| 00:00.07 | *** join/#asterisk Bourrelle (n=Bourrell@132.207.156.100) |
| 00:00.19 | Bourrelle | Anyone know in a conversation between 2 persons, if asterisk by default forwards the RTP-H.263 packet |
| 00:00.26 | Bourrelle | or we have to manually set it ? |
| 00:00.34 | Bourrelle | The RTp-H.263 paquet leave 1 computer arrive to asterisk |
| 00:00.43 | Bourrelle | but asterisk doesnt send them to the other phone |
| 00:01.11 | Bourrelle | it only send the voice RTP packet |
| 00:01.45 | eurowerke | thanks riddlebox |
| 00:01.47 | *** part/#asterisk eurowerke (n=bmrjunk@paul.shepnet.org) |
| 00:02.03 | *** join/#asterisk RoyK (n=roy@ip-233-49-149-91.dialup.ice.no) |
| 00:02.54 | glaz | I have a * server on a public IP, but a phone behind a wrt linksys router, what port should be forwarded beside 5060 ? |
| 00:03.53 | cleone | glaz waw i'm like u i habe public ip and linksys router |
| 00:05.04 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 00:05.40 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:05.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:09.41 | *** join/#asterisk ming_zym (n=ming_zym@220.174.91.241) |
| 00:10.14 | *** join/#asterisk ManxPower (n=manxpowe@94.sub-70-220-252.myvzw.com) |
| 00:11.06 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:11.48 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.128.9) |
| 00:12.05 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 00:14.31 | *** join/#asterisk marksaitis (n=maxxx@80-46-86-99.static.dsl.as9105.com) |
| 00:16.11 | marksaitis | can anybody tell me if it is possible to ssl the data going between two clients speaking with each other in asterisk ? |
| 00:16.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:17.54 | marksaitis | in other words how to make sip go trough ssl |
| 00:18.28 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 00:18.53 | jjshoe | does anyone know how to simulate a condition where asterisk.ctl permissions will change? |
| 00:19.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:19.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:19.44 | jjshoe | or ownership |
| 00:19.58 | jjshoe | as in I set an owner on asterisk.ctl and it changes later on, I'm interested in what triggers the change |
| 00:20.03 | jjshoe | or perhaps a way to see what's changing it |
| 00:20.17 | marksaitis | jjshoe can you tell me how to make sip go trough ssl ? |
| 00:20.32 | jjshoe | marksaitis tunnel it. |
| 00:20.50 | marksaitis | is it not possible to do anything else ? |
| 00:20.55 | jjshoe | dunno. |
| 00:21.20 | marksaitis | tunnel is smth stupid, I wanna just encrypt the traffic going from client to server |
| 00:21.26 | *** join/#asterisk scoates (n=sean@64.15.79.181) |
| 00:21.29 | scoates | hello |
| 00:21.31 | marksaitis | with no other apps |
| 00:21.41 | lmadsen | hello |
| 00:21.58 | scoates | I've been having a problem with some of my phones going UNREACHABLE today. Is there a command-line utility that will check SIP connectivity for me? |
| 00:22.02 | lmadsen | marksaitis: sorry, there is no encryption in released asterisk -- you can check the bug tracker though |
| 00:22.26 | Katty | lmadsen: hai |
| 00:22.32 | lmadsen | Katty: well hello there little lady |
| 00:22.35 | russellb | it's in 1.6 |
| 00:22.35 | marksaitis | lmadsen :( very very bad |
| 00:22.38 | russellb | not on the bug tracker |
| 00:22.44 | lmadsen | russellb: oh did that get closed? |
| 00:22.47 | russellb | yes |
| 00:22.51 | marksaitis | probably the bigest problem |
| 00:23.03 | lmadsen | ah gotcha... merged, or just closed because a new branch is going ot be made? |
| 00:23.13 | russellb | well, a different implementation got merged months ago |
| 00:23.15 | lmadsen | I admit to not staying up on the bug tracker lately |
| 00:23.30 | lmadsen | oh, so there is some encryption stuff in trunk then? |
| 00:24.08 | marksaitis | so ok, 1.6 has ssl ? for both sip an iax ? |
| 00:24.58 | lmadsen | probably would say what was merged in the CHANGES file |
| 00:25.06 | ManxPower | and the upgrade.txt |
| 00:26.09 | russellb | 1.4 has IAX as well |
| 00:26.18 | russellb | (encryption) |
| 00:26.33 | marksaitis | ok |
| 00:27.08 | marksaitis | how about video conferencing with asterisk ? is this possible ? |
| 00:27.22 | lmadsen | you can make video calls |
| 00:27.36 | marksaitis | ok |
| 00:28.44 | marksaitis | can anyone tell me why to use IAX instead of sip ? |
| 00:28.52 | lmadsen | I only use SIP |
| 00:29.45 | lmadsen | 2 reasons: trunking (get more simultaneous calls over the same bandwidth), and easier NAT traversal |
| 00:30.04 | lmadsen | I've personally never had good luck with IAX, but others have fantastic luck -- just depends on your implementation I suppose |
| 00:30.19 | marksaitis | ok got that |
| 00:30.22 | marksaitis | :) |
| 00:30.43 | outtolunc | prefers smoke signals |
| 00:30.57 | lmadsen | I like carrier pigeons |
| 00:32.05 | marksaitis | is it possible somehow to connect some kind of hardware to the main telephone exchnage box in one branch, for all the phones in that branch to be able to reach asterisk server ? |
| 00:33.09 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:34.13 | *** part/#asterisk `paul (n=aldee@203.192.188.226) |
| 00:36.31 | scoates | a few of my phones keep going UNREACHABLE (with a very high qualify: 3000). Pinging the phones' IPs gives good response times. I suspect firewall timeout problems (they're NAT). How can I test to be sure that it's the firewall (from the asterisk side)? |
| 00:39.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:48.02 | *** join/#asterisk Cle0 (n=cleo@adsl196-131-185-206-196.adsl196-6.iam.net.ma) |
| 00:51.48 | scoates | why, if I've got nat=yes in a device in sip.conf, does sip show peers show NAT = N for that phone? |
| 00:53.20 | riddlebox | man I dont understand it, I setup asterisk myself, and blf's work, but in the asterisknow gui, I cannot get them to work |
| 00:53.33 | scoates | however, sip show peer sean has NAT = Always |
| 00:53.41 | scoates | confusing |
| 01:08.36 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 01:10.54 | bluregard | scoates, that N in sip show peers isn't for No it means that that nat is enabled |
| 01:11.05 | scoates | bluregard: ah. ok, thanks. |
| 01:12.34 | *** join/#asterisk objective (n=objectiv@ool-43536c3b.dyn.optonline.net) |
| 01:13.10 | scoates | bluregard: any idea how to determine the culprit in a phone that keeps going UNREACHABLE? |
| 01:13.27 | bluregard | scoates, local phone or is it remote? |
| 01:13.40 | scoates | bluregard: remote |
| 01:13.49 | scoates | some of my phones always work |
| 01:14.04 | scoates | today (or yesterday) 3 of them started going UNREACHABLE, seemingly randomly |
| 01:14.41 | bluregard | have you done network testing? |
| 01:15.00 | scoates | all 3 on the same ISP, at 3 different locations, all sipura ATAs, and they're all going through apple airports (one other phone, not sipura, different ISP is also going through airport).. these phones worked for 2 years, reliably before today (or yesterday) |
| 01:15.41 | scoates | bluregard: I'm not sure where to start. I tried playing with sipsak and sipp but I keep getting invite failures because the phones are expecting traffic from the port that asterisk is using (5060) |
| 01:16.09 | scoates | ping (ICMP) to the IP is all good |
| 01:16.16 | bluregard | well, first thing would be to check the latency between your end points |
| 01:16.45 | scoates | actual latency is very low: rtt min/avg/max/mdev = 28.867/34.968/115.187/7.498 ms |
| 01:16.47 | bluregard | how often are they dropping out? |
| 01:16.59 | scoates | more than once per hour |
| 01:17.08 | bluregard | I shouldn't have said latency, I meant packet loss |
| 01:17.14 | scoates | ah.. no loss either |
| 01:17.20 | scoates | 304 packets transmitted, 304 received, 0% packet loss, time 303624ms |
| 01:17.35 | bluregard | is one of the phones down right now? |
| 01:17.40 | scoates | I'm trying to measure DURING the loss |
| 01:18.00 | scoates | yep. 2 of the 3 are online, but the third is UNREACHABLE |
| 01:18.04 | *** join/#asterisk fedya (n=fedya@c-76-26-182-13.hsd1.fl.comcast.net) |
| 01:18.21 | scoates | arbi/arbi 99.237.x.y D N 39023 UNREACHABLE |
| 01:19.18 | bluregard | can you ping the phone itself |
| 01:20.01 | scoates | bluregard: how? |
| 01:20.15 | JT | with ping? |
| 01:20.22 | bluregard | ping :-/ ? |
| 01:21.00 | scoates | you mean the IP that the phone is at? |
| 01:21.12 | scoates | there's a NAT firewall between the phone and asterisk |
| 01:21.17 | scoates | (the phone is NATted |
| 01:21.18 | scoates | ) |
| 01:21.20 | bluregard | ok first of all, how are you setup |
| 01:21.42 | bluregard | so the phones are at a remote site behind a nat router |
| 01:21.48 | scoates | yes |
| 01:22.01 | bluregard | do you have access to the network at the remote site? |
| 01:22.27 | bluregard | or any other way of verifying that the phone is indeed running and connected to the network |
| 01:22.37 | scoates | indirectly.. I have someone who knows what they're doing on that side, but they're not available right now. |
| 01:22.47 | scoates | yes, he checked and rebooted the phone earlier today |
| 01:24.04 | *** join/#asterisk weazahl (n=jeremy@adsl-66-143-45-223.dsl.ksc2mo.swbell.net) |
| 01:24.19 | bluregard | he tried pinging the phone while it was unavailable to * |
| 01:24.58 | *** join/#asterisk ming_zym (n=ming_zym@220.174.91.241) |
| 01:25.05 | scoates | no.. I have someone else who can test the downed phone right now, though, I think |
| 01:26.55 | scoates | ok, yes, the UNREACHABLE phone is returning local pings (to the private IP) |
| 01:27.06 | *** join/#asterisk PepOSX (n=angeldav@190.72.130.192) |
| 01:27.18 | bluregard | and you can ping the public IP of that site? |
| 01:27.25 | scoates | yes |
| 01:28.39 | scoates | this FEELS like a UDP firewall timeout... but I can't prove it |
| 01:28.51 | scoates | and odd that it would start happening at 2 places at the same time |
| 01:29.34 | bluregard | are you able to watch your network traffic at either end with something like tcpdump? |
| 01:29.48 | scoates | yes, I can watch it at the asterisk end |
| 01:29.52 | scoates | what should I be looking for? |
| 01:30.22 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 01:30.29 | Yourname` | Anyone got any info on how to open 5060, 10000-20000 on SonicWall Pro 2040 Enhanced? |
| 01:31.30 | bluregard | scoates, your sip phones are connecting to UDP 5060 on asterisk right |
| 01:31.41 | scoates | bluregard: yes |
| 01:32.31 | scoates | Reg. Contact : sip:arbi@192.168.1.8:5060 ; Addr->IP : 99.237.x.y Port 39023 |
| 01:34.20 | *** part/#asterisk RoyK (n=roy@ip-233-49-149-91.dialup.ice.no) |
| 01:35.39 | bluregard | sorry, tcpdump on your SIP port and make sure the phones reg packets are getting to the * server |
| 01:38.13 | scoates | while looking up the syntax, another phone went unreachable |
| 01:38.23 | scoates | and I was pinging the IP the whole time without losing a single packet |
| 01:39.09 | bluregard | tcpdump -vi eth0(or whatever) port 5060 |
| 01:41.19 | scoates | right; that's too much traffic.. |
| 01:41.25 | scoates | I got it, though |
| 01:41.36 | scoates | do you want me to look at a phone that is unreachable or reachable? |
| 01:41.54 | scoates | also, two of the 3 phones are at the same location |
| 01:42.11 | bluregard | scoates, the 2 that are going down |
| 01:42.14 | scoates | currently one is unreachable, but the other is OK (41 ms) |
| 01:42.41 | scoates | (the third, at its own location is also unreachable.. probably easier to monitor that one) |
| 01:43.17 | scoates | ok.. packets going from asterisk to the unreachable phone, but no replies |
| 01:43.37 | bluregard | so the phone isn't talking to the * server |
| 01:44.03 | bluregard | talking back to the * server rather |
| 01:44.05 | scoates | well, it was.. it went unreachable ~1 hour ago |
| 01:44.07 | scoates | right |
| 01:44.19 | scoates | this could be UDP timeout |
| 01:44.29 | scoates | (firewall isn't passing the packet along to the originator) |
| 01:44.44 | bluregard | yeah, it would be handy to be able to watch the traffic going out of the remote site |
| 01:44.45 | scoates | what bugs me is that it magically started happening today |
| 01:45.10 | bluregard | what kind of router is doing your NAT and what WAN connection is it on |
| 01:45.12 | scoates | unfortunately, I can't do that.. even on that end, I won't be able to see switched traffic to the phone |
| 01:45.26 | scoates | it's an apple airport extreme (both locations) |
| 01:45.44 | scoates | what do you mean about WAN? |
| 01:45.56 | scoates | (fwiw, I have a third phone that's on the same type of router that's been fine all day) |
| 01:46.01 | scoates | (another location) |
| 01:46.24 | bluregard | what kind of internet connection |
| 01:46.32 | bluregard | cable dsl etc |
| 01:46.41 | scoates | ah.. it's cable (rogers) |
| 01:46.52 | bluregard | have you tried power cycling the airport |
| 01:46.55 | scoates | both failing locations are rogers.com, near Toronto |
| 01:47.04 | scoates | yep.. it's been cycled a few times |
| 01:47.08 | scoates | phones, too |
| 01:47.16 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 01:47.24 | scoates | the phones come back after a cycle, and are then lost 10-60 minutes later |
| 01:47.49 | bluregard | are the phones wireless |
| 01:48.06 | bluregard | being that they're using an Airport |
| 01:48.14 | scoates | might be.. 1s |
| 01:48.28 | scoates | can't check.. |
| 01:48.36 | scoates | I'm pretty sure at least one of them is |
| 01:48.56 | scoates | cabled through an imac via connection sharing |
| 01:49.11 | bluregard | I thought the airport was just an access point and didn't do any nat'ing or anything like that |
| 01:49.13 | bluregard | oooh |
| 01:49.14 | scoates | (like I said, though, this config has worked for months-to-years before today) |
| 01:49.28 | scoates | airport extreme is a nat router |
| 01:50.15 | bluregard | ok, so it's phone--->mac----[wifi]--->airport--->internet----->asterisk |
| 01:50.31 | scoates | one of them is like that, certainly, yes |
| 01:51.06 | scoates | I realize there are a lot of points of failure there, but it's happening at different locations on different phones, so I'd have a hard time pinning it on "the mac" or "wifi at both locations" |
| 01:51.23 | bluregard | yeah |
| 01:51.46 | *** join/#asterisk Katty (n=The@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 01:52.02 | scoates | hmm |
| 01:52.21 | scoates | when the guy's around tomorrow, he should be able to tcpdump on the bridging mac to see if the phone is getting poke packets |
| 01:52.23 | *** join/#asterisk fedya (n=fedya@c-76-26-182-13.hsd1.fl.comcast.net) |
| 01:52.30 | Katty | hai. |
| 01:52.44 | bluregard | hello |
| 01:54.42 | scoates | bluregard: I'll try to capture packet data on the phone side tomorrow. Thanks much for your help. |
| 01:54.48 | scoates | (glad I'm not crazy (-; ) |
| 01:55.32 | bluregard | I don't think so |
| 01:56.59 | bluregard | I'd be interested to know what the problem is when you figure it out |
| 01:57.34 | scoates | bluregard: ok. I'll come back and report.. |
| 01:57.49 | scoates | I highly suspect UDP timeout and a firmware auto-update |
| 01:58.13 | bluregard | firmware on what, the phone |
| 01:58.17 | scoates | router |
| 01:58.27 | bluregard | they do auto updates? |
| 01:58.47 | scoates | I don't think so, but it might be tied to the mac (which DID have an airport update in the last week or so) |
| 01:59.15 | scoates | bluregard: do you know the poke interval? it seems to be 5 packets every 10s on an unreachable phone, and (fewer) every 60s on a reachable phone.. sound right? |
| 01:59.29 | bluregard | not sure |
| 02:00.49 | scoates | I get this http://phpfi.com/305789 |
| 02:01.03 | scoates | pretty sure the first burst is the unreachable phone, and the 2nd burst is the reachable phone |
| 02:01.10 | scoates | first burst happens every ~10s |
| 02:01.19 | scoates | 2nd less often |
| 02:02.43 | scoates | ... and now all 3 phones are reachable |
| 02:02.45 | Treytor | is anyone aware of a list I could add to block all emergency numbers, hospitals, and government agencies? |
| 02:02.55 | scoates | they both came back within 3 mins of each other |
| 02:03.22 | scoates | Treytor: setting up a slave camp? (-: |
| 02:03.30 | bluregard | scoates, does the airport have any kind of logging |
| 02:03.31 | Treytor | haha, yeah something like that ;-) |
| 02:03.46 | Treytor | no, I just want to ensure no calls to any of those numbers leave my asterisk box |
| 02:03.55 | scoates | bluregard: doubtful.. I'll try to find out tomorrow |
| 02:03.56 | Treytor | I'm sure someone has a compiled list somewhere |
| 02:04.04 | scoates | airport is very much dumbed down |
| 02:04.15 | bluregard | never used one |
| 02:04.44 | bluregard | I've been using IPCop for my router for several years now, I love it |
| 02:05.08 | scoates | I have a hacked WRT54g here (-: |
| 02:05.38 | scoates | alright.. I've got to take off. |
| 02:05.39 | bluregard | still haven't gotten around to trying that one out yet either |
| 02:05.41 | scoates | thanks again for the help |
| 02:05.44 | bluregard | good luck |
| 02:08.46 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 02:18.28 | *** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) |
| 02:21.04 | _Sam-- | if im using ASTapi to dial out from MS Outlook, how would i set the caller id? |
| 02:22.02 | _Sam-- | it seems like that thing just connects directly to manager to make the call...and my calls show up with teliax's, and not my own set caller ID. |
| 02:35.59 | *** part/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) |
| 02:37.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:58.03 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 02:59.20 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 02:59.58 | anonymouz666 | anyone has idea on how to stop the damn REINVITE in Asterisk 1.4.15? |
| 03:00.01 | anonymouz666 | even with nat=yes |
| 03:00.04 | anonymouz666 | canreinvite=no |
| 03:01.44 | kyron | Q: I did (for zaptel): `./configure --prefix=/mnt/RAID5/asterisk/ && make && make install` then `./configure --prefix=/mnt/RAID5/asterisk/ --with-zaptel=/mnt/RAID5/asterisk/` for asterisk...problem is, zap_chan options are all XXX in the menu...what am-I doing wrong for *'s config to detect zaptel...? |
| 03:02.09 | drmessano | canreinvite=no |
| 03:02.52 | anonymouz666 | drmessano: it is already configured. |
| 03:03.01 | anonymouz666 | doesn't work. |
| 03:03.33 | drmessano | In sip.conf? |
| 03:03.39 | anonymouz666 | of course. |
| 03:03.39 | drmessano | or for the peer? |
| 03:03.44 | anonymouz666 | GENERAL |
| 03:03.54 | drmessano | CAPS LOCK STUCK? |
| 03:03.58 | anonymouz666 | there's no user/peer sections |
| 03:06.38 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 03:06.38 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta6, 1.4.18.1, 1.4.19-rc3, 1.2.27 (2008/03/18), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies. |
| 03:07.39 | anonymouz666 | jbot: WTF |
| 03:08.43 | bluregard | anonymouz666, what do you mean there's no user/peer section |
| 03:09.03 | *** join/#asterisk enmaca (n=enmaca@201.163.46.3) |
| 03:09.10 | anonymouz666 | it's just [general] |
| 03:09.25 | anonymouz666 | I don't want any user |
| 03:09.51 | bluregard | then what are you worried about sip reinvites if you have no sip users |
| 03:09.57 | bluregard | or peers |
| 03:10.27 | anonymouz666 | someone sends requests to the asterisk |
| 03:10.34 | anonymouz666 | and it is handle in [general] |
| 03:10.37 | anonymouz666 | handled |
| 03:11.08 | drmessano | canreinvite=no is the same in 1.2 and 1.4 |
| 03:11.14 | drmessano | Sure you didn't typo it? |
| 03:11.41 | anonymouz666 | the option is exactly the same. |
| 03:11.45 | anonymouz666 | not the behavior. |
| 03:12.20 | drmessano | .......ok |
| 03:13.52 | *** join/#asterisk suvir (n=suvir@ppp-124-120-245-57.revip2.asianet.co.th) |
| 03:20.10 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 03:21.31 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 03:28.40 | glaz | I am trying to connect two asterix server togheter via IAX, and I get a chan_sip error, I don't get it |
| 03:28.48 | glaz | WARNING[3153]: chan_sip.c:6713 determine_firstline_parts: Bad request protocol |
| 03:29.15 | drmessano | Sounds like an epic fail |
| 03:29.43 | kyron | epic? we have to write an epic?! |
| 03:30.20 | drmessano | No |
| 03:30.28 | drmessano | It was already written |
| 03:30.28 | kyron | :) |
| 03:30.34 | kyron | but we read it ;) |
| 03:30.35 | drmessano | But since you didn't get the joke |
| 03:30.43 | drmessano | Epic fail 2u |
| 03:31.16 | glaz | hehe |
| 03:31.21 | glaz | drmessano: any idea? |
| 03:31.34 | kyron | drmessano, hey, I never get your jokes, I am missing out on quite a lot :P |
| 03:31.39 | glaz | I've followed the manual to create a IAX channel |
| 03:32.41 | drmessano | Did you create it in sip.conf? |
| 03:34.56 | glaz | create what? |
| 03:35.12 | glaz | the manual did not mention anything about sip.conf |
| 03:35.36 | drmessano | Tell me what you did |
| 03:36.32 | glaz | I created iax.conf on both side |
| 03:36.40 | glaz | and restarted * |
| 03:36.44 | kyron | glaz, didn't you tell me you created the trunk under IAX |
| 03:37.18 | glaz | I did. |
| 03:39.52 | glaz | drmessano: http://rafb.net/p/jAN9b240.html |
| 03:39.57 | glaz | you have the config on both sides |
| 03:43.53 | *** join/#asterisk PepOSX (n=angeldav@190.78.216.146) |
| 03:46.09 | glaz | drmessano: any idea? |
| 03:47.26 | [TK]D-Fender | glaz, pastebin both sides of the call attempt at verbose 10 |
| 03:48.51 | glaz | core set debug 10 ? |
| 03:49.19 | kyron | glaz, wasn't even aware this was a call attempt...thought it was while attempting to bring up the trunk |
| 03:49.37 | glaz | it is |
| 03:49.52 | kyron | to [TK]D-Fender ...you have any idea why zaptel options would be XXX out? |
| 03:49.53 | glaz | I get this error message when the trunk try to come online |
| 03:50.13 | [TK]D-Fender | kyron, XXX out of where? |
| 03:50.16 | kyron | glaz, conflicting extensions?...dunnot.. |
| 03:50.25 | kyron | [TK]D-Fender, sorry, make menuconfig |
| 03:50.41 | [TK]D-Fender | kyron, First guess... zaptel is nto completely setup and configured. |
| 03:50.44 | glaz | kyron: I doubt it, there is only one extention on my side |
| 03:50.45 | kyron | channel drivers section |
| 03:50.56 | [TK]D-Fender | glaz, Show us something useful. |
| 03:51.09 | [TK]D-Fender | kyron, Go install Zaptel properly and load the modules, etc. |
| 03:51.27 | kyron | glaz, hey, you set it up... |
| 03:52.09 | glaz | [TK]D-Fender: tell me what you want to see, ill show you. The only thing I get is this error: |
| 03:52.13 | glaz | [Mar 27 23:51:26] WARNING[3317]: chan_sip.c:6713 determine_firstline_parts: Bad request protocol |
| 03:52.18 | kyron | well, weird thing is config.log shows me that the zaptel stuff tested out ok.. |
| 03:52.39 | [TK]D-Fender | glaz, Well, where is something telling you that your IAX setup isn't right? |
| 03:53.02 | glaz | iax2 show peers does not show me the host of my remote * |
| 03:53.13 | kyron | glaz, didn't you have an extension configured to work from your place (the Cisco you have at home)? |
| 03:53.45 | glaz | kyron: no, only an entry in sip.conf |
| 03:54.15 | kyron | glaz, show me the office sip.conf and extensions.conf |
| 03:55.18 | [TK]D-Fender | glaz, reload your remote system's iax.conf |
| 03:55.30 | [TK]D-Fender | glaz, enable iax2 debug and watch for traffic |
| 03:55.39 | glaz | there is, hold on |
| 03:56.37 | glaz | I see the Dcall |
| 03:56.53 | [TK]D-Fender | glaz, And you should have 2 entries on each side, not just 1. |
| 03:57.11 | [TK]D-Fender | glaz, You have no way to call from one to the other. Sure the register might work, but thats it. |
| 03:57.20 | [TK]D-Fender | (based on your pastebin) |
| 03:57.34 | glaz | the register doesnt work |
| 03:57.41 | [TK]D-Fender | glaz, Did you add auth info to your Dial statments directly? |
| 03:57.45 | glaz | REGREQ , but no REGACK on my remote site |
| 03:57.51 | glaz | no |
| 03:57.56 | [TK]D-Fender | glaz, Are you forwarding ports on both sides? |
| 03:58.06 | glaz | no |
| 03:58.15 | glaz | only one side is behind a linksys router |
| 03:58.15 | [TK]D-Fender | glaz, Theres a probelm then. |
| 03:58.31 | glaz | no need to forward ports on the other * server |
| 03:58.33 | [TK]D-Fender | glaz, and the one that is behind the router? |
| 03:58.35 | glaz | its DMZ |
| 03:58.48 | glaz | yes, port 4569 is forwared |
| 03:58.54 | [TK]D-Fender | glaz, pastebin the whole comm attempt |
| 03:59.17 | anonymouz666 | problem solved. |
| 03:59.19 | [TK]D-Fender | glaz, But first for testing, remove the permit/deny |
| 03:59.32 | glaz | ok |
| 03:59.43 | anonymouz666 | NEVER, EVER, put an ANSWER if you try to avoid the REINVITE. |
| 04:00.54 | glaz | http://rafb.net/p/riTH2j28.html |
| 04:01.00 | anonymouz666 | the damn reinvite was sent by Answer() app with nat=yes and canreinvite=no. |
| 04:01.04 | glaz | this is the output of iax set debug |
| 04:01.27 | glaz | its repeating this all the time |
| 04:02.59 | [TK]D-Fender | glaz, Check your firewalls..... |
| 04:03.19 | glaz | hold on, I will DMZ both side to see |
| 04:04.10 | [TK]D-Fender | glaz, why would you have to DMZ both? You jsut said only 1 was behind a router. |
| 04:04.59 | *** join/#asterisk shital (n=shital@122.167.84.153) |
| 04:05.06 | glaz | yeah, but you say check your firewalls.. |
| 04:05.08 | shital | hello all |
| 04:05.25 | glaz | what else I need? I have port 4569 forwarded |
| 04:05.41 | [TK]D-Fender | glaz, can you show me this forwarding? |
| 04:06.13 | [TK]D-Fender | glaz, and verify the firewall on each machine |
| 04:06.21 | glaz | [TK]D-Fender: an other curious thing, when I use sip set debug , I get this : http://rafb.net/p/MQqmnm49.html |
| 04:06.43 | glaz | the firewalls are fine, look at this paste, you will see the connection attemp from the remote * |
| 04:06.58 | glaz | what are those weird characters |
| 04:07.03 | glaz | .. |
| 04:07.21 | glaz | I can't paste those, but they are weird E |
| 04:07.38 | shital | give me the good site address in india from where i can by various Digium Cards |
| 04:08.09 | glaz | shital: www.digium.com , there is a reseller list. |
| 04:08.11 | [TK]D-Fender | glaz, SIP doesn't matter |
| 04:08.33 | glaz | [TK]D-Fender: why do my home * see the IAX connection attemp as a SIP connection attemp? |
| 04:13.52 | [TK]D-Fender | glaz, please pastebin your full sip.conf & iax.conf masking only plasswords |
| 04:13.58 | [TK]D-Fender | passwords |
| 04:14.51 | glaz | I got it |
| 04:15.10 | glaz | its my damn linksys router that does not like the packet |
| 04:15.36 | glaz | I remove it, put the box directly on the net, works fine. |
| 04:15.43 | kyron | glaz, but, if I get this right, you should still not be getting sip messages... |
| 04:16.13 | glaz | kyron: I think that if * doesnt recognize the type of packet that he recieves, he treats it as a SIP packet |
| 04:16.20 | kyron | glaz, hahah, that's one way of cutting out the crap ;) |
| 04:16.32 | kyron | (direct connection) |
| 04:16.57 | glaz | yup |
| 04:17.19 | glaz | so I gess I will bring an other mcable tomorrow dedicated for the * server |
| 04:17.20 | glaz | :) |
| 04:17.31 | shital | i need Digium cards which i can fix to my PC and able to connect analog phones extension 2 to 3 and one incoming line, suggest me which model no i should go for and where from i can buy it |
| 04:17.47 | [TK]D-Fender | glaz, Show me your attempt to forward at the router level. |
| 04:18.02 | glaz | [TK]D-Fender: you want a screenshot ^ |
| 04:18.03 | glaz | ? |
| 04:18.15 | [TK]D-Fender | glaz, Yes |
| 04:18.54 | [TK]D-Fender | shital, None. Get a Linksys SPA-2102 & SPA-3102. Far cheaper and mroe flexible |
| 04:19.47 | *** join/#asterisk Arkenazy (n=Admin@rrcs-76-79-244-108.west.biz.rr.com) |
| 04:20.00 | Arkenazy | hi can someone please tell me how to format a sound file for asterisk? |
| 04:20.20 | glaz | www.glaciuz.com/tmp/screen_fw.jpg |
| 04:21.18 | glaz | [TK]D-Fender: the thing is that I see the connection attemp in the logs |
| 04:23.04 | [TK]D-Fender | glaz, you pointed it to 5060! |
| 04:23.09 | [TK]D-Fender | glaz, WTF! |
| 04:23.11 | *** join/#asterisk coolblade (n=gregoryr@72.8.73.204) |
| 04:23.19 | shital | Fender: can i able to achieve all functionalities those i can achieve from digium cards with Linksys SPA-2102 & SPA-3102?? |
| 04:23.22 | [TK]D-Fender | glaz, thats so completely wrong its not funny |
| 04:23.37 | [TK]D-Fender | shital, that combo offers you 1 FXO, 3 FXS as you described |
| 04:24.02 | [TK]D-Fender | glaz, You are sending your incoming IAX packets to your * servers SIP port. |
| 04:24.04 | glaz | [TK]D-Fender: omg, please kill me |
| 04:24.06 | [TK]D-Fender | glaz, INSANITY |
| 04:24.15 | glaz | I did not do that on purpose... |
| 04:24.23 | glaz | hahahaha, my god. |
| 04:24.53 | glaz | I deserve to die. |
| 04:24.54 | kyron | glaz, O_o.... vat coucher dude... |
| 04:24.56 | kyron | LOOOL |
| 04:24.57 | coolblade | Hi guys, I am trying to get my first configuration of asterisk running but it has the sample dialplan loaded (as shown be dialplan show), I have replaced the sample dialplan with my own and run dialplan reload and then run dialplan show and it still shows the original dial plan, I looked at the file it says it is parsing and it has my new dialplan (/etc/asterisk/extensions.conf), can anybody give me a clue? |
| 04:24.58 | kyron | pwahahaha |
| 04:25.00 | [TK]D-Fender | glaz, No no no.... I only offer LIFE threats..... |
| 04:25.14 | kyron | [TK]D-Fender, that's even WORSE! |
| 04:25.18 | glaz | [TK]D-Fender: hahaha |
| 04:25.20 | glaz | DAMN IT |
| 04:25.24 | [TK]D-Fender | kyron, Indeed. |
| 04:25.26 | glaz | I am so shy... |
| 04:25.37 | kyron | HAHAHAHA |
| 04:25.38 | glaz | runs |
| 04:25.48 | shital | thank you Fender: i can able to fix Linksys SPA-2102 & SPA-3102 to my PC PCI slots rt? |
| 04:25.49 | kyron | glaz, I won't tell them where you work and what you do |
| 04:25.55 | kyron | ....for a small fee...PWahahahahaha |
| 04:26.01 | kyron | bon...really, good night all |
| 04:26.07 | glaz | that is because the first time i tried to configure it, I wanted to make the phone directly connect to the other * server |
| 04:26.15 | glaz | haha |
| 04:26.20 | glaz | good night. |
| 04:26.21 | [TK]D-Fender | coolblade, pastebin "cat /etc/asterisk/extensions.conf" from linux CLI, then add the CLI attempt of you trying to reload the dialplan, and then add "dialplan show" |
| 04:26.41 | [TK]D-Fender | shital, No. These are not PCI cards. There are SIP ATAs |
| 04:26.47 | shital | i need to test these with asterisk PBX |
| 04:27.21 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 04:27.43 | [TK]D-Fender | shital, you plug them into your network swith and you will use SIP to have them communicate with * |
| 04:28.14 | shital | so i need PCI cards itself, what u say about TDM400P card?? |
| 04:28.38 | jameswf-home | ~tdm400p |
| 04:28.38 | jbot | it has been said that tdm400p is http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
| 04:28.52 | [TK]D-Fender | shital, Get the newer TDM410P instead |
| 04:28.55 | shital | its also having 3 fxs and 1 fxo morever its PCI card |
| 04:28.57 | glaz | [TK]D-Fender: everything works fine trought the firewall now, thanks :S |
| 04:29.22 | glaz | I apologyze for being such an idiot tonight. I would had laugh at any of my friends for doing this. |
| 04:29.30 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:29.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:29.41 | [TK]D-Fender | glaz, bienvenu, pis cherche-toi donc un cafe :p |
| 04:29.42 | coolblade | [TK[D-Fender: http://pastebin.com/d4a21fad3 |
| 04:29.43 | shital | can u give me good site address from india where i can get that? |
| 04:29.59 | glaz | [TK]D-Fender: I'll go to sleep instead |
| 04:30.07 | glaz | thanks, later. |
| 04:30.34 | [TK]D-Fender | shital, www.voip-info.org <- go look for a reseller |
| 04:30.45 | shital | ok |
| 04:31.37 | [TK]D-Fender | coolblade, looks fine. Whats the problem? |
| 04:32.19 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:32.47 | coolblade | Hmm, i thought the dial plan show is showing allot more then my 10 line dial plan, is it just allot of meta stuff I dont understand then? I am trying to connect my phone to my tdm fxs card but not getting a dial tone |
| 04:33.01 | coolblade | i am a complete novice |
| 04:33.53 | jameswf-home | ls -R | grep ":$" | sed -e 's/:$//' -e 's/[^-][^\/]*\//--/g' -e 's/^/ /' -e 's/-/|/' |
| 04:33.57 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 04:34.49 | [TK]D-Fender | coolblade, you have an extensions.ael that is getting loaded by "pbx_ael.so" if you don't need AEL (nobody does really), just add "noload => pbx_ael.so" to modules.conf |
| 04:35.59 | _gm | how pstn works |
| 04:36.11 | [TK]D-Fender | coolblade, There goes your "dialplan" question. Now as for not getting dialtone, pastebin * CLI "zap show channels", linux CLI : "cat /proc/zaptel/*" , "cat /proc/interrupts", "cat /etc/asterisk/zapata.conf" , "cat /etc/zaptel.conf" |
| 04:36.20 | [TK]D-Fender | _gm, Well. |
| 04:36.24 | _gm | :) |
| 04:36.44 | shital | Fender: Digium TDM410P Base PCI Card, i think this do not have FXS and FXO modules we need to fix it up right? |
| 04:36.46 | _gm | i have to present "IP Telephony" tomorrow |
| 04:36.55 | _gm | a detailed presentation |
| 04:36.57 | [TK]D-Fender | shital, not by that name it doesn't. |
| 04:37.01 | *** join/#asterisk fedya (n=fedya@c-76-26-182-13.hsd1.fl.comcast.net) |
| 04:37.19 | coolblade | [TK]D-Fender, ok this will take a second to get all of those, I do appreciate you taking the time to help a newb. |
| 04:37.29 | [TK]D-Fender | _gm, How can you explain IP telephony without understanding conventional telephony? |
| 04:37.52 | _gm | yeah |
| 04:37.56 | _gm | [TK]D-Fender: hm |
| 04:38.18 | _gm | i do understand stuff, but u know i have to prepare slides and speak, |
| 04:38.30 | [TK]D-Fender | _gm, I wish *I* could get paid to talk about things I know nothing about.... |
| 04:38.33 | shital | Fender :means i need to buy seperate FXO and FXS cards rt? |
| 04:38.47 | [TK]D-Fender | shital, probably |
| 04:38.58 | _gm | i dont want anything wrong in my slides, not a single point ;) |
| 04:39.11 | [TK]D-Fender | _gm, Get to work then.... |
| 04:39.11 | _gm | [TK]D-Fender: btw, i m not getting paid for this, hehe |
| 04:39.35 | [TK]D-Fender | _gm, Then tell them that if you're wrong you'll give them DOUBLE their money back :o |
| 04:39.37 | _gm | [TK]D-Fender: reading this: http://en.wikipedia.org/wiki/Time-division_multiplexing |
| 04:39.44 | [TK]D-Fender | _gm : |
| 04:39.44 | _gm | haha |
| 04:39.45 | [TK]D-Fender | ~101 |
| 04:39.46 | jbot | hmm... 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 04:39.49 | [TK]D-Fender | ^^^^^^^ |
| 04:40.09 | _gm | nice |
| 04:40.58 | _gm | [TK]D-Fender: thanks |
| 04:41.20 | shital | Fender: plz tell me card which has inbuilt required modules |
| 04:41.46 | coolblade | [TK]D-Fender: http://pastebin.com/m26674a36 |
| 04:41.50 | [TK]D-Fender | shital, Seriously go look at that reseller and if you can't figure it out, CALL THEM. |
| 04:44.32 | coolblade | shital have you looked at digums website? |
| 04:44.45 | coolblade | they should list all the different model numbers that have which modules |
| 04:44.51 | [TK]D-Fender | coolblade, permanently remove EVERY commented out line from zaptel.conf & zapata.conf and repastebin them along with * CLI's "reload chan_zap.so" |
| 04:45.10 | coolblade | [TK]D-Fender, you got it, give me a min |
| 04:48.18 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:52.59 | coolblade | [TK]D-Fender: umm... wow your a genius... |
| 04:53.02 | coolblade | it fixed it |
| 04:53.06 | coolblade | lol |
| 04:53.32 | CCFL_Man2 | my $132 2U rackmount case comes tomorrow |
| 04:53.39 | [TK]D-Fender | coolblade, I'm betting you needed to reload chan_zap after adding some of your config options for them to take effect... |
| 04:53.49 | [TK]D-Fender | CCFL_Man2, Cool. retail? |
| 04:54.05 | CCFL_Man2 | [TK]D-Fender: yes, from newegg |
| 04:54.08 | coolblade | reload chan_zap? |
| 04:54.19 | [TK]D-Fender | CCFL_Man2, link it. |
| 04:54.29 | [TK]D-Fender | coolblade, nevermind. Glad its working. |
| 04:54.42 | tengulre | how to using crontab running a command per half of hour |
| 04:55.06 | coolblade | lol |
| 04:55.43 | coolblade | crontab -e |
| 04:55.50 | CCFL_Man2 | [TK]D-Fender: it's a cheaper unit |
| 04:55.53 | CCFL_Man2 | $132 |
| 04:55.54 | *** join/#asterisk SwK (n=SwK@m875f36d0.tmodns.net) |
| 04:56.17 | coolblade | 30 * * * * command (IIRC) |
| 04:56.56 | CCFL_Man2 | [TK]D-Fender: http://www.newegg.com/Product/Product.aspx?Item=N82E16811192045 |
| 04:57.37 | CCFL_Man2 | shortest i could find with a 3.5" drive bay and a 5.25 drive bay |
| 04:59.03 | coolblade | [TD]D-Fender, not sure what fixed it but I really do appreciate your help. |
| 04:59.25 | [TK]D-Fender | coolblade, You're welcome |
| 04:59.40 | [TK]D-Fender | CCFL_Man2, How it does the job. What are you doing with it? |
| 05:01.30 | CCFL_Man2 | [TK]D-Fender: well, i have an 800MHz emachines with 64mb ram that runs freebsd, i'm tired of having the silly case on the floor |
| 05:02.28 | [TK]D-Fender | CCFL_Man2, well if you've got the rack, more power to you... |
| 05:03.17 | shital | coolblade: yaa i looked |
| 05:03.47 | CCFL_Man2 | [TK]D-Fender: i screwed rack rails to the side of a 1970s home audio rack |
| 05:04.04 | CCFL_Man2 | works pretty well |
| 05:05.16 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:05.21 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 05:05.46 | CCFL_Man2 | for stuff that needs to be mounted in the back i'll use a shelf for support |
| 05:06.36 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 05:06.50 | CCFL_Man2 | because it's not long enough |
| 05:12.34 | *** part/#asterisk SwK (n=SwK@m875f36d0.tmodns.net) |
| 05:13.22 | *** join/#asterisk dongs (i=500@l212168.ppp.asahi-net.or.jp) |
| 05:14.11 | dongs | umso someone is calling me using softphone possibly behind nat, and it sounds like their voice is like rrrrrrrrrrrrreeeeeeeeeeeeeaaaaaaaaaaallllllllllllllyyyyyyyyyyy slow. the pitch is ok, but its like mega slowed down. whats the issue? (this worked like.. yesterday). |
| 05:17.42 | *** join/#asterisk Olobola (i=Olobola@60.sub-75-210-210.myvzw.com) |
| 05:18.58 | dongs | is asterisk officially dead |
| 05:20.16 | _gm | dongs, ??? |
| 05:24.15 | Olobola | asterisk is dead, check it out: * -rv !! |
| 05:26.59 | [TK]D-Fender | ok, checking out, later all |
| 05:27.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:31.31 | Nasra | !help |
| 05:47.47 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 06:00.16 | *** join/#asterisk BeeBuu (n=beebuu@219.132.191.239) |
| 06:04.28 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 06:05.58 | *** join/#asterisk nighty^ (n=nighty@210.188.173.249) |
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| 06:25.22 | *** part/#asterisk dongs (i=500@l212168.ppp.asahi-net.or.jp) |
| 06:27.28 | *** join/#asterisk nighty^ (n=nighty@210.188.173.249) |
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| 06:36.31 | *** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net) |
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| 06:38.52 | awk | is anyone else having issues with ringinuse=no on queues, still passing calls to an agent with state inuse? |
| 06:41.48 | *** join/#asterisk nirz (n=nir@80.74.118.81) |
| 06:44.34 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 06:44.55 | santoshr | Has anybody been able to get g723 to run on asterisk |
| 06:45.09 | santoshr | my show translation shows a "-" for 723 |
| 06:45.36 | *** join/#asterisk fedya (n=fedya@c-76-26-182-13.hsd1.fl.comcast.net) |
| 06:46.07 | BBHoss | santoshr, g723 is another one of those patented codecs |
| 06:46.42 | santoshr | yes but they are available for download if u dont want to use them for commercial purposes |
| 06:46.44 | BBHoss | you can get it here: http://asterisk.hosting.lv/#bin |
| 06:47.01 | santoshr | i was able to get 729 to run.. but 723 .. no luch |
| 06:47.04 | BBHoss | its actually illegal even for personal use |
| 06:47.14 | BBHoss | its a really gray area |
| 06:47.17 | santoshr | s/luch/luck/ |
| 06:47.51 | santoshr | the bot is cool. what all does i do.. ? |
| 06:49.08 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 06:49.27 | BBHoss | dunno, the replace thing and some definitions |
| 06:49.35 | BBHoss | ~grandstream |
| 06:49.35 | jbot | extra, extra, read all about it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 06:49.46 | santoshr | hehe |
| 06:49.52 | BBHoss | ~yousuck |
| 06:50.23 | santoshr | whts the difference between ast14-gcc4 and ast14icc |
| 06:50.32 | BBHoss | different compilers |
| 06:50.40 | BBHoss | gcc4 and intel's compiler |
| 06:50.46 | santoshr | this is mine.. which one should i useLinux mail002 2.6.11 #7 SMP Mon Dec 4 12:41:53 IST 2006 i686 unknown unknown GNU/Linux |
| 06:51.02 | BBHoss | i always use gcc |
| 06:51.20 | santoshr | <PROTECTED> |
| 06:53.43 | *** join/#asterisk airjump (n=zielonka@62.159.95.82) |
| 06:57.22 | santoshr | BBHoss: and it works thanks bro |
| 06:57.46 | BBHoss | np |
| 06:57.57 | BBHoss | have fun "experimenting" :) :) |
| 06:59.22 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:06.43 | santoshr | ata -- > gnugk --> asteriks --> remote IP in this scenario will be able to achieve reinvite and send /recieve rtp from ata to remote ip |
| 07:07.25 | BBHoss | santoshr, you using gnugk for a reason, h323? |
| 07:07.30 | santoshr | i meant is this possible reinvite twice.. gnugk does it bit and i c packets from ata to * |
| 07:08.05 | santoshr | yes. and registrations are the issue need a gatekeeper in between |
| 07:08.48 | BBHoss | why? do you have like 100000 registrations? why not just use asterisk to register? |
| 07:09.48 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 07:11.19 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:14.13 | santoshr | but i read on the wiki that asterisk cannot be used as a gatekeeper. it can only be used as a gateway |
| 07:15.11 | *** join/#asterisk sucram2g (n=markus@41.222.0.61) |
| 07:15.38 | santoshr | i have both h323 and sip ata. sip ata's register to asterisk and h323 ata's register to gnugk |
| 07:16.05 | sucram2g | Hi, i want to interconnect asterisk with a cisco router |
| 07:27.34 | *** join/#asterisk shinao1 (n=shinao1@35-230.rv.ipnxtelecoms.com) |
| 07:29.04 | *** part/#asterisk sucram2g (n=markus@41.222.0.61) |
| 07:29.21 | *** join/#asterisk adjohn (n=adjohn@i219-167-226-232.s05.a013.ap.plala.or.jp) |
| 07:37.48 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 07:46.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:46.32 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:55.06 | pa | is it possible to set up iax.conf or sip.conf with an extension without username and password, but which instead do database (or ldap, or radius) lookup for autentication? |
| 07:55.21 | pa | i saw only for password and only with asteriskdb |
| 07:58.39 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 07:59.21 | *** join/#asterisk Dextorion (n=dex@fingerbottom.tekproj.bth.se) |
| 08:00.41 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:04.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 08:05.31 | santoshr | realtime sip |
| 08:05.33 | *** join/#asterisk Tuxofred (n=Fred@ip-80-236-192-102.dsl.scarlet.be) |
| 08:05.46 | Tuxofred | Hello |
| 08:05.50 | Tuxofred | how are you |
| 08:06.04 | Tuxofred | is it possible to refuse anonymous call? |
| 08:06.08 | santoshr | pa: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 08:09.31 | mort_gib | Morning |
| 08:10.07 | mort_gib | Is there any reason why busylimit=1 is not working in sip.conf ?? |
| 08:11.14 | pa | thanks |
| 08:13.02 | pa | sorry, if i get this: Rejected connect attempt from 88.33.148.98, request '44@default' does not exist, does it mean i authenticated successfully? while if i get Rejected connect attempt from 88.33.148.98, who was trying to reach '11@', does it mean i didnt autenticate? |
| 08:16.24 | *** join/#asterisk Grash (n=grash@244.47.217.87.dynamic.jazztel.es) |
| 08:29.52 | yang | What kind of GUI has asterisknow ? |
| 08:29.57 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 08:33.08 | *** join/#asterisk GBLRA77 (n=GBLRA77@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:33.21 | *** join/#asterisk amaache (n=maache76@41.221.27.181) |
| 08:36.23 | Tuxofred | i'd like to refuse anonymous calls, is it possible? |
| 08:36.42 | Tuxofred | i'm using *71 on sip phone, but it can't work |
| 08:38.34 | *** join/#asterisk dcmwai (n=dcmwai@60.54.46.150) |
| 08:57.32 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:57.44 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
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| 09:22.08 | *** join/#asterisk Dextorion (n=dex@fingerbottom.tekproj.bth.se) |
| 09:24.55 | Dextorion | i only what to use sip and local softphones. only internal numbers. Do i still need a service provider? |
| 09:27.29 | mvanbaak | Dextorion: no |
| 09:32.05 | pa | yang, it has asterisk-gui |
| 09:32.44 | *** join/#asterisk stony (n=oloch@p57B39AD2.dip0.t-ipconnect.de) |
| 09:32.45 | stony | hi |
| 09:32.47 | pa | is it possible to do callerID blocking based on a custom database instead of the default one? |
| 09:32.54 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:32.59 | stony | is there a way to test the scripts/macros without calling the pbx each time ? |
| 09:33.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:33.40 | stony | pa: just check the callerid against your database before answering the cannel and play a "caller not available" if it matches your list |
| 09:34.14 | Dextorion | okay mvanbaak, however, im trying out asterisknow, and trying to create a calling rule. But there i have to enter a service provider to place the call through. Any ideas what is needed here? |
| 09:36.07 | pa | stony, but to check againist your database you mean using a kind of script called through System()? |
| 09:36.40 | stony | pa: f.e. or do an sql query in the dialplan |
| 09:36.48 | pa | ah! |
| 09:36.51 | stony | pa: or run a agi script ... |
| 09:36.52 | *** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 09:36.52 | pa | is that possible |
| 09:36.53 | pa | ? |
| 09:36.58 | stony | pa: sure |
| 09:37.01 | pa | wow |
| 09:37.02 | pa | thanks |
| 09:37.12 | stony | pa: f.e. check mysql() on voip-info.org |
| 09:39.44 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:40.10 | madduck | can someone explain to me the relation of my extensions.conf file from the 1.2 installation and the extensions.ael file from 1.4? |
| 09:40.37 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:41.48 | JT | ael is optional |
| 09:41.55 | JT | it's a different way of writing dialplans |
| 09:41.58 | JT | it's in 1.2 too |
| 09:42.07 | JT | afaik |
| 09:42.21 | madduck | i like the syntax better. |
| 09:42.39 | *** join/#asterisk razu__ (n=razu@razu.data.ee) |
| 09:42.43 | JT | just beware it doesn't have any inherent performance benefit |
| 09:42.52 | JT | ael is parsed bacl to standard syntax internally by asterisk |
| 09:42.56 | madduck | of course not. but it seems more logical |
| 09:43.00 | JT | s/bacl/back/ |
| 09:43.06 | madduck | ok |
| 09:43.10 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:45.54 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
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| 09:49.37 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 09:49.53 | ice_croft | hi all |
| 09:50.09 | ice_croft | how can i comment a section of code in sip.conf? |
| 09:50.26 | ice_croft | not a single string, but a several of them? |
| 09:51.31 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 09:52.02 | sysadmin-lb22 | hi all..I need around 15 fxs lines for my asterisk box..what digium card must I use ? |
| 09:55.18 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 10:00.34 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 10:08.47 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:09.37 | *** join/#asterisk bminish (n=bminish@78.152.234.102) |
| 10:09.50 | madduck | so i have a conceptual problem i think. in sip.conf, i understand user authenticates by credentials and host by IP |
| 10:10.06 | madduck | but i can't get anything else to work than peer |
| 10:10.23 | madduck | and "host" by IP should have been peer |
| 10:10.29 | madduck | i can't get anything else to work than peer |
| 10:10.37 | madduck | and that works even if host=dynamic |
| 10:11.15 | *** join/#asterisk viperdude (n=user@87-127-248-101.no-dns-yet.enta.net) |
| 10:13.15 | madduck | so with type=user and host=dynamic, I am told |
| 10:13.28 | madduck | Registration from '"martin" <sip:piper@madduck.net>' failed for '192.168.14.3' - No matching peer foun |
| 10:13.54 | madduck | and so i set it to type=peer and get |
| 10:13.55 | madduck | Registration from '"martin" <sip:piper@madduck.net>' failed for '192.168.14.3' - Peer is not supposed to register |
| 10:16.14 | *** join/#asterisk Scrounch (n=Scrounch@unaffiliated/scotchi) |
| 10:16.21 | Scrounch | hi all, salut |
| 10:16.52 | Scrounch | anyone here ? |
| 10:16.57 | madduck | so the only way i can make it work is with type=friend and host=dynamic |
| 10:17.49 | Scrounch | Si quelqu'un parle francais... car ce n'est pas simple |
| 10:18.38 | madduck | Scrounch: c'est mieux essayer que demander |
| 10:19.01 | Scrounch | i would like to use my fai phone (tele2) with asterisk for voip with a modem classic phone |
| 10:19.56 | Scrounch | en fait le telephone n'as qu'une fiche de telephone "normal" et j'aimerais qu'un pc remplace cette box pourrie de tele2 |
| 10:21.30 | madduck | so you want to use a PC/softphone instead? |
| 10:22.38 | Scrounch | i have this =< |
| 10:22.40 | Scrounch | http://www.tele2.be/nl/internet/adsl4m/easybox/index.html |
| 10:23.29 | Scrounch | i want to use a pc because i need to close all computer for call with telephone (bug?) |
| 10:25.33 | stoffell | is it normal behaviour that RTCP traffic does not use the TOS bit I set in sip.conf ? RTP traffic does set it, so that is ok. |
| 10:25.46 | Scrounch | i want to use tele2 voip with a computer for telephone, but the phone jack is normal phone, then i want to plug the phone in a modem classical |
| 10:28.56 | Scrounch | plug normal telephone in http://i22.ebayimg.com/06/i/000/b9/ff/81d4_1.JPG and call with fai's voip, that's possible ? |
| 10:30.32 | madduck | ah, you want to use a modem to connect a PC to the phone jack of the internet router? |
| 10:31.52 | Scrounch | internet (with fai voip)-> pc router ->pci V90 ->telephone (fai) |
| 10:32.57 | JT | no, you cannot use modems for asterisk |
| 10:32.59 | Scrounch | telephone's not USB, but RJ11 |
| 10:33.40 | Scrounch | okay |
| 10:35.30 | *** join/#asterisk pekdon (n=noosnasc@ips-fw-01.ipsoft.no) |
| 10:35.44 | *** part/#asterisk pekdon (n=noosnasc@ips-fw-01.ipsoft.no) |
| 10:36.17 | *** join/#asterisk maveric1 (n=root@83.240.229.38) |
| 10:36.27 | maveric1 | hello |
| 10:36.32 | Scrounch | hi |
| 10:36.36 | maveric1 | could someone |
| 10:36.49 | maveric1 | tell me if exists tools |
| 10:36.54 | maveric1 | to test asterisk security |
| 10:36.59 | pa | umh.. im using asterisk on ubuntu. i placed a file in gsm format in /var/lib/asterisk/sounds, named both chiamanteanonimo and chiamanteanonimo.gsm |
| 10:37.06 | pa | but i keep getting File chiamanteanonimo does not exist in any format |
| 10:37.54 | JT | pa: what is your extensions.conf line? |
| 10:38.26 | pa | exten => 631522,2,Playback(chiamanteanonimo) |
| 10:38.51 | JT | seems ok |
| 10:38.59 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 10:39.05 | pa | actually, i have the stock asterisk.conf |
| 10:39.24 | pa | and i googled, and seems that sounds by defaults go in /var/lib/asterisk/sounds |
| 10:41.19 | maveric1 | i'm looking for some tools to test asterick security |
| 10:41.48 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 10:42.26 | madduck | pa: on ubuntu, it's surely /usr/share/asterisk something |
| 10:42.32 | madduck | or /etc |
| 10:42.36 | madduck | not /var |
| 10:44.14 | pa | ah thanks! |
| 10:46.11 | pa | yeah now it works |
| 10:47.11 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 10:56.10 | *** join/#asterisk viperdude (n=user@87-127-248-176.no-dns-yet.enta.net) |
| 10:59.02 | pa | stupid question: does openwengo work well with asterisk sip accounts? |
| 11:01.37 | stony | anyone got a link to the documentation of "$[ ]" ? |
| 11:02.28 | stony | ah got it |
| 11:07.35 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:11.06 | madduck | pa: yes |
| 11:11.34 | madduck | if i have asterisk running on a gateway with a public IP and some NATted phones behind it, do I need to set nat=yes for them? |
| 11:12.03 | madduck | I am thinking no because there is no NAT between them and asterisk, but there is between them and the peer when they use RTP for calls. |
| 11:14.42 | Dextorion | how can i add user from the asterisk CLI? |
| 11:15.21 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 11:16.12 | _gm | Dextorion: you can't |
| 11:16.17 | mvanbaak | hhmm |
| 11:16.21 | _gm | Dextorion: you can add exten though |
| 11:16.25 | mvanbaak | if I have a PRI |
| 11:16.31 | mvanbaak | and everything is setup correctly |
| 11:16.31 | Dextorion | oh.. |
| 11:16.36 | mvanbaak | but.... |
| 11:16.42 | mvanbaak | on incoming calls I loose the first 0 |
| 11:16.44 | mvanbaak | how to fix it |
| 11:17.17 | mvanbaak | <PROTECTED> |
| 11:17.27 | mvanbaak | it should be 070....... |
| 11:19.43 | Dextorion | _gm, very very new to all this. A user and extension is two different things then.. A user can have more then one extension? |
| 11:21.14 | *** join/#asterisk duckz (n=duckz@81-180-102-217.etth.opensys.ro) |
| 11:22.57 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:23.40 | madduck | ok, so rtp goes via asterisk, not past it, if two phones talk to each other, right? |
| 11:25.58 | stoffell | madduck, depends on reinvite parameter in the sip config |
| 11:26.38 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:26.52 | madduck | stoffell: ah, so only if canreinvite=no? if i set that to yes, then the RTP packets will go directly? |
| 11:27.17 | oej | it also depends on the call configuration |
| 11:27.27 | madduck | what's the call configuration? |
| 11:27.30 | oej | If Asterisk needs to listen for DTMF, we can't run external bridge |
| 11:27.39 | madduck | hm |
| 11:27.43 | oej | Like you add a "T" in the dial string, or enable features in features.conf |
| 11:28.55 | madduck | so for an internal set of phones, which are primarily used to receive and make external calls, going via asterisk on the gateway, it's best to set reinvite=no to make them all go via asterisk? |
| 11:28.59 | mvanbaak | someone ? |
| 11:29.01 | madduck | for then I don't have to worry about NAT? |
| 11:29.14 | madduck | mvanbaak: check dialplan. search for EXTEN |
| 11:29.52 | oej | yes, absolutely, madduck |
| 11:29.58 | madduck | great |
| 11:30.19 | oej | the config name is "canreinvite" btw, not "reinvite" |
| 11:30.24 | madduck | yeah. :L) |
| 11:30.39 | madduck | so another quick question... i am restarting from scratch with the * config |
| 11:30.45 | madduck | because i had weird problems with one-way |
| 11:30.55 | madduck | and decided to understand it all rather than poking at it blindly |
| 11:31.04 | madduck | so now i have asterisk 1.4 running again and two phones on it |
| 11:31.15 | madduck | but already if i make a call from one to the other, it's one way. wtf? |
| 11:31.30 | madduck | they're both in the same network and directly connected to asterisk |
| 11:31.41 | madduck | and identically configured |
| 11:31.45 | madduck | one is a twinkle softphone |
| 11:31.49 | madduck | and the other a siemens c450ip |
| 11:31.59 | madduck | the siemens can't hear what twinkle transmits |
| 11:32.04 | madduck | the other way works |
| 11:32.39 | madduck | though curiously only if twinkle calls the siemens |
| 11:32.50 | madduck | if siemens calls twinkle, it's bidirectional alright |
| 11:33.17 | madduck | (and Packet2Packet bridging SIP/piper-0816a5e0 and SIP/c450ip-08168e68 means that asterisk is being an rtp relay, right?) |
| 11:42.42 | loompek | exten => 1,1,Dial(SIP/${EXTEN},60,rm) |
| 11:42.58 | loompek | this should generate both dial tone and music on hold.. right? |
| 11:45.28 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:58.35 | _gm | Dextorion: it's all up to your dialplan (extensions.conf)'s routing, you can have anynumber of extension's for a user |
| 11:59.38 | _gm | Dextorion: you define an extension in extensions.conf, and tell it what to do, either call a sip user, either execute a php script or play a file or music on hold , queues, conferences |
| 12:00.16 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 12:00.31 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 12:01.06 | Zeeek | it is now 8 AM EDT. That means the VoIP Users Conference (formerly asterisk user conference) starts in ummmmmm 4 hours at 12 Noon EDT |
| 12:02.13 | *** join/#asterisk MattJ (n=MattJ@88-110-69-232.dynamic.dsl.as9105.com) |
| 12:04.56 | viperdude | Zeeek: what is the URL for this conference, i am interested in it |
| 12:05.12 | Zeeek | why I am glad uyou asked that question |
| 12:05.23 | Zeeek | http://VoipUsersConference.org |
| 12:05.26 | viperdude | lol |
| 12:05.55 | Zeeek | This week is about astlinux |
| 12:05.59 | viperdude | hmm interesting, thanks for the tip |
| 12:06.18 | Zeeek | fascinating subject and comes with a free download |
| 12:06.24 | Zeeek | several free downloads in fact |
| 12:06.31 | Zeeek | thousands of free downloads even |
| 12:06.45 | viperdude | is there a RSS feed of the recordings? |
| 12:07.03 | Zeeek | I'm glad you asked that questions, there is a RSS feed, several in fact |
| 12:07.27 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 12:07.30 | Zeeek | http://feeds.feedburner.com/AstUSer |
| 12:08.08 | Zeeek | several others can be found here: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 |
| 12:08.27 | madduck | any idea why a siemens c450ip and a twinkle softphone, both directly connected to the same asterisk and on the same net with canreinvite=no, can talk alright if the siemens calls the softphone, but if the softphone calls the siemens, then only voice from siemens (callee) to softphone (caller) is transmitted, not the other way around? |
| 12:08.27 | viperdude | thanks |
| 12:08.30 | Zeeek | also the countdown timer says the conference is now in 3h49m30s |
| 12:09.05 | madduck | they can talk fine in all scenarios if canreinvite=yes |
| 12:09.36 | Zeeek | viperdude no roundup of info would be complete without saying the PSTN (724) 444-7444 and the entry code 22622# 1# |
| 12:10.27 | Zeeek | madduck a wild guess, check with sip debug, but the phone is sending the wrong ip |
| 12:11.22 | madduck | good idea |
| 12:11.52 | Katty | yawn. |
| 12:12.00 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:12.13 | Zeeek | Katty at least cover your mouth! |
| 12:12.21 | Katty | too sleepy |
| 12:12.23 | Katty | can't move |
| 12:12.31 | *** join/#asterisk Angel-D (n=angeldav@190.72.130.192) |
| 12:12.37 | Zeeek | typing in a horizontal position? |
| 12:12.43 | Katty | no |
| 12:12.48 | Katty | i'm zombied on the couch |
| 12:12.59 | Zeeek | core show zombies |
| 12:13.03 | Zeeek | I don't see you |
| 12:13.12 | Katty | of course you can't |
| 12:13.16 | Katty | all the curtains are drawn |
| 12:13.28 | Zeeek | you really like red that much? |
| 12:13.43 | Katty | mew? |
| 12:13.55 | Zeeek | we say "miao" |
| 12:14.52 | Katty | i was mewing before it was loled. |
| 12:15.14 | Katty | i should get redbull. yes. |
| 12:19.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:20.05 | *** join/#asterisk the_5th_wheel (n=edd@41.223.244.9) |
| 12:20.29 | shasta | is BLF possible with dynamic realtime? |
| 12:20.36 | madduck | Zeeek: which phone would be sending the wrong IP? the caller or the callee? |
| 12:22.03 | Zeeek | either could be. That's why there's sip debug ;) |
| 12:22.14 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 12:22.54 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 12:23.08 | madduck | well, i really just cannot find any problem in there |
| 12:23.17 | madduck | the routing is via the asterisk bridge |
| 12:23.25 | madduck | and asterisk seems to know both sides |
| 12:23.28 | madduck | Peer audio RTP is at port 192.168.14.30:8482 |
| 12:24.36 | madduck | Peer audio RTP is at port 192.168.14.3:8000 |
| 12:27.04 | the_5th_wheel | i have a sip phone, where the phone can make external phonecalls, can receive external phonecalls, but can hear any audio on local phone calls |
| 12:27.26 | Zeeek | I had a very odd problem once. THe audio was one way for exactly 10 seconds! Then the call was normal. |
| 12:28.14 | the_5th_wheel | nah, even after a minute there is nothing |
| 12:28.38 | the_5th_wheel | the phone is outside our office, connection via a DSL |
| 12:29.25 | madduck | Zeeek: nah, here it stays like that forever |
| 12:29.35 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:29.56 | the_5th_wheel | oh, that wasnt aimed at me |
| 12:30.14 | Zeeek | it could have been |
| 12:30.15 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 12:30.18 | the_5th_wheel | true |
| 12:30.25 | Zeeek | pôint being, I had 1way for the first 10 secs of a call |
| 12:30.46 | Zeeek | that was in IAX though. It seems to have been the phoe's IAX implementation |
| 12:36.07 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 12:37.20 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 12:41.30 | santoshr | i have dial statement like this dial(H323/12345678@ip) .. is this syntax wrong because this works for SIp. |
| 12:41.39 | *** join/#asterisk southtel (n=southtel@24-240-24-20.dhcp.gwnt.ga.charter.com) |
| 12:41.43 | santoshr | i get " -- Requested transfer capability: 0x00 - SPEECH" and the call disconnects |
| 12:41.47 | santoshr | am i doing something wrong |
| 12:42.09 | *** join/#asterisk southtel (n=southtel@24-240-24-20.dhcp.gwnt.ga.charter.com) |
| 12:43.07 | *** join/#asterisk rafiks (n=rafiks@c-68-56-151-58.hsd1.fl.comcast.net) |
| 12:43.34 | coppice | do you feel guilty? that's usually an indication you are doing something wrong :-) |
| 12:45.11 | santoshr | i have no clue. I mean i am not able to see any packets headed to the IP in the dial statement.. |
| 12:46.58 | madduck | Zeeek: i may be getting closer. while alaw is being selected if A calls B (which works bidirectionally), GSM is being selected if B calls A (and I just get one-way) |
| 12:47.25 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 12:48.21 | rafiks | hi! is there a fix to zaptelnot building on 2.6.24 |
| 12:48.36 | madduck | Zeeek: and in fact, if I disable gsm in the softphone, it works. |
| 12:48.56 | madduck | the c450ip actually does not have GSM support |
| 12:49.14 | madduck | so now the only question I have left is why it can apparently send data to the softphone |
| 12:49.22 | madduck | even though it can't do GSM |
| 12:49.37 | madduck | aha, it's *not* |
| 12:49.50 | madduck | softphone sends GSM, phone answers with ulaw |
| 12:49.52 | madduck | omg |
| 12:51.12 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 12:51.18 | madduck | wireshark for the win! |
| 12:51.47 | *** join/#asterisk PepOSX (n=angeldav@190.72.130.192) |
| 12:53.10 | *** join/#asterisk solaries (n=wr@kagu.c-lab.de) |
| 12:53.54 | madduck | any reason not to just limit allowed codecs to *law? G.711? |
| 12:54.45 | *** join/#asterisk karlh627 (n=kharris@IP-66.249.227.5.indigital.net) |
| 12:54.46 | Zeeek | I good start is using ONLY ulaw and alaw and then adding gradually |
| 12:55.27 | Zeeek | once that works |
| 12:57.28 | madduck | the weird thing is that the softphone still insists on using GSM |
| 12:57.33 | madduck | let me check the SDP |
| 12:57.39 | madduck | it looks like a bug |
| 12:57.49 | Zeeek | what softphone? |
| 12:58.08 | madduck | twinkle |
| 12:58.16 | madduck | (i wish i had a different one at my disposal) |
| 12:58.20 | madduck | x-lite is !amd64 |
| 12:58.30 | Zeeek | never hoid of it |
| 12:58.35 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 12:58.48 | madduck | can you suggest one? ekiga is also buggy |
| 12:58.55 | Zeeek | zoiper? |
| 12:59.42 | madduck | will check it out |
| 12:59.43 | coppice | if it ain't buggy, it ain't VoIP :-) |
| 12:59.50 | madduck | but it looks like the bug is elsewhere |
| 12:59.54 | Zeeek | statement of the day! |
| 13:00.14 | *** join/#asterisk jamshid (i=dce3806f@gateway/web/ajax/mibbit.com/x-0746ee15696690d7) |
| 13:00.14 | madduck | twinkle sends a SDP announcing speex, alaw, ulaw, gsm |
| 13:00.17 | madduck | in that order |
| 13:00.32 | Zeeek | tell it to change the order |
| 13:00.37 | madduck | the 200 OK SDP sends gsm ulaw alaw in that order |
| 13:00.41 | madduck | *but* |
| 13:00.46 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:00.48 | madduck | the phone on the other hand doesn't *do* gsm |
| 13:00.59 | madduck | and i also told asterisk to disallow=all and only support g.711 |
| 13:01.13 | Zeeek | tell the phone to get over itself and stop using gsm |
| 13:01.13 | jamshid | hi can i use an ordinary dial up modem to work with asterisk in anyway? |
| 13:01.14 | madduck | now links into the siemens and packet captures |
| 13:01.27 | Zeeek | jameswf-home probably not |
| 13:01.37 | Zeeek | jamshid prolly not |
| 13:01.38 | madduck | nope |
| 13:02.09 | karlh627 | Advice Please... I am trying to get asterisk to trigger an outside program when an extension answers an incoming call. The cdr records are updated when a call ends but I want something that logs and entry when the call is answered. Any suggestions? |
| 13:02.33 | jamshid | or is there any idea to make a small circuit that can have a minimum functionality to work with asterisk..like to have a caller id or so? |
| 13:03.04 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 13:03.09 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:03.29 | [TK]D-Fender | jamshid: Go buy an X100P off ebay or something then. |
| 13:04.31 | madduck | i think this may be a bug in asterisk... |
| 13:04.32 | jamshid | [TK]D-Fender: :) |
| 13:04.48 | jamshid | thanks anyway ill do more googling and come back:) |
| 13:04.58 | *** part/#asterisk jamshid (i=dce3806f@gateway/web/ajax/mibbit.com/x-0746ee15696690d7) |
| 13:05.00 | madduck | on the other side, asterisk announces ulaw, alaw, g726-32 (why?) |
| 13:05.09 | madduck | and the siemens answers with ulaw only |
| 13:05.15 | yang | [TK]D-Fender: hey do you use Grandstream phones? |
| 13:05.39 | [TK]D-Fender | yang: lol |
| 13:05.41 | [TK]D-Fender | ~gs |
| 13:05.41 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:05.44 | [TK]D-Fender | ~grandstream |
| 13:05.44 | jbot | rumour has it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 13:05.52 | yang | haha Yugo |
| 13:06.03 | madduck | hardly politically correct |
| 13:06.18 | yang | We had lots of Yugos in Slovenia a few years ago, now with the catalyser rule, many got wasted |
| 13:06.57 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 13:07.09 | yang | It was really good for someone who could afford a Yugo in Yugoslavia :), but the quality of Zastava, well we could discuss that |
| 13:07.25 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:07.31 | Katty | mmm, breakfastseses. |
| 13:07.52 | yang | They are still making Zastava 101 in Serbia, however the company is going to be sold to some other Automobile company |
| 13:08.32 | yang | [TK]D-Fender: I just thought you said once, that you were using gs, maybe it was someone else then |
| 13:08.45 | coppice | probably Tata :-) |
| 13:09.13 | yang | coppice: Tata could be one of the buyers yes, hehe they bought Land Rover and Shaguar |
| 13:09.39 | coppice | Tata make some really nasty vehicles |
| 13:10.13 | yang | coppice: you mean the small one with a Vespa wheels, which they will sell at 1500 Eur ? |
| 13:10.55 | coppice | I've no idea about that. I've never been in one. the only one I'd choose to go in is their large 4x4 |
| 13:11.31 | yang | coppice: I think their new hit is called "nano" and was expossed at that CES in Las Vegas |
| 13:12.03 | solaries | is the following possible with asterisk: |
| 13:12.03 | solaries | 1) same carrier for home telephone and mobile phone. I can call for free from mobile to home |
| 13:12.03 | solaries | 2) from home I can call for free to other "normal" telephones, but from my mobile it costs something |
| 13:12.03 | solaries | 3) With my mobile I call something like <home number>#<desired number> and asterisk (sitting before my phone telephone) forwards the call |
| 13:12.19 | yang | http://en.wikipedia.org/wiki/Tata_Nano |
| 13:12.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.05 | *** join/#asterisk RoyK (n=roy@ip-233-49-149-91.dialup.ice.no) |
| 13:13.08 | yang | solaries: you need a GSM gateway to work with your asterisk for this kind of functionality |
| 13:13.46 | yang | solaries: Portech makes some bad ones, and 2n makes some better ones if you got the money to afford it |
| 13:13.59 | *** join/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 13:14.05 | yang | ~portech |
| 13:14.22 | yang | ~poortech |
| 13:14.51 | solaries | url? |
| 13:14.59 | yang | portech.com.tw |
| 13:15.07 | *** join/#asterisk riddlebox (n=JamesMid@159.251.13.2) |
| 13:15.35 | yang | solaries: from my personal experience the thing works, but their Web GUI crashes a lot |
| 13:15.46 | solaries | why do I need such a GSM gateway? I thought about setting up a cheap PC with only a phone card in it, that is connected to my telephone access |
| 13:16.33 | yang | solaries: with GSM gateway you can do mobile<->mobile calls or mobile<->voip etc. |
| 13:16.42 | Linker3000L | Solaries: FreePBX with DISA will let you call in and dial out again. Some mobile carriers in the UK allow you to designate a number of free-to-call numbers so you can specify your home one. We do this with a biz account so mobile users can call in free and then dial local/national at VoIP rates |
| 13:17.12 | Linker3000L | UK Vodafone also allows free voda-voda calls within a grouped biz account |
| 13:17.39 | yang | Linker3000L: Have you tried if your vodafone blocks 5060 port? |
| 13:17.39 | solaries | I see I'm lacking a lot of knowledge in this area. Is there a decent tutorial to this? googling for "asterisk + call forward" has only led to a few asterisk configuration sites |
| 13:18.19 | Linker3000L | I haven't tried VoIP over GSM yet |
| 13:18.51 | mocker | solaries: It's because yiou are trying to do everything on the same carrier |
| 13:19.01 | mocker | Basic call forwarding is easy. |
| 13:20.05 | solaries | is DISA an astersik feature or something carrier related? |
| 13:20.26 | Linker3000L | No - it's generic but implemented easily with something like FreePBX |
| 13:20.40 | Linker3000L | ..others will probably have other Asterisk solutions for this too |
| 13:20.51 | solaries | will it be a problem that I have an analog telephone access? |
| 13:22.40 | Linker3000L | Not really |
| 13:23.30 | [TK]D-Fender | solaries: DISA is a single dialplan app, which you don't even need for this, though its makes implementing it a little easier. |
| 13:24.09 | solaries | ok, thanks |
| 13:24.13 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 13:24.16 | solaries | http://dumbme.voipeye.com.au/trixbox/trixbox_without_tears.pdf had a nice explanation |
| 13:24.22 | Zeeek | Hi [TK]D-Fender, feeling good? |
| 13:24.33 | [TK]D-Fender | Zeeek: Doing OK |
| 13:24.56 | Zeeek | I don't feel too bad myself (play on an old Traffic song) |
| 13:25.23 | solaries | the last question left: If I use something like freepbx I need a PC which will have to run day and night. Has anyone experience with fanless PC devices? Keyword GreenIT ;-) |
| 13:27.57 | madduck | Zeeek: if you are curious, see the -users list. This seems like a weird bug. |
| 13:28.06 | Linker3000L | Well,if you want a VoiP/GSM/Analogue gateway without a PBX, have a look here: http://www.burnsideeverwell.com/da/27251 I've not used these but they came up as possible solution elsewhere |
| 13:28.22 | [TK]D-Fender | solaries: you'll need 2 phone lines for your setup. Greenest easiest way : 2 x Linksys SPA-3102 for your 2 lines, and a Linksys WRT54G router running OpenWRT and running *. |
| 13:28.40 | [TK]D-Fender | solaries: And you don't need FreePBX. |
| 13:28.46 | mocker | [TK]D-Fender: Greenest == most energy efficient? :) |
| 13:29.02 | [TK]D-Fender | mocker: No, easiest to paint. Mi solution takes up less space :p |
| 13:29.06 | [TK]D-Fender | My* |
| 13:29.40 | [TK]D-Fender | solaries: But yes, it'll be small and very low power. |
| 13:31.04 | Zeeek | greenest solution is not to use telecommunications or the internet at all |
| 13:31.12 | Zeeek | turn off all TV sets and lamps |
| 13:31.27 | yang | i think there is some sort of small asterisk built also for openwrt |
| 13:31.31 | [TK]D-Fender | load chan_congodrums.so |
| 13:31.36 | Zeeek | do not take jet plane to Berlin |
| 13:31.47 | solaries | thank you all. it was great help |
| 13:32.27 | solaries | my carrier offers ISDN instead of analog for +2 ?. Then it I have 2 lines and it should work with only one PC connected to the telephone access, shouldn't it? |
| 13:32.31 | madduck | Zeeek: zoiper is also binary/proprietary and compiled for x86, not amd64 |
| 13:33.23 | Zeeek | someone was asking about controlling home devices like doorbells a few weeks ago. Apparently there's a device described in Make magazine that does it all for like $40 |
| 13:33.40 | Zeeek | madduck ask zoa to look into this, they might |
| 13:33.57 | Zeeek | ask on the list about a version of other softphones |
| 13:34.15 | [TK]D-Fender | solaries: Yes, but that means you need a device witha PCI slot for a compatible card. That hurts your "green" issue |
| 13:34.27 | Zeeek | HEAT! |
| 13:34.35 | Zeeek | heat isn't green |
| 13:34.42 | Zeeek | sweat is green |
| 13:34.44 | yang | Zeeek: hey ! Doorbells i was thinking of that just yesterday, do you have an URL ? |
| 13:35.04 | Zeeek | what like the doorbell button phones you? |
| 13:35.17 | coppice | Zeeek: if your sweat is green see a doctor right away |
| 13:35.17 | yang | Zeeek: exactly you answer doorbell on phone |
| 13:35.20 | Zeeek | look for make magazine |
| 13:35.28 | yang | Zeeek: and optionally speak over it |
| 13:35.43 | Zeeek | I'd rather answer phone on doorbell. With doorbell in Hawaii :) |
| 13:35.56 | Zeeek | or Tahiti |
| 13:36.20 | yang | Zeeek: what is the url for that magazine, maybe tehy have the online article about that |
| 13:36.47 | Zeeek | your google is as good as mine, if I may coin a phrase (IOW, I have no idea, sorry) |
| 13:36.58 | Zeeek | Heard about it in a podcast |
| 13:37.11 | Zeeek | A device that will do almost anything and is connected to a computer |
| 13:37.40 | Zeeek | Google says http://makezine.com |
| 13:37.52 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 13:38.44 | Zeeek | ok, I'm getting close |
| 13:38.50 | Zeeek | http://store.makezine.com/SearchResults.asp?Cat=43 |
| 13:38.54 | yang | Zeeek: from our buldint installation we only have a 12Volt plug on doorbell |
| 13:39.25 | Zeeek | http://store.makezine.com/ProductDetails.asp?ProductCode=MKBBARD |
| 13:39.46 | Zeeek | a relay is a wonderful 19th century invention |
| 13:40.49 | Zeeek | definitely Arduino boards |
| 13:41.04 | yang | Zeeek: this is some sort of a controller....but how to enable that on a PC |
| 13:41.14 | glaz | Which softphone software do you recommend for Mac OSX ? |
| 13:41.15 | Zeeek | http://store.makezine.com/ProductDetails.asp?ProductCode=MKARUSBAD |
| 13:41.24 | Zeeek | enough of the URLs. Go read the page now |
| 13:41.25 | yang | Zeeek: it looks like there is no ready to use device ? |
| 13:42.53 | pa | how good is asterisk LDAP support? |
| 13:43.38 | [TK]D-Fender | pa: what are you expecting * to do via LDAP? |
| 13:43.45 | madduck | what is "asterisk[8898]: rc_avpair_new: unknown attribute 1490026597" in syslog trying to tell me? |
| 13:44.13 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.122.138) |
| 13:46.12 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 13:46.25 | mort_gib | Affternoon |
| 13:46.56 | mort_gib | I need a SIP client to report busy, when in a call |
| 13:47.09 | mort_gib | incominglimit=1 gives me unavail |
| 13:47.11 | mort_gib | ?? |
| 13:47.39 | Juggie | madduck, do you have asterisk patched with some addon? |
| 13:49.05 | madduck | Juggie: i am using the xorcom backport, I don't think it's patched |
| 13:49.06 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 13:49.10 | madduck | there's stuff removed, like IMAP support |
| 13:49.15 | madduck | but not added |
| 13:49.24 | Juggie | hmmm |
| 13:50.45 | mort_gib | Nobody has any idea?? |
| 13:50.58 | pa | [TK]D-Fender, mostly keep iax and sip user database, rather than using directly postgresql |
| 13:52.14 | Katty | hugs anonymouz666 |
| 13:52.26 | Katty | [TK]D-Fender: oh hai |
| 13:52.31 | Juggie | madduck, i cannot find 'rc_avpair_new' in my asterisk source |
| 13:52.46 | madduck | good idea, lemme look |
| 13:52.53 | [TK]D-Fender | pa: as far as I'm aware, you can only use relational DB's for that... |
| 13:52.56 | _ShrikE | dangles the catnip for katty |
| 13:52.57 | [TK]D-Fender | Katty: Mew. |
| 13:53.09 | pa | ok.. |
| 13:53.12 | pa | i'll use that |
| 13:53.15 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:53.27 | anonymouz666 | Katty! |
| 13:53.37 | [TK]D-Fender | Anybody got a link to a place selling 2U-3U short depth rack-mount chassis for PC servers? |
| 13:54.26 | mort_gib | [TK] US?? |
| 13:55.29 | Dextorion | btw.. is recording included in version 1.2(win emulated)? |
| 13:55.42 | Dextorion | you can record calls, right? |
| 13:56.01 | _ShrikE | [TK]D-Fender: http://mypccase.stores.yahoo.net/2urashd3ba2f.html |
| 13:56.06 | madduck | weird, Juggie, me neither. |
| 13:56.24 | mort_gib | Please I need hand with busy unavail |
| 13:56.26 | *** join/#asterisk real0ne (n=Ismail@41.251.98.101) |
| 13:56.41 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-179-153-198.bltmmd.east.verizon.net) |
| 13:58.19 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
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| 13:58.30 | [TK]D-Fender | mort_gib: Ship-able (reasonably) to North America |
| 13:58.31 | boblutz | Does anyone have a remedy for phantom rings? |
| 13:59.28 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:59.28 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:59.42 | Dextorion | its possible to record calls? |
| 13:59.54 | mort_gib | Uhm, Wim from the OPenBSD gang is shipping some custom rack cases with Soekris 5501 bords |
| 14:01.00 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-179-153-198.bltmmd.east.verizon.net) |
| 14:01.05 | SteveTotaro | turn ringer off |
| 14:03.14 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-179-153-198.bltmmd.east.verizon.net) |
| 14:04.08 | [TK]D-Fender | _ShrikE: Thanks.... looking for a new case for my work * server. I dislike having it in my big server rack and want it on my patch-panel relay rack... |
| 14:04.52 | _ShrikE | NP. I've have on of the 2 unit ones. They are pretty decent. |
| 14:04.58 | _ShrikE | s/on/one/ |
| 14:06.17 | madduck | so my sip domain answers on echo@ with Answer()/Echo()/Hangup(). I would like to execute the same sequence for another extension in another context. Can I somehow include a sequence and bind it to another extension? |
| 14:06.59 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 14:08.33 | riddlebox | hrmm I wonder why blf's work when I edit the conf files myself, but if I create the system with the gui, they dont work...... |
| 14:09.48 | [TK]D-Fender | riddlebox: Obviously the conf files are different. setup eac and slowly transform 1 into the other and see where the break-point is. |
| 14:10.16 | sysadmin-lb22 | hey all does asterisk support jingle in any possible way ? |
| 14:13.30 | lmadsen | there used to be a chan_jingle... |
| 14:13.39 | lmadsen | never used it though |
| 14:13.51 | lmadsen | should be listed in menuselect on 1.4 |
| 14:17.16 | riddlebox | [TK]D-Fender: thats what I started to do last night |
| 14:17.35 | boblutz | Why do they call it "Kewlstart" ? |
| 14:17.52 | riddlebox | because its kewl |
| 14:17.52 | [TK]D-Fender | boblutz: 'cause its teh shiznit y0! |
| 14:17.56 | boblutz | LOL |
| 14:18.17 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:18.19 | boblutz | 10/10 |
| 14:18.46 | *** join/#asterisk SwK (n=SwK@70-46-62-90.atl.fdn.com) |
| 14:19.51 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 14:19.58 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:19.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:22.20 | ZPertee | anyone know how call transfer works if the call is passed through a legacy pbx before getting to the phone? |
| 14:22.51 | [TK]D-Fender | ZPertee: DETAILS please.... |
| 14:26.53 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 14:33.28 | ZPertee | [TK]D-Fender, never mind I figured out an alternative solution. |
| 14:35.20 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 14:40.55 | boblutz | Would a telco let me purchase a PRI channel for use at my house (non-business related) ? |
| 14:41.40 | maveric1 | who knows security check tools for asterisk ? |
| 14:41.43 | scoates | I'm having trouble with this phone periodically going unreachable: http://phpfi.com/305858 any idea why asterisk is sending "No NAT"? and why I'd get 481? (this was captured while the phone was reachable) |
| 14:41.48 | maveric1 | security resources for asterisk |
| 14:41.55 | Kobaz | boblutz: you can't get just one channel |
| 14:42.00 | Kobaz | boblutz: pri is all or nothing |
| 14:42.01 | maveric1 | help !!! |
| 14:42.37 | boblutz | Kobaz: Ok, so 23 channels would be overkill for me to use at my house to call grandma, got it |
| 14:42.40 | florz | maveric1: I guess a text editor should do. |
| 14:42.51 | Katty | lmadsen: i haven't heard from you this morning |
| 14:42.57 | Kobaz | boblutz: pri will run you around 300-400 a month (depending on the provider) |
| 14:43.30 | Kobaz | boblutz: you should just get either voip from some itsp or just a regular analog line from the local telco |
| 14:44.07 | boblutz | Kobaz: The underlying situation is I am trying to determine whether this office has BRI or PRI channels (did I say that right?) |
| 14:44.16 | Kobaz | we're paying 220 per pri for our lines, but they come from the office next door |
| 14:44.17 | boblutz | it has 4 lines ... I am thinking that is BRI ? |
| 14:44.29 | Kobaz | bri would be good then |
| 14:44.48 | Kobaz | oh you already have a line, yeah |
| 14:44.49 | [TK]D-Fender | boblutz: How many connectors ? |
| 14:44.55 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:45.07 | boblutz | [TK]D-Fender: connectors? |
| 14:45.10 | Kobaz | boblutz: do you have four seperate lines comming in? |
| 14:45.15 | boblutz | Kobaz: Yes |
| 14:45.15 | Kobaz | or is it on one span |
| 14:45.37 | boblutz | there are 4 little boxes hanging on the wall with one phone number written on each box |
| 14:45.41 | Kobaz | are the lines comming off of a csu/dsu? |
| 14:45.43 | [TK]D-Fender | boblutz: how many jacks are these 4 "lines" delivered over? |
| 14:45.44 | Kobaz | or straight from the wall |
| 14:45.57 | [TK]D-Fender | Kobaz: And why kind? |
| 14:46.02 | boblutz | [TK]D-Fender: 4 |
| 14:46.03 | [TK]D-Fender | what* |
| 14:46.10 | [TK]D-Fender | boblutz: rather. |
| 14:46.12 | [TK]D-Fender | askhdashdal |
| 14:46.14 | Kobaz | [TK]D-Fender: what kind of what? |
| 14:46.14 | Kobaz | oh |
| 14:46.17 | boblutz | rj-14 |
| 14:46.33 | [TK]D-Fender | boblutz: Well 4 jacks for 4 lines sure sounds like analog to me. |
| 14:46.47 | Kobaz | unless they are split off of a channel bank or something in the closet |
| 14:47.04 | boblutz | Dont think so, we rent this office space |
| 14:47.08 | Kobaz | but yeah, what you have are some pots lines |
| 14:47.14 | [TK]D-Fender | boblutz: So where do they come in from? And local equipment before those jacks? |
| 14:47.27 | boblutz | [TK]D-Fender: 1 sec |
| 14:47.41 | sysadmin-lb22 | any possibility to send traffic from jingle to pstn gw using asterisk |
| 14:48.46 | boblutz | cables come out from the ceiling, it looks like those cables connect to a punch block |
| 14:49.05 | Kobaz | you have yourself some regular standard analog phone lines |
| 14:50.21 | boblutz | Actually, I think those cables I saw lead to the jacks ... Would a telephony DSL modem make sense? |
| 14:50.51 | boblutz | you could fit 4 lines in 1 cat 5 (4 twisted pairs) ? |
| 14:51.18 | Kobaz | for transport, sure, but you have to break them baack out at the endpoints |
| 14:51.34 | boblutz | hence a punch block? |
| 14:51.49 | Kobaz | yeap |
| 14:51.56 | boblutz | Kobaz: [TK]D-Fender: thanks |
| 14:52.54 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-179-153-198.bltmmd.east.verizon.net) |
| 14:53.40 | patrick-- | Evening! I was wondering if its possible to generate a certain ringtone on picking up the phone... some sort of internal ring, and when you press a 0 you get a constant tone indicating a ready line to the outside.. |
| 14:53.48 | Kobaz | yeap |
| 14:55.00 | patrick-- | [TK]D-Fender: will you save me this time? :D |
| 14:56.36 | *** join/#asterisk zepmantra (i=theman@210.23.188.172) |
| 14:59.23 | lmadsen | Katty: oh no... I'm getting fast busy when I dial it today.... |
| 14:59.49 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 14:59.58 | Katty | lmadsen: mew? |
| 15:02.44 | lmadsen | ya... not sure if it's my phone or what |
| 15:02.52 | lmadsen | but when I try to redial, I get fast busy |
| 15:03.21 | patrick-- | how can i generate an internal dialtone`? |
| 15:03.37 | lmadsen | DISA() |
| 15:03.46 | lmadsen | if that's what you're asking... |
| 15:03.56 | patrick-- | not sure, ill look it up |
| 15:04.17 | *** join/#asterisk galeras (n=galeras@201.244.195.99) |
| 15:04.53 | patrick-- | what i want is, when i pickup the phone, it should play the "beep beep beep <pause> beep beep beep <pause> etc" tone until a 0 or an internal number is dialled |
| 15:05.14 | patrick-- | when you press 0 you get a constant tone signalling a ready outside line |
| 15:05.27 | patrick-- | or you simply dial an internal number and it puts through |
| 15:05.56 | lmadsen | analog? |
| 15:05.59 | patrick-- | isdn |
| 15:06.06 | lmadsen | hrmmm... no idea |
| 15:06.11 | lmadsen | I don't do hardware... |
| 15:06.18 | tzanger | heh |
| 15:06.48 | patrick-- | gah |
| 15:06.48 | patrick-- | trying to figure this out for ages |
| 15:06.48 | tzanger | lmadsen: how was the taug meeting? I couldn't make it |
| 15:06.48 | lmadsen | tzanger: for good reason too ^^^ |
| 15:06.55 | lmadsen | tzanger: it was alright.. I recorded it on my camera, but no idea how it came out because a guest who was staying over stole my USB cable :) |
| 15:07.06 | tzanger | ha |
| 15:07.24 | lmadsen | so I have to wait until IT360 to have ti returned |
| 15:07.34 | tzanger | or just go buy another damned $5 cable |
| 15:07.46 | Juggie | a mini usb cable is worth like 3 cents |
| 15:07.54 | Juggie | dell throws like 4 of them in with every laptop |
| 15:08.04 | lmadsen | I know, but then I'd still have to go find one at a store somewhere |
| 15:08.26 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:08.26 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:08.31 | tzanger | yeah, it's not like you're in one of the biggest cities in canada or anything |
| 15:08.42 | patrick-- | anthm: any fermilliar with isdn? |
| 15:08.42 | Qwell | tzanger: well, it is canada |
| 15:09.06 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 15:09.26 | tzanger | Qwell: indeed; he'd have to buy a jug of maple syrup to keep the cable's canadian content :-) |
| 15:09.51 | Katty | lmadsen: i've not changed anything. |
| 15:09.56 | Katty | lmadsen: i think you've borked. |
| 15:10.00 | lmadsen | ya, it's probably my phone :) |
| 15:10.11 | lmadsen | I need new firmware... this firmware is teh screwed |
| 15:10.38 | tzanger | lmadsen: I've been sitting in william's drinking coffee and using free bandwidth since 9:15 waiting for the car to get fixed :-) |
| 15:10.45 | tzanger | yay for long battery life and powertop |
| 15:11.00 | boblutz | wait |
| 15:11.00 | lmadsen | tzanger: handy! I gotta go to the grocery store, bank, and LCBO soon-ish |
| 15:11.05 | lmadsen | I refuse |
| 15:11.09 | boblutz | this is deja vu |
| 15:11.22 | lmadsen | ya, me and tzanger say this everyday |
| 15:11.29 | boblutz | lol |
| 15:11.42 | *** join/#asterisk daniev (n=psycke@201.245.223.90) |
| 15:12.29 | tzanger | hahaha |
| 15:12.48 | [TK]D-Fender | patrick--: make a single digit extension and what you hit it, just make an IVR with your outbound extens in it Background the new "tone" you want them to hear. |
| 15:14.38 | mvanbaak | someone here that can run 'df /usr' on centos 4.5 for me ? |
| 15:14.49 | mvanbaak | I want to know the output format of that an centos |
| 15:18.58 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.52.66) |
| 15:19.10 | Kobaz | mvanbaak: it's the same as any other df on any other distro of linux |
| 15:19.52 | patrick-- | right [TK]D-Fender you got that for beginners? |
| 15:20.31 | Kobaz | patrick--: get the asterisk book |
| 15:20.37 | patrick-- | ive got it |
| 15:20.37 | Kobaz | patrick--: and hit the wiki |
| 15:21.01 | [TK]D-Fender | patrick--: Make an IVR. Background a soundfile with this other tone you want. |
| 15:21.09 | Kobaz | look up playing sounds |
| 15:21.18 | mvanbaak | Kobaz: that cannot be |
| 15:21.30 | Kobaz | mvanbaak: sure it can |
| 15:21.34 | patrick-- | cant you just set a standart sound somewhere? |
| 15:21.49 | mvanbaak | well, I'm porting some scripts that run fine on centos4.5 |
| 15:21.52 | Kobaz | patrick--: there are a number of sounds that do come with asterisk, yeah you can use one of those |
| 15:21.54 | mvanbaak | and on ubuntu it's not |
| 15:22.08 | Kobaz | mvanbaak: make sure you're using the same options for df |
| 15:22.20 | Kobaz | and don't just run 'df' |
| 15:22.24 | Kobaz | use the full path |
| 15:22.25 | mvanbaak | Kobaz: it's just 'df /usr' |
| 15:22.31 | Kobaz | to make sure you're not running an alias |
| 15:22.35 | patrick-- | Kobaz: where can i check them out? |
| 15:22.52 | Kobaz | patrick--: the default directory is /var/lib/asterisk/sounds |
| 15:23.17 | mvanbaak | Kobaz: using /bin/df gives me the same error in the script |
| 15:23.35 | boblutz | ~ubuntu |
| 15:23.44 | madduck | does anyone know of a service that can be used to test receiving sip calls? like a website where i enter my sip ID and then it calls me? |
| 15:24.06 | mvanbaak | it's this perl function: http://pastebin.ca/960914 |
| 15:24.12 | mvanbaak | it works on centos |
| 15:24.19 | mvanbaak | but wont work on ubuntu |
| 15:25.05 | mvanbaak | and the output of ubuntu's df: |
| 15:25.08 | mvanbaak | http://pastebin.ca/960916 |
| 15:26.14 | boblutz | mvanbaak: Is that function ..err sub routine returning false? |
| 15:26.38 | mvanbaak | it is |
| 15:26.46 | mvanbaak | but as you can see, I have plenty diskspace |
| 15:26.52 | mvanbaak | so it should return true |
| 15:27.13 | boblutz | mvanbaak: Does line 4 actually execute `df /usr` ? |
| 15:27.27 | mvanbaak | that's why I think the output format of df on centos differs from ubuntu |
| 15:27.46 | mvanbaak | yes |
| 15:27.59 | galeras | SIRS: My asterisk is behind a nat and some of my sip clients too, is STUN requiered iin this scenario ? |
| 15:28.04 | mvanbaak | I tested with: print $diskfree |
| 15:28.11 | mvanbaak | ~sipnat |
| 15:28.12 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:29.13 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:29.15 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:29.24 | Kobaz | mvanbaak: looks like a standard df output |
| 15:29.43 | boblutz | yea, my RH server shows the same |
| 15:29.44 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 15:29.49 | mvanbaak | hhmm |
| 15:29.59 | mvanbaak | then why is it not working on ubuntu ? |
| 15:30.11 | Qwell | mvanbaak: spacing? |
| 15:30.11 | mvanbaak | stupid perl |
| 15:30.14 | Kobaz | mvanbaak: if the other box gives you something different, either a) it's using different command line args, b) it's a way different version c) there are env variables that are affecting the output |
| 15:30.31 | Kobaz | patrick--: do you have an ivr already? |
| 15:30.40 | mvanbaak | it must be spacing indeed |
| 15:31.03 | davevg-btwtech | try splitting on \s+ then Dump the output of @disk |
| 15:31.34 | Kobaz | patrick--: http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu |
| 15:31.59 | Kobaz | patrick--: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Background |
| 15:32.08 | Kobaz | i found those in two seconds of searching |
| 15:32.18 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 15:32.20 | boblutz | Has anyone experienced their fxo module "starting simple switch" after hanging up (no one is calling) ? |
| 15:32.24 | Kobaz | patrick---: google is your friend |
| 15:32.34 | boblutz | Zeeek: We are having a conference today at noon |
| 15:33.02 | Kobaz | boblutz: your card/driver isn't properly handling disconnect supervision, or your line doesn't have any |
| 15:33.12 | Kobaz | boblutz: what fxo card? |
| 15:33.20 | boblutz | tdm410p - x100m |
| 15:33.21 | [TK]D-Fender | boblutz: Perhaps your telco is using a wink to indicate disconnect, etc. |
| 15:33.40 | boblutz | I never had this issue with my tdm400p though |
| 15:33.49 | Kobaz | maybe the telco changed something |
| 15:33.57 | Kobaz | did you change any software/drivers? |
| 15:33.59 | [TK]D-Fender | boblutz: And what card is this now? using the same lines? |
| 15:34.32 | Kobaz | patrick--: i know it's rough starting out, but i learned a ton just looking through the wiki |
| 15:34.40 | boblutz | [TK]D-Fender: Kobaz: Everything is the same, I just swapped out the cards, and loaded wct4xxp rather than wctdm |
| 15:35.00 | davevg-btwtech | mvanbaak, does this help? http://pastebin.ca/960924 |
| 15:35.06 | boblutz | well.. I load wctdm, wct4xxp, and wct24xxp |
| 15:36.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:36.25 | [TK]D-Fender | boblutz: Just load the one yuo NEED |
| 15:36.42 | [TK]D-Fender | boblutz: and WCT4XXP is for T1 cards |
| 15:37.36 | boblutz | edits /etc/sysconfig/zaptel |
| 15:39.32 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:40.38 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 15:41.55 | Yourname`` | How can I make one sip peer show his own caller ID while everybody else just does whatever the default is? I tried setting callerid= in sip.conf for the peer, and it's not making a difference. |
| 15:43.17 | *** join/#asterisk ar3dam (n=ar3dam@189.156.233.99) |
| 15:43.18 | patrick-- | Kobaz: what wiki, exactly? |
| 15:43.36 | Kobaz | patrick--: check out the links, the voip-info wiki |
| 15:43.40 | patrick-- | right |
| 15:43.42 | patrick-- | im on that one |
| 15:43.59 | Kobaz | they have some good info on ivr's and sounds and all that |
| 15:44.16 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 15:44.53 | ar3dam | hi there .. somewhere are use card zapmicro? |
| 15:45.05 | Kobaz | all you really need to do is play the sound you want in the background, and then wait for dtmf from the user |
| 15:45.26 | Zeeek | I must leave to start the VUC: http://voipusersconference.org if you would like to be a part of it |
| 15:45.40 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
| 15:45.58 | ar3dam | hi there, i have little problem, i have many echo from ata grandstream |
| 15:52.18 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:52.27 | ar3dam | any advice? |
| 15:53.18 | jasonwoot | management wants multiple voicemail boxes per extension, possible? |
| 15:53.52 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:53.52 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:55.12 | shido6 | like a distribution list, jasonwoot? |
| 15:56.43 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 15:56.55 | *** part/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 15:57.11 | jasonwoot | no, individual vm boxes on one extension, for shared desks, "press 1 to leave msg for Joe Blow, 2 for Ted Nugent, .." etc |
| 15:57.34 | jasonwoot | already told them no, but when you guys help me figure out how to do it, I come off like hero |
| 15:57.41 | shido6 | so you need an autoattendant mailbox? |
| 15:58.07 | jasonwoot | trying to figure out how to do it without complicated routing table |
| 15:59.18 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 15:59.23 | mvanbaak | davevg-btwtech: thank you :) |
| 15:59.31 | flujan | hi all. :D |
| 15:59.38 | [TK]D-Fender | jasonwoot: make a little IVR to pick the VMbox. |
| 15:59.47 | flujan | guys, I am trying to configure snmp and asterisk to work using a article of mark spencer... |
| 16:00.15 | [TK]D-Fender | jasonwoot: or you could do a small loop |
| 16:01.15 | *** part/#asterisk maveric1 (n=root@83.240.229.38) |
| 16:01.18 | flujan | http://pastebin.com/d480b826d here it goes... |
| 16:01.25 | flujan | asterisk is outputting nothing... :D |
| 16:01.39 | shido6 | something like this http://pastebin.ca/960951 , jasonwoot ? |
| 16:06.25 | tzanger | http://xkcd.com/242/ |
| 16:06.29 | tzanger | that is one of my favourites |
| 16:08.59 | flujan | someone here using snmp to monitor asterisk? |
| 16:10.38 | boblutz | What is wink timing? |
| 16:11.10 | coppice | typically 140ms |
| 16:11.30 | boblutz | I mean, what is the definition of it? |
| 16:11.42 | boblutz | ~wink |
| 16:11.43 | jbot | ACTION winks at boblutz |
| 16:11.46 | boblutz | ou! |
| 16:16.28 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 16:17.44 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 16:17.49 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 16:18.32 | ar3dam | hi there .. i have installed trixbox, and the extensions is 2 grandstream tonebudge 101, and 1 ata grandstream. |
| 16:18.40 | *** join/#asterisk tomcontr3 (n=gcontrer@165-150-222-201.adsl.terra.cl) |
| 16:18.46 | tomcontr3 | Hi does anyone knows why the person that starts a call can not trasnfer the call using ## ? |
| 16:19.25 | ar3dam | tomcontr3, check the features.conf .. |
| 16:20.38 | flujan | I am following a document from mark spencer to use cacti and the res_snmp from version 1.4 without success... dunno if it is a problem with the asterisk mibs or with my configuration. |
| 16:20.51 | *** join/#asterisk ManxPower (n=manxpowe@98.sub-70-223-42.myvzw.com) |
| 16:21.26 | tomcontr3 | what should I look for there? I know that there is where you configure all the features keys,.. but my problem is that the person who starts the call can trasnfer it, only the person that I called can. |
| 16:24.01 | ManxPower | tomcontr3: WHO can use the feature is determined by the t/t/W/w options |
| 16:24.34 | tomcontr3 | where are those options seted? |
| 16:24.44 | ManxPower | In the Dial line |
| 16:25.20 | tomcontr3 | ohh I see... |
| 16:25.20 | ManxPower | I assume "features keys" means "DTMF specified in /etc/asterisk/features.conf" |
| 16:25.29 | tomcontr3 | yep |
| 16:25.43 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 16:25.53 | ManxPower | tomcontr3: Does the phone you are using NOT have a transfer button? |
| 16:27.45 | tomcontr3 | yes it does... |
| 16:27.53 | tomcontr3 | but it doesnt work very well |
| 16:27.54 | ManxPower | tomcontr3: then why are you not using that? |
| 16:27.59 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 16:28.09 | tomcontr3 | so I figured that it was better use the pbx method.. |
| 16:28.29 | ManxPower | "doesn't work very well" term is banned under International Support Treaties for being too vague. |
| 16:29.01 | ManxPower | what make/model of phone are you using? |
| 16:29.46 | tomcontr3 | this are test phones.. we test them for the factory |
| 16:30.09 | tomcontr3 | I have ton run... Ill came back later |
| 16:36.39 | *** join/#asterisk smallfoot- (i=smallfoo@unaffiliated/smallfoot-) |
| 16:37.34 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 16:38.58 | Seggy | alright, I have an aastra 480i that I'm using SIP to asterisk (SVN-trunk-r110621M) All the prompts sound horrible, muffled, little bit of chop... it seems to clear up after the prompts are done (echo conference, etc) Anyone else run across it? |
| 16:39.36 | jjshoe | seggy takes the phones echo can to kick in |
| 16:39.37 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 16:39.38 | Seggy | Oh, and as a second test I'm using SJPhone (softphone) and everything sounds excellent across it |
| 16:39.45 | jjshoe | takes time* |
| 16:40.02 | jjshoe | do you experience it on a very long voice prompt? does it clear up part way through or no? |
| 16:40.04 | Seggy | jjshoe: Yeah, but I'm talking like 20 seconds for the demo echo conference announcement and it never ets better |
| 16:40.10 | Seggy | err gets |
| 16:40.14 | Seggy | and it's only on the aastra |
| 16:40.19 | jjshoe | Seggy yuck. time to play with the gain in the config files |
| 16:40.31 | Qwell | Seggy: http://bugs.digium.com/view.php?id=11243 |
| 16:40.34 | Qwell | Seggy: that, maybe? |
| 16:41.21 | Seggy | Qwell: hmm, let me see if SJphone is connecting gsm |
| 16:41.37 | Qwell | doesn't matter how the phone is connecting |
| 16:42.16 | Qwell | verify the compiler that was used to build asterisk |
| 16:43.36 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 16:43.38 | Seggy | oh look |
| 16:43.39 | Seggy | 4.2.3 |
| 16:43.40 | Seggy | ... |
| 16:43.59 | jjshoe | brb |
| 16:44.24 | Qwell | Seggy: try with 4.1 (or 4.3), and if it works, please try the patch I posted there with 4.2 :D |
| 16:44.46 | Qwell | or, actually, just use DONT_OPTIMIZE with 4.2, and it should go away - then test the patch |
| 16:44.58 | Seggy | Qwell: the reason I was gonna look what sjphone was doing as it was mentioned that it was just in transcoding, and sjphone sounds beautiful, so figured it was probably a transcode issue |
| 16:45.50 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 16:45.51 | Seggy | and... it is in gsm on sjphone. I'll rebuild with DONT_OPTIMIZE |
| 16:46.58 | *** part/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 16:47.39 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:50.43 | *** part/#asterisk solaries (n=wr@kagu.c-lab.de) |
| 16:50.57 | *** join/#asterisk RoyK (n=roy@tmi212017138038.mobil.telenor.no) |
| 16:54.13 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 16:54.50 | EmleyMoor | Are there any "truly free" softphones for Linux that accept status notification? |
| 16:54.57 | CanWood | hey folks. I'm compiling asterisk 1.4.18.1 on an ubuntu box to replace the 1.4.10 that went in using apt. I'm needing to install quite a few "dev" packages to meet the dependencies first. has anyone else been through this that I can comapre notes with? |
| 16:58.37 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 16:59.26 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 16:59.26 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:59.41 | *** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 17:03.55 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:06.37 | Katty | mmm, chocolate. |
| 17:07.55 | shazaum | ¬¬ |
| 17:10.35 | boblutz | What does "RoHS" mean on my x100m ? |
| 17:11.14 | Katty | it means you need chocolate. |
| 17:11.40 | boblutz | o yea? |
| 17:12.06 | EmleyMoor | Reduction of Hazardous Substances |
| 17:12.20 | boblutz | EmleyMoor: thanks |
| 17:12.30 | shazaum | Restriction of Certain Hazardous Substances |
| 17:12.32 | shazaum | lol |
| 17:12.49 | boblutz | shazam! |
| 17:13.06 | shazaum | hehehehe |
| 17:13.55 | shazaum | Katty: i have a box of chocolate |
| 17:14.02 | Katty | oh? |
| 17:14.17 | boblutz | i got a pickle for a nickel |
| 17:14.39 | EmleyMoor | Better than a blob for a bob |
| 17:14.45 | shazaum | Katty: want to buy? Brazilian chocolate |
| 17:14.46 | shazaum | :P |
| 17:14.48 | boblutz | :-/ |
| 17:15.09 | Katty | no |
| 17:15.35 | Katty | i like chocolate from Sweden the best. |
| 17:15.41 | Katty | the dark stuff |
| 17:16.06 | boblutz | LOL "the dark stufF" |
| 17:17.16 | shazaum | :P |
| 17:19.22 | patrick-- | Hey, im using 2 BeroNET isdn cards and im trying to figure out how i can transfer calls so that when i receive an incoming call, i can press the Flash button, dial a target number and put down the phone so the caller is automatically transferred to the target number |
| 17:20.55 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 17:22.10 | ManxPower | patrick--: the answers you seek are in the manual for your phone. |
| 17:22.19 | patrick-- | i doubt it |
| 17:22.32 | ManxPower | patrick--: Well it won't be in Asterisk's docs. |
| 17:22.45 | ManxPower | You would be looking for Blind Transfer feature. |
| 17:23.00 | ManxPower | patrick--: in ISDN almost everything is controlled on the phone. |
| 17:23.29 | *** join/#asterisk grEvenX (n=even@pc107-102.ktv.no) |
| 17:23.34 | ManxPower | And if you want to use the Flash button you must find that info in the phone manual |
| 17:26.15 | ManxPower | This is because Asterisk does not even see anything from the phone until the final stage of the transfer |
| 17:27.13 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:27.21 | generalhan | hey all !~ |
| 17:28.11 | generalhan | now that i have upgraded from 1.2 to 1.4 ive noticed that when checking voicemails it now says the number that the voicemail came from but no longer tells the date that it was left |
| 17:28.20 | generalhan | is that an option that i need to set ? or is that just the new way ? |
| 17:29.04 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 17:29.36 | patrick-- | ManxPower: no |
| 17:29.37 | patrick-- | well |
| 17:29.39 | patrick-- | its a |
| 17:29.39 | boblutz | how would you define kewlstart signalling for fxo modules on ports 0 and 3 ? "fxsks=1-4" ? |
| 17:29.43 | patrick-- | asterisk thing |
| 17:30.33 | boblutz | nm, `genzaptelconf` fixed it |
| 17:30.40 | Yourname`` | How can I make one sip peer show his own caller ID while everybody else just does whatever the default is? I tried setting callerid= in sip.conf for the peer, and it's not making a difference. |
| 17:31.20 | [TK]D-Fender | Yourname`pastebin....... |
| 17:31.29 | *** join/#asterisk hohum (n=dcorbe@70.0.246.40) |
| 17:37.48 | *** join/#asterisk Guggemand (i=guggeman@80.198.131.46) |
| 17:39.37 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 17:41.11 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 17:49.01 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:50.50 | CanWood | Still trying to meet dependencies to build 1.4.18.1 |
| 17:51.01 | CanWood | http://pastebin.ca/961097 shows an issue if anyone has any input |
| 17:52.17 | bkruse | CanWood: You might want to make distclean, or make clean, or both, but it looks like it might be trying to compile against an older/newer library? Did you apt-get build-dep asterisk, or get another h323 library? |
| 17:53.00 | bkruse | Are you looking to specifically build h323? If not, make menuselect -> channel drivers -> chan_h323, unmark it with space, hit x, make; make install |
| 17:53.20 | CanWood | originally, asterisk was installed via apt, so I have a bunch of 1.4.10 .so files |
| 17:54.17 | Qwell | CanWood: make uninstall |
| 17:54.24 | Qwell | that'll clean out the old stuff |
| 17:54.34 | CanWood | I am trying to bring it up to 1.4.18.1 from source since the package is not new enough, and installed the h323 via apt |
| 17:54.55 | CanWood | it is a working asterisk system. will make uninstall blow away my current setup? |
| 17:55.17 | Qwell | CanWood: just the binaries |
| 17:55.26 | Qwell | configs will stay - just DO NOT do make uninstall-all |
| 17:55.30 | CanWood | heh |
| 17:56.04 | CanWood | at this point, I have four old .so files and the rest has all worked, but I want to bring them up to latest |
| 17:56.24 | bkruse | CanWood: mkdir /backup; cd /backup; tar -czvf asterisk.configs.`date +%s`.tar.gz /etc/asterisk ; echo "just incase" |
| 17:56.28 | bkruse | if you feel that strongly about it |
| 17:56.42 | bkruse | CanWood: If you know specifically which ones, you can rm them from /usr/lib/asterisk/modules |
| 17:57.07 | CanWood | right'o |
| 17:57.10 | *** join/#asterisk hugohagogo (n=cleber@189.23.20.10) |
| 17:57.40 | CanWood | I'll move'm out for now, but I still want to work at getting the patched ones in |
| 17:57.47 | CanWood | tks for the tips |
| 17:59.43 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.122.138) |
| 18:04.40 | madduck | check_auth: username mismatch, have <piper>, digest has <00413251xxxxx> |
| 18:04.49 | madduck | why do i get this and how can i solve it? |
| 18:05.04 | bkruse | np |
| 18:05.16 | bkruse | madduck: Do you really want to know why? Or how to solve it |
| 18:05.27 | madduck | bkruse: i really want to know why |
| 18:05.30 | bkruse | the digest field in the sip header (username@ip) is different than the username field |
| 18:05.34 | madduck | then i can hopefully solve it myself. |
| 18:05.44 | madduck | is this something i have to solve with insecure=? |
| 18:05.48 | bkruse | This happens if, persay, you have a username set in sip.conf, but your context name is different |
| 18:06.01 | bkruse | madduck: Um, I believe that will help port. Is this on a phone? Or another asterisk box? or... |
| 18:06.13 | Yourname`` | [TK]D-Fender: http://pastebin.ca/961117 |
| 18:06.57 | [TK]D-Fender | Yourname``: Well all calls from that peer will have the CID listed.. |
| 18:07.07 | madduck | bkruse: Proxy-Authorization: Digest username="0041325119511", realm="madduck.net", ... |
| 18:07.18 | madduck | [insphone] |
| 18:07.22 | madduck | fromdomain=sip05.insphone.ch |
| 18:07.22 | madduck | fromuser=0041325119511 |
| 18:07.25 | bkruse | madduck: Again, is this coming from a phone or what? |
| 18:07.31 | bkruse | and you are trying to register as something? |
| 18:07.40 | bkruse | where is "piper" in that context? |
| 18:07.43 | Yourname`` | [TK]D-Fender: But they aren't, good sir. |
| 18:07.52 | [TK]D-Fender | Yourname`PASTEBIN. |
| 18:07.54 | madduck | bkruse: this is me calling from a softphone to sip:41325119511@sip05.insphone.ch |
| 18:08.00 | madduck | where my asterisk has registered |
| 18:08.08 | Yourname`` | [TK]D-Fender: This time what do you want to see good sir? |
| 18:08.11 | madduck | insecure=invite is already set |
| 18:08.31 | [TK]D-Fender | Yourname`Show me the call & backup. Noop callerid and enable sip debug and show me a call. |
| 18:08.33 | bkruse | madduck: Where is the word "piper", it is obviously somewhere... |
| 18:08.43 | bkruse | The softphone has it in its account configuration somewhere, for sure. |
| 18:08.58 | madduck | bkruse: oh, right. the softphone is actually registered with my asterisk as "piper" |
| 18:09.14 | bkruse | :] |
| 18:09.15 | Yourname`` | [TK]D-Fender: Uh oh. There's SO much traffic going on there. I wish there was something like cli filtering for this. I'll revisit this tmrw, since it's the weekend.. :) |
| 18:09.28 | bkruse | change piper to the number, of vice versae |
| 18:09.39 | bkruse | (I would recommend piper to the number, as that is how your context is laid out) |
| 18:09.48 | bkruse | Yourname``: For sip debug? |
| 18:09.54 | [TK]D-Fender | Yourname`blah, blah, blah... Yeah I'm used to it :) NEVER ready to "show me the money" |
| 18:10.00 | madduck | bkruse, but i have multiple softphones and i can't change them all to the same number, and i cannot change the number per se |
| 18:10.15 | bkruse | madduck: ok, then change the context fromuser to piper, instead of the number |
| 18:10.17 | madduck | there must be a better way? |
| 18:10.30 | [TK]D-Fender | Yourname``: Maybe next time you'll ask when you're actually able to provide useful information and be prepared to do something about it :P |
| 18:10.38 | Yourname`` | [TK]D-Fender: LOL, I swear I will. I just thought it'd be easier if it was a conf thing, and with SO much cli stuff going on at this time.. it just takes a lot of time for a newb like me :D |
| 18:10.40 | madduck | bkruse: but what if someone else calls me on that number? |
| 18:10.48 | Yourname`` | bkruse: Yeah, and CLI output |
| 18:11.07 | *** join/#asterisk CVirus (n=GoD@196.205.191.81) |
| 18:11.08 | bkruse | Yourname``: sip set debug peer <peername> helps a ton |
| 18:11.09 | Yourname`` | bows to [TK]D-Fender, the sith lord of Asterisk |
| 18:11.29 | Yourname`` | bkruse: Oh shoot, that sounds neat. One sec.. |
| 18:11.35 | [TK]D-Fender | fries the channel with Force Lightning |
| 18:11.38 | Yourname`` | [TK]D-Fender: WE might just have a solutin, thanks to bkruse |
| 18:11.41 | bkruse | madduck: There is no problem at all....You just have it call piper (the fromuser only matters on the registration), asteirsk will call the endpoint that registered to it |
| 18:11.52 | bkruse | [TK]D-Fender is the bomb at helping, agreed |
| 18:12.08 | [TK]D-Fender | goes thermo-nuclear |
| 18:13.11 | Yourname`` | _THE_ bomb, might I add. He's also good at castrating me for obvious stupidity in conf files. He reminds me of Miss June, the strictest teacher I had in elementary. |
| 18:13.55 | Yourname`` | [TK]D-Fender: Great, 46 is off to lunch. |
| 18:13.57 | Yourname`` | :)( |
| 18:13.59 | Yourname`` | :( |
| 18:23.34 | CanWood | well, I found my problem: https://bugs.launchpad.net/ubuntu/+source/openh323/+bug/159307 |
| 18:23.53 | boblutz | ~digium |
| 18:23.53 | jbot | i guess digium is the creator, primary developer, and maintainer of Asterisk. They have a full-time team dedicated to open source Asterisk development which carries the majority of the load. They also sell various hardware and software products, as well as provide support and development services. http://www.digium.com/ |
| 18:29.14 | *** join/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 18:30.15 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:35.33 | Seggy | Qwell: That resolved it, I'll try your patch next |
| 18:37.58 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 18:40.50 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:40.58 | rupa | hi, I'm using asterisk Asterisk 1.4.18.1~dfsg-1 and trying to get followme working to my satisfaction. The problem I'm running into is getting it to dial multiple numbers simultaneously. If I have "number=>101,20,1&102,20,1" only the first extension is called. Help? |
| 18:41.31 | *** join/#asterisk riddlebox (n=JamesMid@159.251.13.2) |
| 18:42.46 | boblutz | clear |
| 18:42.48 | boblutz | damn! |
| 18:43.02 | mwalling | .w 17 |
| 18:44.14 | jer | anyone here on babytel with 1.4.18.1 ? i keep getting a probably a dns error for registration; but i've got the appropriate entry in /etc/hosts to point sip.babytel.ca to the ip of nat.babytel.ca... i've also tried commenting that out, and using the outboundproxy settings in sip.conf to point at nat.babytel.ca, no dice. i can paste stuff if i need too, my config worked fine on 1.4.11 |
| 18:44.55 | [TK]D-Fender | rupa: pastebin to actual complete line of dialplan you are debugging |
| 18:45.00 | rupa | if I put two lines for number=> and set priority of the second to 2, then it does a hunt style ring just fine. |
| 18:45.06 | rupa | [TK]D-Fender, ok |
| 18:48.24 | ZPertee | I have my ata hooked up my to an fxo port. why is the digium tdm card going nuts. switching between chan_zap.c:6685 handle_init_event: Detected alarm on channel 8: No Alarm and chan_zap.c:6678 handle_init_event: Alarm cleared on channel 8 |
| 18:49.04 | rupa | [TK]D-Fender, http://pastebin.com/d37413824 |
| 18:51.25 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 18:51.36 | [TK]D-Fender | rupa: number=>101,20,1&102,20,1 |
| 18:51.53 | [TK]D-Fender | rupa: You don't put ","'s between items in your & |
| 18:52.07 | [TK]D-Fender | rupa: number=>101&102,20,1 |
| 18:52.15 | [TK]D-Fender | rupa: go read the sample |
| 18:52.21 | rupa | reading again |
| 18:52.50 | [TK]D-Fender | rupa: ; number=> <number to call[&2nd #[&3rd #]]> [, <timeout value in seconds> [, <order in follow-me>] ] |
| 18:53.01 | rupa | bah, ok, read it wrong |
| 18:54.30 | rupa | nice |
| 18:55.10 | riddlebox | ZPertee: try running fxotune |
| 18:55.38 | rupa | now a real basic question, I'll go research, but... if there is a "no" answer it'll save time. Is it possible to do Followme type behavior without picking up the incoming call? That is, I want to not supervise the incoming call until either 1) someone picks up or 2) I hand off to voicemail |
| 18:55.43 | riddlebox | ZPertee: you have run "stop now" from the asteirsk cli |
| 18:55.56 | ManxPower | ZPertee: Search the mailing list archives |
| 18:56.24 | riddlebox | hey ManxPower :) |
| 18:56.30 | ManxPower | rupa: that is the default if you don't run some app that answers the line like Answer(), Background(), etc |
| 18:56.44 | rupa | so, I guess FollowMe by default answers. :( |
| 18:57.04 | ManxPower | rupa: You're not running a GUI or using config files from a GUI, are you? |
| 18:57.11 | rupa | no, no gui |
| 18:57.18 | ManxPower | rupa: usually show application followme would tell you |
| 18:58.41 | rupa | ManxPower, doesn't specify that it answers, but I guess it is implied 'cause it normally plays status msgs to the caller while finding the callee. can't send status info to caller without answering... |
| 18:59.14 | rupa | bummer, guess I'll have tol use the 1.2 examples for doing followme without the followme module then. |
| 19:00.16 | rupa | thnx |
| 19:00.29 | ManxPower | rupa: playing messages does not require an answer if you are on a PRI |
| 19:00.53 | *** join/#asterisk talntidwrk (n=swarm@66.208.251.162) |
| 19:01.01 | ManxPower | rupa: every single app that I've seen in Asterisk that would by default answer the call has an option to not answer. |
| 19:01.42 | ManxPower | followme could be an exception, but if it is, I'd consider it a bug |
| 19:02.00 | ManxPower | rupa: are you using PRI or VoIP? |
| 19:02.00 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 19:05.21 | scoates | bluregard: adjusting some NAT settings on the sipura boxes seems to have fixed it... I'm almost certain it's a firewall UDP timeout |
| 19:11.12 | *** join/#asterisk corrupt (i=81079d09@gateway/web/ajax/mibbit.com/x-04de2d309151fae8) |
| 19:11.27 | ZPertee | is the channelredirect app supported? |
| 19:12.06 | ManxPower | ZPertee: define "supported" |
| 19:12.14 | corrupt | how do i set up a 1 800 number using asterisk? |
| 19:14.44 | madduck | bkruse: i still just don't understand. i will take a moment and write up what I am trying to do... |
| 19:16.27 | riddlebox | corrupt: we need more info than that, do you want to dial an 800 number, or do you want to have an 800 number brought into your Asterisk system so people can call in on it? |
| 19:17.07 | generalhan | can anyone tell me how i change the context that you enter when transfering a call |
| 19:17.29 | Yourname`` | If I want the incoming phone number to be set as callerID, should it be Answer(), and then Set(CALLERID(number)=${EXTEN})? |
| 19:17.49 | corrupt | riddlebox, i want people to call in on it. |
| 19:18.36 | riddlebox | corrupt: do you have lines already coming into the asterisk machine? if so you can order an 800 number to piggy back on one of those, or you can order an 800 number from an ITSP |
| 19:19.45 | EmleyMoor | <PROTECTED> |
| 19:21.01 | ZPertee | ManxPower, it seems like I can use transfer app or channelredirect app to do the same thing. just wondering which to use so that if I have problems I don't get chewed out for using, such as when I used users.conf |
| 19:21.23 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 19:21.40 | The_X | Anyone else ever had DTMF issues when talking to acmepacket boxes? |
| 19:22.01 | corrupt | phmm |
| 19:22.03 | The_X | one of my customers is running switchvox and it just won't work when talking SIP with provider |
| 19:22.14 | The_X | I upgraded my own asterisk to 1.4 and it works fine now |
| 19:22.36 | The_X | called switchvox and they told me they can't upgrade their box so we're kinda screwed |
| 19:23.34 | corrupt | from my asterisk box could i then route the incoming calls to any destination using voip to side step any long distance fees? |
| 19:24.25 | [TK]D-Fender | corrupt: You can send any call arriving into your system anywhere you cofigure it to be able to go out through |
| 19:24.36 | ZPertee | The_X, it would be kind of a pain but if you absolutely had to could possibly put asterisk 1.4 in a virtual machine and then connect asterisk and switchvox together. |
| 19:24.42 | madduck | if someone has a moment to look over http://scratch.madduck.net/__tmp__vit.CrG29332, that would be great |
| 19:24.45 | madduck | i just don't get it. |
| 19:24.48 | EmleyMoor | is about to use "reverse hold with arbitrary return extension" in "anger" |
| 19:24.50 | madduck | bkruse: ^ |
| 19:25.26 | The_X | Zpertee, is it a known asterisk bug? |
| 19:25.40 | The_X | If so, I'll tell switchvox to fix their software |
| 19:26.31 | corrupt | so i can send any call anywhere i configure it to go, aye. that would take bandwidth, right? i would have to get a voip provider? |
| 19:27.06 | ZPertee | The_X, not sure on the asterisk side of things. no matter what will probably be a pain as most likely switchvox will take a while to get the bug worked out |
| 19:27.25 | robeph | wow someone has the strangest ideas about a dialplan, when an extensions is called, it rings 3 other extensions for 25 seconds, which is normally ok, but when no one answers it simply rings the same 3 extensions, forever. |
| 19:27.38 | ZPertee | The_X are you sure you have sip configured correctly? |
| 19:27.41 | The_X | so my customer is fucked! |
| 19:27.43 | robeph | seems a bit odd and redundant |
| 19:27.57 | The_X | Z, pretty sure, my asterisk 1.4 boxes work fine with the same type of setting |
| 19:28.12 | The_X | wasn't working on 1.2 |
| 19:29.08 | ZPertee | The_X, ah, I see. sorry don't know what to tell you. I've never had a problem with getting sip to work, so it seems weird |
| 19:29.30 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 19:29.36 | The_X | DTMF is the problem, not sip |
| 19:29.45 | The_X | working well when you send the call over a PSTN line |
| 19:29.54 | robeph | ZPertee: never used a netgear switch/router? |
| 19:30.00 | robeph | sip problems abound! |
| 19:30.23 | ZPertee | I don't use netgear routers but thanks for the heads up |
| 19:30.29 | [TK]D-Fender | The_X: Switchvox = GUI over *. If your "standalone *" box isn't working, then you didn't set it up right. |
| 19:30.33 | robeph | even with nat on, this one for a customer was mangling the packets like mad. |
| 19:30.51 | robeph | I mean it was passing them correctly, just, what was passed was no longer "sip" it was just trashed |
| 19:31.04 | The_X | Fender, my standalone boxes work fine on 1.4 |
| 19:31.04 | robeph | went to the right place though ;p |
| 19:31.15 | ZPertee | then it might as well have not passed them on then |
| 19:31.20 | robeph | yeah I know heh, |
| 19:31.21 | The_X | just that switchvox won't upgrade their box and since it's < 1.4 it just won't work properly |
| 19:31.40 | The_X | I'm sure I'm not the 1st guy ever to notice that problem |
| 19:32.16 | robeph | spent hours trying to figure wth was going on, cos I mean we were getting traffic, then had the bright idea to actually look AT the packets... which looked a total mess |
| 19:32.31 | ZPertee | The_X, yeah you sure wouldn't think so. who is the provider? sometimes there is problems with the provider running a newer verion of asterisk while a customers box is the older version |
| 19:32.45 | madduck | hey, who is 24.106.241.121 (rr.com)? |
| 19:35.00 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 19:35.04 | *** join/#asterisk kannan (n=kann@123.201.60.116) |
| 19:38.09 | The_X | I'm the provider |
| 19:38.13 | The_X | I have an acmepacket SBC |
| 19:38.15 | The_X | it just routes calls |
| 19:39.09 | EmleyMoor | A call I'd already been holding on for 12 minutes has come back from a 10 minute period on reverse hold, with me STILL in the queue! |
| 19:44.55 | generalhan | ohhh ... ive got a good one ! how can i set something up in extensions.conf so that i can play a sound file to a caller that im still connected with ? |
| 19:45.30 | generalhan | like i know i can transfer to an extension that is setup to play a sound, and then send the caller back to me. but i want to be on the phone with the caller while they are listening to the clip |
| 19:46.23 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 19:46.31 | The_X | you could easily conference ... |
| 19:46.40 | kombi | how do I easiest install app_jack? |
| 19:46.55 | generalhan | The_X: explain ! |
| 19:47.03 | kombi | is it in 1.6? |
| 19:47.20 | boblutz | The_X: ? |
| 19:47.24 | [TK]D-Fender | generalhan: go read up on "dynamic features" on the WIKI under "features.conf" |
| 19:47.34 | generalhan | [TK]D-Fender" perfect ! thanks ! |
| 19:47.47 | [TK]D-Fender | generalhan: It is "live" |
| 19:48.54 | kombi | or is it even in the tree and I just didn't see it? (app_jack that is) |
| 19:49.59 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:50.16 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 19:50.48 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 19:51.02 | generalhan | [TK]D-Fender" "audio output is only sent to one party (the caller or callee). " thats a shame ! cause that is SIMPLE ! |
| 19:52.02 | galeras | Dear Sirs, I'm looking for a paid help to resolv a nat issue, can someone help me please ? |
| 19:53.07 | kannan | a couple days back, {TK}D-Fender, said to me first ask in the main room, i did, and the issue was resolved, after a lot of googling tho |
| 19:53.27 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:53.39 | kannan | galeras -> i had also asked for paid support first thing |
| 19:54.07 | jameswf-home | is rebuilding the kernel in trixbox.... uh oh god just killed a kitten |
| 19:54.15 | galeras | kannan: my last resource :( |
| 19:55.15 | kannan | galeras -> i was also at wits end, maybe if you ask, it may be helped out by the good folks here |
| 19:56.55 | kombi | found the souce of app_jack now, do I just compile and throw in? |
| 19:57.14 | [TK]D-Fender | galeras: Read up : |
| 19:57.16 | [TK]D-Fender | ~sipnat |
| 19:57.17 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:57.19 | boblutz | kombi: `module load app_jack.so` after compilation! |
| 19:57.31 | madduck | check_auth: username mismatch, have <insphone>, digest has <0041325119511> -- how can i change this "insphone" thing? only by renaming the entire stanza? |
| 19:57.32 | [TK]D-Fender | I love Apple Jacks! |
| 19:57.35 | madduck | ([insphone]) |
| 19:57.38 | boblutz | LOL |
| 19:57.40 | kombi | boblutz: thanks! |
| 19:57.41 | boblutz | 9/10 |
| 19:58.16 | kombi | boblutz: and..erm, what was to command to compile c code again..? |
| 19:58.30 | boblutz | kombi: Depends on whether you have a makefile or not |
| 19:58.48 | madduck | insecure=invite |
| 19:58.55 | kombi | hmm, found it in the trunk, looks like the plain source |
| 19:59.38 | kombi | boblutz: so no makefile as I see it |
| 19:59.48 | lirakis | is away (leaving..."the internets" are safe ... for now) |
| 20:00.00 | boblutz | kombi: Is app_jack in the Asterisk source tree? |
| 20:00.25 | jameswf-home | asterisk would not perform my loop of death.... bastards |
| 20:00.37 | Katty | runs around causing trouble |
| 20:00.40 | kombi | boblutz: nope, only in svn |
| 20:01.50 | boblutz | well this just went from simple to tricky |
| 20:02.05 | kombi | sigh.. |
| 20:02.18 | Qwell | what? it's in asterisk |
| 20:02.46 | kombi | Qwell: really? where? did locate on that box a couple of times.. |
| 20:02.58 | Qwell | in a version that you need to have in order to run it |
| 20:02.58 | kombi | I'll do a find now.. |
| 20:03.07 | Qwell | 1.6, for example |
| 20:03.18 | Qwell | it's not gonna run without a bunch of other stuff |
| 20:03.42 | kombi | Qwell: I'll get 1.6 then..;) |
| 20:04.32 | boblutz | cs 1.6? |
| 20:05.02 | jameswf-home | I wonder how big I can make this kernel...... |
| 20:05.28 | kombi | Qwell: can I just install over 1.4? |
| 20:05.45 | jameswf-home | build the worlds fattest kernel.... weeeeeee |
| 20:06.19 | boblutz | jameswf-home: is that healthy? |
| 20:07.22 | jameswf-home | sure kernels can me fat... |
| 20:07.27 | jameswf-home | *be |
| 20:08.01 | *** join/#asterisk joejaxx (i=joejaxx@fluxbuntu/founder/joejaxx) |
| 20:08.19 | jameswf-home | I should put zaptel in to the kernel tre and static compile it that would add some size... loadable mdules pshh pohleaze |
| 20:09.57 | jameswf-home | thinks he wants a $5 sub... |
| 20:10.06 | joejaxx | Good Evening All |
| 20:10.29 | boblutz | joejaxx: So we meet again... |
| 20:10.45 | joejaxx | ? |
| 20:10.46 | joejaxx | lol |
| 20:10.56 | robeph | that reminded me for some reason of a bond movie |
| 20:11.07 | *** join/#asterisk Tuxofred (n=Fred@ip-80-236-192-102.dsl.scarlet.be) |
| 20:11.21 | jameswf-home | bondage.... james bondage........ |
| 20:11.28 | robeph | O.o |
| 20:11.38 | robeph | He likes to go by jim. |
| 20:11.41 | jameswf-home | double oh oh ohhhhh seven |
| 20:11.44 | robeph | that james stuff he says, shows his age |
| 20:12.16 | boblutz | yea, he tried to pull that on me one time but i didnt fall for it |
| 20:16.35 | ddunavant | Does anyone have any idea if ${IF($[ ${CallerID} = _7XX]?30:15)} would work? |
| 20:16.40 | ddunavant | CallerID is defined |
| 20:17.22 | boblutz | ddunavant: I may be wrong, but I think that is illogical |
| 20:17.36 | boblutz | its like seeing if NULL == NULL |
| 20:17.51 | _ShrikE | ${CALLERID} is incomplete I believe |
| 20:17.55 | outtolunc | a macro isn't a lookup <G> |
| 20:17.59 | _ShrikE | ${CALLERID(num)} |
| 20:18.13 | _ShrikE | or ${CALLERID(name)} |
| 20:18.18 | ddunavant | No, I made var CallerID |
| 20:18.25 | _ShrikE | ahh.. ok |
| 20:18.28 | ddunavant | and I defined it to something |
| 20:18.37 | ddunavant | for example 727 |
| 20:18.56 | ddunavant | I want it to try to test if the var is 700 something |
| 20:18.58 | boblutz | hates prime numbers! |
| 20:19.28 | ZPertee | I am trying to get asterisk to do a channelredirect or transfer without any success. My setup is as follows: POTS >> Asterisk FXO >> SIP ATA >> legacy pbx >> phone. I can't do a direct call transfer from the actual line so what I was wanting to do is put the call on hold (on the legacy pbx), dial an extension on the legacy pbx which asterisk is connected dial an extension and get asterisk to grap ahold of the call and send it |
| 20:19.29 | ZPertee | <PROTECTED> |
| 20:19.30 | _ShrikE | I dont think you can do an extensions pattern match in your expressions |
| 20:19.53 | ddunavant | would I be able to do any patter matching? maybe 7** or something? |
| 20:20.10 | boblutz | ddunavant: I would --> ${CallerID} > 700 && ${CallerID} < 800 |
| 20:20.26 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 20:20.34 | ddunavant | ok, that could work |
| 20:20.36 | _ShrikE | boblutz has an excellent idea |
| 20:20.42 | boblutz | realy? |
| 20:20.46 | ddunavant | thanks boblutz and ShrikE |
| 20:21.00 | boblutz | wow, i was expecting to get flamed and/or humiliated over that |
| 20:22.09 | *** join/#asterisk docelmo (n=vircuser@h44.196.180.12.cable.rstb.cablerocket.net) |
| 20:22.09 | _ShrikE | why? it accomplishes exactly what he was asking for :) |
| 20:22.39 | boblutz | haha |
| 20:22.46 | *** join/#asterisk southtel (n=southtel@24-240-24-20.dhcp.gwnt.ga.charter.com) |
| 20:22.54 | Katty | hugs _ShrikE |
| 20:22.56 | *** part/#asterisk southtel (n=southtel@24-240-24-20.dhcp.gwnt.ga.charter.com) |
| 20:23.12 | [TK]D-Fender | ZPertee: Because ZAP/8 isn't a channel |
| 20:23.24 | [TK]D-Fender | ZPertee: Go look in "core show channels for a clue. |
| 20:23.25 | Juggie | however the proper answer is 7XN as your pattern. |
| 20:24.41 | _ShrikE | hey Katty :) |
| 20:24.59 | Katty | allo |
| 20:25.02 | Katty | hugs grumpy [TK]D-Fender |
| 20:25.02 | boblutz | shakes fist |
| 20:25.14 | ZPertee | [TK]D-Fender, is this Zap/8-1 a channel? |
| 20:25.25 | ZPertee | [TK]D-Fender, thats what it showed |
| 20:25.47 | Katty | ZPertee: yes, that could be a channel |
| 20:26.15 | ZPertee | Katty, thanks. how you doin? |
| 20:26.28 | docelmo | anyone in here know why AGI would return a 200 code with -1 for the response? The command I sent was a EXEC DIAL command |
| 20:26.42 | Katty | ZPertee: slight headache. |
| 20:26.54 | Katty | docelmo: because you didn't MEW at it yet. |
| 20:27.17 | [TK]D-Fender | ddunavant: ${IF($["${CallerID:1}" = "7" | ${LEN(${CallerID})}= 3]?30:15)} |
| 20:27.20 | ZPertee | Katty, thanks for the update :-). its working now |
| 20:27.35 | [TK]D-Fender | Katty: not grumpy yet.... |
| 20:27.37 | [TK]D-Fender | TGIF! |
| 20:27.40 | Katty | oh hey |
| 20:27.42 | Katty | it IS friday |
| 20:27.47 | Katty | i'm so getting a mojito after work |
| 20:27.53 | Katty | with SUGAR CANE in it |
| 20:28.01 | [TK]D-Fender | Katty: Is that what the kids are calling it these days? ;) |
| 20:28.14 | Katty | [TK]D-Fender: go stand in the corner. |
| 20:29.03 | boblutz | Im so scared to run Asterisk as non-root |
| 20:29.11 | Katty | it's going to asplode |
| 20:29.26 | boblutz | I think the trick is to create /var/run/asterisk, and edit asterisk.conf to tell it where the run directory is |
| 20:29.29 | Katty | chmod -R 777 / |
| 20:29.32 | Katty | ^- that'll fix it ;) |
| 20:29.50 | boblutz | rm -rf * / ? |
| 20:30.36 | [TK]D-Fender | boblutz: there is a chapter in the book on this... |
| 20:30.48 | Katty | speaking of chapters in books, i've been meaning to read up on hotdesking |
| 20:30.54 | boblutz | [TK]D-Fender: I know this |
| 20:31.02 | [TK]D-Fender | ok, checkout time, heading home, later all |
| 20:31.08 | Katty | later fender |
| 20:31.12 | boblutz | may the force be with you |
| 20:32.41 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-87-253.vif.net) |
| 20:32.44 | boblutz | I need to convert these wu-tang songs so i can use them for MOH |
| 20:32.49 | boblutz | sucks |
| 20:33.52 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 20:35.50 | docelmo | Ill MEW At it.. I will kick it in the ovaries.. |
| 20:36.23 | docelmo | anyone in here know the underlying API for AGI? |
| 20:36.57 | boblutz | docelmo: Asterisk? |
| 20:38.00 | boblutz | docelmo: res/res_agi.c in Asterisk source tree might be of interest |
| 20:39.05 | ManxPower | docelmo: define "underlying API" |
| 20:39.31 | ManxPower | The AGI API is pretty well documented. |
| 20:40.00 | ManxPower | the API that res_agi.c uses to communicate with Asterisk is well documented |
| 20:41.49 | *** join/#asterisk metfan2007 (n=metfan20@201.103.91.164) |
| 20:42.27 | metfan2007 | Hi all!!! Do you know how many simultaneous Agents can Asterisk handle in 1 server?? Last Asterisk 1.4 release |
| 20:42.34 | docelmo | ManxPower ya I know.. I have looked at it.. I am trying to understand why when I do a request I get the following.. lemme pastebin |
| 20:43.09 | *** join/#asterisk hugohagogo (n=cleber@189.23.20.10) |
| 20:43.15 | ManxPower | docelmo: but that is not what I asked for. |
| 20:43.30 | docelmo | http://pastebin.com/d40afa51f |
| 20:43.30 | jameswf-home | how cool is that I drove to subway, sat down, ate a footlong club, drove back, and my kernel is still compiling modules |
| 20:43.44 | boblutz | jameswf-home: LOL |
| 20:43.48 | boblutz | did you get teh veggie-max? |
| 20:44.02 | jameswf-home | always orders "the works" |
| 20:44.10 | boblutz | is a vegetarian |
| 20:44.12 | docelmo | ManxPower well when I say underlying API I mean how it acts.. Cause its doing some really funky shit |
| 20:44.18 | docelmo | MEW! |
| 20:44.27 | jameswf-home | jared would not have lost weight the way I eat subway |
| 20:44.28 | docelmo | Im a little mind boggled.. |
| 20:44.34 | docelmo | MEW.... |
| 20:44.38 | ManxPower | docelmo: I suppose you won't answer this question either: Are you using the PHPAGI library or are you trying to handle it manually? |
| 20:44.40 | Katty | what? |
| 20:44.40 | Katty | oh |
| 20:44.46 | Katty | good job. |
| 20:44.46 | jameswf-home | ~moo |
| 20:44.47 | jbot | ACTION mooooooooo! I am cow, hear me moo, I weigh twice as much as you. I am cow, eating grass, methane gas comes out my ass |
| 20:44.50 | Katty | jbot: mew? |
| 20:44.51 | jbot | A MIME mail reader for Emacs/XEmacs. URL: http://www.mew.org/ |
| 20:44.54 | docelmo | I am using what I helped write.. :) phpagi |
| 20:45.03 | jameswf-home | ~pew |
| 20:45.07 | Katty | pewpew! |
| 20:45.08 | jameswf-home | ~obama |
| 20:45.12 | Katty | ~Katty |
| 20:45.13 | jbot | i guess katty is the only girl in the channel, so be nice to her |
| 20:45.20 | Katty | wow, surely that's not true |
| 20:45.24 | ManxPower | docelmo: I don't see any reading of STDIN |
| 20:45.27 | jameswf-home | ~girls |
| 20:45.28 | jbot | girls are sweet, wonderful and necessary :) |
| 20:45.35 | jameswf-home | ~women |
| 20:45.37 | defsdoor | Girls ? here ? |
| 20:45.38 | docelmo | Katty haha.. girl? hmmm |
| 20:45.42 | ManxPower | If you don't do that to get all the info Asterisk sends the app on startup BAD THINGS will happen |
| 20:45.44 | jameswf-home | ~porn |
| 20:45.45 | jbot | Porn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type. |
| 20:45.46 | drmessano | Theres probably other women hiding as men's nicknames |
| 20:45.49 | drmessano | Thats how it goes |
| 20:46.01 | defsdoor | drmessano: usually is the other way around |
| 20:46.02 | docelmo | ManxPower tell me about it.. Im trying to resolve one of my clients issues now |
| 20:46.04 | ManxPower | There's nothing wrong with playing with the bot, as long as you do it in private and wash your hands after. |
| 20:46.07 | jameswf-home | we have a girl tech here |
| 20:46.16 | Katty | docelmo: are you questioning my feminininininiity |
| 20:46.21 | docelmo | yes |
| 20:46.23 | Katty | i so can even say that word |
| 20:46.27 | drmessano | defsdoor thinks you're a man |
| 20:46.34 | boblutz | ~drmessano |
| 20:46.35 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway |
| 20:46.36 | Katty | really? |
| 20:46.40 | boblutz | just let him look |
| 20:46.43 | ManxPower | ~manxpower |
| 20:46.44 | jbot | you are, like, Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. |
| 20:46.52 | defsdoor | I don't - I'm just shocked that there is a girl present and yet you guys are still typing |
| 20:46.54 | drmessano | Shameless plug? |
| 20:46.56 | boblutz | ~boblutz |
| 20:46.57 | defsdoor | one handed maybe ? |
| 20:47.00 | boblutz | a straight g |
| 20:47.12 | drmessano | Who said I was typing |
| 20:47.20 | ManxPower | boblutz: she'd castrate anyone that hit on her. |
| 20:47.23 | drmessano | Dragon naturally speaking (TM) FTW |
| 20:47.36 | boblutz | ManxPower: ? |
| 20:47.52 | Katty | defsdoor: i don't count. i'm always here. |
| 20:48.08 | ManxPower | boblutz: sorry, that was for defsdoor |
| 20:48.13 | boblutz | o lol |
| 20:48.53 | ManxPower | docelmo: also in some versions of Asterisk you can't get Asterisk created variables, but I thought that was fixed in 1.2. |
| 20:49.09 | drmessano | I tried to hit on Katty once...I woke up with Ferret bites all over me |
| 20:49.31 | docelmo | Ya.. I am using 1.4 It was working fine for days but now they are running about 240 calls on average over fastagi and I think fastagi is puking |
| 20:50.27 | docelmo | ~docelmo |
| 20:50.34 | docelmo | damn Im nothing |
| 20:50.35 | docelmo | :( |
| 20:50.37 | jameswf-home | ~jameswf-home |
| 20:50.37 | jbot | when -home is added it means he is on his own time dont call his boss |
| 20:50.45 | jameswf-home | ~jameswf |
| 20:50.46 | jbot | jameswf loves unsolicited technical support |
| 20:50.46 | jasonwoot | the female of the species is more deadly than the male |
| 20:50.57 | ManxPower | ~zeeek |
| 20:50.57 | jbot | from memory, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 20:51.06 | drmessano | jameswf, I need help with this Sangoma card.... |
| 20:51.15 | jameswf-home | ~bacon |
| 20:51.16 | jbot | bacon is probably a filling substance which will make you even fatter than you already are... Take my advice, bacon is a food for the gods or for underweight models such as that McBeal chick... |
| 20:51.33 | jameswf-home | ~sangoma |
| 20:51.33 | jbot | rumour has it, sangoma is a Canadian based company that makes PRI and Analog cards. See their site at http://www.sangoma.com/ |
| 20:51.44 | boblutz | I dont eat meat though |
| 20:51.48 | jameswf-home | canada psh |
| 20:51.50 | docelmo | how did you guys get the bot to recgonize those code words? |
| 20:52.03 | boblutz | speaking of which, toronto fc is getting crushed tomorrow |
| 20:52.04 | Qwell | docelmo: send Tim Riker lots and lots of money |
| 20:52.05 | _ShrikE | ~botsnack |
| 20:52.05 | jbot | _ShrikE: thanks |
| 20:52.05 | jameswf-home | bacon isnt meat.... ita more of an animal fat |
| 20:52.08 | drmessano | It's a secret that only Asterisk 1.8 users know |
| 20:52.12 | Qwell | fat is meat too.. |
| 20:52.30 | denon | drmessano: you're still on 1.8? we just moved all our prod to the secret 2.0beta |
| 20:52.44 | drmessano | Crap.. Guess I got behind :( |
| 20:52.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:52.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:52.54 | ManxPower | denon: Is that the Windows 2008 server edition of Asterisk? |
| 20:52.56 | denon | maybe you haven't heard about it .. it is secret, after all |
| 20:53.12 | denon | ManxPower: nope, it's compiled directly into JunOS |
| 20:53.13 | jameswf-home | 2.5 hours to put kernel debuger on a 1.5 intel |
| 20:53.14 | drmessano | It's hard to keep up with all the forking |
| 20:53.43 | drmessano | Microsoft Live AsteriskWin32 2008 Enterprise? |
| 20:53.54 | jameswf-home | dares drmessanoto go ask in #freeswitch why an asterisk fork doesn't configure like asterisk |
| 20:53.55 | boblutz | Where would you guys place 2 fxo modules on a tdm410 card? |
| 20:53.57 | drmessano | I'll take 2 copies, thank you |
| 20:54.03 | boblutz | 1-3? 2-4? 1-4? |
| 20:54.09 | boblutz | 1-2 ? |
| 20:54.12 | drmessano | I would put them on the bottom |
| 20:54.15 | boblutz | oic |
| 20:54.16 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:54.20 | Katty | HAI |
| 20:54.35 | ManxPower | boblutz: Unless you have 1.0.x zaptel and a rev of carrier card before Rev F (I think) It does not matter. |
| 20:55.24 | boblutz | ai ai |
| 20:56.13 | boblutz | picks 1-4 ftw |
| 20:56.58 | ManxPower | "Asterisk Business Edition for Windows" -- Coming on April 1 2008 |
| 20:57.34 | Qwell | ManxPower: that press release hasn't gone out yet |
| 20:57.35 | denon | ManxPower: you know, I still dont understand why people care what OS their PBX runs on |
| 20:57.41 | denon | it should really just be an appliance .. |
| 20:57.47 | denon | thing could run on BeOS and it shouldn't really matter |
| 20:58.00 | drmessano | I want my PBX to go down because of a virus |
| 20:58.05 | drmessano | Thats my goal |
| 20:58.07 | denon | and yet we hear the whole debian/ubuntu/mac/etc thing |
| 20:58.11 | denon | drmessano: run it on a mac |
| 20:58.19 | drmessano | I want 100 phones going full bore |
| 20:58.21 | drmessano | Then ZIP |
| 20:58.25 | drmessano | and check the server |
| 20:58.28 | drmessano | Bam, Happy99 |
| 20:58.29 | boblutz | asterisk32.dll |
| 20:58.31 | denon | drmessano: run it on a mac |
| 20:58.32 | ManxPower | denon: mainly people want what they are familiar with |
| 20:58.33 | denon | :) |
| 20:58.52 | denon | ManxPower: yeah .. but why? to install the kernel and edit asterisk configs? |
| 20:58.55 | denon | nothing else should be happening .. |
| 20:59.49 | jameswf-home | heh http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-backdoor-china-what-heck |
| 21:00.08 | denon | isnt that incredibly old news? |
| 21:00.16 | denon | oh, china.. heh |
| 21:00.18 | jameswf-home | no different subject |
| 21:00.28 | jameswf-home | dude got owned |
| 21:00.46 | denon | hah |
| 21:00.49 | denon | yeah |
| 21:01.08 | drmessano | Wow, I can't imagine a trixbox ever being hacked |
| 21:01.21 | drmessano | It's too green for that |
| 21:01.24 | Katty | roful mow |
| 21:01.26 | Katty | do do be do do |
| 21:01.29 | Katty | roful mow do do be do |
| 21:01.31 | Qwell | ~roflmao |
| 21:01.32 | jbot | hmm... roflmao is rolling on the floor laughing my arse off, or painful |
| 21:01.35 | Qwell | aww, no link |
| 21:01.40 | Qwell | jameswf-home: video, now |
| 21:01.41 | jameswf-home | ~fleamarket |
| 21:01.41 | jbot | Fleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E |
| 21:01.58 | Katty | Qwell: http://angela.sleekgeek.org/2008/03/08/roful-mow/ |
| 21:02.32 | Qwell | Katty: close enough |
| 21:02.46 | Katty | roxorz boxors... omg? hax. |
| 21:03.00 | jameswf-home | ~roflmao is also http://www.youtube.com/watch?v=iEWgs6YQR9A |
| 21:03.01 | jbot | jameswf-home: okay |
| 21:03.09 | Qwell | jameswf-home: thank you |
| 21:03.59 | Katty | Qwell: that's my ringtone on my blackberry |
| 21:04.04 | Qwell | heh |
| 21:04.05 | riddlebox | drmessano: nice |
| 21:04.05 | Katty | Qwell: the rofulmow |
| 21:04.10 | Qwell | mp3 link? :D |
| 21:04.15 | Katty | no. |
| 21:04.18 | Qwell | lame |
| 21:04.21 | Katty | i flved it. |
| 21:04.25 | Katty | then turned it into avi |
| 21:04.29 | Qwell | that'd work |
| 21:04.32 | Katty | then into wav and snippeted |
| 21:04.40 | Katty | and then into mp3, and emailed it to myself to save |
| 21:04.43 | Katty | keepvid.com |
| 21:05.09 | Qwell | mencoder <link> -o /mnt/myphone/Music/roflmao.mp3 |
| 21:06.44 | jameswf-home | http://www.youtube.com/watch?v=9alejPWHboc heh |
| 21:06.52 | jameswf-home | ~spiderpig |
| 21:06.59 | jameswf-home | ~spiderpig is http://www.youtube.com/watch?v=9alejPWHboc |
| 21:07.00 | jbot | okay, jameswf-home |
| 21:07.00 | Qwell | ha |
| 21:07.02 | Qwell | that's my current ringtone |
| 21:07.05 | russellb | you guys are so off topic, it's a bit out of hand. |
| 21:07.08 | russellb | looks at Qwell |
| 21:07.13 | jameswf-home | I have it set on my email :)) |
| 21:07.16 | Katty | gives russellb a cookie |
| 21:07.18 | file | Qwell: back to work with you |
| 21:07.25 | jameswf-home | Qwell: started it |
| 21:07.31 | jameswf-home | he should know better |
| 21:07.42 | russellb | yes, I am blaming Qwell :) |
| 21:08.29 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:10.18 | rupa | Dial requires a Channel/#. Is there a way to tell it to use an extension? |
| 21:10.36 | rupa | (so it understands one level of indirection) |
| 21:10.44 | russellb | use a Local channel |
| 21:10.58 | russellb | Dial(Local/extension@context) |
| 21:11.03 | Seggy | Qwell: regarding the chopping/cruddy audio, my svn tree already had your patch applied and did it without the optimization turned off, so it appears the patch does not resolve it. |
| 21:11.06 | rupa | oh, was missing @context |
| 21:11.36 | Qwell | already had it applied? |
| 21:12.16 | Seggy | yeah, when I tried to patch it it said it appeared it was already applied so I compared the diff to codecs/log2comp.h and they matched |
| 21:12.32 | Qwell | my patch wasn't for log2comp.h |
| 21:13.22 | Qwell | 11243-maybe-asm.diff |
| 21:13.35 | Seggy | http://bugs.digium.com/view.php?id=11243#84195 |
| 21:13.36 | Seggy | ah |
| 21:14.19 | boblutz | Is that the real allison or a TTS allison on the Digium IVR? |
| 21:15.13 | boblutz | And this Digium MOH is really inspirational |
| 21:15.30 | ManxPower | What doe sit inspire you to do? Kill yourself? |
| 21:15.37 | ManxPower | Most hold music does |
| 21:15.54 | boblutz | makes it count |
| 21:16.34 | Seggy | compiling... |
| 21:18.17 | *** join/#asterisk unbkbl (n=work@static-adsl201-232-88-87.epm.net.co) |
| 21:18.22 | unbkbl | clear |
| 21:18.49 | [TK]D-Fender | defibrulates the channel |
| 21:19.12 | mwalling | unshockable rythem |
| 21:20.35 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 21:21.11 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 21:22.02 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 21:22.30 | kombi | addons 1.6 fail to compile at res_config_mysql.o, 1.4 compiled fine, what must I do? |
| 21:26.34 | kombi | ..compiling without mysql now.. |
| 21:26.59 | kombi | ..same thing! weired.. |
| 21:27.09 | joejaxx | boblutz: why do you say that? |
| 21:27.10 | joejaxx | :P |
| 21:27.23 | boblutz | say what? |
| 21:27.26 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-74-210.vif.net) |
| 21:27.36 | joejaxx | that we meet agian :P |
| 21:27.38 | joejaxx | again* |
| 21:28.05 | boblutz | haha I was just messing around |
| 21:28.11 | joejaxx | oh lol |
| 21:28.29 | *** join/#asterisk _Vile (n=vile@208.100.152.234) |
| 21:28.38 | jameswf-home | 3hours and running on kernel watch |
| 21:29.27 | joejaxx | boblutz: i was going to ask whether my visit to this channel was that memorable haha |
| 21:29.30 | joejaxx | lol |
| 21:29.44 | boblutz | haha |
| 21:30.14 | joejaxx | bah |
| 21:30.17 | joejaxx | this is annoying |
| 21:30.28 | joejaxx | this 7960 will not register with asterisk |
| 21:30.30 | joejaxx | :P |
| 21:30.38 | jameswf-home | ~7960 |
| 21:30.44 | boblutz | this tdm410 will not stop making phantom calls! |
| 21:30.50 | jameswf-home | doesnt work with jbot either |
| 21:31.03 | joejaxx | i ran into this the last time i setup a 79X0 based voip network |
| 21:31.09 | Katty | boblutz: in sleep he calllleddd to me.... |
| 21:31.13 | joejaxx | and i do not remember what i did to resolve it |
| 21:31.13 | Katty | boblutz: in dreamsss he rang!!! |
| 21:31.16 | joejaxx | Katty: loll |
| 21:31.24 | joejaxx | jameswf-home: :P |
| 21:31.36 | Katty | jbot: Phantom |
| 21:31.36 | jbot | well, phantom is the BYU UUG server, recently equiped with a new hard drive which now holds several mirrors |
| 21:31.39 | Katty | jbot: Phantom of the Opera |
| 21:31.43 | Katty | :< |
| 21:32.20 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 21:33.41 | *** part/#asterisk riddlebox (n=JamesMid@159.251.13.2) |
| 21:34.55 | *** part/#asterisk MattJ (n=MattJ@88-110-69-232.dynamic.dsl.as9105.com) |
| 21:35.48 | *** join/#asterisk hugohagogo (n=cleber@189.23.20.10) |
| 21:36.38 | *** join/#asterisk Goldfisch (n=gregturn@user-142gb70.cable.mindspring.com) |
| 21:36.50 | *** part/#asterisk Goldfisch (n=gregturn@user-142gb70.cable.mindspring.com) |
| 21:39.10 | Seggy | Qwell: Yup |
| 21:39.14 | Seggy | patch seems to solve the issue |
| 21:39.15 | Qwell | yup what? |
| 21:39.29 | Qwell | compiled without DONT_OPTIMIZE? |
| 21:39.43 | Seggy | yes |
| 21:39.47 | Qwell | great |
| 21:39.48 | Seggy | did a make distclean |
| 21:39.51 | Seggy | applied the patch |
| 21:39.54 | Seggy | configured and compiled |
| 21:39.58 | Qwell | same compiler? |
| 21:40.00 | Seggy | without DONT_OPTIMIZE |
| 21:40.01 | Seggy | yup |
| 21:40.02 | Seggy | 4.2.3 |
| 21:40.09 | Qwell | perfect, committing |
| 21:40.10 | Seggy | gcc version 4.2.3 (Debian 4.2.3-2) |
| 21:41.35 | Qwell | 1.4? |
| 21:41.50 | Seggy | latest trunk |
| 21:41.58 | Seggy | well, 3/26 trunk |
| 21:42.04 | Qwell | k |
| 21:42.11 | Seggy | Asterisk SVN-trunk-r110621M |
| 21:46.02 | Qwell | and fixed |
| 21:46.33 | *** join/#asterisk DarWin_vcch (i=yellow@unaffiliated/darwin) |
| 21:46.35 | *** part/#asterisk DarWin_vcch (i=yellow@unaffiliated/darwin) |
| 21:50.11 | boblutz | asterisk? |
| 21:50.21 | jer | if i've got an extension which comes in via a reinvite, and i want calls into that extension to automatically call some other number out another sip channel, but i want to preserve the caller id, how can i do that? exten => 500,1,Set(CALLERID(num)=${CALLERID(num)}) doesn't work as i'd expect |
| 21:51.14 | kombi | ast_yyerror(): syntax error: syntax error, unexpected '=' <- formely working dialplan, what might it be? |
| 21:54.14 | ManxPower | jer: that is like saying 1=1 |
| 21:54.24 | ManxPower | It should preserve the callerid by default. |
| 21:54.37 | ManxPower | Look at the "o" option of "show application dial |
| 21:54.40 | jer | well, in the console it shows Set("SIP/500-...", "525") |
| 21:54.43 | jer | calling from exten 525 |
| 21:54.48 | [TK]D-Fender | kombi, pastebin the whole mess |
| 21:54.50 | jer | ManxPower, ok |
| 21:55.05 | [TK]D-Fender | jer, You too |
| 21:55.13 | boblutz | boblutz: you three! |
| 21:57.13 | generalhan | [TK]D-Fender: so i setup my dynamic features thing ... but for some reason i can only get it to work if im doing it for the callee and not the caller |
| 21:57.15 | jer | ManxPower, problem is, i'm using a macro to dial out (incase one of the channels is unavailable), and 'o' isn't practical in that macro |
| 21:57.19 | jer | [TK]D-Fender, http://pastebin.com/m2e7fdadc |
| 21:57.33 | generalhan | it just send the tones to the caller: http://pastebin.com/d637c4beb <---- this is my setup ... seems right to me 1 |
| 21:57.34 | jer | relevant part only, i can paste the macro if needed, but it works fine |
| 21:58.03 | kombi | Fender: and here's me.. http://pastebin.se/193969 |
| 21:58.53 | *** join/#asterisk bminish (n=bminish@brenbox.home.minish.org) |
| 21:59.55 | [TK]D-Fender | jer, I said everything. |
| 22:00.09 | [TK]D-Fender | jer, that measn every device used, and ever bit of dialplan it passes through |
| 22:00.24 | jer | oh lord, that's like 100kb worth of config files |
| 22:00.46 | jer | fuck it, i'll just write a separate macro for this one single extension |
| 22:00.51 | boblutz | WHOA! |
| 22:02.23 | *** join/#asterisk MazeMiami (i=MazeMiam@98.211.197.119) |
| 22:02.30 | kombi | could it be that 1.6 needs every variable to be defined beforehand (as opposed to 1.4)? |
| 22:02.41 | MazeMiami | Hello People |
| 22:04.04 | [TK]D-Fender | kombi, pastebin the failed call attempt |
| 22:04.28 | kombi | http://pastebin.se/193969 |
| 22:05.49 | [TK]D-Fender | kombi, exten => 99,4,Gotoif($[${NEXTOK}=yes]?5:7) |
| 22:06.07 | [TK]D-Fender | kombi, At this point in time ${NEXTOK} is completely blank. |
| 22:06.25 | [TK]D-Fender | kombi, remeber how dumb * is with dialplan variables & parsing. |
| 22:06.39 | [TK]D-Fender | kombi, exten => 99,4,Gotoif($["${NEXTOK}"="yes"]?5:7) ; <- better |
| 22:06.55 | [TK]D-Fender | kombi, do this for ALL of your text-comparisons. |
| 22:07.18 | [TK]D-Fender | kombi, exten => 99,7,Gotoif($[${count}=0]?8:9) ;<- bad |
| 22:07.20 | MazeMiami | having an issue with Free PBX |
| 22:07.25 | [TK]D-Fender | kombi, exten => 99,7,Gotoif($["${count}"="0"]?8:9) ;<- better |
| 22:07.30 | [TK]D-Fender | ~freepbx |
| 22:07.30 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:07.34 | kombi | ;).. all right.. thanks Fender! so 1.6. is more strict there.. |
| 22:07.35 | MazeMiami | I would like to set up 100s of number to all ring for x amount of times, then go to there own voice mail |
| 22:07.45 | MazeMiami | All these numbers are not connected to a soft phone or any other type of phone i would like them to ring virtually. |
| 22:07.51 | MazeMiami | Maybe just have the system play back a .wav that sound as if it was ringing ? |
| 22:07.53 | [TK]D-Fender | kombi, No, ALL vers of * have that effect |
| 22:07.54 | galeras | i suspect my nat issues is because my iptables conf (http://www.pastebin.ca/961434), please, could you tell me if you see something wrong on it? |
| 22:08.03 | kombi | Fender <- great guy! |
| 22:08.23 | [TK]D-Fender | MazeMiami, FreePBX is not supported here, please ask in their channel. |
| 22:08.49 | MazeMiami | darn, no one in there knows what there doing :) |
| 22:09.03 | drmessano | They dont? |
| 22:09.03 | *** join/#asterisk hubguruJR (n=hubguruJ@208.81.48.254) |
| 22:09.17 | MazeMiami | not really |
| 22:09.25 | drmessano | or maybe your idea is stupid and no one wants to help |
| 22:09.26 | MazeMiami | not in this particluar issue at least |
| 22:09.55 | MazeMiami | maybe |
| 22:10.21 | drmessano | But going to #Asterisk to insult the users in #freepbx because they didnt help you is pretty lame |
| 22:10.25 | drmessano | Epic fail |
| 22:11.03 | MazeMiami | i wasn't insulting them, they were just unsure of the solution to my question |
| 22:11.19 | drmessano | No |
| 22:12.11 | [TK]D-Fender | MazeMiami, What you want to do is perfectly easy in *. |
| 22:12.30 | drmessano | He wants to play a ringing to them |
| 22:12.38 | [TK]D-Fender | MazeMiami, "show application ringing" |
| 22:12.42 | [TK]D-Fender | \o/ |
| 22:12.46 | drmessano | lol |
| 22:12.47 | [TK]D-Fender | yippy-kai-yay |
| 22:13.22 | boblutz | .||.| |
| 22:13.32 | drmessano | Playing ringing to a caller so they don't know the users softphone is offline before sending it to VM, versus direct to VM because the softphone is offline |
| 22:13.38 | [TK]D-Fender | (o)(o) ? |
| 22:13.44 | boblutz | LOL |
| 22:13.49 | lmadsen | (. )(. ) |
| 22:13.55 | drmessano | O.o |
| 22:14.04 | MazeMiami | exactly |
| 22:14.23 | [TK]D-Fender | ($)($) <- Pamela Anderson's ;) |
| 22:14.31 | drmessano | I fail to see the usefulness of such a task |
| 22:14.45 | lmadsen | who said I was useful? |
| 22:14.50 | drmessano | "Oh, it's ringing.. he must be there... " *BAM* voicemail |
| 22:14.51 | lmadsen | I'll kill them! |
| 22:14.58 | drmessano | Versus |
| 22:15.02 | drmessano | *BAM* voicemail |
| 22:15.03 | lmadsen | faking ringing seems useless to me |
| 22:15.10 | drmessano | Exactly |
| 22:15.18 | drmessano | "So they dont know my softphone is offline" |
| 22:15.27 | lmadsen | maybe a msg saying, "pls wait while we try to contact the party you are dialing" |
| 22:15.27 | *** part/#asterisk hubguruJR (n=hubguruJ@208.81.48.254) |
| 22:15.41 | drmessano | Sounds like "I don't want to the boss to know I've left the office" to me |
| 22:15.41 | *** part/#asterisk galeras (n=galeras@201.244.195.99) |
| 22:15.55 | drmessano | Sounds like "I don't want the boss to know I've left the office" to me |
| 22:15.59 | lmadsen | I don't have that problem... I *am* the boss ) |
| 22:16.03 | drmessano | lol |
| 22:16.35 | lmadsen | hrmmm... it looks like beer time |
| 22:16.37 | drmessano | Playing someone a fake ring before sending them to an eventual VM box so they'll think the user is there, just busy, is lame |
| 22:16.56 | lmadsen | ok, I think we've killed that conversation :) |
| 22:16.58 | drmessano | So I refused to help for the 2 days hes been spamming #freepbx for help.. |
| 22:17.03 | drmessano | lol |
| 22:17.05 | drmessano | So, I mock |
| 22:17.10 | lmadsen | mocka mocka mock |
| 22:17.12 | lmadsen | a |
| 22:17.14 | lmadsen | heh |
| 22:17.19 | drmessano | Remember "Spies like us" |
| 22:17.24 | *** join/#asterisk hubguruJR (n=hubguruJ@208.81.48.254) |
| 22:17.26 | drmessano | "We mock what we don't understand" |
| 22:17.27 | lmadsen | nope, never heard of it |
| 22:17.29 | drmessano | Damn |
| 22:17.30 | lmadsen | I'm not that old |
| 22:17.36 | lmadsen | oh burn |
| 22:17.41 | drmessano | Dan Aykroyd, Chevy Chase |
| 22:17.45 | drmessano | Great movie |
| 22:17.50 | lmadsen | sounds like an 80s movie |
| 22:17.53 | generalhan | very great movie |
| 22:17.57 | drmessano | Oh, it's VERY 80s |
| 22:18.00 | lmadsen | I was no more than 9 y/o in the 80s |
| 22:18.21 | drmessano | It's not only set in the 80's, but it's funny and cheesy as hell |
| 22:18.27 | lmadsen | heh |
| 22:18.28 | drmessano | Worth watching A++++++ |
| 22:18.40 | *** join/#asterisk doolph (n=doolph@200.75.192.45) |
| 22:18.40 | lmadsen | nothing from the 80s is worth watching |
| 22:19.06 | _ShrikE | lmadsen: not even the A Team? |
| 22:19.12 | drmessano | shenanigans |
| 22:19.15 | lmadsen | _ShrikE: especially the A Team |
| 22:19.19 | drmessano | Airwolf |
| 22:19.23 | drmessano | Knight Rider |
| 22:19.26 | *** part/#asterisk hubguruJR (n=hubguruJ@208.81.48.254) |
| 22:19.33 | drmessano | The Greatest American Hero |
| 22:19.37 | lmadsen | even though I have the first season of the A-Team on DVD on my shelf |
| 22:19.44 | generalhan | MacGyver !!!!!!!!!!!! |
| 22:19.48 | drmessano | I pity the fool |
| 22:19.55 | boblutz | holy shite |
| 22:19.59 | boblutz | i was only 3 in 1990 |
| 22:20.07 | lmadsen | seriously? |
| 22:20.11 | boblutz | ya |
| 22:20.16 | lmadsen | wow... that makes me feel old, and I'm only 27 :) |
| 22:20.21 | drmessano | MacGyver could build a working PBX with a pen and some shaving cream.. if you want voicemail, he's gonna need a toothpick |
| 22:20.27 | MazeMiami | any known ways to accomplishes this ? |
| 22:20.55 | drmessano | MazeMiami: Try a Wizard |
| 22:21.00 | Juggie | lmadsen, you were born in 81 though right? |
| 22:21.01 | drmessano | ~wizard |
| 22:21.01 | jbot | wizard is, like, enchancement to howto's |
| 22:21.04 | lmadsen | Juggie: yep |
| 22:21.23 | lmadsen | the only other year that might compete for coolness is 1984 |
| 22:21.26 | MazeMiami | a wizardd ? |
| 22:21.30 | drmessano | My wife was born in 80.. what a difference 4 years makes |
| 22:21.49 | Juggie | lmadsen, i have to book my travel for * con, wed night if nothing else happens can we do supper? and i'll book like the last flight off the island. |
| 22:22.00 | lmadsen | Juggie: you mean IT360? :) |
| 22:22.04 | Juggie | oh ya |
| 22:22.06 | Juggie | woops :) |
| 22:22.08 | Juggie | brain fart. |
| 22:22.12 | lmadsen | Juggie: haha.. np, ya, I'll be around |
| 22:22.28 | lmadsen | if all else fails you can come to my place and drink some beers |
| 22:22.36 | boblutz | what kind of beer? |
| 22:22.40 | Juggie | you live close to the island airport |
| 22:22.48 | lmadsen | Juggie: lol, ya, you can walk to it from my place |
| 22:23.04 | Juggie | at 8:30 in toronto, will you escort me? :P |
| 22:23.11 | lmadsen | I live at Bathurst and Lakeshore, and the ferry for the airport is just south of Lakeshore and Bathurst |
| 22:23.23 | lmadsen | Juggie: I'll point at the airport from my balcony for you :) |
| 22:23.36 | Juggie | ok, but you have to watch me walk there :) j/k |
| 22:23.43 | lmadsen | I could actually :) |
| 22:24.26 | *** join/#asterisk ITiliti (n=ITiliti@mail.itility.us) |
| 22:24.36 | Juggie | hmmmm |
| 22:24.46 | Juggie | when i dont need a static bag, i have too many |
| 22:24.46 | ITiliti | I am having an issue with an Ifexec statement, anyone have a second to help me iwth it? |
| 22:24.53 | Juggie | and now that i need one, i cant find one |
| 22:25.04 | lmadsen | ITiliti: sure |
| 22:25.11 | ITiliti | sweet |
| 22:25.19 | lmadsen | I usually do: Exec(${IF($[...]?foo:bar)}) though |
| 22:25.21 | ITiliti | here it goes. |
| 22:25.37 | lmadsen | I don't like the format of ExecIf() |
| 22:25.43 | ITiliti | exten => s,1,ExecIf($["${KEEPCID}" = "TRUE"])|SipAddHeader(Diversion:<sip:7733096498@sipconnect.chi0.cbeyond.net\;user=phone>) |
| 22:25.45 | mvanbaak | I usually do: Agi(dialplan.agi) |
| 22:26.11 | ITiliti | here is what I am getting: |
| 22:26.18 | lmadsen | ITiliti: you need to escape some of those characters |
| 22:26.22 | Juggie | i usually do, AGI(agi://mysuperfastagihost/dialplan.agi) |
| 22:26.31 | ITiliti | ExecIf("Local/3124988982@from-internal-33cf,2", "1)|SipAddHeader(Diversion:<sip:7733096498@sipconnect.chi0.cbeyond.net;user=phone>") in new stack |
| 22:26.46 | mvanbaak | Juggie :) |
| 22:27.05 | lmadsen | ITiliti: why are you closing the ExecIf() after the "TRUE") |
| 22:27.07 | ITiliti | something similar to Executing [s@macro-dialout-trunk:15] SIPAddHeader("Local/3124988982@from-internal-678b,2", "Diversion:<sip:7733096498@sipconnect.chi0.cbeyond.net;user=phone>") in new stack |
| 22:27.24 | lmadsen | ITiliti: typo |
| 22:27.26 | drmessano | exten s,269u,IfUser($lameidea = "true")|$Mock |
| 22:27.29 | ITiliti | it is like it is trying to add it as a string, instead of a command. |
| 22:27.41 | lmadsen | ITiliti: you are closing the execif() too early |
| 22:27.58 | ITiliti | exten => s,1,ExecIf($["${KEEPCID}" = "TRUE"]|SipAddHeader(Diversion:<sip:7733096498@sipconnect.chi0.cbeyond.net\;user=phone>)) |
| 22:28.02 | ITiliti | like this then? |
| 22:28.26 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:28.28 | mvanbaak | hhmm |
| 22:28.35 | ITiliti | Then I get |
| 22:28.39 | mvanbaak | doesn't ExecIf use ? and : |
| 22:28.44 | ITiliti | ExecIf("Local/3124988982@from-internal-33cf,2", "1|SipAddHeader(Diversion:<sip:7733096498@sipconnect.chi0.cbeyond.net;user=phone>") in new stack |
| 22:29.28 | lmadsen | Exec(${IF($["${KEEPCID}" = "TRUE"]?SipAddHeader(Diversion:<sip:7733096498@sipconnect.chi0.cbeyond.net\;user=phone>):NoOp())}) |
| 22:29.30 | boblutz | it uses ? but one can omit the : |
| 22:29.38 | lmadsen | That's how I'd do it -- however, you might have to escape some of the chars |
| 22:29.53 | lmadsen | you're making it possibly more complicated than you need though |
| 22:29.56 | boblutz | "A char is really just a small number" - Dennis Ritchie |
| 22:30.13 | lmadsen | GotoIf($["${KEEPCID}" = "TRUE"]?:skip_sipheader) |
| 22:30.17 | lmadsen | SipAddHeader(Diversion:<sip:7733096498@sipconnect.chi0.cbeyond.net\;user=phone>) |
| 22:30.26 | lmadsen | exten => blah,n(skip_sipheader),NoOp() |
| 22:31.18 | ITiliti | wow, you rock.. |
| 22:31.25 | *** join/#asterisk RoyK (n=roy@ti211110a081-5006.bb.online.no) |
| 22:31.42 | lmadsen | of course I do |
| 22:31.50 | lmadsen | everyone here knows it too |
| 22:32.00 | lmadsen | although I'd never skip to my loo |
| 22:32.04 | mvanbaak | ITiliti: lmadsen rox. We all know that |
| 22:32.37 | boblutz | bought his book |
| 22:33.10 | mvanbaak | boblutz: we all did |
| 22:33.18 | mvanbaak | otherwise we would be banned from this channel ;) |
| 22:33.20 | lmadsen | lies! :) |
| 22:33.37 | boblutz | well.. i bought it and then found out it was freely available |
| 22:33.44 | mvanbaak | well, I can cover for someone because I have two |
| 22:34.05 | lmadsen | I have 10... now try and catch up! |
| 22:34.06 | mvanbaak | boblutz: it being freely available does not remove the need to fund the author |
| 22:34.12 | mvanbaak | lol lmadsen |
| 22:34.26 | lmadsen | if we don't get sales, then the next version can't be free |
| 22:34.40 | boblutz | the irony |
| 22:34.41 | lmadsen | it's free because we've previously gotten enough sales |
| 22:34.41 | mvanbaak | that's what I mean |
| 22:34.56 | lmadsen | and who the hell wants to only read the online version anyways :) |
| 22:35.01 | mvanbaak | same reason goes for a lot of free stuff |
| 22:35.08 | boblutz | lmadsen: having "author" on your resume must look tits, no? |
| 22:35.16 | mvanbaak | I buy the OpenBSD cd set every 6 months |
| 22:35.34 | mvanbaak | and yes, I can download it for free, but I want to support the project so I buy them |
| 22:35.36 | lmadsen | boblutz: yes, the chicks dig it |
| 22:35.39 | boblutz | LOL |
| 22:35.43 | mvanbaak | makes a nice collection on my shelf |
| 22:36.04 | mvanbaak | lmadsen: you are going to share those for free as well ? |
| 22:36.06 | mvanbaak | hides |
| 22:36.23 | lmadsen | mvanbaak: I don't share my girls, sorry :) |
| 22:36.29 | lmadsen | well... at least not with guys... :) |
| 22:36.35 | mvanbaak | hahahahahahaha |
| 22:36.45 | mvanbaak | correct answer there :) |
| 22:36.52 | lmadsen | yep, I'm smart :) |
| 22:36.58 | lmadsen | passes teh dutchy upon da left |
| 22:37.07 | lmadsen | who said that?! |
| 22:37.08 | mvanbaak | ;) |
| 22:38.39 | Juggie | the secret is lmadsen does not have any girls :) |
| 22:39.05 | lmadsen | haha.. lately I've been having some pretty amazing luck |
| 22:39.17 | boblutz | pwn |
| 22:39.18 | lmadsen | one is coming over tonight actually |
| 22:39.34 | lmadsen | and I'm going on a date with the other one tomorrow |
| 22:39.53 | Juggie | haha they would be happy to know they are named 'one' & 'other one' i bet :) |
| 22:40.18 | mvanbaak | hhmm, I think the next version of the book will have a whole character on the 'anti-ex-girlfriend feature' |
| 22:40.20 | lmadsen | well, I'm not gonna tell you names :) |
| 22:40.27 | lmadsen | someone on here might start stalking or something |
| 22:40.35 | boblutz | and the fact you got 21 y/o boblutz geekin on them in irc |
| 22:40.40 | boblutz | "omg girlz!" |
| 22:40.45 | lmadsen | lol, exactly |
| 22:40.47 | boblutz | "zomg bewbz!" |
| 22:41.04 | mvanbaak | *droooooooooooooooooooooool* |
| 22:41.22 | boblutz | lol |
| 22:43.15 | drmessano | zomg a heartbeat |
| 22:43.39 | mvanbaak | *hump* *hump* |
| 22:43.56 | *** part/#asterisk doolph (n=doolph@200.75.192.45) |
| 22:44.42 | lmadsen | lol |
| 22:45.36 | *** join/#asterisk C4colo (n=DJpyro@67.41.154.214) |
| 22:45.53 | C4colo | I may have found a bug, anyone care to try and reproduce it? |
| 22:46.03 | boblutz | C4colo: explain |
| 22:46.06 | Juggie | if your confident, post it to the tracker. |
| 22:46.13 | C4colo | or tell me that it's not a bug and I'm just stupid |
| 22:46.46 | boblutz | C4colo: Does it have to do with a tdm402b making phantom calls? |
| 22:46.54 | C4colo | ok, I have an AGI script that creates a call file as part of what it does... |
| 22:47.02 | C4colo | no |
| 22:47.12 | Juggie | C4colo, thats silly, but continue |
| 22:47.24 | C4colo | if I structure the call file so it calls a SIP URI first, and then connects it to a local extension I get one-way audio |
| 22:47.31 | Juggie | unless you are scheduleing calls in the future then it would make sense. |
| 22:47.42 | C4colo | (no other one-way audio issues with my system) |
| 22:48.14 | Juggie | try a rtp debug to see if rtp is going both ways |
| 22:48.15 | C4colo | if I call the local extension first, and then call the external by way of calling a custom extension second, it works fine but uses 5 channels |
| 22:49.05 | mvanbaak | wow |
| 22:49.07 | C4colo | well, three for that plus two for the other call the AGI script makes |
| 22:49.14 | [TK]D-Fender | C4colo, do "canreinvite=no" under [general] and each other entry in sip.conf |
| 22:49.26 | ITiliti | okay. I screwed up.. |
| 22:49.38 | C4colo | I have TtWw or whatever, it won't reinvite regardless of the canreinvite |
| 22:49.39 | ITiliti | That code you gave me will only work in asterisk 1.4, right? |
| 22:50.33 | C4colo | at least that's what I picked up from voip-info site, if you have t or T then it won't release the rtp audio stream |
| 22:50.47 | ITiliti | lmadsen: that will only work on asterisk 1.4 |
| 22:50.54 | C4colo | I"ll do an RTP debug |
| 22:51.02 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 22:51.54 | drmessano | People still use 1.2? Isn't 1.8 out yet? |
| 22:52.00 | C4colo | hah |
| 22:52.24 | ITiliti | I am ussing it on one client because we are using gnudialer on it.. |
| 22:52.39 | C4colo | there are probably people using 0.1.13 still on some production server somewhere |
| 22:53.03 | C4colo | don't fix it if it ain't broken, yet |
| 22:53.17 | *** join/#asterisk mattzerah (n=matt@pool2-134.dyn.winshop.com.au) |
| 22:53.41 | drmessano | My three favorite words |
| 22:53.46 | drmessano | "It still works!" |
| 22:54.09 | C4colo | haha |
| 22:54.21 | ITiliti | will this work on asterisk 1.2? ;[macro-dialout-trunk-predial-hook] |
| 22:54.21 | ITiliti | ;exten => s,1,Exec(${IF($["${KEEPCID}" = "TRUE"]?SipAddHeader(Diversion:<sip:8156763062@sipconnect.chi0.cbeyond.net\;user=phone>):NoOp())}) |
| 22:54.29 | drmessano | Long live Redhat 6.2 |
| 22:54.50 | C4colo | and Juggie, the reason I make the call file happen is that I am connecting a different user into a meetme conference and then putting the active caller into that same conference |
| 22:56.12 | C4colo | and I'm getting no output at verbose 10 for the rtp debug, does that only write to the log and not the console? |
| 22:56.51 | Juggie | no it writes to the console |
| 22:57.11 | Juggie | C4colo, you should use the manager interface' |
| 22:57.27 | Juggie | read up on Action: Originate |
| 22:57.35 | C4colo | and 'tail -5000 /var/log/asterisk/full | grep rtp' returns 0 lines |
| 22:57.42 | Juggie | then something is amiss |
| 22:57.50 | boblutz | damn you broken tdm410! |
| 22:58.04 | Juggie | set canreinvite=no globally in your sip.conf and try again |
| 22:58.10 | C4colo | ok |
| 22:58.41 | C4colo | would it even reinvite if I'm connecting the user to a meetme conference? |
| 22:59.02 | C4colo | it seems, because of the nature of the way meetme works, reinvite would not even be a possibilty |
| 22:59.12 | Juggie | it should not be, but try anyway |
| 22:59.17 | C4colo | ok, I'll try |
| 22:59.26 | Juggie | as not seeing any rtp output from rtp debug is sketchy |
| 22:59.39 | Juggie | did you go into the conference and speak? |
| 23:01.12 | drmessano | boblutz: I told you a tdm410p is going to cost more than $11.30 on eBay |
| 23:01.14 | C4colo | oh, nevermind, iax2 doesn't use rtp, I just made a quick call and just realized that it is an IAX2 extension |
| 23:01.27 | C4colo | let me test with a sip extension |
| 23:02.24 | Juggie | i have to go but keep playing with it and see if you can nail down the broken scenario |
| 23:03.01 | drmessano | YAY - NEW RELEASES |
| 23:03.33 | lmadsen | ITiliti: show function IF |
| 23:03.36 | lmadsen | does that show up? |
| 23:03.43 | *** join/#asterisk Katty (n=Angela@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 23:03.44 | drmessano | is stalking russellb on twitter |
| 23:03.48 | lmadsen | ITiliti: like I said... using the GotoIf() method is going to be easier to use |
| 23:03.54 | C4colo | DAMN, that's a lot fo debug output |
| 23:04.30 | russellb | drmessano: yay |
| 23:04.32 | Juggie | just make sure its going both ways... |
| 23:04.34 | drmessano | lol |
| 23:05.05 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.18.1, 1.2.27 (2008/03/18) 1.4.19-rc4, 1.6.0-beta7 (2008/03/28), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies. |
| 23:05.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:05.35 | *** join/#asterisk also (n=also@76.10.180.9) |
| 23:05.41 | C4colo | interesting note, I have rtp.conf limit the ports to 10001 - 10100 and it is using 16xxx and 17xxx for the ports |
| 23:05.56 | drmessano | outbound |
| 23:05.59 | *** part/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 23:06.09 | drmessano | rtp.conf is the asterisk listeners |
| 23:06.24 | jameswf-home | making linux telephony exciting http://www.youtube.com/watch?v=Bc1uUN4Nt8I |
| 23:07.06 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 23:07.06 | *** join/#asterisk ITiliti (n=ITiliti@mail.itility.us) [NETSPLIT VICTIM] |
| 23:07.06 | C4colo | oh |
| 23:07.06 | drmessano | 1 hour long? |
| 23:07.06 | Juggie | russ, menuselect.makeopts: menuselect/menuselect menuselect-tree makeopts doesnt work for me, i can still run make menuselect before ./configure |
| 23:07.06 | drmessano | ZzZzZzZzZzZzZzZz |
| 23:07.09 | jameswf-home | you only need like the first 2 minutes |
| 23:07.25 | Juggie | confirmed with lmadsen too. |
| 23:07.41 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 23:07.41 | *** join/#asterisk ITiliti (n=ITiliti@mail.itility.us) [NETSPLIT VICTIM] |
| 23:07.55 | C4colo | ok, let me do some fiddling and I'll let you know my results |
| 23:08.03 | also | My ITSP les.net just implemented free peer to peer phone calls I just phone my home telephone using Ekiga softphone!!!! |
| 23:08.47 | also | Bell Canada never gave me extra stuff for free |
| 23:09.19 | Juggie | local calls are free :) |
| 23:09.27 | Yourname` | Hi. I had exten => 12349,n,GotoIfTime(10:30-23:59|mon-sat|||?sandiego,100,1) -> But now I dont have sandiego context, but I'd like to do a dial/retry dial. How can I do so? |
| 23:09.43 | drmessano | wow jameswf-home |
| 23:10.00 | drmessano | That was.. riv.... ZzZzZzZz |
| 23:11.48 | *** join/#asterisk Katty (n=The@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 23:13.25 | jameswf-home | he said digium sangoma and openvox... bastard |
| 23:13.38 | jameswf-home | he says dont use pstn its overpriced and breaks |
| 23:13.45 | Qwell | who's "he"? |
| 23:13.55 | Katty | jbot: he? |
| 23:13.56 | jbot | rumour has it, he is not |
| 23:13.58 | jameswf-home | some german |
| 23:14.03 | Yourname` | Hello hello hello. I had exten => 12349,n,GotoIfTime(10:30-23:59|mon-sat|||?sandiego,100,1) -> But now I dont have sandiego context, but I'd like to do a dial/retry dial. How can I do so? |
| 23:14.05 | Katty | he is not some german? |
| 23:14.07 | Katty | *hee* |
| 23:14.22 | drmessano | Larz Snorzovich who presented the aforementioned video |
| 23:14.23 | Katty | jbot: recipes |
| 23:14.24 | jbot | i hope you have some good ones ;), or but if you don't, go here for some good ones: http://astrecipes.net/ |
| 23:14.24 | C4colo | ok Juggie, this call has one way audio, I am receiving the audio from the remote pbx, but the audio is not being received on the far end |
| 23:14.47 | Katty | those aren't real recipes |
| 23:14.50 | jameswf-home | paul kroschnikoff |
| 23:14.51 | Katty | you silly bot |
| 23:15.40 | jameswf-home | qwell http://www.linuxsymposium.org/2007/view_bio.php?id=1139 |
| 23:15.59 | C4colo | this sample rtp output seems to indicate that audio is being sent both directions |
| 23:16.00 | C4colo | Sent RTP packet to x.x.x.x:18492 (type 03, seq 007805, ts 165296, len 000033) |
| 23:16.00 | C4colo | Got RTP packet from x.x.x.x:58105 (type 03, seq 054095, ts 165664, len 000033) |
| 23:17.28 | *** join/#asterisk RoyK (n=roy@ti211110a081-5006.bb.online.no) |
| 23:18.05 | C4colo | the odd thing is that if I call that sip uri directly, or via a custom extension it works fine, but when I use a call file it fails to have two-directional audio |
| 23:18.13 | jameswf-home | found jared http://www.youtube.com/watch?v=hjFyhmRM6lg |
| 23:19.56 | *** join/#asterisk weazahl (n=jeremy@12.53.40.34) |
| 23:20.03 | C4colo | testing on another server |
| 23:21.39 | lmadsen | hey! I got mentioned! :) |
| 23:22.33 | tzanger | lmadsen: yeah but how many swear words were beside the mention |
| 23:22.36 | ITiliti | hey lmadsen, |
| 23:22.42 | lmadsen | tzanger: surprisingly none |
| 23:22.48 | ITiliti | will that line you helped me ith work on 1.2? |
| 23:22.56 | lmadsen | usually my name is prefixed with "that fuckin" |
| 23:23.06 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 23:23.07 | tzanger | hahaha |
| 23:23.15 | lmadsen | ITiliti: the GotoIf() method will -- I don't know about 1.2 because I haven't used it in almost 2 years |
| 23:23.37 | ITiliti | I know, I am just running gnudioaler on it and it does ike 1.4 too much... |
| 23:24.02 | ITiliti | you dontr have it in your histoory do you? I had to reconnect, and I lost it.. |
| 23:24.09 | ITiliti | the three lines you gave me... |
| 23:29.01 | lmadsen | exten => blah,n,GotoIf($["${KEEPIT}" = "TRUE"]?:skip_sipheader) |
| 23:29.06 | lmadsen | (sipheader line here) |
| 23:29.16 | lmadsen | exten => blah,n(skip_sipheader),NoOp() |
| 23:29.55 | Juggie | lmadsen, do you live close to the intercontinental on front which i guess is pretty much part of the MTCC? |
| 23:30.50 | lmadsen | Juggie: I'm a 20-25 min walk |
| 23:30.59 | lmadsen | or you can take the streetcar to Union |
| 23:31.04 | lmadsen | which is a 5 min walk away |
| 23:31.30 | Juggie | its 20-25min from Union Station -> the intercontinental? |
| 23:31.32 | ITiliti | thank you sooooo much.. I have been pulling my hair out with that since earlier.. |
| 23:32.01 | mvanbaak | bbl |
| 23:32.05 | Juggie | i dont remember it being that far, and union is like next to the airport barge thingy so far as i recall? |
| 23:32.07 | Juggie | or is it a bit of a walk |
| 23:32.10 | lmadsen | Juggie: http://tinyurl.com/37njzp |
| 23:32.13 | Katty | hai |
| 23:32.15 | Katty | hugs lmadsen |
| 23:32.20 | lmadsen | Juggie: no, it's 5 mins from Union |
| 23:32.25 | lmadsen | 25 min walk from my place |
| 23:32.29 | Juggie | ic |
| 23:32.41 | Juggie | i thought union was like ontop of the airport barge pretty much |
| 23:32.42 | lmadsen | 20-25 min walk from my place to toronto centre |
| 23:32.48 | lmadsen | Juggie: nope... it's downtown |
| 23:32.57 | joejaxx | grrr |
| 23:32.59 | lmadsen | the airport is south of bathurst |
| 23:33.00 | joejaxx | blasted ciscos |
| 23:33.07 | lmadsen | hugs Katty |
| 23:33.11 | Juggie | ah ic |
| 23:33.11 | lmadsen | evening girly |
| 23:33.14 | Juggie | well the taxi will be on me :) |
| 23:33.23 | Juggie | er, goverment. |
| 23:33.31 | lmadsen | scroll the map down and you can see how close I am to the airport |
| 23:34.03 | lmadsen | I'm only 2km from the Intercontinental :) |
| 23:34.05 | Juggie | ya i see it, i guess i was remembering incorrectally i had this vision of the airport being to the left of union station if you were facint it |
| 23:34.16 | Juggie | 2km is taxi worthy :) |
| 23:34.16 | lmadsen | nope, to the right :) |
| 23:34.33 | lmadsen | $8-$10 taxi ride from my place to Union |
| 23:34.50 | lmadsen | depends if you tip or not I suppose |
| 23:35.02 | Juggie | i'm not tiping, the feds are :) |
| 23:35.29 | *** join/#asterisk Budah (n=IceChat7@201.3.255.174) |
| 23:35.30 | lmadsen | heh |
| 23:35.34 | lmadsen | so you mean I am then |
| 23:35.48 | Juggie | supposedly, the taxi driver is supposed to write the receipt for the fare ammount |
| 23:35.51 | Juggie | and the tip is your money |
| 23:35.59 | Budah | hi... can i record all calls in asterisk using a parameter in features.conf? |
| 23:36.02 | Juggie | but as you know most taxi drivers are pretty much crooked |
| 23:36.08 | Juggie | so they all write the fare + tip |
| 23:36.16 | Juggie | and the majority just give you an empty receipt |
| 23:36.41 | lmadsen | Juggie: yep, I usually just get the business card blank |
| 23:36.49 | lmadsen | *sometimes* they tell me the taxi number :) |
| 23:37.01 | Juggie | lmadsen, my hotel room comes w/ complimentary limo drop off downtown |
| 23:37.04 | joejaxx | anyone here ever tried registering a cisco 7960 with asterisk? |
| 23:37.11 | lmadsen | Juggie: from where? |
| 23:37.14 | joejaxx | Juggie: haha nice |
| 23:37.16 | Juggie | from the hotel i guess |
| 23:37.20 | lmadsen | joejaxx: yep.. check thebook |
| 23:37.25 | lmadsen | ~thebook |
| 23:37.25 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 23:37.34 | lmadsen | should be in chapter 3 or 4 I think |
| 23:37.51 | Juggie | it just says 'complimentary limo drop off in the downtown core' |
| 23:38.10 | lmadsen | probably means they drive you somewhere in the downtown core |
| 23:38.22 | lmadsen | if it's really a limo, we might use that to get to a club one night :) |
| 23:38.23 | Juggie | like the airport, or your building :) |
| 23:38.33 | lmadsen | Kristian is crashing at my place |
| 23:38.34 | Juggie | its prob a towncar |
| 23:38.44 | lmadsen | ya |
| 23:39.03 | Juggie | gotta love hotels |
| 23:39.13 | Juggie | the cheapest rate if you just search online for the intercontinental |
| 23:39.15 | Juggie | 273 |
| 23:39.25 | Juggie | goverment rate, 150 with free limo, better room, etc. |
| 23:39.45 | lmadsen | welp, I'm off to the living room to drink my beer and watch baseball before brit gets here |
| 23:39.56 | Juggie | tap her once for me |
| 23:40.02 | lmadsen | will do sir |
| 23:40.07 | lmadsen | hey, you still seeing that chick? |
| 23:40.10 | Juggie | ya |
| 23:40.12 | Juggie | over 10months now |
| 23:40.13 | lmadsen | crazy |
| 23:40.17 | lmadsen | you're married! |
| 23:40.18 | Juggie | indeed |
| 23:40.20 | lmadsen | she didn't move in did she? |
| 23:40.21 | Juggie | and i've done alot of stupid stuff |
| 23:40.25 | lmadsen | a lot* |
| 23:40.29 | Juggie | and she still keeps comming back |
| 23:40.36 | Juggie | ya she moved in. |
| 23:40.42 | lmadsen | yer ridiculous |
| 23:40.57 | Juggie | i cant complain |
| 23:40.59 | lmadsen | you made it past the 8 month window... next one is the 14 month window |
| 23:41.12 | lmadsen | if you make it past that you should be good until at least the 2 year window |
| 23:41.20 | Juggie | heh |
| 23:41.26 | lmadsen | I'm telling you man... it's true |
| 23:41.27 | *** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com) |
| 23:41.28 | Juggie | i have to go meet all her family this summer |
| 23:41.32 | lmadsen | you haven't done that yet? |
| 23:41.33 | Juggie | then she is comming home with me in sept |
| 23:41.37 | lmadsen | and you've been dating for 10 mths? |
| 23:41.43 | Juggie | her family lives in but fuck no where |
| 23:41.49 | Juggie | northern ontario |
| 23:41.52 | lmadsen | I see |
| 23:41.55 | Juggie | i met some of them when she convicated |
| 23:42.03 | Juggie | and her sister lives in ottawa |
| 23:42.04 | lmadsen | convicated? |
| 23:42.06 | lmadsen | I don't know that word |
| 23:42.11 | joejaxx | lmadsen: lol |
| 23:42.17 | Juggie | umm... when you graduate from university :) |
| 23:42.17 | lmadsen | sounds like a cross between a convict and a convent |
| 23:42.19 | joejaxx | lmadsen: i am WAY past that |
| 23:42.22 | lmadsen | neither of which bodes well |
| 23:42.29 | Budah | lmadsen brother.. can you help me about a feature for recording calls? |
| 23:42.31 | joejaxx | lmadsen: i have everything setup already |
| 23:42.31 | lmadsen | joejaxx: 3+ yrs? |
| 23:42.44 | Juggie | lmadsen, i figure we are good, she is too in love to go anywhere |
| 23:42.45 | lmadsen | Budah: it's friday night... I'm off the clock |
| 23:42.54 | joejaxx | lmadsen: it is just not registering with the asterisk server |
| 23:42.55 | lmadsen | Juggie: that's what they all say... just don't get lazy |
| 23:43.01 | Juggie | nah i'm not. |
| 23:43.04 | Budah | lmadsen ow.. i'm working a lot |
| 23:43.17 | lmadsen | sorry.. I worked a lot this week and I usually get paid to do consulting... |
| 23:43.25 | lmadsen | (and no, I don't want your money... I just want my beer) |
| 23:43.34 | Juggie | but he is really a smut peddler |
| 23:43.48 | joejaxx | lmadsen: it is ok |
| 23:43.50 | Juggie | home made movies in fact. |
| 23:43.52 | Budah | lmadsen owwww.. don't do this.. eheh i'm freak for a beer |
| 23:43.58 | Juggie | brit could be the next big star :) |
| 23:44.00 | lmadsen | and to chat with my good buddy Juggie :) |
| 23:44.04 | joejaxx | i will just keep looking around |
| 23:44.06 | lmadsen | Juggie: lol |
| 23:44.10 | lmadsen | Juggie: I already got one of those |
| 23:44.24 | Budah | lmadsen i understand.. ;) it's ok |
| 23:44.24 | Juggie | well i am off, i just dropped into work to get my confirmation email for it360 and pick up something |
| 23:44.28 | Juggie | i have been off sick since wed |
| 23:44.31 | Juggie | i think i have the plague |
| 23:44.40 | lmadsen | gross... don't be sick when you're here |
| 23:44.45 | lmadsen | I refuse to hang out with sick people |
| 23:44.51 | lmadsen | I avoid them like the plague :) |
| 23:44.55 | Juggie | nah, i was lying in bed wed evening coughing (makes your blood pressure go messy) |
| 23:44.58 | Juggie | then i got up real fast |
| 23:45.02 | Juggie | also makes your blood pressure go messy |
| 23:45.06 | lmadsen | indeed |
| 23:45.07 | lmadsen | I've done that |
| 23:45.08 | Juggie | and i made it about 10 steps and faited, first time ever |
| 23:45.14 | Juggie | right onto the bathroom floor |
| 23:45.37 | lmadsen | crazy.. I've never fainted.. just passed out when I went over the handle bars on my bike and smashed my face into the pavement with no helmet on |
| 23:45.50 | Juggie | i looked it up its common so nothign to worry about |
| 23:45.53 | lmadsen | Juggie: ya, of course it wasn't onto the carpet |
| 23:46.08 | lmadsen | ya, I've almost passed out lots of times doing what you just described, but never actually passed out |
| 23:46.11 | Juggie | but i had not eaten barely anything, heavy coughing messes up your blood pressure and then getting up fast does too |
| 23:46.15 | lmadsen | i don't faint/pass out easily |
| 23:46.21 | Juggie | me either i guess |
| 23:46.25 | Juggie | but that was a first |
| 23:46.31 | lmadsen | yep, you just drained all the blood from your head really fast |
| 23:46.33 | Juggie | drove my chin into the floor |
| 23:46.36 | lmadsen | ouch |
| 23:46.44 | Juggie | ya, tile, its a little red :) |
| 23:46.48 | lmadsen | anyways, go home and drink lots of fluids... I'm off to watch the Jays play baseball |
| 23:46.56 | Juggie | ya i did, been being careful since. |
| 23:46.57 | Juggie | later. |
| 23:47.02 | lmadsen | peas out homie |
| 23:48.27 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 23:48.30 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 23:49.50 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 23:51.50 | Budah | can anyone help me with a record calls in asterisk? |
| 23:52.49 | ZPertee | Budah, how are you trying to do it |
| 23:53.04 | Budah | ZPertee i need to record all calls |
| 23:53.44 | Budah | ZPertee at the moment.. i record with one touch.. *1 |
| 23:55.06 | ZPertee | Budah, what is your setup? pots, sip, iax? |
| 23:55.21 | Budah | ZPertee SIP |
| 23:57.03 | ZPertee | Budah, not sure if I can help you or not. I'm trying to figure it out myself :-) |
| 23:57.29 | Budah | ZPertee okz |
| 23:58.25 | lmadsen | Budah: I think you want to use MixMonitor() |
| 23:59.35 | Budah | lmadsen hmm.. i try to do this.. thank |
