| 00:02.40 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 00:03.06 | apocn | I have registered skype to a softswitch, how can I let a sip user make calls through it? |
| 00:03.09 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 00:07.00 | mosty | apocn, what softswitch? |
| 00:14.33 | apocn | mosty: its a class 4 |
| 00:14.52 | apocn | I registered with my softphone (same user/pass) and I could make international calls |
| 00:15.06 | apocn | now I registered my asterisk cause I want my users to make calls through it too |
| 00:15.46 | apocn | how can I tell the Dial to go through it? making a [my_provider] on sip.conf as peer |
| 00:16.04 | mosty | apocn, what make/model softswitch? |
| 00:16.14 | mosty | what software? |
| 00:17.28 | apocn | mosty: its a veraz |
| 00:17.31 | apocn | dont know the model |
| 00:18.25 | chisefu|afk | how are you guys accessing your sip.conf files? |
| 00:18.57 | mosty | chisefu|afk, we use our favorite text editor. try #asterisk-gui for help with asterisknow |
| 00:19.22 | mosty | apocn, so you are trying to figure out how to send calls to the softswitch from asterisk? |
| 00:21.22 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 00:22.21 | apocn | mosty: right |
| 00:22.32 | *** join/#asterisk chuck64 (n=nospam@74.aa.425e.cidr.airmail.net) |
| 00:23.08 | apocn | mosty: I want to know how to make calls through it from my softswitch (to an external phone or cellphone). |
| 00:23.21 | mosty | apocn, setup a peer in sip.conf then Dial(SIP/thepeernameyouchose/thenumbertodial,timeout) |
| 00:23.43 | mosty | then you'll have to figure out how to configure the softswitch |
| 00:28.12 | chuck64 | Hello. Joe Newbie here... I'm just hanging out hoping to soak up some knowledge. |
| 00:29.25 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 00:31.06 | chuck64 | Can anyone suggest a good ATA and a reliable VOIP-to-PSTN provider? |
| 00:31.34 | jwh | country? |
| 00:31.51 | apocn | mosty : yes the softswitch is setup |
| 00:31.52 | chuck64 | USA. I'll eventually add Asterisk to the equation, but not at first. |
| 00:32.15 | jwh | hm, not sure then |
| 00:32.23 | jwh | if you find a reasonable carrier let me know :P |
| 00:32.31 | chuck64 | uh oh... |
| 00:33.08 | chuck64 | I'm guessing Skype and Vonage are curse words in this crowd? |
| 00:33.18 | jwh | guess so ;p |
| 00:33.55 | chuck64 | well ok... I'll take my chances with a no-name provider. Any suggestions on an ATA? |
| 00:34.35 | jwh | can't recommend any, just buy/acquire a sensible voip phone |
| 00:34.42 | chuck64 | I saw the Systm episode where the guy used a Sipura. How are they now that Linksys bought them? |
| 00:35.18 | mosty | ata's are cheap, but you won't get as good call quality as with a real sip phone |
| 00:35.27 | mosty | and less features |
| 00:35.33 | chuck64 | ok |
| 00:35.36 | mosty | but the linksys ata's are ok |
| 00:35.59 | chuck64 | well, OK. Maybe I'm attacking this from the wrong end. |
| 00:36.01 | *** part/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com) |
| 00:36.11 | apocn | mosty : now I have setup a peer in sip.conf, but... how does it know that it will have to go through the "register" ? |
| 00:36.13 | chuck64 | What's a cheap way to get set up with VOIP phone service? |
| 00:36.26 | apocn | what parameter should I give this "peer" in sip.conf? |
| 00:36.40 | apocn | its fromdomain? |
| 00:37.02 | mosty | apocn, no, that's done in the dialplan, extensions.conf - sounds like you should read the book |
| 00:37.04 | mosty | ~thebook |
| 00:37.05 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:37.23 | mosty | chuck64, get a sip phone or ata, and get an account with a voip provider |
| 00:45.09 | apocn | mosty: you didnt get me |
| 00:45.38 | apocn | I know in extensions.conf you specify where you go, but... Im talking about the sip.conf |
| 00:45.40 | mosty | apocn, read that free online book jbot mentioned, it will get you started |
| 00:45.43 | apocn | when I create my [peer] |
| 00:46.04 | apocn | tr/[]/""/ |
| 00:46.19 | apocn | how will the peer know that it must go through the "register" ?? |
| 00:46.25 | mosty | apocn, the dialplan says where to send calls, sip.conf just defines what peers you can send calls to |
| 00:46.27 | apocn | and not for example a local extension |
| 00:46.48 | mosty | apocn, see the syntax of the dial command |
| 00:46.57 | apocn | I know that |
| 00:47.07 | apocn | I think you are not understanding me... |
| 00:47.16 | mosty | can you rephrase your question? |
| 00:47.46 | apocn | ok, in sip.conf I have [my_provider] type=peer, dtmfmode=rfc2833, etc... |
| 00:48.24 | apocn | I also have my register => user:pass@mydomain |
| 00:48.29 | mosty | yes, and? |
| 00:48.51 | apocn | in my extensions I have _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@my_provider) |
| 00:49.16 | apocn | now, how does asterisk know that he must route this my_provider through the registry? |
| 00:49.29 | mosty | i think you misunderstand what sip registration is |
| 00:49.45 | apocn | ok |
| 00:49.46 | mosty | the sip register command just tells "my_provider" where to send incoming calls |
| 00:49.55 | apocn | ohh... |
| 00:50.33 | mosty | it's like saying to my_provider "i'm logging in from here, so you know how to send incoming calls to me" |
| 00:50.47 | apocn | I see, thats why there is a /extension on the register line |
| 00:50.57 | apocn | ... ok |
| 00:52.06 | mosty | you only need to register to your provider if your provider thinks you have a dynamic ip |
| 00:52.28 | apocn | now, if I have a: host=xxx.xxx.xxx.xxx, type=peer, username=myuser, pass=mypass |
| 00:52.45 | apocn | and I Dial(SIP/ext@my_provider), it should work? |
| 00:52.48 | mosty | yes |
| 00:53.03 | apocn | where username and pass are the ones used to authenticate to the softswitch |
| 00:53.07 | apocn | s/pass/secret/ |
| 00:53.13 | mosty | apocn, yes |
| 00:53.26 | apocn | I like jbot! hehe |
| 00:53.42 | apocn | mosty: ok, thanks a LOT |
| 00:53.52 | mosty | no prob |
| 00:57.37 | *** join/#asterisk mechanicus01 (n=none@189.149.60.15) |
| 01:00.26 | *** join/#asterisk mechanicus01 (n=none@189.149.60.15) |
| 01:02.22 | WilliamK | Can one make asterisk listen on multiple ports? (5060/5061) |
| 01:04.46 | mosty | WilliamK, if not, try port forwarding |
| 01:19.00 | WilliamK | anyone know what the default duplex setting is on an SPA-2002? |
| 01:20.04 | mosty | WilliamK, duplex? |
| 01:20.19 | WilliamK | on the ethernet side |
| 01:22.03 | mosty | i would imagine that any ethernet device made in the last ten years is either autosense or full duplex by default |
| 01:22.30 | WilliamK | actually I found it... 10/half |
| 01:23.04 | WilliamK | which is surprising considering we are talking about a voip adapter |
| 01:23.14 | chuck64 | yeah... pretty amazing |
| 01:23.41 | mosty | WilliamK, maybe it's autosensing that setting |
| 01:25.07 | WilliamK | I finally googled enough docs and also looked at the cisco switch |
| 01:26.43 | WilliamK | what I find very amusing is the postage machine works but the fax machine won't on the SPA-2002 |
| 01:29.08 | chuck64 | Does anyone have an opinion/suggestion on a WiFi VOIP phone? |
| 01:45.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:46.59 | *** join/#asterisk nny_1 (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 01:47.08 | nny_1 | ~ grandstream |
| 01:47.09 | jbot | grandstream is probably the Yugo of VoIP hardware. Run. Run away now. |
| 01:47.20 | nny_1 | lol |
| 01:47.23 | nny_1 | ~ budgetone |
| 01:47.28 | nny_1 | :( |
| 01:48.55 | riddlebox | nny_1, I have three grandstream gxp2000 phones and have no problems at all |
| 01:49.35 | nny_1 | riddlebox: looking at something for a lrage deployment for users who need nada |
| 01:49.42 | riddlebox | true |
| 01:50.21 | nny_1 | either the soundpoint 320 or the budgetone, and the budgetone,wekk.. i don't have high hopes |
| 01:50.24 | nny_1 | well* |
| 01:50.40 | riddlebox | I hear polycom's are the best choice |
| 01:50.44 | nny_1 | indeed |
| 01:53.22 | mosty | snom and linksys are also ok |
| 01:54.06 | nny_1 | yeah |
| 01:54.10 | riddlebox | I am interested in getting a linksys wifi cordless |
| 01:54.10 | nny_1 | have a client with snoms |
| 01:54.14 | nny_1 | <PROTECTED> |
| 01:54.15 | nny_1 | scripts/Makefile.build:46: *** CFLAGS was changed in "/usr/src/zaptel-1.4.7.1/Makefile". Fix it to use EXTRA_CFLAGS. Stop. |
| 01:54.17 | nny_1 | ? |
| 01:54.30 | riddlebox | nny_1, ubuntu? |
| 01:54.44 | nny_1 | riddlebox: indeed |
| 01:54.59 | riddlebox | nny_1, did you apt-get install build-essential? |
| 01:55.02 | nny_1 | riddlebox: hehehe hardy (don't shoot me) trying to get snmp working on this test box |
| 01:55.21 | nny_1 | riddlebox: yes |
| 01:55.44 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 01:55.48 | riddlebox | hrmm I am not sure then |
| 01:55.58 | nny_1 | riddlebox: yeah np i'll figure it out |
| 01:56.11 | nny_1 | had dapper, and the net-snmp version had linked library issues |
| 01:56.18 | riddlebox | I upgraded my asterisk box from feisty to gutsy with no problem at all |
| 01:56.41 | *** join/#asterisk pikos (n=pikos@77.49.48.107) |
| 01:56.49 | riddlebox | actually it is my mythtv and asterisk box |
| 01:56.54 | nny_1 | riddlebox: yeah this was dapper LTS, just installed teh hardy alpha in hopes to take a shot.. seems it needs libsnmp15 (AFAIK) |
| 01:56.55 | nny_1 | nice |
| 01:56.57 | pikos | hello all |
| 01:59.30 | pikos | does any one knows if i can alter the priority proprty in realtime into a varchar rather than an int? |
| 01:59.47 | pikos | for the extentions |
| 02:04.31 | pikos | ok then .. next question .. is there a way i can dynamicly alter the <number> into a context ? lets say "exten => ${MYVAR},1,Answer" ?? |
| 02:05.38 | riddlebox | is there a way to have a conference call and set it to record when I press * and to stop # whenever I want |
| 02:09.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 02:13.29 | mosty | riddlebox, enable one-touch recording in features.conf |
| 02:14.17 | riddlebox | I figured it was possible |
| 02:19.12 | *** join/#asterisk __freedom__lover (n=eduardo@201-42-187-19.dsl.telesp.net.br) |
| 02:20.04 | __freedom__lover | hey, can anyone say me if "ASTERISK-MIB::astNumChannels.0" is the number of activies channels? |
| 02:22.19 | Nugget | http://forums.macrumors.com/showthread.php?p=4644102 <-- the Programmer Hieracrchy |
| 02:22.27 | Nugget | Hierarchy, even. |
| 02:25.37 | __freedom__lover | Nugget that's great |
| 02:27.24 | nny_1 | how would you compile zaptel to not include hotplugging support |
| 02:32.43 | mosty | nny_1, what's wrong with hotplugging support? |
| 02:32.59 | nny_1 | mosty: make[2]: Entering directory `/usr/src/linux-headers-2.6.24-2-server' |
| 02:32.59 | nny_1 | scripts/Makefile.build:46: *** CFLAGS was changed in "/usr/src/zaptel-1.4.7.1/Makefile". Fix it to use EXTRA_CFLAGS. Stop. |
| 02:33.11 | nny_1 | mosty: can't seem to satiate the makefile changes it is asking for |
| 02:33.40 | mosty | what does that have to do with hotplug? |
| 02:34.06 | nny_1 | eh maybe i assumed that was the issue as it stated make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.4.7.1 HOTPLUG_FIRMWARE=yes modules |
| 02:34.17 | nny_1 | but either way, still trying to get around the issue |
| 02:35.16 | mosty | are you using a clean zaptel source? |
| 02:35.44 | nny_1 | mosty: yes |
| 02:35.57 | nny_1 | mosty: 1.4.7 |
| 02:36.05 | nny_1 | mosty: well.. i have hpec in it |
| 02:36.45 | nny_1 | er 1.4.7.1* |
| 02:36.55 | mosty | you applied a patch to zaptel? does it compile without that patch? |
| 02:39.40 | nny_1 | mosty: no patch |
| 02:39.50 | nny_1 | mosty: well maybe not considered a patch |
| 02:39.56 | nny_1 | i can try without anything in hpec dir |
| 02:43.59 | nny_1 | hmmph nope |
| 02:44.45 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 02:47.18 | *** join/#asterisk adjohn (n=adjohn@p5087-ipad408marunouchi.tokyo.ocn.ne.jp) |
| 02:50.06 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:57.56 | *** join/#asterisk mog (n=mog@nat/digium/x-ac62de182e868059) |
| 02:57.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:58.21 | *** join/#asterisk pirulo (n=andres_p@70.56.223.76) |
| 02:59.04 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 02:59.13 | teknoprep | dial plans are crazy complex |
| 02:59.21 | teknoprep | my brain hurts from learning this crap |
| 03:02.23 | *** part/#asterisk pirulo (n=andres_p@70.56.223.76) |
| 03:03.16 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
| 03:03.52 | nny_1 | teknoprep: heh we are hopefully going to be doign one for 2k phones here |
| 03:03.57 | nny_1 | imagine that |
| 03:04.34 | jwh | rather you that me ;p |
| 03:05.21 | teknoprep | its not the amount of phones that makes it complicated |
| 03:05.40 | teknoprep | its when you get into queues / IVR's / recording / dial plans |
| 03:05.51 | teknoprep | phones just take time to add as an extension |
| 03:06.09 | teknoprep | if you have 2k phones on 1 queue that would be very simple |
| 03:07.41 | nny_1 | lol |
| 03:07.49 | nny_1 | well yeah true |
| 03:08.55 | teknoprep | i have been setting up my systems using freepbx |
| 03:09.07 | teknoprep | but i am tired of using a web gui and not knowing what is going on |
| 03:09.16 | teknoprep | but i am tired of reading right now |
| 03:09.35 | teknoprep | i need to get my box in my basement for my house setup on a purely asterisk machine |
| 03:09.41 | teknoprep | i think i will do this on sunday |
| 03:10.02 | teknoprep | i want to setup a colocated server for small business's we are working with |
| 03:10.10 | teknoprep | instead of selling them a one time solution for VoIP |
| 03:10.36 | teknoprep | we want to just install phones and maby an embeded PfSense router running on a small embeded device for QoS |
| 03:10.55 | teknoprep | then sell them there voip connections which are colocated elsewhere |
| 03:11.01 | teknoprep | since we have the customer base already |
| 03:11.16 | teknoprep | we would make monthly charges over the hi price installation charges |
| 03:11.27 | teknoprep | only thing is you can't run freepbx for this type of situation |
| 03:11.35 | teknoprep | so here i am learning asterisk |
| 03:11.56 | teknoprep | i do have to say tho i do not see many limitations of freepbx when using a single server for a med sized business |
| 03:12.25 | teknoprep | the only real limitation i have found with freepbx is when trying to run multiple sites on a single asterisk box using freepbx as a configuration point |
| 03:14.35 | nny_1 | yeah freepbx wouldn't really give you much in the way of config for that |
| 03:14.40 | nny_1 | i started with asterisknow |
| 03:14.55 | teknoprep | that wouldn't work either |
| 03:15.50 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 03:16.07 | *** join/#asterisk angom (n=Angel@201.170.49.106) |
| 03:16.12 | teknoprep | where is the asterisk db ? |
| 03:16.23 | teknoprep | when i tell it to write something to the DB where does it go? |
| 03:20.50 | nny_1 | yeah indeed it wouldn |
| 03:20.51 | nny_1 | 't |
| 03:24.48 | *** join/#asterisk jaxxan (n=jaxxan@202.70.125.111) |
| 03:24.55 | jaxxan | hey guys |
| 03:25.55 | jaxxan | any of you have any experience sending/receiving sms with Asterisk via SS7? |
| 03:26.43 | jaxxan | I find myself with two SMSC's, neither of which I can really use to dissect the contents of SMS and redirect to another server or database. |
| 03:27.34 | jaxxan | I'm thinking about using Asterisk to receive an SMS from a mobile user, verifying the contents of the message somehow and send back an acknowledgement of some sort. |
| 03:53.16 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 03:55.55 | *** part/#asterisk nny_1 (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 03:58.06 | *** join/#asterisk [gnubie] (n=[gnubie]@cm71.gamma182.maxonline.com.sg) |
| 04:04.47 | *** join/#asterisk saftsack (n=saftsack@p4FC75AE9.dip.t-dialin.net) |
| 04:17.09 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 04:19.37 | *** join/#asterisk IgorG (n=FeedomPa@host-90-188-188-119.pppoe.omsknet.ru) |
| 04:23.48 | piper69 | i was able to downgrade my Cisco ATA186 to v3.1.0 atasip from v3.1.1 atasccp. i am also able to access http://192.168.1.100/dev i am not able to get H.323 to work , can someone help me please |
| 04:34.14 | [gnubie] | anyone here uses the voipuser.org inbound service? |
| 04:50.30 | Majost | <PROTECTED> |
| 04:50.49 | Majost | is there another command I should use? |
| 04:56.23 | Nugget | Majost: SendDTMF |
| 04:59.02 | Majost | ah |
| 04:59.07 | Majost | thanks |
| 05:07.54 | piper69 | Majost: is it gated community |
| 05:08.16 | Majost | Apartment complex, but yes. |
| 05:08.37 | piper69 | Majost: but if you called the gate number you can't open it |
| 05:08.45 | piper69 | Majost: i tried that |
| 05:09.02 | piper69 | when someone call me i press 9 to open |
| 05:09.05 | Majost | Yeah.. I want to setup an access code bypass in case I forget my keys |
| 05:09.06 | Majost | hehe |
| 05:09.42 | piper69 | Majost: i lost you ? |
| 05:10.16 | Majost | As in... if I dial a valid access code, it will open the gate |
| 05:10.21 | piper69 | Majost: i have mine routed to my cellphone |
| 05:10.52 | piper69 | Majost: maybe your is diffrent ?! |
| 05:11.29 | Majost | nah... I just forget my keys and phone in the apartment from time to time, and the gate closes behind me. |
| 05:11.30 | Majost | heh |
| 05:14.46 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 05:14.48 | piper69 | hire someone to open it for you |
| 05:14.49 | piper69 | lol |
| 05:14.54 | Majost | hah |
| 05:16.14 | piper69 | will be like that other joke, that someone hired a miged and put him inside his car just incase if he locked his keys |
| 05:17.01 | Majost | never heard it. heh |
| 05:18.27 | piper69 | or the guy that bought a new car and was scared someone will steal the new car, so he bought a car alarm and wire it to a bomi3 |
| 05:19.32 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 05:20.38 | piper69 | btw , do you guys think if an ATA works with 12DCV 1A , and i connected with 12DCV 500mA will work fine |
| 05:24.18 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
| 05:26.12 | Majost | Probably wont power on... or will do weird things |
| 05:26.38 | Majost | I have done that with small switches with those results. |
| 05:26.45 | Majost | but you never know. |
| 05:59.47 | *** part/#asterisk [gnubie] (n=[gnubie]@cm71.gamma182.maxonline.com.sg) |
| 06:21.43 | *** join/#asterisk vn (n=nostalge@modemcable083.222-80-70.mc.videotron.ca) |
| 06:22.01 | vn | hi, anyone knows of a smartphone that supports SIP? |
| 06:22.22 | vn | or one on which I can use voip with SIP while connected on a 802.11b/g wlan |
| 06:23.26 | vn | and uhm does asterisk has some ATS-like functionnality? |
| 06:34.55 | *** join/#asterisk Maliuta (n=nikolai@119.11.102.69) |
| 06:48.48 | *** join/#asterisk Maliuta_ (n=nikolai@119.11.98.164) |
| 06:54.54 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 07:02.33 | *** join/#asterisk Maliuta_ (n=nikolai@119.11.98.164) |
| 07:07.25 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 07:31.15 | *** part/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 08:02.20 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 08:26.49 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 08:46.37 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 09:02.06 | *** join/#asterisk af_ (n=getsmart@88-149-240-60.dynamic.ngi.it) |
| 09:17.01 | *** join/#asterisk Maliuta (n=nikolai@119.11.98.84) |
| 09:20.39 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 10:03.51 | *** join/#asterisk mamep (i=fallen@helios.edu.uoc.gr) |
| 10:04.33 | mamep | how can i connect two numbers using asterisk ..... like a conference |
| 10:04.35 | mamep | ? |
| 10:05.06 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 10:05.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 10:12.17 | *** join/#asterisk Maliuta (n=nikolai@119.11.101.2) |
| 10:18.56 | *** part/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 10:20.22 | *** join/#asterisk becks` (n=sdlfkjfs@41-183.5-85.cust.bluewin.ch) |
| 10:23.21 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 10:25.16 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 10:28.46 | becks` | somebody knows why there is no good open-source sip client? |
| 10:30.43 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 10:37.13 | *** join/#asterisk Psychobilly (n=Fuzz@online1.ioa.forthnet.gr) |
| 10:51.13 | Psychobilly | hello, can anyone suggest me some h323 providers? |
| 10:55.54 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 11:02.51 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 11:21.27 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 11:21.34 | *** join/#asterisk cjk (n=cjk@d90-129-39-85.cust.tele2.lu) |
| 11:22.18 | cjk | hi, i would like when my phone rings to play a sound to the calling party after X seconds (please hold the line), but my phone should keep ringing. any hint to put me on the right track? |
| 11:23.34 | mvanbaak | use a queue |
| 11:30.30 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 11:44.22 | *** join/#asterisk MacWinner (n=chatzill@74-33-175-68.dsl1-merch.roc.ny.frontiernet.net) |
| 11:45.29 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 11:46.05 | cjk | mvanbaak, ok, thanks |
| 11:49.34 | *** join/#asterisk guillote_GNU (n=guillote@host216.200-82-56.telecom.net.ar) |
| 12:03.39 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:06.45 | MacWinner | what's the best iax softphone? |
| 12:08.48 | *** join/#asterisk arctanx (n=VK7NML@pdpc/supporter/base/arctanx) [NETSPLIT VICTIM] |
| 12:09.38 | *** join/#asterisk squigly (n=bdeluca@cm58.psi170.maxonline.com.sg) |
| 12:18.44 | *** join/#asterisk shido6 (n=shido6@74-130-55-168.dhcp.insightbb.com) |
| 12:23.13 | *** join/#asterisk ronr_ (n=ron@82-170-109-196-static.dsl.ip.tiscali.nl) |
| 12:24.03 | ronr_ | hi, I'm getting a Error: missing /dev/zap error, I already ran make install-udev and restarted udev (debian etch), but no succes, any ideas? |
| 12:33.47 | *** join/#asterisk Winkie (n=urmom@general-kt-195.t-mobile.co.uk) |
| 12:35.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 12:41.00 | *** join/#asterisk Schumie (n=Steve@212.183.134.208) |
| 12:44.58 | MacWinner | in my demo install of asterisk, when i dial an unknown extension, the "s" extension is not matching.. but if I dial the 1000 demo extension, it jumps to the "s" extension fine |
| 12:45.14 | MacWinner | is the "s" extension supposed to match all unknown extensions? |
| 12:45.45 | ronr_ | found it (I was compiling zaptel with kernel headers that were slightly off) |
| 12:46.04 | ronr_ | MacWinner: no, s means 'the next priority' |
| 12:47.02 | MacWinner | oh, is the info at http://www.voip-info.org/wiki/index.php?page=Asterisk+s+extension innacurate? |
| 12:47.07 | MacWinner | or am i reading it incorrectly? |
| 12:48.48 | ronr_ | MacWinner: no, I read it wrong and was confused with n |
| 12:49.17 | MacWinner | oh.. ok.. damn |
| 12:50.42 | *** join/#asterisk ozus (n=ozus@202.77.104.197) |
| 13:00.19 | ronr_ | I got a BBned ISDN 15 E1, anyone here knows what my zaptel.conf should look like and how I can test if zaptel is configured correctly? |
| 13:09.22 | *** join/#asterisk Washy (i=Washy@gateway/tor/x-4728cb7ac6fe2d81) |
| 13:09.24 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
| 13:09.41 | Washy | can you tunnel VoIP through ssh or a VPN? |
| 13:11.04 | Greek-Boy | ofcourse |
| 13:11.14 | Greek-Boy | its actually adviseable to do so Washy |
| 13:11.24 | Greek-Boy | encryption |
| 13:12.06 | Washy | Well unfortunately I'm calling PTSN phones so that's not avail but that's not Y I wanna |
| 13:13.04 | Greek-Boy | well if you are interconnecting asterisk boxes via IAX2 or using a SIP provider, VPN tunnels are a plus I would say |
| 13:14.25 | Washy | I am using a SIP provider |
| 13:14.36 | Washy | for SIP to PTSN service |
| 13:15.17 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 13:15.30 | Greek-Boy | I think that most SIP providers do not offer VPN access |
| 13:15.35 | Greek-Boy | but u can try your luck |
| 13:17.34 | Washy | I know, I wanted to route my calls through another host |
| 13:26.25 | shido6 | go for it washy. :) |
| 13:26.41 | shido6 | go to another host then that host terminates the call through another host? |
| 13:34.58 | *** join/#asterisk bantu (n=Miranda@p54A33205.dip0.t-ipconnect.de) |
| 13:40.21 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 13:42.16 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 13:55.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 13:58.27 | squigly | so imtrying to figure out why my phone keeps on hanging up |
| 13:58.34 | squigly | i keep on getting lines like |
| 13:58.51 | squigly | Retransmitting #5 (NAT) to XXX.XXX.XXX.XXX |
| 13:58.52 | *** join/#asterisk adjohn (n=adjohn@p5087-ipad408marunouchi.tokyo.ocn.ne.jp) |
| 13:59.22 | squigly | when i run sip debug |
| 13:59.28 | squigly | any idea where to start to figure this out |
| 14:00.50 | mvanbaak | ~sipnat |
| 14:00.51 | jbot | [~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 |
| 14:01.07 | mvanbaak | go read that |
| 14:01.18 | squigly | thanks |
| 14:06.21 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:09.02 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 14:15.39 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 14:19.04 | squigly | mvanbaak, so i possibly need to add the qualify directive to my sip.conf for those handsets behind nat? |
| 14:22.43 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:24.05 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:33.20 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:44.07 | MacWinner | teliax or vitelity for origination? |
| 14:48.46 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 14:56.54 | mosty | squigly, with qualify=yes asterisk will send packets periodically so the nat doesn't timeout |
| 14:58.29 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 14:58.51 | squigly | do you some times have to fiddle the quallify value? |
| 14:59.39 | mosty | i've never had to, i just use the default qualify=yes |
| 15:00.27 | squigly | i have handsets on the local lan and i am getting unreachable messages from them |
| 15:01.49 | squigly | is a call thats going out through asterisk getting routed to the asterisk server? |
| 15:03.08 | squigly | <PROTECTED> |
| 15:03.15 | squigly | i see these? what does that mean |
| 15:05.04 | mvanbaak | that means the phone had a sip ping time of more then 2000 ms |
| 15:05.16 | mvanbaak | if that happens, asterisk marks them as UNREACHABLE |
| 15:05.23 | squigly | thats asterisk talking to it directly? |
| 15:05.34 | mvanbaak | that you cannot see by that line |
| 15:05.47 | mvanbaak | it does not tell what ip it talks to etc. you can only see that in sip debug |
| 15:05.50 | squigly | what should i look for? |
| 15:05.56 | squigly | im in sip debug |
| 15:06.08 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:06.08 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:07.18 | squigly | the peer 9001 is a local peer its siting on the desk next to my server |
| 15:08.02 | mvanbaak | ok |
| 15:08.24 | mvanbaak | then your network is borked, because on a lan you should never see sip pingtimes that high |
| 15:08.29 | mvanbaak | or is it a wifi phone ? |
| 15:08.37 | *** join/#asterisk SKu||LL (n=Pada-@196.203.51.37) |
| 15:08.43 | squigly | its a ethernet attached phone |
| 15:08.49 | SKu||LL | hi there |
| 15:08.51 | SKu||LL | :) |
| 15:08.51 | squigly | can i show you the sip debug? |
| 15:09.01 | squigly | im begining to think this is a broke switch |
| 15:09.08 | squigly | (this all worked a little while ago) |
| 15:09.09 | mvanbaak | sure, but wont matter |
| 15:09.32 | mvanbaak | because if it's on a lan and switching between reachable/unreachable something is wrong with your lan |
| 15:09.47 | squigly | i have another switch |
| 15:09.48 | squigly | brb |
| 15:09.50 | mvanbaak | either the lan is saturated (virus or worm or something) or the connection is borked |
| 15:12.44 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 15:13.00 | squigly | hmm im thinking there is some thing wrong with the voip device, its a billion modem |
| 15:13.07 | tzafrir_home | or the cable is bad |
| 15:13.10 | squigly | as soon as i hang up it goes good again |
| 15:13.18 | squigly | yeah let me swap cables |
| 15:13.41 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 15:14.17 | blitzrage | morning all |
| 15:14.42 | mvanbaak | hey blitzrage |
| 15:14.50 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 15:14.52 | SKu||LL | hi blitzrage |
| 15:15.19 | blitzrage | hey hey. How goes this fine Saturday? |
| 15:15.24 | squigly | are some voip devices just not able to handle the request that comes from qualify? |
| 15:15.51 | mvanbaak | squigly: only really weird devices |
| 15:15.55 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:15.57 | blitzrage | squigly: some... but it's rare |
| 15:16.10 | blitzrage | asterisk just cares about a response... not even what it really gets back |
| 15:16.25 | blitzrage | even a 489 Bad Event is ok |
| 15:16.30 | *** join/#asterisk ManxPower (n=manxpowe@94.sub-75-201-42.myvzw.com) |
| 15:16.53 | squigly | ok swapped cables |
| 15:17.40 | SKu||LL | blitzrage are you a core team developer in the @ team ? |
| 15:17.52 | blitzrage | no, I write documentation and do implementations |
| 15:18.06 | SKu||LL | that's good nice too meet you |
| 15:18.10 | blitzrage | I'm one of the few who actually know how to use Asterisk :) |
| 15:18.20 | blitzrage | you too |
| 15:18.51 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:18.51 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:19.05 | lmadsen | gah.... Q is too close to Tab |
| 15:19.35 | SKu||LL | i know a little but thing i am doin' those days is tryin to integrate a win app to administrate @ i know that it's exist many ready to use things but it's just an idea like a lot evoluating around the @ project |
| 15:20.34 | SKu||LL | also interested in the new app_chanspy in the 1.4 but still not try the -w -W |
| 15:20.44 | SKu||LL | v V |
| 15:21.53 | Qwell | @ project? |
| 15:22.50 | SKu||LL | i was just meaning asterisk project |
| 15:23.11 | squigly | so, all i can think of is that the hardware here is broken, damnit |
| 15:24.41 | *** join/#asterisk masus (n=ethemc@88.248.14.186) |
| 15:25.06 | mvanbaak | squigly: try qualify=4000 |
| 15:25.56 | squigly | i have a sip on my mobile ill try with that tomororw but tonight i cant handle typing alll the details |
| 15:26.07 | masus | hi all, does anyone know how to replace a string in extensions. EXAMPLE SET(TEST=0049214?0049:0) so the result will be 0214 |
| 15:26.38 | masus | can anybody help ? |
| 15:28.36 | ManxPower | masus: read doc/channelvariables.txt in the Asterisk source directory. |
| 15:28.45 | squigly | do local extention settings override global settings? |
| 15:28.58 | ManxPower | notice the :x and :x:y format of channel variables |
| 15:29.08 | ManxPower | squigly: your question is too vague |
| 15:29.21 | masus | ok i'll see |
| 15:29.32 | masus | thanks |
| 15:32.34 | masus | i have only /doc/README.variables |
| 15:32.43 | masus | and there is no information about replace |
| 15:33.14 | ManxPower | masus: in 1.2 it is called README.variables. You do not want to replace, you want to remove parts of the variable |
| 15:34.02 | masus | hmm |
| 15:34.07 | masus | so we cant replace ? |
| 15:34.12 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 15:34.35 | masus | because the length is not always the same |
| 15:34.50 | masus | so i cant cut the first 4 characters |
| 15:35.02 | masus | sometime it's 5 sometime 4 |
| 15:35.50 | masus | so i need something like this -> SET(TEST=replace(0049214,0049,0)); and the result will be 0214 |
| 15:35.51 | masus | .P |
| 15:35.53 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 15:36.35 | ManxPower | Set(TEST=${BOB:3:4}) |
| 15:36.39 | *** join/#asterisk FlatFoot (n=chatzill@80.88.218.4) |
| 15:37.03 | ManxPower | you are removing 0049 |
| 15:37.15 | ManxPower | so it would be Set(TEST=${BOB:4}) |
| 15:37.19 | ManxPower | this is not rocket science |
| 15:38.26 | ManxPower | then you will have to look at much more complex functions. |
| 15:38.41 | ManxPower | "show applications" and "show functions" |
| 15:38.50 | masus | ok |
| 15:39.11 | masus | REGEX |
| 15:39.18 | masus | i have found what i want thanks |
| 15:42.02 | *** part/#asterisk masus (n=ethemc@88.248.14.186) |
| 15:47.30 | *** part/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:52.02 | *** part/#asterisk ManxPower (n=manxpowe@94.sub-75-201-42.myvzw.com) |
| 15:54.17 | squigly | so it was buggy software on my router, thats connecting to the asterisk, which is now fixed combined with the loop number i was calling not being a loop and it hanging up on me! |
| 16:09.23 | rob0 | If Asterisk isn't rocket science, how come Digium is based in a NASA town? |
| 16:09.36 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 16:13.19 | *** join/#asterisk maache (i=maache@81.52.163.202) |
| 16:17.24 | squigly | is there a howto on setting up registration? |
| 16:17.34 | squigly | so people can dial into my DID? |
| 16:28.59 | *** join/#asterisk timeshell (n=Khoja@206.248.136.108) |
| 16:29.02 | timeshell | Greetings |
| 16:29.43 | timeshell | I am a new user of asterisk-gui as up until now I've been using CLI and .conf files to manage my asterisk server 1.2 |
| 16:29.55 | timeshell | I just upgraded to 1.4 and am trying to install/use asterisk=gui |
| 16:30.16 | timeshell | Does the asterisk-gui use your previous .conf files without modification? |
| 16:30.55 | timeshell | And second, although my permissions in the manager.conf are set correctly, the webpage keeps telling me they are not when I try to go to the basiccfg.html page |
| 16:31.04 | timeshell | So, I'm looking for a little help here. |
| 16:31.11 | timeshell | Please. :) |
| 16:36.01 | lmadsen | jbot: tell squigly about book |
| 16:36.37 | mosty | timeshell, we don't support asterisknow here, try #asterisk-gui |
| 16:36.57 | timeshell | I don't have asterisknow |
| 16:37.00 | timeshell | :p |
| 16:37.02 | timeshell | just asterisk |
| 16:37.07 | timeshell | self compiled |
| 16:37.16 | timeshell | but thanks |
| 16:37.19 | mosty | i thought you said you were trying to use asterisk-gui |
| 16:37.23 | timeshell | Yes |
| 16:37.32 | timeshell | I installed it long after I installed asterisk |
| 16:37.33 | mosty | that's asterisknow, isn't it? |
| 16:37.44 | timeshell | Isn't asterisknow a prebuilt appliance? |
| 16:37.48 | timeshell | I dunno |
| 16:37.56 | timeshell | I haven't done anything specifically for asterisknow |
| 16:37.56 | mosty | i think they're the same thing |
| 16:38.05 | timeshell | ok |
| 16:38.15 | *** join/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 16:38.19 | mosty | well beware that asterisk-gui will probably overwrite all your config files later if you use that |
| 16:38.29 | piper69 | good morning all |
| 16:38.32 | timeshell | I've already backed em up |
| 16:38.35 | timeshell | BUt so far it hasn't |
| 16:39.16 | mosty | re manager.conf, are the file permissions ok? and the permissions inside the file? |
| 16:39.32 | piper69 | what is a good free voip for incoming calls |
| 16:39.34 | timeshell | I was wondering about that |
| 16:39.34 | mosty | also check the ownership |
| 16:39.42 | timeshell | Inside the file they are correct |
| 16:39.47 | timeshell | Everything is running as root |
| 16:39.52 | timeshell | And the file permissions are root |
| 16:39.57 | lmadsen | piper69: free world dialup -- there is no free providers for connecting to the telephone network |
| 16:40.06 | timeshell | (been meaning to change that btw) |
| 16:40.22 | mosty | timeshell, what user is asterisk running as? |
| 16:40.35 | timeshell | piper69: Depends on where you are |
| 16:40.37 | piper69 | lmadsen: i meant to call from my landline to voip |
| 16:40.43 | piper69 | timeshell: usa |
| 16:40.46 | lmadsen | that's what I just said |
| 16:40.50 | lmadsen | doesn't exist |
| 16:40.56 | lmadsen | telephony network == money |
| 16:40.56 | timeshell | piper69: none that I know of there. I have a free UK and Rome number |
| 16:41.19 | lmadsen | get a pre-paid account for 1.1 cents a minute |
| 16:41.40 | piper69 | timeshell: what about free digits |
| 16:41.41 | timeshell | To call into VOIP there are free gateway numbers |
| 16:42.10 | timeshell | And then you call the VOIP number |
| 16:42.24 | mosty | piper69, it will cost you money to connect to the regular telephone network |
| 16:42.49 | timeshell | mostly: That really depends on haow |
| 16:42.51 | timeshell | how* |
| 16:43.07 | timeshell | Skype Pro is good for outgoing very cheap |
| 16:43.10 | timeshell | With chanskype |
| 16:43.24 | Qwell | chan_skype is junk |
| 16:43.27 | timeshell | Incoming you can use a PSTN gatway |
| 16:43.36 | mosty | show me a telephone company that will give you free access to the regular company, and i will show you a company going out of business |
| 16:43.37 | timeshell | And then dial the SIP # from there |
| 16:43.57 | mosty | regular telephone network, rather |
| 16:44.01 | timeshell | Qwell: I use it and it works very well for me. All my home phones go through it and most of my calls are long distance through Skype |
| 16:44.16 | piper69 | mosty: i don't want to call i want people to be able to call me |
| 16:44.21 | timeshell | Qwell: It is the only suitable asterisk channel to skype right now so I'd hardly call it junk |
| 16:44.34 | Qwell | except that it isn't even suitable |
| 16:44.42 | Qwell | it's complete junk |
| 16:44.45 | mosty | piper69, that still costs the telco money- they wont give that to you for free |
| 16:45.03 | timeshell | Qwell: Again, I say you're wrong. It works which is good enough for me. The call quality is acceptable. |
| 16:45.10 | lmadsen | I think it's ridiculous people aren't willing to pay the 1.1 cents a minute several good ITSP's provide |
| 16:45.24 | Qwell | acceptable doesn't make it good, by any means |
| 16:45.28 | lmadsen | timeshell: you can't say he's wrong -- you can disagree though |
| 16:45.41 | lmadsen | just because you think you're right doesn't make Qwell wrong (and vice-versa) |
| 16:45.54 | timeshell | Imadsen: Same diffference. |
| 16:46.01 | Qwell | lmadsen: Qwell is always right :p |
| 16:46.36 | timeshell | Qwell: Show me an alternative the chanskype that works as an asterisk channel and I'll happily consider it. |
| 16:46.36 | lmadsen | timeshell: not really, unless you really are wrong |
| 16:46.52 | Qwell | timeshell: there isn't one |
| 16:46.55 | lmadsen | and 'same difference' is a misnomer |
| 16:46.57 | timeshell | lmadsen: Well in this case it's not really relevent |
| 16:47.22 | timeshell | I don't concern myself with symantics when you really already understand what I meant |
| 16:47.22 | piper69 | have you tried sipnumber.com |
| 16:47.32 | piper69 | its free to recive phone calls |
| 16:48.01 | lmadsen | Qwell: well... I can't say you're not a newb :) |
| 16:48.08 | timeshell | Qwell: Hence I rest my case. I agree it's not the best way to implement a channel, however, it does the job. |
| 16:48.08 | lmadsen | err... can* |
| 16:48.37 | Qwell | timeshell: can it run without X? |
| 16:48.41 | timeshell | Qwell: And since nothing else does the job, it in itself is the best available. |
| 16:49.02 | lmadsen | X on an Asterisk server... ya that *must* make it good |
| 16:49.11 | timeshell | Qwell: not to my knowledge, I use X on it anyway. |
| 16:50.19 | timeshell | lmadsen: I only run a single main server at home. I'm an energy mizer. While in practice it's not the best, it works with minimal resource impact. |
| 16:50.39 | Qwell | you'd use less energy without X |
| 16:50.48 | mosty | piper69, i'm skeptical |
| 16:50.49 | piper69 | lmadsen: timeshell : let me put it this way , can i get a free SIP and program my ata to use it |
| 16:51.01 | piper69 | mosty: what do you mean |
| 16:51.04 | lmadsen | "a free SIP" does not compute |
| 16:51.06 | Qwell | piper69: to make/receive calls? no |
| 16:51.14 | timeshell | Qwell: I use the server for other things as well, not just asterisk. X will be on it anyway therefore that's not really an argument |
| 16:51.15 | piper69 | Qwell: to recive |
| 16:51.42 | mosty | piper69, that site does not say specifically what sort of telephone number they will give you |
| 16:52.22 | piper69 | mosty: yes, and the only downside is that you will get a number that is not in you local area |
| 16:52.53 | mosty | if they give you a mongolian mobile number, will your friends be willing to pay the cost of calling you? |
| 16:53.12 | timeshell | piper69, let's go private |
| 16:53.24 | lmadsen | ugh.... smells like sex in here |
| 16:54.11 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:55.01 | mosty | piper69, the website is so light on details, i would be very skeptical |
| 17:05.05 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:07.02 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:31.56 | MacWinner | so i setup a voipjet peer to their test server, but how exactly do you see if it peered correctly? iax2 show peers says it's unmonitored |
| 17:36.03 | timeshell | show sip registry |
| 17:47.58 | MacWinner | so the default playback of audio file is a little choppy.. and pointers on tweaks to make this better? is it just codec/bandwidth related? |
| 17:53.21 | carrar | Make sure nothing else is running on the machine |
| 17:53.38 | carrar | make sure your network is fast between both points and not oversaturated with packets |
| 17:54.02 | carrar | Make sure the file is good to begin with |
| 17:54.30 | carrar | Best to keep the phone audio format the same as the file being played so there is no transcoding |
| 17:54.38 | carrar | or vs versa |
| 17:55.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:58.02 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 18:00.20 | MacWinner | cool.. i was just testing out the default asterisk welcome prompt |
| 18:00.37 | MacWinner | bandwidth should be fine.. maybe it's PC issue |
| 18:01.02 | MacWinner | i tried out voipjet and it seems to work.. but it seems to bill a fraction of a cent for calls that are initiated but not picked up |
| 18:08.01 | vn | does asterisk has some ATS-like functionnality? |
| 18:11.10 | *** join/#asterisk ariel_ (n=ariel_@server.onesteppapers.com) |
| 18:11.37 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 18:11.39 | teknoprep | hey |
| 18:11.49 | teknoprep | does anyone know how to turn down the input gain on a 7940g cisco phone |
| 18:15.40 | *** join/#asterisk abatista (n=ariel_@70-46-87-154.ftl.fdn.com) |
| 18:22.42 | teknoprep | anyone? |
| 18:22.50 | teknoprep | cisco 7940 input gain adjustment? |
| 18:22.57 | teknoprep | using the SIPDefault.cnf file would be great |
| 18:24.42 | piper69 | guys i still need help configure my Cisco ATA 186 |
| 18:26.08 | ariel_ | wow I have not seen a ATA 186 in along time. Don't even remember them much. But there is allot of info on the wiki about them |
| 18:26.27 | ariel_ | teknoprep, can't help you with the Cisco. I stopped using them over 3 years ago. |
| 18:28.39 | piper69 | ariel_: i don't think so, i am having hard time with this ata |
| 18:29.40 | piper69 | i was able to upgrade/downgrade from v3.1.1 atasccp to v3.1.0 atasip |
| 18:31.14 | piper69 | ariel_: this is my first time to use Cisco , i am able to access the web config , it only a one page not tabs . no? |
| 18:32.20 | ariel_ | don't remember it much like I said last time I used them was over 3 years ago. |
| 18:32.54 | teknoprep | piper69, i would recommend using tftp server to configure your cisco devices with the proper xml / .cnf file |
| 18:33.02 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 18:33.44 | piper69 | teknoprep: like i said i was able to upgrade/downgrade from v3.1.1 atasccp to v3.1.0 atasip |
| 18:34.21 | piper69 | teknoprep: i used a tftp to do so, i am having a hard time to get the H.323 |
| 18:34.49 | teknoprep | hey piper69 do you know the config for input gain on the 7940 |
| 18:34.59 | teknoprep | through the .cnf |
| 18:36.57 | *** join/#asterisk abatista (n=ariel_@server.onesteppapers.com) |
| 18:39.14 | vn | anyone knows of a smartphone that supports SIP? |
| 18:39.27 | vn | or how can I use some voip on a smartphone? |
| 18:40.30 | *** join/#asterisk ToTo (n=ToTo@87.2.138.122) |
| 18:40.31 | abatista | smartphone? |
| 18:41.00 | Qwell | vn: some of the Nokia's, I think |
| 18:45.00 | teknoprep | there is software you can run on smartphones for sip connections |
| 18:47.00 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 18:49.45 | _x86_ | can someone send me a test fax? |
| 18:49.54 | _x86_ | 1-309-693-6737 |
| 18:55.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 18:56.50 | teknoprep | yeah sure |
| 18:57.25 | teknoprep | its on its way |
| 18:57.55 | teknoprep | its on page 2 |
| 19:00.50 | *** join/#asterisk asdx (n=diego@adsl-146-86.click.com.py) |
| 19:00.57 | teknoprep | so not one person here knows |
| 19:01.01 | teknoprep | damit |
| 19:01.11 | teknoprep | i'll be back later |
| 19:01.18 | mvanbaak | mail cisco support |
| 19:10.05 | vn | Qwell: thanks |
| 19:13.40 | piper69 | sorry i was away |
| 19:16.57 | drmessano | Hey piper69, I have a solution to your ATA problem |
| 19:31.04 | *** join/#asterisk katsuodo (n=musashi@pool-71-187-107-7.nwrknj.east.verizon.net) |
| 19:31.09 | katsuodo | hallo |
| 19:31.50 | katsuodo | have pc phone, ip phone, and asterisk server behind nat |
| 19:32.24 | katsuodo | asterisk tdm400p card and digital phone attached |
| 19:33.33 | katsuodo | sip phones are registered on asterisk (rtp) traffic and I am able to dial extensions internally from analog phone to other phones |
| 19:33.44 | katsuodo | however sip to sip direct dial phone rings |
| 19:33.49 | katsuodo | but no voice |
| 19:34.14 | katsuodo | checked ports and all well |
| 19:34.19 | katsuodo | any suggestions |
| 19:38.13 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 19:54.32 | katsuodo | hallo [TK]D-Fender |
| 20:01.01 | [TK]D-Fender | ~sipnat |
| 20:01.02 | jbot | [~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 |
| 20:01.37 | katsuodo | yes I read this |
| 20:01.57 | [TK]D-Fender | ~pb |
| 20:01.58 | jbot | [~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 |
| 20:01.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 20:02.22 | [TK]D-Fender | katsuodo, Show your sip.conf masking only passwords |
| 20:02.32 | katsuodo | okay |
| 20:02.37 | katsuodo | one moment |
| 20:09.34 | katsuodo | [TK]D-Fender http://pastebin.ca/828880 |
| 20:11.01 | [TK]D-Fender | katsuodo, if your * is behind NAT you have shown NONE of the settings the quide tellso you to do. Go read again. You didn't even show a complete [general] section. |
| 20:11.26 | katsuodo | one moment |
| 20:17.09 | [TK]D-Fender | ajksdhjksladhf |
| 20:17.15 | [TK]D-Fender | can't type for beans today... |
| 20:17.25 | katsuodo | yes * is behind nat and what you see at the top is the general section just did not place the [general] context on the paste |
| 20:18.37 | [TK]D-Fender | katsuodo, then you have not followed the guide at all. |
| 20:19.06 | [TK]D-Fender | katsuodo, Go read it again till your eyes bleed because it looks like you haven't done ANY of what it tells you to do. |
| 20:19.08 | [TK]D-Fender | ~sipnat |
| 20:19.09 | jbot | [~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 |
| 20:19.12 | katsuodo | okay let me go back and read and again, thanks "Sensei" |
| 20:19.43 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:19.43 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:23.26 | Majost | For some reason or another, when I use SendDTMF(9) I am not hearing the tone for 9 |
| 20:24.16 | Majost | Do I need to use a specific DTMF mode for SIP to do this? |
| 20:33.44 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 20:34.19 | WilliamK | anyone have a good place to find templates for the cisco 79xx series phones? |
| 20:42.03 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:42.59 | *** join/#asterisk nirz (n=nir@bzq-79-183-137-211.red.bezeqint.net) |
| 20:43.50 | lmadsen | Majost: would have to be inband dtmf to hear it. If it's sent out of band (info,rfc2833), then you don't hear it |
| 20:44.02 | Majost | ahhh |
| 20:44.16 | Majost | Thanks. =) |
| 20:45.09 | lmadsen | WilliamK: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
| 20:45.18 | lmadsen | google is your friend |
| 20:49.42 | *** join/#asterisk JerJer[mobile] (n=jj@207.203.89.1) |
| 20:53.10 | piper69 | how can i get H.323 in my ATA 186 |
| 20:53.50 | mvanbaak | get the correct firmware |
| 20:58.40 | WilliamK | yeah that page isn't showing what I'm wanting from what I saw |
| 20:58.58 | WilliamK | I just reflashed the phone from 7.4 to 8.2 and still can't make it talk to * |
| 21:01.55 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 21:02.02 | teknoprep | yo |
| 21:02.19 | teknoprep | does anyone here know how to lower the input gain of cisco 7940 phones ? |
| 21:02.21 | teknoprep | using sip |
| 21:04.17 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:04.18 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 21:08.49 | teknoprep | jfc i am about to go crazy with these echo on the cisco phones |
| 21:09.04 | teknoprep | i am pretty sure that all i need to do is lower the input gain |
| 21:09.21 | *** join/#asterisk zuez (i=steve@66.103.132.86) |
| 21:10.44 | [TK]D-Fender | teknoprep, Do you get echo directly between 2 of these phones? |
| 21:16.49 | zuez | It's possible to store a dynamic dialplan using ODBC/Realtime, correct? if I want to dynamically assign/update/remove extensions? |
| 21:17.03 | mvanbaak | correct |
| 21:17.17 | zuez | ok, cool |
| 21:17.25 | zuez | thanks mvan |
| 21:17.29 | mvanbaak | no problem |
| 21:18.37 | piper69 | mvanbaak: will you help me please , today is my 4th day trying to get it to work |
| 21:19.19 | teknoprep | [TK]D-Fender, sometimes |
| 21:19.38 | teknoprep | [TK]D-Fender, it never happens all the time |
| 21:19.51 | teknoprep | [TK]D-Fender, all the phones i had on the network are doing the same thing |
| 21:20.07 | [TK]D-Fender | teknoprep, and happens between 2 SIP phones in direct contact? |
| 21:20.22 | teknoprep | [TK]D-Fender, every phone on the network besides the cisco phones i was able to turn down the input gain and the echo was resolved |
| 21:20.28 | teknoprep | [TK]D-Fender, sometimes not always |
| 21:20.47 | teknoprep | [TK]D-Fender, and sometimes cisco phones don't echo when going outbound to the real world ... sometimes they do |
| 21:20.50 | teknoprep | [TK]D-Fender, i don't get it |
| 21:21.18 | [TK]D-Fender | teknoprep, Still sounds like a dodgy answer. These are calls only between direct SIP phones? |
| 21:21.46 | teknoprep | [TK]D-Fender, it happens between internal phones... |
| 21:21.59 | teknoprep | [TK]D-Fender, it also happens on outside lines to outside phones |
| 21:22.08 | [TK]D-Fender | teknoprep, Oh well. |
| 21:22.11 | teknoprep | [TK]D-Fender, but it doesn't happen all the damn time which is was is driving me nuts |
| 21:22.31 | teknoprep | [TK]D-Fender, i don't understand how that isn't exactly what you wanted to hear? |
| 21:23.05 | mvanbaak | piper69: I have no experience with h323 |
| 21:23.08 | [TK]D-Fender | "teknoprep> [TK]D-Fender, sometimes not always" |
| 21:23.18 | teknoprep | <teknoprep> [TK]D-Fender, it happens between internal phones... |
| 21:23.18 | teknoprep | <teknoprep> [TK]D-Fender, it also happens on outside lines to outside phones |
| 21:23.29 | drmessano | piper69: sell that thing on eBay.. 4 days is enough |
| 21:23.40 | drmessano | piper69: I suggest "Buy it now" |
| 21:23.49 | dezenten | h323 is the future |
| 21:23.57 | [TK]D-Fender | teknoprep, I like definitive answers like "Yeah, 2 x 7940's in seperate offices echo between each other" |
| 21:24.13 | [TK]D-Fender | dezenten, lol |
| 21:24.18 | dezenten | :) |
| 21:24.28 | teknoprep | [TK]D-Fender, right now... this problem only happens on the 7940's since i was able to lower the input gain on ALL the other phones in the office that are not 7940 |
| 21:24.36 | drmessano | yes, h323 is the next big thing |
| 21:24.45 | dezenten | h323 and fax |
| 21:24.50 | drmessano | I am working on making FreePBX work with GopherD |
| 21:25.07 | drmessano | Gopher.. Not dead yet |
| 21:26.26 | teknoprep | [TK]D-Fender, do you have another idea on the idea why i am getting echo on these phones? |
| 21:27.07 | [TK]D-Fender | teknoprep, If its direct between them, its their gains. Tweak your firmware |
| 21:27.20 | [TK]D-Fender | teknoprep, Or stop buying trouble hardware |
| 21:28.20 | teknoprep | how do i tweak the firmware ? |
| 21:28.46 | teknoprep | or even change the input gain |
| 21:29.01 | dezenten | lower the volume |
| 21:29.17 | teknoprep | dezenten, on the 7940... how would this be accomplished? |
| 21:29.33 | dezenten | teknoprep: i have no ide |
| 21:29.39 | teknoprep | dezenten, thanx |
| 21:29.49 | dezenten | but its usually the way to fix echo |
| 21:30.13 | dezenten | is that cisco 7940 ? |
| 21:30.30 | teknoprep | i don't know anyone on IRC or on google that knows how to lower the input gain on a cisco 7940 unless its using cisco call manager |
| 21:30.40 | dezenten | http://uwadmnweb.uwyo.edu/InfoTech/Services/departments/voip/79407960userguide.htm |
| 21:30.42 | teknoprep | and then to lower it you have to doit on the port your phone is plugged into |
| 21:30.47 | dezenten | first image button "12" |
| 21:31.01 | dezenten | press the left button |
| 21:31.04 | teknoprep | jfc |
| 21:31.09 | teknoprep | thats not input gain |
| 21:31.11 | teknoprep | thats output |
| 21:31.20 | dezenten | sorry |
| 21:31.25 | dezenten | but that might work aswell |
| 21:31.30 | teknoprep | no it doesn;t |
| 21:31.40 | *** join/#asterisk Bananaskin (n=Banana@user-5af0486a.wfd99.dsl.pol.co.uk) |
| 21:33.33 | WilliamK | hmmm, asterisk and cisco aren't playing nice |
| 21:33.43 | WilliamK | * is giving the phone a 401 unauthorized |
| 21:35.09 | teknoprep | nat=never |
| 21:35.13 | teknoprep | qualify=no |
| 21:35.16 | teknoprep | try that |
| 21:36.40 | WilliamK | doesn't work either |
| 21:37.28 | teknoprep | i am switching to polycom phones only |
| 21:37.32 | teknoprep | i am tired of cisco crap |
| 21:37.38 | WilliamK | I ordered 3 more for this client |
| 21:37.42 | teknoprep | have fun |
| 21:37.44 | WilliamK | I'm VERY sick of the 7900 |
| 21:37.46 | teknoprep | which phone are you using? |
| 21:37.52 | teknoprep | 7940 ? |
| 21:37.53 | WilliamK | 320 series |
| 21:37.56 | WilliamK | 7940g |
| 21:38.02 | teknoprep | yeah they are easy to setup |
| 21:38.13 | WilliamK | we have like 5 polycom 320s already |
| 21:38.18 | teknoprep | do you like them ? |
| 21:38.20 | WilliamK | no issues at all |
| 21:38.35 | WilliamK | we did do an upgrade to the latest firmware though |
| 21:38.59 | teknoprep | how are your speakerphone quality on those phones? |
| 21:39.10 | teknoprep | are they really that nice? |
| 21:39.11 | WilliamK | very good |
| 21:39.17 | teknoprep | how about any echo problems? |
| 21:39.21 | WilliamK | I can use them half way accross an office |
| 21:39.22 | WilliamK | none |
| 21:39.26 | teknoprep | ggz |
| 21:39.37 | teknoprep | what internet connection you using? |
| 21:39.47 | teknoprep | or are you using analouge trukns? |
| 21:40.18 | WilliamK | hey tek, I may have found something for you in regards to the gain |
| 21:40.22 | teknoprep | w0ot |
| 21:40.25 | teknoprep | show me |
| 21:40.26 | teknoprep | lol |
| 21:40.34 | WilliamK | covers dtmf |
| 21:40.38 | WilliamK | lemme just give you the URL |
| 21:40.44 | teknoprep | ok |
| 21:40.46 | WilliamK | http://www.trixbox.org/forums/vendor-specific-unmoderated/linksys-cisco/urgent-help-needed-please-weird-cisco-7940-tftp-com |
| 21:40.51 | WilliamK | I've been copy pasting from it |
| 21:40.54 | WilliamK | it works so far |
| 21:42.20 | WilliamK | I definitely see a diff in the registration from polycom to cisco |
| 21:42.41 | WilliamK | doesn't display the "extn" in quotes in the register statement |
| 21:42.46 | teknoprep | yeah thats DTMF |
| 21:43.25 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:44.03 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 21:44.14 | teknoprep | i am going crazy lol |
| 21:45.49 | WilliamK | :) |
| 21:51.47 | WilliamK | all I can say is the Cisco and * are fighting constantly, even on our working phones * is giving 401 unauthorized msgs in the SIP debug |
| 21:54.15 | WilliamK | oh nice, google shows I'm not the only user with this issue |
| 21:58.19 | Bananaskin | As a matter of intrest why not look at sccp for the ciscos under asterisk, thats what I do here |
| 21:59.33 | teknoprep | is sccp hard to setup for cisco under asterisk ? |
| 21:59.40 | WilliamK | I'd rather throw the phone out the window than change over to SCCP |
| 21:59.42 | WilliamK | :) |
| 21:59.48 | WilliamK | SIP or die |
| 21:59.50 | WilliamK | :) |
| 21:59.57 | Bananaskin | ar5e |
| 22:00.37 | Bananaskin | sccp sound quality is better and allows the retention of the functions which are lost when u use sip on the ciscos |
| 22:01.21 | *** join/#asterisk Coder365_ (n=me@wrlsmdm025.cbpu.com) |
| 22:01.44 | WilliamK | Poly fix Crisco come Monday |
| 22:01.46 | WilliamK | :) |
| 22:02.01 | WilliamK | it would be NICE however if * played nice with the Cisco's |
| 22:02.13 | teknoprep | Bananaskin, do you have a sccp how-to for asterisk with 7940 phones? |
| 22:02.13 | Coder365_ | I reloaded my asterisk config via "reload" on the asteriskCLI, and now everything quit workign |
| 22:02.23 | Coder365_ | is there any way to find out what I did and how to fix it |
| 22:02.40 | Bananaskin | teknoprep I can point you in the right direction and sup;lly the config etc |
| 22:02.49 | Bananaskin | what release of asterisk for a start ? |
| 22:02.52 | Bananaskin | 1.2 or 1.4 ? |
| 22:02.55 | Coder365_ | hold on |
| 22:03.46 | Coder365_ | 1.2.24 |
| 22:04.21 | Bananaskin | sorry Coder365_ my questions were directed at teknoprep |
| 22:04.25 | Coder365_ | oh |
| 22:04.27 | Coder365_ | okay :) |
| 22:07.36 | rob0 | In CCCP phone throw YOU out window |
| 22:07.50 | WilliamK | golly, I finally got it to register |
| 22:07.54 | WilliamK | too bad I can't get it to dialout |
| 22:08.06 | teknoprep | Bananaskin, 1.4 |
| 22:08.36 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 22:08.57 | teknoprep | Bananaskin, if you throw me a pastebin.com of the conf and tftp file that would be great |
| 22:09.18 | Bananaskin | k, nps, need to grab the 1.4 sccp file from a mate, as I run 1.2 here 2 secs |
| 22:09.27 | teknoprep | ok |
| 22:09.31 | Bananaskin | 2 mins |
| 22:10.09 | teknoprep | ok np |
| 22:14.18 | teknoprep | hey Bananaskin i am heading out soon |
| 22:14.26 | teknoprep | can you /query me with the info then |
| 22:14.29 | teknoprep | i will be back in about an hour |
| 22:14.47 | MacWinner | are there any example php scripts that i can use to make asterisk dial 2 numbers and cross connect them? |
| 22:14.59 | MacWinner | just need a simple web form to get going |
| 22:15.50 | WilliamK | looks like the nice callerid= line in sip.conf goofed it |
| 22:19.44 | *** join/#asterisk lesouvage (n=lesouvag@a80-126-75-151.adsl.xs4all.nl) |
| 22:25.49 | piper69 | who was looking for a Cisco 7940 / 7960 SIP Firmware |
| 22:27.29 | Majost | Is there a setting to increase the length of a DTMF tone over SIP? From what I have read, it sounds like I am going to have to do something incredibly hacky and just record a long inband tone and then play it back as needed. heh |
| 22:28.09 | WilliamK | piper, I got a few vers... what ver u got? |
| 22:29.12 | piper69 | WilliamK: me i am looking for the H.323 for Cisco ATA 186. but i thought someone here today was looking for the 7940 firmware and i found it while looking for mine |
| 22:29.24 | piper69 | WilliamK: do you have anything for the ATA 186 |
| 22:29.35 | WilliamK | nope |
| 22:29.46 | WilliamK | I quit using the 186 back when I had vonage |
| 22:29.50 | WilliamK | long time ago |
| 22:29.50 | Bananaskin | I have the sccp firmware for the 186 :) |
| 22:29.51 | WilliamK | :) |
| 22:32.06 | Bananaskin | piper69 what firmware you after for the ata-186? |
| 22:32.20 | Bananaskin | H323 ? |
| 22:34.28 | drmessano | piper69: eBay |
| 22:44.07 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:44.07 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:48.02 | *** part/#asterisk Coder365_ (n=me@wrlsmdm025.cbpu.com) |
| 23:16.39 | *** join/#asterisk cesar_CR (n=cr@celord.ice.co.cr) |
| 23:28.15 | *** join/#asterisk implicit (i=implicit@gateway/tor/x-3eb1f35e63d682f8) |
| 23:49.33 | *** join/#asterisk droemel (n=droemel@p548EAD7E.dip0.t-ipconnect.de) |
| 23:51.37 | *** join/#asterisk droemel (n=droemel@p548EAD7E.dip0.t-ipconnect.de) |