| 00:01.06 | TheIzkabola | i'm sorry could you eleberate more? I typed it in and got a list of installed and avaliable packages. thanks for your help! |
| 00:01.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:04.19 | TheIzkabola | outtolunc: and I do not see the kernel-smp-devel listed... |
| 00:06.02 | outtolunc | on centos i think they changed it to pae (checking) |
| 00:06.49 | legis | Is the asterisk online book link working for you guys? |
| 00:07.42 | outtolunc | yeah, pae for x86-only |
| 00:07.54 | TheIzkabola | legis: I'm using it right now and I just hit a roadblock with one of the packages I need, but so far its pretty smooth |
| 00:09.01 | xuser | TheIzkabola: Actually I mean that I can't open the pdf, it won't download for me, is there another link? |
| 00:10.13 | TheIzkabola | outtolunc: pae? is there somthing else I need to install? |
| 00:10.19 | TheIzkabola | xuser: http://www.asteriskdocs.org |
| 00:10.35 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 00:11.12 | Katty | i just ate 3 chicken tacos. |
| 00:11.12 | *** join/#asterisk eltech (n=eltech@ool-4576feda.dyn.optonline.net) |
| 00:11.24 | Katty | oww, the pain. |
| 00:12.01 | _ShrikE | says more hot sauce please |
| 00:12.14 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 00:12.15 | Katty | it was real mexican. not taco bell. |
| 00:12.46 | _ShrikE | I cant really eat taco bell unless I have been drinking |
| 00:12.56 | outtolunc | actually, just the 'kernel' now handles multiple cpus |
| 00:12.58 | _ShrikE | then for some reason it is soooo good |
| 00:13.24 | xuser | TheIzkabola: thx |
| 00:13.28 | Katty | drunken munchies are the worst. |
| 00:13.29 | outtolunc | pae is for 64gb mem |
| 00:13.33 | xuser | _ShrikE: Like ugly chicks :D |
| 00:13.35 | Katty | we have something here in cape called a Slinger |
| 00:14.00 | Katty | it's from one of the gas stations... it's biscuits, with chili, and cheese, and tator tots, and eggs |
| 00:14.02 | TheIzkabola | any idea how I can get the kernel-smp-devel to install? The book said I need it to compile zaptel |
| 00:14.23 | outtolunc | its just kernel-devel |
| 00:14.23 | kmshanah | hi all, I'm looking for some help with an strange ISDN issue where the channels all end up busy: Problem: |
| 00:14.36 | kmshanah | er, http://pastebin.com/d14fb76ea |
| 00:14.38 | Katty | oh and gravy |
| 00:14.45 | TheIzkabola | ok thanks |
| 00:14.45 | Katty | it's just nasty. |
| 00:15.09 | outtolunc | says sorry the box i was looking at earlier had 6gb ram |
| 00:15.29 | JT | Katty: tator tots? |
| 00:15.35 | TheIzkabola | outtolunc: thanks for the assistance |
| 00:15.42 | outtolunc | np |
| 00:15.51 | Katty | JT: umm. bits of hash brown in a cylindrical object |
| 00:16.03 | JT | o_O i see |
| 00:16.25 | Katty | JT: http://www.recipe4living.com/uploadedImages/Recipes/Recipes/tater%20tots.jpg |
| 00:16.43 | JT | ah yeah |
| 00:16.49 | JT | we have the same things with different names |
| 00:16.59 | TheIzkabola | afk |
| 00:17.23 | Katty | what do you call them? |
| 00:20.09 | *** join/#asterisk moy (n=moy@189.169.61.29) |
| 00:21.29 | *** join/#asterisk koenigjm (n=koenigjm@75.150.72.222) |
| 00:22.13 | koenigjm | I am new and having trouble parsing this line: ${IF($[${LEN(${CALLERID(num)})} > 6]?${CALLERID(all)}:${ARG2})})....how does this read? |
| 00:22.57 | outtolunc | IF is a function, not a var |
| 00:23.40 | koenigjm | Is it return CALLERID(all) if CALLERID(num) >6 else the value of $ARG2? |
| 00:23.53 | JT | Katty: i'm really not sure, i've never bought them, but i think at least one major fast food chain has them too |
| 00:24.12 | CrashSys | if length callerid > 6 then return callerid(all) else return arg2 |
| 00:24.14 | JT | it might've been kfc |
| 00:24.37 | angryuser | forget about fastfood eat something decent ;) |
| 00:25.11 | CrashSys | I think you got and extra ) in the else statement |
| 00:25.12 | JT | i don't recall advocating eating fast food |
| 00:25.34 | Katty | JT: it's not very healthy for you. |
| 00:25.39 | CrashSys | You opened with ${ so you should end with a } |
| 00:25.45 | angryuser | it was a global statement |
| 00:26.30 | Katty | you're a global statement. |
| 00:26.39 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 00:26.52 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 00:27.24 | angryuser | huh ? |
| 00:28.14 | Katty | nevermind. |
| 00:30.48 | mosty | i'm having trouble with meetme pin numbers, it does not appear to recognise my dtmf, however this works fine in other parts of the dialplan |
| 00:31.09 | mosty | is there any reason why this might be happening? |
| 00:31.53 | Katty | mosty: pastebinning your meetme.conf and the cli of what's happening when you do it would be helpful |
| 00:33.51 | mosty | the conference is defined as: conf => 1,1234,5678 and the dialplan shows <SIP/340-0079f340> Playing 'conf-getpin' (language 'en') followed by <SIP/340-0079da80> Playing 'conf-invalidpin' (language 'en') after a timeout (regardless of how many digits i enter) |
| 00:34.30 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6e4df4bbe49765ec) |
| 00:36.41 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279551595.dsl.bell.ca) |
| 00:36.48 | mosty | if i remove the two pin numbers, it simply drops me into the conference |
| 00:38.08 | JT | mosty: do you enter a # after the pin? |
| 00:38.27 | mosty | jt: i have tried that, it makes no difference |
| 00:38.59 | JT | you could see if pins work with DISA |
| 00:39.40 | Katty | you could pastebin some cli. |
| 00:41.23 | mosty | ahh, i figured it out, my sip localnet setting had been changed |
| 00:41.58 | mosty | it didn't break rtp but it did break dtmf :/ |
| 00:45.47 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 00:47.35 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 00:50.44 | serialthrilla | is anyone else having a prob with GXP2000 phones not setting the layer 2 priority? |
| 00:50.57 | koenigjm | am I right in thinking that the best way to block outbound callerid is to call Set(CALLERID(all))=unknown before calling Dial()? |
| 00:52.09 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:54.00 | CrashSys | set it to null or 0000000000 |
| 00:54.12 | CrashSys | unknown isn't a valid NPA-NXX |
| 00:54.25 | CrashSys | AND, you can crash some switches/PBX's/CO's :D |
| 00:54.58 | CrashSys | just like trying to dial over a PRI with the phone number w5551212 (on a PRI the w doesn't wait, it sends it as the dial string) |
| 00:55.09 | angryuser | nice thing to know ;) |
| 00:55.36 | CrashSys | Yeah, I found that out the hard way one time |
| 00:55.41 | CrashSys | ofcourse, I didn't let them know |
| 00:56.25 | angryuser | its was on the pri line ? |
| 00:56.39 | CrashSys | yeah |
| 00:56.43 | *** join/#asterisk jpastore (n=jpastore@69.65.65.40) |
| 00:56.59 | CrashSys | CID is only digits, not names |
| 00:57.15 | CrashSys | the name resolution is done by the DIP at the terminating provider |
| 01:02.16 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 01:06.56 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-134-114.dsl.stlsmo.sbcglobal.net) |
| 01:11.46 | *** join/#asterisk sucituanbo (n=blah@c-24-21-121-148.hsd1.mn.comcast.net) |
| 01:14.57 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 01:24.54 | *** join/#asterisk ecrist (n=ecrist@snipe.secure-computing.net) |
| 01:26.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:30.09 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 01:31.33 | Katty | fskrotzki_: hai! |
| 01:34.55 | *** join/#asterisk SilentSounD (n=blayh@c-76-22-178-102.hsd1.tn.comcast.net) |
| 01:37.46 | Katty | so quiet. |
| 01:37.56 | Katty | pokes people. |
| 01:38.01 | SilentSounD | screams |
| 01:38.06 | Katty | :< |
| 01:38.45 | SilentSounD | :] |
| 01:39.21 | outtolunc | checks to see if my lint is still in my pockets |
| 01:41.25 | *** join/#asterisk LemensTS (i=LemensTS@adsl-70-238-134-114.dsl.stlsmo.sbcglobal.net) |
| 01:41.32 | outtolunc | yeah thats right, i also hold others lint too <G> |
| 01:42.37 | Katty | mmmmkay then. |
| 01:42.54 | outtolunc | it has been a long day |
| 01:43.27 | koenigjm | Ok, so in my outbound context I defined the following: |
| 01:43.27 | koenigjm | exten=_*811XXXXXXXXXX!,1,Set(CALLERID(all))=0000000000 |
| 01:43.27 | koenigjm | exten=_*811XXXXXXXXXX!,2,Macro(trunkdial,${broadvox}/${EXTEN:3},"unknown"). I activated the changes then tried to call using *811<DiD> and my caller id still showed up.. what am I missing here? |
| 01:45.08 | _ShrikE | Set(CALLERID(all)=0000000000 <0000000000>) |
| 01:46.12 | LemensTS | What are the 0's for? |
| 01:46.51 | koenigjm | was trying to block out the number |
| 01:47.23 | LemensTS | does it send the cid num as 0000000000 then, or does it make it unknown |
| 01:48.15 | koenigjm | I was hoping for unknown...but 000(etc) would do the trick as well |
| 01:48.23 | koenigjm | just kinda going for anything right now :P |
| 01:48.41 | _ShrikE | Have you tried sending an privacy header? |
| 01:49.13 | koenigjm | no...I don't even know what that is *is really green to all of this* |
| 01:50.09 | unpaidbill | oh crap |
| 01:50.25 | unpaidbill | i lost 2 ports on my server (ZAP FXS!) |
| 01:50.36 | unpaidbill | it is giving me power up issues.. but the other 2 ports on the card work fine |
| 01:50.38 | unpaidbill | i am sad. |
| 01:50.51 | LemensTS | what brand card? |
| 01:51.01 | unpaidbill | TDM2400P |
| 01:52.13 | *** join/#asterisk salaud (n=coates@smtp.jjamusements.com) |
| 01:52.41 | salaud | hello there... anyone have any experience compiling / making NVFaxDetect work under debian/ubuntu? |
| 01:53.58 | salaud | Also... anyone have any experience with multiple public IP's bound to one outside interface with asterisk on debian? |
| 01:54.35 | salaud | I have had no luck trying to compile NVFaxDetect... I'm also using debian packages for asterisk... but, I need to add NVFaxDetect... any pointers? |
| 01:56.11 | mosty | salaud, for sip at least, asterisk can only use one ip address per route. i don't know about iax |
| 01:56.43 | *** part/#asterisk TheIzkabola (n=TheIzkab@c-67-171-143-153.hsd1.or.comcast.net) |
| 01:56.44 | mosty | in other words, you can't have a single instance of asterisk listen on multiple public ip addresses |
| 01:56.57 | salaud | mosty: I don't want it to listen to multiple IP's |
| 01:57.10 | koenigjm | exten=_*811XXXXXXXXXX!,1,Set(CALLERID(all)=0000000000 <0000000000>) |
| 01:57.10 | koenigjm | exten=_*811XXXXXXXXXX!,2,Macro(trunkdial,${broadvox}/${EXTEN:3},"unknown") |
| 01:57.20 | koenigjm | those are the lines I changed them too and still nothing |
| 01:57.21 | salaud | the issue is that there are two IP's... when it sends packets upstream they appear to come from the wrong IP |
| 01:57.39 | koenigjm | is there a place that I can read up on sending privacy headers? or did I do something wrong? |
| 01:57.48 | salaud | so.. the upstream provider (who authenticates based on IP) won't accept the calls. |
| 01:58.03 | salaud | when I remove the alias eth0:0 with the other IP... everything works fine |
| 01:58.11 | mosty | salaud, yeah, as far as i know asterisk will just pick the first public ip |
| 01:58.45 | salaud | The problem is that I need asterisk to listen to two out of three IP's |
| 01:59.05 | salaud | I need it to listen to 192.168.1.x and pu.bl.ic.ip |
| 01:59.31 | salaud | but.. not not the eth0:0 virtual public ip.. |
| 01:59.45 | mosty | salaud, listening on multiple ip's is ok, but each has to have separate routes |
| 02:00.14 | salaud | right... But, I can't even find the syntax for binding to multiple ip's... |
| 02:00.23 | salaud | bindaddr= ??? |
| 02:00.30 | mosty | bindaddr=0.0.0.0 will listen on all interfaces |
| 02:00.39 | salaud | I tried bindaddr = this, that ... but that doesn't work |
| 02:00.41 | mosty | or i think you can specify bindaddr multiple times |
| 02:00.47 | salaud | I tried that too ;) |
| 02:00.59 | salaud | bindaddr only seems to take the last one.. :( |
| 02:01.01 | mosty | and look at the localnet and externip settings in sip.conf |
| 02:01.29 | mosty | use bindaddr=0.0.0.0 and set those other vars i mentioned |
| 02:01.31 | salaud | I did look at those... externip does a good job with NAT... it puts the right thing INSIDE the sip message... but, has no effect on routing |
| 02:02.18 | salaud | mosty: you are confirming my fears that the issue is with one route per nic/IP... This might need to be iptables magic... |
| 02:02.30 | mosty | it's one ip per route |
| 02:02.59 | mosty | if there are multiple possible source addresses when sending packets to some host, asterisk will always use the first of those addresses (ie the others won't work) |
| 02:03.32 | salaud | right... so it can LISTEN to one IP... but when it SENDS it's always the first one.. |
| 02:03.52 | mosty | yes, which means that any sip peer that connects to the second ip won't work |
| 02:03.54 | salaud | so it could bind to the second address... can it can only send to the first.. |
| 02:03.58 | mosty | yes |
| 02:04.04 | mosty | it's a design flaw, i'm told |
| 02:04.29 | salaud | bummer... that's what I thought... this problem makes using heartbeat and High Availability a problem... out of the box |
| 02:04.30 | jaytee | a design flaw in Asterisk or in the SIP protocol? |
| 02:04.51 | salaud | hearbeat adds the shared address as an alias... a SECOND IP.. |
| 02:05.04 | mosty | jaytee, in asterisk |
| 02:05.28 | jaytee | so what about using something like sipX or openSER as a proxy? |
| 02:05.48 | salaud | but... since my provider authenticates based on IP... instead of maybe credentials of some kind at ANY.. IP.. I'm screwed unless I can do some iptables magic... or mod the heartbeat resource scripts |
| 02:05.56 | salaud | jaytee: I thought of just NAT'ing... |
| 02:06.14 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 02:06.17 | salaud | but... I was TRYING to avoid having to get into that with this server.. |
| 02:06.52 | [TK]D-Fender | salaud: Act as thogh you're behind NAT. Bind to a single IP, and force it through externip/localnet |
| 02:06.52 | salaud | If I set the correct public IP up in a NAT .. and send it back to a private IP that moves around... that should work... I was trying to avoid packet mangling to remove latency |
| 02:07.19 | salaud | [TK]D-Fender: that's an interesting ID |
| 02:07.30 | salaud | s/ID/idea.... |
| 02:08.16 | jaytee | salaud, I'd try what he say.....he knows things! |
| 02:08.36 | salaud | hmmm... the problem is that the packets going upstream to the provider have to come from the correct IP.... not INSIDE the SIP message? |
| 02:09.00 | salaud | I can't have both servers (high availability) bound to the correct IP. |
| 02:09.37 | jaytee | are both server functioning or is one in a standby mode |
| 02:09.56 | kmshanah | anyone got any ideas what I can do about my asterisk getting stuck in this state: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion), see http://pastebin.com/dbb2ff89 for more details |
| 02:10.03 | salaud | I believe that externip make sure that certain headers in the SIP tell the other side how to correctly get back to it... but, the address inside (obviously for NAT) doesn't have to match |
| 02:10.03 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:10.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:10.09 | salaud | jaytee: standby.. |
| 02:10.44 | jaytee | salaud, if you cluster them then they'll share the same virtual IP and then HA will handle which one is in active mode if it's done properly. |
| 02:10.45 | salaud | but... heartbeat requires (with scripts in the box) that the interfaces are at least up and bound to the same subnet as the alias IP (eth0:0) when it fails over.. |
| 02:11.23 | salaud | jaytee: I'm doing that correctly... the problem is that the virtual IP is the SECOND IP bound to the NIC... so asterisk is sending from that IP. |
| 02:12.05 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:12.28 | salaud | for instance... eth0 is 78.78.78.01 (the non-shared IP)... when this server is primary it gets eth0:0 alias (78.78.78.05) |
| 02:12.52 | rukus | "requested/capability 0x4/0xf804 incompatible with our capability 0xe008" |
| 02:12.56 | rukus | any pointers ? |
| 02:13.10 | salaud | asterisk is sending packets looking like the come from 78.78.78.01 because this is the first IP (according to mosty).. not from 78.78.78.05 |
| 02:13.28 | [TK]D-Fender | rukus: pick codec that match and next time pastebin the complete failed call. |
| 02:13.53 | [TK]D-Fender | kmshanah: enable PRI debug and try your call again. |
| 02:13.56 | salaud | [TK]D-Fender is saying to change the externip and NAT stuff... but, I don't think this will overcome the fact that packets come from the first public IP address bound to the NIC... and it is this address that my upstream provider is using to authenticate |
| 02:15.02 | [TK]D-Fender | salaud: Hrm.I think I see the issue. |
| 02:15.45 | salaud | It sounds like I need to 1) Proxy with sipX,SER (but this is a point of failure) 2) NAT at firewall (deal with asterisk behind NAT.. ugh) 3) modify the heartbeat scripts to deal with interface never being up in the first place and not adding an alias.. but taking the interface |
| 02:15.57 | *** part/#asterisk ecrist (n=ecrist@snipe.secure-computing.net) |
| 02:17.04 | kmshanah | [TK]D-Fender: it is enabled, intense, nothing is happening |
| 02:17.16 | *** part/#asterisk koenigjm (n=koenigjm@75.150.72.222) |
| 02:17.54 | [TK]D-Fender | kmshanah: pri debug span [number] |
| 02:18.12 | salaud | Other people have done high-availability... but seems that they have mostly done it with PRI or the like... Not VOIP... now "dynamic T1".. ie VOP between you and your provider directly across T1 is what we have... if this was PRI... no problem |
| 02:18.25 | [TK]D-Fender | salaud: * behind NAT is no issue |
| 02:18.55 | salaud | [TK]D-Fender: It SHOULD be as simple as setting up externip and localnet .. right? |
| 02:18.57 | [TK]D-Fender | salaud: You could go "crazy" and run a VM with * in it SNAT-ing its IP :) |
| 02:19.29 | salaud | [TK]D-Fender: you are not suggesting running * in a VM? I thought * couldn't run in a VM? |
| 02:19.30 | mosty | salaud, are you using sip or iax? |
| 02:19.35 | [TK]D-Fender | salaud: In theory, but you're right about the outbound packets. As its stateless it'll go out your uplink even though you'd like it to source as a routed IP |
| 02:19.45 | [TK]D-Fender | salaud: Perhaps some IPTABLES tricky could do? |
| 02:19.47 | salaud | mosty: SIP... |
| 02:19.57 | salaud | [TK]D-Fender: that's right... SNAP in iptables would work |
| 02:20.03 | mosty | salaud, have you investigated using openser in front of asterisk as your HA solution? |
| 02:20.16 | [TK]D-Fender | salaud: I think you could 'prroute it. |
| 02:20.20 | [TK]D-Fender | preroute |
| 02:20.39 | salaud | mosty: I haven't investigated that... what do you mean "as my HA solution"? |
| 02:20.47 | jameswf-home | openser is no longer openser |
| 02:20.54 | Qwell | O.o |
| 02:20.57 | russellb | o.O |
| 02:21.07 | jaytee | huh? |
| 02:21.09 | salaud | I'm trying to make sure that the asterisk server itself doesn't go down... it's running as a PBX... not just a SIP proxy |
| 02:21.13 | Qwell | russellb: remember; 11:42am |
| 02:21.28 | [TK]D-Fender | Qwell: The name jsut changed for it. |
| 02:21.32 | jameswf-home | jbot: openser is also now known as Kamailio |
| 02:21.33 | jbot | okay, jameswf-home |
| 02:21.33 | Qwell | to? |
| 02:21.36 | Qwell | huh |
| 02:21.40 | russellb | sure enough ... http://openser.org/ |
| 02:21.50 | Qwell | wonder why |
| 02:21.53 | Qwell | or does it explain? |
| 02:21.58 | russellb | RTFA |
| 02:22.03 | jameswf-home | http://openser.org/pipermail/users/2008-July/018712.html |
| 02:22.05 | Qwell | ahh |
| 02:22.13 | salaud | zaptel isn't zaptel anymore... right? I heard some rumor |
| 02:22.14 | mosty | salaud, perhaps it would help if you tell us what your design goals for this setup are |
| 02:22.23 | jameswf-home | ~dahdi |
| 02:22.23 | jbot | [DAhdi] Digium/Asterisk Hardware Device Interface. The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi, and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav, |
| 02:22.30 | mosty | salaud, eg are you looking for something that will not drop calls if asterisk segfaults? |
| 02:22.36 | [TK]D-Fender | Qwell: "Kamailio". Because all the more blatantly faggoty named like trixbox & dahdi were taken ;) |
| 02:22.42 | jaytee | Is Kamailio Asterisk's DAHDI? |
| 02:22.50 | jameswf-home | huh |
| 02:22.57 | russellb | I like that name ... |
| 02:23.02 | salaud | mosty: I just need to make sure that all the 100 phones on the system work and that the client is happy.... |
| 02:23.09 | Qwell | It's FAR better than some of the other suggestions. |
| 02:23.11 | Qwell | ... |
| 02:23.26 | salaud | if asterisk segfaults... or if some hardware in the server fails... or if the moon shines the wrong way on the CPU... etc. |
| 02:23.27 | jameswf-home | salaud: if you want to ensure 100 phones work install 150 |
| 02:23.39 | salaud | jameswf-home: ;) |
| 02:23.43 | [TK]D-Fender | russellb: dahdi & "comedian" mail. How the hell is the corporate world supposed to take * seriously if we keep using flakey names for everything? The joke is over. Let it die. |
| 02:23.57 | mosty | salaud, make sure you use safe_asterisk, at least it will try to restart if it segfaults |
| 02:24.09 | jaytee | comedian mail is definitely a no-sell name |
| 02:24.17 | [TK]D-Fender | ^ |
| 02:24.18 | Qwell | russellb: senseofhumor=no |
| 02:24.23 | russellb | Qwell: +1 |
| 02:24.27 | salaud | mosty: I use that.... and I run safe op-panel... this is using freepbx on the front-end... so the client feels cozy.. |
| 02:24.32 | jameswf-home | I am going to start a sohn and sihster project |
| 02:24.39 | jaytee | lol |
| 02:24.45 | russellb | we acutally had a large OEM enter an issue on our issue tracker requesting that Comedian be removed |
| 02:24.53 | mosty | salaud, how often are you seeing asterisk die? |
| 02:25.03 | russellb | so I suggested that we add the option, senseofhumor=no, which would disable it |
| 02:25.20 | kmshanah | [TK]D-Fender: Yeah man, if only it were that simple - fry*CLI> pri show debug => Span 1: Debug: YesIntense: No |
| 02:25.20 | salaud | mosty: well... I didn't see asterisk die much... but.. we did see a NIC die... that sucked |
| 02:25.40 | [TK]D-Fender | kmshanah: Its dialing out a PRI, right? |
| 02:25.42 | salaud | the idea is more HA for the server .. than for asterisk itself... |
| 02:25.51 | [TK]D-Fender | kmshanah: pastebin your zapata.conf |
| 02:26.01 | salaud | asterisk seems to putt right along... |
| 02:26.42 | mosty | salaud, i see, but NIC's dying are relatively rare in my experience |
| 02:27.00 | Qwell | hard to avoid it though |
| 02:27.02 | salaud | mosty: yep... that's just an example... but, the client is paying for the "insurance"... |
| 02:27.25 | salaud | mosty: you had any experience getting NVFaxDetect working? |
| 02:27.52 | mosty | salaud, it might be simpler to setup some hardware monitoring, if anything fails then you get a warning of some sort, and can investigate manually after that |
| 02:27.56 | salaud | I have seen a few things online about how to get it compiled on debian.. but, I have had no success... and I need to work for FreePBX which uses it... |
| 02:28.06 | mosty | i don't do fax with asterisk, so no |
| 02:28.11 | salaud | mosty: unfortuantely... I'm manually 3000 miles away a lot |
| 02:28.42 | kmshanah | [TK]D-Fender: http://pastebin.com/d67aac4e4 |
| 02:28.48 | salaud | anyone here ever got NVFaxDetect compiled on debian? |
| 02:29.07 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 02:29.18 | salaud | mosty: really appreciate the help by the way... |
| 02:29.36 | russellb | So, if not "Comedian Mail", what should we have instead? |
| 02:29.43 | russellb | "Asterisk Mail", "Asterisk Voicemail" ? |
| 02:29.47 | salaud | "Serious Mail" ? |
| 02:30.06 | [TK]D-Fender | kmshanah: ok, "pri debug span 1" should have somwn some traffic |
| 02:30.12 | salaud | "Comedianne Mail" ? |
| 02:30.15 | jaytee | salaud, have you read this? http://www.voip-info.org/wiki/view/NewmanTelOnAsterisk14 |
| 02:30.30 | [TK]D-Fender | russellb: Ask yourself this... how else's VM system even NEEDS a name? |
| 02:30.38 | [TK]D-Fender | who* |
| 02:30.53 | russellb | it's free damnit, we're going to have our name in there :-p |
| 02:31.04 | salaud | jaytee: crap... how come the internets failed all my google searches to find that nugget... you rock |
| 02:31.05 | russellb | you can go buy a prompt if you want it to be something else |
| 02:31.12 | Qwell | [TK]D-Fender: all of them. |
| 02:31.13 | [TK]D-Fender | russellb: the more varied names you throw around to more you dilute the "brand". |
| 02:31.21 | jaytee | salaud, I use this thing called Google |
| 02:31.33 | jaytee | although there is now http://goosh.org |
| 02:31.51 | jaytee | for which I have no idea why it exists but it also works |
| 02:31.52 | salaud | jaytee: I used it too... couldn't find this doc... found a lot of others... I should go back to lycos |
| 02:32.08 | jaytee | Google is my buddy, Google is my friend |
| 02:32.12 | Qwell | [TK]D-Fender: what's the mail system for Nortel? |
| 02:32.18 | jaytee | CallPilot |
| 02:32.21 | Qwell | erm, Avaya I guess it was |
| 02:32.31 | salaud | I can't figure out why FreePBX is not packaged well (and maintained) for debian |
| 02:32.38 | jaytee | and used to be Meridian Mail when it was native running the Unix kernel |
| 02:32.41 | Qwell | salaud: it's incredibly difficult to package well |
| 02:32.44 | Qwell | jaytee: yes, Meridian |
| 02:32.46 | russellb | [TK]D-Fender: Asterisk Voicemail dilutes the brand? |
| 02:32.55 | russellb | i don't buy it |
| 02:32.58 | Qwell | That one said "Meridian Mail" when you connected |
| 02:33.16 | salaud | Qwell: I can imagine...... |
| 02:33.17 | jaytee | Qwell, that's running native. CallPilot replaced it and runs embedded Win2K server. |
| 02:33.30 | [TK]D-Fender | russellb: No, "Comedian Mail" does. "Astrerisk Voicemail" is certainly better. But then do look at the rest of the world. Who else names theirs? |
| 02:33.42 | Qwell | [TK]D-Fender: ^^^ |
| 02:33.51 | russellb | Nortel :) |
| 02:33.52 | [TK]D-Fender | Qwell: And Meridian is antiquated |
| 02:34.02 | [TK]D-Fender | #$&^ Norhell :p |
| 02:34.15 | Qwell | wait, I'm confused. Nortel or Avaya? |
| 02:34.27 | Qwell | I could've sworn we had Avaya phones there... |
| 02:34.33 | russellb | Well, if people are concerned about selling Asterisk voicemail, then they can probably afford the $10 to replace the prompt |
| 02:34.38 | russellb | so, i'm not that worried about it |
| 02:34.43 | jaytee | Nortel is Meridian, Avaya is Avaya Messaging System |
| 02:34.45 | Qwell | and, I know I remember Avaya trucks coming every few weeks |
| 02:34.49 | Qwell | erm, vans |
| 02:34.55 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 02:34.57 | Qwell | (they were a very large customer) |
| 02:35.04 | salaud | jaytee: just for grins... http://www.google.com/search?q=nvfaxdetect+debian+compile&ie=utf-8&oe=utf-8&aq=t&rls=com.ubuntu:en-US:unofficial&client=firefox-a |
| 02:35.06 | Qwell | ((where they == "us")) |
| 02:35.07 | [TK]D-Fender | russellb: Thing is, even if we did, would you ensure its being the new "permanent" replacement? |
| 02:35.25 | russellb | No ... I'm saying people can buy a custom replacement for themselves if they want |
| 02:35.26 | salaud | jaytee: this document not on the first three pages of hits... nvfaxdetect debian compile |
| 02:35.26 | jaytee | although Meridian is what they used to sell running a Unix kernel co-resident on the PBX from the main PBX CPU and kernel |
| 02:35.29 | Qwell | [TK]D-Fender: it would be a config option |
| 02:35.35 | jameswf-home | hmmm .07% of my blog readers used a playstation 3 werd |
| 02:35.39 | jameswf-home | *wierd |
| 02:35.41 | russellb | to replace it, we'd have to get it in english, spanish, and french ... |
| 02:35.51 | jaytee | CallPilot is the new Nortel voicemail running on an embedded Win2K system. |
| 02:35.56 | [TK]D-Fender | russellb: lol. Now we find a better supporting name, and you tel me you wouldn't even prop it up :) |
| 02:35.58 | russellb | wonders if the spanish and french prompts say "comedian mail", heh |
| 02:36.01 | *** join/#asterisk moy (n=moy@189.169.61.29) |
| 02:36.12 | [TK]D-Fender | russellb: they do. |
| 02:36.27 | russellb | [TK]D-Fender: sure, if you get all 3 languages in high quality format ... then yes, we would at least add it as an option, if not a complete replacement |
| 02:36.28 | Qwell | That joke probably doesn't carry over well into Spanish.. |
| 02:36.29 | russellb | you can quote me |
| 02:36.33 | salaud | Maybe esperanto for "voicemail" |
| 02:36.33 | russellb | Qwell: yeah ... |
| 02:36.41 | salaud | voice is voki... |
| 02:36.48 | jaytee | salaud, all I did was search for NVFaxDetect and I saw a link from Digium regarding anyone having problems compiling it for Asterisk. That's the link I got. |
| 02:36.49 | Qwell | we need Klingon prompts. |
| 02:37.33 | russellb | i want an adult prompt set |
| 02:37.36 | jaytee | the Spanish version of Asterisk calls it Carlos Mencia Mail |
| 02:37.49 | jaytee | "K'Plak!!!" |
| 02:37.51 | salaud | jaytee: you are right... the more general search worked better.... I don't want google anymore... I want a research assistant |
| 02:38.10 | russellb | Instead of the jingle when someone joins a conference ... "Some Mother F**ker just joined the conference." |
| 02:38.14 | jaytee | salaud, have you tried http://goosh.org ? |
| 02:38.18 | Qwell | jaytee: terrible |
| 02:38.22 | salaud | ever wonder how people survived before the internet? Assistants.. |
| 02:38.29 | jaytee | Qwell, what the Carlos or goosh? |
| 02:38.32 | Qwell | russellb: have to change how that prompt is done... |
| 02:38.34 | Qwell | it's sad |
| 02:38.41 | jaytee | what? |
| 02:38.45 | Qwell | (unless Mark actually ended up fixing it.. he was talking about it once) |
| 02:38.50 | Qwell | jaytee: Carlos - he's bad. |
| 02:38.51 | russellb | Qwell: nope, it's still in the code |
| 02:38.53 | Qwell | terrible even |
| 02:38.56 | jaytee | Qwell, ok |
| 02:38.58 | salaud | jaytee: not yet.. check it out... does it work from my iphone? ;) |
| 02:38.59 | jaytee | I agree |
| 02:39.01 | Qwell | russellb: it's silly..heh |
| 02:39.05 | russellb | who wants to buy me an iphone |
| 02:39.12 | russellb | that would be greaaat |
| 02:39.13 | jaytee | goosh.org is a google shell |
| 02:39.20 | salaud | russellb: Tim O'Reilly ;) |
| 02:39.21 | Qwell | jaytee: oh, yeah, I think I saw that |
| 02:39.25 | Qwell | pretty cool |
| 02:39.48 | Qwell | jaytee: And to call him a Comedian... |
| 02:39.53 | salaud | russellb: Tell him you will write something cool and show it off at E-Tel. |
| 02:39.54 | Qwell | is a huge blow to comedians everywhere. |
| 02:40.03 | jaytee | Qwell, he steals most of his jokes |
| 02:40.10 | Qwell | and they STILL aren't funny. |
| 02:40.11 | russellb | and he's still not funny |
| 02:40.13 | salaud | Qwell: don't forget Comedianne's too |
| 02:40.21 | russellb | salaud: heh ... too bad E-Tel doesn't exist anymore |
| 02:40.32 | Qwell | bites his tongue regarding etel |
| 02:40.33 | salaud | well... ok.. maybe there is that |
| 02:40.44 | Qwell | no, you know what, I'm gonna say it. |
| 02:40.45 | jaytee | well, if you want "still not funny" just stick around for some of my really bad puns. |
| 02:40.48 | salaud | I hadn't heard that... didn't go the last year (or two) |
| 02:40.53 | russellb | stares at Qwell |
| 02:41.00 | salaud | waits ... breathlessly |
| 02:41.05 | Qwell | They scheduled 3 conferences the same week... |
| 02:41.13 | russellb | heh |
| 02:41.17 | Qwell | well, they scheduled 2, but it was at the same time as a third |
| 02:41.24 | Qwell | and etel died because of it |
| 02:41.38 | russellb | i never went, so i don't feel like i'm missing out |
| 02:41.38 | jaytee | the best comedian in my opinion passed away last month |
| 02:41.42 | Qwell | etel *was* actually pretty decent |
| 02:41.58 | salaud | I haven't talked to surj in a while... he's here in Portland.. and I still never see him... |
| 02:42.13 | salaud | at least I think he is still here... I liked E-tel ok... learned a few things... |
| 02:42.16 | Qwell | salaud: Surj was an interesting person. |
| 02:42.22 | Qwell | s/was/is/ |
| 02:42.31 | russellb | anyone going to Astricon? |
| 02:42.36 | Qwell | :( |
| 02:42.38 | salaud | perhaps he has become boring |
| 02:42.39 | jaytee | I wish |
| 02:42.42 | russellb | Qwell: i know, i'm sorry :( |
| 02:42.42 | file | hugs Qwell |
| 02:42.52 | russellb | Qwell: wasn't my call :( |
| 02:42.54 | Qwell | glares at Polycom. |
| 02:43.05 | file | wishes we had infinite currency |
| 02:43.26 | jameswf-home | file: do what i do just write a check |
| 02:43.36 | russellb | I'd probably still be working on Asterisk if I had infinite money, heh |
| 02:43.39 | Qwell | jameswf-home: write me a check, kthx |
| 02:43.52 | Qwell | buy my plane ticket, and let me crash at your pad - it'll be all good |
| 02:43.57 | salaud | goosh is pretty cool... |
| 02:43.57 | russellb | that's really all I would do ... is work on open source, specifically asterisk most of the time |
| 02:43.59 | jaytee | if I had semi-inifinite currency I'd charter a jet and a full floor of suites at a 5 star and drinks would be on me. |
| 02:44.11 | file | russellb: who would be our leader! |
| 02:44.21 | Qwell | file: I suspect aliens. |
| 02:44.22 | russellb | file: i said i'd still be working on asterisk! |
| 02:44.53 | *** part/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 02:44.54 | salaud | thanks for the help folks... I'm going to try to get some sleep now.. third all-nighter in a row to get off TrixBox and onto Debian/FreePBX/HighAvailability |
| 02:44.59 | file | russellb: true |
| 02:45.07 | jaytee | aliens? the guys that mow the lawn or the little grey men hangin at Roswell? |
| 02:45.12 | salaud | waves |
| 02:45.13 | jameswf-home | my wife says no coders are allowed to come play anymore.. there was this whole microwave incident |
| 02:45.31 | Qwell | pfft, excuses |
| 02:45.34 | Qwell | see msg :p |
| 02:45.46 | russellb | [TK]D-Fender: you going to astricon/ |
| 02:45.48 | russellb | ? |
| 02:45.54 | Qwell | russellb: he needs to |
| 02:45.58 | russellb | indeed |
| 02:46.30 | [TK]D-Fender | russellb, Qwell : It's be nice, but thats a fair amount of $ and I don't have my passport yet |
| 02:46.47 | Qwell | yeah...stupid cost |
| 02:46.55 | [TK]D-Fender | &^#%$&^#$ DHS |
| 02:46.58 | russellb | [TK]D-Fender: get passport, and make work send you |
| 02:47.14 | [TK]D-Fender | russellb: We run a supported GUI... no point :) |
| 02:47.21 | jameswf-home | [TK]D-Fender: there are folks here that can get you papers for $50 |
| 02:47.23 | russellb | O.O |
| 02:47.29 | russellb | [TK]D-Fender: what do you run?! |
| 02:47.33 | *** mode/#asterisk [-o [TK]D-Fender] by Qwell |
| 02:47.37 | [TK]D-Fender | russellb: ScopServ. |
| 02:47.37 | Qwell | You used to be cool, man. |
| 02:47.39 | [TK]D-Fender | :( |
| 02:47.40 | LemensTS | FUNDRAISER FOR TKD-FENDER |
| 02:47.41 | russellb | all that GUI bashing ... |
| 02:47.55 | file | Qwell: man, such hate |
| 02:47.57 | Qwell | :D |
| 02:48.04 | Qwell | fortunately, I'm not chanserv |
| 02:48.27 | LemensTS | Digium should pay for you as much free help as you give out |
| 02:48.29 | Qwell | [TK]D-Fender: never seen that one...how is it? |
| 02:48.32 | *** mode/#asterisk [-o [TK]D-Fender] by russellb |
| 02:48.34 | [TK]D-Fender | russellb: I bash people who expect US to support their GUI's :) |
| 02:48.50 | russellb | i'm going to troll on you for forever about that |
| 02:49.00 | russellb | LemensTS: he tells people to buy non-digium products, though :) |
| 02:49.04 | [TK]D-Fender | Qwell: As GUI's go, pretty good actually. The developers are in Quebec and I've met them several times :) |
| 02:49.05 | jameswf-home | GUI's arent bad it is what they produce and atract that is bad |
| 02:49.15 | jaytee | I'd kick in a few buck for [TK]D-Fender to go if we actually started a collection |
| 02:49.29 | [TK]D-Fender | russellb: I'm sure you will. |
| 02:50.08 | jameswf-home | I could fly to astricon... like 3 minutes in a Helocopter... that could be cool |
| 02:50.08 | [TK]D-Fender | russellb: How often do you see me ask for help on just about ANYTHING in here? |
| 02:50.16 | jaytee | I've been messing with the Asterisk GUI because I wanted something that wouldn't screw up my dialplan like that Waring Blender PBX called Trixbox |
| 02:50.31 | Qwell | [TK]D-Fender: probably less often than me :p |
| 02:50.49 | [TK]D-Fender | Qwell: Potentially accurate :) |
| 02:51.04 | jameswf-home | jaytee: technicaly trixbox doesnt do configs freepbx does |
| 02:51.06 | jaytee | and all I really wanted was the ability to control the dialplan logic from the configs through a command line editor but let some admin type add extensions and SIP or IAX phones. |
| 02:51.14 | Qwell | jameswf-home: trixbox just adds remote roots |
| 02:51.19 | Qwell | coughs and shuts up |
| 02:51.23 | jaytee | jameswf-home, yeah, that's kinda what I meant. |
| 02:51.26 | Qwell | did they ever fix that yet? |
| 02:51.39 | [TK]D-Fender | goes back to commandeering Qwell's chan_skinny botnet... |
| 02:51.53 | [TK]D-Fender | zing! |
| 02:51.56 | jameswf-home | Qwell: did you see my video..... text was kinda crappy but timeline was dead on |
| 02:52.02 | Qwell | jameswf-home: yep |
| 02:52.17 | jameswf-home | heh my mac at work just died... damnit |
| 02:52.22 | file | [TK]D-Fender: noooooo! now we can't get infinite currency |
| 02:52.25 | russellb | [TK]D-Fender: every second!!11 |
| 02:52.39 | jaytee | chan_skinny is bad! your server will lose weight and start having fainting spells like one of the Olsen twins |
| 02:52.42 | jameswf-home | I want some of that internet money |
| 02:52.54 | *** part/#asterisk SilentSounD (n=blayh@c-76-22-178-102.hsd1.tn.comcast.net) |
| 02:53.23 | Qwell | internet money <3 |
| 02:53.56 | jaytee | I don't understand why we can't print our own like the government does when they run out of it. It's not like they have anything to back it up with either. |
| 02:54.56 | jameswf-home | the government is 9876 trillion in debt and the banks still cash their checks...... awesome |
| 02:55.31 | jaytee | yeah! and watch what happens if you write a check for 10 bucks when overdrawn |
| 02:56.03 | Qwell | jaytee: it goes through - they just charge $35 |
| 02:56.06 | jameswf-home | well thats where most of the national debt came from.... overdraft fees |
| 02:56.15 | jaytee | lol |
| 02:56.53 | [TK]D-Fender | Qwell: Oh, and the only reason I'm running that GUI at work is that it was the only way I could sell * as a solution and avoid getting locked in to an Avaya IP Office or ---OMFG~NO--- Nortel BCM |
| 02:56.56 | jameswf-home | without overdraft fees the govt would only be down a hundred bucks |
| 02:57.28 | jaytee | OMFG-NO---Nortel BCM is why I'm in here |
| 02:57.39 | jameswf-home | ~bcm |
| 02:57.44 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 02:57.54 | jameswf-home | ~spm |
| 02:57.54 | jbot | [spm] South Park Mexican |
| 02:57.55 | [TK]D-Fender | jaytee: You poor poor sap |
| 02:57.57 | jameswf-home | yay |
| 02:58.15 | LemensTS | jaytee: the government is not in controll of the money, the federal reserve is...thats our problem |
| 02:58.27 | jaytee | [TK]D-Fender, no because I avoided it and I'm using * to move to VOIP but I'm doing it in phases. |
| 02:58.29 | angryuser | i am searching for a reliable 4xT1 gateway, thank you |
| 02:58.42 | jameswf-home | your welcome ? |
| 02:58.51 | [TK]D-Fender | angryuser: AudioCodes Mediant |
| 02:59.23 | angryuser | ok, what do you think about patton ? |
| 02:59.26 | jaytee | and out of all the possible solutions ONLY Asterisk was flexible enough to enable me to have 4 digit dial between the two PBXs and migrate users over a 2 year period. |
| 02:59.41 | jaytee | that and Digium's T1 cards |
| 02:59.58 | [TK]D-Fender | angryuser: "OK" from what I've heard. Very little word however. Not very commonly used by people here. |
| 03:00.06 | jameswf-home | how do the gyros do under machine gun fire http://www.crazynfunny.com/wp-content/uploads/2008/07/chinesecops.jpg |
| 03:00.21 | jaytee | AudioCodes gets really high marks but they are very pricey |
| 03:01.01 | angryuser | [TK]D-Fender ok, i knew about them, but price ... |
| 03:01.08 | jameswf-home | could use a quad t1 card with a simple dialplan |
| 03:01.44 | angryuser | i cant it's not for asterisk based ;( |
| 03:02.58 | mosty | angryuser, gateway to what? |
| 03:03.15 | angryuser | someting like this http://www.voipsupply.com/product_info.php?products_id=1874 |
| 03:04.32 | *** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-34.syrcny.fios.verizon.net) |
| 03:05.09 | jaytee | [TK]D-Fender, did you know that BCM is dead and that Nortel is now using the code base for sipXecs and calling it Software Communications Server 500 ? |
| 03:05.09 | jameswf-home | angryuser: for under 3000 you can make an asterisk based gateway |
| 03:05.15 | jameswf-home | put it inline like any other gateway |
| 03:05.19 | [TK]D-Fender | jaytee: Nope, news to me. |
| 03:06.03 | jaytee | yeah, they've been doing it since April. They've even contributed code to SIPfoundry for their OSS version. |
| 03:06.43 | angryuser | jameswf-home that's true, but it wont be such a solid as that device in exemple, no mov parts ...etc |
| 03:07.26 | angryuser | i can get ssd, cut logs ect , but not the same thing |
| 03:09.14 | *** join/#asterisk Segnale007 (n=Segnale0@host188-251-dynamic.26-79-r.retail.telecomitalia.it) |
| 03:09.32 | jaytee | cool!!! Statins may reduce dementia risk. |
| 03:10.02 | jaytee | maybe all that money I pay for Lipitor will pay off |
| 03:17.44 | *** join/#asterisk d-k-t-2 (n=dt@vpn21.witopia.net) |
| 03:21.25 | *** join/#asterisk minthome (n=mintee@68.45.231.166) |
| 03:24.19 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 03:30.29 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:41.40 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3101f100b1ad955d) |
| 03:42.28 | baliktad | can anyone have a look and tell me what I can do to allow asterisk to boot? Right now it is hanging when I start with 100% cpu usage: http://pastebin.ca/1085927 |
| 03:47.28 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 03:48.13 | [T]ank | I still have some T1 cards to sell. I have another Sangoma A104D. Any one interested? $500 shipped to US addresses |
| 03:49.04 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 03:49.15 | [T]ank | Or I have some digium cards but they are not going as cheap as the sangoma |
| 03:50.21 | [TK]D-Fender | baliktad: add "noload => chan_skinny.so" to modules.conf |
| 03:50.51 | [TK]D-Fender | baliktad: And you probably shouldn't be running beta software in production if you even have to ask that. |
| 03:51.10 | baliktad | This is my test box to learn on :) |
| 03:51.57 | [TK]D-Fender | baliktad: Try learning where you aren't going to run into stuff you didn't need to break yourself :p |
| 03:52.51 | jaytee | good advice |
| 03:53.04 | jaytee | learning on beta is just increasing the learning curve |
| 03:53.22 | baliktad | I only have windows platform at the moment and the only instructions I could find for running on cygwin were on the 1.6 beta |
| 03:53.39 | jaytee | ewwwwwww! no freakin wonder |
| 03:53.54 | [TK]D-Fender | no *&#%^$ comment. |
| 03:54.00 | jaytee | lol |
| 03:54.16 | [TK]D-Fender | goes off to oil & hone his sword |
| 03:54.18 | LemensTS | baliktad: run vmware with a linux image in it |
| 03:54.22 | *** join/#asterisk rfernandez (n=rfernand@189.136.64.128) |
| 03:54.35 | rfernandez | hiya! its fine to use stun server to fix sip nat issues? |
| 03:54.37 | baliktad | I know it doesn't live up to everyone's pristine image of what a * box should be but I have to start somewhere |
| 03:54.53 | [TK]D-Fender | baliktad: And you chose THERE. |
| 03:55.06 | jaytee | and that's definitely the WRONG place |
| 03:55.10 | [TK]D-Fender | rfernandez: No, * does not even support it, let alone need it. |
| 03:55.17 | [TK]D-Fender | rfernandez: Go read the guide : |
| 03:55.19 | [TK]D-Fender | ~sipnat |
| 03:55.20 | 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 |
| 03:55.21 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 03:55.30 | baliktad | every time I come in here and ask I question I take a beating about something |
| 03:55.40 | baliktad | I spent literally all day reading everything I could |
| 03:55.52 | baliktad | just to avoid asking a question about how to get up and running |
| 03:56.07 | rfernandez | [TK]D-Fender, ah ok but, how can i solve the issue of the sipphardphone in the house of my boss and the * in the office? =S he can listen but cant be listened in remote way =S |
| 03:56.10 | baliktad | the only instructions I could find that were halfway useful were using the 1.6 beta |
| 03:56.20 | baliktad | I'm sorry a windows box is the only box I have at the moment |
| 03:56.22 | [TK]D-Fender | rfernandez: Read. The. Guide. |
| 03:56.25 | jaytee | what LemensTS suggested would be a better alternative if you're stuck with only one machine running Windows. A VM will work better than cygwin |
| 03:56.30 | [TK]D-Fender | baliktad: Run a VM. |
| 03:56.36 | rfernandez | [TK]D-Fender, alright |
| 03:56.40 | LemensTS | baliktad: vmware workstation with linux image, and use asterisk 1.4. |
| 03:56.42 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:56.54 | jaytee | workstation isn't free, server is |
| 03:56.59 | rfernandez | [TK]D-Fender, good guide thanks! =D! |
| 03:57.08 | jblack | Dude, just do Xen. |
| 03:57.14 | baliktad | *sigh* |
| 03:57.19 | jaytee | jblack, he's running Windows |
| 03:57.34 | LemensTS | Its free for 30 days, or run vmware server is free but it is too confusing for its own good. |
| 03:58.00 | jaytee | I've never found VMWare server to be at all confusing. |
| 03:58.30 | LemensTS | jaytee: the new web based version is very confusing compared to the previous one |
| 03:58.47 | jaytee | I run XP in one on my workstation at work that runs Ubuntu Gutsy and I run sipX as a proxy in a VM on a Win2K3 server. |
| 03:59.29 | jaytee | version 1.0.5 |
| 03:59.30 | baliktad | before I go off and set up a brand new VM for this, please everyone think of the worst platform possible |
| 03:59.38 | baliktad | then direct your snide remarks at me for choosing that platform |
| 03:59.46 | jaytee | Vista |
| 03:59.58 | jaytee | with cygwin |
| 04:00.02 | baliktad | because I will inevitably choose that to run 1.4 in |
| 04:00.03 | LemensTS | lmao |
| 04:00.27 | jaytee | or better yet, Windows 95 with Microsoft Bob |
| 04:00.27 | LemensTS | Just use your XP, XP is the best windows |
| 04:01.09 | baliktad | been there, see above |
| 04:01.43 | jaytee | baliktad, how much free drive space do you have on that system? |
| 04:02.11 | baliktad | 50 GB or so |
| 04:03.25 | jaytee | download the SystemRescueCD, burn the ISO and boot to it, use GParted to resize your NTFS partition but run Disk Defragmenter about 10 times first. Then you can set up the free space as linux for dual boot. |
| 04:03.31 | [TK]D-Fender | LemensTS: XP = Deluxe crap :) |
| 04:05.10 | jaytee | Vista = Ultimate Crap |
| 04:05.33 | baliktad | a VM I can do |
| 04:05.37 | jaytee | "oooh! look!!! I have transparent title bars!!!" |
| 04:05.43 | baliktad | dual boot not so much |
| 04:06.07 | [TK]D-Fender | baliktad: Why not? |
| 04:06.08 | jaytee | dual boot with Debian 4.0 or Ubuntu is a breeze |
| 04:06.24 | baliktad | still need that xp box running :S |
| 04:06.30 | jaytee | if you have XP already on there it'll detect it and setup the GRUB bootloader for dual boot for you. |
| 04:06.32 | [TK]D-Fender | baliktad: .... because? |
| 04:06.50 | jaytee | games |
| 04:07.07 | jaytee | he won't be able to play Halo without it. |
| 04:07.15 | [TK]D-Fender | jaytee: what.. playing them SIMULTANEOUS to learning *? |
| 04:07.43 | jaytee | [TK]D-Fender, hey! I'm just guessing here! |
| 04:08.57 | jaytee | hell, I'm still cleaning the spew off my monitor from when I had a mouthful of Coca Cola and saw the words Windows XP with cygwin. |
| 04:09.10 | jaytee | damn near choked |
| 04:09.12 | baliktad | the why is less important than the fact that I can't be rebooting into another OS all the time, I just have to use what I can |
| 04:09.49 | jaytee | then I'd suggest blowing away windows, installing a linux distro like Debian or CentOS and running XP in a VM. |
| 04:10.10 | jaytee | unless you have to have advanced graphics capabilities for games. |
| 04:10.23 | [TK]D-Fender | jaytee: wonder how * would do under VirtualBox indside of Windows.... |
| 04:10.46 | jaytee | probably crappy but I've never run Virtualbox |
| 04:11.07 | [TK]D-Fender | jaytee: I tried it a handful of times. |
| 04:11.55 | jaytee | I've tested the same server hardware with VMWare Server running on Win2K3 Server and CentOS 5.1 and any VM I run runs better on a linux host using VM. |
| 04:12.10 | jaytee | [TK]D-Fender, and how'd it run? |
| 04:12.32 | unpaidbill | linux is much better with xen + multi core procs from what i hear |
| 04:12.41 | [TK]D-Fender | jaytee: Slow, but functional. |
| 04:12.41 | unpaidbill | i havent tried it yet though, unfortunately |
| 04:12.51 | [TK]D-Fender | jaytee: Only actually tested with Ubuntu mind you. |
| 04:12.58 | [TK]D-Fender | jaytee: And not with * |
| 04:13.02 | jaytee | [TK]D-Fender, I couldn't run * in a VM due to the TE212P card |
| 04:13.10 | unpaidbill | i actually do testing with two VMware servers running ubuntu and asterisk |
| 04:13.13 | [TK]D-Fender | jaytee: Yeah.. that would suck :0 |
| 04:13.39 | unpaidbill | i can do like 150+ calls (playback 10-30 sec audio) between them |
| 04:13.41 | jaytee | I wouldn't trust my PRI's to a VM even with a Digium written virtual driver for VMWare. |
| 04:14.22 | unpaidbill | anyone know why when i try to load my TDM2400P it bitches about no power to the first 2 ports, but the other 6 work fine? |
| 04:14.26 | *** join/#asterisk zoid_99 (n=zoid_99@24.96.150.105) |
| 04:14.28 | unpaidbill | it just started ahppening after a reboot :/ |
| 04:15.01 | jaytee | Xen has made some serious strides and I've been meaning to play with it some but unfortunately I'm busy with working on LumenVox stuff right now. |
| 04:15.39 | unpaidbill | ooh lumenvox! how do you like it |
| 04:15.47 | unpaidbill | i havent used it for 1.5 years but it was pretty decent then |
| 04:15.55 | jaytee | it's really not bad at all |
| 04:15.58 | Yourname`` | [TK]D-Fender: Remember we talked earlier about agents logging into multiple queues? What do you suggest is the best way to handle that? |
| 04:16.05 | unpaidbill | does it do realtime voice recognition now? |
| 04:16.18 | jameswf-home | Asterisk runs okay in virtualbox under windows |
| 04:16.21 | jaytee | and even though I have the starter kit which comes with no support their rep has called me and given me some help. |
| 04:16.24 | unpaidbill | or is it still waiting for X ms of silence before it tries to do the recog |
| 04:16.27 | jameswf-home | s/windows/linux/ |
| 04:16.38 | unpaidbill | man if it could do realtime voice recognition i'd be all over it |
| 04:16.50 | jaytee | unpaidbill, by realtime you mean no lag? I don't notice any. |
| 04:16.53 | [TK]D-Fender | Yourname``: you never said whether this agent was only to join this other queue as some sort of "guest", not as a regular member |
| 04:17.25 | unpaidbill | jaytee as in, if i say hello my name is bill, will it recognize each word as i say it, or will it wait till i finish speaking and then analyze the audio |
| 04:17.44 | jameswf-home | unpaidbill: isnt that what humans do |
| 04:17.56 | [TK]D-Fender | I can do realtime voice recognition. Works almost 100% |
| 04:17.56 | jaytee | I believe it recognizes each word as it's spoken now. |
| 04:17.58 | jameswf-home | store and foreward |
| 04:18.00 | unpaidbill | i'm trying to integrate asterisk into my love doll |
| 04:18.24 | unpaidbill | i want it to react to my sweet nothings as i whisper them |
| 04:18.32 | [TK]D-Fender | unpaidbill: Don't overcomplicate 1/3 of her orifices like that! |
| 04:18.37 | unpaidbill | haha |
| 04:18.45 | unpaidbill | KISS applies in so many ways, here! |
| 04:19.14 | jaytee | unpaidbill, there are programmable audio chips that are only like 10 bucks at Radio Shack that could fit the "ooooh, ooooh, Oh!!!! Unpaidbill, you're the BEST!!!!" in a repeating loop. |
| 04:19.28 | unpaidbill | no you misunderstand me, i want her to react to MY cooing! |
| 04:19.56 | unpaidbill | i'm going to install a soekris board in one of her boobs or buttcheeks |
| 04:20.04 | unpaidbill | ok i should stop now |
| 04:20.14 | jameswf-home | unpaidbill: only women that do that real or fake are hookers |
| 04:20.26 | jaytee | I was making out one time and I started whispering "zero, zero, zero, zero" in her ear and she says, "What are you doing?" "I'm whispering sweet nothings in your ear!" |
| 04:20.46 | unpaidbill | seriously though, my TDM2400P is jerking me around :/ and i cant connect to my work network because i f'd up the PIX before i left |
| 04:20.47 | unpaidbill | awesome! |
| 04:20.54 | unpaidbill | i am on a roll today |
| 04:21.03 | jaytee | ouch, so no VPN? |
| 04:21.10 | unpaidbill | hrmm |
| 04:21.11 | unpaidbill | actually |
| 04:21.15 | unpaidbill | i didnt even try that |
| 04:21.37 | jaytee | how many modules on your TDM2400P? |
| 04:22.08 | unpaidbill | hah |
| 04:22.10 | unpaidbill | it works! |
| 04:22.15 | unpaidbill | now i can fix the PIX and ssh in |
| 04:22.21 | Yourname`` | [TK]D-Fender: Yeah but I didn't quite understand what you meant by guest or permanent member. Basically, there is going to be these 2 queues where 1 agent is logged in to. He has a Polycom phone, at times he will log in only to queue1 and sometimes both queues. He should also be able to log out of these two queues. I would love to have the beep-in-ear AgentLogin functionality but you said... |
| 04:22.22 | Yourname`` | ...there's no alternative. |
| 04:22.33 | jaytee | unpaidbill, how many modules? |
| 04:22.35 | unpaidbill | i didnt even consider VPN, im kind of amused that it works considering i changed the IP block today and didnt update the vpn stuff |
| 04:22.40 | jaytee | FXS or FXO? |
| 04:22.41 | unpaidbill | 2 modules, 4 FXS on each |
| 04:22.50 | unpaidbill | 8 ports total |
| 04:22.53 | unpaidbill | ports 1 and 2 are fubar |
| 04:22.55 | jaytee | ah, does each module have it's own power connector? |
| 04:22.59 | unpaidbill | nope |
| 04:23.03 | unpaidbill | TDM2400P has one power connector |
| 04:23.24 | jaytee | ah. then the first module is probably crap. try moving them around. |
| 04:23.27 | unpaidbill | and i know it's getting power because the other ports work on it:/ |
| 04:23.33 | unpaidbill | it worked fine before my reboot |
| 04:23.38 | *** part/#asterisk zoid_99 (n=zoid_99@24.96.150.105) |
| 04:23.41 | unpaidbill | but i did push on the pci card lightly while the system was up |
| 04:23.49 | jaytee | AHA! |
| 04:23.51 | unpaidbill | damn thing is screwed in the chassis so tight i seriously doubt i moved anything though |
| 04:24.00 | unpaidbill | i dunno |
| 04:24.08 | unpaidbill | i just hope i didnt bust the FXS module |
| 04:24.12 | jaytee | down the bitch and reseat the card |
| 04:24.13 | Yourname`` | unpaidbill: Try a yank. Solves anything! |
| 04:24.25 | unpaidbill | i'm fond of lifting and dropping myself |
| 04:25.01 | jaytee | "just spit in the back and whack it a good one!" - Andy Griffith in No Time For Sergeants |
| 04:25.09 | unpaidbill | haha |
| 04:25.13 | unpaidbill | ok im going to fix my damn PIX |
| 04:25.16 | unpaidbill | vpn == no internet fun |
| 04:25.17 | unpaidbill | bbl |
| 04:25.22 | jaytee | later |
| 04:25.43 | jaytee | I think I'm gonna turn in. |
| 04:26.02 | jaytee | nite [TK]D-Fender |
| 04:26.18 | Yourname`` | I think I'm going to be one of the last helpees of [TK]D-Fender |
| 04:26.33 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:26.57 | [TK]D-Fender | Yourname``: Agents don't log into queues. Agents LOG IN. |
| 04:27.45 | Yourname`` | [TK]D-Fender: Ok, cool. So how can we do this you think? |
| 04:27.52 | Yourname`` | I mean the best way that you think is best. |
| 04:28.03 | [TK]D-Fender | Yourname``: You need to actually understand queues & members. |
| 04:28.50 | Yourname`` | No no, I mean regarding the best way to do the one agent logging into multiple queues part. |
| 04:29.03 | unpaidbill | hey i tricked the vpn |
| 04:30.15 | [TK]D-Fender | Yourname``: AGENTS DON'T LOG INTO QUEUES. |
| 04:31.40 | Yourname`` | [TK]D-Fender: Members do. |
| 04:31.42 | Yourname`` | Right. |
| 04:31.50 | [TK]D-Fender | Yourname``: No. |
| 04:32.06 | Yourname`` | Ok, lol.. omg! You know what I mean tho |
| 04:32.30 | [TK]D-Fender | Yourname``: You seem to have lost the most basic perceptual skills. |
| 04:32.45 | Yourname`` | lol it's been some time :P |
| 04:34.02 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 04:34.16 | [TK]D-Fender | Yourname``: Queues have members. Agents can be members as can other CHANNEL TYPES. AGENTS log into THEMSELVES. You can have an agent log and and if no queue INCLUDES him as a member and you don't otherwise specifically reference the agent then it is meaningless. |
| 04:34.52 | [TK]D-Fender | Yourname``: the Agent has to be a MEMBER of both queues. When it becoes logged in, ita can process calls from either queue based on distribution. |
| 04:35.14 | [TK]D-Fender | Yourname``: they do NOT log to a queue. They are either a memeber of one or more, or AREN'T. |
| 04:35.27 | [TK]D-Fender | Yourname``: And they are either logged in, or not. |
| 04:35.27 | Yourname`` | Gotcha. |
| 04:35.46 | Yourname`` | I meant to say when you use AddQueueMember, isn't that kind of like "logging into a queue"? |
| 04:36.29 | [TK]D-Fender | Yourname``: No. Loggin in involves some sense of identity or membership. That there is perhaps even a PASSWORD or some other controls placed on it. |
| 04:37.18 | Yourname`` | Ah, so this is not really logging in since you made them a member explicitly. |
| 04:37.26 | [TK]D-Fender | Yourname``: that add just adds a member on the fly. You COULD technically add an AGENT as a memeber and one who was previously only a member of one queue will now be available for two |
| 04:37.28 | Yourname`` | Thanks a lot for that. :) |
| 04:37.48 | jameswf-home | uytrew11234567890-==qwertyuioppp[[[asdfghjkl;' |
| 04:37.56 | jameswf-home | oops crap |
| 04:38.51 | Yourname`` | [TK]D-Fender: So then, if I explicitly define member=>Agent/100 under [salesq] and [helpq] how can I get Agent 100 to show up in both queues easily and then kind of "not show up" when he's done for the day? |
| 04:39.30 | [TK]D-Fender | Yourname``: he won't get calls when he's not logged in. |
| 04:39.45 | [TK]D-Fender | Yourname``: Or paused. |
| 04:40.11 | unpaidbill | i'd like to kick whoever thought it was a good idea to make -P the port option for scp and -p the port option for ssh. thank you for listening to me whine. |
| 04:40.50 | Yourname`` | [TK]D-Fender: Ok so how can I not log him in, and how can I log him in? |
| 04:41.08 | [TK]D-Fender | Yourname``: Go read the application listing. |
| 04:41.25 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 04:41.41 | Yourname`` | [TK]D-Fender: AQM/RQM and AgentLogin and AgentCallBackLogin are the ones we can use. Am I on the dot with those ones? |
| 04:42.21 | [T]ank | I also have about 100 Linksys spa942 phones that I need to sell. A paid $130 a piece for them, but would probably let them go for $50 a piece |
| 04:42.42 | [TK]D-Fender | Yourname``: Read their instructions and go DO SOMETHING. |
| 04:42.57 | unpaidbill | you selling those in bulk or one at a time? i may be interested in one for testing tank |
| 04:42.57 | [TK]D-Fender | [T]ank: Why so much to get rid of? |
| 04:43.29 | [T]ank | [TK]D-Fender: growth.. going to an all ip softphone set up |
| 04:43.32 | Yourname`` | [TK]D-Fender: LOL that's the point I was asking what's the best way you think for logging them in and out? |
| 04:43.51 | [TK]D-Fender | Yourname``: I've done entirely too much thinking for you already. |
| 04:43.55 | [T]ank | unpaidbill: I would be willing to sell you just one. You would have to cover the shipping expenses |
| 04:44.01 | [T]ank | pm me |
| 04:44.58 | *** join/#asterisk sucituanbo (n=blah@c-24-21-121-148.hsd1.wa.comcast.net) |
| 04:45.11 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d00dcfc495bf0b42) |
| 04:46.03 | [T]ank | one down, 99 to go ;-) |
| 04:46.07 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 04:46.28 | jblack | I'm tempted to get 3 of 'em from ya. |
| 04:46.45 | Yourname`` | [T]ank: If you did this maybe 10 days ago, I'd get 50% of them. If they were IP 330s, I'd get all of 'em. |
| 04:46.56 | [TK]D-Fender | jblack: You mean you've recovered from your ex's phone bills and investment in that gateway? ;) |
| 04:47.15 | unpaidbill | haha |
| 04:47.21 | jblack | Mostly, yeah. |
| 04:47.38 | jblack | The only reason I'm not jumping is I'd have to haul her fat ass back in to run more lines. |
| 04:49.13 | jblack | Somebody would be smart to make cheap desk phones w/ 802.11g. |
| 04:49.39 | unpaidbill | ugh |
| 04:49.41 | unpaidbill | no please |
| 04:50.29 | unpaidbill | too easy to take down all your phones :/ |
| 04:51.04 | Yourname`` | Nothing wrong with fatasses. |
| 04:51.06 | jblack | fair enough. But then I could have a phone in my attic. I could even name it the 'batphone' |
| 04:51.13 | unpaidbill | ok well |
| 04:51.18 | unpaidbill | that sounds pretty sweet |
| 04:51.44 | unpaidbill | fiberglass + hot attic though.. you'd be better off digging a hole in your basement |
| 04:51.54 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 04:51.57 | jblack | Oh, that one, I'd name the ratphone |
| 04:52.07 | scooby2 | ~centos52bug |
| 04:52.07 | 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 |
| 04:52.33 | jblack | Ohhh. And following the theme, I could put one in my kid's room, and name it the bratphone... |
| 04:53.05 | jblack | and the one in my bedroom could be the napphone... |
| 04:53.22 | jblack | I'll give you two guesses on what I'd call the one in the bathroom, and "#1" doesn't count. |
| 04:53.46 | ManxPower | and your wife's phone? |
| 04:53.56 | jblack | ex-wife, if you don't mind. |
| 04:54.31 | jblack | evilcheatingbitchfromhellindebttohereyeball phone? |
| 04:54.33 | unpaidbill | that would be the ricinphone |
| 04:54.36 | jameswf-home | exwife the bat phone |
| 04:55.39 | jblack | when I ran ethernet a while back, I did it for two reasons. One was that I didn't want to snake walls alone. The other, was it did a great job of putting her in her place. |
| 04:55.53 | jblack | Got her help, that is. |
| 04:58.19 | jblack | Don't know why I bothered. she's the biggest caller to the house, so you can imagine how many calls I get. |
| 05:01.43 | unpaidbill | i bet it's a lot |
| 05:01.52 | unpaidbill | how right am i |
| 05:03.49 | Yourname`` | If I have Goto(context,s,1), what are the good chances that it will go to the next priority? |
| 05:03.56 | Yourname`` | Since there's no timeout.. |
| 05:04.24 | jblack | If you have x,1,Goto(context,s,1) x,2,Dosomething().... |
| 05:04.34 | jblack | the chances that DoSomething will happen are basically 0. |
| 05:04.38 | [TK]D-Fender | Yourname``: if it doesn't exist? Pretty high. |
| 05:04.56 | jblack | Oh, yeah, in that case. [TK]D-Fender caught me. |
| 05:04.56 | Yourname`` | [TK]D-Fender: it = goto timeout? |
| 05:05.15 | [TK]D-Fender | Yourname``: there is no such things as TIMEOUT with a Goto. |
| 05:05.27 | Yourname`` | Exactly.. but who knows, you're the guru here. |
| 05:05.45 | [TK]D-Fender | Yourname``: You are spouting random ravings here. |
| 05:05.51 | [TK]D-Fender | Yourname``: Go somehting to show us? |
| 05:06.00 | Yourname`` | So I can't have a Goto and a retrydial and expect the retrydial to work out. Dammit. |
| 05:06.23 | jblack | Try Gosub if you want to come back |
| 05:06.29 | [TK]D-Fender | Yourname``: What, now you you're going to introduce 3 other things we didn't know that you were doing? |
| 05:06.45 | [TK]D-Fender | Yourname``: unload chan_scatterbrain.so |
| 05:06.49 | Yourname`` | lololol |
| 05:07.39 | unpaidbill | lol is the bane of my existence. |
| 05:07.43 | unpaidbill | i felt dirty typing it even then |
| 05:07.51 | *** join/#asterisk ZX81 (n=matt@120.89.80.110) |
| 05:08.09 | Yourname`` | You must feel real filty if you had to type 'lololol' |
| 05:08.22 | unpaidbill | there was a time, long ago, where lol had no place on the internet. |
| 05:08.30 | unpaidbill | then cablemodems came. and dsl. |
| 05:08.33 | ZX81 | does anyone know a way to have agents which are in multiple queues, each with round robin and still have the ability to make outgoing calls? |
| 05:08.33 | Yourname`` | I remember those times. |
| 05:08.39 | jblack | if lol makes you feel dirty, what happens when you rofl? |
| 05:08.43 | Yourname`` | I actually like these times better than those times. |
| 05:08.59 | unpaidbill | jblack i cut myself. i cut myself just to know a better time existed. |
| 05:09.00 | Yourname`` | Yeah, when you actually rofl, you're uber dirty. |
| 05:09.21 | unpaidbill | i duno, the fact that i'm disgusted by having a T1 at work these days is kind of sad |
| 05:09.26 | jameswf-home | if anyone was curious http://www.mcs.csuhayward.edu/~tebo/GoogleStreetViewVan/ |
| 05:09.30 | unpaidbill | i used to be so excited to go to work because i could get a decent speed |
| 05:09.32 | jblack | ZX81: I suppose either have your agents log out to make calls, or have them suffer race conditions with your customers. |
| 05:09.37 | unpaidbill | now i'm pissed because my speed is terrible at work. |
| 05:09.40 | jblack | ZX81: Give 'em 2 phones. :) |
| 05:09.55 | Yourname`` | jameswf-home: Nice link, thank you.. |
| 05:10.06 | Yourname`` | Is it robot driven? :P |
| 05:10.14 | ZX81 | :) yeah I was thinking of registering multiple accounts per agent (with multiple identities on the snom 360s) |
| 05:10.17 | jblack | Someone rich please buy the crap on woot that I don't want, so that I can buy the stuff I do? |
| 05:10.19 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 05:10.20 | ZX81 | but don't really want to |
| 05:10.21 | ZX81 | :) |
| 05:10.29 | unpaidbill | although we got our t1 upgraded today, now it uses this crazy adtran that does VOIP on the backend and handles all my data/voice QoS stuff, so i get the full t1 speed if nobody is using a phone in the office |
| 05:10.40 | jblack | Oh, I know. Do those phones have a DND button? |
| 05:10.40 | unpaidbill | kind of nice.. got back 8*64Kb/s |
| 05:11.31 | jblack | unpaidbill: Bah. put everyone on ilbc and tell 'em the phone co sucks. ;) |
| 05:11.37 | unpaidbill | hah |
| 05:11.46 | unpaidbill | we could go g711 |
| 05:11.51 | unpaidbill | but im afraid faxing will suffer greatly |
| 05:11.58 | unpaidbill | and im waiting for 1.6 and T.38 |
| 05:12.03 | ManxPower | When I started using G726 on some of my links, people thought we were doing "HD Voice". |
| 05:12.37 | unpaidbill | err wait i didnt mean g.711 did i.. i meant g.whateverthepayforfastoneis |
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| 05:12.40 | bkw_ | G726-32 is pretty good on CPU and bandwidth |
| 05:12.59 | ManxPower | bkw_: that's why I chose it. |
| 05:13.06 | bkw_ | G726-24 is also good.. G726-16 sounds a bit tinnie |
| 05:13.10 | bkw_ | or metalic |
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| 05:14.14 | ManxPower | I'd pay for quite a bit of money for a HD Cell phone |
| 05:14.22 | bkw_ | but honestly you can't mistake G722 and G726 |
| 05:14.29 | ManxPower | At least I would have full quality all the way |
| 05:14.39 | bkw_ | ManxPower: it might happen some day |
| 05:14.50 | bkw_ | AMR-WB is pertty awesome |
| 05:14.54 | ManxPower | well, the max quality of the equipment, no loss on the "last mile" |
| 05:15.01 | unpaidbill | yeah some day.. maybe they'll stop charging 20 cents per text message in the USA some day too |
| 05:15.08 | unpaidbill | just kidding, it wont ever happen :P |
| 05:15.32 | bkw_ | ManxPower: thats what amr-wb should solve if we ever get cellphone companies to pull their heads out of their asses |
| 05:15.51 | Yourname`` | Does anyone know if autofill is disabled when a queue sees one of its members is a member of another queue and taking calls too? |
| 05:17.03 | [TK]D-Fender | Yourname``: unrelated. |
| 05:17.20 | Yourname`` | ok |
| 05:20.33 | Yourname`` | Weird because earlier today I had Agent 1, 2, 3 and 4 online on salesq and helpq. Agent 1 is on call, but 2 3 4 are available in BOTH queues.. yet the calls that came in were piling up. Like 4 calls in queue.. and only 1 aent was in ringing state although multple agents were avilable |
| 05:22.30 | unpaidbill | man this is sweet |
| 05:22.54 | unpaidbill | the clec i use gave me FULL access to CID info, so i can put any name/number i desire in and they'll pass it along |
| 05:22.54 | [TK]D-Fender | Yourname``: "Its 3am in the Wihitehouse and Yourname`` still has no backup to lend credence or coherene to anything he says" |
| 05:22.57 | unpaidbill | fap fap fap. |
| 05:24.03 | jblack | [TK]D-Fender: You're pissed at the natural gas company I use. |
| 05:24.06 | jblack | Just thought you should know. |
| 05:24.16 | ManxPower | unpaidbill: we all hate you now. |
| 05:24.21 | [TK]D-Fender | jblack: ... Taco Bell? |
| 05:24.31 | jblack | No. UGI. |
| 05:24.33 | unpaidbill | haha manx |
| 05:25.12 | jblack | "America has an 80 year supply of natural gas. canada has 40 years of natural gas. Therefore, there is a 120 year supply of natural gas for the US |
| 05:29.39 | *** part/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 05:33.16 | unpaidbill | i've got a surplus of natural gas i wish i could offload |
| 05:33.23 | unpaidbill | i dont think they deal in methane though |
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| 05:37.13 | unpaidbill | oh shit old school |
| 05:37.19 | unpaidbill | nick collisions boo yah |
| 05:37.28 | bbryant | lol |
| 05:39.14 | [TK]D-Fender | Nostalgia isn't what it used to be.... |
| 05:39.41 | unpaidbill | i learned everything i know from BBSes and the newsgroups .... probably why i'm so demented. |
| 05:39.46 | *** part/#asterisk bbryant (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 05:40.21 | unpaidbill | well this is a bit of a pickle im in here |
| 05:40.33 | unpaidbill | i need to change the subnet mask of the WAN interface on the PIX |
| 05:40.52 | unpaidbill | but if i do a no ip address outside... then i'll be kicked off the network |
| 05:40.55 | unpaidbill | hmm. |
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| 05:47.18 | [TK]D-Fender | ok, checkout time. LAter all |
| 05:48.30 | d-k-t-2 | unpaidbill, make a new config and load it in |
| 05:48.44 | obnauticus | Err, is this a very good deal? http://img242.imageshack.us/my.php?image=0728082050rh4.jpg |
| 05:48.56 | obnauticus | The headset is normally 99.99 :\ I'm just wondering ifa nyone here has it |
| 05:49.55 | unpaidbill | d-k-t-2 im cisco retarded. i'll just wait until i am in the office tomorrow |
| 05:50.09 | unpaidbill | i'm a scratch the surface and move on to something else kind of guy |
| 05:50.23 | unpaidbill | you know, the guy who fucks everything up |
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| 05:50.46 | AJayMN | OK.. If i call into my system the sound stutters..but if i access its website or anything on the system at the same the, the stutter goes away? whats with that? |
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| 05:51.34 | d-k-t-2 | unpaidbill, well, at least you knew it would screw up before doing it... I've encountered quite a few 'experts' that didn't think beyond their next step and ran straight into the sort of problem you knew would happen... |
| 05:53.36 | *** join/#asterisk hyphenex (n=scoobr@203.63.45.23) |
| 05:54.27 | hyphenex | G'Day, if I've got a bunch of VoIP phones, and there are computers sitting next to each of those terminals, is it possible to have a program tell the phone to dial a phone number through software? (or transfer an existing call through software?) |
| 05:55.08 | jblack | hyphenex: Sure. Use callfiles. |
| 05:55.31 | jblack | The website uses javascript to hit a page that generates a callscript, and bada-bing |
| 05:55.52 | jameswf-home | I prefer originate via asterisk -rx in an endles while loop |
| 05:56.02 | hyphenex | callfiles hey? |
| 05:57.11 | jameswf-home | I trued to do this fancy forking thing to make asterisk implode and it failed but I typod a context and poof asterisk went in to a loop of death |
| 05:57.17 | hyphenex | and SIP can do all that? (so the phone doesn't even need to start the phone call? You can just tell the phone you are now calling x?) |
| 05:57.18 | jameswf-home | *tried |
| 05:58.18 | jblack | jameswf-home: I'm replacing that rhino you recommended. |
| 05:58.41 | jameswf-home | umm okay |
| 05:59.22 | jameswf-home | has the memory of a goldfish |
| 05:59.35 | hyphenex | jblack, Will this work with any VoIP phone? |
| 05:59.50 | jblack | It was a few months ago, so I can understand you forgetting. |
| 05:59.55 | jblack | hyphenex: Yes. |
| 06:01.37 | jameswf-home | careful of web based file generation ( as trixbox learned) |
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| 06:25.27 | oilinki | hi. |
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| 06:25.56 | oilinki | if the network limitation is 64kbps. what codecs could be used with voip? |
| 06:26.28 | unpaidbill | any of them |
| 06:26.42 | JT | unpaidbill: except G.711 and G.722 |
| 06:26.46 | unpaidbill | you could have one call placed with ulaw/alaw, but nothing else |
| 06:27.02 | unpaidbill | oh wait that's 80kb/s isnt it |
| 06:27.02 | oilinki | there will be overhead as well? |
| 06:27.09 | unpaidbill | nevermind jt is right |
| 06:27.14 | unpaidbill | im an incorrect person. |
| 06:27.33 | JT | G.711 iwth 20ms packetisation on SIP will run around 85kbit/s |
| 06:28.13 | unpaidbill | best bet.. g726 |
| 06:28.18 | oilinki | true is offering free wifi network in bangkok, but the limitation is that 64kbps. |
| 06:28.29 | unpaidbill | that will get you like.. 2 or 3 calls |
| 06:28.47 | unpaidbill | simultaneous |
| 06:28.53 | oilinki | basically I would need only 1 call from the terminal |
| 06:28.54 | hyphenex | so just to be sure, I can tell asterisk to tell a phone "Hey, you're not dialing this number on speakerphone?" |
| 06:28.56 | JT | G.729 and GSM are the most compressed |
| 06:29.04 | unpaidbill | err g729 i meant, jesus i need to quit |
| 06:29.07 | JT | speex and iLBC are fairly compressed too |
| 06:29.45 | oilinki | g729 could be the best as it seems to be quite well supported by voip-proxies as well. |
| 06:30.02 | unpaidbill | yup, everyone supports g729 unless they're terrible human beings |
| 06:31.01 | oilinki | btw. how should I setup the remote voip-server and local client registration for 729 if I wish to use the dial-tones as well? |
| 06:32.03 | JT | oilinki: is this free 64kbit/s wireless 64kbit/s each way, of 64kbit/s total of both directions? |
| 06:32.28 | oilinki | JT: they don't say it. I suppose it 64 for each way |
| 06:32.48 | oilinki | http://www.truewifi.net/en/promotion/green_bangkok.htm |
| 06:33.20 | JT | you should keep in mind that wifi is awful for voip |
| 06:33.29 | JT | so results may not be the best |
| 06:33.36 | unpaidbill | oilinki your biggest problem is packet loss and latency heh |
| 06:34.25 | oilinki | wifi itself is quite ok for voip (as in home usage for example), but the public networks can be a pain. |
| 06:34.28 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 06:34.59 | unpaidbill | i'd be surprised if they passed UDP |
| 06:35.12 | unpaidbill | no reason to |
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| 06:38.35 | JT | oilinki: again debatable, wifi isn't that great for voip, even in home networks |
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| 06:41.33 | hyphenex | how would I go about telling Asterisk that I want to transfer phone calls? (so a piece of software can show you your calls you have open, and click and drag to transfer the calls) |
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| 06:52.09 | oilinki | JT: naturally there is latency, but not too much. |
| 06:52.27 | oilinki | I have been using voip/wifi in home for years. |
| 06:52.39 | creativx | hyphenex: the asterisk manager interface is your friend. AMI |
| 06:52.58 | JT | oilinki: marvelous, fixed or mobile? |
| 06:53.01 | oilinki | and the latency comes from the internet actually (well the servers are 100-350ms away) |
| 06:53.26 | oilinki | JT: mobile. |
| 06:53.38 | unpaidbill | do the asterisk mailing lists queue mail that isnt immediately deliverable or throw it out? |
| 06:53.47 | unpaidbill | i dont seem to have gotten any mail from today when my server was down |
| 06:53.50 | JT | mobile wifi leads to jitter and random packet loss |
| 06:53.51 | unpaidbill | :( |
| 06:54.00 | JT | you won't notice that much when doing web browsing and email |
| 06:54.09 | JT | but it will make quite a difference on voip |
| 06:54.15 | JT | also, wifi is half duplex |
| 06:54.29 | oilinki | JT: not too much really. |
| 06:54.31 | unpaidbill | 200ms is crap for voip |
| 06:54.32 | JT | voip send and receives data simultaneously |
| 06:54.40 | JT | which also degrades data |
| 06:54.43 | unpaidbill | 150 max i says |
| 06:55.00 | JT | oilinki: it depends on the codec too, some are more forgiving than others |
| 06:55.02 | oilinki | well naturally there is a way to talk to a voip calls.. a bit slower reponses |
| 06:55.15 | oilinki | but then again. from Asia to Europe, all the calls will be the same |
| 06:55.19 | unpaidbill | i find that to be extremely annoying |
| 06:55.20 | JT | g.726 is completely unforgiving codec, don't use it on unreliable networks |
| 06:55.49 | oilinki | I'm using g729 for calls from thailand to singapore and finland. works quite well. |
| 06:55.59 | unpaidbill | if i notice the latency when im talking it isnt acceptable.. and 150+ is definitely noticable |
| 06:56.08 | JT | look, you either want advice or not |
| 06:56.19 | JT | if you want to refute everything i say, sure, go right ahead and do it |
| 06:56.20 | oilinki | eg. Nokia voip-phone - home asterisk - voip server in finland - breakout to CS. |
| 06:56.26 | JT | but if you are the expert, why are you asking us? |
| 06:56.36 | JT | G.726 is a delta codec |
| 06:56.45 | JT | if you miss a frame it will sound awful |
| 06:56.47 | oilinki | I did ask for the codec. |
| 06:57.05 | oilinki | but then again. all the advice is wellcome. |
| 06:57.51 | oilinki | however it's virtually impossible to reduce the internet latency.. |
| 06:58.32 | creativx | the solution is to keep people from talking. |
| 06:58.47 | jblack | Oh _man_. I am so tempted to buy http://woot.com |
| 06:59.16 | jblack | Imagine hooking elvis up to chanspy |
| 07:01.05 | dominic1 | hello guys |
| 07:01.19 | dominic1 | can anybody tell me for what I need membername in the queues table? |
| 07:01.42 | miloux | In the perl module, Asterisk:AGI get_database('family', 'key').... what does it mean by "family"? |
| 07:02.31 | dominic1 | /dundi/secret |
| 07:02.35 | dominic1 | so dundi is the family |
| 07:03.00 | dominic1 | it's like files in folders |
| 07:03.13 | miloux | ok |
| 07:03.21 | dominic1 | /dundi/secret2 can be a second key in the same family |
| 07:04.56 | miloux | so if i wanted to get a specific queue_table post i do this?: get_database('/queue_table/name', 'support'); ? |
| 07:05.37 | unpaidbill | woot is BS |
| 07:05.46 | unpaidbill | they dont ship to hawaii even though you can place an order with a hawaii address |
| 07:05.50 | unpaidbill | assholes. |
| 07:06.16 | dominic1 | you mean with realtime? |
| 07:07.09 | miloux | y |
| 07:07.12 | miloux | yea |
| 07:07.47 | dominic1 | then get_database is wrong. You can only get values from the asterisk internal database |
| 07:07.50 | dominic1 | called astdb |
| 07:08.05 | dominic1 | if you are using odbc use this to get you values |
| 07:08.19 | miloux | so for the realtime database i have to you another implementation to talk to the db? |
| 07:09.01 | dominic1 | that's correct. If you want to get information in the dialplan from the database look at func_odbc.conf |
| 07:10.17 | miloux | ah, thanks a bunch mate |
| 07:10.33 | unpaidbill | when using AMI i should be able to set callerid as such: callerid: "blah blah" <1111111111> ? or is there some other syntax i need? i'm not sure if i'm doing this wrong or if i'm having the name stripped, currently i only see the number i specify but not the name |
| 07:11.04 | unpaidbill | asshole clec is stripping the name i think |
| 07:11.11 | dominic1 | manager show commands |
| 07:11.54 | dominic1 | I think there is no function to set the callerid |
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| 07:12.03 | creativx | originate |
| 07:12.08 | creativx | can set callerid |
| 07:12.24 | unpaidbill | yeah i can set callerid |
| 07:12.32 | unpaidbill | for the number, the name isnt coming through |
| 07:12.33 | dominic1 | okay, then I was wrong, didn't do that before with ami |
| 07:12.38 | unpaidbill | must be my carrier stripping it |
| 07:12.52 | unpaidbill | this makes me sad. |
| 07:13.08 | creativx | my itsp replaces our callerid if i set it outside our did range |
| 07:13.09 | dominic1 | callerid: "blah blah <1111111111>" isn't that better? |
| 07:13.21 | unpaidbill | you know, i didnt try that |
| 07:13.24 | unpaidbill | let me see |
| 07:13.52 | dominic1 | can I store my agents in the database? |
| 07:15.00 | tzafrir_laptop | http://cuil.com/ is nice. But they "borrow" some definitions from wikipedia, and for some strange reason forget to mention that on their site |
| 07:16.56 | unpaidbill | no that doesnt work at all dominic |
| 07:17.12 | unpaidbill | damnit |
| 07:18.20 | unpaidbill | even no quotes is a crapshoot |
| 07:18.40 | unpaidbill | maybe it's a cell phone thing.. i hope :/ arg |
| 07:19.18 | Strom_C | unpaidbill: have you tried just setting the caller ID manually from within extensions.conf and testing to make sure it works that way? |
| 07:19.33 | unpaidbill | strom that's what i'm trying now |
| 07:19.39 | unpaidbill | i'll let you know in a min or two |
| 07:19.53 | unpaidbill | exten => _91NXXNXXXXXX,2,Set(CALLERID(all)=BILLY <4157346404>) |
| 07:19.57 | unpaidbill | that should be suficient |
| 07:20.00 | unpaidbill | oh boy i gave you my cell # |
| 07:20.02 | unpaidbill | haha |
| 07:20.06 | creativx | hehe |
| 07:20.07 | unpaidbill | :( |
| 07:20.08 | creativx | 10 points. |
| 07:20.16 | creativx | let the terror reign |
| 07:20.47 | Strom_C | unpaidbill: set name and number separately |
| 07:20.57 | unpaidbill | oh really |
| 07:20.59 | unpaidbill | ok |
| 07:21.38 | unpaidbill | exten => _91NXXNXXXXXX,2,Set(CALLERID(name)=BILLY) |
| 07:21.38 | unpaidbill | exten => _91NXXNXXXXXX,n,Set(CALLERID(number)=4157346404) |
| 07:21.40 | unpaidbill | here we gooo... |
| 07:21.58 | unpaidbill | i will high five you SO HARD if this works |
| 07:22.01 | unpaidbill | prepare yourself |
| 07:22.02 | Strom_C | i don't even know why you're bothering with name; mobile phones don't have CNAM service, and most wireline CNAM service is done with a database lookup on the terminating end |
| 07:22.13 | Strom_C | also, it's CALLERID(num), not CALLERID(number) |
| 07:22.24 | unpaidbill | god damnit strom! that's what i was just wondering. i was calling my cell |
| 07:22.29 | unpaidbill | that's probably why the name was stripped |
| 07:22.33 | unpaidbill | retarded. |
| 07:22.40 | Strom_C | ,,, |
| 07:22.43 | Strom_C | come on |
| 07:22.48 | Strom_C | you've had a mobile phone for how long now? |
| 07:22.54 | creativx | heheh |
| 07:22.56 | unpaidbill | i use it like.. once a month if that |
| 07:23.01 | Strom_C | have you EVER seen a name show up that you hadn't already entered into your phone book? |
| 07:23.06 | *** join/#asterisk oej (n=olle@cust-IP-10.data.tre.se) |
| 07:23.10 | unpaidbill | now that you mention it, no |
| 07:23.15 | Strom_C | ~brain |
| 07:23.16 | jbot | methinks brain is a very useful device. It is advised that you ensure your brain is switched to the "on" position before continuing. If you need assistance, please consult your directory. This is a recording. |
| 07:23.28 | unpaidbill | what, is that too much fucking data for them to attach to a friggen call |
| 07:23.38 | unpaidbill | do i have to pay 25 cents a call to get that info |
| 07:23.43 | unpaidbill | goddamn retarded cell networks |
| 07:23.45 | unpaidbill | worthless. |
| 07:23.47 | Strom_C | unpaidbill: it's not part of the spec |
| 07:23.52 | unpaidbill | oh |
| 07:23.56 | unpaidbill | well thank you for educating me |
| 07:24.21 | unpaidbill | i appreciate you people from cali, ca, whatever you like to call it |
| 07:26.10 | unpaidbill | strom, i dig you. |
| 07:26.21 | unpaidbill | in a hetero way of course. |
| 07:26.48 | Strom_C | why? are you afraid I'm going to get turned on? |
| 07:27.03 | unpaidbill | absolutely not |
| 07:27.12 | unpaidbill | a little dry humping never killed anyone. |
| 07:29.36 | Strom_C | unpaidbill: maybe you should get a better idea of how all this stuff works... |
| 07:29.37 | Strom_C | ~101 |
| 07:29.38 | jbot | extra, extra, read all about it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 07:29.56 | unpaidbill | i probably should |
| 07:30.03 | unpaidbill | but now i know the little cell network bullshit |
| 07:30.17 | unpaidbill | and that is good enough for me. thanks for the tip! |
| 07:30.28 | Strom_C | "cell network bullshit" |
| 07:30.30 | Strom_C | are you 14? |
| 07:30.33 | unpaidbill | yes indeed |
| 07:30.39 | unpaidbill | 15 in a few days |
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| 07:30.51 | Strom_C | idiot |
| 07:30.58 | unpaidbill | send me a polycom for my birthday |
| 07:31.04 | unpaidbill | i want one of those ninja star ones |
| 07:34.03 | unpaidbill | so are you going to send me a present or what |
| 07:34.07 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 07:34.13 | unpaidbill | are you checking amazon.com for the perfect gift? |
| 07:42.34 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-6-49.revip2.asianet.co.th) |
| 07:43.10 | *** join/#asterisk MaliutaLap (n=nikolai@124-171-128-169.dyn.iinet.net.au) |
| 07:47.11 | *** join/#asterisk oej_ (n=olle@80.251.192.3) |
| 07:47.31 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:53.49 | *** join/#asterisk Clumpco (n=chatzill@88.191.78.134) |
| 07:54.21 | Clumpco | Good morning all |
| 07:54.46 | Clumpco | <<--- total Asterisk n00b |
| 07:55.54 | jblack | you must have been a treat in elementary school. |
| 07:56.13 | jblack | kindergarten: "I don't know how to draw in the lines". First grade "I don't know how to add".... |
| 07:56.18 | foexle | hi guys, i'm a study from germany and i have a questtion. Which card would u advice for a BRI and a PRI system ? We use in Germany to run ISDN, dss1-protocol |
| 07:56.47 | JT | you'll need different cards |
| 07:56.55 | JT | bri cards are a bit annoying in asterisk |
| 07:56.56 | foexle | dont say digium pls other prducers |
| 07:57.01 | jblack | foexle: I'd avoid rhino. |
| 07:57.10 | JT | my main recommendation for bri would be a BRI to SIP external gateway |
| 07:57.18 | JT | otherwise Sangoma or Junhanns card |
| 07:57.32 | Strom_M | digium |
| 07:57.34 | Strom_M | OH NO, I SAID IT |
| 07:57.40 | Strom_M | runs around in circles |
| 07:57.59 | foexle | its only for ISDN, i dont need a connection to VoIP |
| 07:58.23 | JT | i know that, you can use a BRI to SIP gateway to connect to asterisk |
| 07:58.29 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:58.35 | JT | how many BRIs? |
| 07:58.43 | foexle | i think SIP is a VoIP protocol |
| 07:58.47 | foexle | 4 |
| 07:58.54 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 07:59.05 | jblack | foexle: He's not misleading you. |
| 07:59.12 | JT | yes it's a voip protocol |
| 07:59.15 | JT | what's the problem? |
| 07:59.30 | JT | BRI <> Gateway <> SIP <> Asterisk |
| 07:59.42 | foexle | ah hmmm .... |
| 08:00.01 | foexle | ok that is a new idea |
| 08:00.27 | foexle | i would BRI <> Asterisk |
| 08:00.28 | JT | same way ATAs like the Linksys/Sipura units are basically analogue to SIP gateways |
| 08:00.48 | JT | well using a BRI gateway is the most reliable solution |
| 08:00.55 | JT | cards may be a bit cheaper |
| 08:01.05 | *** join/#asterisk gilli (n=gilli@p57A1B6C7.dip.t-dialin.net) |
| 08:01.10 | JT | but you'll find setting up BRI cards in asterisk can be a hassle |
| 08:01.32 | foexle | hmmm |
| 08:01.41 | foexle | ok thank u |
| 08:03.37 | foexle | PRI gateways are better as pri cards, too ? |
| 08:04.49 | JT | yeah but significantly more expensive |
| 08:05.07 | JT | pri cards also tend to work better than bri cards in asterisk |
| 08:05.18 | foexle | ok |
| 08:05.39 | JT | but if you find a cheap second hand AS5300 of AS5400 with all the correct modules, that can be a good way to go PRI gateway |
| 08:05.40 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:06.28 | foexle | i'll research that :> thank u again... and sry for my bad engl ;) |
| 08:06.45 | JT | haha no probs i've seen much worse english before :P |
| 08:07.04 | foexle | ;) |
| 08:11.28 | *** join/#asterisk hyphenex (n=scoobr@203.63.45.23) |
| 08:12.05 | hyphenex | Hey, my current phone system consists of 7 white boxes, all leading into a NEC PBX system |
| 08:12.18 | hyphenex | those white Telecom boxes would mean I'm using ISDN right? |
| 08:13.23 | MaliutaLap | the white boxes could be anything |
| 08:13.31 | MaliutaLap | they could be your voicemail |
| 08:13.47 | hyphenex | hehe, nahh, I don't think so :P |
| 08:13.54 | hyphenex | they have telephone numbers written on them :P |
| 08:14.33 | MaliutaLap | hyphenex: and tha rules out voicemail why? |
| 08:14.50 | MaliutaLap | we can't see them, so we can't do anything more than speculate |
| 08:14.51 | Strom_M | hyphenex: "white boxes" means nothing |
| 08:14.57 | Strom_M | you have to be more specific |
| 08:14.57 | MaliutaLap | and it's not * related anyhow |
| 08:15.14 | Strom_M | perhaps provide a photo |
| 08:15.22 | Strom_M | describe the physical interface |
| 08:15.23 | Strom_M | etc |
| 08:15.43 | creativx | disconnect all the white boxes and see if your telephony still works |
| 08:15.50 | hyphenex | Ahh, well, they lead into a Xen Alpha PBX system |
| 08:16.19 | JT | hyphenex: do they say Telstra on them? |
| 08:16.26 | hyphenex | JT, yes they do |
| 08:16.58 | JT | i bet they say NT1 Network Termination on the bottom |
| 08:17.05 | JT | means you have BRI ISDN |
| 08:17.18 | hyphenex | JT, Thanks, I'll go check |
| 08:17.41 | Clumpco | jblack>you must have been a treat in elementary school.<<< Sorry was afk. Yes I was a treat, they called me "greensleeves" |
| 08:18.25 | Clumpco | I was just warning folks that I know v little of Asterisk, before asking what will no doubt be a stupid question |
| 08:18.30 | Clumpco | which is.... |
| 08:19.17 | Clumpco | Where do I find a detailed explanation of outgoing rules? All that I seem to be able to find is "use NXXXNXXX" etc. |
| 08:19.19 | JT | hyphenex: 7 also means they are on drugs, PRI is much cheaper than 7 BRIs |
| 08:19.39 | Strom_M | Clumpco: what do you mean, exactly? |
| 08:20.00 | unpaidbill | allow me, strom. |
| 08:20.01 | unpaidbill | ~astbook |
| 08:20.10 | Strom_M | you fail |
| 08:20.12 | Strom_M | ~book |
| 08:20.13 | 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 |
| 08:20.13 | unpaidbill | horribly |
| 08:20.25 | unpaidbill | i guess im not getting that bday present |
| 08:20.29 | Clumpco | Well I'm wanting to set up outgoing rules for a v small system with two PSTN |
| 08:21.05 | hyphenex | JT, It says S/T on the bottom |
| 08:21.24 | Clumpco | Just need al that starts with 06 to go to line1, everything else to the other |
| 08:21.38 | Strom_M | Clumpco: read the book in that link |
| 08:21.41 | unpaidbill | _06. |
| 08:21.46 | Clumpco | Will do Strom |
| 08:22.28 | Clumpco | unpaidbill>_06. <<<You are assuming that I know where to put that ;-) |
| 08:22.42 | unpaidbill | it's dirty and bad |
| 08:22.43 | *** join/#asterisk _sc0tty_ (i=Sc0ttY@217.144.147.41) |
| 08:22.46 | unpaidbill | the book will shed light |
| 08:23.44 | Clumpco | OK - that's cool |
| 08:24.31 | tzafrir_laptop | foexle, what do you mean by dss1? Standard ISDN/BRI line? |
| 08:25.00 | foexle | yeah |
| 08:25.04 | Clumpco | Tell me before I start trying to do soemthing that is impossible... Can I set it up so that no prefix is need for outside lines? |
| 08:25.11 | Strom_M | Clumpco: yes |
| 08:25.16 | Clumpco | Cool |
| 08:25.26 | unpaidbill | you can do...whatever you want |
| 08:25.30 | unpaidbill | as long as strom approves |
| 08:25.33 | Clumpco | Got to be Mrs Clumpton-friendly :-) |
| 08:25.36 | unpaidbill | if he doesnt.. god help you |
| 08:25.42 | foexle | but the idea with a gateway is perfect |
| 08:25.52 | tzafrir_laptop | I wouldn't say that BRI is such a hassle. If you're looking for a simple experimention, get an HFC-S based card |
| 08:26.16 | tzafrir_laptop | quite cheap and well-supported in Asterisk |
| 08:26.49 | foexle | HFC-S ? |
| 08:26.50 | *** join/#asterisk ToTo (n=ToTo@207.176.6.199) |
| 08:27.02 | foexle | sry im a verry noob :> |
| 08:27.06 | unpaidbill | high fructose corn syrup |
| 08:27.08 | foexle | +big -.- |
| 08:27.13 | unpaidbill | it's found in nearly every food product that is processed |
| 08:27.19 | Clumpco | Final question - will an Epia 800 MHz be fast enough for 2 lines/2phones on a TDM400 card? |
| 08:27.33 | unpaidbill | more than enough |
| 08:27.39 | Clumpco | w00t! |
| 08:27.56 | tzafrir_laptop | Clumpco, sure. But think what else you want to do with the box |
| 08:27.58 | hyphenex | Got to dash! |
| 08:28.00 | tzafrir_laptop | Voip? |
| 08:28.26 | tzafrir_laptop | transcoding and such? THough if you're only have up to 2 calls at once, that box will do |
| 08:28.35 | Clumpco | May add wifi voip later, yes |
| 08:29.00 | Clumpco | But for the moment POTS in/out |
| 08:29.00 | unpaidbill | even with transcoding my 500mhz geode can do 10 calls no problem with transcoding |
| 08:29.05 | unpaidbill | err... drunky |
| 08:29.17 | tzafrir_laptop | foexle, many of the cheaper ISDN (BRI) cards are based on the Cologne HFC chipset |
| 08:29.28 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 08:29.31 | kamanashisroy | hi, the inband and rfc2833 dtmf mode is not working in my asterisk 1.4 .. |
| 08:29.48 | Clumpco | <tzafrir_laptop>Clumpco, sure. But think what else you want to do with the box <<< Not much else... already got 4 other boxes doin stuff ;-) |
| 08:29.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:30.06 | kamanashisroy | do I need to install any special thing for that ? |
| 08:30.33 | foexle | ah ok ... i'll refer that |
| 08:30.36 | foexle | thank you |
| 08:30.37 | tzafrir_laptop | kamanashisroy, not working ==? How can you tell? |
| 08:30.43 | miloux | how many calls would you think a quad core 2200mhz AMD could handle? |
| 08:31.22 | unpaidbill | a lot |
| 08:31.42 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 08:31.53 | unpaidbill | probably over 100 |
| 08:32.05 | unpaidbill | http://www.voip-info.org/wiki-Asterisk+dimensioning check this site though |
| 08:32.13 | kamanashisroy | tzafrir_laptop: I sent call from asterisk server to asterisk server .. both are the same version .. and I set the same dtmf mode along 3 call tests(one for each) .. And I read dtmf using Read() application .. sent DTMF Using sendDtmf .. |
| 08:32.14 | miloux | ok thanks |
| 08:33.08 | Clumpco | OK, got the book, got the info - Thanks v much all of you |
| 08:33.11 | unpaidbill | you take big hits doing conferencing and whatnot |
| 08:33.16 | Clumpco | I may be back though ;-) |
| 08:33.53 | unpaidbill | the book is the best thing about asterisk sir. |
| 08:33.59 | Clumpco | OK |
| 08:35.15 | Clumpco | Oops, another question. Is there a problem using FF3 for the GUI? I can't select a user, it selects the whole list. (IE7 is ok, just dreadfull slow) |
| 08:35.25 | unpaidbill | asterisknow? |
| 08:35.29 | JT | tzafrir_laptop: you expect foexle to have 4 HFC cards in his server? |
| 08:35.31 | Clumpco | yep |
| 08:35.34 | JT | and a PRI card? |
| 08:35.38 | unpaidbill | i would say dont use that |
| 08:35.44 | Clumpco | ah |
| 08:35.48 | unpaidbill | asterisknow is good for simple things |
| 08:35.51 | MaliutaLap | Clumpco: have you read the topic? |
| 08:36.03 | unpaidbill | but if you're learning, the best way to go is pure unadulterated sexy commandline |
| 08:36.10 | unpaidbill | also, asteriskgui sucks (sorry!( |
| 08:36.13 | Clumpco | erm, should I be in another channel? APologies |
| 08:36.15 | unpaidbill | err, asterisknow |
| 08:36.42 | Clumpco | I had assumed that "now" was just a preconfigured install... |
| 08:36.50 | unpaidbill | no |
| 08:37.02 | unpaidbill | it's a very simple and limited gui interface |
| 08:37.03 | tzafrir_laptop | err... he has 4 connections, well, that might be a bit different. |
| 08:37.33 | Clumpco | OK - I'll fiddle with it a bit and then prolly go for the real thing then |
| 08:37.39 | Clumpco | Thanks for the heads up |
| 08:37.52 | unpaidbill | i highly recommend you read the book and stick to compiling it and using it from command line only |
| 08:37.57 | Clumpco | okis |
| 08:38.01 | unpaidbill | asterisknow is good for end users if you dont need anything special |
| 08:38.28 | unpaidbill | if you try to do anything fun with your dialplan, asterisknow is going to make you kill yourself. |
| 08:38.50 | Clumpco | Ah... exactly my case |
| 08:39.38 | unpaidbill | haha, yeah dude.. read book, mess around, search google. you will be where you need to be in no time |
| 08:39.53 | unpaidbill | then come here and ask questions and receive stroms fury |
| 08:39.58 | Clumpco | hehe |
| 08:40.05 | unpaidbill | but his fury comes with bits of information |
| 08:40.09 | unpaidbill | so pick through them :P |
| 08:40.33 | Clumpco | bows before the asterisk gods |
| 08:41.10 | unpaidbill | i picture strom as agdistis |
| 08:41.29 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 08:41.31 | Clumpco | trembles |
| 08:41.52 | unpaidbill | i really love almonds. |
| 08:42.04 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 08:42.14 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 08:42.33 | Clumpco | sends virtual almonds to unpaidbill |
| 08:42.53 | JT | sends cyanide |
| 08:42.57 | Clumpco | OK, got to go and do some "real" work |
| 08:42.58 | JT | smells like almonds |
| 08:43.00 | JT | must be almonds |
| 08:43.01 | unpaidbill | im totally eating them |
| 08:43.02 | JT | :D |
| 08:43.04 | Clumpco | Thanks again all of you |
| 08:43.05 | unpaidbill | jt you are so nice! |
| 08:43.23 | JT | apparently anyway, i've never smelt cyanide as far as i know |
| 08:43.24 | Clumpco | cyas |
| 08:43.35 | unpaidbill | i havent smelled it either, but i have heard the same, jt |
| 08:44.02 | *** part/#asterisk Clumpco (n=chatzill@88.191.78.134) |
| 08:44.23 | *** join/#asterisk oej (n=olle@cust-IP-10.data.tre.se) |
| 08:44.29 | unpaidbill | i kind of get a rush when i eat an almond.. is this the last bite i take? i wonder every time. |
| 08:45.05 | unpaidbill | i wish that sweet ass gui digium sold was open source :/ |
| 08:45.29 | tzafrir_laptop | unpaidbill, switchvox? |
| 08:45.32 | unpaidbill | yeah |
| 08:45.39 | unpaidbill | the devices they sell look really nice |
| 08:45.47 | unpaidbill | i've been building a crappy ajax interface |
| 08:45.48 | unpaidbill | slowly |
| 08:45.55 | tzafrir_laptop | unpaidbill, Druid look also nice. http://voiceroute.org/ . |
| 08:46.11 | tzafrir_laptop | And is free |
| 08:46.28 | *** part/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 08:46.53 | unpaidbill | no kidding.. i'll check that out, thanks! i have a monestary i am helping out that is currently using asterisknow.. of course i had to strip out most of the functionality because half the time you'll break the damn system if you try to modify a dialplan with it |
| 08:47.49 | unpaidbill | it's nice proof of concept though, i guess |
| 08:48.41 | unpaidbill | can you drop druid into an existing system tza |
| 08:48.54 | unpaidbill | that would be damn sweet |
| 08:48.54 | tzafrir_laptop | unpaidbill, no |
| 08:48.58 | unpaidbill | ahh |
| 08:49.26 | tzafrir_laptop | they seem to be working on it, though |
| 08:49.37 | unpaidbill | the screenshots look promising |
| 08:49.55 | unpaidbill | not as nice as the interface digium sells though |
| 08:50.13 | unpaidbill | switchvox is pretty damn nice looking. |
| 08:50.52 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 08:56.44 | *** join/#asterisk rethus (n=rethus@xdsl-84-44-224-114.netcologne.de) |
| 08:57.07 | rethus | anyone experiance with phpagi here? |
| 08:57.53 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 08:58.18 | rethus | have i to put all selfcoded phpagi-scripts into external.conf, or can i call them with other php-scripts? |
| 08:58.30 | rethus | i don't know how to start with phpagi |
| 09:01.24 | tzafrir_laptop | unpaidbill, my policy with this is simply not to use non-free software and not to tie myself to one. |
| 09:01.50 | kamanashisroy | does sendDTMF can send inband or rfc2933 way ? |
| 09:02.22 | unpaidbill | yea pretty much tza, except you use asterisk do you not |
| 09:02.37 | unpaidbill | or do you mean just for interfaces :P |
| 09:02.38 | *** join/#asterisk Dovid (n=Dovid@mail.matan-noar.org) |
| 09:03.06 | unpaidbill | i regret the install of asterisknow |
| 09:03.07 | unpaidbill | hugely |
| 09:03.45 | tzafrir_laptop | unpaidbill, asterisk is free software |
| 09:03.56 | unpaidbill | ooh, i misread. |
| 09:04.35 | unpaidbill | anyway, i tried to coach them on using asterisknow for simply adding extensions |
| 09:04.38 | unpaidbill | then what happens |
| 09:04.45 | unpaidbill | the next day some ass goes and fucks with the dialplan |
| 09:04.51 | unpaidbill | and breaks the system horribly |
| 09:05.04 | unpaidbill | so i removed that from the interface |
| 09:05.21 | unpaidbill | then they went and did several other dumb things |
| 09:05.26 | *** join/#asterisk bird_of_Luck (i=melifaro@secured.by.ipfw.ru) |
| 09:06.13 | unpaidbill | guis are bullshit, sure they may make some tasks easier, but they also give idiots the ability to point and click and break everything! sweet! |
| 09:06.44 | unpaidbill | still, switchvox looks sweet. |
| 09:09.36 | dominic1 | hi, does the ndc in e.164 include the local area code or the area code only? |
| 09:11.22 | dominic1 | is it possible to dialout with e.164 numbers over isdn? |
| 09:11.22 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 09:14.28 | *** join/#asterisk fuzzy (n=fuzzy@c-67-171-182-99.hsd1.or.comcast.net) |
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| 09:25.51 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:26.02 | rethus | how can i send a command via php-website to asterisk? |
| 09:26.32 | tzafrir_laptop | unpaidbill, and idiots can't ruin anything just as easily through text files? |
| 09:27.19 | jblack | no, because they don't know how to run vim |
| 09:28.14 | *** join/#asterisk blq (i=Bl@dslb-088-067-043-100.pools.arcor-ip.net) |
| 09:28.27 | *** join/#asterisk blq (i=Bl@dslb-088-067-043-100.pools.arcor-ip.net) |
| 09:28.54 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 09:30.29 | bird_of_Luck | hi all, I've got several cisco 7911G using sip and i need to disable DND button on them. Tried to set dndControl,settingsAccess to 0 in config, without any success. Can anybody point to right config option? |
| 09:33.07 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 09:33.51 | rethus | can anyone red this text from me, or is my irc-client broken? please give a short answer. |
| 09:34.14 | jblack | hi retus |
| 09:34.17 | miloux | rethus: yea |
| 09:34.52 | tzafrir_laptop | vim '+/^\[' +x:x /etc/asterisk/extensions.conf |
| 09:35.15 | tzafrir_laptop | after that reload and see what happens :-) |
| 09:36.02 | tzafrir_laptop | rethus, yes, I read your messages |
| 09:36.38 | tzafrir_laptop | though this does not mean that your IRC client is not broken :-p |
| 09:37.27 | tzafrir_laptop | the conclusion from the above vim trick is that the dialplan is just as sensitive to changes of a single chanracter |
| 09:37.54 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 09:38.47 | jblack | If you wish to use trixbox, be my guest. |
| 09:38.51 | rethus | miloux: thnaks... now i know, it works, and noone here have experiance with php and asterisk |
| 09:39.22 | rethus | thanks a lot. and have a nice day :-) |
| 09:39.29 | *** part/#asterisk rethus (n=rethus@xdsl-84-44-224-114.netcologne.de) |
| 09:40.06 | kamanashisroy | I see using g729 fixes the dtmf problem :) |
| 09:41.16 | michael-i | Hi everyone. Can anyone point me to the right parameter in zapata/zaptel which sets the timeout for entering a number? I want to stretch the input time out a bit since sometimes 3 seconds (estimated) timesout and dials before I'm finished. |
| 09:41.59 | jblack | that's controlled by your phone or ata. |
| 09:42.34 | jblack | I think. I'm not too conversant on FXS devices |
| 09:42.36 | michael-i | I'm pretty sure the analog phone doesn't actively "send" the number. |
| 09:42.50 | *** part/#asterisk redax (i=redax@r6.hu) |
| 09:43.06 | jblack | I'm used to ATA's, which turn analog phones into sip devices. It's built into the ATA for me. Sorry to misdirect. |
| 09:44.00 | michael-i | no problem, just seeing if anyone can point me in the right direction |
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| 10:13.12 | *** join/#asterisk patrick-- (n=patrick@sam.openroot.de) |
| 10:13.57 | patrick-- | hey, im having trouble with asterisk and certain hotline IV's that require keyboad inputs from the phone |
| 10:14.07 | patrick-- | my sipphones dont seem to function properly |
| 10:17.16 | hi365 | how can i announce a doller and cents amount? saynumber seem to ignor the . and everything after :( |
| 10:24.26 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 10:30.13 | kamanashisroy | Hi, how do I know the channel duration ?? |
| 10:30.28 | kamanashisroy | when I am not dialing outgoing call ?? |
| 10:38.15 | *** join/#asterisk X-dark (n=X-dark@wan.agentspace.com) |
| 10:40.24 | X-dark | Hello, I want to disable authentication between two of my sip phones, I've added the option insecure=invite but the authentication is still requested. What do I have to add ? |
| 10:41.57 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 10:44.01 | *** join/#asterisk dikdust (n=dikdust@77.43.42.95) |
| 10:50.34 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
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| 10:51.34 | Shotygun | Hi. Is it possible to execute a context upon sip registration? I understand regcontext (sip.conf) "activates" a dialplan by adding priority 1 but can it also be executed after the addition of the priority ? |
| 10:53.51 | dominic1 | what's the difference between an american areanumber and a mobilenumber? |
| 10:54.28 | Shotygun | dominic1: You mean landline to mobile I believe. As far as I know there is no way to tell the difference, but I could be wrong on this one. |
| 10:55.47 | hi365 | Shotygun: there *shouls* be something that can be doen on registration - check the documentaion on sip (sip.conf.sample?) |
| 10:57.05 | Shotygun | That's what I am thinking, but can't find it, only a way to activate/disactive an extension |
| 10:57.14 | Shotygun | (by playing with priority 1) |
| 10:58.30 | X-dark | I want to disable authentication between two of my sip phones, I've added the option insecure=invite but the authentication is still requested. What else I have to do ? |
| 10:58.51 | Shotygun | There is invite and there is also port I believe |
| 10:59.02 | Shotygun | If it's 1.2 then you could use insecure=very |
| 11:04.43 | X-dark | no it's 1.4.13 |
| 11:06.37 | X-dark | adding port doesn't change anything |
| 11:07.15 | tzafrir_laptop | To rephrase Shotygun' question: is there a manager even sent upon SIP registration? |
| 11:07.54 | hi365 | anyone doing regex's in dialplan? |
| 11:08.45 | hi365 | saynum can handel decimals, aka money :( (i.e. $55.33) |
| 11:08.50 | hi365 | can = cant |
| 11:13.13 | *** join/#asterisk telenieko (i=marc@telenieko.com) |
| 11:13.17 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.121) |
| 11:14.06 | telenieko | Hi. Until my last upgrade I had a single sip peers table in MySQL and "sip show peers" in every asterisk box would only show those in that box. Now if I do "ChanIsAvail" on any box it says "yes" to any peer no matter where it got registered. What changed in 1.4.21 ¿?¿ |
| 11:14.39 | Shotygun | I'm aware it's possible to do it with AMI, but I rather avoid writing a daemon for that purpose and was wondering if there is more elegant way to do it. |
| 11:15.01 | Shotygun | shalom tzafrir =) |
| 11:15.04 | l0verb0y | hey hows it going |
| 11:20.04 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 11:20.04 | *** mode/#asterisk [+o russellb] by ChanServ |
| 11:22.29 | miloux | why does my perl agi scripts always return 0 no matter what i set with return();? doing return(1); still returns 0.... |
| 11:22.42 | hi365 | anyone doing regex's in dialplan? saynum cant handel decimals, aka money :( (i.e. $55.33) - im trying to get the cents part |
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| 11:53.12 | miloux | how do i reload the config for queue.conf? |
| 11:54.24 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-133-231.red.bezeqint.net) |
| 12:00.18 | russellb | miloux: module reload app_queue.so |
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| 12:00.42 | miloux | russellb unfortunately it doesnt work :/ |
| 12:02.20 | russellb | well ... that's how you do it ... |
| 12:02.44 | miloux | The agents are still in the queue |
| 12:02.51 | miloux | do i have to remove them manually? |
| 12:02.53 | russellb | yes |
| 12:02.58 | miloux | how do i do that? |
| 12:03.04 | miloux | from cli pref. |
| 12:03.16 | russellb | how were they added in the first place |
| 12:03.21 | russellb | queues.conf or using addqueuemember |
| 12:03.32 | miloux | quues.conf |
| 12:03.37 | miloux | queue.conf |
| 12:03.51 | russellb | Ah, then I guess a reload should take them out ... |
| 12:04.01 | miloux | it doesnt :/ |
| 12:04.07 | russellb | shrugs |
| 12:04.20 | russellb | i'm not awake enough to go look at the code |
| 12:04.22 | miloux | how do i remove them if addqueuemember was used then? |
| 12:04.27 | miloux | :p |
| 12:04.29 | creativx | removequeuemember |
| 12:04.30 | creativx | :) |
| 12:04.33 | miloux | :D |
| 12:04.42 | miloux | in cli? |
| 12:04.59 | russellb | "queue remove member" |
| 12:12.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:24.22 | miloux | oh btw thanks russellb :) |
| 12:24.29 | miloux | whats wrong with this query: insert into queue_member_table (queue_name, interface) values ( "test", "SIP/9999" ); |
| 12:25.46 | Shotygun | miloux: That depends on the error message of course. |
| 12:25.48 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 12:26.52 | miloux | im trying to insert data into queue_name interface, the data is "test" for queue and "sip/9999" for interface...i get the error message that "test" doesnt exist... |
| 12:27.06 | miloux | but thats what im trying to achive...adding test |
| 12:27.29 | miloux | (this is from CLI in psql) |
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| 12:30.26 | l0verb0y | does anyone know how I can install ramFS on centos? |
| 12:32.09 | [TK]D-Fender | l0verb0y: try asking in ##linux |
| 12:32.29 | [TK]D-Fender | l0verb0y: But my guess it, that with linux being linux... the answer is probably "yes" |
| 12:33.03 | l0verb0y | thx |
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| 12:39.18 | *** part/#asterisk X-dark (n=X-dark@wan.agentspace.com) |
| 12:44.14 | Illarane | [TK]D-Fender: Heh... Yeah, they can be a bit obtuse in there. :/ |
| 12:47.50 | Shotygun | l0verb0y: Not a problem. |
| 12:49.02 | Shotygun | in grub append to the 'kernel' line ramdisk_size=2000000 (2gb), it will create for you /dev/ramdisk, the rest is just mkfs'ing & mounting. |
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| 13:16.30 | *** mode/#asterisk [+o mog] by ChanServ |
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| 13:24.56 | zapp-branigan | i have a problem can i do transfers in the same internal net using stun ? |
| 13:25.05 | zapp-branigan | for the sip |
| 13:25.41 | zapp-branigan | if the stun rewrite the ip then who now the client where to conect in the internal net ? |
| 13:27.15 | [TK]D-Fender | ~sipnat |
| 13:27.16 | 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 |
| 13:27.18 | [TK]D-Fender | zapp-branigan: ^^^^^ |
| 13:27.28 | [TK]D-Fender | zapp-branigan: * does not use STUN and this is a non-issue |
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| 13:29.38 | zapp-branigan | thanks for all :D |
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| 13:37.11 | jaytee | I got this message on my console repeated three times over a couple minutes and then it stopped. The phone shows it's registered when I do a SIP SHOW PEERS. Should I be concerned? http://pastebin.com/m1594164 |
| 13:39.28 | jaytee | sorry, never mind that last question |
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| 14:06.32 | edwin_quijada | Hi! |
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| 14:07.42 | edwin_quijada | I have configured a T1 card normal ,24 channels, and when I try to call to my lines I get fast busy in my line |
| 14:08.18 | LemensTS | If im calling out to a cell phone, is thier any reason to use GSM codec over my normal codec? |
| 14:08.51 | LemensTS | (since it will be changed at the pstn anyways) |
| 14:09.01 | brodiem | LemensTS: not at all |
| 14:09.19 | coppice | its very unlikely your cell phone is using that GSM codec |
| 14:09.58 | *** join/#asterisk bildo (n=tobbe@bildo.tk) |
| 14:10.49 | edwin_quijada | this is my zaptel and zapata http://pastebin.com/m201896a3 |
| 14:10.50 | [TK]D-Fender | LemensTS: and how exactly are you "calling out" to your cell? |
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| 14:12.19 | [TK]D-Fender | edwin_quijada: You have no dchan specified, no signalling (since I get the impression you might actually be using PRI. So far your link has no identity <- |
| 14:12.33 | LemensTS | TKD-Fender: From one asterisk to another asterisk that is connected to an E1 from the telco. Im told the landlines sound good but the cell phones dont. It would probably be a carrier issue right? |
| 14:12.57 | LemensTS | (dialing out to landlines and to cell phones over the E1..) |
| 14:12.58 | *** join/#asterisk jack_sparo (n=jack_spa@dxb-b1310.alshamil.net.ae) |
| 14:13.21 | edwin_quijada | [TK]D-Fender: I am not using a PRI so I dont have a dchannel |
| 14:13.49 | [TK]D-Fender | edwin_quijada: Well you are listing bchans (which imply PRI), and you did not specify signalling in zapata.conf at all. |
| 14:13.52 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:13.56 | [TK]D-Fender | edwin_quijada: So far it says you have nothing at all |
| 14:15.22 | [TK]D-Fender | edwin_quijada: span=1,0,0,d4,ami <- you also calim to be SETTING timing to your telco instead of GETTING it from them (1,1,0) |
| 14:16.03 | [TK]D-Fender | LemensTS: if You're E!, anything other than ALAW is a waste |
| 14:16.05 | edwin_quijada | [TK]D-Fender: I read that is the way to have slave channels use TELCO |
| 14:16.10 | edwin_quijada | timing |
| 14:16.16 | [TK]D-Fender | LemensTS: EVERYTHING gets converted to that at the PSTN point. |
| 14:16.36 | [TK]D-Fender | edwin_quijada: but you aren't using the telco for timing when you do "1,0,0" |
| 14:16.52 | edwin_quijada | [TK]D-Fender: No? |
| 14:17.14 | [TK]D-Fender | edwin_quijada: the 2nd number is the timing. 0 means that * PROVIDES it. 1+ prioritizes GETTING it from the other end |
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| 14:23.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:26.38 | jack_sparo | sorry for the paste guys |
| 14:26.39 | jack_sparo | Description Alarms IRQ bpviol CRC4 |
| 14:26.40 | jack_sparo | Wildcard TDM400P REV I Board 1 UNCONFIGUR 0 0 0 |
| 14:26.47 | jack_sparo | how to configure zaptel? any idea? |
| 14:27.04 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 14:27.39 | LemensTS | YOu seupt zaptel.conf yet |
| 14:28.08 | LemensTS | You setup zaptel.conf yet |
| 14:28.23 | jack_sparo | what shall i do abt it LemensTS |
| 14:29.18 | LemensTS | http://en.wikipedia.org/wiki/Genkernel |
| 14:29.30 | LemensTS | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf |
| 14:29.34 | LemensTS | ignore that first link heh |
| 14:30.39 | *** join/#asterisk puzzled (n=patrick@62-177-241-95.dsl.bbeyond.nl) |
| 14:31.08 | LemensTS | jack: you basicly need your loadzone, defaultzone, and fxoks and fxsks set |
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| 14:35.03 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 14:35.43 | l2cache | Is there an option in queues.conf to not ring agents when they are on a call. It keeps ringing their lines because they have call-waiting |
| 14:36.02 | l2cache | And no, disabling call-waiting is not an option |
| 14:37.55 | tzafrir_laptop | jack_sparo, but if you want a name similar to genkernel, zaptel includes a script called "genzaptelconf" :-) |
| 14:38.18 | [TK]D-Fender | l2cache: What do you use as members? |
| 14:39.00 | l2cache | dynamic logins |
| 14:39.56 | edwin_quijada | [TK]D-Fender: The signalling for T1 normal card 24, channels is signalling=fxo_ks |
| 14:40.00 | [TK]D-Fender | l2cache: If you go through the dialplan, check if they are on a call already |
| 14:40.25 | Yourname`` | Hi I want agent 10 to be a member of two queues. However agent10 has context=testq where in extensions.conf testq hs the relevant adqueuemember, amongst other things. For the member to login to the other queue, how can I make 10 use context of [salesq as well? |
| 14:40.37 | [TK]D-Fender | edwin_quijada: taht is not what you configured |
| 14:41.20 | [TK]D-Fender | Yourname``: Now you're calling DEVICES as if they were AGENTS. |
| 14:41.31 | edwin_quijada | i change but now I get 24 error with ztcfg I will paste it all |
| 14:42.02 | Yourname`` | [TK]D-Fender: Ah ok. So how can I do so? |
| 14:42.07 | [TK]D-Fender | Yourname``: Agents don't have ^&%$# contexts. |
| 14:43.18 | tzafrir_laptop | edwin_quijada, please paste the output of 'cat /proc/zaptel/*' as well as your /etc/zaptel.conf |
| 14:43.49 | [TK]D-Fender | edwin_quijada: Of course you'll get errors. You also tried telling it that if had BCHAN's. Thats a PRI thing. your configuration is suffering from acute multiple personality disorder. |
| 14:44.00 | Yourname`` | So if agent 10's device username is 10 whose context is set to [testq], but agent10 will be logging in to salesq which has a context [salesq] as well, how can I make agent 10 use the stuff in [salesq] [TK]D-Fender ? |
| 14:44.17 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 14:44.28 | [TK]D-Fender | Yourname``: "So if agent 10's device username is 10 whose context is set to [testq], but agent10 " <- this is NOT AN AGENT. |
| 14:44.29 | edwin_quijada | [TK]D-Fender:http://pastebin.com/m3b0acbfb |
| 14:45.39 | [TK]D-Fender | edwin_quijada: line 83 : bchan=1-24 <- Yyou are sitll telling it that its a PRI! |
| 14:46.00 | Yourname`` | [TK]D-Fender: I call him Agent because he's defined as member=>Agent/10 in queues.conf under [testq] |
| 14:46.03 | [TK]D-Fender | Yourname``: "logging in to salesq" <-- huh? |
| 14:46.04 | *** join/#asterisk spokra (n=spokra@host093-179-146.sea0.speakeasy.net) |
| 14:46.18 | Yourname`` | [TK]D-Fender: AQMing into salesq |
| 14:46.40 | [TK]D-Fender | Yourname``: pastebin the exact bloody line. |
| 14:46.48 | brodiem | heh |
| 14:46.53 | edwin_quijada | [TK]D-Fender: i cant use this for this t1 |
| 14:46.56 | [TK]D-Fender | Yourname``: terminology and understanding are seriosly F'd up. |
| 14:47.11 | l2cache | I have dynamic queue member logins setup, is there any way to have the queue not ring the agent's phone when they are on a call (call - waiting) |
| 14:47.16 | l2cache | There has to be a queues.conf setting |
| 14:47.18 | edwin_quijada | How as u said I have "acute multiple personality disorder" :( |
| 14:47.22 | [TK]D-Fender | l2cache: I just told you where. |
| 14:47.32 | l2cache | You said look in the dialplan |
| 14:47.45 | brodiem | l2cache: ringinuse=no |
| 14:47.53 | l2cache | Thank you! |
| 14:47.54 | Yourname`` | [TK]D-Fender: I know.. but you do understand what I'm trying to get help with. |
| 14:48.04 | edwin_quijada | [TK]D-Fender : I took this example from wiki config |
| 14:48.30 | [TK]D-Fender | edwin_quijada: http://pastebin.com/m76251bcf <- Look for my arrow in there |
| 14:48.58 | edwin_quijada | ok |
| 14:49.01 | [TK]D-Fender | edwin_quijada: You took EVERY kind of sample and just mashed it together |
| 14:49.09 | edwin_quijada | I put off here and try again |
| 14:49.16 | l2cache | so ringinuse=no in queues.conf and I'll be good? |
| 14:50.05 | Yourname`` | [TK]D-Fender: http://pastebin.ca/1086156 |
| 14:50.21 | brodiem | l2cache: if you're using chan_agent yes, but if using chan_local then you may still run into problems |
| 14:50.53 | [TK]D-Fender | Yourname``: that has abso-fucking-lutely NOTHING TO DO WITH AQM! |
| 14:50.59 | brodiem | lol |
| 14:51.12 | [TK]D-Fender | ,/embolism> |
| 14:51.20 | l2cache | ok. What should I do if I'm using chan_local |
| 14:51.21 | [TK]D-Fender | </embolism> |
| 14:51.36 | brodiem | l2cache: bridging to sip channel? |
| 14:51.46 | [TK]D-Fender | l2cache: check if they are on a call FIRST <- "core show application chanisavail" |
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| 14:52.37 | Yourname`` | [TK]D-Fender: ok well, print mistake.. I meant to say AgentLogin.. just woke up |
| 14:54.01 | edwin_quijada | [TK]D-Fender: I put off so when i call to my line I get nothing just silence |
| 14:54.48 | Yourname`` | [TK]D-Fender: So you see what I mean now? |
| 14:55.12 | *** join/#asterisk |dennis| (n=Dennis@200.32.231.18) |
| 14:55.33 | Yourname`` | [TK]D-Fender: I made them member but they can only use 4 to AgentLogin if they are assigned that context... but [10] is context=salesq. Unless I could do context=salesq,helpq |
| 14:55.51 | [TK]D-Fender | Yourname``: AgentLogin does not log you into a QUEUE. It logs you in as an AGENT and whchever queues that agent is a member of can beep him. |
| 14:56.13 | [TK]D-Fender | Yourname``: AgentLogin has NOTHING to dowith the friggen DIALPLAN. |
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| 14:57.49 | [TK]D-Fender | Yourname``: You just sit there and wait for a beep. There is no CONTEXT. There is no DIALPLAN. There is no logging into a QUEUE. The Agent is a member wherever the hell he's a member of and calls don't go to the dialplan so long as he's using that app |
| 14:57.52 | jaytee | I'm on the phone with Time Warner telecom's NOC trying to resolve an issue where we can't dial one particular 800 number but any other 800 number we dial seems to work. |
| 14:58.14 | jaytee | I'm sure they'll try and blame Asterisk even though it's going out on a PRI span |
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| 14:58.50 | Yourname`` | [TK]D-Fender: Ah, so one AgentLogin will "log" him into all the queues he's member of? |
| 14:59.01 | brodiem | ding ding |
| 14:59.12 | [TK]D-Fender | Yourname``: you don't long into a ^&$#$##& QUEUE! |
| 14:59.15 | [TK]D-Fender | log* |
| 14:59.18 | mort_gib | What does chan_sip.c: Remote host can't match request BYE to call ' mean?? |
| 14:59.25 | Yourname`` | Ok what do you do then lol |
| 14:59.29 | [TK]D-Fender | Yourname``: Last time. You don't get to CHOOSE what you are member to! |
| 14:59.51 | Yourname`` | If the command itsellf is called "AgentLogin" you should take that up with asterisk.. it implies an agent logging in, no? |
| 15:00.04 | [TK]D-Fender | Yourname``: AgentLogin <- can't say "I don't want sales calls!". No ^&$# choice. |
| 15:00.36 | *** join/#asterisk thomas (i=tm@tm.muc.de) |
| 15:00.37 | thomas | hello |
| 15:00.43 | thomas | i try to set a variable |
| 15:00.50 | thomas | but i have an error: [Jul 29 17:00:17] WARNING[23120]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for extension (eingehend-sipgate, 1954655, 2) |
| 15:00.51 | Yourname`` | [TK]D-Fender: Yes but I'm just confirming that as long as AgentLogin is called by Agent/10 who is a member of salesq and testq, he will start taking calls from both the queues, corrrecto? |
| 15:00.56 | [TK]D-Fender | Yourname``: the INSTRUCTIONS tells you what it does. "Implies" NOTHING. It tells you what it does and you just don't seem to get it. |
| 15:01.00 | thomas | try with: exten => 1954655,n,SetVar(absender=123456) |
| 15:01.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:01.09 | thomas | is it wrong, if yes, why? |
| 15:01.25 | [TK]D-Fender | Yourname``: "AgentLogin is called by Agent/10" <- OMG. |
| 15:01.33 | *** join/#asterisk Talnakh (n=Talnakh@c180-25.i02-6.onvol.net) |
| 15:01.38 | [TK]D-Fender | thomas: SetVar was deprecated in 1.2 for "Set" |
| 15:01.50 | edwin_quijada | [TK]D-Fender: this is my new zapata.conf zaptel.conf |
| 15:01.53 | thomas | [TK]D-Fender: have the version 1.4. |
| 15:01.53 | [TK]D-Fender | Yourname``: It isn't called by an AGENT! |
| 15:02.01 | [TK]D-Fender | thomas: So you should be using SET. |
| 15:02.08 | thomas | [TK]D-Fender: ah, ok. |
| 15:02.08 | edwin_quijada | now when I call I get nothing just SILENCE! |
| 15:02.09 | Yourname`` | OMG!! |
| 15:02.14 | Yourname`` | Then wtf is it then?!?!!? |
| 15:02.19 | Yourname`` | How would you term it?? |
| 15:02.49 | *** join/#asterisk basement_troll (i=hwa@c-24-13-109-243.hsd1.il.comcast.net) |
| 15:02.49 | brodiem | Yourname``: when you successfully login with AgentLogin(), then the agent ID will be a member of any queues where you define that agent ID as a member to |
| 15:02.53 | thomas | [TK]D-Fender: works now! thank you very much, have a nice day! |
| 15:02.57 | Yourname`` | exten,4,n,AgentLogin() -> If Agent10/device10 or whatever the F "10" calls 4, doesn't it go to AgentLogin? |
| 15:03.02 | [TK]D-Fender | Yourname``: DEVICES places calls. SIP/10 is a device. Zap/1 is a device. Devices place calls. |
| 15:03.20 | Yourname`` | brodiem: Thank you. [TK]D-Fender doesn't like me to say "login" and "agent" in the same line. |
| 15:03.26 | brodiem | hehe |
| 15:03.48 | [TK]D-Fender | Yourname``: No, you're the one trying to pour gas into a radiator all the time. NONE of your words match up. |
| 15:04.11 | Yourname`` | lol |
| 15:05.27 | jaytee | god, I need a Xanax |
| 15:06.14 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 15:06.14 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 15:09.05 | twisted | jaytee: i have some |
| 15:09.10 | twisted | actually, I have a lot |
| 15:09.23 | [TK]D-Fender | Yourname``: And if you have a device that points to that context, or is included in some chain to it, then yes, they can dial that exten. |
| 15:09.38 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 15:09.48 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 15:09.57 | hsv-al | hello fellow internet addicts - are we all looking forward to another long & glorious week of irc & internet addiction? :) |
| 15:10.22 | jaytee | twisted, I'd say you're lucky but then I could be wrong |
| 15:10.41 | jaytee | my doc is one of those, "just suck it up!" assholes |
| 15:11.34 | hsv-al | this is bullcrap i have to hack my iphone to access the dam file system in linux and bsd |
| 15:11.41 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 15:11.42 | Qwell | openmoko.org |
| 15:11.42 | hsv-al | cant even access the photos, usb driver = encrypted |
| 15:12.03 | hsv-al | heh |
| 15:12.29 | *** join/#asterisk fogo (n=fogo@69.169.132.121) |
| 15:12.55 | mort_gib | <PROTECTED> |
| 15:13.52 | *** join/#asterisk ccesario (n=ccesario@189.20.219.10) |
| 15:14.09 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 15:14.13 | b11d` | hello all |
| 15:14.47 | b11d` | how many .call files can asterisk handle at a time? Say I have 200 .call files, and I drop them all into the appropriate place for asterisk to pick them up, will Asterisk choke on that? |
| 15:15.34 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:17.30 | brodiem | b11d`: just treat it the same as any normal simultaneous calls... and depends what kind of channels the endpoints are, transcoding, etc.. |
| 15:18.32 | *** join/#asterisk eth01 (i=eth01@gentoo/user/eth01) |
| 15:19.30 | mort_gib | Nobody can help on the chan_sip.c: Remote host can't match request BYE to call ??? |
| 15:20.20 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 15:21.30 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:24.00 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:24.00 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:31.08 | *** join/#asterisk spokra (n=spokra@host093-179-146.sea0.speakeasy.net) |
| 15:34.06 | wasabi | so is making honest to go sip: calls through asterisk work normally? And who might I call? :) |
| 15:34.09 | wasabi | s/is/does/ |
| 15:35.04 | Qwell | wasabi: that still makes no sense |
| 15:35.47 | wasabi | heh. what I mean is, if you were to dial sip:bob@somedomain.com, through a phone, connected to asterisk, is there anything special that you'd need to do to get it to get to somedomain.com? |
| 15:36.05 | Qwell | the phone wouldn't go through Asterisk for that call. |
| 15:36.11 | Qwell | you're dialing that URI directly. |
| 15:37.36 | wasabi | oh the phone just dials it directly? hmm |
| 15:37.45 | Qwell | well, yeah |
| 15:37.48 | edwin_quijada | Somebody knows why asterisk doestn answer when I call it just I get silence in my T1 line |
| 15:37.49 | wasabi | so the phone makes a decision to use the proxy based on what? |
| 15:37.56 | Qwell | who knows |
| 15:38.03 | Qwell | what proxy? |
| 15:38.12 | wasabi | the proxy i have configured on the phone: name of asterisk box |
| 15:38.22 | Qwell | when you ask to dial through it |
| 15:38.58 | edwin_quijada | this is my zap*.conf http://pastebin.com/m48eeab24 |
| 15:39.08 | *** join/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com) |
| 15:39.22 | Hydrant | Hey all, is it possible to receive SMS messages over voip ? |
| 15:39.30 | mort_gib | edwin: so you use zap for T1 |
| 15:39.57 | edwin_quijada | why ztcfg -vv always - 24 channels to configure. |
| 15:40.10 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 15:40.11 | Qwell | edwin_quijada: that's poorly worded |
| 15:40.22 | Qwell | well, sort of.. it makes sense if you understand how it works |
| 15:40.34 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 15:40.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:40.44 | Qwell | first, it goes through and finds all the channels it needs to configure, then it says "I need to configure X channels"...then it configures them |
| 15:41.12 | edwin_quijada | Qwell: Ok but where I can configure I think this is in zapata.conf |
| 15:43.16 | *** part/#asterisk eth01 (i=eth01@gentoo/user/eth01) |
| 15:44.04 | Yourname`` | Hmm, one wonders. AgentLogin.. but no AgentLogout? |
| 15:44.12 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:44.12 | *** join/#asterisk FabiOne (n=FabiOne@151.13.190.20) |
| 15:44.20 | Qwell | Yourname``: one could just...you know...hang up |
| 15:44.21 | FabiOne | hi all |
| 15:44.29 | Yourname`` | you know. |
| 15:44.30 | Yourname`` | hehe |
| 15:45.48 | FabiOne | someone can help me to make cdr_addon_mysql? |
| 15:46.14 | FabiOne | i got an error message when i launch make |
| 15:46.40 | _Sam-- | is there a way to make a script echo something to the cli using asterisk -rx 'something' ? |
| 15:46.44 | jameswf-home | pastebin the error |
| 15:47.00 | jameswf-home | yes sam |
| 15:47.10 | _Sam-- | what would i put in the 'something' part |
| 15:47.25 | edwin_quijada | mort_gib: Yes! I am using zap |
| 15:47.26 | jameswf-home | whatever you want the CLI command to be |
| 15:47.33 | _Sam-- | i want to echo something to the cli |
| 15:47.43 | _Sam-- | echo $rand |
| 15:47.50 | FabiOne | http://pastebin.com/d689f5e13 |
| 15:47.51 | Hydrant | I want to have a voip number that forwards SMS messages to another number... anyone know if such is possible / common ? |
| 15:48.04 | FabiOne | i thik that's a mysql lib problem |
| 15:48.47 | FabiOne | goggling i find many solution |
| 15:48.55 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 15:48.57 | FabiOne | but none of them works |
| 15:50.14 | [TK]D-Fender | Yourname``>Hmm, one wonders. AgentLogin.. but no AgentLogout? <- No. |
| 15:50.28 | [TK]D-Fender | (there isn't, and no need of one) |
| 15:50.37 | *** join/#asterisk emiller (n=ed@static-98-117-7-2.hrbgpa.fios.verizon.net) |
| 15:50.40 | [TK]D-Fender | edwin_quijada: Your signalling is backwards <- |
| 15:51.09 | [TK]D-Fender | edwin_quijada: You should not be "fxoks". You should be "fxsks" |
| 15:51.15 | emiller | hello. is there a command to grab the serial number off of a digium card without physically opening the machine? [shot in the dark here] |
| 15:51.21 | Qwell | emiller: no ;/ |
| 15:51.25 | emiller | drat. |
| 15:51.30 | emiller | worth a try. thank you. |
| 15:51.32 | Qwell | we'd love it if there were |
| 15:51.55 | emiller | trying to get licences for the echo cancel modules from digium |
| 15:52.06 | emiller | they asked for the serials. good thing i documented them so well. |
| 15:52.30 | edwin_quijada | [TK]D-Fender: Ok |
| 15:52.34 | Qwell | yeah, ideally, you should be able to look through the records that you diligently kept ;) |
| 15:52.39 | edwin_quijada | I change! |
| 15:52.40 | jameswf-home | hmmm flash the SN in to the card.... wouldnt be to hard |
| 15:52.47 | jameswf-home | kinda cool actualy |
| 15:52.55 | Qwell | jameswf-home: not terribly hard, no |
| 15:53.07 | FabiOne | someone help me? |
| 15:53.10 | FabiOne | http://pastebin.com/d689f5e13 |
| 15:53.10 | Qwell | but...what are you going to flash it to? :) |
| 15:53.25 | *** join/#asterisk eaglexl (n=dsfds@bzq-82-81-105-46.red.bezeqint.net) |
| 15:53.38 | jameswf-home | Qwell: the firmware.... the processor |
| 15:53.50 | jameswf-home | many options in our case |
| 15:54.06 | jameswf-home | ok a few |
| 15:54.08 | jameswf-home | sorry |
| 15:54.53 | eaglexl | Hey Guys.. I have installed asterix 1.4.21 on debian, and its running without any problems. However, even though the asterisk box is located in a DMZ, people from the outside cannt connect to my server using egika (only sip). What am I doing wrong? |
| 15:55.04 | Qwell | jameswf-home: processor doesn't store information |
| 15:55.09 | _Sam-- | Qwell: is there a command that would echo something to the cli, such as , asterisk -rx 'echo blah', and blah would show on cli? |
| 15:55.10 | Qwell | firmware can be reflashed |
| 15:55.19 | [TK]D-Fender | eaglexl: DMZ is overkill for forwarding, and not enoug for SIP to function behind NAT. Go read the guide : |
| 15:55.21 | [TK]D-Fender | ~sipnat |
| 15:55.22 | 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 |
| 15:55.23 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 15:56.05 | jameswf-home | Qwell: but you put a routine in the firmware flash to capture and rewrite.... sprinkle on some DRM for good measure |
| 15:56.19 | Qwell | or...we could not |
| 15:56.49 | jameswf-home | your right they can simply open the box |
| 15:57.20 | jameswf-home | I write such things down prior to install |
| 15:57.28 | Qwell | liar :p |
| 15:58.01 | jameswf-home | well I do here just because our inventory people are nazis |
| 15:59.10 | jaytee | I've got a weird issue with anyone dialing a particular 800 number. my telco has given me call data from testing that indicates my name is being passed in the callerid info over the PRI. I'm trying to set any callerid info to null. |
| 15:59.59 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 16:03.03 | zeeesh | in queues.conf if i set"monitor-type = MixMonitor" which directory or path save all the files? |
| 16:03.53 | tzafrir_laptop | FabiOne, hmm... /usr/include/asterisk/lock.h:534: error: âPTHREAD_MUTEX_RECURSIVEâ undeclared |
| 16:04.06 | [TK]D-Fender | zeeesh: under "monitor" in whatever spool folder is in asterisk.conf |
| 16:04.12 | tzafrir_laptop | FabiOne, can you try: make NOISY_BUILD=yes |
| 16:04.12 | eaglexl | D-Fender, as much as Im ashemd of myself, I have read most of the ways and solutions, but has no idea how to implent them. Im in my local network, the asterisk server is in DMZ, I can connect. outside of the network, a friend, also in DMZ.. cant connect.. I even tried dyndns.. |
| 16:04.16 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 16:04.39 | [TK]D-Fender | eaglexl: the guide tells you EXPLICITLY what paramters to set. |
| 16:06.08 | eaglexl | well, the ports are all opened.. |
| 16:06.25 | [TK]D-Fender | eaglexl: And I told you its not just about port forwarding. READ THE GUIDE |
| 16:07.10 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:07.10 | *** join/#asterisk hardwire (n=hardwire@rdbk-12639.mtaonline.net) |
| 16:07.39 | eaglexl | I didnt understand how to use STUN |
| 16:08.17 | [TK]D-Fender | eaglexl: And you don't need it. You are doing everything exceppt folowing the guide I linked you to. |
| 16:08.37 | FabiOne | tzafrir_laptop: same error |
| 16:08.45 | jaytee | is exten => 1234,1,Set(CALLERID(all)="") the correct way to set all the CID info to null? |
| 16:09.01 | eaglexl | crap, I just found out that I kept opening voip-info link, sry :\ |
| 16:09.02 | [TK]D-Fender | jaytee: Most 1-800's won't ALLOW you to call them without a number |
| 16:09.14 | tzafrir_laptop | FabiOne, sure. But now the gcc line is more explicit, right? |
| 16:09.25 | FabiOne | sorry |
| 16:09.25 | tzafrir_laptop | Coulw you please pastebi it? |
| 16:09.33 | jaytee | [TK]D-Fender, somehow it's passing my name which is what TW Telecom thinks is screwing up the call |
| 16:09.56 | edwin_quijada | [TK]D-Fender: I change to fks but I get just silence |
| 16:10.11 | [TK]D-Fender | jaytee: Set(CALLERID(all)="" <1234567890>) <- probably better |
| 16:10.53 | [TK]D-Fender | edwin_quijada: Go ask your telco to send you a detailed spec sheet of waht signalling they are sending and provide it to us |
| 16:10.58 | jaytee | [TK]D-Fender, thanks, I'll give that a shot. So far since we went live with * on our outbound T1 this is the only 800 number that's done this. |
| 16:11.18 | FabiOne | tzafrir_laptop: http://pastebin.com/d79af4773 |
| 16:11.41 | jaytee | and the tech at their NOC says it's because their end wants to open the B channel before the call is connected to play an announcement. |
| 16:11.51 | zeeesh | is it possible to use virtaul server for call conference .. logically it works or not ? |
| 16:12.30 | wasabi | so are there any phones that support smart cards? :) |
| 16:13.02 | [TK]D-Fender | zeeesh: If you can somehow get Zaptel working, yes. But that isn't very likely. |
| 16:15.08 | *** join/#asterisk Segnale007 (n=Segnale0@host188-251-dynamic.26-79-r.retail.telecomitalia.it) |
| 16:15.17 | zeeesh | how to check ztdummy installed or not ? |
| 16:15.54 | eaglexl | D-Fedner, I have finally read the right guide, and my config was something like that anyway. My friend still cant connect. If I am behind NAT and DMZ, should I write nat=yes? |
| 16:15.54 | [TK]D-Fender | zeeesh: "zap show channels" and then try to USE IT |
| 16:16.33 | [TK]D-Fender | eaglexl: the guide tells you a PILE of settings and you come back asking as if there was a single yes/no to fill in. Read the ENTIRE guide and pay attention. |
| 16:17.00 | [TK]D-Fender | eaglexl: After trying AGAIN, please PASTEBIN your sip.conf masking only passwords. |
| 16:17.02 | [TK]D-Fender | ~pb |
| 16:17.03 | 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 |
| 16:17.04 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 16:17.48 | eaglexl | I know what pasebin is :P |
| 16:20.46 | zeeesh | <[TK]D-Fender> No such command 'zap show' (type 'help' for help)? |
| 16:21.12 | eaglexl | http://pastebin.com/f34d7e8b7 |
| 16:21.13 | [TK]D-Fender | zeeesh: You need to have compiled and configured zaptel BEFORE compiling *. |
| 16:21.57 | [TK]D-Fender | eaglexl: externip=eaglex.no-ip.org <-- this needs to be externhost, not externip. And you'll need an externrefresh set as well |
| 16:22.25 | *** join/#asterisk legis (i=estar@unaffiliated/legis) |
| 16:22.39 | eaglexl | not changing my password was smart. |
| 16:23.01 | [TK]D-Fender | SMRT |
| 16:23.26 | legis | Hi, any recommended ITSP to peer with asterisk? for calls to US/Canada. |
| 16:23.46 | edwin_quijada | [TK]D-Fender: if i cant use "zap show channels" it is not installed? |
| 16:24.18 | [TK]D-Fender | edwin_quijada: try "load chan_zap.so" |
| 16:24.35 | [TK]D-Fender | legis: les.net is decent. |
| 16:24.37 | LemensTS | legis: flat planet phone company has been good to me so far |
| 16:25.00 | eaglexl | Fender, Sadly, its still not working :\ |
| 16:26.20 | edwin_quijada | [TK]D-Fender: I get 2 warnings |
| 16:26.27 | *** part/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 16:26.32 | [TK]D-Fender | eaglexl: now pastebin a CALL at verbose 10, SIP DEBUG enabled. |
| 16:26.40 | jameswf-home | ~itsplist-us |
| 16:26.41 | 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 |
| 16:27.17 | edwin_quijada | i cant use this command zap show channels |
| 16:27.22 | LemensTS | james: is thier a list of ~ references i can see all at once? |
| 16:27.27 | edwin_quijada | No such command |
| 16:27.32 | jameswf-home | which means chan_zap is not loaded |
| 16:27.44 | [TK]D-Fender | edwin_quijada: Then it FAILS to load. |
| 16:28.02 | jameswf-home | LemensTS: no but you can /msg jbot and play with him |
| 16:28.04 | legis | LemensTS: [TK]D-Fender: thx |
| 16:28.50 | *** join/#asterisk |||Mad||| (n=mad@mail.rubbusa.com) |
| 16:28.51 | jameswf-home | I wonder who owns jbot... |
| 16:29.06 | [TK]D-Fender | jameswf-home: Tim Riker |
| 16:29.37 | jameswf-home | [TK]D-Fender: does he come in ? |
| 16:29.54 | LemensTS | yea how do we update them? |
| 16:29.59 | eaglexl | Call at verbose 10? |
| 16:30.01 | [TK]D-Fender | jameswf-home: Never here that I've seen. I tried e-mailing him once. |
| 16:30.19 | [TK]D-Fender | eaglexl: do "set verbose 10" and "sip debug" at * CLI and place your call |
| 16:30.23 | LemensTS | like zaptel should say that it is now known as dahdi in 1.6 |
| 16:30.29 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-077-155.pools.arcor-ip.net) |
| 16:30.53 | |||Mad||| | Hello, all! Got my Asterisk box up and working pretty well, just gotta quick question: how can I tell if an extension dialled on an analog line is busy so I can sent the caller to voicemail? |
| 16:31.18 | jameswf-home | ~bdhit |
| 16:31.48 | |||Mad||| | I seems ${DIALSTATUS} doesn't work |
| 16:33.15 | tzafrir_laptop | FabiOne, what version of asterisk do you have installed? What version of asterisk-addons do you try building? |
| 16:33.15 | [TK]D-Fender | |||Mad|||: PASTEBIN what you're doing now. |
| 16:33.17 | [TK]D-Fender | ~pb |
| 16:33.17 | 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 |
| 16:33.18 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 16:33.44 | eaglexl | wow, D-fender, thats a lot of text, should I paste it all? |
| 16:34.29 | [TK]D-Fender | eaglexl: yes |
| 16:35.05 | jameswf-home | wow tim ricker big mormon... no note of jbot or asterisk in his bio |
| 16:35.31 | tzafrir_laptop | tim riker, you mean? |
| 16:35.51 | jameswf-home | yeah him too |
| 16:35.53 | eaglexl | http://pastebin.com/m56ce4ad0 enjoy! :) |
| 16:35.54 | tzafrir_laptop | jbot, tell jameswf-home about timriker |
| 16:36.40 | jameswf-home | jbot seen timriker |
| 16:36.44 | jbot | timriker is currently on #bzflag (1h 34m 19s) #utah (1h 34m 19s) #openmoko (1h 34m 19s) #infobot (1h 34m 19s) #elinux (1h 34m 19s), last said: 'bike ride?'. |
| 16:37.08 | Qwell | jameswf-home: I don't think he does too much with Asterisk |
| 16:37.34 | jameswf-home | odd I looked in bzflag and didnt see him. |
| 16:37.50 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 16:38.06 | |||Mad||| | D-Fender: http://pastebin.com/d15b7993c |
| 16:38.08 | [TK]D-Fender | eaglexl: ..... So far that doesn't look like a NAT involved call at all. |
| 16:38.15 | jameswf-home | my client sucks |
| 16:38.25 | |||Mad||| | That's the relevant portion of it |
| 16:38.51 | eaglexl | Hmm, well, according to my router im behind a nat.. |
| 16:38.52 | edwin_quijada | how can activate de full debug to asterisk? |
| 16:39.14 | [TK]D-Fender | eaglexl: But you aren't NAT'd RELATIVE to Asterisk |
| 16:39.30 | [TK]D-Fender | eaglexl: And right now I'd blame your SOUND CARD. |
| 16:39.39 | [TK]D-Fender | eaglexl: Go verify your mic & speakers |
| 16:40.09 | eaglexl | wait, let me understand... how does my mic&speakers has anything to do with this? |
| 16:40.18 | [TK]D-Fender | |||Mad|||: [macro-voicemail] <- has no "1" priority. BAD. Also you combine 2 things on 1 line here : exten=_1[0-4]X,1,Macro(voicemail,${EXTEN})exten=s,1,Dial(Zap/1/${ARG1},10) <- totally not valid |
| 16:40.46 | [TK]D-Fender | eaglexl: Ekiga is a soft phone. If your sound card settings are bad it isn't LISTENING to your MIC at all. |
| 16:41.14 | [TK]D-Fender | eaglexl: Or maybe it is picking up from the mic, but you can't hear the echo because your speakers are cut off, etc |
| 16:41.43 | |||Mad||| | Whoops, something didn't paste right there, the [macro-voicemail] is a line down from where it should be |
| 16:41.48 | |||Mad||| | Lemmee redo it |
| 16:42.00 | eaglexl | I think I lost you... my problem is that people from outside the network cant CONNECT to my asterisk server, nothing involving speaking.. |
| 16:42.32 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 16:42.34 | [TK]D-Fender | eaglexl: And yo aren't SHOWING me a call originating from outside your network in that pastebin. |
| 16:42.56 | [TK]D-Fender | eaglexl: <--- SIP read from 10.0.0.4:5064 ---> <-- INSIDE your LAN |
| 16:43.04 | Blackvel | wasn't there any special bristuff irc channel long time ago? what was it's name? |
| 16:43.11 | eaglexl | Because he cant connect! :| |
| 16:43.25 | |||Mad||| | D-Fender: http://pastebin.com/d292cf9eb |
| 16:43.32 | [TK]D-Fender | eaglexl: Show me his FAILED attempt to connect. have him restart his phone and pastebin the debug for it |
| 16:43.51 | eaglexl | he his always using ekiga now, for testing purpoes. nothing show in asterisk cli... |
| 16:43.58 | [TK]D-Fender | |||Mad|||: now pastebin a call. |
| 16:44.14 | [TK]D-Fender | eaglexl: Then the packets aren't reaching your box at all. |
| 16:44.22 | [TK]D-Fender | eaglexl: Maybe he screwed up all of his settings |
| 16:44.26 | |||Mad||| | OK, lemmee make one with a busy line |
| 16:44.34 | [TK]D-Fender | eaglexl: Go check your server's firewall wihle you're at it |
| 16:44.40 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 16:45.12 | eaglexl | as of now, no firewall at all, until I get it fully up and running |
| 16:45.25 | [TK]D-Fender | eaglexl: Go prove it while you're at it. |
| 16:45.49 | Alan_Hicks | I've got a strange intermittent problem on my simple asterisk box. I had a TDM400 card with four FXO modules, only two of which are ever in use (testing box). I'm using polycom sip phones. Most of the time everything works fine, but every few days it's like the TDM card locks up. Phone calls can't be placed into or out of the card. |
| 16:46.42 | Alan_Hicks | SIP calls between the IP phones work fine, even through asterisk, but any calls into my POTS lines receive a busy signal. Attempting to call out from the SIP phones results in a loud buzzing sound, not a busy or fast busy sound, just sort of like static. |
| 16:47.17 | Alan_Hicks | Restarting asterisk alone doesn't help. I have to remove the wctdm module from the kernel and reload it. Since this only happens once every few days, I'm having a devil of a time troubleshooting it. |
| 16:47.38 | |||Mad||| | D-Fender: CLI output is here: http://pastebin.com/d51402635 |
| 16:48.17 | Alan_Hicks | I don't _think_ there's a problem with my asterisk configuration as all calls work some of the time and some of the calls work all of the time. Is this most likely a hardware issue? What files should I pastebin here to help troubleshooting? |
| 16:49.04 | mags2 | Alan_Hicks a long while ago it was recommended to prefer not to use port 1 or port 4 of that card. not sure it's still true... |
| 16:49.35 | [TK]D-Fender | |||Mad|||: -- Zap/1-1 answered Zap/3-1 <----- answered |
| 16:49.40 | eaglexl | Im starting to think it is something with my no-ip update client... |
| 16:49.45 | |||Mad||| | D-Fender: What happens is the call plays busy for the 10 second Dial timeout... I'd like the phone to try ringing for 10 seconds, then if unanswered or busy to go to voicemail |
| 16:49.52 | [TK]D-Fender | eaglexl: Go resolve your host and verify |
| 16:49.58 | Alan_Hicks | I've switched the ports in use between 1 and 2 to 3 and 4. Haven't tried 2 and 3 by themselves. |
| 16:50.06 | [TK]D-Fender | |||Mad|||: -- Zap/1-1 answered Zap/3-1 <----- answered |
| 16:50.50 | |||Mad||| | Yes, I let the busy signal run for 10 seconds... but most people won't listen to a busy signal that long |
| 16:51.53 | Alan_Hicks | I am wondering if this problem may be related to my test configuration. We have just two PSTN lines at my office- one for voice and one for fax. I have these two lines also plugged into my asterisk box in addition to several analog phones and the fax. Might this be causing problems? |
| 16:52.02 | [TK]D-Fender | |||Mad|||: Right now I see no "busy" |
| 16:52.48 | |||Mad||| | Can Asterisk detect a busy line on an analog port? |
| 16:52.53 | BBHoss | can someone explain this: [Jul 29 11:52:19] WARNING[6904]: app_voicemail.c:6294 vm_authenticate: Couldn't read username |
| 16:53.06 | mags2 | Alan_Hicks you plugged phones and fax into fxo modules? |
| 16:53.14 | BBHoss | DTMF works otherwise, its just like VM is not picking it up |
| 16:53.21 | Alan_Hicks | mags2: no no no no |
| 16:53.58 | Alan_Hicks | They are terminated at my demarc on the same PSTN number that my FXO cards are terminated to. |
| 16:54.08 | mags2 | o ok |
| 16:55.23 | scooby2 | anyone using redundant/failover asterisk servers? Any way to automatically switch the T1's over or does that have to be done by hand. |
| 16:56.03 | [TK]D-Fender | |||Mad|||: if you do "callprogress=yes" in zapata.conf, set the right zone information and are INCREDIBLY lucky, yes |
| 16:56.23 | [TK]D-Fender | BBHoss: No entry <- |
| 16:56.54 | BBHoss | [TK]D-Fender: do i need to do mailbox=106@default in sip.conf? |
| 16:57.12 | *** join/#asterisk oilinki (n=oil@ppp-124-120-10-53.revip2.asianet.co.th) |
| 16:57.16 | [TK]D-Fender | BBHoss: Nothing to do with. they did not ENTRY a box # at all. |
| 16:57.19 | [TK]D-Fender | ENTER* |
| 16:57.41 | BBHoss | ok well asterisk is not reading the dtmf then |
| 16:57.43 | *** join/#asterisk kb1_kanobe (n=kris@bdr2.fieldrd.scrd.ca) |
| 16:57.49 | |||Mad||| | D-Fender: well, it's worth a shot... where do I set the right zone information? |
| 16:57.54 | Blackvel | who can help me combine two dial commands with old asterisk 1.0 and bristuff for ZAP dailing? http://pastebin.com/d335cd011 |
| 16:58.16 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 16:58.18 | [TK]D-Fender | |||Mad|||: zaptel / zapata. Go read |
| 16:58.35 | Blackvel | both DIAL commands work alone, but the 2nd always gets a busy and invalid hangup channel message |
| 16:58.52 | *** join/#asterisk tekryan (n=Anonymou@209.133.228.121) |
| 16:58.52 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 16:59.21 | tekryan | Hi. Is this channel for asteriskNOW as well? |
| 16:59.49 | JerJer | no |
| 17:00.08 | JerJer | tekryan: #asterisknow |
| 17:00.10 | Blackvel | I understand that on timeout priority gets 1+1 = 2 and the 2nd DIAL would be executed. I can see that the channel gets a hangup, but for some reason it fails. did i forget something after the 1st DIAL command or is priorty 2/3 for 2nd DIAL wrong? |
| 17:00.13 | tekryan | Is there a channel for that? |
| 17:00.26 | tekryan | Thanks... |
| 17:00.30 | tekryan | Didn't see that till now |
| 17:01.56 | tzanger | Blackvel: that is why you never ever rely on that |
| 17:02.02 | tzanger | use Dial() with 'g' flag |
| 17:02.10 | tzanger | and check ${DIALSTATUS} and branch based on that |
| 17:02.19 | tzanger | never ever rely on +1 or +101 (j flag) |
| 17:02.29 | Qwell | well, +1 is fine |
| 17:03.51 | tzanger | no +1 sucks ass |
| 17:04.37 | *** join/#asterisk swiftkick (i=eeeeeeee@mail.beanproducts.com) |
| 17:04.48 | tzanger | especially if you want to do post-call processing |
| 17:04.50 | tzanger | but anyway |
| 17:05.14 | Blackvel | its my internal pbx |
| 17:05.28 | Blackvel | and i want to kick ass all consulting firms and present IVR |
| 17:05.53 | Katty | so. stuffed. |
| 17:05.55 | swiftkick | can someone point me to the specs of how to specify outbound caller ID? e.g. not only specs on calling the CALLERID function but also sip.conf and users.conf callerid = and cid_number = |
| 17:06.10 | Blackvel | and after pressing 4, i want to call my normal pots telephone, and if i am not there, I want to run 2nd DIAL and call my mobile phone over POTS/PBX :) |
| 17:06.31 | Blackvel | tzanger: ok, going that way then |
| 17:06.36 | swiftkick | in fact i'd love to see some docs on users.conf vs sip.conf for that matter. |
| 17:07.21 | Blackvel | its a while ago since i last configured / tuned my local asterisk pbx...hehee...having to find all docs again |
| 17:07.45 | Katty | [TK]D-Fender: herro? |
| 17:08.28 | Qwell | pokes Katty |
| 17:08.34 | Katty | squeaks. |
| 17:08.45 | Blackvel | if i have two dial commands (1st prio 1, 2nd prio 2), wouldn't 1st dial hangup the 2nd zap channel before 2nd dial command creates a new zap channel for my mobile phone? I seem not to understand why DIAL gets busy channel |
| 17:09.10 | tzanger | hmm, I am having some troubles with HPEC |
| 17:09.15 | tzanger | it emits noise instead of cancelled echo |
| 17:09.21 | Qwell | tzanger: O.o |
| 17:09.29 | tzanger | seems to be at any tap setting, from 64 (which is technically invalid) through to 1024 |
| 17:09.32 | Blackvel | changing this to using DIALSTATUS and Dial() with g option probably will not change too much on it? |
| 17:11.36 | *** join/#asterisk jpcansa (n=jpbenavi@201.206.229.34) |
| 17:13.05 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:15.05 | jpcansa | HI, any idea why my zap channels got stucked like this: http://imagebin.ca/view/tj1x8r.html ??? |
| 17:15.08 | FabiOne | wow |
| 17:15.45 | FabiOne | recompiling entire bristuff package after installing libmysqlclient-dev everything goes ok |
| 17:16.06 | FabiOne | now i've cdr on mysql |
| 17:16.41 | Qwell | jpcansa: please don't upload errors like that as images. :) |
| 17:16.44 | Qwell | that site is terribly slow |
| 17:17.10 | jpcansa | didnt know, where can i upload images? |
| 17:17.27 | [TK]D-Fender | Katty: Mew. |
| 17:19.31 | jpcansa | @Qwell: do you have any idea on my problem? |
| 17:19.59 | Qwell | is away after being idle for 3 minutes. |
| 17:20.29 | *** join/#asterisk fogo- (n=fogo@rs-69-169-132-121-0003.broadweave.net) |
| 17:20.54 | edwin_quijada | how can activate debug of asterisk? |
| 17:21.39 | *** join/#asterisk kc2tnk (n=fskrotzk@host.textwise.com) |
| 17:21.39 | edwin_quijada | My telco told me that my signaling is wink and my trunk lines are E&M |
| 17:21.42 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-8-115.nyc.res.rr.com) |
| 17:22.31 | edwin_quijada | I change the zaptel and zapata but I am not getting anything |
| 17:22.45 | edwin_quijada | just silence when I call |
| 17:22.58 | [TK]D-Fender | edwin_quijada: pastebin your configs |
| 17:23.09 | edwin_quijada | What should be the output for ztcfg -vvv |
| 17:23.11 | edwin_quijada | ? |
| 17:23.16 | [TK]D-Fender | edwin_quijada: pastebin your configs <----- |
| 17:23.49 | edwin_quijada | [TK]D-Fender:Ok |
| 17:27.26 | edwin_quijada | [TK]D-Fender: http://pastebin.com/m131231c1 |
| 17:27.37 | Katty | [TK]D-Fender: i think i found my problem. |
| 17:27.46 | Katty | [TK]D-Fender: http://pastebin.ca/1086262 |
| 17:28.17 | [TK]D-Fender | edwin_quijada: e&m = bad. e&mw = good. You are WINK start. |
| 17:28.24 | |||Mad||| | D-Fender: Got it, sort of... I had to add "busydetect=yes" to zapata.conf... now it detects the busy, but reports the extension as "Unavailable" |
| 17:28.33 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 17:28.35 | [TK]D-Fender | Katty: lol |
| 17:28.50 | *** join/#asterisk FabiOne (n=FabiOne@151.13.190.20) |
| 17:29.10 | [TK]D-Fender | edwin_quijada: signalling=em <- not even a valid choice. |
| 17:29.21 | [TK]D-Fender | edwin_quijada: Pay attention to the allowed values. |
| 17:29.24 | Katty | is aunty em a valid choice? |
| 17:29.50 | [TK]D-Fender | Katty: You've insaned. |
| 17:30.11 | Katty | [TK]D-Fender: been there. done that. |
| 17:30.13 | Katty | [TK]D-Fender: got the shirt. |
| 17:30.22 | [TK]D-Fender | edwin_quijada: And as if that wasn't enough, you tried declaring your signalling AFTER declaring your channels. |
| 17:30.37 | [TK]D-Fender | Katty: They ran out of stock for mine. |
| 17:30.38 | Katty | [TK]D-Fender: i've also had too much lunch. nap required. |
| 17:31.05 | [TK]D-Fender | Katty: kat-nap tyme |
| 17:31.26 | Katty | [TK]D-Fender: pls to provide pillow. |
| 17:31.38 | Katty | i actually have a pillow and blanket at work. |
| 17:32.11 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 17:32.36 | Katty | scampbell: ohai |
| 17:33.15 | edwin_quijada | [TK]D-Fender: i am reading the zaptel.comf and I dont see e&mw but e&m yes |
| 17:33.40 | edwin_quijada | the capitalize first letter? |
| 17:34.22 | *** join/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 17:34.36 | pabelanger | anybody have a polycom soundpoint 301 handy? |
| 17:35.36 | Katty | if i did, i'd punt it. |
| 17:35.47 | Katty | and then punt whoever ordered it (= |
| 17:36.39 | *** join/#asterisk The-Bat (i=HydraIRC@221.134.199.14) |
| 17:37.23 | [TK]D-Fender | edwin_quijada: http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf.sample |
| 17:38.22 | *** join/#asterisk mrnick (n=ubugo@88.197.232.204) |
| 17:38.28 | mrnick | hi |
| 17:38.54 | Katty | ohai. |
| 17:39.47 | [TK]D-Fender | pabelanger: Just ask your actual question. |
| 17:40.31 | jameswf-home | ~ask |
| 17:40.32 | jbot | ask is probably Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:40.53 | pabelanger | I need to confirm the power adapter is 12V DC 400mA for the SoundPoint 301 |
| 17:40.56 | edwin_quijada | [TK]D-Fender: in this example they use em and em_w which use |
| 17:40.57 | edwin_quijada | ? |
| 17:40.58 | Katty | oh boy. on topic. |
| 17:41.08 | Katty | jbot: will you go out to dinner with me? |
| 17:41.14 | Katty | jbot: harsh. |
| 17:41.32 | jameswf-home | jbot: hug Katty |
| 17:41.33 | jbot | ACTION hugs Katty tightly until Katty turns slightly blue |
| 17:41.33 | [TK]D-Fender | edwin_quijada: You said they use WINK, so use your brain. |
| 17:42.06 | Katty | jameswf-home: ohai. |
| 17:42.14 | Katty | jameswf-home: how's rhino |
| 17:42.38 | jaytee | has anyone ever seen a problem with dialing an 800 number over a PRI using Asterisk 1.4 and Digium TE212P card where the number would not answer? I can dial a bunch of 800's that all work fine but this one number for Northwest Air Cargo. |
| 17:42.38 | *** join/#asterisk drumkilla (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:42.38 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 17:42.42 | jameswf-home | chuggin along |
| 17:42.47 | Katty | jaytee: i did. |
| 17:42.52 | jameswf-home | is happy it is taco tuesday |
| 17:42.53 | Katty | jaytee: but it turned out to be a telco problem. |
| 17:43.05 | Katty | jameswf-home: oh? taco tuesday? do tell |
| 17:43.19 | jameswf-home | dell taco 3 tacos for $1 |
| 17:43.24 | jameswf-home | *del |
| 17:43.29 | Katty | del? |
| 17:43.34 | jaytee | my telco is still working on it but he says this 800 number wants to "open" my B-channel before the call is connected. wthell is up with that? |
| 17:43.35 | Katty | Del Taco(tm)? |
| 17:43.40 | jameswf-home | si |
| 17:43.45 | Katty | fascinating. |
| 17:43.47 | jaytee | we don't have del tacos here |
| 17:43.54 | Katty | we don't have them here either. |
| 17:44.02 | Katty | jaytee: i've come to learn telcos are stupid. |
| 17:44.05 | jaytee | I used to eat there in Oklahoma City back in the day. |
| 17:44.22 | Katty | jaytee: i presume the cli is not helpful? |
| 17:44.23 | scooby2 | jaytee: sounds like a telco issue. Make up any excuse to get out of fixing the problem. |
| 17:44.38 | [TK]D-Fender | pabelanger: 12v is right. 400mw sounds right from memory. |
| 17:44.39 | *** join/#asterisk Levonk (n=lk@adsl-76-230-111-146.dsl.lsan03.sbcglobal.net) |
| 17:44.52 | *** join/#asterisk DSpair (n=D-Spare@163.muaa.syrc.chcgil24.dsl.att.net) |
| 17:44.57 | Katty | jaytee: i don't suppose you have an analog line sitting around somewhere, like a fax line... |
| 17:44.59 | DSpair | Hello again all... |
| 17:45.04 | Katty | jaytee: where you could actually test it. |
| 17:45.09 | DSpair | Quick question on relax DTMF. |
| 17:45.10 | |||Mad||| | D-Fender: this is odd... in the call log, that extension is being reported as busy, yet for some reason the macro is executing the "unavailable" option - any ideas? |
| 17:45.13 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 17:45.16 | |||Mad||| | <PROTECTED> |
| 17:45.25 | |||Mad||| | <PROTECTED> |
| 17:45.26 | jaytee | Katty, I've tested from cell and a POTS line and they both answer. |
| 17:45.33 | Katty | mad. |
| 17:45.37 | LemensTS | . |
| 17:45.39 | Katty | sounds telco for sure. |
| 17:45.40 | LemensTS | .. |
| 17:45.46 | pabelanger | [TK]D-Fender: Thanks... want to confirm 400mA before plugging into phone. Don't want it toasted. |
| 17:45.48 | LemensTS | hello |
| 17:45.52 | Katty | jaytee: same problem we had. |
| 17:45.54 | DSpair | My boss called an AT&T auto-attendant today, and consistently during a portion of the recording, the call terminates automatically. When we call from a cell phone, the call does not drop. |
| 17:45.56 | LemensTS | whoop sorry. |
| 17:46.06 | [TK]D-Fender | pabelanger: lemme see if its marked on the phone itself... |
| 17:46.23 | DSpair | I have already turned on relaxed DTMF as suggested in various forums. Can anyone make another suggest on how to resolve this? |
| 17:46.28 | edwin_quijada | I still get 24 channels to configure form ztcfg -vvv |
| 17:46.31 | Katty | jaytee: what kind of outgoing callerid do you have? |
| 17:46.38 | edwin_quijada | this is OK? |
| 17:46.45 | Katty | jaytee: well nevermind, you can call other numbers to that company, right? |
| 17:46.49 | |||Mad||| | Katty: telco? I don't understand |
| 17:47.00 | Katty | |||Mad|||: Telephone Company |
| 17:47.12 | pabelanger | [TK]D-Fender: Ok, thanks. Mine does not list it. Could you check your adapter if you have one? |
| 17:47.13 | jaytee | Katty, I've tried to just send the number but I'm not sure how to use NoOp to show what's actually being sent. |
| 17:47.18 | kb1_kanobe | DSpair: what is your connection to the telco? |
| 17:47.30 | DSpair | kb1_kanobe, PRI T1 |
| 17:47.43 | |||Mad||| | Katty: Right, but the calls are never leaving the building... it's to an analog port on our PBX |
| 17:47.55 | kb1_kanobe | DSpair: does the call terminate when a dtmf key is pressed, or at the same point in time after the call starts each time? |
| 17:48.18 | DSpair | kb1_kanobe, We never press a key, it's part of the recording playing that must be doing it. |
| 17:48.22 | Katty | jaytee: exten => s,1,NoOp(${CALLERID(all)}) ? |
| 17:48.29 | *** join/#asterisk Andares (n=root@unaffiliated/jacco) |
| 17:48.32 | kb1_kanobe | DSpair: about how long into the recording? |
| 17:48.34 | Andares | Hi guys. |
| 17:48.37 | Katty | jaytee: i think? |
| 17:48.40 | |||Mad||| | Katty: The line really is busy, and the log shows it being detected as busy, yet the macro sends people to the unavailable box of viocemail |
| 17:48.43 | Katty | |||Mad|||: Pardon? |
| 17:48.46 | DSpair | kb1_kanobe, Perhaps 30-45 seconds? |
| 17:48.54 | Katty | |||Mad|||: i have no idea what you're talking about. |
| 17:48.59 | Andares | Is there any way I can get asterisk to initiate a dialout on one of the phones? |
| 17:49.14 | *** join/#asterisk ruinous (n=ruinous@74.202.196.195) |
| 17:49.15 | jameswf-home | |||Mad|||: pb your dialplan |
| 17:49.16 | Katty | Andares: call file? |
| 17:49.19 | Andares | like to make an outgoing call show up on a random phone before they dile. |
| 17:49.20 | Andares | *dial |
| 17:49.28 | |||Mad||| | http://pastebin.com/d51402635 |
| 17:49.30 | Andares | Katty: what's that? Please excuse my ignorance. :( |
| 17:49.32 | file | I do not accept calls. |
| 17:49.37 | Katty | file: you so do. |
| 17:49.37 | Andares | haha file. |
| 17:49.38 | jameswf-home | Andares: on 1.4 originate |
| 17:49.40 | |||Mad||| | Oops, that was incorrect |
| 17:49.48 | |||Mad||| | http://pastebin.com/d292cf9eb |
| 17:49.48 | Katty | Andares: sec. i'll get you my blog page. |
| 17:49.50 | |||Mad||| | That one |
| 17:49.53 | Andares | Katty: thanks. |
| 17:50.06 | Andares | jameswf-home: so use that command on the console? cool. |
| 17:50.06 | mrnick | is there a way to hang up but continue the dialplan with the peer? |
| 17:50.19 | Katty | Andares: http://angela.sleekgeek.org/2008/03/13/e911/ |
| 17:50.22 | jameswf-home | |||Mad|||: that is the CLI I wanna see dialplan |
| 17:50.27 | kb1_kanobe | DSpair: I've seen that behaviour when the call isn't properly answered. If the call isn't all the way 'up' it will eventually get disconnected by the CO. |
| 17:50.30 | [TK]D-Fender | pabelanger: have a PoE-only one here. I'll have an answer for 5pm. |
| 17:50.32 | ruinous | question for you guys: I have some phones that I moved to new extensions, they can dial out to external numbers, but are unable to be called from other extensions |
| 17:50.33 | [TK]D-Fender | (est) |
| 17:50.35 | Andares | Thanks Katty. |
| 17:50.40 | Katty | Andares: you basically make a text file and move it to the ougoing spool. |
| 17:50.52 | Katty | Andares: when it hits outgoing, the server makes a call. |
| 17:50.54 | DSpair | kb1_kanobe, I'll try tracing the call and see what the reason is for the disconnect. |
| 17:51.07 | Katty | Andares: that particular example makes the phones auto answer to a conference room, but you can do whatever. |
| 17:51.07 | DSpair | kb1_kanobe, Thanks for the ideas!!! |
| 17:51.13 | kb1_kanobe | DSpair: you're welcome. |
| 17:51.16 | jameswf-home | mrnick: exten => 111,h,example |
| 17:51.36 | mrnick | thx |
| 17:51.37 | |||Mad||| | jameswf: dialplan is http://pastebin.com/d292cf9eb |
| 17:51.46 | |||Mad||| | jameswf: current CLI is http://pastebin.com/d292cf9eb |
| 17:51.52 | |||Mad||| | Argh |
| 17:51.52 | Andares | Katty: so I could, say, make phone #1002 show that it's making an outgoing call once it hits outgoing? |
| 17:52.00 | mrnick | i'll try it right away |
| 17:52.01 | |||Mad||| | jameswf: current CLI is http://pastebin.com/d24b67dd3 |
| 17:52.06 | Katty | Andares: you do not parse. reword. |
| 17:52.36 | Katty | Andares: with cron, you could schedule it |
| 17:52.41 | Katty | Andares: if that's what you mean |
| 17:52.59 | Andares | Katty: basically I have a program on some computers that has an option for dialing a number inside it (a stupid enterprise software thing) |
| 17:53.00 | unpaidbill | is there something different about configuring a T1 card for pri_cpe and national signaling rather than e&m wink as far as echo is concerned? |
| 17:53.05 | kb1_kanobe | DSpair: fwiw, not all telcos do the disconnect. Eg. in my area calls between Telus endpoints don't care about answer state (they can be half-up all the time and just never generate a bill), but others, such as Rogers, enforce answer timers on all inter-carrier calls. |
| 17:53.13 | Andares | Anyway, you can make it launch an external dialing app with the phone number as an argument. |
| 17:53.15 | unpaidbill | i'm noticing echo and i never did before, just switched from e&m to pri |
| 17:53.30 | Andares | So I wanted to write a program that would tell asterisk to dial that number out, and hook the call up to the phone next to the user's computer. |
| 17:53.39 | Andares | anddd I'm way over my head. :( |
| 17:53.39 | [TK]D-Fender | jameswf-home: lol... NO |
| 17:54.06 | DSpair | kb1_kanobe, The first thing I am going to try is to use an IAX2 soft-phone instead of a SIP handset. |
| 17:54.06 | Katty | well you could take that arguement and make a text file out of it |
| 17:54.14 | jameswf-home | |||Mad|||: this strikes me as freepbx |
| 17:54.18 | jameswf-home | ~freepbx |
| 17:54.19 | 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 |
| 17:54.25 | Andares | Katty: then send that to outgoing? |
| 17:54.33 | |||Mad||| | It is Asterisk |
| 17:54.37 | Katty | yeah, but i don't know how you'd connect it to the sending party |
| 17:54.39 | DSpair | kb1_kanobe, That way I can enable debugging on IAX and watch the complete call |
| 17:54.39 | kb1_kanobe | DSpair: you could issue a 'pri debug' on the console and trace the disconnect there. That will tell you quickly if you're issuing the hangup or the CO is. |
| 17:55.00 | Katty | Andares: unless you can send the number as an arguement too |
| 17:55.03 | Katty | Andares: the extension, i mean |
| 17:55.05 | [TK]D-Fender | jameswf-home: And no, that clearly ISN'T FreePBX. |
| 17:55.07 | jameswf-home | |||Mad|||: did you write the dialplans with freepbx naming conventions? |
| 17:55.08 | Katty | Andares: that'd get so messy. |
| 17:55.13 | |||Mad||| | Well, the macro-voicemail script came from the Asterisk book |
| 17:55.13 | DSpair | kb1_kanobe, That would generate more output than I could easily parse. We have 46 PRI channels and our fax server uses them heavily. |
| 17:55.28 | [TK]D-Fender | |||Mad|||: next smart thing you can do : LOOK at the variable before you GOTOIF based on it <- |
| 17:55.33 | *** part/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 17:55.43 | jameswf-home | |||Mad|||: well it seems something is missing contextualy |
| 17:55.44 | kb1_kanobe | DSpair: enable log-to-disk? still better to look at the CO facing interface first, then track backwards. :-) |
| 17:55.59 | [TK]D-Fender | jameswf-home: He's got variable issues he needs to loko at. |
| 17:56.02 | DSpair | kb1_kanobe, How would I enable log-to-disk? |
| 17:56.08 | kb1_kanobe | rummages |
| 17:56.12 | mrnick | jameswf-home: are you sure about the example you gave me, it doesn't seem to work |
| 17:56.20 | [TK]D-Fender | mrnick: Its wrong |
| 17:56.20 | Katty | Andares: cause all the call file does is make the server go to a particular part of the context. |
| 17:56.21 | Andares | Katty: hm. It does sound messy. :| But is there any way I can get my outgoing call to show up on a particular sip? |
| 17:56.26 | mrnick | thought so |
| 17:56.31 | kb1_kanobe | DSpair: see /etc/asterisk/logger.conf |
| 17:56.38 | |||Mad||| | D-Fender... OK, I will do some more digging, thank you very much |
| 17:56.49 | jameswf-home | |||Mad|||: if [TK]D-Fender is on the right path I would NOOP the hell out of your dialplan everywhere you use ${} |
| 17:56.57 | [TK]D-Fender | |||Mad|||: NoOp(It gave me ${DIALSTATUS}) <-------- |
| 17:56.58 | Katty | Andares: how many extensions do you have to do this for? |
| 17:57.03 | DSpair | kb1_kanobe, Can changes to that be done without a restart? Using a reload is OK. |
| 17:57.05 | Andares | Katty: only about 4. |
| 17:57.08 | Katty | Andares: oh. |
| 17:57.11 | Katty | Andares: well... |
| 17:57.15 | [TK]D-Fender | jameswf-home: :p |
| 17:57.16 | Katty | yeah, you could. |
| 17:57.27 | Katty | Andares: can you get the extensions as a variable too? |
| 17:57.36 | Andares | Would I have to make a call file for each one? :( |
| 17:57.40 | Katty | pretty much |
| 17:57.43 | kb1_kanobe | unpaidbill: signalling e&m vs. PRI shouldn't, in and of itself, affect echo unless the switch at the CO handles the channels differently. |
| 17:57.47 | Katty | they're'd be 4 template files |
| 17:57.48 | Andares | Katty: nope, but I can make a dictionary of ip address -> extension # |
| 17:57.56 | Katty | well that might work |
| 17:58.01 | mrnick | so anybody knows the answer, how to continue a dialplan after the caller hangs up |
| 17:58.04 | Katty | and depending on which ip came in.. |
| 17:58.05 | |||Mad||| | D-Fender: Thanks, I haven't used NOOP yet, will do that.... thank you too jameswf |
| 17:58.06 | Andares | And run that as a service on the pbx server. |
| 17:58.16 | Katty | you'd have 4 call files. |
| 17:58.20 | Katty | and each would go to a different context. |
| 17:58.27 | [TK]D-Fender | mrnick: exten => h,1,DoStuff |
| 17:58.27 | Blackvel | wow cool |
| 17:58.35 | jameswf-home | [TK]D-Fender: why was "h" wrong? |
| 17:58.45 | kb1_kanobe | DSpair: I think so, yes but I'm not 100% certain |
| 17:58.46 | [TK]D-Fender | jameswf-home: You tried treating it as a PRIORITY <- |
| 17:58.46 | Blackvel | now it works to have two DIAL commands. i just have to call at least Wait(2) in between them |
| 17:58.48 | Andares | Katty: can I somehow send the phone number to call to the call file? |
| 17:58.49 | jameswf-home | wtf what did i put |
| 17:58.50 | Katty | Andares: ithink you'd have to put it into a conference room |
| 17:59.07 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:59.08 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:59.10 | jameswf-home | oh dislexia |
| 17:59.12 | kb1_kanobe | DSpair: Almost everything reconfigures on reload. |
| 17:59.13 | jameswf-home | damnit |
| 17:59.15 | Blackvel | if I don't do ZAPTEL devices do not reset hangup/available status fast enough. now it works even with old priority +1 concept |
| 17:59.19 | Katty | Andares: you could make the file each tmie using an echo thing i guess |
| 17:59.26 | [TK]D-Fender | jameswf-home: Dog Named it! |
| 17:59.46 | implicit | 9 |
| 17:59.55 | edwin_quijada | [TK]D-Fender: I get an error form zaptel.conf with this value e&mw |
| 17:59.56 | Katty | Andares: echo, mv, del, everytime. |
| 18:00.09 | [TK]D-Fender | edwin_quijada: Read the link I gave you AGAIN |
| 18:00.33 | mrnick | doesn't work either although I though that one was more plausible |
| 18:00.39 | Katty | Andares: there's probably a better way to do it with the manager api, but i dunno how you'd do it |
| 18:01.22 | Andares | Katty: hm okay. Thanks :p |
| 18:01.33 | DSpair | kb1_kanobe, Can you help me to decode this? http://www.pastebin.org/58240 |
| 18:01.36 | Andares | Katty: do I need to reload anything to make it see the call file? |
| 18:01.41 | Andares | Or to run the call file? |
| 18:01.44 | Katty | Andares: if you could create a text file template from your software... |
| 18:01.59 | Katty | Andares: you could do it with meetme |
| 18:02.08 | |||Mad||| | D-Fender and jamessw: D'oh!! It was a stupid missing { !!! Thanks again for your help!!! |
| 18:02.27 | Andares | Sounds complicated. D: |
| 18:02.34 | Katty | Andares: well you have to add in to extensions conf where your call file is executing it's stuff at. |
| 18:02.38 | Katty | Andares: that will require you to reload. |
| 18:02.50 | Katty | Andares: but just dumping a call file into the outgoing folder you don't need to reload it each time |
| 18:03.02 | Katty | Andares: it executes anything in out going automatically |
| 18:03.11 | edwin_quijada | I read thks, but I say in zaptel.conf is e&m not zapata in zapata is em_w , BTW: How can i put asterisk in debug mode? |
| 18:03.26 | kb1_kanobe | DSpair: That is the final message in the hangup sequence - the confirmation that the B channel has been released for call with ID 273 |
| 18:03.36 | Andares | Katty: so it executes it the minute it's put into outgoing? :o |
| 18:03.37 | kb1_kanobe | rummages |
| 18:03.51 | Katty | Andares: pretty much |
| 18:04.22 | Katty | Andares: the call file just says GO HERE in the extensions.conf |
| 18:04.28 | Katty | Andares: go HERE and NOW |
| 18:04.43 | Andares | Katty: and the call file specifies stuff like number to dial, what phone to show it on etc? |
| 18:05.02 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-182-169.dsl.tul2ok.sbcglobal.net) |
| 18:05.10 | DSpair | kb1_kanobe, Hrrrmmmm . . . No DTMF was detected when I ran a debug on the channel. |
| 18:05.13 | kb1_kanobe | DSpair: I wrote something up for someone a while back on this. let me find it. |
| 18:05.23 | DSpair | kb1_kanobe, Thanks much!! |
| 18:05.26 | Katty | Andares: no. |
| 18:05.36 | Katty | Andares: look at the post again. |
| 18:05.44 | jameswf-home | I like originate zap/$random1 application dial zap/$random2 |
| 18:06.26 | Andares | Ahhhh cool. Katty, I found a good article describing how to do what I want with the manager API. |
| 18:06.31 | Andares | http://www.voip-info.org/wiki/view/Asterisk+manager+dialout |
| 18:06.33 | Andares | :D |
| 18:06.44 | kb1_kanobe | DSpair: This thread: http://www.tek-tips.com/viewthread.cfm?qid=1455378 should help with debugging. |
| 18:06.50 | Katty | Andares: good. much easier that way i'm sure. |
| 18:06.51 | Andares | Thanks. |
| 18:07.02 | Andares | Yeah, now all I have to do is write an asterisk client. :p |
| 18:07.09 | DSpair | kb1_kanobe, I'll take a look right now... Thanks much!!! |
| 18:07.10 | mrnick | i can't use the goto with the h priority |
| 18:07.18 | kb1_kanobe | DSpair: You're welcome |
| 18:08.17 | mrnick | could it be because of festival because the debug says: /asterisk/festival.conf': Found |
| 18:08.17 | mrnick | [Jul 29 20:05:39] WARNING[7947]: app_festival.c:217 send_waveform_to_channel: Null frame == hangup() detected |
| 18:10.20 | kb1_kanobe | DSpair: another quick tip: each d channel message contains a line like 'Call Ref: len= 2 (reference 273/0x111)'. That reference number will be the same for all messages related to a particular call over its lifetime. |
| 18:10.48 | DSpair | OK, that's very helpful. |
| 18:11.32 | kb1_kanobe | DSpair: so, if I'm reading this right and that's the span that faces the CO then the message in the pastebin is your box confirming the call has been taken down at the request of the CO. |
| 18:11.46 | Andares | Hey Katty, is there a way I can reload manager.conf without restarting? I'm kind of nervous about restarting the phone server as it's pretty unstable. :( |
| 18:11.49 | kb1_kanobe | DSpair: look for the preceeding RELEASE message |
| 18:12.22 | DSpair | kb1_kanobe, Here's a complete debug from start to finish: http://www.pastebin.org/58246 |
| 18:12.25 | *** join/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com) |
| 18:12.27 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 18:12.36 | Katty | Andares: module reload manager |
| 18:13.04 | mrnick | is it a festival problem, anybody knows? |
| 18:13.10 | DSpair | kb1_kanobe, Looks to me like the Originator (Me) is sending a "DISCONNECT". |
| 18:13.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:14.20 | kb1_kanobe | DSpair: so pulling out the Message Type fields, >SETUP <STATUS <CALL PROCEEDING <PROGRESS >DISCONNECT <RELEASE >RELEASE COMPLETE |
| 18:14.37 | kb1_kanobe | DSpair: What sequence do you get if you call a known good number? |
| 18:14.50 | Andares | Katty: thanks. |
| 18:14.53 | DSpair | kb1_kanobe, I'll let you know in a moment. |
| 18:16.55 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 18:16.58 | Andares | mrnick: you got asterisk to work with festival? :o |
| 18:17.31 | mrnick | yep |
| 18:17.41 | kb1_kanobe | DSpair: btw, the <STATUS 'Info. element nonexist or not implemented' is likely the CO complaining that you sent it your calling name in the 'Display' element of the >SETUP message and they don't accept that data. should be safe to ignore. |
| 18:17.46 | Andares | does it sound like a robot with a hoarse throat, mrnick? |
| 18:17.59 | LemensTS | ~festival |
| 18:18.00 | jbot | extra, extra, read all about it, festival is a general multi-lingual speech synthesis system developed at CSTR. See http://www.cstr.ed.ac.uk/projects/festival/, or festival lite a much more compact festival http://www.speech.cs.cmu.edu/flite/index.html |
| 18:18.06 | mrnick | I don't care i just use it for debugging |
| 18:18.10 | Andares | heh, k. |
| 18:18.24 | mrnick | and it's not my native langue |
| 18:18.30 | mrnick | language |
| 18:18.31 | DSpair | kb1_kanobe, I tried a test to my cell, and hanging up on the cell gives me a <DISCONNECT. Hanging up from the SIP handset gives me a >DISCONNECT |
| 18:18.50 | DSpair | So, it looks like the Asterisk end is hanging up on the call. |
| 18:19.05 | Blackvel | what best way is to create a busy tone for the caller when asterisk noticed the ringing device as busy? 102,Busy? BUSY Cmd doesn't seem to create the busy tone? or is CONGESTED the right one? |
| 18:19.09 | kb1_kanobe | DSpair: one sec. |
| 18:19.41 | mrnick | but to answer your question, yes |
| 18:19.45 | *** join/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 18:21.23 | pabelanger | anybody had anyluck with wct4xxp using DAHDI? |
| 18:21.55 | pabelanger | Board: TE210P, OS: Ubuntu 8.04 |
| 18:22.10 | DSpair | kb1_kanobe, BRB, need a cigarette... |
| 18:22.27 | kb1_kanobe | DSpair: what protocol is the PRI? |
| 18:27.20 | Hydrant | can I receive SMS messages via Asterisk ? |
| 18:27.25 | DSpair | kb1_kanobe, b8zs esf |
| 18:28.01 | mrnick | the h extension doesn't work |
| 18:28.06 | DSpair | Wow, can't believe I remembered that. |
| 18:28.52 | kb1_kanobe | DSpair: that's the underlying framing setup (well remembered!) |
| 18:29.02 | mrnick | anybody got a better idea, i've been searching for a long time, can anybody please help me? |
| 18:29.07 | kb1_kanobe | DSpair: but what PRI variant? eg. national, NI2 etc. |
| 18:29.18 | Andares | Hey, I'm having issues with the manager API. |
| 18:29.28 | Andares | I enabled it and connect to the right port. |
| 18:29.34 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 18:29.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:29.36 | DSpair | kb1_kanobe, defaultzone=us, I believe that makes it national... |
| 18:29.41 | Katty | hai mog |
| 18:29.53 | Andares | Then I do "Action: Login\r\n ActionID: 1\r\nUsername: <username>\r\n Secret: <password>\r |
| 18:29.56 | DSpair | Yes, national. |
| 18:29.57 | Andares | *\n\r\n" |
| 18:30.05 | Andares | But it says "Message: Missing action in request." |
| 18:30.08 | DSpair | kb1_kanobe, switchtype=national |
| 18:30.11 | Andares | What am I doing wronG? |
| 18:30.14 | Andares | *wrong |
| 18:31.02 | Katty | i'm bored. |
| 18:31.05 | *** join/#asterisk foexle (n=heiko@88-134-242-24-dynip.superkabel.de) |
| 18:31.09 | Katty | [TK]D-Fender: where did you go? |
| 18:31.10 | jpcansa | any idea why i get SIP/2.0 603 Declined |
| 18:31.23 | [TK]D-Fender | Katty: On camera research right now... |
| 18:31.27 | mog | hi Katty |
| 18:31.30 | Katty | [TK]D-Fender: ah. |
| 18:31.32 | jpcansa | http://pastebin.com/m2013b6e9 |
| 18:31.38 | [TK]D-Fender | jpcansa: You're tilted to far forward |
| 18:32.00 | Katty | [TK]D-Fender: you're just tilted in general. |
| 18:32.05 | jpcansa | Fender: what do you mean? |
| 18:32.42 | Katty | wanders off. |
| 18:32.53 | kb1_kanobe | DSpair: alright, I'm running NI-2 so the messaging isn't entirely the same. Suffice it to say my calls go >SETUP <STATUS <CALL PROCEEDING <ALERTING {calling party hears ringing in band} {called party answers} <CONNECT >CONNECT ACKNOWLEDGE {both parties chat} {called party hangs up} <DISCONNECT >RELEASE <RELEASE COMPLETE |
| 18:33.12 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 18:33.45 | DSpair | I'm going to have to consider writing an AMI/AGI application to allow for tracing call information. |
| 18:34.35 | DSpair | kb1_kanobe, Either way, I'm seeing >DISONNECT, which indicates my end is doing the hang up. |
| 18:34.39 | kb1_kanobe | DSpair: yeah, parsing out the logs on a busy server is an ugly thing. It would be nice if there was glue to allow wireshark to read the call trace logs. |
| 18:35.13 | *** part/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 18:35.21 | kb1_kanobe | DSpair: Sorry, yes. Back on topic: you're requesting the disconnect by the looks of it. It usually indicates the call isn't fully 'up' somewhere and times out. |
| 18:35.35 | kb1_kanobe | try the different client and see what happens :-) |
| 18:35.47 | DSpair | kb1_kanobe, K, will do. |
| 18:35.59 | kb1_kanobe | DSpair: gotta run - good luck. |
| 18:36.38 | *** part/#asterisk kb1_kanobe (n=kris@bdr2.fieldrd.scrd.ca) |
| 18:38.27 | *** join/#asterisk jadeisfalling (n=ruinous@66.194.15.236) |
| 18:40.49 | Blackvel | how can i send a busy tone to the caller when all my ZAP channels are busy/congested? |
| 18:41.04 | Blackvel | it's only ringing no matter of busy :( |
| 18:46.50 | [TK]D-Fender | Blackvel: PASTEBIN is your friend. |
| 18:48.06 | DSpair | OK, I've got a lead here. When I call this automated call service, it doesn't ring even once before it picks up. That means the channel state never reaches "UP". How can I account for this type of situation? |
| 18:48.15 | *** join/#asterisk Bananaskin (n=Bananask@78-105-249-152.zone3.bethere.co.uk) |
| 18:48.36 | *** join/#asterisk PeterFA (n=Peter@unaffiliated/peterfa) |
| 18:48.36 | jameswf-home | chacha |
| 18:48.42 | PeterFA | I'm going to install this software ^^; |
| 18:48.46 | PeterFA | Not for a few days though. |
| 18:48.54 | b11d` | how many .call files can asterisk handle at a time? Say I have 200 .call files, and I drop them all into the appropriate place for asterisk to pick them up, will Asterisk choke on that? |
| 18:49.26 | Qwell | b11d`: try it |
| 18:49.49 | b11d` | i dont want to call all those phones right now :) |
| 18:50.00 | mrnick | this is not working, anybody got an idea why? h,1,ChannelRedirect(${CHANNEL}|SAVEDB|s|1) |
| 18:50.02 | b11d` | i guess you're right though, its the only way to find out |
| 18:50.25 | marc7 | b11d`: we have a system queue 20 calls in the spool directory at once |
| 18:50.38 | marc7 | as calls complete, files disappear, more .call files get moved into the directory |
| 18:51.03 | eaglexl | People can finally connect from the outside, but when I try to call somebody - Yate says noroute, and the CLI: http://pastebin.com/m2d286810 |
| 18:51.10 | b11d` | basically i need to call all of my internal extensions at the same time.. |
| 18:51.19 | b11d` | and play a message.. |
| 18:51.38 | b11d` | using call files seemed appropriate |
| 18:51.39 | marc7 | there's no transcoding, no network limitations, really you should be able to call everybody given a fast enough system |
| 18:51.45 | jaytee | arrrrgh!!! my telco is trying to point a problem where a single specific 800 number won't answer back at the configuration on my end with our PRI circuit. and he's quoting from BellCore specs from 1998 on ISDN for connecting the B-channel when the CPE recieves a PROGress message with progress indicator#1 "call is not end-to-end ISDN; further call progress infromation may be available inband" |
| 18:52.02 | b11d` | no transcoding.. its 100mbit.. dual cpu 2.8ghz box w/ 2gb of RAM.. |
| 18:52.11 | b11d` | i guess i'll find out :0 |
| 18:52.13 | b11d` | :) |
| 18:52.17 | marc7 | best bet |
| 18:52.24 | marc7 | start with 20 or 50 |
| 18:52.34 | marc7 | have it call you first, and listen to the audio quality |
| 18:52.46 | b11d` | sure.. |
| 18:52.52 | marc7 | build the recording dialplan so that you can press 1 to replay the message |
| 18:52.58 | marc7 | keep replaying the message as you feed more .call files in |
| 18:53.02 | Hydrant | can someone give me the link to the asterisk book ? |
| 18:53.06 | Qwell | ~book |
| 18:53.06 | 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 |
| 18:53.23 | marc7 | as long as you don't hear any call quality problems, you win! |
| 18:53.43 | Hydrant | thx |
| 18:53.43 | [TK]D-Fender | eaglexl: Looking for 102 in home (domain 10.0.0.2) SIP/2.0 404 Not Found <-- Dialplan error. You don't have a match for 102 in [home] |
| 18:54.06 | b11d` | haha |
| 18:54.51 | [TK]D-Fender | b11d`: what kind of message? Fixed recording? Something like a school-bell ringer? Pre-recorded "page"? |
| 18:54.56 | eaglexl | fender, you are right yet again, thanks :) |
| 18:55.41 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:56.42 | DSpair | Does anyone have a suggestion for fixing call states when the distant end doesn't ring before answering? |
| 18:57.21 | russellb | DSpair: i suggest a hammer |
| 18:58.00 | Blackvel | i was just working with the busy cmd in the dailplan (found Playtones(busy) on voip-info.org). oh well looks like setVar(PRI_CAUSE=17) followed by HANGUP does it's job. looks like to be my pots pbx which won't handle the correct busy tone. |
| 18:59.49 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:05.06 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 19:09.16 | *** join/#asterisk gjw86 (n=gwong@69.43.72.131) |
| 19:09.26 | mrnick | what i need to do is connect a called phone to an ivr after having a chat with that person, i proposed several solutions but never got a working answer, is it impossible with *? |
| 19:09.55 | gjw86 | Hey everyone. Is anyone familar integrating Asterisk and Cisco Call Manager Express? I am trying to use Asterisk for IVR and Voicemail functions but am running into some issues. |
| 19:10.01 | mrnick | transfer, channelredirect, h extension, ...? |
| 19:10.14 | jameswf-home | Anyone seen cuil.com ... I am not sure how it happened as I thought working at google meant signing a NDA from hell |
| 19:10.26 | jameswf-home | and I also imagine a non compete |
| 19:12.58 | *** join/#asterisk ylem1925 (n=xyz@121.97.202.155) |
| 19:13.25 | *** join/#asterisk ener_uno (n=renet@121.97.202.151) |
| 19:15.09 | Blackvel | [TK]D-Fender: http://pastebin.com/d44bdc53b. it shows non-working busy with improved hangup with busy tone on pstn. do I even need prio 4 congestion and 5 hangup after dial (I read about Playtones(congestion) before sending congestion) |
| 19:17.16 | ylem1925 | hi everyone |
| 19:17.19 | ylem1925 | i'm new here |
| 19:17.39 | [TK]D-Fender | Blackvel: Congestion will sit there FOREVER. You can tell it a TIMEOUT as a parameter however |
| 19:19.47 | Blackvel | i mean i am reading thru the voip-info docs and trying to understand what I knew in 2005-2006 :) but.... doesn't DIAL just STOP when the parties hangup up? I mean no n+1 gets executed after dial? so congestion, hangup won't be called after dial? and if dial gets busy it even jumps to 102 and not 4 |
| 19:20.38 | Blackvel | probably I should just remove it? |
| 19:24.03 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 19:24.35 | [TK]D-Fender | Blackvel: after a call it goes to exten "h" (hangup). If you specify the "g" parameter to Dial, then it will CONTINUE onwards if the CALLEE hangs up. In all other cases it goes to "h" |
| 19:24.49 | [TK]D-Fender | Blackvel: And that +101 BS is HISTORY |
| 19:27.14 | exothermc | [TK]D-Fender: is there anyway to have a call go to exten "h" from a queue, or does it do that by default? |
| 19:27.50 | [TK]D-Fender | exothermc: It will. |
| 19:28.06 | exothermc | [TK]D-Fender: cool thanks |
| 19:28.24 | *** part/#asterisk mrnick (n=ubugo@88.197.232.204) |
| 19:32.21 | DSpair | How do I set up a group of DIDs to ring the next available IAX2 peer in a group? |
| 19:35.21 | Blackvel | [TK]D-Fender: i read about that :) but my current local asterisk system is really old. probably I am going to upgrade (if time). AHHHH! i have this h exten. NOW I understand. much thanks |
| 19:35.56 | [TK]D-Fender | DSpair: what makes them a "group"? |
| 19:37.07 | DSpair | [TK]D-Fender, Hmmm . . . Well, basically, I want any inbound DID ending in 2XX to go to a bank of IAX modems. The first one available should pick up |
| 19:37.28 | [TK]D-Fender | DSpair: just dial them back to back |
| 19:37.50 | DSpair | Let me pastebin what I have. |
| 19:38.22 | [TK]D-Fender | DSpair: jsut do Dial(IAX2/modem1), Dial(IAX2/modem2), Dial(IAX2/modem3), etc back to back |
| 19:38.58 | Blackvel | anyone using bristuff? |
| 19:40.14 | Blackvel | has something much improved in quality with newest driver versions and asterisk 1.4 (not features) with a BRI in NT mode? |
| 19:40.16 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:40.56 | Blackvel | I am kinda satisfied. any reason to work on upgrade? :) |
| 19:43.33 | DSpair | [TK]D-Fender, It's still hit and miss. Every 4-5 tries it answers, and other times it's RNA. |
| 19:44.01 | [TK]D-Fender | DSpair: pastebin.... |
| 19:44.51 | DSpair | [TK]D-Fender, http://www.pastebin.org/58272 - And thanks for the help. |
| 19:45.35 | [TK]D-Fender | DSpair: show me the call |
| 19:46.01 | DSpair | [TK]D-Fender, Debug the channel or debug iax? |
| 19:46.14 | tzafrir_laptop | Blackvel, the driver (any specific driver?) or libpri/chan_zap? |
| 19:46.37 | *** join/#asterisk kschreyack (n=kschreya@mail.wirelesscapital.com) |
| 19:46.38 | [TK]D-Fender | DSpair: basic CLI at verbose 10 |
| 19:46.54 | DSpair | [TK]D-Fender, Too many calls going through for that to be useful. |
| 19:47.13 | [TK]D-Fender | DSpair: If you can't isolate it there isn't much I can say |
| 19:47.35 | DSpair | [TK]D-Fender, When I debugged the channel, I see " chan_iax2.c:7736 socket_process: Call rejected by 127.0.0.1: Busy" |
| 19:48.07 | [TK]D-Fender | DSpair: then it should fall to the next |
| 19:48.26 | kschreyack | Is there a way to get phone service on Asterisk without POTS or a PRI line? If so, who provides the service? |
| 19:48.50 | DSpair | kschreyack, Yes, there are many SIP and IAX2 providers. |
| 19:48.50 | Qwell | ~itsp |
| 19:48.51 | 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 |
| 19:49.28 | jameswf-home | I think I am going to throw up 9 tacos BAD idea |
| 19:49.31 | DSpair | [TK]D-Fender, Thanks, I think that I found it. |
| 19:49.40 | pkunkra | ~itsplist-us |
| 19:49.41 | 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 |
| 19:49.59 | pkunkra | i use teliax, not too bad. |
| 19:50.11 | DSpair | I've used teliax with success as well. |
| 19:50.12 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 19:50.37 | pkunkra | yeah, never had any complaints with them at all |
| 19:50.44 | [TK]D-Fender | jameswf-home: Indeed. Throwing up more than 5 tacos might lead to dehydration. Try to minimize your regurgitation. |
| 19:50.46 | kschreyack | awesome, thanks guys. I'm thinking on running asterisk in a VM ;) |
| 19:51.23 | DSpair | kschreyack, Bad idea!!! |
| 19:51.40 | DSpair | kschreyack, I've had MANY timing problems with asterisk in a VM. |
| 19:52.06 | pkunkra | actually, i think that depends on the type of vm you're using |
| 19:52.26 | jameswf-home | virtual box rocks |
| 19:52.38 | DSpair | pkunkra, I don't think so... I've tried with VMware ESX, VMware Server, VirtualBox, and Virtuozzo. |
| 19:52.51 | kschreyack | DSpair: Timing problems? How loaded was your host machine? |
| 19:52.55 | pkunkra | i've been using * on openvz successfully for quite some time.... |
| 19:52.58 | DSpair | kschreyack, Not at all. |
| 19:53.00 | jameswf-home | DSpair: what was the host OS |
| 19:53.09 | DSpair | jameswf-home, Ubuntu |
| 19:53.12 | pkunkra | i can't use conferences though since i can't get a ztdummy driver in there. |
| 19:53.21 | eaglexl | How can I automaticly start asterisk in debian (/etc/init.d/asterisk) with sip set debug on? |
| 19:53.24 | pkunkra | i'm trying out a xen based asterisk now. |
| 19:53.46 | kschreyack | Xen at least will get you a 20% performance boost over Vmware |
| 19:54.07 | kschreyack | I run vmware on a quad core, dual processor setup with 16GB of memory |
| 19:54.28 | kschreyack | I guess I'll find out in the end if I'll have issues |
| 19:54.48 | kschreyack | if so, I can always setup a box - I've got lots of spares |
| 19:55.19 | kschreyack | Does anyone else have Virtualization of Asterisk stories to share? |
| 19:55.23 | Blackvel | tzafrir_laptop: heya, well more in general. this old bristuffed asterisk 1.0 seems working with old zaphfc drivers (chan_zap/libpri). if going for asterisk/bristuff 1.3 or 1.4 I expect it can't be much better than "just working"? :) maybe it's more stable, better audio quality and stuff like that? why one should upgrade an almost 99% working system (besides trying and having fun, if time)? :) |
| 19:55.41 | pkunkra | kschreyack, not any interesting ones. |
| 19:55.42 | pkunkra | hmmm |
| 19:56.05 | pkunkra | bump up your bean counters *alot* on any virtuozzo/openvz based one... |
| 19:56.07 | pkunkra | that's about it. |
| 19:56.34 | pkunkra | i do have asterisk stories though. but those are related to jitter in multipath networks |
| 19:56.57 | tzafrir_laptop | Blackvel, bristuff 0.2: asterisk/zaptel 1.0 . bristuff 0.3: asterisk/zaptel 1.2 . bristuff 0.4: asterisk/zaptel 1.4 |
| 19:57.14 | basement_troll | I was able to run asterisk in ESX |
| 19:57.19 | tzafrir_laptop | asterisk has greatly changed between 1.0 and 1.4 |
| 19:57.26 | basement_troll | Now that ESXi is free, you can probably try that kschreyack |
| 19:57.29 | tzafrir_laptop | I wouldn't run asterisk 1.0 today |
| 19:57.42 | basement_troll | but it didn't work in vmware free |
| 19:57.55 | basement_troll | The voice was horrible |
| 19:58.06 | kschreyack | bummer |
| 19:58.18 | kschreyack | I run Vmware Server 1.0.6 |
| 19:58.32 | basement_troll | I haven't tried in ESXi yet since it hasn't been free that long |
| 19:58.44 | kschreyack | its still worth my time to give it a trial run |
| 19:58.56 | kschreyack | hmm, I didn't even know they offered a free ESX! |
| 19:59.38 | Qwell | kschreyack: only as of like...yesterday |
| 19:59.40 | basement_troll | I think they did it to compete with hyper-v |
| 19:59.45 | tzafrir_laptop | Blackvel, and the driver of zaphfc has improved over time |
| 19:59.57 | Blackvel | tzafrir_laptop: what you say it's working to upgrade on a kinda production system just for zaphfc/libpri patches? |
| 20:00.01 | Blackvel | hmm i see |
| 20:00.04 | tzafrir_laptop | Specifically, latest versions of the bristuff patch apply the florz patch |
| 20:00.14 | Blackvel | probably I just just give it a try |
| 20:00.16 | tzafrir_laptop | (but strangely enough: still as a separate patch) |
| 20:00.24 | Blackvel | found something about that on voipinfo |
| 20:00.30 | Blackvel | do you use this florz patch? |
| 20:00.49 | tzafrir_laptop | I don't have a HFC-S card |
| 20:00.59 | pkunkra | whoa. didn't know about ESXi... i can really use that! |
| 20:00.59 | Blackvel | well hfc and nt mode is just for playing around in non-commercial systems :) |
| 20:01.10 | Blackvel | tzafrir_laptop: do you use any bristuff? |
| 20:01.38 | Blackvel | how much has asterisk software improved from 1.0 to 1.4? only features or also quality? |
| 20:01.49 | tzafrir_laptop | both |
| 20:02.21 | Blackvel | maybe I should just give it a try on weekends and do a backup before :) |
| 20:02.37 | Blackvel | I cant believe how long I haven't been here :) |
| 20:02.58 | tzafrir_laptop | Blackvel, see also http://bristuff.org/ |
| 20:03.15 | Blackvel | saw the link today |
| 20:03.58 | DSpair | [TK]D-Fender, Would a Queue work better on those fax modems? |
| 20:04.21 | [TK]D-Fender | DSpair: yuck.... |
| 20:04.34 | [TK]D-Fender | DSpair: more channel interfaces to pass through... |
| 20:04.39 | DSpair | [TK]D-Fender, So, I take that as a NO. |
| 20:04.42 | Andares | Hey, how do I figure out what channel I should use for outgoing calls? |
| 20:04.43 | DSpair | :) |
| 20:05.05 | DSpair | :q |
| 20:06.21 | Andares | Like I see some things where they use like 6159bfb47b9 for channel number. |
| 20:07.08 | [TK]D-Fender | Andares: ...huh? |
| 20:07.37 | bkruse | [TK]D-Fender: wow.. heh :] |
| 20:07.44 | [TK]D-Fender | Andares: What means of calling out have yuo set up? |
| 20:07.55 | [TK]D-Fender | bkruse: "wow" on what? |
| 20:08.29 | DSpair | So, does anyone have an idea how to fix call status for a remote end which does not ring before it answers? I still cannot get this AT&T autoattendant to stop hanging up. |
| 20:08.35 | Andares | [TK]D-Fender: I have a couple zaptel modems set up. |
| 20:08.36 | bkruse | [TK]D-Fender: @ trying to use a channel with a unique ID tod ial out |
| 20:09.19 | Levonk | anybody have any opinion on using a Linksys 3102 for a home PBX FXO interface? Any drawbacks I should be aware of? |
| 20:09.22 | tzafrir_laptop | Andares, zaptel modem? X100P card? |
| 20:09.47 | Andares | tzafrir_laptop: not sure what brands the cards are. I think they're a type of winmodem or something though. |
| 20:09.54 | Andares | They're like regular modems actually. |
| 20:11.01 | tzafrir_laptop | right |
| 20:11.10 | Hydrant | where can I find a list of dialplan functions ? |
| 20:11.33 | bkruse | Hydrant: core show functions |
| 20:11.46 | tzafrir_laptop | Andares, you have more than 1? |
| 20:11.49 | [TK]D-Fender | bkruse: Now you lost me... |
| 20:11.50 | Hydrant | where do I do that? I meant more an online reference :-) |
| 20:12.17 | tzafrir_laptop | Generally: Zap/1/<number> to dial <number> through the first one |
| 20:12.39 | bkruse | [TK]D-Fender: nvm nvm, don't mind me :] |
| 20:13.04 | Andares | tzafrir_laptop: yeah, I have 2. |
| 20:13.07 | [TK]D-Fender | Levonk: Probably a decen choice. |
| 20:13.11 | Andares | Also, I'm getting an Originate failed error. :( |
| 20:13.14 | Andares | Trying to dial out. |
| 20:13.32 | [TK]D-Fender | Andares: Don't use Originate. setup a softphone or something at least. |
| 20:13.42 | Levonk | [TK]D-Fender: Any gotchas that you know I should be aware of (Thanks for responding BTW) |
| 20:14.10 | Levonk | seems like the cheapest FXO interface at ~$58+s/h |
| 20:14.14 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 20:14.18 | [TK]D-Fender | Levonk: You can pretty much certainly count out against using it for any kind of faxing. But thats most of it. Echo Cancellation can be spotty. YMMV |
| 20:14.28 | [TK]D-Fender | Levonk: But its a good value overall |
| 20:14.35 | tzafrir_laptop | Andares, what command did you use? |
| 20:15.45 | Levonk | [TK]D-Fender: it will have problems with the FXO interface, the FXS interface or both for faxing? |
| 20:16.25 | [TK]D-Fender | Levonk: jsut forget about faxing with it on EITHER |
| 20:16.46 | *** join/#asterisk nny_1 (n=Scott@64.203.244.146) |
| 20:16.50 | Andares | tzafrir_laptop: "Action: Originate\r\nChannel: SIP/1000-9293\r\nExten: 1000\r\nContext: test\r\nPriority:1\r\n\r\n" |
| 20:17.07 | nny_1 | anyone tried to set up a way for a park call returned to play a distinctive ring? |
| 20:17.41 | [TK]D-Fender | Andares: Channel: SIP/1000 <--- |
| 20:17.46 | Levonk | [TK]D-Fender: that's a bummer.... I don't fax often but some lame companies haven't gotten the concept of a web form yet. What's the only option to support faxing through asterisk the digium + echo cancellation cards, or would any card do like a zapmicro? |
| 20:18.04 | nny_1 | i figure the ring is a feature of the phone, so maybe a way to notify the reciever the call they are on is from the parking lot being returned cause of a timeout |
| 20:18.05 | [TK]D-Fender | Andares: no "-blah" at the end. |
| 20:18.15 | tzafrir_laptop | Andares, a simpler thing: use: originate from the CLI |
| 20:18.26 | Andares | [TK]D-Fender: thanks. |
| 20:18.31 | [TK]D-Fender | tzafrir_laptop: his channel format was wrong |
| 20:18.37 | Andares | tzafrir_laptop: but eventually this will be run from a network computer outside of the server. |
| 20:18.44 | Andares | So I think I have to use the manager. |
| 20:19.00 | [TK]D-Fender | Andares: only depends on how you want to trigger things |
| 20:19.32 | Andares | [TK]D-Fender: I'd prefer not to create an ssh session on the box and log into asterisk every time. :( |
| 20:19.42 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:19.54 | Andares | It'd be nice to tunnel it through ssh somehow, but I don't think it's really necessary. |
| 20:19.56 | nny_1 | kind of figuring this out with research (gasp) I can use parkinglot context to do *something* before ringing the return line |
| 20:21.20 | DSpair | tzafrir_laptop, Another question for you. I have an AT&T auto-attendant that I am dialing. It answers before even a single ring and as such the call never reaches a state of "UP" and disconnects after about 45 seconds. Any ideas on working around this? |
| 20:21.32 | unpaidbill | is there any small application, something that runs in the taskbar, that will pop up a little window that displays the CID for an incoming call for SIP phones with no LCD? |
| 20:21.43 | unpaidbill | i wrote one a while back using yahoo widgets but it was kind of crappy |
| 20:22.21 | unpaidbill | something like the samba method but all pretty looking |
| 20:22.24 | tzafrir_laptop | What channel is it? |
| 20:22.29 | tzafrir_laptop | sip? zap? |
| 20:22.37 | *** join/#asterisk ibnolqaiyem (n=ibnolqai@41.196.251.156) |
| 20:23.12 | [TK]D-Fender | unpaidbill: plenty of little apps out there for this. Go check the WIKI |
| 20:23.29 | DSpair | tzafrir_laptop, SIP->Asterisk->Zap |
| 20:23.31 | unpaidbill | yeah i'm browsing it now |
| 20:23.43 | unpaidbill | i was hoping there was one that was better than all the rest |
| 20:23.47 | unpaidbill | and then i wouldnt have to pick :P |
| 20:25.31 | Illarane | Hmm... What's the frequency range for your average telephone speaker? |
| 20:25.42 | Qwell | Illarane: not very high |
| 20:26.05 | Illarane | That much I know already. :p |
| 20:26.12 | DSpair | tzafrir_laptop, My signaling is national, and when I debug the channel I see a >DISCONNECT coming from my Asterisk box even though I never pressed a button or hung up. |
| 20:26.17 | [TK]D-Fender | ok, checkout time. Heading home.... |
| 20:26.39 | tzafrir_laptop | DSpair, what version of libpri? |
| 20:26.55 | DSpair | tzafrir_laptop, How do I tell? |
| 20:27.17 | tzafrir_laptop | When did you build it? |
| 20:27.36 | DSpair | tzafrir_laptop, Yesterday. |
| 20:27.43 | DSpair | Looks like it's 1.0 |
| 20:28.04 | DSpair | ldd on the asterisk binary returns "libpri.so.1.0 |
| 20:28.54 | *** join/#asterisk chandoo (n=chandra@ool-4353b4c7.dyn.optonline.net) |
| 20:29.44 | *** join/#asterisk antonyo14 (n=tony@206.135.21.162) |
| 20:29.58 | tzafrir_laptop | "1.0"? strange. It means libpri 1.4.0-1.4.3 . |
| 20:30.50 | DSpair | tzafrir_laptop, Is libpri included in zaptel or asterisk? |
| 20:30.51 | antonyo14 | hello all, I was wondering if anyone had any recommendations for a SIP phone with wireless headset? |
| 20:31.26 | DSpair | antonyo14, Many SIP handsets can handle wireless headsets. It mainly depends on what type of connector it uses. |
| 20:31.51 | DSpair | We use the PolyCom IP550, and it can handle several models of wireless headsets. |
| 20:32.15 | DSpair | antonyo14, Also, the sound quality on the IP-550 is EXCELLENT |
| 20:32.29 | antonyo14 | Well, we are using cisco 7940's and it looks to me that you have to attach some sort of hokey handset lift that I just dont want to bother with |
| 20:32.56 | antonyo14 | Dspair, thank you. |
| 20:33.17 | DSpair | antonyo14, From what I have seen, that is pretty standard. |
| 20:33.37 | jetlagmk2 | DSpair, have you used presense? |
| 20:33.51 | DSpair | jetlagmk2, Nope, I don't even know what it is. |
| 20:33.59 | antonyo14 | Dspair, i was kind of hoping there would be a phone that would auto switch between wired handset and wireless headset |
| 20:34.30 | antonyo14 | Are you using a lift with your IP550? |
| 20:34.58 | DSpair | antonyo14, Then I'm afraid I can't help you. I don't use wireless on my handset because of the expense. I just use an IAX soft-phone with a bluetooth headset. |
| 20:35.31 | antonyo14 | Thats a good idea. |
| 20:35.37 | tzafrir_laptop | DSpair, sorry, misunderstood what you wrote. I need to go now, but I guess you should paste a PRI debug trace |
| 20:35.44 | DSpair | antonyo14, It works well with both iaxcomm and zoiper. |
| 20:35.54 | jetlagmk2 | That's a great idea. |
| 20:35.56 | tzafrir_laptop | maybe along side with a standard trace |
| 20:36.13 | DSpair | tzafrir_laptop, K, I'll try to catch up with you tomorrow. |
| 20:36.19 | DSpair | tzafrir_laptop, Thanks again! |
| 20:36.24 | jetlagmk2 | buys bluetooth dongle |
| 20:37.50 | antonyo14 | DSpair, is there a reason for using IAX over SIP for the softphone? Or is that just the protocol you are working with. |
| 20:38.44 | DSpair | antonyo14, No real reason. |
| 20:38.57 | russellb | IAX is teh awesome |
| 20:39.18 | DSpair | Laters ya'll!!! |
| 20:39.20 | DSpair | exit |
| 20:42.15 | Andares | == Starting SIP/1000-b980 at default,7062547942,1 failed so falling back to exten 's' |
| 20:42.23 | Andares | Hey, can someone tell me why this is happening when I try to originate? :( |
| 20:42.34 | Qwell | antonyo14: There is an Aastra phone that include a DECT handset, I believe |
| 20:43.00 | antonyo14 | Qwell, thanks I will look into that |
| 20:43.01 | Qwell | not sure if that would be quite what you're looking for, but it seems to fit the requirements |
| 20:43.08 | Qwell | (of course, so would a Polycom with a lifter) |
| 20:43.15 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 20:47.38 | *** join/#asterisk hi365_m (n=hi365@213.151.61.251) |
| 20:47.56 | *** join/#asterisk hi365_m (n=hi365@213.151.61.251) |
| 20:49.23 | *** join/#asterisk spokra (n=spokra@host093-179-146.sea0.speakeasy.net) |
| 20:50.37 | jameswf-home | I hope to someday be cool enough to go to comicon in costume |
| 20:51.21 | legis | What good lite SIP softphones are out there? for linux |
| 20:52.09 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:53.37 | jblack | twinkle works reliably for me. |
| 20:54.03 | Blackvel | have a good evening :) |
| 20:54.08 | Blackvel | cu soon |
| 20:54.11 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 20:55.41 | *** join/#asterisk moy (n=moy@nat/ibm/x-647c7937c4fc5ddb) |
| 20:55.53 | Andares | Hey. |
| 20:56.02 | Andares | So, for originating calls with the manager API. |
| 20:56.17 | Andares | For providing the channel, how should I tell if it's busy? |
| 20:56.23 | legis | jblack: K, gonn check it out. |
| 20:57.44 | Andares | Like for channels I was using Zap/1/<number> |
| 20:57.56 | Andares | But what if Zap/1 is busy? |
| 20:58.45 | jaytee | jameswf-home, did you go to comic-con? I have a friend who went this year |
| 21:01.45 | *** join/#asterisk linuxstb_ (n=linuxstb@rockbox/developer/linuxstb) |
| 21:04.31 | *** part/#asterisk telenieko (i=marc@telenieko.com) |
| 21:05.21 | edwin_quijada | I cant load my T1 card I done everything and I get nothing just silence |
| 21:09.29 | edwin_quijada | I am giving up |
| 21:09.33 | edwin_quijada | http://pastebin.com/m7c274c55 |
| 21:10.24 | LemensTS | ~alochol |
| 21:10.30 | LemensTS | ~alcohol |
| 21:10.31 | jbot | i heard alcohol is the answer to most of lifes problems.., or a good excuse to alias rm='rm -i', or a cause of "rm -rf . / "'s, or the cause of, and solution to, all of lifes problems, or ask me about ambrosia |
| 21:10.58 | edwin_quijada | i need alcohol!! :( |
| 21:11.32 | hubguruJR | edwin_quijada: you need to put 'context=from-pstn' above 'channel' in zapata.conf |
| 21:12.04 | hubguruJR | never mind, it's already up there |
| 21:12.17 | LemensTS | edwin: whats the cli show when you try to cal |
| 21:12.23 | *** join/#asterisk linuxstb__ (n=linuxstb@rockbox/developer/linuxstb) |
| 21:12.25 | hubguruJR | what does your lsmod show |
| 21:12.39 | edwin_quijada | LemensTS: Nothing |
| 21:12.46 | hubguruJR | what about 'ztcfg -vvv' |
| 21:13.05 | LemensTS | zap show status at teh cli? |
| 21:13.25 | edwin_quijada | zap show status tell me that NO such command |
| 21:13.44 | hubguruJR | then zaptel is not loaded becuase of configuration issue |
| 21:13.58 | edwin_quijada | hubguruJR: let me see |
| 21:14.19 | hubguruJR | start asterisk with 'asterisk -vvvvdgc' and you will see why zatel is not loading |
| 21:15.19 | LemensTS | you can try 'load chan_zap.so' i think that shows you what things are wrong too |
| 21:15.55 | edwin_quijada | LemensTS: I tried but said me the same |
| 21:17.26 | LemensTS | did you try what hubguruJR said about asterisk -vvvvdgc |
| 21:17.31 | edwin_quijada | no |
| 21:17.35 | LemensTS | try it |
| 21:17.50 | edwin_quijada | LemensTS: hubguruJR: this is my lsmod http://pastebin.com/m698d4ab1 |
| 21:18.28 | nny_1 | tuning a pri install. I had the client transfer me to "311" (my millwatt app) and got a read out of: |
| 21:18.31 | nny_1 | #########################* Rx: 0 ( 0) Tx: 4716 ( 4716) |
| 21:18.40 | edwin_quijada | hubguruJR: here is my ztcfg -vvv |
| 21:18.42 | edwin_quijada | <PROTECTED> |
| 21:19.09 | edwin_quijada | LemensTS: stop asterisk and restart again with this options? |
| 21:19.19 | nny_1 | this is with no tx or rx gain |
| 21:19.37 | LemensTS | edwin i think that last pastebin was an old link |
| 21:20.05 | edwin_quijada | but it is my new config and ztcfg -vvv |
| 21:20.27 | edwin_quijada | as I said "I need alcohol if I live |
| 21:20.38 | *** join/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com) |
| 21:21.34 | hubguruJR | looks like you have the wcte11xp module loaded ok, but is this the correct module for your card? |
| 21:21.59 | nny_1 | wth |
| 21:22.01 | nny_1 | seen this a lot |
| 21:22.07 | hubguruJR | do a 'ztcfg -vvv' and send the resilts |
| 21:22.08 | nny_1 | getting -- Remote UNIX connectionbose 5 -- Remote UNIX connection disconnected oevr and over |
| 21:22.23 | nny_1 | how do i fix it is annoying as hell |
| 21:23.19 | nny_1 | nm styop wait start seems to have fixed it for now |
| 21:24.13 | edwin_quijada | hubguruJR:http://pastebin.com/m46902d94 |
| 21:24.34 | LemensTS | http://pastebin.com/m46902d94 |
| 21:24.39 | LemensTS | gotta put a space in thier |
| 21:25.51 | edwin_quijada | LemensTS: i dont understand |
| 21:26.06 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:26.18 | Hydrant | I'm following Example 2 from here: http://www.voip-info.org/wiki-Asterisk+cmd+DISA however I'm not sure how to use the right context I have as an argument to DISA() |
| 21:26.47 | LemensTS | edwan: i just mean between the link you pasted and hubguruJR: you cant click it like you posted it... |
| 21:27.51 | edwin_quijada | ok |
| 21:28.34 | *** join/#asterisk blq (i=Bl@dslb-088-067-043-100.pools.arcor-ip.net) |
| 21:28.42 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:28.47 | *** join/#asterisk blq (i=Bl@dslb-088-067-043-100.pools.arcor-ip.net) |
| 21:29.23 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:29.35 | hubguruJR | ok, that looks ok |
| 21:30.24 | LemensTS | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf maybe this will help you edwin |
| 21:32.04 | edwin_quijada | I read this document too |
| 21:34.17 | nny_1 | anyone have experience with eliminating echo on a digium pri card with the hardware module? |
| 21:34.44 | russellb | digium tech support does :) |
| 21:35.02 | nny_1 | yeah figures |
| 21:35.13 | Hydrant | I'm trying to understand DISA()... when it forwards to a context... it's an outbound rule or something? |
| 21:35.33 | nny_1 | Hydrant: it drops the caller in the context specified |
| 21:35.47 | nny_1 | Hydrant: so if you want them to dial out, you would put them in the context for that |
| 21:36.28 | Hydrant | hrrm... does it matter which file the context is in? |
| 21:37.33 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 21:38.26 | nny_1 | i am not sure i understand what you mean |
| 21:39.15 | [TK]D-Fender | Hydrant: what file do you think it refers to? |
| 21:39.17 | Hydrant | just that I'm defining this in the extensions_custom.conf file |
| 21:39.28 | [TK]D-Fender | oh God.... |
| 21:40.00 | hubguruJR | edwin_quijada: your going to have to start asterisk 'asterisk -vvvvdgc' to see what the error is when loading zaptel |
| 21:40.03 | Hydrant | [TK]D-Fender: ? |
| 21:40.09 | [TK]D-Fender | Hydrant: its ANY context in your dialplan. |
| 21:42.28 | [TK]D-Fender | Hydrant>hrrm... does it matter which file the context is in? <- Lets say that if you're talking about a file on a USB-key in your pocket then the answer would be "no". |
| 21:42.59 | Hydrant | I wasn't sure which files get included... if the dialplan is just a concatenation of all files there |
| 21:43.36 | [TK]D-Fender | Hydrant: the dialplan is extensions.con + extensions.ael and anything they link to. |
| 21:43.55 | edwin_quijada | hubguruJR: I start asterisk as u said but nothing about zaptel |
| 21:44.01 | [TK]D-Fender | Hydrant: Not even knowing your own config is a very bad sign about your state. |
| 21:44.23 | Hydrant | [TK]D-Fender: yup... I've used all the graphical stuff to do the config :-( |
| 21:44.28 | [TK]D-Fender | edwin_quijada: PASTEBIN. FFS always include a pastebin. |
| 21:44.36 | [TK]D-Fender | Hydrant: what "graphical stuff"? |
| 21:44.38 | Hydrant | I don't presume to be a telephony expert |
| 21:44.42 | hubguruJR | can you pastbin what the output was? |
| 21:44.44 | Hydrant | FreePBX |
| 21:44.47 | edwin_quijada | hubguruJR: I didnt see any error |
| 21:45.10 | edwin_quijada | [TK]D-Fender: about what? zaptel, zapata , what do u want? |
| 21:45.21 | [TK]D-Fender | Hydrant: I know. That was a rhetorical question as extensions_custom.conf is a dead giveaway. I just wanted to see it. |
| 21:45.33 | hubguruJR | i have your configs already |
| 21:45.38 | [TK]D-Fender | Hydrant: Don't expect any support for that flaming POS GUI. |
| 21:45.54 | [TK]D-Fender | (here anyways) |
| 21:45.58 | Hydrant | I'm starting to learn the actual dialplan stuff... not expecting support for the GUI |
| 21:46.22 | Hydrant | I'm normally not a GUI person, but at the time I just wanted my phones to work with minimal learning curve :-) |
| 21:47.01 | [TK]D-Fender | Hydrant: Getting any further will be even WORSE with that as a starting point. |
| 21:48.00 | Hydrant | how do outbound routes play into contexts ? |
| 21:48.59 | edwin_quijada | hubguruJR" do u want the output? |
| 21:49.13 | [TK]D-Fender | Hydrant: they don't |
| 21:49.27 | [TK]D-Fender | Hydrant: FreePBX's stupid macro does crazy shit. |
| 21:49.32 | hubguruJR | yes, i want to see what asterisk did when you started it |
| 21:49.41 | edwin_quijada | ok |
| 21:49.49 | [TK]D-Fender | Hydrant: Dialparties.agi |
| 21:50.00 | *** join/#asterisk linagee (n=jalton@about/linux/staff/linagee) |
| 21:50.02 | Hydrant | so I have DISA()... and I want it to let me know make a call, which requires a context correct ? |
| 21:50.16 | linagee | is it really worth it running your own asterisk server? :( |
| 21:50.16 | [TK]D-Fender | Hydrant: FrrePBX can onlyl teach you how NOT to build an * setup. |
| 21:50.16 | linagee | just for 1 or 2 phone numbers? |
| 21:50.32 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-139-75.telkomadsl.co.za) |
| 21:50.39 | [TK]D-Fender | Hydrant: Yes it needs a context, and no I can't tell you what will do. |
| 21:50.55 | [TK]D-Fender | linagee: You tell us. |
| 21:50.55 | linagee | i've had mine for a while, but this is the nth+1 time that I have to go :( because I end up having to restart the server, or my voip loses billing somehow. very very annoying. :( |
| 21:51.13 | Hydrant | [TK]D-Fender: we agree that FreePBX is a POS... I just don't want to start from scratch right now... I have to finish some other projects first |
| 21:51.14 | linagee | [TK]D-Fender: people email me and they're a bit :-/ because they can't call me |
| 21:51.30 | linagee | [TK]D-Fender: it's a problem you have when you get called so infrequently. things can stay broke a long time until they are needed. |
| 21:52.03 | linagee | [TK]D-Fender: maybe if i had like nagios calling in/out every day or something and alerting me if voip goes down it wouldn't be so annoying |
| 21:52.07 | [TK]D-Fender | linagee: You'll have to answer this question yourself. If you service goes down, look at what service you are paying for and the hardware thats supporting it on your end. |
| 21:52.12 | [TK]D-Fender | linagee: Versions, etc |
| 21:52.26 | linagee | [TK]D-Fender: oh yes and sometimes it's my colo. hahaha. that's just great. |
| 21:52.31 | linagee | [TK]D-Fender: too many points of failure.... |
| 21:52.42 | [TK]D-Fender | Hydrant: Well, simply"good luck" finding the right place to drop a caller into.... |
| 21:52.47 | linagee | [TK]D-Fender: if i had a cheap $99 cell from a grocery store it wouldn't be down half as often |
| 21:53.15 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 21:53.24 | edwin_quijada | hubguruJR:http://pastebin.com/m28e19d7 |
| 21:53.29 | linagee | [TK]D-Fender: the only problem there is that cell phones cannot handle a dozen different numbers on one cell. |
| 21:53.34 | Hydrant | How long would it take me as an asterisk n00b to start from scratch writing my own dialplan for 3 phone lines each with their own context, a ring group, and a couple DISA things |
| 21:53.57 | [TK]D-Fender | linagee: You said 1-2 numbers, now its a dozen? Make up your mind |
| 21:54.03 | linagee | Hydrant: this long: " " |
| 21:54.13 | [TK]D-Fender | linagee: :) |
| 21:54.21 | linagee | [TK]D-Fender: hrm. ok. right now it's 3. but i used to have 6. (i disconnected the DIDs to save money) |
| 21:54.49 | linagee | [TK]D-Fender: one of those three is not technically mine, i just don't have the balls to pull the number on the non-paying friend |
| 21:54.54 | Hydrant | I've forgotten how painful it is to be a n00b at something :-P |
| 21:54.58 | [TK]D-Fender | linagee: You know what you want, you know where the weak points are. Do the math yourself. I can't tell you what things are worth to you. |
| 21:55.02 | hubguruJR | edwin_quijada: is that the whole output? |
| 21:55.11 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.142) |
| 21:55.12 | hubguruJR | you are missing some modules |
| 21:55.13 | xacatecas | hi all, is it possible to capture just the raw stream being transmitted into, or received from, a specific channel? |
| 21:55.32 | linagee | [TK]D-Fender: hrm true. i guess i could have a $900/mo colo that is up 99.9999% of the time and a voip provider that costs tons a month for 99.9999% |
| 21:55.34 | [TK]D-Fender | Hydrant: Run facing forward long enough and you'll never know the wolf is right behind you. |
| 21:55.57 | hubguruJR | edwin_quijada: I don't see zaptel loading into asterisk, that is the problem |
| 21:56.03 | linagee | [TK]D-Fender: lots of customers -> lots of complaining when things get broken -> things get fixed rather quickly. |
| 21:56.05 | xacatecas | i've got a problem with an incoming gsm channel being transcoded to g729 sounding really bad on the receiving end and would like to trouble-shoot where it's going belly up. |
| 21:56.29 | hubguruJR | edwin_quijada: pastebin your modules.conf file |
| 21:56.30 | linagee | [TK]D-Fender: land line -> FXO -> never sent over the interweb. that would probably be a way more stable system for asterisk. |
| 21:56.39 | [TK]D-Fender | xacatecas: Only low-end codec to another. BAD conversion |
| 21:57.04 | Qwell | xacatecas: yeah, gsm>g729 doesn't really work |
| 21:57.08 | [TK]D-Fender | edwin_quijada: Zaptel was loading beofre. What did you do? |
| 21:57.10 | xacatecas | perhaps, but it should still work. I only generally allow gsm for testing purposes. |
| 21:57.15 | [TK]D-Fender | Qwell: It does... like shit :) |
| 21:57.21 | Qwell | xacatecas: can you put an example up somewhere? |
| 21:57.31 | Qwell | just to make sure things are no worse than they should be |
| 21:57.35 | xacatecas | i'm trying to get something that consistently fail. |
| 21:57.58 | hubguruJR | edwin_quijada: if you are using putty or secureCRT, set your buffer to about 1500 or higher so you get the full console output on the asterisk restart |
| 21:57.58 | edibrac | can i get to the asterisk shell remotely? like asterisk -rvvvvvvvv I can do locallly, can I get to that from another machine? |
| 21:58.10 | linagee | -rvvvvvvvvvvvvvvv |
| 21:58.20 | linagee | very very very verbose |
| 21:58.24 | Qwell | edibrac: ssh user@host :) |
| 21:59.01 | edibrac | Qwell: ah yeah, i mean i'm wondering how I'd configure nagios to check for the output of "zap show status" |
| 21:59.13 | Qwell | edibrac: you could try using the manager interface |
| 21:59.33 | edibrac | Qwell: oh wait nevermind, i can do that with ssh ..with a one off command....though i'd have to setup ssh-keys |
| 21:59.54 | edwin_quijada | hubguruJR: Buut lsmod says that load the module |
| 22:00.09 | edibrac | Qwell: "manager interface" as in zttool? |
| 22:00.20 | Qwell | no, as in the manager interface |
| 22:00.23 | Qwell | ~manager |
| 22:00.23 | jbot | from memory, manager is a thing that should be killed |
| 22:00.26 | Qwell | ~asterisk manager |
| 22:00.29 | Qwell | ~AMI |
| 22:00.29 | jbot | it has been said that ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 22:00.33 | Qwell | close enough |
| 22:00.39 | hubguruJR | edwin_quijada: Here is what mine looks like, you will se zaptel loading about half way down: http://pastebin.com/m4372d3ef |
| 22:01.22 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:02.07 | riddlebox | is the fax exten part of the latest release of asterisk 1.4.21.2? or do i need to install something else? |
| 22:02.13 | edibrac | hmm for nagios, i'm thinking it might be easier if i do this: write the output of "zap show status" to a file where nagios can check it via HTTP |
| 22:02.42 | edibrac | rather than go through the hassle of setting up authentication for nagios |
| 22:03.01 | hubguruJR | edwin_quijada: send me the modules.conf file |
| 22:03.34 | edibrac | or if i were cool enough, somehow put the results of "zap show status" somewhere query'able in SNMP |
| 22:03.43 | edibrac | is that a common thing to do? |
| 22:04.20 | [TK]D-Fender | riddlebox: Try since pre-1.0 :p |
| 22:04.48 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:05.23 | riddlebox | [TK]D-Fender, hrmm I wonder why its not working, well i will keep digging |
| 22:05.46 | edwin_quijada | hubguruJR: ok |
| 22:06.59 | *** part/#asterisk nny_1 (n=Scott@64.203.244.146) |
| 22:10.48 | edwin_quijada | hubguruJR:http://pastebin.com/m6c73166c |
| 22:11.46 | hubguruJR | edwin_quijada: well, that looks ok too |
| 22:11.59 | hubguruJR | edwin_quijada: now this is confising |
| 22:12.04 | hubguruJR | confusing |
| 22:12.26 | edwin_quijada | that's confuse for me too |
| 22:13.17 | hubguruJR | do you have a chan_zap.so file in /usr/lib/asterisk/modules/ ? |
| 22:14.09 | edwin_quijada | ?? |
| 22:14.44 | edwin_quijada | let me see |
| 22:14.47 | hubguruJR | look in directory /usr/lib/asterisk/modules/ |
| 22:16.44 | edwin_quijada | hubguruJR: I dont have that module |
| 22:16.57 | hubguruJR | aaah haaaa |
| 22:17.02 | hubguruJR | that is the problem |
| 22:17.24 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 22:17.37 | hubguruJR | did you so a ./configure when you compiled asterisk? |
| 22:17.49 | edwin_quijada | yes |
| 22:17.57 | edwin_quijada | for zaptel too |
| 22:17.59 | hubguruJR | do a make menuselect before you make asterisl |
| 22:18.09 | edwin_quijada | uhmmm |
| 22:18.15 | edwin_quijada | i dont think so |
| 22:18.21 | hubguruJR | verify the chan zap is set to load |
| 22:18.35 | edwin_quijada | so what can i solve this probelm |
| 22:19.35 | hubguruJR | go back into your asterisk source tree, do a ./configure, then do a 'make menuselect', verify chan_zap is selected to load, save config, do a 'make', then 'make install' |
| 22:19.51 | *** join/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-863ae1d6678859d5) |
| 22:20.10 | edwin_quijada | ok |
| 22:20.12 | riddlebox | [TK]D-Fender, I found this example on a website, but I get the error on the bottom when I have a fax come in, http://pastebin.ca/1086568 |
| 22:20.17 | kensuke_ | hi, a question the "make sample" command asterisk..... what file created? |
| 22:20.20 | edwin_quijada | Here we go |
| 22:20.36 | edwin_quijada | hubguruJR: the configuration file willbe deleted |
| 22:20.37 | edwin_quijada | ? |
| 22:20.52 | hubguruJR | after you do a make install, check the /usr/lib/asterisk/modules/ directory and make sure you have chan_zap.so |
| 22:22.19 | hubguruJR | don't worry about the config file, it will be re-created when you do the ./configure |
| 22:23.18 | kensuke_ | hubguruJR: hi, the make samples delete old config asterisk?¿ |
| 22:23.30 | hubguruJR | don't so the make samples |
| 22:23.38 | kensuke_ | i don't care /etc/asterisk, but yes the agi-bin |
| 22:23.40 | hubguruJR | don't do |
| 22:23.48 | exothermc | When asterisk waits for a response. if the response is an exact match and there could be no other possible option within that context then shouldn't it go to that exten? |
| 22:24.02 | kensuke_ | hubguruJR: ok |
| 22:24.13 | kensuke_ | hubguruJR: i'm migrate 1.2 to 1.4 |
| 22:26.01 | hubguruJR | did you compile the new 1.4 zaptel source or are you still using the 1.2 zaptel source? |
| 22:26.31 | edwin_quijada | hubguruJR: where i can find that module in menuselect |
| 22:26.36 | edwin_quijada | what option? |
| 22:26.52 | hubguruJR | <PROTECTED> |
| 22:26.52 | exothermc | [TK]D-Fender: When asterisk waits for a response. if the response is an exact match and there could be no other possible option within that context then shouldn't it go to that exten? |
| 22:27.29 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:30.50 | xacatecas | ok, why specifically is gsm -> g729 a bad idea other than the fact that they're both "complex" codecs and transcoding between complex codecs is bad for quality? |
| 22:31.43 | Qwell | xacatecas: that's pretty much it |
| 22:31.52 | Qwell | that, and the low quality of each to begin with |
| 22:32.27 | xacatecas | ok, so not being able to hear the other party at all (sounds like water on the receiving side) is not quite it? |
| 22:32.41 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 22:32.55 | xacatecas | [Jul 30 00:29:23] WARNING[5881]: codec_gsm.c:144 gsmtolin_framein: Invalid GSM data (1) <-- this also does not look promising. |
| 22:36.35 | xacatecas | Qwell, are you aware of any way to from a rtp stream extract the g729 data? |
| 22:38.00 | tzanger | hmm, has anyone here played with the HPEC before? I seem to be getting noise instead of echo cancellation |
| 22:38.22 | xacatecas | Qwell, sorry, something else is bugging me now, gsm == not my favorite codec, it's quality leaves much to desire, however, time upon time g729 comes up as the recommended codec - yet you say it has low quality - what's the argument there? |
| 22:38.54 | Qwell | xacatecas: it's tiny. |
| 22:39.44 | xacatecas | ok, i'd say that's a good thing for bandwidth constrained links? |
| 22:41.35 | *** join/#asterisk koenigjm (n=koenigjm@75.150.72.222) |
| 22:42.08 | koenigjm | Anyone here have experience with Linksys PAP2s? |
| 22:43.10 | koenigjm | In particular I am interested in finding out if they remove *-codes from dialed strings prior to sending the invite request to asterisk. |
| 22:47.19 | edwin_quijada | At last !! I can install my T1 :) Thks to [TK]D-Fender and hubguru for you help !! I owed to red winw |
| 22:47.25 | edwin_quijada | bottle!! :) |
| 22:48.17 | edwin_quijada | Just I have a little question when I call to my T1 this is not ring and come in to messages into extension |
| 22:54.47 | exothermc | is there anyway to edit the from header for asterisk? |
| 22:55.38 | exothermc | right now it says: "From: "asterisk" <sip:asterisk@<IP>>;tag=as663478b0" It would be nice to replace "asterisk" with something custom. |
| 23:01.46 | xacatecas | koenigjm, yes. |
| 23:02.14 | xacatecas | but no, i can't answer your question :) |
| 23:02.37 | koenigjm | fair enough ;) |
| 23:02.57 | xacatecas | i'd recommend watching your asterisk CLI to see if they do, but I can't imagine they would - not unless your dialplan is configured to do so. |
| 23:03.19 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 23:03.41 | MrTelephone | does anyone have a pri from a telco with multiple dids? |
| 23:04.00 | jameswf-home | ~ask |
| 23:04.00 | jbot | somebody said ask was Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:04.22 | MrTelephone | i pay a 911 charge for each t1 channel.. do any telco companies allow you to have different addresses for different dids? |
| 23:04.55 | xacatecas | i ran into a function once that would read a file, say /foo.wav and write it to another, say /foo.gsm ... however, i seem unable to locate this again. |
| 23:05.09 | jameswf-home | ~sox |
| 23:05.10 | jbot | extra, extra, read all about it, sox is Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 23:05.17 | koenigjm | I did check out the CLI, and I was confused by the results...the first invite I see from my pap2 is 1<area_code><prefix><number> even though I dialed *81 1 <area_code><prefix><number>. |
| 23:05.40 | rob0 | k, you're a 'sip set debug peer' and a test call, away from finding your answer. Also see Linksys documentation/help. |
| 23:06.25 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:06.25 | koenigjm | I am looking at the dialplan from my pap2 and I am having trouble parsing it...(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.), would that cause the *-code to be removed? |
| 23:06.54 | xacatecas | i don't think so no. |
| 23:07.23 | xacatecas | however, that dialplan wouldn't make a heck of a lot of sense where I live... |
| 23:11.01 | jameswf-home | koenigjm: http://www.netphonedirectory.com/pap2_dialplan2.htm |
| 23:12.47 | *** part/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:13.02 | MrTelephone | why can't voip providers just use their upstream 911 service that is provided with the t1? |
| 23:13.15 | *** join/#asterisk NickKoji (n=nicholas@168-215-215-195.static.twtelecom.net) |
| 23:16.47 | xacatecas | ok, any ideas on how I'm going to obtain the actual g729 stream that was transmitted in the rtp stream? |
| 23:17.36 | NickKoji | Asterisknow is telling me that there is no hardware detected(analog and digital) and i am using a tdm400p card. My ztsscan.conf reads active=yes alarms=UNCONFIGURED description=ZTDUMMY/1 (source: HRtimer) 1 name=ZTDUMMY/1 manufacturer= devicetype=Zaptel Dummy Timing Driver location= basechan=1 totchans=0 irq=0 |
| 23:18.30 | pkunkra | ~wifivoip |
| 23:18.30 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 23:19.58 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 23:21.27 | exothermc | ok that is odd, when I call in via my trunk, then initiate a call back out the same trunk I get one way audio, If I initiate it or terminate it on network then it works fine. |
| 23:21.55 | *** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 23:27.35 | voxter | anyone know why i might be getting "Received trunked frame before first full voice frame" when using IAX trunking after every call, and when i do an iax2 reload, etc? it flies by about every 20 seconds here. |
| 23:35.36 | *** join/#asterisk Firass (n=Firass@ead224-222.housing.wwu.edu) |
| 23:41.19 | voxter | is there a way to select codec through the dial plan? Like if i have g729 and g711 enabled on my phones, can i use the dialplan to select which codec should be used for the call? |
| 23:42.21 | xacatecas | exothermc, sounds like your re-invites are getting stuck. |
| 23:42.33 | xacatecas | try can canreinvite=no for that trunk and try again. |
| 23:43.05 | xacatecas | voxter, if you find a way - let me know. |
| 23:43.27 | exothermc | xacatecas: I was just looking into that, testing now |
| 23:43.30 | exothermc | xacatecas: thanks |
| 23:43.59 | xacatecas | it's def not ideal, but it'll at least get you going to where exactly the problem really is. |
| 23:45.06 | *** join/#asterisk TJNII (n=knoppix@209.234.91.14) |
| 23:47.06 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 23:48.34 | exothermc | bingo it works now |
| 23:51.42 | JayTee52 | [TK]D-Fender, I figured out my problem with that one troublesome 800 number. My Dial statement had the ring timeout set and the r flag. Removing them fixed it. |
| 23:54.06 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 23:56.24 | *** join/#asterisk simon (n=simon@78.129.201.121) |
| 23:56.52 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 23:57.20 | simon | hi. am I in the right place for turning my analog phone line and a computer modem into an answering machine, among other things? |
| 23:57.54 | NickKoji | How do i stop the ztdummy from loading and load wctdm module |
| 23:58.05 | simon | it seems that asterisk is so versatile that I can't even figure out if a regular 56k modem will do. |
| 23:58.12 | voxter | simon: it wont. |
| 23:58.45 | simon | voxter, can you suggest a piece of software that will; or is the hardware incompatible? |
| 23:59.21 | voxter | simon: the hardware is incompatible. |
| 23:59.36 | voxter | simon: get a sipura 3102 or something. |
| 23:59.48 | voxter | voxter: www.voip-info.org is a good resource |
