| 00:02.51 | blitzrage | Qwell: hey... when you get a chance, can you check out my comments in 11437 and tell me if that is a "fix" or not? |
| 00:06.00 | Qwell | funky...logger stuff? |
| 00:06.30 | blitzrage | Qwell: ya... I tried putting logger.h back in below the #ifdef, but it didn't help |
| 00:08.58 | Qwell | I would expect that to happen always - it doesn't look related to imap at all |
| 00:09.07 | *** join/#asterisk mog (i=mog@nat/digium/x-8173594dedc301f2) |
| 00:09.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:09.19 | blitzrage | Qwell: only happens when I do --with-imap though |
| 00:09.47 | Qwell | trunk? |
| 00:09.53 | blitzrage | yes trunk |
| 00:09.55 | blitzrage | doesn't happen in 1.4 |
| 00:10.05 | blitzrage | (because of the way it is structured I guess... ?) |
| 00:12.24 | Qwell | latest trunk, no patches? |
| 00:12.29 | blitzrage | latest, no patches |
| 00:12.32 | blitzrage | as of... 10 mins ago |
| 00:13.10 | blitzrage | ./configure --with-imap=/usr/local/src/imap-2006k |
| 00:13.29 | blitzrage | tried --with-imap=system as well on RHEL5, and it gave me the same results |
| 00:13.34 | blitzrage | which is 2004g |
| 00:14.28 | Qwell | neat the top, it tries to #include 3 imap headers |
| 00:14.30 | Qwell | near* |
| 00:14.41 | Qwell | see if one of those are defining LOG_WARNING |
| 00:15.43 | Qwell | it is...that's awesome |
| 00:16.45 | blitzrage | Qwell: yah -- that's what Paul and I found |
| 00:17.00 | blitzrage | both asterisk and imap are defining LOG_WARNING, so they are stepping on each other |
| 00:17.00 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 00:17.44 | Qwell | yeah, that'll work, but it may not be the right fix. and the warning on 4287 is something completely different |
| 00:18.21 | blitzrage | indeed... which I wasn't getting before |
| 00:18.32 | blitzrage | I wasn't sure what the right fix was... but moving the ifdef at least made it compile without the morning |
| 00:19.05 | *** part/#asterisk scottpayne (n=sp@n128-227-11-151.xlate.ufl.edu) |
| 00:19.07 | Qwell | bbiab |
| 00:19.10 | blitzrage | cool |
| 00:19.21 | JT | Found peer '09185590' |
| 00:19.21 | JT | Reliably Transmitting (no NAT) to 125.213.160.81:5060: |
| 00:19.21 | JT | SIP/2.0 403 Forbidden |
| 00:19.49 | JT | what's the workaround for when asterisk matches the wrong peer/registration with an incoming call? |
| 00:19.59 | JT | it's just going with the last registration |
| 00:20.03 | JT | not the best match |
| 00:20.09 | JT | (ie. the correct number) |
| 00:21.45 | blitzrage | JT: do you just have peers defined? |
| 00:22.02 | blitzrage | or are they friends? (i.e. users and peers) |
| 00:22.21 | blitzrage | it should try to match on the user names first, followed by the IP address of the peers, starting from the bottom of the file |
| 00:23.02 | JT | friends |
| 00:23.14 | JT | it's going from the bottom |
| 00:24.53 | *** join/#asterisk nhuisman_work (n=nhuisman@dhcp54.IfA.Hawaii.Edu) |
| 00:25.00 | blitzrage | right... do you have a [09185590] in sip.conf? |
| 00:25.14 | nhuisman_work | is there a best way to handle failover for asterisk servers? |
| 00:25.23 | nhuisman_work | besides just rsyncing the configs and restarting the secondary |
| 00:25.58 | nhuisman_work | keep the configs on shared storage? But then are there any host specific items in the config that would screw it up? |
| 00:26.23 | JT | blitzrage: sure do |
| 00:26.34 | JT | thing is the number that is in the To: field is 09185589 |
| 00:27.53 | blitzrage | that might be why it isn't matching on the user, and is instead matching on the peer (ip address) |
| 00:28.04 | blitzrage | insecure=invite help any? |
| 00:28.29 | JT | sorry, what might be why? |
| 00:28.47 | blitzrage | JT: the To: not matching the section name |
| 00:28.52 | nhuisman_work | no one? |
| 00:29.10 | blitzrage | nhuisman_work: I use SVN to keep my configs synced, the just keep the unique files for each system out of SVN |
| 00:29.36 | nhuisman_work | yeah but how do you handle the server crashing? You manually fail over? |
| 00:29.56 | blitzrage | nhuisman_work: LinuxHA |
| 00:30.03 | blitzrage | i.e. heartbeat |
| 00:30.23 | nhuisman_work | so when you change anything you change it twice? |
| 00:30.25 | *** join/#asterisk killfill (n=killfill@200.55.220.3) |
| 00:30.27 | killfill | hi! |
| 00:30.36 | blitzrage | nhuisman_work: no, I change it once, commit to SVN, then sync to the SVN server |
| 00:31.14 | nhuisman_work | have you ever had your system fail and go into failover mode? |
| 00:31.19 | nhuisman_work | well I assume you've tested it |
| 00:31.21 | nhuisman_work | but in real life |
| 00:31.38 | *** join/#asterisk flujan (n=flujan@201-68-192-6.dsl.telesp.net.br) |
| 00:31.48 | fujin_ | Anyone got any tutorials for using perl to interface with AMI, over a local socket? |
| 00:31.49 | killfill | when doing AMI, the ActionID is something the server set?.. or its the client? |
| 00:32.14 | JT | blitzrage: |
| 00:32.33 | JT | To: <sip:09185589@ |
| 00:32.34 | JT | Found peer '09185590' |
| 00:32.47 | JT | yes insecure=invite allows the call to come in |
| 00:32.50 | JT | on the wrong peer |
| 00:32.51 | killfill | becouse im playing with a .NET library, and in one of the computer, i see negative values for it, and it fails to login. |
| 00:32.55 | killfill | :S |
| 00:32.55 | blitzrage | nhuisman_work: yes, I've tested it -- but I use the HA for OpenSER which sits in front of the Asterisk boxes, then a script pings the Asterisk boxes and sees if they are up, and if not, removes them from the dispatcher.list file --OpenSER does the distribution to multiple boxes (cluster). But if an OpenSER box goes down, yes, it actually works. |
| 00:33.01 | killfill | why set the actionid? |
| 00:33.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:33.29 | *** part/#asterisk flujan (n=flujan@201-68-192-6.dsl.telesp.net.br) |
| 00:33.32 | nhuisman_work | openser is a sip load balancer eh |
| 00:33.40 | blitzrage | yes... my topology is different than yours |
| 00:33.44 | blitzrage | but the HA does work |
| 00:34.01 | JT | openser is a sip proxy. |
| 00:34.04 | blitzrage | OpenSER 1 dies.... IP address is taken over by OpenSER 2 |
| 00:34.17 | blitzrage | OpenSER 2 then continues to take calls |
| 00:34.36 | JT | blitzrage: any ideas on why it's picking the wrong peer? |
| 00:34.42 | [hC] | this is very strange. I have hints in asteriskserver1 that say exten => 3101,hint,SIP/3101@asteriskserver2 |
| 00:34.44 | killfill | why/who.. |
| 00:34.54 | [hC] | so i can provide hints on asteriskserver1 based on the status of an exten @ asteriskserver2 |
| 00:34.57 | [hC] | this used to work, now it does not. |
| 00:35.02 | blitzrage | JT: ok... the To: is 09185589 (not 90) -- you have the 'user' defined? |
| 00:35.21 | JT | it's a friend |
| 00:35.31 | blitzrage | JT: other than that... I have no idea |
| 00:35.42 | blitzrage | it's obviously not finding the username |
| 00:35.47 | JT | i seem to remember hearing this is an asterisk bug |
| 00:35.57 | blitzrage | ya well, that's what everyone seems to hear |
| 00:36.05 | JT | that it matches calls from the one host with the most recent registration |
| 00:36.15 | JT | not the correct to: field |
| 00:36.27 | blitzrage | open a bug? |
| 00:36.51 | JT | if it hasn't been resolved, this bug has been open for over 3 years |
| 00:37.00 | blitzrage | JT: I still don't know |
| 00:37.09 | killfill | hm.. negative has nothign to do with it.. :S |
| 00:37.14 | JT | hmm |
| 00:37.24 | killfill | i see "Missing action in request" from the server.. :S |
| 00:37.34 | fujin_ | Anyone know if you can echo " blah blah" shit into the asterisk.ctl, as a manager socket? |
| 00:37.36 | killfill | im sengin action: Challenge... |
| 00:37.41 | fujin_ | or do you actually have to do it over the real socket |
| 00:39.20 | killfill | http://pastebin.ca/805058 <--- that the data i see passing |
| 00:39.43 | killfill | looks like its doing two requests? |
| 00:40.31 | killfill | hm.. the first one is 3 bytes.. thats the wrong one.. |
| 00:43.51 | JT | blitzrage: it looks like no-one has bothered to fix the bug, people just rely on insecure= and dialplan matching..... |
| 00:44.46 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 00:45.27 | [hC] | im sure this worked at one point, but should it even be possible to define a hint like exten => 123,hint,SIP/123@some_other_asterisk_box ? |
| 00:45.29 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ced59bcb431097b2) |
| 00:45.30 | *** join/#asterisk salzh (n=salzh@124.77.15.177) |
| 01:03.39 | *** join/#asterisk BadHorsie (n=sebas@ip254-10.ct.co.cr) |
| 01:04.14 | BadHorsie | i wonder if anybody knows where the FOP svn repository located... |
| 01:05.43 | [TK]D-Fender | [hC], nope |
| 01:06.40 | BadHorsie | wait, FOP is not open source? |
| 01:07.26 | nhuisman_work | anyone here got any reasons why asterisk might be better then SIPxchange EC? |
| 01:07.49 | *** join/#asterisk RoyK (n=roy@ip-26-2-149-91.dialup.ice.no) |
| 01:08.07 | [hC] | [TK]D-Fender: thats so weird, i implemented it in early 1.2, and im sure it worked..... it seems the only way to do it currently is using func_devstate in asterisk 1.4 |
| 01:09.38 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 01:10.19 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 01:11.21 | mosty | is there a priority like the n priority, but for the "current" priority? ie the same as n except it doesn't increment? |
| 01:11.48 | [hC] | mosty: yes, "n" with nothing after it. |
| 01:11.49 | [hC] | :) |
| 01:12.06 | mosty | [hC], if i do that, then it increments |
| 01:12.09 | DarkRift | "s" |
| 01:12.23 | [hC] | mosty: what are you actually trying to do? your description is kind of confusing. |
| 01:12.52 | mosty | [hC], i want to create a bunch of extensions with the same priority, and i don't want to explicitly number them |
| 01:13.10 | [hC] | mosty: i cant understand why you would want to do that. |
| 01:13.27 | [hC] | mosty: because then you couldnt address them. |
| 01:13.31 | [hC] | unles you mean n(label) |
| 01:13.37 | mosty | i am using labels |
| 01:13.38 | *** join/#asterisk RoyK (n=roy@ip-26-2-149-91.dialup.ice.no) |
| 01:13.53 | mosty | i don't want these to be addressed, i don't want to use explicity numbering |
| 01:14.09 | [hC] | so how do you pick which one you want? |
| 01:14.23 | mosty | however, i don't see how i can create a bunch of extensions of the same priority without explicitly numbering them |
| 01:14.23 | [hC] | if you just have them all the same priority, which one gets executed? |
| 01:14.36 | mosty | the one that matches |
| 01:14.49 | [hC] | can you actually explain what you are trying to accomplish, then i might understand what you need |
| 01:15.04 | mosty | eg exten => 123,X,Foo , exten => 456,X,Foo |
| 01:15.41 | [hC] | and why does X=1 not work here? |
| 01:15.55 | mosty | because i'm trying to remove explicitly numbered priorities |
| 01:16.01 | [hC] | yes i get it. why? |
| 01:16.18 | [hC] | thats what i dont get, i cannot understand why you would want that, or how it could help anything |
| 01:16.20 | mosty | because explicitly numbered priorities are a pain to maintain |
| 01:16.38 | [hC] | thats what n(label) is for. if you dont use numbers, you have to use SOMETHING to be able to say which one to go to |
| 01:17.05 | [hC] | you either step through them sequentially with numbers, or sequentially with n and n(label) |
| 01:17.35 | mosty | so you're saying there's no priority like n that doesn't increment? |
| 01:17.43 | DarkRift | mosty use "s" instead of "n" |
| 01:18.03 | [hC] | DarkRift: unless im misisng something, you cant use 's' there |
| 01:18.07 | mosty | DarKnesS_WolF, there's an s priority? |
| 01:18.11 | DarkRift | ; The priority "same" or "s" means the same as the previously specified |
| 01:18.11 | DarkRift | ; priority, again regardless of whether the previous entry was for the |
| 01:18.11 | DarkRift | ; same extension. |
| 01:18.46 | DarkRift | taken directly from the extension.conf example |
| 01:18.51 | DarkRift | sample |
| 01:19.05 | mosty | DarkRift, that is exactly what i wanted, thankyou |
| 01:19.39 | [hC] | thats messed up. I cant understand where you'd want that |
| 01:19.41 | [hC] | but ill have to go look now |
| 01:20.21 | mosty | [hC], have you ever had a dialplan with multiple extensions at the same priority? |
| 01:20.44 | BBHoss | anyone know what 481 Call Leg/Transaction Does Not Exist is? |
| 01:21.35 | JT | it means it's responding to sip wrt a call that it cannot find |
| 01:21.38 | *** join/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz) |
| 01:21.45 | JT | sip traffic received from the far end |
| 01:22.06 | iamthelostboy | hi.. can someone point me in the right direction of an explaination of how to setup voicemail to email? |
| 01:22.26 | *** join/#asterisk BCS-Satori (n=BCS-Sato@68.34.28.26) |
| 01:22.39 | iamthelostboy | i have the email address' in voicemail.conf, but i dont know how to get it to access the smtp server to send them.. |
| 01:23.05 | BBHoss | JT, wrt? |
| 01:23.23 | mosty | iamthelostboy, it uses the smtp server on localhost |
| 01:23.31 | JT | with respect to |
| 01:23.48 | BCS-Satori | Evening, I am having a slight issue with conference channels (moh) When a user connects to a conference room and is the only person there or Wait for leader is set, they hear a quick 1 second blurp of music and then it is pure silence. MoH for phone calls works though. Any ideas? |
| 01:23.54 | BBHoss | well if i recieve that from an itsp, what might the problem be? |
| 01:24.13 | JT | that they're not getting some of your traffic |
| 01:24.22 | JT | or you are referencing bogus calls |
| 01:24.32 | BBHoss | hmm, that would explain no audio then |
| 01:24.45 | iamthelostboy | so i'll have to figure out how to set that up.. :) |
| 01:24.46 | iamthelostboy | cheers |
| 01:25.09 | BBHoss | the call works fine, just no audio on either end, AND i have nat set up |
| 01:25.35 | fujin_ | Is the correct way to use iax channels IAX/ or IAX2/? |
| 01:26.12 | mosty | iax2 |
| 01:28.32 | fujin_ | hrmp |
| 01:28.36 | fujin_ | this is odd |
| 01:28.51 | fujin_ | have configured a friend in iax.conf, but can't originate iax2/bla |
| 01:29.00 | [hC] | mosty: you mean exten => 123,4,blah and exten => 345,4,blah? yes of course, but i just use numbers. do you simply want to simplify your dialplan so that at priority 4 or 5 or whatever, you can use the same dial plan entry between multiple extens so you dont have to duplicate code? |
| 01:30.21 | mosty | [hC], i want to remove numbers. i use the n priority almost everywhere, except in this situation i use n for the first extension and s for the other extensions (those that should be the same priority) |
| 01:31.07 | mosty | it makes the dialplan a lot easier to maintain. didn't you ever learn to program in BASIC? remember how shitty it was remembering what line number you jump to everywhere? |
| 01:31.23 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 01:31.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:37.56 | fujin_ | Eh |
| 01:37.59 | fujin_ | to use originate |
| 01:38.13 | fujin_ | can I do originate IAX2/peer/number application Playback tt-monkeys? |
| 01:38.17 | fujin_ | doesn't seem to work properly |
| 01:41.51 | [TK]D-Fender | fujin_, pastebin everything |
| 01:43.54 | fujin_ | [TK]D-Fender: http://rafb.net/p/RR6gLg73.html |
| 01:44.35 | [TK]D-Fender | fujin_, not patebin your full AMI data and CLI output |
| 01:44.51 | fujin_ | I'm just using 'originate' on the CLI. |
| 01:44.56 | fujin_ | http://rafb.net/p/mmVJac39.html |
| 01:44.58 | fujin_ | is cli output |
| 01:45.04 | fujin_ | 006121921741 is my cellphone number |
| 01:46.33 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 01:47.15 | *** join/#asterisk BeeBuu (n=beebuu@61.145.77.158) |
| 01:48.24 | fujin_ | [TK]D-Fender: any ideas? |
| 01:48.37 | [TK]D-Fender | fujin_, IIRC that uses DSP for it, check to see if a channel is lingering, and you should try to use AMI instead... |
| 01:48.55 | fujin_ | It doesn't use DSP - I can do it on my other system and it works |
| 01:49.06 | [TK]D-Fender | fujin_, though it does LOOK like its doing it. Now does it appear that it plays weasels INSTANTLY after the dial? |
| 01:49.15 | fujin_ | yes |
| 01:49.20 | fujin_ | I'm wondering if it's an issue with the peer |
| 01:49.25 | fujin_ | unfortunately i'm remote to it, it's a little hard to test |
| 01:49.49 | [TK]D-Fender | fujin_, It could be that your ITSP is counting the call as answered the moment they accept if from you |
| 01:50.01 | [TK]D-Fender | fujin_, probably so they can bill you for "ringing time" as well |
| 01:50.42 | fujin_ | heh |
| 01:50.43 | fujin_ | yeah |
| 01:50.44 | fujin_ | looks like it |
| 01:50.47 | [TK]D-Fender | fujin_, which if thats the case this idea is dead in the water |
| 01:50.49 | fujin_ | getting a different error now when I type it differently |
| 01:51.04 | fujin_ | if I do originate IAX2/006121921741@voipstreet application Playback tt-monkeys |
| 01:51.07 | fujin_ | [Dec 3 20:36:13] WARNING[21784]: chan_iax2.c:7194 socket_process: Call rejected by 64.136.174.24: No authority found |
| 01:51.51 | fujin_ | I'm not too familiar with IAX2. |
| 01:52.51 | fujin_ | [TK]D-Fender: I will use AMI, eventually; just wanted to try and get calls running first |
| 01:52.56 | fujin_ | it'd appear there's some sort of error with the itsp |
| 01:54.06 | [TK]D-Fender | fujin_, I always advise "tech/peer/number" as opposed to "tech/number@peer" for reasons like the "error" you just noticed |
| 01:54.16 | [TK]D-Fender | fujin_, and its not the ITSP./ |
| 01:54.27 | fujin_ | Oh, it's tech/peer/number? damn |
| 01:54.29 | fujin_ | sec |
| 01:55.37 | fujin_ | cool |
| 01:55.40 | fujin_ | looks like an itsp issue somehow |
| 02:01.04 | *** join/#asterisk PepOSX (n=pepOSX@190.72.149.98) |
| 02:02.13 | *** join/#asterisk bryanfe2 (n=chatzill@74.10.9.114) |
| 02:02.29 | fujin_ | [TK]D-Fender: wrong dialing format, that's why |
| 02:02.49 | bryanfe2 | question, in SIP, whats an example of a totally uncompressed, or not very compressed, codec? On the opposite end of uLaw which I think is very compressed.. |
| 02:02.56 | fujin_ | alaw is uncompressed |
| 02:03.10 | bryanfe2 | cool thank you |
| 02:03.15 | blitzrage | bryanfe2: ulaw/alaw are uncompressed |
| 02:03.21 | blitzrage | they are "companded" |
| 02:03.24 | bryanfe2 | i thought ulaw is compressed |
| 02:03.30 | [TK]D-Fender | bryanfe2, ulaw/alaw (G711u/G711a) are the baseline codec of digital telephony at the telco level |
| 02:03.40 | [TK]D-Fender | bryanfe2, You are mistaken. |
| 02:03.57 | blitzrage | bryanfe2: gsm and g.729 would be examples of compressed codecs |
| 02:04.01 | [TK]D-Fender | bryanfe2, these are the "native" PSTN codecs depending on your region |
| 02:04.20 | [TK]D-Fender | blitzrage, Not sure what I'd count G.726 as :) |
| 02:04.31 | blitzrage | ulaw == North America, alaw == pretty much everywhere else |
| 02:04.35 | bryanfe2 | thank you Fender, not using PSTN, this is all SIP. I have a network with relatively high UDP packet loss (or latency), not sure which, and I am trying to adjust the codec away from ulaw and to "something else", to see if it improves audio quality. |
| 02:04.37 | fujin_ | Anyone have any perl for AMI access? |
| 02:04.37 | blitzrage | [TK]D-Fender: lightly compressed? :) |
| 02:04.40 | bryanfe2 | can you recommend a "something else" to try? |
| 02:04.42 | [TK]D-Fender | blitzrage, Time-sliced, but not "compressed"... |
| 02:04.47 | fujin_ | Just need to originate a few calls per minute |
| 02:04.50 | fujin_ | is AMi best way? or callfiles? |
| 02:04.52 | [TK]D-Fender | bryanfe2, What is your goal? |
| 02:05.28 | [TK]D-Fender | fujin_, Either would do the job honestly, but I like AMI for not having to muck with the filesystem and to capable of being remotely issued |
| 02:05.40 | [TK]D-Fender | fujin_, Makes your solution more portable |
| 02:06.17 | blitzrage | fujin_: I'm sure there has to be a perl example on the web somewhre |
| 02:06.29 | bryanfe2 | Fender not really a lan, it's a WAN with high packet loss/latency |
| 02:06.35 | [TK]D-Fender | bryanfe2, choppy = jitter or bandwidth issues |
| 02:06.44 | [TK]D-Fender | bryanfe2, that'd nuke it |
| 02:06.45 | bryanfe2 | bandwidth is good |
| 02:06.56 | blitzrage | bryanfe2: then probably jitter |
| 02:07.05 | JT | bryanfe2: ulaw/alaw is companded, which you could say is a form of compression |
| 02:07.06 | blitzrage | if you're on a satellite connection... that is very high latency |
| 02:07.14 | [TK]D-Fender | bryanfe2, PL can mangle th reassembly of your audio |
| 02:07.33 | *** join/#asterisk techie (n=techie@adsl-76-214-24-128.dsl.lsan03.sbcglobal.net) |
| 02:07.34 | bryanfe2 | yes probably very analogous to satellite |
| 02:07.35 | JT | but it's a form of compression that gives greater dynamic range than it would otherwise have |
| 02:07.45 | JT | bryanfe2: use iLBC |
| 02:07.53 | bryanfe2 | should I be messing with the codec, or the jitter settings? |
| 02:07.57 | Daejeo | JT: MY FRIEND HOW TR U? |
| 02:08.08 | bryanfe2 | FYI it's audio *from* asterisk, *to* the ATA, which is choppy |
| 02:08.19 | blitzrage | is the ATA remote? |
| 02:08.24 | bryanfe2 | yes ATA is remote |
| 02:08.24 | blitzrage | or on the same network? |
| 02:08.28 | bryanfe2 | Asterisk is on 'net |
| 02:08.37 | blitzrage | is it only that one ATA? maybe the other network is bad |
| 02:08.49 | blitzrage | bryanfe2: turn off bittorrent :) |
| 02:08.57 | bryanfe2 | nice idea |
| 02:09.38 | bryanfe2 | should I try a higher network jitter level? |
| 02:12.10 | bryanfe2 | . |
| 02:15.42 | bryanfe2 | i tried "low" and "very high" jitter buffer settings, neither seems to help |
| 02:17.15 | bryanfe2 | any other ideas? |
| 02:17.19 | bryanfe2 | think satellite |
| 02:17.41 | bryanfe2 | i don't mind if there's a delay, as long as the audio is intact |
| 02:18.39 | *** join/#asterisk bryanfe2 (n=chatzill@74.10.9.114) |
| 02:21.24 | *** join/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz) |
| 02:21.48 | iamthelostboy | awesome.. voicemail to email is now up and running.. seriously cool stuff :) |
| 02:22.59 | *** join/#asterisk thinko (i=jdoe6alp@smaug.rackdragon.com) |
| 02:23.47 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177809381.dsl.bell.ca) |
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| 02:28.53 | fujin_ | Is the perl Telnet module the best way to AMI, via Perl? |
| 02:28.53 | Nugget | telnet is eeeeeeevil! |
| 02:30.13 | *** join/#asterisk RoyK (n=roy@ip-26-2-149-91.dialup.ice.no) |
| 02:32.28 | *** join/#asterisk techie (n=techie@adsl-76-247-11-252.dsl.lsan03.sbcglobal.net) |
| 02:38.24 | iamthelostboy | ive set callgroup=1 for all my sip users, and in features.conf i have pickupexten = *8, though it wont let the line get picked up |
| 02:39.04 | iamthelostboy | on one phone, it actually shows the number trying to call when *8 is dialed, it just doesnt pick it up.. |
| 02:44.23 | *** join/#asterisk ReD-MaN (i=root-rox@172-220.static.golden.net) |
| 02:46.04 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:47.47 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-574ec49a288148c0) |
| 03:02.00 | Corydon76-dig | fujin_: there's an Asterisk::Manager module |
| 03:07.24 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 03:09.49 | Daejeo | any good sip/iax provider? usa/canada unlimited $10tp 15$? |
| 03:10.01 | Daejeo | to* |
| 03:10.11 | Daejeo | per month |
| 03:10.45 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 03:13.30 | Daejeo | any good sip/iax provider? usa/canada unlimited $10to 15$? permonth |
| 03:14.49 | Corydon76-dig | Don't know of any that will give you unlimited |
| 03:15.10 | Corydon76-dig | but some will give you good per-minute rates |
| 03:15.40 | Corydon76-dig | I get .02/minute for my toll free number |
| 03:20.15 | *** join/#asterisk chesty (n=chesty@unconcerned.org) |
| 03:22.39 | Daejeo | Corydon76-dig: per min plan is very expensive |
| 03:23.14 | Corydon76-dig | flat rate, unlimited can be very expensive to your provider, because that's what the provider's vendors use |
| 03:23.49 | Daejeo | www.voicestick.com home unlimited $ 19.99 |
| 03:24.01 | Daejeo | looking for more cheaper one |
| 03:24.09 | Corydon76-dig | Read the fine print. You'll find that "unlimited" has a limit |
| 03:24.24 | Qwell | usually an unstated one, at that |
| 03:24.37 | Daejeo | really? |
| 03:24.41 | Qwell | always |
| 03:24.45 | Daejeo | oh |
| 03:25.18 | Daejeo | why people use "unlimited" |
| 03:25.35 | Daejeo | eye catching thing? |
| 03:25.35 | Qwell | because people let them get away with it |
| 03:26.08 | Corydon76-dig | Daejeo: yep, that provider limits you to 4800 minutes per month |
| 03:26.27 | Corydon76-dig | See provision 11.1 in their TOS |
| 03:26.50 | Corydon76-dig | Also see provision 8, where additional fees may be levied |
| 03:27.31 | Qwell | wait, i2? |
| 03:27.40 | Corydon76-dig | So that comes out to $.004/minute |
| 03:27.50 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-d9c3ca147b017d89) |
| 03:27.54 | Corydon76-dig | half a penny a minute |
| 03:27.58 | Daejeo | right |
| 03:28.05 | Daejeo | got it |
| 03:28.31 | Corydon76-dig | or considerably more, if you don't use the full 4800 minutes per month |
| 03:28.42 | Qwell | wow |
| 03:28.52 | Qwell | they're charging $50 for a gs ata286 |
| 03:29.00 | Qwell | aren't those like...$30? |
| 03:29.17 | Corydon76-dig | Sure, wholesale |
| 03:29.25 | Corydon76-dig | $50 is retail |
| 03:29.28 | Qwell | oh |
| 03:29.54 | Corydon76-dig | Then again, who on the Internet ever pays retail anymore? |
| 03:30.59 | [TK]D-Fender | Qwell, http://www.telephonydepot.com/product_p/105-056-286.htm |
| 03:31.16 | Qwell | oh no you didn't just link me to a gs |
| 03:32.57 | *** part/#asterisk BeeBuu (n=beebuu@61.145.77.158) |
| 03:34.09 | blitzrage | Corydon76-dig: where are your manners? |
| 03:34.37 | [TK]D-Fender | Corydon76-dig, I just liked their clause where they say they can teminate your account for any or no reason (guy thinks he found a loophole? CANNED!) |
| 03:34.54 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 03:38.56 | Corydon76-dig | blitzrage: eh? |
| 03:39.05 | blitzrage | gs link |
| 03:44.12 | Corydon76-dig | blitzrage: wasn't me |
| 03:44.56 | blitzrage | oops.. you both showed up as the same colour, so I misread |
| 03:53.23 | [TK]D-Fender | blitzrage, Sure.... us coloreds all look alike to you now don't we! Racist swine! |
| 03:53.44 | nhuisman_work | laugh |
| 03:54.25 | nhuisman_work | anyone got any good reasons why asterisk is better then sipx |
| 03:55.10 | Corydon76-dig | You're comparing apples and oranges |
| 03:55.13 | [TK]D-Fender | nhuisman_work, Better for WHAT is the question |
| 03:55.24 | Corydon76-dig | Asterisk is a B2BUA. SIPX is a proxy |
| 03:55.31 | nhuisman_work | that's part of the question I guess |
| 03:56.09 | [TK]D-Fender | nhuisman_work, * can do a LOT more than just SIP. |
| 03:56.40 | nhuisman_work | hmm for my application only sip really matters |
| 03:56.40 | [TK]D-Fender | nhuisman_work, What can you build with sipX? Then try and compare to *. |
| 03:56.53 | nhuisman_work | all my phones are sip |
| 03:56.58 | [TK]D-Fender | nhuisman_work, Are you happy with sipX? |
| 03:57.06 | Corydon76-dig | What about all your trunks? |
| 03:57.08 | nhuisman_work | i don't use it |
| 03:57.25 | fujin_ | heh |
| 03:57.29 | fujin_ | Anyone got a perl example for AMI? |
| 03:57.30 | Corydon76-dig | What happens if a provider you want to use won't provision SIP? |
| 03:57.31 | nhuisman_work | i'm on cisco call manager but we are upgrading and realizing it's stupid expensive to use cisco |
| 03:57.35 | nhuisman_work | we have a t1 pri |
| 03:57.38 | fujin_ | trying to do some outbound dialing with application: playback |
| 03:57.40 | nhuisman_work | we don't need a provider |
| 03:57.49 | nhuisman_work | in that sense |
| 03:57.54 | Corydon76-dig | Then Asterisk makes sense to you |
| 03:58.07 | [TK]D-Fender | nhuisman_work, and what will let you use the T1 with sipX? |
| 03:58.21 | Corydon76-dig | Being able to control your PRI directly would be important |
| 03:58.33 | nhuisman_work | tkd-fender good point |
| 03:58.39 | nhuisman_work | i wonder if the enterprise version of sipx does that |
| 03:59.01 | [TK]D-Fender | nhuisman_work, sipX is software. What hardware can you use with it to use your T1? |
| 03:59.11 | Corydon76-dig | Sounds like you already know the answer, but you're looking for an excuse not to use Asterisk |
| 03:59.41 | nhuisman_work | no i just want to see what people have to say about it, but it seems like it's comparing apples to oranges |
| 03:59.47 | [TK]D-Fender | Corydon76-dig, Oh snap, its MONDAY! Guess Asterisk is out of the question! |
| 03:59.49 | nhuisman_work | ina sense |
| 03:59.54 | nhuisman_work | in a |
| 04:00.26 | [TK]D-Fender | nhuisman_work, it is. * can talk in multiple protocols, use a wide variety of hardware, allow you very comprehensive call processing. |
| 04:00.34 | [TK]D-Fender | nhuisman_work, what exactlye do YOU want it for? |
| 04:00.44 | [TK]D-Fender | nhuisman_work, My system used to make me COFFEE. |
| 04:00.48 | nhuisman_work | hehe |
| 04:00.53 | Qwell | [TK]D-Fender: now make me some |
| 04:01.17 | nhuisman_work | yeah I just want it to do sip and interface with a t1 pri and provide some analog fxo's. |
| 04:02.00 | [TK]D-Fender | Qwell, I would, but its -17 out there.... it'd arrive solid :) |
| 04:02.08 | Qwell | I've got a blender |
| 04:02.54 | nhuisman_work | coffee popsickle |
| 04:02.56 | nhuisman_work | mmmm |
| 04:20.19 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:20.19 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 04:48.58 | UnixDog | [Dec 3 20:37:26] WARNING[2352]: loader.c:413 load_dynamic_module: Error loading module 'chan_zap.so': /usr/local/lib/asterisk/modules/chan_zap.so: Undefined symbol "ast_smdi_interface_find" |
| 04:50.05 | UnixDog | it looks like if you put noload => res_smdi.so in modules.conf zaptel wont load |
| 04:50.13 | *** join/#asterisk lwizardl (n=lwizardl@adsl-71-155-203-106.dsl.klmzmi.sbcglobal.net) |
| 04:50.15 | lwizardl | Hi |
| 04:51.40 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 04:52.40 | UnixDog | so we have either found a bug |
| 04:53.17 | lwizardl | I want to build a pbx box out of a 733mhz p2 w/ 256mb ram and 20gb hdd but how do i decide on what cards i need? |
| 04:53.36 | dec | pr upi |
| 04:53.39 | dec | whoops. |
| 04:55.20 | [TK]D-Fender | lwizardl : well.. what do you think you need cards for? |
| 04:56.42 | lwizardl | [TK]D-Fender, well I want to have phones in different rooms (2 bedrooms, living room, kitchen) so I'd need atleast 4 ports for the phone lines out |
| 04:57.13 | [TK]D-Fender | lwizardl, so you want to have 4 phones that all work 100% independantly of each other? |
| 04:57.16 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 04:57.19 | JT | lwizardl: i think you mean PIII. |
| 04:57.30 | lwizardl | jt: yeah its a p3 |
| 04:57.44 | lwizardl | [TK]D-Fender, if possible |
| 04:57.51 | [TK]D-Fender | lwizardl, Ok, what else? |
| 04:57.59 | JT | lwizardl: well you said P2 |
| 04:58.07 | lwizardl | [TK]D-Fender, just voice mail etc |
| 04:58.27 | lwizardl | JT, yeah i know hit wrong key |
| 04:58.36 | [TK]D-Fender | lwizardl, well so far you've talked about using 4 PHONES (analog I presume) with Asterisk. What about LINES? |
| 04:58.45 | phix | hey, I am having bad echo problems on my asterisk box, both from my ATA and softphones |
| 04:59.05 | lwizardl | [TK]D-Fender, I was thinking about just using voip |
| 04:59.07 | [TK]D-Fender | phix, What's on each end of an echoy call? |
| 04:59.36 | [TK]D-Fender | lwizardl, In that case I'd suggest 2 x Linksys SPA-2102's (+/- $65 USD each) |
| 04:59.51 | [TK]D-Fender | lwizardl, 2 ATA's, and no cards. |
| 05:00.18 | [TK]D-Fender | lwizardl, http://www.telephonydepot.com/product_p/105-054-212.htm |
| 05:00.24 | phix | Analog phone -> Sipura 312something -> asterisk box -> SIP provider, and softphone (xlite) -> asterisk -> SIP provider or another softphone |
| 05:00.51 | [TK]D-Fender | phix, So FXS through * to your ITSP? |
| 05:00.57 | phix | [TK]D-Fender: What echo filtering options are available in asterisk? (or are there third party plugins available?) |
| 05:01.04 | phix | yes |
| 05:01.08 | JT | phix: pretty much none if not zaptel |
| 05:01.10 | phix | ITSP == SIP Provider? :) |
| 05:01.22 | [TK]D-Fender | phix, then its your provider, and there's nothing you can do about it except switch |
| 05:01.26 | [TK]D-Fender | ~itsp |
| 05:01.27 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 05:01.46 | phix | JT: ok, hmmm I have echo cancelling enabled on Sipura and some jitter buffer enabled but I think that is just for PSTN line |
| 05:02.22 | lwizardl | [TK]D-Fender, ok I have a vonage linksys rtp-300 that i read I can make work with other voip services would this work for 2 for a test run? |
| 05:02.22 | phix | !itsplist-au |
| 05:02.45 | phix | [TK]D-Fender: hmmmm, I guess I will give them a call then |
| 05:02.59 | [TK]D-Fender | lwizardl, if you can unlock it, it might be usable |
| 05:03.00 | phix | they have a SIP number so they will beable to hear it hopefully, unless it is just my end |
| 05:03.13 | [TK]D-Fender | phix, Correct, only for the FXO port. |
| 05:03.33 | phix | [TK]D-Fender: aawww, so there is no third party plugins for asterisk? |
| 05:03.44 | phix | I thought I saw one on a digium site |
| 05:03.53 | phix | for x amount of y units of currency |
| 05:03.57 | [TK]D-Fender | phix, if 2 of your SIP devices don't echo between each other within the scop of your * server, there's your answer |
| 05:04.06 | lwizardl | [TK]D-Fender, yeah I have had it unlocked before I had to redo the config on it for vonage to work again but I'm no longer a subscriber of theres |
| 05:04.10 | [TK]D-Fender | phix, No, there isn't |
| 05:04.26 | phix | [TK]D-Fender: well I can't be sure if it is the softphone doing that (I am using a headset) |
| 05:04.33 | [TK]D-Fender | lwizardl, Well Vonage uses SIP, so it should be every bit as functional with * with any luck. |
| 05:04.50 | lwizardl | ok thanks |
| 05:04.57 | phix | [TK]D-Fender: how is your C? you want to help me write one then :P |
| 05:05.02 | [TK]D-Fender | lwizardl, Then again I just quoted you a comlpete solution for $130 so I'd say "no big deal" either way |
| 05:05.10 | JT | phix: have you used only the one headset? |
| 05:05.14 | [TK]D-Fender | phix, You're screwed, go stock up on KY |
| 05:05.28 | phix | [TK]D-Fender: :( KY? |
| 05:05.34 | phix | JT: I have 4 of them |
| 05:06.15 | lwizardl | [TK]D-Fender, yeah I just figure maybe i can setup the pbx within the next couple of days to get a test run with out spending anything to see how I like the software |
| 05:06.51 | [TK]D-Fender | lwizardl, Certainly. Go download a soft-phone, set it up on 2 systems, go learn * place some calls, set up VM,e tc. |
| 05:06.52 | lwizardl | whats the most common voip provider that is used with asterisk? |
| 05:07.04 | [TK]D-Fender | lwizardl, For what area? |
| 05:07.14 | lwizardl | Michigan, USA |
| 05:07.16 | [TK]D-Fender | ~itsplist-us |
| 05:07.16 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com |
| 05:07.18 | [TK]D-Fender | ^^^ |
| 05:07.39 | lwizardl | thanks again [TK]D-Fender, |
| 05:07.59 | [TK]D-Fender | lwizardl, np. Here, save this link to a quick guide |
| 05:08.03 | [TK]D-Fender | ~jerjerguide |
| 05:08.03 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 05:08.22 | [TK]D-Fender | lwizardl, and buy or download THE BOOK |
| 05:08.24 | [TK]D-Fender | ~book |
| 05:08.25 | jbot | book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 05:09.05 | lwizardl | wow this has to be the most helpful channel I have found on freenode so far |
| 05:09.10 | lwizardl | thanks again |
| 05:09.12 | lwizardl | :) |
| 05:09.53 | phix | KY? |
| 05:09.57 | JT | phix: yes, have you tried all the headsets? |
| 05:10.08 | JT | phix: you must be living under a rock if you don't know what K-Y is |
| 05:12.08 | flenders | phix: http://en.wikipedia.org/wiki/K-Y_Jelly |
| 05:13.08 | phix | oh, lube |
| 05:16.14 | lwizardl | lol |
| 05:20.50 | lwizardl | how is broadvoice service i think i'd use them for the + package |
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| 05:30.44 | lwizardl | anyone here use broadvoice ? |
| 05:31.48 | *** join/#asterisk Landon (n=Landon@unaffiliated/landon) |
| 05:31.56 | [TK]D-Fender | lwizardl : honestly I'd probably go with another company. |
| 05:32.33 | lwizardl | why? |
| 05:32.38 | *** part/#asterisk Landon (n=Landon@unaffiliated/landon) |
| 05:33.41 | lwizardl | i doubt they could be worse then vonage was after feb 07 my service started getting horrible |
| 05:33.45 | [TK]D-Fender | lwizardl, Broadvoice's proxies have a habit of winking out and their support is pretty dim-witted. |
| 05:33.55 | lwizardl | hmm |
| 05:33.59 | [TK]D-Fender | lwizardl, I'd put them in the same boat |
| 05:34.09 | lwizardl | sounds similar |
| 05:34.58 | *** join/#asterisk zubzero (i=sub_zero@cpe-024-088-080-200.sc.res.rr.com) |
| 05:35.02 | zubzero | hey guys |
| 05:35.04 | lwizardl | I just like the monthly fee versus the per minute/per hour deal |
| 05:37.35 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 05:37.44 | zubzero | neone know of a provider that has at least 4 simultanious connections and unlimited minutes? |
| 05:39.20 | zubzero | Teliax does but it says theres a limit of 1500 minutes and its .02 cent for every minute after |
| 05:39.27 | lwizardl | zubzero, not me I'm new to this (day 1) |
| 05:41.43 | zubzero | im pretty new to this also |
| 05:41.46 | *** join/#asterisk d-k-t (n=dt@125.120.137.80) |
| 05:41.52 | zubzero | im settin up voip for my church |
| 05:42.01 | lwizardl | on that site only one i see with 4 Simultaneous Calls is the corp package |
| 05:42.24 | zubzero | yea and thats fine but the fine print says limited to 1500 or 2500 minutes |
| 05:42.24 | lwizardl | i'm planing on using asterisk in my home |
| 05:42.38 | UnixDog | it looks like if you put noload => res_smdi.so in modules.conf zaptel wont load. anyone else have this issue |
| 05:42.52 | lwizardl | yeah 2500 minutes and then charged per min |
| 05:42.58 | zubzero | lwizardl well check this out http://revision3.com/systm/asterisk/ |
| 05:43.16 | JT | minutes cost money in america. perhaps you'll have to actually pay money. |
| 05:43.49 | lwizardl | that show never seems to stop amazing me they always have something i'm trying :) |
| 05:43.50 | zubzero | JT mayb theres no unlimited provider in US |
| 05:44.30 | JT | not truly unlimited, that would just be uneconomical |
| 05:44.37 | [TK]D-Fender | zubzero, ANCIENT link |
| 05:44.53 | zubzero | [TK]D-Fender just tryin to help him |
| 05:47.08 | zubzero | neone know if broadvoice supports multiple calls? |
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| 06:06.05 | fujin | Anyone very familiar with AMI? |
| 06:06.52 | fujin | http://rafb.net/p/j4jYim12.html |
| 06:06.54 | fujin | I'm getting a little issue |
| 06:06.57 | fujin | can't quite work out why |
| 06:09.47 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:10.34 | fujin | It's like it's not seeing the application one |
| 06:11.45 | fujin | hrm |
| 06:11.47 | fujin | nevermind, got it |
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| 07:14.54 | mkl1525 | Hi, having problem with queue agents that login after a caller has joined the queue. retry time is set to 1 second, agent phones are called with "exten => s,1,Dial(SIP/${ARG1},15)" , ARG1 is phone number of the agent. after 15 seconds ringing the phones wait 11 seconds till all phones ring again. if I increase the retry time to 10 seconds phones ring 20 seconds later - so there is a 10 second span that seems fix and I don't know where it comes from - any s |
| 07:15.08 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 07:16.59 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 07:25.38 | mosty | mkl1525, are you using Queue()? |
| 07:26.22 | *** join/#asterisk dlynes (n=chatzill@d154-20-45-103.bchsia.telus.net) |
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| 07:36.10 | mkl1525 | mosty, yes |
| 07:41.58 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
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| 07:49.58 | mkl1525 | mosty thanks for the hint had a look again at queues.conf and read the voip-info wiki page again - misunderstood the behavior of the timeout setting |
| 07:51.36 | zeekmor | ok i need basic help please |
| 07:51.48 | zeekmor | how do i RTFM when i cant find the manual? |
| 07:53.20 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
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| 07:54.24 | zeekmor | please don't tell me the forum is the only spot for docs |
| 07:56.06 | kaldemar | ~thebook |
| 07:56.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
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| 07:56.35 | kaldemar | zeekmor: that's a pretty good start. |
| 07:56.44 | zeekmor | thank you |
| 07:56.52 | zeekmor | i didnt know the book was pdf |
| 07:56.57 | fujin | the source tarball has lots of nice documentation. |
| 07:57.52 | zeekmor | sorry this can be a bit overwhelming @.@ |
| 07:58.01 | kaldemar | and you can get some application/function documentation in the cli with 'core show application <app>' and 'core show function <func>'. |
| 07:58.19 | zeekmor | nope i did not know that |
| 07:58.31 | zeekmor | in fact the cli scared the $##@@ out of me when i went into it |
| 07:58.52 | kaldemar | the cli is your friend. get used to it and learn how to use it. |
| 07:59.42 | zeekmor | nods, usually it is, i mean my irc client at the moment is epic ;) but help wasnt as clarivoyant as i would have liked |
| 08:00.53 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 08:00.53 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 08:00.56 | zeekmor | and you have to know the application name to show core on it |
| 08:01.08 | kaldemar | there is also http://www.voip-info.org that has lots of useful information. |
| 08:01.30 | kaldemar | core show applications lists your applications. |
| 08:02.34 | zeekmor | wow |
| 08:02.56 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 08:03.57 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 08:04.11 | J4zen | Hi there |
| 08:04.12 | J4zen | What's the most commonly used (low-budget) billing interface for asterisk? |
| 08:05.45 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
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| 08:12.09 | zeekmor | kaldemar: thank you |
| 08:22.26 | salzh | hey. who know the consume power of tdm400p |
| 08:26.04 | mosty | salzh, depends what's pluggedin |
| 08:27.04 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:31.26 | zeekmor | last silly question, i can have asterisk interface with exchange (2007), jabber, and my phones all in one? |
| 08:31.53 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 08:36.07 | *** join/#asterisk tiav (n=tiav@inv75-3-82-241-117-16.fbx.proxad.net) |
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| 08:39.34 | mosty | probably. not sure about exchange |
| 08:39.34 | J4zen | There are so many billing solutions, yet i cannot seem to find any that actually look solid enough and are low-budget. Does anyone have any recommendations? |
| 08:40.00 | dudes | yup |
| 08:40.02 | dudes | pay someone |
| 08:40.13 | J4zen | lol |
| 08:40.27 | J4zen | how about not-so-low budget ? |
| 08:40.41 | dudes | I don't know what you're asking sorry |
| 08:40.47 | dudes | I was just being a dick |
| 08:40.52 | dudes | for fun =) |
| 08:41.31 | dudes | on a serious note, you could try using asterisk DB CDR and create a frontend to that for billing purposes |
| 08:41.45 | dudes | that may provide enough info maybe |
| 08:42.19 | J4zen | hehe fun times |
| 08:42.30 | J4zen | yeah that thought occured to me, was just hoping for some funky popen source solution |
| 08:42.41 | J4zen | as a last meassure i'll code a frontend myself |
| 08:42.56 | dudes | well there you go |
| 08:43.00 | dudes | question answered |
| 08:43.14 | J4zen | lol |
| 08:43.19 | J4zen | Watch out, i didn't have my coffee yet |
| 08:43.42 | dudes | but I got a 750 of whiskey |
| 08:43.54 | J4zen | that worsk |
| 08:43.55 | J4zen | works* |
| 08:44.13 | dudes | f'n bars closing at 2AM |
| 08:44.15 | dudes | what is that |
| 08:44.22 | zeekmor | its only 12:45 |
| 08:44.29 | J4zen | make that 09:45 |
| 08:44.31 | zeekmor | still got an hour and half |
| 08:44.34 | dudes | it's 2:45AM |
| 08:44.45 | zeekmor | ... someone here isnt on a global clock |
| 08:45.03 | dudes | it's call local time |
| 08:45.20 | zeekmor | .. i think its my server, it is telling me that it is 12:15 O.o? |
| 08:46.06 | dudes | it was slim picking at thebar |
| 08:46.16 | dudes | stupid Tuesdays |
| 08:46.59 | dudes | Wend is better though |
| 08:47.07 | zeekmor | wow i lagged myself ... mental note, do not change time while on irc |
| 08:47.17 | zeekmor | wow that sucked |
| 08:47.23 | dudes | haha |
| 08:47.36 | dudes | Tue Dec 4 07:44:40 GMT 2007 <--- Server in Ireland |
| 08:48.12 | zeekmor | why are you using an irish server? |
| 08:48.59 | dudes | I do work with people in Ireland |
| 08:49.05 | zeekmor | ahh |
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| 08:49.22 | dudes | the only other reason I'm still up |
| 08:49.47 | zeekmor | i am still up because i am trying to figure out if asterisk is a good answer for my problem |
| 08:50.13 | dudes | that can depend on the question |
| 08:50.53 | dudes | but my answer is always, "It may not be the best answer --- but it's the easiest answer." |
| 08:50.56 | zeekmor | i have ... telepacific for my voip phones, would it be more effective to bring them all in house or find another telco company |
| 08:51.21 | zeekmor | hehe |
| 08:51.37 | dudes | what do you mean "in house" |
| 08:52.10 | zeekmor | literally in house |
| 08:52.16 | zeekmor | you know asterisk in the office |
| 08:52.52 | dudes | you're looking for a VOIP PBX? |
| 08:53.02 | zeekmor | yeah |
| 08:53.21 | dudes | well you're certainly not going to find anything that works as easy as asterisk |
| 08:53.39 | zeekmor | well there is also meterics that is lacking at the moment, and managability for it |
| 08:54.04 | dudes | It's amazing what safe_asterisk does for a PBX =p |
| 08:54.33 | dudes | if in doubt and a daily cron job to cycle everything |
| 08:55.01 | zeekmor | i am far from that stage |
| 08:55.15 | zeekmor | ... i am at stage pre1 ... can i get it to work with my phones and voicemail :-D |
| 08:55.51 | dudes | voicemail is simple --- turn it off on your provider and use asterisks voicemail |
| 08:56.12 | dudes | most phones will like good --- if in doubt google it |
| 08:56.49 | zeekmor | nods, voicemail seems to be the easiest part, just need to make adjustments for storage of messages |
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| 08:58.36 | zeekmor | le sigh i should go to sleep and let someone else worry about this |
| 08:59.38 | dudes | sounds like a good idea |
| 09:02.04 | zeekmor | atleast i got it to start :-D |
| 09:02.23 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 09:06.16 | *** join/#asterisk regurgitate (n=regurita@20.Red-80-35-165.staticIP.rima-tde.net) |
| 09:06.17 | regurgitate | hi |
| 09:06.33 | regurgitate | i have a problem before updte |
| 09:06.45 | regurgitate | sip show users is empty |
| 09:06.51 | dudes | sry, I have to regurgitate --- f'n boozze! |
| 09:07.00 | regurgitate | XD |
| 09:07.29 | dudes | if users is empty try peers |
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| 09:15.23 | BBHoss | dudes, cheap beer! |
| 09:15.39 | regurgitate | also is empty |
| 09:15.54 | regurgitate | i suffered this before freepbx update |
| 09:16.11 | regurgitate | users are files |
| 09:16.17 | regurgitate | but are not showed at asterisk |
| 09:19.39 | agx | did you guys experienced problem with Alaw? it seems that Ulaw perform better then Alaw (sound quality) even if it should be viceversa; maybe its poorly implemented onto the phones? |
| 09:20.19 | fujin | uh |
| 09:20.23 | fujin | they're exactly the same |
| 09:20.29 | fujin | (well, not exactly, but mostly) |
| 09:20.44 | fujin | agx: depending on where you are int he world, you should use ONE or the OTHER |
| 09:20.46 | fujin | not switch between both |
| 09:20.52 | fujin | i.a.; USA => ulaw |
| 09:20.57 | fujin | everywhere else (mostly?) => alaw |
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| 09:24.52 | regurgitate | please help me with this |
| 09:24.53 | regurgitate | chalet*CLI> sip show users |
| 09:24.53 | regurgitate | Username Secret Accountcode Def.Context ACL NAT |
| 09:24.53 | regurgitate | chalet*CLI> sip show peers |
| 09:24.53 | regurgitate | Name/username Host Dyn Nat ACL Port Status |
| 09:24.53 | regurgitate | 0 sip peers [0 online , 0 offline] |
| 09:25.01 | regurgitate | and i have many users registered |
| 09:26.47 | tzafrir | regurgitate, registered? do you mean: entries in sip.conf? |
| 09:26.59 | tzafrir | or 'register=>' lines? |
| 09:27.59 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
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| 09:44.37 | dudes | sip show registry ? |
| 09:44.50 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:45.45 | *** part/#asterisk dudes (n=nixtux@74-60-94-233.stc.clearwire-dns.net) |
| 09:48.27 | J4zen | Does anyone have the SNOM320/360 or the expansion module for the 360 (keypad) ? |
| 09:49.27 | *** join/#asterisk BeeBuu (n=beebuu@218.13.66.79) |
| 09:49.37 | BeeBuu | hello,all. |
| 09:50.10 | BeeBuu | how can i anwser a in queues with press a key? |
| 09:50.44 | BeeBuu | how can i anwser a all that in queues with press a key? |
| 09:50.54 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 09:52.40 | regurgitate | sorry |
| 09:53.25 | BeeBuu | how can i anwser a call that in queues with press a key? |
| 09:53.56 | regurgitate | i had added |
| 09:53.57 | regurgitate | #include sip_nat.conf |
| 09:53.57 | regurgitate | #include sip_custom.conf |
| 09:53.57 | regurgitate | #include sip_additional.conf |
| 09:53.59 | regurgitate | to sip.conf |
| 09:54.08 | regurgitate | and i had fixed it |
| 09:58.24 | RoyK | http://picayune.uclick.com/comics/wpldl/2007/wpldl071204.gif |
| 09:58.45 | BBHoss | regurgitate, freepbx stores the config in a mysql database and when you reload it writes to the asterisk conf files |
| 09:59.05 | regurgitate | yes |
| 09:59.22 | regurgitate | but before update any user appears with sip show users |
| 09:59.29 | regurgitate | and doing it i had solved it |
| 09:59.37 | regurgitate | if you know another way, please let me know |
| 10:01.50 | *** part/#asterisk BeeBuu (n=beebuu@218.13.66.79) |
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| 10:12.03 | J4zen | oh shit |
| 10:12.04 | J4zen | im fucked |
| 10:12.30 | J4zen | we recently applied at a VOIP provider to import all our analogue lines to a VOIP-trunk |
| 10:12.51 | J4zen | so we recieved a letter yesterday , from our analogue phone provider |
| 10:12.54 | fujin | O RLY? |
| 10:13.02 | J4zen | stating that our lines will be transferred etc etc, no problem |
| 10:13.03 | J4zen | however |
| 10:13.08 | J4zen | we also have ADSL with them |
| 10:13.14 | J4zen | which they chose to cancel as well |
| 10:13.18 | fujin | pwnt |
| 10:13.22 | BBHoss | heh, telco :) |
| 10:13.22 | J4zen | appying for new ADSL takes between 10~30 days |
| 10:13.35 | fujin | That's pretty gay. |
| 10:13.37 | fujin | Get fibre? :0 |
| 10:13.38 | J4zen | 10~30 days of no ADSL = gone |
| 10:13.40 | J4zen | brb phone |
| 10:13.49 | fujin | heh, getting *anything* provisioned will take time. |
| 10:21.34 | mvanbaak | J4zen: that sux |
| 10:27.45 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 10:31.24 | *** join/#asterisk Le_Vert (n=gandalf@adsl02.metz.linbox.com) |
| 10:31.38 | Le_Vert | hello :) |
| 10:31.56 | Le_Vert | could someone advice me a good asterisk monitor software ? |
| 10:32.29 | Le_Vert | all I need is to see all extensions, their status and who they are talking with |
| 10:32.39 | fujin | no such thing |
| 10:32.54 | Le_Vert | really ? |
| 10:32.55 | J4zen | Back |
| 10:32.59 | J4zen | yes it does suck mvanbaak |
| 10:33.13 | J4zen | KPN + Xs4all |
| 10:33.14 | J4zen | drama |
| 10:33.33 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-f3bdd8f49cabf100) |
| 10:42.44 | JT | fujin: not quite true |
| 10:42.58 | JT | there's Flash Operator Panel and perhaps a couple of others |
| 10:42.59 | *** join/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 10:43.06 | JT | or you could make your own using AMI |
| 10:43.21 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:43.49 | *** join/#asterisk tarantul (n=tarantul@hacer.buildnet.ru) |
| 10:44.37 | tarantul | hello! Ppl, can asterisk use radius for sip account registration/authorization? |
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| 11:16.00 | zeeesh | at asterisk console when i give command reload or module reload .. will it effect the current calls on server or not? |
| 11:19.41 | *** join/#asterisk Rapani (n=Cartman@u211.ip1.netikka.fi) |
| 11:19.45 | Rapani | hello |
| 11:23.31 | Rapani | I have googled for hours now, and I just can't get my asterisk (with freepbx) to work.. Call always goes to voice mail, here is debug: http://pastebin.com/m1fa3f7c0 |
| 11:23.37 | Rapani | what am I missing? |
| 11:25.40 | *** join/#asterisk myiagy (n=myiagy@200.215.59.133) |
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| 11:30.20 | Rapani | someone? |
| 11:30.52 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 11:32.43 | *** join/#asterisk defswork (n=andy@77.44.54.26) |
| 11:33.00 | tzafrir | Rapani's issue was: Executing NoOp("SIP/0003-081f1098", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack |
| 11:33.03 | Sonderblade | anyone know of a free phone number tracking service for bulgaria? |
| 11:33.04 | tzafrir | A freepbx issue |
| 11:33.35 | tzafrir | Sonderblade, doesn't your government do that for you? |
| 11:34.15 | Sonderblade | tzafrir: not for free :) |
| 11:34.42 | tzafrir | Sonderblade, not for free? wow, that's even worse |
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| 12:11.27 | dijungal | hello |
| 12:11.38 | dijungal | how do i log what's on the CLI |
| 12:11.39 | dijungal | ? |
| 12:11.48 | dijungal | all that scrolling info |
| 12:12.14 | dijungal | or can i increase the verbosity/detail of messages log |
| 12:12.14 | agx | dijungal, change /etc/asterisk/logger.conf then watch inside /var/log/asterisk/ directory |
| 12:12.15 | dijungal | ? |
| 12:12.24 | dijungal | ok thanks |
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| 12:50.13 | stimpie | I have experienced a few 'stuck' sip calls a bye message was sent to asterisk but the call was never ended. Someone know what might cause this and how to prevent it? |
| 13:01.41 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:12.32 | *** part/#asterisk salzh (n=salzh@124.77.15.177) |
| 13:12.49 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 13:14.36 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
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| 13:20.55 | mosty | i have two asterisk servers, sip clients register to the front facing server, which delivers calls to a second asterisk server to be terimated via PRI. this works fine for regular calls, but when i set a redirect on a sip phone, calls to that phone aren't redirected properly |
| 13:22.23 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 13:23.15 | mosty | the call goes to the asterisk server with the PRI lines ok, but i believe that the call is rejected by the PRI provider because callerid or something is wrong. i can see the sip extension's number in pri debug traces. however CALLERID(num) is definitely not set to the sip extension, it's set to a valid DID for that PRI line. what else could be wrong? the sip phone can dial the same forwarding number perfectly |
| 13:31.12 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:33.34 | *** join/#asterisk ming_zym (n=ming_zym@124.14.234.210) |
| 13:55.26 | mosty | what other information is sent out a PRI line besides callerid? i'm trying to figure out why calls redirected from my phone are rejected by my pri provider, i can see my sip extension in the pri debug, but i can't find where this might be set and/or where i can overwrite this |
| 13:56.28 | *** join/#asterisk Delvar (n=Delvar@77.240.56.17) |
| 13:57.47 | [TK]D-Fender | mosty: So you see your sip extensions used as your outbound CID? |
| 13:57.59 | [TK]D-Fender | (devices rather) |
| 13:59.01 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:59.02 | mosty | no, but i can see sip extensions in the pri debug of forwarded calls |
| 13:59.41 | mosty | and those calls are rejected by the pri provider. the callerid is set to a valid number for that pri, and the phone can dial that number successfully |
| 13:59.57 | *** join/#asterisk dreamind (n=dreamind@p54A79377.dip0.t-ipconnect.de) |
| 14:00.01 | dreamind | Hi folks :) |
| 14:00.19 | [TK]D-Fender | mosty: You should always set your CID before you dial. |
| 14:00.44 | dreamind | can anybody tell me if there is an easy way to identify all events which belong to an Action Originate, I sent over the manager interface? |
| 14:00.49 | mosty | [TK]D-Fender, i NoOp(CALLERID(num)) before i dial, and it's correct. do i need to set it again anyway? |
| 14:00.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:00.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:01.04 | dreamind | because the UniqueID always changes, so IMHO its not good to use it for identifying the events |
| 14:01.08 | mosty | ${CALLERID(num)} rather |
| 14:01.28 | dreamind | do I have to set a channel variable and fetch it with GetVar? is there no easier way? |
| 14:01.32 | [TK]D-Fender | mosty: pastebin a complete call including that & PRI debug & zapata. If I can't see whats the matter someone else likely will |
| 14:02.19 | [TK]D-Fender | dreamind: Nope, that soulnds to be about it, and its pretty good that you came to that yourself... |
| 14:03.07 | dreamind | [TK]D-Fender: oh well, it seems more and more to me like the manager interface sucks badly :( |
| 14:04.02 | [TK]D-Fender | dreamind: Well its not "full interactive" when you throw off an Originate is all. Its really a seperate process. |
| 14:04.30 | [TK]D-Fender | dreamind: Wan'ts meant to stay kind of tethered to your AMI connection. |
| 14:04.35 | [TK]D-Fender | wasn't |
| 14:04.40 | [TK]D-Fender | God its cold in here... |
| 14:04.59 | dreamind | [TK]D-Fender: well I don't thought its interactive. - I use a homebrew perl interface for it. |
| 14:05.07 | dreamind | err, didn't thought |
| 14:05.20 | [TK]D-Fender | s/thought/think ;) |
| 14:05.27 | dreamind | ;) |
| 14:05.40 | [TK]D-Fender | What do you want to track it and do? |
| 14:06.07 | dreamind | I want to create a dialing software for callcenters, and I need to track specific calls |
| 14:06.19 | dreamind | so I need to track the channel I originated |
| 14:07.03 | dreamind | maybe GetVar helps me enough... |
| 14:07.25 | dreamind | I thought I could use uniqueid, but then I realized how the uniqueid works :( |
| 14:07.58 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.24) |
| 14:08.24 | [TK]D-Fender | dreamind: Slippery bugers... |
| 14:08.27 | [TK]D-Fender | buggers* |
| 14:08.44 | [TK]D-Fender | dreamind: SetVar really would be the best. |
| 14:08.46 | deeperror | how can i extend the timeout or post-dial delay on a zap channel? asterisk seems to miss the last digit dialed sometimes |
| 14:08.47 | mosty | [TK]D-Fender, http://pastebin.ca/805569 <- note the 340340, that is my sip extension |
| 14:08.48 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 14:09.05 | dreamind | [TK]D-Fender: ok, oh well... |
| 14:09.12 | *** part/#asterisk emist (n=emist@unaffiliated/emist) |
| 14:09.26 | dreamind | [TK]D-Fender: besides all that I really would prefer an RPC alike interface... and not this string parsing stuff ;) |
| 14:10.16 | [TK]D-Fender | mosty: PITA to read, next time grep out the "[Kskutter1*CLI>" :) |
| 14:10.29 | mosty | one sec |
| 14:10.34 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 14:11.28 | [TK]D-Fender | mosty: > Called Number (len= 9) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '131166' ] <--- I think you need to set pridialplan |
| 14:12.33 | mosty | [TK]D-Fender, when i dial that number from the sip client, the call goes through. it's when i set the sip phone to redirect calls that it fails |
| 14:13.13 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:13.20 | *** join/#asterisk DJF5 (n=irc@84-105-201-37.cable.quicknet.nl) |
| 14:13.29 | mosty | here's the cleaned up log: http://pastebin.ca/805574 |
| 14:14.25 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 14:14.53 | [TK]D-Fender | mosty: Where do I see sip? |
| 14:15.51 | mosty | [TK]D-Fender, see the 340340 on line 62? that is my sip extension |
| 14:16.31 | lirakis | ... hmm asterisk is running.. but i cant connect via asterisk -vvvvr ... |
| 14:16.37 | mosty | sip/340340 registers to asterisk machine A, asterisk A sends outgoing calls to asterisk machine B which terminates via PRI |
| 14:16.59 | mosty | this log is from asterisk machine B, when sip/340340 is set to redirect calls |
| 14:17.12 | lirakis | and /var/run/asterisk/asterisk.ctl does exists.. its 0 bytes |
| 14:17.34 | [TK]D-Fender | mosty: so SIP > IAX -IAX -> ZaptelPRI, right? |
| 14:17.43 | mosty | [TK]D-Fender, yes |
| 14:17.53 | [TK]D-Fender | lirakis: Who are you running as, and who is * running as? |
| 14:18.36 | lirakis | [TK]D-Fender: i am root, asterisk is running as root |
| 14:19.31 | [TK]D-Fender | mosty: I'm thinking that IAX2 carries some extra stuff in it that may be interfering. This is outside of my experience, and I'd advise that you be ready to repastebin it (just resubmit) when one of the more IAX/PRI experienced people arrives |
| 14:20.17 | mosty | [TK]D-Fender, i agree that iax2 is carrying extra info. perhaps i can try using sip between the two asterisk machines |
| 14:20.31 | mosty | thanks for looking it over, at least |
| 14:20.42 | [TK]D-Fender | mosty: np, keep taht window ready to refresh... |
| 14:22.31 | lirakis | <PROTECTED> |
| 14:22.39 | lirakis | <PROTECTED> |
| 14:22.45 | [TK]D-Fender | lirakis: Not sure what to say... |
| 14:23.01 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 14:27.01 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 14:27.39 | lirakis | [TK]D-Fender: is the ctl file supposed to be 0 bytes? i assume its just a stdin/stdout communication socket? |
| 14:28.44 | [TK]D-Fender | lirakis: No, should have the PID # for the * process |
| 14:29.01 | lirakis | [TK]D-Fender: i thought that was the .pid file |
| 14:29.07 | [TK]D-Fender | lirakis: if the file isn't locaked you could always try to forge it :) |
| 14:29.16 | [TK]D-Fender | locked* |
| 14:29.25 | [TK]D-Fender | lirakis: Mines 5b |
| 14:29.26 | lirakis | [TK]D-Fender: just rm and touch? |
| 14:29.49 | lirakis | hmm .. okay i didnt know it had the pid inside it |
| 14:29.53 | lirakis | ill poke around.. |
| 14:29.55 | lirakis | thanks |
| 14:30.01 | [TK]D-Fender | lirakis: no I jsut told you it has to hold the PID #. So no, a 0 byte touched file will not be better than your 0b CURRENT file :) |
| 14:31.42 | lirakis | [TK]D-Fender: yeah i got that... hmm i get permission denied when i try to open it with vim .. and i can see the file with ls, but when i try to cat the file, i get "no such device or address" |
| 14:35.22 | *** join/#asterisk funxion (n=x@63.214.236.140) |
| 14:35.53 | funxion | does anyone know what the character limit is in the callerid(num) field over an * PRI? |
| 14:39.06 | De_Mon | why does * pad incoming CID? "SOME ONE " <mynum> |
| 14:46.09 | lirakis | [TK]D-Fender: i think i have several instances of * running... and that has messed up the ctl file |
| 14:46.19 | lirakis | <PROTECTED> |
| 14:47.14 | [TK]D-Fender | lirakis: what OS? |
| 14:47.25 | lirakis | [TK]D-Fender: centos 4 |
| 14:47.48 | [TK]D-Fender | lirakis: I'd say kill them all and restart |
| 14:48.45 | lirakis | [TK]D-Fender: thats the idea... im waiting for a dip in call volume since this a call center box |
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| 14:53.14 | *** join/#asterisk Hastalavi (n=kumar@mail.netvita.com) |
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| 14:56.27 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 14:56.42 | cjk | hi, does asterisk support any format for voicemail etc... that is readable by default under windows? |
| 14:58.15 | mosty | cjk, wav |
| 14:58.26 | mosty | or mp3 |
| 14:58.54 | lirakis | [TK]D-Fender: done and fixed |
| 14:59.10 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 14:59.23 | cjk | mosty, ok thanks. even though i have problems with the .wav |
| 14:59.41 | [TK]D-Fender | mosty: No, not MP3 |
| 15:00.03 | [TK]D-Fender | cjk: I've never had issues with it. VLC & Winamp both play fine |
| 15:00.28 | cjk | [TK]D-Fender, yep but both are not available under windows by default |
| 15:00.33 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:00.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:01.46 | [TK]D-Fender | cjk: "default" is a very dirty word. ActiveX is avaiable "by default" with MSIE. thats not a GOOD thing... |
| 15:02.22 | cjk | [TK]D-Fender, yes, you are right but mnost of the time i dont control the users desk |
| 15:02.27 | mosty | i was sure i had asterisk configured to save voicemail as mp3 in the past, i must have setup a wrapper of some sort. in any case i remember that it sounded terrible |
| 15:02.50 | [TK]D-Fender | mosty: probably wrote an "externnotify" script to hack it in. |
| 15:02.58 | [TK]D-Fender | mosty: but * can't encode MP3 itself. |
| 15:05.51 | *** join/#asterisk _pepo_ (n=c9eea608@190.10.187.20) |
| 15:05.52 | agx | (whinemode on)looks like snom has released firmware v7.x.x on all new phones and it does not even register while 6.5.10>12 was great and zero problems(whinemode off) |
| 15:06.06 | _pepo_ | hi friends |
| 15:06.48 | r3zon8 | is switchvox worth the $? |
| 15:07.13 | mosty | agx, http://provisioning.snom.com/from7to6/downgrade.php put that in the provisioning setting on the web interface |
| 15:07.35 | agx | mosty, that's the issue :) 80 phones x 10 mins = 800 mins lots |
| 15:08.02 | mosty | agx, setup local provisioning, and push that setting out to the phones |
| 15:08.25 | agx | mosty, next time i buy phones i better ask the distributor for specific firmware version |
| 15:08.44 | r3zon8 | or snom for better firmware |
| 15:09.56 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:10.40 | [TK]D-Fender | agx: Or just get better phones :) |
| 15:10.48 | mosty | if you have 80 phones, you should setup a local provisioning server anyway |
| 15:11.09 | [TK]D-Fender | r3zon8: As for Switchbox... depends how you value what it offers you. What have you compared it to, and what do you want? |
| 15:11.36 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 15:11.57 | Qwell | r3zon8: there is a free version that you can trial - see if you like it, and if so, upgrade to the full version |
| 15:11.58 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:13.31 | r3zon8 | i got it, and it works well, im just not completely sold on it just yet...does anyone here use it? |
| 15:14.27 | r3zon8 | im comparing it to asterisk now, and now is also working fone for me, i just get some occasional lockups |
| 15:15.04 | r3zon8 | switchvox, if i read correctly, also wants you to buy their pc hardware |
| 15:15.08 | r3zon8 | not just the cards |
| 15:15.40 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 15:15.40 | *** join/#asterisk oej (n=olle@193.126.30.214) |
| 15:20.00 | Corydon76-dig | r3zon8: correct, because it eliminates possible hardware conflicts for their support department |
| 15:29.43 | r3zon8 | i think ill go with asterisk first, see how it works out, and if i cant get it working, or not as easy to admin ill change to switchvox |
| 15:30.50 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 15:34.01 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 15:35.12 | *** join/#asterisk blepsoaf (n=pbaker@167.206.216.189) |
| 15:36.03 | blepsoaf | hello all, is there a way to do in call announcements with queues. IE if you want to pop in every 30 seconds and offer the ability leave a message instead of waiting on hold. |
| 15:37.59 | mosty | i believe there's a periodic-announcement setting |
| 15:38.05 | blepsoaf | I understand you can do the announcement with periodic-announce, but how to break out is another thing |
| 15:38.05 | *** join/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 15:38.08 | [TK]D-Fender | blepsoaf: Yes, and its well commented in the sample queues.conf that came with your source tarball |
| 15:38.35 | [TK]D-Fender | blepsoaf: "context=" in your queue. Read the SAMPLE. its all in there |
| 15:40.25 | geminidomino | Another google failure. Whenever I try to make an IAX2 call out, I get this in my full log: "Warning[xxx]: app_dial.c: Unable to create channel of type 'IAX2' (cause 3 - no route to destination)." Anyone ever seen that before? AFAIK, I've set everything up to connect to VP all right. |
| 15:40.47 | blepsoaf | great thank you - was looking at the wiki |
| 15:41.06 | errr | Im having some trouble getting the VoiceMailMain() to pickup in asterisk 1.4.15 http://pastebin.ca/805668 any ideas? |
| 15:44.24 | De_Mon | geminidomino its a pretty self explanitory error |
| 15:44.41 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 15:44.50 | De_Mon | maybe you should show us the whole CLI output of the call attempt on pastebin |
| 15:46.41 | geminidomino | de_mon: http://www.pastebin.ca/805672 The odd part is I can connect to VP directly with Kiax all right... |
| 15:47.17 | mosty | errr, looks like codec issue. do you have g729 licences? |
| 15:47.57 | errr | mosty: no I just found that answer on the list, thanks for the tip though |
| 15:48.22 | errr | some day I will learn I am not the first one to have problems, and to google first :) |
| 15:48.36 | mosty | errr, what was the issue? |
| 15:49.54 | De_Mon | omg |
| 15:50.58 | De_Mon | looks like your problem is with VP-IAX2-A in iax.conf |
| 15:51.08 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 15:51.26 | errr | mosty: in my database I created I told it to use g729;ilbc;gsm;ulaw;alaw and I removed the g729;ilbc; and that fixed it |
| 15:51.29 | De_Mon | (you missed a phone number substitution btw) |
| 15:52.30 | De_Mon | maybe not, its hard to tell with all that macro crap going on |
| 15:52.32 | hesco | In my logs, I'm getting an outgoing phone number on calls which fail during or immediately after the Dial command, but not on the successful calls. How can I configure CDR to ALWAYS record the phone number for an outgoing phone call? |
| 15:52.50 | geminidomino | De_Mon: Tell me about it... |
| 15:53.10 | De_Mon | geminidomino do you have a VP-IAX2-A decliration in iax.conf? |
| 15:53.41 | geminidomino | it's in iax_additional.conf, actually |
| 15:53.50 | [TK]D-Fender | De_Mon: FREEPBX |
| 15:53.56 | [TK]D-Fender | geminidomino: please note : |
| 15:53.58 | [TK]D-Fender | ~freepbx |
| 15:53.59 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:54.02 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:54.10 | geminidomino | ok, so it's a freepbx problem? |
| 15:54.21 | geminidomino | or rather, thier CF dialplans? |
| 15:54.25 | [TK]D-Fender | geminidomino: Its your configuration of FreePBX. |
| 15:54.44 | [TK]D-Fender | geminidomino: go read some of their setup guides again for your ITSP |
| 15:54.48 | geminidomino | Voicepulse's, not mine. :) Followed thier instructions. :) |
| 15:55.09 | De_Mon | freepbx is YUCK |
| 15:55.09 | geminidomino | I just wanted to make sure it wasn't an asterisk build issue or something like that. Thanks |
| 15:55.20 | [TK]D-Fender | geminidomino: Never is. |
| 15:55.31 | geminidomino | De_Mon: Agreed. I hate it with a passion. Unfortunately I don't make the rules, I'm just the server monkey. :P |
| 15:55.38 | [TK]D-Fender | geminidomino: * works, VP works, your configs don't. |
| 15:55.56 | De_Mon | you |
| 15:56.02 | hesco | How can I configure CDR to ALWAYS record the phone number for an outgoing phone call? |
| 15:56.24 | De_Mon | you'll have to talk to freepbx, they jump through a few extra hoops we're not familiar with |
| 15:56.41 | geminidomino | Thanks |
| 15:57.00 | geminidomino | One last question, about the * CLI... if I do "IAX2 show channels", is it normal for the channel column to say (none)? |
| 15:57.27 | [TK]D-Fender | geminidomino: entirely possible. Those do not necessarily relate to active calls |
| 15:57.43 | geminidomino | ok, just making sure that wasn't my problem. :) |
| 15:57.51 | *** join/#asterisk JayTee52 (n=jforde05@207-67-84-185.static.twtelecom.net) |
| 15:57.57 | geminidomino | I'll hit #FreePBX then. Thanks guys. |
| 15:58.54 | muiro | woah, trying to load lumenvox blew my asterisk. Not as painless as I had anticipated. |
| 16:00.21 | muiro | I'll bet it's something to do with running asterisk as non-root |
| 16:01.08 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 16:01.21 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 16:01.58 | *** part/#asterisk tarantul (n=tarantul@hacer.buildnet.ru) |
| 16:09.45 | De_Mon | eh, is it normal for the ? to not work in the * cli? |
| 16:12.27 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 16:12.30 | ussrback | hi all |
| 16:12.56 | ussrback | in meetme.conf there is [room] section. is it a context? |
| 16:13.00 | *** join/#asterisk mog (i=mog@nat/digium/x-01c9c2aa9f3b284c) |
| 16:13.00 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:13.10 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-d8c16df9bc9a9188) |
| 16:13.56 | De_Mon | ussrback no, they are pre-made rules |
| 16:13.58 | De_Mon | rooms |
| 16:14.27 | [TK]D-Fender | ussrback: there are no user-definable "sections" for meetme.conf. |
| 16:14.28 | ussrback | sooo cant i add other rules and use it? |
| 16:14.41 | ussrback | [TK]D-Fender: ahh ok |
| 16:14.46 | [TK]D-Fender | ussrback: [general] and [rooms] are fixed. There is not other functionality |
| 16:15.09 | De_Mon | rules? |
| 16:15.28 | De_Mon | I corrected myself! |
| 16:15.36 | *** join/#asterisk pepo--- (n=pepOSX@190.79.246.105) |
| 16:15.43 | ussrback | [TK]D-Fender: thanks |
| 16:15.48 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 16:19.22 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:21.44 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 16:25.11 | ussrback | anybody can give me some information abut meetme realtime? |
| 16:26.11 | muiro | question: anyone have much experience with LumenVox? |
| 16:29.13 | muiro | when I put the lumenvox module file into asterisk's modules directory, asterisk fails to load. On the lumenvox site I was told to add environment variable if I was going to run asterisk through a script and I have. Asterisk fails to load when the module .so file is in the modules directory. |
| 16:30.23 | file | muiro: a pastebin of what happens when you try to start it would be helpful... |
| 16:30.37 | *** part/#asterisk harpal (n=Harpal@124.125.255.24) |
| 16:31.31 | muiro | hold on, that's on tty9 |
| 16:33.35 | ussrback | is realime function supported for meetme? |
| 16:36.06 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 16:36.11 | ussrback | is realime function supported for meetme? |
| 16:37.24 | _Sam-- | o app_setcidname.o |
| 16:37.29 | _Sam-- | er damn right click |
| 16:39.33 | ussrback | how can i make realtime meetme? |
| 16:40.36 | file | ussrback: on my screen... I have counted you have asked 4 times, and that's without scrolling up... either 1. Nobody knows, 2. People are busy, or 3. It can be answered by a simple Google |
| 16:40.58 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 16:40.58 | file | that being said it is supported in Asterisk 1.4 |
| 16:41.44 | ussrback | @file: i think that its an only place to get answers. google says that its possible but no reall examples. |
| 16:43.03 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:43.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:43.08 | twisted | russellb |
| 16:43.22 | *** join/#asterisk aikanaro79 (n={aikanar@89-180-91-152.net.novis.pt) |
| 16:43.24 | De_Mon | is this considered a bug? http://pastebin.ca/805723 |
| 16:43.46 | De_Mon | I dialed an extension and from the console and the console disconnected |
| 16:43.50 | aikanaro79 | how can I enable logging in asterisk? I need it to log everything...including SIP register attempts |
| 16:44.04 | De_Mon | aikanaro79 /etc/asterisk/logger.conf |
| 16:44.13 | aikanaro79 | De_Mon, thanks |
| 16:44.20 | tzafrir | aikanaro79, look at the "full" sample in logger.conf |
| 16:44.23 | *** join/#asterisk aurax (n=axaxax@192.115.235.250) |
| 16:44.31 | aikanaro79 | good tip |
| 16:44.32 | aikanaro79 | :) |
| 16:44.53 | *** part/#asterisk aikanaro79 (n={aikanar@89-180-91-152.net.novis.pt) |
| 16:45.44 | muiro | file: http://pastebin.com/m7c1a68a7 |
| 16:45.48 | De_Mon | eeeeeeh |
| 16:46.07 | file | muiro: that's the environment variables, they aren't set |
| 16:46.20 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:46.55 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 16:47.39 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:48.30 | muiro | They're set. Set echo them from the safe_asterisk script and from both the user's bashrc and root's bashrc |
| 16:48.32 | jameswf | anyone play with jeOS |
| 16:48.33 | muiro | they're set |
| 16:48.50 | file | muiro: are you starting it in the console? |
| 16:49.03 | file | well, I mean try it that way... |
| 16:49.22 | De_Mon | whhhat the crap? |
| 16:49.28 | De_Mon | [Dec 4 11:48:38] NOTICE[20957]: chan_sip.c:13605 handle_request_invite: Failed to authenticate user "Jon Webster" <sip:office2@pbx.elephantoutlook.com>;tag=9ACA4F97-6FF4A10C |
| 16:49.30 | muiro | file: no, it starts automaticalyl through through init with safe_asterisk |
| 16:49.45 | De_Mon | thats what I get when I try to dial [Dec 4 11:48:38] NOTICE[20957]: chan_sip.c:13605 handle_request_invite: Failed to authenticate user "Jon Webster" <sip:office2@pbx.elephantoutlook.com>;tag=9ACA4F97-6FF4A10C |
| 16:49.50 | De_Mon | craptastic |
| 16:49.56 | De_Mon | dialplan add extension 4001,1,NoOp(${IF(true:false)}) into local replace |
| 16:49.57 | muiro | file: that part of the pastebin was from when I interrupted it with service asterisk restart |
| 16:50.22 | *** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 16:50.38 | muiro | file: I've double checked that the environment variables are actually correct as well, but I'll check again |
| 16:50.41 | file | muiro: confirm the environment variables are set using env in your console, if so then start it there... if it doesn't work then they may be set to the wrong values... if it does start then you know the variables aren't set before starting in safe_asterisk |
| 16:50.49 | file | and by "start it there" I mean start it in your console |
| 16:51.03 | *** join/#asterisk Op3r (n=edwin@203.177.210.124) |
| 16:51.15 | *** join/#asterisk aikanaro79 (n={aikanar@89-180-91-152.net.novis.pt) |
| 16:52.47 | aikanaro79 | I...I'm trying to connect to asterisk with SIP but it's never successful...I'm developing my own app but I've tried with another and it went well |
| 16:53.01 | aikanaro79 | I answer the challenge with an authorization header |
| 16:53.10 | aikanaro79 | and I'm lost as to what may be the problem |
| 16:53.15 | aikanaro79 | can anyone help me? |
| 16:53.36 | mamep | anyone experienced with chan_h323? |
| 16:54.08 | aikanaro79 | mamep, is that for me? |
| 16:54.25 | mamep | not aikanaro79 |
| 16:54.34 | aikanaro79 | mamep, sorry then :) |
| 16:54.35 | mamep | i'm looking for some help too |
| 16:54.38 | mamep | np |
| 16:58.50 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:58.53 | *** join/#asterisk smash- (n=smash@c-71-59-163-135.hsd1.wa.comcast.net) |
| 16:59.07 | smash- | hey anyone know a place to get royalty free music .. |
| 16:59.17 | smash- | that is actually free to download |
| 16:59.24 | [TK]D-Fender | aikanaro79: Next time you ask something like that, come in with a pastebin of a failed call attempt at verbose 10 and sip debug enabled in hand. |
| 16:59.35 | muiro | smash-: google the "internet archive". They have a lot of CC music |
| 17:01.51 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:02.38 | smash- | k |
| 17:02.49 | smash- | huge site hard to find usefull content |
| 17:03.45 | muiro | smash-: keep looking. You'll find it. |
| 17:05.18 | FlatFoot | [TK]D-Fender: why would you prefer verbose set at 10 ? no larger ? |
| 17:06.07 | [TK]D-Fender | FlatFoot: it has a nice meaningful feel to it without looking obscene or ridiculous. |
| 17:06.26 | FlatFoot | aha makes sense , just interested in prefs |
| 17:07.31 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
| 17:07.35 | nephfl | hello |
| 17:08.17 | muiro | 10 seems like a square number |
| 17:08.25 | muiro | even though, mathematically, speaking... |
| 17:08.26 | Qwell | square of 100 |
| 17:08.45 | FlatFoot | root 100 |
| 17:08.53 | muiro | FlatFoot's got it |
| 17:09.36 | nephfl | what is the easiest auto dialer to setup for asterisk? |
| 17:09.38 | muiro | because 10 isn't a square number |
| 17:10.14 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:10.26 | Qwell | muiro: sure it is |
| 17:10.34 | FlatFoot | 3.16227766 ^2 |
| 17:10.36 | Qwell | it just isn't a square number of a whole |
| 17:10.42 | Qwell | ;) |
| 17:10.44 | nephfl | lol |
| 17:11.33 | muiro | poo on you qwell because... |
| 17:11.37 | *** join/#asterisk RoyK (n=roy@sbs.hardrockcafe.no) |
| 17:11.37 | muiro | because it is so |
| 17:11.51 | FlatFoot | an aside . Anyone tried to get a Wii console for christmas ? the term rocking horse sh*t comes to mind |
| 17:11.56 | muiro | file: used connector b18 instead of b19 and the env variable I had set worked |
| 17:12.47 | file | interesting, the connector itself doesn't care about it... just the Lumenvox libraries |
| 17:12.50 | nephfl | so? any really simple autodialers for asterisk? |
| 17:13.15 | FlatFoot | nephfl: try a wife |
| 17:13.37 | *** join/#asterisk disa-help (n=phobosd@shell.intarwebnetorg.com) |
| 17:13.46 | nephfl | what? |
| 17:13.56 | FlatFoot | get the wife to do it |
| 17:14.03 | FlatFoot | when shes finished the cleaning |
| 17:14.12 | De_Mon | what is supposed to happen if you try ${IF(condition:false)} and omit the ? |
| 17:14.28 | lirakis | nephfl: call spool is the "simplest" |
| 17:14.39 | aikanaro79 | [TK]D-Fender, I have those...I just didn't know if I was allowed to paste it here |
| 17:14.40 | lirakis | nephfl: but can also be the biggest nightmare |
| 17:14.51 | disa-help | does anyone know of any good queue metrics for asterisk? |
| 17:14.52 | [TK]D-Fender | aikanaro79: pastebin <------ |
| 17:14.53 | disa-help | http://queuemetrics.com/ |
| 17:14.54 | [TK]D-Fender | ~pb |
| 17:14.55 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:14.56 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 17:15.00 | disa-help | not taht though... |
| 17:15.02 | jer | any particular way i can set up my ivr so that if someone dials say, number 12, they're given an option to enter in a telephone number, and have * grab those digits and dial that number over another sip connection? |
| 17:15.04 | disa-help | tomcat. ugh |
| 17:15.17 | lirakis | nephfl: http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out |
| 17:15.19 | [TK]D-Fender | Jer yes |
| 17:15.21 | De_Mon | jer many ways read the book |
| 17:15.27 | jer | De_Mon, which book? |
| 17:15.31 | De_Mon | ~book |
| 17:15.32 | jbot | [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 http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:15.37 | jer | okay, thanks |
| 17:16.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:16.36 | nephfl | i just want to have asterisk call a list and connect to ivr... i tried VICIdial, but it has a ton of options and it didnt want to dial for me |
| 17:17.23 | lirakis | nephfl: dude .. are you listening? |
| 17:18.34 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 17:18.52 | [TK]D-Fender | lirakis: Or you can assume he saw that link and has no intention or writing his own |
| 17:18.56 | De_Mon | how do you call a list? |
| 17:19.01 | [TK]D-Fender | lirakis: and wants something more ready to run |
| 17:19.47 | lirakis | [TK]D-Fender: " just want to have asterisk call a list and connect to ivr" seems like that would take a few minutes to implement with call files |
| 17:20.22 | [TK]D-Fender | lirakis: Ummmm yeah, then there's the idea of queueing them so the system doesn't freak, etc |
| 17:20.27 | [TK]D-Fender | lirakis: more than a few minutes |
| 17:20.50 | aikanaro79 | [TK]D-Fender, sorry but how do I get a pastebin from sip debug? |
| 17:20.57 | [TK]D-Fender | ~pb |
| 17:20.58 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:20.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:21.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:21.12 | disa-help | whats pastebin? |
| 17:21.33 | De_Mon | someone please shoot me I don't deserve to live |
| 17:21.36 | aikanaro79 | [TK]D-Fender, I saw that...problem is that asterisk -r is a command line (or so to speak) |
| 17:21.52 | De_Mon | aikanaro79 COPY.. and pastebin! |
| 17:22.03 | FlatFoot | De_Mo: have more beer |
| 17:22.04 | [TK]D-Fender | aikanaro79: Scroll-back. Copy & paste. End of story |
| 17:22.10 | lirakis | sigh |
| 17:22.19 | aikanaro79 | thanks...was trying to use wgetpaste...my mistake |
| 17:22.39 | [TK]D-Fender | aikanaro79: read the BIG PRINT |
| 17:23.36 | *** part/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 17:24.12 | aikanaro79 | my output when I try to connect: http://pastebin.com/d2053a693 |
| 17:25.23 | [TK]D-Fender | aikanaro79: might be nice to see your sip.conf |
| 17:25.43 | aikanaro79 | in a sec |
| 17:26.17 | aikanaro79 | here: http://rafb.net/p/uIbrny37.html |
| 17:26.33 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:26.33 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:30.54 | aikanaro79 | [TK]D-Fender, any clues? |
| 17:31.03 | *** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-18-82.w82-120.abo.wanadoo.fr) |
| 17:32.08 | nephfl | is there an asterisk polling script? |
| 17:32.08 | [TK]D-Fender | aikanaro79: Check your pass.. all I can suggest |
| 17:32.14 | [TK]D-Fender | nephfl: Polling for what? |
| 17:32.18 | aikanaro79 | otherwise all seems good? |
| 17:32.57 | nephfl | like telephone polling, as in a dialer with ivr specifically designed for polling |
| 17:33.08 | [TK]D-Fender | aikanaro79: Well its a simple register an "401". Kinda self explanitory |
| 17:33.20 | *** join/#asterisk kkjoe (n=kkjoe@p578b42d3.dip0.t-ipconnect.de) |
| 17:33.27 | kkjoe | im getting poll.c:283: Assertion while getting a call on an ptp port with overlap reciving, im using misdn, does someone had this before ? |
| 17:33.30 | *** join/#asterisk mav3rick (n=t0m@trankil.biz) |
| 17:33.40 | [TK]D-Fender | nephfl: possibly, but I wouldn't bet on it |
| 17:34.06 | aikanaro79 | [TK]D-Fender, self-explanatory? so where's the problem? I've gonne through the RFC several times and I can't put my finger on it |
| 17:34.26 | mav3rick | hello |
| 17:34.27 | aikanaro79 | I've compared outputs with another app that registers ok |
| 17:34.34 | [TK]D-Fender | aikanaro79: looks like the apsswrod is wrong. thats all. Make sure of how each end is dealing with hasing, etc. |
| 17:34.36 | aikanaro79 | and apart from that everything else seems to be ok |
| 17:38.14 | *** join/#asterisk riddlebox (n=JamesMid@68.143.86.98.nw.nuvox.net) |
| 17:42.16 | *** join/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 17:44.50 | Zuchmir | if i have mutiple for example 3 DIDs comming in on 23 channels (on a T1), is there any way of knowing which DID the user dialed (so that I can send it off to a unique extension), or can i only see which physical channel it came in on? |
| 17:46.25 | De_Mon | [TK]D-Fender you been drinking? |
| 17:46.52 | mav3rick | hi guys. I have a strange problem with /dev/zap/pseudo not correctly closed by Asterisk when I use my Digium card timer, and meetme conferences. When I unload chan_zap from Asterisk, everything works fine : all pseudo-channels are closed when the conferences are empty. I check the pseudo devices left open with "lsof". I am doing SIP-only conference (that explains the pseudo channel created). I tried Asterisk 1.4.14 and 1.4.15. I use zaptel 1.4.7. |
| 17:47.02 | [TK]D-Fender | De_Mon: No, its freezing ass cold in here and my typing is suffering.... |
| 17:47.02 | *** join/#asterisk bmcghee (n=brentmcg@d66-183-250-149.bchsia.telus.net) |
| 17:47.03 | Qwell | mav3rick: funny you should bring that up |
| 17:47.08 | Qwell | mav3rick: that was just fixed yesterday :) |
| 17:47.09 | bmcghee | Heya Qwell! |
| 17:47.16 | mav3rick | really ? |
| 17:47.18 | Qwell | yep |
| 17:47.31 | mav3rick | wow, my boss will love me :) |
| 17:47.37 | Qwell | the fix was tiny too - let me get the bug number, you can apply the patch |
| 17:47.45 | Qwell | 11383 I think |
| 17:47.54 | Qwell | http://bugs.digium.com/bug_view_page.php?bug_id=11383 |
| 17:49.16 | mav3rick | haaaa that's so great ! |
| 17:49.17 | mav3rick | thanks ! |
| 17:49.42 | smash- | If im running a remote phone into my pbx does it need all ports 10000-20000 open? |
| 17:49.44 | smash- | and 5060? |
| 17:49.47 | smash- | udp |
| 17:50.42 | [TK]D-Fender | smash-: Go read this : |
| 17:50.44 | [TK]D-Fender | ~sipnat |
| 17:50.45 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:50.46 | [TK]D-Fender | ^^^ |
| 17:50.51 | bmcghee | where can i quiz someone on why asterisk-addons are fooooking up |
| 17:51.12 | [TK]D-Fender | bmcghee: pastebin your failed attempt |
| 17:51.14 | [TK]D-Fender | ~pb |
| 17:51.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:51.16 | [TK]D-Fender | ^^^^^ |
| 17:51.35 | *** join/#asterisk bantu (n=Miranda@p54A32A5D.dip0.t-ipconnect.de) |
| 17:52.15 | bmcghee | http://pastebin.com/d5b9ea2e9 |
| 17:57.15 | bmcghee | =( |
| 17:57.39 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:57.53 | Qwell | bmcghee: you aren't telling it where to find asterisk |
| 18:01.05 | geminidomino | Anyone have a quick guide to IAX2 troubleshooting? I'm gonna nuke FreePBX and try to get it running raw. |
| 18:02.55 | RoyK | geminidomino: I think Yate has a pretty good IAX2 stack :P |
| 18:03.08 | geminidomino | ~yate |
| 18:03.09 | jbot | yate is probably a pbx written in c++, something more a solid framework than asterisk, possibly better, who knows. Try #yate if you want to know more. |
| 18:04.31 | bmcghee | Quell: what do you mean |
| 18:07.56 | *** join/#asterisk bmg505 (n=leon@196.209.178.145) |
| 18:09.16 | *** join/#asterisk sid (i=unstable@tor/regular/sid) |
| 18:09.21 | sid | I have 3 people in my company, I only need one phone line right now, I want to use this Aastra sip phone, I have a mini-itx box, with one PCI slot..what card should I get? |
| 18:12.27 | *** join/#asterisk riddlebox (n=JamesMid@68.143.86.98.nw.nuvox.net) |
| 18:12.46 | riddlebox | man I have been working with a guy all morning, all I had to do was plug the PMS cable into the back of the Hitachi to get the PMS system working but he is making me test the cable from the PMS to the switch, he wont believe me that it is his PMS vendor that has the |
| 18:16.45 | Zuchmir | is there any way to know which phone number the caller dialed? |
| 18:17.45 | hesco | If I were to set up to log to my postgresql backend, instead of to /var/log/asterisk/cdr*, where might I find the CREATE TABLE statement to start that ball rolling? |
| 18:20.12 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 18:21.18 | docelmo | Does anyone know of any issues with 1.4.14 and RFC2833? |
| 18:21.30 | docelmo | I am experiencing a lot of issue with RFC2833 |
| 18:23.22 | *** join/#asterisk saftsack (n=saftsack@pD9E04F88.dip.t-dialin.net) |
| 18:24.50 | bmcghee | QWell: how do i tell it where to look. |
| 18:24.56 | Qwell | ./configure --help |
| 18:25.29 | twisted | docelmo: tell asterisk to read the rfc. |
| 18:26.18 | docelmo | true.. but is there a known issue in 1.4.14 with RFC2833 that is fixed in 1.4.15? |
| 18:26.50 | twisted | *shrug* i don't use 1.4 |
| 18:27.40 | docelmo | Anyone? |
| 18:27.51 | file | docelmo: what exactly is wrong... and is it receiving or sending, and from what/to what |
| 18:28.27 | *** part/#asterisk RoyK (n=roy@sbs.hardrockcafe.no) |
| 18:29.01 | twisted | oh no |
| 18:29.20 | twisted | can't let him loose |
| 18:29.22 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:29.57 | bmcghee | Qwell: i did ./configure --localedir= and it still fails on gmake |
| 18:35.03 | *** join/#asterisk techie (n=techie@adsl-76-247-11-252.dsl.lsan03.sbcglobal.net) |
| 18:37.29 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:37.39 | *** join/#asterisk saftsack (n=saftsack@p4FC773FA.dip.t-dialin.net) |
| 18:41.24 | *** join/#asterisk grantm (n=grant@kolob.wingateservices.com) |
| 18:41.24 | riddlebox | twisted, why dont you use 1.4? |
| 18:42.55 | hesco | ~release |
| 18:42.56 | jbot | When is release ... the official answer: "Debian has a strict policy of never announcing release dates; therefore, the release date for ``sarge'' is not yet known.", or Stable release is "woody" (3.0), Testing release is "sarge", following that will be something not yet named |
| 18:43.55 | hesco | Is this channel specifically about the Debian version of Asterisk? And if so, is there a different channel for the upstream? |
| 18:44.17 | blitzrage | hesco: the debian version of asterisk? |
| 18:44.25 | blitzrage | asterisk is just asterisk... |
| 18:44.40 | hesco | jbot just gave me an answer about the Debian release. |
| 18:44.52 | Qwell | hesco: it's a shared bot |
| 18:44.53 | hesco | I know that Debian packages * in apt-get |
| 18:44.59 | hesco | ok, thanks. |
| 18:45.06 | Qwell | what are you trying to find out? |
| 18:45.10 | blitzrage | just compile... you'll be more up to date, and it isn't that hard |
| 18:46.35 | hesco | I'm still on an initial learning curve, having read the open sourced book from OReilly's and building a tweaking out an * server for the first time. I've had many questions, many of them still pending. At the moment I'm stumped on a postgres permissions issue. But that was really a distraction. |
| 18:46.47 | hesco | My current priorty is to figure out how . . . |
| 18:46.58 | *** part/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 18:47.01 | hesco | to log the outgoing phone number in CDR |
| 18:54.17 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 18:56.28 | brodiem | Can anyone help with a prob I'm having with queues sending calls to the local channel? The problem is that if a recipient at the local channel (bridging to a SIP channel) transfers the call elsewhere, the initial local channel stays open and therefore keeps two active calls open - the transferer and transferee. I think if I add the queue members without the "/n" option on the local channel it would work properly, but then the queue can't ret |
| 18:56.49 | Qwell | can't re.. |
| 18:57.09 | Qwell | brodiem: you got cut off there |
| 18:57.17 | brodiem | ah, where at? |
| 18:57.20 | Qwell | can't re |
| 18:57.47 | brodiem | can't retain any kind of status on the channel (i.e. to know that the channel is in use and not send calls there) |
| 18:59.13 | brodiem | am I making sense? |
| 19:01.17 | Yourname`` | Hi. What do I look out for in Manager events that'll tell me about SIP responses? |
| 19:01.17 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 19:02.18 | [TK]D-Fender | Yourname`There isn't any |
| 19:03.05 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 19:03.21 | deeperror | anyone know of any phones that would have 1 line sip 1 line analog? |
| 19:04.12 | [TK]D-Fender | deeperror: SPA-3102 |
| 19:04.14 | Qwell | deeperror: get an ATA |
| 19:04.29 | deeperror | got several ata's but i need a phone that coudl do both |
| 19:04.43 | grandpapadot | fa la la la la, la la la laaaaa .. |
| 19:04.57 | *** join/#asterisk Laureano (n=Laureano@200.59.172.38) |
| 19:05.11 | brodiem | Anyone have any ideas, qwell, fender? I know I can't be the only one using the local chan driver for a replacement to chan_agent |
| 19:05.49 | *** join/#asterisk nny (n=nny@64.203.237.227.static-pool-2.pool.hargray.net) |
| 19:05.55 | nny | hmm i love NTP |
| 19:06.17 | grandpapadot | Today on SlashDot: AT&T To Decommission Pay Phones |
| 19:06.31 | Qwell | I wonder if they're gonna be selling them |
| 19:06.35 | deeperror | if i get an ATA then i still need to purchase 2 line phones |
| 19:06.41 | [TK]D-Fender | deeperror: get a 2-line phone or run a single line phone on an SPA-3102 in passthrough mode. |
| 19:06.42 | grandpapadot | My blue box hasn't worked in years, guess it's about time ... |
| 19:06.55 | nny | so if i do an ntptime on the server, i getan ok. I copied these configs from a previous install that works. Using a polycom 501, with SNTP sett to the server IP and I still get the flashing date of doom, any way to troublehoot? |
| 19:07.15 | nny | sorry typing on an eeepc, makes me type like a 4 year old |
| 19:07.22 | [TK]D-Fender | nny: First pastebin your configs and phone logs |
| 19:07.29 | nny | [TK]D-Fender, roger that |
| 19:07.56 | Yourname`` | Hello, again. What event do I need to look for in manager that gives me SIP responses? |
| 19:08.12 | brodiem | heh |
| 19:08.43 | [TK]D-Fender | Yourname``I already answered you. You can't. It wasn't designed for that. |
| 19:08.50 | [TK]D-Fender | uload chan_brokenrecord.so |
| 19:10.06 | Yourname`` | [TK]D-Fender: Sorry, didn't get highlighted. Ok, so you sy it's not possible? Basically, I'm trying to use manager to log these SIP events on a web gui.. so, without looking at the CLI, I can be "alerted" when these SIP responses occur. Like 408, 508, etc.. |
| 19:10.23 | [TK]D-Fender | Yourname`You can't. |
| 19:10.38 | nny | [TK]D-Fender, first ntp.conf |
| 19:10.40 | nny | http://pastebin.com/m25836157 |
| 19:11.05 | [TK]D-Fender | Yourname``: And for highligh, autocomplete stops on the first `. And suffixs like "`" suck |
| 19:11.24 | *** join/#asterisk Tourinho (n=tourinho@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 19:11.27 | Qwell | [TK]D-Fender: it's stopping because he has two clients in here |
| 19:11.32 | Yourname`` | [TK]D-Fender: Wow, seriously? So there is no way I can look remotely to see if we're getting SIP errors other than having to go into the CLI? |
| 19:11.37 | Tourinho | good day all |
| 19:11.42 | Yourname`` | Qwell: LOL, I was just going to type that. Thank you. |
| 19:11.52 | Qwell | (which is often not recommended, for that very reason) |
| 19:12.06 | nny | [TK]D-Fender, which log files for phone? App or Boot? |
| 19:12.25 | [TK]D-Fender | Yourname``: * is not a diagnoses friendly SIP proxy. |
| 19:12.32 | [TK]D-Fender | nny: YES. |
| 19:12.37 | nny | [TK]D-Fender, lol k |
| 19:12.47 | Yourname`` | I know, I know.. I like screen but it gets cumbersome for me at times with ircii, so I login from home and work with Chatzilla. |
| 19:13.20 | [TK]D-Fender | nny: and your sip.cfg |
| 19:13.27 | Yourname`` | [TK]D-Fender: Ok, so I'm guessing Asterisk isn't capable of doing so. How about, in your experience, what could I use to help me out with it? |
| 19:13.29 | Qwell | Yourname``: irc proxy |
| 19:14.04 | Yourname`` | Qwell: Used to use irssi, but that got cumbersome for me too :( |
| 19:14.13 | Qwell | irssi isn't a proxy |
| 19:14.15 | tzafrir | Yourname``, I see people are more happy with bitchx and irssi |
| 19:14.25 | Yourname`` | tzafrir: I know, haven |
| 19:14.34 | Yourname`` | haven't tried irssi in over 3 years :( |
| 19:14.44 | Yourname`` | I hear it's gotten much better |
| 19:14.51 | Yourname`` | Qwell: You can use irssi as a proxy.. |
| 19:15.04 | tzafrir | I only tried ircii once. It badly lacked support for other charsets |
| 19:15.41 | Tourinho | Guys, an AGI script is able to monitor a channel after it executes a Dial application? Or when Dial is used * gets the controler and just return it after Dial ends? |
| 19:15.50 | Yourname`` | I liked it before irssi, heh |
| 19:16.02 | Tourinho | Using Asterisk Manager inside the same AGI script |
| 19:16.36 | twisted | so in asterisk now |
| 19:16.45 | twisted | why does everything suck? :P |
| 19:16.59 | Yourname`` | Because it's a beta, and a GUI. |
| 19:17.14 | grandpapadot | I just installed AsteriskNOW in a VM to play with it. Don't like it, but "A" for effort. |
| 19:17.21 | *** join/#asterisk ACiDV (n=acidv@97-147.dr.cgocable.ca) |
| 19:18.54 | *** join/#asterisk swampfox0866 (n=frankb@166.70.132.97) |
| 19:20.03 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:20.31 | Yourname`` | Oh, man. This would rock if manager included sip events.. so I could log it. :( |
| 19:21.14 | grandpapadot | Yourname: likely your web server would be the issue (unending persistent connection). |
| 19:21.57 | grandpapadot | Or were you planning on running a looping script, parsing, and stuffing the results into a db? |
| 19:22.02 | Yourname`` | grandpapadot: It's a script. |
| 19:22.17 | grandpapadot | php? |
| 19:22.20 | gr0mit | is 1.4.x branch now stable enough for production use? |
| 19:22.23 | Yourname`` | There we go, that's what's happening right now with most other manager events showing up on an AJAX GUI. |
| 19:22.42 | Yourname`` | grandpapadot: The scripts are in PHP, yes. |
| 19:22.45 | grandpapadot | That's what we're doing, ajax to AMI. |
| 19:22.51 | grandpapadot | via php |
| 19:23.09 | grandpapadot | Yourname: internal or commercial product/service? |
| 19:24.47 | blitzrage | gr0mit: I've been using it for over a year :) |
| 19:25.09 | Yourname`` | grandpapadot: Both. |
| 19:25.19 | gr0mit | is it as stable or more stable than 1.2.x ? |
| 19:25.21 | grandpapadot | Cool. Website? |
| 19:25.23 | Yourname`` | grandpapadot: Primarily internal. Commercial for friends. |
| 19:25.44 | blitzrage | gr0mit: always test |
| 19:25.52 | grandpapadot | gr0mit: I find 1.2.24 very stable. |
| 19:26.06 | gr0mit | i am running 1.2.24 on my boxes at the moment |
| 19:26.16 | grandpapadot | gr0mit: Why updating? curious |
| 19:26.24 | gr0mit | new machine |
| 19:26.29 | grandpapadot | Ahh. |
| 19:26.36 | Yourname`` | grandpapadot: There's times when our callcenter calls us saying, "umm, something is not right".. and we have to login to atleast 3-4 dialers to see if our provider(s) are messing up and giving us stupid SIP errors. To streamline that, I was thinking of putting that into manager and then on the GUI, so all we do is look. |
| 19:26.55 | gr0mit | dont want to install a load of trouble though |
| 19:27.27 | ACiDV | someone here use Asterisk 1.4.x + Queue + AddQueueMember + Local Channel (with or without /n ??) + Call Transfer ? |
| 19:28.07 | grandpapadot | gr0mit: I would say unless there is a compelling reason to upgrade, wait until there is one. |
| 19:29.27 | gr0mit | prob is that the time there is a compelling reason to upgrade, it will be an emergency |
| 19:29.57 | gr0mit | so i would rather get it over and done with before it becomes critical |
| 19:30.11 | grandpapadot | gr0mit: manage your managers .. lol |
| 19:31.14 | brodiem | ACiDV: I have EXACTLY the same question |
| 19:31.18 | ACiDV | When I use /n in local channel, I can see channel status (In Use, Not InUse) in 'show queues' but agent that initiate a call transfer cannot receive new call until 2nd agent hangup. When remove /n, call transfer work perfectly but 'show queues' always report 'Not InUse' |
| 19:31.19 | grandpapadot | My list of 'reasons to go to 1.4' is growing but not quite there yet. |
| 19:32.03 | brodiem | ACiDV: what I have to do for the time being is set ringinuse=yes in queues.conf |
| 19:32.19 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 19:32.21 | brodiem | ACiDV: although its causing server load to get out of hand with queuemetrics getting millions of attempts.... |
| 19:32.36 | brodiem | ACiDV: err reading the attempts from queue_log |
| 19:33.29 | ACiDV | brodiem But what is recommanded... use /n in local channel + addqueuemeber ? what are you using |
| 19:34.30 | gr0mit | gradpapadot: what reasons do you have so far? |
| 19:35.11 | brodiem | ACiDV: the problem is that when using /n, transferring calls doesn't release the initial local channel, so to the queue there are 2 active calls. Removing /n corrects that but of course you lose the status info. |
| 19:35.20 | brodiem | ACiDV: at least that's _my_ problem |
| 19:36.07 | ACiDV | brodiem: I presume I must simply ignore warning message on CLI :( |
| 19:36.41 | ACiDV | brodiem: by chance, * 1.4.15 remove 'CDR not posted' warning |
| 19:37.24 | brodiem | ACiDV: turn off notice warnings to console nad just log them maybe |
| 19:38.04 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 19:39.30 | mocker | ihateavaya |
| 19:41.06 | codefreeze | ACiDV: we've put a fix for that into 1.4 svn; |
| 19:41.48 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:41.55 | Zuchmir | is there any way to know which phone number caller dialed? |
| 19:42.42 | brodiem | codefreeze: which fix are you referring to? |
| 19:42.49 | codefreeze | Zuchmir: ${EXTEN}, perhaps? |
| 19:43.04 | codefreeze | brodiem: the "not posted" message |
| 19:43.28 | brodiem | codefreeze: ah, thought maybe it was something related to the problem of queues + local channel + transfers =/ |
| 19:43.30 | Zuchmir | EXTEN tells me the extension routed to |
| 19:44.27 | Zuchmir | i want to tell apart users who dialed primary line (and were rolled over to secondary line), from those that dialed secondary line |
| 19:54.14 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 19:54.33 | rantsh | Hi guys |
| 19:54.52 | *** join/#asterisk jtexter3 (n=jamest@ip67-90-136-204.z136-90-67.customer.algx.net) |
| 19:55.47 | rantsh | quick question, I've never used SVN and an issue that got worked on SVN revision 41810 is affecting me, I'd like to know if there's a way to know which revision went to which main release |
| 19:56.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:56.49 | rantsh | This is because my asterisk version is very old and my boss doesn't want to upgrade |
| 19:57.09 | grandpapadot | rantsh: What's the issue? |
| 19:57.25 | rantsh | voicemail files permission |
| 19:57.36 | _x86_ | that's easy :) |
| 19:57.55 | grandpapadot | rantsh: You can just modify the .c file yourself, recompile. |
| 19:57.56 | rantsh | I want to be able to set modes to these files without using umask or chmod thru the command line |
| 19:58.22 | grandpapadot | rantsh: if memory serves, it's documented towards the top of the file. Assuming 1.2 branch ... |
| 19:58.30 | grandpapadot | er.. code base |
| 19:58.33 | rantsh | grandpapadot, I know I could... if I could convince my coward boss to do it |
| 19:58.54 | rantsh | grandpapadot, yup actually 1.2.3 |
| 19:59.29 | Yourname`` | Hmm, are there any words like "mark" or "alex" in the sounds dir? |
| 19:59.47 | grandpapadot | You could probably just download voicemail.c from 1.2.24 and recompile... backup your source tree first... |
| 20:00.05 | _x86_ | Yourname``: no proper nouns afaik |
| 20:00.20 | jtexter3 | Is it just me, or does app_amd not honor the total_analysis_time setting? |
| 20:00.49 | Yourname`` | Thanks _x86_ |
| 20:02.50 | rantsh | grandpapadot, I could probably do that or even upgrade to 1.2.24 altogether, panics everytime he hears the word recompile or upgrade on the prod server... hence I'm looking for a different approach, maybe there's a solution that makes him happy |
| 20:03.07 | grandpapadot | rantsh: also, just copy the voicemail.so file instead of doing a "make install". |
| 20:03.29 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 20:03.47 | rantsh | either way I'd like to know (just for knowing actually) if there's a way to match an svn revision to a stable release |
| 20:03.49 | luke-jr | Anyone know why Asterisk 1.4 doesn't handle timezones properly when sending VM emails? |
| 20:04.23 | grandpapadot | dunno, I personally would never run an SVN release in production, only formal releases... |
| 20:05.11 | rantsh | grandpapadot, you mean: get voicemail.so from a newer version and replace mine, no reloading, restarting or recompiling implied? |
| 20:05.47 | grandpapadot | rantsh: You'll probably have to at lease unload and load voicemail. I would personally just backup my old one, stop asterisk, copy the new one, start asterisk, but that's me. |
| 20:05.58 | rantsh | grandpapadot, me neither, let me explain myself: I just want to know if there is way to know what svn version went to which release |
| 20:06.21 | grandpapadot | Ahh.. No clue here, but somebody is bound to know the answer. |
| 20:07.47 | file | rantsh: svn info http://svn.digium.com/svn/asterisk/tags/<version> and look at the last changed rev |
| 20:07.54 | rantsh | grandpapadot, mmm... that might do it, I'll try that and see what happens, I'll have to do it during the night while off peak hours :p (2 to 3 am) |
| 20:08.15 | rantsh | file, thanks |
| 20:11.13 | brodiem | since others have joined I'll ask again.. |
| 20:12.05 | brodiem | Anyone know how to solve the problem of queue -> local channel -> transfer_call_elsewhere can be corrected so that the queue will stop reporting the local channel as "in use" even though the call was transferred (bridged) elsewhere? |
| 20:12.07 | grandpapadot | brodiem: No, member enlargement doesn't really work despite the emails. |
| 20:13.06 | *** part/#asterisk myiagy (n=myiagy@200.215.59.133) |
| 20:14.40 | brodiem | grandpapadot: huh? |
| 20:15.44 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
| 20:16.52 | nephfl | i always to have the most basic problems... i am trying to compile asterisk, and it is telling me i am missing ncurses (trying to use menuselect) svn release of 1.4 |
| 20:17.07 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 20:17.21 | grandpapadot | nephfl: Install ncurses? |
| 20:17.37 | nephfl | i installed ncurses with yum |
| 20:19.00 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:19.26 | jtexter3 | nephfl - Did you install ncurses-devel? |
| 20:19.36 | nephfl | yeah |
| 20:20.22 | nephfl | version 5.5 |
| 20:20.36 | *** join/#asterisk Op3r (n=Op3r@203.177.223.21) |
| 20:21.40 | brodiem | grandpapadot: was that supposed to be for me? |
| 20:22.04 | nephfl | any ideas why make is confused? |
| 20:23.20 | Qwell | nephfl: did you rerun configure after installing it? |
| 20:23.31 | nephfl | yeah |
| 20:23.48 | Qwell | make distclean, and do it again |
| 20:23.59 | Qwell | I think I know what the problem is |
| 20:24.29 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:24.36 | fujin_ | I believe you have to re-run ./configure |
| 20:24.36 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 20:24.41 | fujin_ | oh |
| 20:24.42 | fujin_ | nm |
| 20:24.45 | grandpapadot | make clean |
| 20:24.50 | Qwell | no, make distclean |
| 20:24.58 | grandpapadot | What he said |
| 20:25.00 | nephfl | looks like that was it...i just didnt clean it enough..lol |
| 20:25.33 | *** join/#asterisk icewaterman (n=immagine@i577BBFED.versanet.de) |
| 20:39.38 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:45.10 | *** join/#asterisk s0lid (n=_freq@210.213.240.151) |
| 20:45.38 | nephfl | simple question, do i need to load the zaptel drivers if i have no zap device on a system? |
| 20:52.11 | nephfl | hello? |
| 20:52.17 | *** join/#asterisk mirco (n=mirco@tmo-106-1.customers.d1-online.com) |
| 20:53.21 | _x86_ | nephfl: no |
| 20:53.33 | _x86_ | nephfl: but you might want ztdummy for timing |
| 20:53.39 | _x86_ | (part of the zaptel drivers) |
| 20:53.42 | grandpapadot | I think if you're using meetme you'll need ztdummy for timing purposes |
| 20:54.32 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 20:55.17 | nephfl | i guess it cant hurt to load the zaptel drivers |
| 20:55.27 | denon | for meetme or moh |
| 20:57.15 | *** join/#asterisk Sebb (n=sebastia@einstein.f0o.de) |
| 20:57.53 | Sebb | hi, i have a question about func_odbc: is it possible to set more than one variable at once from one table, without doing more than one select? |
| 21:01.58 | *** join/#asterisk datachomper (n=russ@75.146.194.61) |
| 21:02.22 | datachomper | What ports do I need to open in my firewall for iax2 to work? |
| 21:02.38 | datachomper | 4569 udp ? |
| 21:03.32 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 21:05.41 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 21:07.43 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 21:08.52 | _Sam-- | hey im having a problem using 'safe_asterisk' to start my asterisk. when i use asterisk -vvvvv & it starts and runs fine in the background. if i use safe_asterisk, i cant make it start -- it keep dying and restarting immediately. |
| 21:08.54 | nephfl | lol..the simplest things today just arent workin' for me... the tar is tellin me the zaptel drivers arent in gzip format |
| 21:10.21 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
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| 21:26.59 | hmmhesays | hmm what mime type should i use to attach a wav file to an email |
| 21:32.44 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:36.37 | nephfl | if i am not using zap is ztdummy the only driver i should load from the zaptel drivers? |
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| 21:43.45 | *** join/#asterisk minaguib (n=mina@modemcable137.196-82-70.mc.videotron.ca) |
| 21:44.26 | minaguib | Is there some sort of concensus on what's a good hardware home phone to get ? |
| 21:44.39 | Qwell | ~phones |
| 21:44.39 | jbot | rumour has it, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. ... |
| 21:44.47 | minaguib | The voip-info hardware wiki is overloaded to the point of uselessness |
| 21:45.01 | minaguib | well hot damn. Thank you Qwell. |
| 21:45.20 | Qwell | ~gs |
| 21:45.20 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:45.54 | _Sam-- | my phone system is no less reliable or robust as a result of my use of grandstream phones. im not getting into this debate, but will insist they work fine for US. |
| 21:46.58 | _Sam-- | for a time, and on the early firmwares, i do acknowledge they were a challenge to make work reliably. |
| 21:47.34 | _Sam-- | anywho |
| 21:47.34 | Qwell | firmware isn't going to make them feel less junky |
| 21:47.35 | JT | or sound less crappy |
| 21:47.35 | _Sam-- | if the person answering it is an 8 dollar an hour employee, then they are getting what they deserve. |
| 21:47.35 | Qwell | my son has a fisher price phone that feels more real than a gs |
| 21:47.38 | _Sam-- | the firmware does make it sound less crappy, however. |
| 21:48.00 | JT | it's false economy to waste employee time just because they're low paid |
| 21:48.12 | _Sam-- | it doesnt waste any time by having them use them, but thank you for your concern. |
| 21:48.25 | JT | actually it does, it's just hard to measure |
| 21:48.29 | _Sam-- | the phone rings, they answer it. |
| 21:48.32 | _Sam-- | plain and simple. |
| 21:48.40 | _Sam-- | im not saying im Cisco or ATT here. |
| 21:48.55 | JT | if only everything in life could be simplified that much |
| 21:49.59 | minaguib | ga. polycom's all conference room or microsoft-usb stuff. on to next :) |
| 21:50.06 | _Sam-- | i feel somewhat of a contradiction since i live by the adage ' good things arent cheap, and cheap things arent good'...but i can tell you these grangstreams have been good, and cheap, for ME. |
| 21:50.25 | Qwell | _Sam--: what business are you in? |
| 21:50.49 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 21:50.52 | _Sam-- | i own one of the 500 largest ecommerce businesses in the world (at least, as of last year) |
| 21:51.01 | fujin_ | O WOW |
| 21:51.08 | fujin_ | surely you could afford better than crapstream |
| 21:51.09 | fujin_ | grandsuck |
| 21:51.15 | _Sam-- | i couldnt at the time. |
| 21:51.21 | Qwell | _Sam--: so, these people are doing...customer service? |
| 21:51.21 | _Sam-- | luckily, business has grown :) |
| 21:51.29 | _Sam-- | customer service, sales. |
| 21:51.34 | fujin_ | ugh, I'd hate to be a csr/sales on a gs |
| 21:51.42 | _Sam-- | id hate to be a CSR on anyything. |
| 21:51.57 | _Sam-- | the phones, actually, are the most reliable part of our * :) |
| 21:52.17 | _Sam-- | ever since 1.4 anyway. |
| 21:53.18 | _Sam-- | do you own polycom or something? what is your interest in denegrading a brand that has shown its willing to work with its customers to improve the products. |
| 21:53.33 | Qwell | _Sam--: quality is NOT a software issue. |
| 21:53.39 | _Sam-- | its just ludicrous that you take someone's selection of phones so personally. |
| 21:53.45 | _Sam-- | my shit works fine for ME. thats all im saying. |
| 21:54.01 | Qwell | sure, I'll be it's great for your apparent bottom-line |
| 21:54.03 | Qwell | bet* |
| 21:54.04 | _Sam-- | when my 1.4 every version of * crashes in 3 days... |
| 21:54.11 | _Sam-- | its easy to have phones more reliable than * |
| 21:54.50 | *** join/#asterisk techie (n=techie@adsl-76-214-29-238.dsl.lsan03.sbcglobal.net) |
| 21:55.11 | file | and you have filed a bug report for this with all available information and backtraces? |
| 21:55.12 | Qwell | outsourcing support is also great for a bottom-line. |
| 21:55.29 | _Sam-- | Qwell : if i wanted your business advice, id ask? |
| 21:55.35 | _Sam-- | but , just for your info...we dont outsource. |
| 21:55.48 | _Sam-- | file: ive tried to have tk-fender do it...but i dont think he has. |
| 21:56.07 | file | okay, we can not fix what we do not know about |
| 21:56.13 | EnigmaCurry | So where's a good place to find reviews for hard phones? |
| 21:56.27 | _x86_ | polycom.com ;) |
| 21:56.29 | Qwell | EnigmaCurry: in here, usually |
| 21:56.31 | grandpapadot | EnigmaCurry: here, buy Polycom. |
| 21:56.39 | _x86_ | polycom++ |
| 21:56.58 | _x86_ | i like them so much, i've got (3) IP601's at home just to play with |
| 21:57.10 | grandpapadot | The new 330 is a great phone for the $$. |
| 21:57.36 | _Sam-- | file: when my asterisk segfaults, and dies and leaves nothing, what is there to gather so you can trace it? |
| 21:57.52 | *** part/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 21:57.54 | EnigmaCurry | I'll check em out. So far I have a sipura connected to cordless.. but would be nice to have a hard phone at the office |
| 21:57.58 | file | _Sam--: if configured properly it will leave a core dump |
| 21:58.09 | file | properly depends on how you run it |
| 21:58.19 | _Sam-- | it left no cores. it was run as safe_asterisk previously. |
| 21:58.28 | file | and is it running as root? |
| 21:58.35 | _Sam-- | it was running as asterisk. |
| 21:58.39 | grandpapadot | running as root is bad, m-key |
| 21:58.43 | grandpapadot | m-kay |
| 21:58.48 | _Sam-- | now its running as root, because safe_asterisk exits with exit code 127 now. |
| 21:59.06 | _Sam-- | and im just running asterisk -vvvvvcg or somesuch |
| 21:59.33 | file | then a core dump should exist in /tmp or the directory where you started it |
| 22:00.02 | _Sam-- | ive tried find and updatedb -- there are no cores to be found. |
| 22:01.08 | tzanger | _Sam--: if your shell has core generation turned off you won't see 'em |
| 22:01.11 | tzanger | ulimit -c |
| 22:01.17 | De_Mon | rawr my ? key doesn't work on the console |
| 22:03.40 | _Sam-- | tzanger : ulimit -c returns - |
| 22:03.43 | _Sam-- | er returns 0 |
| 22:03.55 | tzanger | yeah see cores are turned off |
| 22:03.59 | tzanger | ulimit -c unlimited turns them on |
| 22:04.06 | tzanger | at least for that shell session |
| 22:04.23 | tzanger | depending on your distro, you may have to turn it on in /etc/profile or somewhere similar |
| 22:06.17 | mosty | i have an error when a sip phone redirects a call (though that sip phone can call the number just fine), the call path is sip -> iax -> iax -> pri, i think the problem is something to do with the fact that the sip extension appears in the pri trace, what could be wrong? http://pastebin.ca/806018 |
| 22:06.33 | _Sam-- | up until this 1.4.15 which i upgraded to today, i had been able to start my asterisk with 'safe_asterisk'....now i when i try that, i get "asterisk exited with code 127" and it tries to restart repeatedly. |
| 22:06.38 | _Sam-- | asterisk -vvvvvgc works fine. |
| 22:06.49 | grandpapadot | _Sam--: permissions |
| 22:07.08 | _Sam-- | i dont see it , though. everything is owned by asterisk, which is what safe_asterisk has always run as. |
| 22:07.20 | _Sam-- | nothing in any logs. |
| 22:07.22 | grandpapadot | _Sam--: Did you check in /var/run and your asterisk.ctl file? |
| 22:08.00 | _Sam-- | it exists, and is owned by asterisk |
| 22:08.14 | minaguib | Linksys SPA901-NA seems like a nice simple home phone. Any warnings/recommendations ? |
| 22:08.32 | grandpapadot | minaguib: How much $$? |
| 22:08.47 | minaguib | grandpapadot: 79.99 CAD |
| 22:08.58 | grandpapadot | _Sam--: Reset all your asterisk permissions, then start with asterisk -vvvgc -U asterisk -G asterisk (or whatever user/group you're using). |
| 22:09.41 | Yourname`` | Do all the calls get closed or something when you do a 'reload'? Or am I too tired? |
| 22:10.07 | grandpapadot | Yourname: No, channels won't be reset. |
| 22:12.38 | shido6 | the calls do not drop when you issue a reload |
| 22:12.46 | shido6 | lucky us |
| 22:13.29 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:13.45 | r0d3nt | how is #asterisk ? my fine african american brethrens... |
| 22:14.14 | De_Mon | oh cool, I found a crash bug |
| 22:14.25 | *** join/#asterisk Nix (n=Nix@opensuse/developer/nix) |
| 22:14.53 | _Sam-- | grandpapadot : thanks for your help. close but not there yet. asterisk -vvvvvcg -U asterisk works fine |
| 22:14.58 | _Sam-- | safe_asterisk still no go |
| 22:15.31 | *** join/#asterisk bmg505 (n=leon@196.209.178.145) |
| 22:16.03 | JT | _Sam--: what distro? |
| 22:16.10 | _Sam-- | debian etch |
| 22:16.29 | JT | i know safe_asterisk has issues with ubuntu |
| 22:16.40 | *** part/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 22:17.30 | _Sam-- | i wouldnt worry about running it the way i am, except for my stuff crashes, and i really need the safe_asterisk restarts. |
| 22:18.58 | Yourname`` | grandpapadot: Weird, because somehow about 112 calls just went bloop as soon as I did a reload, heh |
| 22:20.35 | De_Mon | Yourname`` whoops |
| 22:21.16 | grandpapadot | Yourname: that is weird and not normal behavior unless something in your reload caused channels to be reset, what did you change? |
| 22:27.17 | grandpapadot | Yourname: ?? |
| 22:30.57 | *** join/#asterisk centrex (i=centrex@nat/digium/x-141e1f8550a9041a) |
| 22:32.13 | De_Mon | M11468 |
| 22:32.26 | De_Mon | aww what the crap |
| 22:32.31 | De_Mon | jbot? |
| 22:32.46 | De_Mon | ~jbot |
| 22:32.47 | jbot | jbot is, like, a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 22:32.55 | De_Mon | ~M11468 |
| 22:34.12 | Qwell | De_Mon: don't do that then |
| 22:34.13 | _Sam-- | ~jbot seen [TK]D-Fender |
| 22:34.17 | jbot | [tk]d-fender <n=joe@MTRLPQ02-1177745839.sdsl.bell.ca> was last seen on IRC in channel #asterisk, 3h 20m 57s ago, saying: 'nny: and your sip.cfg'. |
| 22:34.52 | grandpapadot | Shout out if you were ever a SysOP |
| 22:37.01 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 22:37.02 | _Sam-- | im the sysop of a broken * installation. |
| 22:37.10 | fujin_ | owned |
| 22:37.16 | fujin_ | nuke it and reinstall it in the late night hour |
| 22:37.38 | _Sam-- | yeah you may be right. i just upgraded it from sarge to etch this morning. |
| 22:37.49 | grandpapadot | in-place upgrade? |
| 22:38.21 | _Sam-- | yeah, worked fine though. aptitude uprade aptitude install new-kernel aptitude dist-upgrade |
| 22:38.58 | hesco | I need to log the outgoing phone number in CDR, can someone point me to where that might be documented please? |
| 22:39.08 | grandpapadot | hesco: userfield |
| 22:39.30 | De_Mon | Qwell yeah I almost set the severity to minor, but eventually talked myself out of it |
| 22:39.31 | JT | why aptitude? |
| 22:39.42 | De_Mon | because aptitude is better |
| 22:39.52 | hesco | I saw those two functions in the appendix. Thanks. But what do I call the phone number itself in order to put it into the userfield? |
| 22:39.59 | JT | better than what? |
| 22:40.02 | De_Mon | apt-get |
| 22:40.06 | JT | how so? |
| 22:40.07 | _Sam-- | JT: because the debian guide recommended it. |
| 22:40.07 | De_Mon | dselect... |
| 22:40.11 | _Sam-- | it breaks less things. |
| 22:40.19 | De_Mon | JT it is smarter |
| 22:40.22 | JT | everything is better than dselect |
| 22:40.39 | fujin_ | Isn't update-manager the debian way to distupgrade? |
| 22:40.41 | De_Mon | JT its like playing chess with a 8bit computer or a 64bit computer |
| 22:40.44 | De_Mon | its just better! |
| 22:40.53 | fujin_ | aptitude is crapo |
| 22:41.11 | _Sam-- | my debian etch has no upgr4ade-manager |
| 22:41.16 | _Sam-- | er update-manager |
| 22:41.34 | hesco | I had apt-get break a working hylafax server once. It was down for nearly a week before the Debian maintainer for hylafax responded to my bug reports and walked me through recovering the installation. |
| 22:41.44 | hesco | a frustrating week, to be sure. |
| 22:41.49 | JT | De_Mon: err sure, i'll keep that in mind for reasons "better" |
| 22:42.03 | JT | hesco: i don't think aptitude would've fixed that |
| 22:42.06 | _Sam-- | hesco: make a backup. |
| 22:42.17 | De_Mon | :( |
| 22:42.17 | _Sam-- | (my hylafax was upgraded fine!) P) |
| 22:42.19 | hesco | Some guy named Giuseppe from Italy. He was great, very helpful. |
| 22:42.36 | _Sam-- | i heart hyla |
| 22:42.42 | hesco | this was nearly three years ago. |
| 22:44.04 | mosty | JT, i have sip extension/callerid "leaking" into a pri call when my sip phone is redirecting calls, which causes the pri to reject my call. do you know what else i might need to set besides callerid? |
| 22:44.24 | *** join/#asterisk Greek-Boy (n=Greek-Bo@41.221.58.2) |
| 22:44.43 | hesco | so, grandpapadot: where might I find that userfield documented? |
| 22:45.09 | stimpie | mosty, not sure if its relevant but a 302 redirect goes to context 'default' |
| 22:45.11 | hesco | I've already added two lines to my call plan to account for its use, |
| 22:45.33 | hesco | but am not sure how to get my outgoing phone number into that user field. |
| 22:46.10 | mosty | stimpie, the redirected call traverses the dialplan as i expect, but my sip extension appears in the pri trace, even though it's not the callerid set for the call |
| 22:46.46 | JT | perhaps it's the callerid name? |
| 22:47.39 | mosty | JT, callerid name and number are both set to a regular telephone number that is valid for the pri line |
| 22:48.02 | hesco | I sure wish I could get the text from my CID to display on the receiving phone, but only the phone number seems to get through. Is there some secret syntax that only insiders seem to know? |
| 22:48.11 | mosty | JT, the sip phone can call the number ok, it's only when it's set to redirect that the redirected call fails |
| 22:49.15 | _x86_ | hesco: you passing the call out over a PRI? |
| 22:49.15 | *** join/#asterisk wcuestas (n=wcuestas@201.230.213.82) |
| 22:49.29 | wcuestas | hi |
| 22:49.49 | wcuestas | anyone knows a company from chile or argentina offering asterisk services ? |
| 22:50.03 | *** part/#asterisk jtexter3 (n=jamest@ip67-90-136-204.z136-90-67.customer.algx.net) |
| 22:50.35 | stimpie | mosty, you should look at the redirect as if it was a call originated at the sip phone |
| 22:51.11 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:51.47 | mosty | stimpie, that's what i am doing. the sip extension "leaks" into the pri leg when it's redirected, but not when it's a simple call |
| 22:52.04 | *** part/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 22:52.20 | *** join/#asterisk rvhi (n=chatzill@udp182407uds.hawaiiantel.net) |
| 22:52.45 | mosty | the path out to the pri is this: sip -> iax host A -> iax host B -> pri, so i don't see how the iax2 host B should know anything about the sip extension |
| 22:53.01 | rvhi | anyone uses aastra 480i-ct phones? not sure why, it only register once and stop renew registration. any special settings? |
| 22:54.39 | stimpie | mosty, very strange indeed |
| 22:55.03 | *** part/#asterisk wcuestas (n=wcuestas@201.230.213.82) |
| 22:56.05 | mosty | is there a way i can get a list of all variables set on a channel? |
| 22:57.07 | stimpie | mosty, core show channel |
| 22:57.19 | centrex | mosty, What do you mean by "leaking"? So the call going out the pri retains the original sip's callerid? Can you pastebin your /etc/asterisk/zapata.conf ? |
| 22:58.24 | Qwell | De_Mon: ...upgrade |
| 22:58.34 | Qwell | I'm pretty sure that's been fixed |
| 22:59.28 | Qwell | yeah, it has |
| 23:00.10 | mvanbaak | I should upgrade as well |
| 23:00.17 | mvanbaak | still running 1.0.9 in production |
| 23:00.25 | centrex | nice |
| 23:00.35 | _Sam-- | mvanbaak : i had a production box from that era, and have not had much luck with any 1.4's being stable. |
| 23:00.57 | mvanbaak | _Sam--: thinking about latest 1.2 here |
| 23:01.04 | mosty | centrex, the pri trace is here, note the 340340 on line 62 (that's the sip ext): http://pastebin.ca/806018 and the zapata.conf is here: http://pastebin.ca/806083 |
| 23:01.08 | Qwell | mvanbaak: trunk |
| 23:01.14 | mvanbaak | lol Qwell |
| 23:01.16 | Qwell | you can handle it |
| 23:01.21 | mvanbaak | Qwell: at home I run trunk indeed |
| 23:01.44 | mvanbaak | because I'm too lazy to maintain the MWI backport for chan_skinny |
| 23:01.49 | Qwell | :D |
| 23:02.00 | mosty | centrex, a regular call from the sip phone goes through fine, however if the phone is set to redirect to a number and i call the sip phone, the sip ext of the redirecting phone appears in the pri trace and the call fails |
| 23:02.16 | mvanbaak | trunk on openbsd 4.2 |
| 23:02.20 | mvanbaak | works like a charm |
| 23:07.00 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:07.50 | mosty | centrex, is there any other info i can give you that might help? |
| 23:09.59 | centrex | mosty, not that I can think of, it looks pretty weird. |
| 23:10.56 | mosty | centrex, ok then, i will try the userlist |
| 23:11.02 | mosty | mailing list, that is |
| 23:11.25 | centrex | mosty, Try just setting callerid in dialplan before it goes out the trunk. |
| 23:11.35 | centrex | And that should take care of the issue for you. |
| 23:11.58 | centrex | As far as why it's forwarding the sip extension, I have no clue. |
| 23:12.16 | mosty | centrex, i do NoOp(${CALLERID(num) - ${CALLERID(name)}) right before the call, neither is set to 340340 |
| 23:12.25 | centrex | What are they set to? |
| 23:13.17 | mosty | centrex, they're set to a valid number for that pri line, see line 11 in the pri trace |
| 23:13.39 | centrex | mosty, http://lists.digium.com/pipermail/asterisk-dev/2006-October/023642.html seems to be related. |
| 23:16.38 | centrex | mosty, are you sure this is an issue with the pri line? what exactly do you hear on the call? The pri debug says that the number passed screening and went out. |
| 23:17.38 | centrex | one way audio perhaps? |
| 23:17.42 | mosty | centrex, i hear congestion tones |
| 23:19.22 | centrex | mosty, from looking at this, it almost seems like the problem is with sip and we're sending a disconnect to the pri side. |
| 23:20.03 | centrex | Look at lines 184-190. |
| 23:20.17 | *** join/#asterisk RipeR-81 (n=ircap8@190.53.33.10) |
| 23:20.32 | mosty | also, when trying a few things last night, i was somehow getting a recorded message about having too few digits (which must have been coming from the pri provider) |
| 23:21.09 | JT | how many were you sending? |
| 23:21.24 | hesco | _x86_: my outgoing calls are riding an IAX2 server, I'm dropping .call files into the spool, which use an IAX account. At least that's how I'm doing it on the workbench. Once in production, I'm not sure how that will play out. I'm looking for some unlimited voip options. |
| 23:21.56 | centrex | mosty, From looking at this, I would start checking nat issues with the phone that is redirecting, seeing if there's anything weird like canreinvite=yes what nat= is set to, etc... |
| 23:22.02 | hesco | sorry, distracted by a phone call for a bit. You asked for clarification about a question I had related to logging outgoing phone numbers. |
| 23:22.06 | centrex | in the configuration, that is. |
| 23:23.15 | mosty | centrex, hmm, i will take a look, but i think that part of the trace is after i had hung up, having heard the congestion tones |
| 23:23.54 | centrex | mosty, Well if you look there, you can see that right after the call goes out, asterisk tries to bridge zaptel and the iax channel, complains of an error with the iax channel, and hangs up both lines. |
| 23:25.12 | centrex | The part where it says the extension that forwarded the line, it only says that it was forwarded unconditionally from them. |
| 23:25.24 | mosty | centrex, what i don't get is why 340340 would appear at all in the pri trace, the sip peer is not registered to this iax box |
| 23:25.45 | centrex | It just says that is who transferred the call. |
| 23:26.14 | centrex | I'm not a sip expert, but it would like that information gets recorded in the sip headers, but it's not stopping the call from going out. You can literally see that the pri switch passed the call for screening. |
| 23:26.23 | centrex | s/like/look like |
| 23:26.39 | mosty | centrex, the iax box with the pri doesn't use sip at all |
| 23:27.18 | centrex | mosty, It's recording it in the call, but that doesn't look like it's causing your problem. |
| 23:27.20 | mosty | the path is sip -> iax host A -> iax host B -> pri, and it's on host B that i see this |
| 23:28.00 | centrex | mosty, It's not like all original call information never gets passed through... |
| 23:28.31 | mosty | centrex, it never appears when the sip client makes a regular call- only when the sip client is redirecting |
| 23:28.46 | centrex | mosty, Right. It's saying that it was forwarded from that number. |
| 23:28.54 | centrex | It wouldn't say that if it wasn't ever forwarded. |
| 23:29.04 | mosty | hmm ok |
| 23:29.14 | *** part/#asterisk riddlebox (n=JamesMid@68.143.86.98.nw.nuvox.net) |
| 23:30.34 | *** join/#asterisk bmg505 (n=leon@196.209.178.145) |
| 23:31.29 | centrex | mosty, Try this, as a test. on the box with the iax channel, put an iax softphone on it, and try again, see if you get the same problem trying to call that phone. |
| 23:31.43 | centrex | I mean on the box with the zap channel, |
| 23:31.59 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:32.23 | JT | centrex: pri switch passed the call for screening? what do you mean? |
| 23:32.27 | mosty | ok, i can try that |
| 23:33.14 | hmmhesays | hmm is it possible to only record so many seconds off the beginning of a call? |
| 23:33.20 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 23:33.37 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-187.pskn.east.verizon.net) |
| 23:33.41 | centrex | JT: based on the pri dialplan, the switches screen the calls for where they should go... I don't know the specifics, but you have to set it to unknown a lot of times, because if not, it might try to send a local call as an international, append digits, etc... but you can tell whether or not they blocked it in the pri debug. |
| 23:34.10 | centrex | If it's wrong it won't say number permitted like it does in his log. |
| 23:36.10 | centrex | JT, In his logs, it's lines 45-64, and it says the presentation was permitted. |
| 23:36.16 | JT | ... |
| 23:36.54 | JT | that has NOTHING to do with whether or not a call is permissable |
| 23:37.05 | JT | that means far end presentation of CID is permitted |
| 23:37.17 | JT | as that is what asterisk has indicated |
| 23:37.33 | centrex | JT, no, it means the outgoing number matches its dialplan, I'm pretty sure. |
| 23:37.36 | JT | wrong |
| 23:37.44 | JT | it means that callerid is not restricted |
| 23:38.02 | JT | read Q.931 etc if you want proof |
| 23:38.31 | outtolunc | where is the pri debug? |
| 23:38.45 | mosty | http://pastebin.ca/806018 |
| 23:40.41 | outtolunc | cute |
| 23:40.50 | outtolunc | any way to set the dialplan type |
| 23:41.15 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 23:41.21 | outtolunc | (its set unknown currectly) so when the redirect happens.. and the redir number is trimmed...it doesn't know what dialplan to look in |
| 23:41.55 | mosty | i can set the dialplan type to something else, but not until after business hours |
| 23:42.03 | RipeR-81 | hey guys im looking for a voip termination service... do you reccomend any bussiness? |
| 23:42.55 | outtolunc | that would be my test as the number you dialed was 0390118400, that was unconditionally redir'd as 340340 and the pri is set as TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) |
| 23:42.58 | centrex | JT, http://lists.digium.com/pipermail/asterisk-users/2005-May/102837.html |
| 23:44.45 | mosty | outtolunc, that's not the number that was dialled, in that example i dialled 131166 |
| 23:44.55 | mosty | well, i redirected to 131166 |
| 23:45.46 | outtolunc | sorry that was the 'callerid' of that dialed number |
| 23:45.59 | mosty | correct |
| 23:49.35 | JT | centrex: umm i know what NPI and TON is |
| 23:49.46 | JT | centrex: the discussion was about number presentation |
| 23:53.53 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
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