| 00:02.02 | *** part/#asterisk agx (n=badpengu@81-174-47-57.dynamic.ngi.it) |
| 00:10.46 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
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| 00:24.46 | *** part/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 00:29.59 | TrentCreek | Since we can desinate CID, what about CID name? |
| 00:31.49 | bjweeks | Depends... |
| 00:32.07 | TrentCreek | on what |
| 00:32.37 | bjweeks | Set(CALLERID(name)=Asterisk Rocks) should do the trick if your service supports it |
| 00:33.02 | bjweeks | VoIP or PSTN, different VoIP providers may or may not support it |
| 00:33.32 | TrentCreek | VOIP IAX |
| 00:33.43 | bjweeks | What provider? |
| 00:34.01 | TrentCreek | rapidvox |
| 00:34.38 | bjweeks | Not sure about them, you can ask them or just try it yourself |
| 00:34.51 | TrentCreek | i dont know how |
| 00:34.57 | bjweeks | Set(CALLERID(name)=Asterisk Rocks) |
| 00:35.07 | TrentCreek | thanks |
| 00:35.19 | bjweeks | or Set(CALLERID(all)="Asterisk Rocks" <18005551212>) |
| 00:35.42 | bjweeks | I'm assuming 1.4 but that *should* work with 1.2 |
| 00:35.55 | JT | i don't think you need quotes |
| 00:36.21 | bjweeks | good point |
| 00:38.13 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
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| 00:48.44 | TrentCreek | what if I have multiple DIDs and want * to answer any incoming call unless it is specified to go to another extention? |
| 00:48.50 | TrentCreek | If this correct? exten => _X.,1,Answer |
| 00:48.50 | TrentCreek | exten => _X.,2,Wait,2 |
| 00:50.28 | *** join/#asterisk jks (n=jks@0x503e4c12.abnxx10.adsl-dhcp.tele.dk) |
| 00:51.55 | jks | anyone has an opinion on whether to prefer alaw or ulaw encoding for fax reception? |
| 00:53.03 | TrentCreek | outlaw would be best ;-) |
| 00:53.32 | bjweeks | _X. => Answer(); (do stuff later) |
| 00:53.45 | bjweeks | _NUMBER => Answer() (do stuff) |
| 00:53.51 | jks | TrentCreek, hehe? |
| 00:54.08 | bjweeks | number being the other number, or I think that is what you meant |
| 00:59.08 | TrentCreek | well lets say I have 100 DIDs, for example, that would be a tedious task of punching in all those humbers |
| 00:59.29 | TrentCreek | and each one have its on dial plan |
| 00:59.48 | JT | use a pattern match? |
| 01:00.08 | TrentCreek | _NxNXXXXXX? |
| 01:00.38 | JT | no lower case x |
| 01:01.10 | TrentCreek | so then exten => _X.,1,Answer |
| 01:01.10 | TrentCreek | exten => _X.,2,Wait,2 |
| 01:01.10 | TrentCreek | exten => _X.,3,DeadAGI,a2billing.php |
| 01:01.10 | TrentCreek | exten => _X.,4,Wait,2 |
| 01:01.10 | TrentCreek | exten => _X.,5,Hangup |
| 01:01.35 | bjweeks | JT: caps always worked for me :/ |
| 01:01.39 | TrentCreek | okay..so I change those to NxxxNxxxxxxx |
| 01:01.57 | JT | wtf |
| 01:02.02 | JT | do you know what lower case is |
| 01:02.03 | TrentCreek | huh? |
| 01:02.16 | JT | since when do pattern matches ever use lower case in asterisk |
| 01:02.47 | TrentCreek | i am using examples, not precise |
| 01:03.06 | JT | your "examples" keep getting less precise... |
| 01:03.37 | TrentCreek | so I change _X. to NXXXNXXXXXXX > |
| 01:03.38 | TrentCreek | ? |
| 01:04.08 | JT | i guess, if that's how your numbers come in |
| 01:06.15 | TrentCreek | Or to NXXXXXXXXXX? |
| 01:06.42 | JT | i have no idea on the format of your numbers |
| 01:07.59 | TrentCreek | whatever the iax line sends in |
| 01:08.12 | JT | which i have still no idea about |
| 01:08.17 | JT | anyway, back later. |
| 01:08.30 | TrentCreek | ok |
| 01:08.35 | TrentCreek | thanks |
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| 01:30.53 | *** part/#asterisk putnopvut (n=putnopvu@c-76-27-145-6.hsd1.al.comcast.net) |
| 01:42.01 | TrentCreek | This is strange... |
| 01:42.25 | TrentCreek | I made the caller ID name in my exentions.conf file |
| 01:43.21 | TrentCreek | and CallerID....instead of the name I put in, it seems the Service I called to, did a reverse loojup and instead listed the name whos account it is instead of what I put |
| 01:43.58 | Maliuta | TrentCreek: that's not strange at all, alot of providers force the callerid |
| 01:44.16 | Maliuta | I know my 2 sip providers that push my stuff on the PSTN do |
| 01:45.05 | Maliuta | TrentCreek: you can make you asterisk box override the incoming callerid too |
| 01:45.15 | TrentCreek | i guess no way to over ride unless I use a plain ol DID number? |
| 01:45.35 | TrentCreek | oh you can? that is neat |
| 01:45.44 | Maliuta | it's just data |
| 01:46.09 | TrentCreek | I'd really rather override the outgoing CID |
| 01:46.15 | TrentCreek | name that is |
| 01:47.10 | Maliuta | considering it will be being re-written on the far end of the connection you have shit all luck of doing that |
| 01:47.43 | TrentCreek | i am trying a DID number now, before it was my cell number |
| 01:48.40 | Maliuta | unless the provider is going to let you abitrarily set your callerid |
| 01:49.31 | TrentCreek | caller ID is not the problem..its the Name I want |
| 01:50.59 | Maliuta | who said callerid had to be a number? |
| 01:51.05 | Maliuta | it |
| 01:51.22 | TrentCreek | asterisk |
| 01:51.30 | Maliuta | it's still going to be re-written at the far end unless the provider allows you to do otherwise |
| 01:51.36 | TrentCreek | and it does.. |
| 01:51.44 | Maliuta | callerID is a string |
| 01:51.50 | TrentCreek | I dont care about number..I am trying to fix callerID name |
| 01:52.11 | TrentCreek | number is easy nough to change |
| 01:53.54 | *** join/#asterisk CVirus (n=GoD@82.201.222.194) |
| 01:54.29 | TrentCreek | I changed the number and the service I called did not want to accept the number I put in |
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| 02:23.04 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 02:23.28 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 02:23.51 | *** join/#asterisk emacsen (n=serge@pdpc/supporter/sustaining/emacsen) |
| 02:24.18 | emacsen | hey, this may be a dumb question but can someone tell me how to use sox or ffmpeg to turn the .gsm files into g711? |
| 02:25.38 | Dovid | eamcsen: not a dumb question at all ;) |
| 02:25.41 | Dovid | ~sox |
| 02:25.41 | jbot | it has been said that sox is Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 02:25.53 | emacsen | yeah I already asked about sox |
| 02:26.14 | Dovid | http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 02:26.30 | Dovid | http://www.voip-info.org/wiki/view/sox |
| 02:26.31 | *** join/#asterisk TJNII (n=TJNII@209.234.89.226) |
| 02:26.42 | Dovid | Anyone here going to be at VON israel? |
| 02:27.04 | emacsen | what's the difference between wav and g711? |
| 02:27.14 | emacsen | because I'm still not seeing g711 explicitly |
| 02:27.29 | Dovid | http://www.asteriskguru.com/audio_conversion.php |
| 02:27.33 | Dovid | yea. didnt see that area |
| 02:27.37 | Dovid | wav is a different format |
| 02:27.49 | Dovid | asterisk will need to do transcoding which can put a load on ther server if u do it a lot |
| 02:27.56 | Dovid | http://www.nch.com.au/wavepad/index.html |
| 02:27.57 | emacsen | that url is bad |
| 02:28.02 | Dovid | http://www.germanixsoft.de/ |
| 02:28.17 | emacsen | I don't have Windows or use non-FS software |
| 02:29.36 | Dovid | argh !!1 |
| 02:29.40 | Dovid | this seems to do uLaw |
| 02:29.40 | Dovid | http://www.freedownloadmanager.org/downloads/AudioCommander_15284_p/ |
| 02:31.20 | emacsen | <PROTECTED> |
| 02:31.21 | emacsen | hrm |
| 02:31.51 | Dovid | so try that ;) |
| 02:31.56 | Dovid | i never used sox much |
| 02:32.12 | emacsen | well everything you pasted was for Windows |
| 02:32.23 | emacsen | and as I explained- I don't use non-Free Sotware |
| 02:33.50 | Dovid | ok np |
| 02:33.57 | Dovid | sorry i cant help |
| 02:34.35 | Dovid | i dont like M$ much but i need it for "other" things. its 4:34 AM and i need to be up in 1.5 hours for VON. night Y'Allllllll |
| 02:34.45 | emacsen | ok |
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| 02:47.38 | linagee | what do i use the browser in my polycom phone for? |
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| 03:29.04 | Twister | hey all, quick question on setting up dundi, do i need to generate keys on every server im peering with and put them in the keys directory on each server, like say i have serverA and serverB, do i need to generate keys on both servers then put A's keys in B and B's keys in A? |
| 03:36.53 | TrentCreek | Give |
| 03:36.53 | TrentCreek | ol |
| 03:37.28 | TrentCreek | croc |
| 03:38.02 | TrentCreek | Keyboard |
| 03:38.04 | TrentCreek | got |
| 03:38.05 | TrentCreek | water |
| 03:38.15 | TrentCreek | shit |
| 03:39.20 | TrentCreek | a |
| 03:40.29 | Twister | TrentCreek: calm down..have some dip |
| 03:47.38 | halogen8 | [TK]D-Fender: yes, running trixbox |
| 03:50.53 | TrentCreek | 74my |
| 03:50.58 | TrentCreek | my |
| 03:51.01 | TrentCreek | key |
| 03:51.05 | TrentCreek | 3.2board |
| 03:51.10 | TrentCreek | wet |
| 03:52.42 | *** join/#asterisk bmg505 (n=leon@196.209.183.36) |
| 03:55.51 | TrentCreek | who was |
| 03:55.54 | TrentCreek | aski |
| 03:55.58 | TrentCreek | shit |
| 03:56.33 | Nugget | heh |
| 03:56.52 | TrentCreek | stil |
| 03:56.54 | TrentCreek | wet |
| 04:00.14 | *** join/#asterisk asdx (n=diego@adsl-138-116.click.com.py) |
| 04:00.49 | Nugget | I don't think any of us really care. |
| 04:06.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 04:31.38 | TrentCreek | why |
| 04:32.43 | JT | TrentCreek: if your keyboard is stuffed, stop using it and stop flooding the channel |
| 04:33.54 | TrentCreek | its |
| 04:33.58 | TrentCreek | dryer |
| 04:34.20 | JT | your typing still hasn't improved |
| 04:34.50 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 04:42.04 | TrentCreek | sure it has |
| 04:46.00 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
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| 05:10.45 | *** join/#asterisk Cyford (i=geegs1@c-24-99-118-189.hsd1.ga.comcast.net) |
| 05:11.18 | Cyford | how can i raise the volume for IVR's ? |
| 05:12.07 | JT | record the audio at higher volume |
| 05:12.32 | Cyford | and the defualt system volume? |
| 05:12.49 | JT | what |
| 05:12.54 | JT | what system volume? |
| 05:13.06 | coppice | de-wax your ears |
| 05:13.15 | Cyford | the defualt voice like voicemail |
| 05:13.31 | JT | Cyford: what system volume? |
| 05:14.10 | *** join/#asterisk samdell3 (n=antispam@nap-mt2a.airnet.net.nz) |
| 05:14.45 | Cyford | thats what i am asking, is there a way to raise the volume for the voice promps that came with asterisk? |
| 05:14.57 | JT | re record them. |
| 05:15.24 | samdell3 | prompts are already normalised well. you have a problem elsewhere. |
| 05:16.02 | Cyford | ok, |
| 05:17.23 | samdell3 | On what 'channel' is you audio too quiet? Zap / SIP / other ? |
| 05:17.43 | Cyford | sip |
| 05:17.53 | Cyford | but only on 1 device |
| 05:17.54 | samdell3 | what is your ATA hardware ? |
| 05:18.07 | Cyford | soft phone |
| 05:18.14 | Cyford | on my pda |
| 05:18.33 | Cyford | rings loud but the ivr is low |
| 05:18.54 | JT | ... |
| 05:19.00 | JT | so you're using some crappy softphone |
| 05:19.04 | JT | and blaming asterisk? |
| 05:19.23 | lowlevel | i never had any luck with softphones |
| 05:19.27 | samdell3 | ..most likely local mic adjustments / software related |
| 05:19.30 | lowlevel | apparently the cisco softphone isn't bad |
| 05:19.40 | Cyford | cell phone has volume control so i cant really tell if the level is ok, computer softphone has volume control as well |
| 05:20.08 | TJNII | Heh. I just wrote a AGI to crontrol MPD. I'm proud of myself. |
| 05:20.57 | Cyford | pda is using sjphone |
| 05:21.07 | Cyford | works well with vonage |
| 05:24.33 | Cyford | jt not blaming asterisk only asking a simple question |
| 05:24.53 | JT | softphones suck |
| 05:25.25 | Cyford | yes, i hacked the linksys rtp and that works wonders |
| 05:26.04 | JT | eh |
| 05:26.37 | Cyford | for my fax and analag line |
| 05:29.24 | samdell3 | Anyone got any idea how well Asterisk scales as far as SIP registration goes ? EG forget transcoding, and call volume etc, I am just trying to establish how many concurrent SIP clients can be registered to one (big) machine at any given time before * starts to strain, with a SIP client re-registration time of 60 secs. Anyone here taken it past, say 1000 clients ? |
| 05:35.36 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
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| 05:36.47 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:37.27 | Cyford | why you dont load balance * |
| 05:43.33 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 05:48.52 | Cyford | how do i transfer calls |
| 05:48.53 | samdell3 | ...will end up load balancing and/or using SER, just hoping to get first hand feedback as to how far * can go as a SIP registrar server |
| 05:49.54 | samdell3 | dial(SIP/xxxxxxx,60,Tt) then press hash button during call |
| 05:52.02 | samdell3 | make sure canreinvite=no as per bug 10647 :-) |
| 05:52.48 | Cyford | i dont understand |
| 05:52.53 | Cyford | the dial |
| 05:53.01 | Cyford | where do i place that |
| 05:53.11 | Cyford | in the extensions. |
| 05:53.16 | samdell3 | use T and or t as dial options |
| 05:54.02 | samdell3 | exten => _XX.,1,Dial(SIP/${EXTEN},60,Tt) |
| 05:57.43 | Cyford | does that only transfer to extensions or will it work with outside lines too |
| 05:58.42 | JT | samdell3: you don't usually both use t and T |
| 05:59.09 | samdell3 | Cyford: you can transfer anywhere asterisk can talk |
| 05:59.57 | samdell3 | JT: T and t was for example... you can use them together but a good idea as the other end can transfer by pressing hash also |
| 06:00.30 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:01.23 | JT | samdell3: using both is a bad idea |
| 06:01.28 | JT | usually you want one or the other |
| 06:01.30 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 06:02.54 | samdell3 | sorry typo, meant 'but not a good idea....' |
| 06:04.41 | Cyford | ok, i have added it to my dial plan but when i tested it with a call and hit # nothing happened |
| 06:11.33 | Cyford | sould this work with asterisknow too |
| 06:18.12 | Cyford | is this true http://sipx-wiki.calivia.com/index.php/AsteriskNOW_vs._sipXecs_-_Comparing_User_Features |
| 06:19.56 | *** part/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
| 06:36.27 | Cyford | ok call transers are working |
| 06:38.02 | Cyford | but it doesnt work when an outside line calls in |
| 06:44.01 | JT | Cyford: using # for transfers is a hack |
| 06:44.09 | JT | sip has built in ability to transfer |
| 06:48.56 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:48.59 | Cyford | how do i use that built in ability |
| 06:49.08 | remmo | # |
| 06:49.14 | remmo | call features |
| 06:50.39 | *** join/#asterisk Jubalint (n=HoBob@adsl-072-148-059-225.sip.ard.bellsouth.net) |
| 06:50.59 | Cyford | yes Remmo i do that and it works from ext to ext, but when i call from my cell to an ext. i cant transfer the call to another ext |
| 06:54.49 | *** join/#asterisk bintut (n=bintut@cm2.gamma181.maxonline.com.sg) |
| 06:54.53 | bintut | hello all.. |
| 06:55.08 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 06:55.17 | _pepo_ | hi friends |
| 06:55.57 | bintut | anyone familiar with the chan_gtalk on asterisk? i can call to a gtalk buddy from my analog phone but my gtalk buddy cannot call me.. you can find the debug message at http://www.privatepaste.com/7b1NBFsv4p |
| 06:58.43 | *** join/#asterisk AJaymn (i=TJ14@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 06:58.56 | AJaymn | Can you spoof callerid on a ISDN line? |
| 07:01.37 | Jubalint | Simple question from a asterisk neophite. Is the following possible (what basically i want asterisk for) -> Have someone call to asterisk box via VoIP. get an option to leave a message or forward the call to another number, and have the caller ID from the original person show up on the cell phone? How about the caller ID plus an appended something like "Mike via Asterisk"? |
| 07:04.48 | Cyford | yes |
| 07:10.59 | SwK | Jubalint, part one 1 is possible, adding "mike via asterisk' is not possible |
| 07:15.32 | Cyford | i can do that with asterisknow, it (allens Laptop -404-424-82xx) |
| 07:16.48 | *** part/#asterisk AJaymn (i=TJ14@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 07:17.42 | Cyford | but my cell phone doesnt display names unless i added the person to my address book. but it does display on my land line, but i believe it really depends on your sip provider |
| 07:20.40 | Cyford | finally got the call transfers to work :) |
| 07:22.18 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 07:22.23 | _pepo_ | hi friends |
| 07:23.51 | _pepo_ | I am trying to make my voicemail system. Please, How do I know the extension that dial the voicemail number (*97 |
| 07:25.52 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 07:31.44 | Cyford | exten=8500,1,VoicemailMain |
| 07:32.03 | Cyford | mine is set in the extensions file under defualt |
| 07:32.18 | Cyford | i set it too 8500 |
| 07:33.47 | Cyford | i am using asterisknow though, and can be set in the gui |
| 07:55.40 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
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| 08:08.57 | agx | Hi, i there any branches/trunk revision that has STUN support in chan_sip.c ? |
| 08:18.16 | *** join/#asterisk syneus (n=syneus@host79-27-dynamic.8-87-r.retail.telecomitalia.it) |
| 08:56.21 | linagee | wtf? bush was wiretapping as soon as he stepped into office? (well before sept 11). how did ma bell allow this? |
| 08:58.12 | SwK | hah |
| 08:58.44 | SwK | like you think mabell is not going to ignore the law to make a few million dollars? |
| 08:59.45 | SwK | i will never spend a dollar with ATT (or what was bellsouth) because they did some really stupid shit like that years ago to me directly and cost me my ISP business |
| 09:05.27 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 09:06.15 | sheppard | oh? |
| 09:10.34 | SwK | http://www.washingtonpost.com/wp-dyn/content/article/2007/10/12/AR2007101202485.html |
| 09:10.45 | SwK | thats what linagee was refering too |
| 09:26.19 | *** join/#asterisk fed0ra (n=iplab@203.82.225.127) |
| 09:29.57 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 09:30.51 | *** join/#asterisk CVirus (n=GoD@82.201.222.194) |
| 09:37.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:41.46 | lesouvage | With zaptel config I got this output http://www.pastebin.be/5838 Is this the normal output when there are no cards in the system? |
| 09:56.50 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 09:58.03 | *** join/#asterisk _adrin (n=adrin@chello084010032216.chello.pl) |
| 09:58.15 | _adrin | hello guys |
| 09:58.38 | _adrin | can I have a question? |
| 09:59.25 | _adrin | how can i setup email2fax(TIFF/PDF) service using T38 asterisk/other software? |
| 09:59.33 | _adrin | is there a way to do this? |
| 09:59.42 | *** join/#asterisk newbie`` (i=nouser@117.102.56.98) |
| 10:04.01 | The_Ball | How can I enable any sip client to call our inbound extension? or any iax client as well for that matter |
| 10:07.43 | The_Ball | _adrin, iaxmodem and hylafax |
| 10:09.41 | coppice | The_Ball: How exactly does that meet his needs? |
| 10:10.48 | The_Ball | coppice, hylafax can receive emails that will be converted to faxes which can be sent out on the analog line using iaxmodem |
| 10:11.08 | coppice | he said he wants T.38 |
| 10:11.34 | The_Ball | thought that was a fax encoding, guess it's that voip fax encoding then? |
| 10:12.17 | agx | The_Ball, http://www.voip-info.org/wiki-T.38 |
| 10:12.37 | _adrin | hmm |
| 10:12.40 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 10:12.42 | _adrin | and what about t38modem? |
| 10:13.37 | _adrin | thanks The_Ball |
| 10:13.50 | _adrin | but i need to send those faxes via t.38 to my provider |
| 10:14.04 | _adrin | which uses SIP (IAX too maybe ?) |
| 10:14.17 | coppice | t38modem should do what you want. It only worked with H.323 until recently, but I believe it works with SIP now |
| 10:14.52 | lesouvage | I have installed asterisk many times but today with asterisk version 1.4.13 and zaptel 1.4.5.1 zaptel doesn't load and even doesn't seem to be available after make make install. Any suggestions? |
| 10:15.36 | The_Ball | wow Current Bounty amount: USD 12,250 for t.38 in asterisk, impressive |
| 10:15.46 | _adrin | heheh |
| 10:15.53 | _adrin | that is good |
| 10:15.54 | coppice | that's good for a laugh |
| 10:15.58 | _adrin | t38 termination? |
| 10:16.09 | The_Ball | _adrin, http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty |
| 10:16.19 | _adrin | yeah i've seen |
| 10:16.20 | _adrin | it :-) |
| 10:16.31 | _adrin | not for me anyway :/ |
| 10:18.03 | coppice | some of the responses people had to that bounty were amusing. people who had never seen the spec, or knew anything about it said they were going to do it for the money. :-) |
| 10:18.43 | The_Ball | yeah, how hard could it be ;) |
| 10:18.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:18.54 | _adrin | why is it that hard? |
| 10:19.16 | coppice | most of those bounty entries don't actually say what they want. |
| 10:19.59 | *** join/#asterisk jonathonfletcher (n=jonathon@OFSfb-01p1-159.ppp11.odn.ad.jp) |
| 10:42.43 | *** join/#asterisk bagpuss_thecat (i=bagpuss@lodge.glasgownet.com) |
| 10:54.00 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 10:59.47 | The_Ball | can someone recommend a very simple softphone which will allow a windows user to dial a sip:// url without registering a account on the softphone? |
| 11:00.39 | agx | The_Ball, X-Lite free version, perhaps? |
| 11:06.29 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 11:11.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 11:15.10 | *** join/#asterisk kkn088 (n=kikoun@77.205.38.178) |
| 11:18.25 | The_Ball | i thought that needed a proper sip user account, but I will download it and try it thanks |
| 11:20.10 | lesouvage | Is /lib/modules/2.6.9-55.0.9.EL/extra/ztdummy.ko the normal location for ztdummy and zaptel? I can't get it to load and I checked everything I could think of and could find. |
| 11:21.02 | The_Ball | lesouvage, what about insmod /lib/modules/2.6.9-55.0.9.EL/extra/ztdummy.ko |
| 11:21.15 | *** part/#asterisk agx (n=badpengu@81-174-47-230.dynamic.ngi.it) |
| 11:21.42 | The_Ball | lesouvage, my zap modules go into misc not extra though |
| 11:22.19 | lesouvage | The_Ball: thanks I check now |
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| 11:28.03 | *** part/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 11:29.32 | *** join/#asterisk bintut (n=bintut@cm2.gamma181.maxonline.com.sg) |
| 11:31.38 | ThoMe | Hallo und guten Tag! |
| 12:02.37 | *** join/#asterisk bantu (n=Miranda@p54A33F3C.dip0.t-ipconnect.de) |
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| 12:12.06 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:13.53 | Daejeo | ARI- Version 00.10.02) can anyone tell why sometimes recording does not play? if i click 6 times it plays only one time. |
| 12:14.43 | *** join/#asterisk russo (n=russo@about/goats/goatjockey/russo) |
| 12:15.13 | russo | hey guys |
| 12:15.37 | russo | i run asterisk 2.1 on debian... but the init script doesn't seem to be starting it |
| 12:15.54 | russo | are there any falgs i might have set that keep the init script from starting asterisk |
| 12:15.59 | russo | if i start it through cli its fine |
| 12:21.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:28.57 | *** join/#asterisk russo (n=russo@about/goats/goatjockey/russo) |
| 12:29.36 | *** join/#asterisk asterisk4ever_ (n=asterisk@62-90-140-7.barak.net.il) |
| 12:29.57 | asterisk4ever_ | hello guys :) is anyone awake or alive now and have 2 minutes for a question ? |
| 12:30.41 | asterisk4ever_ | anyone ? |
| 12:30.55 | asterisk4ever_ | just 1 question ;) |
| 12:31.37 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:33.27 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:33.27 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:34.01 | tzafrir | gee, he asked 2 questions |
| 12:35.41 | *** join/#asterisk ming_zym (n=ming_zym@124.254.57.46) |
| 12:36.51 | k31th | im the uk is it alaw or ulaw #? |
| 12:38.40 | Daejeo | alaw |
| 12:38.46 | k31th | Daejeo: thansk |
| 12:38.50 | k31th | thanks* |
| 12:38.57 | Daejeo | us- ulaw |
| 12:46.43 | *** join/#asterisk Penggu (n=me@203.213.102.59) |
| 12:47.51 | Penggu | hi all. i've got a call file going to a local channel, which rings a phone (with auto-answer) and plays to it an audio message (PA system) |
| 12:48.06 | Penggu | prob is, in the console, im getting all these error messages for the duration of the call: |
| 12:48.20 | Penggu | [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for gsmtolin |
| 12:48.40 | Penggu | heaps of them, really fast, at only 3x verbosity level |
| 12:49.05 | Penggu | the sound plays alright. i was testing on vmware, so i can't judge reliably the impact of those error messages. |
| 12:49.15 | Penggu | it's always a bit scruncy on vmware |
| 12:49.20 | Penggu | scrunchy |
| 12:51.13 | *** join/#asterisk russo (n=russo@about/goats/goatjockey/russo) |
| 12:52.23 | *** join/#asterisk blq (n=Bl@dslb-088-064-150-021.pools.arcor-ip.net) |
| 12:53.47 | JT | vmware for you... |
| 12:54.42 | Penggu | is it any better on other vms ? |
| 12:55.02 | JT | not really |
| 12:56.34 | Penggu | ive been trying to post my question to asterisk-users@ for the past 3 days, but i haven't been able to get my mail through... no bounces, even. |
| 12:57.15 | Penggu | read somewhere about others experiencing the same problems |
| 12:58.47 | Penggu | running 1.4.13 on linux 2.6.16 |
| 12:58.57 | Penggu | 2.6.18 i mean |
| 12:59.05 | Penggu | might try 1.2... |
| 12:59.19 | Penggu | since that's our production system |
| 13:02.52 | tzafrir | Penggu, maybe you need to reply to someone else's message |
| 13:03.09 | tzafrir | That seems to be one way around the spam filters there |
| 13:03.10 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 13:07.24 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 14:01.36 | *** join/#asterisk circas (n=dom_paq@CPE0015e985d53c-CM0011aec7a4c6.cpe.net.cable.rogers.com) |
| 14:01.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 14:02.17 | circas | hi everyone! |
| 14:03.06 | circas | anyone now if theres a way to insert a caller in a queue at a specific position? |
| 14:09.05 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 14:21.43 | tzafrir | no way to do that? Nobody uses Asterisk in Israel? |
| 14:21.53 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 14:23.32 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
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| 14:31.02 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 14:31.44 | hi365 | is it posible for a call file to do intercom/paging (using sip headers)? |
| 14:32.11 | tzafrir | hi365, it can go into a custom dialplan extension |
| 14:32.23 | tzafrir | It can also provide custom variables |
| 14:32.45 | hi365 | hmm, should i set the sip headers in the call file then? |
| 14:33.15 | hi365 | tzafrir: where you at von 2day? |
| 14:35.04 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:35.24 | tzafrir | At work :-( |
| 14:37.01 | hi365 | it is pretty nice - tell them your sick or something ... |
| 14:37.05 | hi365 | :-} |
| 14:37.06 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 14:38.01 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:38.59 | hi365 | can i string varibale in a call file? foo=bar, foo2=bar2, etc? |
| 14:39.54 | hi365 | seem so |
| 14:40.05 | *** join/#asterisk dynamicproxy (n=chatzill@122.167.93.145) |
| 14:43.32 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:43.45 | hi365 | tzafrir: next step - have a q call an agent play a mesage to the agent and then disconnect <--- is this even posible with a call file? |
| 14:44.31 | tzafrir | I don't know queues well (and don't like to wait...) |
| 14:46.52 | Cyford | why when i have two nics, one for public and one on the private network. Only the on with the gateway works |
| 14:47.51 | Cyford | i cant use multiple gateways on differnt subnets |
| 14:48.40 | Cyford | but sometime the other does work |
| 14:48.54 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:55.40 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 14:56.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:58.33 | dynamicproxy | Anyone here from India ? I have concerns about replacing a regular EPABX with asterisk. |
| 15:02.35 | *** join/#asterisk gardo (n=gardo@121.97.247.235) |
| 15:09.10 | circas | anyone know if its possible to put a caller at a specific position in a call queue? |
| 15:10.56 | [TK]D-Fender | circas, No. |
| 15:11.12 | [TK]D-Fender | dynamicproxy, What does being in Indai have to do with replacing your PBX? |
| 15:11.19 | [TK]D-Fender | India* |
| 15:12.34 | dynamicproxy | [TK]D-Fender: There's this rule that mandates that in India, we should have a "logical partitioning" in place -> Which is explained by various ISPs to be a separate of IP traffic from PSTN lines. |
| 15:13.34 | dynamicproxy | [TK]D-Fender: Going by that interpretation, I would perhaps be breaking the law by using xlite/a SIP phone to call my asterisk server, and then have that place a call over the PSTN via Digium. |
| 15:14.06 | [TK]D-Fender | dynamicproxy, That should mean that you not use * as a termination or origination server. You should be able to use SIP inside an office as a way to connect your phones, and for a remote office extension. but you can not offer ITSP services using it. |
| 15:14.19 | [TK]D-Fender | dynamicproxy, No, that should be fine |
| 15:14.44 | [TK]D-Fender | dynamicproxy, You are using thier PSTN lines and only using the internet as the way to get there. |
| 15:15.03 | [TK]D-Fender | dynamicproxy, they basically don't want you hurting their PSTN business. |
| 15:15.21 | dynamicproxy | [TK]D-Fender: Hmm in fact, I'd be using the office network to reach *... |
| 15:15.58 | dynamicproxy | [TK]D-Fender: Could you expand the acronym ITSP ? |
| 15:16.08 | [TK]D-Fender | dynamicproxy, Same thing. But you are not bypassing the telco, you are using * as a way to GET to your telco |
| 15:16.11 | [TK]D-Fender | ~itsp |
| 15:16.12 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 15:16.40 | [TK]D-Fender | dynamicproxy, Like if you wanted to use an ITSP to PLACE calls over the internet. |
| 15:16.56 | dynamicproxy | [TK]D-Fender: Could you expand the acronym ITSP ? |
| 15:17.01 | [TK]D-Fender | dynamicproxy, But what you're doing is using the internet to get to lines you PAY FOR from them |
| 15:17.19 | [TK]D-Fender | dynamicproxy, Look up for ITSP! Pay attention to the BOT! |
| 15:17.35 | dynamicproxy | [TK]D-Fender: Hey, thanks for that interpretation.. We're indeed using * to GET to the PSTN operator. |
| 15:17.55 | dynamicproxy | [TK]D-Fender: And I'm new to IRC (been a yahoo IM user all these years).. this bot stuff is cool :) |
| 15:18.05 | [TK]D-Fender | dynamicproxy, its ok. |
| 15:20.47 | circas | fender : do you know if someone tried to de this before? |
| 15:21.03 | circas | add in a specific pos in a queue |
| 15:21.46 | [TK]D-Fender | circas, No, see the moment you look at who is in what position, it'll CHANGE on you. |
| 15:22.28 | [TK]D-Fender | circas, We have "weights" to help speed up higher priority calls. |
| 15:23.35 | circas | so i could use weights to have a caller move in front of everyone else |
| 15:30.14 | *** join/#asterisk keith4_ (n=erekson@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 15:33.27 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:33.29 | keith4_ | is anyone aware of a decent cost comparison of typical proprietary PBX solutions with asterisk-based ones? |
| 15:34.32 | Cyford | 20000 - 0 |
| 15:35.13 | Cyford | not including your hardware |
| 15:35.38 | Cyford | for me total was $150 |
| 15:35.39 | circas | but 20000 it includes the hardware right |
| 15:35.45 | Cyford | yes |
| 15:36.14 | circas | what are you refering to |
| 15:37.05 | circas | what does it do, takes a t1 line and splits it to 23 analog |
| 15:37.12 | *** join/#asterisk Twister (n=twister@mail.positech.com) |
| 15:37.15 | circas | with IVR |
| 15:38.36 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:39.53 | Cyford | i use sip |
| 15:40.17 | Cyford | but you can add a card for that |
| 15:41.33 | circas | I was just wondering what that 20000$ does |
| 15:41.43 | circas | voip? |
| 15:41.47 | Cyford | i see with other pbx systems it can go as high as 50k, depending on your options with nortel avaya,nec , toshiba |
| 15:43.22 | blitzrage | do not look to Asterisk just for cost -- look to it for the feature set and flexibility -- it still costs money to develop, unless of course your time is worth nothing. |
| 15:43.53 | *** join/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
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| 15:46.49 | Cyford | thats true but what ever system you go with your going to need to calculate the time it takes you to set it up. unless you ay someone eles to do it. |
| 15:46.57 | circas | of course blitzage, i agree, but if your trying to sell an asterisk system to your custommer, you need probably want to know the features/price of potential competitors |
| 15:49.31 | Cyford | i think with alot of other systems you may be forced to use there phones as well |
| 15:49.46 | circas | I |
| 15:50.38 | Cyford | i can use any analag phone or any sip phone |
| 15:52.09 | newbie`` | oops wrong window .. sorry |
| 15:52.15 | Cyford | lol |
| 15:52.17 | circas | for 150$ what do you run asterisk on? |
| 15:52.22 | circas | a p4? |
| 15:52.31 | Cyford | dual core |
| 15:52.36 | Cyford | yep |
| 15:52.45 | Cyford | only 3 users though |
| 15:52.59 | Cyford | no problems or complaints |
| 15:53.04 | circas | you got a dual core for 150? good price I guess lol |
| 15:53.29 | Cyford | yep, well got it used dell optiplex |
| 15:53.39 | *** join/#asterisk shtoom (n=shtoom@59.93.116.15) |
| 15:53.48 | circas | you could prob have more than 3 users |
| 15:53.54 | circas | i would think |
| 15:54.09 | Cyford | yes, just started the company though, i will |
| 15:54.39 | Cyford | im running it in 64bit |
| 15:54.47 | Cyford | im using asterisknow |
| 15:54.57 | *** join/#asterisk lmoreira (n=lmoreira@201008224042.user.veloxzone.com.br) |
| 15:55.13 | circas | never tried asterisknow |
| 15:55.24 | Cyford | i like it |
| 15:55.37 | bjweeks | Writing dialplans is fun :( |
| 15:55.38 | circas | whats the difference again |
| 15:55.39 | tzafrir | Cyford, does it run Asterisk as root? |
| 15:55.55 | lmoreira | Folks, How Do I supose to Dial on a Unicall Channel? |
| 15:55.56 | Cyford | no |
| 15:55.57 | tzafrir | I'm not sure if it still does |
| 15:56.06 | *** part/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
| 15:56.12 | Cyford | doesnt have a root |
| 15:56.19 | Cyford | need to sudo |
| 15:56.24 | circas | I had problems with unicall before |
| 15:56.37 | tzafrir | Cyford, asterisk itself (the process) runs as root? |
| 15:56.43 | circas | I installed that for a cust in colombia I think |
| 15:56.45 | bjweeks | tzafrir: NovceGuru |
| 15:56.48 | bjweeks | er |
| 15:56.49 | bjweeks | NO |
| 15:57.03 | bjweeks | Never, ever, ever, run asterisk as root |
| 15:57.09 | Cyford | i dont know, i know i tryed to log in as root, and the user didnt exsist |
| 15:57.21 | circas | theres a patch somewhere for unicall |
| 15:57.37 | tzafrir | bjweeks, wow, you expect your IRC client to give a correct tab-completion for the answer :-) |
| 15:57.43 | hi365 | any way to start the clock on queue members? (i.e. to start the count scince the last call?) |
| 15:57.44 | lmoreira | looks like is was installed ok. Astreisk 1.4.11 + Unicall 0.0.4 |
| 15:57.46 | Cyford | in the forum it says you can only sudo for security, and it works for me |
| 15:58.02 | lmoreira | I did the patch |
| 15:58.26 | circas | cool! |
| 15:58.28 | lmoreira | I can see the channels by CLI> UC show channels |
| 15:58.41 | bjweeks | tzafrir: No, it replaces my answer with peoples names :( |
| 15:59.08 | circas | i run asterisk as root lol |
| 15:59.12 | Cyford | 1moreira your using *Now |
| 16:00.02 | Cyford | *Now installs on Rpath so i never had to configuire the user it runs on |
| 16:00.15 | lmoreira | circas: for Dial, I supose use "Dial(Unicall/g1/${EXTEN}|Tr)", writh? |
| 16:01.30 | tzafrir | lmoreira, the second parameter for Dial is the timeout, not the options, right? |
| 16:01.34 | Cyford | who said all sip providers suck? |
| 16:02.09 | lmoreira | circas: for simple example: "Dial(Unicall/g1/${EXTEN})" |
| 16:03.02 | Cyford | jbot why you say sip providers suck, what type of service do you recomend? |
| 16:05.13 | lmoreira | circas, do you have the Unicall.conf file? |
| 16:05.18 | lmoreira | Can I see it? |
| 16:06.37 | Cyford | do you need to be a programer to know how to write these calls, i mean how do yall learn this stuff? is there a book |
| 16:07.31 | circas | no, its on my cust server... |
| 16:07.45 | circas | dident you say you wrote the patch? |
| 16:07.47 | hi365 | can you specify that a q call only a specific agent (even if tthere are other agents around)? |
| 16:08.23 | lmoreira | circas, I got it from a Unicall installation package |
| 16:08.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:09.13 | lmoreira | from here http://www.moythreads.com/astunicall/downloads/astunicall-1.4.9-0.1.tar.gz |
| 16:09.18 | circas | i know we had probs with that... ahh bad memories |
| 16:09.36 | *** join/#asterisk Dovid (n=Dovid@bzq-79-182-99-49.red.bezeqint.net) |
| 16:09.52 | circas | we had finally got this guy we know from mexico, and he was used to dealing with unicall |
| 16:10.13 | circas | and we gave him remote access to the server and he installed the patch for us |
| 16:10.46 | lmoreira | so, you have the unicall.conf file? |
| 16:11.12 | circas | nope, i told you its on my cutomers server |
| 16:11.35 | lmoreira | ok |
| 16:11.55 | circas | I can check if we used the same patch thought |
| 16:12.16 | circas | wait give me a couple of minutes ok |
| 16:12.27 | lmoreira | Are you using Ast 1.4.11 or 1.4.9? |
| 16:14.02 | Cyford | will asterisk work with meridian phones if i have a sip - analag bridge |
| 16:14.14 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:16.21 | hi365 | can anyone think of an orginal way to force a call from to queue to a specific extension? |
| 16:17.43 | Cyford | while there are other agentsin the Queue |
| 16:17.52 | hi365 | yes |
| 16:18.29 | Cyford | based on caller id, or period |
| 16:18.35 | hi365 | Cyford: or any other way to get the agent idle timer started |
| 16:18.54 | hi365 | with a call file i guess |
| 16:19.40 | Cyford | not familiar with call file |
| 16:19.50 | hi365 | what do you suggest? |
| 16:19.51 | Cyford | any of these options |
| 16:19.52 | Cyford | Strategy:This option sets the Ringing Strategy for this Queue. The options are: |
| 16:19.52 | Cyford | <PROTECTED> |
| 16:19.52 | Cyford | <PROTECTED> |
| 16:19.52 | Cyford | <PROTECTED> |
| 16:19.52 | Cyford | <PROTECTED> |
| 16:19.54 | Cyford | <PROTECTED> |
| 16:19.56 | Cyford | <PROTECTED> |
| 16:20.18 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:20.38 | hi365 | these options ar global queue options. thay cannot be changed on a call by call basis |
| 16:20.52 | hi365 | all i need is one call to get the timer started |
| 16:20.55 | circas | lmoreira, the patch we used is called unicall-mfcr2 |
| 16:21.18 | lmoreira | circas, I'll google it. |
| 16:21.30 | circas | ok |
| 16:21.39 | lmoreira | thanks |
| 16:22.47 | *** join/#asterisk dynamicproxy (n=chatzill@122.167.93.145) |
| 16:23.32 | Cyford | nope well beyond my skill level |
| 16:23.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:23.54 | *** join/#asterisk celord]cR (n=Cesar@celord.ice.co.cr) |
| 16:27.03 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.142.13) |
| 16:33.46 | *** join/#asterisk lonekazoo (i=lonekazo@207.173.74.70) |
| 16:35.34 | *** join/#asterisk felix_da_catz (i=felix_da@c-76-31-59-88.hsd1.tx.comcast.net) |
| 16:40.50 | lonekazoo | anyone else here using the aa50 asterisk appliance? If so, are you having the same flash memory problems I've had on the last two? |
| 16:46.14 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 16:47.58 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.142.13) |
| 16:55.38 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 17:12.53 | lmoreira | circas, Unicall+MFCR2 working ok. Thank you! |
| 17:14.38 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.22) |
| 17:15.49 | *** join/#asterisk Cyford (n=allen@12.22.184.3) |
| 17:16.54 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 17:24.12 | *** join/#asterisk putnopvut (n=putnopvu@c-76-27-145-6.hsd1.al.comcast.net) |
| 17:25.35 | *** join/#asterisk hi365_m (n=hi365@213.151.59.7) |
| 17:28.53 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.105.176) |
| 17:33.16 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-235-73.lv.lv.cox.net) |
| 17:38.14 | Penggu | tzafrir: that error... it had nothing to do with call files... |
| 17:38.30 | Penggu | tzafrir: even if i just dial the ext num with a soft phone |
| 17:38.38 | Penggu | tzafrir: to an ivr, those errors come up |
| 17:39.19 | Penggu | it doesn't seem to show on 1.2 (our production system) |
| 17:39.21 | tzafrir | Penggu, please remind me what the problem is |
| 17:39.27 | Penggu | might have messed up some things.. |
| 17:39.28 | Penggu | umm |
| 17:39.37 | Penggu | [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for gsmtolin |
| 17:39.39 | Penggu | during a call |
| 17:39.46 | Penggu | repeatedly... |
| 17:40.29 | Penggu | i got a msg out to asterisk-users (thanks for the tip) |
| 17:40.33 | Penggu | so we'll see what happens |
| 17:41.19 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 17:41.26 | Penggu | might rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk and try again |
| 17:41.57 | Penggu | ah, and /var/log/asterisk .. |
| 17:42.15 | Penggu | anywya, ill be auff |
| 17:42.19 | Penggu | cyas! |
| 17:42.33 | Penggu | 03:42:32 ! |
| 17:45.01 | *** join/#asterisk ManxPower (n=manxpowe@100.sub-75-202-119.myvzw.com) |
| 17:48.37 | keith4_ | what are common makes of SIP phones that people use? |
| 17:50.01 | *** join/#asterisk agx (n=badpengu@81-174-47-230.dynamic.ngi.it) |
| 17:50.38 | agx | On a 4 BRI, can i connect a cable from a port 1 TE to port 2 NT and same on port 3 and 4 to create a loopback for testing? |
| 17:50.51 | ManxPower | ~phones |
| 17:50.52 | jbot | from memory, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. ... |
| 17:53.50 | Daviey | jbot: +1 |
| 17:53.51 | jbot | 1 is a number, silly |
| 17:55.26 | *** join/#asterisk Runlvl (n=juan@128-19-235-201.fibertel.com.ar) |
| 18:08.15 | *** join/#asterisk Cyford (i=geegs1@c-24-99-118-189.hsd1.ga.comcast.net) |
| 18:10.48 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 18:17.50 | keith4_ | how is the PDF of "the book" licensed? |
| 18:19.22 | celord]cR | hello guys I'm from costa rica :d |
| 18:20.46 | celord]cR | hello guys I'm from Costa Rica |
| 18:22.32 | celord]cR | can I use asterisk with only Ekiga sofphones before buy any hardware? |
| 18:22.32 | macTijn | they have phones there ?! |
| 18:22.35 | Runlvl | celord]cR, Hello, for asterisk community support in spanish http://www.asterisk-la.org |
| 18:22.54 | Runlvl | celord]cR, yes, of course |
| 18:27.10 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 18:28.33 | tzafrir | celord]cR, what about #asterisk-es or any similar IRC channel? (not that I speak spanish) |
| 18:29.19 | celord]cR | mmm thanks tzafrir, I did not know that! |
| 18:29.40 | tzafrir | celord]cR, sure you can. |
| 18:30.13 | tzafrir | celord]cR, though Ekiga may not be my favorite soft phone. Are you on Linux? |
| 18:30.34 | celord]cR | yes, which one do you use ? |
| 18:31.06 | tzafrir | I like twinkle |
| 18:33.52 | tzafrir | Mind you, that a you may find it is preferable to use a dedicated phone |
| 18:34.14 | tzafrir | It tends to be more available |
| 18:45.43 | k31th | out of interest for feature code stuff like talking clock what do you guys use? 4 digit? and for extension 4 digit |
| 18:45.46 | k31th | ? |
| 18:47.13 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 18:49.15 | TrentCreek | wake up |
| 18:49.22 | k31th | ? |
| 18:56.48 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 19:02.50 | *** join/#asterisk hi365_m (i=HydraIRC@213.151.59.7) |
| 19:10.20 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 19:16.00 | Nivex | jbot: chan_mobile |
| 19:16.23 | Nivex | ~chan_mobile |
| 19:16.30 | Nivex | guess it doesn't know |
| 19:16.52 | Mw3 | do you know about any working digium ftp server? |
| 19:17.05 | Nivex | They took them all away |
| 19:17.44 | Mw3 | its quite hard to download asterisk to a server, because their http site uses some kind of weird javascript stuff. so its not working from console browser |
| 19:17.58 | Nivex | yeah, I was just trying to dl asterisk 1.4.13 |
| 19:18.10 | Nivex | guess you have to get it with a GUI and sftp it up |
| 19:18.17 | Nivex | (which is a PITA) |
| 19:18.21 | Mw3 | yes :( |
| 19:18.41 | Nivex | I can understand wanting to do download tracking |
| 19:18.47 | Nivex | but that JS thing is... |
| 19:19.06 | Mw3 | not the best choice :) |
| 19:19.55 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.22) |
| 19:22.00 | Nivex | Bluetooth Mobile Device Channel Driver: Depends on: bluetooth(E) |
| 19:22.15 | Nivex | says I haven't met the depedencies, but I have the bluetooth dev package installed |
| 19:22.19 | Nivex | what am I missing? |
| 19:22.28 | Mw3 | run ./configure again |
| 19:22.37 | Mw3 | if you just installed bluetooth dev pakcage |
| 19:22.47 | Nivex | I just ran ./configure against a clean unpack |
| 19:23.09 | Nivex | and I had chan_mobile compiled a looooong time ago, so the libs are still on the box |
| 19:25.28 | Nivex | I've grabbed chan_mobile.c and put it in channels/ |
| 19:28.15 | *** join/#asterisk ToTo (n=ToTo@host75-142-dynamic.8-87-r.retail.telecomitalia.it) |
| 19:30.42 | Nivex | ah, that could be the problem. I tried to drop it in to a 1.4 branch, and it needs trunk. |
| 19:30.52 | ManxPower | There you go. |
| 19:31.30 | *** join/#asterisk rhombus (n=rhombus@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 19:32.39 | Nivex | I'm debating getting one of these for my parents' new place: http://www.cellantenna.com/Dockingstations/dockntalk.htm |
| 19:32.59 | Nivex | chan_mobile might be tad cheaper :) |
| 19:43.19 | *** join/#asterisk luni-sama (i=lunix@gateway/tor/x-1851f60c363b0bb2) |
| 19:48.33 | *** join/#asterisk rtasterisk (n=mozveren@se167-1-82-242-148-65.fbx.proxad.net) |
| 19:48.36 | rtasterisk | hello all |
| 19:48.47 | rtasterisk | I need help about realtime sip configuration |
| 19:48.56 | rtasterisk | I followed tutorials on voip-info |
| 19:49.01 | rtasterisk | installed a asterisk |
| 19:49.04 | rtasterisk | postgresql |
| 19:49.24 | rtasterisk | the command "realtime load sipusers name 203" |
| 19:49.36 | rtasterisk | load the sip buddie |
| 19:49.54 | rtasterisk | but when with a sip client (sjphone) I try to register |
| 19:50.00 | rtasterisk | its doesn't work |
| 19:50.07 | rtasterisk | Host not found error |
| 19:50.24 | tzafrir | where do you see that error? |
| 19:50.43 | rtasterisk | In asterisk console |
| 19:51.07 | rtasterisk | The problem is asterisk don't try to lookup the sip device in database |
| 19:51.21 | rtasterisk | the connection with postgresql is ok |
| 19:51.34 | rtasterisk | 'realtime load ...' works |
| 19:51.58 | rtasterisk | I have inserted ' sipusers => pgsql,asterisk,sip_buddies' in extconfig.conf |
| 19:52.17 | rtasterisk | but registration is not ok |
| 19:52.23 | rtasterisk | what is the problem ? |
| 19:52.30 | tzafrir | What is the exact error you get? |
| 19:52.38 | rtasterisk | [2007-10-14 21:54:52] NOTICE[4734]: chan_sip.c:14839 handle_request_register: Registration from '<sip:203@88.191.32.36>' failed for '82.242.148.65' - No matching peer found |
| 19:53.42 | rtasterisk | ;iaxusers => odbc,asterisk |
| 19:53.42 | rtasterisk | ;iaxpeers => odbc,asterisk |
| 19:53.42 | rtasterisk | sipusers => pgsql,asterisk,sip_buddies |
| 19:53.42 | rtasterisk | ;sippeers => odbc,asterisk |
| 19:53.42 | rtasterisk | ;voicemail => odbc,asterisk |
| 19:53.43 | rtasterisk | ;extensions => odbc,asterisk |
| 19:53.45 | rtasterisk | ;queues => odbc,asterisk |
| 19:53.47 | rtasterisk | ;queue_members => odbc,asterisk |
| 19:53.50 | rtasterisk | its the extconfig.conf file |
| 19:53.51 | tzafrir | ~pb |
| 19:53.51 | jbot | A Pastebin is a place to paste your stuff 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 |
| 19:54.23 | tzafrir | "peer" as in 'sip show peers' |
| 19:54.55 | tzafrir | you should probably need an entry of type 'friend' for that phone |
| 19:55.08 | tzafrir | have you set there host=dynamic ? |
| 19:55.14 | rtasterisk | Name/username Host Dyn Nat ACL Port Status |
| 19:55.14 | rtasterisk | 101 (Unspecified) D N 0 Unmonitored |
| 19:55.14 | rtasterisk | freephonie-out/myphone 212.27.52.5 5060 Unmonitored |
| 19:55.32 | rtasterisk | I am confused because I declared 2 devices in sip.conf |
| 19:55.48 | rtasterisk | and I want to use the realtime architecture too |
| 19:56.20 | rtasterisk | Its possible to use simultany the both architectures (text file and DB) ? |
| 19:57.12 | tzafrir | rtasterisk, you have there just the "sipusers" table? Maybe you also need an entry in sippeers ? |
| 19:57.17 | tzafrir | (Not really sure) |
| 19:57.40 | rtasterisk | I have created a table 'sip_buddies' |
| 19:57.54 | rtasterisk | *CLI> realtime load sipusers name 203 |
| 19:57.54 | rtasterisk | <PROTECTED> |
| 19:57.54 | rtasterisk | <PROTECTED> |
| 19:57.54 | rtasterisk | <PROTECTED> |
| 19:57.54 | rtasterisk | <PROTECTED> |
| 19:57.55 | rtasterisk | <PROTECTED> |
| 19:57.57 | rtasterisk | <PROTECTED> |
| 19:57.59 | rtasterisk | <PROTECTED> |
| 19:58.03 | rtasterisk | <PROTECTED> |
| 19:58.05 | rtasterisk | <PROTECTED> |
| 19:58.07 | rtasterisk | <PROTECTED> |
| 19:58.09 | rtasterisk | <PROTECTED> |
| 19:58.11 | rtasterisk | <PROTECTED> |
| 19:58.13 | rtasterisk | <PROTECTED> |
| 19:58.15 | rtasterisk | <PROTECTED> |
| 19:58.17 | rtasterisk | <PROTECTED> |
| 19:58.19 | rtasterisk | <PROTECTED> |
| 19:58.21 | rtasterisk | <PROTECTED> |
| 19:58.23 | rtasterisk | <PROTECTED> |
| 19:58.25 | rtasterisk | <PROTECTED> |
| 19:58.27 | rtasterisk | I can retrieve information from console |
| 19:58.35 | rtasterisk | but with sjphone, registration is impossible |
| 19:59.18 | rtasterisk | ??? |
| 20:03.03 | rtasterisk | no idea ? |
| 20:07.43 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 20:09.34 | rtasterisk | its works with sippeers !!! |
| 20:09.41 | rtasterisk | thanks you !!!! |
| 20:09.49 | rtasterisk | but I dont understand why |
| 20:10.06 | rtasterisk | for me sippeers refer to sip devices of type peer ! |
| 20:10.15 | rtasterisk | ?? |
| 20:12.59 | rtasterisk | what is the usage of sipusers ? |
| 20:16.35 | rtasterisk | very ambigus configuration file |
| 20:16.36 | rtasterisk | ... |
| 20:34.10 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 20:38.08 | *** join/#asterisk naif (n=xyz@81-208-60-192.ip.fastwebnet.it) |
| 20:38.09 | naif | http://incredibledirectory.com/rss/bigtits/mariah.jpg |
| 20:38.11 | *** part/#asterisk naif (n=xyz@81-208-60-192.ip.fastwebnet.it) |
| 20:38.22 | bjweeks | haha? |
| 20:38.45 | fujin_ | what the shit |
| 20:41.38 | Greek-Boy | lol |
| 20:46.09 | *** join/#asterisk russo (n=russo@about/goats/goatjockey/russo) |
| 20:58.08 | *** join/#asterisk Sorikan (n=sorikan@208.52.160.70) |
| 20:59.59 | Sorikan | Is there anyone here who can possibly help with easyvoxbox install issue? |
| 21:04.00 | k31th | what is easyvoxbox ? |
| 21:04.21 | Sorikan | www.easyvoxbox.com |
| 21:04.24 | Sorikan | like trixbox |
| 21:07.45 | fujin_ | probably the maintainers |
| 21:07.50 | fujin_ | we don't support such silly things |
| 21:08.14 | Sorikan | feel the love |
| 21:08.18 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-59-108.pskn.east.verizon.net) |
| 21:08.52 | fujin_ | not love, just resistance to stupidity |
| 21:08.59 | fujin_ | tried #easyvoxbox ? :] |
| 21:10.50 | Sorikan | Yes - doesnt exist |
| 21:11.27 | fujin_ | what an awesome project |
| 21:12.23 | fujin_ | why not install Your Favourite Linux(TM) and proceed to install and configure asterisk by hand? |
| 21:12.42 | fujin_ | anything that easyvoxbox/trixbox can do would be relatively easy to do by hand |
| 21:12.51 | Sorikan | no time |
| 21:12.57 | fujin_ | actually, I retract that statement, being that you can't even install it |
| 21:18.13 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-92-214-219.dsl.hstntx.swbell.net) |
| 21:34.36 | hi365_m | fujin_:without starting a flame war - ide really like to know how you can configure grandstream phones "by hand" with the same relitive ease as with trixbox |
| 21:43.45 | fujin_ | ~gs |
| 21:43.46 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:43.51 | fujin_ | I'm not even going to dignify that |
| 21:43.57 | fujin_ | phones and pbx's are completely different, I hope you realise |
| 21:47.51 | *** join/#asterisk agx (n=badpengu@81-174-47-230.dynamic.ngi.it) |
| 21:49.02 | *** join/#asterisk Runlvl (n=juan@128-19-235-201.fibertel.com.ar) |
| 21:49.25 | Runlvl | We are looking for partners in Argentina or Latin America, http://www.asterisk-la.org |
| 21:50.58 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.117) |
| 21:55.54 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 21:56.30 | hmmhesays | This polycom ip-601 config file is making me feel retarded |
| 22:00.05 | *** join/#asterisk russo (n=russo@about/goats/goatjockey/russo) |
| 22:04.02 | JT | Sorikan: time is what you lose when you find no-one will support the package you are using |
| 22:04.14 | JT | so the "no time" thing is pretty shortsighted |
| 22:04.30 | CBU[^_^]M`` | waaa |
| 22:05.02 | JT | hi365_m: also, wrong channel for trixbox |
| 22:07.10 | *** join/#asterisk Maan (n=maan@c-24-34-119-183.hsd1.ma.comcast.net) |
| 22:08.00 | hi365_m | yeh - didnt expect anyone to answer that one! |
| 22:22.03 | *** join/#asterisk jsaunders (n=super@S0106006008145635.vs.shawcable.net) |
| 22:23.13 | ManxPower | What kind of idiot would ask for trixbox support here? |
| 22:24.46 | Nivex | ManxPower: The kind who didn't read the /topic ? |
| 22:25.25 | ManxPower | Nivex: *nod* |
| 22:26.12 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 22:27.25 | lirakis | i dont know wtf just happened.. my home phone just stopped working. I get no audio either way, I can send out calls, but asterisk seems to be unable to reach my phone and goes to vm for inbound calls... |
| 22:27.52 | Strom_C | what is your home phone connected to? which voicemail answers? |
| 22:28.10 | *** join/#asterisk stafai (n=kamaji@resnet-186224.resnet.bris.ac.uk) |
| 22:28.13 | bjweeks | are you using a GUI of some sorts? |
| 22:28.21 | ManxPower | did you reboot the asterisk box? |
| 22:28.29 | stafai | Could someone please explain to me the different between asterisk and yate ;_; |
| 22:28.33 | Strom_C | did you twiddle the thingy? |
| 22:28.48 | bjweeks | pfft, this is Linux, rebooting is for Windows :P |
| 22:29.01 | Strom_C | i hate linux zealots |
| 22:29.17 | ManxPower | stafai: yate was started by some Romanian woman that was very unhappy with the design of Asterisk |
| 22:29.35 | Qwell | Strom_C: I'm a linux zealot :p |
| 22:29.39 | jsaunders | stafai: Asterisk kicks ass and Yate is made by gypsies. |
| 22:29.45 | stafai | lol |
| 22:30.00 | stafai | so Asterisk works as a VoIP client, too? |
| 22:30.05 | ManxPower | Asterisk has a parge community -- that helps. |
| 22:30.07 | jsaunders | Indeed |
| 22:30.08 | stafai | it has too many features, they confuse me :( |
| 22:30.16 | ManxPower | stafaiparge == large |
| 22:30.26 | stafai | ManxPower: I was hoping so :P |
| 22:30.26 | jsaunders | asterisk is a pbx, yate is a softswitch. |
| 22:30.43 | Strom_C | Qwell: every system has its problems. zealots seem to magnify the problems of everything else and pretend like their preferred system's problems don't exist |
| 22:30.59 | ManxPower | jsaunders: any yet YATE supports Zaptel |
| 22:31.02 | Qwell | yeah, I'm a zealot then :P |
| 22:31.12 | stafai | jsaunders: a softswitch is only for voip and a PBX is for everything? |
| 22:31.14 | Strom_C | Qwell: bad bad critical thinking and argument skills :( |
| 22:31.30 | Qwell | I'm on the internet. I don't need those skills. |
| 22:31.35 | jsaunders | freeswitch is crushing yate anyways. |
| 22:31.38 | Qwell | I JUST USE CAPSLOCK, AND I WIN EVERY ARGUMENT |
| 22:31.56 | Strom_C | I JUST USE BONERS AND GAGHLGHALHAHGALAH AND I WIN MORE THAN YOU |
| 22:31.57 | Qwell | jsaunders: does freeswitch even have a release yet? |
| 22:32.14 | ManxPower | stafai: You should be asking that question to the YATE people. They are the ones with the smaller community and so would be more interested in people using their stuff. |
| 22:32.18 | bjweeks | Strom_C: In the last two weeks only one of my machines crashed, and it was using Windows. I will keep making fun on Windows until it stops crashing |
| 22:32.23 | bjweeks | of* |
| 22:32.23 | ManxPower | As far as I'm concerned if you don't like Asterisk then don't use it. |
| 22:32.43 | Strom_C | bjweeks: i'm not saying linux requires reboots like windows does, but there are times when you need to reboot liinux |
| 22:32.49 | Strom_C | i.e. upgrading to the kernel of the week |
| 22:32.54 | Strom_C | :) |
| 22:33.08 | stafai | ManxPower: well I tried, but there's about 6 people in #yate. I really wanted something I could use to 'bridge' my phone line with voip sorta thing |
| 22:33.18 | bjweeks | True, I was making fun of the "reboot it!" logic that Windows users use |
| 22:33.18 | Qwell | stafai: so use asterisk |
| 22:33.22 | ManxPower | stafai: well that tells you one thing, doesn't it. |
| 22:33.39 | stafai | Qwell: yes sir |
| 22:33.46 | stafai | ManxPower: it does? |
| 22:33.50 | ManxPower | bjweeks: *I* know you don't normally have to reboot Linux, but it is easier than stepping people thru making sure Zaptel is unloaded. |
| 22:34.09 | ManxPower | stafai: yes. #asterisk has 261 people, #yate has 6 |
| 22:34.16 | Strom_C | Qwell: zealotry |
| 22:34.19 | Strom_C | :) |
| 22:34.45 | ManxPower | So Asterisk is 43.5 times as popular as YATE. |
| 22:34.51 | bjweeks | If you can't read the source of Yate, Asterisk seems like a better choice |
| 22:34.57 | stafai | ManxPower: oh, sorry, I'm slow :P |
| 22:35.08 | bjweeks | ManxPower: or 43.5 time as broken :P |
| 22:35.21 | Cyford | if it doesnt need to rebooted it probably wasnt built by man |
| 22:35.26 | ManxPower | do not dispute my statements with logic! |
| 22:35.30 | ManxPower | 8-) |
| 22:35.43 | bjweeks | Cyford: *Battlestar joke here* |
| 22:35.45 | Qwell | Cyford: so, we need an OS that evolved? |
| 22:36.10 | ManxPower | At this point I'm too commited to Asterisk to switch to something else, even if that something else was easier. |
| 22:36.23 | *** join/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com) |
| 22:36.25 | Cyford | I love asterisk |
| 22:36.29 | Qwell | there's no reason software couldn't evolve, eh? |
| 22:36.32 | *** part/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com) |
| 22:36.42 | Qwell | (and yes, I do mean by itself, over millions of years :p) |
| 22:36.45 | ManxPower | of course, I said that about Digium cards until we had so many problems with them we were forced to switch to Sangoma. |
| 22:36.47 | Cyford | everything evolves |
| 22:36.58 | bjweeks | ManxPower: Well, I don't think anybody says their software is easier than Asterisk, just better |
| 22:37.04 | Cyford | especially technology |
| 22:37.21 | stafai | rocks don't |
| 22:37.29 | bjweeks | You say that now... |
| 22:37.30 | ManxPower | If Digium had fixed those motherboard compat issues and IRQ latency issues a year earlier we would still be using Digium cards. |
| 22:37.30 | Cyford | lol |
| 22:37.51 | Cyford | they do though |
| 22:37.58 | stafai | into what? |
| 22:38.00 | Cyford | they get bigger |
| 22:38.02 | stafai | or are they in their final form already |
| 22:38.10 | Cyford | change shapes |
| 22:38.15 | ManxPower | Asterisk has major design issues -- they ARE being addressed. |
| 22:38.15 | stafai | gain powers? |
| 22:38.18 | stafai | can you battle them? |
| 22:38.19 | Cyford | turn into sand |
| 22:38.48 | Cyford | can you battle a rock? |
| 22:38.53 | Qwell | you could |
| 22:38.54 | Cyford | are you asking me that |
| 22:38.57 | Qwell | just bring a lot of paper |
| 22:39.16 | Qwell | eep |
| 22:39.17 | Cyford | and sizzors |
| 22:39.24 | Cyford | paper rock sizzors |
| 22:39.30 | ManxPower | files Asterisk-foo is strong. |
| 22:43.47 | Cyford | im having the hardest time getting 2 nic cards to work on differnt subnets only the one with the gateway is consistant. the other goes in and out |
| 22:48.33 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:48.59 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 22:52.03 | *** join/#asterisk Twister (n=bob@71-213-215-72.sxcy.qwest.net) |
| 23:00.12 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 23:08.19 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 23:12.20 | *** part/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:17.19 | *** join/#asterisk jsaunders (n=super@S0106006008145635.vs.shawcable.net) |
| 23:17.23 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:17.59 | jsaunders | From my reading it looks as if there has been some major improvements to inband dtmf detection in 1.4 branch? |
| 23:20.09 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 23:20.18 | jsaunders | My current 1.2 setup w/ analog lines on a tdm2400 is a little too inconsistant with inaccurate tone detection. Tried looking at voip-info, asterisk guru, googled'd... found the regular try relaxdtmf, adjust gain, etc... but none of these have fixed our issues. |
| 23:20.45 | jsaunders | Guess I'll have to give 'er a try. |
| 23:20.50 | *** join/#asterisk hijacked (i=0VpK@66.255.220.17) |
| 23:23.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:24.25 | *** join/#asterisk mihinomenest (n=argh@66.255.220.22) |
| 23:26.52 | *** join/#asterisk CVirus (n=GoD@82.201.222.194) |
| 23:30.43 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.141.235) |
| 23:33.42 | *** join/#asterisk TrentCreek (i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com) |
| 23:34.30 | Freman | I don't suppose anyone knows if the polycom's conferencing is compatible with asterisk (ie: voIpProt.SIP.conference.address="??") |
| 23:37.21 | *** join/#asterisk HarryR (n=harryr@cpc1-lamb3-0-0-cust695.bmly.cable.ntl.com) |
| 23:37.35 | HarryR | is there any standard limit defined for the maximum length of the caller id? |
| 23:37.43 | HarryR | or would 16 be a good guestimate |
| 23:41.28 | *** join/#asterisk fholmes (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 23:54.53 | Strom_C | HarryR: the name, or the number? |
