| 00:08.20 | mchou | LemensTS: well, if you write anything that's really usable make sure you let me know too. I'm involved in the same effort (appt. reminders). :) I just looked at the lay of the land to understand what works and what doesnt. |
| 00:10.10 | mchou | LemensTS: If you are in the US I think you also have to make sure the client can opt out of pre-recorded calls. i.e. "press 1 now to not get pre-recorded reminders." |
| 00:10.51 | mchou | LemensTS: tere are some corner cases of the law that you need to make sure you're compliant with. |
| 00:17.44 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 00:18.18 | irieKen | Anyone know if there is a security setting in Asterisk that prevents pinging from the asterisk box to outside of the network? |
| 00:18.53 | HumanCell | irieKen: probably not in Asterisk, but maybe in the underlying distro ... |
| 00:18.53 | irieKen | ie. I am trying to ping from the asterisk box to test connectivity, but nothing is coming back. |
| 00:19.36 | HumanCell | irieKen: have you tried "route" to ensure that the routing is set-up properly? |
| 00:19.57 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 00:20.00 | pcrane | irieKen: iptables firewall setting? |
| 00:20.03 | irieKen | HumanCell: I haven't, but I will try that. |
| 00:20.05 | HumanCell | then 'iptables -L' can also tell you what rules are set ... |
| 00:20.13 | irieKen | pcrane: Ok. |
| 00:20.19 | pcrane | or, some other firewall between your asterisk box and the network |
| 00:20.27 | pcrane | or the other machine doesn't respond to pings |
| 00:20.55 | HumanCell | pcrane: good point ... making sure to try and ping *several* machines ... |
| 00:21.28 | pcrane | traceroute/tracepath should help in identifying where the pings stop |
| 00:23.31 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:25.06 | irieKen | Hmm. |
| 00:25.13 | irieKen | iptables -L shows no rules set. |
| 00:25.34 | irieKen | I can ping in-network, but I can't ping anything outside (though, DNS resolution does appear to be working). |
| 00:30.45 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:34.47 | *** join/#asterisk ManxPower (n=manxpowe@adsl-156-119-179.msy.bellsouth.net) |
| 00:34.59 | irieKen | Ah, just need to add a default route entry. Pinging issue resolved:) |
| 00:35.16 | pcrane | ;) |
| 00:35.26 | HumanCell | irieKen: yep ... |
| 00:39.34 | *** join/#asterisk micander (n=Michael_@68.121.201.170) |
| 00:40.33 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 00:42.01 | *** join/#asterisk shidan (n=chatzill@CPE000d483807fd-CM001371871af0.cpe.net.cable.rogers.com) |
| 00:42.28 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 00:42.58 | bad_duck | Hi, how do I know wich version of asterisk I installed ?? I installed the trunk version from svn |
| 00:43.08 | Qwell | you installed trunk |
| 00:43.17 | bad_duck | :) |
| 00:43.26 | Qwell | I fail to understand the question |
| 00:43.26 | bad_duck | and is more like 1.4 or 1.6 ? |
| 00:43.30 | Qwell | it's trunk |
| 00:43.40 | pcrane | # asterisk -r |
| 00:43.40 | pcrane | Asterisk SVN-branch-1.4-r118558M, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 00:43.57 | pcrane | the first line tells you 1.4 or 1.6 |
| 00:44.06 | Qwell | trunk is neither |
| 00:44.10 | pcrane | bad_duck: what svn checkout command did you use? |
| 00:44.22 | bad_duck | Qwell: I would like now to install chan-mobile, but there are two different way depends on your asterisk version |
| 00:44.43 | Qwell | You shouldn't be running trunk if you don't know the answer to that question.. |
| 00:44.55 | bad_duck | svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 00:45.22 | bad_duck | root@obelix:~# asterisk -r |
| 00:45.22 | bad_duck | Asterisk SVN-trunk-r140246, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 00:45.28 | pcrane | ok |
| 00:45.31 | pcrane | so it's trunk ;) |
| 00:46.38 | pcrane | if you want a particular version use: |
| 00:46.40 | jblack | lol. The manpage to wipe is a better melodrama than what most high school cliques can produce. |
| 00:46.42 | Qwell | bad_duck: install 1.4 or 1.6.0, you don't need to be running trunk |
| 00:46.45 | pcrane | svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 00:46.58 | pcrane | bad_duck: http://www.asterisk.org/developers/get-source |
| 00:47.22 | bad_duck | ok, and for libpri and zaptel ? (I also install trunk versions) |
| 00:47.38 | Qwell | there is no such thing as zaptel trunk |
| 00:47.45 | bad_duck | yes |
| 00:47.50 | bad_duck | libpri on trunk |
| 00:47.53 | bad_duck | zaptel 1.4 |
| 00:48.16 | Qwell | why did you install zaptel when using trunk? |
| 00:48.31 | Qwell | it's not going to work |
| 00:48.36 | bad_duck | yes I think |
| 00:48.53 | bad_duck | hmm.. ok |
| 00:49.01 | Qwell | don't use trunk. |
| 00:49.49 | bad_duck | so.. asterisk1.4, zaptel1.4 and libpri 1.4 is right ? |
| 00:49.55 | pcrane | yep |
| 00:49.58 | Qwell | if you want 1.4 |
| 00:50.27 | bad_duck | can I run juste a "make clean" to uninstall ? |
| 00:50.27 | HumanCell | Hmmm ... I'm having trouble getting the CallerID to pass through to VoicePulse ... |
| 00:50.29 | tvirus | Has anyone been able to setup HUDLite on a non-trixbox machine? I'm just getting it set up and trying to call from softphone to softphone inside of the office. Asterisk says 'cannot create channel SIP/user'. HUDLite tosses out some random error. |
| 00:50.37 | jblack | humancell: How are you testing? |
| 00:50.45 | pcrane | bad_duck: make distclean |
| 00:50.55 | bad_duck | pcrane: ok, thanks I'll try |
| 00:51.01 | Qwell | distclean doesn't uninstall |
| 00:51.01 | pcrane | that just removes all the compiled stuff in the directory |
| 00:51.06 | pcrane | it doesn't actually uninstall |
| 00:51.19 | pcrane | should read a bit before answering ;) |
| 00:51.25 | HumanCell | I'm calling into my box, and then am using this command: |
| 00:51.40 | HumanCell | exten=s,1,Set(CALLERID(all)=Distinguished <5555550269>) |
| 00:51.40 | HumanCell | exten=s,n,Dial(SIP/5000&IAX2/5000&SIP/15555557456@trunk_3,20) |
| 00:51.44 | Qwell | tvirus: #hudlite |
| 00:51.55 | tvirus | Lies :( |
| 00:52.04 | Qwell | You'll also need to know how to run an IRCd, and a bunch of other junk |
| 00:52.10 | tvirus | It's up and running |
| 00:52.18 | tvirus | I can connect to it, see other users, just can't call them. |
| 00:52.24 | Qwell | tvirus: #hudlite |
| 00:52.24 | drmessano | Hudlite sux |
| 00:52.43 | jblack | HumanCell: What is 155555557456 ? |
| 00:52.48 | HumanCell | jblack: the SIP phone at x5000 gets the callerID ... but VoicePulse at trunk_3 isn't ... |
| 00:52.59 | tvirus | drmessano: Probably, but there's no other decent call manager that shows what people are talking. |
| 00:53.07 | tvirus | At least none that are free/cheap |
| 00:53.17 | HumanCell | jblack: trunk_3 is voicepulse ... I'm having the call go out to an external number ... |
| 00:53.30 | Qwell | does voicepulse let you set callerid? |
| 00:53.38 | jblack | I didn't ask what trunk_3 is. Who owns 7456 ? |
| 00:53.55 | HumanCell | Yeah ... it does work when I call from my Soft phone out ... the callerID is set-properly ... |
| 00:54.26 | jblack | is 7456 a cell phone? A landline? ipkall? |
| 00:54.46 | HumanCell | jblack: it's a desk phone POTS line at another office ... |
| 00:55.16 | jblack | So you're relying on another person to tell you what the other phone says. YOu checked that they can see callerid? |
| 00:55.40 | jblack | by calling them with a different phone. |
| 00:55.50 | HumanCell | jblack: yes ... but I'm also going to a cell phone ... |
| 00:56.07 | *** part/#asterisk Gershwin (n=fake@63.250.233.162) |
| 00:56.08 | HumanCell | jblack: I've changed to to a variety of phones for testing purposes ... |
| 00:56.11 | jblack | And in both cases, callerid is broken through voicepulse. |
| 00:56.16 | HumanCell | Yeah ... |
| 00:56.27 | jblack | Ok. voicepulse breaks callerid. |
| 00:56.37 | HumanCell | jblack: BUT ... if I call out from my XLite soft phone ... and set the callerID ... THAT works ... |
| 00:56.45 | HumanCell | So it's NOT VoicePulse ... |
| 00:57.08 | HumanCell | It's only when I have an inbound call, that I then route to that particular "ring group" ... |
| 00:57.12 | bad_duck | pcrane: do I need to do something to uninstall libpri trunk |
| 00:57.26 | jblack | So do a sip debug, and verify callerid is being set in the call to voicepulse. |
| 00:57.45 | jblack | If it is, either your callerid format is broken, or voicepulse is ignoring it. |
| 00:58.09 | HumanCell | jblack: Ok ... so my newbie question ... how do I debug/trace the SIP call? |
| 00:58.10 | ManxPower | Would this be CallerID NUMBER or CallerID NAME you are having problems with? |
| 00:58.21 | jblack | sip set debug on, I believe. |
| 00:59.01 | HumanCell | ManxPower: I've tried setting both ... CALLERID(all) or just CALLERID(num) ... |
| 00:59.11 | jblack | HumanCell: I think he means which are you not receiving. name, num, or both |
| 00:59.20 | HumanCell | In both cases ... nothing goes through ... |
| 00:59.36 | HumanCell | Sorry ... both do not go through .,.. |
| 01:00.12 | HumanCell | I was wondering if I somehow didn't define something properly somewhere ... like in my trunk_3 definition in user.conf ... |
| 01:00.19 | ManxPower | HumanCell: You cannot send a name. Well you can, but the telco will ignore the name. With the number if you are in the USA/Canada the number would be 10 digits, no leading 1, no special chars, NO QUOTES |
| 01:00.25 | *** join/#asterisk Levonk (n=lk@76.237.15.224) |
| 01:00.45 | Qwell | What IS being sent? |
| 01:01.00 | Qwell | or received, as it were |
| 01:01.17 | HumanCell | ManxPower: So If I use: exten=s,1,Set(CALLERID(num)=5554440269) |
| 01:01.25 | ManxPower | that works. |
| 01:01.27 | HumanCell | ManxPower: that would be the correct syntax? |
| 01:01.30 | HumanCell | Ok ... |
| 01:01.37 | ManxPower | perhaps your carrier just doesn't allow you to send the number. |
| 01:01.53 | jblack | yeah, getting back to whre I was before I got interrupted.... |
| 01:02.05 | HumanCell | ManxPower: That is what is strange ... it does get sent when I call out from my XLite phone on my laptop ... |
| 01:02.12 | jblack | HumanCell: Some providers out there have instructions on unchecking a box to allow callerid, or blanking out a field with a number. |
| 01:02.59 | [TK]D-Fender | HumanCell: pastebin your SIP peer masing only your password. |
| 01:03.02 | [TK]D-Fender | masking |
| 01:03.04 | [TK]D-Fender | ~pb |
| 01:03.05 | 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 |
| 01:03.06 | [TK]D-Fender | ^^^^^^^^^^ |
| 01:03.11 | ManxPower | [TK]D-Fender: he's using users.conf |
| 01:03.14 | ManxPower | poor sod |
| 01:03.50 | HumanCell | Yes ... I be a poor sod using users.conf ... :-( |
| 01:04.03 | ManxPower | HumanCell: I can't help you with it. |
| 01:04.12 | ManxPower | most people here would have problems helping you with it. |
| 01:04.39 | anonymouz666 | if users.conf sucks, why is it there? |
| 01:04.43 | anonymouz666 | heh. |
| 01:04.51 | HumanCell | And in my extension definition in users.conf I have cid_number=5554446446 ... and when I dial out from my extension it works ... :-( |
| 01:05.55 | [TK]D-Fender | HumanCell: PASTEBIN |
| 01:06.29 | HumanCell | [TK]D-Fender: you still want the sip.conf? Or the users.conf section? |
| 01:06.32 | [TK]D-Fender | anonymouz666: Because of the GUI. |
| 01:06.41 | [TK]D-Fender | HumanCell: users.conf section |
| 01:06.50 | anonymouz666 | why after you set the iax2 peers in iax.conf just removing them from the file and 'iax2 reload' does not remove the peers from 'iax2 show peers', why? just if i do "restart now" |
| 01:07.24 | [TK]D-Fender | anonymouz666: File a bug |
| 01:08.28 | anonymouz666 | that's so basic that I can even imagine this as a bug... maybe I am missing something here... I think... |
| 01:08.43 | *** join/#asterisk bkw_ (n=bkw_@freeswitch/developer/bkw) |
| 01:09.02 | HumanCell | [TK]D-Fender: http://pastebin.com/d35006c9 |
| 01:09.45 | anonymouz666 | "can't" |
| 01:09.46 | [TK]D-Fender | HumanCell: "fromuser=GMd45obn87" <-- comment this out, reload, test |
| 01:09.48 | amessina | is away: Gone away for now. |
| 01:09.57 | *** join/#asterisk n3hxs (n=HAMming@host-64-65-218-80.har.choiceone.net) |
| 01:10.38 | HumanCell | testing ... |
| 01:11.58 | HumanCell | [TK]D-Fender: now it fails to make the outbound call ... |
| 01:15.32 | [TK]D-Fender | HumanCell: change the insecure to "no" |
| 01:16.34 | HumanCell | I had to add the fromuser back in ... |
| 01:16.47 | *** join/#asterisk MrNaz (n=naz@ppp118-208-173-20.lns10.mel4.internode.on.net) |
| 01:17.50 | [TK]D-Fender | don't |
| 01:17.54 | HumanCell | [TK]D-Fender: Hmm ... now that killed my inbound calling ... |
| 01:18.22 | [TK]D-Fender | HumanCell: change to "port,invite |
| 01:18.34 | [TK]D-Fender | HumanCell: then fromuser back in. |
| 01:18.40 | [TK]D-Fender | (after the nest test |
| 01:19.27 | HumanCell | [TK]D-Fender: So insecure=port,invite? |
| 01:19.32 | HumanCell | no fromuser |
| 01:19.46 | [TK]D-Fender | HumanCell>[TK]D-Fender: So insecure=port,invite? <- correct |
| 01:20.50 | HumanCell | [TK]D-Fender: still failed to make the outbound call ... |
| 01:20.58 | HumanCell | add fromuser back? |
| 01:21.08 | [TK]D-Fender | HumanCell: Ok thats as much as I'm going through for this |
| 01:21.34 | anonymouz666 | [TK]D-Fender needs to go to the gym! |
| 01:21.48 | [TK]D-Fender | AnonJust got back from there :0 |
| 01:21.59 | [TK]D-Fender | anonymouz666: Off to see "The Dark Knight" |
| 01:22.02 | ManxPower | hugs his PRIs |
| 01:22.22 | HumanCell | [TK]D-Fender: THanks! |
| 01:23.02 | *** part/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 01:24.05 | *** join/#asterisk koppernet (i=kopperne@119.111.140.226) |
| 01:28.40 | ManxPower | Damn Gustave, d heading for New Orleans. |
| 01:29.06 | _ShrikE | lets hope not |
| 01:31.56 | ManxPower | _ShrikE: most models show it impacting New Orleans, but it is 7 days away so things could easily change. On the other hand, they tend to have a pretty good track record about predicting the path (recent events not withstanding) |
| 01:32.15 | ManxPower | The whole city is abuzz with about about Gustave |
| 01:32.30 | _ShrikE | Indeed |
| 01:33.56 | *** part/#asterisk washburnello (n=washburn@h46.114.31.71.dynamic.ip.windstream.net) |
| 01:35.26 | jblack | Yeah. By New Orleans. Again. |
| 01:35.33 | *** join/#asterisk dwayne (i=dwayne@76.29.245.9) |
| 01:35.52 | jblack | I bet the people that can't afford gas, plane or train walk this time. |
| 01:37.54 | *** join/#asterisk vonkleist (n=gcontrer@189.155.100.170) |
| 01:38.08 | ManxPower | Some friends of my BF got out of New Orleans by mostly walking on the elevated I-10. |
| 01:39.04 | ManxPower | I hear rents are finally starting to fall in New Orleans |
| 01:40.04 | *** join/#asterisk haryv (n=hary@S0106001d7e52cc78.vs.shawcable.net) |
| 01:40.59 | _ShrikE | Jindal is already talking about pulling the trigger on contraflow saturday if it looks even close to a landfall here |
| 01:41.05 | _ShrikE | in new orleans that is |
| 01:41.36 | haryv | Has anyone seen this error before? I found this one on google that is the same as mine. The part of the error I am concerned wih is Maximum PBXstack exceeded http://pastebin.ca/raw/250681 |
| 01:42.08 | CoffeeIV | I'm helping someone who apparently has this bug: http://bugs.digium.com/view.php?id=8565 and worked around it by adding "insecure=port,invite" to their conf files. What security implications might that have ? |
| 01:42.11 | haryv | Shrike, are you in NO? |
| 01:43.36 | brodiem | Can someone confirm that the 3com-branded Polycom phones are identical, and use the same polycom firmware? |
| 01:43.50 | Qwell | brodiem: probably not |
| 01:44.11 | Qwell | I mean, the hardware probably is identical |
| 01:46.01 | _ShrikE | haryv: yes |
| 01:46.10 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:46.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8e380a2d4b38ce5f) |
| 01:46.22 | _ShrikE | well.. a subarb really |
| 01:47.33 | ManxPower | _ShrikE: Are you in New Orleans? |
| 01:47.43 | ManxPower | is in the Quarter at the moment |
| 01:47.50 | _ShrikE | Im in covington |
| 01:47.57 | brodiem | ah well best to just stay away I suppose |
| 01:47.59 | _ShrikE | I work downtown |
| 01:48.06 | ManxPower | brodiem: What in the WORLD would make you think that? |
| 01:48.34 | haryv | Manx, are you from there |
| 01:48.38 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 01:48.39 | brodiem | ManxPower: ?? |
| 01:48.41 | ManxPower | The only 3Com IP phones I know of only work with their own PBX, using some form of hacked sip protocol. |
| 01:49.01 | ManxPower | brodiem: Where did you hear that the 3Com phones are really rebranded Polycom phones? |
| 01:49.21 | ManxPower | haryv: No. I come to New Orleans to work on site at clients |
| 01:49.23 | *** join/#asterisk moy (n=moy@216.13.156.194) |
| 01:49.39 | ManxPower | _ShrikE: I work in Covington. |
| 01:50.47 | ManxPower | haryv: At first glance, I'd say you had a loop in your dialplan or a major syntax error like not having a [global] sections |
| 01:50.57 | haryv | Manx, how much Telco and Data infrastructor needs to be replaced since those floods? |
| 01:51.20 | brodiem | ManxPower: 1) by looking at them, and 2) the tons of press releases about 3com/Polycom partnering for joint offerings. |
| 01:51.35 | _ShrikE | ManxPower: Really? How many clients do you have around here? |
| 01:51.47 | ManxPower | _ShrikE: 1 client currently |
| 01:51.52 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 01:51.57 | ManxPower | brodiem: go for it and report your results. |
| 01:54.20 | haryv | mm |
| 01:56.09 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 01:56.27 | ManxPower | haryv: In the French Quarter the cable tv still goes out for a few hours per week |
| 01:56.56 | ManxPower | and there's frequently flicker outages on the CATV |
| 01:57.09 | _ShrikE | We put in the asterisk system for Kpauls down in the quarter, they have terrible power issues still. |
| 01:57.17 | ManxPower | *nod* |
| 01:57.34 | ManxPower | _ShrikE: you work for a consulting company? |
| 01:57.57 | _ShrikE | CLEC |
| 01:58.01 | ManxPower | Ah. |
| 01:58.25 | _ShrikE | We mainly do oil and gas / vsat type stuff. Not sure how we got turned on to kpauls. |
| 01:58.26 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:58.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:58.36 | ManxPower | If you say XFone I'll have to smack you. |
| 01:58.52 | _ShrikE | Nope |
| 01:59.47 | ManxPower | Do you have a presence in the Birmingham, AL area? |
| 02:00.21 | ManxPower | I/ll bet you guys picked up a lot of new clients when Madison River decided to pull out of NOLA |
| 02:00.21 | _ShrikE | No. Just a physical presence New Orleans, and Houston. |
| 02:00.48 | _ShrikE | Actually yeah, got quite a few from the pan am building. They had a large footprint in that building. |
| 02:05.56 | jblack | http://voices.washingtonpost.com/capitalweathergang/2008/08/gustav_could_become_a_giant.html?hpid=news-col-blogs looks cool, at least |
| 02:10.32 | *** join/#asterisk mirrorcolor (n=mirrorco@unaffiliated/mirrorcolor) |
| 02:18.50 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 02:19.23 | bad_duck | to install chan_mobile (and use bt phone to call), which version of asterisk is the best ? |
| 02:20.20 | anonymouz666 | do you trust chan_mobile? |
| 02:20.35 | anonymouz666 | I don't. |
| 02:20.58 | ManxPower | bad_duck: what versions are your options? |
| 02:21.14 | ManxPower | I thought it was only in the as yet unreleased 1.6 |
| 02:21.21 | bad_duck | ManxPower: all I guess, I have a fresh debian install |
| 02:21.57 | bad_duck | anonymouz666: do you have a better solution ? |
| 02:22.11 | ManxPower | Have you read this: http://www.voip-info.org/wiki/view/chan_mobile |
| 02:23.11 | bad_duck | ManxPower: I'm reading it right now |
| 02:23.15 | ManxPower | or this http://www.chan-mobile.org/?page_id=4 |
| 02:23.40 | bad_duck | also :) |
| 02:29.44 | *** join/#asterisk WimpMan (n=wimpy@gw.fl.yeti.dk) |
| 02:29.59 | bad_duck | It seems that I can compile it on asterisk 1.4 |
| 02:30.09 | bad_duck | with little modifications |
| 02:30.14 | bad_duck | I'll try |
| 02:30.18 | *** join/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com) |
| 02:30.27 | WimpMan | Hi all |
| 02:31.25 | WimpMan | I wonder if someone can give me a clue with dundi. I've got it running and it works well for all unique numbers. |
| 02:32.11 | WimpMan | But it doesn't seem to answer requests for numers that are matched by wildcarded extensions. What is it I'm missing there? |
| 02:33.28 | drdrain | anybody know what gcc will do if you try to init an unsigned char with the value axaf? |
| 02:34.52 | drdrain | So guy from rhino changed "0x0f" in an array init to "axaf" in the driver source on my machine today trying to fix a problem |
| 02:35.32 | Alton2 | 0x is the prefix for a hex character in many languages, |
| 02:35.42 | drdrain | Yeah I know |
| 02:35.46 | Alton2 | I'm surprised ax would be understood at all, probably cause a compiler error. |
| 02:36.05 | drdrain | No, apparently it compiled |
| 02:36.23 | drdrain | I don't think ax is a prefix |
| 02:36.26 | Alton2 | funny, compiler isn't very careful, trying to think what else it might have understood it as. |
| 02:36.29 | WimpMan | Is it #defined by chance? |
| 02:36.46 | drdrain | I don't think so but I'll check |
| 02:36.59 | drdrain | what is the prefix for octal ... don't remember |
| 02:37.09 | drdrain | it aint ax I'm sure |
| 02:37.10 | WimpMan | It's the only thing I can think of ATM, that makes sense. |
| 02:37.15 | WimpMan | 0 |
| 02:37.44 | drdrain | Dang ... haven't pulled out the white book in a decade |
| 02:38.07 | Alton2 | K&R, the Bible |
| 02:38.34 | WimpMan | Alton2: Don't make me feel old. |
| 02:38.35 | drdrain | LOL ... yep. Once upon a time I was a Preacher |
| 02:38.35 | Alton2 | fellows smart enough to dream up the rules as they went along... and have them come out right! |
| 02:38.56 | drdrain | I'm a backslider now though |
| 02:38.56 | Alton2 | only as old as I feel..... |
| 02:39.21 | Alton2 | I like lots of languages, but Perl kinda offends me. |
| 02:39.46 | drdrain | Was seduced by the Dark Side: Stroustoup |
| 02:39.53 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 02:39.54 | Alton2 | I use PHP with Asterisk and am pretty happy about that. Good generic language. |
| 02:39.59 | WimpMan | A good typing execise tho |
| 02:40.01 | Alton2 | What's that? |
| 02:40.09 | Qwell | Alton2: s/Good // |
| 02:40.26 | Alton2 | hah |
| 02:40.32 | Qwell | PHP makes it too easy to write bad code |
| 02:40.52 | WimpMan | That's why so many people like it. |
| 02:40.54 | Alton2 | You'd prefer... ADA? |
| 02:41.01 | Alton2 | 20 lines for hello world! |
| 02:41.10 | Alton2 | ok, 10, but you get the idea. |
| 02:41.13 | Qwell | Alton2: would you rather 1 line for a SQL injection? |
| 02:41.15 | drdrain | Or Modula maybe |
| 02:41.39 | Alton2 | lots of good languages out there, I guess everything's a tradeoff |
| 02:41.46 | drdrain | How bout CHILL? |
| 02:41.50 | Qwell | but where's the tradeoff on PHP? |
| 02:41.51 | drdrain | God I am old |
| 02:41.52 | Alton2 | I liked cobol, stuff read like English to me. |
| 02:42.39 | Alton2 | I'm just referring to the normal tradeoffs, loose variable typing versus less to define/code, that sort of thing. |
| 02:43.10 | Alton2 | I don't make _that_ many mistakes coding. :) I can live with PHP, C, etc.. |
| 02:44.40 | Alton2 | I used to say about the music biz, a good sound system cannot fix a bad band. Similarly, a good language cannot make up for poor coding. |
| 02:45.21 | florz | but a bad language can make good coding impossible |
| 02:45.48 | Alton2 | Oh, I don't know, wrote many decent programs in Basic in the past. I'd prefer everyone use Pascal, but what do I know? :-) |
| 02:46.31 | drdrain | You haven't lived until you coded in 6502 machine language |
| 02:46.39 | WimpMan | pleads guilty as well. Since brandy appeared I'm using Basic again from time to time. |
| 02:46.42 | Alton2 | I have actually, and kidna prefer it. |
| 02:46.45 | drdrain | And I do mean "machine language" |
| 02:46.59 | drdrain | No wimpy assembler |
| 02:47.16 | drdrain | Yeah I have too. Hand coded! |
| 02:47.21 | Alton2 | I know, used to pore over hex displays on Commodore computers. |
| 02:47.28 | WimpMan | So did I, even if I can't imagine today how we did it with only threee 8-bit registers. |
| 02:47.32 | drdrain | Try a KIM-I |
| 02:47.41 | Alton2 | KIM-I? |
| 02:48.01 | drdrain | Hex keypad and 6 seven segment LEDS |
| 02:48.11 | Alton2 | I was going to guess, breadboard computer.... |
| 02:48.21 | drdrain | Yeah ... google it |
| 02:48.40 | drdrain | A real man's computer |
| 02:48.49 | Alton2 | yup, hah |
| 02:48.52 | WimpMan | beer mat :-) |
| 02:49.55 | drdrain | I remember steppin up to 8085 ... wow! |
| 02:50.05 | Alton2 | hah |
| 02:50.06 | florz | well, one can certainly disagree about the quality of languages, and it really depends a lot on the purpose it's being used for ... but PHP? I mean, even if you ignore the fact that the PHP code base itself is ... well ... not exactly well-coded |
| 02:50.40 | Alton2 | I don't know, I don't find many bugs in everyday use of it. So it seems to go well enough. |
| 02:51.58 | florz | well, I for one have somewhat higher standards for compilers/interpreters than "usually works" |
| 02:52.15 | Alton2 | KBFVFPFP |
| 02:52.21 | Alton2 | HOLD |
| 02:52.37 | florz | I mean, what can you do about security of a program if the interpreter is full of security problems? |
| 02:52.56 | Alton2 | <PROTECTED> |
| 02:53.19 | ManxPower | florz: People say the same about Asterisk. |
| 02:53.21 | Alton2 | well, golly, at some point you have to use something. |
| 02:53.35 | florz | ManxPower: Well ... and they are right, obviously, so what? |
| 02:53.46 | Alton2 | Yes, not a bad parallel, I have noticed that ASterisk is not particularly well written, but we're still using it. |
| 02:54.00 | Alton2 | so it suffices, and it's important to have a standard of sorts... as long as it's not Windows. |
| 02:54.02 | florz | for a lack of alternatives, I would assume |
| 02:54.16 | florz | which isn't quite the case for programming languages and PHP =:-) |
| 02:54.34 | Alton2 | What would you prefer for the sort of use that we put PHP to? |
| 02:54.43 | Alton2 | just curious. please don't say Perl. |
| 02:54.47 | ManxPower | Alton2: PARTS of Asterisk may not be written as well as it could be, but remember many people contribute to Asterisk |
| 02:55.10 | Alton2 | Yes, same for PHP I assume, and I am quite grateful for those contributions, which seem to work well enough on the whole. |
| 02:55.34 | florz | well, I would certainly prefer Perl over PHP - but Ruby, Python, pretty much anything? |
| 02:55.36 | ManxPower | I tend use Perl just because it is a good general purpose scripting language |
| 02:56.14 | florz | ManxPower: erm ... where have you seen well-written code in asterisk? |
| 02:56.18 | ManxPower | and there is SO much written in Perl. My Perl programming style tends to look like a cross between Perl, C, and COBOL. 8-) |
| 02:56.20 | Alton2 | I like the ++ increment operator in C, etc., but I don't tend to go for too very much of the cryptic nature of Perl. |
| 02:56.39 | Alton2 | Yes, I am the master of the 25-character-long variable name. :-) |
| 02:56.45 | *** join/#asterisk JenniferAkemi (n=akemi@MTLXPQAK-1178074603.sdsl.bell.ca) |
| 02:56.48 | florz | Alton2: Well, writing Perl cryptically really is your own choice |
| 02:57.13 | florz | it's really useful to be able to do that for one-shot stuff, though |
| 02:57.22 | Alton2 | I don't write cryptically, I just see a lot of cryptic Perl code. Tends to give me an opinion about it. Its operators apparently encourage that sort of "cleverness". |
| 02:58.17 | ManxPower | Give me the string handling of Perl in C and I would never have to use anything except C |
| 02:58.56 | Alton2 | true, darnit, those darned trade-offs again |
| 02:59.35 | florz | Alton2: which isn't exactly an argument against Perl =:-) - plus, the completely inconsistent naming scheme of PHP functions easily makes up for it ... |
| 03:00.28 | Alton2 | I know what you mean there. Oh well, we PHP people get by somehow. :-) |
| 03:01.33 | florz | but the best language ever will be Perl6, in case you didn't know it yet ;-) |
| 03:01.38 | mchou | anyone here happen to use grandcentral? |
| 03:02.08 | Alton2 | Oh, another version of Perl, I can't wait. ;-) |
| 03:02.19 | Alton2 | more cpan errors! |
| 03:02.28 | riddlebox | ManxPower, what do you think of python? |
| 03:02.31 | florz | CPAN errors? uh? =:-) |
| 03:02.37 | mchou | perl code is unmaintainable |
| 03:02.45 | mchou | python ftw! |
| 03:03.04 | Alton2 | Old programmer friend of mine recommends python. need to check it out. |
| 03:03.22 | mchou | perl is just a mess |
| 03:03.28 | ManxPower | riddlebox: never used it. |
| 03:03.38 | WimpMan | diesn't really want to interrupt the language wars, but does anyone have some advice for my dundi/wildcard problem? |
| 03:03.41 | ManxPower | Not much stuff that I would use is written in it. |
| 03:04.54 | mchou | python is obeject C with better perl type string handling. |
| 03:04.59 | mchou | object* |
| 03:05.23 | Alton2 | Not a war, merely an "exchange of ideas". :-) |
| 03:05.24 | riddlebox | sorry WimpMan never setup dundi |
| 03:05.35 | ManxPower | Not many of us have set up Dundi. |
| 03:05.47 | mchou | no one here use grandcentral??? |
| 03:05.55 | riddlebox | mchou, I have a number |
| 03:06.04 | mchou | riddlebox: you use it much? |
| 03:06.15 | mchou | riddlebox: or only for testing? |
| 03:06.54 | riddlebox | haha I forgot the number, its in my cell, I think its a great tool, I just dont really use it yet, I am sure I will find something to do with it |
| 03:07.18 | *** join/#asterisk AutumnLeaves (n=AutumnLe@123.127.252.82) |
| 03:07.25 | WimpMan | Well, I'll try another time of day then. |
| 03:07.33 | ManxPower | My main client has Asterisk, Toshiba, Comdial and Nortel PBXs |
| 03:07.40 | riddlebox | WimpMan, you can try the mailing list |
| 03:07.54 | mchou | riddlebox: for the life of me I can't figure out how to get GC to programmatically accept a "1" DTMF generated by asterisk |
| 03:07.55 | riddlebox | ManxPower, Comdial, ewwww |
| 03:08.04 | ManxPower | riddlebox: I don't touch them. |
| 03:08.32 | riddlebox | ManxPower, Nortel's arent bad, I havent messed with the cs1000 to much |
| 03:08.35 | ManxPower | They have less and less of them as their ports get blown by random power surges that other PBXs would not even notice. |
| 03:08.42 | mchou | riddlebox: that's like the last obstacle to world domination! |
| 03:09.10 | ManxPower | riddlebox: The problem with Nortel is the greatest strength of Asterisk -- being able to do custom stuff |
| 03:09.37 | ManxPower | With Nortel the words "You can't do that on the Nortel" happen disturbingly often. |
| 03:09.51 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:09.53 | riddlebox | mchou, I need to test it |
| 03:10.11 | drdrain | Being an ex Nortel dork I can verify that one! |
| 03:10.11 | WimpMan | With * you hear that for the standard features. |
| 03:10.16 | *** join/#asterisk Levonk (n=lk@adsl-76-243-64-231.dsl.lsan03.sbcglobal.net) |
| 03:10.26 | riddlebox | ManxPower, yeah, Nortel has the power to do alot but you have to buy a license to do any of it |
| 03:10.51 | ManxPower | WimpMan: Asterisk isn't really a PBX (even though everyone says it is). Asterisk is a PBX TOOLKIT, that lets you build a PBX |
| 03:11.05 | mchou | lol |
| 03:11.13 | mchou | what's the difference? |
| 03:11.15 | riddlebox | WimpMan, what do you mean, the only standard feature I know that it doesnt do is SLA and I know asterisk can do it, but it doesnt work |
| 03:11.22 | pcrane | looking at asterisk CDRs, how do you tell if a call has been transfered using the builtin functions? |
| 03:11.23 | ManxPower | Trixbox and similar ilk could, arguably, be called Asterisk based PBXs |
| 03:11.50 | ManxPower | mchou: PBXs come with a standard set of ready to use features. Asterisk does not, you build features into it. |
| 03:11.57 | ManxPower | In your dialplan, AGIs, etc. |
| 03:12.19 | WimpMan | ManxPower: Well, yes and no. Yes it's not a PBX, but no it lacks quite some things to be a real toolkit. |
| 03:12.37 | ManxPower | WimpMan: Such as? |
| 03:12.42 | riddlebox | WimpMan, like what? |
| 03:12.47 | drdrain | pcrane: If you look at the CDR data in FreePBX it shows an 's' in one of the cplums |
| 03:12.54 | ManxPower | What sorts of things can you not do using the dialplan, AGI, or Manager interfaces? |
| 03:13.15 | pcrane | what about other asterisk stuff? |
| 03:13.15 | WimpMan | Yes, but you can't interface sensibly with ISDN equipment. That's a major issue if you're used to comfort phones. |
| 03:13.20 | pcrane | presumably it'll be using s too? |
| 03:13.25 | ManxPower | drdrain: "s" is usually what the destination of FXO calls is, since Asterisk does not know the dialed number. |
| 03:13.37 | ManxPower | WimpMan: I think you mean ISDN BRI. |
| 03:13.50 | drdrain | pcrane: I've never looked at the raw cdr data |
| 03:14.02 | ManxPower | WimpMan: And ISDN BRI was never popular for voice in the USA and most of the developers seem to be in the USA. |
| 03:14.12 | drdrain | Let me go look and see |
| 03:14.13 | WimpMan | BRI or PRI. It's the signalling. There is virtually no support for supplementary services. |
| 03:14.22 | ManxPower | WimpMan: Such as? |
| 03:14.30 | WimpMan | ManxPower: Yes, that's rather obvious. |
| 03:14.48 | ManxPower | Asterisk doesn't even really support BRI without having to add 3rd party software and drivers. |
| 03:15.03 | ManxPower | WimpMan: Where are you located? |
| 03:15.09 | WimpMan | ManxPower: Call forwarding, conferences, charging information. |
| 03:15.23 | WimpMan | ManxPower: de (surprise) |
| 03:16.28 | ManxPower | WimpMan: When you could easily get BRI 2B+D in the USA it was around $100/month in most places, where an analog line would cost $30/month. Also you have to buy your own NT1 |
| 03:17.18 | ManxPower | not so fondly remembers the days of installing Ascend Pipeline BRI routers. |
| 03:18.19 | WimpMan | Oh and the most anoying everyday thing is that it can't signal a call having been picked up on another phone. |
| 03:18.30 | ManxPower | For a few years before DSL was available the most common "better than dialup, not as good as T-1" service was 128K via BRIs. The telcos were swamped with orders and they barely sold any BRIs before. |
| 03:18.41 | WimpMan | But I have to admit I'm not sure SIP supports that at all. |
| 03:18.59 | ManxPower | WimpMan: Asterisk's BRI support is very sub par, compared to other systems with BRI support. |
| 03:19.01 | mchou | WimpMan: support what exactly? |
| 03:19.41 | WimpMan | Yes, i keep hearing that ISDN is obsolete since there is DSL and cable, but just not the point. ISDN is for comfortable telephone use, not internet. |
| 03:19.42 | ManxPower | mchou: in europe (even today) it is very common for a business to have an ISDN PBX. ISDN BRI going to the telco and ISDN BRI ports going to the ISDN BRI desk phones. |
| 03:19.56 | florz | Asterisk's SIP support is sub par, compared with other systems with SIP support ... so? =:-) |
| 03:20.18 | mchou | WimpMan: my question was in response to this: [20:18] <WimpMan> But I have to admit I'm not sure SIP supports that at all. |
| 03:20.37 | WimpMan | mchou: I get signalled missed calls on all telephones, even when I actually took it. |
| 03:21.29 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 03:21.32 | florz | WimpMan: SIP does support it ... there is even an asterisk patch flying around somewhere ... |
| 03:21.32 | bad_duck | is it possible to install the asterisk-addons-1.4 for cdr and the trunk version only for channel mobile on asterisk 1.4 ? |
| 03:22.47 | WimpMan | florz: Ah, nice to hear. Unfortunaletly I'm as bad as using both SIP and ISDN telephones, so that would probably only solve part of the problem. |
| 03:22.51 | jaytee | not sure if that's possible without using pry-bar-1.4.3 and shoe-horn-1.4.11 |
| 03:22.59 | *** join/#asterisk afink_ (n=afink@ip68-13-89-102.om.om.cox.net) |
| 03:23.21 | mchou | florz: I just wish * supported multiple simultaneous sip registrations (with same credentials). Life would be good then. |
| 03:23.39 | florz | WimpMan: if the ISDN phones are connected to asterisk, that should even work, I think |
| 03:23.58 | florz | mchou: I just wish, asterisk supported SIP ... =:-) |
| 03:24.31 | mchou | florz: huh? What's so bad re * support of sip that you arent happy about? |
| 03:24.57 | ManxPower | mchou: The general idea in much of the the Asterisk community is that each device should have it's own auth information and SIP user IDs are NOT the same as an extenstion. Thinking that they are is like thinking the earth if flat -- it severely limits your view and creativity. |
| 03:25.12 | WimpMan | florz: They are. Might take a try. But the mext thing to try is your miracle shaper. Found that by chance yesterday and that looks like a great thing. |
| 03:25.14 | ManxPower | We use the MAC of the device with -a for the first line appearance, -b for the second, etc |
| 03:25.21 | florz | mchou: well, it starts with the fact that it can't even reliable parse and encode display names and SIP URIs? |
| 03:25.33 | AutumnLeaves | ./install_amp -bash:/var/lib/asterisk/bin/retrieve_conf: Permission denied |
| 03:25.40 | AutumnLeaves | howto resolve? |
| 03:25.40 | florz | WimpMan: it _is_ a great thing! ;-) |
| 03:25.56 | mchou | florz: huh?? I havent encountered that TTBMK |
| 03:26.11 | ManxPower | florz: the great thing about open source is you are free to try to fix it and submit the fix to the code base |
| 03:26.40 | mchou | ManxPower: what creativity? I want real "follow me." :) |
| 03:26.45 | ManxPower | About 4 lines of my code is in Asterisk. 8-) |
| 03:26.51 | WimpMan | ManxPower: I wish I would find the time to dig through the * sources. |
| 03:27.11 | florz | ManxPower: erm, yeah - but how does the fact that I am free to do so change the quality of the code? =:-) |
| 03:27.12 | ManxPower | mchou: So do real followme, having different userids for each device does NOT prevent you from doing that. |
| 03:27.23 | mchou | ManxPower: I dont want to have to admin overhead |
| 03:27.28 | Juggie | Someday sip will support multiple registrations |
| 03:27.30 | mchou | to deal* |
| 03:27.35 | Juggie | feel free to add it |
| 03:27.43 | ManxPower | Juggie: what a horrid idea. |
| 03:27.47 | mchou | Juggie: it's already there |
| 03:27.58 | Juggie | multiple registrations from the same device |
| 03:27.58 | mchou | it's called openser :) |
| 03:28.00 | Juggie | same id. |
| 03:28.10 | ManxPower | Juggie: but not from different devices? |
| 03:28.22 | Juggie | you know what i mean :) |
| 03:28.32 | Juggie | its late i'm tired |
| 03:28.37 | ManxPower | mchou: imagine if users could just TELL the pbx where they are. |
| 03:28.38 | Juggie | multiple sip phones use the same username/pass |
| 03:28.49 | ManxPower | regardless of device userid |
| 03:28.53 | Juggie | and they all ring if the sip username is called. |
| 03:29.09 | Juggie | oej does have plans for it someday |
| 03:29.19 | mchou | ManxPower: w are you talking about? Sip tells ther server where they are. It's the IP address:port |
| 03:29.26 | mchou | the* |
| 03:29.32 | ManxPower | mchou: no, what device they are at. |
| 03:29.35 | florz | mchou: well, I can't remember all the gory details in which ways the asterisk sip parser is broken - but for example, IIRC, it won't properly encode display names containing quotes, or local parts containing pounds |
| 03:29.44 | mchou | ManxPower: say again?? |
| 03:29.46 | ManxPower | via a web browser of dialing an extension and logging in. |
| 03:29.57 | ManxPower | s/of/or |
| 03:30.04 | mchou | ManxPower: what device they are at? What exactly do you mean? |
| 03:30.30 | florz | mchou: plus, it's not able to correctly disect the URI sip:foo%40bar@foo |
| 03:30.37 | *** part/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com) |
| 03:30.56 | mchou | ManxPower: screw web browser |
| 03:31.18 | ManxPower | lets say you are at your desk phone, when you come in for the day you dial a code to tell the phone system where you are. It routes calls to that device. When you leave for the day, you could call it from your cell phone to tell it, or just dial a code on the desk phone. |
| 03:31.35 | mchou | ManxPower: why bother? |
| 03:31.52 | mchou | ManxPower: you have lappy, ata, let's say |
| 03:31.52 | WimpMan | Indeed. Why not just call them all? |
| 03:32.00 | ManxPower | mchou: hey, you can always just have it ring multiple SIP userIDs instead of only one. |
| 03:32.05 | mchou | you travelling with lappy |
| 03:32.22 | mchou | wtf do I need a new AUTH token?? |
| 03:32.29 | mchou | just for that? |
| 03:32.54 | ManxPower | Dude, you put it in once and never worry about it again. |
| 03:33.10 | mchou | and sometimes I might be at lappy, and sometimes I might be by a hard phone |
| 03:33.22 | ManxPower | yes? |
| 03:33.26 | mchou | put it in once?? |
| 03:33.33 | mchou | wtf you talking about? |
| 03:34.14 | ManxPower | mchou: I am talking about ideas for the specifics of a follow me feature that does not require all the devices to register to one account userid/password to the PBX. |
| 03:34.31 | WimpMan | I really don't see what's the problem about giving each device it's login and put them all in the dial command. Seems perfectely logical to me. |
| 03:34.50 | ManxPower | (10:26:30 PM) mchou: ManxPower: what creativity? I want real "follow me." :) |
| 03:35.17 | florz | mchou: erm, and the worst part probably is that asterisk uses the request URI of incoming INVITEs as the number to match against the dialplan ... that really gives you headaches ... |
| 03:35.21 | mchou | ManxPower: you're severely stretching if you say multiple regs is not a desirable feature which would ring many locales (with the same auth tokens) |
| 03:35.44 | mchou | cause that would cut down both on admin and user overhead |
| 03:36.02 | WimpMan | ManxPower: Doing a follow me shouldn't be that hard, I guess. Crrate an extension that writes it's caller to a variable or astdb and use that in the dial command. |
| 03:36.56 | ManxPower | It would be a nice feature, but I just don't see it cutting down on admin all that much. |
| 03:37.07 | mchou | ManxPower: tell my why that's not considered "creative" |
| 03:37.19 | ManxPower | If you have enough adds/moves/changes for it to be an issue you would have automated it anyway. |
| 03:37.21 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 03:37.35 | mchou | lol |
| 03:37.59 | mchou | no we are talking "automated?" like writing scripts and whatnot? |
| 03:38.07 | mchou | now* |
| 03:38.20 | JT | anyone know of a way to set a different vlan on multiple ports of a cisco catalyst at once? |
| 03:38.24 | ManxPower | mchou: once we set up a phone, we never have to touch the phone config again. |
| 03:38.28 | mchou | which part of "no admin overhead" dont you grok? |
| 03:38.47 | lanning | creativity is a waste. destruction is fun! |
| 03:38.51 | mchou | lol |
| 03:38.56 | ManxPower | You understand how much time is saved by not having to reconfigure phones for an add/move/change |
| 03:39.14 | mchou | who the hell needs to reconfig phones? |
| 03:39.30 | ManxPower | you do as the phone moves between accounts. |
| 03:39.33 | WimpMan | doesn't get the context, either. |
| 03:39.45 | ManxPower | In an office phones are moved and changed out all the time., |
| 03:39.59 | mchou | yeah so? |
| 03:40.14 | JT | jesus chris mchou is a grump boy |
| 03:40.15 | lanning | I seem to remember it being called TFTP |
| 03:40.21 | mchou | lol |
| 03:40.26 | ManxPower | with your idea, phone A is given to Marcy and well, Marcy is going to get the previous owner's calls unless the auth info is changed. |
| 03:40.31 | WimpMan | Why chance the phones login? You know the phone the rest is dialplan. |
| 03:40.47 | mchou | JT: I'm not grumpy. Merely dont see what ManxPower is arguing about? |
| 03:41.08 | ManxPower | mchou: How long have you been working with Asterisk? |
| 03:41.34 | mchou | ManxPower: long enough to understand there are shortcomings with * |
| 03:41.54 | ManxPower | See how you'll feel after using Asterisk for a few years. |
| 03:42.21 | mchou | ManxPower: so how is that any different with Marcy succeeds JoeBob? |
| 03:42.43 | ManxPower | Until then fight Asterisk's oddities while I accept Asterisk's oddities. |
| 03:43.12 | WimpMan | ManxPower: You just have to change the phones callerid in sip.conf and the extension in extensions.conf and if you don't like much work, you create both from a singe config. |
| 03:43.39 | mchou | WimpMan: bingo!! |
| 03:43.41 | ManxPower | I do see how if phones are not changing hands all the time (ISTP situation) I can see how having multi-device/same auth info would be handy. |
| 03:44.01 | ManxPower | WimpMan: You understand what I'm saying. |
| 03:44.33 | WimpMan | ManxPower: Sure? As I don't see a problem there. |
| 03:44.39 | mchou | lol |
| 03:44.59 | mchou | wonders who is trying to be intentionally dense |
| 03:45.18 | lanning | I am! |
| 03:45.29 | mchou | yeah. "limits creativity" |
| 03:45.30 | afink_ | wonders if there is an #asterisk-trolls channel |
| 03:45.54 | mchou | that's some orwellian doublespeak if I've ever heard any |
| 03:46.41 | lanning | The Ministry of Information |
| 03:46.56 | mchou | lanning: aka ManxPower :) |
| 03:48.14 | afink_ | Hey well were talking about it... ManxPower thanks for all the help you have given me and the countless others I have seen you help. ;) |
| 03:52.25 | bad_duck | does somebody uses chan_mobile ? |
| 03:53.27 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 03:53.43 | WimpMan | is only on wlan. |
| 03:54.58 | *** join/#asterisk Levonk (n=lk@adsl-75-62-132-111.dsl.lsan03.sbcglobal.net) |
| 04:12.52 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 04:14.11 | ReDNeQ | if you have 2 locations for 1 company.. is it better to run 2 systems seperated by IAX and Diff extensions, or 1 server everyone sips |
| 04:15.35 | ManxPower | ReDNeQ: each has it's advantages and disadvantages |
| 04:16.06 | ManxPower | If you have a dedicated link between the two sites like a T-1, then it would be cheaper to have only one system. |
| 04:16.16 | WimpMan | 'd take two, just to be on the safe side if either end has some sort of trouble, but that's probably more a question of personal taste. |
| 04:16.22 | ReDNeQ | we have a VPN between the shops |
| 04:16.45 | ManxPower | That is not a dedicated link. In that case, I recommend two different systems with local lines at each location |
| 04:16.51 | ReDNeQ | currently we have 2 systems using iax2 trunk |
| 04:16.51 | WimpMan | Number of extension would be a factor. |
| 04:17.28 | ReDNeQ | ManxPower: thats the issue thoughg.. how can i save money with the phone costs @ both locations plus the internet connections |
| 04:17.34 | ReDNeQ | they want me to reduce costs even more |
| 04:17.40 | ManxPower | I doubt you will. |
| 04:17.45 | WimpMan | So why do you consider getting rid of one? |
| 04:18.00 | ReDNeQ | the second shop is just estimators and they have their cells. |
| 04:18.13 | ReDNeQ | if the connection is down, at least they have those phones |
| 04:18.30 | ReDNeQ | i can save the $200 per month by getting rid of the phone lines |
| 04:18.36 | JT | too bad if they can't call emergency services during a disaster |
| 04:18.37 | ReDNeQ | and dropping the internet costs |
| 04:18.45 | ManxPower | If the internet is down and they can't recieve calls, they will nail you to the wall until it's fixed. |
| 04:18.52 | WimpMan | $200 for phone lines??? |
| 04:19.03 | ReDNeQ | ManxPower: We have been there trust me |
| 04:19.16 | ReDNeQ | this VPN and with 2 systems still hasnt made them happy |
| 04:19.25 | ReDNeQ | they bitch about the qualiyt between locations |
| 04:19.31 | ReDNeQ | saying it sound robotic sometimes |
| 04:19.50 | JT | what sort of vpn is it? |
| 04:20.03 | ManxPower | You're going over a VPN over the internet, chances are over consumer cheap DSL or cable modems with cheap linksys routers |
| 04:20.13 | ReDNeQ | i was thinking a sip extenions would use less bandwidht since its not using iax2 trunk |
| 04:20.18 | WimpMan | Do you use some sort of traffic control? |
| 04:20.21 | ManxPower | Be surprized it works as well as it does. |
| 04:20.37 | ReDNeQ | ManxPower: we have T1 at one location and 15x2 at other |
| 04:20.38 | ManxPower | ReDNeQ: IAX2 trunking will save you the most bandwidth by far. |
| 04:20.45 | ReDNeQ | we are using Netgear VPN routers |
| 04:20.56 | ManxPower | even worse than linksys. |
| 04:21.08 | ReDNeQ | hrmm.. |
| 04:21.14 | ManxPower | I'm a router snob. I even buy Cisco for my HOME router. |
| 04:21.19 | ReDNeQ | they are all made in same plant in china |
| 04:21.43 | ReDNeQ | ManxPower: I used to work @ Cisco |
| 04:22.26 | ReDNeQ | its just a matter of what costs you want to spend. I mean Jordache, levis or wrangler |
| 04:22.30 | ReDNeQ | they are all jeans |
| 04:22.52 | ManxPower | ReDNeQ: My clients use 2621/2621XM at almost all of their sites with one 1850 and a couple of linksys routers. The switches at the larger offices are all Catalyst 55xx switches |
| 04:22.54 | ReDNeQ | any thats another fight, the routers were in place |
| 04:23.35 | ManxPower | Got the Catalysts for very, very, very cheap on eBay, cheap enough we bought some spares and still saved money |
| 04:29.46 | *** join/#asterisk DirtyD (n=DirtyD@ool-4571b24d.dyn.optonline.net) |
| 04:29.56 | DirtyD | Hola good people. |
| 04:33.56 | *** join/#asterisk Cresl1n (n=matt@c-68-62-170-117.hsd1.al.comcast.net) |
| 04:33.56 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 04:34.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:42.12 | mchou | wth. Why is my ITSP messing up my caller ID? |
| 04:45.36 | *** join/#asterisk irieKEN (n=chatzill@netblock-68-183-115-231.dslextreme.com) |
| 04:46.31 | irieKEN | HumanCell: I don't know if my thank you message from before went through earlier. |
| 04:46.47 | irieKEN | HumanCell: Thanks for the help figuring out my routing issue:) |
| 04:47.06 | HumanCell | irieKEN: np ... |
| 04:47.20 | HumanCell | I've got some assistance here lately ... glad to give some back ... |
| 04:47.39 | irieKEN | Got to love the community effort;) |
| 04:48.07 | HumanCell | Yeah ... now if only I can figure out my CallerID issue ... it's a weird one ... |
| 04:48.32 | irieKEN | What's the issue? |
| 04:49.28 | *** join/#asterisk slider750 (n=Slider@ip68-96-75-158.oc.oc.cox.net) |
| 04:49.33 | Chicago | Hi, I am trying to figure out howto support 'MSN Messenger Audio Codec' or which ever codecs MSN Messenger 5.1 permits. |
| 04:50.13 | Chicago | Anybody covered this before? About 30 minutes ago I had 2 way audio, but after I broke something setting the codec to u-law, I think I have don't it wrong. |
| 04:50.58 | *** join/#asterisk AutumnLeaves (n=AutumnLe@123.127.252.82) |
| 04:51.04 | irieKEN | Chicago: Unfortunately, I've never tackled this particular problem. |
| 04:51.20 | *** join/#asterisk s0lid (n=s0lid@60.52.253.84) |
| 04:51.24 | irieKEN | What do you guys think of Flowroute? |
| 04:52.57 | Chicago | irieKEN: Well, I did have a breakthrough where calls from my Verizon wireless phone pass through asterisk and get answered by MSN messenger 5.1 using SIP. Now I want to tune it up to work correctly. |
| 04:54.14 | irieKEN | Hmm. |
| 05:06.10 | [TK]D-Fender | irieKEN: We think the only reason anyone mentions flowroute is because of that stupid kiddie video using them to unmask blacked calls and that those inquiring about them only ask because of having heard about it. |
| 05:06.30 | [TK]D-Fender | In other words : target audience acheived |
| 05:07.38 | irieKEN | [TK]D-Fender: Yep. Saw the vid, looked at the site, was interested.... Just thought that it was weird that there is NO information about it anywhere online:) |
| 05:08.23 | irieKEN | What provider would you recommend? |
| 05:08.32 | [TK]D-Fender | ~itsplist-us |
| 05:08.32 | 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, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 05:09.29 | *** join/#asterisk Rico29 (n=Rico@ARennes-257-1-166-80.w86-214.abo.wanadoo.fr) |
| 05:11.02 | slider750 | broadvoice ;( argh |
| 05:11.10 | slider750 | <- just me |
| 05:11.21 | slider750 | I need to give teliax another shot |
| 05:13.47 | irieKEN | Hmm. |
| 05:14.09 | irieKEN | I just need good PRI service:) |
| 05:16.22 | irieKEN | Any recommendations there? |
| 05:23.23 | slider750 | Yea, UCN |
| 05:23.47 | slider750 | They have been around a long time, they have a tech side to the business (hosted PBX/Call Center) but they do both PRI and their VOIP offering is pretty solid. |
| 05:24.14 | slider750 | I'll get flamed for this but I never do VoIP over Public Internet for my clients... Just local and/or dedicated circuits |
| 05:24.56 | drmessano | sets slider750 on fire |
| 05:24.57 | JT | that makes perfect sense |
| 05:25.02 | JT | dunno why you'd get flamed |
| 05:26.17 | slider750 | lol... some people think SIP is the answer for everything... |
| 05:26.38 | slider750 | JT: Just need PRI for the office? |
| 05:26.45 | [TK]D-Fender | slider750: Yes, we call them "retards" |
| 05:27.10 | drmessano | SIP isn't a bad solution for very small business.. The same crowd that you could get by with setting them up with 2 or 4 lines of Vonage |
| 05:27.17 | [TK]D-Fender | slider750: SIP makes sense if you need a massive amount of channels and burning up a lot of minutes. |
| 05:27.29 | slider750 | Word Up Doc |
| 05:27.35 | drmessano | Beyond that you're scaling up b/w to account for internet latency where the money could be better spent with dedicated circuits |
| 05:27.37 | irieKEN | slider750: Yeah, a PRI for our little office. |
| 05:27.41 | [TK]D-Fender | drmessano: Yup, pretty small or really large. Not so much inbetween |
| 05:27.49 | slider750 | Fender: Agreed, we burn a ton of minutes with SIP outbound |
| 05:28.09 | slider750 | Level 3 crapped on us today |
| 05:28.20 | slider750 | anybody else experience that? |
| 05:28.47 | slider750 | was a pretty major outage with their session border controllers and such |
| 05:29.00 | drmessano | Oh shit |
| 05:29.04 | WimpMan | gets stuck at l3 when trying to reach ebay, and that's for at least three weeks. |
| 05:29.06 | drmessano | Did it go down? |
| 05:29.48 | drmessano | I was playing with a few BGP exploits on my Netzero line.. no clue I could take down L3 with it |
| 05:29.50 | drmessano | Crap.. |
| 05:29.56 | slider750 | oh yea we went down and when we were up we had echo and no DTMF pass thru it really sucked.. switched all my traffic away for the day... |
| 05:29.58 | drmessano | Guess I will call them and apologize |
| 05:30.11 | slider750 | dr: nice.... nah no need... |
| 05:30.36 | drmessano | No no, I feel bad now |
| 05:30.40 | slider750 | good for their network ops team... they should be able to handle it... |
| 05:31.10 | drmessano | Yeah, about that |
| 05:31.22 | drmessano | I sent them to the zoo for the day |
| 05:31.25 | drmessano | :( |
| 05:31.30 | drmessano | bad timing |
| 05:31.36 | slider750 | the tech on the phone was like "hold on, let me ask so and so engineer" and finally i was like, dood put me on the phone with the damn engineer, no offense but I need to talk to someone that can give me a solution |
| 05:32.14 | slider750 | They were at the penguin exhibit all day |
| 05:32.20 | slider750 | saw them on the web cam |
| 05:32.31 | drmessano | All you need to do is sent a Level 4 engineer into the back room, have him unplug the WRT54G at the core of L3 for 5 seconds, and plug it back in... |
| 05:32.36 | slider750 | <-- hmm... that wasn't funny... maybe a drink will help |
| 05:32.38 | *** part/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 05:33.11 | drmessano | I am completely going to start asking level 3 and 4 techs to do that shit |
| 05:33.17 | slider750 | Yes, sir... We are rebooting our core router now... Please hold the line. My Screen says the device will be back up in "60 Seconds..." |
| 05:33.24 | drmessano | "Yes sir, we are experiencing a wide area outage" |
| 05:33.34 | drmessano | "Can you just power cycle your router?" |
| 05:33.41 | drmessano | "Wha?" |
| 05:34.00 | drmessano | lol |
| 05:34.34 | slider750 | Today I realized calling the RESPORG dept. of any carrier gets you someone you can talk to |
| 05:35.10 | slider750 | I friggin hate going thru Level 1, to level 2, oops i transferred you to accounting, is your bill paid, oh you need tech support, level 1, dammit let me talk to your manager, oh he's not in.... |
| 05:35.32 | drmessano | I called a tech yesterday for an issue I was having with a customer using scanning software on Ricoh copiers |
| 05:35.42 | slider750 | omg ricoh |
| 05:35.47 | slider750 | and? |
| 05:36.01 | slider750 | send an IKON tech |
| 05:36.14 | drmessano | Tech told me "Oh, thats on a windows domain controller? The software uses MSDE, and we dont support running it on a domain controller because MSDE can corrupt the AD database" |
| 05:36.33 | drmessano | ZOMG BECUZ DAY BOT HAVE DATABASE IN THEM |
| 05:36.34 | [TK]D-Fender | ok, bed time, later all. |
| 05:36.38 | slider750 | sweet, back door hack, i'll install MSDE on a Best Buy computer next time I'm there |
| 05:36.53 | drmessano | I was livid |
| 05:37.05 | slider750 | brb |
| 05:37.15 | drmessano | Apparently.... |
| 05:37.24 | drmessano | and this may be good for your Asterisk DB guys |
| 05:38.05 | drmessano | Running two applications on the same server that happen to store their info in any kind of database is bad. Apparently the tubes get crossed up. |
| 05:38.40 | drmessano | DB + DB = ZOMG CROSSED THE STREAMS OF A UNLICENSED NUCLEAR ACCELERATOR |
| 05:40.26 | irieKEN | Night, D-Fender |
| 05:40.26 | slider750 | senator from alaska said the same thing, internet is a bunch of tubes... they have a youtube song, quite funny. |
| 05:40.35 | slider750 | d-fenda long gone |
| 05:41.38 | irieKEN | Meh. |
| 05:42.29 | drmessano | It just occurred to me we really have had a successful test of this equipment |
| 05:42.32 | drmessano | I blame myself |
| 05:42.33 | drmessano | Me too |
| 05:42.59 | drmessano | Whats to worry about, each of us is wearing an unlicensed nuclear accelator on their back |
| 05:43.06 | drmessano | switch me on |
| 05:43.10 | drmessano | Love that scene |
| 05:44.53 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 05:45.20 | slider750 | anybody know of any good SEO channels? |
| 05:50.26 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 05:50.32 | *** join/#asterisk miloux (n=miloux@213.88.194.120) |
| 05:52.51 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 05:54.28 | *** join/#asterisk pids (n=pids@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 06:02.10 | *** join/#asterisk [pyro] (n=Pyro@tor/regular/bracketed-pyro) |
| 06:04.55 | [pyro] | anyone here had exp with digium 410p cards? just wondering if i need to set dip switchs to "terminate" or not when connected to an NT1 |
| 06:10.07 | pids | Anyone know why when I set my dtmf to INFO asterisk sees the dtmf tone but then ignores it? |
| 06:11.08 | mosty | how do you know asterisk sees dtmf? |
| 06:11.17 | pids | shows on the console |
| 06:11.51 | mosty | can you pastebin the relevant context in your dialplan? |
| 06:13.57 | pids | [Aug 25 12:12:50] DTMF[16456]: channel.c:2193 __ast_read: DTMF begin '1' received on SIP/xxx.xxx.xxx.xxx |
| 06:13.57 | pids | [Aug 25 12:12:50] DTMF[16456]: channel.c:2203 __ast_read: DTMF begin ignored '1' on SIP/xxx.xxx.xxx.xxx |
| 06:14.24 | pids | It sees it then ignores it. |
| 06:15.16 | mosty | the dialplan needs to be configured to listen for dtmf. can you pastebin the context that the call was in when that message was printed to the console? |
| 06:16.06 | pids | not easily it goes into a perl script |
| 06:16.37 | pids | however if I set it to IAX from the same provider it works great or I set the dtmf to rfc2833 it works |
| 06:17.06 | pids | It has to be set in the dialplan to catch dtmf? |
| 06:17.42 | pids | Were switching from IAX to SIP, IAX has been working for several months |
| 06:17.54 | mchou | pids: really?? |
| 06:18.01 | pids | yes |
| 06:18.13 | mchou | pids: I'm imagine most ppl prefer iax |
| 06:18.22 | pids | thats the confusing part. Or if I set it to SIP/rfc2833 it works |
| 06:18.26 | mchou | pids: why the switch? |
| 06:18.35 | SplasPood | Heh, most people needing to link to asterisk servers :) |
| 06:18.38 | mosty | pids, ok so why don't you use rfc2833? |
| 06:18.56 | pids | mchou, our provider is dropping iax support |
| 06:19.12 | mchou | pids: which provider is that? |
| 06:19.28 | pids | mosty, asterisk has an issue with dropping dtmf under sip under rfc2833 |
| 06:19.40 | pids | mchou, voxbone |
| 06:19.42 | mosty | which version of asterisk? |
| 06:20.16 | pids | <PROTECTED> |
| 06:20.41 | mchou | pids: are you manually sending a dtmf 1 by pressing 1 on handset? |
| 06:20.51 | pids | yes |
| 06:21.15 | pids | thats whats frustrating, it sees the correct dtmf, then ignores it |
| 06:21.55 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-68ceb93cea4c942e) |
| 06:22.26 | slider750 | i use voxbone ;) |
| 06:23.57 | mchou | crap. someone tell me what I'm doing wrong. I set 'callerid="My Name" <My Number>' in my sip.conf. Whenever I call another phone some 'mystery' CallerID shows up! |
| 06:24.23 | mchou | I mean it's totally random |
| 06:25.25 | mchou | that command is supposed the callerID on outbound calls, correct? |
| 06:25.46 | pids | slider750, SIP or IAX? |
| 06:25.56 | slider750 | SIP |
| 06:26.04 | mchou | at least as far the phone number is concerned, correct? |
| 06:26.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:26.12 | pids | mchou, does your carrier support setting CID? |
| 06:26.29 | baliktad | why would * start denying sip registrations with 503 server error? |
| 06:26.40 | mchou | pids: it's supposed to |
| 06:26.57 | mchou | pids: it has a we page for setting the callerID |
| 06:27.02 | mchou | web* |
| 06:27.06 | baliktad | I just restarted my phone and 2 of the 4 extensions are fine, 2 of the 4 are getting rejected with 503 |
| 06:27.36 | mchou | pids: and I've done the same thing on the web page..... |
| 06:27.40 | pids | mchou, you mean you can enter a number on the webpage? |
| 06:27.47 | mchou | yeah |
| 06:27.52 | mchou | and a name |
| 06:28.04 | mchou | but no guarantees on the name, of course :) |
| 06:28.14 | pids | mchou then you cant send it via sip, its static to the one on the web page. |
| 06:28.38 | pids | mchou, which carrier? |
| 06:28.41 | mchou | pids: so you're saying my provider has it messed up? |
| 06:29.15 | pids | mchou, no some providers dont allow setting the cid at dial out. They dont want people using them to spoof CID |
| 06:29.17 | mchou | cause the web page caller ID setting is "mine" |
| 06:29.58 | mchou | I mean the info the the provider web page is the same as my caller ID info in SIP |
| 06:30.43 | mchou | but I STILL get some different #s |
| 06:31.15 | mchou | it makes no sense at all |
| 06:31.28 | pids | then either the providers page is messed up or it may be getting confused that you are sending it CID as well. |
| 06:31.56 | mchou | hmm, interesting |
| 06:32.14 | pids | mosty, any ideas on the dtmf issue? |
| 06:34.02 | mosty | pids: what is the specific issue of using rfc2833? it seems to work for me |
| 06:34.43 | *** part/#asterisk [pyro] (n=Pyro@tor/regular/bracketed-pyro) |
| 06:35.27 | pids | if the dmtf are sent to fast then it drops some dtmf's |
| 06:37.14 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-be837dd2a9cae639) |
| 06:45.16 | *** join/#asterisk hackbanger (n=hackbang@213.209.114.6) |
| 06:46.25 | baliktad | something is wrong with my poor * box |
| 06:46.32 | baliktad | and I don't know what it is :( |
| 06:46.41 | baliktad | this always happens after 3-4 days |
| 06:46.58 | mchou | yeah, there's something foul in demark |
| 06:47.28 | mchou | I beginning to think asterisk is a ball of duct tape and twine |
| 06:47.36 | mchou | I'm* |
| 06:48.55 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 06:49.21 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 06:49.31 | DarKnesS_WolF | good morning |
| 06:50.12 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 06:52.30 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 06:52.36 | mandh | Hello |
| 06:52.49 | mandh | i get lot of error |
| 06:52.52 | mandh | chan_zap.c: CallerID returned with error on channel 'Zap/4-1' |
| 06:53.56 | mandh | i guess that make lot of "Calls Abandoned" to me |
| 06:54.05 | mandh | how to fix the problme |
| 06:54.09 | mandh | problem |
| 06:57.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:15.55 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 07:19.12 | *** join/#asterisk Levonk (n=lk@adsl-75-62-139-153.dsl.lsan03.sbcglobal.net) |
| 07:25.31 | *** join/#asterisk At|eT (n=chatzill@193.2.139.253) |
| 07:25.54 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 07:34.45 | *** join/#asterisk Levonk (n=lk@adsl-75-62-139-153.dsl.lsan03.sbcglobal.net) |
| 07:39.55 | *** join/#asterisk At|eT (n=chatzill@193.2.139.253) |
| 07:40.23 | *** join/#asterisk ToTo (n=ToTo@209.8.41.110) |
| 07:41.36 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 07:43.59 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:45.29 | At|eT | hello |
| 07:45.40 | At|eT | i have problems setting up Pickup for IAX |
| 07:45.46 | At|eT | can anybody help me? |
| 07:47.53 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-c7fd4a7357bdc77b) |
| 07:50.39 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 07:59.11 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:00.09 | *** join/#asterisk bazilek (n=bazil@mail.generation-p.com) |
| 08:04.55 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:21.32 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:28.10 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:38.20 | *** join/#asterisk lzhang (n=lzhang@209-20-86-89.slicehost.net) |
| 08:45.53 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 09:04.00 | *** join/#asterisk slider750 (n=Slider@ip68-96-75-158.oc.oc.cox.net) |
| 09:05.29 | *** join/#asterisk Great_Anta_Baka (i=c4249986@gateway/web/ajax/mibbit.com/x-bf55a015b021f5ff) |
| 09:06.59 | Great_Anta_Baka | sigh google isnt helping.. dtmf key presses arent being detected from analog lines but are detected on calls from isdn and mobile calls??!?? o_0. Please help someone |
| 09:20.42 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 09:29.14 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 09:32.19 | mchou | pids: well, well. Figured out the callerid issue. Turns out my ITSP requires 11 digits. 1-NXX-NXX-XXXX. Originally put only 10 digits. |
| 09:32.59 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 09:34.28 | *** join/#asterisk ToTo (n=ToTo@207.176.6.200) |
| 09:43.12 | Great_Anta_Baka | anyone? its just incomming calls from analog lines.. are these dtmf tones sent inband? |
| 09:45.59 | *** join/#asterisk Andre101 (n=a@123-243-77-135.tpgi.com.au) |
| 09:46.36 | Andre101 | Hello people, I have a VoIP provider with 2 DID's, how do I setup asterisk to tell the difference between the two numbers that are called? |
| 09:47.39 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 09:48.21 | mchou | Andre101: you arent providing enough info. What protocol does said provider use? |
| 09:48.35 | Andre101 | mchou: SIP |
| 09:49.45 | mchou | in sip.conf you can do register=>username1:password1@provider1/exten1 |
| 09:49.56 | mchou | and for the 2nd s/1/2 |
| 09:50.41 | Andre101 | s/1/2? |
| 09:50.53 | mchou | sed |
| 09:51.19 | Andre101 | mchou: sorry, you have lost me |
| 09:51.35 | mchou | Andre101: forget it |
| 09:51.47 | mchou | if you dont know sed dont mess with asterisk |
| 09:54.13 | qp | how can I make a trunk re-register? stuck our * box behind cisco in our farm, and the trunk times out after a while |
| 09:55.11 | mchou | qp: enable appropriate firewall rules |
| 09:55.37 | mchou | regestering is not the way to solve that |
| 09:55.50 | mchou | re-register* |
| 09:55.52 | Andre101 | mchou: an you breifly explain what i need to do to get this working? i'll read whatever I have to |
| 09:56.01 | mchou | Andre101: google |
| 09:56.11 | Andre101 | what terms? |
| 09:56.13 | qp | iv enabled udp 5060, 20000:20010, 4569 and 5038 and created static routes for them |
| 09:57.23 | mchou | qp: enabled those ports where and how exactly? |
| 09:57.51 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 09:58.23 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 09:58.50 | qp | on the cisco pix .. |
| 10:00.33 | *** join/#asterisk ph8 (i=henri@85.234.155.91) |
| 10:00.58 | ph8 | Hi all, i've just started using asterisk; set it up with gradwell - just waiting my phone to arrive - it looks pretty nice so far, nice one! |
| 10:01.00 | qp | what would be nice would be a "heres a working set of cisco rules for your * box using a sip trunk" :) |
| 10:01.02 | ph8 | But i bet you hear that a lot :) |
| 10:01.11 | mchou | qp: lol |
| 10:02.18 | qp | surely its just a few access-list and static commands :( |
| 10:03.04 | *** part/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 10:04.07 | mchou | qp: google 'cisco pix asterisk' turns up lots of good stuff |
| 10:04.18 | Andre101 | mchou: still same problem, they seem to all be coming in on the main SIP register statement. The problem is, I didn't get a username2/password2 with the second DID number |
| 10:04.23 | *** join/#asterisk Levonk (n=lk@adsl-75-62-129-208.dsl.lsan03.sbcglobal.net) |
| 10:04.23 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 10:04.59 | mchou | Andre101: that's even better |
| 10:05.21 | mchou | all you need is one registration statement |
| 10:05.32 | mchou | I thought you had two accts |
| 10:05.54 | mchou | if you only have 1 acct w/ multiple DID that's the way to go |
| 10:06.09 | Andre101 | all the calls to the second number seem to be detected as the first DID.. |
| 10:06.55 | Andre101 | if i just do register=>username1:password1@provider1 it doesn't match anything DID |
| 10:06.56 | mchou | Andre101: I doubt it. pastebin output of 'dialplan show' |
| 10:07.36 | *** join/#asterisk padski (n=padski@host217-45-159-149.in-addr.btopenworld.com) |
| 10:07.49 | mchou | Andre101: it's not supposed to. It matches the 's' extension |
| 10:12.08 | Andre101 | mchou: It's FreePBX, so there's alot of crap in dialplan show |
| 10:12.33 | mchou | screw that |
| 10:12.42 | mchou | ho to #FreePBX |
| 10:12.46 | mchou | go* |
| 10:13.06 | Andre101 | the channel is normally dead.. |
| 10:13.10 | Andre101 | i'll give it a go anyway |
| 10:14.16 | Great_Anta_Baka | where do i look if asterisk is having problems detecting dtmf from analog lines |
| 10:14.24 | padski | Hi! I am looking at a trixbox that sometimes (approx 1 in 10 calls) has a ~10 sec delay in repsonding to incoming sip calls. I have set sip debug for the ip and even the sip packets are logged late (compared with a tcpdump I also have running). any suggestions ? |
| 10:14.56 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-17-133.revip2.asianet.co.th) |
| 10:15.00 | kaldemar | padski: yeah, ask in #trixbox |
| 10:15.08 | Great_Anta_Baka | oi delete that trixbbox crap |
| 10:15.13 | Great_Anta_Baka | its a fail |
| 10:15.13 | mchou | lol |
| 10:15.19 | padski | kaldemar, thanks. here on freenode, yes ? |
| 10:15.24 | kaldemar | padski: yes. |
| 10:15.26 | Great_Anta_Baka | i did it and havent looked back |
| 10:15.41 | padski | Great_Anta_Baka, what do you recommend instead ? |
| 10:15.48 | mchou | wonders why ppl wont use pedal to the metal straight asterisk |
| 10:15.51 | Great_Anta_Baka | ast_now_gui so much better IMHO |
| 10:16.05 | Great_Anta_Baka | and doesnt mess up the config files |
| 10:16.11 | kaldemar | padski: most people here don't use GUI's. |
| 10:16.14 | padski | Great_Anta_Baka, thanks |
| 10:16.15 | Andre101 | mchou: Coz my configs get too messy (my own fault) |
| 10:16.16 | Great_Anta_Baka | so its still possible to mess in it |
| 10:16.44 | Great_Anta_Baka | padski gimme a sec |
| 10:16.50 | Great_Anta_Baka | i will give you url |
| 10:16.55 | mchou | Andre101: your configs would never be messier than trixbox :) |
| 10:17.30 | Andre101 | mchou: haha.. probably right.. hmm, thinking about switching back to asterisk now.. |
| 10:17.50 | Andre101 | i'm not using the whole trixbox thing.. just FreePBX side of things |
| 10:18.14 | Great_Anta_Baka | thats the problem andre |
| 10:18.15 | *** join/#asterisk Levonk (n=lk@75.62.134.244) |
| 10:18.23 | Great_Anta_Baka | its really really messy in there |
| 10:18.37 | Great_Anta_Baka | padski http://svn.digium.com/svn/asterisk-gui/branches/2.0/ |
| 10:18.51 | Andre101 | yea, it does a lot of stuff.. most of it crap, use about 10% of it.. |
| 10:18.56 | Great_Anta_Baka | you need subversion |
| 10:19.05 | Andre101 | whats the asterisk-gui like? |
| 10:19.13 | Great_Anta_Baka | maybe that tells you that you shouldnt be using it Andre101 |
| 10:19.28 | padski | Great_Anta_Baka, thanks. |
| 10:19.31 | Great_Anta_Baka | tis cool.. i like the new interface |
| 10:19.37 | defswork | mchou: asterisk and freepbx is a nice solution - but it doesn't have an endpoint manager yet |
| 10:19.53 | mchou | defswork: maybe |
| 10:19.55 | defswork | otherwise my trixbox days are over |
| 10:20.07 | Andre101 | is going to give asterisk-gui a go... |
| 10:20.13 | defswork | I installed from source over the weekend and am quite happy with the results |
| 10:20.28 | Andre101 | anyone know how to restore the configs back to defaults? |
| 10:20.35 | mchou | haha |
| 10:20.46 | Great_Anta_Baka | using trixbox just puts in another layer that can cause crap |
| 10:20.48 | Great_Anta_Baka | debugging |
| 10:20.52 | Great_Anta_Baka | is a bitch |
| 10:20.55 | mchou | Andre101: rm -rf /etc/asterisk :) |
| 10:21.35 | mchou | Andre101: make sample config |
| 10:21.42 | mchou | or some such :) |
| 10:22.00 | Andre101 | using ubuntu packages |
| 10:22.09 | padski | I'm not really much concerned or interested in the gui myself. my task is to fix the problems that can't be fixed from such a gui. |
| 10:22.31 | Great_Anta_Baka | aptitude remove asterisk |
| 10:22.41 | Great_Anta_Baka | and then reinstall |
| 10:22.46 | Great_Anta_Baka | ok i am out peace |
| 10:22.49 | defswork | Great_Anta_Baka: I've got 4 trixbox installs and for speed of setup and deploy I've no regrets. |
| 10:23.00 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 10:25.51 | padski | pretty quiet on #trixbox. another question then ... |
| 10:26.10 | padski | can I strace a production asterisk without messing up the calls ? |
| 10:26.34 | mchou | heh |
| 10:26.54 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 10:26.57 | mchou | padski: why exactly do you want to do that? |
| 10:27.13 | padski | mchou, I want to know what it is doing in those 10 seconds. |
| 10:27.58 | padski | I might get a clue from an strace |
| 10:28.13 | mchou | I'm dubious |
| 10:28.16 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 10:28.40 | mchou | you'd be much better off turning on the verbose level up |
| 10:28.56 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 10:28.59 | mchou | to find out what's going on |
| 10:28.59 | padski | mchou, how do I turn it to full ? |
| 10:29.26 | Andre101 | meh, freepbx is looking to be a pain in the ass to uninstall |
| 10:29.47 | padski | mchou, what can it log before the sip packet ? |
| 10:30.29 | mchou | just do asterisk -rv (with as many v's at the end as necessary) |
| 10:31.46 | padski | "Verbosity was 3 and is now 10" |
| 10:32.16 | defswork | padski: sounds like a networking issue to me |
| 10:32.38 | padski | defswork, I see the packets come in on a tcpdump 10 seconds earlier. |
| 10:32.53 | defswork | aah I thought you said later :) |
| 10:32.56 | mchou | padski: what protocol? |
| 10:33.02 | padski | defswork, what sort of networking issue do you have in mind ? |
| 10:33.13 | padski | mchou, sip, udp on port 5060 |
| 10:33.19 | mchou | like dns |
| 10:33.23 | mchou | for one |
| 10:33.38 | defswork | does * do reverse dns checks ? |
| 10:33.43 | padski | mchou, possible yes, ptr lookup |
| 10:34.20 | padski | defswork, would seem like a silly thing to block on |
| 10:34.45 | defswork | unless you are doing other lookups |
| 10:35.06 | defswork | padski: dns timeout - you can't do nothing else but wait |
| 10:35.13 | mchou | I don't think asterisk does reverse lookups, but dont quote me on that |
| 10:35.50 | mchou | padski: try IP addrs, if it works dandy you know it's dns :) |
| 10:35.55 | defswork | with trixbox though it could be doing a whole host of things in the dial plan - but you say you see nothing for 10 seconds ? |
| 10:36.10 | mchou | oh shit |
| 10:36.22 | mchou | he's using trixbox?? |
| 10:36.24 | defswork | aye |
| 10:36.26 | mchou | haha!! |
| 10:36.39 | mchou | it might not be DNS after all :) |
| 10:36.44 | *** join/#asterisk EugenA (n=eugen@213.179.145.226) |
| 10:37.10 | EugenA | i'm tring to get running web-gui |
| 10:37.49 | EugenA | i've done everything from README, but i cannot connect to http://localhost:8088/gui/static/config/cfgbasic.html |
| 10:38.20 | EugenA | have i change also my apache config? |
| 10:39.51 | padski | it is trying to look up some crazy stuff in dns ... |
| 10:40.04 | padski | A? 5060-b7d26590.example.com |
| 10:40.15 | padski | A? 203-b7d2a508.example.com |
| 10:40.20 | mchou | haha!! |
| 10:40.26 | padski | this for a call to extension 203 |
| 10:40.46 | mchou | padski: that's some trixbox config cruft, no doubt |
| 10:41.03 | *** join/#asterisk pcrane (n=pcrane@125-238-255-60.broadband-telecom.global-gateway.net.nz) |
| 10:41.21 | padski | mchou, heh, reckon you're probably right there |
| 10:41.40 | pcrane | ~centos52bug |
| 10:41.41 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 10:42.52 | kaldemar | EugenA: apache has nothing to do with asterisk-gui. check that asterisk is in fact listening to port 8088 and try http://127.0.0.1:8088/asterisk/static/config/index.html |
| 10:43.41 | defswork | trixbox forked freepbx - that's a little unsportmanlike |
| 10:43.59 | mchou | wtf!! |
| 10:44.34 | mchou | I'm really puzzled by MWI for pap2 and asterisk |
| 10:45.10 | mchou | why is there a stutter tone even when there are no new messages? |
| 10:45.28 | mchou | I sthis some pap2/asterisk bug? |
| 10:45.37 | mchou | Is this* |
| 10:46.08 | mchou | it's bugging the shit out of me |
| 10:47.05 | padski | even with Verbose at 10 the first thing logged is the VERBOSE[2738] logger.c: <--- SIP read from |
| 10:49.29 | pcrane | I need a hand compiling asterisk addons |
| 10:49.40 | mchou | padski: then try 'help core set debug' |
| 10:49.42 | pcrane | I need mysql support (for realtime stuff) |
| 10:50.05 | pcrane | when I do make menuselect and look under applications the mysql option is XXX'd out |
| 10:50.09 | pcrane | complaining that it Depends on: mysqlclient(E) |
| 10:50.15 | padski | mchou, thanks |
| 10:50.40 | defswork | pcrane: install mysqlclient-dev |
| 10:50.45 | pcrane | yum list | grep mysql |
| 10:50.50 | pcrane | shows mysql-deve is installed |
| 10:50.51 | pcrane | ah |
| 10:51.08 | pcrane | # yum install mysqlclient-dev |
| 10:51.08 | pcrane | Loading "installonlyn" plugin |
| 10:51.08 | pcrane | Setting up Install Process |
| 10:51.09 | pcrane | Setting up repositories |
| 10:51.09 | pcrane | Reading repository metadata in from local files |
| 10:51.09 | pcrane | Parsing package install arguments |
| 10:51.11 | pcrane | Nothing to do |
| 10:51.13 | pcrane | nope |
| 10:51.15 | mchou | wtf |
| 10:51.16 | defswork | typically called something like libmysqlclient14-dev |
| 10:51.25 | mchou | pcrane: use pastebin dude |
| 10:51.31 | defswork | use brain |
| 10:51.33 | pcrane | kk |
| 10:52.09 | defswork | libmysqlclient15-dev on debian |
| 10:52.40 | pcrane | know what it is for Centos by any chance? |
| 10:52.53 | defswork | yum search |
| 10:53.35 | *** join/#asterisk Levonk (n=lk@adsl-75-62-140-73.dsl.lsan03.sbcglobal.net) |
| 10:54.31 | defswork | hmm - according to the internet on centos the client lib dev has been rolled up into mysql-dev |
| 10:55.07 | padski | "Core debug was 99 and is now 2". was fun, but ... |
| 10:55.47 | pcrane | which is strange, cause I've got it installed... yet it complains that it can't find mysqlclient |
| 11:07.20 | pcrane | I know what the problem was |
| 11:07.34 | pcrane | there was some i386 libs for zlib and open ssl messing things up |
| 11:07.52 | yang | In Grandstream phones, there is a setting LOCAL RTP PORT = 5004 , does this value needs to match the rtp port range from /etc/asterisk/rtp.conf ? |
| 11:07.55 | pcrane | remove those, making sure that mysql-devel.x86_64 is installed |
| 11:08.02 | pcrane | voila, problem solved |
| 11:11.25 | EugenA | i cannot connect to * gui |
| 11:12.14 | EugenA | telnet: connect to address 127.0.0.1: Connection refused |
| 11:12.14 | Nugget | telnet is eeeeeeevil! |
| 11:13.34 | *** join/#asterisk angryuser (i=c392fa5f@gateway/web/ajax/mibbit.com/x-95bde39c17130460) |
| 11:13.42 | yang | hi angryuser |
| 11:13.52 | angryuser | hello yang |
| 11:15.14 | padski | at core debug 99 and verbose 10, the first sign of the sip packet in the logs is still >10 seconds late. |
| 11:15.54 | padski | no sign of any dns traffic that would account for it at the time. |
| 11:17.10 | padski | box is otherwise idle |
| 11:18.20 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 11:18.44 | *** join/#asterisk Levonk (n=lk@adsl-75-62-137-199.dsl.lsan03.sbcglobal.net) |
| 11:20.32 | EugenA | do i have to start web-gui-server or something? why i cannot connect to it on port 8088? |
| 11:20.58 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.5) |
| 11:21.18 | *** join/#asterisk n3hxs (n=HAMming@host-64-65-218-80.har.choiceone.net) |
| 11:24.28 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 11:24.39 | EugenA | Nugget, hi |
| 11:25.05 | kaldemar | EugenA: did you do what i said? |
| 11:28.33 | EugenA | kaldemar, i checked port 8088 with telnet - no success |
| 11:30.53 | kaldemar | EugenA: pastebin your asterisk's http.conf and double check that you have webenabled=yes in manager.conf |
| 11:32.39 | kaldemar | EugenA: and show "http show status" in CLI |
| 11:33.34 | *** join/#asterisk dream_th (n=dream_th@91.187.96.48) |
| 11:33.44 | dream_th | Hi |
| 11:34.15 | dream_th | i am having some problems with my asterisk server, i have a local network and static IP which is configured in a router, i have setup all the configuration of asterisk/freepbx and it works perfect |
| 11:34.33 | dream_th | but i have some issues with it whenever the internet connection is down, phones just stop working cant call or receive any calls |
| 11:34.52 | dream_th | this is my sip_nat.conf -> http://pastebin.com/m765c1ff4 |
| 11:35.01 | dream_th | i need to make sure that even if there is no internet connection to be able to make local calls |
| 11:35.11 | dream_th | any ideas where im doing wrong here |
| 11:35.38 | kaldemar | dream_th: ask in #freepbx |
| 11:36.16 | dream_th | why not here? |
| 11:36.33 | dream_th | after all its asterisk right? |
| 11:36.34 | kaldemar | because no one here uses freepbx nor wants to degub it. |
| 11:36.38 | kaldemar | debug... |
| 11:37.01 | dream_th | but i believe this is asterisk related issue not freepbx |
| 11:37.13 | gr0mit | kaldemar, you were right the first time ;-) |
| 11:38.16 | kaldemar | dream_th: plain asterisk doesn't stop working if your outbound connection drops, so it's hardly asterisk. |
| 11:38.38 | kaldemar | gr0mit: ummm.. first time? |
| 11:38.44 | gr0mit | degub |
| 11:38.51 | kaldemar | oh :) |
| 11:39.12 | gr0mit | if something is all gubbed up, it means it is clogged up, got lots of mess around it |
| 11:39.26 | gr0mit | which describes freepbx rather well ;-) |
| 11:39.36 | kaldemar | that was a new word for me. :) |
| 11:39.39 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 11:39.40 | *** join/#asterisk s0lid (n=s0lid@219.93.152.130) |
| 11:39.40 | gr0mit | hehe! |
| 11:40.01 | dream_th | so what do you prefer over freepbx |
| 11:40.05 | gr0mit | it is British English. not sure if it translates over to USA or wherever you are? |
| 11:40.09 | amessina | is away: Gone away for now. |
| 11:40.13 | kaldemar | plain asterisk with no GUI. |
| 11:40.29 | gr0mit | nods at kaldemar |
| 11:40.50 | dream_th | and with gui? |
| 11:40.51 | beek | Wonders -- Asterisk has a GUI? |
| 11:41.08 | jblack | beek: Third party ones. |
| 11:41.22 | kaldemar | people here don't prefer GUI. |
| 11:41.27 | dream_th | i see |
| 11:41.44 | beek | I was being facetious |
| 11:41.48 | TJNII | ~freepbx |
| 11:41.48 | 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 |
| 11:41.49 | kaldemar | that's why i told you to go to #freepbx in the first place. |
| 11:42.03 | EugenA | kaldemar, allright, everything works :-) i've set prefix in http.conf |
| 11:42.31 | dream_th | thanks anyway |
| 11:42.34 | dream_th | :) |
| 11:42.54 | *** part/#asterisk dream_th (n=dream_th@91.187.96.48) |
| 11:43.01 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 11:43.04 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 11:43.08 | kaldemar | gr0mit: i'm from finland. not a native english speaker. |
| 11:45.21 | EugenA | kaldemar, i don't see in gui my current asterisk config. why? |
| 11:46.22 | At|eT | i have problems setting up Pickup for IAX |
| 11:46.25 | At|eT | can anybody help me? |
| 11:46.53 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 11:47.02 | EugenA | kaldemar, i don't know, but should i see anything from current config in this gui? |
| 11:47.04 | kaldemar | EugenA: i'm the wrong person to ask that. i've only started it to see how it looks, not really used it. :) |
| 11:47.41 | TJNII | At|eT: You mean call picking? |
| 11:47.44 | waverly360 | Hey guys, I'm using asterisk 1.2 (working on upgrading to 1.4 soon) and I'd like some suggestions for preventing a queue from dialing a user if they're already on the phone. Currently, I'm using a local channel tied to the queue which uses the Chanisavail function in order to determine whether a user is on the phone or not before it routes the call to them. I realize this isn't what that function was designed for, and that's why I'm asking for sugg |
| 11:47.55 | At|eT | TJNII: yes |
| 11:49.49 | TJNII | At|eT: Last I checked it was buggy. I had to use pickupmark to get it working, but for (IIRC) a SIP bug. |
| 11:51.16 | At|eT | TJNII: I folowed this guide http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup |
| 11:51.19 | At|eT | but with no suces |
| 11:51.42 | At|eT | i recieve No channel found |
| 11:51.47 | At|eT | any idea? |
| 11:52.34 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:53.39 | TJNII | At|eT: Like I said, it's buggy and doesn't always work. Try using PICKUPMARK. |
| 11:53.44 | TJNII | has to go to work |
| 11:53.49 | TJNII | Good luck |
| 11:53.57 | At|eT | ok, tnx |
| 11:56.44 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 11:58.14 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:59.00 | *** join/#asterisk jazzmann (n=chatzill@cpc1-lutn9-0-0-cust163.lutn.cable.ntl.com) |
| 12:00.48 | jazzmann | hi can someone please clear one confusion.With ATA adapter can I use my analog line in asterisk.So anyone calling to my landline number will be picked up by asterisk or it is just acts as a voip thanks |
| 12:01.22 | jblack | I thought ATAs just took phones, not phone lines. |
| 12:01.50 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:02.11 | jazzmann | yeah thanks that was i was confuesed.So if I need to connect my phone line,I need a card no other option |
| 12:02.44 | jblack | You can get one for a couple hundred dollars. It's a one time purchase. |
| 12:03.45 | jblack | That can feel like a lot up front, but six months later, most people wonder why they cared. |
| 12:04.17 | jblack | Alternatively, you can just get a DID over voip for about 3 bucks a month. |
| 12:05.00 | jazzmann | it is possible that I can integrate asterisk in webapp in a way that clients can connect their phone lines to my remote asterisk box through user name password |
| 12:05.12 | jblack | play with a did for a few months, decide you like your new toys, and get a FXS adaptor for your phone line if you want, then. |
| 12:05.28 | jblack | jazzmann: Look for web based sip clients. |
| 12:05.40 | jblack | there are some out there. |
| 12:06.06 | ph8 | in an extensions file with a line like this: exten => s,n,Background(thanks) <--- how do i define 'thanks' - i have to relate it to an mp3? |
| 12:06.27 | jazzmann | so with that sip clients clients phone line will be connected in a way that any call to there number will be picked up by asterisk or I have to give them new numbers |
| 12:06.52 | jazzmann | I was trying at present chan_mobile to test asterisk |
| 12:07.03 | jblack | ph8: s,n,Background(/file/path/thanks). Don't give the extension, * searches for all the extensions it can handle. |
| 12:07.16 | jblack | goes for a smoke. |
| 12:07.20 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 12:07.28 | ph8 | cool, if i don't specify a path where does it look do you know? |
| 12:07.32 | mvanbaak | jblack: good idea |
| 12:08.46 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:08.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:09.38 | jazzmann | jblack smoking is only good for manufacturers of ciggerattes |
| 12:09.43 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-670fc9c7534cb20a) |
| 12:10.19 | jazzmann | many thanks for the help,hoping to catch up later with you for more advise |
| 12:10.57 | Great_Anta_Baka | sigh @ me loosing wallet and cell phone at the same time |
| 12:11.08 | Great_Anta_Baka | didnt know my biggest pbx was down :( |
| 12:11.45 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:12.17 | mvanbaak | ouch |
| 12:12.23 | jblack | Let me get this right jazzmann... |
| 12:12.43 | mvanbaak | maybe it's time to implement some redundancy Great_Anta_Baka ? |
| 12:12.48 | jblack | I took the time to help you, and so in way of thanks, you're gonna deride me at 8:12 am my time? |
| 12:12.50 | *** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it) |
| 12:12.57 | Great_Anta_Baka | like another cell phone!!!! |
| 12:12.59 | padski | I just saw a call that noone could pick up. the CDR says it was answered by vmu205. what is that ? |
| 12:13.26 | jblack | padski: Sounds like the name of your sip/iax2 phone to me. |
| 12:14.10 | padski | jblack, nope. we have a 205 |
| 12:14.25 | jblack | No, the account, not the model number. |
| 12:14.34 | ph8 | i don't have my SIP phone yet, so to test I was just trying to get a connection with some background music playing - at the moment i have num,1,Ringing; num,n,Wait,2 <-- and I have a 2 ring then hang-up working with that, I then tried to add --> num,n,WaitExten, num,n.Background(/home/pbx/sounds/musicfile) |
| 12:14.38 | jblack | I bet if you look at sip.conf, you called the phone vmu205 |
| 12:14.40 | ph8 | but that doesn't play musicfile.mp3 which resides in that dir |
| 12:14.42 | ph8 | any ideas? |
| 12:14.56 | padski | jblack, in the CDR calls answered on 205 appear as "205", this appears "vmu205" |
| 12:15.03 | jblack | ph8: Perhaps you didn't install mp3 support for *, or you didn't strip the id3 tags from the mp3. |
| 12:15.12 | ph8 | i havent' stripped any tags |
| 12:15.16 | ph8 | but i haven't conciously installed mp3 support either |
| 12:15.21 | ph8 | let me look for a package |
| 12:15.39 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:15.40 | jblack | ph8: Ok. It could be either of those. * is very particular about the mp3s it plays. |
| 12:15.57 | ph8 | ok i've just installed the mp3 package |
| 12:15.59 | ph8 | will try again |
| 12:16.02 | jazzmann | hey sorry if I offended you jblack,hope you will find some room in your big generous heart to forgive me |
| 12:16.06 | jblack | don't forget to restart *. |
| 12:16.08 | ph8 | but that should just play the song indefinitely right jblack ? |
| 12:16.20 | ph8 | yep i was going to ask that as well, every time i change extensions.conf - do i have to restart, there's no reload? |
| 12:16.32 | jblack | jazzmann: Me? Nah. I'm heartless and cruel. Now you'll have to protect your dog against me kicking it. Forever!! |
| 12:16.40 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:17.04 | jblack | You can reload most of asterisk by reloading a subsystem. asterisk -r, "extensions reload" in that particular case. |
| 12:17.12 | Great_Anta_Baka | mvanbaak: redundancy is a great idea but some businessmen are just too stingy to fork out cash to secure their stuff |
| 12:17.24 | jblack | Technically, you could probably do something funky like "module load mp3" or something, but I'm not sure of the naming. For that one, just restart * |
| 12:17.45 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:17.55 | jblack | Great_Anta_Baka: Thats ok for them to make that decision, just so long as you're clear, and redundant, that that's the choice they've made. |
| 12:18.13 | [TK]D-Fender | jblack: "module load format_mp3.so |
| 12:18.33 | jblack | ph8: See what tk wrote? That was for you. |
| 12:18.42 | jblack | [TK]D-Fender: Morning bud. How's it going? |
| 12:19.17 | ph8 | erm where? |
| 12:19.28 | jblack | The "module load format_mp3.so" part. |
| 12:19.28 | ph8 | ah thanks |
| 12:19.44 | padski | jblack, thank-you for your help, think I understand whats happening there now |
| 12:20.12 | jblack | You can reload most anything, if you know the right name. Some things are modules, some things are applicatons, others are specific. (extensions reload, sip reload, iax2 reload are the three most common) |
| 12:20.43 | russellb | format_mp3 is the devil |
| 12:20.52 | russellb | I strongly recommend not using it |
| 12:21.05 | ph8 | Please ensure that the file 'manager.d/*.conf' |
| 12:21.08 | ph8 | in /etc/asterix i assume? |
| 12:21.13 | ph8 | russellb: How come? |
| 12:21.20 | jblack | touch -p /etc/asterisk/manager.d/empty.conf |
| 12:21.24 | ph8 | ty |
| 12:21.27 | [TK]D-Fender | russellb: No, it is merely a tool. It is not inherently evil. I just has consequences you'd better understand when choosing it :) |
| 12:21.27 | russellb | because the actual mp3 handling code has blown up many a time |
| 12:21.50 | jblack | actually, mkdir /etc/manager.d ; touch /etc/asterisk/manager.d/empty.conf" |
| 12:21.51 | russellb | and fyi, if that code blows up, we're not fixing it |
| 12:21.52 | russellb | :) |
| 12:21.54 | jazzmann | I google it and it seems sipbase web clients are only good for interent.I am looking that peopl can connect their phone line via there fxo card to my asterisk server.So when their phone rings it is picked up by asterisk box |
| 12:22.01 | ph8 | jblack: Yeh i got you :-) |
| 12:22.12 | jblack | touch should have a -p option |
| 12:22.18 | jazzmann | http://npramesh.wordpress.com/2008/06/05/web-based-sip-phone-with-flex-java-and-red5-server this is the link of one sip web base client |
| 12:22.50 | [TK]D-Fender | jazzmann: Linksys SPA-3102 <--- |
| 12:23.00 | ph8 | right i added load => format_mp3.so to /etc/asterisk/modules.conf |
| 12:23.05 | Great_Anta_Baka | jblack: yeah but they crap on me for some reason, which i ignore of course, because they couldnt get hold of me and had to call this other company in |
| 12:23.05 | jblack | jazzmann: That's further than this channel can take you. If you get anywhere, please put a nicely written doc up at voip-info.org |
| 12:23.08 | Great_Anta_Baka | to fix it |
| 12:23.20 | ph8 | and i still see WARNING[12560] file.c: Unable to open /home/pbx/sounds/comeonover (format 0x2 (gsm)): No such file or directory |
| 12:23.27 | jblack | Great_Anta_Baka: You lost me. Who craps on you? Does it turn you on? |
| 12:23.28 | ph8 | so should i try stripping id3 tags? |
| 12:23.37 | [TK]D-Fender | ph8: default "autoload => yes" should already have taken care of it unless you're paranoid and/or a minimalist |
| 12:23.40 | [TK]D-Fender | ph8: Yes |
| 12:23.52 | ph8 | oh right |
| 12:23.53 | Great_Anta_Baka | no the company owner that the pbx is installed at jblack |
| 12:24.05 | jblack | ph8: if you insist on using mp3s, yes. A better option is to convert the mp3 to gsm (the sox tool will help, and you can find info by googling sox mp3 gsm) |
| 12:24.22 | ph8 | ah sounds better |
| 12:25.19 | jblack | Great_Anta_Baka: Ahh, ok. Well, you want something warm and fuzzy like "Don't stand up to crap!" ? |
| 12:25.20 | *** part/#asterisk EugenA (n=eugen@213.179.145.226) |
| 12:25.49 | Great_Anta_Baka | yes yes :p |
| 12:26.02 | Great_Anta_Baka | its all good though SLA is ending soon |
| 12:26.16 | jblack | Ok. Don't accept crap. kick 'em in the balls, and say you have expectations that MUST be met. |
| 12:26.17 | Great_Anta_Baka | so wont have to listen to their nonsense no more |
| 12:26.26 | Great_Anta_Baka | lol |
| 12:26.38 | jblack | by the way, I have no idea what you're talking about. |
| 12:26.46 | Great_Anta_Baka | thats good |
| 12:26.48 | jblack | stats his second cup of coffee |
| 12:26.58 | Great_Anta_Baka | i was lost anyway |
| 12:27.00 | jblack | bah. half empty |
| 12:27.12 | mvanbaak | hands jblack more coffee |
| 12:27.24 | jblack | my kid went off to her first day of high school today. |
| 12:27.27 | styelz | hands jblack a Penguin mint |
| 12:27.40 | jblack | I had those once. I think they taste nasty |
| 12:27.49 | styelz | yes |
| 12:27.52 | jblack | at least when you eat 10-20 at once. |
| 12:27.52 | styelz | i agress |
| 12:28.24 | styelz | eww |
| 12:28.25 | styelz | lol |
| 12:28.32 | styelz | heart attack man |
| 12:28.34 | jblack | Penguin mints taste nasty. Mints taste good. Therefore, penguins must taste nasty. |
| 12:28.39 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:28.52 | jblack | styelz: 2 pots of coffee a day. =) |
| 12:29.35 | mvanbaak | jblack: gheh :) I'm at 4 to 5 liters a day |
| 12:29.56 | styelz | clarity |
| 12:30.19 | jblack | I know what 2 litre is. That's a lot. |
| 12:30.35 | jblack | That's more than me, if you're drinking 4. |
| 12:31.18 | jblack | Oh, some of you guys that are new, you'll probably want to start testing sip soon. There's a channel conference you can use to conect to, to try and get sip working. |
| 12:31.24 | jblack | ~conference |
| 12:31.24 | jbot | from memory, conference is The channel conference can be reached via IAX2/asterisk.linuxguru.net , SIP/asterisk.linuxguru.net , and PSTN 1-206-424-2834. Contact jblack with any concerns. |
| 12:32.19 | mvanbaak | jblack: cool. thanks |
| 12:32.36 | jblack | Nobody is likely to be there unless someone says "Why is nobody there" |
| 12:32.56 | mvanbaak | whehehe |
| 12:33.02 | mvanbaak | they are all out getting more coffee |
| 12:33.06 | *** join/#asterisk guillote_GNU (n=guillote@host58.201-253-57.telecom.net.ar) |
| 12:33.18 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:33.20 | jblack | Nah. It's just a great irony of life. Asterisk admins hate hanging on the phone. |
| 12:33.28 | styelz | heh |
| 12:33.30 | russellb | you usually have people hanging out there? |
| 12:33.43 | styelz | thats what voicemail is for |
| 12:33.47 | russellb | for a telephony community, we don't use conferences very much, heh |
| 12:33.53 | *** join/#asterisk Talnakh (n=Talnakh@c180-25.i02-6.onvol.net) |
| 12:33.55 | jblack | russellb: No, I don't think it's being used very much at all. |
| 12:34.00 | ManxPower | I did telephone tech support for 5 years. After that I no longer like talking on the phone. |
| 12:34.15 | russellb | yeah, i'm not a fan of the phone either, most of the time |
| 12:34.18 | jblack | Not enough people are interested to get critical mass. |
| 12:34.32 | russellb | it's good for explaining complicated stuff sometimes |
| 12:34.35 | ph8 | haha works thanks jblack |
| 12:34.36 | russellb | because i'm too lazy to type it out |
| 12:34.41 | ph8 | it's awesome having your own phone system |
| 12:34.52 | Talnakh | hi guys, i am using asterisk with voice menus and the voice is jerky and very slow. what resource i have to look into to make the voice sound natural? |
| 12:34.59 | jblack | ph8: Welcome |
| 12:35.20 | jblack | Talnakh: Stuttering and stuff? |
| 12:35.26 | ph8 | :p thanks |
| 12:35.26 | Talnakh | yes |
| 12:35.44 | jblack | Talnakh: Where is the server in relation to you? Same network, across town.. ? |
| 12:35.46 | ph8 | do you think i'll have any issues taking this back to university? their firewall is massively anal. My PBX is on a virtual server but the phone will be on a router in my uni room |
| 12:36.13 | [TK]D-Fender | Qapf: Please kill or disable the away script for this channel... |
| 12:36.15 | jblack | ph8: That depends upon your network-fu skills. |
| 12:36.16 | Talnakh | jblack, it the same subnet. It is just like the voicemenu is on slow mode |
| 12:36.29 | ph8 | jblack: I need to tunnel udp -> tcp? |
| 12:36.35 | jblack | Talnakh: I'd look really closely at whether or not sound is workign well on your server. |
| 12:36.41 | jblack | ph8: Look at pptp. |
| 12:36.43 | ph8 | I think i read something to that effect last night |
| 12:36.45 | ph8 | cheers again |
| 12:36.54 | jblack | which is a ppp over tcp tunnel. |
| 12:37.04 | [TK]D-Fender | Talnakh: Who made the recordings? What format? What are you listening to it on? |
| 12:37.13 | Talnakh | jblack, any resource you could point me to to check |
| 12:37.21 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:37.21 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:37.49 | jblack | Talnakh: Not offhand, no. It just sounds like a local sound playing issue. That's based on my gut reaction, not fact. |
| 12:38.40 | jblack | Man. I hope that everyone in new orleans learned how to swim since the last hurricane |
| 12:38.57 | jblack | Think they'll rebuild, again? |
| 12:39.07 | russellb | stares at jblack |
| 12:39.08 | [TK]D-Fender | jblack: Like ants.... |
| 12:39.19 | Talnakh | [TK]D-Fender, i installed asterisk on faster PC first and everything worked perfectly. but then i am using slower pc now and voicemenu is jerky. it used to be p4 3 ghz and now it is p3 800 MHz. but i thought that 800MHz should be really enough to syntesize voice and playback menus |
| 12:39.37 | jblack | russellb: New Hurrican, Gustav, aiming straight at new orleans. http://www.stormpulse.com/ |
| 12:39.42 | [TK]D-Fender | jblack: Its like people who are starving and refuse to stop living in the desert. Go where the food is dammit!!! |
| 12:39.57 | ManxPower | jblack: Yes, we know. |
| 12:40.09 | [TK]D-Fender | Talnakh: And you have failed to answer ANY of my questions. |
| 12:40.13 | jblack | ManxPower: I know you know. He was staring at me. Figured he didn't know. |
| 12:40.21 | ManxPower | If New Orleans shut down a large part of shipping in the USA would stop. |
| 12:40.26 | russellb | jblack: harsh |
| 12:40.30 | [TK]D-Fender | jblack: He wants you to repent :) |
| 12:40.54 | jblack | Repent for what? I make a lot of people miserable, but causing hurricanes is not my forte! |
| 12:41.12 | jblack | what? Did we decide not to discuss that or something? |
| 12:41.26 | Talnakh | <PROTECTED> |
| 12:41.28 | [TK]D-Fender | jblack: For making people miserable and then reminding them that more is on the way |
| 12:41.39 | jblack | Ohhh, I see. I'm sorry. |
| 12:41.39 | [TK]D-Fender | Talnakh: "man ls" |
| 12:43.05 | ph8 | Sorry to be noob, i thought i had this down - after my Background music (which works) and my WaitExten, i have exten => 1,1,Goto(hc), then i have another section [hc] exten => s,1,Dial(SIP/201) - which should dial my phone no? |
| 12:43.15 | ph8 | eventually i probably won't have an extension system, but for now |
| 12:43.27 | jblack | Goto(hc,s,1) |
| 12:43.30 | lmadsen | ph8: you need to tell it what extension and priority to goto in context hc |
| 12:43.33 | ManxPower | ph8: you don't have a destination priority or extension. |
| 12:43.39 | ph8 | ahhh |
| 12:43.41 | ManxPower | Goto(hc,s,1) |
| 12:43.52 | ph8 | so s,1 <- the 1 is like an ordering? |
| 12:44.00 | jblack | what manxpower wrote is much better than what I wrote. Listen to him, not me. |
| 12:44.02 | lmadsen | ph8: yes |
| 12:44.02 | ManxPower | 1 is the PRIORITY |
| 12:44.07 | ph8 | cool |
| 12:44.12 | lmadsen | s,1,NoOp() |
| 12:44.14 | lmadsen | s,n,NoOp() |
| 12:44.19 | ph8 | so it's a way of listing actions o nconnection |
| 12:44.19 | lmadsen | s,n,NoOp(another line!) |
| 12:44.24 | ManxPower | ph8: "show application goto" in the CLI |
| 12:44.28 | jblack | ph8: Have you downloaded the book yet? |
| 12:44.29 | ph8 | and n is like a generic background thing |
| 12:44.35 | ph8 | the book? |
| 12:44.36 | jblack | ph8: Or better yet, bought a copy? |
| 12:44.38 | jblack | ~book |
| 12:44.39 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 12:44.41 | lmadsen | ph8: 'n' is a priority that auto-increments |
| 12:44.43 | ph8 | :o |
| 12:44.53 | ph8 | lmadsen: ever better! |
| 12:44.57 | jblack | Grab that pdf. There's a whole book there. There's a chapter just on dialplans, that will sort you out on all the details. |
| 12:44.59 | ph8 | can you start with n or do you have to start with 1? |
| 12:45.05 | lmadsen | jblack: actually 2 chapters |
| 12:45.07 | [TK]D-Fender | ph8: no, "n" is an auto-numbering feature, as are "labels" for priority. |
| 12:45.15 | lmadsen | ph8: always have to start with '1' -- the book will give you all this info |
| 12:45.21 | jblack | lmadsen: I'm trying to get someone to buy your book. Off my case! |
| 12:45.21 | [TK]D-Fender | ph8: Needs to start with 1 |
| 12:45.23 | ph8 | i'll buy a copy |
| 12:45.36 | jblack | ph8: That'll make me happy, if you do that |
| 12:45.36 | lmadsen | jblack: I wasn't try to "get on your case" |
| 12:45.44 | lmadsen | trying* |
| 12:45.48 | *** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it) |
| 12:46.00 | jblack | lmadsen: You're just trying to keep me down, man! Well, I've had it it will all people "l"! Time to revolt! |
| 12:46.09 | ph8 | so instead of Goto(hc,s,1) can i Goto(DIAL(Sip/201)) somehow? |
| 12:46.13 | ph8 | or do i have to have that seperate context? |
| 12:46.22 | ph8 | or do i take out goto and just write dial |
| 12:46.22 | lmadsen | ph8: nope, that's not how Goto() works |
| 12:46.24 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:46.27 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:46.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:46.31 | ph8 | like a function pointer |
| 12:46.32 | ManxPower | ph8: Now you are asking questions you should be reading the book to learn |
| 12:46.37 | lmadsen | :) |
| 12:46.40 | jblack | ph8: YOu can goto any (context, extension, priority) you want. Please hit the book up on this before you ask too many questions, ok? |
| 12:47.20 | ManxPower | ph8: Goto works like...Goto |
| 12:47.20 | ph8 | ok will do :) |
| 12:47.20 | Talnakh | [TK]D-Fender, the voice menus seems to be in gsm format |
| 12:47.20 | jblack | thanks. appreciated much. |
| 12:47.20 | [TK]D-Fender | Talnakh: and for the rest of what I asked? |
| 12:47.40 | jblack | puts down his "down with l's" poster |
| 12:47.49 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:48.08 | jblack | ? |
| 12:48.11 | Talnakh | i am listening to them from voIP phone and they were done by Digium |
| 12:48.16 | jblack | Did I offend him? |
| 12:48.55 | [TK]D-Fender | Talnakh: .. DETAILS.... |
| 12:49.00 | *** join/#asterisk arpu (n=arpu@chello084114025240.14.vie.surfer.at) |
| 12:50.48 | jblack | Real men use rotary. |
| 12:51.16 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 12:53.07 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 12:53.11 | Dr-Linux|home | hi |
| 12:53.16 | jblack | hi |
| 12:53.35 | Great_Anta_Baka | ih |
| 12:54.35 | Dr-Linux|home | my provide gave me SIP account which works fine on softphone, but i want to use same account on my asterisk server as sip trunk, my provider says you can use the same account |
| 12:55.30 | Great_Anta_Baka | JUST DO IT!!! |
| 12:55.48 | Dr-Linux|home | so my question is what's the parameter to do that? actually i registered is sucessfully on my asterisk server but problem is when i dial/debug i get: |
| 12:55.50 | jblack | Dr-Linux|home: You can. Btw, I'd avoid the term "sip trunk". It annoys some. |
| 12:55.56 | Dr-Linux|home | SIP/2.0 403 Forbidden |
| 12:56.24 | jblack | Dr-Linux|home: You'll set up a sip account in /etc/asterisk/sip.conf with a context to send it to, probably a register as well. There should be good examples in the book. |
| 12:56.28 | Great_Anta_Baka | on the phone? |
| 12:56.44 | Great_Anta_Baka | or the cli |
| 12:57.20 | Dr-Linux|home | on the CLI |
| 12:57.24 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 12:57.44 | Great_Anta_Baka | what is your dial command? |
| 12:58.07 | jblack | odds are he didn't register. |
| 12:58.53 | Dr-Linux|home | see here: |
| 12:58.54 | Dr-Linux|home | [Aug 27 17:58:34] NOTICE[18336]: chan_sip.c:12517 handle_response_peerpoke: Peer '0428315509' is now Reachable. (75ms / 2000ms) |
| 12:59.07 | Dr-Linux|home | It is registered but at the same time: |
| 12:59.16 | Dr-Linux|home | <--- SIP read from 10.64.1.13:5060 ---> |
| 12:59.16 | Dr-Linux|home | SIP/2.0 403 Forbidden |
| 12:59.26 | jblack | ok. throw your dialplan and your sip.conf (with passwords replaced with XXX) on pastebin.ca |
| 12:59.56 | Dr-Linux|home | jblack: i'll on phpfi.com |
| 12:59.57 | jblack | btw, that notice indicates you're using quality. That doesn't mean you're doing a register => |
| 13:00.26 | jblack | qualify, not quality |
| 13:00.26 | Dr-Linux|home | s/qualify/qualify? |
| 13:00.33 | Dr-Linux|home | oo ok |
| 13:02.34 | *** join/#asterisk r0land (n=r0land@212.36.208.1) |
| 13:02.37 | r0land | hello all |
| 13:02.43 | r0land | hi [TK]D-Fender |
| 13:02.59 | r0land | im facing a lil issue with sip to sip (softphone to softphone) calls |
| 13:03.15 | r0land | even though all peers are properly added in sip.conf with the appropriate extensions |
| 13:03.28 | r0land | and even though the softphones actually register with asterisk, and can make outgoing calls |
| 13:03.51 | r0land | but some peers cant recieve calls! the dialplan immediatly skips answering (considering peer to b offline) and jumps to voicemail |
| 13:04.06 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:04.10 | r0land | i double check sip.conf for context fault... or any other thing tht may b related |
| 13:04.36 | r0land | whts weird is tht i can call out from my softphone but cant recieve calls! |
| 13:05.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:05.33 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:05.36 | Great_Anta_Baka | not that weird |
| 13:05.52 | jblack | we can't cover softphone to softphone here, as asterisk isn't involved. |
| 13:05.52 | Dr-Linux|home | jblack: go there: http://phpfi.com/347728 |
| 13:05.54 | Great_Anta_Baka | had that quite a lot |
| 13:06.00 | jblack | SIR YES SIR! |
| 13:06.12 | jblack | I am at the site, SIR |
| 13:06.36 | jblack | Yeah. It's just like I said. You don't have a register. In fact, you don't even have a general section |
| 13:06.50 | r0land | jblack how is asterisk isnt involved! |
| 13:07.00 | jblack | r0land: YOu said softphone to softphone, right? |
| 13:07.02 | r0land | jblack both softphones are registerd to asterisk as their target domain |
| 13:07.12 | jblack | Did you mean softphone <-> asterisk <-> softphone? |
| 13:07.17 | r0land | yes |
| 13:07.22 | r0land | sorry didnt explain correctly |
| 13:07.28 | jblack | Ahh. gotcha then. |
| 13:07.32 | jblack | rereads more carefully |
| 13:07.44 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:07.56 | Great_Anta_Baka | what does sip show peers say |
| 13:08.08 | jblack | Ok. Throw up your sip.conf and your extensiosn.conf please. Make sure you tested after you did a reload of sip.conf and extensions.conf. Don't forget to replace your passwords with xxxxx |
| 13:08.51 | jblack | Dr-Linux|home: You catch what I said about the missing general section, and the missing register? |
| 13:08.57 | r0land | ok jblack |
| 13:08.57 | *** join/#asterisk zydoon (n=zydoon@41.225.156.60) |
| 13:09.04 | r0land | Great_Anta_Baka shows all peers |
| 13:09.22 | jblack | r0land: While you're at it, know how to do a sip debug? |
| 13:09.57 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:10.14 | Dr-Linux|home | jblack: my general section is just fine, but i didn't setup register stuff? :S |
| 13:10.28 | jblack | so, you didn't give me your entire sip.conf, and your entire extensions.conf, please? |
| 13:10.32 | Dr-Linux|home | jblack: can you tell me example how can i setup register line? |
| 13:10.42 | Great_Anta_Baka | r0land: what is your dial command? |
| 13:10.48 | Great_Anta_Baka | in extensions.conf |
| 13:11.07 | jblack | In the future, when someone asks for a file, please give the entire file. Register goes in the general section, and looks like register => user:pass@host |
| 13:11.13 | Great_Anta_Baka | waaaa.. i really want to buy coffee |
| 13:11.17 | *** part/#asterisk zydoon (n=zydoon@41.225.156.60) |
| 13:11.23 | Great_Anta_Baka | but no wallet :((((( |
| 13:11.26 | jblack | Great_Anta_Baka: You can come down here and have a cup. |
| 13:11.38 | Dr-Linux|home | jblack: ohh my extensions.conf is about 1800 lines, it's a production server, all is going good |
| 13:11.45 | Dr-Linux|home | context is not an issue |
| 13:11.48 | Great_Anta_Baka | you mean up... SA is at the bottom of the world |
| 13:11.54 | Dr-Linux|home | SIP port / codec is not an issue |
| 13:12.09 | Dr-Linux|home | jblack: but register => .......... i don't have this |
| 13:12.21 | Great_Anta_Baka | mebe you should |
| 13:12.26 | Great_Anta_Baka | ... |
| 13:12.31 | jblack | Dr-Linux|home: That's a good reason not to paste the whole thing. Regarding the resgister, you saw my instructions? |
| 13:13.03 | Dr-Linux|home | opss sorry you already instructed |
| 13:13.05 | Dr-Linux|home | thanks |
| 13:13.08 | jblack | welcome |
| 13:13.15 | r0land | jblack http://pastebin.com/d9544f5d :) |
| 13:13.22 | jblack | looking |
| 13:13.40 | jblack | wow that's long. |
| 13:14.02 | Dr-Linux|home | jblack: just asking for leraning purpose, what it does with the user >> register => user:pass@host ? |
| 13:14.14 | jblack | r0land: a word of advice? In the future, say "sure, I'll paste the whole thing, but it's quite long. You sure you want it?" |
| 13:14.22 | r0land | jblack 104, 181 facing probs |
| 13:14.27 | jblack | Dr-Linux|home: what does register do? It authenticates you to the server. |
| 13:14.34 | jblack | r0land: 104 and 181. Ok. |
| 13:14.39 | Dr-Linux|home | hhm... |
| 13:14.42 | jblack | What are these phones, btw? You said soft phones, right? |
| 13:14.42 | Dr-Linux|home | ok lemme put this |
| 13:15.01 | r0land | jblack sorry bout tht |
| 13:15.04 | r0land | jblack yep |
| 13:15.16 | jblack | r0land: My fault. You gave me exactly what I asked for. |
| 13:15.21 | r0land | jblack using an open source called outcall... and Eyebeam |
| 13:15.36 | jblack | r0land: If you have 200 working sip accounts, I'm inclined to blame the 2 that don't work. |
| 13:16.07 | r0land | jblack i agree which is why am here |
| 13:16.10 | jblack | are these two soft phones unique? Different software than the rest? |
| 13:16.21 | r0land | jblack on the contrary.. exactly the same as others |
| 13:16.38 | jblack | everything's the same, yet they're somehow different. Smells fishy. |
| 13:16.40 | r0land | jblack if i use any other peer on the same softphone/pc ! it works like a charm |
| 13:16.57 | jblack | How are these two different than the rest? Different network? Local firewall perhaps? |
| 13:17.12 | [TK]D-Fender | jblack>Dr-Linux|home: what does register do? It authenticates you to the server. <- nope. |
| 13:17.23 | jblack | [TK]D-Fender: Ok, what does it do? |
| 13:17.26 | r0land | jblack not at all all in the same network.. and as i said if i choose ANY other peer lets say 200! it owrks like a charm |
| 13:17.44 | jblack | r0land: My naive bet is a firewall is in the way. |
| 13:17.53 | Dr-Linux|home | i do sip authentication but i never used that before |
| 13:18.07 | [TK]D-Fender | jblack: It informs the server as to what IP address to send calls to. Does not have any inherent role in authing calls. |
| 13:18.15 | [TK]D-Fender | jblack: thats what your peer si for. |
| 13:18.35 | r0land | jblack but would it b possible for other peers to register?! on the same pc?! |
| 13:18.40 | [TK]D-Fender | Dr-Linux|home: Get off your ass and go emulate settings like those used by other ISP's |
| 13:18.56 | jblack | r0land: Sure. I'd put them on different ports, though. |
| 13:19.23 | Dr-Linux|home | :S |
| 13:19.30 | r0land | jblack i tried to troubleshoot.. tried 4 different peers on the same pc/softphone they all worked except those 2 |
| 13:19.40 | Dr-Linux|home | [TK]D-Fender: what's wrong? |
| 13:19.43 | jblack | [TK]D-Fender: Fair enough. Not using a registry line can give authentication problems though, since you're not where the server thinks. :) |
| 13:20.12 | Dr-Linux|home | [TK]D-Fender: why you always behave me like this? i appologize if i did something wrong |
| 13:20.19 | jblack | Which is r0land's problem here. Failure to authenticate. |
| 13:20.45 | [TK]D-Fender | jblack: I make no assumption as to what the server is thinking. I know ISP's where yuo need to give them an IP/host and you don't ever register to. |
| 13:20.47 | jblack | Dr-Linux|home: [TK]D-Fender Is just a busy person that tries to help a lot. Just be patient with his brusqueness, and he'll see you through. |
| 13:21.16 | [TK]D-Fender | jblack: He's been here for years and is simply clueless & lazy. |
| 13:21.25 | r0land | jblack the prob is! tht they DO authenticate.. (i think) since they can call out like every1 else... but they cannot recieve calls |
| 13:21.26 | Dr-Linux|home | If busy then don't answer my question this way |
| 13:21.29 | jblack | if he says that your problme isn't because you're not registering, then I'll stand with him, and say I don't know why it's not working for you. |
| 13:21.42 | jblack | r0land: Yeah. That definitely sounds like there's a firewall in the way. |
| 13:21.57 | [TK]D-Fender | jblack: He showed a 403, but to WHAT? We didn't even get to see that did we? |
| 13:22.02 | r0land | jblack oh ok gr8 then :) thanks for the help! |
| 13:22.23 | jblack | [TK]D-Fender: I'm on your side. I know my place is to stand back and just agree with you. :) |
| 13:22.58 | r0land | jblack one more quest.. exten =>_04,1,Dial(SIP/$(EXTEN)@306) <<-- if i dial 04, i get a tone! i tried changing "04" to "#4" but it doesnt work! any idea why ? |
| 13:23.05 | [TK]D-Fender | Dr-Linux|home: I'm available for those currently worth helping. |
| 13:23.15 | jblack | regarding the 403, it's been a while since he pasted the error, but if I remember correctly, on call initiation, he's getting a 403 that the call isn't authorized. |
| 13:23.19 | [TK]D-Fender | r0land: and I ansered your #4 question YESTERDAY. |
| 13:23.36 | r0land | [TK]D-Fender sorry didnt notice it |
| 13:24.22 | [TK]D-Fender | r0land: put "pedantic=yes" under [general] |
| 13:24.29 | r0land | Dr-Linux|home a peice of advice bout [TK]D-Fender... as much as hes for a lack of a better word! RUDE:P he does wht hes here for.. and tht to help ppl out! i for one always have appreciated hes advices though not his personality... though its not tht were both here to make friends! were here to get help/help eachother and thts exactly wht hes doing |
| 13:24.31 | Dr-Linux|home | jblack: but call is autherized from the softphone |
| 13:25.10 | r0land | [TK]D-Fender ok i will, need to change anything under the peers acounts?! |
| 13:25.33 | [TK]D-Fender | r0land: People expect to be spoon-fed and never get off their asses around here, and then when I ^#%$ing hand them the answer they don't follow it and aren't even paying attention. |
| 13:25.42 | jblack | wanders off for a bit |
| 13:25.46 | [TK]D-Fender | r0land: Do you like it when people waste your time? |
| 13:25.51 | r0land | [TK]D-Fender i agree as im one of them :) |
| 13:26.11 | r0land | [TK]D-Fender of course not! which is why i said i understood why u act in such a way and i never feel offended |
| 13:26.17 | r0land | on the contrary.. i always thank u for ur help |
| 13:26.50 | Dr-Linux|home | r0land: he always help good, I like him ... he is the one i respect much, but he is the one who always behave me like this .. even i don't ask him anymore for help. I also help alot of people here in my country what i know, but this is what if they are here and they say how i'm being treated |
| 13:27.03 | [TK]D-Fender | r0land: Learn from yesterdays example and split your contexts up. You have everything mashed into the crazy [spa] context still. |
| 13:27.48 | r0land | [TK]D-Fender ya still didnt figure out wht u did for me .. im trying to understand it b4 implementing it! so if i ever get an error to b able to know why it occured instead of looking for u to ask |
| 13:28.08 | r0land | [TK]D-Fender i saved it and trying to get my mind around it |
| 13:28.36 | *** join/#asterisk riddlebox (n=james@IP-16-235.wustl.edu) |
| 13:29.34 | Talnakh | [TK]D-Fender, found the issue with asterisk, slower pc didnt work well with ACPI and I just forced it in the kernel and voila |
| 13:29.37 | [TK]D-Fender | r0land: And your priorities are still a broken mess in your last pastebin. |
| 13:29.49 | [TK]D-Fender | Takapa: Congrats |
| 13:29.55 | [TK]D-Fender | Talnakh: rather |
| 13:29.59 | r0land | [TK]D-Fender k ill see wht i can do |
| 13:30.00 | Talnakh | thanks dude |
| 13:30.13 | r0land | in the meanwhile still getting my headaround the authentican advice u gave me |
| 13:30.37 | jblack | r0land: Try adding the register => line in general like I suggested. Tell me how it goes. |
| 13:30.58 | r0land | jblack k i will thanks :) |
| 13:31.07 | Andre101 | [TK]D-Fender: I don't know if you remember my issues with sip/nat, but it's fixed now.. Was the DD-WRT router |
| 13:31.31 | [TK]D-Fender | Andre101: as I hinted at. Found a suitable replacement? |
| 13:31.33 | jblack | It was kernel mustard, in the server room, with the firewall? |
| 13:31.40 | jblack | pardon, colonel mustard |
| 13:31.53 | jblack | wtf is "kernel" spelled colonel, anyways? |
| 13:32.05 | [TK]D-Fender | grabs a candle-stick & lead pipe and goes akimbo |
| 13:32.17 | *** join/#asterisk riddlebox (n=james@IP-119-142.wustl.edu) |
| 13:33.11 | r0land | [TK]D-Fender the "#" issue worked... thanks :) |
| 13:33.39 | Andre101 | [TK]D-Fender: Yep, using my linux box as the firewall/gateway now |
| 13:34.01 | jblack | looks up the pedantic thing |
| 13:35.00 | [TK]D-Fender | jblack: forces * to do a full decode on the inbound invite for special chars. |
| 13:35.22 | jblack | Got it. Thanks |
| 13:35.24 | [TK]D-Fender | jblack: # gets written in hex by a lot of devices |
| 13:35.39 | [TK]D-Fender | "%23" IIRC |
| 13:35.42 | jblack | I suppose it should, if it's an url |
| 13:35.47 | jblack | That makes sense. |
| 13:36.29 | Andre101 | [TK]D-Fender: What are your thoughts on the asterisk-gui? |
| 13:36.39 | jblack | seems like urldecode would be a more descriptive than "pedantic". |
| 13:36.51 | r0land | [TK]D-Fender one more q, i have 4 lines! if i call 01, to take al ine! and tht line was busy.. i want asterisk to immediatly jump to the next line till i can find one to dial out from! such an issue can b done with priorities?! like first line priority 1, 2nd line priority 2.. and so on.. ? |
| 13:36.54 | jblack | real men use bash. |
| 13:37.16 | *** join/#asterisk riddlebox (n=james@IP-119-142.wustl.edu) |
| 13:37.18 | jblack | r0land: That's exactly how to do it. |
| 13:37.18 | Andre101 | lol... just installed it, i dunno if i like it or not |
| 13:37.18 | shine | uh |
| 13:37.29 | r0land | jblack gr8 thanks :) |
| 13:37.42 | jblack | exten => out,1,Dial(first) exten => out,n,Dial(second) .... |
| 13:38.05 | jblack | shine: real women use bash too. |
| 13:38.17 | [TK]D-Fender | r0land: if your Dial fails it will go on to the next priority in yrou extension. |
| 13:38.29 | [TK]D-Fender | r0land: And jblack just wrote |
| 13:38.31 | r0land | gr8 :) |
| 13:38.32 | [TK]D-Fender | As* |
| 13:38.43 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 13:38.56 | r0land | oh ok ill set N then instead of specifying the priority.. |
| 13:39.00 | bad_duck | Hi |
| 13:39.13 | jblack | r0land: You can speicfy it if you want, but it's generally seen as poor form. |
| 13:39.23 | r0land | jblack i understand.. thanks :) |
| 13:39.34 | [TK]D-Fender | r0land: No, use hard-numbering. It will reinforce the importance of basic math. |
| 13:39.41 | r0land | lol |
| 13:39.59 | [TK]D-Fender | r0land: And believe me, you need it. |
| 13:40.06 | jblack | heh. "Just do what I say, damnit" and sarcasm, are incompatible. |
| 13:40.16 | r0land | Aug 27 16:39:44] NOTICE[2867]: rtp.c:1279 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' <<-- wht might b the cause of htis! |
| 13:40.31 | bad_duck | does anybody uses chan_mobile ? I would like to know if I can install asterisk 1.4 + asterisk-addons 1.4 for cdr_mysql and then the trunk version of asterisk-addons to install chan_mobile ? |
| 13:40.38 | jblack | never seen that one before. |
| 13:40.57 | [TK]D-Fender | jblack: feh |
| 13:41.43 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net) |
| 13:42.04 | LemensTS | is the gs-102/gs-200 poe capable? |
| 13:43.22 | r0land | jblack anyway i could know which peer talkign to which peer at one point of time! |
| 13:43.46 | drako | gs is capable to be throwed away |
| 13:44.49 | LemensTS | Someone ordered 65 of these and i am needing to make sure if they are poe capable or not, i dont see anywhere that points to that they are |
| 13:45.05 | LemensTS | ...just for hotel rooms... |
| 13:45.13 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 13:45.15 | jblack | r0land: show channels ? |
| 13:45.32 | [TK]D-Fender | LemensTS: www.grandstream.com |
| 13:45.37 | waverly360 | [TK]D-Fender: Hey man. awhile back you mentioned methods available in asterisk 1.4 that would allow me to restrict queues from sending calls to agents who were already on the phone. I haven't found anything better than the Chanisavail function thus far...could you point me in another helpful direction? :) |
| 13:45.48 | LemensTS | http://www.grandstream.com/consumerphones.html |
| 13:45.54 | [TK]D-Fender | waverly360: Asterisk 1.4 <- |
| 13:46.05 | LemensTS | yea im on there, i dont see where it is, thought i would get second opinion in here |
| 13:46.28 | waverly360 | [TK]D-Fender: yeah I know..I'm making the move...but I haven't found any documentation on 1.4 specifically that resolves that problem. |
| 13:46.56 | [TK]D-Fender | LemensTS: Because they AREN'T. |
| 13:47.01 | r0land | jblack thanksssss |
| 13:47.31 | [TK]D-Fender | waverly360: UPGRADE.TXT |
| 13:47.49 | [TK]D-Fender | waverly360: when I tell you the answer is there, it is. Don't drag on this... |
| 13:48.05 | waverly360 | [TK]D-Fender: you mean look at the docs in the actual tar file...now why didn't I think of that.. :P |
| 13:48.30 | waverly360 | [TK]D-Fender: I'll check it out. I keep expecting the answers to appear in google. |
| 13:48.36 | waverly360 | [TK]D-Fender: sorry. |
| 13:48.50 | *** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net) |
| 13:48.53 | *** join/#asterisk n3hxs (n=HAMming@c-71-57-230-234.hsd1.pa.comcast.net) |
| 13:49.00 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:51.04 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:51.10 | *** join/#asterisk mltlnx (n=mltlnx@96.232.20.139) |
| 13:51.27 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 13:58.10 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 13:58.56 | *** join/#asterisk Knightfa| (n=jjj@66.178.134.235) |
| 14:00.42 | Knightfa| | Can I get some direction on Asterisk Call Queues? I would like to transfer a call off to another PBX if the caller has been waiting for > 5 min. Can this or something similar be done? |
| 14:01.28 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-f4368e15bc275a39) |
| 14:01.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:01.31 | [TK]D-Fender | Knightfa|: "core show application queue" |
| 14:02.10 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 14:04.57 | Knightfa| | Alright |
| 14:05.11 | Knightfa| | So I think I understand a bit of what I saw there |
| 14:05.20 | Knightfa| | But Im not sure I am seeing what I want |
| 14:05.50 | Knightfa| | You have any good links that explain what Im looking for TK? |
| 14:06.39 | Knightfa| | What I currently have set up for my queue is a custom ext that dials a zap channel |
| 14:07.15 | Knightfa| | Problem is when the asterisk agents are on a call it dials the zap channel |
| 14:07.15 | bad_duck | is the trunk version of asterisk stable ?? (I would like to use chan_mobile and it seems easier with asterisk trunk) |
| 14:07.40 | Knightfa| | I know its supposed to do that but like I said I want the calls to wait in queue before routing our the zap |
| 14:07.55 | ph8 | ~book |
| 14:07.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:09.10 | Great_Anta_Baka | anyone here tried freeswitch? |
| 14:09.14 | [TK]D-Fender | Knightfa|: then your zap channel should not be part of the queue and instead something done after they LEAVE the queue |
| 14:10.01 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:11.06 | Knightfa| | TK: Im not sure Im following.. A.Customer enters queue B. all agents on call C. Customer in queue for >x time D.transfer customer out |
| 14:11.17 | *** part/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net) |
| 14:11.32 | Knightfa| | TK: The after they LEAVE is where ya lost me |
| 14:11.38 | C4away | Knightfal: it just goes to the next line in the dialplan |
| 14:12.19 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 14:12.21 | C4away | when the queue times out it just moves on |
| 14:12.56 | Great_Anta_Baka | this http://www.freeswitch.org/ |
| 14:13.25 | Knightfa| | When the queue times out? So you mean when there are no agents avail it just hits he next linein the dial plan? |
| 14:13.27 | C4away | Great_Anta_Baka: you will probably have better luck getting questions about freeswitch answered in #freeswitch |
| 14:13.35 | C4away | as freeswitch != asterisk |
| 14:13.55 | C4away | that's like asking in #linux if anyone has tried windows server 2003 |
| 14:13.55 | Great_Anta_Baka | i know |
| 14:14.00 | Great_Anta_Baka | just want to know if anyone |
| 14:14.04 | Great_Anta_Baka | here has tried it |
| 14:14.10 | Great_Anta_Baka | and why they still use * |
| 14:14.11 | [TK]D-Fender | Knightfa|: "core show application queue" <--------- |
| 14:14.37 | bradleyprice86 | Is there any way I can set my company asterisk system to allow me to call from home into the asterisk system and from there dial out long distance. Trying to setup a method for our managers to use this to keep long distances charges down while having conferences. |
| 14:14.39 | Knightfa| | TK I saw that |
| 14:14.58 | C4away | Knightfal: it is a QUEUE it will queue them until the timeout is reached |
| 14:15.07 | C4away | you set the timeout in the queues.conf file |
| 14:15.25 | Knightfa| | Ahhh |
| 14:15.28 | jblack | bradleyprice86: of course. |
| 14:15.46 | [TK]D-Fender | Knightfa|: Read. The. INSTRUCTIONS. |
| 14:15.48 | Knightfa| | Is the timeout per call? |
| 14:15.50 | bradleyprice86 | jblack: how would I go about setting that up? |
| 14:15.51 | jblack | bradleyprice86: Set them up with soft sip phones, or ip phones. |
| 14:16.02 | jblack | bradleyprice86: The same way you would any other sip account. |
| 14:16.05 | [TK]D-Fender | bradjust like any other call. |
| 14:16.05 | C4away | bradleyprice86: check out DISA or give them a phone at home |
| 14:16.17 | jblack | Or, have them dial into the office, enter a pin, then the number they want to call. |
| 14:16.18 | C4away | or have them take their phone home with them |
| 14:16.24 | padski | I think I've found my problem. the system has another sip trunk configured and we are querying dns for SRV? _sip._udp.sip.voipcheap.com. which is taking 20 secs to return. during that time asterisk is effectively blocked: it does not poll the fd that the sip connection is coming in on. |
| 14:16.27 | C4away | (if it is configured properly) |
| 14:17.03 | bradleyprice86 | I do know I can do that. One of the managers requested being able to use his regular house phone (cordless) so he wouldn't have to sit at his computer for 2 hours. |
| 14:17.08 | jblack | bradleyprice86: It's not very complicated for anyone that's been managing asterisk for any reasonable length of time. |
| 14:17.33 | jblack | Ok. So, have him dial into the office, set up an extention that reads a pin, then dials whatever number he punches in. |
| 14:17.47 | *** join/#asterisk zydoon (n=zydoon@41.225.155.169) |
| 14:17.59 | [TK]D-Fender | bradleyprice86: make an extension in your inbound processing that he can dial that will let him dial out. Hopefully with a PIN or something requested first. |
| 14:18.01 | *** part/#asterisk zydoon (n=zydoon@41.225.155.169) |
| 14:18.09 | [TK]D-Fender | bradleyprice86: DISA is a fine example |
| 14:18.29 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:19.53 | [TK]D-Fender | bradleyprice86: Or you can just shove an exten into your main IVR with a long coded prefix so people don't guess it or know its there. |
| 14:20.31 | [TK]D-Fender | bradleyprice86: exten => _8675309.,1,Dial(Zap/G1/${EXTEN:7}) |
| 14:21.10 | *** join/#asterisk angryuser (n=angryuse@95.250.146.195.dynamic.adsl.abo.nordnet.fr) |
| 14:21.33 | bradleyprice86 | Gotcha. I appreciate it. This gives me several instances of how I can set it up. |
| 14:22.06 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-76-250.vif.net) |
| 14:26.30 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:29.47 | *** join/#asterisk Dead_Assclown (n=frederic@modemcable124.92-203-24.mc.videotron.ca) |
| 14:29.52 | *** join/#asterisk Levonk (n=lk@adsl-75-62-133-9.dsl.lsan03.sbcglobal.net) |
| 14:31.35 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 14:36.46 | aliver | If I want to create a "generic" extension where anyone can call in, input _their_ extension, and then get their voicemail, will I need to setup an IVR for that or does Voicemailmain() have a way to do it? I didn't see anything along these lines when I did a "show application voicemailmain()" |
| 14:37.29 | lowtek | If you call voicemailmain it will do exactly what you're asking. |
| 14:37.40 | C4away | depending on contexts |
| 14:37.47 | lowtek | ++ |
| 14:37.53 | C4away | if they are all in the default voicemail context then yes |
| 14:38.11 | C4away | if you have a context set up in voicemail.conf for each "customer" then no |
| 14:38.19 | aliver | Okay, I'll try that. |
| 14:38.28 | aliver | Everyone in my setup is in the default voicemail.conf context. |
| 14:38.59 | C4away | is it a multi-customer setup? |
| 14:39.05 | C4away | or more of a corporate pbx? |
| 14:39.10 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-65f406a2fd3703d4) |
| 14:39.10 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:39.40 | C4away | oh |
| 14:39.41 | C4away | heh |
| 14:39.58 | C4away | I should actually read, I thought you said "everyone is in a different context" |
| 14:40.03 | C4away | default, got it |
| 14:40.08 | C4away | carry on |
| 14:41.01 | aliver | :-) |
| 14:41.25 | Knightfa| | C4away: so when I set the time out on the queue to say 5 min can I use the failover destination (in freepbx) to route them out |
| 14:41.41 | Knightfa| | Is that what that is for? |
| 14:41.54 | C4away | oh freepbx? |
| 14:41.55 | C4away | heh |
| 14:41.57 | Knightfa| | ya |
| 14:42.06 | Knightfa| | Im trying to just get the hang of this |
| 14:42.14 | C4away | you should really say that to begin with, since you can configure all of that crap without even needing to know how it works |
| 14:42.22 | C4away | yea |
| 14:42.24 | Knightfa| | Id like to know how |
| 14:42.26 | Knightfa| | :) |
| 14:42.34 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 14:42.38 | C4away | the timeout destitanion will send them somewhere after the timeout has expired |
| 14:42.47 | Knightfa| | gottcha |
| 14:43.13 | C4away | freepbx has such a convoluted dialplan it doesn't help to know how things work |
| 14:43.31 | C4away | I have been writing asterisk dialplans from scratch for almost 2 years and I still can't understand half of what freepbx is doing |
| 14:43.46 | bkw_ | haha |
| 14:43.49 | bkw_ | C4away: you're funny |
| 14:44.04 | C4away | lol, it's true though |
| 14:44.09 | padski | is there a way to check whether srvlookup is on ? |
| 14:44.37 | C4away | yea, check sip.conf and see if you have it enabled |
| 14:45.45 | C4away | also you should start getting messages every 30 seconds or so that say: some.host.net matched to _sip._udp.some.other.host.net |
| 14:45.57 | C4away | or something like that, I have seen them often enough I should know what they say by now |
| 14:46.07 | bkw_ | padski in sip.conf |
| 14:46.17 | bkw_ | doh C4away you beat me |
| 14:46.22 | C4away | lol |
| 14:46.34 | C4away | I do that sometimes |
| 14:46.49 | padski | bkw_, it is not in my sip.conf |
| 14:47.10 | C4away | are you running freepbx? |
| 14:47.31 | C4away | are you running asterisk 1.4 or later? |
| 14:47.40 | bkw_ | srvlookup=no |
| 14:47.42 | bkw_ | add that to general |
| 14:48.56 | padski | bkw_, yes, that helps, but sip.conf is not the current conf in the running asterisk. |
| 14:49.17 | C4away | huh? |
| 14:50.45 | *** join/#asterisk Whitor (n=Whitor@mail.audiovideocorp.com) |
| 14:51.04 | C4away | how is it that you are not using sip.conf? |
| 14:51.10 | C4away | are you not running sip at all? |
| 14:52.03 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 14:53.04 | padski | C4away, no I can add this to sip.conf. all I meant is that changing a setting this way is not the same as interogating the asterisk from the console. the contents of sip.conf might have been changed and not reloaded, or in this case may not yet contain a srvlookups line at all. |
| 14:54.03 | C4away | do you want to make sure it is, or is not running? |
| 14:54.22 | C4away | or do you want to know how to tell from the CLI? |
| 14:54.33 | ph8 | ~book |
| 14:54.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:55.17 | padski | C4away, from the console interface "asterisk -r" |
| 14:55.37 | C4away | I don't know if there is a way |
| 14:55.48 | *** join/#asterisk s0lid (n=s0lid@219.93.152.130) |
| 14:56.13 | C4away | if you want it on add srvlookup=yes / if you want it off add srvlookup=no ... then run: sip reload |
| 14:56.18 | C4away | that way you know if it is on or off |
| 14:56.54 | C4away | I don't know if there is a way to tell if they are enabled other than the fact that srv lookups will start to show on your screen if they are enabled and the hosts you are connecting/registering to are srv-enabled |
| 14:58.13 | padski | C4away, thanks. setting srvlookup=no will be enough :-) |
| 14:59.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:00.04 | riddlebox | can anyone tell me what to enter in for jbot to tell me about the centos zaptel compile bug? |
| 15:00.28 | [TK]D-Fender | riddlebox: read the TOPIC |
| 15:00.30 | Qwell | what it says in the topic |
| 15:00.32 | Qwell | ~centos52bug |
| 15:00.33 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 15:00.34 | riddlebox | crap sory |
| 15:00.36 | riddlebox | sorry |
| 15:04.50 | *** join/#asterisk Levonk (n=lk@76.237.13.164) |
| 15:05.01 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:07.01 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:07.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:07.18 | *** join/#asterisk Cresl1n (n=matt@nat/digium/x-4bfc976c2a3bf46a) |
| 15:07.19 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:07.51 | Qwell | bad_duck: I said last night that you shouldn't run trunk |
| 15:08.06 | bad_duck | :p |
| 15:08.24 | bad_duck | Qwell: i'm tring to make chan_mobile works with asterisk 1.4 now |
| 15:08.26 | bad_duck | will see |
| 15:09.57 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 15:11.08 | viraptor | hi - is there any way to report / drop channels that should pass rtp via asterisk, but stopped sending anything? |
| 15:11.14 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 15:11.36 | Qwell | viraptor: there is an rtp timeout config option in chan_sip.. sec |
| 15:11.54 | WimpMan | Yes. Wait 15Sek. |
| 15:11.57 | brodiem | I wish I could think of a practical use for chan_mobile, but just can't think of any reason why I would want to make my mobile phone immobile. |
| 15:12.05 | Qwell | yeah, it's just rtptimeout=<number of seconds> |
| 15:12.14 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 15:12.23 | viraptor | Qwell: cheers :) |
| 15:12.24 | Qwell | brodiem: presence, or to replace a landline |
| 15:12.38 | Qwell | or for easy sms stuff.. |
| 15:13.15 | brodiem | ah forgot about SMS support |
| 15:14.44 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 15:15.49 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 15:24.26 | *** join/#asterisk oej (n=olle@cust-IP-10.data.tre.se) |
| 15:28.14 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 15:28.48 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:30.11 | jeev | bambolayooooooooo, bambolaya |
| 15:35.57 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-b07a29a83b096499) |
| 15:36.21 | ph8 | what's the cheapest anyone's found SIP Trunks? |
| 15:36.28 | ph8 | I've got one for £3/mo, and a friend who can do them for £1.50/mo |
| 15:36.38 | ph8 | is that expensive? |
| 15:37.08 | ph8 | what kit do i need to provide my own SIP trunks? Some sort of agreement with the phone networks? |
| 15:37.24 | *** join/#asterisk IsUp (n=nocturne@88.255.77.200) |
| 15:37.29 | jeev | you mean a DID? i personally like lugosoft.com but you need to be a big minute user, they dont take little people |
| 15:37.38 | ph8 | a DID? |
| 15:37.55 | ph8 | I just googled a company that sold incoming SIP trunks :p |
| 15:38.03 | jeev | dunno |
| 15:38.15 | ph8 | what's DID stand for? |
| 15:38.26 | jblack | direct inward dialing |
| 15:38.26 | Nugget | ~DID |
| 15:38.27 | jbot | from memory, did is Direct Inward Dialing, or just a phone number |
| 15:38.35 | ph8 | ah then yes |
| 15:39.24 | *** join/#asterisk JenniferAkemi (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 15:39.30 | *** join/#asterisk JenniferAkemi- (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 15:39.52 | [TK]D-Fender | dribbles JenniferAkemi- and shoots for the 3-pointer |
| 15:40.03 | [TK]D-Fender | *swoosh* |
| 15:40.33 | *** join/#asterisk seanmh (i=seanmh@216.31.101.107) |
| 15:41.43 | jeev | heh |
| 15:42.27 | *** join/#asterisk dhill (n=dhill@ipv6.mindcry.org) |
| 15:42.35 | ph8 | what does everyone pay for their trunks then? Anyone know the PSTN-interfacing hardware i'd need to provide my own? |
| 15:42.38 | dhill | anyone use asterisk to dial outside world? |
| 15:43.08 | ph8 | dhill: I'm planning to |
| 15:43.18 | ph8 | and do you mean actual world, or just like local phone numbers |
| 15:43.26 | dhill | yes, 800 number |
| 15:44.04 | ph8 | i'm not an asterisk pro but i don't see why it wouldn't work if other numbers do, unless it's something to do with your outbound provider |
| 15:44.04 | dhill | there is a certain 800 number I call that asterisk doesn't do mark the state as UP.. even though it is working |
| 15:44.31 | dhill | wonder if anyone can reproduce? |
| 15:44.58 | jblack | few things in life are free. |
| 15:45.09 | *** join/#asterisk Rico29 (n=Rico@ARennes-257-1-166-80.w86-214.abo.wanadoo.fr) |
| 15:46.16 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 15:48.39 | *** join/#asterisk JenniferAkemi (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 15:48.42 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 15:50.33 | IsUp | anyone able to help about chanSS7? i am using Sangoma cards. |
| 15:50.58 | ph8 | Anyone ever clustered asterisk? Are there any issues? |
| 15:51.27 | lowtek | ph8: Ellaborate what you mean by cluster? |
| 15:51.28 | *** join/#asterisk mltlnx (n=mltlnx@cpe-68-175-59-236.nyc.res.rr.com) |
| 15:51.35 | IsUp | IAX or something else? |
| 15:51.37 | lowtek | fail-over or load-balancing? |
| 15:51.39 | ph8 | a highly scalable asterix solution |
| 15:51.51 | ph8 | so presumably a load balanced cluster |
| 15:52.01 | ph8 | say 2 machines distributing traffic to N machines behind them |
| 15:52.12 | jblack | ph8: You can do stuff like that, yes. |
| 15:52.22 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 15:52.31 | lowtek | You would likely use OpenSER as the front-end, asterisk on the back for that kind of solution. A better choice might be to build a federated network using dundi and cleaver dial-plans ... |
| 15:52.46 | jblack | At least with voip. You set the providers to fail over to one another, and the phones to try both * servers. |
| 15:53.02 | lowtek | cleaver=clever |
| 15:53.15 | ph8 | interesting |
| 15:53.18 | ph8 | just something i'm curious about |
| 15:53.50 | lowtek | The guys I referred you to do just that, their entire solution is federated, load balanced, automatically fails over, etc. |
| 15:54.16 | *** join/#asterisk tkbeat (n=tk@p54B968BD.dip.t-dialin.net) |
| 15:54.21 | tzafrir_laptop | IsUp, try asking your question, maybe? |
| 15:54.42 | tzafrir_laptop | IsUp, trunk has chan_dahdi |
| 15:54.50 | tzafrir_laptop | it only builds iwth dahdi |
| 15:55.02 | jblack | By the time you're loading a pair of machines down so much that they can't handle the load, you should have long since split the system up. |
| 15:55.07 | tzafrir_laptop | I have no idea if Sangoma has dahdi versions of its drivers yet |
| 15:55.20 | IsUp | I am using Sangoma A108 + asterisk 1.4 and chan_ss7 |
| 15:55.26 | IsUp | when i type wanrouter status |
| 15:55.37 | IsUp | it says "Connecting" for all ports |
| 15:55.40 | padski | ph8, the provider I'm working with has a bit of kit called the 'M6'. I'm guessing it may be one of these: http://www.broadsoft.com/m6/ |
| 15:55.46 | tzafrir_laptop | BTW: trunk has support for ss7 through chan_dahdi/libss7 |
| 15:56.00 | IsUp | is this means i got a psyhical problem or something about with drivers? |
| 15:56.26 | IsUp | when i type ss7 link status |
| 15:56.46 | IsUp | it says: NOT_ALIGNED, rx: 0 tx: 1/4, tx counting but theres nothing in rx |
| 15:57.20 | IsUp | and RED alarm: ON on some ports |
| 15:58.06 | IsUp | i tried so many combinations, changing spans, timeslots etc. |
| 15:58.07 | Cresl1n | IsUp: I could tell you about libss7 setup, but I don't know much about chan_ss7 |
| 15:58.30 | NoxIn- | IsUp: not aligned usually mean that you and the provider at the other end do not count the same way |
| 15:58.56 | IsUp | hmm, what about RED alarms and Connecting state? |
| 15:59.07 | IsUp | this means a problem with cables? |
| 15:59.21 | NoxIn- | for instance if you have 30 channels ... maybee for you it's 1-30 and 0 for signalisation, while the other ends thinks it's 0-14 , 16-30 and 15 used for signaling |
| 16:00.03 | IsUp | signalisation on 16 |
| 16:00.33 | IsUp | link 1: channels => 1-15,17-31 __ schannel => 16 |
| 16:00.57 | IsUp | link 2 ... link 8: channels => 1-15,17-31 __ schannel => (blank) |
| 16:01.17 | IsUp | in my zaptel.conf |
| 16:01.27 | IsUp | span=1,0,0,ccs,hdb3#,crc4 |
| 16:01.30 | IsUp | bchan=1-31 |
| 16:02.21 | Cresl1n | your zaptel.conf is wrong if it's in red alarm |
| 16:02.24 | Cresl1n | or your cable is not setup |
| 16:02.28 | Cresl1n | fix the red alarm first |
| 16:02.35 | IsUp | i should say |
| 16:02.49 | IsUp | i have another Sangoma cards working at another locations |
| 16:02.55 | IsUp | i am setupping a system to new telco |
| 16:03.06 | IsUp | i checked out other conf files from other systems |
| 16:03.11 | IsUp | they are working fine with it |
| 16:03.16 | dhill | can anyone use asterisk to call 1-800-205-5000? When the recording picks up, does core show channels show the state as Up? |
| 16:03.54 | IsUp | i think, it's about cables. coz i've tried so many configurations |
| 16:03.57 | lowtek | dhill: Yes. |
| 16:03.59 | dhill | cdr marks it as NO ANSWER.. even though it worked |
| 16:04.20 | dhill | lowtek: are you using zaptel or giving your call to another provider over sip? |
| 16:04.33 | angler | dhill, it maybe using early audio and thats why it is not up. usually after some time or pressing a DTMF on the IVR it will send a CONNECT if using PRI |
| 16:05.21 | dhill | i am giving my calls to both a 5300 which sends it out a PRI.. and to a sip provider.. both ways still show call as Down |
| 16:05.22 | dhill | weird |
| 16:05.24 | lowtek | dhill: over pri |
| 16:05.41 | *** join/#asterisk Levonk (n=lk@76.238.251.50) |
| 16:05.51 | lowtek | It's AT&T support or something |
| 16:06.04 | dhill | yes |
| 16:06.18 | dhill | going through the menus does not change the state |
| 16:06.35 | lowtek | If the same people that do support handle their internal phone systems I'm surprised asterisk doesn't core dump when you try to call them ... |
| 16:06.56 | dhill | :P |
| 16:06.58 | errr | heh |
| 16:07.39 | IsUp | also when i loads chan_ss7 it says |
| 16:07.41 | IsUp | config.c:836 load_config_host: Missing interface entries for host 'Sys1' |
| 16:08.14 | C4away | what exactly is chan_ss7 ? is it production ready? |
| 16:08.22 | IsUp | my server not using a local ip like 10.x.x.x or 192.x.x.x |
| 16:08.27 | C4away | is that related to the openSS7 project? |
| 16:08.36 | angler | dhill, you may need to execute the Answer app in your dial plan when calling that specific number if you have a problem with the call disconnecting after some time |
| 16:08.53 | Cresl1n | C4away: no, it's not related to OpenSS7 |
| 16:09.05 | C4away | then I must investigate this |
| 16:09.11 | Cresl1n | I know a lot of people that have been using libss7 and chan_ss7 in production environments |
| 16:09.16 | dhill | it doesn't disconnect.. it is a normal call.. asterisk just never marks it as being "Up".. keeps it in Down/Ring |
| 16:09.27 | Qwell | chan_ss7? |
| 16:09.35 | C4away | where is this module documented? |
| 16:09.56 | IsUp | i am using chan_ss7 on my all systems :> |
| 16:09.57 | lowtek | To be clear, my call went from OpenSER -> asterisk 1.4 -> Plexus 9000 -> pri |
| 16:10.04 | Cresl1n | libss7/chan_dahdi is documented in the libss7 README file |
| 16:10.11 | angler | dhill, turn on pri debug on your Asterisk system and see if the far end ever sends a CONNECT |
| 16:10.19 | Cresl1n | and in the ss7.txt in the documentation directory in Asterisk |
| 16:10.30 | C4away | our ss7 equipment is having problems with some switches and their support is clueless as to why the calls are failing and unable to come up with a proper solution |
| 16:10.50 | dhill | angler: i am not using pri |
| 16:11.05 | C4away | opensource is always nice |
| 16:11.08 | Cresl1n | well, if you use libss7 and a Digium card, you can feel free to bug me all you want about your SS7 problems :-) |
| 16:11.09 | angler | dhill, oh... then what? |
| 16:11.15 | dhill | sip |
| 16:11.19 | C4away | is libss7 provided by digium? |
| 16:11.22 | Cresl1n | Yeah |
| 16:11.24 | Cresl1n | I wrote it |
| 16:11.35 | dhill | i have tried sip .. one to a sip provider.. and one to a cisco 5300 which puts it on a PRI |
| 16:11.38 | C4away | nice |
| 16:11.47 | angler | dhill, then check the sip debug for a 200 OK msg |
| 16:12.12 | C4away | Cresl1n: can I PM you briefly? |
| 16:12.16 | Cresl1n | C4away: Sure |
| 16:12.25 | Cresl1n | you should probably join the asterisk-ss7 mailing list too |
| 16:12.37 | dhill | angler: just INVITE |
| 16:12.38 | C4away | I will now that I know it exists =) |
| 16:13.03 | rwaite | okay. asterisk and echo. what are the best resources i can read to figure out where echo is coming from and how to eliminate it? |
| 16:13.20 | rwaite | i've been trying to fix this for a few weeks now by trial and error and i'm getting nowhere |
| 16:13.21 | angler | dhill, there should be more msgs than just INVITE |
| 16:14.39 | dhill | and ACK |
| 16:14.48 | dhill | CANCEL when i hangup |
| 16:16.44 | angler | dhill, so thats the issue, the sip device that Asterisk sends the call to never sends the 200 OK msg indicating that the far end answered |
| 16:17.14 | angler | dhill, which I imagine is due to early audio |
| 16:18.02 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 16:18.52 | dhill | ok |
| 16:19.03 | NoxIn- | IsUp if it's a new telco maybe the configuration is not the same, but you can at least test your config by putting a physical loop |
| 16:20.50 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.14) |
| 16:22.00 | *** part/#asterisk dhill (n=dhill@ipv6.mindcry.org) |
| 16:23.21 | *** join/#asterisk JenniferAkemi (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 16:23.24 | *** join/#asterisk JenniferAkemi- (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 16:24.02 | *** join/#asterisk Ziaeon (n=Ziaeon@75.149.177.2) |
| 16:24.31 | *** join/#asterisk JenniferAkemi- (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 16:28.29 | *** join/#asterisk r0land (n=r0land@212.36.208.1) |
| 16:28.32 | r0land | hello all |
| 16:28.45 | r0land | hi [TK]D-Fender |
| 16:29.31 | r0land | may some1 guide me to a wiki or something tht explains how to turn on logging for users calls! in other words, i want to know how many minutes each user spoke to wht numbers on a weekly basis |
| 16:30.49 | Ziaeon | Hey, how would I go about receiving calls from one trunk (friend/peer), and sending it to my dialplan before it goes to an outbound trunk? I was mucking about with custom dest that would Goto (outbound-allroutes) but I don't think I'm going the right path. |
| 16:31.37 | IsUp | r0land check CDR |
| 16:31.47 | r0land | IsUp abrv of wht?! |
| 16:32.15 | [TK]D-Fender | ~cdr |
| 16:32.16 | jbot | extra, extra, read all about it, cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 16:32.27 | [TK]D-Fender | r0land: Chapter 13 of the BOOK. |
| 16:33.02 | r0land | oh ok thanks [TK]D-Fender ill get to it :) |
| 16:33.33 | [TK]D-Fender | Ziaeon: Calls ALWAYS go through your dialplan. calls do not come right in one device and out another. |
| 16:33.54 | pids | any clue as to why my dtmf tones show as 0ms? |
| 16:35.00 | heedly | Ziaeon: sounds right to me. |
| 16:35.06 | heedly | What do you think you're doing wrong? |
| 16:35.30 | Ziaeon | It's actually a pri that I have connected to my provider |
| 16:35.41 | Ziaeon | I want any calls coming from that pri to go to my asterisk box first, so I can keep a CDR |
| 16:35.52 | Ziaeon | otherwise it goes straight from the PRI to my provider and I cant keep track of their calls. |
| 16:36.07 | [TK]D-Fender | Ziaeon: Then plug your PRI into an interface that talks to * |
| 16:36.14 | Ziaeon | I already have incoming calls going to my asterisk box, then out to the pri, works great, thanks btw [TK]D-Fender |
| 16:36.24 | *** join/#asterisk RandalSchwartz (i=merlyn@p3m/member/merlyn) |
| 16:37.30 | Ziaeon | [TK]D-Fender: it is, it's on an IAD |
| 16:38.06 | Ziaeon | and we have a trunk between the IAD and the asterisk |
| 16:38.07 | [TK]D-Fender | Ziaeon: Well calls coming into * don't just go out another device. Everything passes through the dialplan. |
| 16:38.17 | RandalSchwartz | Interviewing Mark Spencer on live.twit.tv at 2pm Pacific Time today... I'll be watching here for questions during the interview. |
| 16:38.26 | RandalSchwartz | (for FLOSS Weekly - twit.tv/floss) |
| 16:41.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:42.01 | rwaite | does anyone here have experience with SoftEcho? |
| 16:42.33 | *** join/#asterisk sant0sk1 (n=sant0sk1@wsip-68-15-235-89.om.om.cox.net) |
| 16:43.03 | *** join/#asterisk max_______ (n=mochoa@chi-pat.cashnetusa.com) |
| 16:43.13 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 16:44.23 | *** join/#asterisk eric256 (n=ejh@72.1.122.62) |
| 16:46.54 | eric256 | hello |
| 16:47.00 | *** part/#asterisk Ziaeon (n=Ziaeon@75.149.177.2) |
| 16:47.55 | eric256 | any ideas if i could geta n inter-tell 5000 to chat with Astrisk over SIP trunks? |
| 16:47.55 | [TK]D-Fender | eric256: Don't see why not. Go ahead and try |
| 16:48.57 | eric256 | if i only have phone services on the Inter-tel can the Asterisk use that? i'm new to the whole voip thing especialy the asterisk side, so any tips are appreciated |
| 16:49.38 | [TK]D-Fender | eric256: "phone services"? Could please try to be a little more descriptive. Maybe even a lot... |
| 16:50.11 | eric256 | lol sorry, access to the outside world.. in this case a T1 and a PRI (one for long distance and one for local) |
| 16:50.32 | eric256 | but i want to be able to setup remote IP phones to chat with asterisk and use those lines. |
| 16:50.53 | hi365 | anyone using email-to-fax with hylafax? Im looking for suggestions for an imap email client... (to use with google apps) |
| 16:50.58 | eric256 | idealy i just want simultaneous ring so that an outside number can ring an IP phone and my blackberry |
| 16:51.40 | [TK]D-Fender | eric256: all depends on what the InterTel lets you do. |
| 16:52.24 | eric256 | [TK]D-Fender: yea thats why i'm trying to integrate with Astrisk, figure if i can get it working i can probably get alot more features than through intertel |
| 16:53.44 | IsUp | i am getting error: "Operation not permitted" reading key "net.ipv6.route.flush" on Sangoma install |
| 16:53.47 | eric256 | is currently reading tons of documentation on the asterisk website ;) and hoping this will all work in the end. if not it looks like i might be able to re-use intertel phones connected directly to *, that would be interesting |
| 16:54.00 | IsUp | cat: /etc/inittab: No such file or directory - Warning: Failed to determine init boot level, assuming 3 |
| 16:54.02 | IsUp | any ideas? |
| 16:54.30 | [TK]D-Fender | eric256: If they are digital sets, then no, they will not be reusable |
| 16:54.56 | eric256 | they arn't the 5000 is all IP |
| 16:55.24 | [TK]D-Fender | IsUp: pastebin the attempt again.... |
| 16:55.48 | [TK]D-Fender | eric256: If they work with SIP and aren't locked, sure. |
| 16:56.31 | NoxIn- | hi365: I tried mail2fax from smtp, but not from imap , you mean a solution like a process check if there is something in an imap mailbox and then get it to pass it to hylafax ? |
| 16:57.18 | hi365 | NoxIn-: yes. using the imap idle command, the delivery will be in real time |
| 16:59.02 | NoxIn- | hi365: I would use something like fetchmail to get the mail and feed it to a local script that decode the needed part (fax number, document, ...) and passe it to hylafax |
| 16:59.27 | hi365 | NoxIn-: does fetchmail supprot imap, idel and ssl? |
| 17:00.11 | NoxIn- | imap and ssl yes |
| 17:00.15 | NoxIn- | idle I don't know |
| 17:00.57 | NoxIn- | hi365: yes it can (just checked on the website) |
| 17:01.01 | WimpMan | Let's see if I have more success at this time of day. Anyone awake with dome dundi experience? I can only get it working for uniqe numbers, but not for wildcarded dialplan entries. Any hints |
| 17:01.08 | hi365 | NoxIn-: seems like yes, http://fetchmail.berlios.de/fetchmail-features.html "Fetchmail can optionally use the RFC 2177 IDLE extension on an IMAP server that supports it." |
| 17:01.09 | WimpMan | appreciated. |
| 17:01.34 | hi365 | NoxIn-: your quicker! |
| 17:01.43 | hi365 | hands NoxIn- a cookie |
| 17:01.50 | NoxIn- | :) |
| 17:01.54 | hi365 | thanks! now for the configuring.... |
| 17:01.54 | rwaite | maybe i am confused. this softecho echo canceller. it installs as an echo canceller in the zaptel lib. does that mean that it will only apply to call going thru the zap card? |
| 17:02.28 | rwaite | i am experiencing echo/bg noise on sip->iax2(provider)->pstn |
| 17:02.44 | rwaite | so would this not help me at all since im not going through a zap card for the call? |
| 17:03.04 | hi365 | NoxIn-: have you actualy done it befor? |
| 17:03.09 | tzafrir_laptop | rwaite, no, it won't help you. |
| 17:03.19 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:03.32 | tzafrir_laptop | Get it installed on the IAX->PSTN provider ;-) |
| 17:03.57 | rwaite | :( what options do i have to handle echo/bg noise? |
| 17:04.11 | WimpMan | Change provider. |
| 17:04.57 | NoxIn- | hi365: imap->hylafax->asterisk no, smtp->hylafax->asterisk we are trying but still having some problems, asterisk->hylafax->smtp works |
| 17:05.02 | *** join/#asterisk mattx86 (n=matt@static2073.uctnwd.ken-tennwireless.com) |
| 17:05.08 | NoxIn- | (we = the compagny I work for) |
| 17:05.17 | hi365 | NoxIn-: what kind of issues? |
| 17:05.41 | hi365 | and what command do you use to run your script after your recive the email? |
| 17:05.41 | eric256 | does asterisk include a softphone? or do you have to buy those? |
| 17:05.45 | NoxIn- | encoding issue, the fax received is garbage |
| 17:06.03 | hi365 | eric256: there are pleant that can be downloaded for free |
| 17:06.32 | mattx86 | just curious.. does anyone else use Voxitas/NetLogic with IAX2? I think they're having problems right now. |
| 17:07.13 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:07.28 | NoxIn- | hi365: a shell script provided by the hylafax package mail2fax.sh |
| 17:07.51 | hi365 | NoxIn-: yeh, but how do you tell fetchmail to pass the email to that script? |
| 17:08.40 | NoxIn- | hi365: I don't use fetchmail because I don't get mails from a mailbox, I receive the mail on a postfix using SMTP |
| 17:08.45 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 17:08.55 | NoxIn- | (hi365: by the way we are a little offtopic here) |
| 17:08.56 | exothermc | How does DTMF=auto work? |
| 17:09.38 | exothermc | errr dtmfmode=auto |
| 17:10.17 | exothermc | oh nm it does rfc2833 and fails back to tones |
| 17:10.31 | exothermc | or inband |
| 17:11.16 | *** join/#asterisk zwt (n=test@dsl51B644AC.pool.t-online.hu) |
| 17:11.19 | zwt | hey all |
| 17:11.37 | zwt | I have just installed asterisk with asterisk gui but after I logged in I have no option |
| 17:12.11 | [TK]D-Fender | zwt: #asterisk-gui <----- |
| 17:12.13 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 17:12.28 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:12.53 | zwt | oh |
| 17:12.54 | zwt | sry |
| 17:12.55 | Whitor | Is that x based or curses based ? |
| 17:13.14 | [TK]D-Fender | Whitor: WEB <- |
| 17:13.19 | Whitor | is using astriasknow |
| 17:13.31 | Whitor | [TK]D-Fender, thanks |
| 17:13.42 | jameswf-home | ~asterisknow |
| 17:13.42 | jbot | from memory, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 17:13.48 | [TK]D-Fender | Whitor: thats the GUI that *Now uses |
| 17:14.09 | Whitor | not looknig for support for asterisk now :) ... just trying to learn stuff about the 'real' asterisk |
| 17:14.17 | jameswf-home | rumor has it *NOW can use freepbx |
| 17:14.18 | zwt | but |
| 17:14.24 | Whitor | OH !! Thats really good to know [TK]D-Fender |
| 17:14.27 | zwt | can I install asterisknow on a vps? |
| 17:14.43 | [TK]D-Fender | zwt: Its a distro <- |
| 17:14.44 | Whitor | Asterisknow has its own channel |
| 17:14.57 | [TK]D-Fender | zwt: Pretty much like any other. |
| 17:15.04 | zwt | [TK]D-Fender: hm |
| 17:15.06 | jameswf-home | ~book |
| 17:15.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:15.09 | jameswf-home | ^^^^^All you need to know about Asterisk ^^^^^^^^^^^^^^ |
| 17:15.18 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 17:15.30 | zwt | yes but maybe anybody can give me a howto install asterisknow on a vps? |
| 17:15.31 | Whitor | Nice link Thansks jameswf-home |
| 17:15.45 | Whitor | What is a vps ? |
| 17:15.46 | jameswf-home | zwt #asterisknow |
| 17:15.49 | [TK]D-Fender | zwt: this isn't #vps |
| 17:15.52 | jameswf-home | virtual server |
| 17:15.54 | zwt | pff |
| 17:16.03 | jameswf-home | ~pff |
| 17:16.04 | *** join/#asterisk _jeff_o (n=jeff@c-24-63-95-52.hsd1.ma.comcast.net) |
| 17:16.15 | Whitor | you will be limited with the hardware you can interface with in a Virtual environment |
| 17:16.27 | Whitor | though, you could run a strictly sip and IP based server |
| 17:16.40 | zwt | I see |
| 17:16.43 | [TK]D-Fender | zwt: Does a car salesman have to teach you how to DRIVE as well? |
| 17:16.47 | jameswf-home | i use virtual box for devel |
| 17:17.07 | zwt | [TK]D-Fender: yes of course without that I wont buy that fucking card |
| 17:17.07 | zwt | lol |
| 17:17.10 | jameswf-home | [TK]D-Fender: depends does he want the sale |
| 17:17.23 | rwaite | well, ulaw sounds a hell of a lot better than g729 |
| 17:17.34 | [TK]D-Fender | rwaite: ... DUH <- |
| 17:17.38 | *** join/#asterisk jpcansa (n=jpbenavi@190.10.2.87) |
| 17:17.47 | rwaite | lol |
| 17:17.49 | [TK]D-Fender | jameswf-home: ok, fine, sure. |
| 17:17.57 | rwaite | but the background noise almost sounds ... louder |
| 17:18.05 | _jeff_o | Hi, I'm having a problem with echo. I'm using ekiga to make calls through an asterisk computer which sends outgoing call through voicepulse.com. When I call somebody from an SIP and they're using a cell phone the get an echo (I haven't tested yet with a landline). I don't get an echo on my end. Is there a solution for this? |
| 17:18.16 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 17:18.26 | jameswf-home | ~ekiga |
| 17:18.26 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 17:18.26 | [TK]D-Fender | _jeff_o: No. |
| 17:18.48 | [TK]D-Fender | jameswf-home: How crap cut it out with the instant bot-sayings for everything... |
| 17:18.49 | aliver | Do I have to do anything special to enable music-on-hold? |
| 17:18.52 | [TK]D-Fender | holy* |
| 17:18.56 | _jeff_o | [TK]D-Fender, Is it my setup or a problem with cell phones? |
| 17:19.14 | [TK]D-Fender | _jeff_o: you are in a pure SIP environment. Its the fault of your endpoints. |
| 17:19.14 | aliver | besides put someone on hold, that is. |
| 17:19.29 | jameswf-home | [TK]D-Fender: I thought ekiga was a "Womans excersize" I was curious |
| 17:19.35 | [TK]D-Fender | aliver: configure your class, have sounds for it, and put someone on hold. |
| 17:19.42 | Whitor | What is a good source for FXO cards and and FXS gateway devices ? |
| 17:19.49 | _jeff_o | [TK]D-Fender, I can't use a software echo canceler? |
| 17:19.56 | Whitor | jameswf-home, no, that keegles, and its for men too :) |
| 17:20.01 | [TK]D-Fender | Whitor: www.telephonydepot.com |
| 17:20.11 | jameswf-home | voipsupply.com too |
| 17:20.11 | [TK]D-Fender | _jeff_o: No EC for VoIP |
| 17:20.15 | Whitor | thx [TK]D-Fender |
| 17:20.34 | jameswf-home | on VOIP its not echo its delay |
| 17:21.00 | [TK]D-Fender | _jeff_o: Ekiga has been known to cause delay all by itself. test another soft-phone |
| 17:21.19 | jameswf-home | i use moziax nice small simple |
| 17:21.27 | _jeff_o | [TK]D-Fender, ok, any recommendation? |
| 17:21.46 | [TK]D-Fender | _jeff_o: Zoiper / X-Lite |
| 17:22.43 | _jeff_o | [TK]D-Fender, ok, so my asterisk setup should theoretically work? |
| 17:23.25 | [TK]D-Fender | _jeff_o: * plays no part in your problem. |
| 17:23.36 | jameswf-home | so pbx in a flash bumped their kernel to 5.2 and didnt account for the xpp bug. :)) |
| 17:23.55 | jameswf-home | s/kernel/centos/ |
| 17:23.58 | [TK]D-Fender | jameswf-home: 5.2 Kernel? Cool... What OS is that for? :) |
| 17:24.07 | _jeff_o | [TK]D-Fender, ok, so what I'm trying to do should be possible without echo? |
| 17:24.13 | jameswf-home | [TK]D-Fender: its beta |
| 17:24.35 | [TK]D-Fender | _jeff_o: it is the fault of the ENDPOINTS involved. |
| 17:24.52 | _jeff_o | [TK]D-Fender, ok |
| 17:25.14 | eric256 | laughs at the bot correcting people, lmao |
| 17:25.52 | beek | thinks its hilarious too. |
| 17:26.08 | Whitor | Is this the card I would need for interfacing with an analog line(s) from my PBX (with ring gen). http://www.telephonydepot.com/product_p/105-050-tdm410p.htm Currently I am using an x100p and I would like to support more lines ... (with better quality too) |
| 17:26.50 | [TK]D-Fender | Whitor: Digium TDM410P or Sangoma A200d |
| 17:27.22 | jameswf-home | cough or Rhino cough |
| 17:29.54 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:30.13 | Whitor | nice! the 410p is what I'm looking at ... I just like to confirm before I go spending my companies money. on the wrong stuff. thanks ! |
| 17:30.27 | *** join/#asterisk angryuser (n=sldf@95.250.146.195.dynamic.adsl.abo.nordnet.fr) |
| 17:30.34 | Whitor | I havn't used any Sangoma products yet ... |
| 17:30.39 | angryuser | fender you here ? |
| 17:30.44 | Whitor | Diginetworks has been good to me |
| 17:30.45 | [TK]D-Fender | Whitor: I use them exclusively. |
| 17:30.53 | Whitor | mmm, good to know |
| 17:32.16 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 17:32.37 | Whitor | Wow! sangoma cards are like 4x the cost of a digi networks equivilant !! |
| 17:32.48 | Whitor | I hope their performance is that much better too :) |
| 17:32.59 | jameswf-home | yeah 4X the cost probably same support :) |
| 17:33.09 | *** join/#asterisk lowtek (n=anonymou@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:33.14 | [TK]D-Fender | Whitor: link please... |
| 17:33.23 | Whitor | doesn't mind spending money when you get better performance |
| 17:33.37 | Whitor | http://www.telephonydepot.com/SearchResults.asp?Search=sangoma+a200d&Search.x=0&Search.y=0 |
| 17:33.40 | Whitor | top card |
| 17:34.04 | Whitor | vs: http://www.telephonydepot.com/product_p/105-050-tdm410p.htm |
| 17:34.06 | [TK]D-Fender | Whitor: I KNOW the Sangoma's pricing. Link me to this "other" card. |
| 17:34.14 | Whitor | gotcha |
| 17:34.16 | jameswf-home | Whitor: price at voipsupply.com as well they are a huge distributor of everyone may be cheaper |
| 17:34.47 | [TK]D-Fender | Whitor: You didn't compare LIKE SPEC'd cards. The TDM410P didn't have the HWEC module included by default <- |
| 17:34.52 | angryuser | i have installed a new provider yesterday, i dont need to incoming call that much, but in the matter of test tryed to set them, and i got this when someone calling in http://pastebin.ca/1186549 (outgoing are fine) |
| 17:34.57 | [TK]D-Fender | Whitor: pay attention |
| 17:35.32 | angryuser | i dont need incoming working that much * ;)) |
| 17:36.03 | angryuser | i did unsecure very, without |
| 17:36.47 | angryuser | any effect, also insecure=yes, same result, auth mismatch |
| 17:36.59 | [TK]D-Fender | Whitor: and that's DIGIUM, not "digi networks" |
| 17:39.55 | jaytee | angryuser, are you running * 1.4 |
| 17:40.21 | angryuser | jaytee yes |
| 17:40.22 | jaytee | try insecure=port,invite |
| 17:41.05 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:41.34 | angryuser | jaytee same result, same string |
| 17:41.35 | Whitor | dang . |
| 17:41.42 | Whitor | <=- newb, obviously |
| 17:41.56 | Whitor | sorry, boss just walked in |
| 17:42.59 | angryuser | there is something with the string also * take the calling person's number as user ... |
| 17:43.32 | angryuser | going to goggle more |
| 17:43.45 | oilinki3 | hi, |
| 17:44.27 | oilinki3 | how can I put realm to the realtime peer? |
| 17:45.02 | *** join/#asterisk ShaunWing (n=chatzill@dsl-241-200-131.telkomadsl.co.za) |
| 17:46.48 | oilinki3 | eg. I try to call out via another proxy, but asterisk/phone tries to authenticate with my default realm |
| 17:46.59 | ShaunWing | say, I'm getting very confused. I've got a PAPP2T registering via NAT to a remote asterisk server. The PAP2T is configured with SIP port 5060. On the Asterisk server it shows 919/919 192.168.10.215 D N 15227 UNREACHABLE. Firstly why PORT 15227 and not 5060 and secondly why the unreachable? please help..... |
| 17:49.00 | ShaunWing | On top of it all I get one way speach!!!! |
| 17:49.51 | *** join/#asterisk ElSonico (n=tav@213-157-93-129.localloop.fi) |
| 17:50.24 | angryuser | ShaunWing it is remote port |
| 17:50.25 | WimpMan | SIP doesn't like NAT. STUN might help. |
| 17:51.06 | ShaunWing | Are there any disadvanteages to STUN? |
| 17:51.16 | lowtek | SIP likes NAT just fine ... |
| 17:51.26 | angryuser | ùy config is working kust fine with nat |
| 17:51.28 | *** join/#asterisk dlynes (n=daniel@d206-116-189-12.bchsia.telus.net) |
| 17:51.32 | ShaunWing | Thats what I thought |
| 17:51.51 | ShaunWing | And I've been using it fine for ages in this setup and now this problem |
| 17:51.54 | [TK]D-Fender | ShaunWing: You didn't configure * properly to operate from behind NAT. Go read the guide : |
| 17:51.56 | [TK]D-Fender | ~sipnat |
| 17:51.57 | 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 |
| 17:51.58 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 17:52.12 | [TK]D-Fender | WimpMan: * works jsut fine when you configure it properly. |
| 17:52.13 | TJNII | Hmmm... A basic Qwest home phone line is on-par with price for some of these ITSPs... |
| 17:52.31 | [TK]D-Fender | TJNII: "basic".... |
| 17:52.50 | TJNII | Yes, basic |
| 17:52.59 | TJNII | As in not payinf for features I get from * |
| 17:53.06 | WimpMan | [TK]D-Fender: Well if you're on your way across the internet it's quite a bot hit or miss. |
| 17:53.16 | WimpMan | bit |
| 17:53.17 | TJNII | Only on call at a time, though, but that's a givven for POTS. |
| 17:53.20 | [TK]D-Fender | TJNII: And jsut about every "residential" serive allows you 2 channels (3-way calling etc). Through * those channels are INDEPENDANT and is like having 2 lines as it is SIMULTANEOUS. |
| 17:53.23 | ShaunWing | tx. But why the funny port no? |
| 17:53.40 | [TK]D-Fender | WimpMan: What about across the street internet? :) |
| 17:53.56 | [TK]D-Fender | shauTake a networking 101 class. This is perfectly normal. |
| 17:54.29 | [TK]D-Fender | WimpMan: When do you NAT that ISN'T concerning the "internet" anyways? |
| 17:54.58 | *** join/#asterisk Levonk (n=lk@adsl-76-238-250-65.dsl.lsan03.sbcglobal.net) |
| 18:03.33 | *** join/#asterisk ShaunWing (n=chatzill@dsl-241-200-131.telkomadsl.co.za) |
| 18:03.43 | ShaunWing | Say how can I kill an active call? |
| 18:04.11 | kaldemar | soft hangup in CLI |
| 18:05.56 | ShaunWing | tx but whats the command |
| 18:07.45 | kaldemar | i just gave it to you. |
| 18:07.52 | IsUp | just type "hangup" |
| 18:08.00 | IsUp | or "soft hangup" |
| 18:08.07 | IsUp | to see |
| 18:08.09 | ShaunWing | tx |
| 18:08.22 | [TK]D-Fender | ShaunWing: "soft hangup [channelwithoutbraces]" |
| 18:08.23 | kaldemar | IsUp: hangup only hangs up console channel. and it's deprecated in 1.4. |
| 18:08.33 | ShaunWing | tx |
| 18:08.37 | IsUp | yea, i see |
| 18:09.38 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:10.50 | Whitor | dang... sorry... weekly IT status update meeting with management |
| 18:11.20 | Whitor | Thanks for all the info [TK]D-Fender , I need some time to read up now. |
| 18:11.52 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 18:12.12 | pigpen | hi all, I have an urgent one: |
| 18:12.28 | pigpen | Our server was knocked down due to a power outage, now I am getting: |
| 18:12.29 | pigpen | [Aug 27 13:06:33] WARNING[5168]: chan_sip.c:1784 __sip_xmit: sip_xmit of 0x2cd0f8007b10 (len 507) to 192.168.100.182:5060 returned -2: Bad file descriptor |
| 18:12.33 | ShaunWing | Any ideas why in my sitiation I get a half a ring but no voice |
| 18:12.43 | pigpen | Any ideas??? |
| 18:13.14 | [TK]D-Fender | ShaunWing: You didn't configure * properly to operate from behind NAT. Go read the guide <----- |
| 18:13.35 | IsUp | NAT = Never Activate This |
| 18:13.40 | ShaunWing | tx, I'm reading the guide |
| 18:13.40 | Whitor | heh |
| 18:13.42 | IsUp | lol |
| 18:13.53 | angryuser | hm strange, check_auth: username mismatch, have <ovh2>, digest has <anonymous> , with unsecure=very or yes or whatever, same string, (incoming call) outgoing are fine, maybe i forgot something ? (i got type=friend in sip.conf) |
| 18:14.27 | pigpen | [TK]D-Fender, I try not to bug you directly, but have you ran across my issue noted above? |
| 18:17.21 | Whitor | I love support with a bitter edge. You learn so much more. |
| 18:18.10 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552046.dsl.bell.ca) |
| 18:18.34 | Whitor | pigpen, Have you tried bringing the system down gently an then back up ? |
| 18:18.35 | pigpen | [TK]D-Fender, nm, got it. |
| 18:18.43 | Whitor | pigpen, what was it ? |
| 18:19.09 | pigpen | best I could figure asterisk came up "dirty" if that is possible. |
| 18:19.19 | Whitor | so a reboot fixed it ? |
| 18:19.28 | pigpen | I knocked down all services on the box, and brought everything up. |
| 18:19.31 | Whitor | could be becasue it went down dirty |
| 18:19.34 | pigpen | yeah, probably equiv to a reboot. |
| 18:19.44 | pigpen | but since I am 200 miles away.... |
| 18:19.49 | Whitor | mmm |
| 18:20.01 | pigpen | first time for everything I guess... |
| 18:20.03 | Whitor | remote power strips ftw |
| 18:20.47 | Whitor | of course ... a remote power strip would bring the system down dirty too :) |
| 18:21.06 | Whitor | would need to be software initiated |
| 18:21.30 | Whitor | you did good - obviously by your result :) |
| 18:21.41 | pigpen | yeah, we just had a nice power outage for 3 hours.... |
| 18:21.55 | pigpen | but yeah, not everybody wants to buy some nice WTI's. |
| 18:22.02 | Whitor | too long for my UPS |
| 18:22.15 | pigpen | much better than APC's Network Power Switch imho. |
| 18:22.27 | *** join/#asterisk Atel (n=hary@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:22.50 | pigpen | Asterisk is pretty solid for things like this, but oh well, could have been worse. |
| 18:22.57 | pigpen | thanks for the moral support. |
| 18:24.24 | Whitor | he was a bad duck |
| 18:26.17 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 18:26.49 | sant0sk1 | anybody know the best way to troubleshoot this error? "Unable to handle indication 8 for 'SIP/209.62.1.2-08183018'" |
| 18:26.56 | *** join/#asterisk bbryant (n=brett@68.208.65.34) |
| 18:27.58 | Knightfa| | Any one have any reccomendations on what type of processor is best to process call for *... I have been reading since * is I/O that AMD is the way to go and that fewer cores is better. |
| 18:29.19 | Knightfa| | We are wanting to process ~ 200 calls both sip and zap |
| 18:30.02 | Atel | Knightfal there is one problem. MAKE sure that there is a supply of those CPUs in your area in the event yours does fail. |
| 18:30.13 | lowtek | We easily process 100+ calls on a quade-core AMD system w/2GB of ram and g729 transcoding, rarely goes about 20% total CPU |
| 18:30.14 | Atel | I have the opteron 940 and no one has them here. |
| 18:30.41 | lowtek | about=above |
| 18:31.02 | Knightfa| | http://www.newegg.com/Product/Product.aspx?Item=N82E16819105142 |
| 18:31.17 | Knightfa| | x2 |
| 18:40.02 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 18:46.57 | Atel | two much datacom gear need to audit. |
| 18:51.03 | jeev | is it possible to debug not an ip but a specific channel ? |
| 18:55.36 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 18:58.23 | Whitor | whats hte problem ? |
| 18:58.50 | Whitor | you can debug -anything- with enough time |
| 18:59.32 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 19:02.08 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 19:03.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:05.38 | [TK]D-Fender | jeev: peer / ip /all. |
| 19:08.12 | tvirus | I setup my * box with 2 users using their NAT IP. We are trying to use X-Lite to call each other but nothing happens. sip debug throws some SIP/2.0 404 and a ton of text. I can probably search to find a similar issue. Any idea why the users couldn't call anyone? We're on the same network 192.168.x.x and have no provider, just *. Is setting the user's host to their IP and not dynamic an issue? |
| 19:10.37 | kaldemar | tvirus: have you configured the extensions? |
| 19:11.09 | tvirus | Yeah |
| 19:11.20 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 19:11.22 | tvirus | I'm not at work, so I can't paste it, but let me fake it on rafb.net |
| 19:11.23 | tvirus | sec |
| 19:11.28 | kaldemar | well, pastebin your sip.conf and a trace of a call. |
| 19:13.13 | [TK]D-Fender | tvirus: Forget faking it. We need to see whats REALLY going on. |
| 19:13.14 | tvirus | http://rafb.net/p/0YDrI413.html |
| 19:13.21 | tvirus | =| |
| 19:13.34 | kaldemar | sounds like a dialplan problem anyway. |
| 19:13.42 | tvirus | Hang on, let me VPN to work and snag the 2 files. |
| 19:13.48 | [TK]D-Fender | kaldemar: No, sounds like notihng at all right now. |
| 19:14.23 | [TK]D-Fender | tvirus: And what do you mean "NAT IP"? thats a local IP, and phones should be REGISTERING to * anyways and left on "dynamic" |
| 19:14.42 | tvirus | I don't know how to register a softphone. |
| 19:14.46 | kaldemar | [TK]D-Fender: well 404 sounds like it. but without any configs to show it's a needle in a haystack. |
| 19:14.48 | tvirus | That was my only way to solve that issue. |
| 19:14.57 | [TK]D-Fender | tvirus: It should be trying to register all b y itself <- |
| 19:15.21 | tvirus | And if it doesn't, is it possible to do it manually? |
| 19:15.22 | [TK]D-Fender | kaldemar: He's mentioning NAT and showing nothing that idicates he's set it up properly. |
| 19:15.28 | *** join/#asterisk Levonk (n=lk@adsl-76-238-248-74.dsl.lsan03.sbcglobal.net) |
| 19:15.40 | [TK]D-Fender | tvirus: Stop with the conjecture and explain your networking in detail between * and your phones. |
| 19:15.45 | tvirus | [TK]D-Fender: I'm sure I left out some important bits, let me VPN, I'll have to disconnect, sadly. |
| 19:18.43 | waverly360 | [TK]D-Fender: ok. I've done some homework. I've read the upgrade.txt file, installed asterisk 1.4 (the latest stable revision), and have 3 phones setup in a queue. |
| 19:19.37 | waverly360 | upgrade.txt doesn't say anything about new options that will make a queue be smart about who it sends calls to. |
| 19:19.47 | [TK]D-Fender | waverly360: It does. |
| 19:20.39 | waverly360 | [TK]D-Fender: In relation to sending calls out to multiple agents at the same time, it does. |
| 19:20.57 | [TK]D-Fender | waverly360: KEEP READING. |
| 19:22.24 | waverly360 | [TK]D-Fender: Are you referring to the limitonpeer and call-limit settings? Because those don't accomplish what I need. |
| 19:22.43 | jeev | fender, but no rtp debug for peer, eh? |
| 19:22.48 | [TK]D-Fender | waverly360: You need it not to send calls to agents already on the phone. |
| 19:22.58 | [TK]D-Fender | jeev: You said SIP.... |
| 19:23.11 | [TK]D-Fender | actually |
| 19:23.16 | waverly360 | [TK]D-Fender: ONLY for queues. They need to be able to make calls out, and receive calls from people outside of the queue. |
| 19:23.19 | [TK]D-Fender | jeev: Sorry, you were non-descript |
| 19:23.36 | [TK]D-Fender | jeev: I'm presuming it works the same but get CLI help to see. |
| 19:23.53 | [TK]D-Fender | waverly360: And there is a section concerning queues. |
| 19:24.30 | jeev | yea i did say sip |
| 19:24.39 | *** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net) |
| 19:24.48 | tvirus | Back... I think. |
| 19:24.56 | [TK]D-Fender | jeev: must have been higher up than the point I saw it stated |
| 19:27.07 | jeev | yea, i was just curious |
| 19:27.28 | *** join/#asterisk mog (n=mog@nat/digium/x-105898003e7a2ed0) |
| 19:27.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:27.37 | Cresl1n | thmpfs mog |
| 19:27.38 | jeev | i'm calling wells fargo, enter my card number, last 4 digits. it's like please hold, i either get hung up or it just sits there |
| 19:27.47 | mog | ow |
| 19:27.54 | jeev | itsp told me it could be the fact that i'm faking a callerid or restricting it to an 8** number |
| 19:27.57 | jeev | toll free number |
| 19:28.10 | Cresl1n | How's IAXTEL doing mog? |
| 19:28.24 | mog | it needs some loving this weekend |
| 19:28.35 | mog | should have time given the long weekend |
| 19:28.42 | mog | hows ds iaxphone |
| 19:28.48 | Cresl1n | Is monday holiday? |
| 19:28.59 | [TK]D-Fender | jeev: Should stop in the middle |
| 19:29.02 | [TK]D-Fender | shouldn't* |
| 19:29.05 | mog | ya |
| 19:29.32 | file | Cresl1n: yes, we have it off so don't come into the office :P |
| 19:29.40 | Cresl1n | I might |
| 19:29.41 | Cresl1n | :-) |
| 19:29.51 | mog | holidays are the best time to come in |
| 19:30.07 | Cresl1n | yeah, get more done on holidays than any other day |
| 19:30.30 | tvirus | http://rafb.net/p/c3PloI27.html <-- sip.conf |
| 19:30.57 | tvirus | http://rafb.net/p/yDh00251.html <-- extensions.conf |
| 19:31.11 | tvirus | Also note, I tried extensions with and without the _ |
| 19:33.05 | kaldemar | you're trying to dial SIP/kurt ..dimitry and ..john in your dialplan but the sip users are caller user1, user2 and user3. that's one problem. |
| 19:33.26 | tvirus | Ah |
| 19:33.30 | tvirus | I was trying to mask our names |
| 19:33.32 | tvirus | =| |
| 19:33.34 | tvirus | fail++ |
| 19:33.35 | [TK]D-Fender | tvirus: And you didn't describe how they are networked relative to * like I asked |
| 19:33.48 | tvirus | I was disconnected |
| 19:34.11 | tvirus | * is 192.168.123.230. We're all on the same network. |
| 19:34.11 | [TK]D-Fender | 15:15]<[TK]D-Fender>tvirus: Stop with the conjecture and explain your networking in detail between * and your phones. |
| 19:34.13 | [TK]D-Fender | [15:15]<tvirus>[TK]D-Fender: I'm sure I left out some important bits, let me VPN, I'll have to disconnect, sadly. |
| 19:34.25 | tvirus | Oh, missed that line then, my mistake. |
| 19:34.35 | [TK]D-Fender | tvirus: Ask again when you're in a position to really test it |
| 19:34.41 | jeev | fender, any suggestions? |
| 19:34.53 | [TK]D-Fender | jeev: For a random dropout? Nope. |
| 19:35.21 | jeev | wow, now dtmf wont even go through |
| 19:35.29 | [TK]D-Fender | tvirus: Keep in mind your sip users indeed do not match what you're dialing at all. |
| 19:35.35 | tvirus | What specifics do you need to know about how they are connected? They're all on a switch |
| 19:35.46 | tvirus | [TK]D-Fender: I know they don't match, I was trying to remove our names to protect the innocent. |
| 19:35.55 | tvirus | I just forgot to change the names in extensions.conf |
| 19:36.17 | [TK]D-Fender | tvirus: I am not dealing with a fake scenario. Come and ask when you can actually show us the problem. |
| 19:37.10 | tvirus | http://rafb.net/p/FKutUm66.html Changed it to user1-user3 to match sip.conf |
| 19:37.23 | jeev | Fender, now.. it hasn't dropped the call, the call is still active but nothing is going on, someone should've answered by now |
| 19:37.45 | jeev | and boom, drop |
| 19:37.51 | Damin | Hmm.. |
| 19:37.51 | Damin | This check verifies the ability of the UA to authenticate REGISTER requests. |
| 19:37.52 | Damin | Recommendation |
| 19:37.53 | Damin | It appears that the target UA does not authenticate REGISTER requests using UDP. This configuration allows unauthorized users to generate malicious messages in order to hijack user registration and ultimately divert calls or perform other undesired functions. Change the configuration of the SIP component to require authentication of REGISTER requests. |
| 19:38.09 | Damin | Any way to force Asterisk to authenticate UDP register requests? |
| 19:38.14 | Damin | Or block them entirely? |
| 19:38.15 | [TK]D-Fender | jeev: maybe they are using VAD and its getting dropped for RTP timeout? tweak rtp.conf <- |
| 19:38.55 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:39.12 | waverly360 | [TK]D-Fender: I've read this document multiple times...I've re-read the parts concerning queues even more. The only two features mentioned in that file that pertain to what I'm needing are the autofill and the limitonpeer/call-limit options. Unless I'm missing something, those features alone won't prevent queues from trying to send a call to an agent who's already on the phone. At least not in a way that will let that same agent call someone outsi |
| 19:39.37 | jeev | fender, looking at debug file, what would tell me about a rtp timeout? |
| 19:39.44 | jeev | ?? pb |
| 19:40.01 | jeev | ~ pb |
| 19:40.01 | 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 |
| 19:40.05 | jeev | [Aug 27 12:39:15] DEBUG[39059] rtp.c: Got RTCP report of 88 bytes |
| 19:40.05 | jeev | [Aug 27 12:39:16] DEBUG[39059] chan_iax2.c: Immediately destroying 8807, having received hangup |
| 19:40.05 | jeev | [Aug 27 12:39:16] DEBUG[39059] channel.c: Didn't get a frame from channel: IAX2/jupiter-8807 |
| 19:40.08 | jeev | not 3 lines! |
| 19:40.10 | jeev | over |
| 19:41.57 | pids | why does asterisk report that it got a dtmf and that the length was 0ms? How can it detect it if its 0ms long? |
| 19:43.19 | jeev | ahh fender |
| 19:43.22 | jeev | it is the caller id. |
| 19:43.51 | *** join/#asterisk mikeones (n=mikeones@pool-71-97-107-3.dllstx.dsl-w.verizon.net) |
| 19:44.06 | mikeones | hello |
| 19:44.49 | mikeones | anyone know if I can walk into a frys/bestbuy and find a compatible fxo fxo/fxs pci card? |
| 19:46.30 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 19:47.18 | tvirus | http://dl.getdropbox.com/u/21932/network.png That's how the network is setup. We're not dealing with anything from the internet or Digium card... yet. We're just trying to call softphones from different computers in the office. |
| 19:47.36 | [TK]D-Fender | jeev: umm... now non-applicable... thats **IAX2** |
| 19:47.39 | pids | mikeones, I would seriously doubt it. You cant even buy an adapter that isnt locked for vonage,skype or msn. If you asked a sales person at Frys for an FXO adapter his head would probably explode. |
| 19:47.49 | tvirus | Not sure how much more detail you would need. |
| 19:48.33 | mikeones | pids: true, just wondering if I can find something local. maybe an fxo only? |
| 19:49.50 | [TK]D-Fender | waverly360: waverly360 Yes, it WILL. |
| 19:50.07 | [TK]D-Fender | waverly360: and all your phones should now be type=peer |
| 19:50.18 | pids | mikeones, depends on what local is. If your in Silicon Valley possibly, If your in Idaho probably not. |
| 19:50.41 | [TK]D-Fender | mikeones: this is NOT consumer tech. You aren't going to find this in normal stores |
| 19:51.01 | waverly360 | mikeones: surprisingly enough, I've found fxo/fxs gateway devices in strange places. Those might be easier to come across than a pci card and could provide similar functionality in a pinch. I did actually find a couple of Grandstream fxs devices in Goodwill once... |
| 19:51.17 | waverly360 | mikeones: That's just random happenstance though... |
| 19:51.23 | tvirus | brb |
| 19:51.44 | pids | ouch, grandstream at goodwill. |
| 19:52.44 | waverly360 | [TK]D-Fender: hmm..I'm using type=friend for everything..let me try that. |
| 19:52.51 | mikeones | hmm, is there a quick and dirty way to tell if a pci card is an fxo/fxs card? without knowing the model number? |
| 19:53.02 | pids | mikeones, really the best bet is to order one fedex if you need it asap. You would spend at least 24 hours unlocking anything you found laying around. |
| 19:53.11 | waverly360 | mikeones: really depends on what kind of card it is. |
| 19:53.38 | waverly360 | mikeones: sangoma cards have small daughterboard which are color coded |
| 19:53.42 | [TK]D-Fender | mikeones: Dunno, does it have a DVI connector on it? |
| 19:54.12 | waverly360 | heh |
| 19:54.18 | pids | hehe |
| 19:54.27 | [TK]D-Fender | Seriosly people, W-T-F |
| 19:55.20 | pids | good thing that coffee wasnt to hot or my nose would be burned. |
| 19:55.20 | waverly360 | harmless |
| 19:55.30 | mikeones | heh, just new to pstn stuff and was wounding how to find comething to start on |
| 19:56.37 | [TK]D-Fender | mikeones: Go buy a compatible piece of hardware. Linksys SPA-3102 is my first recommendation. www.telephonydepot.com |
| 19:58.39 | pids | mikeones, buy the right card or you will soon hate pstn stuff. |
| 20:00.16 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 20:02.26 | tvirus | Ok, back at work, [TK]D-Fender, hit me with your questions, por favor. |
| 20:02.59 | [TK]D-Fender | tvirus: SHOW US the problem. CLI output with SIP debug. |
| 20:03.06 | tvirus | Sure |
| 20:03.09 | tvirus | 1 moment. |
| 20:04.04 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:05.24 | tvirus | http://rafb.net/p/FcJcIr97.html <-- when X-Lite connects. |
| 20:07.32 | [TK]D-Fender | tvirus: I don't see the problem |
| 20:07.54 | tvirus | Line 20-26 |
| 20:07.58 | [TK]D-Fender | tvirus: that isn't a call OR a regitser |
| 20:08.21 | tvirus | That's when it connects. I'm copying the call right now, had to adjust text buffer. |
| 20:09.08 | [TK]D-Fender | tvirus: that is a PRESENCE subsciption. |
| 20:09.11 | [TK]D-Fender | not a CALL. |
| 20:09.26 | tvirus | Ok, just making sure that isn't a major issue. |
| 20:11.12 | tvirus | Ugh, copying this sucks, hang on. |
| 20:11.44 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:12.02 | *** join/#asterisk buzzyd (n=psp-man@194.164.81.21) |
| 20:14.02 | buzzyd | hi all, I've downgraded by gcc to 4.1 to compile asterisk however just wondering if it's safe for me to reinstall gcc 4.2 now thats done or will it cause issues? |
| 20:14.10 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 20:14.57 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 20:16.45 | EI5GTB | hi guys, whats the deal with system phones for asterisk... i.e having a programmable phone showing the status of extensions and having an extension onn each button....i,e the 'old fasioned' systemphones |
| 20:18.07 | WimpMan | Was that a question? |
| 20:18.33 | tvirus | http://rafb.net/p/OA7spS25.html <-- that's X-Lite dialing. It's huge. and I'm not sure what all that means. |
| 20:18.52 | buzzyd | alternatively can both gcc 4.2 and 4.1 coexist on the same server? |
| 20:19.15 | [TK]D-Fender | tvirus: Found no matching peer or user for '192.168.123.202:23520' |
| 20:19.21 | [TK]D-Fender | tvirus: It isn't matching your user. |
| 20:19.48 | EI5GTB | WimpMan, well, yea... i just want to know can the be bought....are there special types for asterisk does asterisk work with any...etc...etc.. |
| 20:20.15 | [TK]D-Fender | tvirus: pastebin your sip.conf masking only passwords |
| 20:20.25 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 20:20.35 | lirakis | hey everyone |
| 20:20.42 | WimpMan | EI5GTB: I've done it with Snom and Linksys. |
| 20:20.53 | rwaite | well well well. |
| 20:20.55 | EI5GTB | hmm, i see |
| 20:20.59 | [TK]D-Fender | EI5GTB: www.telephonydepot.com <- Polycom recommended, Aastra is an option. |
| 20:21.00 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:21.06 | lirakis | ... im poking around looking to do a meetme conference and make use of the admin features... but im not sure "how" the admin can use them while in the conf ... |
| 20:21.08 | EI5GTB | so it can be done :P tnx |
| 20:21.30 | rwaite | i like polycom except for this horrible headset issue |
| 20:21.47 | lirakis | does a meetme conf go to a specific context (or can you tell it where to look) for commands being dialed while in the conference? |
| 20:21.47 | WimpMan | Sure. Try to search for busy lamp feature. |
| 20:22.01 | EI5GTB | [TK]D-Fender, tnx :0 |
| 20:22.29 | lirakis | ... any pointers on how the admin cmds get executed, or a sample dialplan some where to look at would be great |
| 20:24.12 | [TK]D-Fender | lirakis: that isn't DIALPLAN. its internal to the room you're in. |
| 20:24.35 | tvirus | Ok 1 moment. I'm on the phone with an angry person :D |
| 20:24.50 | rwaite | wait im not on the phone |
| 20:25.05 | *** join/#asterisk Levonk (n=lk@adsl-76-243-67-198.dsl.lsan03.sbcglobal.net) |
| 20:25.26 | *** join/#asterisk JenniferAkemi (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 20:27.09 | lirakis | [TK]D-Fender: .. okay .. so .. where are the admin commands listed??? are they all just built into the meetme() app? b/c i know there is that meetmeadmin app |
| 20:27.19 | tvirus | http://rafb.net/p/Vp2hlM45.html <-- sip conf with majority of the ; comments deleted and proper usernames and masked passwords. |
| 20:28.01 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 20:33.16 | [TK]D-Fender | tvirus: From: "Kurt"<sip:6111@asterisk>;tag=2c042547 you don't HAVE a SIP peer called [6111] |
| 20:33.30 | [TK]D-Fender | heading home, later all |
| 20:34.08 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:35.42 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:37.28 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 20:38.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:40.10 | RandalSchwartz | FLOSS Weekly (twit.tv/floss) interviewing Mark Spencer in 25 minutes - watch live.twit.tv for the feed... I'll be watching here for questions |
| 20:42.12 | lesouvage | I have search te internet for an example for configuring a hotline in Asterisk. The idea is that when the phone is picked up the operator is dialed. Does any of you has a pointer or suggestion. |
| 20:42.58 | lesouvage | I know it is pssible with a sipura ata but I would like to configure it in Asterisk itself. |
| 20:43.37 | lirakis | does anyone have an idea of how many callers can be in a meetme? |
| 20:43.41 | lirakis | i know it depends on hardware |
| 20:43.49 | lirakis | .. but for no transcoding .. g711 ... |
| 20:43.52 | lirakis | any ideas? |
| 20:45.09 | M1s3ry | lirakis, there is no hard set limit. however if your looking for a table to help determine the amount based on hardware specs or whatnot, I'm unaware of one |
| 20:45.26 | lesouvage | lirakis: if you have enough lines/bandwidth for the lines than my guess is that you can have 50 participants, one lousy phone with echo etc. and you are in trouble. |
| 20:46.04 | lirakis | yeah |
| 20:46.05 | lirakis | okay |
| 20:46.08 | lirakis | sounds good |
| 20:46.09 | lesouvage | it is just a guess |
| 20:46.15 | lirakis | im hoping to get ~30 |
| 20:46.19 | lirakis | we will see |
| 20:46.21 | lirakis | :) |
| 20:47.04 | M1s3ry | i've been in conferences with about 30, not sure the exact number, but it was nearly one entire department |
| 20:48.00 | lirakis | cool |
| 20:48.05 | lirakis | how can i find the hardcoded admin commands for meetme as an admin? |
| 20:48.26 | lirakis | .. so that as an admin i can dial *XX .. or whatever .. to mute .. unmute .. etc.? |
| 20:48.33 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:48.41 | lirakis | im not finding them in ATFOT or searching via "the googles" |
| 20:48.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:48.54 | lesouvage | lirakis: if the participants use proper phones and there are no problems with thel phone lines that generate echo 30 must be possible. There are parameters that can be set for just listening mode and other modes. |
| 20:49.06 | WimpMan | What about an web based conf admin? |
| 20:49.41 | M1s3ry | lesouvage, Ctrl+f for immediate |
| 20:49.44 | M1s3ry | http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
| 20:51.15 | lesouvage | lirakis: see p://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe |
| 20:52.35 | Assid | is there a way to have the musiconhold to start from the middle of a file? |
| 20:52.50 | lirakis | .. found the s param allows you to hit * to get a menu |
| 20:53.21 | WimpMan | Assid: Cut of the part you don't want? |
| 20:53.31 | Assid | everytime i start it.. it starts at the begginining of the trck |
| 20:53.53 | Assid | WimpMan: nah.. i want the track to play in background and this listens in on it |
| 20:53.55 | lirakis | lesouvage: yeah im on that page... but im looking for the command that you can dial while you are in the conf ... not the parameters to pass to the meetme app |
| 20:54.09 | WimpMan | If you get on hold repeatedly? Or what? |
| 20:54.12 | lesouvage | WimMan: I make use of a sip privder so I don't have a zaptel.conf to configure (I guess) |
| 20:54.53 | Assid | WimpMan: whenever its played |
| 20:55.47 | WimpMan | Still not exactely sure what you want. |
| 20:56.09 | WimpMan | Is it just to add some randomness or something? |
| 20:56.14 | Assid | you know how you call call centers? they have some promo crap going on |
| 20:56.41 | Assid | i want it similar.. the track should keep playing. but once anyone gets MoH, they hear whatever is playing that time |
| 20:56.52 | Assid | im thinking i gotta do this using shoutcast or something? |
| 20:56.57 | WimpMan | So you want the Music interrupted and then continue afterwards? |
| 20:57.09 | Assid | no |
| 20:57.17 | RandalSchwartz | FLOSS Weekly (twit.tv/floss) interviewing Mark Spencer at 2pm pacific - watch live.twit.tv for the feed... I'll be watching here for questions |
| 20:57.30 | Assid | i want the music to continue on.. and if a person is put on hold , they hear whatever is playing at that time |
| 20:57.43 | WimpMan | Ok, so everyone is to hear the same thing at the same time? |
| 20:58.35 | _ShrikE | Assid: use mpg123 |
| 20:58.38 | WimpMan | Well, you can use external streams as MOH, so you could det up a player that runst contingously and connect to that, I guess. |
| 20:59.22 | *** join/#asterisk Arck-FR (n=Arck-FR@cvl92-2-82-228-145-232.fbx.proxad.net) |
| 21:00.09 | lesouvage | Assid: just start moh by calling an extension that start it. from what I understand everybody entering moh is hearing the same music. |
| 21:00.14 | esaym | anyone got tips on preventing voices from generating dtmf? It is getting really annoying... |
| 21:00.16 | Assid | yep |
| 21:00.17 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:01.28 | aliver | If I want to "forward" calls which come into my switch from the outside back to an outside call can I do it like this: |
| 21:01.30 | aliver | exten => 799,n,Transfer(SIP/+19706908268@bandwidth.com_outbound) |
| 21:01.46 | aliver | I guess "n" should be "1" there, but you get the idea. |
| 21:01.50 | *** join/#asterisk angryuser (n=angryuse@95.250.146.195.dynamic.adsl.abo.nordnet.fr) |
| 21:01.52 | aliver | bandwidth.com is my outbound provider. |
| 21:02.02 | aliver | and that's the context in my sip.conf |
| 21:02.05 | troubled | RandalSchwartz: oooh? :) |
| 21:02.26 | RandalSchwartz | heh |
| 21:02.31 | troubled | RandalSchwartz: i live here ;) |
| 21:02.35 | RandalSchwartz | FLOSS Weekly (twit.tv/floss) interviewing Mark Spencer at 2pm pacific - watch live.twit.tv for the feed... I'll be watching here for questions |
| 21:02.58 | [TK]D-Fender | aliver: No. |
| 21:03.05 | lesouvage | Nobody has a suggestion for the hotline. It is a phone hanging in an elevator. It may not be used for normal phone calls, instead of that one has to be connected with an operator that can send out the troups to save you. Any suggestion? |
| 21:03.36 | troubled | RandalSchwartz: so will this be the primary question channel, or the twitlive one? |
| 21:03.38 | [TK]D-Fender | aliver: aliver aliver a transfer would send the call over to bandwidth.com with ITS auth, not YOURs. |
| 21:03.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:03.52 | [TK]D-Fender | damn autocomplete... |
| 21:04.37 | [TK]D-Fender | lesouvage: an analog phone hooked up to an ATA / FXS port on "immediate mode" |
| 21:04.52 | RandalSchwartz | I'm watching both |
| 21:04.59 | RandalSchwartz | two windows in emacs :) |
| 21:05.19 | WimpMan | hardcore |
| 21:06.55 | lesouvage | [TK]D-Fender: I know that that is possible but I was hoping there was a way to trigger the dialplan by the "picked up" status of the phone. |
| 21:07.30 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 21:08.07 | [TK]D-Fender | lesouvage: done on the ATA or with "immediate=yes" for zaptel FXS. |
| 21:09.08 | *** join/#asterisk UdontKnow (i=evaldo@freenode/staff/udontknow) |
| 21:09.23 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:09.30 | lesouvage | [TK]D-Fender: I'm using a sip provider so I guess the ata option is the only option to go. No way to detect "the phone is picked up" status of the phone? |
| 21:09.48 | implicit | does anyone know how to reduce the amplitude of ringback in indications.conf ?? its way overmodulated for me |
| 21:09.56 | [TK]D-Fender | lesouvage: W-T-F?! |
| 21:10.27 | [TK]D-Fender | lesouvage: "done on the ATA" <- what part of this is not clear? |
| 21:10.28 | troubled | UdontKnow meet RandalSchwartz, RandalSchwartz meet UdontKnow :) (regarding email/cacert.org) |
| 21:11.02 | RandalSchwartz | hi |
| 21:11.09 | troubled | okay, i spammed #defocus. hopefully can get the viewer count up as well :) |
| 21:11.16 | aliver | [TK]D-Fender What you recommend using (application wise) to do what I'm suggesting? |
| 21:11.23 | troubled | hits up a few other channels |
| 21:11.38 | [TK]D-Fender | aliver: just dial like normal. |
| 21:11.49 | aliver | [TK]D-Fender Dial() then? |
| 21:11.55 | UdontKnow | RandalSchwartz: hi |
| 21:11.56 | [TK]D-Fender | aliver: just dial like normal. <------ |
| 21:12.11 | UdontKnow | RandalSchwartz: actually, you met me in FISL 8.0 in Porto Alegre, Brazil |
| 21:12.15 | RandalSchwartz | cool! |
| 21:12.18 | *** part/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 21:12.30 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088810630.dsl.bell.ca) |
| 21:12.36 | troubled | sorry, I neglected to mention that :) |
| 21:12.38 | aliver | [TK]D-Fender Maybe I wasn't clear. I want to have calls coming into my * from outside forwarded outside my *. |
| 21:13.05 | aliver | [TK]D-Fender I can't tell people calling into my * to "just dial like normal" |
| 21:13.16 | *** join/#asterisk Bryan[STLH] (n=Bryan@unaffiliated/stlhack) |
| 21:13.25 | [TK]D-Fender | aliver: It isn't "forwarding". * sits in the middle of calls. every call into * is JUST ANOTHER CALL |
| 21:13.54 | aliver | [TK]D-Fender okay, I get that, but I used the term "forwarding" loosly. |
| 21:14.03 | [TK]D-Fender | aliver: Now stop. |
| 21:14.26 | aliver | [TK]D-Fender Do I need to Answer() the incoming call, then Dial() the number I want them to "forward" to ? |
| 21:14.28 | lesouvage | [TK]D-Fender: It is perfectly clear, I was just asking if there are options to do this in Asterisk. I consider W-T-F?! as a no. Just relax and cool down (just an advice) |
| 21:14.53 | [TK]D-Fender | aliver: Exactly that. |
| 21:15.05 | aliver | [TK]D-Fender Okay, thanks. |
| 21:15.10 | [TK]D-Fender | lesouvage: no its a "holy shit what don't you get?" |
| 21:15.27 | [TK]D-Fender | lesouvage: and this is don on the interface that the phone is plugged into. |
| 21:17.35 | aliver | http://pastebin.com/d6b8ae4a8 <---- From googling it seems like this means I'm running out of open files. |
| 21:17.44 | aliver | I'll jack it up and hope that works. |
| 21:17.54 | aliver | Anyone seen that error output before? |
| 21:18.19 | WimpMan | man ulimit |
| 21:18.25 | [TK]D-Fender | aliver: maybe you should look at what CAUSED it. |
| 21:18.28 | *** join/#asterisk jtodd (i=_squid@c-76-27-193-118.hsd1.or.comcast.net) |
| 21:18.31 | aliver | I don't need to man ulimit. |
| 21:18.57 | WimpMan | So how many calls are you forwarding? |
| 21:19.07 | aliver | [TK]D-Fender I would if I could scroll up high enough ;-) |
| 21:20.05 | aliver | Yep. Open files. Fixed. BTW, altering your ulimit isn't going to do shit if your system wide ulimits are lower than what you are trying to set to. |
| 21:20.26 | aliver | but you are welcome go chest-thump and yell "man" an RTFM as you wish. |
| 21:20.46 | tvirus | [TK]D-Fender: I was logged in as 6111. I logged in as kurt and it still did the same thing. |
| 21:21.03 | [TK]D-Fender | PASTEBIN <---------- |
| 21:21.20 | tvirus | One moment. |
| 21:21.43 | jeev | ~ pb |
| 21:21.43 | 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 |
| 21:24.01 | tvirus | http://rafb.net/p/uGt5MQ53.html <-- me (kurt) called 6112 (dimitry) |
| 21:25.27 | *** join/#asterisk phrozen77__ (n=phrozen7@unaffiliated/phrozen77) |
| 21:25.46 | [TK]D-Fender | tvirus: sip.conf please |
| 21:26.23 | tvirus | http://rafb.net/p/Vp2hlM45.html <-- sip conf with majority of the ; comments deleted and proper usernames and masked passwords. |
| 21:26.56 | tvirus | Heh, woops. I commeted out the username and not the secret. |
| 21:27.09 | tvirus | username is the same as [name] |
| 21:27.56 | tvirus | http://rafb.net/p/oIZnku75.html properly commented, for the anal retentive. :) |
| 21:28.49 | [TK]D-Fender | tvirus: remove ALL of the "username=" lines from your phone entries |
| 21:29.55 | tvirus | Ok, done. |
| 21:30.13 | *** join/#asterisk ShaunWing (n=chatzill@dsl-241-200-131.telkomadsl.co.za) |
| 21:30.46 | ShaunWing | Say, how can I dial from CLI and play a messge so that I can hear the sound |
| 21:31.19 | tvirus | Getting number not found. |
| 21:31.26 | WimpMan | originate |
| 21:31.37 | [TK]D-Fender | PASTEBIN |
| 21:31.55 | [TK]D-Fender | ShaunWing: Set up a soft-phone |
| 21:32.38 | ShaunWing | I need to make a few simultaneous calls. Want to just make calls from teh cli |
| 21:32.48 | tvirus | http://rafb.net/p/Rl5Ryq89.html <-- sip errors when I call 6112. |
| 21:33.32 | *** part/#asterisk phrozen77__ (n=phrozen7@unaffiliated/phrozen77) |
| 21:33.34 | WimpMan | ShaunWing: use originate, call files or the management interface. |
| 21:33.52 | [TK]D-Fender | tvirus: Looking for 6112 in tutorial (domain asterisk) SIP/2.0 404 Not Found <--- clearly not there. |
| 21:34.15 | [TK]D-Fender | tvirus: At least now your calls are coming in. |
| 21:34.22 | tvirus | All I did was comment out the username=. It would at least attempt to call before, not it just dies instantly. |
| 21:34.30 | tvirus | now* |
| 21:34.54 | ShaunWing | originate number@context ... but how do I get it to play a sound byte? |
| 21:35.33 | WimpMan | originate <dest> application playback |
| 21:36.00 | WimpMan | Saying that, I have never tried anything that needs a parameter. |
| 21:36.48 | [TK]D-Fender | tvirus: your calls are now passing AUTH, which is good. What I showed you is you now have DIALPLAN errors |
| 21:37.02 | WimpMan | But it works, as I just tried :-) |
| 21:37.11 | tvirus | Nice :) Let me snag my extensions.conf |
| 21:37.25 | *** join/#asterisk Psychobilly (n=moi@adsl144-112.kln.forthnet.gr) |
| 21:41.15 | Psychobilly | if i want to check a string indise the dialplan if tis is numeric or not, exept from the REGEX function is there any other way? Can i use the _XXNNZZ patterns inside an expression or they work only for extension matching? |
| 21:43.12 | *** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com) |
| 21:43.32 | _ShrikE | Psychobilly: you just want to make sure there are only numbers in your string? |
| 21:43.38 | Psychobilly | yes |
| 21:43.45 | _ShrikE | core show function FILTER |
| 21:43.51 | tvirus | http://rafb.net/p/Dg71IX67.html <-- there's extensions.conf without all the fluff that I deleted earlier. |
| 21:44.15 | Psychobilly | i dont want to filter it, i want to check if its numeric, i know about the filter func |
| 21:44.46 | Psychobilly | i was just wondering if i can use the NXZ.! patterns inside an expression |
| 21:45.25 | _ShrikE | Dont think so. |
| 21:46.22 | *** join/#asterisk _khan (n=shariq@124.29.194.207) |
| 21:47.09 | Psychobilly | thx _ShrikE , i also think thats not possible but i want to be sure |
| 21:47.45 | _khan | what are the tonezone settings in indications.conf for pakistan country. i have problem with tdm2400p when calling it, cli shows phone answered while actually it is on ringing |
| 21:49.21 | implicit | does anyone know hwo to reduce the amplitude of tones in indications.conf? |
| 21:50.16 | lesouvage | Psychobilly: You can check it using numbermatching |
| 21:51.12 | lesouvage | Psychobilly: if you set up the dialplan in a clever way. |
| 21:51.24 | Psychobilly | lesouvage can u make it a bit more clear? |
| 21:51.35 | [TK]D-Fender | tvirus: pastebin "dialplan show" |
| 21:51.36 | Psychobilly | actually i want to check the callerid(num) |
| 21:51.47 | Psychobilly | and make sure it is a number |
| 21:52.23 | lesouvage | Psychobilly: what do you want to check in the callerid(num) |
| 21:52.31 | lesouvage | OK |
| 21:52.38 | Psychobilly | that it is really a numeric string |
| 21:52.47 | [TK]D-Fender | Psychobilly>if i want to check a string indise the dialplan if tis is numeric or not, exept from the REGEX function is there any other way? Can i use the _XXNNZZ patterns inside an expression or they work only for extension matching? <- No. |
| 21:52.57 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:53.19 | Psychobilly | thx [TK]D-Fender |
| 21:54.15 | lesouvage | Psychobilly: give me a minute |
| 21:54.17 | Psychobilly | is there any trick to do it without using regex funct? i want to avoid loading another module in my embedded asterisk |
| 21:54.32 | [TK]D-Fender | Psychobilly: large amounts of dialplan. |
| 21:54.41 | Psychobilly | xm |
| 21:56.29 | tvirus | http://rafb.net/p/ymOxHM21.html I dunno where those other contexts came from after tutorial. I did restart now in asterisk after I removed the other crap in extensions.conf. |
| 21:57.05 | Psychobilly | [TK]D-Fender i dont know whats worse, another module and less free ram or a bigger dialplan with ugly code :P |
| 21:58.03 | [TK]D-Fender | Psychobilly: I know whats worse... another stingy twit trying to save a few byes :) |
| 21:59.12 | Psychobilly | hehehe |
| 21:59.36 | [TK]D-Fender | tvirus: Looking like changes haven't been applied properly. Do a "reload" and retest |
| 22:00.51 | Psychobilly | and another thing that makes me wonder, which is the correct way to send CID info in sip? some pbxs send me "number" <name> and some others "name" <number> |
| 22:01.10 | Psychobilly | this way callerid(num) some times returns me a name and other a number |
| 22:01.22 | mvanbaak | Psychobilly: that's just plain wrong |
| 22:01.31 | mvanbaak | Psychobilly: it should be: "name" <number> |
| 22:02.14 | tvirus | Now it just dials like before, let me paste the sip debug. One moment. |
| 22:02.19 | Qwell | no quotes |
| 22:02.37 | aliver | Can I alter the way caller ID from outside calls appears on my SIP phones using asterisk? (PSTN trunked from a commercial SIP provider) |
| 22:02.43 | mvanbaak | Qwell: even if there's a space in it ? |
| 22:02.46 | Qwell | yep |
| 22:02.50 | mvanbaak | hhmm |
| 22:03.07 | mvanbaak | gheh, indeed |
| 22:03.15 | mvanbaak | my callcontrol agi strips them ;) |
| 22:03.18 | lesouvage | Psychobilly: I'm almost there. |
| 22:03.23 | aliver | Or is my upstream SIP provider the only one who can change the CID? |
| 22:03.54 | tvirus | http://rafb.net/p/WLwid744.html sip debug output from latest attempt at me (kurt) dialing 6112 (dimitry) |
| 22:03.56 | mvanbaak | aliver: you can overwrite them |
| 22:03.58 | _ShrikE | aliver: you can change it, but you provider may not honor that change. |
| 22:04.02 | tvirus | all that crap is still in my show dialplan. |
| 22:04.26 | aliver | Well, I don't care about how my provider shows my CID to others, I'm talking about what shows up when I get calls coming in. |
| 22:04.48 | aliver | For example it shows "CITY, ST" then "PHONE NUMBER" I'd rather see the phone number first. |
| 22:05.07 | Bryan[STLH] | Does this work on *BSD? |
| 22:05.18 | aliver | Bryan[STLH] asterisk? |
| 22:05.20 | mvanbaak | Bryan[STLH]: does what work on *BSD ? |
| 22:05.23 | Bryan[STLH] | yes |
| 22:05.24 | Bryan[STLH] | asterisk |
| 22:05.26 | aliver | yes |
| 22:05.31 | Bryan[STLH] | awesome |
| 22:05.34 | Bryan[STLH] | thank you |
| 22:05.34 | aliver | FreeBSD ports has 1.4 |
| 22:05.40 | aliver | NetBSD pkgsrc has 1.2 |
| 22:05.52 | aliver | OpenBSD, who knows, it's probably too "secure" for asterisk. |
| 22:06.07 | mvanbaak | it runs fine on OpenBSD |
| 22:06.12 | aliver | Well, there ya go. |
| 22:06.14 | mvanbaak | there are 1.4 packages |
| 22:06.22 | mvanbaak | and trunk and 1.6 run on it as well :) |
| 22:06.37 | aliver | good deal. |
| 22:06.57 | aliver | However, if you want to have G729 working you'd better stick with Linux. |
| 22:07.15 | aliver | Since Digium only sells that module as a Linux shared object. |
| 22:07.34 | aliver | and their license rig works with linux's ethernet MAC. |
| 22:07.42 | *** join/#asterisk brown10101 (n=celso@cpc1-walt6-0-0-cust156.popl.cable.ntl.com) |
| 22:08.01 | aliver | I tried to get them to sell me a G729 for NetBSD and they acted like I was insane. |
| 22:08.38 | mvanbaak | same goes for dahdi hardware |
| 22:08.43 | WimpMan | likes companies who think that way about cusomers. |
| 22:08.43 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 22:09.01 | aliver | mvanbaak they acted as if you were insane? |
| 22:09.03 | lesouvage | Psychobilly: ${CALLERID(num)}= ${ARG1} see http://www.pastebin.be/13436 This should work and takes just 15 lines. |
| 22:09.29 | jeev | wow |
| 22:09.32 | jeev | lesouvage, what's that do? |
| 22:09.38 | Psychobilly | lesouvage let me see |
| 22:09.46 | jeev | it sets a caller id ? |
| 22:09.53 | Psychobilly | hehehe |
| 22:09.59 | Psychobilly | nice |
| 22:10.08 | jeev | ah, it checks caller id |
| 22:10.08 | Psychobilly | it looks cute, thats for sure |
| 22:10.20 | jeev | Psychobilly, is that what it's supposed to do? check / |
| 22:10.46 | Psychobilly | jeev check if the calleridNum) is a numeric string |
| 22:10.51 | mog | aliver, sorry for your problems porting zaptel back to bsd is a lot of work for such a small customer base, but the code is there to be ported, and there is/was a zaptel-bsd port project |
| 22:10.54 | Psychobilly | jeev check if the callerid(num) is a numeric string |
| 22:11.26 | tvirus | How can I clear out what's in dialplan show in the CLI. I stopped, started, reloaded, restarted, and it's still there even though my extensions.conf is only 1 context. |
| 22:11.43 | mvanbaak | aliver: they told me: "if you want dahdi on OpenBSD you know where to get the source, and you know how to handle vim" |
| 22:11.49 | lesouvage | Psychobilly: it just test a cid if it is nummeric up to a length of 14 positions. If there isn't a match yet it seems save to asume that th cid is alphanummeric. For the rest it's doing nothing. It's up to you what it really has to do. |
| 22:12.00 | jeev | hm, i want to be able to set an outgoing caller id number, i'm sure thta's possible.. right |
| 22:12.14 | mog | time to go home though |
| 22:12.16 | mog | bye everybody |
| 22:12.20 | Psychobilly | yes lesouvage i got it |
| 22:13.14 | TJNII | ~pb |
| 22:13.15 | 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 |
| 22:13.22 | Psychobilly | thanks, as i said earlier i dont know what is worse, to fill my dialplan with code like this or just use the func_strings module |
| 22:13.27 | Psychobilly | :) |
| 22:13.54 | mvanbaak | lesouvage: nice dialplan magic |
| 22:14.09 | Psychobilly | yes very clever |
| 22:14.15 | *** join/#asterisk Levonk (n=lk@adsl-75-62-130-228.dsl.lsan03.sbcglobal.net) |
| 22:14.17 | lesouvage | mvanbaak: thanks |
| 22:14.40 | mvanbaak | for stuff like that I always do: Agi(voodoo.py) |
| 22:14.41 | mvanbaak | ;) |
| 22:14.51 | Psychobilly | heh |
| 22:15.00 | aliver | mog is the G729 codec dependent on the zaptel driver? |
| 22:15.13 | aliver | The rest of asterisk works fine in NetBsd |
| 22:15.23 | aliver | even zaptel. |
| 22:16.25 | mvanbaak | aliver: I dont think G729 is dahdi dependent |
| 22:16.53 | lesouvage | mvanbaak: I only do Asterisk script and the funny thing is that I can do all the dialplan magic needed (except setting up a hotline for the elevator) |
| 22:17.29 | TJNII | Okay IAX gurus, what am I missing here? http://pastebin.com/m4923eff5 |
| 22:17.37 | TJNII | Outgoing works, incoming gets rejected. |
| 22:19.07 | aliver | mvanbaak they probably just don't have development environments setup for *BSD, Solaris, etc... so they just stick to linux. |
| 22:19.15 | aliver | for G729's .so |
| 22:21.55 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:23.40 | lesouvage | TJNII: Are you sure you have a line somewhere like exten => 5155982226,3,Dial(SIP/<sip number linkt with incoming line>,40). If you don't dial an inter phone for an incoming line no phone will ring. |
| 22:24.23 | TJNII | lesouvage: It is in the pastebin. |
| 22:25.06 | TJNII | It never gets to exten => 5155982226 though |
| 22:25.14 | *** join/#asterisk ManxPower (n=manxpowe@6.sub-70-218-131.myvzw.com) |
| 22:28.02 | mvanbaak | [lesouvage] s/linkt/linked/ |
| 22:28.05 | mvanbaak | ;) |
| 22:28.42 | [TK]D-Fender | '5155982226@' <- missing context. |
| 22:29.10 | *** join/#asterisk daniev (n=ganbarim@190.144.60.154) |
| 22:29.11 | TJNII | I figures it out. I needed to have [vitel-inbound] as the section name instead of using username=vitel-inbound |
| 22:30.01 | *** join/#asterisk lalmia (n=cmrahman@64.129.97.77) |
| 22:30.18 | *** join/#asterisk Pedro51 (i=Pedro@88.157.91.40) |
| 22:30.40 | lesouvage | mvanbaak: thanks |
| 22:30.46 | Pedro51 | hi, can someone answer me a question about asterisk? |
| 22:31.07 | WimpMan | Pedro51: Only if you ask. |
| 22:31.30 | Pedro51 | it's really simple, i just came to ask if you can PC to phone calls for free... |
| 22:31.32 | lesouvage | Pedro51: give it a shot (and don't ask to ask) |
| 22:31.38 | lalmia | anybody know why "agent show" command all of suddent disappear ? |
| 22:31.57 | tvirus | I don't get it. If i call a random number it says it's not found. If I dial 6111 or 6112 or 6113 it just says 'calling', but nothing happens. No popups nothing. :'( |
| 22:32.14 | lalmia | No such command 'agent show' |
| 22:34.28 | mvanbaak | Pedro51: so you want asterisk to pay your phonebills ? |
| 22:34.31 | *** join/#asterisk Whitor (n=Whitor@cpe-74-76-185-31.nycap.res.rr.com) |
| 22:34.53 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:35.12 | mvanbaak | hey anthm |
| 22:35.34 | Pedro51 | i guess so... really, don't take my question as offensive because i didn't mean it, i just want to know if it works for what i want so i can get free calls... if not, its ok anyway |
| 22:36.09 | mvanbaak | Pedro51: some ITSP's provide free national calls |
| 22:36.41 | mvanbaak | Pedro51: but asterisk is not some magic money machine. It all depends what voip links you get with ITSP's |
| 22:36.44 | Pedro51 | hmmm nice, but can you tell me one or two of them that do it? |
| 22:36.46 | mvanbaak | ~itsp |
| 22:36.46 | 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 |
| 22:37.21 | mvanbaak | Pedro51: nope. in my experience the ones that provide free calls are the worst quality |
| 22:37.29 | mvanbaak | ad the real good ones ask some money |
| 22:37.49 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 22:37.52 | Pedro51 | oh ok then |
| 22:37.58 | Pedro51 | i think ill just stay as i am |
| 22:38.00 | lesouvage | Pedro51: If you manage to get Asterisk up and running you can make free calls with others with an asterisk box or other sip pbx, sip hardphone or a sip softphone (iax2 is also an option). So you can use Asterisk to set up your own infrastructure with friends and family. |
| 22:38.04 | Pedro51 | but thanks anyway |
| 22:38.13 | mvanbaak | Pedro51: but I'm in .nl, so I have no idea how it works in other countries |
| 22:38.33 | mvanbaak | lesouvage: you forgot Skinny in there ;) |
| 22:38.35 | Pedro51 | ok, i really gotta go now, but thanks, and good bye :) |
| 22:38.36 | mvanbaak | hides |
| 22:38.40 | *** part/#asterisk Pedro51 (i=Pedro@88.157.91.40) |
| 22:38.53 | WimpMan | or h.323? |
| 22:39.06 | mvanbaak | yeah |
| 22:39.06 | *** join/#asterisk rasterix (n=IceChat7@host81-155-2-150.range81-155.btcentralplus.com) |
| 22:39.07 | mvanbaak | and mgcp |
| 22:39.39 | mvanbaak | and vocp (voice over carrier pigeon) |
| 22:40.02 | WimpMan | Is there an rfc for that? |
| 22:40.19 | mvanbaak | there is. it's not specific for voip |
| 22:40.30 | mvanbaak | but there's an rfc for network traffic over cp |
| 22:40.36 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 22:40.43 | mvanbaak | there's even an event where they implemented it |
| 22:41.05 | outtolunc | those *dropped* packets can get messy |
| 22:41.12 | lesouvage | Does Asterisk support MELPe. There has been some efforts in the past. |
| 22:41.14 | WimpMan | Yes, BTW is there a list of just those funny rfcs somewhere? |
| 22:41.24 | mvanbaak | WimpMan: http://en.wikipedia.org/wiki/IP_over_Avian_Carriers |
| 22:41.44 | florz | rfc 1st april should get you there with pretty much any search engine ... |
| 22:42.12 | WimpMan | Did they all originate on April 1st? |
| 22:42.21 | florz | yeah |
| 22:42.31 | rasterix | hi people |
| 22:42.33 | florz | at leas I don't know of any that didn't |
| 22:42.40 | tvirus | If I wanted to test dialing from CLI, would I do dial extension@context? |
| 22:42.58 | WimpMan | Ok, that makes it easier. Never notived the date... |
| 22:43.14 | rasterix | im doing a simple help page for extensions.conf and realised i dont realise what the autofallthrough property does |
| 22:43.23 | mvanbaak | tvirus: yeah: console dial exten@context |
| 22:43.39 | tvirus | Thought so, nothing rings :( |
| 22:43.41 | tvirus | God damnit. |
| 22:43.47 | rasterix | read up a lil bit and wrote this does it sound right? In this context you can specify general configuration options for the dial plan. As of version 1.4 three configuration options can be set.<br/><br/><b>static=yes|no</b><br/><br/><b>writeprotect=yes|no</b><br/><br/><b>autofallthrough=yes|no</b><br/>If set when the last priority for an extension is reached with no following priority Asterisk will hang up the call. |
| 22:45.35 | mvanbaak | rasterix: have a look at the sample extensions.conf in asterisk-source/configs/extensions.conf.sample |
| 22:45.40 | mvanbaak | it's documented there |
| 22:46.38 | mvanbaak | rasterix: http://svn.digium.com/view/asterisk/trunk/configs/extensions.conf.sample?view=markup |
| 22:46.52 | rasterix | i did maybe i should have said i dont understand what it means by "it will wait for another extension to be dialled" |
| 22:47.20 | mvanbaak | it will just sit there, and you can pound in another valid extension in the context you are in |
| 22:47.24 | lalmia | what will be the reason when I run command "show agents" it says command not found? |
| 22:48.21 | mvanbaak | lalmia: it is: agent show |
| 22:49.16 | lalmia | ipbx*CLI> agent show |
| 22:49.16 | lalmia | No such command 'agent show' (type 'help' for help) |
| 22:49.27 | lesouvage | mvanbaak: show agents is still working (at least at my box) |
| 22:49.30 | rasterix | so if the caller enters an extension "23" it will run extension 23 in the context its hung in? |
| 22:50.05 | mvanbaak | rasterix: if autofallthrough is set to no, the call wont be hangup |
| 22:50.12 | mvanbaak | lesouvage: 1.4 ? |
| 22:50.39 | mvanbaak | lalmia: that means you dont have chan_agent loaded |
| 22:50.53 | rasterix | ok i still dont understand... it hangs and waits for more input from the caller? |
| 22:50.55 | mvanbaak | lesouvage: it does not work on my box. but I'm running trunk |
| 22:51.02 | mvanbaak | rasterix: yup |
| 22:51.12 | rasterix | woah thats dangerous |
| 22:51.13 | mvanbaak | rasterix: try it :) |
| 22:51.19 | rasterix | no wonder its says set it to yes! |
| 22:51.35 | mvanbaak | rasterix: nah. it will only work for the extens you have in the current context |
| 22:51.41 | [TK]D-Fender | rasterix: Be smart and END your calls YOURSELF |
| 22:51.48 | mvanbaak | that too |
| 22:51.49 | mvanbaak | ;) |
| 22:51.51 | rasterix | lol i do |
| 22:51.57 | mvanbaak | hurray for Hangup() |
| 22:52.03 | rasterix | yeah i understand that |
| 22:52.04 | [TK]D-Fender | I never use Autofallthrough=yes. |
| 22:52.22 | bradleyprice86 | I am setting up my extensions_custom.conf for the ability to dial in and make a call through the asterisk system (for long distance purposes) but I can't seem to get my dial command to work. How exactly are the arguments suppose to be passed? I have tried dial(Zap/1/XXXXXXXXXX) but that doesn't work. |
| 22:52.23 | rasterix | im trying to what understand what happens if you omit to call hangup() |
| 22:52.38 | mvanbaak | rasterix: only way to understand it is to try it |
| 22:52.38 | tvirus | [TK]D-Fender: May we resume our previous help session? |
| 22:53.41 | rasterix | why not fender? |
| 22:54.09 | rasterix | autofallthrough = yes is the recommended setting... why not use it? |
| 22:54.12 | *** join/#asterisk Tako-san (n=Tako-san@24.108.192.144) |
| 22:54.26 | [TK]D-Fender | rasterix: because I use "s" for IVR's the old fashioned way and I end my extensions when I say they end. |
| 22:54.54 | rasterix | aha! |
| 22:54.56 | [TK]D-Fender | rasterix: "Waitexten" was a waste of time mostly and now morons run IVR's off Friggen NUMBERED extens which can lead things in cirecles. |
| 22:55.33 | rasterix | im pretty sure ive got some "s" on our ivr... brb... |
| 22:56.41 | rasterix | oh one last question, three options, static, writeprotect and autofallthrough... anyone know what versions of asterisk each of these is valid for? |
| 22:57.28 | Tako-san | Hey, anyone have any recommendations for a 24 port PoE switch? |
| 22:57.43 | rasterix | yah |
| 22:57.54 | rasterix | we got a d-link one which works just fine |
| 22:57.55 | *** join/#asterisk Levonk (n=lk@adsl-76-237-12-55.dsl.lsan03.sbcglobal.net) |
| 22:57.59 | rasterix | and cheap |
| 22:58.12 | [TK]D-Fender | Tako-san: D-Link DES-1128P |
| 22:58.16 | [TK]D-Fender | 1228* |
| 22:58.25 | rasterix | yeah wot fender said |
| 22:58.31 | rasterix | i was just lookin up the model |
| 22:58.35 | rasterix | but hes too damn fast |
| 22:58.41 | Tako-san | [TK]D-Fender & rasterix : Thank you. |
| 22:59.23 | rasterix | blah fender stole my chance to answer a question |
| 22:59.45 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 23:00.16 | scooby2 | DES-1228P |
| 23:00.46 | tvirus | rasterix: You can help me out if it'll make you feel better :) |
| 23:00.58 | rasterix | someone on here said there is an xml syntax for extensions being worked on anyone know where it is? |
| 23:01.18 | mvanbaak | rasterix: that's far from being coded |
| 23:01.41 | mvanbaak | rasterix: we are working on xml documentation inside the asterisk source |
| 23:01.46 | rasterix | k im trying to work out the syntax for different versions of asterisk |
| 23:01.55 | WimpMan | wonders if everything needs to be xml these days. No matter if it makes sense or not. |
| 23:02.00 | rasterix | is running the help on different installs safe? |
| 23:02.02 | mvanbaak | and if that works out, and I have some more time, I'll have a stab at using libxml2 for extensions.xml |
| 23:02.44 | rasterix | i.e. if i install 1,1.2,1.4 and run the help from cli will that give me the correct syntax for dialplan apps on each version... or not reliable? |
| 23:02.48 | WimpMan | Just being curious: How many of you use extensions.ael? |
| 23:03.11 | ManxPower | rasterix: the application docs in the CLI are the ONLY official docs for applications. |
| 23:03.15 | mvanbaak | WimpMan: I am |
| 23:03.34 | rasterix | so u dont think they are a reliable souce? |
| 23:03.42 | rasterix | source* |
| 23:03.43 | ManxPower | WimpMan: I like the idea of AEL quite a bit, but at this point it is not an option for me. |
| 23:04.07 | ManxPower | Any other application docs is likely to be outdated. |
| 23:04.16 | ManxPower | Especially voip-info.org |
| 23:04.27 | rasterix | yeah i figured that manx |
| 23:04.56 | rasterix | i was wondering if the cli help that installs with different versions would be a starting point |
| 23:07.01 | tvirus | [TK]D-Fender: Everything from the * console looks fine, but the calls never connect. The dialplan is basic, it should work, shouldn't it? http://rafb.net/p/WLwid744.html |
| 23:07.13 | rasterix | the only part im really interested in at the moment is [[context,]extension,]priority etc.... |
| 23:07.52 | mvanbaak | rasterix: for dialplan commands that is ? |
| 23:07.57 | rasterix | yah |
| 23:08.00 | jeev | i have serious issues, i need to really implement billing into my systems already |
| 23:08.13 | mvanbaak | rasterix: then you will like what we are doing with the appdocsxml branch |
| 23:09.02 | mvanbaak | rasterix: there compiling asterisk will generate an xml file with all the documentation |
| 23:09.08 | rasterix | mvanbaak: yah when i heard someone mention it my ears pricked... but it doesnt help me with 1.0,1.2,.... |
| 23:09.13 | mvanbaak | rasterix: let me upload the generated xml so you can see it |
| 23:09.48 | rasterix | its kind of a shame that extensions.conf language is so badly structured... |
| 23:10.05 | rasterix | i guess it cant be helped with a project thats grown like asterisk though |
| 23:10.12 | mvanbaak | rasterix: http://michiel.vanbaak.info/temp/core-en_US.xml |
| 23:10.42 | [TK]D-Fender | tvirus: Set all of your phone tries to "type=peer", "host=dynamic", and verify that each registers OK |
| 23:12.15 | mvanbaak | rasterix: <application name="Dial"> is your best reference there |
| 23:12.22 | rasterix | im looking at that |
| 23:12.24 | rasterix | right now |
| 23:12.30 | rasterix | it looks to have potential |
| 23:12.47 | mvanbaak | rasterix: I started work on a php file to parse that and make nice html docs |
| 23:12.54 | [TK]D-Fender | s/tries/entries/ |
| 23:13.01 | rasterix | sweet |
| 23:13.03 | mvanbaak | but there's more important stuff in that branch to be fixed first |
| 23:13.27 | rasterix | all i wanted to do was find a way |
| 23:13.36 | rasterix | of listing all the dial plan functions |
| 23:13.41 | mvanbaak | I think we are close to have a basic system working, so we can start to convert every application and function to have xml docs |
| 23:13.45 | rasterix | with the difference in syntax indicated |
| 23:13.48 | mvanbaak | then we can finish the php |
| 23:13.53 | rasterix | like a kind of truth table |
| 23:13.54 | mvanbaak | and be done with it |
| 23:14.15 | tvirus | Ok, one moment. |
| 23:14.25 | mvanbaak | and now I'm going to bed |
| 23:14.28 | mvanbaak | latero all |
| 23:14.39 | rasterix | mvanbaak: bye.. thanks |
| 23:14.41 | *** join/#asterisk Levonk (n=lk@75.62.137.255) |
| 23:14.46 | WimpMan | rasterix: Do you really want to go back to 1.0? |
| 23:14.46 | mvanbaak | ur welcome |
| 23:14.56 | *** join/#asterisk sarascle (n=chatzill@63.160.230.16) |
| 23:15.10 | rasterix | i have no way of analysing the number of live 1.0,1.2,1.4 systems in use |
| 23:15.19 | WimpMan | rasterix: Or are you trying to document the evolution? |
| 23:15.26 | sarascle | what do i need for home phone set up |
| 23:15.37 | rasterix | i wanted to do a "cheat sheet" |
| 23:15.52 | rasterix | a quick reference for application differences between the versions |
| 23:15.53 | [TK]D-Fender | sarascle: Depends exactly what you want to interface with |
| 23:16.48 | sarascle | can interface with packet 8 or the normal phone system |
| 23:16.53 | tvirus | host=dynamic works. Why wouldn't it work if I set the host's IP? |
| 23:17.02 | rasterix | evolution is fine but im bet there are a lot of people with working 1.0,1.2 systems that see no reason to touch them |
| 23:17.12 | [TK]D-Fender | tvirus: in case its pointed the wrong way. if it works, great |
| 23:19.03 | WimpMan | rasterix: Possibly. I started right at the point where the first 1.4 was released (still beta) but didn't want to bother with the changes so went staright to 1.4 and it did what it was supposed to do |
| 23:19.12 | rasterix | general question.. is it safe to say asterisk 1.0 is now largely outdated and in use by very few legacy systems? |
| 23:19.43 | WimpMan | However at home I had to upgrade a few times to get around some issues I experienced. |
| 23:20.28 | rasterix | wimpman: i installed 1.2 first and just upgraded to 1.4... had a few problems because fairly large dialplan but largely ok |
| 23:21.18 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 23:21.20 | WimpMan | wonders if it would be time to take a look at 1.6. |
| 23:21.50 | rasterix | i will address 1.6 later |
| 23:21.57 | rasterix | still dont know the release date |
| 23:22.08 | drmessano | 1.0 is extinct |
| 23:22.15 | drmessano | 1.2 is soon to be |
| 23:22.31 | rasterix | i think changes from 1.4 to 1.6 will be better documented than previous upgrades |
| 23:22.44 | mchou | lol |
| 23:22.47 | drmessano | o.O |
| 23:22.57 | drmessano | All the changes have been documented |
| 23:22.57 | mchou | that's hardly saying much :) |
| 23:23.05 | drmessano | Are you just spewing crap? |
| 23:23.16 | rasterix | they may have been documented |
| 23:23.19 | mchou | documented where exactly? |
| 23:23.25 | rasterix | im just having trouble finding it? |
| 23:23.46 | *** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 23:23.59 | *** join/#asterisk StanAccy (n=npratt@pool-71-183-32-116.nycmny.fios.verizon.net) |
| 23:24.02 | WimpMan | has had quite some troube to find up to date documentation. |
| 23:24.10 | *** join/#asterisk ivarv (n=ivar@d66-183-61-180.bchsia.telus.net) |
| 23:24.14 | StanAccy | Is there an Admin tool for Asterisk/freepbx that will give me a summary of the current status of all extentions (who has call forwarding configured, who has voicemail setup, number of voicemails waiting etc.)? |
| 23:24.24 | sarascle | i would like what hardware and software i can use in my home to replace my land line |
| 23:24.27 | drmessano | It's hidden deep inside the asterisk source inside a file called ChangeLog |
| 23:24.28 | _-Jon-_ | Evening all.. I'm having a slight problem with configuring my Asterisk box for use with a new provider. Wondering if anyone can lend a hand |
| 23:24.37 | rasterix | no drmessano if a reasonably experienced end user struggles to find the relevant documentaion then it is NOT properly documented... maybe its you whos spewing **** |
| 23:24.50 | [TK]D-Fender | ~freepbx |
| 23:24.51 | 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 |
| 23:24.53 | drmessano | lol if you say so |
| 23:24.53 | [TK]D-Fender | StanAccy: ^^^^^^^^^^^^^ |
| 23:24.54 | mchou | rasterix: yeah. I agree |
| 23:25.08 | drmessano | All the changes are in the tarball you fucking moron |
| 23:25.12 | drmessano | Go look |
| 23:25.31 | StanAccy | I see. Let me rephrase: Is there an Admin tool for Asterisk that will give me a summary of the current status of all extentions (who has call forwarding configured, who has voicemail setup, number of voicemails waiting etc.)? |
| 23:25.43 | *** part/#asterisk sarascle (n=chatzill@63.160.230.16) |
| 23:25.49 | rasterix | ok drmessano your attitude is fairly immature ill talk to others... thanks |
| 23:25.53 | StanAccy | Im assuming this data is all in the DB |
| 23:26.14 | StanAccy | before I go write the SQL, I thought I'd check if there was such an admin summary tool around |
| 23:26.31 | drmessano | No problems, Rasterix.. my IQ is higher thanks to you |
| 23:26.49 | ivarv | are there any opensource asterisk compativle clients written in flash ? I hate flash but... |
| 23:26.53 | rasterix | just being in my presence raised your iq... glad to hear it :P |
| 23:27.05 | StanAccy | cutting wit |
| 23:27.08 | mchou | ivarv: wtf?? |
| 23:27.19 | drmessano | rasterix: Yeah, next to you, I look like I have a 500 IQ |
| 23:27.26 | drmessano | rasterix: So thank you |
| 23:27.32 | StanAccy | 30-15 |
| 23:27.34 | WimpMan | ivarv: You try to set up some "value added" web stuff? |
| 23:27.41 | _-Jon-_ | I'm not picking sides, but is drmessano 12? |
| 23:27.48 | rasterix | lol |
| 23:27.51 | ivarv | WimpMan, that's the idea.. yeah |
| 23:28.01 | mchou | WimpMan: to call flash "value-add" is beyond a sick joke |
| 23:28.07 | jblack | He's closer to 14 than 12. |
| 23:28.10 | drmessano | _-Jon-_: No, 15 |
| 23:28.12 | drmessano | HEY |
| 23:28.12 | WimpMan | immediately got the smell :-) |
| 23:28.14 | drmessano | I'm 15 |
| 23:28.15 | mchou | ivarv: go look at grandcentral.com :) |
| 23:28.29 | jblack | I didn't say you were 14. |
| 23:28.39 | jblack | 14 is closer to 15 than 12, no? |
| 23:28.40 | drmessano | Oh |
| 23:28.45 | drmessano | Indeed |
| 23:28.46 | mchou | ivarv: plenty of flash there for your "value add" |
| 23:29.11 | ivarv | mchou, I was hoping for something opensource.. there's also ribbit.com |
| 23:29.21 | [TK]D-Fender | StanAccy: All of the forwarding, etc is FreePBX specific. Voicemails are not necessarily related to phone devices, and there is no general GUI I know of that will cover all of what you're looking for. |
| 23:29.30 | StanAccy | Ah. Thanks |
| 23:30.00 | ivarv | also, grandcentral seems US centered |
| 23:30.03 | WimpMan | ivarv: I only know one companie doing such a thing and I'm sure they will want to keep it secret. |
| 23:30.06 | drmessano | I am not the smartest guy in the world, but I pride myself on not going into a channel with a nickname like rasterix that sounds like I know something about asterisk, then making lame statements about changes not being documented |
| 23:30.12 | drmessano | But hey, that's me :) |
| 23:30.40 | rasterix | so getting back to my original point... i dont think i can ignore 1.2... so if i install 1.2 and 1.4 and use the cli to get the syntax will i be close? |
| 23:30.41 | mchou | ivarv: open source and flash in the same breath, you're brave |
| 23:30.57 | Assrisk | in case you have a lisp |
| 23:30.58 | Asstorisc | Whats a PBX? |
| 23:31.01 | ivarv | mchou, well.. adobe is opening up a bit... |
| 23:31.23 | Asstorisc | If I install 1.2 and 1.4, will I learn something about asterisk by osmosis? |
| 23:31.28 | mchou | ivarv: haha, that's a joke too, right? |
| 23:31.30 | Asstorisc | Do I need 1.8 too? |
| 23:31.54 | drmessano | Loser |
| 23:32.02 | ivarv | mchou, and adobe is creating their own native voip/sip extension to flash too.. |
| 23:32.07 | mchou | ivarv: adobe is very aggressive with coyrights and patents |
| 23:32.19 | mchou | copyrights* |
| 23:32.42 | rasterix | drmessano please just ignore me i have not time to fight with a 14 year old |
| 23:32.43 | *** join/#asterisk pids (n=pids@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 23:32.50 | mchou | with that kind of attitude I'm dubious adobe and "open source" will get very far |
| 23:32.53 | drmessano | rasterix: Ignore is a CLIENT SIDE function |
| 23:33.02 | drmessano | rasterix: Not that I would expect you to know that |
| 23:33.11 | drmessano | rasterix: Since your nick isn't eyearesee |
| 23:33.54 | ivarv | mchou, no http://pac.ifica.net/ |
| 23:34.12 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
| 23:34.38 | ivarv | mchou, also see http://www.mocaedu.com/mt/archives/000340.html |
| 23:35.11 | ivarv | it seems like something as important as voip client delivered via the web would be oss |
| 23:35.22 | mchou | screw it |
| 23:35.39 | mchou | I dont understand the value proposition |
| 23:35.39 | drmessano | There's always red5 |
| 23:35.41 | *** join/#asterisk Psychobilly (n=moi@adsl144-112.kln.forthnet.gr) [NETSPLIT VICTIM] |
| 23:35.44 | drmessano | Open source flash VOIP client |
| 23:35.48 | drmessano | Well |
| 23:35.51 | drmessano | Its more than that |
| 23:35.56 | drmessano | But thats one of its features |
| 23:36.11 | mchou | voip isn't complicated enough by itself, you need to layer flash on top?? |
| 23:36.50 | mchou | I mean even the iphone doesn't cope well with flash |
| 23:36.55 | WimpMan | mchou: It's for people who are unable to install software. |
| 23:37.03 | mchou | so what exactly is the point? |
| 23:37.17 | rasterix | its designed for drmessano lol |
| 23:37.20 | rasterix | JOKE |
| 23:37.32 | mchou | WimpMan: a wifi phone has software pre-installed :) |
| 23:37.34 | WimpMan | Just type in an url into your browser and start talking. |
| 23:37.41 | drmessano | rasterix: Get a clue about asterisk, and stop asking stupid questions, then you can make fun of me |
| 23:37.56 | drmessano | rasterix: Until then, you're just a moron |
| 23:38.09 | ivarv | red5 looks promising |
| 23:38.11 | drmessano | rasterix: Please, move on |
| 23:38.19 | drmessano | ivarv: Red5 is pretty stable too |
| 23:38.20 | mchou | WimpMan: I sure hope that day never comes to pass (at least as far as flash is concerned) |
| 23:38.21 | rasterix | drmessano: i didnt claim to be an expert... you decided that... im just here for advice... |
| 23:38.37 | drmessano | rasterix: HA.. yeah |
| 23:38.51 | WimpMan | mchou: As you can just read. it's already there. |
| 23:39.14 | drmessano | There's lots of Flash based voip clients out there.. WTF |
| 23:39.17 | mchou | WimpMan: than god it's not in wide use :) |
| 23:39.18 | drmessano | Most of the click to call stuff is |
| 23:39.23 | drmessano | LOL |
| 23:39.27 | drmessano | It is in wide use |
| 23:40.10 | drmessano | Do a quick search of Facebook apps and you'll see how rare those flash based voice apps are.. by big names |
| 23:40.19 | drmessano | Click to Call is huge right now |
| 23:40.46 | mchou | there is no reason click to call NEEDs to be flash |
| 23:41.13 | drmessano | Yeah, I mean.. it can be JAVA (hah) or silverlight (HAH) |
| 23:41.27 | WimpMan | Well, it would probably work much better (and more often) if it wasn't. |
| 23:41.48 | ivarv | drmessano, java might make a comeback.. the jvm is sweet, and it's being GPLed |
| 23:42.05 | mchou | ivarv: java will never make a comeback |
| 23:42.17 | WimpMan | But as long as telephony urls are hardly supported, there's no option for Mr Dumbuser. |
| 23:42.17 | drmessano | Java can't come back from something it never was |
| 23:42.51 | mchou | ivarv: the language sucks. A billion situps for "hello world" |
| 23:43.10 | _-Jon-_ | Is there a way to use a shoutcast stream for MOH in 1.4 seeing as mpg123 isn't used anymore? |
| 23:43.13 | mchou | no developer in their right mind can be productive with java |
| 23:43.28 | drmessano | _-Jon-_: It's been done.. Google it |
| 23:43.59 | WimpMan | _-Jon-_: Maybe not by default, but you certainly can use it. |
| 23:44.05 | rasterix | why would you want to use a stream for MOH? |
| 23:44.06 | [TK]D-Fender | _-Jon-_: mpg123 is used plenty |
| 23:44.37 | rasterix | isnt native better to keep processor overhead down? |
| 23:45.02 | _-Jon-_ | One reason for me is mpg123 sucks at randomizing the order of the songs |
| 23:45.18 | _-Jon-_ | Also I've had it where it doesn't play certain mp3s |
| 23:45.37 | rasterix | convert all your moh to gsm? |
| 23:45.37 | drmessano | Shoutcast isn't going to use that much CPU |
| 23:45.41 | drmessano | You're just decoding |
| 23:45.51 | rasterix | ok drmessano i admit i dont know this |
| 23:45.56 | rasterix | before you pounce |
| 23:46.23 | rasterix | moh for me was a quick read... it seemed gsm was the best choice |
| 23:46.23 | drmessano | Good boy |
| 23:46.27 | _-Jon-_ | If this system running Asterisk can't decode mp3s, I think I have bigger issues :) |
| 23:46.28 | rasterix | so thats what i went with |
| 23:46.30 | drmessano | hands rasterix a biscuit |
| 23:47.11 | drmessano | _-Jon-_: There's a food good writeups on using Shoutcast for MOH... if that's what you want to do. |
| 23:47.52 | _-Jon-_ | drmessano, Yes that's what I want :) |
| 23:48.05 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 23:48.16 | Defraz | Has anyone been playing with Chan_mobile? |
| 23:48.22 | drmessano | Defraz: Yes |
| 23:48.33 | drmessano | Defraz: Using it on 1.4 ATM |
| 23:48.46 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 23:48.52 | Defraz | nice I was going to play with that tonight. |
| 23:48.52 | rasterix | anyone know an approx release date for 1.6? |
| 23:49.18 | drmessano | Defraz: Make sure your BT dongle doesn't suck.. it will slow the process |
| 23:49.22 | Defraz | I have a gum stix computer and was going to get asterisk and that running and link my phone to it. |
| 23:49.28 | Defraz | and see what I could come up with. |
| 23:49.41 | Defraz | I have a few different ones. |
| 23:49.45 | drmessano | Also dont get a POS phone tied to it |
| 23:49.55 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) |
| 23:50.00 | drmessano | One where you cant answer it independant of Asterisk lol |
| 23:50.03 | Defraz | Oh it is a Sprint Mogul |
| 23:50.28 | drmessano | I have a shitty samsung I was using, and I could only answer calls via Asterisk once i teamed it |
| 23:50.38 | drmessano | If I tried otherwise, I had no audio |
| 23:50.50 | drmessano | So I would need to wait for the call to time out, then call back lol |
| 23:51.08 | Defraz | Why not just answer the phone tied to the asterisk |
| 23:51.14 | Defraz | IP phone or whatever. |
| 23:51.20 | drmessano | I didnt always want to answer my * phone |
| 23:51.24 | *** part/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 23:51.35 | Defraz | oh I see |
| 23:51.38 | drmessano | Especially if the cell was on my side from coming home ffrom work |
| 23:51.53 | drmessano | But I am gonna use a fixed cell now |
| 23:52.02 | drmessano | Perm power, dedicated to * |
| 23:52.15 | drmessano | I have a deactivated cell on there now for 911 use |
| 23:52.20 | drmessano | Works BRILLIANTLY |
| 23:52.38 | *** join/#asterisk s0lid (n=s0lid@219.93.152.130) |
| 23:54.26 | drmessano | My next trick is gonna be tying into some one of the new ham radios with bluetooth |
| 23:55.12 | rasterix | night all |
| 23:55.39 | *** part/#asterisk ivarv (n=ivar@d66-183-61-180.bchsia.telus.net) |
| 23:56.45 | *** join/#asterisk Tuari (n=Tuari@cpe-76-183-79-199.tx.res.rr.com) |
