IRC log for #asterisk on 20120313

00:05.46datruthwhat ports are needed to be open for asterisk to function properly?
00:05.56Kobazdepends what you want it to do
00:06.01Kobazand what protocols you're using
00:06.17datruthahh I see
00:06.27Kobazyou don't need any ports open if you're using only isdn and analog
00:06.57Kobazif you want to use it for voip, you'll probably want to use sip, which is port 5060 and media ports (generally something like 20000-21000
00:07.01Kobaz)
00:07.21datruthI have asterisk on a server I dont have physical phone I was going to do everything via the internet.
00:07.40Kobazsip most likely then
00:08.46datruthI wanted to use this as a internet conference i'm not sure if it'll work how I want but worth a shot
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00:13.23pguillemhi all
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00:13.44pguillemanybody with audiohook API knowledge?
00:14.14mythicalboxis there a trick to get Set to save a variable that's pushed out via AMI (using the VarSet packet) for a new channel? the docs say to put two underscores "__" in front of the var name, but that is not working
00:14.34mythicalboxI'm wondering if this might just be an obscure bug/use case
00:14.46pguillemThe __ means the cariable inheritance..
00:14.48pguillemvariable
00:15.10pguillemso it will be available in macros and subs in the dialplan..
00:15.21pguillemwhat are you trying to do?
00:16.34mythicalboxpguillem, here is my AMI packet capture: http://pastebin.com/0QduyYJS
00:16.55mythicalboxi have an app monitoring AMI for a crmpath to load a webpage when the callee answers the phone
00:17.15mythicalboxmy call to set is being assigned to the caller channel, not the callee
00:17.29pguillemi understand
00:18.26pguillemyou could code a small socket in javascript to plug the web into the AMI.. a simple event from the AMI could trigger a page change
00:18.46pguillemyou would have to build a parser thou...
00:19.00pguillemi have one written in AS3 in case you need it
00:19.11pguillemi'm not familiar with CRMPATH
00:19.18mythicalboxi'd be interested in looking
00:19.22mythicalboxthat's my var i'm looking for
00:19.28pguillemcool
00:19.38pguillemi totally undertand what you are trying to acomplish
00:19.55mythicalboxI obviously see that data in the AMI, but it's not directly tied to the called channel
00:20.20pguillemgive me your email and i will send you the AS3 library i wrote to plug Flash into the AMI
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00:20.50mythicalboxsent
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00:30.37p3nguindatruth: If your phones have to pass through a firewall to get into asterisk, you'll want to allow ports 5060 UDP and whatever range is configured in rtp.conf (usually 10000-20000, also UDP) to reach asterisk.
00:31.05datruthahh gotcha
00:31.50p3nguinIf your phones are on the same network as the server, there probably isn't a firewall in between.
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00:35.33pguillemYou will have
00:35.40pguillemSorry
00:45.36kessiushi, what is the difference between install asterisk, with  (./configure make and make install ...) and simply yum install asterisk16
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00:49.59pguillemGuys.. when i Dial() with the L option.. i need warning messages to be listened by both bridged channels at the same time... this is not happening
00:50.46pguillembasically, the channel.c function that plays the audio file reads the warning one channel at a time... making a huge silence for the other party.
00:51.23pguillemAnybody knows how should i fork a new thread to play the warning messages simultaneously (or almost) ?????
00:53.19pguillemchannel.c already includes utils.c and audiohook.c ... i know i could use ast_pthread_create() to do that, but i'm not an expert in handling a live channel. So what considerations should i take when firing threaded events from channel.c ???
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01:00.11kessiusdifference in which to install simply yum install asterisk16  / and compile the source
01:01.12[TK]D-Fenderkessius: Tink about the obvious nature of what YUM implies you'll get.
01:01.15[TK]D-FenderThink*
01:07.23pguillemAnybody?
01:10.23[TK]D-Fenderpguillem: I would just call dial with M()  then have that lauch a backgouned script passing it the 2 channel names.  That script would AMI Originate() local channels that would bridge to each end to play the messages
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01:24.20kessiusbut case , we need to install any module - case   need recompile ? - I will not be able because  yum install asterisk18  ?
01:27.56p3nguinYou'll either use yum to install everything, or you'll compile from source.  Don't try to do both.
01:28.20p3nguinIf you insist on compiling from source, use checkinstall to make your own package so you can manage it properly with rpm.
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01:39.42pguillemthanks TD-Fender
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01:40.46pguillem[TK]D-Fender: What cmd should i use to launch the background script?, do you mean an AGI?
01:41.54[TK]D-Fenderpguillem: Doesn't have to be AGI.  That script would want to check that the channel(s) are still up before launching the new channel to bridge in the announcement as it's process will be independant and will need to know when to terminate
01:42.33p3nguinits
01:46.01pguillem[TK]D-Fender: Good advise. I hadn´t think about it that way. I had tried to spawn a macro on answer, but the channels won´t bridge untill the macro is finished. I have no idea how to launch the macro in the background and let the call bridge.
01:46.40[TK]D-Fenderpguillem: No, the macro launches immedaite.  you use System() to launch a completely independant script with &
01:47.41pguillem[TK]D-Fender: Totally got the idea. I'll give it a try :)
01:47.46[TK]D-Fenderp3nguin: sig heil mein herr!
01:48.06pguillem[TK]D-Fender: Thanks for the light ;)
01:48.33[TK]D-Fenderpguillem: You're welcome...
01:50.07pguillem[TK]D-Fender: Would you rather use ChanSpy() to play on whisper mode ? or Background() on each channel?
01:54.08pguillemI think they coded the B option in ChanSpy() to
01:54.10[TK]D-Fenderpguillem: As you are starting a new channel you need to hook into the other ones.  This will have Chanspy on one end, and Playback on the other.
01:54.42kessiusbut when I try to compile dahdi-tools - Centos appears erro kernel - then  send command  -   yum -y install kernel-devel   _because keep getting erro
01:57.19kessiusbut when I try to compile dahdi-tools - Centos appears erro kernel - then  send command  -   yum -y install kernel-devel   _because i getting erro
01:57.45pguillem[TK]D-Fender: Ok. So from the async script i originate() a Local channel hooked to the bridge, and exec a context where ChanSpy is called with the Barge option and then Playfile?
01:58.05[TK]D-FenderPlayback.  Yes
01:58.49pguillem[TK]D-Fender: I owe you a beer!
01:59.22[TK]D-Fenderpguillem: You're welcome.
02:00.16[TK]D-Fenderpguillem: it is kinda hackish, but releives you from having to mod C directly and complicate upgrades, etc
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02:05.42jdoeanyone here logging rtcp data to for call quality stats? (and if so, what are you logging?)
02:15.44kessiusRTCP is a good question - how to it works
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03:11.03ChannelZSo how does one work around a remote peer whose behind NAT and giving out its LAN IP, when that IP happens to match the same LAN network of the Asterisk box it's connecting to which is also behind a firewall?
03:11.40ChannelZAsterisk sees the 192.168.x.x IP and says 'Hey that's MY local net' and happily sends RTP to it.
03:12.19[TK]D-Fenderyou set the peer to nat=yes and * will send to the IP it is sending from, not what it claims
03:12.36ChannelZBut it doesn't.
03:13.09ChannelZUnless it's an oddity with it being an anonymous call as opposed a specifically defined peer.
03:13.25ChannelZnat=yes is in [general]
03:14.07[TK]D-FenderAnd details are only slowly emerging... without actual configs and debug.
03:18.04ChannelZhttp://pastebin.com/3Xg2FYK8
03:18.26BuenGenioJust found the best hold music
03:19.09ChannelZSent RTP packet to      192.168.1.5:8222 (type 08, seq 062547, ts 004800, len 000160)
03:19.17ChannelZetc.  Forgot that.
03:23.51ChannelZBuenGenio: Morning zoo show on the mexican radio station?
03:23.59BuenGeniolol
03:24.06BuenGeniohttp://www.youtube.com/watch?v=83F9ek8J25M
03:24.59BuenGenioneed at least SPEEX for it though
03:25.02BuenGeniocheck out that bassline
03:25.16ChannelZAh.  Malfunctioning CD player.
03:25.21[TK]D-FenderChannelZ: Odd indeed.. because it responds to the SIP packet properly as NAT'd.  You are several versions behind.  This may have been fixed and I think I saw something similar a few weeks ago...
03:25.33ChannelZExactly.
03:25.55ChannelZI'll try the other direction tomorrow from work to my home which is running *10
03:40.21kessius<PROTECTED>
03:42.50[TK]D-Fenderkessius: First he wants guests.  Second his IP is set correctly for 1.8.  Allowguest=NOT.... "not" is not a valid value.  And finally he isn't matching a peer at all.... so this has nothing to do with them.
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03:52.56woleiumHas anyone successfully paid license fees for g729 without using Digium's (factor of 10 too expensive) option?
03:53.19[TK]D-Fenderwoleium: Where do you come up with "factor of 10"?
03:53.34woleiumit's a secret
03:53.37woleium:p
03:54.33woleiumI know someone that works for an OEM. it was their comment :-)
03:54.34[TK]D-Fenderwoleium: Oh.. you mean "delusionally derived".  Gotcha!
03:55.25woleiumBut to all intents and purposes it could be "fat bloke down the pub told me"
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03:56.50[TK]D-Fenderwoleium: Just found out Sipro is actually located right where I live... maybe I'll drop by one day ;)
03:57.35woleiumit kinda makes sense though. I'd not expect handset manufacturers to include a license if it adds $6 or so to the cost - but then they are probably just paying a flat fee for unlimited use.
03:57.50[TK]D-Fenderwoleium: http://www.sipro.com/g729_licterms.php
03:58.27woleiumthat looked good until i saw "Initial Fee: $15,000"
03:58.32[TK]D-Fenderwoleium: woleium Note the "initial fee" and "minimum prepaid" sections.  These are the "gotchas"
03:58.41[TK]D-FenderYup :p
03:59.17[TK]D-Fenderwoleium: So if you're rolling out a ton... I suppose it could be cheaper.  Till then "good luck".
03:59.28woleiumI guess digium's offering is the most logical then
04:01.32woleiummaybe if we all go stand outside Polycom's HQ with speex banners…
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04:07.02[TK]D-Fenderwoleium: G.711, G.729, G.722, and iLBC aren't enough?
04:08.16woleiumIt's not a bad list
04:10.58woleiumbut 711 is to big, 729 is too expensive. I guess 722 & iLBC are OK, but my current SIP provider doesn't talk either :-(
04:12.20[TK]D-FenderG.722 would be too big then
04:13.33woleium?? i thought polycoms did 722.1c @~32k
04:13.42kessius[TK]D-Fender you're correct - externip was used * 1.4 -
04:14.15[TK]D-Fenderwoleium: Last I heard it was 64kbps standard
04:16.47woleiumhttp://www.polycom.com/company/about_us/technology/siren14_g7221c/index.html
04:17.04woleium24, 32, and 48 kbps
04:18.11woleiumbut it's probably proprietary. Looks like it's based in their SIREN codec.
04:19.48[TK]D-Fenderwoleium: I'm referring to *'s support
04:19.52[TK]D-Fender^^
04:21.06woleiumindeed.
04:22.13woleiumwonders if there is a way to customise the "unable to connect" to say "out of 729 licenses" when appropriate...
04:22.40[TK]D-Fenderwoleium: I'd sooner have * fallback on SDP offering G.729 once you've run out....
04:22.56[TK]D-Fenderwoleium: Because it still lets you make promises you can't keep
04:23.40woleiumyou are probably right. Maybe something in the log then
04:24.09woleiumat the moment i don't think it logs anything when it runs out.
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05:25.55[TK]D-FenderAlrighty... bed time, back tomorrow....
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07:58.43schmidtsgood morning
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09:40.42faithlovehi. i'm currently developing a project based on asterisk and i'm wondering if there's any tool which can simulate calls so I can test the functionality of my tool.
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10:40.27dxd828Does anyone know if any of the Cisco video hardphones could work with Asterisk?
10:48.54vltHello. When connecting two asterisks via IAX is it possible to use two different voice codecs for each direction? I want the data sent in g711 and the data received in gsm? Possible?
10:52.59kaldemarvlt: no.
10:53.52bulkorokvlt: you have to put a third machine in the middle for transcoding
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10:58.05vltbulkorok: A third machine? What exactly would that machine "C" do that not "A" or "B" can do?
11:00.55kaldemarenable a session between A and B with A using gsm and B using G.711.
11:01.11kaldemarbut what's the point in using different codecs anyway?
11:01.40bulkoroka calls c with g711 ---- c calls b with gsm so you have a translator...
11:01.49vltkaldemar: limited bandwidth in only _one_ direction.
11:02.41vltbulkorok: Hmm, I don't understand why A can't call B directly using gsm.
11:04.06vltI want A to call B, send all voice data in gsm but receive B's voice in alaw/ulaw.
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11:05.10vltI'd understand "impossible", but fail to see how a third box in the middle could solve this.
11:06.11bulkorokI have to check, but a SIP-session can only use one codec for both UAs. So you can not say UA 1 uses codec A and UA 2 uses codec B in the same session...
11:07.30fakhir~itsp
11:07.30infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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.
11:07.44fakhir~itsplist-us
11:07.44infobotHere 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
11:09.59kaldemarvlt: it would not solve it. it would only enable the A and B to use different codecs, but not two codecs at the same time in a single session.
11:12.05bulkorokvlt: right... it's only one codec per session possible. The codec is handled in the SDP-part of INVITE and the response-pakets from the other UA. And there it's only one possible per session...
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11:15.14bulkorokso again: set up one session from a to c with g711 and then a second session with GSM from c to b ... two sessions but two different codecs...
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11:19.34vltkaldemar: Ok, thank you.
11:19.52vltbulkorok: Ok, only one codec per session.
11:20.09vltbulkorok: So machine C in the middle doesn't make sense at all ;-)
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11:27.21faithlovehi. i'm currently developing a project based on asterisk and i'm wondering if there's any tool which can simulate calls so I can test the functionality of my tool.
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11:29.50kaldemarfaithlove: what kind of calls?
11:30.02faithloveinbound / outbound
11:30.10faithloveanything which would help
11:30.53kaldemaras in technology
11:31.28kaldemaruse a soft phone, another asterisk box or sipp for example
11:38.39bowzaki'm having problems with some SPA525 phones running over a VPN client.  They will work for a while, but occasionally they fall into a loop where they cannot complete registration.  If I change the SIP port, they come right back up..  but eventually the problem recurrs.  Ideas?
11:38.40faithloveaha. well, i throught that there might be something like a simulator app. thanks anyway :)
11:47.29kaldemarfaithlove: sipp can be thought as a simulator app.
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11:53.32pjmHi, I wonder if someone might be able to give me a pointer re voicemail. I have a few SIP phones on internal LAN, all using ulaw, voice mail is fine, I can hear all the user recorded announcements etc. Dialling in via dahdi or SIP peer, the VM announcement starts to play but then stops after a few seconds, call remains but there is just silence, what can I check?
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12:50.08mahiti-irchi
12:50.53mahiti-ircis it possible to setup a single asterisk box with a STM-1 connection ?
12:51.09mahiti-ircif yes how many concurrent calls can this box handle??
12:53.24mahiti-irci was thinking of using STM-1 OC3 PCI card like this http://www.iphase.com/products/product.cfm/PCI%20Express/472
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13:04.59[TK]D-Fendermahiti-irc, That card does not mention any support of Asterisk
13:07.20[TK]D-Fendermahiti-irc, I also can't imagine the system surviving the EC load that many channels could place....
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13:08.23[TK]D-Fendermahiti-irc, And their "Applications" doesn't seem to list telephony too directly in any way.
13:08.29mahiti-irc[TK]D-Fender, EC ?
13:08.36[TK]D-FenderEcho Cancellation
13:08.42mahiti-ircok
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13:09.10[TK]D-Fendermahiti-irc, As it is there still aren't any DS3 cards out there that are compatible with * yet
13:09.14qakhanhi all
13:09.20mahiti-ircok
13:09.43mahiti-irc[TK]D-Fender, actually i am setting up a project where the client requires around 1200 calls simultaneously
13:09.55mahiti-ircwhat architecture would u suggest for it
13:09.57qakhani have a queue with 5 agents. i want to setup a alarm on queue after 3 mins if not agent answer the call
13:10.05mahiti-irci thought STM-1 handles it
13:10.12[TK]D-Fendermahiti-irc, * could do that if you use a gateway to do the conversion to a voip protocol
13:10.50mahiti-irc[TK]D-Fender, can u please elaborate it?
13:10.52[TK]D-Fendermahiti-irc, They mention no support of *, no drivers, and no application layer to handle EC, etc.  Looks dead to me.
13:11.32[TK]D-FendermahGet a gatway appliance that can convert to SIP.  They have these for DS1, and I think I saw a DS3 one once (never implemented by anyone I know)
13:13.00mahiti-irc[TK]D-Fender, You mean to say the STM-1 will be terminated in the gateway appliance and the asterisk speaks to this appliance
13:13.15[TK]D-Fendermahiti-irc, yes
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13:13.54mahiti-irc[TK]D-Fender, ok can you suggest me one such appliance, would help in the right direction
13:14.09[TK]D-Fendermahiti-irc, I imagine this would have to have a gigabit interface to handle the packet overhead, and would probably be VERY expensive... but conceivable
13:14.27[TK]D-Fendermahiti-irc, As I said... I'm not sure I've ever even seen one this big before...
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13:14.36[TK]D-Fendermahiti-irc, get googleing....
13:15.13mahiti-irc[TK]D-Fender, oh but the question is wht do i google for ? "Gateway appliance for STM-1"
13:15.22fpriorHi, http://pastebin.com/YKSHkztS : what cause this messages when debug is on ?
13:15.25eduzimrsanyboody knows this issue > WARNING[5328] chan_sip.c: Matched device setup to use SRTP, but request was not!      im trying to use MizuPhone with SRTP and this appears i cant make calls
13:15.31[TK]D-FenderSIP  STM-1 OC3
13:15.34[TK]D-Fender^
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13:16.08[TK]D-Fendereduzimrs, As it says, the device didn't request SRTP.... and the perr says it's supposed to.
13:16.12[TK]D-Fendereduzimrs, Fix your device
13:17.26eduzimrs[TK]D-Fender: im using SRTP option at the device, (Encrypt Media [SRTP])
13:17.39qakhani have a queue with 5 agents. i want to setup a alarm on queue after 3 mins if not agent answer the call
13:17.40[TK]D-Fendereduzimrs, * doesn't seem to see that in the incoming request....
13:18.32eduzimrs[TK]D-Fender: maybe a problem with MizuPhone?
13:18.48eduzimrs[TK]D-Fender: do u konow this softPhone?
13:19.00[TK]D-Fenderqakhan, * has no mechanism for this.  Before dumping the caller into the queue, launch some background external process to watch the caller's channel and act on a timeout.
13:19.07[TK]D-Fendereduzimrs, No, I do not.
13:19.35eduzimrs[TK]D-Fender: do u know other that uses TLS and SRTP ?
13:19.59[TK]D-Fendereduzimrs, Not offhand .... I try to never use softphones...
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13:21.19mahiti-irc[TK]D-Fender, ok i got a media gateway link which takes care of all what we wanted to do only on asterisk.
13:21.26mahiti-ircis that what you mention?
13:22.15qakhan[TK]D-Fender you mean some agi script?
13:33.14fpriorHi, what cause this boring messages  http://pastebin.com/YKSHkztS ?
13:34.51jkroonhi guys, is it a problem to call ast_timer_set_rate multiple times on the same (active) timer?
13:35.22jkroonin chan_iax2 there is an issue where trunkfreq from the config file does not take effect, and the simplest fix would be to just call ast_timer_set_rate on the existing timer.
13:35.47jkroonnot sure if this can break other things
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13:51.41mahiti-irc[TK]D-Fender, or anyone, can you tell me if this can be used with  asterisk box ?
13:51.42mahiti-irchttp://www.teleprime.com/whatis/ss7sip.html
13:54.51jkroonmahiti-irc, looks sane
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13:56.57mahiti-ircthanks jkroon , but can i know what u found in there?
13:57.14jkroonit talks SIP.
13:57.33jkroonso as long as you can get it to bridge properly between SS7 or C7 and SIP you should be fine.
13:57.55McBoingBoLooking at the SNTP section of my phone config, do I always need to manually adjust the daylight savings or is there some automagical way of setting this up?
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14:00.23jkroonshould be able to deduce it based on the timezone which you will need to set manually.
14:01.11[TK]D-Fenderqakhan, No, not an AGI.  I said BACKGROUND process.
14:01.48[TK]D-Fendermahiti-irc, That says it talks SIP.  that is what I'm suggesting already...
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14:02.45mahiti-ircok thanks a lot jkroon  and [TK]D-Fender :)
14:02.59mahiti-ircbtw
14:03.32mahiti-irchave u guys any idea of a USSD gateway that works with asterisk?
14:04.37jkroonmahiti-irc, i've seen the question before, I still don't understand what USSD has to do with voice.  it's a textual protocol from what I understand.
14:07.33mahiti-ircok, but cant asterisk be used to redirect a USSD transaction to a USSD gateway??
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14:18.26leifmadsenwhat is USSD?
14:18.48leifmadsenand since Asterisk is a B2BUA I suspect Asterisk would have to somewhat understand how to handle USSD
14:19.32leifmadsenoh ya, this doesnt' seem like anything Asterisk should really handle
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14:31.20Dovidwith libss7 is there any variable that will let me know if an incoming call is a national or international call?
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14:47.47mahiti-ircthanks for your support :)
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15:16.09sgtpepperhi there... quick question.. I'm trying to mask an early media message with a ringing tone since I might retry the call through another route
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15:16.29sgtpepperand I don't want the customer the hear a messge like "The subscriber you're trying to reach blah..."
15:16.39sgtpepperI've tried with R,r in the dial command
15:16.50sgtpepperswitching progressinband in the peeer in sip.conf
15:16.58sgtpepperstill, I cannot find a way to mask that message
15:17.09sgtpepperI'm using asterisk 1.8.3.2
15:17.21sgtpepper(one of the latest)
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15:23.08kaldemarsgtpepper: 1.8.3.2 is actually quite old for its branch. it was released a year ago. have you tried setting prematuremedia=yes in addition to progressinband=never?
15:24.07sgtpepperlet me try that combination specifically kaldemar
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15:27.13sgtpepperkaldemar: I'm trying those on the caller side... not on the trunk side... is that correct?
15:31.31kaldemarprematuremedia is a general option.
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15:39.23sgtpeppergrrr I hate early media
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15:43.45sgtpepperany other way you can think of to mask that early media message?
15:46.48Qwelltzafrir_laptop: I'm sad that nobody ever recommends using JPAH.  I bet it's the least expensive echo can there is.
15:47.16jacc0sgtpepper: let the callee pers a dtmf to confirm his/he pickup and then bridge maybe?
15:47.24jacc0*pres
15:48.07tzafrir_laptopQwell, it can also quarntee the same quality regardless of the jitter and such, which is also important
15:48.23Qwellexactly
15:48.30Qwellguaranteed results - that's the Qwell way
15:49.19Qwelltzafrir_laptop: I couldn't contain my laughter when committing that.  There was actually a useful reason for it, too.
15:49.29QwellI'm sure nobody has actually used it since, though
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15:55.02sgtpepperjacc0: mmm... thats not going to work
15:57.07jacc0ok, it was just a suggestion
15:57.29dxd828are there asterisk repo's for centos 6.2?
15:57.33Qwelldxd828: no
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15:58.44dxd828Qwell, are there for Debian?
15:58.54Qwellthere are a bunch for debian
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16:02.04p3nguinYay for stupid people!  Another set of faxes went out today from that place...with my number on it again.
16:02.16p3nguinFCC Form 1088A has been filed.
16:05.50mcrownoveri know this will most likely be a difficult question but for a novice building a new PBX today would anyone recommend going with Asterisk 10 over Asterisk 1.8.10.0
16:06.10p3nguinDepends on your use and expectations.
16:06.12Qwellmcrownover: Do you need features in 10?
16:06.31p3nguinI only deploy LTS branches in business production.
16:06.58mcrownoverthis would be for business use - about 150 extensions
16:07.17mcrownoverand 5 locations in the enterprise
16:07.18p3nguinDo you mean 150 phones?
16:07.33mcrownoveryes 150 phones
16:07.43p3nguinThat's a lot different from 150 extensions.
16:08.04mcrownover<----"novice"
16:08.29p3nguinLearn the difference early.
16:08.59mcrownoveri'm trying
16:09.10p3nguinTake this for comparison: on a system I have with six phones, I have 63 extensions.
16:11.12p3nguinIn Asterisk, extensions are the dialing rules which match numbers called.
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16:13.25mcrownoverwell we have about 150 desk phones with unique DIDs and about 75 or more that will route calls out via followme - plus 10 or so unique numbers inbound
16:13.56p3nguinOkay, so that's 235 extensions right there.
16:14.08p3nguinPlus you'll have more with subroutines and whatnot.
16:14.36p3nguin~book
16:14.37infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
16:15.30woleium~istplist-ca
16:15.54woleium~itsplist-ca
16:15.54infobotit has been said that itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca
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16:16.59mcrownoveri guess i am most confused on afer implementation on what to upgrade, what not to, how long the life expetancy is for the pbx
16:17.29p3nguinLife for the system as a whole, or life of the software?
16:18.09mcrownoverfor the software
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16:18.35p3nguinhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
16:19.14mcrownoverlooked at that earlier - so my current 1.6.2.20 is going to EOL in April
16:19.47p3nguin<p3nguin> I only deploy LTS branches in business production.      <---- important
16:20.15[TK]D-Fender^ For p3nguin
16:20.30mcrownoverso there is my easy answer to go with the 1.8.x vein
16:20.42[TK]D-Fendermcrownover, Not a bad choice
16:20.55[TK]D-FendermcrI'd recommend it unless there is something special in 10 you need
16:21.56mcrownoverof course all of this would be easier if I were lucky enough in my organization to only work on one item like the PBX - (i'm sure you guys know how it is to be responsible for EVERYTHING in the building)
16:22.57[TK]D-Fendermcrownover, Yes... I'm in the middle of moving 1/3 of my IT infrastructure around this week...
16:23.02bulkorokhi... where can I see all the status strings for FAXOPT(status) and FAXOPT(statusstr)!?
16:23.18p3nguinIf you are responsible for everything else, too, maybe you don't have time to learn Asterisk.  Is outsourcing an option?
16:26.04mcrownoveroutsourcing would be great if the cost weren't an issue - we used to host our pbx with a local source but it was upwards of $50k a year
16:26.54drfreezeIs there a way to specify a moh file for callers?
16:27.06p3nguinWhat kind of PSTN connectivity are you going to have on your in-house PBX?
16:27.23drfreezeI have callers dialing in - in context inbound
16:27.28p3nguindrfreeze: musiconhold.conf and extensions.conf
16:27.30mcrownoverour pbx is working, but i know it is not optimal - and i need to fix little issues like xml phone directories on the desk phones
16:27.45drfreezeThere are two offices - each with a DiD
16:28.02drfreezeso, some callers dial in 1000 and others 2000
16:28.06mcrownoverwe only have SIP from our provider
16:28.34drfreezeI would like to have the callers who dialed 1000 to hear a different moh that the callers who dialed 2000
16:28.53p3nguindrfreeze: Define two classes in musiconhold.conf.
16:28.53drfreezeI'm guessing, that I will need to create a context for each
16:29.08p3nguinNo, you can use a single context for two extensions.
16:29.13drfreezep3nguin: can you point to some examples?
16:29.27mcrownoverback in a bit - small issue down the hall
16:29.28p3nguinThe sample musiconhold.conf has examples.
16:31.07drfreezep3nguin: didn't see anything that would work for this scenario
16:31.33p3nguinAfter you have defined TWO classes, one for each moh you want, then you specify which moh to use in each extension (extensions.conf).
16:31.50p3nguinLet's call them class1 and class2.
16:32.42p3nguinIn one extension, you use Set(CHANNEL(musicclass)=class1)
16:32.50p3nguinIn the other, use Set(CHANNEL(musicclass)=class2)
16:33.01p3nguinThat's all there is to it.
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16:33.37p3nguinI have to take care of something for about 15 minutes.  I'll be back to help if you aren't done with it.
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16:35.24drfreezep3nguin: cool. I'll give that a try. Thanks
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16:40.09DanFromUKhi, if a user has an analog sip adapter, whats the best way for allowing the user to place a call on hold, and make a second code, and then allow the user to return to the original call?
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16:51.26*** join/#asterisk devmikey (~irc@unaffiliated/devmikey)
16:53.56p3nguindrfreeze: Back.  Did you get it?
16:56.09*** join/#asterisk cadmium (1000@adsl-99-121-236-244.dsl.sfldmi.sbcglobal.net)
16:57.07cadmiumHi, i'm using exten => _1NXXNXXXXXX,n,Set(FROM_DID=12142421782)  but its not setting that as the caller iD
16:57.18cadmiumany idea what statment I need to use to sed the CID properly?
16:58.00navaismoSet(CALLERID(num)=)
16:58.24cadmium;exten => _1NXXNXXXXXX,n,SetCallerID(12142421782)
16:58.30cadmiumthis causes an error
16:58.45Andee16:58 < navaismo> Set(CALLERID(num)=)
16:58.52AndeeSet(CALLERID(num)=12142421782)
17:00.00cadmiumk pbx.c:4232 pbx_extension_helper: No application 'SetCallerID' for extension (MITUS, 12149013232,
17:00.19navaismo¬¬'
17:00.59*** join/#asterisk macroevolve (~macroevol@c-98-234-125-202.hsd1.ca.comcast.net)
17:01.10Qwellcadmium: Don't use SetCallerID.  Set(CALLERID(num)=1234)
17:01.35cadmiumAsterisk 10.0.0-rc2, Copyright (C) 1999 - 2011 Digium, Inc. and others.
17:01.58macroevolvehi i was wondering. when customer calls a customer service number.  when the agent picks up and decides he needs to transfer the customer to a more experienced agent, my thoughts are there are two ways to do this.  Either the experienced agent is already waiting on some sort of queue line and can be hot transferred in to or the agent can initiate a separate call to the experienced agent's
17:01.58macroevolveextension and when connected can bridge the calls (experienced agent wasn't waiting on queue line).  If in the latter case, the experienced agent's extension had auto-answer turned on, is there a difference in connection speed between the two approaches, or would both approaches take < 1 s to connect the experienced agent into the call?
17:02.07p3nguinYou don't set things FROM DIDs, anyway.  DIDs are for inward dialing, hence "Direct Inward Dialing."
17:02.59cadmiumQwell thanks!
17:03.00cadmiumworking now!
17:03.32cadmiumand thanks also Andee, i had already tried that one
17:03.36cadmiumi guess i don't have that application?
17:04.06[TK]D-Fendercadmium, that app was deprecated something like FIVE YEARS ago.
17:04.18[TK]D-Fenderin * 1.2
17:04.36[TK]D-Fender6 branches ago
17:05.09cadmiumahh
17:06.21cadmiumcan you send text messages with asterisk ?
17:06.53[TK]D-FenderNo
17:07.57cadmiumany recomendations for wholesale sms carrier gw ?
17:08.38*** join/#asterisk flan (~flan@23.17.142.228)
17:09.36flanDoes anyone know how, inside of Bridge(), to suppress DTMF pass-through, so that the other party doesn't get keypress data?
17:10.06*** join/#asterisk its_jeremy_ (~omghax@gateway/tor-sasl/itsjeremy/x-75806909)
17:12.58*** join/#asterisk vinhdizzo (~vinh@dhcp-v006-207.mobile.uci.edu)
17:13.22*** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr)
17:15.10cadmiumflan maybee a sip peer setting?
17:16.44flancadmium, any idea what it might look like? I haven't noticed anything helpful-looking in sip.conf's documentation, but I have noticed that combinations of settings make things work in unexpected ways.
17:17.19cadmiumdtmfmode=inband.
17:18.09mcrownoverIf $$ wasn't an issue would any of you choose a Cisco pbx over an asterisk one??
17:18.12flanI've currently got it at 2833 (and have both parties configured the same way). I want to take the event out of the stream, not keep it in there.
17:18.52flanThat would depend on what you need it to do, mcrownover. Asterisk is probably a lot more flexible, if more involved to configure.
17:19.32cadmiumi don't see an option to disable it completely for a peer
17:19.45p3nguinmcrownover: No.
17:20.22cadmiummcrownover never
17:20.25*** join/#asterisk rossand (~aross@foundation-yow.eclipse.org)
17:20.45drfreezep3nguin: yes, got it working
17:20.45drfreezethanks
17:20.49cadmiumi thought call managers were for suckers :P
17:21.45p3nguinmcrownover: For agents waiting in queue, the agent calls the login number and logs in.  He/she then sits in the queue listening to either silence or moh.  Calls entering the queue are then connected to the agent who is waiting.  The time to connect can be adjusted with dial plan.
17:23.42flancadmium, thanks for confirming that. I'll probably have to weigh the cost of hacking RTP (or doing aggressive firewall-filtering) versus the possible annoyance factor.
17:23.51p3nguinThe time can also be adjusted with the member delay value in the queue.
17:24.53cadmiumflan I would think you could disable it ... or chose which channel was the receiver in a bridge call
17:24.59cadmiumsomtiems i lose dtmf when i conference
17:25.07cadmiumand dtmf only responds to the person i conference
17:25.12cadmiumand they have to do all the number pressing
17:25.52flanI know ConfBridge hides it. Looking at the Asterisk debug log, AMI events, tcpdump, and some debug lines I added to the code, it's doing an explicit retransmission of the event.
17:26.31flanBut there doesn't seem to be a simple flag or channel variable that can be set (documented, at least) to turn off that behaviour.
17:26.48cadmiumright.. you can always hack the source :P
17:27.15cadmiumwish i knew
17:27.20cadmiumsure some guys could tell you more
17:27.29flanYeah, which is what I'm considering doing. I just don't want to redo someone else's work and possibly provide two implementations for the exact same thing if I publish the patch, though.
17:27.57cadmiumoh .. if i hacked the source i wouldn't publish the patch i'd just be happy it was working :P
17:28.49flanSure, until someone comes along after you and tries to maintain your fix. =P
17:29.04flanHence weighing the costs.
17:29.45*** join/#asterisk Reenigne_Esrever (~reenignee@mail.trmcopycenters.com)
17:30.54*** join/#asterisk rossand (~aross@foundation-yow.eclipse.org)
17:31.13Reenigne_EsreverHello, I'm thinking of registering for the Digium class coming up in April in Vegas but was hoping to get some feedback from someone who has attended the class whether or not they think it is worth it.  I'm paying for it out of my personal account to further my knowledge and opportunities with asterisk
17:41.07*** join/#asterisk mintos (~mvaliyav@114.143.162.200)
17:42.19*** join/#asterisk blizzow (~jburns@67.50.165.58)
17:46.20[TK]D-FenderReenigne_Esrever, Depends what topics they are going to cover that yuo care about, how much you already know, how long it would take you to acquire any knowledge you'd get from there by other means & times...
17:46.39[TK]D-FenderReenigne_Esrever, This is a key reason I've never left town for one fo these before...
17:46.58[TK]D-FenderReenigne_Esrever, Your needs may be different  Judge accordingly
17:49.47*** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net)
17:51.50joesuffcerenI have two fractional T1s (pri_cpe,esf,b8zs) connected to a single box (two digium single span T1 cards). When I do dahdi show status, but cards show up as ok (no alarm). When I do PRI show span 1, it is up, provisioned, and active. PRI show span 2 show down, provisioned, active. Where should I start troubleshooting why span 2 is down? Span 2 is new and has not been functional previously....
17:51.52joesuffceren...Span 1 is existing and is still functioning properly
17:53.10Reenigne_Esrever[TK]D-Fender, I was going to sign up for both classes they offer if I'm going to take the time to fly out of town, I have minimal knowledge on asterisk, I didn't setup our last system only maintained the configuration etc... after it was installed
17:53.49Reenigne_EsreverI have since then installed asterisk from source on my home pc and setup a simple dial plan, I want to take it to the next level and build a solution that would be reliable in an enterprise environment
17:54.27*** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com)
17:54.31Reenigne_Esreverthe $5k price is a little high and that is why I was hoping to hear from someone who has taken any class from them if they thought it was worth it
17:54.52jayteeThe Advanced class is worth it just for the swag. The backpack laptop bag is awesome. I also got a Polycom 331, a TDM410 card and a TE110 card along with an orange * Geek t-shirt.
17:55.52Reenigne_Esreverjaytee, did you have any asterisk knowledge before taking the advanced class?
17:57.51*** join/#asterisk e7e5 (~rudenko@188.134.2.33)
18:01.51mcrownoveri am considering going the same classes in May in Huntsville
18:10.25[TK]D-FenderReenigne_Esrever, Well you're in "general" territory... unless they really target your needs across the board I'd save up on it and just spend the time getting your hands dirty and drilling specific bits towards your goal.
18:14.38*** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
18:15.43bmoraca_workAsterisk 10 should be able to detect faxes over a sip channel by default, correct?
18:16.31p3nguinI wouldn't think so -- you have to enable it with the faxdetect setting.
18:17.22bmoraca_workwell i understand that...but i mean if i have that enabled, there shouldn't be anything else I have to do (any other modules or anything)
18:18.35p3nguinI don't know of anything else.  Just enable it and create the necessary fax extension within the context where the call lands.
18:19.47mjordanbmoraca_work: there are other modules, but it depends on your configuration.  Are you using Fax For Asterisk?  Or res_fax_spandsp?
18:20.06p3nguinFor fax detection, no other modules are needed.
18:20.07bmoraca_workno, i just want to use app_fax and sip fax detection to redirect
18:20.13bmoraca_worki didn't believe so
18:20.23p3nguinIt's built into chan_sip.
18:20.23bmoraca_worki think i know what the problem is, though
18:20.59mjordanp3nguin: true, but it isn't going to do a whole lot without something to receive the fax, act as a T.38 gateway, etc.
18:21.34bmoraca_workapp_fax receives faxes well enough
18:21.39*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
18:21.41bmoraca_workand i don't need t.38
18:24.01*** join/#asterisk eject_ck (~eject@h82.241.159.dialup.iptcom.net)
18:24.07eject_ckHi akk
18:24.09eject_ckall
18:25.06eject_ckhow can I get SIP peer external IP address from cli ? When I do sip show peer 100, then I see Reg. Contact with internal IP address (peer is behind NAT)
18:26.56p3nguin"sip show peers" will show the peers' public addresses if the peers are behind a NAT somewhere else, or the local address if they are on the same LAN as Asterisk.
18:27.43p3nguinAnd sip show peer 100 would show Addr->IP:  followed by the public address.
18:28.30p3nguinThe Reg Contact field is the only one containing the private address.
18:28.56jdoedoes automon/automixmon record the call from its start and delete unless the feature code is hit, or does it start recording after?
18:29.42p3nguinIt begins recording either when the app is run or when the call goes answered, whichever is later.
18:30.38jdoehmm.
18:31.48p3nguinIf you start mixmonitor on a channel which is not up, it does not start recording until it goes "Up."
18:32.08*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
18:32.18p3nguinIf the channel is already answered when you start mixmonitor, it starts recording immediately.
18:32.56jdoep3nguin: this is the feature-code triggered version though, not MixMonitor() itself.
18:33.24p3nguinIs it not the exact same application?
18:33.29jdoeand what I was asking was more if I punch *1 in a call, does it record from that point onward, or does it record *every* call and only save flagged ones or similar
18:33.42p3nguinIt records the current call.
18:33.52p3nguinfrom the point you press the keys to start.
18:33.57jdoeright, thanks.
18:34.08p3nguinWhen the call ends, the recording ends.
18:34.18jdoeyeah.
18:34.37p3nguinIf you want to record every call for its entirety, use MixMonitor() in dial plan.
18:35.14jdoeyeah, I think that's what I'll be doing... basically I want to be able to record calls, but I may not care until something of importance has already been said.
18:35.34jdoeso I may need to use mixmonitor in a dial plan, and have it auto-deleted, unless someone hits a feature code or something.
18:37.06p3nguinThat could work.  Make a followup after the call where the agent has to press one key to save the recording or another to delete it.
18:37.28jdoeyeah that could work too.
18:40.36p3nguinOr have a batch job run every night to delete recordings.  Anything that needs to be saved could have already been copied or moved to a safe place.
18:40.50*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
18:44.11leifmadsenI had a cronjob nightly that moved files to an ftp server remotely then deleted the files from the HD
18:51.36*** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
19:04.20*** part/#asterisk cadmium (1000@adsl-99-121-236-244.dsl.sfldmi.sbcglobal.net)
19:04.31*** join/#asterisk pa (~pa@unaffiliated/pa)
19:07.32eject_ckp3nguin: thank you!
19:07.51eject_ckI'm getting messages in console: [2012-03-13 21:58:38] NOTICE[1786]: chan_sip.c:9581 copy_header: No field 'CSeq' present to copy
19:08.28*** join/#asterisk bowzak (~bowzak@95.170.203.162)
19:10.54drfreezeAnyone have a config setup to connect a Gigaset S675 IP to asterisk?
19:11.18drfreezeI keep getting 'registratation failed' or 'server not accessible'
19:12.17*** join/#asterisk corretico (~luis@190.211.94.6)
19:12.43eject_ckdrfreeze: what about debugging problem ?
19:13.28eject_ckcheck sip traffic with wireshark to see registration info
19:15.05*** join/#asterisk tdillon (~travis@208.85.166.100)
19:15.53tdillonI have asterisk version 1.4.22 and i am having dtmf out of order issuse
19:15.57tdillonany ideas?
19:19.55bowzakany good softphones that support launching an external URL when calls come in?
19:19.55[TK]D-FenderUpgrade.  You over 20 releases behind on a branch that is 5 branches old.
19:22.27navaismobowzak, zoiper
19:23.29tdillonIf I upgrade will that fix the dtmf issue?
19:25.00*** join/#asterisk pigpen (~mark@fw.seamans.cc)
19:28.56*** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net)
19:31.37Guggetdillon: if it doesnt, you will at least have a better chance getting help :)
19:32.07tdillonAlright. That is what I was afraid of.
19:34.49*** join/#asterisk pa (~pa@unaffiliated/pa)
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19:44.56*** join/#asterisk salviadud (~blasko@187.162.141.20)
19:46.39bowzakthanks navaismo.  Voiper is popping up the call log window in my smartertrack.  there are query values associated with the ap...  what's the stringname for incoming DID ?
19:48.02drfreezeeject_ck: I've got some debug data
19:48.26drfreezedo you have recommendations for tcpdump command to best collect SIP registration?
19:49.11*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
19:49.28salviadudquick question
19:49.47salviadudthose digium phones, do they support IAX2?
19:50.32salviadudcan't believe the spec sheet only mentions SIP
19:51.49pabelangersalviadud: no
19:51.55pabelangerthey are SIP phones
19:52.02salviadudthat's odd
19:52.27eject_ckdrfreeze: sure
19:52.54eject_cktcpdump -ni eth0 -s 65500 port 5060
19:53.23eject_ckwhere eth0 is your corresponding NIC
19:53.49eject_cktcpdump -w dump.pcap -ni eth0 -s 65500 port 5060
19:55.00eject_ckdrfreeze: check this post ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/
19:55.59[TK]D-Fendersalviadud, Inter Asterisk eXchange.  Not Phone To Asterisk.
19:56.13drfreezeeject_ck: cool
19:56.26eject_ckand this http://wiki.wireshark.org/SIP
19:56.43[TK]D-Fendersalviadud, IAX was not built with phones in mind and Digium wants to make sure not to scare off customers on a statistically proprietary protocol that wasn't built with phones in mind
19:57.05eject_ck[TK]D-Fender: AFAIK somme phones support IAX
19:57.22[TK]D-Fender"some".  99% or which are cheap crap
19:57.29[TK]D-Fenderof*
19:57.38[TK]D-FenderThe rest = meh
19:57.43[TK]D-Fendernothing worthwhile
19:57.53*** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com)
19:58.03eject_ckto be honest  - I've never used it
19:58.09eject_ckhttp://www.voip-info.org/wiki/view/Asterisk+IAX+clients
19:58.22*** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com)
19:58.25eject_ckSNOM should be good, no ?
19:58.28salviadudyeah, I guess you got a point
19:58.45salviadudif I got 2 offices, and want to handle them with the same extensions
19:58.59salviadudit be easier to just trunk them through IAX and locally set a bunch of sip phones
19:59.32eject_cksalviadud: you can do that with sip as well :)
20:00.00eject_ckjusify your dialplan :)
20:01.11salviadudwell, I think its a lot more clean to trunk with iax
20:01.18salviadudless ports to worry about
20:04.35*** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl)
20:19.09*** join/#asterisk wonderworld (~ww@dsdf-4db54bf5.pool.mediaWays.net)
20:19.58*** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-dajbovqorrkhydqt)
20:20.33rrittgarnAfternoon. Having an issue with localzed park. I have 3 separate parkinglots defined (the default, and two separate named parkinglots). I am able to dial the parking number, and then see my BLFs /  pick up the call. If i do a supervised transfer to park, it works, however if I blind transfer to it, I cant pick it back up. Any ideas?
20:22.50jdoe... I can make channel variables inherited by setting adding underscores... is there a way to get reverse inheritance?
20:22.55*** part/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr)
20:23.16jdoeie have a channel variable set in a ... sub-channel propagate up to parents?
20:26.19jdoeah, I can use SHARED.
20:32.43*** part/#asterisk eject_ck (~eject@h82.241.159.dialup.iptcom.net)
20:34.21bowzakuff.. stupid trixbox.  finally worked for like 4 days after a hard reboot, but now i'm getting retransmits from hell again
20:43.49*** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
20:45.52*** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net)
20:47.42*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
20:52.28*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
20:54.42*** join/#asterisk timahvo1 (~rogue@41.80.66.51)
21:01.18*** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com)
21:25.56*** join/#asterisk dijib (~root@bas10-kitchener06-1176001626.dsl.bell.ca)
21:26.09dijibanybody familier with shorewall?
21:26.24*** part/#asterisk libryder (~libryder@libryder.com)
21:27.56dymdijib: to an extent - shoot your question and cut the metaquestions.
21:29.07dijibsimpe question. is the network zone fw behind firewall or net facing?
21:30.04dijiblike should i be using my 'net' zone for my internal or external NIC?
21:30.37*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
21:30.55dijibeverytime i start shorewall i get locked out. i believe i have the firewall rules correct and a other basic configurations
21:31.07dijib2nic system, one inside one out
21:39.52dymdijib: thats up to you. the zone "fw" is the machine the firewall runs on itself.
21:40.03dymwhatever you name the other zones is completely up to you
21:45.01mcrownoverif i were needing to deploy phones at a remote office (about 30 or more lines) would it be better to set up a second asterisk pbx in some sore of cluster or just let them register on the one via the site to site VPN?
21:45.59dymmcrownover: both could be done really. depends on your call capacity
21:46.39mcrownoverdym: i am using g.729 and have 40 sip trunks at the main office
21:46.56mcrownoveron a 12mb pipe
21:47.05mcrownoverthe second office has a 6mb pipe
21:47.45dymthen calculate the capacity
21:49.19mcrownoverhttp://www.asteriskguru.com/tools/bandwidth_calculator.php - according to this calc i need 4.6mb to support 100 alls in and out
21:49.31mcrownoverusing SIP and g.729
21:49.55mcrownoversorry 100 calls
21:51.47*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
21:52.49jdoeanyone here logging rtcp data for call quality stats? (and if so, what are you logging?)
21:54.11navaismojdoe, what i do is run tcpdump for a while, then open the cap file with wireshark, select voipcalls and check the completed calls with the player using the rtp time stamp
21:56.47jdoenavaismo: I'm looking to do something a bit more automated :P
21:57.14*** join/#asterisk serafie (~erin@nat/digium/x-zxxxxffawgjnprom)
22:04.26*** join/#asterisk gonewage (~gonewage@72.2.130.205)
22:07.41rrittgarnanyone have any experience on how asterisk would handle a Blind Transfer differently than a transfer? Specifically on an Aastra?
22:09.13navaismonope but you can enable the sip debug to see it
22:14.00dymdijib: and?
22:16.45dijiband.
22:16.49dijibuhm sorry i was afk
22:16.50dymokay then
22:26.22dijibi dont know whats up. anytime i enable shorewall im locked out
22:28.03dymdijib: locked out ssh wise?
22:28.16dymyou must have an error in your config.
22:28.32dijibeverything wise and 22 is defined in my firewall ruiles
22:28.57dymShorewall Rule #1: Never disconnect a host after a major rulechange, without trying to connect to it on a seperate shell and testing that still works.
22:29.01*** join/#asterisk wonderworld (~ww@dsdf-4db557b8.pool.mediaWays.net)
22:29.13dymdijib: There must be an error.
22:29.38dymDid you set shorewall to log and then try ssh connect and see what it drops?
22:31.07dijibno
22:31.12dijib/var/log/shorewall?
22:31.14dymNo
22:31.22dymYou need to enable logging in the policy file
22:31.42dymdo you have four columns on there?
22:31.45dymor just three?
22:33.06dijibno clue i cant see that file right now as im on the box itself
22:33.18dijibon an external ssh screen instance
22:33.25dymyou are on the box?
22:33.27dymconnected to it?
22:33.28dijibyes
22:33.33dymwell - how can you not see the file?
22:33.42dijibbut since im playing with firewall screen irssi on another one
22:34.06dymExplain the situation.
22:34.10dymAre you locked out?
22:34.16dymAnd only connected via screen?
22:34.23dym(a session that runs screen)
22:35.00dymLets not get on these guys nerves -> /q me
22:36.07*** join/#asterisk mdiehl (~mdiehl@173-10-242-193-Albuquerque.hfc.comcastbusiness.net)
22:36.21dijiblocked out
22:36.37dijibive got iptables filters set to all allow
22:36.43dymAre you even reading me?
22:36.45dymQuery...
22:37.18*** part/#asterisk mdiehl (~mdiehl@173-10-242-193-Albuquerque.hfc.comcastbusiness.net)
22:37.23*** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net)
22:37.40dijibi'll be back in a bit
22:37.51dymErr
22:37.58dymOkay, whatever.
22:42.13*** join/#asterisk mdiehl (~mdiehl@173-10-242-193-Albuquerque.hfc.comcastbusiness.net)
22:43.13mdiehlHi all.  I'm looking to sign up to contribute some code to Asterisk and I have a few questions.
22:43.38*** part/#asterisk mjordan (~mjordan@nat/digium/x-ncdgfxunqmlqzqlw)
22:44.23jgowdyWhat's the best I/O scheduler for use with Asterisk?  I'm running 2.6.37
22:45.16jdoemdiehl: may have better luck in #asterisk-dev
22:45.33mdiehlThanks.
22:47.35dym:O jdoe Are you THE J. Doe? (:
22:47.43dympokes ChrisInSydney
22:47.45dymevening lad
22:51.23jdoedym: depends on who's asking.
22:51.49dijib<PROTECTED>
22:52.04dijiblet me change computers hgere
22:52.32dijibk bak
22:52.39dymjdoe: i mean "John Doe" :D
22:52.47dymThe one on all the dummy credit cards :P
22:52.51dijibmay i show you the confs from my simple toplogy?
22:53.03*** part/#asterisk mdiehl (~mdiehl@173-10-242-193-Albuquerque.hfc.comcastbusiness.net)
22:53.11dymdijib: If you cared to read what I wrote, you'd know I asked you to query me an hour ago
22:53.19dijibdoes iptabes have an impact on shorewalls operation? its a wrapper so iptables dependent?
22:53.52dijibim thinking this could be fixed by adjusting my simple confs
22:53.56dijibits a new setup
22:53.59dymAre you kidding me?
22:54.24dijibyou think logging is simpler?
22:54.57dymFor the last time: This is an asterisk channel - if you'd like some pointers on shorewall QUERY me now. If you dont. Your loss.
22:56.51dymOr dont you know what a query is?
23:00.56ChrisInSydneyhey dym
23:01.51dymHey there
23:02.08ChrisInSydneygot a whole building out. Ni lines, no providers !
23:02.28ChrisInSydneypanic stations
23:02.40dymWell now that sounds like fun...
23:03.35dymDont you have PSTN fallback?
23:07.13ChrisInSydneyno PSTN, no ISDN, no dialtone, no nothin
23:07.20ChrisInSydneysomeone has cut a line
23:07.42dymi can just picture the employees
23:08.09Russhas civilization reverted to the stone age?
23:08.29ChrisInSydneyThere is the Bat and Ball around the corner. They'll all be there
23:08.37ChrisInSydneypub
23:08.45ChrisInSydneyits after 10am
23:09.10dym:D
23:17.24ChrisInSydneyI wa about to rush out on site, then I said, just check next door and see, they did a door knock and no no one has a dial tone
23:17.39ChrisInSydneyLooks like i have just sold a big GSM gateway :-)
23:19.57*** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net)
23:31.38dymChrisInSydney: So whats the issue? SIP provider having PSTN connectivity problems?
23:32.19ChrisInSydneyI recon some one has put a saw through a cable
23:32.29ChrisInSydneyor a digger
23:32.51ChrisInSydneyhttp://1100.com.au/default.aspx
23:32.56*** part/#asterisk bowzak (~bowzak@95.170.203.162)
23:33.00dymSounds like fun
23:33.23ChrisInSydneynot my fun anymore. handed it off. Now I can sell a 16 or 32 channel GSM gateway
23:33.50ChrisInSydneyone of the clients is a small 25 seat call centre
23:34.47dymprofit (:
23:34.56dymbuilders failure => admins win :D
23:38.04ChrisInSydneyI really cant charge for the last hour I have spent chasing this up.
23:38.11ChrisInSydneycomes as part of the maintenance
23:38.26dymBut the GSM Gate?
23:39.05ChrisInSydneyyep. I need to get a deposit first. Lets see if they will pay
23:48.16*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)

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