 16:42.13 | beek | ~itsp-us |
| 16:42.14 | *** join/#asterisk [netman] (n=netman@193.153.155.169) |
| 16:42.26 | beek | ~itsp |
| 16:42.27 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:42.34 | beek | ~itsplist-us |
| 16:42.35 | infobot | [itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 16:42.43 | beek | highvoltz: ^^^^ |
| 16:42.45 | highvoltz | thx |
| 16:43.22 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 16:43.27 | beek | highvoltz: I use both teliax and vitelity with no problems. |
| 16:43.42 | ManxPower-work | waves to all the n00bs and all the people that won't read the docs. |
| 16:44.00 | _abc_ | waves back wondering what docs ManxPower-work had in mind |
| 16:44.48 | highvoltz | I've been using nexvortex, no problems just looking for alternatives that might be cheaper for a new locatio |
| 16:45.15 | Kobaz | mmm |
| 16:45.18 | Kobaz | sexy new water cooler |
| 16:45.59 | Kobaz | vitelity is cool since you can drop a box in their datacenter so that it's a few dozen feet from pstn termination |
| 16:46.45 | highvoltz | wouldn |
| 16:46.55 | highvoltz | 't need anythign like that, only about 15 users or so |
| 16:47.01 | highvoltz | but that is cool |
| 16:48.27 | *** join/#asterisk funtoo_nbu (i=seb@jet.bayhost.net) |
| 16:48.31 | funtoo_nbu | HAIL!!! |
| 16:49.41 | funtoo_nbu | im a total asterisk newb, its a miracle I even got it functioning. |
| 16:50.02 | [TK]D-Fender | funtoo_nbu: Got any loaves & fishes for the rest of us? |
| 16:50.04 | funtoo_nbu | For local calls, I would like it to autocomplete the area code, to do this I do it in extensions.conf ? |
| 16:50.16 | funtoo_nbu | i hate fish |
| 16:50.20 | funtoo_nbu | its bad for you |
| 16:50.30 | [TK]D-Fender | funtoo_nbu: Make the local pattern and tack on the area code in your dial command |
| 16:51.49 | funtoo_nbu | googling all of that |
| 16:51.49 | funtoo_nbu | thx |
| 16:52.47 | [TK]D-Fender | funtoo_nbu: Whats to google? shorten your full pattern, add the AC directly into the dial |
| 16:54.41 | p3nguin | funtoo_nbu: It is as simple as exten => _NXXXXXX,1,Dial(SIP/1202${EXTEN}@voipms) |
| 16:56.06 | funtoo_nbu | oh snap |
| 16:56.09 | funtoo_nbu | it already has a 1 there |
| 16:56.15 | _abc_ | so really how do people here cope with sccp ? |
| 16:56.15 | funtoo_nbu | assuming the 202 is area code? |
| 16:56.19 | _abc_ | skinny ? |
| 16:56.20 | p3nguin | funtoo_nbu: right |
| 16:56.23 | funtoo_nbu | kewl |
| 16:56.24 | funtoo_nbu | thx |
| 16:56.34 | funtoo_nbu | i was wondering why i ddint have to put a 1 |
| 16:56.37 | funtoo_nbu | now it makes sense |
| 16:56.38 | funtoo_nbu | :D |
| 16:56.54 | p3nguin | funtoo_nbu: Some ITSPs will dial with and without the preceeding 1. |
| 16:57.49 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 16:58.10 | ManxPower-work | _abc_: Usually we just point and laugh at the SCCP/Skinny people and they go off on their own. |
| 16:58.36 | _abc_ | thanks ManxPower-work |
| 16:58.48 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:58.58 | _abc_ | no really these cisco people must have been smoking some bad stuff |
| 16:59.28 | fenrus_ | my 7940 and 7960 work quite well with SIP :) |
| 17:00.04 | _abc_ | fenrus_: Usually I just pont and laught at the SIP/Cisco people and they go off on their own ... |
| 17:00.07 | p3nguin | Mine too, but I've still been contemplating switching to skinny to see what features I will regain. |
| 17:00.12 | _abc_ | chalks up the score as 1:1 |
| 17:00.22 | fenrus_ | _abc_, ;) |
| 17:00.26 | _abc_ | p3nguin: you will regain nothing |
| 17:00.35 | p3nguin | _abc_: That's not what I've heard. |
| 17:00.45 | fenrus_ | p3nguin, you get an unstable pbx using sccp/skinny :D |
| 17:00.50 | _abc_ | but upgrading phones without a contract or in var mode (just testing the things) is not an option |
| 17:00.56 | p3nguin | I can't auto-answer for doing paging/intercom with SIP, for example. |
| 17:01.09 | _abc_ | p3nguin: you can script it i think |
| 17:01.17 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 17:01.22 | _abc_ | but my concern is the cisco people's network magic |
| 17:01.43 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:01.45 | _abc_ | they do things like set VLAN=1 and then ignore the setting on responses so it works on a VALN-less raw if |
| 17:02.05 | _abc_ | my jaw dented the floor when i saw it on a network packet analyzer |
| 17:02.24 | _abc_ | has anyone seen this before? is it a 'feature' ? |
| 17:02.36 | fenrus_ | what do you mean |
| 17:02.59 | Qwell | _abc_: it's a Ciscoism. Take that as you will. |
| 17:03.21 | _abc_ | exactly what i wrote. phone is say 10.0.0.10 802.11q virtual circuit 1 and it talks fine with a pbx at 10.0.0.1 which has no vlan. in both directions. |
| 17:03.34 | ManxPower-work | _abc_: The main issue seems to be less "Cisco phones are crap" and more "Cisco actively discourages people from using their phones with anything but Cisco servers". If figure if they don't want my business there are many other companies that do want my business. |
| 17:03.39 | _abc_ | Qwell: i was just checking whether it was not my sanity that was faulty |
| 17:03.49 | ManxPower-work | Polycom is happy to sell me phones. |
| 17:04.02 | Qwell | when Cisco is involved, your sanity is already in question. |
| 17:04.04 | _abc_ | ManxPower-work: in my situation the pbx is used by a var to test and setup phones |
| 17:04.04 | fenrus_ | vlan1 is considerd the native-vlan |
| 17:04.09 | _abc_ | it has to take them as they come |
| 17:04.21 | _abc_ | fenrus_: 'considered' by the rfc or by cisco? |
| 17:04.27 | fenrus_ | the appliances append no vlan-tags to that packets. |
| 17:04.42 | p3nguin | The native vlan should be whatever vlan you set as the native vlan. |
| 17:04.44 | fenrus_ | _abc_, by default in cisco, can be reconfigured at least in switches. |
| 17:05.17 | funtoo_nbu | do cisco phones work with asterisk? |
| 17:05.22 | p3nguin | yes |
| 17:05.43 | p3nguin | I use 7900 series phones with Asterisk 1.4. |
| 17:05.55 | funtoo_nbu | ah, i got polycoms |
| 17:06.04 | p3nguin | Those work, too. |
| 17:06.11 | funtoo_nbu | yea i made sure :D |
| 17:06.21 | funtoo_nbu | before I spent dollars |
| 17:06.27 | funtoo_nbu | thanks for your help btw |
| 17:06.44 | fenrus_ | hm, i have a cisco conference phone, i've been told it only ran SCCP, been thinking i should set it up some day :) |
| 17:06.48 | p3nguin | If a device talks SIP or IAX2, it should be compatible with Asterisk. |
| 17:08.13 | funtoo_nbu | i assumed they used some proprietary protocol |
| 17:08.43 | funtoo_nbu | likely out of my price range anyway |
| 17:09.09 | fenrus_ | most cisco phones run both SIP and SCCP depending on firmware in the phone |
| 17:09.28 | titter | I prefer snom. |
| 17:09.34 | _abc_ | funtoo_nbu: it is proprietary *.- |
| 17:10.02 | _abc_ | say a good pc based sip phone that is free and supports more than 1 sip line at the same time ? |
| 17:10.12 | fenrus_ | _abc_, windows ? |
| 17:10.15 | _abc_ | i ran through some lists and i am not happy |
| 17:10.18 | _abc_ | yes windows |
| 17:10.21 | _abc_ | xlite does one line |
| 17:10.44 | _abc_ | nch has a good sip phone but one must register it after 30 days and it has no chat (text chat) |
| 17:11.06 | _abc_ | wxcommmunicator crashed someone's pc |
| 17:11.07 | drmessano | How are you gonna chat with the others? |
| 17:11.10 | _abc_ | (not mine) |
| 17:11.18 | fenrus_ | xlite works quite well on macosx |
| 17:11.34 | _abc_ | how much is a full xlite license ? |
| 17:11.45 | p3nguin | _abc_: zoiper is a good one. |
| 17:12.02 | _abc_ | zoiper does not support more than one line afaik |
| 17:12.09 | p3nguin | Sure it does. |
| 17:12.15 | p3nguin | I think it supports 6 or 8. |
| 17:12.15 | _abc_ | people seem to botch the more than one line thing with boring regularity |
| 17:12.26 | _abc_ | p3nguin: are you sure? |
| 17:12.31 | fenrus_ | _abc_, http://www.counterpath.com/xlite-comparison.html |
| 17:12.33 | _abc_ | i uninstalled zoiper after testing it |
| 17:12.38 | p3nguin | Load it up. The line buttons are right on the screen. |
| 17:12.40 | p3nguin | oh |
| 17:12.43 | p3nguin | Let me find a pic. |
| 17:13.34 | p3nguin | http://www.zoiper.com/softphone/ |
| 17:13.42 | p3nguin | It shows 6 line buttons. |
| 17:14.46 | p3nguin | Zoiper has had less problems than all the other free ones I tested. |
| 17:15.15 | Kobaz | twinkle ftw |
| 17:15.28 | p3nguin | I never tried twinkle on Windows. |
| 17:15.30 | Kobaz | zoiper free is annoying with it's two account limit |
| 17:15.41 | p3nguin | "its" |
| 17:17.11 | kc8pxy | softphones? |
| 17:17.22 | kc8pxy | i like sipdroid :) |
| 17:17.22 | voipmonk | its free, Kobaz |
| 17:17.24 | *** part/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 17:17.27 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 17:18.28 | kc8pxy | p3nguin: even xlite? |
| 17:18.29 | Kobaz | voipmonk: twinkle is free too, and it doesn't have any limits |
| 17:18.36 | drmessano | hates any phone that announces its name |
| 17:18.47 | p3nguin | kc8pxy: X-Lite is really terrible. |
| 17:18.52 | drmessano | DROOOOID... PLAAAAAYEDOUT |
| 17:18.55 | ManxPower-work | hates any phone that relies on Windows for everything. |
| 17:19.05 | Kobaz | yeah the configuration for xlite is a bit annoying |
| 17:19.24 | drmessano | X-lite just works |
| 17:19.32 | drmessano | Not fancy, but it works |
| 17:19.33 | Kobaz | except when it crashes |
| 17:19.36 | p3nguin | X-Lite is very resource intensive, for who knows what reason. |
| 17:19.42 | drmessano | Never had that problem |
| 17:20.06 | ManxPower-work | p3nguin: if they removed that crappy eye candy interface it might run fast. |
| 17:20.19 | kc8pxy | has not found a better softphone for android. |
| 17:20.24 | p3nguin | I've tried it on several computers, and it has been bad on all. |
| 17:20.46 | drmessano | 3.x or 4.x? |
| 17:20.47 | p3nguin | manxpower-work: The older version was less of a hog than the new "black" one. |
| 17:21.03 | drmessano | Oh we're talking about 4 |
| 17:21.05 | p3nguin | I guess both. |
| 17:21.10 | ManxPower-work | p3nguin: I'll stick to Polycom hardphones and just avoid all those issues |
| 17:21.10 | drmessano | 3.x works fine |
| 17:21.35 | p3nguin | I think 4 is the black one that brings explorer.exe to a crawl. |
| 17:22.02 | _abc_ | why don't i see a windows link here: http://www.zoiper.com/download_list.php ? |
| 17:22.17 | drmessano | Never used 4... the whole "BETA" thing made me feel like it may be in BETA |
| 17:22.27 | p3nguin | manxpower-work: Sounds like a good plan to me. It's certainly cheaper to by a phone than a computer to use a free softphone. |
| 17:22.50 | Qwell | _abc_: because you're running Linux, and aren't seeing the link to show other OSes |
| 17:23.16 | p3nguin | That's retarded. |
| 17:23.28 | drmessano | So are end users |
| 17:23.31 | ManxPower-work | p3nguin: and I don't even need my computer powered up to make phone calls! |
| 17:23.33 | _abc_ | Qwell: yeah 'clever' interfaces |
| 17:23.40 | p3nguin | They shouldn't hide links based on what your browser sends as the OS. |
| 17:23.48 | _abc_ | clicks on the user agent setting button in ff |
| 17:24.09 | _abc_ | bingo |
| 17:24.09 | p3nguin | http://www.zoiper.com/download_list.php?os=All |
| 17:24.23 | p3nguin | You could have just clicked the link right below the Linux link. |
| 17:24.33 | drmessano | Yeah |
| 17:24.34 | drmessano | lol |
| 17:24.42 | drmessano | "Zoiper for other operating systems" |
| 17:24.53 | drmessano | Damn them for making that sound so cryptic |
| 17:25.03 | _abc_ | umm i only get zoiper communicator. i want the classic |
| 17:25.12 | Qwell | drmessano: They should have listed the other OSes |
| 17:25.26 | drmessano | "Zoiper 2.0" --> SCROLL |
| 17:25.34 | p3nguin | Make sure you use Zoiper 2.0 and not communicator (unless you want the full communicator instead of just the phone). |
| 17:26.32 | _abc_ | i just need the phone. does it have chat? text chat? |
| 17:26.44 | p3nguin | The phone does not, but communicator does. |
| 17:26.49 | _abc_ | sigh |
| 17:26.52 | _abc_ | ok, whatever |
| 17:27.06 | p3nguin | I thought you wanted to talk on the phone, not type... |
| 17:27.19 | Qwell | who talks on phones anymore? |
| 17:27.21 | _abc_ | sometimes i need to send urls and such when debugging stuff, this is not for me |
| 17:27.38 | _abc_ | i just need a pc sip phone that works, period. zoiper seems to be it |
| 17:27.41 | *** join/#asterisk [netman] (n=netman@193.153.155.169) |
| 17:28.15 | drmessano | I dont want a softphone that pretends to be an IM client and vice versa |
| 17:28.35 | drmessano | Just means it will do one thing well, the other thing well, or neither well |
| 17:28.38 | _abc_ | i want a softphone that can do the equivalent of sms, not more |
| 17:28.48 | drmessano | Now you want SMS |
| 17:28.56 | _abc_ | it's the state of the art. we are in 2010 not 1910 |
| 17:28.57 | Qwell | SMiShes drmessano |
| 17:29.36 | coppice | I wonder which art SMS is the state of? :-\ |
| 17:30.07 | coppice | but it is on pretty much all phones these days |
| 17:30.15 | _abc_ | quit joking. i want to see you do support and 'sending' a url to a luser using pidgin nato phonetic alphabet (albert berta charlie ...) |
| 17:30.22 | Qwell | coppice: Van Gogh |
| 17:30.42 | coppice | Salvador Dali sounds closer |
| 17:30.58 | _abc_ | coppice: he did not have a cut ear |
| 17:31.16 | _abc_ | but i think that Gaudi was involved with the GUI on many phones ... |
| 17:31.20 | drmessano | I guess my pidgin is broken.. URLs work fine |
| 17:31.20 | coppice | _abc_: this is why everyone needs wideband voice |
| 17:31.45 | _abc_ | coppice: so you can scream louder and they can hear when you whisper to your colleague 'what an idiot' ? |
| 17:32.14 | _abc_ | drmessano: how do you dictate a url to a lady who is looking for the [Enter] key and cannot find it ? |
| 17:32.42 | Qwell | How do you get a lady who can't find the Enter key to use a softphone? |
| 17:32.56 | Naikrovek | get her a real phone |
| 17:32.57 | Naikrovek | ? |
| 17:32.59 | drmessano | You said PIDGIN |
| 17:33.06 | drmessano | You PASTE it in the window |
| 17:33.10 | _abc_ | it comes up when she turns on the computer. ignore the old lady it was just a bad example |
| 17:33.19 | _abc_ | drmessano: not that pidgin |
| 17:34.14 | drmessano | Pigeon? |
| 17:34.30 | _abc_ | pidgin <language> is an expression. look it up |
| 17:34.56 | _abc_ | on google: "pidgin latin" |
| 17:35.22 | drmessano | My google is broken and Bing keeps trying to sell me Vista upgrades. Will take your word for it |
| 17:36.15 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:36.57 | _abc_ | drmessano: well then use that chinese boing or whatever ;) |
| 17:37.13 | _abc_ | baidu |
| 17:38.13 | p3nguin | So what is the real goal? A softphone, an IM, an SMS? |
| 17:38.58 | _abc_ | p3nguin: my goal has been met, i am fishing for a freebie sip phone which preferrably has chat (sms) |
| 17:39.16 | p3nguin | and NATO phonetic alphabet is alpha bravo charlie. |
| 17:39.24 | p3nguin | (jsyk) |
| 17:39.31 | raden_work | *yawn* |
| 17:39.35 | coppice | haha. www.百度.com does not resolve to www.baidu.com. So much for i18n domain names |
| 17:39.49 | _abc_ | by the way asterisk -rx "logger rotate" will keep creating new suffixes forever? is there a config option to make it stop at a certain number? |
| 17:40.11 | _abc_ | p3nguin: the *pidgin* nato phonetic alphabet is as i showed |
| 17:40.13 | p3nguin | Let logrotate purge them. |
| 17:41.08 | p3nguin | I guess I got lost on the same road as drmessano when you said pidgin. |
| 17:41.14 | _abc_ | coppice: baidu.cn ? |
| 17:41.47 | _abc_ | what does baidu mean? p0wn3d by the ccp ? |
| 17:42.40 | coppice | www.百度.cn goes to yet another web site which is not the real baidu |
| 17:42.58 | _abc_ | coppice: that's where they install the party compliance software |
| 17:43.31 | *** part/#asterisk dustybin (i=dustybin@thinkdebian.org) |
| 17:44.45 | _abc_ | back to asterisk: using a call file, can i start a context *without* creating a channel and calling it? perhaps by using a http://www.voip-info.org/wiki/view/Asterisk+local+channels ? |
| 17:45.09 | _abc_ | and put a Dial command there, after I run my scripts ? |
| 17:45.42 | Qwell | _abc_: mmhmm |
| 17:46.23 | _abc_ | so when i specify a call file with a Local target then i still have to give it the extension and all the data but it won't call, right ? |
| 17:46.44 | _abc_ | is there a cleaner way to do it? callfile that does not start a call but starts a context ? |
| 17:46.47 | sassyn | tzafrir, 10x 4 the help |
| 17:48.25 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 17:49.23 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 17:49.24 | *** mode/#asterisk [+o file] by ChanServ |
| 17:50.07 | ManxPower-work | Local/extension@context is how you "dial" a local channel |
| 17:51.16 | ManxPower-work | _abc_: It's a fscking CALL FILE. It will always generate a call. |
| 17:52.34 | *** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:53.07 | _abc_ | i hear your loud voice ;) |
| 17:54.17 | ManxPower-work | _abc_: BTW, the docs I was referring to are in /path/to/src/asterisk/doc/ |
| 17:55.13 | _abc_ | i know |
| 17:59.59 | p3nguin | Contexts aren't started. Contexts are parts of your extensions.conf that group rules together. |
| 18:00.44 | _abc_ | ok well making a 'call' to a context/extension/priority in Local will do what I need |
| 18:01.04 | p3nguin | great |
| 18:01.07 | p3nguin | That reminds me: I need a good bread recipe. |
| 18:01.25 | _abc_ | what is the highest vlan circuit ever seen ? i think i saw 15 once |
| 18:01.32 | _abc_ | should to to 65535 no ? |
| 18:01.34 | ManxPower-work | I usually use vlan 42 |
| 18:01.37 | _abc_ | or 32767 |
| 18:01.48 | benngard | vlan id is 12 bits |
| 18:01.52 | _abc_ | p3nguin: what triggered that association |
| 18:02.00 | _abc_ | benngard: thanks, integer or unsigned |
| 18:02.08 | benngard | unsigned |
| 18:02.11 | _abc_ | is preparing for more 'cisco' like 'logic' |
| 18:02.12 | benngard | 0-4095 |
| 18:02.15 | _abc_ | ok |
| 18:02.30 | *** join/#asterisk netpro25_ (n=mmanning@64-238-176-248.ksg.apt.gru.net) |
| 18:02.31 | fenrus_ | 4094 vlans is available |
| 18:02.40 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 18:02.40 | *** mode/#asterisk [+o file] by ChanServ |
| 18:02.51 | netpro25_ | Anyone know how to get the message light working on the spa941? |
| 18:02.55 | benngard | the rest 4 bits are for qos (3 bits) cant remember the fourth |
| 18:02.58 | netpro25_ | It was working then it stopped. |
| 18:03.01 | [TK]D-Fender | _abc_: He uses Asterisk. Asterisk consulting generates income. Income = money. Bread is a slang term for money. |
| 18:03.17 | [TK]D-Fender | _abc_: You such at charades. Remind me not to pick you as my parter :p |
| 18:03.21 | [TK]D-Fender | suck* |
| 18:03.31 | benngard | dont use 0 and 4095 normally resrved |
| 18:03.50 | fenrus_ | benngard, CFI |
| 18:04.01 | carrar | VLAN 69 is where all the action happens! |
| 18:04.18 | [TK]D-Fender | carrar: Doesn't that cause a loopback? |
| 18:04.23 | carrar | haha |
| 18:04.29 | carrar | yeah |
| 18:04.46 | fenrus_ | benngard, used for tokenring shiznit.. not in use that much anymore :) |
| 18:05.04 | netpro25_ | Nm got it |
| 18:05.31 | ManxPower-work | 68 is better. That is "do me and I'll owe you 1" |
| 18:05.42 | benngard | :) |
| 18:06.33 | _abc_ | [TK]D-Fender: i don't suck in general, and i'm good at charades in languages other than english, which is not my native language |
| 18:07.05 | benngard | _abc_: be careful when playong around with vlan, both cisco and hp (small switches) have very well known limitations of how many vlan u can use per switch |
| 18:07.17 | _abc_ | benngard: thanks |
| 18:07.20 | [TK]D-Fender | _abc_: What is? |
| 18:07.42 | _abc_ | romanian is one of them |
| 18:07.56 | [TK]D-Fender | _abc_: What others? |
| 18:08.20 | _abc_ | asr check ? |
| 18:08.29 | benngard | normally 31 vlan per switch, normally not a problem but if u really "vlan up" your network u can ran into som e nasty problem |
| 18:10.00 | fenrus_ | if you really need 31 vlans, you probably need something a little bit more competent than the cheapest 2900XL-Switch |
| 18:10.22 | _abc_ | i can go higher than that, catalyst and all that |
| 18:10.27 | _abc_ | but not for this project |
| 18:10.44 | benngard | thats very correct but i have seen (to often) people running int problem with that stuff |
| 18:11.10 | bmoraca_work | woohoo! fixed by blank callerid routing problem! |
| 18:11.19 | _abc_ | i am trying to keep it as simple as possible, i just need to cope with crazy settings on inoming phones before they are configured |
| 18:11.25 | benngard | and its not fun to go to your boss and ask fort a 35xx or so |
| 18:11.45 | _abc_ | well just today i was *offered* to use those if needed ;) |
| 18:12.08 | bmoraca_work | 24pt 3550PWRs are like $220 on ebay |
| 18:12.18 | benngard | take the 35xx with ppoe support |
| 18:12.23 | benngard | poe* |
| 18:12.47 | benngard | i use them for alot of projects, costs like shit but do the trick |
| 18:14.26 | benngard | i guess _abc_ just "changed" switch! ;) |
| 18:14.32 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 18:14.32 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 18:15.06 | _abc_ | so who likes pjproject-1.5.5 cli phone for debugging? i think it's lame? |
| 18:15.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.19 | _abc_ | the cli phone works but sometimes it does not play nice with * |
| 18:17.46 | _abc_ | anyway setting debug options high enough yields a huge amount of useful data which can be captured as a text file and read |
| 18:17.49 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 18:18.03 | _abc_ | so client side sip registration and the like can be debugged |
| 18:20.15 | benngard | btw, read a pretty fun story (in swedish) gonna try to translate it for u: bark = rapa (swedisj) a guy (swedish) had a beatiful blond at his table, ofc he did bark, so he sad: sorry ma'm i raped u |
| 18:23.31 | fenrus_ | and in swedish? ;) |
| 18:23.34 | p3nguin | I'm guessing it loses effect in the translation. |
| 18:24.35 | benngard | ursäkta jag rapade/våldtog dig |
| 18:25.13 | fenrus_ | jag som brukar gilla ordvitsar.. |
| 18:25.21 | benngard | lol |
| 18:25.56 | benngard | atleast 2 seds here ;) |
| 18:26.00 | benngard | sweds |
| 18:29.07 | netpro25_ | Well I thought I fixed the message waiting light but I did not. How can I get this working on an SPA941? |
| 18:29.24 | ManxPower-work | netpro25_: it works by default. |
| 18:29.37 | netpro25_ | hmmm, must have screwed something up |
| 18:29.44 | ManxPower-work | Obviously |
| 18:29.47 | netpro25_ | haha |
| 18:30.00 | netpro25_ | anything in voicemail.conf that needs to be set? |
| 18:30.33 | ManxPower-work | nope, just in sip.conf mailbox=vmbox@vmcontext for the peer. |
| 18:30.34 | fenrus_ | anyone got good experience with linksys pap2t ? :) |
| 18:31.05 | fenrus_ | need to configure it to sound swedish ;) |
| 18:31.24 | ManxPower-work | fenrus_: Some sub-models are locked. |
| 18:32.05 | netpro25_ | ManxPower-work: hah, that was the missing piece |
| 18:32.10 | netpro25_ | thanks |
| 18:35.09 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 18:35.18 | _abc_ | fenrus_: just jumble the characters in the text |
| 18:41.10 | [TK]D-Fender | BORK! BORK! BROK! |
| 18:41.17 | [TK]D-Fender | BORK!* |
| 18:41.53 | p3nguin | Thought you were mutating into a chicken, for a second. |
| 18:43.04 | leifmadsen | BRAWWWK! |
| 18:43.31 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 18:44.51 | ManxPower-work | netpro25_: Um, virtually every single document on the internet about Asterisk voicemail would have mentioned the mailbox option |
| 18:45.06 | netpro25_ | yea, I feel like a dumbass |
| 18:45.15 | netpro25_ | heh |
| 18:46.18 | netpro25_ | While you are here I have a question |
| 18:47.34 | netpro25_ | ManxPower-work: With my SPA941 and PAP2's whether the ones I have here or at some one elses house. When the dynamic ISP ip changes they loose connection and go to registration failure. I have to power then down for about 5 minutes before they work again. Does this sound familiar? |
| 18:48.54 | netpro25_ | or do you have an idea on what might be happening |
| 18:49.27 | ManxPower-work | stop running your server on a dynamic ip? |
| 18:52.58 | ManxPower-work | Does anyone have any ideas if you can access the content of the Facility messages that asterisk might receive after call setup on a PRI? |
| 18:53.15 | netpro25_ | ManxPower-work: not the server the clients |
| 18:53.34 | ManxPower-work | netpro25_: do the PCs have internet access issues too? |
| 18:53.39 | netpro25_ | they are behind a NAT, nope |
| 18:53.46 | ManxPower-work | remember the phones don't know the address changed. |
| 18:53.50 | netpro25_ | right |
| 18:53.54 | netpro25_ | STUN? |
| 18:53.57 | ManxPower-work | do you have ANY nat stuff enabled on the phones themselves? |
| 18:54.03 | netpro25_ | yes |
| 18:54.31 | ManxPower-work | TURN IT OFF! (You should be able to enable any "NAT Keepalive" options, other than that disable all NAT stuff on the phones. |
| 18:54.42 | ManxPower-work | netpro25_: Are you doing any port forwarding on the phones? |
| 18:54.45 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 18:55.00 | netpro25_ | no, just using NAT enabled settings |
| 18:55.03 | ManxPower-work | sorry, port forwarding on the nat router FOR the phones. |
| 18:55.25 | ManxPower-work | make sure you turn off any SIP NAT support on the NAT router as well. |
| 18:55.31 | bmoraca_work | ManxPower-work, i've never seen anything that suggests you can access facility messages after setup on a PRI |
| 18:55.45 | bmoraca_work | using asterisk, that is |
| 18:55.46 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 18:56.04 | ManxPower-work | bmoraca_work: Me neither. However, some carriers send the callerid name message after the call setup and asterisk receives that and deals with it. |
| 18:56.48 | bmoraca_work | ManxPower-work, i've only ever had callerid name work when passed in setup. maybe wanrouter can do it, but i don't believe dahdi can. |
| 18:57.12 | bmoraca_work | ManxPower-work, my feeling is that this is because most of the facility messages after setup are proprietary |
| 18:57.13 | ManxPower-work | bmoraca_work: you know how some people have to add a Wait(1) or similar when using a PRI or they won't get the clidname? |
| 18:57.24 | bmoraca_work | ahh |
| 18:57.27 | bmoraca_work | makes a little sense |
| 18:57.40 | ManxPower-work | when you need to do that, then you are getting the name in a message other than the setup message |
| 18:57.50 | netpro25_ | ManxPower-work: router does not have any options like that so they are disabled. |
| 18:58.05 | devmod | has anyone here successfully transcoded video to be used in asterisk with ffmpeg? These options dont seem to work with Playback() : http://pastebin.com/m4200b366 - any ideas? |
| 18:58.40 | bmoraca_work | best solution, i suspect, is to call the provider and have them send it in setup. however, it might be nice to be able to access those messages in order to access some of the proprietary features other PBXes might use |
| 18:58.54 | ManxPower-work | bmoraca_work: in theory a "number changed" cause code might have a message after it indicating what the number was changed to. Just thought it would be cool if I could access the info if available. |
| 18:59.11 | ManxPower-work | bmoraca_work: I totally agree with you. |
| 18:59.13 | bmoraca_work | ahhh, yeah, that'd be neat, too |
| 19:00.04 | ManxPower-work | netpro25_: configure the phones for a 60 second registration interval |
| 19:00.39 | netpro25_ | ManxPower-work: yea read that somewhere but it said that there are side effects |
| 19:01.05 | netpro25_ | I enabled stun, we will see if that helps |
| 19:01.08 | ManxPower-work | netpro25_: you will NOT have a "perfect" setup when you have dynamically changing IP addresses. |
| 19:01.14 | netpro25_ | and then if now I will try that |
| 19:01.21 | netpro25_ | if not that is |
| 19:01.31 | netpro25_ | ManxPower-work: yea noticed that |
| 19:02.18 | bmoraca_work | netpro25_, if you can, you can set up a dynamic DNS service with a very low TTL and direct the endpoints to that hostname instead of your IP directly |
| 19:02.26 | bmoraca_work | it's hokey, but it works |
| 19:02.48 | netpro25_ | bmoraca_work: thanks I will give that a shot |
| 19:02.59 | bmoraca_work | good luck, lol |
| 19:03.07 | bmoraca_work | static ip is simpler by far :) |
| 19:03.13 | netpro25_ | it is |
| 19:03.21 | netpro25_ | *that it is |
| 19:03.29 | ManxPower-work | netpro25_: you're going to generate network traffic to keep the NAT translations open. You can set a low registration interval or not. |
| 19:04.01 | netpro25_ | what about not being able to make calls during registration? |
| 19:04.15 | bmoraca_work | netpro25_, don't need to be registered to make calls |
| 19:04.24 | netpro25_ | so enable that option |
| 19:04.25 | bmoraca_work | all registration does is tell the proxy how to find you |
| 19:04.30 | netpro25_ | oh ok |
| 19:04.43 | netpro25_ | makes more sense now |
| 19:04.50 | ManxPower-work | Yes. So when your IP address changes, Asterisk won't know about it until the next time the phone registers..... |
| 19:04.58 | netpro25_ | got it |
| 19:05.05 | netpro25_ | so what should I set it to? |
| 19:05.09 | netpro25_ | 60 seconds? |
| 19:05.14 | bmoraca_work | and if asterisk's ip changes, the phones don't know about that until you tell them |
| 19:05.16 | ManxPower-work | That's what I'd do. |
| 19:05.23 | netpro25_ | wonderfull |
| 19:05.44 | bmoraca_work | which is where dynamic dns can help |
| 19:05.45 | ManxPower-work | netpro25_: 60 second registration generates extra traffic, but not any more so than qualify=yes or STUN. |
| 19:06.20 | netpro25_ | would that happen to be Reg Retry Long Intvl: or Reg Max Expires |
| 19:06.35 | ManxPower-work | netpro25_: I have no idea. |
| 19:06.56 | netpro25_ | I will google it |
| 19:06.58 | ManxPower-work | I don't deal with dynamic IPs much anymore so I don't have to know those config options on phones |
| 19:07.30 | bmoraca_work | dynamic ips are aweful |
| 19:07.40 | netpro25_ | they are so 90's |
| 19:07.46 | ManxPower-work | netpro25_: another option is VPN between the two locations. Just be sure to REALLY understand the issues you might have with different types of VPN. |
| 19:08.01 | bmoraca_work | vpn doesn't solve the dynamic ip issue either |
| 19:08.02 | netpro25_ | ManxPower-work: yea delays etc |
| 19:08.22 | ManxPower-work | bmoraca_work: no, but VPN endpoints tend to EXPECT IP address changes. |
| 19:08.22 | bmoraca_work | netpro25_, not delays, but 10-20 bytes of extra traffic per packet |
| 19:08.30 | netpro25_ | ah ok |
| 19:09.14 | *** join/#asterisk [netman] (n=netman@193.153.155.169) |
| 19:09.15 | bmoraca_work | ManxPower-work, maybe some of the SOHO ones...but Cisco ASAs and routers need at least one end to be static |
| 19:09.24 | ManxPower-work | I'm suprized at how well my SIP over IPSec works. |
| 19:09.36 | ManxPower-work | bmoraca_work: netpro's server is on static |
| 19:10.10 | bmoraca_work | SIP over IPSec works great...but IPSec does add some bytes to the header, requiring more bandwidth per call |
| 19:10.19 | netpro25_ | bmoraca_work: yes, thanks so much for the help. I gotta run |
| 19:10.20 | netpro25_ | ttyl |
| 19:10.20 | ManxPower-work | obviously you do NOT a VPN over SSL or VPN over TCP |
| 19:10.21 | bmoraca_work | ManxPower-work, ahh...simplicity in itself then. |
| 19:10.28 | netpro25_ | ManxPower-work: thank you also |
| 19:10.48 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:11.34 | bmoraca_work | i run some SIP traffic through an unencrypted GRE tunnel, and that works just fine. i've also installed systems which have 4-5 offices working off a single PBX over IPSec VPNs...they work great |
| 19:11.55 | Naikrovek | same. sip over ipsec works very well |
| 19:12.26 | ManxPower-work | bmoraca_work: I loved GRE tunnel wrapped in IPSec VPN when I used that. You can OSPF routing info over such links |
| 19:12.29 | sassyn | I need some help here |
| 19:12.36 | sassyn | I have asterisk 1.6.20 |
| 19:12.38 | sassyn | working |
| 19:12.41 | sassyn | everything is ok |
| 19:12.47 | ManxPower-work | sassyn: I think you are missing a . |
| 19:12.53 | sassyn | but when dailing a wrong number |
| 19:13.18 | sassyn | Playing 'cannot-complete-as-dialed.ulaw' can't be heard on the phone |
| 19:13.28 | bmoraca_work | ManxPower-work, that's exactly why I'm using GRE. i'd like to encrypt it, but I can't take the routers down long enough to install the IOS which has crypto features...i didn't set them up... |
| 19:14.36 | ManxPower-work | I mainly wanted encryption, but also wanted to send routing updates over the link. |
| 19:15.05 | bmoraca_work | GRE is fun |
| 19:15.26 | bmoraca_work | routing is fun |
| 19:15.38 | bmoraca_work | more fun than voice, anyway :P |
| 19:15.41 | sassyn | any idea anyone? |
| 19:17.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.32 | fenrus_ | bmoraca_work, just add the correct ios and next time they reboot you have it.. :) |
| 19:17.59 | fenrus_ | gre does the job, aswell as IPIP, but i like some encrypted solution better.. |
| 19:18.12 | bmoraca_work | they don't reboot...one of them was up for 4 years |
| 19:18.29 | fenrus_ | ipsec+gre is nice, but if you need some capacity you need aim-vpn cards.. and that's expensive. |
| 19:18.44 | fenrus_ | bmoraca_work, then you have the correct ios in a couple of years then ? |
| 19:18.51 | bmoraca_work | lol |
| 19:18.53 | fenrus_ | or just create a planned-maintainence-window |
| 19:19.15 | fenrus_ | we have routers at work with uptimes over 10 years. |
| 19:19.58 | *** join/#asterisk jro (n=jaredo@216.145.224.138) |
| 19:20.41 | jro | If I need to make 10 concurrent outbound calls, do I need 10 sip trunks or can they all go over one? |
| 19:20.55 | fenrus_ | one trunk is enough |
| 19:21.00 | fenrus_ | the bandwidth is the limit |
| 19:21.06 | jro | Say if x company sells me an unlimited trunk for $30/mo |
| 19:21.47 | jro | I know the bandwidth on my end wouldnt be an issue but I'm not sure of the sip provider |
| 19:21.53 | [TK]D-Fender | jro: it means you'd better get more details on what that service offers you |
| 19:22.05 | jro | They told me I should have a sip trunk for each concurrent call lol |
| 19:22.15 | [TK]D-Fender | jro: How the hell is it you should be worried about your PROVIDER'S bandwidth? |
| 19:22.16 | fenrus_ | sounds odd |
| 19:22.41 | jro | [TK]D-Fender: I'm on a 100mb opteman |
| 19:22.57 | jro | hmm |
| 19:22.58 | [TK]D-Fender | jro: plenty. BW isn't an issue then |
| 19:22.59 | *** join/#asterisk [netman] (n=netman@193.153.155.169) |
| 19:23.11 | bmoraca_work | jro, you do need a sip trunk for each concurrent call. they're not going to let you make more simultaneous calls than you're paying for |
| 19:23.38 | jro | How can they monitor that if only 1 did? |
| 19:23.59 | bmoraca_work | jro, seriously? it's not hard |
| 19:24.00 | jro | ah nm |
| 19:24.16 | [TK]D-Fender | [14:23]<bmoraca_work>jro, you do need a sip trunk for each concurrent call. they're not going to let you make more simultaneous calls than you're paying for <- we don't know how many he's paying for .. thats the point |
| 19:24.27 | [TK]D-Fender | jro: Go read what that product they are selling PROVIDES |
| 19:24.30 | *** join/#asterisk dgit (n=chatzill@67.200.59.158) |
| 19:24.52 | jro | Can you recommended any known good sip trunk providers? |
| 19:25.04 | bmoraca_work | [TK]D-Fender, agreed...but by his comments, it seems he's trying to get away with one $30/mo unlimited channel to handle multiple simultaneous calls |
| 19:25.29 | [TK]D-Fender | mbranca: Which may be possibly |
| 19:25.32 | [TK]D-Fender | e |
| 19:25.35 | [TK]D-Fender | akshdash |
| 19:25.37 | jro | bmoraca_work: I'm not, I just wanted to make sure they weren't trying to charge me 20 trunks if it was unnecessary |
| 19:25.59 | [TK]D-Fender | jro: Why don't you show us what they SAY they are selling you |
| 19:26.12 | bmoraca_work | jro, if you need 20 simultaneous calls, you probably will need to pay 20 * $30/mo |
| 19:26.21 | [TK]D-Fender | bets on otherwise |
| 19:27.21 | bmoraca_work | [TK]D-Fender, would you allow a customer to place 20 simultaneous calls while only paying for a single $30/mo channel? |
| 19:27.21 | jro | bmoraca_work: thanks, was just checking |
| 19:27.47 | [TK]D-Fender | jro: no, KEEP checking. You haven't provided anything solid yet |
| 19:28.00 | ManxPower-work | jro: most of the time per-min plans are cheapest |
| 19:28.40 | [TK]D-Fender | jro: And I don't see a statement of your expected usage so far... |
| 19:29.40 | ManxPower-work | jro: also remember Unlimited plans are NOT actually Unlimited. |
| 19:30.54 | jro | ManxPower-work: could you elaborate please? |
| 19:31.42 | ManxPower-work | jro: they all have caps. IIRC Teliax is 1,500 (res) or 3,000 (business) mins month. Other carriers have similar caps on their Unlimited plans. |
| 19:32.24 | ManxPower-work | Read the fine print. |
| 19:32.47 | [TK]D-Fender | jro: Go link us to their product page or something. We are getting tons of hearsay and unvalidated advice. |
| 19:33.22 | jro | [TK]D-Fender: http://www.bandwidth.com/sip-trunking/ |
| 19:34.05 | *** join/#asterisk profxavier (n=chatzill@unaffiliated/neverblue) |
| 19:34.37 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:34.41 | profxavier | i just purchased a SoundsStation2, I noticed the POE device has three plugs, 2 ethernet and one phone |
| 19:35.04 | profxavier | can I use a VOIP system to connect, or is it strictly for 'regular' phone service ? |
| 19:35.31 | jro | ManxPower-work: Is that inbound outbound or both? |
| 19:35.38 | ManxPower-work | bandwidth.com has a 1,500 min cap on their Unlimited Inbound service |
| 19:35.41 | [TK]D-Fender | jro: they do not expressly say. Go ask them for full details of services supported including "call-waiting", etc |
| 19:36.00 | ManxPower-work | http://www.bandwidth.com/assets/pdf/legal/SIP_Trunking/Bandwidth_SIP_trunking_inbound_terms.pdf |
| 19:36.00 | jro | ah |
| 19:36.22 | ManxPower-work | jro: where are you located? |
| 19:36.32 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
| 19:36.38 | titter | Anyone have or know of a tool that will let me edit my conf files within windows without having to use WinSCP |
| 19:36.44 | jro | Assuming they'd pool all the trunks, if we had 10 trunks I'd assume 15000 minutes? |
| 19:36.54 | [TK]D-Fender | ManxPower-work: Not applicable |
| 19:37.01 | ManxPower-work | jro: why would you assume they pool the mins? |
| 19:37.02 | killfill | does the wct4xxp driver of dhadhi works with a TE110P?.. |
| 19:37.03 | jro | ManxPower-work: chicago |
| 19:37.09 | [TK]D-Fender | titter: PuTTY <- |
| 19:37.20 | ManxPower-work | killfill: Does the DAHDI README say it does? |
| 19:37.35 | ManxPower-work | 'cause that's where all the card/driver combos are listed. |
| 19:37.46 | killfill | ooh sorry. |
| 19:38.14 | bmoraca_work | /etc/dahdi/modules.conf also will tell you generally which module goes with which card |
| 19:38.28 | ManxPower-work | jro: You are using this channel as a sales person for bandwidth.com service. Stop. Go ask a bandwidth.com sales rep about their service. |
| 19:38.45 | titter | [TK]D-Fender: thats what I use now ssh then vim the files, but I rather have a better editor that is more windows friendly ... I need to copy and past large chunks of my dialplan |
| 19:39.06 | titter | vim works, but just wondering if anyone had a preference |
| 19:39.07 | ManxPower-work | titter: jedit and many others support open/save over SSH/SFTP. |
| 19:39.16 | titter | ManxPower-work: thanks |
| 19:39.29 | ManxPower-work | you need the FTP plugin (!) for SSH/SFTP open/save with jEdit. |
| 19:39.31 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 19:40.04 | titter | ManxPower-work: thanks again, will check it out |
| 19:40.08 | ManxPower-work | I dislike jEdit because it's in Java. I like jEdit because it runs on Windows AND Linux and it works well. |
| 19:40.15 | bmoraca_work | titter, what's wrong with winscp? |
| 19:41.06 | titter | bmoraca_work: nothing, but I like a put/get system similar to dreamweaver. |
| 19:41.18 | jro | ManxPower-work: Actually I was just simply asking if it was necessary to have a sip trunk per concurrent channel, any references to the company you mentioned were in response to questions I was asked |
| 19:41.34 | ManxPower-work | jro: and the answer to that is provider and service specific |
| 19:41.36 | jro | I have the info I need though thanks everyone |
| 19:41.47 | jro | That would have been a better answer :) |
| 19:42.13 | dgit | We are trying to use the monitor software in asterisk 1.4.22 to transcribe calls. But when we hit pause on the Cisco 7960 because we are listening to them speak it stops recording. Any ideas? |
| 19:42.30 | dgit | Only on outbound calls. |
| 19:43.58 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:44.07 | [TK]D-Fender | titter: XMing + GEDIT |
| 19:44.53 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:47.34 | [TK]D-Fender | jro: Ask them in writing what they are offering exactly. |
| 19:48.06 | avleen | <PROTECTED> |
| 19:48.34 | ManxPower-work | Most providers are not actually "evil". They are incompetent. That just sometimes looks like evil. |
| 19:49.07 | *** join/#asterisk moy (n=moy@189.162.219.76) |
| 19:49.20 | minotaur01 | lol |
| 19:50.09 | ManxPower-work | It LOOKS like every night Verizon goes out and screws up a few lines at our customers. I'm sure they don't, but it does seem that way. |
| 19:50.24 | titter | believe it or not ... I never thought of checking for SFTP in Dreamweaver CS4 ... it works. |
| 19:50.53 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 19:51.02 | rue_mohr | hello }:] |
| 19:51.06 | titter | by default it doesn't open the conf files, but if you edit the Extensions.txt file for Dreamweaver ... it works. |
| 19:51.07 | rue_mohr | how are we all doing? |
| 19:51.36 | rue_mohr | why I'm here: anyone know reasons why record() wont stop recording after pressing #? |
| 19:51.37 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 19:51.42 | rue_mohr | (sip phone) |
| 19:52.10 | [TK]D-Fender | rue_mohr: wrong DTMF mode |
| 19:52.45 | rue_mohr | hmm |
| 19:53.10 | rue_mohr | I cant remember why I'm using the mode I'm using but I know the other one dosn't work |
| 19:53.29 | rue_mohr | wait, is this a phone/asterisk mismatch problem? |
| 19:54.07 | rue_mohr | hmm worked (kinda) on a analog set |
| 19:54.11 | *** join/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 19:54.28 | [TK]D-Fender | rue_mohr: Connected to what? |
| 19:54.32 | rue_mohr | same system |
| 19:54.48 | [TK]D-Fender | rue_mohr: How do you connect an analog set to a PIECE OF SOFTWARE? |
| 19:54.53 | rue_mohr | remember? that terrible digium card that has never stopped giving me echo grief? |
| 19:55.04 | [TK]D-Fender | rue_mohr: You got FXS on that thing too now? |
| 19:55.11 | rue_mohr | always did |
| 19:55.16 | [TK]D-Fender | rue_mohr: And not its been a year and a half :) |
| 19:55.20 | rue_mohr | put the fax and a cordless on it |
| 19:55.34 | [TK]D-Fender | now* |
| 19:55.43 | rue_mohr | gee, I'm surprised you were able to forget me :) good work! |
| 19:55.45 | [TK]D-Fender | rue_mohr: Well if that works... it'd better :p |
| 19:56.19 | profxavier | i just purchased a SoundsStation2, I noticed the POE device has three plugs, 2 ethernet and one phone :: can I use a VOIP system to connect, or is it strictly for 'regular' phone service ? |
| 19:56.24 | rue_mohr | so I need to set asterisk to the same dtmf mode as the phone dont I? |
| 19:56.31 | [TK]D-Fender | rue_mohr: I don't remember the FXS ever coming in to question.... more like the retarded SLA attempts and echo issues because you refused to ditch taht POS :p |
| 19:56.39 | *** join/#asterisk mcr (n=mcr@132.213.238.4) |
| 19:56.40 | profxavier | 2 phone and 1 ethernet* |
| 19:57.00 | [TK]D-Fender | profxavier: What does the manual say? |
| 19:57.21 | ManxPower-work | profxavier: Ask Polycom |
| 19:57.42 | profxavier | is there another type of phone someone can recommend ? |
| 19:57.43 | mcr | hi, just moved from 1.2 to 1.4, and trying to updated the VoiceMailMain(s${CALLERIDNUM}) to something newer... we think it should be VoiceMailMain(${CALLERID(num)},s), but it seems that no number gets passed by that. do we need updates to sip.conf, or??? |
| 19:57.43 | ManxPower-work | Usually if it's not a SoundPoint IP model then it's not SIP |
| 19:57.50 | ManxPower-work | ~phones |
| 19:57.50 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones. Do not consider Cisco phones. Ever. |
| 19:58.17 | ManxPower-work | mcr: read the UPGRADE*.txt files. They are there to help you migrate. |
| 19:59.35 | mcr | ManxPower-work, yeah, the text is not that helpful. It says that they are replaced by their respective dialerplan *functions*, but I'm not clear what those are. |
| 19:59.43 | rue_mohr | [TK]D-Fender, it uses bridged mode, so the fxs isn't an issue, I tried digium for support and when it came to it, there wasn't any with digium for the card or polycom for the phones |
| 19:59.59 | rue_mohr | it all took so long that nobody will help now anyhow |
| 20:00.01 | ManxPower-work | mcr: "core show functions" functions are case sensitive |
| 20:00.04 | mcr | we are not trying to *SET* values, but rather just to use them, and from reading I did it seemed that CALLERID(num) |
| 20:00.20 | rue_mohr | so for all the money we put into it, were jsut living with the remaining echo |
| 20:00.22 | [TK]D-Fender | rue_mohr: you were getting DTMF from the FXS.. thats the poitn. You should, and you do. Now go deal with the device taht IS causing issues |
| 20:00.23 | ManxPower-work | mcr: pastebin the output of a failed call |
| 20:00.55 | [TK]D-Fender | mcr: Alonge with your dialplan |
| 20:00.58 | [TK]D-Fender | ~pb |
| 20:00.59 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit |
| 20:01.01 | [TK]D-Fender | ^^^^^^^^^ |
| 20:01.11 | mcr | pastbin'ing. |
| 20:01.25 | rue_mohr | IT IS NOT POSSABLE go get polycom to answer any questions about how the volume on their phones work, from what I can telll, when you adjust th volume control on it, it adjusts the ear AND mic, which throws the echo training to hell |
| 20:01.59 | rue_mohr | but thats not todays issue |
| 20:03.00 | ManxPower-work | rue_mohr: you read the admin guide? |
| 20:03.29 | dgit | Maybe the 'Do not consider Cisco phones, ever' is why my pushing the mute button stops asterisk from recording the inbound person as well lol. I can't figure out who to ask about this one as I would think it would happen on other phones. |
| 20:03.36 | rue_mohr | its been a long time since I touched this system |
| 20:03.48 | rue_mohr | aside from horrid audio quality, its been flawless |
| 20:04.02 | ManxPower-work | rue_mohr: *nod* Get a Sangoma card and get rid of the issue. |
| 20:04.07 | mcr | http://www.pastie.org/776916 <- it's for automagically jumping into voicemail. |
| 20:04.15 | rue_mohr | we dont have any more money for it |
| 20:04.27 | rue_mohr | and I have NO credibility with them anymore |
| 20:04.29 | [TK]D-Fender | dgit: Someone twisted that infbot snippit. It should say do not touch GRANDSTREAM ever. |
| 20:04.45 | [TK]D-Fender | rue_mohr: What ar yo talking about.... you still have some here? ;) |
| 20:04.45 | ManxPower-work | It wasn't me. |
| 20:04.46 | rue_mohr | I got digium cause I thought they were a good company with good product and good support |
| 20:04.48 | rue_mohr | WRONG |
| 20:05.09 | rue_mohr | sorry, thorn in my side |
| 20:05.18 | ManxPower-work | rue_mohr: *nod* Same reason I originally went with Digium, then used Sangoma for a many years, tried Digium again and was still not impressed. |
| 20:05.18 | dgit | [TK]D-Fender: lol ok. |
| 20:05.40 | rue_mohr | hey IS there a channel log? |
| 20:05.54 | [TK]D-Fender | rue_mohr: infobot logs everything while its here |
| 20:05.55 | ManxPower-work | rue_mohr: we put external t-1 echo canceling hardware on most of our Digium cards. |
| 20:07.38 | ManxPower-work | At this point I consider any claim that Digium's new cards are good to be just as believable as your wife telling you she won't cheat on you a 4th time. |
| 20:08.02 | rue_mohr | yea If I were into antoher $500 I'd get a 2 port T1 and use the telco calceler I have |
| 20:11.16 | rue_mohr | ok to get the dtmf fixed on the sip phones I need to change the mode asterisk is using for digiti detection, right? |
| 20:11.35 | ManxPower-work | rue_mohr: you want RFC2833 set in Asterisk and on the phone |
| 20:11.56 | rue_mohr | I have to look this all up again |
| 20:12.15 | rue_mohr | I think I'm using inband cause something didn't work if I didn't |
| 20:12.28 | rue_mohr | possibly issue with uesrs not being able to hear their digits |
| 20:13.48 | rue_mohr | everything else on the system works, dtmf wise tho |
| 20:14.03 | rue_mohr | even it detecting ..... ## as 'line flash' |
| 20:14.05 | rue_mohr | ... |
| 20:15.12 | dgit | Where should I go to figure out why pushing mute on our phone doesn't record the incoming call either? I tried a bunch of Google posts but they didn't end up being the same issue. I'm thinking it may have something to do with silence detection and shortening recordings. TY. |
| 20:16.28 | ManxPower-work | dgit: look for "VAD" options for the cisco. If VAD is enabled that would explain it |
| 20:16.31 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:19.20 | [TK]D-Fender | rue_mohr: would be nice to see a broken call.... |
| 20:20.29 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 20:20.31 | dgit | ManxPower-work: I went to the tftp directory and found it is sending enable_vad: "0" in phone.conf the line for silencesupression=yes is commented out(grep found VAD above it) |
| 20:21.03 | ManxPower-work | make sure silence supression is dsabled |
| 20:21.58 | *** join/#asterisk Alagar (n=Administ@122.164.42.222) |
| 20:22.15 | dgit | ManxPower-work: Ok, I'll set that to silencesuppression=no tonight after everyone leaves and see if that fixes it. I'm not sure what being commented out defaults to. |
| 20:22.36 | ManxPower-work | hugs his Polycoms |
| 20:22.55 | bmoraca_work | mmmm...open face polycom sandwich |
| 20:23.50 | hardwire | bmoraca_work: motor boat it |
| 20:24.05 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 20:24.07 | bmoraca_work | gotta boil them down first so they're soft |
| 20:24.40 | nny | anyone here know of a wireless handset that is shaped like a traditional handset? It's an odd request, not an entire phone, just a handset... |
| 20:26.41 | *** join/#asterisk retentiveboy (n=pdugas@adsl-221-170-128.pns.bellsouth.net) |
| 20:26.50 | mcr | problem solved: we were doing it exactly right, but it was done in [internal] (IAX connections), and [sip-internal], and we didn't remember that. |
| 20:28.10 | bmoraca_work | lol...looks like diablo got the whole goat-man thing right afterall... http://english.pravda.ru/science/mysteries/12-01-2010/111621-sheep_human_face-0 |
| 20:32.06 | *** join/#asterisk Lord-Rahl (n=Lord-Rah@216.109.206.210) |
| 20:34.44 | Lord-Rahl | quick question for anyone what is the proper syntax for queue timeout in asterisk 1.4? here is what I have now - (queues,250,1|Tt|||60|) |
| 20:34.47 | _abc_ | what is so good about polycoms? |
| 20:35.27 | bmoraca_work | _abc_, they're designed to be open, so configs are not difficult, and they have some of the best hardware |
| 20:35.34 | titter | they are nice to configure |
| 20:35.35 | ManxPower-work | _abc_: they have good phones, incredible speakerphones, incredible amounts of documentation and they don't hate their customers. |
| 20:35.46 | titter | what those guys said |
| 20:36.11 | ManxPower-work | Polycom also does not require you to pay or have a support contract to be able to legally get the firmware. |
| 20:36.36 | _abc_ | ManxPower-work: so when are they going bankrupt ? |
| 20:37.40 | Lord-Rahl | And they are inexpensive as well basic phone is about 80.00 buck |
| 20:37.54 | _abc_ | umm $80 is not so cheap |
| 20:38.00 | _abc_ | for a phone |
| 20:38.13 | bmoraca_work | for an IP phone, yes it is |
| 20:38.15 | ManxPower-work | I'm not a big fan of Polycom's lowest end phones. I'd rather have a Linksys phone than the really cheap polycoms |
| 20:38.16 | fenrus | that's cheap |
| 20:38.17 | titter | compared to the other ip phones on the market ... it is |
| 20:38.59 | _abc_ | what is 'license and product activation' on their web site ?! |
| 20:39.00 | bmoraca_work | ManxPower-work, you don't like the 330s? |
| 20:39.32 | bmoraca_work | _abc_, probably their proprietary apllications they have on their phones that are not necessary and you don't need |
| 20:39.39 | _abc_ | how can one compare a $80 polycom phone with a $100-150 cisco 7960 from eay? |
| 20:39.42 | _abc_ | *b |
| 20:39.48 | ManxPower-work | _abc_: They have a few features (corporate type) that require additional money to purchase. LDAP Directory and Call Recording to a USB flash drive are the two that I remember |
| 20:40.23 | ManxPower-work | bmoraca_work: No, I don't like their 330s, but they ARE much, much better than the 300/301s |
| 20:40.34 | Lord-Rahl | I am using a 330 Polycom around the 80 dollar range it is solid and it sounds is great. |
| 20:40.59 | bmoraca_work | i can see why someone wouldn't like them, though...they aren't as easy to use as the 550s |
| 20:41.09 | ManxPower-work | Lord-Rahl: all polycoms are solid and sound great. It's the small display and lack of dedicated buttons that I don't really like. |
| 20:41.16 | _abc_ | ManxPower-work: which have, cisco ? |
| 20:41.30 | ManxPower-work | _abc_: which have what? |
| 20:41.34 | _abc_ | what you wrote |
| 20:41.39 | _abc_ | 'they' |
| 20:41.48 | ManxPower-work | They are Polycom |
| 20:41.51 | _abc_ | ah |
| 20:42.23 | ManxPower-work | bmoraca_work: Users also get confused by the slightly different UI on the 330s compared to the higher end models. |
| 20:42.54 | bmoraca_work | I get confused byt he slightly different UI on the 330s, lol |
| 20:43.06 | bmoraca_work | menu navigation isn't all that intuitive on them |
| 20:44.03 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-243-188.telstraclear.net) |
| 20:44.21 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-243-188.telstraclear.net) |
| 20:47.14 | benngard | try configure an avay 9650 with sip, u gonn alove it NOT! |
| 20:47.17 | *** join/#asterisk retentiveboy (n=pdugas@adsl-221-170-128.pns.bellsouth.net) |
| 20:47.37 | *** join/#asterisk ip-route (n=iproute@unaffiliated/contraventor) |
| 20:47.41 | ip-route | hi folks |
| 20:47.44 | benngard | but the sound is perfect |
| 20:47.57 | ip-route | how test if my remote asterisk works? |
| 20:48.12 | benngard | try register? |
| 20:48.43 | nny | remote asterisk? |
| 20:48.55 | ip-route | benngard but i'm contains logins... |
| 20:49.13 | ip-route | my problem is : how test if asterisk ports is bind? |
| 20:49.20 | ip-route | telnet sip.server.com 5060 ? |
| 20:49.39 | ip-route | if port is open, asterisk work? |
| 20:49.50 | ManxPower-work | ip-route: Sip uses UDP. telnet uses TCP |
| 20:49.51 | nny | well |
| 20:50.09 | ip-route | ManxPower-work hmmm |
| 20:50.17 | ip-route | ManxPower-work but how test ? |
| 20:50.19 | benngard | u can always "nmap" to se what is answering |
| 20:50.24 | ManxPower-work | ip-route: send a call to it |
| 20:50.29 | ip-route | how test if sip.tmais.com.br is down? |
| 20:50.35 | ManxPower-work | ping it |
| 20:50.41 | ip-route | ManxPower-work ping? |
| 20:50.45 | ManxPower-work | ping |
| 20:50.55 | ip-route | but if host drop icmp? |
| 20:50.57 | benngard | but that doesnt mean your * is "working and feeling well" |
| 20:50.57 | ip-route | O-o |
| 20:51.13 | ManxPower-work | You can't just magically check if "SIP is working". If you want to find out if SIP is working, then use SIP to make a call. |
| 20:51.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:51.51 | ManxPower-work | You can simulate calls using several tools, sipsak being the most common |
| 20:52.56 | Lord-Rahl | Any one know of the proper syntax for 1.4 queue time out to move next step. is this right? > (queues,250,1|Tt|||60) |
| 20:53.19 | ip-route | ManxPower-work ok... |
| 20:53.29 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 20:53.34 | ip-route | do you don't understand my problem |
| 20:53.58 | ManxPower-work | What is your specific problem? |
| 20:54.33 | *** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it) |
| 20:54.41 | [TK]D-Fender | [15:40]<ManxPower-work>bmoraca_work: No, I don't like their 330s, but they ARE much, much better than the 300/301s |
| 20:55.01 | Lord-Rahl | I am try to have a time so if a person is in the queue they are not stuck there for ever. may I will drop them to a VM after 5 min |
| 20:55.05 | [TK]D-Fender | ManxPower-work: 30X = disco. 320/330 = disco. 321/331 = current, 350 = better still. |
| 20:55.28 | ManxPower-work | I know the 30x are discontinued. Doesn't mean nobody is using them. |
| 20:55.42 | ManxPower-work | I don't think I've seen the 350 |
| 20:55.55 | bmoraca_work | never heard of em either |
| 20:56.11 | Lord-Rahl | ManPower-work: That is the way I used on asterisk 1.2 but it is not working on 1.4 |
| 20:56.24 | ManxPower-work | Lord-Rahl: I was not helping you. |
| 20:56.30 | bmoraca_work | maybe he means the 450 |
| 20:56.32 | ip-route | ManxPower-work how test if my sip server works? |
| 20:56.48 | ip-route | works == "accept connections" |
| 20:56.55 | nny | ip-route: try xlite + the proper crendentials |
| 20:57.00 | nny | credentials( |
| 20:57.03 | nny | meh |
| 20:57.06 | [TK]D-Fender | ManxPower-work: 350 = 331 + RJ9 + backlight + G.722 |
| 20:57.08 | ManxPower-work | ip-route: Go research "sipsak" |
| 20:57.09 | Lord-Rahl | sorry |
| 20:57.39 | [TK]D-Fender | ip-route: place calls <- |
| 20:57.42 | ManxPower-work | [TK]D-Fender: That's the other think I don't like about polycoms. Their rj9 headset ports |
| 20:57.56 | [TK]D-Fender | ManxPower-work: Industry standard for amps |
| 20:57.56 | ip-route | ManxPower-work grats |
| 20:58.05 | ip-route | greats* |
| 20:58.11 | bmoraca_work | [TK]D-Fender, i think that's the 335... |
| 20:58.50 | nny | heh |
| 20:58.52 | nny | http://www.youtube.com/watch?v=FR_2dnBEapM |
| 20:59.03 | nny | too bad someone doesn't sell something like this, i have a client who wants one |
| 20:59.13 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 20:59.54 | [TK]D-Fender | bmYeah, made that typo before too... thanks |
| 21:00.00 | [TK]D-Fender | bmoraca_work: Yeah, made that typo before too... thanks |
| 21:00.06 | [TK]D-Fender | ManxPower-work: correction, IP 335 |
| 21:00.27 | ManxPower-work | [TK]D-Fender: I'm just glad my cell phone apparently has a built in amp. |
| 21:00.39 | ManxPower-work | It would suck to have to plug an rj9 headset into my cell phone. |
| 21:02.11 | [TK]D-Fender | ManxPower-work: You can always use the 321/331 with 2.5mm |
| 21:02.51 | ManxPower-work | I'd just be happy to have a headset that doesn't cost more than the phone. |
| 21:04.01 | [TK]D-Fender | ManxPower-work: So make sure you get IP670's ;) |
| 21:06.40 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 21:06.50 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 21:10.44 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 21:11.32 | Lord-Rahl | quick question for anyone what is the proper syntax for queue timeout in asterisk 1.4? here is what I have now - (queues,250,1|Tt|||60|) on time out it would move to VM |
| 21:12.09 | [TK]D-Fender | Lord-Rahl: "core show application queue" <- go read the instructions and see if the parameters match |
| 21:13.43 | Lord-Rahl | [TK]D-Fender: Thanks I will try that |
| 21:15.54 | *** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) |
| 21:18.09 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 21:18.50 | *** join/#asterisk nix8n82 (n=nate@63.162.27.14) |
| 21:20.15 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:21.26 | *** join/#asterisk deeperror (n=deeperro@adsl-99-33-114-101.dsl.sfldmi.sbcglobal.net) |
| 21:22.19 | [TK]D-Fender | Lord-Rahl: You've got 5 mins of help before I'm out. Don't waste them |
| 21:23.17 | rue_mohr | hah, silence |
| 21:23.50 | deeperror | if a client wants caller id name to be seen when making outbound calls what would be a simplified/best way to say it will not work for every call? |
| 21:24.02 | Lord-Rahl | [TK]D-Fender: exten=2,1,Goto(queues,251,1|Tt|||6)exten=2,2Voicemail(151,u) |
| 21:24.58 | ManxPower-work | deeperror: it won't work for any calls. |
| 21:25.10 | deeperror | yea i found the cid faq haha i'm on there now |
| 21:25.35 | ManxPower-work | deeperror: the idea is flawed. |
| 21:25.44 | Lord-Rahl | [TK]D-Fender: my question is or 251,1 count as separate options |
| 21:25.55 | ManxPower-work | you are not looking for CallerID. You are looking for Called Party ID, which is not supported in Asterisk. |
| 21:26.04 | deeperror | ManxPower-work: yea i agree, i explain but they don't get it |
| 21:26.37 | ManxPower-work | deeperror: I can't help you with customers that are too stupid to understand "no, it is not supported" |
| 21:27.56 | deeperror | ok thanks :)) |
| 21:28.16 | ManxPower-work | It's called CalleRID and not CallEEID for a reason. 8-) |
| 21:29.07 | carrar | calleeeYo |
| 21:29.20 | [TK]D-Fender | Lord-Rahl: that is a GOTO.. that is not even a call to QUEUE. Horrible mixup |
| 21:29.49 | ManxPower-work | Lord-Rahl: you have a basic misunderstanding about Asterisk. |
| 21:29.59 | [TK]D-Fender | Lord-Rahl: GOTO does not have more than 3 parameters. You seem to be using apps and not even reading which ones you are calling or the instructions associated with them |
| 21:30.04 | [TK]D-Fender | And its checkout time. |
| 21:30.07 | [TK]D-Fender | Problem ID'd |
| 21:30.09 | [TK]D-Fender | later all |
| 21:30.54 | Lord-Rahl | [TK]D-Fender: duuhh :) you are right. |
| 21:30.55 | p3nguin | deeperror: I think you were thinking of CNAM, possibly, which is a function of the recipient's telco. |
| 21:31.05 | carrar | Thats really the best way to use Asterisk, just make up options for each app |
| 21:31.13 | ManxPower-work | p3nguin: not "possibly", "is" |
| 21:31.26 | ManxPower-work | ~answers |
| 21:31.27 | infobot | somebody said answers was Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 21:31.27 | deeperror | p3nguin: yea CNAM is what i was looking for... |
| 21:31.28 | p3nguin | manxpower-work: huh? |
| 21:31.47 | p3nguin | deeperror: Yeah, CNAM isn't something you send, but something the receiving end must lookup. |
| 21:31.48 | ManxPower-work | p3nguin: CNAM is always looked up by the terminating carrier. |
| 21:31.58 | p3nguin | manxpower-work: I'm fully aware. |
| 21:32.07 | p3nguin | manxpower-work: " which is a function of the recipient's telco." |
| 21:32.21 | p3nguin | English fail. |
| 21:32.30 | Lord-Rahl | [TK]D-Fender: sorry it has been awhile since I had to work with this stuff |
| 21:32.48 | ManxPower-work | If you stay "on-net" i.e. totally within your carrier's network, depending on the carrier, you may be able to send Caller*ID Name, but that's an edge case. |
| 21:33.00 | ManxPower-work | p3nguin: sorry, my english parser broke. 8-) |
| 21:33.42 | p3nguin | deeperror: Can you explain to the client how CNAM lookups work and satisfy them? |
| 21:33.43 | deeperror | ManxPower-work: that's a good idea there... i think the issue is because the option is available to change names it should work and then trying to tell the client that it's there for our network but not other networks |
| 21:34.01 | deeperror | p3nguin: yea def will bring this up on the next call |
| 21:34.08 | deeperror | thanks for the input! |
| 21:34.08 | ManxPower-work | deeperror: "option is available to change names" ? |
| 21:34.26 | deeperror | in asterisk it has the (name) in there |
| 21:34.28 | p3nguin | CALLERID(all)? |
| 21:34.35 | deeperror | yea CALLERID(name) or something |
| 21:34.49 | ManxPower-work | deeperror: *nod* |
| 21:34.50 | p3nguin | (name) is part of (all) |
| 21:35.10 | deeperror | i know but this is a client that reads in forums and online and thinks they are pro telling me what is possible |
| 21:35.32 | p3nguin | This brings up a thought... I have seen that there is CALLERID(ANI), but when should it be used? |
| 21:35.44 | deeperror | HAHA |
| 21:35.56 | p3nguin | Or when does it do any good to use, rather. |
| 21:36.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 21:36.13 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:36.19 | ManxPower-work | deeperror: you can tell them that Asterisk IS sending out the correct callerid info (chances are it is), but the terminating carrier always overwrites the name portion (also true) |
| 21:36.28 | p3nguin | Maybe it never works for anything, heck, I don't know. |
| 21:37.03 | ManxPower-work | you should be able to demonstrate this with pri debug or sip debug |
| 21:37.08 | deeperror | yea i think i'll end the conversation when they see CNAM |
| 21:38.09 | p3nguin | The receiving telco does a dip into the database, matches the number you send, and displays the name associated with that number, even if it is not what you want it to be or expect it to be. |
| 21:38.59 | p3nguin | Or you could end the conversation like Ron White... |
| 21:39.17 | p3nguin | "Yeah, well fuuuuuuuckk youuuuuu!" |
| 21:40.23 | p3nguin | That could end more than just the conversation, though. |
| 21:40.40 | p3nguin | Proceed with CAUTION! |
| 21:40.56 | carrar | Throw caution to the wind |
| 21:41.59 | _abc_ | is this some sort of perverted godwin clause? |
| 21:42.37 | ManxPower-work | _abc_: Just wait until the Squirrel Lady shows up. |
| 21:42.51 | p3nguin | lol |
| 21:43.00 | p3nguin | I've noticed she's MIA today. |
| 21:43.24 | carrar | mmm Squirrel |
| 21:43.28 | carrar | lady |
| 21:43.49 | carrar | and Elephant zoo keeper should be showing up too |
| 21:44.05 | carrar | the owner of the SNAKE PHONE |
| 21:44.41 | _abc_ | is the Squirrel Lady the aserisk version of the Fat Lady? |
| 21:44.57 | carrar | no |
| 21:45.12 | _abc_ | i need to lurk more here |
| 21:45.53 | nny | for anyone curious here is my results for the re-park from star code and seamless xfer to cellphone stuff over the last couple of days |
| 21:45.54 | nny | http://pastebin.com/m24b19500 |
| 21:45.56 | _abc_ | snake phone? elephant zoo keeper? |
| 21:46.16 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 21:47.09 | _abc_ | you are talking code here |
| 21:47.34 | carrar | printf("CODE TALK HERE\n"); |
| 21:50.33 | p3nguin | "The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.'" If I use _X. like they suggest, that will match all numbers, but is there anyway to match letters and * and #? Should I make a new exten for each? |
| 21:50.41 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 21:50.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:52.22 | p3nguin | I guess it would take like five separate extensions that way. No so bad, I suppose. |
| 21:56.10 | bmoraca_work | p3nguin, it'll match all extensions that begin with a number. the problem with _. is that it matches ALL extensions, even those that begin with * and # and blank extensions |
| 21:56.16 | p3nguin | Or what about using _[*#a-z]. for the *, the #, and any single lower-case letter? |
| 21:56.35 | bmoraca_work | p3nguin, what are you trying to match? |
| 21:56.57 | p3nguin | any extens that are NOT the standard built-in extens |
| 21:57.24 | p3nguin | Could be numbers, could be * or #, or could even be a letter. |
| 21:57.31 | bmoraca_work | what do you mean "standard built-in extens"? |
| 21:57.41 | p3nguin | h, H, s, i, t, T |
| 21:57.41 | bmoraca_work | you mean like a, h, and o? |
| 21:57.43 | p3nguin | o |
| 21:57.51 | p3nguin | all the standard built-in extens. |
| 21:58.08 | bmoraca_work | just be absolutely sure of the order of your matching, because _. will match those standard extensions as well |
| 21:58.16 | p3nguin | I KNOW THIS. |
| 21:58.31 | p3nguin | Actually, I don't even care about lettered extens, to nevermind about those. |
| 21:58.49 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 21:58.53 | *** join/#asterisk danj1980 (n=dan@91.109.125.222) |
| 21:59.14 | p3nguin | I'll just make one exten with _X. for numbers, and something with _[*#]. in it for things to match those. |
| 21:59.21 | p3nguin | disregard everything. |
| 22:00.49 | *** join/#asterisk Shooters (n=shooters@bas10-montrealak-1128580753.dsl.bell.ca) |
| 22:01.29 | p3nguin | _[*#0-9]! might work properly. |
| 22:01.37 | danj1980 | Does anyone know how to set up a provisioning server for the linksys pap2t devices? xml samples? |
| 22:01.51 | Shooters | I'm getting a 401 Unautorized error message when I try to do an outgoing call with my aastra 55i, but it works if I do it through a softphone. Incoming calls work though. Any idea what is the problem? |
| 22:02.44 | Corydon76-dig | 401 means you don't have a password set |
| 22:03.13 | *** join/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 22:03.29 | *** part/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 22:03.47 | Shooters | Corydon76-dig, I do have a "secret" set for all my extensions. My logs are here: http://goo.gl/c2nA . The register works, but the outgoing call doesnt. |
| 22:08.55 | fenrus | danj1980, well, i'm a little bit into it |
| 22:09.19 | fenrus | have just been surfing around to gather some xml-data |
| 22:09.44 | danj1980 | fenrus, have you found a pdf of the administration guide anywhere? |
| 22:09.48 | fenrus | nope |
| 22:09.59 | fenrus | have been grinding bits and pieces |
| 22:10.16 | fenrus | will probably build some mysql+php based provisioning-server when i can find the time |
| 22:10.36 | fenrus | but i'll gladly share the xml i've found |
| 22:11.37 | danj1980 | Ive seen this site:- http://www.ataconfig.com/ |
| 22:12.15 | Shooters | Corydon76-dig, any other idea? |
| 22:12.20 | danj1980 | I'm thinking of signing up for a month, creating some profiles, downloading the xml to see what its like and then making my own provisioning server for my pap2ts |
| 22:13.06 | fenrus | danj1980, http://pastebin.com/m32356ee9 |
| 22:13.32 | *** part/#asterisk nny (n=scott@64.203.239.83) |
| 22:13.39 | fenrus | i've got three ata's working with this config |
| 22:14.54 | fenrus | danj1980, perhaps time to create some project to build this, to not reinvent the wheel 100 times |
| 22:14.57 | fenrus | =) |
| 22:15.23 | danj1980 | or wiki everything online somewhere |
| 22:15.33 | danj1980 | for others |
| 22:15.33 | fenrus | yea |
| 22:16.08 | fenrus | perhaps voip-info.org :) |
| 22:16.30 | danj1980 | yup |
| 22:16.50 | danj1980 | thanks for the sample. ill take a look at it and see what I can add. |
| 22:17.20 | fenrus | feel free to drop me an pm if you find anything that can be useful |
| 22:17.28 | fenrus | now i'm off for bed :) |
| 22:19.04 | _abc_ | me too |
| 22:19.10 | danj1980 | thanks |
| 22:19.46 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:19.47 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:20.36 | *** join/#asterisk fuxu2 (i=iconicfl@www.kevinlynn.com) |
| 22:28.56 | danj1980 | if one was going to provide a SIP service like vonage, what gateway would be suitable for a userfriendly experience? im looking for just plug and play, with no end user configuring required. |
| 22:29.00 | *** join/#asterisk dennis00 (i=dennis00@unaffiliated/dennis00) |
| 22:29.09 | dennis00 | Can I use the newest dahditools with Asterisk 1.6.0? |
| 22:29.16 | tzafrir_laptop | yes |
| 22:29.18 | danj1980 | Maybe with an Lcd display like vonage's current gateway to help with tech support. |
| 22:33.23 | dennis00 | the compiling thing takes a lot of time |
| 22:33.39 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 22:34.18 | dennis00 | tzafrir_laptop: Toda |
| 22:36.06 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 22:38.03 | dennis00 | What is dahdi? |
| 22:38.35 | tzafrir_laptop | The new name of Zaptel |
| 22:38.45 | dennis00 | What does it do? |
| 22:39.26 | Corydon76-dig | It provides an interface between Asterisk and the physical telephony hardware |
| 22:40.00 | mltlnx | Hello....What is the best way to determine which SIP trunk a call comes in on....via the dialplan. |
| 22:40.09 | cusco | hi |
| 22:40.26 | cusco | I'm trying to setup the atxfer in features.conf |
| 22:40.34 | cusco | atxfer => *8 |
| 22:40.43 | cusco | that alone should work, right? |
| 22:40.55 | cusco | (and obviously having the t in the Dial() |
| 22:40.57 | cusco | ) |
| 22:41.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:44.14 | p3nguin | cusco: Make sure you reload for the changes to take effect. |
| 22:44.26 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.137.221.dsl.dyn.forthnet.gr) |
| 22:46.20 | cusco | ok it works :D |
| 22:46.22 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 22:50.37 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:51.58 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 22:52.01 | hardwire | is there a handy list somewhere of all situations where asterisk will return a certain sip response code? |
| 22:52.19 | hardwire | I don't need it.. but I think it would help educate a very idiotic coworker of mine. |
| 22:53.28 | *** join/#asterisk etfonhomey (n=etfonhom@12.4.217.194) |
| 22:53.39 | hardwire | if only he could be somebody elses problem for a day. |
| 22:56.25 | p3nguin | which person? |
| 22:59.01 | *** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com) |
| 22:59.40 | p3nguin | Oh, not an IRCer. Nevermind. |
| 22:59.55 | p3nguin | I was going to trout him. |
| 23:00.47 | carrar | give him the task to research them |
| 23:03.14 | hardwire | p3nguin: I need somebody to salmon him, IRL |
| 23:03.23 | hardwire | alaskans don't trout.. we salmon. |
| 23:11.42 | dennis00 | What does it mean if it says 10 sec registering and then time-out? |
| 23:14.48 | netpro25_ | How do you tell asterisk to answer calls on all extensions? |
| 23:14.56 | netpro25_ | I forgot the config option |
| 23:17.28 | *** join/#asterisk delrocas (n=delrocas@209-181-45-189.spkn.qwest.net) |
| 23:19.38 | delrocas | Are there distros asise from SUSE that work well with ISDN BRI cards? |
| 23:20.16 | delrocas | aside from SUSE? |
| 23:25.49 | Qwell | any? |
| 23:25.54 | Qwell | Linux is Linux |
| 23:27.14 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 23:28.33 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
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| 23:30.22 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 23:30.52 | dennis00 | I get Forbidden messages, how to troubleshoot? |
| 23:30.56 | delrocas | So no particular distro is considered more "ISDN ready" than the next? |
| 23:34.15 | *** join/#asterisk sevv (i=sev@216.164.6.24) |
| 23:34.53 | *** join/#asterisk mr_ian (n=mr_ian@S0106001b63f49383.du.shawcable.net) |
| 23:35.17 | dennis00 | Can Asterisk work without an extensions.conf? |
| 23:35.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:35.21 | dennis00 | I have no idea how to write extensions.conf. |
| 23:35.45 | mr_ian | you won't get many calls |
| 23:35.46 | sevv | I'm looking for a free sip provider (no PSTN termination) that's compatible with gizmo5 (they've closed registrations). I'm on my mobile and am having troble lcating the list. Can anyone direct me |
| 23:36.02 | dennis00 | mr_ian: It would work with outgoing calls? |
| 23:36.21 | sevv | dennis00: there are pretty detailed guides out there for extensions configurations. |
| 23:36.48 | mr_ian | sevv: ideaSIP, antiSIP, brujula & ekiga are all quality free providers |
| 23:36.59 | dennis00 | Yes, but I dont know what they mean with 'SIP number', the extension number? |
| 23:37.05 | mr_ian | sevv: & sip2sip.info |
| 23:37.18 | sevv | mr_ian: i actually found antisip but can't find the domain/stun server settings anywhere |
| 23:37.43 | sevv | it could be that for soome reason it's presented iin a java or flash app which my mobile doewsn't support |
| 23:38.17 | sevv | dennis00: an extension number is unrelated to the sip number. |
| 23:38.38 | dennis00 | sevv: I see. I keep on getting 'not found' while dialing. |
| 23:39.24 | sevv | you should check out the guides, your phone registers with your server and your dialing patterns must be matched within a context. |
| 23:40.02 | dennis00 | I have checked out so many guides related to my VOIP provider. |
| 23:41.10 | sevv | you should be checking out an Asterisk guide. i'd direct you to a nerdvittles guide but i'm on my mobile, a google search would help you |
| 23:41.49 | sevv | you know your sip phone can register straight to your sip provider without an asterisk pbx, right? |
| 23:43.59 | dennis00 | Yes, of course. |
| 23:44.10 | dennis00 | But I decided I was interested in Asterisk 2 days ago. |
| 23:44.21 | dennis00 | It made my head hurt and haven't got it working yet, but I won't give up. |
| 23:44.29 | dennis00 | I made some stupid decisions before. |
| 23:44.35 | dennis00 | http://pastebin.ca/1750373 < extensions.conf, correct? |
| 23:45.25 | carrar | dennis00 |
| 23:45.27 | carrar | ~book |
| 23:45.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 23:45.29 | carrar | read that |
| 23:45.31 | sevv | somebody else is going to have to pick up here, my webbrowser on my mobile is now totally munched |
| 23:46.10 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 23:46.32 | mr_ian | dennis00: (A) pbxes.com you can get a functional PBX running |
| 23:47.01 | dennis00 | I am a student, I don't have a budget. |
| 23:47.03 | mr_ian | dennis00: (B) try SwitchVOX free, run it on a Sun VirtualBox. easy way to build a play system, get it all running |
| 23:47.07 | mr_ian | both are free |
| 23:47.17 | mr_ian | ;) |
| 23:47.32 | dennis00 | I had already used FreePBX, then AsteriskNow |
| 23:47.51 | dennis00 | I don't want to use some stupid Asterisk alternative |
| 23:47.58 | mr_ian | ok, cool |
| 23:48.53 | bmoraca_work | dennis00, you're on the right track, but you're missing a few key syntactical details that you'll really only get by reading |
| 23:49.03 | bmoraca_work | dennis00, as already suggested, the book is a great place to start |
| 23:49.28 | dennis00 | I see, will do. |
| 23:49.33 | ruben23 | hi anyone have idea on this warnings..---> http://pastebin.com/m715d93d0 |
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| 23:50.58 | *** part/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 23:51.00 | rue_mohr | there's gettin too many ways of doing menus anyon have an example of the "right" way? |
| 23:51.50 | carrar | my way |
| 23:51.55 | carrar | is the right way |
| 23:51.57 | carrar | for everything |
| 23:52.00 | carrar | everytime |
| 23:54.26 | *** join/#asterisk hluesea (n=hulusika@88.239.32.245) |
| 23:55.23 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096590431.dsl.bell.ca) |
| 23:56.39 | *** join/#asterisk hhkahya (n=hulusika@88.239.32.245) |
| 23:59.28 | hhkahya | slaps voipmonk around a bit with a large trout |