| 00:00.01 | KavanS | *snap* |
| 00:00.57 | bmoraca | Micc: I receive every single SIP packet from every single UA on the network EXCEPT the responses to the keepalives. |
| 00:01.54 | field64 | so from asterisk to the phones - then a sniffer trace at the server interfaces, or the private VPN gateway would be the 1st step, presume you've done this,, and the packets never leave the asterisk server ? |
| 00:01.57 | Alfio | but with the cost of that you can get paid support for asterisk, take yourself the asterisk advanced course and save some money |
| 00:01.59 | Micc | bmoraca, are the packets actually making it to the machine? |
| 00:02.26 | bmoraca | according to tcpdump, yes. |
| 00:03.08 | field64 | so if tcpdump is on the right interface, and showing the frames inbound from all phones, and outbound to all phones, they're going astray somewhere along the path |
| 00:03.14 | Micc | bmoraca, and what port are they being sent to? And are the being received on the same IP as the keepalive request? |
| 00:04.04 | Micc | bmoraca, something must be keeping asterisk from correlating the packets to the keepalive request packets. |
| 00:04.30 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 00:04.40 | bmoraca | you think? i'd be nice if the packets showed up in the sip debug, however asterisk has decided that it doesn't want to show incoming packets. |
| 00:05.01 | [TK]D-Fender | Micc: Nope. global SIP debug dumps the whole packet on-screen regardless of what it cares to do with it |
| 00:05.33 | [TK]D-Fender | bmoraca: If you can't see the packet in tcpdump on the * server itself then its not getting there |
| 00:05.38 | grandpapadot | Hey guys, I'm connecting peers from a remote asterisk box and dialing them as members of a queue defined as "SIP/801@remotepeer". It works the first time just fine, but then the member just stays in a "ringing" state until I reload app_queue, any ideas? |
| 00:05.48 | bmoraca | [TK]D-Fender: well, that's a problem, because asterisk isn't displaying ANY inbound packets, and I CAN see it come in on tcpdump |
| 00:06.08 | [TK]D-Fender | bmoraca: Check the interfaces you have bound <--- |
| 00:06.14 | Micc | bmoraca, then it must be on an ip or port that its not listening to. |
| 00:06.21 | bmoraca | SIP is bound to 0.0.0.0 |
| 00:06.29 | [TK]D-Fender | bmoraca: Again the dual-homing you had before is a classic source of crap like this |
| 00:06.43 | field64 | question is - what kind of dual homing ? |
| 00:06.51 | bmoraca | [TK]D-Fender: as I stated before, problem exists whether or not both NICs are enabled. |
| 00:06.53 | Micc | bmoraca, is the vpn ip being created after asterisk starts? |
| 00:06.59 | field64 | if it's bonding mode 0 - that will not work |
| 00:07.07 | field64 | not work well that is |
| 00:07.54 | field64 | or mode 6 |
| 00:07.59 | bmoraca | each NIC is on a separate subnet, the box only has a single route out of the local network, and the VPN is hardware VPN, not software. |
| 00:08.18 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 00:08.46 | Micc | bmoraca, either the packet shows up on the wrong ip or port in tcpdump or it would be printed in asterisk sip debug. |
| 00:09.11 | bmoraca | one would expect so, but alas that is not the case. |
| 00:09.35 | [TK]D-Fender | Ok, I'm not seeing anything here.... I'm stepping back from this... |
| 00:09.50 | Micc | bmoraca, you are telling me it comes in on the same ip and port as the request went out? |
| 00:10.35 | Micc | bmoraca, one of your assumptions or data must be wrong. |
| 00:10.45 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 00:10.45 | bmoraca | Micc: that is correct. only one interface can access the VPN, and the packets are being received by the phone. the phone is unsuccessful in its attempt to send the packets back. |
| 00:11.07 | bmoraca | Micc: I am not making any assumptions. everything I have stated is the result of testing or configuration. |
| 00:12.12 | Micc | bmoraca, why do you phrase it like that? I don't think you have really done the tcpdump and seen the packet come back. Otherwise you would say, the packet does get sent back but asterisk doesn't see it. |
| 00:12.26 | grandpapadot | Hey guys, I'm connecting peers from a remote asterisk box and dialing them as members of a queue defined as "SIP/801@remotepeer". It works the first time just fine, but then the member just stays in a "ringing" state until I reload app_queue, any ideas? |
| 00:12.37 | Micc | bmoraca, it seems like your skirting the question. |
| 00:13.08 | Micc | bmoraca, probably because you believe it to be so and theres no need to actually see it. |
| 00:13.10 | bmoraca | Micc: there's no difference. phone receives packet, asterisk doesn't. how many more times do I have to say it? |
| 00:13.26 | Alfio | its more easy to blame asterisk |
| 00:13.40 | bmoraca | Alfio: if asterisk is what is having the problem, yes, it is. |
| 00:13.58 | Micc | bmoraca, the difference is huge. either it makes it to the box or it doesn't. |
| 00:15.18 | Micc | bmoraca, forget about asterisk for a moment. Just find out if the packet makes it to the box. |
| 00:15.58 | Alfio | asuming you are "networking specialist" asterisk is what you know less, thats why its more easy to blame * |
| 00:18.07 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 00:18.17 | field64 | anyone here know DAHDI ? |
| 00:19.21 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.69) |
| 00:19.38 | Qwell | bmoraca: What distro is Asterisk running on? |
| 00:19.40 | [TK]D-Fender | field64: Yes, married to MAHMI |
| 00:19.50 | bmoraca | centos 5.2 |
| 00:20.09 | field64 | ah - so MAHMI's shoe shopping trips are what is making DAHDI test show 4% ? |
| 00:20.13 | Qwell | well, let's see a pcap from the Asterisk box |
| 00:20.18 | [TK]D-Fender | bmoraca: Show us the tcpdump, show us the firewall dump. The routing table, etc... |
| 00:20.44 | Micc | bmoraca, I can't even count the number of times I've thought for sure asterisk had a bug but it was some minor configuration error. |
| 00:21.15 | [TK]D-Fender | Micc: But he's assured us 1000 times. And he's a Cisco engineer! |
| 00:21.52 | bmoraca | Micc: i don't care whether it's a bug or a config error...i just need it fixed. |
| 00:22.10 | Micc | bmoraca, then show us the tcpdump. |
| 00:22.38 | [TK]D-Fender | And everything else. |
| 00:22.43 | Micc | bmoraca, do you have all your ip's and host names in the domains section of sip.conf? |
| 00:22.59 | [TK]D-Fender | Proof * is bound to the interfaces & ports. Firewall dump..... |
| 00:24.36 | Micc | bmoraca, the only way to fix it is to care what the problem is. It won't just fix itself. |
| 00:30.52 | bmoraca | http://www.pastebin.org/3676 <-- packets being sent to and processed, but not displayed by asterisk debug. |
| 00:31.57 | Qwell | Asterisk wouldn't be responding if it got nothing. Where is the first packet? |
| 00:33.11 | bmoraca | http://www.pastebin.org/3677 <-- debug of asterisk sending keepalives, phone receiving and responding, and asterisk getting nothing |
| 00:33.59 | ltd_wk | Since everyone's around. Does anyone know the semantics of what would happen if you're simultaenously dialling say 2 sip peers and one returned a 302? |
| 00:34.01 | bmoraca | Qwell: ignore the retransmits in the first PB. the meat starts at line 74 where asterisk responds to a packet it doesn't record as having received. |
| 00:35.28 | Qwell | So where are you seeing that machine getting packets back from the phone? |
| 00:36.18 | bmoraca | in the tcpdump |
| 00:36.28 | Qwell | These are different timeframes. |
| 00:36.53 | bmoraca | no they are not. the tcpdumps and the debugs in the first PB were done at exactly the same time |
| 00:37.06 | Qwell | Then show me a full pcap... |
| 00:37.25 | [TK]D-Fender | I'd like to see that this was GLOBAL SIP debug, and not peer/ip |
| 00:38.26 | bmoraca | Qwell: those are the only packets that tcpdump will capture, despite receiving and matching as many as asterisk sends keepalives. |
| 00:38.50 | bmoraca | [TK]D-Fender: it doesn't show any more information. |
| 00:38.57 | [TK]D-Fender | . |
| 00:39.08 | [TK]D-Fender | Now I REALLY don't trust what I see |
| 00:39.28 | Qwell | Then why don't I see the OPTIONS packets in the pcap?.. |
| 00:39.32 | [TK]D-Fender | bmoraca: We'd have seen the register packet come in... |
| 00:39.46 | bmoraca | [TK]D-Fender: but you DIDN"T! and that's the PROBLEM! |
| 00:40.05 | bmoraca | Qwell: good question. i'll be sure to ask the people who wrote tcpdump when I see them next. |
| 00:40.06 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 00:40.27 | Qwell | so it's a bug in tcpdump too?... |
| 00:40.31 | field64 | so this capture is taken on the asterisk serer interface, where asterisk is bound, and this interface has connectivity and router to reach the phones ? and there is no filter set ? |
| 00:40.44 | [TK]D-Fender | I call "bullshit". I am done with "tunnel-vision" debugging for the day |
| 00:40.57 | field64 | i guess it works different for cisco |
| 00:41.14 | bmoraca | Qwell: no, if you had read what I stated a minute and a half ago, you'd have seen that tcpdump increments the "matched" packet counter as many times as asterisk sends out keepalives, and yet only "captures" the registers. |
| 00:41.33 | bmoraca | Qwell: I pastebinned the results to that effect, as well as the command I used. |
| 00:41.58 | Qwell | Where is the REGISTER message from the phone? where are the outgoing OPTIONS in the pcap? |
| 00:42.36 | bmoraca | Qwell: I didn't capture outbound traffic, only inbound. and the registers didn't show up in asterisk debug for some reason that I cannot fathom...hence why I'm in here trying to figure it out. |
| 00:43.06 | *** join/#asterisk ArchGT (n=ArchGT@190.149.38.134) |
| 00:43.14 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 00:43.35 | Qwell | show us *everything*, from both sides. |
| 00:43.43 | field64 | ok, so i solved my DAHDI issue - backdated, wondering where is a good place to post this little adventure, perhaps save someone else the headache ? |
| 00:44.14 | bmoraca | Qwell: you've already seen everything that asterisk and the phone give me. i will pcap both directions for you...but it'll be much of the same. |
| 00:44.55 | Qwell | field64: mailing lists are indexed by Google, and are the answer in many cases |
| 00:45.21 | field64 | @Qwell - not in any asterisk list, perhaps i should join and post this type of thing ? |
| 00:45.35 | Qwell | field64: there's also the wiki too, depending on the issue |
| 00:46.50 | field64 | @Qwell - yes, i've searched almost everything i know, just solved by downgrading DAHDI, now i'd like to post somewhere to save someone else the headache |
| 00:46.58 | Qwell | field64: what was the issue? |
| 00:47.14 | Qwell | (if a downgrade worked...it's probably a bug) |
| 00:47.20 | bmoraca | http://www.pastebin.org/3680 <-- outgoing keepalives from asterisk. |
| 00:47.28 | field64 | DAHDI test showing 4%, asterisk conf, directory - the automated speech broken |
| 00:47.48 | field64 | used to work, then stopped, might be this great PC i have, the IRQs are "Special" |
| 00:49.47 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-70dad05472d322de) |
| 00:49.59 | bmoraca | http://www.pastebin.org/3681 <-- everything into and out of asterisk to my test phone |
| 00:51.02 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 00:52.00 | Qwell | Now show me that Asterisk is listening on that IP:port |
| 00:53.10 | bmoraca | Qwell: the fact that registers and every other SIP packet reach asterisk just fine isn't enough for you? |
| 00:53.26 | Qwell | show me where the register reaches Asterisk |
| 00:53.26 | *** join/#asterisk memph (n=memph@sd1438.sivit.org) |
| 00:54.14 | bmoraca | http://www.pastebin.org/3676 <- the phone registers and immediately loses connectivity because asterisk loses the keepalives |
| 00:54.55 | grandpapadot | Hey guys, I'm connecting peers from a remote asterisk box and dialing them as members of a queue defined as "SIP/801@remotepeer". It works the first time just fine, but then the member just stays in a "ringing" state until I reload app_queue, any ideas? |
| 00:54.58 | Qwell | There is no register there. |
| 00:55.02 | bmoraca | or if you would prefer: http://www.pastebin.org/3682 |
| 00:55.14 | bmoraca | Qwell: there absolutely is. Asterisk just doesn't show it in the debug. |
| 00:55.41 | [TK]D-Fender | bmoraca: And you've aluded that you aren't doing GLOBAL SIP debug |
| 00:55.43 | Qwell | and how are you enabling debug? |
| 00:56.27 | bmoraca | are you really going to make me prove that the global debug will make absolutely no difference in this instance? |
| 00:56.31 | Qwell | Yes |
| 00:56.35 | [TK]D-Fender | YES |
| 00:56.45 | bmoraca | get ready for a huge load of absolutely nothing. |
| 00:57.07 | [TK]D-Fender | expects to see 2-3 specific lines minimum |
| 00:57.39 | Qwell | and you also haven't shown us the register from the phone |
| 00:58.18 | bmoraca | Qwell: yes I have. several times. or at least I've shown you everything that asterisk will let me show you. i've even pcaped the register. |
| 00:58.26 | Qwell | I want to see the debug from the phone. |
| 00:58.42 | [TK]D-Fender | I want to see debug from *... ALL of it. Proof of what I'm seeing. |
| 01:01.12 | bmoraca | http://www.pastebin.org/3683 <-- unregister is around line 2478, register is at the very bottom...cut off by pastebin, i guess |
| 01:01.21 | bmoraca | and yes, i do see the sip reads for the registers. |
| 01:01.39 | [TK]D-Fender | WOW, a miracle. |
| 01:01.42 | bmoraca | however, the fact remains that i still get nothing back from the keepalives. |
| 01:01.52 | Qwell | <--- SIP read from 192.168.254.23:52895 ---> |
| 01:01.52 | Qwell | REGISTER sip:192.168.163.29 SIP/2.0 |
| 01:04.22 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 01:04.32 | [TK]D-Fender | bmoraca: I'd like to see the entire register. So far you aren't showing us anything that involves more that 1 packet going out. |
| 01:04.50 | [TK]D-Fender | bmoraca: Then I'd want tos ee a chanllenged call. |
| 01:05.26 | [TK]D-Fender | bmoraca: Because * could be sending the 200 OK to other side and THAT gets lost TOO. Because doesn't EXPECT an answer, unlike OPTIONS <- |
| 01:05.40 | [TK]D-Fender | Trust = 0 |
| 01:06.08 | [TK]D-Fender | I swear nobody knows how to do a sane test in here.... |
| 01:06.21 | bmoraca | my phone gets every packet that asterisk sends it. |
| 01:06.49 | [TK]D-Fender | bmoraca: Show something more elaborate including the softphone test you refused a few dozen times. |
| 01:06.50 | Qwell | show me the 100 trying and the 200 ok to the register |
| 01:07.13 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 01:08.01 | guax | tension is in the air |
| 01:08.24 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 01:08.41 | bmoraca | Qwell: from the same register that i just pastebinned, but not cut off this time: http://www.pastebin.org/3686 |
| 01:09.00 | Qwell | from the phone.. |
| 01:09.15 | bmoraca | little more difficult to get...one moment. |
| 01:09.18 | Qwell | I know what Asterisk is sending/receiving (which is more than you said it was...) |
| 01:09.44 | bmoraca | sending and receiving everything but the 200 OK response to the keepalive. |
| 01:10.55 | bmoraca | http://www.pastebin.org/3688 <-- register starts on line 41 |
| 01:11.03 | Qwell | well I don't see them in your pcap either... |
| 01:11.46 | bmoraca | see what? the registers are there, the 200 and 100, no, but again, it's not my fault that tcpdump doesn't capture every packet that matches the pattern |
| 01:12.29 | Qwell | how do you know the 200 OK to the OPTIONS is getting received by that machine? |
| 01:12.49 | bmoraca | because every single other sip message sent from that and every other phone is? |
| 01:13.09 | bmoraca | and because the counter for tcpdump increments the appropriate number of times, even if it doesn't capture it |
| 01:13.54 | *** join/#asterisk Magicblaze0071 (n=sony@fl-67-235-208-62.dhcp.embarqhsd.net) |
| 01:14.29 | bmoraca | if i set qualify=no, i can make calls all day long into and out of every single phone in the building. |
| 01:14.45 | bmoraca | well, even with qualify=yes, i can still make calls out of every phone |
| 01:15.43 | [TK]D-Fender | bmoraca: http://www.pastebin.org/3686 <-- I don't see * sending the 200 OK for the register either |
| 01:16.14 | bmoraca | [TK]D-Fender: yeah, i pulled it off too early...it's hte next packet. need me to PB it? |
| 01:16.33 | [TK]D-Fender | bmoraca: you are breaking every attempt to debug this. |
| 01:16.38 | bmoraca | http://www.pastebin.org/3690 |
| 01:17.15 | [TK]D-Fender | bmoraca: place a call from the phone that * challenges |
| 01:17.45 | bmoraca | http://www.pastebin.org/3691 <--- here's a fun one...a notify and a 200 back from the phone to that notify |
| 01:18.25 | bmoraca | [TK]D-Fender: asterisk challenges all of the phones, but unfortunately, i do not have physical access to them at this time so i cannot place a call. i am going to set up a second vpn for testing with identical equipment tomorrow. |
| 01:18.29 | Magicblaze0071 | Can I make the IVR in a way that the user can press "0" and extension "11" is called? etc...? |
| 01:18.56 | [TK]D-Fender | bmoraca: So far the only packet ever getting TO * looks like the register. |
| 01:19.03 | [TK]D-Fender | bmoraca: place the call <- |
| 01:19.18 | [TK]D-Fender | Magicblaze0071: With * you can do whatever you want |
| 01:19.35 | [TK]D-Fender | Magicblaze0071: With FreePBX... bug them if you're playing with their GUI crap menus |
| 01:20.02 | bmoraca | [TK]D-Fender: i can show you call logs of placed calls, but they do not have sip debugs on them |
| 01:20.25 | [TK]D-Fender | bmoraca: Then go make a NEW one. |
| 01:20.36 | *** join/#asterisk Iamnach0 (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 01:20.47 | Magicblaze0071 | [TK]D-Fender: What file should I play with in asterisk / ivr to do what I want to? |
| 01:20.50 | bmoraca | i can't until i set up a second vpn to test from. these phones are all at a customers location that i do not currently have access to.\ |
| 01:21.03 | [TK]D-Fender | Magicblaze0071: Its dialplan, take a wild guess |
| 01:21.27 | [TK]D-Fender | bmoraca: The judges aren't the ones who are supposed to be double-blind... |
| 01:21.29 | Magicblaze0071 | Sorry, I haven't worked on the configurations for more than a month, my asterisk book is sitting in office. |
| 01:21.46 | jblack | Magicblaze0071: Here's a suggestion. Read the Asterisk book. It's free. |
| 01:21.54 | [TK]D-Fender | Magicblaze0071: Not like you'd had much incentive since going FreePBX last year |
| 01:22.11 | [TK]D-Fender | Magicblaze0071: LONG ago.... extensions.conf <- |
| 01:22.26 | Magicblaze0071 | thanks |
| 01:24.29 | bmoraca | i can place one via telnet. one moment |
| 01:26.22 | field64 | thanks for the chat, good luck with that problem bmoraca |
| 01:26.35 | *** part/#asterisk field64 (n=6667@p5B22F010.dip.t-dialin.net) |
| 01:28.39 | bmoraca | [TK]D-Fender: here's your invite, etc: http://www.pastebin.org/3695 |
| 01:30.16 | [TK]D-Fender | HA! |
| 01:30.24 | [TK]D-Fender | Retransmitting #1 (no NAT) to 192.168.254.23:5060: |
| 01:30.26 | [TK]D-Fender | SIP/2.0 407 Proxy Authentication Required |
| 01:30.32 | [TK]D-Fender | Cisco isn't responding to jack shit |
| 01:30.47 | [TK]D-Fender | not just the answer to options, but the answer to the challenge even |
| 01:31.03 | [TK]D-Fender | Dump your firewall |
| 01:31.08 | Xetrov` | this sounds like a job for the mystery machine! |
| 01:31.29 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 01:31.32 | [TK]D-Fender | Goes out but never comes back |
| 01:31.32 | Qwell | [TK]D-Fender: it *should* send a new invite with the account stuff (user/pass) |
| 01:31.47 | Qwell | (after a 407) |
| 01:31.57 | bmoraca | Qwell: it does. there are several invites in that log i posted. |
| 01:32.00 | [TK]D-Fender | Qwell: Yeah, well.... as we see NOTHING comes back, and I STILL never saw the firewall. I only asked a half dozen times. |
| 01:32.12 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 01:32.39 | *** join/#asterisk Kumbang (n=beng@rusnas.paume.itb.ac.id) |
| 01:33.17 | bmoraca | or, i should say...there are 2 invites |
| 01:33.24 | bmoraca | one right before the 407 and one right after. |
| 01:33.42 | bmoraca | in fact, asterisk responds to the second with a 491 and the call goes through |
| 01:33.59 | bmoraca | not sure why asterisk decides to resend the 407. |
| 01:35.23 | bmoraca | seems to me that the phone is doing what it's supposed to |
| 01:35.37 | Alfio | 66.242.170.2 |
| 01:36.20 | bmoraca | what about it? |
| 01:36.51 | *** join/#asterisk dynaram (n=dynaram@rrcs-97-76-19-94.se.biz.rr.com) |
| 01:37.33 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 01:38.47 | bmoraca | you know what's fun? asterisk even resends the 491, despite having received an ACK for it... |
| 01:39.19 | dynaram | hi group before i get involve in reading and reading for hours end. i would like to ask a question to see if asterisk will do what i need. I will like to make phone calls like skype using my pc to make the call and my vonage ot t-mobile voip hardware to make calls |
| 01:39.58 | [TK]D-Fender | dynaram: Don't bet on unlocking your vonage hardware for open use. Might work, might not. |
| 01:40.42 | dynaram | meaning my voip box is in one location and using any wifi re-routing my call to my home voip box to make free calls |
| 01:40.42 | [TK]D-Fender | dynaram: * can work with jsut about any commodity hardware with common open protocols |
| 01:41.48 | dynaram | from what you said is possible. so that means i will be reading alot. thx |
| 01:43.15 | *** join/#asterisk BadHAL (n=nn@173-112-182-76.pools.spcsdns.net) |
| 01:44.39 | bmoraca | well, time to go home. hopefully i can figure this out tomorrow |
| 01:49.00 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 01:52.14 | *** join/#asterisk b14ck (n=quassel@cpe-24-24-136-239.socal.res.rr.com) |
| 01:52.46 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:52.52 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:53.46 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 02:09.45 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:11.36 | MT`AwAy | http://pastebin.com/m4aa8b338 <- got this kind of errors when receiving a call I route via Dial() to something else after Answer()ing it |
| 02:11.45 | MT`AwAy | anyone knows how I can fix this? |
| 02:17.34 | *** join/#asterisk grabiarz (n=ptr@cpe-24-193-83-133.nyc.res.rr.com) |
| 02:19.50 | [TK]D-Fender | MT`AwAy: pastebin the ENTIRE call with SIP DEBUG enabled |
| 02:22.02 | MT`AwAy | [TK]D-Fender> in fact I could fix that by not Answer()ing the call, just need to forget about using music on hold |
| 02:28.14 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 02:32.59 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:36.56 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:37.14 | Micc | TKD-Fender, was bmoraca ever able to prove the packet was getting to the machine like he claimed? |
| 02:38.44 | [TK]D-Fender | Micc: Well an INVITE made it in, but no answer to any challenge. |
| 02:39.16 | [TK]D-Fender | (failure) |
| 02:45.29 | Micc | I'm convinced his firewall is eating or munging the packet. |
| 02:46.49 | Micc | I need a way to check if an extension in the dialplan exists using a dialplan command, does such command exist? I can't seem to find any indication that it does except some references to ISEXTEN, which doesn't seem to exist in my version of asterisk. |
| 02:47.19 | Micc | I'm hoping I'm just missing some simple thing and I'm just really stupid. |
| 02:47.24 | *** join/#asterisk blkry (n=blkry@96.37.27.72) |
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| 02:50.40 | Orbixx | What's the best way to convert wav/mp3 to alaw/ulaw? |
| 02:50.49 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 02:51.10 | [TK]D-Fender | Micc: And after another 1/2 dozen requests I never saw the firewall dump |
| 02:51.28 | [TK]D-Fender | Orbixx: Go lookup "sox convert" on the WIKI |
| 02:51.30 | [TK]D-Fender | ~wikis |
| 02:51.30 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
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| 02:54.33 | stope | TK you put a lot of time in here |
| 02:59.36 | [TK]D-Fender | yup |
| 02:59.38 | blkry | does one have to edit the dial plan on a Polycom to enable ISN dialing? |
| 02:59.59 | [TK]D-Fender | Inter-Stellar Network? |
| 03:00.09 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 03:00.09 | [TK]D-Fender | Check your babCom |
| 03:00.48 | blkry | freenum.org |
| 03:02.08 | [TK]D-Fender | blkry: Their site is wonderfully NON-responsive.... |
| 03:02.49 | blkry | yeah it sure is. |
| 03:02.54 | leifmadsen | eh? |
| 03:03.02 | blkry | yeah |
| 03:03.07 | leifmadsen | oh ya, it is slow right now |
| 03:03.08 | leifmadsen | weird |
| 03:03.24 | leifmadsen | ISN's just started to pick up over the last week or two |
| 03:03.32 | leifmadsen | (pretty much since I started to use it :)) |
| 03:03.39 | leifmadsen | although that is probably just a fluke |
| 03:03.40 | blkry | lol |
| 03:04.07 | leifmadsen | 460 represent! |
| 03:04.55 | blkry | do you have to edit the dial plan on a Polycom to enable dialing. |
| 03:05.12 | blkry | I am not at the office. setting up the ISN |
| 03:05.33 | blkry | I am not at the office so I can not test it. |
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| 03:06.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 03:07.29 | leifmadsen | blkry: well my polycom was already setup to allow me to dial 012XXX*YYY |
| 03:11.13 | [TK]D-Fender | WTF is an "ISN"? |
| 03:12.17 | leifmadsen | [TK]D-Fender: ITAD Subscriber Number |
| 03:12.35 | [TK]D-Fender | leifmadsen: And in "makes a difference" terminology? |
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| 03:14.21 | leifmadsen | [TK]D-Fender: it is a method of using numbers to route to sip routes. i.e. it can be used as an interim method of dialing things like sip:leif@leifmadsen.com, because most (all) telephones do not allow you to dial a SIP URI |
| 03:14.51 | citywok | I'm attempting to do some custom CDR work, and am curious what the recommended method is. write a script that tails CSV output to do the processing, or use something like the not possible to find any more CDR_SHELL that is referenced on voip-info |
| 03:15.00 | [TK]D-Fender | leifmadsen: So a proxy that takes a flat # into a URI? |
| 03:15.33 | leifmadsen | so you are assigned an ITAD number by IANA (IP Telephony Adminstrative Domain), and then you create numbers to dial within that domain. So for example, a school campus can have ITAD 256, and all extensions could be dialed via something like, 1234*256 |
| 03:15.45 | leifmadsen | 1234*256 would then map to a SIP URI |
| 03:15.59 | [TK]D-Fender | leifmadsen: \o/ |
| 03:16.06 | leifmadsen | [TK]D-Fender: basically - it is an interim solution while we continue to have telephones that still only allow dialing by numbers |
| 03:16.15 | [TK]D-Fender | leifmadsen: Ok, so if the phone only has to dial std DTMF... whats the fuss? :) |
| 03:16.16 | leifmadsen | which we'll have for a very long time |
| 03:16.38 | leifmadsen | [TK]D-Fender: there is no fuss, it's a simple way of mapping numbers to SIP URIs which currently work |
| 03:17.06 | [TK]D-Fender | leifmadsen: I meant blkry's |
| 03:17.18 | leifmadsen | huh? |
| 03:17.22 | [TK]D-Fender | blkry: Indeed if all you have to do is dial a #, whats the problem>? |
| 03:17.51 | [TK]D-Fender | blkry: [23:04]<blkry>do you have to edit the dial plan on a Polycom to enable dialing. <--- technically no, desirably, yes |
| 03:17.52 | leifmadsen | [TK]D-Fender: he's basically saying the default dialplan on a polycom seems to fail when he gets to the * part of the number |
| 03:18.16 | blkry | from what I was readin some phones (Aastra's) you have to edit the dial plan on it |
| 03:18.20 | leifmadsen | I suppose you could edit it, but I have no idea how you would know how to make the polycom "auto dial" an ISN |
| 03:18.28 | [TK]D-Fender | blwell DUH, phone dialplans will control what you are allowed to dial. thats the POINT of them. Mind you you can bypass that by dialing ONHOOK anyway |
| 03:21.07 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 03:21.33 | blkry | the onhook trick is good to know |
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| 03:59.18 | mmattice | how do you set up zapata.conf to deal with a pri_cpe and a pri_net at the same time? |
| 03:59.58 | [TK]D-Fender | leifmadsen: MmCare to clarify that... |
| 04:00.07 | [TK]D-Fender | mmattice: rather |
| 04:00.09 | [TK]D-Fender | wow |
| 04:00.47 | shido6 | :) |
| 04:01.11 | mmattice | we have a two channel card, one connected to the telco, another to our pbx |
| 04:01.26 | mmattice | the stuff in zapata.conf seems to be more global than specific to a span |
| 04:01.44 | *** join/#asterisk javb (n=javb@190.94.121.208) |
| 04:02.04 | [TK]D-Fender | mmattice: the signalling is specific down to the channel on the card. |
| 04:02.15 | [TK]D-Fender | mmattice: X channels on a span. |
| 04:02.25 | [TK]D-Fender | MmYour sense of heirarchy is a bit skewed |
| 04:03.40 | mmattice | eh, two port card. TE220B. |
| 04:04.01 | javb | i have a call comming from another pbx, connected using iax2 trunk.. im receiving the call in the "default" context, eventhought the exten is not called form that context... and so i get the error "non extension found" .. any idea. ? |
| 04:04.13 | Orbixx | Can anybody shed some light on what might be going wrong here? [Jul 23 06:03:55] NOTICE[24591]: chan_iax2.c:9305 socket_process: Rejected connect attempt from 66.54.140.46, request 's@incomingkall' does not exist |
| 04:04.49 | javb | I have the same box that is sending me the call, with another iax trunk, using the same syntax, and the call goes OK |
| 04:05.20 | mmattice | [TK]D-Fender: how is it skewed? I'm looking at the example configurations. |
| 04:05.34 | [TK]D-Fender | mmattice: And each port is a span, and each span has channels, and you could have split signalling on a given span. |
| 04:05.55 | [TK]D-Fender | Orbixx: that exten does not exist in that context. Just like it says |
| 04:06.04 | javb | Orbixx, 66.54.140.46, requested the extension "s" on the context "incomingkall", either u dont have that context created, nor the exten "s" |
| 04:06.46 | javb | [TK]D-Fender, have any idea on my issue..? it a little strage.. and i have some time trying to figure it out. |
| 04:07.09 | [TK]D-Fender | javb: I'm not seeing much... |
| 04:07.40 | javb | Jul 22 18:24:51 NOTICE[2103]: chan_iax2.c:6994 socket_read: Rejected connect attempt from 10.10.10.2, request '3220@default' does not exist |
| 04:07.45 | javb | Thats what i get... |
| 04:08.02 | javb | But, nobody is telling the calling party to dial on the context "default" |
| 04:08.03 | mmattice | [TK]D-Fender: that still doesn't answer the question. zapata.conf doesn't seem to have a way to set different signalling for each span |
| 04:08.48 | [TK]D-Fender | mmattice: chages settings. Declare channels. changes settings to something different. Declare channels <0 |
| 04:08.51 | javb | mmattice, zapata.conf does have a way to do that |
| 04:09.09 | [TK]D-Fender | mmattice: When you do "channel => |
| 04:09.26 | [TK]D-Fender | mmattice: When you do "channel =>" that takes whatever the running story is and applies it to them. |
| 04:10.04 | [TK]D-Fender | mmattice: signaling=pri_cpe |
| 04:10.14 | [TK]D-Fender | mmattice: channel => 1-23 |
| 04:10.17 | [TK]D-Fender | mmattice: signaling=pri_net |
| 04:10.25 | [TK]D-Fender | mmattice: channel => 25-47 |
| 04:10.30 | [TK]D-Fender | \o/ |
| 04:10.41 | [TK]D-Fender | mmattice: This ain't Raw-Cat Sigh Hence |
| 04:11.32 | Orbixx | This is my config: http://pastebin.com/d526984fd |
| 04:11.33 | [TK]D-Fender | goes to the Peking House for som more Raw Cats |
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| 04:11.48 | mmattice | [TK]D-Fender: that's unintuitive. |
| 04:11.52 | Orbixx | I do not see what is going wrong or why s@ is being requested. |
| 04:12.09 | [TK]D-Fender | Orbixx: I do not see an exten named "s" in there. do YOU? |
| 04:12.17 | Orbixx | No. |
| 04:12.29 | Orbixx | I even put an exten with s in there, it still fails. |
| 04:12.40 | Orbixx | But I don't understand why it's requesting s anyway. |
| 04:12.43 | [TK]D-Fender | Orbixx: O RLY? Please show |
| 04:13.01 | [TK]D-Fender | mmattice: welcome to the tip of the iceberg :) |
| 04:14.10 | Orbixx | [TK]D-Fender: http://pastebin.com/d51ce993a |
| 04:15.20 | Orbixx | oh, now it works |
| 04:15.38 | [TK]D-Fender | :O |
| 04:15.50 | [TK]D-Fender | </shock> |
| 04:15.53 | Orbixx | Must've done it wrong the first time when I put the 's' in. |
| 04:15.57 | [TK]D-Fender | [00:04]<Orbixx>Can anybody shed some light on what might be going wrong here? [Jul 23 06:03:55] NOTICE[24591]: chan_iax2.c:9305 socket_process: Rejected connect attempt from 66.54.140.46, request 's@incomingkall' does not exist |
| 04:16.03 | [TK]D-Fender | Orbixx: maybe SPELLING too. |
| 04:16.12 | [TK]D-Fender | 's@incomingkall' |
| 04:16.17 | [TK]D-Fender | [incomingipkall] |
| 04:16.24 | Orbixx | Yes, I fixed that after the first pastebin. |
| 04:16.43 | [TK]D-Fender | NEXT!@!@!!@!!@ (c) BKW |
| 04:16.45 | Orbixx | [TK]D-Fender: A different trunk I use expects the exten named by the incoming number, whilst the other expects just 's'. |
| 04:16.58 | Orbixx | Was just something I didn't see coming. |
| 04:17.16 | [TK]D-Fender | Orbixx: And it expects "s" because you didn't tell them any different |
| 04:17.37 | Orbixx | Tell them how? |
| 04:18.07 | *** join/#asterisk thegoat (n=jircii@c-71-224-180-83.hsd1.pa.comcast.net) |
| 04:18.39 | [TK]D-Fender | Orbixx: Your register statement |
| 04:18.41 | thegoat | can asterisk do intercom....i want to be able to pick up the phone and say cleanup on aisle 6, and have it broadcast out all the phones |
| 04:19.12 | [TK]D-Fender | thegoat: Its up to your phones to be capable of being told to answer on speaker |
| 04:19.13 | Orbixx | I don't have a register statement? |
| 04:19.18 | [TK]D-Fender | Orbixx: Do you? |
| 04:19.25 | thegoat | i have cisco 7940s |
| 04:19.38 | [TK]D-Fender | thegoat: Tere are ways with those |
| 04:19.42 | Orbixx | Pretty sure of it. |
| 04:19.42 | [TK]D-Fender | there* |
| 04:19.53 | Orbixx | All I have is what you saw in the pastebins. |
| 04:20.01 | Orbixx | Nothing more. |
| 04:20.26 | [TK]D-Fender | Orbixx: Then its in what you gave them to conteact you with |
| 04:20.28 | thegoat | [TK]D-Fender: member the other day i was having issues with callerid with name...it magically started working |
| 04:20.47 | [TK]D-Fender | thegoat: Magic is what people who can't see whats going on call it :) |
| 04:20.56 | Orbixx | All they asked for was a username, password and proxy. |
| 04:21.09 | thegoat | ;-) |
| 04:21.36 | thegoat | i just couldn't understand, the info was in the sip headers, but now showing up in the phones.... |
| 04:22.12 | [TK]D-Fender | Orbixx: And what'd you give them? |
| 04:22.50 | Orbixx | Exactly that. |
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| 04:31.12 | javb | [TK]D-Fender, it sounds interesting the "intercome topic" .. i i have polycom's 550 and Cisco 7940G (running SIP) .. any info? |
| 04:31.48 | thegoat | i found this : http://www.voip-info.org/wiki/view/Cisco+7940-7960+auto-answer+config |
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| 04:35.38 | [TK]D-Fender | javb: Go look up "paging" on the WIKI |
| 04:37.09 | javb | sure i will.. nice talking about the topic. |
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| 04:49.17 | thehar | jetsnoc: oh hai |
| 04:49.31 | jetsnoc | oh hai how are youuu |
| 04:49.45 | thehar | super weird seeing your hostmask be sagonet |
| 04:49.58 | jetsnoc | i usually troll in here and help but for some reason my bnc wasn't identified to nickserv |
| 04:50.00 | thehar | prepping pbx02 full switchover next monday |
| 04:50.06 | jetsnoc | yuck! |
| 04:50.09 | thehar | it's time! |
| 04:50.33 | jetsnoc | ya are you feeling pretty good about it |
| 04:50.42 | thehar | i think so |
| 04:51.01 | thehar | i switched the pri twice .. smashed those few bugs.. pretty sure it should be smooth |
| 04:51.10 | thehar | s/twice/twice during testing/ |
| 04:51.31 | jetsnoc | cool! |
| 04:51.37 | thehar | si |
| 04:51.54 | jetsnoc | i'm doing another big move for syringa wireless next week |
| 04:52.11 | thehar | aren't you at kickback yet? |
| 04:52.23 | jetsnoc | ya on the payroll at both places :) |
| 04:52.29 | jetsnoc | started at kb tuesday |
| 04:52.36 | jetsnoc | i have an upgrade night the 27th |
| 04:52.49 | thehar | ah nice |
| 04:53.12 | jetsnoc | moving 30k mailboxes to app_pmtvm |
| 04:53.15 | thehar | ew |
| 04:54.41 | jetsnoc | ya i'm certain i'll find out how good of a product it is |
| 04:56.17 | jetsnoc | ;) |
| 04:56.25 | jetsnoc | i created nomadtricks after jesse's joke |
| 04:56.27 | thehar | what additional features did you build into app_pmtvm. i've forgotten |
| 04:57.05 | jetsnoc | xml based button mapping per voicemail context....smdi to multiple switches |
| 04:57.21 | jetsnoc | cool web gui (roomtemperature wrote the php for that0 |
| 04:57.22 | frk2 | hey all - anybody been using the new chinese ATCOM phones recently? they used to be crap in the past- wonder if they have gotten any better |
| 04:57.50 | thehar | ah nice |
| 04:58.03 | [TK]D-Fender | frk2: Even thinking about them raises questions |
| 04:59.07 | frk2 | [TK]D-Fender, hahah really? are they still bad? |
| 04:59.35 | frk2 | problem is that my smaller third world clients cant even think of polycoms / ciscos |
| 05:03.11 | frk2 | my neverending quest for a cheap phone |
| 05:05.05 | [TK]D-Fender | frk2: Or as long as your clients don't care they'll probably mostly work. |
| 05:05.25 | [TK]D-Fender | frk2: In far less refined way. |
| 05:07.03 | *** join/#asterisk errotan (n=errotan@62.201.123.235) |
| 05:07.25 | frk2 | [TK]D-Fender, my only problem with them was the phone freeze and random echo. |
| 05:07.38 | frk2 | [TK]D-Fender, have you used one recently? |
| 05:07.40 | [TK]D-Fender | frk2: fun... |
| 05:07.51 | [TK]D-Fender | frk2: are you on crack? :p |
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| 05:08.26 | frk2 | [TK]D-Fender, ? :) using an atcom aint THAT bad lol |
| 05:08.38 | thehar | & |
| 05:09.29 | [TK]D-Fender | frk2: No, not as bad as the though of PAYING for one :D |
| 05:09.51 | frk2 | hahah i understand :) |
| 05:10.08 | frk2 | man i want a review from someone who has used them. they are not as bad as say grandstreams |
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| 05:10.30 | [TK]D-Fender | frk2: I would bet the reverse |
| 05:10.55 | frk2 | you'd think right? |
| 05:11.08 | [TK]D-Fender | frk2: GS pumps out enough volume and has the kind of client base to at least SLIGHTLY kick them in the ass to make them better. |
| 05:11.59 | [TK]D-Fender | frk2: Atcom used generic PA168/1688 chips in theirs. The lowest reference phone design ever. They didn't even try. |
| 05:12.23 | frk2 | [TK]D-Fender, but now they are using infineons and broadcoms |
| 05:13.02 | frk2 | [TK]D-Fender, it sucks as its IMPOSSIBLE to compete with competing seimens and Avaya analog telephony in the third world |
| 05:13.10 | frk2 | price-wise |
| 05:13.24 | frk2 | due to the price of IP phones and gateways - they become much cheaper |
| 05:13.57 | [TK]D-Fender | frk2: So go analog |
| 05:14.13 | frk2 | with gateways? |
| 05:14.32 | [TK]D-Fender | frk2: They do |
| 05:14.43 | frk2 | even thats super expensive. the only gateways I can get are quintums. grandstream gateways really suck :) |
| 05:15.29 | frk2 | [TK]D-Fender, im just trying to increase my client base here :) |
| 05:15.43 | frk2 | but cant deliver them freezing, coughing, echoing phones |
| 05:19.22 | *** join/#asterisk mintos (n=mvaliyav@209.132.188.254) |
| 05:20.43 | [TK]D-Fender | Ok, hceckout time, later all |
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| 06:00.47 | nextime | Hello all. Asterisk 1.6.1. I need to do this ( all is with sip channels ): friend A try to call friend B. If B is available, nothing strange. If B isn't available ( isn't registered to my * server ), i will call it through the C termination setting a specific callerid, but just to send 1 ring to it, it receive the ring, come online and register to the server, so i can call it directly. |
| 06:01.23 | nextime | how can i say to asterisk to dial an extension but to hangup after the first ring and to go next with dialplan? |
| 06:03.22 | *** join/#asterisk arcanine18_2000 (n=saxon_m2@121.58.209.130) |
| 06:03.46 | arcanine18_2000 | hi |
| 06:05.06 | arcanine18_2000 | help pls... i have encountering one-way voice |
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| 06:05.35 | arcanine18_2000 | am using 4 digium 8-port fxo card |
| 06:06.10 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
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| 07:11.47 | WeazelON | is there a way to check what is exipre time of sip registration from asterisk without seeing the countdown - like the real number itself ? |
| 07:13.39 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:13.44 | arcanine18_2000 | help pls... i have encountering one-way voice |
| 07:13.56 | arcanine18_2000 | am using digium 8-port fxo card |
| 07:15.06 | WeazelON | arcanine18_2000: have you checked volume ? (sometimes the simplest things can surprise all) |
| 07:18.00 | arcanine18_2000 | already done w/ the volume.. |
| 07:18.19 | arcanine18_2000 | and still the same |
| 07:29.19 | WeazelON | need to know a bit more on how things are connected to the card, and phones to pbx and which side is the problem |
| 07:35.05 | tzafrir_laptop | arcanine18_2000, from sip? |
| 07:35.17 | tzafrir_laptop | what happens if you call to asterisk directly? |
| 07:35.26 | tzafrir_laptop | e.g. to voicemail |
| 07:35.37 | tzafrir_laptop | or to Echo |
| 07:36.52 | WeazelON | tzafrir_laptop: good morning :D |
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| 08:32.04 | yxa | hi I just downloaded and make asterisk 1.4.26 but realized there is no chan_zap. what gives? |
| 08:39.49 | TommyBJ | They have changed their API |
| 08:39.57 | TommyBJ | It's now called Dahdi |
| 08:40.08 | TommyBJ | It's more or less the same. |
| 08:40.22 | TommyBJ | They changed it after 1.4.21 |
| 08:41.24 | yxa | oh |
| 08:43.17 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 08:43.37 | yxa | TommyBJ thanks for the update. i must have missed it |
| 08:50.22 | yxa | TommyBJ i dont get a /etc/dahdi directory like the docs say. what did i miss? |
| 08:56.28 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 08:58.29 | dwery | I'm trying to hide the outgoing caller id with Set(CALLERPRES()=prohib) but it seems it does not work. according to the telco the service is enabled. I have usecallingpres=yes and the DAHDI channels are those of an astribank 4xBRI. What else can I check? |
| 08:58.54 | *** join/#asterisk stix (n=stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 08:59.59 | stix | Hi guys. I have been digging in to this, but I cannot figure out if it is possible to put a call on hold via AMI. Can anyone tell me? I can park the calls but that is not exactly what I want. |
| 09:04.34 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 09:07.09 | creativx | stix then how do you define put on hold |
| 09:07.22 | creativx | isnt that the same as parking a call at some extension |
| 09:07.29 | creativx | and when taking off hold, picking up that call |
| 09:07.51 | stix | I guess when you park the call, the user who parks it, will get a free line |
| 09:08.25 | stix | but how can I take a call back from the parking lot? The only thing defined is timeout. |
| 09:09.03 | *** join/#asterisk unasi7 (n=unasi7@80-218-32-110.dclient.hispeed.ch) |
| 09:15.33 | creativx | i hacked up some solution |
| 09:15.50 | creativx | where i created dummy extenions that just play music indefinetly |
| 09:16.01 | creativx | and then using our gui the user picks up the call when they are done |
| 09:16.43 | maxagaz | where to get logs of this channel ? |
| 09:21.22 | *** join/#asterisk kiwi_ (n=kiwi@ks359129.kimsufi.com) |
| 09:26.48 | stix | creativx: so you redirect the channel to an extension with music. But after the redirect, your user is available to receiving new calls? |
| 09:31.35 | creativx | stix: yes |
| 09:32.14 | stix | not exactly what I want, but I get the idea with the dummy extensions |
| 09:32.28 | creativx | and the user gets a warning after 10 minutes if they have forgotten someone in the "park" extension |
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| 09:35.58 | skirmisha | guys |
| 09:36.16 | skirmisha | anyone familar with sip_poke_peer |
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| 09:51.35 | skirmisha | anyone alive here |
| 09:51.58 | vicscandl | yes, but total n00b, so i'm of no help. sorry :( |
| 09:51.59 | skirmisha | can someone tell me what is asterisk sending to sip phone to check the status |
| 09:52.27 | skirmisha | i am on second n00b stage :0 |
| 10:04.55 | *** join/#asterisk user10 (n=user10@125.99.106.189) |
| 10:05.18 | user10 | how do i stop FOP asterisk and all related things which is getting started without caring for my permission |
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| 10:22.07 | dwery | when pri debug says 'Presentation: Presentation prohibited of network provided number (35)' an I reasonably be sure that the problem is on the telco side? |
| 10:22.22 | dwery | (caller id is still shown on the called party) |
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| 10:48.27 | tzafrir_laptop | dwery, what version of asterisk? |
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| 10:52.26 | dwery | tzafrir_laptop: 1.6.0.1 |
| 10:52.39 | dwery | tzafrir_laptop: no, 1.6.1.0 |
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| 10:55.03 | dwery | tzafrir_laptop: the SETUP message looks fine to me,calling presentation is 0x35 |
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| 11:35.58 | VooDooNOFX | I'm considering 3cx for my setup, wanted to now if this would be possible: |
| 11:36.42 | VooDooNOFX | A single Asterisk install handling multiple companies calls. Allowing ext 101@companyA, and 101@companyB to share the same asterisk server, with seperate outbound sip trunks for each |
| 11:37.12 | afink | absolutely |
| 11:38.00 | TommyBJ | VooDooNOFX: Just know that there are potential security risks.. as well as some features might be hard to separate |
| 11:38.00 | VooDooNOFX | sweet :) that one answer just started my new business. lol |
| 11:38.52 | VooDooNOFX | most of my clients use Packet8 or vonage now, but i'm sooo tired of them, and want to implement something else where I can become the phone co for them. |
| 11:40.11 | TommyBJ | VooDooNOFX: Why on earth would you consider 3cx? :p |
| 11:40.30 | afink | VooDooNOFX: As long as you are careful with your contexts which ones are included where and separate them the risks really aren't too much. |
| 11:41.06 | VooDooNOFX | excellent. I'm well versed in linux and it's related applications' security, so i'll make sure to be careful with it. |
| 11:41.23 | VooDooNOFX | I ran make progdocs, but where does it place them? |
| 11:42.10 | afink | in the manual |
| 11:42.30 | afink | man asterisk |
| 11:42.39 | VooDooNOFX | ah, i'm used to it making doxys html files :D |
| 11:44.01 | afink | VooDooNOFX: just out of curiousity can 3cx not do what you asked? |
| 11:44.17 | VooDooNOFX | not that i've been able to make it do. |
| 11:44.34 | afink | I laugh every time I see their ads about how you should use a windows based system. Give me a freakin break |
| 11:44.36 | VooDooNOFX | 3cx wants to setup extensions for each client as if they were the same company |
| 11:45.06 | VooDooNOFX | but I want 1 server to handle multiple clients, and have 2 users with extension 101, but in different organizations |
| 11:45.25 | TommyBJ | 3cx is meant to "break from the proprietary world of PBX" ... with yet another proprietary product. |
| 11:46.05 | VooDooNOFX | 3cx's api is sub-par, their software is expensive, and the auto-dialers are tons of money |
| 11:46.25 | afink | right now I have 6 companies on 1 asterisk server they don't have similar extensions but as long as your users(extens) are in the right context you'll be fine. |
| 11:46.53 | VooDooNOFX | can you describe "context" more? |
| 11:47.06 | afink | have you read the book? |
| 11:47.09 | afink | ~book |
| 11:47.10 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 11:47.39 | afink | ~context |
| 11:47.39 | infobot | somebody said context was like LaTeX but less messy and more oriented to DTP instead of academics. |
| 11:47.50 | VooDooNOFX | nope, yesterday I installed PIAF, but prefer to learn each component individually. |
| 11:48.07 | afink | context is a container of sorts that organizes your dialplan |
| 11:48.24 | TommyBJ | It is a recursive group of extensions. May be included within another |
| 11:49.02 | VooDooNOFX | i'll make sure to buy the book today |
| 11:49.12 | afink | ^^ its free online |
| 11:49.19 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
| 11:49.31 | VooDooNOFX | I prefer cutting down trees. My eyes appreciate it more :D |
| 11:49.37 | TommyBotten | hehehe |
| 11:50.06 | afink | btw, I'm not sure if you are considering a GUI or not, but with what you want to do it would be EXTREMELY difficult with a GUI. |
| 11:50.30 | afink | I just converted mine over last weekend and it is like I finally saw the light |
| 11:50.32 | VooDooNOFX | yeah, I figured. GUI's usually just gum up conf's anyway |
| 11:50.37 | afink | shit works now |
| 11:51.13 | afink | have you seen a complete extensions.conf? if not I would be happy to show you mine |
| 11:51.17 | afink | in a pm |
| 11:51.24 | VooDooNOFX | I saw the samples one, that's it |
| 11:57.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 11:57.38 | VooDooNOFX | Also, to setup faxes for them, is there a windows/linux based GUI fax client for paperless faxes without a fax machine? |
| 12:01.15 | afink | I haven't used any. When we got asterisk I switched our fax lines back over to pots lines b/c they work so much better, but we have myfax.com for the extremely lazy people who want to fax from their desk |
| 12:01.18 | dwery | VooDooNOFX: hylafsp works with hylafax |
| 12:01.53 | dwery | VooDooNOFX: it's a windows based cliente. you can integrate hylafax with asterisk using iaxmodem |
| 12:03.30 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:06.37 | TommyBotten | Or SMTP -> hylafax for a more generic solution |
| 12:06.58 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:07.16 | VooDooNOFX | so SMTP -> hylafax -> T.38 -> SIP trunk for outbound faxes? |
| 12:07.23 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:07.46 | dwery | VooDooNOFX: fortet VOIP when you are faxig if you want to avoid headaches l( |
| 12:07.49 | dwery | :D |
| 12:08.30 | VooDooNOFX | I gave up on faxes a long long time ago. My clients however, deal with banks, and are only allowed to fax in papers |
| 12:08.54 | dwery | Just use real lines |
| 12:08.57 | VooDooNOFX | still waiting for the real estate market to get up to speed, although RETS is a good start |
| 12:08.59 | afink | ^^ |
| 12:09.15 | dwery | T.38 might work but it's not easy to setu. Soif you are in an hurry it's better to do it in the old way |
| 12:09.48 | VooDooNOFX | yeah, concern #1 was to setup and secure asterisk. Then i'll setup faxing later |
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| 12:17.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:23.29 | khussein78 | hi |
| 12:23.45 | *** join/#asterisk odenkos (n=odenkos@85-135-173-31.adsl.slovanet.sk) |
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| 12:25.28 | ariel_ | Morning |
| 12:27.32 | khussein78 | transfered calls from one exten to another not recorded? |
| 12:27.51 | khussein78 | how can i fix this problem ? |
| 12:29.12 | TommyBotten | You need to initiate recording earlier in the diaplplan |
| 12:29.15 | TommyBotten | dialplan |
| 12:29.58 | leifmadsen | khussein78: also, which application you use can make a difference -- you may need to use AUDIOHOOK_INHERIT() dialplan function to tell the system to keep recording beyond a transfer |
| 12:30.09 | leifmadsen | Monitor() shouldn't need that, but MixMonitor() will |
| 12:39.42 | *** join/#asterisk aethelrick (n=richard@mail.mexuar.net) |
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| 12:46.09 | *** join/#asterisk voxta (n=voxta@59.97.224.243) |
| 12:46.16 | voxta | hi |
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| 12:49.54 | voxta | hi |
| 12:50.47 | aethelrick | does anyone know what causes a "utils.c write() Broken pipe error" ?? |
| 12:51.12 | aethelrick | I'm seeing this in /var/log/asterisk/mesages |
| 12:52.11 | aethelrick | this is on a box running astmanproxy, and asterisk. I suspect that it may be something to do with AMI but I'm not sure |
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| 12:58.04 | *** part/#asterisk N3tw0rk (n=kris@r74-192-40-158.vctrcmta01.vctatx.tl.dh.suddenlink.net) |
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| 13:00.03 | jaytee | telnettech, morning Brian |
| 13:00.14 | telnettech | mornng....how are you |
| 13:01.35 | ariel_ | aethelrick: have you check this out? https://issues.asterisk.org/view.php?id=15354&nbn=5 |
| 13:01.48 | *** join/#asterisk Iamnach0 (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 13:04.40 | aethelrick | ariel: thanks I'll have a look |
| 13:05.12 | *** join/#asterisk tfrew|afk (n=tfrew@office.neteasyinc.com) |
| 13:06.05 | grandpapadot | Hey guys, I'm connecting peers from a remote asterisk box and dialing them as members of a queue defined as "SIP/801@remotepeer". It works the first time just fine, but then the member just stays in a "ringing" state until I reload app_queue, any ideas? |
| 13:06.47 | *** join/#asterisk N3tw0rk (n=kris@r74-192-40-158.vctrcmta01.vctatx.tl.dh.suddenlink.net) |
| 13:07.20 | N3tw0rk | anyone know why i cant get two diffrent ext. registered to a single polycom phone |
| 13:07.31 | grandpapadot | N3tw0rk: Nope. |
| 13:08.24 | telnettech | N3tw0rk: Watch the CLI and restart device. Asterisk will show the registration process and show why it wasnt successful |
| 13:09.30 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:09.31 | Katty | mew. |
| 13:09.55 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:10.25 | [TK]D-Fender | Katty: Mew. |
| 13:10.43 | N3tw0rk | telnettech: ya it will register the last extension tried but not both, says have <second> but digest has <first> |
| 13:11.54 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 13:12.15 | ariel_ | heading to the lab, hope not to get dropped off the network. Brb |
| 13:14.52 | ThoMe | hello |
| 13:15.04 | ThoMe | is it posible in ael syntax test how long is a string? |
| 13:15.06 | ThoMe | LENGTH ? |
| 13:16.01 | [TK]D-Fender | ThoMe: "core show functions like LEN" |
| 13:16.10 | ThoMe | [TK]D-Fender: merci |
| 13:16.30 | [TK]D-Fender | ThoMe: bienvenue |
| 13:16.53 | *** join/#asterisk bmg505 (n=leon@196-209-79-155-rndf-esr-5.dynamic.isadsl.co.za) |
| 13:17.09 | ThoMe | :-) |
| 13:17.25 | VooDooNOFX | What SIP providers do people in here use in the US? |
| 13:18.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:18.05 | [TK]D-Fender | ~itsplist-us |
| 13:18.06 | infobot | well, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 13:18.07 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 13:18.39 | VooDooNOFX | Anyone have + or - experience with SoTel? |
| 13:23.27 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:24.48 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
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| 13:28.27 | ThoMe | [TK]D-Fender: emm |
| 13:28.41 | ThoMe | [TK]D-Fender: i try a IF but doesnt works: |
| 13:29.04 | ThoMe | http://paste.keks.be/562 |
| 13:29.18 | ThoMe | but is always: |
| 13:29.19 | ThoMe | Zwischen 3 und 5 > 3 |
| 13:29.27 | ThoMe | Zwischen 3 und 5 > 14 |
| 13:30.11 | ThoMe | but 14 is higher as 5.. |
| 13:31.00 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:31.21 | [TK]D-Fender | ThoMe: don't use "", and I don't trust the CONTENTS of that variable. We should see those first AND the output of that function before your IF |
| 13:31.49 | ThoMe | hmmm, do you have an example para mi? |
| 13:31.50 | ThoMe | for me? |
| 13:32.46 | *** join/#asterisk propellerhead (n=yogurt2u@host35.190-137-10.telecom.net.ar) |
| 13:33.47 | ThoMe | ah, [TK]D-Fender, this works now: if (${LEN(${absender})} <= 5) { |
| 13:34.25 | [TK]D-Fender | ThoMe: This is not text you're comparing, and neither side will be NULL. |
| 13:34.25 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 13:34.26 | *** join/#asterisk Amorsen (n=Amorsen@94.127.50.7) |
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| 13:34.41 | VooDooNOFX | ok, thanks for the advice so far guys. I'm sure i'll bbl with more. night |
| 13:34.42 | Amorsen | I found this in queues.conf: |
| 13:34.45 | Amorsen | <PROTECTED> |
| 13:34.55 | Amorsen | What the heck is '33'? |
| 13:35.48 | [TK]D-Fender | Amorsen: 32 + 1 |
| 13:36.09 | Amorsen | Great help, [TK]D-Fender |
| 13:36.12 | ThoMe | [TK]D-Fender: i dont understand. what is better? |
| 13:36.33 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
| 13:36.36 | Amorsen | Nothing more lovely than randomly sprinkled constants |
| 13:37.01 | [TK]D-Fender | ThoMe: Look at what you're comparing.. 2 NUMBERS. you don't use "" when you are guranteed 2 numbers on either side of an operator and are doing a math operator. |
| 13:37.26 | [TK]D-Fender | ThoMe: As soon as you shove non-numeric chars in like " it gets in the way |
| 13:38.11 | [TK]D-Fender | Amorsen: while (*state_interface && *state_interface < 33) state_interface++ <--- non-zero and less than 33 = iterate |
| 13:38.29 | Amorsen | Right, but why 33? |
| 13:38.31 | N3tw0rk | i changed an exten for a phone and i keep getting an authen error..saying the digest still has the old exten...how can i clear it so it will re-authen |
| 13:38.41 | [TK]D-Fender | Amorsen: Because there is a limit of 32 |
| 13:39.04 | Amorsen | A state interface can be a maximum of 32 characters? |
| 13:39.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:39.35 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 13:39.37 | drfreeze | Anyone have recommendations for a PC that will support a Channel 12. I'm considering either the TE121 or TE122 and want the PC to handle the echo cancelation |
| 13:40.23 | jplank | does this make sense to anyone, I have a PRI with XO and calls are dropping. XO is saying that 40 seconds into the conversation the call is getting a suspend and if there is no resume after 6 seconds the call is dropped, and that I need to have the suspend resume timer pushed to 10 seconds or greater |
| 13:41.41 | [TK]D-Fender | drfreeze: ... "Channel 12"? |
| 13:41.44 | drfreeze | [TK]D-Fender: It's what twtelecom calls their fractional 12 with 12 channels, no data |
| 13:41.57 | drfreeze | 11 lines are available for phone, 1 for control |
| 13:42.09 | [TK]D-Fender | Any P4 class PC is more than enough |
| 13:42.13 | drfreeze | *fractionl T! |
| 13:42.23 | drfreeze | can't type this morning. Fractional T1 |
| 13:42.43 | drfreeze | what about TE121 or TE122. Does that matter? |
| 13:44.46 | [TK]D-Fender | drfreeze: Yeah... buying a TE121 for a system without PCI-E slots would be... DUMB :p |
| 13:44.57 | drfreeze | :P |
| 13:45.06 | [TK]D-Fender | drfreeze: And the reverse for the TE122 |
| 13:45.16 | drfreeze | :PP |
| 13:46.11 | drfreeze | Ok, which Version of Asterisk should I consider? 1.6 or 1.4 |
| 13:46.33 | drfreeze | I've done all of my coding in 1.4 and I hear there are some new features with 1.6 |
| 13:46.35 | *** join/#asterisk dymaxion (n=dymaxion@79.123.75.68) |
| 13:47.02 | Amorsen | The new features primarily working T.38, but even that is getting a backport now |
| 13:47.21 | Amorsen | That said, we're on 1.6.1.1+patches for new installs |
| 13:47.30 | drfreeze | I mainly need a stable system with call parking |
| 13:47.32 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 13:47.43 | leifmadsen | drfreeze: if you don't need new features, stick with what you know |
| 13:47.56 | leifmadsen | drfreeze: that being said, I've had some good luck with 1.6.x releases on a couple of clients |
| 13:48.02 | leifmadsen | ~asteriskversioning |
| 13:48.03 | infobot | it has been said that asteriskversioning is Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
| 13:48.09 | leifmadsen | that is probably a useful article for you |
| 13:48.22 | *** join/#asterisk ehsjoar (n=ehsjoar@24.9.91.203) |
| 13:48.26 | drfreeze | leifmadsen: thanks |
| 13:51.16 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 13:53.00 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:53.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:54.27 | *** join/#asterisk bsteele (n=bbsteele@lb1.ihets.org) |
| 13:54.45 | bsteele | Hi, I |
| 13:55.15 | bsteele | I'm an intern for the company I work for and they have me setting up a test asterisk platform. I have asterisk installed and have been doing research on setting up the configuration files |
| 13:55.31 | *** join/#asterisk ingenius (n=alektro@host95.190-229-175.telecom.net.ar) |
| 13:55.38 | drfreeze | Ok, with the support of T.38 in 1.6, does it really work? I mean, can I put a fax machine on a 1.6 system in a produciton environment? |
| 13:56.56 | bsteele | Will I need zaptel / digium cards for this to work? A lot of configurations I've seen use those |
| 13:57.10 | lirakis | hrm ... i seemed to recently encountered a problem where my sip calls die after 20 seconds with "chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission" |
| 13:57.41 | lirakis | i am nat'd, and have that set as a global in my sip.conf (unless overriden) |
| 13:58.16 | [TK]D-Fender | bsteele: for 8 to work? No. For your intended implementation, maybe |
| 13:58.22 | [TK]D-Fender | ~sipnat |
| 13:58.23 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:58.25 | [TK]D-Fender | lirakis: ^^^^^^ |
| 13:58.33 | [TK]D-Fender | bsteele: s/8/*/ |
| 13:58.52 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 13:59.05 | drfreeze | Amorsen: have you used the fax features of 1.6? |
| 14:00.38 | bsteele | thanks |
| 14:00.39 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 14:00.43 | bsteele | what do u mean by s/8/* btw? |
| 14:04.23 | *** join/#asterisk Iamnach0 (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 14:04.52 | lirakis | [TK]D-Fender: just to clarify - my asterisk server is not behind a nat - just the phones. |
| 14:05.23 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 14:05.40 | [TK]D-Fender | lirakis: relevant bits are in there. |
| 14:05.45 | Amorsen | drfreeze: No, we use Hylafax |
| 14:05.58 | [TK]D-Fender | bste|afk: typo'd 8 instead of * |
| 14:06.02 | Amorsen | drfreeze: But I have used T.38 gateway functionality, it works nicely |
| 14:06.16 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 14:06.22 | Amorsen | As well as T.38 passthrough |
| 14:07.03 | drfreeze | Amorsen: does that require a more powerful processor to support that? |
| 14:08.06 | Amorsen | drfreeze: I doubt it, our PRI gateway is a dual-CPU 32-bit Xeon box... |
| 14:08.14 | Amorsen | Not exactly state of the art |
| 14:10.07 | drfreeze | ah. I need to read up more on this. It sounds like the fax signal ultimately leaves the office via analog, but I would prefer to send it out on PRI |
| 14:10.40 | Amorsen | drfreeze: analog? |
| 14:11.01 | Amorsen | drfreeze: We only have SIP interconnects and PRI-lines |
| 14:11.22 | drfreeze | one article was saying that the gateway converts the signal back to analog before sending out |
| 14:11.27 | Amorsen | Well yes |
| 14:11.31 | Amorsen | That's how faxes work |
| 14:11.40 | drfreeze | what if you only have PRI out? |
| 14:12.00 | *** join/#asterisk LemensTS (n=customgt@207.160.41.191) |
| 14:12.13 | LemensTS | hey all, how do you issue an asterisk cli cmd from the dialplan? |
| 14:12.44 | Amorsen | Analog just means it encodes it into 8kHz Alaw or ulaw |
| 14:13.38 | Amorsen | Whether you choose to transport the 8kHz Alaw or ulaw through PRI or through oldfashioned analog copper wire is up to you |
| 14:14.45 | Amorsen | Even if you send it out through analog copper wire it's going to be turned (lossily) back into Alaw or ulaw at the public exchange anyway |
| 14:14.57 | jplank | is there any way to adjust PRI timers? |
| 14:15.18 | *** join/#asterisk dymaxion (n=dymaxion@79.123.75.68) |
| 14:15.25 | [TK]D-Fender | LemensTS: System() |
| 14:16.31 | Amorsen | I hate when I can't reproduce a bug |
| 14:17.53 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:20.12 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 14:30.26 | lirakis | I honestly think that some one !@#% up the network im on .... :\ ill have to do more testing, but im not getting any different results with changes ive tried |
| 14:30.45 | lirakis | the office IT crew is ... uh... not stellar |
| 14:31.41 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:34.03 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
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| 14:36.37 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:39.57 | *** join/#asterisk BigM (n=michel@nl06sr01.targetmedia.nl) |
| 14:40.38 | *** join/#asterisk Failrar (n=Failrar@2001:888:1440:0:2a0:d1ff:fe4e:e802) |
| 14:41.03 | BigM | hi there |
| 14:41.52 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 14:42.08 | BigM | can some help me out with a question about the notation exten => ? |
| 14:42.18 | jetsnoc | Shoot |
| 14:42.20 | jetsnoc | we'll answer! |
| 14:42.50 | BigM | Iḿ just started with asterisk but I cee everywhere "exten => " How do I need to read that? |
| 14:43.24 | BigM | like: exten => s, 1, Verbose (1|Unrouted call handler) |
| 14:43.46 | BigM | No idea how I had to read that.. |
| 14:43.46 | leifmadsen | BigM: you need to read about dialplan basics |
| 14:43.46 | leifmadsen | BigM: http://astbook.asteriskdocs.org |
| 14:44.36 | BigM | can you recommand a chapter? |
| 14:44.46 | BigM | or just look for dialplan in the apendex? |
| 14:44.51 | leifmadsen | exten => is the prefix marker for a line of dialplan (a line of "doing something"), where 'exten' is short for 'extension'. However, for more information about how dialplans work, and asterisk in general, you will need to do some reading |
| 14:45.07 | leifmadsen | BigM: there are 2 big chapters on dialplans in the book -- just look at the table of contents and you'll see it |
| 14:45.17 | [TK]D-Fender | BigM: ... |
| 14:45.19 | [TK]D-Fender | ~book |
| 14:45.20 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:45.22 | [TK]D-Fender | ^^^ chapter 5 |
| 14:45.25 | leifmadsen | [TK]D-Fender: I already gave him a link |
| 14:45.50 | [TK]D-Fender | BigM: Asterisk = 95% Dialplan. Master it... or else. |
| 14:46.09 | BigM | lol, I gonna read :) |
| 14:47.04 | BigM | lol before chapter five they use => and in chapter 5 they explain what it is.. xD |
| 14:47.18 | BigM | but thnx all :D |
| 14:51.52 | drmessano | They do it that way on purpose |
| 14:52.21 | drmessano | So you dont go back after reading two chapters on dialplan and ask shit like "Whats a SIP" and "Do I just plug my phone into my NIC?" |
| 14:52.23 | *** part/#asterisk JohnTeddy (i=unstable@tor/regular/sid) |
| 14:52.38 | drmessano | Apparently thats what IRC is for |
| 14:54.40 | BigM | I read it a serveal times when I was bussy with other scripts |
| 14:54.48 | BigM | with the agi class |
| 14:55.09 | drmessano | "Hi!!!!!!!!!!!!! (obnoxious greeting) I am new to Asterisk (oh god, here we go), and I just finished staring at the Starfish on the cover of the book for 2 or 3 and hours and was wondering.. When I open the gz file in Vista it opens in Incredimail. How do I make a call from there?" |
| 14:55.34 | BigM | windows o///// |
| 14:55.45 | drmessano | boo yah |
| 14:55.58 | BigM | go and wash your mouth ;) |
| 14:56.13 | leifmadsen | drmessano: press the button in the bottom left corner, then click on "Shut Down". Upon reboot, stick in a Linux disk, format your hard drive, install Linux, and start with Chapter 3 after that. |
| 14:56.17 | leifmadsen | likes Windows |
| 14:56.22 | drmessano | Asterisk tarball opening in Incredimail in Vista.. Only two words... "SUP BRO!!!" |
| 14:56.22 | leifmadsen | I just don't use it :) |
| 14:56.31 | drmessano | lol |
| 14:56.51 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:56.57 | ariel_ | vista argh, not on my systems.... |
| 14:56.57 | coppice | I like windows. the room's kinda gloomy without them |
| 14:57.18 | drmessano | I just finished switching my BIOS to native SATA vs IDE emulation and then making Windows happy with the change.. only took me 11 hours |
| 14:57.34 | coppice | Windows 7 Customers 0 |
| 14:58.09 | [TK]D-Fender | coppice: http://punditkitchen.files.wordpress.com/2009/02/political-pictures-bill-gates-campaign-fail.jpg |
| 14:59.26 | drmessano | So all we need is a touchdown and a 2-point conversion to beat Windows 7? |
| 14:59.28 | drmessano | LET DO IT |
| 14:59.37 | drmessano | LETS DO IT* |
| 14:59.54 | BigM | exten => 123, 1, Answer() << use function 123 of the application/command Answer() or am I not right? |
| 15:00.14 | drmessano | Not right at all |
| 15:00.21 | BigM | :/ |
| 15:00.53 | drmessano | That would be extension 123 |
| 15:01.13 | BigM | use the function 123 of the extension 123.. |
| 15:01.21 | BigM | use the function Answer() of the extension 123.. |
| 15:01.58 | coppice | [TK]D-Fender: Rich Hall has a nice monologue about "How can a man with $88B not smile? What does he think? I gotta find myself another $10B and turn my sorry life around" |
| 15:02.00 | drmessano | Are you writing an instruction manual for home electronics? |
| 15:02.13 | drmessano | extension, priority, function |
| 15:02.34 | drmessano | Well technically, extension, priority, command (AKA stuff to do) |
| 15:02.55 | drmessano | exten => 123,1,Answer() |
| 15:03.01 | drmessano | exten => 123,2,DoStuff() |
| 15:03.11 | drmessano | exten => 123,3,DoMoreStuff() |
| 15:03.26 | drmessano | exten => 123,4,Hangup() |
| 15:03.36 | drmessano | or whatever |
| 15:04.11 | BigM | as I sad do I had to read that like: use the DoStuff() of extension 123 or what can I imagine by 123... |
| 15:05.21 | drmessano | Youre not referencing the extension.. that IS the extension |
| 15:05.23 | [TK]D-Fender | BigM: 123 is what you DIALED. the 1,2,3,4 are the STEPS it will go through. Answer(), etc are the ACTIONS it will take |
| 15:06.46 | BigM | so it could 1257692 ? |
| 15:06.55 | BigM | could be* |
| 15:07.00 | *** join/#asterisk ArchGT (n=ArchGT@190.148.91.196) |
| 15:07.18 | drmessano | It can be anything that you need/want |
| 15:07.29 | BigM | okay ^^ |
| 15:07.37 | *** join/#asterisk Airozi (n=user@200-171-41-207.dsl.telesp.net.br) |
| 15:07.55 | drmessano | That is you making an extension.. this isnt some macro code that goes with the callflow of the system.. youre defining that right NOW |
| 15:07.58 | [TK]D-Fender | BigM: You are configuring what you want * to do when you dial different #'s |
| 15:09.01 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 15:09.22 | drmessano | An extension isnt a box with a handset and buttons on someones desk... its something you define in Asterisk, whether its playing a poop sound, recording a memo, or sending a call to a SIP device and bridging a call through |
| 15:09.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) [NETSPLIT VICTIM] |
| 15:09.42 | BigM | okay Thatś clear than :) |
| 15:10.35 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) [NETSPLIT VICTIM] |
| 15:13.09 | *** join/#asterisk Carlos_PHX (n=carlos@ip68-3-162-244.ph.ph.cox.net) |
| 15:13.49 | drmessano | Great... There is a recall on Romaine Lettuce |
| 15:13.54 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 15:13.58 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 15:14.23 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 15:14.28 | drmessano | Now I get to hang out with all the low-life trash at the iceberg lettuce shelf |
| 15:14.30 | coppice | so, don't do as the Romaines do |
| 15:14.46 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 15:14.46 | [TK]D-Fender | drmessano: EW. |
| 15:14.52 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 15:15.01 | *** part/#asterisk Carlos_PHX (n=carlos@ip68-3-162-244.ph.ph.cox.net) |
| 15:15.44 | coppice | will global warming make the iceberg lettuce extinct? |
| 15:16.34 | *** join/#asterisk Laureano (n=Laureano@190.245.101.140) |
| 15:16.38 | drmessano | I hope not.. how are the kind folks at the tractor pull going to top off their burgers? |
| 15:16.42 | Laureano | Amorsen, ping |
| 15:18.10 | Xetrov` | i like burgers and tractor pulls! :( |
| 15:18.44 | drmessano | I could probably pick up some Boston Lettuce, but I feel like it's overly pretentious.. At least with Romaine I didn't have to feel low class, but yet I didn't feel overly stuck up except when sneering at rednecks picking up 98 cent heads of that awful iceberg |
| 15:18.48 | LemensTS | exec('rasterisk -x "file convert input.wav output.g729"'); and i get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?).... i looked into suexec but would rather not do that...is there any other PHP asterisk people here? just wondering how you dealt with this (yes i know #php...checking both places) |
| 15:22.08 | plundra | I have "joinempty = yes" and "leavewhenempty = no" set in my [general] section of queues.conf; but you can't add a caller to it, unless you add a member => to it. |
| 15:22.11 | plundra | Why is this? |
| 15:22.32 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 15:22.54 | *** join/#asterisk galeras (n=galeras@186.80.186.161) |
| 15:23.36 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 15:23.45 | plundra | All options is set in [general], and I have a few queues just defined with [foo_a], [foo_b] etc. No options set at all. And later I will add/remove memebers dynamically. |
| 15:24.03 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 15:25.43 | plundra | Hmm, I just tried adding joinempty = yes to a queue-definition and that worked too. So I guess you have to have SOME sort of setting, be it a static member, an option or whatever to your queue. |
| 15:26.41 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 15:26.57 | plundra | scratches his head. |
| 15:28.10 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:28.59 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 15:29.45 | *** part/#asterisk N3tw0rk (n=kris@r74-192-40-158.vctrcmta01.vctatx.tl.dh.suddenlink.net) |
| 15:30.28 | kn0x | any way to specify a specific music on hold file without having to use an MoH file? |
| 15:31.40 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:32.10 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
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| 15:32.31 | Slade- | hey shouldnt an agi script writing to stderr output in the asterisk console? |
| 15:32.35 | Slade- | or do i need to do something to enable that |
| 15:32.46 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 15:33.19 | [TK]D-Fender | kn0x: No. |
| 15:33.42 | [TK]D-Fender | Slade-: No, it shouldn't. If you want to output to console, thats what VERBOSE is for |
| 15:34.45 | Slade- | VERBOSE? |
| 15:34.58 | [TK]D-Fender | Slade-: You know.. the dialplan command <- |
| 15:35.05 | Slade- | nope dont know ;( |
| 15:35.21 | ManxPower | ~osmosis |
| 15:35.21 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 15:35.22 | Slade- | where does that go |
| 15:35.31 | Slade- | err where does stderr go |
| 15:36.00 | ManxPower | [TK]D-Fender: good morning. Looks like a typical day here. |
| 15:36.21 | ManxPower | Slade-: Are you working with AGI scripts? |
| 15:36.32 | [TK]D-Fender | Slade-: it goes to the AGI process You don't see it. If you want to see what * receives from your AGI then enable AGI DEBUG at * CLI |
| 15:37.16 | ManxPower | remember, AGI debug is pretty useless in Asterisk versions before 1.4 |
| 15:37.17 | *** join/#asterisk sjobeck (n=Adium@32.152.47.156) |
| 15:37.43 | Slade- | Fender: thats what i thoguht, but i dont even see stderr results in there either |
| 15:38.06 | *** part/#asterisk sjobeck (n=Adium@32.152.47.156) |
| 15:38.13 | Slade- | the scripts in the book and samples all see to use stderr tho so i'm sure i'm not looking in the right spot |
| 15:38.29 | ManxPower | Slade-: what version of Asterisk are you using? |
| 15:39.02 | Slade- | 1.4.24.1 |
| 15:39.52 | ManxPower | Slade-: agi debug should work then. In versions before 1.4.x you had to stop Asterisk, then start it with -c in order to keep asterisk in the foreground and see the agi stderr. |
| 15:40.09 | Slade- | i'm using asterisk -r to connect to console |
| 15:40.32 | ManxPower | Slade-: *nod* in versions before 1.4 you would never see stderr when using -r |
| 15:40.38 | ManxPower | I thought that was fixed in 1.4 |
| 15:44.28 | Slade- | the script i'm using came from the book.. http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-9-SECT-4.html |
| 15:45.57 | ManxPower | Slade-: maybe you need to file a report on bugs.digium.com |
| 15:46.44 | Slade- | yea i assume i'm doing something wrong not asterisk tho.. :P |
| 15:48.46 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:50.26 | galeras | please, tell me why Dial(Zap/g1/x||D(103)) sends DTMFs, however, Dial(Zap/g1/x||D(103)G(C^A^9))), jumps to C,A,2 without sending DTMFs. |
| 15:50.58 | galeras | i need to play DTMFS before to jump to C context |
| 15:51.27 | *** join/#asterisk gornati (n=cams@201.87.101.62) |
| 15:52.42 | gornati | hi all |
| 15:53.01 | galeras | Answer: because "You cannot use any additional action post answer options in conjunction with G()" :( |
| 15:54.43 | galeras | in an automated call, which is the best way to play DTMF tones and then play some recording? |
| 15:56.40 | *** part/#asterisk vicscandl (n=Christop@adsl-76-208-155-248.dsl.lsan03.sbcglobal.net) |
| 15:56.56 | gornati | question |
| 15:57.02 | gornati | help |
| 15:57.05 | gornati | -help |
| 15:57.33 | mog | local channels gornati |
| 15:57.45 | mog | one side does the dial other side executes normal dialplan |
| 15:57.50 | Qwell | gornati: ask your question.. |
| 15:57.56 | galeras | answer: Use sendDTMF with G within Dial, thank guys you are a geniuos |
| 15:58.25 | gornati | i was wondering if someone in here, have or had contact with ZRTP from Phil Zimmerman? |
| 15:58.31 | gornati | Zrtp Protocol |
| 15:58.58 | mercutioviz | gornati: yep, i've played with zrtp |
| 15:58.58 | kaii | i had, but i wonder what it has to do with asterisk |
| 15:59.11 | mercutioviz | but not with asterisk, only with freeswitch |
| 15:59.17 | Qwell | mog: hey :D |
| 15:59.27 | mog | hey Qwell |
| 15:59.34 | AlmightyOatmeal | i feel like calling canada just to see what a canadian talks like |
| 15:59.37 | gornati | i want to install it on asterisk to test the cripto. |
| 15:59.41 | Amorsen | Back now |
| 16:00.02 | [TK]D-Fender | AlmightyOatmeal: "A" canadian. Fully qualified statistic you're likely to find there ;) |
| 16:00.07 | Amorsen | Laureano ? |
| 16:00.16 | AlmightyOatmeal | :P |
| 16:00.55 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 16:01.07 | kaii | i chatted with a canadian last night via omegle |
| 16:01.28 | kaii | longest chat i had since years. 2 hours with a canadian girl stranger |
| 16:01.28 | mog | kaii, with someone who said they where from canada |
| 16:01.39 | mog | that was really me |
| 16:01.41 | kaii | i cannot know, right |
| 16:01.45 | kaii | lol |
| 16:01.47 | mog | ^_* |
| 16:01.49 | Kernel_Core | hi all |
| 16:02.05 | kaii | omegle is funny |
| 16:02.18 | mog | omegle needs some filters |
| 16:02.27 | kaii | f/m filter, yes |
| 16:02.28 | mog | like people looking to cyber need their own section |
| 16:02.35 | kaii | definitely |
| 16:02.39 | Laureano | Amorsen, I saw your issue 0015559. I added a note to it. Now I'm setting up my dev machine to test in the version 1.6.1.1 |
| 16:02.42 | mog | as that seems to be 95% of people that i talked to the one time i tried it |
| 16:02.59 | Amorsen | Laureano: Thank you! |
| 16:03.26 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 16:03.36 | murdock_ut | Howdy all. |
| 16:03.46 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:40b7:297b:441f:ee4d) |
| 16:04.16 | kaii | mog: poor you, i was lucky and chatted with a few nice people. |
| 16:04.44 | mog | meh i just did it for grins, someone told me it ran on ejabberd so i thought id look at it |
| 16:05.30 | Amorsen | Laureano: Nice idea, but unfortunately not |
| 16:05.52 | Amorsen | I have a exten => _X!,1,... |
| 16:06.18 | murdock_ut | I've been tasked with an interesting project. My user wants a DND button that will timeout after a certain amount of time. And they want users to be able to subscribe to that status. |
| 16:06.47 | ManxPower | makes the sign of the cross and backs away from murdock_ut |
| 16:06.58 | murdock_ut | ManxPower: hehehe |
| 16:07.02 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:07.10 | Laureano | Amorsen, thank you. I will try to reproduce this issue then in 1.6.1.1 using a dialplan like that. |
| 16:08.17 | murdock_ut | ManxPower: Creating the dnd button and having users subscribe to it shouldn't be a problem I don't think... but having it timeout is the part I'm wondering about. |
| 16:10.05 | [TK]D-Fender | murdock_ut: System(/pat/to/script/stopdnd ${usernumber} 3600 &) |
| 16:10.34 | bsteele | Anybody have experience with configuring Cisco 7960 IP phones? |
| 16:10.42 | [TK]D-Fender | murdock_ut: make a script that you background that will Sleep X and then issue a call file or whatever to execut the "un-DND" |
| 16:11.45 | murdock_ut | [TK]D-Fender: I was thinking I may have to do something like that. :) |
| 16:12.06 | [TK]D-Fender | murdock_ut: You will in order to have the thought of disabling on timeout. |
| 16:12.18 | murdock_ut | [TK]D-Fender: Knowing my luck I'll get this implemented and 3 months down the road they will want it removed. |
| 16:12.25 | [TK]D-Fender | murdock_ut: Not too hard. You could make a central timer, or script each of these. |
| 16:12.58 | murdock_ut | [TK]D-Fender: Like have it run once a minute and clean up extensions right? |
| 16:13.24 | murdock_ut | [TK]D-Fender: have a field in astdb that saves the extension dnd timeout value. |
| 16:13.27 | [TK]D-Fender | murdock_ut: the central way prevents repeat usage issues. like DND w/ 60 min timeout. Un DND 5 min later, and the old process is still lingering... then DND 50 minutes later thinking you have 1 hour, and 5 mins later the FiRST one undoes you |
| 16:14.07 | [TK]D-Fender | murdock_ut: Timeout value is nice.. the CHECK is another matter. Of course you COULD do this entirely in dialplan where you calculate the time from last DND |
| 16:16.16 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
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| 16:25.25 | galeras | please take a look off http://pastebin.ca/1504487 and tell me why dial with D() option works fine but dial with G() and SendDTMF fails. |
| 16:27.51 | ManxPower | G() is pretty limited as to what you can do. |
| 16:28.07 | ManxPower | for example you can't use waitexten in the G() macro. |
| 16:28.51 | galeras | What about M()? |
| 16:29.43 | ManxPower | Same. |
| 16:29.48 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 16:30.22 | ManxPower | you can use Read() |
| 16:33.19 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) |
| 16:37.11 | ManxPower | galeras: are you dialing out analog or digital ports? |
| 16:38.45 | galeras | ManxPower: i'm dialing digital ports,. |
| 16:38.59 | ManxPower | galeras: a PRI? |
| 16:39.59 | galeras | ManxPower: Yep |
| 16:40.08 | *** join/#asterisk pgarcia (n=chatzill@yoda.kanatek.com) |
| 16:40.22 | ManxPower | galeras: OK. |
| 16:40.31 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:43.53 | murdock_ut | [TK]D-Fender: How do I check out the latest svn version of 1.6.1? |
| 16:47.44 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 16:51.13 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:51.47 | pgarcia | Please, anybody here is already using libpri only with BRI ? |
| 16:53.23 | pgarcia | I'm wondering where I can find something saying what is already supported in libpri+bri... |
| 16:53.45 | psilikon | Are there any asterisk streaming audio feeds out there? I wouldn't mind sitting an learning about this stuff while I am in my office |
| 16:53.53 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 16:53.55 | Orbixx | What's the command to show current SIP devices? |
| 16:54.56 | ManxPower | "sip show peers" |
| 16:54.58 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:55.00 | Orbixx | thanks |
| 16:55.33 | *** join/#asterisk grabiarz (n=ptr@cpe-24-193-83-133.nyc.res.rr.com) |
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| 16:57.32 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 17:04.39 | murdock_ut | [TK]D-Fender: NM |
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| 17:09.58 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:15.58 | [TK]D-Fender | psilikon: Doubt it highly. Its hard enought to get really great articles... |
| 17:16.33 | tzafrir_laptop | pgarcia, generally works. What versions? |
| 17:18.01 | jaytee | [TK]D-Fender, at least leif's been putting several out on his blog and believe it or not someone actually updated the Asterisk-CentOS install guide on the wiki for 1.6 with DAHDI. |
| 17:18.16 | memph | hi |
| 17:18.51 | [TK]D-Fender | jaytee: the requiremensts should be the same for jsut about any RH ver & varient |
| 17:19.13 | pgarcia | tzafir_laptop, I'm researching using libpri+bri so I'm not using any version so far. My concern is about what is supported (like msn, overlap, etc...). |
| 17:19.21 | *** join/#asterisk propellerhead (n=yogurt2u@155-38-17-190.fibertel.com.ar) |
| 17:19.23 | jaytee | [TK]D-Fender, agreed |
| 17:19.29 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 17:20.12 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 17:20.37 | ManxPower | Historically BRI has not been well supported in Asterisk. I believe 1.6 changed that. |
| 17:20.58 | tzafrir_laptop | pgarcia, do you have the hardware already? |
| 17:21.13 | tzafrir_laptop | if so, why don't you try my live CD? |
| 17:21.18 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:21.47 | ariel_ | tzafrir_laptop: have you released an update to the 1.3 beta yet? |
| 17:22.33 | tzafrir_laptop | ariel_, yes, 2.0-beta |
| 17:22.49 | ariel_ | argh pandora is putting ads in there music now....yuk... |
| 17:22.57 | tzafrir_laptop | also under http://updaets.xorcom.com/iso/ . and now also available as a USB image |
| 17:22.58 | pgarcia | tzafrir_laptop, where can I find it? |
| 17:23.09 | ariel_ | oh nice |
| 17:23.17 | ariel_ | is off to get some iso's |
| 17:24.40 | pgarcia | ok, I gotcha (clicked enter while I was reading the line above) |
| 17:24.45 | ariel_ | tzafrir_laptop: that page is not coming up |
| 17:26.51 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:26.59 | tzafrir_laptop | http://updates.xorcom.com/iso/ |
| 17:27.10 | tzafrir_laptop | ariel_, pgarcia ==^ |
| 17:27.27 | ariel_ | that works great t/y |
| 17:27.36 | L|NUX | hello |
| 17:27.38 | L|NUX | can some one tell me how can i setup callback using a2billing using DID ? |
| 17:28.15 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 17:28.50 | pgarcia | tzafrir_laptop, thanks, I'll check that now |
| 17:29.50 | [TK]D-Fender | L|NUX: Go read the BOOK : |
| 17:29.52 | [TK]D-Fender | ~book |
| 17:29.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:30.13 | [TK]D-Fender | L|NUX: the key part of the call-back is using "call files" or "AMI Originate" |
| 17:30.29 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 17:30.45 | [TK]D-Fender | L|NUX: Both of which are documented in there |
| 17:32.39 | *** join/#asterisk Talirk81 (n=David@rrcs-67-78-39-22.sw.biz.rr.com) |
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| 17:36.09 | generalhan | hey all, im looking to pick up a new server for my asterisk box. the client refuses to let me build it out myself. my issue is that i need to find a server from a major manufacturer that will allow me to get power back to the PCI cage, without having to use one of those external power supply options. anyone have good experiences with any ? |
| 17:36.15 | Talirk81 | Does anyone have any ideas why in a DialPlan (inside a loop I set Monitor() on each loop,until a successful call is made) why in the h phase when i stop the monitor the in and out merge processes fires off fine, but i am not left with a final combines audio file. It seems if monitor is run on the same file name more than once it cause it not to actually create the file. ( Should each loop be doing something like if ${MONITORIN |
| 17:36.16 | Talirk81 | G} = "TRUE" ? Goto StopMonitorAnd Restart : StartMonitor ? |
| 17:38.44 | ariel_ | generalhan: I use Rackables there great systems. |
| 17:39.50 | generalhan | ariel_: like -- rackable.com ? |
| 17:39.58 | Talirk81 | Im not using ChangeMonitor, as im not changing the filename, i just was to start recoring over again |
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| 17:42.31 | ariel_ | generalhan: yes |
| 17:44.06 | generalhan | ariel_: thanks for the input. ill look into that. i think the owner will still turn that down. hes a real stickler about using HP, IBM, DELL |
| 17:44.51 | ariel_ | generalhan: well other one I have been using is the HP DL160 G5 they are good but expensive |
| 17:45.21 | generalhan | ariel_: and you are getting power to your cards from inside the case ? |
| 17:45.39 | ariel_ | yes |
| 17:46.13 | generalhan | ariel_: wonderful, i will look into that as well. thank you so much for the information |
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| 17:55.53 | Katty | hmm. |
| 17:57.59 | ariel_ | argh meeting time.... |
| 17:58.59 | Katty | just got done with mine |
| 17:59.25 | qu1ckkkkk | hi all |
| 18:04.26 | drmessano | generalhan: Go with a Lenovo workstation.. Get something small, mirrored drives, and buy a rack shelf |
| 18:04.44 | drmessano | Server class hardware is overkill for most PBX installs, IMO |
| 18:06.03 | *** join/#asterisk Lyma (n=Lyma@unaffiliated/lyma) |
| 18:06.23 | generalhan | drmessano: buying a non rackmount was my first choice, but the client doesnt have the room in their rack. they have room for a 2U server, and everything must be in their rack |
| 18:06.53 | generalhan | drmessano: i would need to pick up a slim-type model desktop to allow room in the rack |
| 18:06.56 | drmessano | Fine, then make them pay out the ass for giving you a limited set of conditions |
| 18:07.04 | generalhan | drmessano: lol. |
| 18:07.20 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:08.34 | drmessano | Im dead serious.. They are getting a first class PBX build with a first class telephony platform on a first class OS, and in appreciation, they're telling you "Make it look like a sunflower" |
| 18:08.45 | drmessano | FTMF.. Give them a Linksys PBX and some GS phones |
| 18:09.34 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 18:09.44 | generalhan | lol, make them pay for 100 analog lines directly wired to each phone, and walk away ! haha |
| 18:09.49 | Orbixx | Any idea how I can use queues.conf to ring two SIP phones simultaneously, then after giving up, move onto the rest of the queue one by one in a linear fashion? |
| 18:10.49 | drmessano | That line, BTW is from an interview with one of the hardware guys at Apple |
| 18:11.49 | drmessano | They spend like 2 years developing the successor to the iMAC.. bring the design to Steve Jobs.. and he hates it.. Tells them to scrap it.. 2 years of work.. Tells him it wasnt anything like what he envisioned.. "Make it look like a sunflower" |
| 18:12.56 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:13.03 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 18:13.19 | coppice | just spread it with margarine |
| 18:13.21 | drmessano | http://en.wikipedia.org/wiki/File:IMac_G4_sunflower7.png |
| 18:13.24 | drmessano | and so it goes |
| 18:14.13 | drmessano | He said something about wanting the screen to move to the users face.. like a sunflower does to the sun.. To have the screen drawn to the user |
| 18:14.30 | drmessano | and something about being low on Columbian Gold and calling Pedro |
| 18:14.40 | drmessano | *shrug* |
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| 18:19.26 | generalhan | lol |
| 18:19.31 | [TK]D-Fender | Orbixx: Use a local channel as that memeber and make the dialplan dial both handsets |
| 18:19.58 | Orbixx | [TK]D-Fender: I just tried something like "exten => 200,1,Dial(sip/200&sip/201)" |
| 18:20.02 | Orbixx | But that didn't seem to work. |
| 18:20.29 | [TK]D-Fender | Orbixx: It will. |
| 18:20.37 | Orbixx | It will... Not work? |
| 18:20.42 | [TK]D-Fender | Orbixx: for that member. |
| 18:20.50 | [TK]D-Fender | Orbixx: WILL work. if used by a queue |
| 18:21.27 | Orbixx | Not familiar with "channel", does that group certain extensions together? |
| 18:21.35 | leifmadsen | member => Local/101@agents |
| 18:21.38 | Orbixx | And so allows a channel to become a member of a queue? |
| 18:21.50 | leifmadsen | Orbixx: no, a channel is a method with which end points are connected |
| 18:22.01 | leifmadsen | i.e SIP channel, IAX2 channel, H323 channel, DAHDI channel, etc.... |
| 18:22.04 | Orbixx | I'm not using agents fyi. |
| 18:22.05 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 18:22.07 | Orbixx | I see. |
| 18:22.13 | leifmadsen | Orbixx: the Local channel is a special channel that loops into the dialplan |
| 18:22.21 | leifmadsen | so you can "call" the dialplan |
| 18:22.31 | Orbixx | I see. |
| 18:22.32 | leifmadsen | Dial(Local/101@context) |
| 18:22.44 | leifmadsen | my example had a context named 'agents', but you can call it whatever you want |
| 18:22.46 | leifmadsen | [agents] |
| 18:22.59 | leifmadsen | exten => _1XX,1,Dial(SIP/${EXTEN}) |
| 18:23.02 | leifmadsen | for example |
| 18:23.27 | Orbixx | That's pretty much what I have, except I'm looking to essentially make two SIP phones ring simultaneously when ONE of them is rung. |
| 18:23.28 | Orbixx | i.e. |
| 18:23.32 | leifmadsen | using the Local channel from the Queue() application (i.e. Local channel as a member) allows you to perform dialplan logic prior to dialing a "real" channel |
| 18:23.38 | Orbixx | If 200 rings, so does 201. |
| 18:23.42 | Orbixx | If 201 rings, just 201 rings. |
| 18:24.03 | leifmadsen | Orbixx: use the Local channel like I showed above, and add SIP/200&SIP/201 |
| 18:24.10 | Orbixx | thanks |
| 18:24.20 | *** join/#asterisk explody (n=groggy@gw.gemstone.com) |
| 18:26.01 | *** part/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 18:26.22 | leifmadsen | i.e. Dial(SIP/201&SIP/200) |
| 18:26.40 | [TK]D-Fender | Which is what I said works just fine already |
| 18:27.06 | explody | anyone know if it's possible, using a dual-port PRI card like a TE210P, to use one port for the inbound line from the provider and the second port outbound to another PBX? |
| 18:27.32 | leifmadsen | explody: sure, why not? |
| 18:27.54 | explody | no reason at all, just wanted to make sure I wasn't making a foolish assumption |
| 18:27.55 | [TK]D-Fender | explody: Plenty of people do this just to add call recording to old PBX's, etc |
| 18:28.02 | leifmadsen | explody: nope, enjoy! :) |
| 18:28.06 | [TK]D-Fender | IVR's, and so forth |
| 18:28.07 | explody | perfect, that's pretty much what I need to do |
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| 18:32.43 | afink | are DAHDI and HPEC working well together yet? I had some issues early on when DAHDI first came out |
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| 18:39.48 | vicscandl | [TK]D-Fender: rolling own this morning; outside of the asterisk package and the billing bolt-on, what other items would you suggest for the vanilla centOS server install? |
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| 18:55.53 | [TK]D-Fender | vicscandl: asterisk-addons |
| 18:56.55 | Qwell | speaking of which... |
| 18:57.52 | vicscandl | [TK]D-Fender: 10-4; i'll say it now: thanks for your help |
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| 19:07.38 | eppigy | TRABAJO |
| 19:11.59 | jaytee | hehe, was just thinking that |
| 19:13.15 | afink | anybody know what you pay for caller id name delivery on a ds1? qwest wants $175 one time then $20/month after that just for name on the cid? seems pretty freakin steep to me |
| 19:14.20 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 19:14.21 | Katty | pouts a bit |
| 19:14.25 | Katty | i finally took down my vegan blog. |
| 19:14.44 | Katty | it was up there for 4 years, with a crap load of followers and comments, but...it was time. |
| 19:15.02 | *** join/#asterisk envirocbr (i=enviro@unaffiliated/envirocbr) |
| 19:15.54 | envirocbr | Is there a method where Asterisk can automatically change the voicemail greeting to a pre-recorded temporary greeting when voicemails reach a certain limit? |
| 19:18.32 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 19:20.00 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:27.26 | [TK]D-Fender | envirocbr: You can write your own script to poll your boxes and do the swap |
| 19:27.39 | [TK]D-Fender | envirocbr: But this will be completely outside of * |
| 19:33.35 | envirocbr | Gotcha |
| 19:35.10 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 19:43.06 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 19:45.47 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 19:54.50 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 19:54.51 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:58.41 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 19:59.03 | lirakis | phones magically started working again .... freakin lame network it guys |
| 20:06.09 | *** join/#asterisk toughmarketing (n=toughmar@ip72-199-181-246.sd.sd.cox.net) |
| 20:09.53 | toughmarketing | Hey guys, I am still searching and trying to figure out how to use time defines in the extensions without doing the include. Right now I am doing include => holiday|*|*|1|jan but I cannot put that into my database and my extensions are all in the database. |
| 20:12.04 | Corydon76-dig | toughmarketing: Use GotoIfTime and the IFTIME() dialplan function to conditionally branch |
| 20:12.24 | toughmarketing | Corydon76-dig: thank you, that is exactly the function i was looking for |
| 20:12.51 | ManxPower | toughmarketing: "core show applications" and "core show functions" |
| 20:12.53 | Corydon76-dig | Timed includes have edge cases that make execution sometimes act in weird ways |
| 20:13.44 | toughmarketing | I will play with it, is it the only way to go if your using the db route. I am making a php interface to handle this stuff and I think it would be easiest if I have it in the db. |
| 20:17.37 | *** join/#asterisk faiz (n=otakucon@c-98-221-51-177.hsd1.nj.comcast.net) |
| 20:17.58 | faiz | requesting some help, please |
| 20:18.00 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 20:18.09 | faiz | regarding asterisk 1.6 & zaptel / DAHDI |
| 20:18.32 | *** join/#asterisk andres833 (n=andres83@200.69.100.123) |
| 20:19.09 | ManxPower | ~ask |
| 20:19.10 | infobot | 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. |
| 20:19.32 | *** join/#asterisk SuperBacon (n=Clarvis@unaffiliated/someidiot) |
| 20:21.31 | [TK]D-Fender | checkout time. Later all |
| 20:22.47 | faiz | well, i compiled asterisk-1.6.current with zaptel and libpri |
| 20:23.11 | faiz | and configured the zaptel.conf, zapata.conf, and extensions.conf with respect to the O'Reilly book (which details version 1.4) |
| 20:23.14 | ManxPower | Unfortunately, 1.6 does not support Zaptel. |
| 20:23.21 | faiz | yeah, such is my question, |
| 20:23.38 | faiz | i understand it supports DAHDI now |
| 20:23.40 | leifmadsen | faiz: you need to use DAHDI |
| 20:23.44 | leifmadsen | not zaptel |
| 20:24.03 | ManxPower | earlier versions of 1.4 only supported Zaptel. More recent 1.4 supports both DAHDI and Zaptel, 1.6 and later only supports DAHDI. |
| 20:24.05 | faiz | yes yes, but where can I get an idea on how to configure, and what files to configure? |
| 20:24.37 | ManxPower | I always just followed the docs that comes with the source. |
| 20:24.40 | leifmadsen | svn co http://svn.asterisk.org/svn/dahdi-linux-complete/tags/2.2.0.1+2.2.0 ; cd 2.2.0.1+2.2.0 ; make all ; make install ; make config ; service dahdi start |
| 20:24.53 | leifmadsen | ManxPower: agreed |
| 20:25.04 | leifmadsen | faiz: the docs that come with the DAHDI source tell you exactly what to do |
| 20:25.11 | faiz | ah.. |
| 20:25.16 | faiz | thank you, much appreciated |
| 20:27.27 | faiz | so the sudden change from zaptel to dahdi is really just a name change? |
| 20:27.47 | faiz | because of intellectual property or rights for name usage? |
| 20:30.12 | *** join/#asterisk jshriver (n=jshriver@h252.10.117.75.dynamic.ip.windstream.net) |
| 20:30.13 | Qwell | faiz: more or less, yes |
| 20:30.14 | jshriver | Greetings |
| 20:30.19 | jshriver | how do you truncate a sqlite table? |
| 20:30.23 | *** join/#asterisk qu1ck (n=leonja51@dsl-240-197-252.telkomadsl.co.za) |
| 20:30.34 | jshriver | asterisk keeps giving me a "database is full" error, even though the sqlite db file is 2megs |
| 20:30.56 | Katty | take an axe to it |
| 20:31.59 | ManxPower | jshriver: define "truncate" |
| 20:32.30 | ManxPower | What version of sqlite does Asterisk use anyway? |
| 20:33.09 | ManxPower | I assume they switched to it in 1.6, since every version before that used Berkley DB |
| 20:33.10 | jaytee | sqlite? |
| 20:33.21 | Qwell | ManxPower: res_config_sqlite |
| 20:33.33 | ManxPower | Ah. Nifty. |
| 20:33.36 | Corydon76-dig | ManxPower: if you're talking about astdb, no change has been made |
| 20:34.28 | jaytee | bbiab |
| 20:35.07 | ManxPower | As far as I can tell the only DB maintaince you can do in sqlite2 is dump the DB into a text file and read a text file into a DB. |
| 20:35.14 | ManxPower | I don't know about sqlite3 |
| 20:35.45 | ManxPower | Regardless, this would be totally outside of Asterisk as far as I know. |
| 20:36.06 | Corydon76-dig | ManxPower: same is true, you cannot do an alter of a sqlite table without recreating the table from scratch |
| 20:36.36 | Corydon76-dig | Can't even do a rename, which means any change like that requires downtime |
| 20:36.39 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:36.42 | KavanS | heh, define: sexual relations |
| 20:36.52 | KavanS | bad joke...clinton reference |
| 20:36.57 | KavanS | shouldn't speak of such things here... |
| 20:37.02 | jshriver | truncate as to remove every entry from a table :) |
| 20:37.07 | *** part/#asterisk ehsjoar (n=ehsjoar@24.9.91.203) |
| 20:37.57 | Corydon76-dig | jshriver: only way is to dump the table description (pretty easy), drop the table, create the table again |
| 20:38.08 | jshriver | yikes |
| 20:38.18 | ManxPower | jshriver: in the sqlite2 application, ".output thefilenameyouwant" then ".schema". then delete the db file, then cat thefilenameyouwant | sqlite dbnameyouwant |
| 20:38.28 | jshriver | thanks.. think I might just go ahead and setup a MySQL table and migrate it over there. sqlite seems a bit ugly anyway. |
| 20:38.35 | ManxPower | I've been doing quite a bit of sqlite stuff recently |
| 20:38.41 | jshriver | ManxPower: nifty will try that in the meantime :) |
| 20:38.49 | vicscandl | sqlite = smaller footprint |
| 20:40.17 | Corydon76-dig | jshriver: to get the table structure, it's as easy as: "SELECT sql FROM sqlite_master WHERE type='table' AND tbl_name='%s'" |
| 20:42.30 | jshriver | ty |
| 20:46.14 | *** join/#asterisk lowtek (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 20:46.58 | lowtek | Hey guys, is there any big secret to passing caller id from phones connected to a remote asterisk box? The caller id comeing through is just the name of the asterisk-asterisk sip.conf peer ... |
| 20:47.48 | *** part/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 20:47.54 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 20:49.30 | ayeso | I am stress testing a new asterisk mail server, I am making many simultaneous calls to separate voice mail boxes and leaving messages. once I get to about 180 simultaneous calls, asterisk crashes with a segmentation fault. Is this to be expected? |
| 20:50.15 | ayeso | nothing shows up in dmesg |
| 20:50.47 | *** join/#asterisk gornati (n=cams@201-87-101-62.static-corp.ajato.com.br) |
| 20:51.11 | lowtek | Ahh.. it was just my fromuser |
| 20:51.13 | lowtek | ;) |
| 20:53.40 | *** join/#asterisk toddross (n=toddross@adsl-76-244-69-82.dsl.austtx.sbcglobal.net) |
| 20:53.45 | *** join/#asterisk qu1ck (n=leonja51@dsl-240-222-230.telkomadsl.co.za) |
| 20:55.17 | *** join/#asterisk zdennis (n=zdennis@75.119.89.19) |
| 20:55.23 | zdennis | anyone deploy asterisk changes with capistrano? |
| 20:55.32 | Qwell | What is capistrano? |
| 20:55.44 | zdennis | Qwell: a deploy tool |
| 20:55.52 | ayeso | anyone know how many simultaneous calls an asterisk server can handle? anyone know what happens when you surpass that limit? |
| 20:55.56 | zdennis | http://www.capify.org/ |
| 20:56.10 | Qwell | a ruby deploy thing? O.o |
| 20:56.11 | zdennis | ayeso: depends on hardware and bandwidth |
| 20:56.20 | zdennis | Qwell: is it built in ruby yes, but can be used for anything |
| 20:56.41 | ayeso | zdennis: Right, but what happens when you pass the maximum amount of calls a given system can handle? |
| 20:57.35 | imcdona | according to voip-info.org I should be able to put ChanSpy(Agent) and then use 1234# 1234 being the agent number. to listen to agents calls. It's not working. Any idea's? Is this even possible? |
| 20:57.39 | lowtek | dial exec should just report full since it can't allocate any resources |
| 20:58.20 | ayeso | im getting segmentation faults at 180 calls |
| 20:58.34 | lowtek | ayeso: Only send 179? |
| 20:59.35 | ayeso | lowtek: Thats a weak solution.... |
| 20:59.40 | lowtek | lol |
| 20:59.45 | ayeso | hehe |
| 21:00.08 | lowtek | I would actually size it to 150 and just use groups to make sure you never get near 180 if that's your hardware ceiling. |
| 21:00.32 | lowtek | transcoding g729/ulaw on a dual opteron with 2gb and 1.4.x I only allow 100 simultaneous channels. |
| 21:00.38 | Qwell | imcdona: are you actually using agents, or just queues? |
| 21:00.42 | lowtek | anything above that gets iffy |
| 21:00.48 | lowtek | for my config, ymmv |
| 21:00.59 | imcdona | I have setup dynamic agents, they are logging into the queues |
| 21:01.11 | Qwell | imcdona: define "dynamic agents" |
| 21:01.23 | ayeso | lowtek: groups? |
| 21:02.05 | zdennis | anyone know of a way to wait for a few seconds before playing a recording ? |
| 21:02.09 | lowtek | ayeso: Set(GROUP()) |
| 21:02.14 | zdennis | we're calling in, and missing the first half-second of a recording |
| 21:02.16 | Qwell | zdennis: Wait() |
| 21:02.20 | zdennis | thx Qwell |
| 21:03.22 | imcdona | this is with free pbx....they login by dialing the queue number followed by star.....503* for example |
| 21:03.32 | lowtek | in 3, 2, 1 ... |
| 21:03.46 | ayeso | lowtek: cant this be accomplished by the asterisk.conf file option maxcalls? |
| 21:03.58 | *** join/#asterisk andres833 (n=andres83@200.69.100.123) |
| 21:03.59 | lowtek | ayeso: Not sure, I just use GROUPS() |
| 21:04.17 | *** part/#asterisk toddross (n=toddross@adsl-76-244-69-82.dsl.austtx.sbcglobal.net) |
| 21:04.34 | ayeso | lowtek: so you handle overflow to a different system? |
| 21:04.47 | *** join/#asterisk demiv (n=demiv@190.144.133.98) |
| 21:05.06 | imcdona | 1001,n,ChanSpy("agent/3001") does not work either...chansype answers, I hear the beep but no audio. Chan spy works if I specify a zap channel though |
| 21:05.09 | *** part/#asterisk mphill (n=mphill@174.37.19.92) |
| 21:05.13 | lowtek | ayeso: I have a tree of servers in a macro on each box, when it gets "not-avail" from the maxed out box, it just falls through to the next one. Works like a champ. |
| 21:05.28 | Qwell | imcdona: we don't know if you're using agents. #freepbx |
| 21:05.39 | *** join/#asterisk WhIteSidE (n=chatzill@209.236.250.9) |
| 21:05.43 | *** join/#asterisk Naf (n=annon@209.212.21.169) |
| 21:05.46 | Naf | isk-il |
| 21:05.48 | imcdona | thanks qwell |
| 21:06.17 | WhIteSidE | Hello all, I just installed asterisk (from the tarballs) and asterisk gui is complaining about not finding ztscan, and locate doesn't find ztscan either |
| 21:06.23 | ayeso | lowtek: i see, sounds cool... so you said that your running a opteron with 2g mem and you limit to 100? |
| 21:06.31 | demiv | Hello there, is posible to execute a goto('mycontext',s,1) inside an agi ? |
| 21:06.34 | Qwell | WhIteSidE: Try #asterisk-gui |
| 21:06.54 | lowtek | ayeso: Yea, but I'm transcoding g729->ulaw, i bet if I did straight ulaw I could double that number (theoretically) |
| 21:07.18 | ayeso | lowtek: well cool, sounds good. thx for the info |
| 21:07.31 | lowtek | np |
| 21:07.37 | WhIteSidE | Qwell: Ok |
| 21:07.53 | WhIteSidE | Although, ztscan doesn't appear to have been installed. I cannot locate it |
| 21:10.13 | leifmadsen | dahdi_scan (I had mentioned this in #asterisk-gui already, for anyone else who isn't in that room) |
| 21:10.24 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 21:10.56 | WhIteSidE | Thanks, I forgot thought it was dahdiscan (whoops!) |
| 21:20.30 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:30.29 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 21:30.46 | *** part/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 21:32.59 | mmattice | is Wait() not in 1.6? |
| 21:36.01 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 21:39.40 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 21:40.03 | Peaceful | quickie question: How do you (what's the syntax to) strip off the first digit of a string in the dialplan" |
| 21:41.06 | Peaceful | ${foo:1} <-- ?? |
| 21:46.31 | Jumpie | sup guys |
| 21:46.40 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 21:46.44 | Jumpie | anybody done much messing around with getting a phone to dial from a command/script sent on a computer? |
| 21:46.48 | Jumpie | i mean i know call centers do stuff like that |
| 21:47.01 | Dovid | surte |
| 21:47.02 | Jumpie | i wanna click a # in lets say, outlook or zimbra email message, and have it start dialing out my phone |
| 21:47.04 | Dovid | sue* |
| 21:47.29 | Dovid | digium has one i think for window$ |
| 21:47.40 | Dovid | works with the AMI |
| 21:48.58 | Katty | eppigy: Moo, it's what's for dinner. |
| 21:51.59 | Jumpie | do i have to have bought a digium appliance or something? is it commercial? |
| 21:56.20 | eppigy | :D |
| 21:58.11 | GameGamer43 | Jumpie: http://blogs.digium.com/2008/12/22/asterisk-desktop-assistant-windows-click-to-call-and-more/ |
| 22:05.04 | Jumpie | thanks alot! |
| 22:05.19 | GameGamer43 | no problem |
| 22:05.48 | Slade- | hmm SAY DIGITS * "" doesnt work |
| 22:05.53 | Slade- | it doesnt like * ? |
| 22:08.24 | *** join/#asterisk WhIteSidE (n=chatzill@209.236.250.9) |
| 22:12.36 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 22:12.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:17.11 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
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| 22:24.21 | vicscandl | anyone: best roll-your-own install guide for centOS and asterisk? |
| 22:24.37 | vicscandl | (outside of the readmes) |
| 22:24.38 | Slade- | just install asterisknow? |
| 22:25.12 | vicscandl | yea, i would like to, but the testing hardware (laptop) does not seem to like it. |
| 22:25.28 | vicscandl | (but i also like to do things the hard way cause when i started this game, there was no mouse) |
| 22:28.28 | Qwell | AsteriskNOW doesn't do anything special with the kernel.. if you can install CentOS, you can install AsteriskNOW |
| 22:30.26 | WhIteSidE | Is there a good guide to setting up the TDM400 with DAHDi? |
| 22:30.51 | Slade- | man say digits is a pain |
| 22:31.12 | Slade- | or maybe get data is a pain.. havent figured out which :P |
| 22:31.34 | vicscandl | Qwell: When I was doing the AN install, it kept complaining about a bad package (ntp) not sure if it was from my download or from the iso that was generated. |
| 22:31.52 | Qwell | during the install? |
| 22:31.56 | vicscandl | yes |
| 22:32.00 | Qwell | Your CD is broken |
| 22:32.02 | vicscandl | both text and gui install |
| 22:32.03 | vicscandl | yea |
| 22:32.17 | vicscandl | i downloaded it 4 times from different networks and used 2 different brands of CD's |
| 22:32.21 | vicscandl | same results. |
| 22:32.28 | Qwell | new burner :p |
| 22:32.30 | vicscandl | but i do agree, its on my end |
| 22:32.38 | Qwell | or try burning it slower |
| 22:32.39 | vicscandl | ha, burner is < 2 months new |
| 22:32.44 | vicscandl | did that too, 9x |
| 22:32.48 | vicscandl | even tried 2x |
| 22:32.48 | Qwell | new CD drive then |
| 22:33.00 | vicscandl | wish: brand new in other laptop |
| 22:33.16 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:33.17 | vicscandl | like i said, over the AN package and doing it all by hand.. ;) |
| 22:33.29 | Qwell | well, if you've got CentOS installed...it's easy |
| 22:33.31 | vicscandl | no biggie, just more skillz to pack up |
| 22:33.40 | Qwell | 32- or 64-bit? |
| 22:33.44 | vicscandl | both |
| 22:33.59 | vicscandl | doing testing comparison of each |
| 22:34.00 | Qwell | rpm -Uvh http://packages.asterisk.org/centos/5/current/x86_64/RPMS/asterisknow-version-1.5.0-1_centos5.noarch.rpm; yum groupinstall asterisk |
| 22:34.08 | vicscandl | oh, you effing rawk |
| 22:34.15 | Qwell | I get that a lot ;) |
| 22:34.18 | vicscandl | i was getting a case of jolt to compile it all by hand. |
| 22:34.27 | Qwell | if you don't want freepbx and stuff, you can swap the "groupinstall" with "install" |
| 22:34.27 | vicscandl | (which i don't mind doing) |
| 22:34.50 | Dovid | anyone know which GPL would be good to use ? i had code written and i want to let anyone and ev1 use it |
| 22:35.05 | vicscandl | ok, well the intent is to run a calling card system using a2billing package.... so would you say i do or do not need the freepbx stuff? |
| 22:35.12 | vicscandl | (and only a calling card system) |
| 22:35.16 | Laureano | Amorsen, check your issue. I added a note. |
| 22:35.22 | Qwell | Do you want to learn how to configure Asterisk the right way? |
| 22:35.35 | vicscandl | Qwell: haha, would not mind |
| 22:35.58 | Qwell | then don't install it :D |
| 22:36.12 | vicscandl | Qwell: thank you for the insight |
| 22:36.27 | vicscandl | freepbx != sane? |
| 22:36.35 | Qwell | No comment. |
| 22:36.52 | Dovid | lol |
| 22:37.19 | vicscandl | ok, i'll take that as its a bit thorny. |
| 22:37.20 | vicscandl | :) |
| 22:37.49 | Qwell | What do I know? I just package the stuff. Nobody ever said I used it. |
| 22:38.01 | Jumpie | hey qwell |
| 22:38.03 | Jumpie | i remember you |
| 22:38.22 | Qwell | Jumpie: That's rarely a good thing. |
| 22:38.46 | arcanine18_2000 | help pls... i have encountering one-way voice |
| 22:38.51 | Qwell | ~nat |
| 22:38.51 | infobot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 22:38.51 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 22:38.56 | Qwell | arcanine18_2000: read that |
| 22:39.32 | vicscandl | Qwell: ok, well given that i'm trying to setup a calling card business on this platform, outside of * and the a2billing package what other tools or goodies would you advise me to take a look at? |
| 22:39.32 | *** join/#asterisk Alfio (n=Amunoz@190.94.57.183) |
| 22:39.37 | Jumpie | lol Qwell you helped me out a little over a year ago when i was first dabbling with this |
| 22:39.56 | Qwell | vicscandl: dunno - I'd personally do it all by hand, but that's just me |
| 22:40.24 | vicscandl | Qwell: Yea, i still code HTML/PHP/SQL thru the command line... ha! |
| 22:40.29 | GameGamer43 | vicscandl: go grab Asterisk The Future of Telephony, it will definately help |
| 22:40.38 | vicscandl | GameGamer43: already reading it |
| 22:40.43 | Qwell | vicscandl: it's the only way |
| 22:40.43 | GameGamer43 | even better |
| 22:41.14 | vicscandl | GameGamer43: just have not gotten too much into it... doing R&D on some hardware right now too! gack! |
| 22:42.00 | GameGamer43 | I find the book to be a good guide and great reference.... |
| 22:44.15 | *** join/#asterisk chrisb (n=chrisb@pool-71-175-254-20.phlapa.east.verizon.net) |
| 22:46.06 | faiz | does chan_dahdi.conf need to be configured for dahdi_cfg to say "4 channels configured" ? |
| 22:48.01 | vicscandl | I <3 Qwell |
| 22:48.24 | vicscandl | btw: thanks for not bashing the n00b (yet) |
| 22:49.41 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 22:49.48 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:58.54 | vicscandl | thanks again Qwell, i'm sure i will be back later |
| 22:58.59 | *** part/#asterisk vicscandl (n=vicscand@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |
| 23:00.29 | WhIteSidE | Is there a way to make ael NOT load? |
| 23:01.14 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 23:01.50 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 23:01.52 | carrar | something like |
| 23:01.53 | carrar | noload => pbx_ael |
| 23:01.56 | carrar | in modules.conf? |
| 23:02.14 | howie | does anyone know if theres and cell providers i can use to get unlimited calls with conjunction of asterisk? |
| 23:02.21 | howie | any* |
| 23:02.28 | GameGamer43 | sprint@home |
| 23:02.33 | GameGamer43 | or tmobile fav 5 |
| 23:02.37 | GameGamer43 | but they have limitations |
| 23:02.52 | howie | i just got off the phone with sprint and they told me they dont offer the unlim home calling anymore |
| 23:02.59 | WhIteSidE | Thanks |
| 23:03.01 | GameGamer43 | and reserve the right to modify the term or cut service if you abuse the service |
| 23:03.17 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:03.19 | GameGamer43 | howie: sprint now the the simply everything plan, i thought that had unlimited calls |
| 23:03.51 | howie | well i got 13 lines was trying to cut my cellphone bill :0 |
| 23:04.12 | howie | minium plans and use my box to make calls.. |
| 23:04.28 | GameGamer43 | tmobile fav 5 is you best bet |
| 23:04.45 | howie | do they offer gps? |
| 23:04.58 | GameGamer43 | gps is specific to the phone not the carrier |
| 23:05.13 | howie | GameGamer43: thank you |
| 23:05.27 | GameGamer43 | howie: no problem |
| 23:06.02 | *** join/#asterisk Airozi (n=user@200-171-41-207.dsl.telesp.net.br) |
| 23:12.59 | Slade- | wow, so if i shoot asterisk right to an agi script in the dialplan.. and I don't 'ANSWER' everything will work just normally, except i wont be able to hang up |
| 23:13.36 | Slade- | wonders if thats a bug, or something useful he can exploit |
| 23:15.57 | *** join/#asterisk ChaosBlack (n=bjastles@67.204.63.40) |
| 23:20.41 | *** join/#asterisk ccbbaa (i=no@gateway/gpg-tor/key-0x66435AF5) |
| 23:21.47 | ccbbaa | is anyone tunning sip over tor ? |
| 23:21.51 | ccbbaa | on asterisk |
| 23:23.05 | Chainsaw | ccbbaa: Your magic words for today: deterministic latency |
| 23:23.23 | Chainsaw | ccbbaa: Unless it was a yes or no question. In which case, the short answer is no. |
| 23:27.02 | ccbbaa | Chainsaw: i was thinking about an announcement service, so only one side would talk. think music on hold but not quite |
| 23:27.37 | ccbbaa | what's the slowest media used for sip or iax ? radio modem ? sattelite ip ? |
| 23:28.14 | Chainsaw | ccbbaa: S...l thi... calle... ...ll want... ....ear ...ontinuou... au... |
| 23:29.07 | _abc_ | oh, I see. hmm. |
| 23:29.09 | Chainsaw | _abc_: *Deterministic* latency. |
| 23:29.45 | Chainsaw | _abc_: Satellite is high-latency, but it's constant. To the millisecond. |
| 23:30.00 | _abc_ | true |
| 23:30.04 | Chainsaw | _abc_: An onion router, by the very nature of how it works, can *not* offer you deterministic latency. |
| 23:30.29 | Chainsaw | _abc_: An RTP stream depends on >90% of the UDP packets making it to the other end in something close to the right order. |
| 23:30.43 | Chainsaw | You're *never* going to get thta. |
| 23:30.46 | Chainsaw | s/thta/that/ |
| 23:31.30 | _abc_ | Chainsaw: i am looking for something that would do that, like redundand rtp or such |
| 23:31.42 | *** join/#asterisk troubled_ (n=troubled@unaffiliated/troubled) |
| 23:31.52 | _abc_ | i am familiar with similar things in ax25 broadcast and the like |
| 23:32.09 | Chainsaw | _abc_: Just like we're all still looking for a battery that lasts forever. Best of luck. |
| 23:33.02 | _abc_ | Chainsaw: oh, those protocols work, they reassemble the data at the other end, a large delay results but it works. whenever you use a cell phone you use something very much like that |
| 23:33.19 | *** join/#asterisk ariel_ (n=ariel_@c-71-196-99-26.hsd1.fl.comcast.net) |
| 23:34.11 | Chainsaw | _abc_: That's still predictable-latency packet radio. |
| 23:34.46 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 23:34.50 | Chainsaw | _abc_: I'm not convinced you appreciate the difference between high-latency and non-deterministic (intermittent, highly variable) latency. |
| 23:35.19 | _abc_ | Chainsaw: not when interference zaps packets. there are 2 was to deal with it: retransmit or transmit everything more than once and time-spread from starters |
| 23:35.29 | _abc_ | i am talking about the 2nd way, |
| 23:35.30 | Chainsaw | Indeed, and the latter is how RTP works. |
| 23:36.07 | Chainsaw | However, as the RT (realtime) suggests, there are concessions made to keep the latency of the link to a minimum. |
| 23:36.27 | Chainsaw | That does mean it can not handle out-of-order packets very well (it handles it by throwing those packets away and hoping more will come in). |
| 23:36.41 | _abc_ | heh |
| 23:36.58 | _abc_ | i am just looking at protocols. oh, youre on wireless linux too |
| 23:37.29 | _abc_ | do you know a problem where the builtin wlan card sees certain access points and a usb one stuck in does not ? i have such a problem. |
| 23:38.13 | Chainsaw | Are the access points in question broadcasting on channel 13? |
| 23:39.03 | _abc_ | anyway i have asterisk running in a quemu vm with linux host and netbsd 3.1 client |
| 23:39.14 | _abc_ | Chainsaw: no, 6 or 2 |
| 23:39.35 | Chainsaw | Likely just better antennas on the built-in WLAN card then. |
| 23:39.38 | Chainsaw | (They're in the screen bezel) |
| 23:39.49 | _abc_ | and this is in north america fwiw |
| 23:40.03 | Chainsaw | Ah, right. So channel 13 is a luxury you'd never have. |
| 23:40.07 | _abc_ | i dont think that my devices span ch 13 |
| 23:40.09 | _abc_ | wrt asterisk in qemu: i use a tap interface to network, i am looking for recipes for that |
| 23:40.11 | _abc_ | Chainsaw: the signal is way too good for that to matter imho |
| 23:40.13 | _abc_ | its afunny combination between ndis driver and ESSID |
| 23:40.22 | _abc_ | Chainsaw: who has 13 ? |
| 23:40.50 | Chainsaw | If you're using hacks involving Windows device drivers; I can't support you. Unlikely that anyone else in #linux-wireless can either. |
| 23:41.10 | Chainsaw | _abc_: It's one of those US vs rest-of-the-world things. You know, like paper sizes. And GSM frequency allocations. |
| 23:41.37 | _abc_ | ok, i'm from Europe really. honest :) |
| 23:42.30 | _abc_ | otoh if you do get my broadcom or wl2313 wifi cards going in native mode i'd be more than very glad. ndis has a way to freeze laptops |
| 23:42.41 | Chainsaw | What Broadcom chip is it? |
| 23:44.27 | _abc_ | Atheros/ASUSTEK built 1a3b:1026 |
| 23:44.48 | Chainsaw | First it was a Broadcom. Now it's an Atheros. |
| 23:45.09 | Chainsaw | Please wipe the pixie-dust off it and see if it stops changing brand names now. |
| 23:45.17 | _abc_ | the radio and the chipset is broadcom |
| 23:45.57 | Chainsaw | _abc_: Let's move your WLAN discussion to #linux-wireless; I need a pastebin of lspci -nnv |
| 23:46.25 | _abc_ | ok just a sec. there is just a prob: low battery. |
| 23:46.41 | Chainsaw | I'm sorry, I can't fix that remotely. |
| 23:47.00 | _abc_ | :) |
| 23:52.28 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
