IRC log for #asterisk on 20110801

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02:06.00a1faanyone worked with Voyage linux for small embedded devices to run Asterisk on?
02:06.56WIMPyDoes it count that I own a HorstBox?
02:07.20a1fawhats that
02:07.38WIMPyA plastic router running Asterisk.
02:07.43WIMPyD-Link
02:07.46a1fayou own one, or you own the company?
02:07.52*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)
02:08.04WIMPyHaha. I own such a thing.
02:08.35a1fasettle down
02:08.47ChannelZSimmer down!
02:09.00a1fai am forced to run vanilla asterisk
02:09.18a1fabecause none of these gui schmui distributions have a way to run custom dialplan
02:09.18ChannelZWhat? Vanilla is the most popular flavor.
02:09.28a1fayes, I agree
02:09.31ChannelZScrew GUIs
02:09.54a1faexten => s,1,Noop(Alert-Info -> '${SIP_HEADER(Alert-Info)}')
02:09.58a1fa:0
02:10.30ChannelZfascinating
02:10.31a1fasome genius in here figured out how to distinguish between multi-line presence on broadvoice
02:10.39a1famy hat goes out to you good sir
02:11.06a1faexcept, that I am considering canceling broadvoice, even at $7/month
02:11.33a1faactually, more like $14 with all taxes
02:11.49a1fai am also trying to kill my VPS asterisk box, and bring it home
02:11.57a1fato run on a 433Mhz embeded device
02:12.02a1fano codec transversion
02:12.21a1faG711 or GMS
02:14.32a1fasomeone tell me its not a good idea?
02:15.15WIMPyNot me
02:16.15a1fathe one thing I loose with this configuration is call logging
02:16.48WIMPyYou have no other box running?
02:17.21ChannelZnot even a USB thumbdrive?
02:17.40a1fanope
02:17.48a1faWIMPy: i am trying to consolidate things in the house
02:17.50ChannelZAn old dot-matrix printer.  That'd be cool
02:17.51a1faand cut my spending on vps-es
02:18.23a1fai went from 500W computer running a firewall down to 5W
02:18.30a1faand now 5W fanless asterisk system
02:18.35WIMPyOr a little POS receipt printer.
02:18.46a1fanah; maybe nfs mount ;)
02:19.15WIMPyI just had to retire my catalyst after I found out how much power it takes.
02:20.07a1falol
02:20.10a1fa6500?
02:20.37WIMPyI always said that VOIP phones are extremely expensive in operation on your electricity bill. But if you tahe switchports into account as well, it becomes even worse.
02:20.56WIMPyNah, just a little 3500.
02:22.15a1fa:)
02:23.01WIMPyBut replacing that should save me more than a tenner/month.
02:24.06WIMPyAnd it has become much quieter.
02:24.26a1faelectricity is cheap where i live
02:24.36a1fa5.9 cents
02:24.52a1famy asterisk vps == $20/month
02:25.13WIMPyIf I buld a big noise cancelling box for that old SCSI drive, it should be even better,
02:25.22WIMPyper kWh?
02:25.45a1fayup
02:26.18WIMPyThat's damn cheap. Standard rate here is 23.3 ยข
02:26.21WIMPyEUR Cent that is.
02:26.49WIMPyI'm on 20.5/16.6 day/night.
02:27.51a1fasucks
02:27.53a1fa:)
02:28.01WIMPyYes
02:28.14*** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com)
02:28.23WIMPyIt's all since the invention of CO2 trade.
02:28.50WIMPyThe price has doubled over the last 9 years.
02:29.05ChannelZThink of the savings if we could just plug Harry Reid's hole
02:32.48a1faso 500Mhz For the Win?
02:34.03nightrid3ror 1.2Ghz kirkwood
02:34.32a1fathe goal is not to spend any money
02:35.11nightrid3rok
02:36.08a1falets see how * behaves in read-only enviroment
02:37.01WIMPyNo more voicemail.
02:37.29a1fa;)
02:37.30*** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins)
02:37.32a1fathat's it?
02:37.59a1fawhat are the considerations for having the system be behind NAT?
02:38.00WIMPyAnd no persistant storage in AstDB.
02:38.42a1fawhat ports do I need to open besides 5060, bi-directional high UDP ?
02:39.06WIMPyDepends on your firewall.
02:39.53a1fa<PROTECTED>
02:40.01a1fathis is what i needed.. the RANGE for RTP
02:40.42WIMPyIf it's Linux, you just need to load the SIP conntrack module.
02:41.18a1fanah
02:41.20a1fapf
02:41.38WIMPyMight have something similar.
02:41.42a1fawhat codec should I run, gsm all accross?
02:41.52WIMPyAnd you can configure a smaller range, of course.
02:43.04a1fasip.conf?
02:43.16WIMPyrtp.conf
02:43.17a1fait wont matter, I am going to IPSEC all my voip traffic between the sites
02:43.49WIMPyThen you don't need to open up anything.
02:45.37a1fai do need to open-up some things to my providers
02:45.54a1fabroadvoice, and teliax
02:56.44a1fa] NOTICE[1687]: chan_sip.c:21253 handle_request_invite: Sending fake auth rejection for device "callerid@broadvoice".
02:56.47a1fawth
02:57.16p3nguin_What?
03:00.01a1fagot a strange reject message
03:00.08p3nguin_Looks typical to me.
03:00.30a1fain what aspect?
03:01.13p3nguin_When "callerid@broadvoice" isn't a legitimate peer on your system, and it tries to use your system, a rejection is generated.
03:01.23a1fabut.. it is ;(
03:01.33p3nguin_Show me.
03:04.45a1falet me try to make another call
03:05.37p3nguin_Unless you are "broadvoice," which it is very clear to me that you aren't, there's no reason that I can think of that that should be showing up.
03:06.16p3nguin_Even if you do have a peer by the name of callerid, you aren't broadvoice.
03:06.59a1fai am just telling you what the log says
03:07.08a1fabesides, this is the same working config that came from my VPS
03:07.42p3nguin_I guess if it's working, my job is done.
03:08.08p3nguin_Another satisfied customer.  NEXT!
03:10.31a1fa?
03:10.39a1faits not working
03:17.26a1famy ISP must be messing with me
03:19.35a1fathey block 5060 on random
03:19.41a1fabecause they now offer voip service too
03:20.14p3nguin_Inbound, outbound, or both?
03:21.40a1faboth
03:22.42p3nguin_Several ITSPs are aware that ISPs block the SIP port, so they over non-standard ports for services.  Check your ITSPs for these non-standard ports.
03:26.27WIMPySipgate even offer free tunneling for all. Incl. free accounts.
03:27.28p3nguin_What type of tunnel are they providing?
03:29.01WIMPyCisco-IPSec
03:29.49p3nguin_Does that require the end user to have a Cisco client of some sort?
03:30.13WIMPyI have NFI, what software supports that.
03:30.59p3nguin_Seems like it could be a bother or a gift.
03:31.25WIMPyI don't plan to try it.
03:31.45p3nguin_While I'm thinking of it...
03:32.12*** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45)
03:32.42p3nguin_Do you have any idea why my openvpn won't route to the rest of my LAN?  I have no problem with traffic getting to and from the server running on the asterisk computer, but if I try to get out to another host on the LAN it just doesn't work.
03:33.03p3nguin_IP forwarding is enabled.
03:33.18WIMPyDo you use routing or bridging?
03:33.42p3nguin_It's a p2p vpn... I guess routing.
03:33.51WIMPyYou can have both.
03:34.09p3nguin_I didn't configure bridging, so probably just routing.
03:34.12a1fai dont think my asterisk is honoring outboundproxy=
03:34.17a1fain sip.conf
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03:35.18WIMPyAnd it's not a firewall thing?
03:35.43p3nguin_FORWARD policy is accept and there are no rules in that chain.
03:36.36p3nguin_The way I see it, with a two directly-connected interfaces, routing should Just Work.
03:36.39p3nguin_but it doesn't.
03:37.44WIMPyThat reminds me that I should set up an openvpn to the vps I got.
03:38.42a1faif you use register=> does it honor outboundproxy?
03:38.44p3nguin_I'm pinging another host on the LAN and watching the FORWARD chain packet count.  It's rising.
03:39.11p3nguin_Oh, I think I know the problem... maybe.
03:39.23p3nguin_Could it be that the other hosts don't have a route back over the VPN?
03:39.29WIMPya1fa: I don't think so, but you can make pretty complex register statements.
03:39.51WIMPyp3nguin_: That's definitely a possible expanation.
03:40.11a1fa[Aug  1 03:39:38] WARNING[2451]: chan_sip.c:12317 transmit_register: Probably a DNS error for registration to 1234@sip.broadvoice.com, trying REGISTER again (after 20 seconds)
03:40.16p3nguin_Maybe I can watch a tcpdump on that other host to see if the pings are getting there.
03:44.16a1facluster
03:44.43p3nguin_Hmm.  As I suspected, the ping does reach that other LAN host.
03:45.15WIMPyNo route, no traffic.
03:45.17p3nguin_So do I need to set up RIP or something on the LAN hosts along with the VPN server host?
03:45.28WIMPyWhy?
03:45.59WIMPyThe VPN os not terminated on the targets standard gateway?
03:46.13p3nguin_The ping reaches the other LAN host, but I get no reply.  I assume it is because it doesn't know the route.  If I had RIP, all machines would know all routes.
03:46.39p3nguin_The VPN server is on the same computer as Asterisk, which is a host behind NAT.
03:48.00WIMPyOk, so need to tell the networks gateway that the Asterisk box is the gateway for your net.
03:48.17p3nguin_route add -net 10.1.0.0 netmask 255.255.255.0 gw asterisk.local
03:48.28p3nguin_Now I get ping replies.
03:48.35WIMPyAnd hope that noone filters ICMP redirects.
03:49.24p3nguin_Since adding the route by hand works, I'd rather just share routing info among the hosts.
03:49.57p3nguin_Oh, I see what you're saying.
03:50.42p3nguin_Tell the edge device that traffic coming to it, which is destined for 10.1.0.0/24, should be rerouted through the other computer.
03:51.37p3nguin_I... don't know what that would do if there is traffic for that net on the outside as well.
03:51.45p3nguin_On the inside, that would probably work.
03:51.57WIMPyOn the outside?
03:52.25p3nguin_It's a cable network, so there's traffic on 10/8.
03:52.48p3nguin_Maybe the firewall is enough to stop that.
03:52.53p3nguin_(it should be, anyway)
03:53.08WIMPyFor unauthorized modems?
03:53.23p3nguin_It's there for all modems, regardless.
03:53.49WIMPyugh :-(
03:54.05p3nguin_The cableco talks to all modems on the 10/8 network.
03:54.26p3nguin_I'd have to go dig around to see if they are using the entire class A or not.
03:54.52p3nguin_They could really be using less than 10/8... but I know they use 10.something
03:55.03WIMPyBut I'm pretty sure you don;t want fo forward traffic to or from there.
03:55.30WIMPyHmm
03:55.56WIMPyHaving said that. I think they had their own website on a private IP here.
03:56.08WIMPyBut only if you use their DNS, obviousely.
03:56.43p3nguin_I guess if the firewall is preventing the rfc1918 addresses from coming in on the outside interface, a route for my vpn network should be okay.
03:57.32p3nguin_I don't know why I didn't think of the route on the other host as being the problem.  Maybe sleep deprivation or something.
03:57.32WIMPyYou should configure reject routes for rfc1918 areas.
03:57.53WIMPyknows that one too well :-(
03:59.03p3nguin_Okay, I put that route on the lan gateway instead of the individual host, and pings are successful again.
03:59.12p3nguin_That was too easy.
03:59.20WIMPy:)
03:59.45p3nguin_It just need to know the next hop, and now it's happy.
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04:02.04p3nguin_Now I need to push the LAN's DNS over the VPN so I can use the local hostnames.
04:03.26p3nguin_But it won't matter anyway, I guess.  There's an ACL that isn't allowing my vpn tunnel address to access services in the lan.  If it's not one thing, it's another!
04:03.50WIMPyNAT :-)
04:04.40WIMPyThat also eliminates the need for the route again.
04:04.47p3nguin_Oh no you di-n't!
04:05.40p3nguin_I don't know if I want to NAT the VPN.  That kind of defeats the purpose.
04:05.55WIMPyDoes it?
04:06.03WIMPyWhy?
04:06.21p3nguin_Although I do know that iptables masquerading doesn't ruin SIP and RTP like other NAT implementations.
04:06.54WIMPySo you want to be able to do reINVITEs from one LAN to the other?
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04:07.40p3nguin_The reason I was using the VPN was to run VoIP phones in remote locations with fscked up NATs.
04:07.56p3nguin_But then I wanted to do some other things over the VPN on other hosts in the LAN.
04:08.08p3nguin_That's when I encounted the lack of traffic flow.
04:08.27WIMPyIn both directions? If te VPN is terminated on te Asterisk box, that wouldn't be affected by the NAT.
04:08.42p3nguin_Ah, an excellent point you have made.
04:09.06p3nguin_I didn't even think that would through all the way.
04:09.15p3nguin_err...
04:09.24p3nguin_I didn't even think that one through all the way.
04:09.40p3nguin_Something tells me it's time for bed.
04:09.55WIMPyIt is.
04:10.06WIMPyBut I can't decide to go.
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04:47.35syntaxxdoing sip show peers doesnt show any peers at all even though i see someone connects.. any idea which configuration file to look at? thanks
04:52.16ChannelZwhat do you mean even though you 'see someone connects'?
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04:52.54ChannelZsip show peers   should show all the peers defined in sip.conf, less registrations
05:10.01nnylol what a nightmare
05:10.18nnylove it when a green linux admin gets a hold of asterisk ><
05:13.25*** join/#asterisk gravin (~gravin@61.76.50.60.brf01-home.tm.net.my)
05:13.43ChannelZWell, at least you have a sense of humility.
05:13.47ChannelZZING!  hehe
05:18.18nnyhahaha
05:18.24nnywell yeah considering my makefile issue earlier
05:18.36nnybut apart from that level of my ignorance
05:19.01nnyjust dealing with a deadlock issue + new consultants who think disabling services like smartd are part of the troubleshooting process
05:19.28nnybasically, don't know what's going on, panic, ruin the (simple) normal way of dealing with it ><
05:20.06nnycapture regress, (or regress capture if you can recreate) and have a plan b.. none of which happened
05:20.11nny:S
05:21.15nnyI mean, if asterisk (and asterisk alone) is deadlocking, why blame things outside of asterisk?
05:21.29nnyunless it's hardware/ram/etc that is
05:21.34nnywell ram/cpu..
05:22.15nnybut you see what I mean. If it was an issue with a transcoder (which is a network device) or other, you'd still wanna see the deadlock
05:22.32nnybecause it'd still be in theory an asterisk bug, since it's not dying gracefully at all
05:22.40nny./rant heh
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06:14.46schmidtsgood morning
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06:33.23ChannelZwoof
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06:37.08irrootrawr :P
06:37.38ChannelZThe end of another Top Gear series. *sniff*
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06:52.17cuscohi folks
06:52.23cuscoim in trouble
06:52.32cuscotwo digium cards, each 4 span
06:52.45cuscoone of them is setting the clock source as internall
06:53.07cuscoand and: wct4xxp 0000:04:01.0: All spans in alarm : No validspan to source RCLK from
06:53.14cuscohelp !
06:54.50irrootin the /etc/dahdi/system.conf you can set the clock priority
06:55.03irrootto the one used
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06:56.47cuscoyou mean the line span=X,X,X
06:57.42cuscothe first 4 spans are fine
06:57.58cuscothe second card, next 4 are internallly clocked
06:59.07ChannelZWhat are they set to?
07:02.10cuscoIm ressetting them as we speak, but the behaviour is always the same... they are on span=X,X,0,ccs,hdb3,crc4
07:02.21cuscowhere X is incrementig from 1 to 8
07:02.45cuscowhatever I set the second digit to, doesn't seem to matter
07:02.59irrootif it is not used or if its suppling clock set to 0
07:03.38cuscowell I should be getting clock from telco, right?
07:03.57cuscospan1 span2 span3 and span4 are on telco1
07:04.00cuscoall working good
07:04.03irrootcusco yes on the telco links its 1,2,... but others 0
07:04.13cuscobut span5 and span6
07:04.21cuscoaren't
07:04.55cuscowhat do you mean on the telco links?
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07:05.01cuscoaren't they all telco?
07:05.05ChannelZI'd assume each of the separate physical spans would be independent of each other
07:06.31ChannelZnevermind
07:07.26ChannelZI wonder if something else is wrong that you're not getting a clock from the remote end
07:10.13cuscook behaviour is: as soon as dahdi modules are loaded, spans 1..4 are yellow and 5..8 are OK. I have a script looking at dahdi_scan
07:10.17cuscoand 2 seconds after
07:10.20cusco1..4 are OK
07:10.26cuscoand 5::8 yellow
07:10.43cusco/etc/init.d/dahdi start && dahdi_scan > /tmp/test.txt && sleep 3 && dahdi_scan > /tmp/test2.txt
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07:12.41ChannelZcan you plug any of the 5-8 spans into the 1-4 ones, just to make sure they're functional?
07:13.11cuscohttp://paste.debian.net/124735/
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07:13.33cuscoim not at the place right now, th I can cann a colleague to do it
07:13.41cuscobut im pretty sure they will be ok
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07:15.51ChannelZI'm not
07:16.15ChannelZAre these new lines
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07:22.01cuscono
07:22.31ChannelZSo this was working and suddenly stopped, or..
07:24.11cuscothere was a power cut
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07:26.00ChannelZWell I think the sync is a red herring, in so much as probably more of a physical link problem (or the card got killed.. maybe not likely since it's showing up and all, but who knows)
07:26.08cuscook it works
07:26.20cuscoI moved 2 cables from card2
07:26.22cuscoto card1
07:26.37cuscoand now I have 2 cables of each telcon on card1
07:26.41cuscoand its working
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07:28.03ChannelZAnd moving them back fails?
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07:30.35jacc0good morning all :)
07:30.44ChannelZoh happy day
07:33.04irrootlo there
07:35.41dymMorns
07:36.46jacc0is there anyting in asterisk sceduald for monday morning around 6:00
07:36.48jacc0?
07:37.05jacc0*scheduled
07:37.28ChannelZYes.  Joy and merriment.
07:37.33dym:D
07:37.39jacc0this is the second week an asterisk server is reporting : cpu user usage of 89.1% matches resource limit [cpu user usage>70.0%]
07:37.43dymWhy would there be something scheduled :D
07:37.48jacc0on momday morning
07:38.00jacc0and :  Description: mem usage of 76.6% matches resource limit [mem usage>75.0%]
07:38.04ChannelZWhoa I just saw Astricon is in my city in October.
07:38.06ChannelZI should go.
07:38.12dym:D
07:38.14dymChannelZ: best do
07:38.21dymjacc0: is that in asterisk CLI?
07:38.23dymOr where?
07:38.32irrootcusco you can do a shutdown [complete pull the power] and then re doo have seen similar problem recently where the card is not 100% on start
07:38.33dymDoesnt sound like "out of the box" stuff.
07:38.39jacc0I've monit watching the processes
07:38.51dymjacc0: not much of an answer there, eh?
07:38.59cuscoirroot: I rebooted some times already
07:39.07cuscosorry folks I was just changing some stuff
07:39.08jacc0it is not in the CLI
07:39.15dymjacc0: whereelse?
07:39.18cuscolike DAHDI/g5/ to DAHDI/g3/
07:39.27jacc0monit is reporting it to me by email
07:39.32dymwell
07:39.37dymobviously not something to do with asterisk...
07:39.43jacc0okay
07:39.53irrootcusco another trick i use with E1 is to toggle the crc4 for about a minute seems to tickle at least our local telco into shape
07:40.18cuscowhat do you mean by that?
07:40.50cuscothat might be usefoul later, but right now... I see that the problem keeps being the me being internally sync
07:41.03cuscothe sync source changes fast upon dahdi restart
07:41.08cuscoI dunno why
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07:43.04cuscook im gonna go there now
07:43.08cuscobbl
07:45.18ChannelZI'm going to bed
07:45.25ChannelZtoodles
07:45.33irrootChannelZ catch ya round
07:45.37irroothaz a good one
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08:03.34atanAsterisk 10, seriously? Yikes. I swear it was just the other day I moved to 1.8.
08:03.42atanshakes head
08:03.50atanI wonder what's in 10. Hmm. :-)
08:04.00dymNo presents, thats for sure.
08:04.24irrootatan 1.10 = 10
08:04.31irrootthe 1. is droped
08:04.38irrootthere some nice goodies in 10
08:04.55dymFree Chocolate?
08:05.08dymIf not, im not interested.
08:05.27*** join/#asterisk cookieliu (~cookie@112.64.189.8)
08:05.51atanirroot, I'd kill to know. I wonder if my dialplan would play nice with it from 1.8... :D
08:06.06WiretapSevenhow is Asterisk's H323 support?
08:06.15irrootatan there is not much diff from 1.8 in that regard
08:06.31atanI have had huge troubles getting my shit to sync with a MySQL database. It is killer. Blarg. All I want is to keep my clients and numbers in mysql... is that so much to ask... hah! :D
08:06.32irrootWiretapSeven chan_ooh323 is good includes T38
08:06.50atanroots for changelog
08:06.51WiretapSevenirroot, still thinking about that Avaya phone I linked you a few days ago :P
08:06.59WiretapSevenno SIP firmware, but has H323
08:07.28irrootAvaya H.323 on there switch worked fine so it could be a idea
08:07.54WiretapSevenirroot, sorry what, I don't quite get what you mean?
08:07.57irrooti use gnugk on the server to do the main grunt its overkill i know
08:08.43irrootthere PBX supports H.323 i have used that to put asterisk infront of to add features and routing to the "dumb" switch
08:09.04WiretapSevenyes, Avaya use H323 for all their stuff
08:10.51atanComplete dumbass question here, but SIP is about as secure as FTP is right? Everything flys in plain text?
08:11.24irrootatan you have TLS support
08:11.48atanBut what's the deal with the ATA adapters and phones out there now... (the cheaper ones, like old ciscos and stuff) ?
08:11.50irrootatan you can get a compromise with SRTP
08:12.18irrootatan the SPA series do there own prop. shit
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09:33.17tuxx-hey guys, does anyone know the name of the android sip phones that were displayed at astricon this year?
09:33.43tuxx-it was a normal deskphone with a big touchscreen that runned android. Can't remember the name :-(
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10:18.22jacc0@atan: here is a presentation about some xss holes in the old linksys spa and ata : http://www.owasp.org/images/6/6a/OWASPBeNeLux2010-State-VoipHacking.pdf
10:20.51jacc0I've been there
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10:35.34*** join/#asterisk djuhl30 (~quassel@121.135.82.142)
10:35.43djuhl30what exactly is dids
10:35.45djuhl30??
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11:21.27jacc0Direct Inward Dialing service
11:21.39jacc0http://www.voip-info.org/wiki/view/DID+Service+Providers
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11:25.36djuhl30Cool I called out
11:25.55djuhl30So dids give your smartphone a telephone number
11:25.58djuhl30??
11:26.12djuhl30But I am living in South Korea.
11:26.37djuhl30And I want my korean wife to call relatives, but it looks like I'll get a US number.
11:26.50djuhl30E.164 format, what is it?
11:28.32djuhl30I can call the USA
11:28.35irrootE.164 is the full international telephone number
11:28.45djuhl30But can't call my cellphone locally
11:29.03djuhl30Country code is 82
11:34.17djuhl30I got it
11:34.27djuhl3001182 blah....
11:34.40djuhl30That's in the dialplan
11:35.56djuhl30"Everyone is busy/congested at this time" circuit is busy..
11:36.04djuhl30Did I cause that or is that normal?
11:36.25irrootmight not be in the service plan ??
11:38.22djuhl30As far as I can tell I can make international calls
11:40.21irrootthat prefix may not be in the providers plan
11:40.35irrootif you can call others but not this prefic
11:40.44irroots/prefic/prefix/
11:40.58djuhl30huh?
11:41.54djuhl30KOREA, SOUTH SKT MOBILE82104030 sec6 sec$0.0550
11:42.18djuhl30I have a sk as my mobile provider
11:42.46djuhl30but it is 821064
11:43.20djuhl30I wonder if I just wasted some money
11:43.58djuhl30Received response: "Forbidden" if that sheds some light on things
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11:47.42djuhl30Cool just called my cellphone
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11:53.50djuhl30Cool my wife is speaking with her mother...
11:54.28dymI've seen that behaviour before.
11:58.55djuhl30Alright I can place calls
11:59.26djuhl30Now I want to be able to call my softphone from a land line or cell.
11:59.37djuhl30I am living in South Korea now.
12:00.02djuhl30So how do I do this so I don't get charged for making a international phone call from a cellphone?
12:01.46djuhl30I am confused really
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12:06.09dym~book
12:06.09infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook
12:06.36djuhl30Yeah I downloaded it.
12:06.46djuhl30Still trying to comprehend it.
12:06.56dymwell
12:07.01dymwhere is the asterisk located?
12:07.05dymat your home?
12:07.12djuhl30Yeah in South Korea
12:07.24djuhl30I have two nationalities wanting to call me.
12:07.28dymand you have a trunking line? as in the asterisk is connected to the public phone line?
12:07.29djuhl30US and South Korean
12:07.47djuhl30I bought a dids whatever that means
12:07.59djuhl30I am just doing stuff not understanding it all.
12:08.15dymwell
12:08.27djuhl30Using floweroute.com
12:08.57dymthats none existing
12:09.14djuhl30huh?
12:09.54dymWell anyways. With DID you usually get assigned a phone number
12:10.08dymfor which the calls will be routed to your pbx, if its registered with the DID
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12:13.39djuhl30So basically now I have a US number
12:13.47djuhl30I just called it with skype
12:13.58djuhl30Does nothing for my Korean relatives
12:14.01djuhl30argh
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12:15.50djuhl30So how do I get a South Korean number?
12:15.59dymYou should have with your DID bookment
12:16.02djuhl30This is really confusing..
12:16.11dymElse seek for a sip trunking provider instead
12:16.21djuhl30Huh?
12:16.33dymdjuhl30: read the book from the start.
12:16.44dymand dont just rush over things
12:17.02djuhl30Like beginning to end?  I wish I had more time.
12:17.19djuhl30I wake up at 0330 AM tomorrow
12:19.58dymNo as in TOC
12:20.02dymutilize the internet
12:20.04dymetc :D
12:20.14djuhl30Working it now.
12:20.23djuhl30Just not getting too many results.
12:20.34djuhl30But I guess my parents can call me cheaply now.
12:21.08WebSprocket[PC]http://didlive.com/ for ddis
12:25.25djuhl30But I am living in South Korea
12:26.18*** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista)
12:26.41dymdjuhl30: http://www.sipgate.com/
12:27.49WebSprocket[PC]http://didlive.com/ supply worldwide.
12:27.57WebSprocket[PC]or sipgate
12:28.39WebSprocket[PC]http://www.mydivert.com/virtual-number-voip/south-korea-voipnational-did-numbers.html
12:29.01WebSprocket[PC]Google returned alot of results by searching "SIP DDI South Korean"
12:30.07tuxx-hey guys, does anyone know if there are android deskphones in production yet? I found the andyphone AD-1000, but doesn't seem like its for sale yet. I know there was an android deskphone displayed at astricon2011, just cant remember the name.
12:30.52djuhl30I did DID.. maybe my problem
12:35.13dymdjuhl30: ive been using sipgate pretty successful - maybe worth a shot
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12:59.53ixyd_hi guys is there any documentation about the difference between the Goto() application and the goto statement?
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13:07.35ixyd_or maybe someone can explain? :)
13:08.18dymhttp://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
13:08.21dymhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Goto
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13:10.04ixyd_to me it looks like the same?!
13:10.14krdianhi
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13:12.08gravinis integration between ericsson MSC and asterisk SIP possible>?
13:12.33tuxx-dym: goto() is just an application in the dialplan. So yeah, answering your question, they are the same :)
13:12.50dymtuxx-: cant remember having asked any, but thanks.
13:13.09ixyd_i am asking because of this: "Suggestion: Use the goto statement instead of the Goto() application call in AEL." which appears after loading ael dialplan with goto() application
13:13.12tuxx-oh w8, i'm mixing up nicknames. my bad :-)
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13:20.34catphishwhat is the recommended version of spandsp for fax?
13:24.27catphishi'll stick with the latest pre version
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13:33.01ixyd_@tuxx- so do you have an idea whats the reason for the suggestions to use the statement insteand of the application, btw: the suggestions is displayed as a warning while reloading the dialplan...
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13:39.06cerienjean@gravin - I have a ast1.6.1.20 with chan_ss7 linked to an E/// MSC - no issues at all
13:39.34gravindo you documented that
13:40.06gravinany guides/docs that i can refer
13:40.49cerienjeannot quite....
13:40.53cerienjeanwhat I did:
13:41.08cerienjeanget two digium cards & two boxes, make a cross cable
13:41.15cerienjeanconnect them in E1/ISDN
13:41.16cerienjeanfirst
13:41.25gravinok
13:41.30cerienjeanthen install chan_ss7 & switch to SS7
13:41.39cerienjeanthen went to the operator.
13:41.45cerienjeanthe E1 went up immediately
13:42.00cerienjeanthey then had minor config identical to when they setup anyh E1
13:42.14gravinok
13:42.33gravine1 is connected to MSC, which has identical conf
13:42.37cerienjeanI choose chan_ss7 over lib_ss7 as I had some issues with caller id on outgoing calls not being presented in my test setup
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13:43.09gravinoo
13:43.19cerienjeansystem.conf:
13:43.23cerienjeanspan=1,1,0,ccs,hdb3
13:43.23cerienjeanbchan=1-31
13:43.26cerienjeanfor 1 e1
13:43.34cerienjeanloadzone        = fr
13:43.34cerienjeandefaultzone     = fr
13:44.23gravinok
13:44.38cerienjeanss7.conf
13:44.39cerienjeanhttp://pastebin.com/jTzSuRfH
13:44.47cerienjeanwith my hostname being pollux
13:45.10jacc0hehehe, the new cisco phones accept a pushxml that can controle all phone features. it has a default usename and password sdkapps:sdkapps. haven't found an option to change the username/password
13:45.17cerienjeanin case you havent already, sign up to the asterisk-ss7 mailing list
13:45.48cerienjeanthat was a no brainer to install - but maybe I got lucky !
13:46.08gravinok
13:46.10cerienjeanonly issue, is every 2-3 days, the links reset without dropping the calls
13:46.27gravinwhich means
13:47.05cerienjeanI get a bunch of messages on the CLI saying that the MTP link is down, then goes up again. I havent noticed the calls being interrupted, so I havent
13:47.13cerienjeaninvestigated further
13:47.57cerienjeanI must also say that the operator wasnt to annoying with protocol testing, so if there are some protocol issues, I would not know
13:48.06cerienjeanthe system works & everybody's happy
13:48.13cerienjeanlive for almost 2 yrs
13:48.20gravinnice, are they routing the GSM calls to sip ?
13:48.49cerienjeanactually not, I am using them only for SIP to GSM (and transit trafic to fix lines)
13:48.57cerienjeanI dont do incoming
13:49.21gravinok but its the same
13:49.31cerienjeanI would not expect any surprises !
13:49.41gravinhaha
13:49.54gravincan i pm you?
13:50.02cerienjeansure !
13:50.02*** join/#asterisk shine (~stroll@lamantin.achamo.net)
13:50.30cerienjeanI am new to IRC... can you get my email or u need it ?
13:50.43*** join/#asterisk jkroon (~jkroon@dsl-241-238-75.telkomadsl.co.za)
13:50.55gravinyes
13:51.04gravinemail would be great
13:51.20cerienjeancerien dot jean
13:51.25cerienjeanat gmail
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13:54.14gravingot
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14:08.37WIMPyI can't find the enum switch any more. Has it disapperared?
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14:11.16tzafrir_laptopcerienjean, I see that your email is hidden in your nick registration preferences.
14:12.05tzafrir_laptopThat said, it is not written in the channel logs :-)
14:12.19irrootevening folks
14:12.41WIMPyHi irroot
14:12.50tzafrir_laptopcerienjean, if you don't want suh information public, sent it in a Private Message (PM)
14:13.24tzafrir_laptopWhich in your client is probly simply done by right-clicking on the person to send to
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14:13.51tzafrir_laptopoh, and, hi irroot
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14:17.08cerienjean@tzafrir_laptop: thanks.... I am using pidgin, and could find this
14:18.10cerienjeancouldnt
14:18.26p3nguin_How about "/msg nickname" or "/query nickname" ?
14:19.11cerienjeanok, got it with pdigin.... right click is too sophisticated.... just double click on the name..... I've learnt something today, and will see if spam increases
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14:19.53p3nguin_Spam will increase because you feel like you learned something?
14:20.05cerienjeanno, coz I posted my email on a public place !
14:24.30schmidtsirroot welcome back ;)
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14:49.51Kattyhello
14:50.43jacc0hi katty
14:51.06Kattywhat's the news.
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14:56.03WIMPySomeone stole the enum switch.
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15:33.39ChannelZThis is weird.
15:34.08WIMPyyes
15:34.16drobuddyHi everyone. I have a quick question regarding analog vs. digital ports. If we were to utilize a digital circuit, all we would need for the asterisk box would be a WAN card, correct?
15:34.21ChannelZI'm having massive audio problems between me and my * at work
15:34.52WIMPydrobuddy: What kind of digital?
15:35.11WIMPydrobuddy: That's generally only true for VOIP.
15:35.49drobuddywimpy: I'm still quite new to telephony. But, we just acquired a data center that has DS-3's running to it, so we could split off a t1 or use the DS-3 circuits from what I understand.
15:36.22WIMPydrobuddy: T1 != Ethernet. You need an onterface for that.
15:36.42drobuddySince we just bought the NOC, we're trying to find a cost-effective method (other than CudaTel) to integrate three offices.
15:36.51drobuddyonterface? That's a new one... Let me Google it real quick.
15:37.08WIMPyEither PCI or in form of a VOIP gateway or with something that does proprietart Ethernet transport.
15:38.08WIMPyinterface off course.
15:38.13WIMPyjust can't type.
15:38.33drobuddyThere we go, I thought there was some new jargon in the industry :D
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15:38.56pabelanger~tdm
15:38.56infobothmm... tdm is Time Division Multiplexing. It is a scheme in which numerous signals are combined for transmission on a single communications line or channel. Each signal is broken up into many segments, each having very short duration.
15:39.26drobuddySo, really we would need a TDM line card, instead of a standard T1 network card?
15:40.01WIMPyNe. you need a T1 card, but one that has drivers that can be used with Asterisk.
15:40.09petern_your DS-3s are data circuits, right?
15:40.13drobuddyCorrect.
15:40.29pabelangerdrobuddy: http://store.digium.com/telephony_card_selector.php
15:40.33drobuddywimpy: Okay, are there any recommendations?
15:40.37WIMPyOk, so you're talking IP, not telephony?
15:40.39drobuddyThere we go, thank you! ;)
15:41.08drobuddywimpy: Correct, they are data circuits that we would like to use for VoIP and VoIP to PSTN.
15:41.26*** part/#asterisk catphish (~catphish@gateway.office.atechmedia.net)
15:41.32WIMPyThen forget the whole thing. IP is IP.
15:42.00drobuddyExcellent, that is what I was hoping to hear
15:42.17drobuddyThank you both (and the rest of the community for that matter). I truly appreciate your time and insight.
15:42.18*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
15:43.17drobuddyPCI-E would be the preference, correct?
15:43.31drobuddyNvm, Google answered it
15:43.43petern_...
15:45.08*** join/#asterisk eggsy84 (~james@109.111.202.18)
15:45.29drobuddyIs it possible to have multiple calls use the same DID over a digital circuit (10 people on the same phone number at once)?
15:47.51eggsy84hi all, does anyone know if it is possible to make asterisk only use particular queue members (agents) when adding a call to a queue? (Effectively limiting the agents that could answer the call) or can this only be done by having a different queue?
15:47.54irrootdrobuddy YES
15:49.10irrootdrobuddy have one site with 2X30 [PRI] and 600 [DID] they often have more than 30 calls to there main helpdesk DID
15:49.40irrooteggsy84 look at priority for the members
15:49.47drobuddyirroot: Thanks
15:51.59Kattycannot brain today, i haz the dumb
15:52.32irrootKatty using internet explorer ?? recent study shows it drains IQ
15:55.31eggsy84irroot: thanks do you know of any particular useful docs on priority? (our scenario is that depending upon the number called only certain queue members - which belong to the same queue - should be used)
15:55.35*** join/#asterisk Manu18 (~megairc@lau18-1-82-246-196-145.fbx.proxad.net)
15:55.39Manu18Bonjour la room
15:55.57Manu18Y a t'il des francais pour de l'aide svp ?
15:56.21irrooteggsy84 that is skill based routing and there is a patch for it but for now you will need to have seperate overlaping queues
15:56.41Manu18NOTICE[1978]: chan_sip.c:25370 sip_poke_noanswer: Peer 'freephonie- in' is now UNREACHABLE! Last qualify: 0
15:57.10eggsy84if you have over lapping queues does that mean that potentially to people could be fighting for first place?
15:57.34irrooteggsy84 members can be part of more than 1 queue
15:58.52*** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net)
15:59.39Kattyirroot: no i think it's the people i've been hanging out with lately
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16:00.48eggsy84irroot: I'm probably not understanding this but I can't see how that would help my scenario?
16:00.59*** join/#asterisk MarKsaitis (~MarKsaiti@78-61-148-80.static.zebra.lt)
16:01.35irroothave couple queues with members in each they can overlaped/have priorities link each to a number
16:04.16voipnet-techhi everybody.  quick question about the approriate sip nat to use on my asterisk box.  here's the diagram: http://imageshack.us/photo/my-images/850/examplediagram.png/   customer is requesting RTP go directly from IP to IP phone when on any internal phone call.  customer only wants asterisk to relay the RTP if it is an outside call (call to remote IP phone or call to PSTN)  right now all calls network wide get relayed RTP through the asterisk box an
16:04.28eggsy84ok I'll have a look at that and see if I can get something implemented many thanks for the help :)
16:05.27Manu18Y a t'il des francais pour de l'aide svp ?
16:06.32voipnet-techto add to the above... the asterisk box has 2 NIC's one with a 10.x IP and the other with a public IP that remote phones register to...  when i tried using directmedia and directrtp i had asymetrical RTP.  one path would go direct and the other leg would go thru the server.  it also created problems when internal users transferred eachother we would lose audio
16:07.02voipnet-techfor this network is it best just to have asterisk relay all RTP... right now i'm using NAT=no because I've got the public IP on the box... is there a better way to do this?
16:07.56p3nguin_There's a value for directmedia that should do what you're wanting.
16:08.58voipnet-techhi p3nguin_thanks.... i have directmedia=yes but there are no re-invites happening.  in the SDP messages, asterisk always give itself as the RTP target.
16:10.23*** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net)
16:10.48voipnet-techwhen i use directrtpsetup=yes then asterisk does pass along the first phones IP in the SDP to the second phone, but when the 2nd phone replies with a 200OK with SDP, the server gives phone 1 it's IP address, and the reverse leg of RTP is relayed through asterisk
16:11.48voipnet-techif i can fix the 2nd leg maybe i'll be ok, but not sure why it would be doing that.  also i'm concerned that if i do do this then there will be problems with remote phones being able to communicate to inside phones.  asterisk must somehow be aware which phone is local/private IP and which ones have a public ip?
16:13.02*** join/#asterisk neurosys_ (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net)
16:13.42*** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net)
16:13.49Dovidanyone in here from France?
16:13.52*** part/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br)
16:14.07voipnet-techalso i'm not sure if this makes a difference or not but the router that connects to the internet is a SIP ALG router so the * box gets it PSTN calls from that router on the 10.200.10 IP.  that router will only talk to the * box so i cant let the IP phones send direct RTP for PSTN or for remote IP phones
16:15.25voipnet-techcan anyone confirm that based on these factors it makes the most sense just to have asterisk relay the RTP and use no direct media?
16:16.02voipnet-techi'm sure that if some random bloke in #asterisk confirms this for me and i show the customer they'll believe me :)
16:16.13*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
16:16.30dan__tAlright, I'm at the point now where my dialplan is unreasonably complex and borderline unreadable, to the point where it's extremely difficult and time consuming to troubleshoot..  Now that I've got the basics down, I'm looking to learn more about best practices for writing dialplans, how to break up contexts and priority groups, things like that.  Anyone got any good links or a few tips for me?
16:17.14*** join/#asterisk estranger (russ@russ.trifecta.com)
16:17.48carrarclean it up!
16:17.53dan__t!!
16:18.03carrarstuff what you can into a db
16:18.06*** part/#asterisk madduck (~madduck@debian/developer/madduck)
16:18.16carrarwrite some AGI moojo
16:18.25dan__tpass.  heh
16:18.29carrarsplit out your files
16:18.41carrarAGI is easier then you probably realize
16:18.43dan__tRight now I'm using one context and a lot of same =>
16:18.53dan__tI'm familiar with AGI.  I don't want to do that right now though.
16:19.07carrarthen clean it up tomorrow then
16:19.10carrargo outside and play
16:19.33neurosys_Anyone have a recommendation for a windoze click-to-dial that works and works well?
16:20.20dan__tI want to do it today.
16:20.33dan__tWhat do you mean split out my files
16:20.38dan__tcan I 'include' and stuff?
16:20.58carrarYou need to read the Asterisk book
16:21.01carrarthat will help alot
16:21.02dan__tWhat's a rule of thumb for breaking stuff off in to different contexts?  Good practice to keep using 'same'?
16:21.05carrar~book
16:21.05infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook
16:21.21dan__tI have 1st and 2nd Ed, I've been over them a few times.  I promise.
16:21.33dan__tAnd I hit up the 3rd ed once in a while.
16:21.42carrarthen you wouldn't be asking some of the questions you are asking
16:21.48dan__tI'm looking to learn good practice, not the actual tech.  I'm doin pretty good learning the tech.
16:22.11dan__t(which neither of those books went in to a whole lot of detail on)
16:22.15dan__tSo, yes, I probably would still be asking.
16:22.36dan__tElse I wouldn't be asking some of the questions I am asking.
16:22.51ChannelZIt's hard to recommend what to do if we don't know what you're doing
16:23.24ChannelZThere's 100 ways to skin a cat so suggesting how to clean up something without knowing why it's a mess...
16:23.29dan__tWriting a dialplan?  Trying to keep it organized and sane.
16:23.31dan__tYes, I understand.
16:24.03ChannelZContexts are like privacy screens
16:24.09dan__tright
16:24.33ChannelZIf "incoming" calls don't need access to every internal extension in existence, separate them.
16:24.56dan__tRight, that makes sense.
16:24.57ChannelZIf you have have IVRs, generally they are in separate contexts.
16:24.59*** join/#asterisk manawenuz (~manwe@ec2-67-202-60-57.compute-1.amazonaws.com)
16:25.02ChannelZThat sort of thing.
16:25.22manawenuzHello everyone ,
16:25.27manawenuz:)
16:25.33dan__tWould it be unwise to separate IVRs in to separate contexts, maybe one for each submenu or something like that?
16:26.57ChannelZNo, that's usually how you do it anyway.. if you have "press 1 for english" and then "press 1 for tech support", you can't really have two extension 1's in the same context.
16:27.16dan__toh, right.
16:27.30ChannelZIf it's a REALLY sprawling IVR I'd say do it in AGI in some other language in which case it can remove a ton of clutter from the dialplan
16:27.51dan__tya
16:27.59ChannelZbut even then you could move all that junk into a separate file and #include it to keep the main extensions.conf feel tidier
16:28.02manawenuzin the middle east , and i use asterisk machines in every city , in the country , 31 in total , each with 16 E1s attached to it , and we'd like to start local termination / origination with the government ,but they require us , to provide them with some sort of access , so they'd be able to monitor all the calls and be able to record them , if they needed , as i said , does anybody have any suggestion what is the best approach t
16:28.55dan__tYep.  I'm trying to isolate the known good blocks of my dialplan.  There are a few bugs in them and I'm trying to hammer them out.
16:29.32dan__tAnd I think I should get away from using 'same' and the 'n' priority for a while until I get a better feel for exactly waht I'm doing, so I'm more likely to find line numbers and positions in the dialplan where things exploded etc etc.
16:29.59ChannelZwell, I'd say that not using 'n' is actualy more error-prone
16:30.40ChannelZIf you have a bunch of priorities in an exten and don't number them properly... also makes it hard to move things around while you're making something work the way you want if you have to keep renumbering
16:30.44dan__tReally?
16:31.04dan__tOh, right.  BUt at least gives me a reference point as to where things may have failed, right?  Wouldn't they be mentioned in console debug?
16:31.31ChannelZThe console will show you the actual priority, and you can count to figure out which line it is in your dialplan if it's not obvious
16:32.10dan__toh
16:32.44ChannelZUnless you have 100 priorities in a single extension it's not too tough to find
16:33.54dan__tI have maybe 15-20 I guess, and half of them are Verbose()
16:34.06dan__tJust to get checkpoints on data that's pulled from ODBC etc etc
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16:34.08jayteethis channel feels somehow incomplete without russellb in here :-(
16:34.54*** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net)
16:35.08ChannelZImitation is the sincerest form of flattery...
16:35.53dan__tthanks for the tips, ChannelZ.
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16:37.41ChannelZgood luck
16:37.45dan__tThanks.
16:37.56dan__tI'm going to scrap the dialplan and re-write it a little more logical this time.
16:40.10*** join/#asterisk frawd (~francois@186.Red-217-125-157.staticIP.rima-tde.net)
16:40.40ChannelZrennovation is fun
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16:55.00dan__trefactoring?  can I use that one? heh
16:55.30Qwelljaytee: He'll be around - he's just busy moving and such.
16:58.51*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
17:09.00linuxgeckoI'm trying to convert an office over to asterisk, using google-voice as my link to the PSTN. as i read, i find that google-voice only does G711u-law, does that mean that if my ata connects to my asterisk, using ulaw also, i might have a decent chance of being able to fax over voip?
17:13.03linuxgeckofrom what i read, the biggest enemy of fax over voip is commpression, and ulaw is an uncompressed format, yes?
17:13.48*** join/#asterisk cyborg-one (1000@85-238-125-110.broadband.tenet.odessa.ua)
17:14.52voipnet-techlinuxgecko, i've had good luck with g711u as long as QoS is used to ensure good voice quality
17:15.01voipnet-techg711u for fax that is
17:16.43*** join/#asterisk Cain (~Geek@unaffiliated/cain)
17:17.59*** join/#asterisk neurosys_ (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net)
17:18.24neurosys_Anyone recommend a good working click-to-dial app for windoze?
17:19.03coppiceu-law over IP works great for fax
17:19.05coppice.... if you have perfect synchronisation
17:19.06coppice.... and no echo cancellers doing funky stuff
17:19.08coppice.... and no jitter buffers doing funky stuff
17:19.10coppice.... and no packet loss
17:19.12coppice..................... :-(
17:19.55ChannelZ.... and a cookie
17:26.52*** join/#asterisk mjordan (~mjordan@nat/digium/x-nhbaynrnfdyrdszp)
17:29.43linuxgeckocoppice:  so fax works on voip with a perfect storm of good luck.  or very nearly.
17:30.41coppicesome would say it works OK on a well engineered path, but how often do you meet a good engineer?
17:34.22*** join/#asterisk gogasca (~Adium@nat/cisco/x-hgoultkudsibsgnt)
17:40.54linuxgeckocoppice: i'm basically the network engineer for the office in question. after it goes to the ISP, it's just as much out of my hands ans it is for nearly everyone :)
17:41.50coppiceand there lies the problem. its always somebody else's fault :-)
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17:49.48conveyAnyone using VZB SIP TG's?
17:50.48linuxgeckoTG ?
17:51.02conveytrunk groups
17:51.07conveyIP trunking
17:51.13conveyvia SIP
17:51.29linuxgeckoas soon as you said trunk groups, it made sense :)
17:51.35conveyThere was an outage today.  I was just wondering if anyone else was impacted.
17:51.47convey:)
17:51.51linuxgeckoconvey:  gotta love outages :-(
17:52.20WIMPyIt's the new standard.
17:52.52linuxgeckoWIMPy:  outages??:)
17:52.53conveyAgreed.  But it sure beats TDM :)
17:53.01WIMPyyes
17:56.44linuxgeckoquestion.   i'm trying to setup an office with multiple g-voice lines. that's not the problem.  the issue i don't know enough asterisk to know if i can, or how,  is how to get asterisk to accept whatever credentials are provided for a sip peer registration, without me having to put them in sip.conf,  and then open a jabber/gtalk connection using the sip credentials it was fed..    how nasty of a configuration am i looking
17:57.18ChannelZSpeaking of, I wonder if cusco ever got his other PRIs working last night
17:57.19linuxgeckothe goal i'm hopeful to achieve,  is to only have to put the gvoice credentials on the ATA.
17:57.56jayteelinuxgecko, why wouldn't you want them on the asterisk server?
17:58.42*** join/#asterisk timahvo1 (~rogue@41.212.123.197)
18:00.04linuxgeckojaytee: the agents each gtalk goes to are independent, and i dont want them to worry that i know their password,   on the other hand, i don't want them monkeying in the asterisk config files, or anything included into them, ( most of them have never even head of linux, let alone asterisk).
18:01.29*** join/#asterisk beetlejooz (~beetlejoo@12.237.61.93)
18:01.31ChannelZthen you are screwed
18:01.46linuxgeckoChannelZ:  so i can't do what i want?
18:02.20*** join/#asterisk fenlander (~fenlander@82.152.81.57)
18:02.29ChannelZwell Asterisk is acting as the gtalk client, so no there's no way for it to login without knowing how to login.
18:03.10linuxgeckoChannelZ:  and i can't create dynamically defined channels to gtalk?
18:03.14linuxgeckoon the fly?
18:03.36ChannelZnot that I'm aware of
18:03.44linuxgecko:-(
18:03.54ChannelZThat would also mean they could only make calls if that were even possible
18:05.03linuxgeckoChannelZ:  not possible, or not currently coded so i could?
18:05.40beetlejoozWhat could cause about a 10 second delay before calls go out through my SIP carrier?  I know that is super vague, sorry.
18:06.54beetlejoozI also periodically get this "Unable to create channel of type '' (cause 66 - Channel not implemen ted)"
18:07.12beetlejoozbut get a delay whether I see that error or not
18:07.29linuxgeckoChannelZ:  yes..  it would.     what i want is to put the gtalk creds on the ata,   have the ata send them to asterisk by trying to register a sip channel using them. asterisk accepting it, not having previously known what good and bad creds for gtalk are,  open a gtalk channel with the creds the sip channel registered, ( and hopefully kick an error to the ata if the gtalk fails)
18:08.25WIMPybeetlejooz: Sounds like at least one major error in your dialplan.
18:08.32ChannelZbeetlejooz: you'd have to watch the console and see, is it a delay placing the call (IE Dia() command has executed but nothing is happening) or is it a delay in the dialplan matching the extension
18:09.05*** join/#asterisk SamuraiAlba (~SamuraiAl@pool-96-248-118-131.cmdnnj.fios.verizon.net)
18:09.11SamuraiAlbaGood bacon to all!
18:09.53SamuraiAlbaCan I use a sipgate account and google voice together with asterisk?
18:10.14ChannelZlinuxgecko: not possible that I can't think of.  The gtalk module doesn't provide functions/applications to create new accounts on the fly.
18:12.20*** join/#asterisk Praise (~Fat@unaffiliated/praise)
18:12.40ChannelZYou could possibly devise your own system to re-generate the config file on the fly and then issue a reload on the modules but that, I think, would have an impact on any gtalk calls that happened to be going at the time
18:13.27beetlejoozast_rtcp_read: RTCP Read too short
18:13.31beetlejoozis the phone causing this?
18:14.21beetlejoozsuddenly the delay is gone....that makes sense
18:14.24beetlejoozwtf
18:15.17ChannelZcould be networking issues, hard to say
18:15.54beetlejoozI put a sniffer on the fastEth and didn't see any irregularities
18:16.23beetlejoozfrustrating
18:16.37beetlejoozI'm really not liking digium boards
18:16.54beetlejoozseems like there are too many ways for them to have problems
18:17.04ChannelZWhat are you using?
18:17.17beetlejoozcurrently an AEX2400
18:17.32ChannelZI thought you were using SIP.  Which leg was being delayed?
18:17.38beetlejoozI am using SIP
18:17.45beetlejoozthe phones terminate through the digium board
18:17.53beetlejoozanalog phones
18:18.19beetlejoozplus, the owner doesn't want to go pure SIP so I'll have to setup the POTS lines through the digium as well
18:18.42ChannelZwell either your dialplan is such that Asterisk is sitting there waiting for more digits to be dialed, or something else is awry
18:18.43beetlejoozI was getting dialtone but after the number is dialed, there is a 10 second delay before it connects
18:19.12ChannelZDo you have something silly like an extension of _X.  ?
18:19.19beetlejoozno
18:19.42beetlejoozand now the delay has gone away since chatting here
18:20.06WIMPyWell, you do have to be careful with your dialplan if you want to make it work with dialing.
18:20.34beetlejoozhttp://pastebin.com/5MzkMGsu
18:20.36ChannelZwell again you haven't really said if you've tried to track down where the delay is coming from based on the console output
18:22.32ChannelZcould be your AGI for all I know
18:22.34beetlejoozI don't see any errors on the console...just a time difference
18:22.35WIMPyTwo identical extensions is not going to work.
18:22.55ChannelZwell, the second one might :)
18:23.20WIMPyI think the first, but there we have th problem :-)
18:23.30drobuddyDo I need multiple SIP trunks per DID?
18:23.37ChannelZyes, but where is the time difference coming from?  Does it sit for 10 seconds after you finish dialing before you ever see the 'Set' application execute?  etc
18:23.56WIMPydrobuddy: Depends on your ITSP.
18:24.16drobuddywimpy: Can you recommend an ITSP?
18:24.41gogascaI use Skype SIP lines works pretty cool
18:24.44WIMPydrobuddy: Not for your area. But infobot might.
18:24.51WIMPy~itsplist-us
18:24.52infobotHere 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 , http://voip.ms and http://flowroute.com
18:24.58drobuddyThank you!
18:24.58gogascayou can get DID and 1 channel for around 12 bucks per month
18:25.04beetlejoozChan: after I finish dialing it sits for 10 seconds
18:25.28*** join/#asterisk defsdoor (~andy@cpc15-sutt4-2-0-cust41.perr.cable.virginmedia.com)
18:25.33drudge`bandwidth.com just bought our previous itsp, DashCS
18:26.18ChannelZbeetlejooz: be specific
18:26.44*** join/#asterisk gogasca (~Adium@nat/cisco/x-fveityghdngcrzga)
18:27.38defsdoorare there any known issues with * 1.8.5.0 locking up ? I've got a live site so haven't been able to do any real diagnostics but it's locked up twice now and am trying to check everything before it happens again
18:27.56defsdoorfirst time it locked up * was at 100% cpu
18:28.03defsdoorsecond time everything was idle
18:28.27WIMPy~collectdebug
18:28.28infobothmm... collectdebug is a method of collecting logs allowing others help troubleshoot an issue.  Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
18:28.32WIMPydefsdoor: ^^
18:28.36defsdoornext time I will try to debug it more
18:29.01*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
18:29.12defsdoordifficult to do a thorough job when being screamed at :)
18:29.13WIMPyThere have been many issues causing Asterisk to lock up.
18:32.03linuxgeckodefsdoor: especially with an issue as serious as a lockup, it's important to bring us the debug info, so we know more about what's going on under the hood.  so we CAN help. :)
18:32.14defsdoorI will try
18:32.28defsdoorI've already turned on verbose/debug logging etc..
18:34.13WIMPyThat might help, but the important information is the 'core show locks' thing that needs to be enabled at build time.
18:35.28defsdoorwhere in menuselect is that ?
18:36.20WIMPyCompiler flags -> DEBUG_THREADS
18:36.52defsdoorok done
18:41.12defsdoorI suspect it might be a hardware issue - some odd activity on events/6 while it was locked most recently - but I need to investigate thoroughly
18:44.55SuPrSluGany way to tell if externnotify is executing after i leave a message? i don't see anything at the cli
18:45.37jayteecore set verbose lebenty-leven?
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18:47.02SuPrSluGi have it at 10. more?
18:47.51jayteeSuPrSluG, actually I was being my silly old self. I don't think SIP Notify messages get displayed on the CLI no matter what level of verbosity is used.
18:48.30*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
18:48.48jayteeyou might use sip debug to watch the peer?
18:48.50beetlejoozlooks like congestion at the SIP provider
18:48.51SuPrSluGexternnotify is a sip message. thought it would be system since it calls a script
18:48.56beetlejoozhad to dig for that one
18:50.17*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
19:02.55defsdoorhmm
19:03.08defsdoorit's locked again now - and theres no one in so incoming calls only
19:06.01dan__tChannelZ, did you say I could use CHANDUMP() to get all the information about an active channel?
19:06.17dan__tVariables (global and channel), statuses, anything I might possibly need to help debug?
19:06.36defsdoorhttp://pastebin.com/jjqEWg2V is anything in that pertinent ?
19:07.20*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
19:08.36*** join/#asterisk angryuser_laptop (~angryuser@66.210.20.81.dynamic.adsl.abo.nordnet.fr)
19:12.03defsdoorWIMPy, is that pastebin useful ?
19:13.53WIMPyNot to me, but hopefully to someone else.
19:16.21ChannelZdan__t: I didn't.  Not heard of it.
19:16.49dan__tah
19:17.20ChannelZI see there is a DumpChan application
19:18.51dan__tYea, I just found it.
19:19.16dan__tExecuting [9998@error:2] DumpChan("SIP/desktopsip-00000002", "10000") in new stack
19:19.21dan__tDoesn't do... anything, so far as I can see.
19:19.37dan__toh, i had to increase console verbosity.  oops :)
19:21.47dan__tI read that when doing ODBC, a return value of -1 is reserved for indicating error.  I'm not sure if this is sufficient:   same    => n,GotoIf($[${ODBCROWS} = -1]?error,9998,1)
19:22.08*** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com)
19:22.30defsdoorhmm I can repeatedly lock it now
19:23.55*** join/#asterisk gravin (~gravin@61.76.50.60.brf01-home.tm.net.my)
19:24.10defsdoorhttp://pastebin.com/MEmGjndJ this is another lockup
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19:27.31ChannelZdefsdoor: doing something specific?
19:27.34defsdooryes
19:27.50defsdoora call in a queue that doesnt get answered
19:28.14defsdoor<PROTECTED>
19:28.20defsdoorlogged that then locked
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19:34.52*** join/#asterisk jetlag (jetlag@pool-71-168-246-149.cmdnnj.east.verizon.net)
19:35.04neurosys_Anyone recommend a good working click-to-dial app for windoze/IE?
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19:40.23dkatz333Howdy all, intersting problem here.  Asterisk 1.8.5 with fax for asterisk doesn't respond to SET(FAXOPT(maxrate)=9600) or res_fax.conf settings.
19:41.00dkatz333NoOp shows that it's set but still receives at 14400
19:41.29defsdoorit seems there have been deadlock issues fixed with queues
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20:41.31cerienjeanneurosys: for windows + FF, nojee click....
20:43.13KavanSnojee click for the win
20:43.40HiveIs there a ${FAXOPT(field)} for fax durration?
20:47.21*** join/#asterisk billmania (~bill@38.98.130.98)
20:49.17*** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net)
20:49.49d_preston215My PRI provider says everything we send to them is "presentation restricted" and doesn't pass the number we send on outgoing calls.
20:49.53d_preston215How do we change the presentation of the caller id?
20:51.09ChannelZusecallingpres=yes  ?
20:51.13*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
20:51.32ChannelZYou have to know what the CID of the channel is though which could be not what you want depending on your setup.
20:52.33ChannelZLike if you have local SIP extensions, it might have a CID number of "300" from the device.. so you might want to specifically Set to something else before doing the dial
20:52.42d_preston215ok
20:53.22ChannelZusecallingpres in chan_dahdi.conf BTW
20:53.32d_preston215Its there.
20:53.51ChannelZok.. so chances are your channels just have weird (or no) caller ID
20:58.29d_preston215How could that be changed?
20:58.43d_preston215The channels having a different callerID.
20:59.28ChannelZSet(CALLERID(num)=1115551212)
20:59.32*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net)
20:59.41ChannelZprior to Dial()ing
21:00.25ChannelZYou can also set the callerID of every channel in chan_dahdi.conf
21:00.39*** join/#asterisk timahvo1 (~rogue@41.212.123.197)
21:00.41d_preston215Ok.
21:01.46dan__tPlaying around with a few ways of reading DTMF input from a caller... Read() has a retries variable.  Assuming I get a DTMF input, I can send a caller to that extension.  What happens if the whole time times out and they're out of retries?  Does it just fall through?
21:03.33ChannelZThe easiest way is to use WaitExten
21:04.00ChannelZBasically if you use Background or WaitExten and someone dials an extension number that has a match in the current context, it will jump to that extension
21:04.14dan__toh, right.
21:04.39dan__tHrm, what if I wanted a simple 1 or 2.
21:04.40ChannelZThen you can use the 't' extension as a timeout catcher to do something like play a "you didn't enter anything, try again" and wait again, or send them to a default extension, whatever
21:04.49dan__toh, right, that's important
21:04.58*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
21:05.36ChannelZIf you want to specifically read digits from the caller but not treat it like an extension, then use Read
21:06.02dan__tThat's what I'm looking to do I think?  Only because it would be safe to assume that at the point of using WaitExten, I'd be past extensions 1 and 2
21:06.10dan__tAssuming I wanted either 1 or 2
21:06.19ChannelZI dunno, just depends on what you're trying to accomplish
21:06.41dan__tIf I get a DTMF of 1 I want to go to one context, 2 a different context
21:06.44ChannelZRead stops and waits for them to press something, but then just gives you what they pressed as data.. which you have to then write the logic to do something with
21:07.02dan__tI think that would be the most flexible, yeah
21:07.17ChannelZIf you're meaning to jump to another extension for an IVR or whatever, it's easier to use WaitExten and whatnot unless you have other dialplan logic you need to do in between
21:07.50dan__tRight.  I'm just saying that my DTMF input which is expected to be 1 or 2 does not necessarily represent an extension.
21:08.27ChannelZLike a basic example is my setup.. someone calls in, they get a greeting and "If you know your extension, dial it now. Otherwise please wait."  It waits 3 seconds via WaitExten for them to dial an extension, all of which are defined in the same context and do whatever they do.
21:09.00dan__tunderstood
21:09.08ChannelZA 't' exten catches if they don't enter anything, and sends them to a "default" extension (like a secretary)
21:09.13dan__tyep
21:09.32dan__tok, sounds like read is what I'm looking for, since I'm looking for data moreso than an extension to go to
21:09.34ChannelZIf you're doing "Enter the year you were born", then yeah Read
21:09.42dan__tgot it, thank you
21:10.16ChannelZI believe the 'attempts' and such of Read are more useful if you play a leading sound
21:10.40ChannelZs/play/use it to play/
21:18.58dan__tgot it
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21:25.00*** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net)
21:28.55nnyis there anything similar to sip show channelstats in 1.4?
21:34.21nnynm found it
21:39.26*** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com)
21:44.16nnylooking at this patch https://issues.asterisk.org/view.php?id=10590 to provide better rtcp statistics to asterisk 1.4.42, the voip-info wiki says this patch is meant for 1.4.30, but works with later releases with some manual intervention. Any advice on how this can be done?
21:50.36*** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com)
21:51.08dan__tHey ChannelZ, using Read(), how can I tell if DTMF was read or it just timed out?
21:51.32*** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa)
21:52.42dan__tCheck the length of the variable that data was read in to?  If my maxdigits was set to 4, if this variable has !4 in it, then I know?
21:53.09a1fa[Aug  1 21:52:18] WARNING[7339]: chan_sip.c:12317 transmit_register: Probably a DNS error for registration to 12345678@sip.broadvoice.com, trying REGISTER again (after 20 seconds)
21:53.20a1fahow do I know it's honoring outbound proxy?
21:54.36a1faalright, confirmed with tcpdump, its not using outboundproxy
21:54.46*** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap)
21:55.22a1fasip show peer
21:55.32a1fashows it using the outbound proxy, tcpdump says otherwise
21:56.26ChannelZdan__t: it sets a variable I think
21:56.40ChannelZdan__t: core show application read
21:57.08dan__tShit yes it does.
21:57.08ChannelZ${READSTATUS}
21:57.09dan__tTHank you.
21:57.43a1fadoes register-> statement use the proxy?
21:58.22a1faif outboundproxy is defined?
21:59.52*** join/#asterisk hfb (~hfb@pool-96-247-53-174.lsanca.dsl-w.verizon.net)
22:02.32a1fai have outboundproxy defined in sip.conf, however my registrations are still going directly to the remote host, and not outboundproxy?
22:04.29ChannelZglobally?  I'm not sure
22:05.00WIMPyI don't think that works for register.
22:05.19WIMPyYou need to put that in individually.
22:05.37*** join/#asterisk papa_john (~papa_john@64.39.0.76)
22:07.19a1fai figured
22:22.36*** join/#asterisk Karen_m (~karen@d50-99-60-236.abhsia.telus.net)
22:22.44Karen_mwhen you get a new polycom, do you need to register the phone before it will work?
22:23.14a1fano
22:24.10Karen_mi've put in the settings for voip.ms, and no matter what it always says (NOT REGISTERED) under status for both the server/line
22:24.34Karen_mchicago.voip.ms: (Not Registered) 0
22:29.19a1fa<PROTECTED>
22:30.35Karen_m192.168.1.254, i will try that .. :)
22:30.37Karen_mthanks for the tip
22:36.24ChannelZuhmm. what?
22:38.30*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
22:38.32dymuhmm. what!
22:38.35dym(:
22:44.53*** join/#asterisk nightrid3r (~kvirc@91.176.247.79)
22:44.56ChannelZYou got your peanut butter in my chocolate!
22:45.37dan__tsame    => n,Set(LOOP=$["${LOOP}" + 1])
22:45.40dan__tDOes that look correct?
22:46.40ChannelZno
22:46.41dan__tNope.  Too many quotes I think.
22:46.50dan__tIt's not a string, if it were a string it'd be quoted.
22:54.19*** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net)
22:56.11a1faHow can I Register => using outbound proxy?
22:57.32*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
23:03.11ChannelZuse the proxy address?
23:03.19dyma1fa: a sip peer?
23:03.25a1fayes
23:03.27dymoutboundproxy= within sip.cinf
23:03.30a1faits not using it
23:03.39a1fait says it is, but tcpdump, i am seeing direct traffic to the hosts
23:03.44a1fainstead proxy
23:03.46ChannelZI mean in the register
23:03.59dymare you not?
23:04.02ChannelZregister lines have almost nothing to do with anything else
23:04.37a1fado i put it in register string somewhere?
23:04.44dymwell
23:04.48dymthe host you are registering to obv.
23:05.16a1fadoh, yes
23:05.24dym;)
23:05.26a1fado I put proxy settings in register =?
23:05.44ChannelZpresumably, since that's who you need to register to
23:06.25dym:D
23:06.34dymgawd, my adsl line is dead :(
23:06.44dymim on fallback 3G right now
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23:57.31Karen_mI am reading about how to upgrade the firmware (bootrom) on the polycom 650.  It says to put the files on the boot server in order to upgrade.   They don't mention how to setup a boot server.
23:57.57Karen_mthe phone won't connect, because the firmware is old and has a problem that has been corrected.  I read the forums, someone else has the same problem that was cured by firmware upgrading the phone.
23:59.29nightrid3rKaren_m: tftp server ?

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