| 00:06.05 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
| 00:11.17 | *** join/#asterisk s14ck (n=s14ck@190-76-70-14.dyn.movilnet.com.ve) |
| 00:13.46 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.217) |
| 00:18.23 | *** join/#asterisk iq (n=chatzill@unaffiliated/iq) |
| 00:18.24 | iq | HI |
| 00:18.49 | metfan2007 | any idea? |
| 00:22.51 | WindowsUser | metfan2007: isolate an invalid number? |
| 00:39.26 | metfan2007 | WindowsUser, I have more info, my carrier says that I have 120 sip channels ready, but the only see 12 simultaneous calls, so 10% of the total channels, and I see in Asterisk the 120 attemps, and the 90% returns hangupcause=0 and no errors |
| 00:41.24 | metfan2007 | how do I know if Asterisk cannot handle the 120 simultaneous petitions |
| 00:43.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5187e200f1267a3d) |
| 00:45.48 | [TK]D-Fender | metfan2007: What gives you the impression that * is failing to handle them? |
| 00:59.10 | metfan2007 | [TK]D-Fender, well, I don't have that impression, but how do I know if the problem is in asterisk or in the telco?, I see all the attemps in the CLI, 120 simultaneous attemps, only 12 calls goes ok, the rest of the calls gets an DIALSTATUS=congestion |
| 00:59.36 | metfan2007 | the tleco says that they only see 12 simultaneous tries |
| 01:01.53 | metfan2007 | I see a lot of "[Jun 28 20:00:15] NOTICE[3122]: chan_sip.c:2941 auto_congest: Auto-congesting SIP/nucleum-093e9840" messages |
| 01:02.59 | Greek-Boy | this one is going to be interesting... |
| 01:04.50 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 01:05.08 | [TK]D-Fender | metfan2007: And how many simultaneous calls does your provider say they PERMIT? |
| 01:05.19 | [TK]D-Fender | metfan2007: They could be telling you to "get lost" |
| 01:05.36 | [TK]D-Fender | metfan2007: Which is very believable |
| 01:06.18 | Anth8708 | Fender, may I bug you about a nortel => * pri issue? |
| 01:06.56 | [TK]D-Fender | Anth8708: Just ask out lound and whoever has something to say will say it |
| 01:06.59 | [TK]D-Fender | loud* |
| 01:07.05 | Anth8708 | rgr |
| 01:07.06 | s14ck | someone have asterisk + ldap? |
| 01:08.07 | metfan2007 | [TK]D-Fender, the telco says ther permit 120 sim calls :( is there any way to know what is the message from the telco? |
| 01:09.27 | [TK]D-Fender | metfan2007: Go look at the actual calls SIP debug, and the DIALSTATUS variable as well |
| 01:10.30 | metfan2007 | [TK]D-Fender, the DIALSTATUS is always CONGESTION in that cases |
| 01:13.42 | Greek-Boy | metfan2007: Type "sip set debug" in the CLI |
| 01:14.06 | metfan2007 | is enabled |
| 01:14.18 | Greek-Boy | ok |
| 01:14.20 | metfan2007 | there's a lot of messages!!!! |
| 01:14.42 | Greek-Boy | metfan2007: Do a dump of the CLI and analyze it |
| 01:14.49 | Anth8708 | ok guys, trying to get my * box to sit between a nortel cs1000 and the pstn. pstn works great, the pri to the nortel keeps dropping. i could really use some help if someone has a few minutes. revelant configs and other info: http://pastebin.com/d4dacbfa6 |
| 01:15.03 | Greek-Boy | metfan2007: Put in a pastebin |
| 01:17.53 | metfan2007 | I cannot get the moment of the error with the debug, I cannot see it!!! |
| 01:19.38 | [TK]D-Fender | Anth8708: "pridialplan=unknown" , "prilocaldialplan=unknown" |
| 01:20.01 | Anth8708 | [TK]D-Fender: rgr. testing now |
| 01:20.41 | *** join/#asterisk missinglink (n=missingl@ppp166-229.static.internode.on.net) |
| 01:22.32 | metfan2007 | Greek-Boy, [TK]-Fender, http://pastebin.ca/1477888 |
| 01:23.41 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 01:23.49 | Anth8708 | [TK]D-Fender: this error concerns me more than any other: [Jun 28 20:23:18] WARNING[5825]: chan_dahdi.c:3360 pri_find_dchan: No D-channels available! Using Primary channel 48 as D-channel anyway! == Primary D-Channel on span 2 up |
| 01:24.11 | *** join/#asterisk s14ck (n=s14ck@190.77.88.14) |
| 01:28.01 | metfan2007 | Greek-Boy, [TK]-Fender, Also I get a lot of http://pastebin.ca/1477896 messages |
| 01:29.56 | [TK]D-Fender | metfan2007: Retransmitting #5 (NAT) to 201.149.5.21:5060: <-- they should not be NAT <- |
| 01:30.07 | Greek-Boy | hmmmm |
| 01:32.05 | metfan2007 | [TK]-Fender, is there NAT?? wow, I have the asterisk server in the carrier datacenter |
| 01:32.21 | *** join/#asterisk Kumbang (n=semprul@rusnas.paume.itb.ac.id) |
| 01:32.34 | [TK]D-Fender | metfan2007: No, this is you configuring things wrong |
| 01:32.57 | Greek-Boy | yeah, NAT is turned on |
| 01:33.00 | Greek-Boy | turn it off... |
| 01:33.36 | metfan2007 | Greek-Boy, [TK]-Fender, ok, turned off |
| 01:35.49 | metfan2007 | Greek-Boy, [TK]-Fender, trying.... |
| 01:38.16 | metfan2007 | Greek-Boy, [TK]-Fender, this is my sip user config: http://pastebin.ca/1477906 is that ok? :S |
| 01:39.23 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 01:39.54 | Greek-Boy | yeah it should be ok |
| 01:39.56 | Greek-Boy | give it a try |
| 01:40.11 | Greek-Boy | it should work even if nat=no is not there... |
| 01:40.52 | metfan2007 | Greek-Boy, [TK]-Fender, no luck |
| 01:41.18 | metfan2007 | Greek-Boy, [TK]-Fender, http://pastebin.ca/1477910 |
| 01:42.56 | Greek-Boy | hmmm |
| 01:42.56 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 01:43.00 | Greek-Boy | i dont get the auto congestion |
| 01:43.28 | Greek-Boy | perhaps [TK]D-Fender can enlighten you |
| 01:44.37 | [TK]D-Fender | I dont even see the dial in there. |
| 01:44.40 | [TK]D-Fender | no SIP debug, NOTHING |
| 01:45.02 | [TK]D-Fender | I see people whose eyes are shut and running around like headless chickens |
| 01:49.19 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 01:51.39 | metfan2007 | Greek-Boy, [TK]-Fender, http://pastebin.ca/1477948 |
| 01:53.01 | Anth8708 | mmm. ok. i give. changing back. that's to everyone who has helped throughout the day. this dch dropping thing just can't be solved easily |
| 01:54.03 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-29-102.dsl.lsan03.sbcglobal.net) |
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| 01:57.08 | metfan2007 | Greek-Boy, [TK]-Fender, there's the sip debug |
| 01:58.13 | [TK]D-Fender | metfan2007: And I don't see a single call attempt in there. |
| 01:58.26 | [TK]D-Fender | metfan2007: All I see is cancel crap, not actual INVITE |
| 01:59.11 | metfan2007 | Greek-Boy, [TK]-Fender, ok, I'll send you another pastebin with the attempts part |
| 02:00.51 | Anth8708 | OK . .one more shot. one more idea. here's my original issue:http://pastebin.com/d4dacbfa6 i'm realizing now, the pri is never REALLY coming up to the nortel. i see the pri come up to the pstn (see the b channels come up), but i don't from the nortel. any final ideas before i call it a night? |
| 02:01.12 | metfan2007 | Greek-Boy, [TK]-Fender, http://pastebin.ca/1477950 The Dial attempts |
| 02:05.12 | [TK]D-Fender | metfan2007: All those calls seem to be accepted |
| 02:05.58 | metfan2007 | and I see a lot of http://pastebin.ca/1477955 messages |
| 02:06.28 | metfan2007 | is that network issues? |
| 02:13.29 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 02:15.09 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 02:15.41 | [TK]D-Fender | -- Nobody picked up in 20000 ms |
| 02:15.43 | [TK]D-Fender | no asnwer |
| 02:15.52 | metfan2007 | yes, that's ok |
| 02:16.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:16.05 | metfan2007 | only 1 of 10 calls are "fine" |
| 02:17.27 | L|NUX | is there any way to check either provider is sending T.38 or not ? |
| 02:17.35 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 02:19.35 | nkohh | metfan2007! it's been *forever* dude! how have you been? what're you up to these days? |
| 02:26.12 | Greek-Boy | metfan2007: Did you come right? |
| 02:26.33 | metfan2007 | Greek-Boy, no, same problem :( |
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| 02:41.18 | trippssss | hello. is there software to create call reports from the asterisk logfile? I usually use CDR logs from mysql but the process wasn't running after a reboot for a few days |
| 02:50.35 | [TK]D-Fender | trippssss: Jump import the records |
| 02:50.38 | [TK]D-Fender | just* |
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| 02:52.21 | trippssss | [TK]D-Fender, how would I do that? |
| 02:53.54 | [TK]D-Fender | trippssss: common SQL thing to take the CSVand just import records. Go ask in #mysql |
| 02:54.38 | trippssss | [TK]D-Fender, ah - import from cdr-custom |
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| 03:21.34 | trippssss | in the cdr call duration and billable cdr columns, is that minutes I presume? |
| 03:22.57 | trippssss | hmmm it appears to be seconds? |
| 03:23.50 | carrar | You might actually have to read some documentation |
| 03:24.10 | trippssss | i'm reading it - it doesn't seem to add up though |
| 03:24.24 | carrar | ~book |
| 03:24.25 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 03:24.43 | trippssss | carrar, read it. |
| 03:25.20 | carrar | don't think you did |
| 03:25.41 | [TK]D-Fender | trippssss: Docs are in the source and its be CRAZY to think its in minutes |
| 03:26.13 | trippssss | [TK]D-Fender, yeah that's what I thought -but seconds is way to small for what I'm trying to report on |
| 03:26.24 | carrar | *60 |
| 03:26.41 | [TK]D-Fender | trippssss: And what do you have doing the actual analysis? |
| 03:26.50 | [TK]D-Fender | carrar: /60 |
| 03:26.51 | trippssss | all the billsec columns are like 18, 14, 12 8 etc. my agents didn't complete the calls in that short of time |
| 03:27.24 | carrar | possibly looking at the wrong CDR entry |
| 03:28.28 | carrar | If you plan to do reports off CDR, you best off having Asterisk put them in a database |
| 03:28.30 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 03:31.40 | trippssss | carrar, right, which I have. I'm just trying to recreate the couple of days worth of data that the mysql service wasn't running |
| 03:32.12 | carrar | write a quick perl script to import them into your db |
| 03:32.16 | carrar | cake |
| 03:32.26 | carrar | maybe 5 lines worth |
| 03:38.57 | *** join/#asterisk techman97 (n=myweiner@71-82-141-75.dhcp.roch.mn.charter.com) |
| 03:39.20 | trippssss | any good queue_log analyzers out there? |
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| 04:23.55 | L|NUX | is there any way to check either provider is sending T.38 or not ? |
| 04:24.01 | L|NUX | any one ? |
| 04:39.26 | WindowsUser | T.38 should count as an "audio protocol" as far as asterisk is concerned, look at sip calls or sip packets |
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| 04:55.09 | ReDNeQ | yo |
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| 05:48.39 | L|NUX | WindowsUser: ok |
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| 05:55.58 | WeazelON | Good morning Guys |
| 05:56.22 | WeazelON | anyone around ? |
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| 06:05.00 | plutopath | hey room |
| 06:05.12 | plutopath | can anyone here PLEASE help me with AsteriskNOW ? |
| 06:09.16 | plutopath | anyone please? |
| 06:11.47 | J4nus | how can we help you ? |
| 06:13.20 | plutopath | i installed AsteriskNOW and i am coming up to a cli login promptfor centos, not the prompt that tells me what ip i can access the gui through |
| 06:14.01 | J4nus | ifconfig |
| 06:14.03 | J4nus | to see the ip |
| 06:14.49 | plutopath | i did |
| 06:15.02 | plutopath | whn i put it in the broswer, it doesnt come up |
| 06:15.18 | J4nus | what is the ip ? |
| 06:16.09 | plutopath | 192.168.1.240 |
| 06:16.12 | plutopath | i assigned it manually |
| 06:16.32 | plutopath | ok? |
| 06:16.36 | J4nus | can you ping from this ip your default gw ? |
| 06:16.37 | drmessano | Sounds like you installed CentOS and not AsteriskNOW via the ks |
| 06:16.41 | J4nus | did you put the right mask ? |
| 06:16.53 | drmessano | Did you put any arguments in when you ran the install? |
| 06:16.54 | plutopath | yup... now i get somewhere, but it has a frog |
| 06:17.01 | drmessano | ah |
| 06:17.03 | plutopath | free pbx? |
| 06:17.55 | plutopath | i have 3 links: Voicemail & Recordings (ARI), Flash Operator Panel (FOP), & FreePBX Administration |
| 06:17.58 | plutopath | Any ideas? |
| 06:18.41 | drmessano | Yeah... welcome to AsteriskNOW.. start using it? |
| 06:18.43 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 06:18.44 | WeazelON | i'm getting this wierd error -- Unable to open file '/var/lib/asterisk/mohmp3/fpm-sunshine': No such file or directory <--= when dialing incoming from the queue |
| 06:18.58 | J4nus | sorry i have to go |
| 06:18.59 | J4nus | bye + |
| 06:19.06 | plutopath | drmessano: freepbx = asterisknow? |
| 06:19.38 | plutopath | ohhh! i see... thanks alot guys!! |
| 06:19.38 | drmessano | Asterisk = Centos + MySQL + PHP + Apache + Asterisk + FreePBX |
| 06:19.43 | drmessano | err |
| 06:19.47 | drmessano | AsteriskNOW = Centos + MySQL + PHP + Apache + Asterisk + FreePBX |
| 06:19.55 | plutopath | i have another ques... would this run stable in my business? |
| 06:20.21 | drmessano | Asterisk is pretty stable |
| 06:20.35 | drmessano | How many users? |
| 06:20.44 | plutopath | around 40 |
| 06:20.55 | WeazelON | i'm running asterisk in my business for 5 years now |
| 06:21.03 | plutopath | or does it depend on my hardware stats? |
| 06:21.05 | drmessano | Ah.. too bad.. It only starts having problems around 5000 users |
| 06:21.13 | plutopath | cool |
| 06:21.23 | plutopath | WeazelON: how many phones? |
| 06:21.33 | WeazelON | 300 |
| 06:21.37 | plutopath | wow |
| 06:21.47 | drmessano | How about 120 running on Asterisk 1.2.1? |
| 06:21.52 | drmessano | heh |
| 06:21.55 | plutopath | can i pm you about or you tell me how you have it setup? |
| 06:21.59 | WeazelON | its 300 on 1.2.17 |
| 06:22.26 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 06:22.29 | WeazelON | sure no prob |
| 06:22.40 | WeazelON | drmessano: do u know anything about why asterisk wont play my Moh ? |
| 06:22.43 | drmessano | I think my 1.2.1 install has you beat.. Grade on a curve due to 1.2.1 |
| 06:23.01 | plutopath | have you guys had any downtime? |
| 06:23.12 | plutopath | throughout the years i mean |
| 06:23.27 | WeazelON | another line i've noticed - File /var/lib/asterisk/mohmp3/fpm-sunshine does not exist in any format |
| 06:23.30 | drmessano | before we rebooted last week it was down about 2 years ago for a reboot |
| 06:23.50 | WeazelON | our record is 325 days up |
| 06:23.55 | WeazelON | :D |
| 06:24.07 | plutopath | and what happend when it went down? |
| 06:24.15 | WeazelON | it went back up :D |
| 06:24.20 | plutopath | downtime? |
| 06:24.24 | drmessano | lol |
| 06:24.25 | WeazelON | depends |
| 06:24.28 | drmessano | Yes, of course |
| 06:24.37 | drmessano | If its DOWN.. that is DOWN TIME |
| 06:24.47 | plutopath | but like how long is the longest? |
| 06:24.47 | drmessano | throws a dictionary |
| 06:24.49 | WeazelON | i think he means how long |
| 06:24.53 | plutopath | lol |
| 06:24.57 | WeazelON | its hard to say plutopath |
| 06:25.01 | WeazelON | shit happens |
| 06:25.06 | WeazelON | you cant really estimate shit |
| 06:25.14 | WeazelON | sometimes you go in do ur shit and go out |
| 06:25.18 | WeazelON | sometimes shit messes with u |
| 06:25.21 | drmessano | 2 mins, 2 days, 2 years.. depends how bad you suck |
| 06:25.25 | WeazelON | sometimes shit is not really there |
| 06:25.39 | WeazelON | but there is no shit if u dont eat |
| 06:25.40 | WeazelON | >< |
| 06:25.43 | drmessano | How much downtime does your main fileserver have? What about your microwave oven? |
| 06:25.54 | plutopath | do you guys recommend any good guides? |
| 06:26.03 | drmessano | Google.com is pretty good |
| 06:26.04 | plutopath | to learn the ins and outs of it ? |
| 06:26.10 | plutopath | i know that drmessano |
| 06:26.36 | plutopath | but personal exeprience tends to be better than a search engine |
| 06:26.43 | WeazelON | the best way to really learn it, is to probably start messing with it and playing with it, and ruin it a couple of times |
| 06:26.56 | drmessano | If you had personal experience, you wouldnt need to google |
| 06:27.02 | WeazelON | i myself learned and still am learning from my work place |
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| 06:49.19 | plutopath | haha! |
| 06:49.25 | plutopath | i found a good book |
| 06:49.58 | plutopath | Building Telephony Systems With Asterisk |
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| 07:42.58 | rvhia | hi, i have a queue with rrmemory. If an agent is on the phone, should * try to ring him? |
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| 08:08.24 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:15.56 | *** join/#asterisk steph_ (n=chatzill@ABayonne-157-1-56-62.w86-213.abo.wanadoo.fr) |
| 08:16.13 | steph_ | Hi |
| 08:16.26 | steph_ | Someone is using call files here? |
| 08:20.38 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.112.153) |
| 08:20.41 | DelphiWorld | hello |
| 08:20.46 | DelphiWorld | please what is the stun port? |
| 08:22.33 | *** join/#asterisk downs (n=downs@p4FD57C09.dip.t-dialin.net) |
| 08:22.59 | downs | hi, I'm a newcomer to asterisk, trying to set up simple video conferencing between ekiga clients (2.0.12). |
| 08:23.26 | downs | however, when a friend of mine tries to connect to my local asterisk server, the server denies the video connection |
| 08:23.29 | downs | (h261) |
| 08:23.48 | downs | the relevant extension is an appkonference room |
| 08:24.04 | downs | asterisk is 1.6.1.0. |
| 08:24.14 | downs | It works for me (on local LAN). |
| 08:24.31 | downs | to elaborate: when I say "denies the video connection", I mean "sets the port field to 0". |
| 08:24.40 | downs | (wiresharked) |
| 08:25.02 | *** join/#asterisk joobie (n=joobie@124-168-57-171.dyn.iinet.net.au) |
| 08:25.09 | downs | the log is devoid of useful messages |
| 08:25.12 | downs | even on full logging. |
| 08:25.49 | downs | (debug,verbose) |
| 08:26.28 | downs | is there any way to find out why asterisk rejects the h261 connection? |
| 08:27.10 | tzafrir_laptop | downs, this is sip, right? |
| 08:27.13 | downs | yep |
| 08:27.30 | tzafrir_laptop | I guess you should look at the sip debug |
| 08:27.36 | downs | checking. |
| 08:27.57 | tzafrir_laptop | and see what the codec capabilities of both sides are |
| 08:28.10 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:32.19 | downs | strange |
| 08:32.27 | downs | looking at his channel info, he's listed as "video support: yes" |
| 08:32.34 | downs | and the format as gsm|h261. |
| 08:32.41 | downs | but the server still rejects the connection. |
| 08:32.58 | downs | *video |
| 08:34.07 | Weazel | downs: i would suggest checking if the vidoe ports are open through firewall/router etc on the other side |
| 08:35.01 | downs | Weazel: it's not that the connection can't be established but that the server flat-out says "I won't even try". |
| 08:35.20 | downs | and yes, all relevant ranges are forwarded. |
| 08:35.26 | Weazel | downs: can u make regular calls ? |
| 08:35.33 | downs | Weazel: private server. |
| 08:35.45 | downs | sip only, no outgoing |
| 08:35.48 | downs | echo mode works though :) |
| 08:36.03 | downs | tzafrir_laptop: the channel stats list his codec capabilities as including h261. |
| 08:36.08 | Weazel | downs: voice passing through both ways ? |
| 08:36.16 | downs | Weazel: voice works. |
| 08:36.35 | Weazel | downs: check on the extension if you are allowing the codec |
| 08:36.39 | downs | Weazel: er, me to him. we haven't tested the other way because he doesn't have a mic. |
| 08:36.41 | tzafrir_laptop | downs, does the other side support h261 as well? |
| 08:36.42 | downs | Weazel: yes I am. |
| 08:36.48 | downs | tzafrir_laptop: same version of ekiga. |
| 08:37.01 | downs | tzafrir_laptop: and I checked the packet protocol. he explicitly requests h261. |
| 08:37.51 | downs | normally I would blame routing problems between me and him. |
| 08:38.01 | downs | but the codec negotiation happens before the first video packets are sent. |
| 08:38.30 | downs | (afaik) |
| 08:39.16 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 08:41.29 | *** join/#asterisk yidiyuehan (n=yidiyueh@bb116-14-126-121.singnet.com.sg) |
| 08:42.29 | yidiyuehan | hi guys, I would like to develop an special application based software based on open source software, anybody has good advice regardingopen source soft phone library? |
| 08:43.43 | Weazel | try #asterisk-dev |
| 08:46.31 | yidiyuehan | thanks bro |
| 08:47.43 | steph_ | who uses "call files" in Asterisk plz? |
| 08:57.14 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 08:58.01 | *** join/#asterisk rockhard1981 (n=snusse@78-82-50-224.tn.glocalnet.net) |
| 08:58.07 | rockhard1981 | hello |
| 09:04.58 | DelphiWorld | tzafrir_laptop: stun travairce UDP? |
| 09:05.27 | downs | okay, the beta seems to work |
| 09:05.29 | downs | have fun |
| 09:05.42 | *** part/#asterisk downs (n=downs@p4FD57C09.dip.t-dialin.net) |
| 09:06.48 | tzafrir_laptop | DelphiWorld, stun is a method of overcoming NAT. And yes, it also works with UDP |
| 09:12.06 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.112.153) |
| 09:12.23 | DelphiWorld | tzafrir_laptop: my ISP is blocking SIP over UDP |
| 09:19.22 | b14ck | use asterisk 1.6 with tcp? |
| 09:20.52 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 09:21.32 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-203.gibnet.gi) |
| 09:30.07 | steph_ | Make a VPN |
| 09:30.32 | steph_ | and change the ISP |
| 09:31.42 | steph_ | Someone has some news about call files? I have a congestion error |
| 09:36.36 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 09:37.59 | dwery | hello. anyone has experience on autoprovisioning with the Yealink T22 ? I need to choice an entry level phone and I need complete customization via tftp/dhcp (configuration, ringtones and logo) |
| 09:39.03 | dwery | (other considered phones are the DLink SPA-921 and Thomson Speedtouch 2030) |
| 09:48.34 | dwery | btw there's no reference to the SPA-921 on DLink's web site.. strange way to sell things... |
| 09:49.05 | dwery | d'oh! it's linksys :D |
| 09:55.03 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 09:56.17 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:00.55 | *** join/#asterisk ik_5 (n=ik@95.211.21.45) |
| 10:07.30 | *** join/#asterisk nicola_pav (n=chatzill@83.244.78.241) |
| 10:07.57 | nicola_pav | hello. have two asterisk servers under same LAN 192.168.0.xxx. |
| 10:08.17 | nicola_pav | i want to forward calls generated from server A to server B |
| 10:08.33 | nicola_pav | so far i managed to generate calls but calls do not reach server B |
| 10:08.33 | rockhard1981 | anybody? is there a module / application that playbacks to channel? or I have to write it? |
| 10:08.36 | nicola_pav | any hint? |
| 10:08.47 | rockhard1981 | re: previous question, (asterisk 1.2.x) |
| 10:11.10 | tzafrir_laptop | Playback? what exactly? |
| 10:11.44 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.126.147) |
| 10:12.00 | DelphiWorld | hello |
| 10:15.41 | DelphiWorld | please i'm unable to register my Linksys SPA901 is behin a ISP that block SIP over UDP, any solution? |
| 10:16.48 | b14ck | DelphiWorld, either use asterisk 1.6 and starting using TCP. Or get a new ISP |
| 10:16.59 | b14ck | Getting a new ISP is probably better, or call and complain. |
| 10:18.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 10:19.28 | nicola_pav | how to forward sip calls between two asterisk servers? |
| 10:19.32 | DelphiWorld | b14ck: i have only this ISP in my citi, cool you tel me if any asterisk 1.6.X package for ubuntu server 8.10? |
| 10:20.25 | b14ck | DelphiWorld, i don't know what packages ubutnu has. But if you're on ubuntu why don't you jsut check with the package manger? |
| 10:20.26 | b14ck | ... |
| 10:21.01 | DelphiWorld | i'm using apt-get |
| 10:23.43 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.126.147) |
| 10:23.54 | DelphiWorld | b14ck: please cool you drop me a SIP call? |
| 10:24.24 | b14ck | what? |
| 10:24.24 | b14ck | what does that mean? |
| 10:26.16 | DelphiWorld | b14ck: to test if i can recev call or no |
| 10:26.30 | b14ck | just call yourself, im not going to call you |
| 10:27.08 | b14ck | it doesnt make any sense to have some random person over the internet call you, when you can easily call yourself. not to mention it woul dprobably be an expensive call since you dont speak good english and probably live outside of the usa |
| 10:27.10 | DelphiWorld | b14ck: ok |
| 10:32.52 | creativx | sip call..expencive? |
| 10:34.49 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 10:36.14 | DelphiWorld | creativx: i don't understand u |
| 10:37.03 | creativx | DelphiWorld: question was for b14ck |
| 10:38.29 | DelphiWorld | creativx: ok, sory |
| 11:02.24 | *** join/#asterisk steph__ (n=chatzill@ABayonne-157-1-154-198.w90-16.abo.wanadoo.fr) |
| 11:05.11 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
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| 11:33.10 | *** part/#asterisk steph__ (n=chatzill@ABayonne-157-1-154-198.w90-16.abo.wanadoo.fr) |
| 11:44.10 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 11:54.45 | aiksa[LV] | is there a difference how include=> statements work between 1.4 and 1.6 ? |
| 11:59.58 | *** join/#asterisk Aiatek (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 12:00.04 | aiksa[LV] | oh got it, change in delimiters. |
| 12:06.16 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-203.gibnet.gi) |
| 12:07.36 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 13:04.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:13.02 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 13:25.50 | Katty | scowls. |
| 13:26.06 | Katty | apparently, it is the end of the world this morning here in missouri. |
| 13:27.41 | creativx | well |
| 13:27.42 | *** join/#asterisk dwayne (n=dwayne@76.29.245.9) |
| 13:27.48 | creativx | odd thing Katty |
| 13:27.49 | creativx | same here |
| 13:33.03 | Katty | hires ninjas and pirates. |
| 13:36.12 | Katty | where's [TK]D-Fender and eppigy and jaytee |
| 13:38.09 | [TK]D-Fender | watches Katty's ninjas and pirates duke it out |
| 13:43.15 | Katty | ohhhh, my ninjas, my pirates....i've hungered for a duel!!!!! a longggg lonely time!!!! |
| 13:45.21 | eppigy | 8[] |
| 13:49.09 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:50.09 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:50.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:52.30 | Katty | windows refuses to activate. |
| 13:52.36 | Katty | this is the first time i've seen it. |
| 13:52.39 | Katty | it just... spazes |
| 13:54.01 | *** join/#asterisk monstertruck (i=monstert@c-75-74-122-15.hsd1.fl.comcast.net) |
| 13:54.17 | monstertruck | hi, is there an equivalent to call-limit for iax? |
| 13:59.53 | *** join/#asterisk VoipForces (n=kvirc@firewall.privalodc.com) |
| 14:00.36 | VoipForces | Hi, anyone knows about a script that can spit out a human readable call trace (based on let's say a callerid) using the log file data ? |
| 14:00.39 | Katty | oh god. microsoft tech support. please kill me. |
| 14:01.14 | beek | hands Katty a couple of Valium tabs for her to prep... |
| 14:01.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:06.21 | *** join/#asterisk MmixX (n=mix@61.14.191.137) |
| 14:06.32 | nkohh | Katty: OMG! it's been *forever*! how are you??? what're you up to these days? |
| 14:07.48 | *** join/#asterisk pmhaddad-lappy (n=Phil@adsl-99-169-190-209.dsl.applwi.sbcglobal.net) |
| 14:07.59 | [TK]D-Fender | Katty: Just askt hem for the phone # of the "Windows Activation Center"and they'll give you a direct 800 # that you can explain why you had to reinstall. Actu nice and they'll just hand you an activation code |
| 14:11.17 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 14:12.56 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 14:16.49 | VoipForces | Anyone on an asterisk log call analyser script? |
| 14:17.28 | Katty | nkohh: on the phone with microsoft )= |
| 14:19.38 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:19.43 | ariel_ | Morning everyone |
| 14:19.51 | Katty | weird. |
| 14:19.59 | Katty | they've renamed the wpa.dbl in system32 |
| 14:20.04 | Katty | and then extracted the original from the cd |
| 14:20.12 | Katty | but it still failzors in normal mode. |
| 14:23.58 | eppigy | ms is the worst |
| 14:24.28 | eppigy | I have spent like 2 hours on the phone with them |
| 14:24.33 | eppigy | to reactivate |
| 14:25.48 | *** join/#asterisk moa_ (n=moa_@lab.vision.net) |
| 14:26.16 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:26.16 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:26.30 | eppigy | TRABAJO |
| 14:30.09 | KavanS | travajo? |
| 14:30.27 | KavanS | anyone try that SIP client "siphon" for iphone w/asterisk yet? |
| 14:30.29 | Katty | they just redid some inf file. |
| 14:30.33 | Katty | and reregistered some dlls. |
| 14:30.37 | Katty | and it still bombzors. |
| 14:30.54 | Katty | i'm about ready to just stick a new license on it |
| 14:31.19 | eppigy | DO IT |
| 14:33.28 | *** join/#asterisk ehsjoar (n=ehsjoar@24.9.91.203) |
| 14:35.41 | *** join/#asterisk BobPierce (n=BobPierc@216.36.132.162) |
| 14:40.11 | Katty | they need to get off the phone with me so i can go pee |
| 14:42.13 | VoipForces | Katty: Put them on hold with a MOH that asvertises linux and asterisk LOL |
| 14:42.13 | carrar | Thats why I do all my meetings from a heated pool |
| 14:43.34 | *** join/#asterisk bluregard (n=matt@66.251.248.60) |
| 14:43.37 | bluregard | hi all |
| 14:43.45 | *** join/#asterisk voipheroes (n=voipman@AMontsouris-757-1-28-165.w90-46.abo.wanadoo.fr) |
| 14:44.29 | *** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e) |
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| 14:51.28 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:55.59 | Katty | heh |
| 14:56.02 | Katty | so they issued me a new product id code. |
| 14:56.06 | Katty | and said, reinstall windows! |
| 14:56.11 | *** join/#asterisk serph (n=serph@64.229.41.116) |
| 14:56.11 | Katty | again! |
| 14:56.27 | [TK]D-Fender | Katty: But they gave it to you :) |
| 14:57.40 | Katty | aye. |
| 14:57.55 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:58.38 | Katty | nkohh: Right. SO. hi. and did you use another /nick cause i don't recognize yours |
| 15:00.54 | *** join/#asterisk ingenius (n=alektro@host147.190-229-174.telecom.net.ar) |
| 15:02.56 | *** join/#asterisk UQlev (n=yuriy@87.228.199.125) |
| 15:03.41 | [TK]D-Fender | Katty: No he's jsut running an obnoxious "conversation starter" script that nags people at random |
| 15:04.29 | [TK]D-Fender | nkohh: That shit is SO last-millennium |
| 15:04.36 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 15:07.09 | bluregard | <PROTECTED> |
| 15:07.34 | Katty | what?! people still use those?! |
| 15:07.56 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:08.06 | Katty | hi mort. |
| 15:09.41 | Katty | the product id you entered is not valid :< |
| 15:11.11 | Katty | sobs a bit |
| 15:12.11 | carrar | This is not a crying channel |
| 15:12.23 | carrar | This is a place of happyness |
| 15:12.30 | seanbright | and happiness |
| 15:12.32 | seanbright | both! |
| 15:12.43 | carrar | hai |
| 15:12.56 | [TK]D-Fender | edits Wikipedia to report that Prozac ended the Great Depression |
| 15:14.29 | *** join/#asterisk errotan (n=errotan@62.201.123.22) |
| 15:15.32 | Katty | oh goody, i wrote the number down incorrectly |
| 15:15.42 | Katty | <PROTECTED> |
| 15:15.58 | Katty | in other mews. |
| 15:16.02 | Katty | eppigy: ALMOST LUNCH TIME! |
| 15:16.10 | Katty | eppigy: what're you having |
| 15:16.18 | mort_gib | hi Katty |
| 15:17.55 | *** join/#asterisk ThatKidKel (n=Kelvin@208.110.55.5) |
| 15:23.58 | leifmadsen | Katty: what wrong with mew? |
| 15:24.17 | *** join/#asterisk SlipperyChicken (n=andrew@LONDON14-1168107385.sdsl.bell.ca) |
| 15:25.06 | ThatKidKel | anyone have recommendations for inbound DIDs in Australia and New Zeland? |
| 15:26.55 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:27.34 | angryuser | good day, can someone tell me what was the last version of asterisk offering zaptel compatibility ? thank you |
| 15:27.45 | angryuser | 1.4.12 ? |
| 15:27.53 | carrar | heh |
| 15:28.09 | *** join/#asterisk |pepesz| (n=kvirc@77-120.ipact.nl) |
| 15:28.45 | angryuser | 1.4.22 it is written chan_dahdi (however i am using zaptel) |
| 15:29.08 | angryuser | i need * without any name changes, thank you |
| 15:30.20 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:32.04 | tzafrir_laptop | chan_dahdi in asterisk 1.4.x can be built with zaptel |
| 15:32.19 | ManxPower | IIRC, all 1.4.x support Zaptel. Versions later than *mumble* support both Zaptel and DAHDI. |
| 15:32.26 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:32.43 | ManxPower | 1.6 does not support Zaptel at all. |
| 15:33.22 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:33.52 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:36.32 | *** part/#asterisk doug (i=doug@breakout.telerama.com) |
| 15:37.00 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 15:40.42 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:41.13 | kaldemar | angryuser: 1.4.21 was the last with chan_zap. as ManxPower said, newer ones can be compiled with zaptel also. |
| 15:44.21 | ManxPower | idly notes he is looking for a job. See also: http://www.fnords.org/skillslist.html |
| 15:44.48 | *** join/#asterisk chendy (n=chatzill@121.15.89.170) |
| 15:45.15 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:49.02 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 15:49.14 | *** join/#asterisk bazhang (n=bazhang@unaffiliated/bazhang) |
| 15:51.30 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 15:59.06 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 16:00.20 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 16:00.43 | |pepesz| | Hi all, I'm using Asterisk 1.6.2.0-beta3. Does anyone has " i extension" working ?? |
| 16:01.24 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 16:02.33 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 16:02.57 | [TK]D-Fender | |pepesz|: feel free to show us your actual problem... PAStebIN is yrou friend <- |
| 16:02.58 | [TK]D-Fender | ~pb |
| 16:02.59 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 16:04.10 | |pepesz| | exten => _5[0-46-9],1,Dial(SIP/${EXTEN},30,r) |
| 16:04.10 | |pepesz| | exten => _5[0-46-9],2,VoiceMail(${EXTEN},u) |
| 16:04.10 | |pepesz| | exten => i,1,NoOp(An invalid number ${INVALID_EXTEN} was dialed.) |
| 16:04.10 | |pepesz| | exten => i,2,Answer() |
| 16:04.10 | |pepesz| | exten => i,3,Playback(invalid) |
| 16:04.11 | |pepesz| | exten => i,4,Hangup() |
| 16:04.34 | nny_1 | quick q, passing any faxes through * is frowned upon for various reasons, some resolved. I have a potential client with a couple of T1s and they have 20 fax DIDs over them as well. Whats the most efficient way of handling that with asterisk? I assume FXS ports etc. would be a kludge and can create issues, is there a "breakout" setup. Just looking for advice, thanks |
| 16:05.24 | |pepesz| | when calling 55 asterisk doesn't play "invalid extension" message |
| 16:05.25 | [TK]D-Fender | |pepesz|: PAStebiN, do not spam in here |
| 16:05.51 | [TK]D-Fender | |pepesz|: Is that being dialed raw from a SIP phone, or via an IVR? |
| 16:06.14 | |pepesz| | from a sip phone |
| 16:06.51 | [TK]D-Fender | |pepesz|: "i" is not used i this case, ever. the phone is reported "SIP 404". You'll need a catch-all like _X instead |
| 16:07.01 | [TK]D-Fender | _x. |
| 16:07.02 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 16:07.17 | [TK]D-Fender | |pepesz|: this is normal behavior since the beginning. |
| 16:07.51 | |pepesz| | I see - when 'i' is then used ? |
| 16:08.26 | [TK]D-Fender | |pepesz|: in IVR's, and by Zap/DAHDI FXS channels |
| 16:08.37 | vAd0r | can i limit the number of users in a conference? |
| 16:08.55 | [TK]D-Fender | vAd0r: Its your dialplan, do whatever you want. |
| 16:09.10 | vAd0r | i mean |
| 16:09.20 | vAd0r | if i have a DID pointing to a confrence bridge |
| 16:09.28 | vAd0r | i only want to allow like 5 users per conf |
| 16:09.35 | [TK]D-Fender | vAd0r: DID's don't point to anything |
| 16:09.39 | kaldemar | vAd0r: see GROUP functions |
| 16:09.41 | vAd0r | sry my route |
| 16:09.54 | [TK]D-Fender | vAd0r: And "conference bridge" isn't a really meaningful term |
| 16:09.55 | |pepesz| | [TK]D-Fender: thanks, I followed "the asterisk book" which a little misguided me ;) |
| 16:10.14 | [TK]D-Fender | vAd0r: And "route" doesn't mean anything here either |
| 16:10.33 | vAd0r | neither does your words |
| 16:10.47 | vAd0r | i just asked a simple question if you dont have an answer dont say anything |
| 16:10.54 | nny_1 | anyone have any experience with http://wiki.sangoma.com/t1e1analogfaxing |
| 16:11.03 | [TK]D-Fender | vAd0r: problem is you keep dumping freePBX garbage terminology here. |
| 16:11.27 | [TK]D-Fender | vAd0r: Odds are you're referring to MeetMe, and you can make your own dialplan that counts people going in and choose to limit them |
| 16:12.09 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:13.44 | *** join/#asterisk apeiron (i=apeiron@isuckatdomains.net) |
| 16:14.49 | [TK]D-Fender | vAd0r: "core show function GROUP" |
| 16:18.54 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:19.03 | nny_1 | [TK]D-Fender: have you ever dealt with passing fax through an asterisk system locally? (ex: T1 -> FXS) |
| 16:19.27 | nny_1 | [TK]D-Fender: i understand the general rule is don't do it, just wondering if there are setups that are known to be reliable at all |
| 16:19.56 | *** join/#asterisk BugKhaM (n=BugKhaM@125.25.36.14.adsl.dynamic.totbb.net) |
| 16:20.29 | coppice | nny_1 use sangoma cards, which allow clock syncing, or use a digium card and a channel bank |
| 16:21.22 | nny_1 | coppice: yeah we found the sangoma e1/t1 sync product and considering that, i'll have to look at the channel bank. My concern is reliability, but don't want to assume it will never work |
| 16:21.29 | *** join/#asterisk ArchGT (n=ArchGT@190.148.61.54) |
| 16:21.54 | vAd0r | I wasnt refering to meet me |
| 16:22.13 | vAd0r | yes it is freepbx |
| 16:22.18 | *** part/#asterisk ArchGT (n=ArchGT@190.148.61.54) |
| 16:22.20 | nny_1 | hides |
| 16:23.16 | ManxPower | nny_1: your best bet is to have the fax lines direct to the telco. That way they cannot point to asterisk as the cause of fax issues. |
| 16:23.53 | ManxPower | And believe me they will blame every fax issue on Asterisk. |
| 16:24.02 | nny_1 | ManxPower: yeah that's generally what I have been taught, hmm just wondering how to handle it when the client has a PRI and DIDs for current fax stuff |
| 16:24.30 | ManxPower | nny_1: At my old employer we just had the telco move those DIDs to POTS lines. |
| 16:24.57 | *** join/#asterisk T3nE (n=notte@host153-219-dynamic.1-79-r.retail.telecomitalia.it) |
| 16:25.05 | ManxPower | The slight additional cost was nothing compared to what they were paying me to troubleshoot faxing problems. |
| 16:25.29 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:26.05 | nny_1 | ManxPower: any opinion on the sangoma based timing stuff? |
| 16:26.16 | nny_1 | ManxPower: http://wiki.sangoma.com/t1e1analogfaxing |
| 16:26.26 | ManxPower | nny_1: "timing stuff"? |
| 16:26.37 | nny_1 | ManxPower: bad choice of words, gave clicky |
| 16:27.12 | ManxPower | have no opinion on that at all. |
| 16:27.57 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:28.01 | nny_1 | k.. hmm well. I see both points, yeah I can agree with not wanting to have to play middle man on the fax issues by just avoiding it. I'll think about it thanks for the input |
| 16:28.16 | *** part/#asterisk bazhang (n=bazhang@unaffiliated/bazhang) |
| 16:28.25 | ManxPower | The system we ended up with was setup to faxdetect on DIDs, sent the call to rx_fax if fax tone was detected. If a person had problems receiving faxes via that system, they were instructed to use the office fax machine. |
| 16:28.47 | ManxPower | (the office fax was on a dedicated line) |
| 16:29.36 | ManxPower | We set this all up long before Digium redesigned their cards. I expect faxing would be more reliable with modern cards. Still, I don't need the headaches. |
| 16:30.30 | coppice | FAXing won't be any more reliable with modern Digium cards, as they still can't sync the E1/T1 cards to the analogue ones |
| 16:30.38 | ManxPower | The DID+rxfax seemed to work with about %90 of incoming faxes (using Sangoma card) |
| 16:31.02 | *** join/#asterisk Pazzo (n=ugelt@p549468A7.dip.t-dialin.net) |
| 16:32.12 | ManxPower | coppice: what about T-1 + channel bank? (assuming all ports are on the same card) |
| 16:33.02 | coppice | that should be OK, but I don't think digium have provided any hardware sync between digital and analogue cards. sangoma have |
| 16:33.31 | *** part/#asterisk |pepesz| (n=kvirc@77-120.ipact.nl) |
| 16:34.11 | nny_1 | From a technical standpoint is the timing the main issue, or is there other apsects of passing it through * that can cause trouble? This is assuming no latency other than hardware locally (I.e. over LAN/WAN) |
| 16:37.15 | ManxPower | coppice: Sangoma always seems to be one step ahead of Digium |
| 16:38.45 | coppice | that lack of synchronisation has been pathetic from day 1 |
| 16:38.53 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 16:43.37 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 16:45.04 | *** join/#asterisk makafre (n=makafre@64.86.141.133) |
| 16:47.06 | [TK]D-Fender | vAd0r: Either way this is still jus dialplan that you can limit for yourself. So what is it you're referring to as a "conference bridge", if not "MeetMe"? |
| 16:47.31 | ManxPower | [TK]D-Fender: he's not using Asterisk |
| 16:48.14 | [TK]D-Fender | ManxPower: Yes, FreePBX is :) |
| 16:48.44 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 16:48.49 | ManxPower | [TK]D-Fender: When did you start drinking the Kool-Aid? |
| 16:49.01 | [TK]D-Fender | ManxPower: Which? |
| 16:49.18 | ManxPower | The "FreePBX is Asterisk" kool-aid |
| 16:49.36 | [TK]D-Fender | ManxPower: I was AGREEING with you. HE isn't using Asterisk, FreePBX is. |
| 16:49.49 | ManxPower | [TK]D-Fender: Oh! You had me worried there for a min. |
| 16:49.51 | [TK]D-Fender | ManxPower: Fear not. |
| 16:53.12 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:53.39 | *** join/#asterisk davevg-btwtech (n=davevg__@67.76.177.147) |
| 16:55.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:00.32 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-144-249.lns10.mel4.internode.on.net) |
| 17:01.00 | *** join/#asterisk willianmazzardo (n=willianm@201-11-236-6.smace701.dsl.brasiltelecom.net.br) |
| 17:01.09 | *** part/#asterisk willianmazzardo (n=willianm@201-11-236-6.smace701.dsl.brasiltelecom.net.br) |
| 17:04.32 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 17:05.48 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 17:09.42 | angryuser | hm, i have reinstalled zaptel for a trixbox ce, but looks like they hided zaptel well, after fresh recompile it is still using old one, how cani find "another" zaptel thank you |
| 17:11.52 | *** join/#asterisk pulpster (n=pulpster@p16.eregie.pub.ro) |
| 17:11.52 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 17:14.56 | makafre | guys, any known problem with dtmf and SIP within 1.4.25.1? |
| 17:15.59 | makafre | i keep getting repeated DTMF |
| 17:16.30 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:16.48 | carrar | double dtmf? |
| 17:16.56 | makafre | quadruple |
| 17:17.03 | makafre | :- ) |
| 17:18.24 | carrar | Your phone is sending dtmf in band and rfc2833? |
| 17:18.48 | carrar | that was a bug |
| 17:18.52 | makafre | rfc2833, yes |
| 17:19.06 | Qwell | what kind of phone? |
| 17:19.10 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:19.11 | makafre | zoiper |
| 17:19.35 | makafre | when I switch to IAX2 it goes well |
| 17:19.43 | carrar | https://issues.asterisk.org/view.php?id=13209 |
| 17:19.46 | carrar | add the patch |
| 17:20.21 | *** part/#asterisk bluregard (n=matt@66.251.248.60) |
| 17:21.01 | makafre | I see, but thats from 2008, wasn't included into 1.4.25.1 yet? |
| 17:21.22 | carrar | is it in your source? |
| 17:21.29 | makafre | let me check |
| 17:23.58 | *** join/#asterisk Erol_ (n=x@85.102.196.45) |
| 17:24.31 | RoyCrowder | Anyone have any good articles, PDFs, etc regarding Asterisk and Jabber? |
| 17:26.00 | makafre | carrar: humm, I am not quite, whats the best way to check that out |
| 17:26.12 | makafre | quite sure I meant |
| 17:26.39 | leifmadsen | RoyCrowder: I don't believe anything really exists other than on the wiki, and whatever you can figure out on your own |
| 17:26.49 | Erol_ | anyone know the difference between trixbox and pbxtra? |
| 17:27.10 | RoyCrowder | leifmadsen: thanks, I guess I'll try to pioneer my way through it then. |
| 17:27.24 | Qwell | Erol_: one is crap based on an incredibly old version of Asterisk. the other is crap based on FreePBX and a more recent (but still not latest) version of Asterisk. |
| 17:27.48 | Qwell | sorry, the first should have been "commercial crap" |
| 17:28.00 | leifmadsen | RoyCrowder: I don't remember it being all the difficult to setup -- just follow the sample configuration files |
| 17:31.41 | makafre | DTMF issue patch vs 1.4.25.1: Reversed (or previously applied) patch detected! |
| 17:32.42 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 17:33.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:33.54 | RoyCrowder | leifmadsen: not really worried so much about setup. I am looking more for discussion on features people are implementing, what customers are looking for in features, etc. |
| 17:34.10 | *** join/#asterisk btsteve (n=chatzill@204.10.20.30) |
| 17:34.12 | btsteve | Hey guys i am trying to test CallerID on some international Numbers. Is anyone in the UK, mexico, peru, argentina, austria, brazil, bulgaria, chile, denmark, iseral, new zeland, or japan that could try placing a call so i can see what is displayed? |
| 17:36.07 | carrar | makafre, there is also this: https://issues.asterisk.org/view.php?id=14815 |
| 17:36.23 | makafre | ok |
| 17:36.49 | carrar | might just upgrade to the latest version |
| 17:36.59 | Erol_ | Qwell: which crap is more recent? |
| 17:37.27 | makafre | carrar: well I am already on 1.4.25.1 |
| 17:37.34 | carrar | oh |
| 17:37.54 | carrar | have to ask if that other patch made it in |
| 17:38.04 | carrar | or the change log |
| 17:39.26 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 17:39.48 | Qwell | O.o |
| 17:40.04 | beek | Where does res_cepstral check to see if it can find a voice license? I have Allison license, as well as two ports. They work from the command line but res_cepstral SayText() continues to give me an "Unlicensed" message. |
| 17:40.15 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:40.21 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.25.1 (2009/06/05), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.2.0, dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 17:40.43 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 17:40.49 | [TK]D-Fender | Qwell: PSA :) |
| 17:40.50 | leifmadsen | [TK]D-Fender: updated the dahdi-* versions? |
| 17:41.01 | [TK]D-Fender | leifmadsen: And * 1.4.25 to ".1" |
| 17:41.07 | makafre | carrar: here is a pastebin: http://pastebin.com/d1bec9e7c |
| 17:41.09 | leifmadsen | oh -- I didn't do that? weird |
| 17:41.17 | the_unknown | so.... horribly basic question |
| 17:41.19 | leifmadsen | holy crap my internet just got REAL slow |
| 17:41.21 | the_unknown | how can I set callerid per extension |
| 17:41.39 | the_unknown | (if extension 7000, calls extension 7001, both SIP softphones, how can I make the proper cid show up) |
| 17:41.39 | leifmadsen | the_unknown: callerid= in sip.conf |
| 17:41.43 | [TK]D-Fender | the_unknown: "callerid=" in your device config |
| 17:41.58 | the_unknown | heh... you would think that wouldn've been easier to find via google |
| 17:42.06 | [TK]D-Fender | the_And frebie lesson, don't call "devices" as "extensions". |
| 17:42.13 | [TK]D-Fender | the_unknown: And freebie lesson, don't call "devices" as "extensions". |
| 17:42.37 | leifmadsen | the_unknown: or by reading the sample configuration file and searching for "caller id" :) |
| 17:43.59 | pulpster | hello - I have this problem with cisco 7911 + asterisk (+ chan_sccp.so installed) -> no rtp traffic taking place, no rtp channel allocated, signaling is ok, I call, phone rings, call is established but the is only silance on the line - any ideas what might cause this ? |
| 17:46.28 | Qwell | pulpster: chan_sccp |
| 17:47.15 | pulpster | yes |
| 17:47.24 | Qwell | that wasn't a question |
| 17:47.30 | pulpster | the one from sourceforge |
| 17:47.38 | pulpster | the beta one |
| 17:47.47 | Qwell | you asked what could cause it... |
| 17:47.51 | pulpster | yes |
| 17:47.54 | pulpster | any idea ? |
| 17:47.59 | Qwell | yes, chan_sccp |
| 17:48.00 | ManxPower | pulpster: NAT? |
| 17:48.15 | *** join/#asterisk krondorl (n=chatzill@216.191.33.42) |
| 17:48.19 | pulpster | I am behind a nat, that is correct but all calls take place within local lan |
| 17:48.30 | pulpster | no outsite or incoming call from outside is taking place |
| 17:48.45 | pulpster | I hae no entries in my FW (no iptables entries) |
| 17:48.50 | Qwell | Your problem is chan_sccp. |
| 17:49.10 | ManxPower | pulpster: The reason so few people use SCCP is because Asterisk's SCCP support is not very good. I thought there was a SCCP channel driver included in Asterisk. |
| 17:49.20 | ManxPower | Qwell: What do you suggest he use instead of chan_sccp. |
| 17:49.25 | Qwell | chan_skinny |
| 17:49.45 | ManxPower | Qwell: You must be channeling [TK]D-Fender today. 8-| |
| 17:50.22 | pulpster | how come other people succeded in gettin cisco phone to work using chan_sccp |
| 17:50.27 | Qwell | luck |
| 17:51.47 | krondorl | Greetings all.. We just put a firewall in place, opened all ports to the asterisk box but we are still having problems with the phones registering with asterisk. Anyone have a possible idea what might be the problem? |
| 17:52.27 | pulpster | I know that chan_sccp is not exactly what I've used if I had the money to buy a CME, but do you suggest that using the configuration shown in asterisk book will make my cisco phone comunicate ? |
| 17:52.33 | ManxPower | krondorl: It could be a zillion different causes. My bet is the firewall is NATing all packets running thru it. |
| 17:52.45 | ManxPower | pulpster: stop talking. Start using chan_skinny |
| 17:52.57 | pulpster | ok, I'll try skinny |
| 17:53.46 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 17:53.48 | ManxPower | krondorl: Assuming SIP, then you need to have open whatever ports (UDP) are listed in /etc/asterisk/rtp.conf and port 5060 UDP. If you don't have rtp.conf then Asterisk defaults to 10,000 - 20,000 UDP. |
| 17:54.18 | ManxPower | krondorl: If that's not the problem then a SIP debug posted to pastebin.ca would be needed. |
| 17:54.19 | krondorl | ManxPower: Is that wrong? Our asterisk sip provider told us to turn on the "Consistant NAT" within the firewall. |
| 17:54.37 | ManxPower | krondorl: then you should contact them, as I have no idea what that is. |
| 17:54.58 | ManxPower | krondorl: you really need to read the docs about Asterisk and NAT. |
| 17:55.01 | ManxPower | ~sipnat |
| 17:55.01 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:55.35 | ManxPower | If the server is on a public IP and only the phones are on private IPs then you should just turn nat=yes in sip.conf and it should work. |
| 17:56.05 | pulpster | ...if no entry exists in my FW, does it mean all ports are permitted, regardless of what default ports of asterisk are |
| 17:56.05 | pulpster | ? |
| 17:56.30 | krondorl | Actually, these phone are not connecting to them but to us. all the phones are setup with nat=yes. |
| 17:56.50 | ManxPower | pulpster: check your firewall docs |
| 17:59.13 | [TK]D-Fender | SCCP over NAT = extreme pain |
| 18:04.54 | krondorl | what exactly is pedantic in the sip.conf for? |
| 18:07.37 | ManxPower | krondorl: it should say in sip.conf.sample. |
| 18:08.07 | ManxPower | krondorl: I guess it's time for a SIP trace |
| 18:08.19 | krondorl | <PROTECTED> |
| 18:08.48 | ManxPower | krondorl: In the 8 years of using Asterisk I've never had to use pedantic=yes |
| 18:08.49 | krondorl | Just eading over those docs wight now. |
| 18:10.37 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:10.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:12.30 | krondorl | Ok. NP.. :) |
| 18:12.36 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 18:15.45 | Katty | leifmadsen: YOU |
| 18:15.58 | leifmadsen | Katty: ME! |
| 18:16.06 | Katty | yes'r |
| 18:16.09 | Katty | hugs leifmadsen |
| 18:16.21 | leifmadsen | SYN/ACK HUGS Katty |
| 18:16.37 | Katty | :> |
| 18:17.16 | Katty | whatcha doin |
| 18:20.26 | Katty | wow that much |
| 18:20.38 | leifmadsen | Katty: on a conference call |
| 18:20.43 | Katty | bummer. |
| 18:20.49 | leifmadsen | waiting for my HP DL360 to show up |
| 18:21.56 | Katty | sounds....exciting |
| 18:22.36 | Weezey | leifmadsen: so you found one then? |
| 18:22.48 | leifmadsen | Weezey: aye |
| 18:22.55 | leifmadsen | even getting delivered :) |
| 18:23.03 | Katty | touches up chipped nailposlih |
| 18:23.08 | Weezey | sweet |
| 18:23.18 | krondorl | can I sip trace for a specific sip number? EG: SIP/2209 |
| 18:23.51 | leifmadsen | Katty: wow... me too! |
| 18:24.43 | [TK]D-Fender | ManxPower: I have... |
| 18:24.44 | [TK]D-Fender | ManxPower: It was needed to decode the HEX that Polycom's send for "#" on their invite's |
| 18:25.09 | [TK]D-Fender | krondorl: Little impact for enabling it, I recommend doing so. |
| 18:25.13 | Katty | leifmadsen: what color |
| 18:25.29 | leifmadsen | Katty: metallic electric blue |
| 18:25.30 | Katty | leifmadsen: and brand |
| 18:25.34 | Katty | leifmadsen: NO WAY |
| 18:25.38 | Katty | leifmadsen: i have a blue too |
| 18:27.29 | krondorl | can I get the debug to go to a file instead of my screen?? Way too much information is going past for me to find the one thing I need to find. |
| 18:28.17 | leifmadsen | Katty: http://www.stylebakeryteen.com/2008/11/claires-electric-blue-nail-pol-1.html |
| 18:28.33 | leifmadsen | krondorl: asterisk -cvvvn | tee /tmp/console_output.txt |
| 18:29.41 | krondorl | awesome thanks |
| 18:29.42 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 18:31.40 | ManxPower | leifmadsen: My hair was that color for a short while. |
| 18:31.48 | leifmadsen | nice |
| 18:32.16 | Katty | leifmadsen: that is an awesome color, sir |
| 18:33.06 | leifmadsen | Katty: is sure is! |
| 18:33.07 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:34.17 | *** join/#asterisk aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
| 18:34.52 | aces1up | anyone have a list of packages i need to install asterisk on my centos box? |
| 18:35.16 | leifmadsen | aces1up: see http://astbook.asteriskdocs.org |
| 18:35.24 | leifmadsen | aces1up: it should have a list of all the pkgs necessary to build |
| 18:35.28 | Katty | leifmadsen: i'm wearing a color called Navy Baby, by sally hansen |
| 18:35.39 | Katty | leifmadsen: http://media.photobucket.com/image/sally%20hansen%20navy%20baby/hills78/Indigopic2.jpg |
| 18:35.54 | leifmadsen | Katty: that is a scary colour to me :) |
| 18:36.14 | Katty | leifmadsen: it's a deep deep almost black looking nail lacquer |
| 18:36.20 | Katty | leifmadsen: and i loves it. |
| 18:36.22 | leifmadsen | ya, I see that :) |
| 18:37.53 | Katty | chips pretty quick tho |
| 18:37.55 | Katty | in like 2 days |
| 18:37.56 | Katty | :< |
| 18:37.58 | Katty | el sucko brando |
| 18:38.19 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 18:41.23 | *** join/#asterisk krondorl (n=chatzill@216.191.33.42) |
| 18:41.34 | krondorl | Geez, powere outage there.. |
| 18:41.57 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:44.51 | Katty | bummer, krondorl |
| 18:45.17 | Katty | this top coat needs to dry faster |
| 18:46.30 | *** part/#asterisk ThatKidKel (n=Kelvin@208.110.55.5) |
| 18:47.27 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:52.26 | henk | is it possible to let agents dial something while waiting? |
| 18:53.06 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-aba6b3045860759b) |
| 18:54.33 | henk | i basically want to call asterisk, have some kind of passwort prompt (agentlogin or authenticate are my best choices so far) and be able to take calls from a queue or 'dial-through' asterisk. can anyone give me a hint how to achieve that? |
| 18:58.05 | [TK]D-Fender | henk: If you want to sit around waiting for a call, do your AgentLogin. When they are done they can quit to do toher things like calling out. |
| 18:58.47 | *** join/#asterisk boch (n=fran@200.61.191.9) |
| 18:59.04 | boch | is it possible to register to a SIP provider using an outbound proxy ? |
| 19:00.44 | WindowsUser | like tell bob.com you're someguy@bochs.org? |
| 19:00.48 | WindowsUser | yes |
| 19:01.06 | [TK]D-Fender | boch: * can be globally directed towards a SIP proxy |
| 19:02.11 | boch | when dialing to that peer, im having [Jun 29 15:57:58] WARNING[26390]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 19:02.33 | boch | but i dont see any signaling going out to my provider |
| 19:03.02 | [TK]D-Fender | boch: and I don't see your configs or your complete failed call attempt to see if what you are dialing is even sane <- |
| 19:03.06 | [TK]D-Fender | ~pb |
| 19:03.07 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 19:03.08 | [TK]D-Fender | ^^^^^ |
| 19:04.13 | henk | [TK]D-Fender: i know _that_ is possible, but i want them to dial into asterisk, authenticate, and have a _simple_ choice whether to 'dial-through' or login as an agent. is it possible to login an agent without prompting for the pin perhaps? |
| 19:05.41 | [TK]D-Fender | henk: No. |
| 19:05.49 | [TK]D-Fender | henk: Login always asks for the PIN |
| 19:05.52 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 19:06.06 | WindowsUser | hax it at your own peril? |
| 19:06.14 | [TK]D-Fender | henk: So that means 1 auth to get in at all, then another to enter the queue |
| 19:06.37 | henk | [TK]D-Fender: yes, exactly that's what's bothering me. |
| 19:07.01 | boch | [TK]D-Fender, http://pastebin.com/m2ee799c6 |
| 19:07.28 | [TK]D-Fender | henk: You could probably cheat by having it dial a local channel to pass the DTMF for login. Messgy and it would clutter your CDR, but it'd work |
| 19:07.41 | ManxPower | boch: and the CLI output of a failed call |
| 19:08.08 | ManxPower | [TK]D-Fender: what about a blank password for agent login? |
| 19:08.13 | [TK]D-Fender | boch: and outboundproxy=xxx.xxx.xxx.xxx <- this is not a PEER level option, it is onlly under [general] |
| 19:08.37 | ManxPower | I strongly doubt boch really needs an "outbound proxy" |
| 19:08.42 | [TK]D-Fender | ManxPower: I'd still bet the App is too dumb to allow that. |
| 19:09.16 | WindowsUser | he does if domain.com doesn't match xxx.xxx.xxx.xxx |
| 19:09.17 | ManxPower | [TK]D-Fender: Queues are so complicated, messy, and downright evil that I've always found ways of doing queue-like stuff in other ways |
| 19:09.22 | [TK]D-Fender | boch: And who cares whAt the proxy is... you have NO FRIGGEN HOST |
| 19:09.37 | ManxPower | WindowsUser: no, then he would need SRV records |
| 19:09.56 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 19:10.27 | ManxPower | boch: change outboundproxy= to host= |
| 19:10.53 | ManxPower | and allow=all is just ASKING for trouble. |
| 19:11.12 | ManxPower | Q-1: What's the best way to break SIP? A-1: put in allow=all |
| 19:12.22 | [TK]D-Fender | ManxPower: Nothing wrong with that.. so long as you have all copdecs available :) |
| 19:13.53 | WindowsUser | is there a small app I can run to test a sip call? would sipp be not too too bad to set up and test? I just want to do some probing and see if my SIP port is blocked |
| 19:14.05 | [TK]D-Fender | WindowsUser: X-Lite |
| 19:14.23 | ManxPower | [TK]D-Fender: that's kind of like saying "no need to use a condom when you already have everything you could catch". |
| 19:14.24 | Katty | dear person who's computer i'm fixing, you do realize it will take LONGER for me to fix it if you keep calling me about it every 15 minutes, RIGHT?! |
| 19:14.35 | ManxPower | That's technically true but not a good idea. |
| 19:14.50 | [TK]D-Fender | ManxPower: Only if you don't care about what YOU'LL transmit ;) |
| 19:14.53 | WindowsUser | Fender: I got * locally and want to test from a remote linux box i have access to |
| 19:15.00 | *** join/#asterisk E-Man (n=user@wsip-98-189-241-126.oc.oc.cox.net) |
| 19:15.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:15.39 | [TK]D-Fender | WindowsUser: Same answer or s/X-lite/any other stupid softphone, etc/ |
| 19:16.10 | henk | [TK]D-Fender: yeah, that's a bit too hacky for my taste... |
| 19:16.39 | [TK]D-Fender | henk: Then enter a PIN. |
| 19:16.43 | WindowsUser | henk: then edit your asterisk agent stuff to skip the password? |
| 19:16.55 | [TK]D-Fender | WindowsUser: there is no option. |
| 19:17.23 | [TK]D-Fender | unless its smart and sees if no PW is specified in the agents.conf and it even allows you to |
| 19:18.00 | henk | WindowsUser: i don't trust myself to not make a mistake somewhere and have every caller login to the queue automatically... |
| 19:18.49 | [TK]D-Fender | henk: If you can't be trusted to admin and deploy your own PBX then maybe you should resign. |
| 19:18.56 | WindowsUser | they'd only log in to the queue if thier dialplan path hits the modified agentlogin, dont you have your own auth to avoid that? |
| 19:19.35 | WindowsUser | if he resigns someone even less knowledgeable will be doing it bwahahahahaha |
| 19:19.59 | [TK]D-Fender | WindowsUser: Depends who they give the job to |
| 19:21.01 | *** join/#asterisk luch (n=Dwayne@64.42.227.97) |
| 19:22.19 | luch | i have a sip trunk, audio works fine voip to voip but not when I'm calling pots, i only have one way video. The sip trunk is to a Callmanager which uses 711ulaw, i have only ulaw allowed, I have nat=no there is no nat, what else could be wrong |
| 19:22.53 | WindowsUser | you get video over pots? im jealous |
| 19:23.07 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:23.21 | beek | Is anyone here using res_cepstral with Digium licensed voice/ports who'd be willing to show me the results of: 'ls /var/lib/asterisk/licenses'. |
| 19:23.34 | [TK]D-Fender | wants to know who's GETTING video, even if only 1 way... |
| 19:25.34 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 19:26.54 | Katty | [TK]D-Fender: i'm getting your video. |
| 19:27.01 | henk | [TK]D-Fender: imho it's "good admin practice" to _not_ configure something potentially insecure. and i think having an agent without password on a public is potentially insecure if not used very carefully. |
| 19:27.33 | henk | s/public/public server/ |
| 19:30.26 | [TK]D-Fender | henk: How you auth them is YOUR job. Relying on *'s is not required |
| 19:30.43 | [TK]D-Fender | Katty: Please, there are children in here! |
| 19:31.25 | WindowsUser | henk: i thought you didn't want the agent to ask for a password :) |
| 19:32.37 | henk | WindowsUser: not gennerally. only for this usecase, when the caller already entered a pin |
| 19:32.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:34.09 | [TK]D-Fender | henk: Still perfectly viable to always jsut do your own auth outside of that app. Then again this all hinged on it behaving in the manner taht we guessed MIGHT be allowed. |
| 19:34.21 | WindowsUser | oh so theres other routes to access the AgentLogin command? |
| 19:35.51 | henk | WindowsUser: yes |
| 19:36.53 | henk | [TK]D-Fender: yes, i guess i'll just let the user choose between agentlogin and authenticate+dialthrough... |
| 19:39.31 | *** part/#asterisk E-Man[a] (n=user@wsip-98-189-241-126.oc.oc.cox.net) |
| 19:46.49 | *** join/#asterisk BobPierce (n=BobPierc@216.36.132.162) |
| 19:49.25 | *** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br) |
| 19:51.48 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 19:53.32 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 19:54.00 | VoipForces | what could cause uniqueid to be missing in the mysql cdr entries? |
| 19:57.08 | *** join/#asterisk Dovid (i=David_M_@host-78-158-94-203.wlan-guest.nycmny02.us.sargasso.net) |
| 19:57.15 | *** join/#asterisk hi365 (n=hi365@94.159.177.78) |
| 19:57.35 | VoipForces | I have them in the master.csv but not in the mysql database strange. |
| 19:57.49 | [TK]D-Fender | VoipForces: the fact there are plenty of specific docs on how to ADD them, and you must not have done... |
| 19:58.27 | VoipForces | [TK]D-Fender: :-P just noticed this while writing a script. never noticed it before. |
| 20:04.12 | Katty | [TK]D-Fender: what's for dinner. |
| 20:04.50 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net) |
| 20:05.02 | leifmadsen | VoipForces: field length not long enough? |
| 20:05.14 | [TK]D-Fender | Katty: Samosas |
| 20:05.41 | bryanfe2 | is there a variant of SET or other command, which will do string replacement? (e.g. replace all "hello" with "goodbye" in string "hello johnnie!") |
| 20:05.54 | VoipForces | leifmadsen: No, the MYSQL_LOGUNIQUEID had been removed/overlooked by an other programmer... |
| 20:08.26 | Katty | [TK]D-Fender: deep fried samosa? |
| 20:08.40 | [TK]D-Fender | Katty: Looks more like standard baked |
| 20:08.55 | Katty | [TK]D-Fender: ah right. yum. what variety are you making/getting |
| 20:09.16 | [TK]D-Fender | Katty: Standard potato & veggie |
| 20:09.27 | [TK]D-Fender | Katty: Already picked up a dozen after eating out last night |
| 20:09.27 | Katty | nods |
| 20:09.55 | Katty | a local indian place makes them out here. they're fantabulious |
| 20:10.33 | eppigy | sleepy |
| 20:10.35 | eppigy | PLATANOS |
| 20:10.42 | [TK]D-Fender | Katty: I prefer pakora personally |
| 20:10.57 | [TK]D-Fender | Katty: But samosas are larger & cheaper :) |
| 20:11.07 | Katty | eppigy: your face. |
| 20:14.21 | Katty | [TK]D-Fender: i'm having grilled hamburgers and coleslaw for dinner. |
| 20:15.29 | Pan3D | goes to Katty's for dinner |
| 20:15.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:15.34 | Nugget | huggles katty |
| 20:15.37 | [TK]D-Fender | Katty: So tragically.... white... |
| 20:15.42 | Pan3D | lol |
| 20:15.52 | Katty | [TK]D-Fender: its yummy |
| 20:15.57 | Katty | huggles on Nugget |
| 20:16.27 | [TK]D-Fender | Katty: I can practically see the Baconnaise being put out... |
| 20:16.34 | Pan3D | ewe |
| 20:16.53 | Pan3D | [TK]D-Fender: you watch Daily Show? |
| 20:17.41 | [TK]D-Fender | Pan3D: And The Colbert Report |
| 20:17.59 | Pan3D | haha, yes. Then you know about the Chocolate Chip Pancakes on a Stick :) |
| 20:18.24 | Pan3D | http://www.junkfoodblog.com/2006/07/jimmy-dean-chocolate-chip-pancakes.html |
| 20:18.29 | Pan3D | ph33r |
| 20:23.09 | [TK]D-Fender | \ |
| 20:23.13 | [TK]D-Fender | -+ |
| 20:23.18 | [TK]D-Fender | -+ |
| 20:23.19 | [TK]D-Fender | + |
| 20:23.31 | suma | Why asterisk is not using cgi connection, to send and receive message through http ? |
| 20:26.36 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 20:29.13 | [TK]D-Fender | Checout time, heading home, BBIAB |
| 20:32.08 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 20:54.57 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 20:54.58 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 20:55.22 | tfrew | does asterisk 1.4.x support comfort noise generation? |
| 20:56.04 | Nugget | for enough money they'll send file to your house to make comforting noises. |
| 20:56.21 | tfrew | thats fascinating |
| 20:56.53 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 20:57.30 | tfrew | but, my current voip providor implements silence suppression, so i am looking for a way to have a global comfort noise generator |
| 20:57.50 | tfrew | some ata's i've seen can do it, but i can't seem to find a way to have asterisk do it on every call |
| 21:00.56 | leifmadsen | tfrew: no, I don't believe chan_sip supports comfort noise generation (which is some rfc if I remember correctly) |
| 21:01.46 | tfrew | ok |
| 21:02.03 | tfrew | i'll check out other voip providers then |
| 21:04.12 | *** join/#asterisk BadHAL (n=nn@70.99.106.38) |
| 21:09.53 | Katty | HUNGRY |
| 21:10.58 | Katty | Nugget: will they deliver edibles? |
| 21:11.45 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 21:12.42 | kannan | hello all, Is there a limit to the length of a variable that we can pass while proginating calls thru Asterisk manager Interface , http://pastebin.ca/1478799 |
| 21:12.49 | Nugget | Katty: quite the opposite. file will eat all of your muffins. |
| 21:13.17 | kannan | if , so , any originating , i meant to type |
| 21:13.21 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:14.03 | Katty | file: :< |
| 21:14.13 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 21:15.00 | [TK]D-Fender | Katty: Samosa consumption underway! |
| 21:15.16 | Katty | cries |
| 21:15.18 | Katty | sooo hungry |
| 21:15.51 | *** join/#asterisk RIsa (n=Soap@84-75-148-82.dclient.hispeed.ch) |
| 21:16.03 | *** join/#asterisk luch (n=Dwayne@64.42.227.97) |
| 21:16.17 | RIsa | this is not an asterisk but a general SIP question. has anyone ever used milkfish on dd-wrt here? |
| 21:16.24 | luch | asterisk -> sip trunk -> call manager |
| 21:16.28 | kannan | i am able to pass shorter texts as a variable, but not longer ones , any idea whats the limit to variable length wjhile origination action in manager api |
| 21:16.33 | luch | gettting one way audio when calling pots |
| 21:17.28 | luch | voip calls are fine, but only one ways (can't hear other party, but they can hear me) |
| 21:17.38 | luch | fo pots |
| 21:17.44 | luch | for pots calls |
| 21:18.15 | Katty | goes in search of edibles. |
| 21:18.57 | [TK]D-Fender | luch: how do you get to POTS? |
| 21:19.25 | luch | asterisk -> sip trunk -> call manager -> pots |
| 21:19.50 | luch | i can call a cell phone no problem too |
| 21:19.57 | luch | the call manager uses 711ulaw |
| 21:20.08 | [TK]D-Fender | luch: pastebin a failed call with SIP debug enabled |
| 21:22.46 | luch | is there a way to send the console stuff direct to a file? |
| 21:23.02 | Katty | luch: echo |
| 21:23.57 | luch | from asterisk console |
| 21:24.36 | Katty | system and echo. |
| 21:24.42 | eppigy | BOYYA |
| 21:24.43 | eppigy | BOOYA |
| 21:24.46 | Katty | HI DAVE |
| 21:25.00 | Nugget | Wwhhaatt ddooeess eecchhoo ccaanncceellaattiioonn ddoo?? |
| 21:25.01 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 21:25.09 | Katty | Nugget: idk, i forgot. |
| 21:25.22 | [TK]D-Fender | CANCELS Nugget |
| 21:25.45 | Katty | watches Nugget almost get returned, but alas, the post office shut down first. |
| 21:26.11 | Nugget | do not fold, spindle, or mutilate Nugget. |
| 21:26.22 | Katty | hugging is safe tho |
| 21:26.27 | Katty | and slight skwishing |
| 21:26.28 | RIsa | someone help me with my crappy SIP setup pls ;/ |
| 21:26.59 | Katty | why would i ruin my happy moment of grilled burgers and coleslaw to do that. |
| 21:27.22 | [TK]D-Fender | ... |
| 21:27.25 | [TK]D-Fender | telnet |
| 21:27.26 | Nugget | telnet is eeeeeeevil! |
| 21:27.27 | [TK]D-Fender | :D |
| 21:27.40 | [TK]D-Fender | Nugget: DANCE MAGGOT!!! |
| 21:27.42 | Katty | what are you guys psyhic. |
| 21:28.00 | Katty | SPOOKY ENTANGLEMENT |
| 21:28.10 | Nugget | heh |
| 21:28.23 | tfrew | telnet > ssh |
| 21:28.33 | Katty | :< |
| 21:28.36 | Nugget | ewww |
| 21:28.42 | Katty | tfrew: GET OUT |
| 21:28.56 | Katty | tfrew: <3 |
| 21:29.19 | tfrew | thanks |
| 21:29.29 | tfrew | telnet a sip server |
| 21:29.31 | tfrew | it's fun |
| 21:29.47 | Katty | butbutbut |
| 21:29.53 | Katty | grilled hamburgers :< |
| 21:29.57 | Katty | they have grill lines and everything |
| 21:30.02 | Nugget | SIP is udp. |
| 21:30.04 | luch | im using putty and can't copy the whole call log to pastebin for a failed call |
| 21:30.08 | Nugget | (usually) |
| 21:30.19 | Katty | luch: winscp |
| 21:30.28 | Katty | luch: you will love it |
| 21:30.48 | luch | i use winscp, how can it help me with this? |
| 21:30.54 | Katty | ... |
| 21:31.06 | Katty | okay let me get this straight |
| 21:31.10 | Katty | you have a call log, on a server |
| 21:31.16 | Katty | and a workstation with winscp |
| 21:31.21 | Katty | but you can't figure out how to pastebin the call log?! |
| 21:31.46 | Katty | [TK]D-Fender: where's that bat |
| 21:31.56 | luch | b nice |
| 21:31.58 | luch | :D |
| 21:32.00 | Katty | i am nice. |
| 21:32.22 | Katty | and now full. |
| 21:32.24 | Katty | of hamburger. |
| 21:33.27 | Katty | luch: soooo uh.... |
| 21:33.29 | luch | ok, i'm using putty as console -> how do i put the asterisk cli output into a file |
| 21:33.40 | Katty | luch: why don't you copy the call log from the server to the workstation using winscp |
| 21:33.45 | Katty | luch: and then pastebin it |
| 21:34.18 | ManxPower | I just copy and paste from PuTTY. |
| 21:34.29 | luch | Manpower - its too long |
| 21:34.32 | Katty | also acceptable. |
| 21:34.34 | luch | manxpower |
| 21:34.35 | ManxPower | I might increase the PuTTY scrollback buffer, but other than that. |
| 21:34.47 | ManxPower | I usually set mine to 4096 |
| 21:34.52 | luch | katty i on't have a call log |
| 21:35.07 | ManxPower | luch: /var/log/asterisk controlled by /etc/asterisk/logger.conf |
| 21:35.23 | Katty | luch: yes you do. |
| 21:35.30 | eppigy | NEIN |
| 21:35.37 | Katty | eppigy: your face. |
| 21:35.45 | eppigy | your |
| 21:35.47 | Katty | eppigy: you missed some awesome hamburgers. |
| 21:35.52 | eppigy | u a nein |
| 21:35.58 | eppigy | oh :[ |
| 21:36.00 | Katty | your mom's a nein |
| 21:36.05 | Katty | OH SNAP |
| 21:36.07 | eppigy | that ass is a nein |
| 21:36.11 | eppigy | BOOYA |
| 21:36.14 | Katty | you win. |
| 21:36.24 | eppigy | man |
| 21:36.29 | eppigy | I am starving |
| 21:36.48 | Katty | i know what you want. |
| 21:36.56 | Katty | you want grilled cheese on marbled rye |
| 21:36.58 | Katty | and tomato soup |
| 21:37.08 | luch | manxpower where do i find scrollback buffer |
| 21:37.30 | eppigy | what I want would make you D: |
| 21:37.40 | Katty | eppigy: what doyou want then? |
| 21:37.54 | Katty | cookies and milk?! |
| 21:38.08 | eppigy | that was supposed be to an innapropriate comment |
| 21:38.16 | eppigy | little too well masked |
| 21:38.20 | eppigy | :< |
| 21:38.28 | Katty | do you want to try again? |
| 21:38.29 | ManxPower | luch: see that little scrollbar on the side of your PuTTY window? Use your mouse and move it up. |
| 21:38.35 | ManxPower | on the right side. |
| 21:38.41 | Katty | ManxPower: you are truly awful :P |
| 21:38.53 | ManxPower | Katty: ask a stupid question, get a stupid answer. |
| 21:39.00 | Katty | ManxPower: i likes it :P |
| 21:39.06 | luch | Manxpower - whatever u said u can increase it so i can go back further |
| 21:39.12 | eppigy | girl i want you in a tub of tomatoe soup |
| 21:39.23 | Katty | that sounds horribly messy. |
| 21:39.30 | eppigy | but lots of fun |
| 21:39.33 | Katty | but a much better delivery. |
| 21:39.50 | ManxPower | luch: click on the upper left corner thingy menu and select change options. I could go look and tell you the exact option, but then I'd have to charge you $10 |
| 21:39.50 | eppigy | thanks |
| 21:41.00 | luch | katty do i have to turn debug logging on to get that written to a file |
| 21:41.31 | luch | Manxpower - tahts what i can't find |
| 21:41.33 | ManxPower | luch: stop. pick one. copy/paste from PuTTY scrollback, or copy and paste from /var/log/asterisk/ |
| 21:41.51 | ManxPower | I await your payment via paypal to eric@fnords.org |
| 21:42.31 | eppigy | OPERATE YOUR COMPUTER |
| 21:42.52 | RIsa | oh the americans, always after the money |
| 21:43.21 | ManxPower | RIsa: I don't really care about money until I can't buy the things I want. |
| 21:43.26 | suma | RIsa: As if other country people don't want money ? ! |
| 21:43.49 | RIsa | suma, no, people in bhutan dont want any money |
| 21:44.26 | suma | Do you have enough Power & Technologies around in Bhutan ? |
| 21:44.47 | RIsa | somewhat |
| 21:44.48 | suma | or a good PC running asterisk for thoses are in need ? |
| 21:45.33 | suma | In US we get good power and Technologies, We pay money and country has money to invest on it |
| 21:46.23 | luch | http://pastebin.com/m781d5197 here is the pastebin of a failed call |
| 21:46.33 | *** part/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 21:46.41 | luch | manxpower, please wait - don't eat or drink till you get it lol |
| 21:47.40 | RIsa | suma, ? |
| 21:47.49 | RIsa | your power infrastructure is falling apart? |
| 21:48.09 | RIsa | your investment in alternative energies is a laugh |
| 21:48.28 | Nugget | divert full power to the forward sensor array. reverse the phase of the photon torpedos. |
| 21:48.47 | ManxPower | luch: you have two networks with NO NAT between them, correct? |
| 21:48.59 | luch | yes |
| 21:49.13 | ManxPower | nope, now I see three networks |
| 21:49.54 | ManxPower | 10.2.0.2, 10.4.30.100, and 192.168.0.196 |
| 21:50.22 | suma | RIsa: I'm not american, I'm from a different country though. I can feel the difference |
| 21:50.30 | ManxPower | luch: set canreinvite=no in sip.conf [general] |
| 21:50.39 | ManxPower | then do a "sip reload" |
| 21:50.45 | luch | yes, 10.4.30.0 and 192.168.0.0 are lan |
| 21:51.17 | RIsa | suma, fortunately I'm from a different country too |
| 21:52.19 | suma | RIsa: but not from US, i'm sure |
| 21:52.28 | ManxPower | What is 10.2.0.2? |
| 21:52.28 | RIsa | hell no |
| 21:52.39 | luch | Manxpower - that did it |
| 21:52.43 | luch | the callmanager |
| 21:52.50 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 21:52.58 | luch | my outgoig calls are callerid unknown, how can i change that |
| 21:53.01 | ManxPower | luch: you have nat or a firewall or something messing up reinvites |
| 21:53.22 | suma | luch: CALLERID="Name"<phone> |
| 21:53.25 | luch | is it a problem or will this work |
| 21:53.28 | ManxPower | luch: callerid=Your Name <666> in the sip.conf of the device making the call. |
| 21:53.32 | ManxPower | don't use quotes! |
| 21:54.04 | ManxPower | some phones (specifically Cisco using some versions of their SIP stack) will reject calls with callerid with quotes in it. |
| 21:54.07 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 21:54.19 | suma | ManxPower: oh thanks for the tip. |
| 21:54.25 | ManxPower | luch: I think maybe you need to read The Book |
| 21:54.26 | ManxPower | ~book |
| 21:54.27 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:54.53 | suma | ManxPower: Why CGI is not used instead of AGI ? |
| 21:55.04 | luch | i did that callerid thing alreadt, still showing up unknown |
| 21:55.46 | suma | luch: Are you making call through ISDN ? |
| 21:55.59 | ManxPower | luch: where is it not showing up? |
| 21:56.13 | ManxPower | suma: I have no idea what you are asking. |
| 21:56.54 | suma | ManxPower: I understand, you are familier with the Asterisk Code, asking in terms of functionality wise |
| 21:57.33 | ManxPower | I am not familiar with Asterisk's source code. I am not a programmer. |
| 21:57.51 | suma | ManxPower: Sorry for my question. |
| 21:57.54 | suma | :) |
| 22:03.26 | *** join/#asterisk theflashdrive (n=joshua@99-10-185-166.lightspeed.wepbfl.sbcglobal.net) |
| 22:05.04 | theflashdrive | hey folks, having some trouble getting voicemail to email to work, and I'm out of ideas. can anyone help me troubleshoot? |
| 22:07.07 | ManxPower | theflashdrive: For the most part asterisk will try to hand the message to the local sendmail program, then lets the e-mail system on that host deal with it from there. |
| 22:08.06 | Qwell | suma: what are you asking? your question makes no sense |
| 22:09.31 | theflashdrive | thanks, I understand how that works :) What I'm experiencing though is a bit more odd. sendmail works fine from cli, even typing the same command that asterisk uses along with some arbitrary file as the input. However, nothing seems to happen when I leave a voicemail, the maillog doesn't show activity, and the asterisk logs don't show anything out of the ordinary, even with debug=10 |
| 22:10.19 | theflashdrive | I changed the mailcmd to just cat > /tmp/mailer.out, and mailer.out is zero-length |
| 22:10.34 | theflashdrive | once i leave a voicemail that is |
| 22:10.50 | ManxPower | theflashdrive: so you are sure that the message never actually gets into the mailq? |
| 22:10.58 | theflashdrive | positive |
| 22:11.12 | theflashdrive | as long as I do it outside of asterisk, it works like a charm |
| 22:11.44 | ManxPower | theflashdrive: are you using a real sendmail or a sendmail interface to something like qmail or postfix (both provide a sendmail executable) |
| 22:11.55 | theflashdrive | postfix |
| 22:13.33 | ManxPower | theflashdrive: weird |
| 22:14.04 | theflashdrive | agreed :) |
| 22:15.43 | ManxPower | theflashdrive: the only time I had issues with Asterisk and postfix was after the message was in the mailq |
| 22:15.44 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.217) |
| 22:17.20 | theflashdrive | yeah, that seems to be where most people are pointing. |
| 22:24.41 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 22:25.22 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:26.54 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:27.25 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 22:29.13 | ManxPower | theflashdrive: have you ever done any C programming? |
| 22:30.04 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 22:30.52 | theflashdrive | just modified the exec to "( %s < %s ;# rm -f %s ) &" |
| 22:31.38 | theflashdrive | and recompiled. the astmail-TJ25U0 that was created has 006 permissions |
| 22:31.53 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 22:32.42 | theflashdrive | (that was in app_voicemail.c, btw) |
| 22:33.16 | theflashdrive | to answer your question, yes, I've tinkered with C programming :) |
| 22:34.27 | ManxPower | my suggestion was to take a look at the code and see what you can change to help you debugging. |
| 22:36.31 | theflashdrive | guess we were thinking alike :) |
| 22:38.14 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 22:39.51 | jaytee | ManxPower, hi guy! how've ya been? |
| 22:40.28 | ManxPower | jaytee: unemployed but trying to enjoy summer |
| 22:41.24 | jaytee | wow, that sucks. how long ago did you get laid off? |
| 22:42.56 | ManxPower | jaytee: middle of may. I don't really need to worry until the end of July |
| 22:43.16 | ManxPower | I really hated that job, so I was not all that broke up about it. |
| 22:43.29 | jaytee | yeah, I remember you saying it was boring |
| 22:44.25 | jaytee | I got the impression you were under challenged |
| 22:44.45 | ManxPower | I can always go live at a hippy commune that is between huntsville and Nashville of all else fails. |
| 22:44.52 | alrs | ManxPower: what state are you in? |
| 22:45.04 | ManxPower | I am in Huntsville, AL. |
| 22:45.12 | jaytee | Rocket City |
| 22:45.48 | alrs | ManxPower: they say Texas is the answer |
| 22:46.05 | jaytee | yes, but what is the question? |
| 22:46.11 | ManxPower | alrs: the only people that say Texas is the answer are from Texas. |
| 22:46.26 | alrs | ManxPower: the economy there is supposed to be relatively unscathed |
| 22:46.32 | alrs | I don't like Texas |
| 22:46.39 | alrs | but I don't know that I'd much love Alabama, either |
| 22:47.01 | ManxPower | alrs: Huntsville is the same, quite a bit of nasa and army facilities in huntsville. |
| 22:47.05 | theflashdrive | heh, i've been trying to relocate to tx for the last 2 years |
| 22:47.09 | theflashdrive | without success |
| 22:47.51 | jaytee | yes, Texas is the answer and the question is: "Where can a person like me who exaggerates about the size of everything and thinks they're better'n everyone else find people who thinks like I do?" |
| 22:48.32 | theflashdrive | <- texan. |
| 22:48.40 | theflashdrive | and you ain't better'n me either! |
| 22:48.43 | theflashdrive | ;) |
| 22:49.03 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 22:49.05 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 22:49.42 | ManxPower | I was stuck in a tiny town in rural texas after Katrina. I never disliked texas before that. |
| 22:49.47 | jaytee | I once climbed all 986 feet of Anderson Mountain. Of course a mountain actually is by definition a minimum of 1000 feet height from base to summit but whose really counting? |
| 22:52.21 | theflashdrive | yeah, i grew up in a town whose population sign said "2336." Last time I went back it was 2,100something. It definitely has its small towns. There are also the rule breakers like Austin, which doesn't seem to fit the feel of the rest of the state |
| 22:52.33 | ManxPower | One of the roads I drive occasionally changes elevation 950ish feet in a mile. It is an interesting road. |
| 22:53.22 | jaytee | and humor aside, Dallas is a pretty big city, lots of nice areas around the suburbs. San Antonio is a really nice city, I love the Riverwalk. |
| 22:53.40 | theflashdrive | wouldn't that be nice. we're lucky to see 4 feet here in S. fla |
| 22:54.08 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 22:54.38 | jaytee | ManxPower, wow! 950 ft change in altitude in a mile? that's some steep damn grades. |
| 22:56.06 | ManxPower | jaytee: something like 5 switchbacks |
| 22:56.40 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 22:57.27 | jaytee | sounds like Highway 299 from Eureka, CA on the coast to Redding, CA inland. |
| 22:57.34 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:01.59 | *** join/#asterisk marksman_ (n=marksman@75.81.97.14) |
| 23:03.57 | marksman_ | So I am brand new to asterisk and PBXs in general. My goal is to have an automated system that answers the phone and directs phone calls to where they need to go. It would be nice if this could happen based off of voice interaction. Is this possible with asterisk? |
| 23:05.18 | jaytee | yes |
| 23:06.25 | theflashdrive | marksman_ - haven't tried it myself, but lumenvox is supposed to do what you're looking for. |
| 23:06.35 | jaytee | asterisk can provide IVR functionality but you'd need to use a speech recognition engine that works with asterisk, like Lumenvox. |
| 23:06.44 | marksman_ | what is the difference between asterisk and asterisknow? From what I gather asterisknow is stripped down, smaller, and less robust... Is that accurate? |
| 23:06.57 | jaytee | and Lumenvox is a licensed application |
| 23:07.44 | Qwell | marksman_: one is Asterisk, the other is a distro that includes Asterisk |
| 23:07.46 | theflashdrive | asterisk - the application. asterisknow - operating system with asterisk installed and configured |
| 23:07.59 | *** join/#asterisk PaulTech (i=paultech@66.103.132.86) |
| 23:08.03 | jaytee | asterisk is a non-gui telephony toolkit. asterisknow is a "distro" of asterisk, linux and a gui using either freepbx or asterisk-gui |
| 23:08.18 | marksman_ | so, is there any free pbx software with voice recognition? |
| 23:08.27 | PaulTech | Quick question regarding noojee fax/txfax/rxfax |
| 23:08.32 | PaulTech | Are txfax/rxfax still in use? |
| 23:08.34 | jaytee | it's not stripped down so much as it trades flexibility for ease of use |
| 23:08.44 | PaulTech | marksman_: Asterisk can be that |
| 23:08.57 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-137-227.dsl.stlsmo.sbcglobal.net) |
| 23:09.44 | suma | Is the uniqueid in the CDR is a unique one ? |
| 23:09.53 | PaulTech | strange question |
| 23:09.57 | suma | Yes |
| 23:10.22 | PaulTech | I couldn't give you a complete honest answer. They seem random. No clashes on 2million records on my end |
| 23:11.42 | suma | PaulTech: It is said in the voip-info The uniqueid field is not guaranteed to be unique across the different CDR entries, even though the name suggests exactly that. |
| 23:11.58 | *** join/#asterisk DonAlex (n=DonAlex@glanforn.demon.co.uk) |
| 23:12.15 | DonAlex | Hey Guys.. .Urgent need of someone's expertise.. |
| 23:12.19 | suma | http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 23:12.27 | suma | DonAlex: on what ? |
| 23:12.38 | DonAlex | Have a Asterisk system that need to connect the voicemail system to an out bound call. |
| 23:13.07 | DonAlex | usuaully the client calls into pick up voice mail but there is something wrong with the physical line it is permenantly engaged |
| 23:13.07 | suma | Using Asterisk AMI ? |
| 23:13.09 | PaulTech | Use Orginate command? |
| 23:13.15 | LemensTS | http://pastebin.com/m5383134f How can i send it back to line 3 if its an invalid keypress? |
| 23:13.25 | DonAlex | I have console access and am trying to use originate to send the voicemail to call them |
| 23:13.29 | LemensTS | invalid = not 0,2,or 3 |
| 23:13.39 | DonAlex | Da. teied that but I am used to using it to attach extensions.. |
| 23:13.50 | DonAlex | I am not sure how to connect voicemail the same way |
| 23:14.10 | suma | DonAlex: Are you plain asterisk or any flavors ? |
| 23:14.16 | PaulTech | What extension do they dial to get their voicemail normally DonAlex? |
| 23:14.30 | DonAlex | This is what I use to call out. originate SIP/500/500 extension 0208572xxxx@from-internal |
| 23:14.59 | suma | LemensTS: Use goto |
| 23:15.24 | LemensTS | Suma: im pre 5.3 php |
| 23:15.35 | DonAlex | Can I just replace SIP/500/500 with the voice mail extension? |
| 23:15.42 | DonAlex | I thought I had to use it as an application |
| 23:15.48 | PaulTech | that'll work |
| 23:15.49 | DonAlex | I know I can do that with echo test? |
| 23:15.52 | PaulTech | just Local/Extension |
| 23:15.59 | DonAlex | Local ahhhh |
| 23:16.01 | PaulTech | Local is a valid tech |
| 23:16.11 | marksman_ | So what is the best service to sign up for to get a phone number to test/experiment with a PBX (hopefully one that works well with asterisk)? |
| 23:16.15 | DonAlex | See learn something new every day ;) |
| 23:16.15 | LemensTS | Suma: cuz that would be very nice but most people probably arent on 5.3 so I dont want to restrict it to that.... |
| 23:16.29 | PaulTech | all about learning! |
| 23:16.33 | DonAlex | ok lemme give that a whirl I'll let you know in a sex |
| 23:16.41 | PaulTech | no sex. just a thank you |
| 23:20.03 | PaulTech | (it was a joke people... come back) |
| 23:22.05 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:22.42 | DonAlex | PaulTech: Hmmm not calling me :( originate Local/*98 extension 07956422xxx@from-internal |
| 23:22.52 | DonAlex | PaulTech: Do I need to escape the * ? |
| 23:24.26 | DonAlex | Hello? |
| 23:24.37 | DonAlex | :( |
| 23:25.15 | marksman_ | It is like a knowledge tease fest in here... just enough to get you excited then they are gone.... |
| 23:25.36 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:26.27 | DonAlex | PaulTech: Paul? Any ideas? |
| 23:28.46 | DonAlex | Awww crap does anyone know how I can do this? |
| 23:29.35 | ManxPower | you do not need to escape the * |
| 23:29.47 | ManxPower | other than that I can't help you. |
| 23:31.36 | DonAlex | Ahh no I got it.. |
| 23:31.48 | DonAlex | Local/<number>@context application <name> |
| 23:32.16 | DonAlex | but hang on hwo I geta list of apps.. |
| 23:32.16 | jaytee | marksman_, you can start by looking over this book |
| 23:32.20 | jaytee | ~book |
| 23:32.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 23:32.25 | DonAlex | cause the voicemail app is just asking me to leave voicemail |
| 23:32.33 | DonAlex | not access the system to read back |
| 23:32.43 | marksman_ | jaytee: thanks! |
| 23:32.56 | jaytee | marksman_, and for a SIP account you can check out these ITSPs that work with Asterisk |
| 23:33.05 | DonAlex | Doh |
| 23:33.05 | ManxPower | "core show applications" Notice Voicemail and VoicemailMain |
| 23:33.06 | DonAlex | sorry asleep |
| 23:33.09 | DonAlex | show applications |
| 23:33.11 | DonAlex | Da |
| 23:33.12 | jaytee | ~itsplist-us |
| 23:33.13 | infobot | hmm... itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 23:33.30 | ManxPower | I currently use Vitelity |
| 23:33.31 | DonAlex | I kwon I am actually not a newbie just it is 12:33 here and I have not looked at this in months |
| 23:33.42 | ManxPower | DonAlex: go to bed |
| 23:34.00 | DonAlex | haha |
| 23:34.09 | DonAlex | will do once I connect my client :) |
| 23:35.36 | *** join/#asterisk propellerhead (n=yogurt2u@host47.200-117-135.telecom.net.ar) |
| 23:36.47 | DonAlex | for the record.. (and others.. ) the command needed was this : originate Local/07956422xxx@from-internal application voicemailmain |
| 23:38.01 | DonAlex | Thanks guys.. nice to know there are others around who are more awake than I am... hope I can return the favour someday.. Cheers PaulTech and manxpower :) |
| 23:38.07 | DonAlex | night night all. |
| 23:38.09 | DonAlex | :) |
| 23:38.13 | *** join/#asterisk sigius (n=sigius@93-125-185-45.dsl.alice.nl) |
| 23:41.59 | sigius | Q:not asterisk related per se but im looking for a phone that allows to do 'text chatting (as in icq,irc,msn etc) over a gsm voice connection. Does such a think exists ? |
| 23:44.37 | [TK]D-Fender | sigius: No, but ask your Boy Scout leader if Morse Code is right for you! |
| 23:45.18 | PaulTech | How would you use icq/irc/msn protocol over a voice codec? |
| 23:45.35 | PaulTech | That would be a impressive to see |
| 23:45.54 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:46.42 | sigius | PaulTech: Im mentioning icq/msn to descibe the user experience, not any protocol |
| 23:47.08 | *** join/#asterisk missinglink (n=missingl@ppp166-229.static.internode.on.net) |
| 23:47.16 | sigius | PaulTech: similar to fax over an analog landline |
| 23:52.16 | [TK]D-Fender | sigius: The user experience isn't on the GSM layer its on the one they interact with. If you're going to be aimlessly generic like that you may as well have said "over smoke signals" |
| 23:52.33 | [TK]D-Fender | sigius: For which chan_smokesignal.so is due out of alpha shortly :p |
| 23:52.41 | [TK]D-Fender | BBIAB, VB time.... |
| 23:53.11 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:54.38 | carrar | VB Time? |
| 23:54.42 | carrar | Visual Basic? |
| 23:54.44 | carrar | WTF |
| 23:55.03 | carrar | Veggie Burritos? |
| 23:55.53 | carrar | Virus Bulletin |
| 23:57.16 | sigius | [TK]D-Fender, read back. I did not say I wanted it implement in the gsm layer. |
