| 00:01.25 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-6a17bdbe261bc7a0) |
| 00:19.53 | *** join/#asterisk jeff_phillips (n=fircuser@m460436d0.tmodns.net) |
| 00:23.18 | jeff_phillips | whats the best linux distribution for running asterisk? |
| 00:23.30 | Qwell | jeff_phillips: AsteriskNOW, naturally! |
| 00:23.37 | Qwell | I'm slightly biased. |
| 00:24.09 | jeff_phillips | why are you slightly biased? |
| 00:25.25 | Qwell | well, I didn't write it or anything |
| 00:25.37 | Qwell | wait, no, the other one. I did write it. |
| 00:26.20 | jeff_phillips | other one? |
| 00:26.44 | Qwell | just smile and nod. ignore my banter. |
| 00:28.49 | jeff_phillips | honestly i dont know as much as i'd like, but i want to learn... |
| 00:30.16 | jeff_phillips | i installed trixbox at work before figuring out that the #freepbx guys poke fun at it. |
| 00:31.15 | jeff_phillips | i like the freepbx gui just as a little bit of a comfort zone where i can feel like i know how to use some major components of the overall functionality pretty well, |
| 00:31.34 | jeff_phillips | but i want to learn the nuts and bolts of asterisk behind the scenes |
| 00:32.05 | jeff_phillips | i can generally get most of what i want to do done on my own if i waste enough time googling info on how to do it |
| 00:32.59 | jeff_phillips | i have a few applications i want to run on a dedicated server im renting at a hosting center... of course nicer hardware equals higher rent $... |
| 00:33.34 | jeff_phillips | so im wondering if i should install a more stripped down set of just the components and tools i will likely use |
| 00:34.57 | jeff_phillips | or if i should just load on something thats easy to use even if it limits my capacity on the hardware |
| 00:42.44 | jplank | should asterisk wait till the second ring to pick up a call for caller ID? |
| 00:44.00 | Qwell | jeff_phillips: like I said - AsteriskNOW :p |
| 00:44.00 | jplank | when I call in over a TDM line, adn route the call to my phone, my phone starts ringing before the caller ID is transmitted, and if I pick up quick enough, I can actually hear the caller ID tone |
| 00:44.00 | Doc | jplank: for PSTN or SIP? |
| 00:44.00 | Qwell | jplank: no, but you can add a Wait(2) or whatever |
| 00:44.00 | jplank | PSTN |
| 00:44.00 | jplank | it seems the PSTN is sending the caller ID too late |
| 00:44.00 | Doc | caller-id is sent between the first and second gings |
| 00:44.02 | Doc | so, yes |
| 00:45.42 | jplank | the wait should be before the answer, or could it be after? |
| 00:45.50 | Qwell | before |
| 00:46.25 | jeff_phillips | Qwell: thanks ill download it and play around a bit tonight |
| 00:47.08 | Qwell | jeff_phillips: have you already been to asterisknow.org/downloads/? |
| 00:47.42 | Qwell | nevermind, I answered my own question |
| 00:48.15 | MikeJ_ | Qwell: how could we ignore your banter? |
| 00:48.23 | Qwell | MikeJ_: I don't know. |
| 00:48.31 | Qwell | You'd be silly to, IMO. |
| 00:49.00 | MikeJ_ | heh |
| 00:49.11 | MikeJ_ | of course IYO |
| 00:52.12 | *** join/#asterisk xa0z (n=interex@cpe-68-20-152-147.netwitz.net) |
| 00:54.04 | xa0z | Anyone know any inexpensive voip providers who offer unlimited in/out (nationwide) ? Even better if I can get one with a local number where I am. |
| 00:57.21 | *** join/#asterisk CrazyTux (n=brandon@ip72-211-214-95.oc.oc.cox.net) |
| 00:57.39 | jplank | My wait is AFTER asterisk detects the call, is that ok? |
| 00:57.42 | jplank | [Apr 3 20:57:05] -- Starting simple switch on 'Zap/13-1' |
| 00:57.42 | jplank | [Apr 3 20:57:07] -- Executing [s@from-zaptel:1] Wait("Zap/13-1", "5") in new stack |
| 00:57.44 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:57.51 | *** part/#asterisk harryv (n=harry@vangberg.name) |
| 00:58.41 | jeff_phillips | xa0z: most of the "unlimited" services are locked to a specific device or restricted through the terms of service to not be used with a pbx system, simply because you'll exceed the average usage of a typical end user on a trunk |
| 00:58.55 | xa0z | Oh. |
| 00:59.10 | jeff_phillips | thats not to say you cant get some deals though |
| 00:59.38 | jeff_phillips | Didforsale.com offers incomming calls on 20 channels for $8.99 |
| 01:00.14 | jeff_phillips | call centric i think has an unlimited outbound for $20/month locked to a static ip address |
| 01:00.24 | xa0z | Best bet is to just stick with 8x8 then :/ |
| 01:01.12 | jeff_phillips | i have a couple of t-mobile@home lines for $9.99 unlimited both ways, but it only works with the linksys box they sent me |
| 01:02.00 | jeff_phillips | they dont mind it being used for business calls but i had to hook it up to my fxo card to get it into asterisk via an analog interface |
| 01:04.08 | jeff_phillips | some people have hacked the sip credentials from a magic jack and use it with asterisk, but that is a blatant violation of their terms of service |
| 01:05.43 | xa0z | I need with control over outgoing name only though...I am setting up 3 phones and I want each one to have its own CID(name) for the office it's in, but all 3 to send the same number but phone1 would display "EXT1" and phone2 would display "EXT2" for example... and incoming would ask which extension you want. |
| 01:05.51 | xa0z | It's hard for me to explain. |
| 01:07.59 | *** join/#asterisk brunner (n=chris@75-143-85-118.dhcp.aubn.al.charter.com) |
| 01:17.28 | jeff_phillips | can you do that?, |
| 01:18.01 | jeff_phillips | i always thought that the caller id name was from querying a telco database against the number |
| 01:18.24 | jeff_phillips | if you have the same number i would imagine you would get the same name on all of them |
| 01:20.37 | *** join/#asterisk werdan7 (i=werdan7@freenode/staff/wikimedia.werdan7) |
| 01:37.15 | *** join/#asterisk lordvadr (n=irc@ppp-70-225-171-46.dsl.chmpil.ameritech.net) |
| 01:38.55 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 01:39.32 | KavanS | y0 |
| 01:39.47 | KavanS | I have a zaptel card, and I use the "c" dial option to press pound to connect for a "followme" macro |
| 01:40.08 | KavanS | I'm trying to do the same with SIP, but I notice the "c" dial option does not work....any suggestions for the same "press # to connect" scenario? |
| 01:47.23 | nkohh | jeff_phillips: it is, and he doesn't know what he's talking about |
| 01:47.45 | *** part/#asterisk lordvadr (n=irc@ppp-70-225-171-46.dsl.chmpil.ameritech.net) |
| 01:49.13 | MikeJ_ | unless your the telco itself.. AND other providers are not caching ... and ... and ... |
| 01:50.04 | nkohh | and the stars have to be aligned right ;) |
| 01:50.23 | MikeJ_ | that being said.. there is some support for end to end caller id name in NI2, but I have never seen it supported on any network |
| 01:50.54 | MikeJ_ | but in a completely private isdn network it is technically possible :D |
| 01:51.40 | nkohh | in a completely private isdn network, anything is possible. |
| 01:51.54 | MikeJ_ | hehe :D |
| 01:52.14 | MikeJ_ | of course thats less useful if you actually want to talk to the outside world |
| 01:52.47 | nkohh | somewhat |
| 01:55.03 | KavanS | y0 anyone got any ideas? |
| 01:55.13 | KavanS | read function can't just take a "#" |
| 01:55.26 | KavanS | trying to setup a "press pound to connect" for custom follow me |
| 02:01.30 | Kobaz | http://pastebin.com/m3222719d i'm having problems with originate via the ami with asterisk 1.6... the call kinda goes through, the remote phone rings for about 10ms... looks like a bug |
| 02:04.08 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:04.48 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 02:07.21 | *** join/#asterisk lordvadr (n=irc@ppp-70-225-171-46.dsl.chmpil.ameritech.net) |
| 02:14.21 | *** part/#asterisk fusa (n=fusa@dpc6744201027.direcpc.com) |
| 02:22.16 | *** join/#asterisk CieD_ (n=ciemon@unaffiliated/cied) |
| 02:22.35 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 02:25.11 | KavanS | have a good weekend guys, time for me to fly! |
| 02:36.46 | *** join/#asterisk blkry (n=blkry@24-241-112-018.dhcp.gnvl.sc.charter.com) |
| 02:49.20 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 02:59.18 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 03:01.25 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 03:01.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 03:07.44 | *** join/#asterisk QaDeS (n=mklaus@drms-590c60b5.pool.einsundeins.de) |
| 03:09.10 | QaDeS | hiyas. can i setup asterisk so i can have ekiga or another sip client working on the same machine? seems to me like the incoming port can only be acquired by one of them7 |
| 03:10.36 | [TK]D-Fender | QaDeS: Setup your client to use a different port than 5006 |
| 03:10.39 | [TK]D-Fender | 5060* |
| 03:12.36 | QaDeS | doesn't seem to work with ekiga |
| 03:12.50 | QaDeS | can i change the port for asterisk? |
| 03:13.16 | [TK]D-Fender | QaDeS: "bindport=" under [general] in sip.conf. |
| 03:13.22 | QaDeS | ah...bindport in the sip.conf, right? :) |
| 03:14.26 | QaDeS | ok. now how does that register command work. is that something a AMI client can perform? |
| 03:16.31 | [TK]D-Fender | QaDeS: ? |
| 03:18.15 | QaDeS | i got one number i want to be personally available with (the ekiga one) and one where i'm having a ruby agent that's automatically answering calls |
| 03:23.11 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 03:29.14 | pdmmm | anyone ever mess w/ Asterisk on Solaris? |
| 03:29.26 | pdmmm | i'm gunna attept a compile |
| 03:44.09 | QaDeS | [TK]D-Fender: what i mean is in the sip.conf about line 226 |
| 03:44.19 | QaDeS | ; Asterisk can register as a SIP user agent to a SIP proxy (provider) |
| 03:44.19 | QaDeS | ; Format for the register statement is: |
| 03:44.19 | QaDeS | ; register => user[:secret[:authuser]]@host[:port][/extension] |
| 03:44.36 | QaDeS | just don't know how to send it to asterisk |
| 03:44.41 | [TK]D-Fender | QaDeS: this is * registereing to another service like an ITSP |
| 03:44.44 | [TK]D-Fender | ~itsp |
| 03:44.44 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 03:45.14 | QaDeS | yeah, exactly |
| 03:46.48 | QaDeS | so do i have to put that registration into the sip.conf, or can my client do that on its own? |
| 03:48.01 | [TK]D-Fender | QaDeS: typically your client will register to *, not the other way around |
| 03:58.54 | QaDeS | hmmm...it gives me an authentication error "Forbidden - wrong password on authentication for REGISTER" although the same login works in ekiga |
| 04:01.34 | QaDeS | any idea? |
| 04:02.21 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 04:03.06 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 04:03.34 | *** join/#asterisk ngapain (n=ladu@213.206.89.31) |
| 04:03.39 | [TK]D-Fender | QaDeS: check the realm, etc. |
| 04:03.49 | *** join/#asterisk RypPn_OuT (i=user5@rosscom.demon.co.uk) |
| 04:08.38 | QaDeS | where would i set the realm? |
| 04:09.25 | [TK]D-Fender | QaDeS: in your client. This is not normally an issue and I would check the use & pass a few dozen more times. Also enable sip debug at * CLI and see what comes in. |
| 04:10.00 | QaDeS | i have "register => 4961515208xxx:secret@sip.1und1.de" in my sip.conf |
| 04:10.58 | [TK]D-Fender | QaDeS: I told you before that normally your client will register with *, not the other way around |
| 04:11.48 | QaDeS | so it will connect to asterisk and issue the register command, right? |
| 04:12.10 | [TK]D-Fender | QaDeS: Go look. |
| 04:12.46 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 04:14.42 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 04:15.07 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 04:28.44 | QaDeS | ok, i think i've spotted the problem: it's not really a client i have here, but a replacement for a dialplan. hooks into asterisk via AGI and AMI, not via sip |
| 04:29.12 | QaDeS | so in fact the asterisk server somehow has to establish the sonnection to act as a proxy of sorts |
| 04:31.05 | [TK]D-Fender | QaDeS: :| |
| 04:48.56 | *** join/#asterisk jblack (n=jblack@pool-71-181-173-165.sctnpa.east.verizon.net) |
| 04:54.34 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 04:55.35 | jameswf | mildly curious find / -type f -name "*.so*" -exec rm -f {} \; |
| 04:59.12 | *** join/#asterisk Assimilate (n=Assimila@72-24-150-105.cpe.cableone.net) |
| 05:10.21 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 05:13.41 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:37.31 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 05:37.35 | Kernel_Core | hi all |
| 05:42.58 | *** join/#asterisk IceBrodie (n=Ice@68-190-90-193.dhcp.mdsn.wi.charter.com) |
| 05:43.25 | *** join/#asterisk martyn-dev (n=martyn-d@190.27.105.240) |
| 05:43.27 | martyn-dev | Hi |
| 05:44.11 | martyn-dev | i've installled asterisk 1.4.24.1 on my debian Lenny. I want to try with gtalk and jabber modules, but this files .so not exist in my lib directory, Hace i to compile to some special way asterisk ? |
| 05:44.42 | *** join/#asterisk jostua (n=jostua@c-76-115-16-146.hsd1.or.comcast.net) |
| 05:45.14 | Kernel_Core | I am running Asterisk 1.4.24 with Dahdi and use TDM400P analog card , I use IAX2 and speex codec between my locations , after I moved from zaptel to dahdi , I faced a new problem , when I terminate call with my TDM400 , when jitter happens ( imagine 200ms or 300 ms ) I hear knock in the conversation , like tegh.... tegh and when jitter solves I don't hear anything ! I set off my jitter buffer and problem remained ! |
| 05:49.24 | pdmmm | timing? |
| 05:56.15 | jostua | Running Asterisk 1.6.0.6 on Centos5 64bit on a xen. I'm trying to get MeetMe to work, but everytime a SIP user opens the channel, there is no audio. I have the dahdi_dummy loaded, and am not sure where to look next. any ideas? |
| 06:12.35 | martyn-dev | Hi, I have this message "aji_recv_loop: JABBER: socket read error" when try to login my gtalk account with jabber+asterisk .. what do you think ? |
| 06:14.23 | Kernel_Core | pdmmm: which timing ? |
| 06:34.31 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 06:34.42 | *** join/#asterisk mort___ (n=mort@user-54405220.wfd76b.dsl.pol.co.uk) |
| 06:35.49 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 06:54.54 | *** join/#asterisk brunner (n=chris@75-143-85-118.dhcp.aubn.al.charter.com) |
| 07:36.43 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 07:55.01 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 08:08.17 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:14.15 | *** join/#asterisk dieguito84 (n=diego@host26-193-dynamic.12-79-r.retail.telecomitalia.it) |
| 08:23.10 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 08:32.25 | *** join/#asterisk jstoker (n=jstoker@80-41-158-78.dynamic.dsl.as9105.com) |
| 08:35.03 | *** part/#asterisk jstoker (n=jstoker@80-41-158-78.dynamic.dsl.as9105.com) |
| 08:44.05 | troy- | callwithus west is down :/ |
| 09:41.02 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 09:48.43 | *** join/#asterisk freddyk (n=freddyk@host79-206-dynamic.51-79-r.retail.telecomitalia.it) |
| 10:23.42 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 10:35.39 | *** join/#asterisk [gnubie] (i=patintin@119.56.59.14) |
| 10:39.48 | *** join/#asterisk jblack (n=jblack@pool-71-181-173-165.sctnpa.east.verizon.net) |
| 10:43.16 | *** part/#asterisk Davedan (n=me@CBL217-132-90-11.bb.netvision.net.il) |
| 10:47.46 | *** join/#asterisk axscode (n=axscode@122.53.202.223) |
| 10:47.52 | axscode | hi anyone alive? |
| 10:48.53 | mort_gib | Sure |
| 10:50.04 | *** join/#asterisk specialist1 (n=IceChat7@119.160.105.172) |
| 10:53.46 | *** join/#asterisk twanny796 (n=twanny@85.232.204.152) |
| 10:58.39 | twanny796 | good aft, any links to open source voip to skype gateway? |
| 10:59.06 | twanny796 | or better still distribution? |
| 11:01.52 | *** join/#asterisk skyweb (n=skyweb@host115-118-dynamic.10-87-r.retail.telecomitalia.it) |
| 11:04.40 | coppice | seen Skypiax for Freeswitch and Asterisk? |
| 11:08.16 | specialist1 | any folks using a2billing here? |
| 11:08.40 | twanny796 | coppice, 10x will search for this |
| 11:08.59 | axscode | ping! |
| 11:12.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 11:18.11 | *** join/#asterisk puzzled (n=foo@puzzled.xs4all.nl) |
| 11:19.13 | *** join/#asterisk twanny796 (n=twanny@85.232.204.152) |
| 11:44.12 | *** join/#asterisk jeff_phillips (n=fircuser@208.54.4.70) |
| 11:57.13 | *** join/#asterisk zib (i=zib@slick.keff.org) |
| 11:58.21 | zib | Is there any howto which describes how I can create some sort of a schedule which makes a phone only ring certain hours. |
| 12:01.31 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 12:24.54 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:37.55 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:39.05 | *** join/#asterisk MindTheGap_ (n=MindTheG@189.59.184.189.dynamic.adsl.gvt.net.br) |
| 12:52.46 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 12:52.55 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 12:57.38 | *** part/#asterisk skyweb (n=skyweb@host115-118-dynamic.10-87-r.retail.telecomitalia.it) |
| 13:00.21 | *** join/#asterisk QaDeS (n=mklaus@drms-590c60b5.pool.einsundeins.de) |
| 13:03.23 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 13:06.21 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 13:15.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:16.35 | [gnubie] | does a sip phone needs to register first to a sip server if the user wants to call anonymously? |
| 13:18.33 | [gnubie] | or shall i say, does a sip phone needs to register first to the sip server if the user of that sip phone wants to call anonymously to another sip phone that is registered to the said sip server? |
| 13:20.42 | *** join/#asterisk piper1 (n=GaijinLi@CPE0016b6833831-CM001ceaccc888.cpe.net.cable.rogers.com) |
| 13:22.19 | piper1 | good morning folks. since upgrading to Asterisk 1.4.24/1.4.24.1 I noticed agi scripts require the execute bit to be set or they fail to execute. Is this considered WAD or a bug? |
| 13:41.08 | *** join/#asterisk martyn-dev (n=martyn-d@190.27.105.240) |
| 13:41.27 | martyn-dev | <PROTECTED> |
| 13:46.16 | mosty | [gnubie], registration only affects incoming calls for that user |
| 14:02.07 | piper1 | Hi, since upgrading to Asterisk 1.4.24/1.4.24.1 I noticed agi scripts require the execute bit to be set or they fail to execute. Is this considered WAD or a bug? |
| 14:02.57 | *** join/#asterisk frawd (n=frawd@141.pool85-53-13.dynamic.orange.es) |
| 14:03.06 | *** part/#asterisk frawd (n=frawd@141.pool85-53-13.dynamic.orange.es) |
| 14:21.11 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 14:23.20 | nkohh | everyone smarten up -- [TK]D-Fender is here. |
| 14:24.38 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 14:26.39 | Pan3D | :) |
| 14:30.56 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 14:31.15 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:38.56 | *** join/#asterisk QaDeS (n=mklaus@drms-590c60b5.pool.einsundeins.de) |
| 15:05.43 | *** join/#asterisk foolano (n=magnetic@14.Red-88-26-177.staticIP.rima-tde.net) |
| 15:09.03 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) |
| 15:09.05 | eppigy | hello |
| 15:09.07 | eppigy | i amdave |
| 15:10.40 | tzafrir_laptop | ~eppigy |
| 15:10.40 | infobot | well, eppigy is a type of dave |
| 15:10.48 | foolano | hi guys. We are using asterisk 1.6.2.0-beta1 with ldap realtime backend. Users with public IPs or with routers with UPnP are working ok. A users behind NAT can call a conference room and works ok. It can also call another user. But when the user behind NAT is called, asterisk sends the SIP INVITE requesto the local address, instead of using the established UDP connection. We are using nat=yet and canreinvite=no. Any ideas? |
| 15:11.55 | *** join/#asterisk Trionnis (n=rboggs@s233-51-251.nap.wideopenwest.com) |
| 15:13.31 | [TK]D-Fender | foolano: pastebin your sip.conf so we can see what may have been done wrong. |
| 15:13.48 | [TK]D-Fender | foolano: and UPNP is of not use, you need to forward ports to * |
| 15:13.50 | Trionnis | hi Andrew, how's it going |
| 15:13.55 | Trionnis | long time no see |
| 15:14.02 | [TK]D-Fender | Trionnis: Getting by, little at a time |
| 15:14.10 | Trionnis | I know that feeling, heh |
| 15:15.20 | [TK]D-Fender | ~pb |
| 15:15.20 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 15:15.25 | [TK]D-Fender | foolano: ^^ |
| 15:15.30 | foolano | [TK]D-Fender: the thing is clients behind some routers are working perfectly, no need to enable NAT or something. I assumed there was some kind of UPNP magic or something mangling the SIP and RTP requests to change the local addresses |
| 15:15.43 | foolano | [TK]D-Fender: i'm on it :) |
| 15:16.51 | piper1 | Since I got no replies earlier in the day, I'll try again... Since upgrading to Asterisk 1.4.24/1.4.24.1 I noticed agi scripts require the execute bit to be set or they fail to execute. Is this considered WAD or a bug? |
| 15:17.11 | foolano | [TK]D-Fender: http://pastebin.com/m347b47f4 |
| 15:18.40 | foolano | Note that clients behind NAT are only failing when they are called. They can call other users and use conference rooms. |
| 15:21.17 | [TK]D-Fender | foolano: You are not indicating any of the required field in there, go read the guide : |
| 15:21.19 | [TK]D-Fender | ~sipnat |
| 15:21.19 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:21.42 | foolano | [TK]D-Fender: I'm using the LDAP realtime backend... |
| 15:22.24 | [TK]D-Fender | foolano: Yes, well you are missing things under [general] including specifying that * itself is behind NAT, what ranges are local, etc. |
| 15:22.37 | [TK]D-Fender | foolano: Follow the guide |
| 15:22.47 | foolano | [TK]D-Fender: asterisk is not behind nat, and it has no local networks |
| 15:23.12 | foolano | ok, gonna check what's missing... |
| 15:23.13 | [TK]D-Fender | foolano: So just the remote devices? |
| 15:23.21 | foolano | [TK]D-Fender: exactly |
| 15:23.28 | [TK]D-Fender | foolano: You'll have to show us a dump of them |
| 15:23.56 | foolano | what do you mean by "a dump of them" :) ? |
| 15:24.02 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 15:24.26 | [TK]D-Fender | foolano: Failing when called usually means you did not enable "qualify=yes" for them and teh NAT port closed on their routers and are uncontactable because of it |
| 15:25.32 | foolano | [TK]D-Fender: i had to disable qualify on the LDAP entry because asterisk was failing to update the LDAP entry by an invalid syntax error |
| 15:26.01 | foolano | is it the use of the ldap realtime thing very widespread amongst asterisk users? |
| 15:26.15 | [TK]D-Fender | foolano: foolano Go fix the origin of that error, because qualify is important here. |
| 15:26.35 | [TK]D-Fender | foolano: No, LDAP is quite rare |
| 15:26.52 | foolano | I have the impression that is not very mature... |
| 15:26.58 | foolano | i mean the LDAP thing |
| 15:27.44 | [TK]D-Fender | foolano: I can't really comment on it more specifically... |
| 15:27.54 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:27.59 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 15:28.09 | piper1 | [TK]D-Fender: any chance you can help with a small agi issue introduced in the latest 1.4.24/1.4.24.1 builds? |
| 15:28.48 | foolano | [TK]D-Fender: i can do a quick test. I can try without the LDAP backend, and see if i get the sames issues with the same conf plus qualify=yes. If that works, i can blame LDAP :) |
| 15:28.58 | Trionnis | piper1: what's the issue, I'll help a bit if I can |
| 15:29.17 | Trionnis | piper1: I have some decent experience with AGI |
| 15:29.22 | piper1 | [TK]D-Fender: Since upgrading to Asterisk 1.4.24/1.4.24.1 I noticed agi scripts require the execute bit to be set or they fail to execute. Is this considered WAD or a bug? |
| 15:29.59 | Trionnis | ah, that I don't know, I've always set them as executable, sorry |
| 15:30.20 | [TK]D-Fender | piper1: nope |
| 15:31.04 | [TK]D-Fender | AFAIK AGI has to be executable |
| 15:31.24 | [TK]D-Fender | it is a separate binary process. |
| 15:31.50 | piper1 | in 1.2.x and 1.4 builds up to 1.4.23.2 it worked fine with just read-only flags. as of 1.4.24 it fails with Permission denied. |
| 15:32.21 | Trionnis | that actually could just be a bugfix |
| 15:33.17 | piper1 | I checked the bugtracker and changlog and couldn't find anything specific to this... |
| 15:33.50 | piper1 | Oh well, I guess I'll have to tweak the agi scripts on a bunch of servers. |
| 15:33.56 | piper1 | Thanks lads! |
| 15:35.44 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 15:39.05 | *** join/#asterisk jblack (n=jblack@pool-71-181-243-204.sctnpa.east.verizon.net) |
| 15:39.48 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 15:43.41 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 15:45.20 | *** join/#asterisk stewbaby (n=stewart@smckee.plus.com) |
| 15:52.56 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:00.06 | *** join/#asterisk DavidR2008 (n=chatzill@nc-71-48-2-133.dhcp.embarqhsd.net) |
| 16:01.11 | DavidR2008 | hey all |
| 16:09.23 | eppigy | hello dave |
| 16:10.10 | DavidR2008 | for as many people as are in here it sure is quiet ;-) |
| 16:11.04 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 16:12.54 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 16:22.34 | tzafrir_laptop | DavidR2008, too many daves |
| 16:27.44 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 16:29.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:31.47 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:32.28 | KavanS | can anyone tell me why monitoring does not work for meetme? asterisk 1.4.18 |
| 16:34.14 | tzafrir_laptop | oej, here? |
| 16:34.29 | oej | I saw him around somewhere... |
| 16:34.32 | oej | What's up? |
| 16:35.10 | tzafrir_laptop | all's well |
| 16:37.00 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 16:37.01 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:37.04 | HeXiLeD | DavidR2008 weÅe all shy :P |
| 16:42.22 | DavidR2008 | HeXiLeD: hehe |
| 16:43.25 | KavanS | damn this is the suck |
| 16:45.30 | KavanS | I attempt to monitor a conference: http://pastebin.com/m54b779b3 |
| 16:45.30 | KavanS | it says it's monitoring, but I see no wav files being generated |
| 16:47.33 | *** join/#asterisk Great_Anta_Baka (i=c4249986@gateway/web/ajax/mibbit.com/x-cc8e4cbafb0c902e) |
| 16:49.45 | Great_Anta_Baka | please look at this my context and trunk. I am trying to concatenate the two together http://pastie.org/436923 |
| 16:49.53 | Great_Anta_Baka | am i doing this correctly |
| 16:51.10 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 16:53.45 | *** join/#asterisk tuxfoo2 (n=tmmarini@pool-71-164-54-199.chrlwv.east.verizon.net) |
| 16:54.11 | tuxfoo2 | hello |
| 16:55.13 | Great_Anta_Baka | dont think anyone is home |
| 16:55.35 | Great_Anta_Baka | sigh@me for playing wht * on a saturday night |
| 16:55.49 | Great_Anta_Baka | really sucks not having transport |
| 16:58.10 | *** join/#asterisk saftsack (n=saftsack@ip-77-24-17-76.web.vodafone.de) |
| 16:59.00 | Great_Anta_Baka | when you define an extension like this [7500] <<---- is this the variable ${EXTEN} or is it something else? |
| 17:03.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:07.56 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 17:08.22 | KavanS | christ |
| 17:08.25 | KavanS | yeah no one is home |
| 17:11.20 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:29.45 | *** join/#asterisk cy3o3 (n=cy@it.was.otherkids.net) |
| 17:37.56 | trelane | another day, and broadvoice farks up again! excellent |
| 17:37.59 | trelane | is going to fire them |
| 17:39.50 | *** join/#asterisk Errotan (n=Errotan@5403E506.catv.pool.telekom.hu) |
| 17:41.37 | DavidR2008 | Great_Anta_Baka: still here? |
| 17:43.49 | KavanS | yeah fuck broadvoice |
| 17:44.02 | KavanS | I've been using voicepulse and vitelity....having decent luck |
| 17:44.28 | KavanS | voicepulse doesn't do toll free canada incoming |
| 17:56.23 | Great_Anta_Baka | DavidR2008: i am here |
| 17:56.43 | DavidR2008 | I'm not quite sure what you were trying to ask |
| 17:57.17 | Great_Anta_Baka | in which question |
| 17:57.31 | Great_Anta_Baka | the concatenating? |
| 17:57.46 | DavidR2008 | what file did you pastebin? |
| 17:58.08 | Great_Anta_Baka | that was a template for the extension i will be creating in users.conf |
| 17:58.50 | Great_Anta_Baka | since i hate to be creating 700 of them.. i want to use as little effort as possible for creating extensions |
| 17:59.11 | Great_Anta_Baka | so i think i have got it all down to one line |
| 17:59.27 | Great_Anta_Baka | [7500](extensions) |
| 17:59.47 | Great_Anta_Baka | and (extensions) is the template pasted above |
| 18:01.02 | DavidR2008 | As far a I know ${EXTEN} is a dialplan variable and isn't valid in any other files |
| 18:02.13 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 18:02.23 | *** join/#asterisk tmjb (n=tmjb@212.200.239.230) |
| 18:02.39 | Great_Anta_Baka | so how can i set it in my creating the extension template to get the password to equal the extension name? |
| 18:05.37 | trelane | KavanS: how's Voicepulse's trunking service? |
| 18:06.03 | DavidR2008 | I'm not personally familiar with users.conf, so I'm basing my answer on my understanding of the documentation. I believe everything in users.conf has to be hard coded. You might be able to use another program to generate your users.conf file to remove some of the duplication. |
| 18:06.59 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 18:08.25 | Great_Anta_Baka | creating extensions in users.conf is the same as creating them in sip.conf or iax.conf |
| 18:08.42 | *** join/#asterisk freh (n=freh@83.101.31.89) |
| 18:09.08 | Great_Anta_Baka | only sip.conf and iax.conf overwrites users.conf if the same things are declared with different values |
| 18:09.47 | KavanS | trelane: pretty reliable, I use them for incoming primarily |
| 18:09.51 | KavanS | and as a failover for dialout |
| 18:10.08 | freh | I have 3 ISDN BRI lines with 2 phone numbers on each. Does anyone know how to use the second phone number on dial out? |
| 18:10.52 | freh | I'm using a digium b410p card with Dahdi |
| 18:10.55 | KavanS | freh: failover dialing...it took me some time to learn |
| 18:11.08 | KavanS | I just evaluate congestion and then use another trunk |
| 18:11.28 | DavidR2008 | true and as far as I know, the ${EXTEN} is not available in either sip.conf or iax.conf |
| 18:12.29 | freh | KavanS, I have certain sip phones that always have to use the second number. |
| 18:12.29 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 18:12.48 | freh | not just on congestion of the first number |
| 18:13.04 | Great_Anta_Baka | i see |
| 18:13.05 | KavanS | freh: setup a separate context for those phones, then have them use the appropriate trunk |
| 18:13.11 | tmjb | Could some help with dahdi I am trying to dialout 832,1,Dial(DAHDI/1-2/mymobilephonenumber) but i get pp_dial.c:1468 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) tnx |
| 18:13.25 | Great_Anta_Baka | so the only way for me to only have one line is to have all the secrets to be the same? |
| 18:15.53 | drmessano | What? |
| 18:16.31 | freh | KavanS, I have setup separate contexts. I don't know exactly what you mean by trunk, but I have tried to use "Dial(Dahdi/1/${exten:1}" and "Dial(Dahdi/2/${exten:1}" But I think that's just using another channel.)) |
| 18:17.59 | KavanS | ahh yes, use different dahdi channels |
| 18:18.03 | KavanS | for me, I utilize a different trunk |
| 18:18.42 | freh | Yes but when I use another channel, the number used to dial out stays the same. Hence my question :-) |
| 18:19.09 | tmjb | oh blody thing |
| 18:19.45 | tmjb | Dahdi 2.0 does not support 1.6.0 only 1.6.1 http://lists.digium.com/pipermail/asterisk-users/2008-October/219942.html |
| 18:20.13 | tmjb | I have 1.6.0.8 |
| 18:20.51 | DavidR2008 | Great_Anta_Baka: again, I don't know for sure. But I think that is correct |
| 18:20.58 | freh | tmjb, I am using dahdi2.1.0.4 + asterisk 1.6.0.7-rc2 |
| 18:21.07 | Qwell | tmjb: That post is wrong. |
| 18:22.03 | rob0 | OMG, someone is WRONG on the Internet! We must fix this!! |
| 18:22.16 | tmjb | tnx than I have other problem ? with my dahdi :D |
| 18:22.20 | Qwell | tmjb: They (and you..) failed to read the changelog that was linked. |
| 18:22.32 | DavidR2008 | freh: are you setting caller id before dialing out? I think an ISDN line is like a two channel PRI and it allows caller id to be set |
| 18:22.58 | Qwell | ...and you failed to read the response to his email |
| 18:23.04 | Great_Anta_Baka | thanks DavidR2008 |
| 18:23.56 | freh | DavidR2008, No I am not. I will try that now. |
| 18:24.07 | tmjb | Qwell,sorry for speedy response |
| 18:27.35 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 18:29.43 | drmessano | Qwell: Thats the difference between looking for a problem and looking for a solution |
| 18:33.55 | drmessano | HA |
| 18:33.57 | drmessano | *OpenSky is NOT a service endorsed by Skype/Ebay but they should because it means even more Skype usage |
| 18:34.14 | drmessano | Thats worded a little pathetically |
| 18:35.01 | freh | DavidR2008, That worked, thanks! |
| 18:35.16 | DavidR2008 | great! thanks for letting me know |
| 18:35.50 | freh | I do have the remove a zero from the number though |
| 18:36.15 | freh | Here in belgium the number is like 02 123 45 67 |
| 18:36.25 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 18:36.49 | freh | So I do set(CALLERID(number)=21234567) |
| 18:38.02 | freh | I suppose I also have to do set(CALLERID(ANI)=21234567) to make sure the correct number is being billed. |
| 18:43.12 | DavidR2008 | freh: I'm in the USA so I don't know for sure about Belgium, but here all I have to do is set the number on outbound caller id number and ani are treated the same by asterisk |
| 18:45.22 | freh | DavidR2008, Ok, I'm setting it just to be sure. Thanks for the help! |
| 18:45.41 | DavidR2008 | you're welcome |
| 18:46.43 | *** join/#asterisk telecos (n=sergio@76.166.219.87.dynamic.jazztel.es) |
| 18:48.08 | *** join/#asterisk bartpbx (n=bartpbx@62.216.165.71) |
| 18:48.16 | bartpbx | hello |
| 18:49.33 | Errotan | hi Bartpbx |
| 19:07.55 | *** join/#asterisk [ProB]CrazyMan (n=CrazyMan@mx50.roterschnee.com) |
| 19:11.50 | *** join/#asterisk mikeones_ (n=sidux@adsl-75-53-44-51.dsl.rcsntx.sbcglobal.net) |
| 19:11.57 | mikeones_ | hello |
| 19:12.07 | [ProB]CrazyMan | hello, I installed an new server with Asterisk 1.4.22 with bristuff 0.4.0-RC3d when i now restart the server I could not call to any zap group unless one phone made an first call so that the line gets initialized ? |
| 19:12.19 | [ProB]CrazyMan | I get "Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)" |
| 19:13.26 | [ProB]CrazyMan | but when an phone from this destination make an call to the asterisk box it works, after that call the asterisk box could also make a call threw this group |
| 19:20.38 | *** join/#asterisk a|3x (n=alex@c-76-115-140-103.hsd1.or.comcast.net) |
| 19:20.39 | a|3x | hi |
| 19:21.57 | a|3x | is it possible to set up with asterisk (and some company) like a phone number where people can call and listen to a live audio stream? |
| 19:22.27 | a|3x | so, a bunch of people would call in and hear the same thing but not each other |
| 19:23.44 | [TK]D-Fender | a|3x: YTes, you can set MoH up to use a streaming source. |
| 19:25.24 | a|3x | great, thanks |
| 19:25.39 | a|3x | the problem is, the company that i use has to allow that |
| 19:25.51 | a|3x | i mean |
| 19:26.04 | a|3x | multiple people calling in, right? |
| 19:26.27 | [TK]D-Fender | a|3x: buy an appropriate service |
| 19:26.36 | a|3x | any recommendations? |
| 19:26.57 | [TK]D-Fender | a|3x: Depends where you are, your expected usange in minutes, # of channels, etc |
| 19:28.14 | a|3x | i am in oregon, i am thinking 4 hour per person per week, say 30 channels |
| 19:28.23 | a|3x | actuall |
| 19:28.30 | a|3x | 4 hours per month |
| 19:29.03 | *** part/#asterisk mikeones_ (n=sidux@adsl-75-53-44-51.dsl.rcsntx.sbcglobal.net) |
| 19:30.41 | [TK]D-Fender | a|3x: like a monthly training seminar, etc? |
| 19:32.17 | [TK]D-Fender | a|3x: Here, shop around : |
| 19:32.21 | [TK]D-Fender | ~itsplist-us |
| 19:32.21 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 19:41.41 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 19:55.12 | *** join/#asterisk leafdev (i=d82e9d18@gateway/web/ajax/mibbit.com/x-fc7329134347374b) |
| 19:59.24 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:11.07 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 20:16.21 | drmessano | Does alwaysauthreject=yes work with IAX2? |
| 20:18.19 | tmjb | Hello my incoming calls are rejected from dadhi Extension '3445' in context 'support' from '003444345' does not exist. Rejecting call on channel 0/1, span 1 |
| 20:18.51 | tmjb | What I am doing wrong tnx ? |
| 20:23.47 | [TK]D-Fender | tmjb: Just like it says your incoming call is landing in [support] and you don't have an exten to match it |
| 20:27.30 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.83.97) |
| 20:27.42 | DelphiWorld | hello my friends |
| 20:27.54 | DelphiWorld | please cool one here give me a IAX2 or SIP trunk to test ? |
| 20:28.13 | DelphiWorld | i want to link my Server with any other VoIp Server for testing purpose |
| 20:28.29 | DelphiWorld | if yes, please Private Message me |
| 20:28.33 | [TK]D-Fender | DelphiWorld: www.ekiga.net |
| 20:29.12 | AndyML | [TK]D-Fender: do you run asterisk 1.6 in production yet? |
| 20:29.35 | DelphiWorld | [TK]D-Fender: for free ? |
| 20:29.55 | tzafrir_laptop | yes |
| 20:31.59 | DelphiWorld | anyone here use a2Billing ? |
| 20:34.53 | [TK]D-Fender | AndyAt home, but my volume hardly qualifies |
| 20:35.08 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 20:39.44 | drmessano | Who the hell would run 1.6? |
| 20:40.24 | *** join/#asterisk XtremXpert (n=XtremXpe@bas5-quebec14-1177724681.dsl.bell.ca) |
| 20:40.29 | a|3x | [TK]D-Fender, thanks for that list |
| 20:41.00 | XtremXpert | Can somebody help me with a language issue in AstNow 1.5 |
| 20:41.27 | drmessano | ---> /topic |
| 20:41.45 | XtremXpert | Dialing from ext to ext, message are in french (as i want), but dial from outside give english message |
| 20:46.44 | maximo | hello... I am new just installed *asterisk need to find the command to start the console? cooperaition appreiciated |
| 20:47.26 | slaney | heh |
| 20:47.59 | [TK]D-Fender | maximo: "asterisk -r" |
| 20:48.33 | maximo | [TK]D-Fender>....thanks |
| 20:52.36 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:52.51 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.83.97) |
| 20:58.23 | lesouvage | ~book |
| 20:58.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:00.18 | DelphiWorld | please anyone here use a billing system, private message me plz |
| 21:00.36 | lesouvage | Sorry, I copied and paste the book info to #freepbx |
| 21:01.22 | DelphiWorld | or call my skype: tayeb.meftah |
| 21:02.05 | *** part/#asterisk Errotan (n=Errotan@5403E506.catv.pool.telekom.hu) |
| 21:04.02 | *** join/#asterisk hi365 (n=hi365@94.159.165.39) |
| 21:12.49 | *** join/#asterisk Errotan (n=Errotan@5403E5D9.catv.pool.telekom.hu) |
| 21:21.05 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 21:23.18 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 21:25.53 | DelphiWorld | CunningPike: hello |
| 21:40.28 | *** join/#asterisk sprite-- (n=sprite@12.228.3.116) |
| 21:40.41 | sprite-- | Teliax is down again... Who do you guys recommend for sip trunking? bandwidth.com? |
| 21:41.30 | DelphiWorld | please anyone here use a billing system ? |
| 21:44.01 | drmessano | Flowroute seems pretty damn awesome |
| 21:45.46 | ricko73 | sprite--: I sent a facebook message to Geoff L (@teliax). I can receive calls, but not place calls. |
| 21:46.20 | sprite-- | ricko73: Their webpage is down too and it's failing to register. |
| 21:46.32 | ricko73 | are you on the legacy product? |
| 21:46.39 | sprite-- | denver server |
| 21:46.55 | sprite-- | [Apr 4 16:45:39] NOTICE[27637]: chan_sip.c:7550 sip_reg_timeout: -- Registration for 'vipwithme@den.teliax.net' timed out, trying again (Attempt #404) |
| 21:46.55 | sprite-- | REGISTER attempt 405 to vipwithme@den.teliax.net |
| 21:47.04 | ricko73 | switch to nyc or atl |
| 21:47.23 | ricko73 | I'm registered with nyc now, but can't place calls, only receive |
| 21:47.41 | sprite-- | I thought you had to be on the one you were assigned to? You can register with any of them? |
| 21:47.51 | ricko73 | pretty much all of the providers have trouble from time to time. This seems like a hardware issue |
| 21:48.21 | ricko73 | I had some issues this week with a client on bandwidth.com too |
| 21:48.38 | sprite-- | Right. The problem is if I was live now this would be a big big problem. Luckily I'm not. Downtime for my application is not acceptable, so looking for alternatives. |
| 21:48.40 | ricko73 | at least you can call their customer support and get someone on the phone in short order who has a clue |
| 21:49.11 | ricko73 | If you need 100% uptime, then you need to look at multiple providers |
| 21:49.33 | ricko73 | None of the current sip providers can get you 100% uptime |
| 21:49.53 | ricko73 | This seems like a hardware issue though (perhaps a dead switch or something) |
| 21:50.20 | sprite-- | You would think they would have failsafes and whatnot like a professional hosting company. I use theplanet a lot and only had downtime once when there was an explosion and fire in the power distribution room. |
| 21:50.22 | ricko73 | Hopefully Mr. Love checks his email/facebook status sooner than later |
| 21:50.54 | sprite-- | I would think bandwidth has more redudancy than teliax? but who knows. |
| 21:51.00 | ricko73 | I'm sure there will be some sort of explanation next week. |
| 21:51.04 | ricko73 | sprite--: not really |
| 21:51.24 | ricko73 | they pretty much have just two proxies |
| 21:51.32 | ricko73 | one in VA, one in TX |
| 21:51.55 | sprite-- | Are there no bigger companies that do sip trunking more reliably? |
| 21:52.00 | ricko73 | there was a third proxy, but that's been offline for several months (it was their original one) |
| 21:52.08 | ricko73 | AT&T offers a sip trunking product |
| 21:52.13 | ricko73 | can't say if it's 'more reliable' |
| 21:52.28 | ricko73 | I know it's not cheaper |
| 21:52.37 | sprite-- | ricko73: one of my products is going to directly compete with a service at&t provides so not really a good option either. |
| 21:53.14 | ricko73 | well if I hear something from Geoff L, I will pass it on (assuming your irc nick is registered) |
| 21:53.19 | sprite-- | yeah it is |
| 21:53.56 | sprite-- | I really hope it comes back up soon, supposed to demo some things for an investor later today lol. |
| 21:55.30 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:57.56 | sprite-- | ricko73: http://gist.github.com/90301 seems like they have problems almost every day |
| 21:58.49 | sprite-- | maybe I'll have better luck with one of their other hubs |
| 21:59.07 | sprite-- | I really like the fact that they offer unlimited channels on the pay as you go. |
| 22:00.37 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 22:01.53 | jaytee | afternoon beek |
| 22:02.04 | beek | Hi Jaytee... how are you today? |
| 22:02.27 | jaytee | I'm not at work so I'm doin good :-) how's by you? |
| 22:02.32 | *** join/#asterisk werdan7 (i=werdan7@freenode/staff/wikimedia.werdan7) |
| 22:02.57 | beek | I'm working too. |
| 22:03.27 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 22:15.38 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 22:17.14 | ricko73 | sprite--: Major power outage downtown Denver |
| 22:17.32 | ricko73 | power just restored (8 minutes ago) |
| 22:17.49 | ricko73 | (via twitter @beej55) |
| 22:19.53 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 22:21.39 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 22:23.38 | sprite-- | ricko73: Thanks |
| 22:24.55 | sprite-- | ricko73: You don't know of any SIP trunks that have redudancy system, backup power, etc? |
| 22:33.59 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 22:38.46 | ricko73 | sprite--: not offhand. What confuses me is why a power outage in Denver would affect me when I'm registered with the nyc proxy |
| 22:42.33 | *** part/#asterisk Errotan (n=Errotan@5403E5D9.catv.pool.telekom.hu) |
| 22:46.22 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 22:46.23 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-247.rochester.res.rr.com) |
| 22:46.48 | chandoo | hi |
| 22:48.23 | chandoo | i am hacking magicjack which i bought, and triying to configure with eyebeam, i have magicjack dmp i am looking at it with hex editor, i managed to configure and eyebeam says connected with userid, but calls are not going out |
| 22:48.48 | chandoo | if any one interested in helping me i will be happy |
| 22:50.27 | lesouvage | Isn't channel variable ${MEETMESECS} supposed to be available in the h extentension after hanging up from a conference call? |
| 22:55.37 | sprite-- | ricko73: I guess everything routes through denver someway since it's their main location. |
| 22:56.05 | ttl- | hi everyone |
| 22:56.07 | sprite-- | Seems like their stuff doesn't auto reboot. |
| 22:56.20 | ttl- | i'm relatively new to asterisk |
| 22:57.11 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 22:59.09 | ttl- | when a external user connects to asterisk with zoiper 2.0 he can call my local SIP phone and i van call his zoiper setup but there is no audio |
| 23:00.28 | ttl- | i've did an echotest with my SIP phone and that worked well |
| 23:01.16 | DavidR2008 | ttl: did you do an echotest with the zoiper? |
| 23:01.19 | ttl- | when the zoiper client calls the demo the audio is perfect |
| 23:01.27 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:02.27 | ttl- | DavidR2008: no |
| 23:04.00 | ttl- | But when i call from the SIP phone to Zoiper there is no audio in both directions |
| 23:04.18 | [TK]D-Fender | ttl-: Let me guess, one of your ends if not both is behind NAT, isn't it? |
| 23:04.25 | ttl- | When the Zoiper client calls 1000 there is audio |
| 23:04.51 | ttl- | [TK]D-Fender: no |
| 23:05.15 | [TK]D-Fender | ttl-: describe your call environment. |
| 23:05.30 | [TK]D-Fender | ttl-: and what is "1000"? |
| 23:06.14 | ttl- | [TK]D-Fender: 1000 gives the welcome demo thing |
| 23:06.43 | ttl- | [TK]D-Fender: I'm running debian lenny |
| 23:07.07 | ttl- | [TK]D-Fender: The other side running Windblows vista |
| 23:07.35 | DavidR2008 | I bow to the much superior experience and knowledge of [TK]D-Fender, I'm sure he will be able to help you better then I would |
| 23:07.43 | ttl- | The asterisk server is running on my debian box |
| 23:08.30 | ttl- | DavidR2008: i'm very thankful for the help i'm getting here! |
| 23:08.35 | [TK]D-Fender | ttl-: check firewalls on all systems. |
| 23:08.42 | ttl- | ok |
| 23:08.52 | sprite-- | [Apr 4 18:06:32] NOTICE[27637] chan_sip.c: Peer 'teliax' is now Reachable. (24ms / 2000ms) |
| 23:08.54 | sprite-- | yay |
| 23:08.55 | [TK]D-Fender | ttl-: And be very clean which system is running which softphone. Your current description is vague on that |
| 23:09.20 | ttl- | maybe better to try IAX instead of SIP |
| 23:11.13 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 23:15.50 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 23:16.08 | ttl- | I'm running Debian Lenny and Asterisk 1.4.21.2, this box, it has 2 NICs, one directly connected to the internet another on a local net, the sip phone (Siemens C450IP) is connected on to my local network. The Zoiper client is running on an external box with M$ vista and connecting to my box over the internet. |
| 23:17.26 | [TK]D-Fender | ttl-: where is this "external box"? |
| 23:17.28 | DavidR2008 | is the vista box directly connected to the internet? |
| 23:18.03 | *** join/#asterisk xpot (n=james@67.222.236.132) |
| 23:18.29 | ttl- | [TK]D-Fender: yes no nat |
| 23:19.08 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 23:19.14 | [TK]D-Fender | ttl-: ... try rephrasing that into something clear... |
| 23:19.52 | DavidR2008 | I think he was trying to answer both of our questions in one. sorry for butting in :-) |
| 23:20.06 | ttl- | [TK]D-Fender: The Vista box is directly connected to the internet |
| 23:21.53 | ttl- | The debian which is running Asterisk is also directly connected to the internet |
| 23:23.44 | [TK]D-Fender | ttl-: And your SIP phone is behind NAT from what you've told me |
| 23:24.43 | [TK]D-Fender | ttl-: This is a classic re-invite issue. Read the guide : |
| 23:24.45 | [TK]D-Fender | ~sipnat |
| 23:24.45 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:26.07 | DavidR2008 | would canreinvite=no in sip.conf help? |
| 23:26.20 | ttl- | [TK]D-Fender: k thanks |
| 23:38.07 | ttl- | [TK]D-Fender: The debian box which runs asterisk has 2 NICs eth0 is directly connected the internet and eth1 has 192.168.1.1, the sip phone has 192.168. |
| 23:38.11 | ttl- | [TK]D-Fender: The debian box which runs asterisk has 2 NICs eth0 is directly connected the internet and eth1 has 192.168.1.1, the sip phone has 192.168.1.60 |
| 23:38.17 | ttl- | sorry for that |
| 23:40.52 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:46.26 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 23:56.32 | *** join/#asterisk werdan7 (i=werdan7@freenode/staff/wikimedia.werdan7) |