| 00:00.39 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 00:03.39 | *** join/#asterisk Xano_ (n=Xano_@82.73.250.109) |
| 00:03.44 | *** part/#asterisk Xano_ (n=Xano_@82.73.250.109) |
| 00:07.44 | ChannelZ | Note to self: update graphics drivers with windows closed |
| 00:08.10 | Chainsaw | Or it'll do it for you? |
| 00:08.27 | ChannelZ | well it shoves them all on one screen and resizes everything |
| 00:08.36 | Chainsaw | Oh yuck :( |
| 00:08.43 | ChannelZ | but I'm kind of impressed that I could do it 'live' without even rebooting |
| 00:12.33 | *** part/#asterisk nny (n=scott@64.203.239.83) |
| 00:16.35 | *** join/#asterisk TJNII (n=TJNII@207.189.199.62) |
| 00:26.04 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
| 00:27.20 | ChannelZ | So there seems to be a lot of VoIP equipment companies whose websites all look the same. Coincidence? |
| 00:33.24 | *** join/#asterisk ffmog (n=ffm@dslb-088-067-244-106.pools.arcor-ip.net) |
| 00:37.44 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 00:42.23 | *** join/#asterisk jermey_g (n=root@c-92a1e755.021-183-73746f3.cust.bredbandsbolaget.se) |
| 00:42.32 | jermey_g | hi |
| 00:44.21 | *** join/#asterisk n0cturnal (n=n0c@203.161.119.242.static.amnet.net.au) |
| 00:44.33 | jermey_g | Conferencing feature, a participant wants his audio to be reached not to all but a subset of participants. |
| 00:45.01 | jermey_g | Does * support this? |
| 00:46.21 | n0cturnal | how does asterisk decide whether to do a Native Bridge, Packet2Packet, or (hmm, what's the other one called?)? |
| 00:47.05 | *** join/#asterisk the_limit (n=the_limi@75-150-44-61-Oregon.hfc.comcastbusiness.net) |
| 00:48.20 | jermey_g | n0cturnal: which bridge you are referring to, the one used inside code |
| 00:48.31 | jermey_g | n0cturnal: or conferencing bridge |
| 00:49.16 | n0cturnal | jermey_g: when connecting calls together... eg "Packet2Packet bridging SIP/101-00000022 and SIP/vegastream-00000023" vs "Native bridging SIP/101-0000001e and SIP/vegastream-0000001f" |
| 00:51.55 | [TK]D-Fender | jermey_g: No |
| 00:52.04 | jermey_g | n0cturnal: are you using bridge () application |
| 00:52.14 | n0cturnal | jermey_g: no, just Dial(() |
| 00:52.21 | n0cturnal | woops |
| 00:53.21 | jermey_g | n0cturnal: i am not very updated, but i think a 8 months back, this bridge defaulted to packet2packet and could be changed except in code |
| 00:53.26 | *** join/#asterisk MaliutaLap (n=biteme@96.53.148.53) |
| 00:53.37 | jermey_g | could be changed only in code i meant |
| 00:53.52 | n0cturnal | I'm using a vegastream box to terminate my office ISDN BRI's. dial plan in the vega is same no matter what number I dial, yet depending on the number I dial, it does the bridge differently.. one works, the other fails. i'm not sure where the issue lies |
| 00:54.19 | jermey_g | n0cturnal: which version ? 1.4 |
| 00:54.21 | ChannelZ | curses the hiccups |
| 00:54.33 | n0cturnal | 1.6 |
| 00:54.56 | n0cturnal | 1.6.1.10 |
| 00:56.24 | jermey_g | n0cturnal: your dialpan and sip.conf |
| 00:56.41 | n0cturnal | 2 secs will pastebin |
| 00:56.57 | jermey_g | do you pastebin |
| 00:57.10 | jermey_g | oh ok |
| 01:03.39 | n0cturnal | jermey_g: sip.conf: http://pastie.org/private/obrwek7shluexiiq5cfaka extensions.ael: http://pastie.org/private/9jpb3ggkrqku5da7r8a |
| 01:05.08 | n0cturnal | i honestly dont know if the issue lies in my vegastream box or in asterisk.. i can't make any sense of any of this :( |
| 01:10.59 | *** join/#asterisk teknomega (n=tekalpha@c-76-117-79-94.hsd1.pa.comcast.net) |
| 01:11.14 | teknomega | hey does Asterisk still transcode all calls even if they are the same codec on both ends ? |
| 01:12.47 | ChannelZ | it shouldn't assuming it's in the media path at all |
| 01:12.54 | manxpower | teknomega: not generally |
| 01:13.27 | Katty | is excited |
| 01:13.31 | Katty | boingboingboing |
| 01:13.42 | n0cturnal | Katty: fix my * then :P |
| 01:13.48 | Katty | ^_- |
| 01:13.51 | Katty | wait, what? |
| 01:13.57 | Katty | why would i do that? |
| 01:14.07 | n0cturnal | Katty: Please? |
| 01:14.12 | n0cturnal | Katty: cause i asked nice? :P |
| 01:14.29 | Katty | well what's wrong with it then |
| 01:16.32 | teknomega | ChannelZ, did something change with Asterisk ? |
| 01:16.45 | teknomega | ChannelZ, like does it allow for phones to directly communicate with eachother ? |
| 01:16.49 | n0cturnal | my wonderful (or maybe not so) vegastream box is having hissy fits when talking to *.. i dial some numbers and it works, others connect then immediately die, others connect, ring and then die when answered.. somehow asterisk is changing its mind on how it "Bridges" the calls when connecting them, though I dont think that's the only problem.. but I can't be sure :( |
| 01:17.03 | teknomega | ChannelZ, also allow for proxy of RTP to an ITSP ? |
| 01:17.35 | Katty | what's this 'vegastream box' thingy? |
| 01:17.54 | TJNII | n0cturnal: canreinvite=no |
| 01:18.01 | n0cturnal | sorry.. it's an ISDN BRI <-> SIP termination box |
| 01:18.11 | teknomega | what about g722 between Polycom phones... is that auto negotiated |
| 01:18.48 | Katty | what does the CLI say when you call a sip DID that fails on answering? |
| 01:19.22 | [TK]D-Fender | teknomega: A codec is a codec is a codec. Negotiation is the same |
| 01:20.17 | n0cturnal | rtp.c: -- Packet2Packet bridging SIP/101-00000024 and SIP/vegastream-00000025 |
| 01:20.17 | n0cturnal | pbx.c: == Spawn extension (telstra-dial, s, 4) exited non-zero on 'SIP/101-00000024' |
| 01:21.06 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 01:21.13 | Katty | you might want to turn debugging on then, since that info isn't too helpful. |
| 01:21.32 | n0cturnal | heh.. right.. |
| 01:22.20 | Katty | what? |
| 01:22.49 | Katty | you wanna find your problem or not? |
| 01:24.31 | *** join/#asterisk coppice (n=chatzill@202.64.176.25) |
| 01:24.56 | teknomega | so... when talking to an ITSP and asteirsk is in the middle of the call... does the call media get proxied through asterisk or is it transcoded ? |
| 01:25.54 | n0cturnal | Katty: sorry.. i meant I should have done that before.. i meant right as in "yeah, i should have done that already.. my bad" |
| 01:25.59 | [TK]D-Fender | teknomega: Proxied & transcoded a separate concepts and not tied to each other |
| 01:26.09 | n0cturnal | it seems the vegastream box is sending a hangup... |
| 01:26.28 | [TK]D-Fender | teknomega: If I put jam on my toast, is peanut butter supposed to majically appear because I might be making a sandwich>? |
| 01:27.18 | teknomega | dude |
| 01:27.22 | teknomega | you know what i am asking |
| 01:27.31 | teknomega | SipX... does things a different way than asterisk |
| 01:27.32 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:27.40 | teknomega | i am just wondering how asterisk handles a call that goes to an ITSP |
| 01:27.43 | teknomega | thats my question |
| 01:28.15 | teknomega | if g711u is on both ends of the call... does the asterisk box still have to transcode or work on the media stream? |
| 01:28.18 | etfonhomey | teknomega, * handles a call going to an ITSP no differently than a call between phones as long as they are all using the same codec. |
| 01:28.19 | [TK]D-Fender | teknomega: Also stop making the fact that one leg goes to an ITSP as being something special,. That is also not the case. |
| 01:28.42 | teknomega | does media go directly to the phone ? |
| 01:28.44 | [TK]D-Fender | teknomega: If its the same codec on 2 ends, then what is there to transcode? |
| 01:28.55 | teknomega | well it is a B2BUA |
| 01:29.00 | [TK]D-Fender | transcoding != media PATH |
| 01:29.11 | teknomega | i understand that |
| 01:29.17 | [TK]D-Fender | teknomega: You don't seem to. |
| 01:29.22 | teknomega | no i do |
| 01:29.25 | teknomega | thats why i asked if it did |
| 01:29.29 | TJNII | void *transcode_g711u_to_g711u(void *data) { return data; } <- AMAZING! |
| 01:29.57 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:30.27 | [TK]D-Fender | teknomega: * stays in the media path if it has a REASON to. |
| 01:30.38 | teknomega | ok |
| 01:30.51 | teknomega | does it do anything to the media when it does stay in its path ? |
| 01:30.56 | teknomega | or does it just act as a proxy |
| 01:31.06 | Katty | n0cturnal: :< |
| 01:31.28 | teknomega | i am just trying to figure out why i have call quality issues when using asterisk |
| 01:31.30 | n0cturnal | Katty: may I pm you a link to the debug? i can't work this stupid thing out.. |
| 01:31.33 | teknomega | and i don't with SipX |
| 01:31.39 | [TK]D-Fender | teknomega: It does what it has to. |
| 01:31.41 | etfonhomey | teknomega, it has an RTP media path between * and local phone. And a separate media path between * and ITSP. |
| 01:31.50 | Katty | n0cturnal: just paste the link here... i'm going to be afk for awhile. |
| 01:31.57 | [TK]D-Fender | teknomega: And please stop calling it "proxy"..... |
| 01:32.26 | etfonhomey | teknomega, it just copies the voice payload from one media path to the other if they are the same codec. |
| 01:33.00 | teknomega | ahh |
| 01:33.09 | etfonhomey | teknomega, one advantage of having * stay in the path between phones and ITSP is only having to deal with NAT between * and ITSP rather than each phone and ITSP. |
| 01:33.15 | teknomega | but that still puts latency and CPU load onto the box it is on |
| 01:33.45 | etfonhomey | copy is not as CPU intensive as decode, encode |
| 01:33.48 | teknomega | i understand the benefits.. just trying to figure out call quality issues |
| 01:34.06 | [TK]D-Fender | teknomega: All of this for THAT goal? WRong questions.... |
| 01:34.15 | teknomega | i doubt its QoS |
| 01:34.29 | teknomega | Connection... NAT... ISP... ITSP... phones... setup |
| 01:34.39 | etfonhomey | teknomega, I'm getting into this late. What's your issues with quality? |
| 01:34.55 | teknomega | etfonhomey, very very random issues with call quality |
| 01:35.06 | teknomega | with SipX i have no issues with call quality EVER |
| 01:35.16 | coppice | [TK]D-Fender: you are treating this like its a dumb question, and its not. Some systems do turn everything to linear and back for no good reason. Some turn everything to linear so they can do tone detection or other processing on the signal. It is not obvious that things will pass through transparently. |
| 01:35.17 | teknomega | i am not saying one product is better... i am just at a loss |
| 01:35.20 | etfonhomey | Whether you have canreinvite=no or canreinvite=yes? |
| 01:35.22 | teknomega | i would rather use asterisk |
| 01:35.36 | teknomega | like sometimes the call just starts to get all garbled |
| 01:35.38 | teknomega | for no reason |
| 01:35.52 | teknomega | and it doesn't come back unless the caller or callee makes a new call |
| 01:35.56 | [TK]D-Fender | teknomega: Oh, there's a reason... |
| 01:36.03 | teknomega | i understand |
| 01:36.06 | teknomega | but i can't find it |
| 01:36.15 | [TK]D-Fender | teknomega: Like one end getting hit with other traffic. |
| 01:36.31 | teknomega | [TK]D-Fender, if it doesn't happen on SipX at the same location EVER |
| 01:36.40 | teknomega | its now making me think its asterisk |
| 01:36.45 | [TK]D-Fender | teknomega: And I don't see you looking at the calls in detail. In fact... you've give virtually no details at all. |
| 01:36.54 | teknomega | well i would have to setup asterisk again |
| 01:36.59 | teknomega | i have SipX running right now |
| 01:37.09 | [TK]D-Fender | teknomega: Oh and now you're looking at this posthumously?> |
| 01:37.20 | [TK]D-Fender | teknomega: Soory, I left my Flux Capacitor in the office... |
| 01:37.45 | [TK]D-Fender | teknomega: Well this was a grand waste of time... |
| 01:37.51 | teknomega | negative |
| 01:38.01 | etfonhomey | teknomega, what kind of Internet connection are you working with? |
| 01:38.02 | teknomega | i learned how call paths are copied with RTP streams |
| 01:38.09 | teknomega | and not decoded or encoded |
| 01:38.12 | teknomega | when they are the same codec |
| 01:38.18 | teknomega | media paths |
| 01:38.58 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:39.06 | [TK]D-Fender | teknomega: Yes, but you don't take a physics class to learn if there are water fountains on the floor the class is taught in. |
| 01:39.17 | teknomega | bleh |
| 01:39.29 | teknomega | it takes alot to get info from ppl on IRC |
| 01:39.48 | teknomega | because the question is either asked wrong or what i want to do is stupid |
| 01:39.55 | [TK]D-Fender | teknomega: Either way, there is no way to debug what you've done. This puts you no closer to figuring out what really happened |
| 01:40.05 | etfonhomey | teknomega, what kind of Internet connection are you working with? |
| 01:40.11 | teknomega | Quest T1 |
| 01:40.15 | teknomega | QoS on both Ends |
| 01:40.24 | [TK]D-Fender | teknomega: You have guesses and no system to debug. |
| 01:40.42 | *** join/#asterisk chendy (n=chatzill@119.137.95.140) |
| 01:40.58 | etfonhomey | QoS can mean a ton of different things. In fact, my Cisco QoS exam I'm taking this week defines best-effort as a QoS model along with IntServ and DiffServ. |
| 01:41.13 | teknomega | i understand |
| 01:41.34 | etfonhomey | teknomega, define your QoS. |
| 01:41.44 | coppice | QoS on IP networks is never more than best efforts |
| 01:41.46 | teknomega | yeah setting up inbound QoS on T1's that don't have QoS at the ISP... is more like magic than a science |
| 01:42.15 | teknomega | its DSCP based |
| 01:42.24 | teknomega | on the ISP end for outgoing traffic |
| 01:42.48 | teknomega | and inbound is based on destination IP and Port with origination also |
| 01:43.28 | etfonhomey | teknomega, the only valuable QoS you can do from CE standpoint when going to an ISP that doesn't provide service levels other than Best Effort is to setup traffic shaping on your outbound traffic so that you don't drop voice packets headed to your ITSP when you're uploading a lot of data. |
| 01:43.31 | [TK]D-Fender | teknomega: And did any of your QoS settings change between use of * vs SIPX? |
| 01:44.09 | teknomega | only 1... we had to change port 5060 for the ITSP to 5080 for the ITSP |
| 01:44.35 | teknomega | we have to send SIP signaling from the ITSP on port 5080 |
| 01:44.36 | [TK]D-Fender | teknomega: Sounds like nothing to me. |
| 01:44.39 | [TK]D-Fender | teknomega: Then we can stop wasting time on yet another fruitless direction. |
| 01:44.46 | teknomega | ? |
| 01:44.50 | jermey_g | teknomega: what is DSCP? |
| 01:45.04 | teknomega | tagging packets for QoS |
| 01:45.06 | [TK]D-Fender | teknomega: QoS isn't the issue here, and drilling aimlessly isn't going to get you anywhere |
| 01:45.14 | teknomega | oh i know QoS isn't the issue |
| 01:45.33 | teknomega | its something with either the server, either hardware or software |
| 01:45.37 | etfonhomey | teknomega, who is your ITSP? |
| 01:45.43 | teknomega | Bandwidth.Com |
| 01:45.45 | jermey_g | [TK]D-Fender: why are you so angry all the time..getting mad easily.. |
| 01:46.00 | [TK]D-Fender | jermey_g: Not angry yet..... |
| 01:46.05 | [TK]D-Fender | jermey_g: Gimme a few ;) |
| 01:47.03 | jermey_g | does asterisk support secure rtp and sips |
| 01:47.18 | etfonhomey | teknomega, so you had to tell Bandwidth.com to send SIP signaling to TCP port 5060? |
| 01:47.38 | teknomega | no... UDP 5080... when using SipX |
| 01:47.55 | teknomega | by default its already setup to work with asterisk... UDP 5060 |
| 01:48.08 | etfonhomey | teknomega, what ports does SipX use for RTP? |
| 01:48.15 | teknomega | 30000-31000 |
| 01:48.21 | teknomega | by default |
| 01:48.51 | [TK]D-Fender | etfonhomey: Still doesn't matter |
| 01:50.17 | teknomega | [TK]D-Fender, you never get call quality issues ? |
| 01:50.28 | [TK]D-Fender | teknomega: Generally no. |
| 01:50.34 | teknomega | so thats a yes |
| 01:50.51 | *** join/#asterisk easydone (n=notdone@82-170-179-248.ip.telfort.nl) |
| 01:51.58 | [TK]D-Fender | teknomega: At home if I'm streaming media, etc, sure things can get cut. Its going to happen. Fact of life. BUt I'm also not attempting to compare 2 different things, one of which I no longer have access to and trying to guess what the difference is |
| 01:52.12 | *** part/#asterisk easydone (n=notdone@82-170-179-248.ip.telfort.nl) |
| 01:54.27 | teknomega | [TK]D-Fender, last question.. for me to test this out again... would a dual 3.0ghz xeon work fine for a max or 14 sim. calls |
| 01:54.39 | teknomega | s/or/of |
| 01:54.58 | [TK]D-Fender | teknomega: massive overkill. |
| 01:54.59 | etfonhomey | teknomega, in one location, I've been using * on an ancient IBM Thinkpad (built-in battery backup!) for more than 2 years with an ITSP. I've rarely had call quality issues. |
| 01:55.26 | teknomega | ok i'll give it a go again |
| 01:55.39 | etfonhomey | teknomega, every time I have, it was due to the cable Internet provider's crappy modem needing a reboot. |
| 01:55.40 | [TK]D-Fender | teknomega: Seriously... think about how long Digium has been selling 4-port PRI cards, and think about the PC's used ion those servers way back when... |
| 01:55.53 | teknomega | i mean i setup some really basic dial plans just to see how call quality would be |
| 01:56.11 | teknomega | <--- teknoprep |
| 01:56.15 | teknomega | you know i have been here for awhile |
| 01:56.17 | teknomega | so yeah i know |
| 01:56.25 | etfonhomey | teknomega, I suspect your gateway device. |
| 01:56.36 | etfonhomey | teknomega, what about quality between local phones? |
| 01:56.46 | teknomega | its usually flawless |
| 01:57.03 | teknomega | but i don't know how asterisk deals well with jitter |
| 01:57.33 | etfonhomey | teknomega, jitter is jitter. Getting beyond 30ms of jitter and you're gonna have trouble. |
| 01:57.49 | teknomega | we are usually around 8ms at most |
| 01:58.06 | n0cturnal | Katty: if you're still around; my full call debug; http://pastie.org/private/qirxcijkka2fx4ve5ql4q |
| 01:58.14 | teknomega | its usually within 2ms of jitter |
| 01:59.17 | etfonhomey | teknomega, so calls between internal extensions work flawlessly? What is your gateway/edge device to the Internet? |
| 01:59.32 | teknomega | cisco router i think its a 2650 |
| 01:59.47 | teknomega | i would have to check again.. its a cisco one that came with ordering the T1 |
| 01:59.56 | etfonhomey | teknomega, do you have access to the router config? |
| 02:00.03 | teknomega | uhg |
| 02:00.04 | teknomega | yes |
| 02:00.43 | teknomega | i am actually not worrying anymore... how about this |
| 02:00.49 | teknomega | tomorrow i will setup asterisk |
| 02:00.56 | teknomega | and let you guys know how it went |
| 02:00.56 | etfonhomey | teknomega, if you pastebin the router config (w/o IP's and username/passwords), I can tell you how well the QoS is configured. |
| 02:01.07 | teknomega | i know for a fact its perfect |
| 02:01.12 | jermey_g | n0cturnal: wow, set verbose to 8 and no debug, send |
| 02:01.28 | teknomega | i had myself... CCNA ... and hired a CCIE (NetPrivateer) to check it |
| 02:01.31 | teknomega | also had cisco check it |
| 02:02.38 | etfonhomey | teknomega, well, then it wouldn't hurt to have someone else say it's perfect. |
| 02:02.41 | jermey_g | etfonhomey: which exam you are to take--cisco |
| 02:02.56 | jermey_g | ? |
| 02:03.01 | jermey_g | ccnp |
| 02:03.02 | jermey_g | ccvp |
| 02:03.04 | etfonhomey | jermey_g, QoS this week. |
| 02:03.18 | jermey_g | etfonhomey: which cert you are after |
| 02:03.50 | etfonhomey | Doing CCNP and CCVP in parallel. I've passed BSCI and ONT for CCNP and CVOICE for CCVP. |
| 02:04.37 | etfonhomey | But, * is why I'm in this channel. :) |
| 02:05.27 | teknomega | passing those... you should agree that call handling with * is very different than CCM |
| 02:07.05 | jermey_g | teknomega: ccm u mean ccum |
| 02:07.11 | jermey_g | no cucm |
| 02:07.11 | Katty | returns. |
| 02:07.21 | teknomega | lol unified |
| 02:07.23 | teknomega | everyone wants it |
| 02:07.43 | teknomega | i just setup SipX with Exchange 2k10 and OCS 2k7 R2 |
| 02:07.45 | teknomega | very nice stuff |
| 02:07.51 | teknomega | i love the transcription of voicemail |
| 02:08.15 | jermey_g | i integrated my * box with cucm 7 two months back - it was a breeze - no need to use ncube or anything |
| 02:08.15 | teknomega | ~vm~201@voip.bluecloudconsultants.com |
| 02:08.21 | n0cturnal | wb Katty |
| 02:08.27 | etfonhomey | teknomega, I would have to disagree. SIP is SIP. Calls are handled according to the standard more or less. |
| 02:08.34 | n0cturnal | jermey_g: heh.. too much information? :P |
| 02:08.35 | teknomega | ~~vm~201@voip.bluecloudconsultants.com |
| 02:08.36 | *** join/#asterisk chendy_ (n=chatzill@119.139.170.111) |
| 02:08.51 | jermey_g | n0cturnal: yes |
| 02:09.03 | Katty | this lip gloss is.... |
| 02:09.05 | Katty | sticky |
| 02:09.07 | n0cturnal | so no core debug, still want sip debug? |
| 02:09.20 | Katty | scowls at product. |
| 02:09.35 | Katty | dear Cocoa Fever lip gloss, why must you be so sticky? you make me sad. |
| 02:10.09 | jermey_g | n0cturnal: i ll say, no core or sip debug. just take a tcpdump lan trace and set verbose 8 cli dump (send two files) |
| 02:11.08 | *** join/#asterisk runic (i=4b97cff9@gateway/web/freenode/x-izmnubhynxhxinhn) |
| 02:11.14 | n0cturnal | jermey_g: that would be tcpdump >> file? |
| 02:11.19 | runic | evening |
| 02:11.44 | Katty | howdy. |
| 02:11.47 | jermey_g | n0cturnal: tcpdump -s 0 -w file.cap |
| 02:12.06 | jermey_g | this would include all traffic. |
| 02:12.06 | runic | I'm having a strange issue, unsure if anyone else has experienced the same but was hoping I could throw it out there |
| 02:12.18 | etfonhomey | runic, throw it! |
| 02:12.23 | runic | System() call does not appear to be calling gvoice correctly on my setup |
| 02:13.04 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 02:13.06 | jermey_g | etfonhomey: i think ccnp is easy for a guy who knows IP networking. |
| 02:13.22 | jermey_g | etfonhomey: are you required to work on certain routers and switches? |
| 02:13.23 | runic | I know System() is working because I've tested some shell scripting in the same context to make sure it was happening |
| 02:14.21 | runic | running PBXIAF 1.4, latest scripts/fixes/blah. It used to work fine, but I moved and lost my IPKall number due to inactivity |
| 02:14.24 | *** join/#asterisk etnos (n=etnos@adsl-2-204-26.mia.bellsouth.net) |
| 02:14.39 | runic | so I decided to run the updates, get a sipgate, etc |
| 02:14.49 | runic | running latest pygooglevoice as well |
| 02:15.12 | etfonhomey | jermey_g, the material is easy, I agree. Knowing it pass the questions that are asked on the exam is something else. |
| 02:16.12 | runic | imo, CCNA is easy. CCNP takes a bit more work |
| 02:16.30 | etfonhomey | jermey_g, I work on lots of different equipment while doing consulting. |
| 02:16.35 | jaytee | CCIE is the toughest I've heard |
| 02:16.48 | runic | there is a new cert beyond CCIE as of recent, forget what it is |
| 02:17.51 | teknomega | how many ppl have it? 60 ? |
| 02:18.09 | teknomega | CCISP ? |
| 02:18.18 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 02:19.13 | runic | Anyone have experience with PBXIAF 1.4 using google voice? |
| 02:19.27 | *** join/#asterisk tgrman (n=jcmoore@unaffiliated/tgrman) |
| 02:20.59 | teknomega | Cisco Certified Architect |
| 02:22.33 | jermey_g | runic: invite me to google voice |
| 02:23.17 | runic | jermey_g: thought invites are going out in like 24hrs these days |
| 02:27.44 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 02:28.38 | *** join/#asterisk chendy_ (n=chatzill@119.139.168.158) |
| 02:34.28 | *** join/#asterisk neurosys (n=neurosys@adsl-233-195-130.mia.bellsouth.net) |
| 02:50.57 | *** join/#asterisk timholum1 (n=chatzill@64-91-67-5.stat.centurytel.net) |
| 02:51.25 | timholum1 | hello, dous any one know of a way to set caller id different per extension? |
| 02:53.02 | p3nguin | sure |
| 02:53.04 | teknomega | how would i setup * to accept inbound SIP signaling on port 5080 so i don't have to change it tomorrow |
| 02:53.20 | timholum1 | I have a kind of rare case where I have a call group where I call from Dial( SIP/200&SIP/201&SIP/trunk/number ) The SIP/trunk/number has to have caller id set to my trunks username or it does not work. Is there any way to preserve my username for the SIP/200 and SIP/201? |
| 02:53.25 | teknomega | i owuld like to use both 5060 and 50980 |
| 02:53.34 | jaytee | teknomega, change the bindport in sip.conf and do a sip reload |
| 02:53.34 | p3nguin | exten => 123,1,Set(CALLERID(num)=newnum) |
| 02:53.45 | teknomega | jaytee, can i use both ports ? |
| 02:53.53 | jaytee | teknomega, don't think so |
| 02:54.08 | timholum1 | p3nguin, that will set the caller id for all of the calls :( |
| 02:54.25 | p3nguin | timholum1: You asked to set it per extension. That sets it on extension 123. |
| 02:54.55 | jaytee | timholum, you can define a sip account with callerid in that account's entry in sip.conf |
| 02:55.10 | timholum1 | how to i do that? |
| 02:55.15 | jaytee | callerid="Jabba The Hut" <1234> |
| 02:55.23 | teknomega | lol |
| 02:55.31 | jaytee | for sip account 1234 |
| 02:57.36 | jermey_g | n0cturnal: did you get over with your problems |
| 02:58.16 | timholum1 | unfortunatly that only changes inbound from 1234, I need it to change it when I call user 1234 that it changes the caller id ( Skype for sip requires all outbound calls to have the account id in order to accept the call ) :( |
| 02:59.12 | n0cturnal | jermey_g: heh sorry, i've managed to lock myself out of the box while i was playing with network configs.. gotta drive over there in a while to fix.. |
| 02:59.18 | timholum1 | kind of a hastle, and I would like all inbound calls on my regular pstn line to call 2 of my employee's cell phones |
| 02:59.22 | p3nguin | Like I said, you can set it per extension using the Set() command and CALLERID() function. |
| 03:01.36 | jaytee | timholum, then you'd want to use something like exten=> 1234,1,Set(callerid(all)=${ACCOUNTID} and set the accountid variable in the global section of extensions.conf if you use the same accountid for all calls |
| 03:02.01 | jaytee | and I left out the closing parentheses in the example |
| 03:06.37 | timholum1 | exten => 123,1,Set(CALLERID(num)=newnum) only works if I dial 123, i have s,1,Answer() s,2,Dial( SIP/200&SIP/201&SIP/trunk/number ) and I only want the caller id to change for SIP/trunk/number not for SIP/200 or SIP/201, I do have outbound working on that using SET( CALLERID=.... ) when I dial the trunk directly, but I am guessing what I am trying to do is impossible |
| 03:07.42 | timholum1 | I do know it would work if I did s,2,SET(CALLERID(num)=xxxx) but it would change it for all of the phones |
| 03:08.11 | timholum1 | SIP/200 and SIP/201 would not be able to figure out who is calling :( |
| 03:08.12 | *** join/#asterisk youngmoney (i=3a06133a@gateway/web/freenode/x-vegndnjaozgfrixr) |
| 03:09.51 | Katty | :< |
| 03:09.57 | p3nguin | timholum1: Well, you said "set caller id different per extension" and that is what I told you how to do. |
| 03:10.02 | Katty | i dusted my desk, but my allergies are still going omgwtftwitchtwitch |
| 03:10.42 | youngmoney | Quick question regarding the functionality of Asterisk: I want to be able to take two phone numbers (landlines) and join them in a conference call. I need multiple conference calls going at the one time amongst various numbers (multiple concurrent calls through the one SIP account). Is this possible? :/ |
| 03:10.50 | p3nguin | Does your 's' exten end up dailing those sip devices? |
| 03:11.21 | Katty | youngmoney: if i'm understanding your question properly, the answer is yes. |
| 03:11.33 | p3nguin | timholum1: Does your 's' exten end up dailing those sip devices? |
| 03:11.41 | Katty | youngmoney: you can transfer as many calls into MeetMe as you like, internal sip devicies and external lines/channels. |
| 03:12.14 | Katty | youngmoney: you can have as many meetme 'rooms' as you like. |
| 03:12.29 | youngmoney | Katty: I've never used Asterisk before so just to clarify - "internal sip devicies and external lines/channels." - Does that mean external hardware? Or is it Asterisk software jargon |
| 03:12.33 | timholum1 | they dial the SIP/trunk/number, I have tryed putting in Dial(SIP/200&SIP/201&8xxxxxx) but it tells me that I need to put a technology in :( |
| 03:13.23 | Katty | youngmoney: internal devices, such as sip phones (polycom, etc) and people calling in to cards handling Pots lines, t1s... and of course calls off DID numbers from a sip provider. |
| 03:13.33 | p3nguin | timholum1: Dial(SIP/200&SIP/201&SIP/trunk/number)? |
| 03:13.54 | Katty | youngmoney: you can assign a DID number to dump directly into a meetme conferece, or you can always have a receiptionist transfer to a meetme extension... |
| 03:13.57 | p3nguin | timholum1: I really can't understand what you're trying to do. |
| 03:14.07 | Katty | youngmoney: you have several options to get your callers into conference rooms. |
| 03:14.10 | p3nguin | timholum1: Is this related to your previous question about the caller ID? |
| 03:14.26 | youngmoney | Katty: thanks for the help. I'll download Asterisk and have a play around |
| 03:14.30 | Katty | youngmoney: as far as i know, there isn't a limit to the number of callers or meetme rooms, except for the limitations of your computer. |
| 03:14.47 | Katty | wlell.... bye then. |
| 03:14.52 | Katty | HAVE A NICE EVENING |
| 03:15.06 | p3nguin | You'll probably use up all your bandwidth before reaching the limit of the computer. |
| 03:15.26 | Katty | yeah probably |
| 03:15.47 | p3nguin | Unless it is ancient hardware, such as what I use. :) |
| 03:16.36 | timholum1 | I am sorry I am not very good at explaining myself |
| 03:16.57 | jaytee | twss |
| 03:17.17 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:18.21 | timholum1 | I would like inbound calls on my pstn line to have 2 different caller id's depending on if it is dialing out my sip trunk or if it is going directly to one of my sip extensions |
| 03:18.58 | p3nguin | SIP doesn't have extensions, but extensions can Dial() SIP devices. |
| 03:18.59 | timholum1 | or if there was a way to dial an extension from the Dial() command that could probably work as well |
| 03:19.59 | p3nguin | You can Dial() local extensions: exten 123,1,Dial(Local/321@context) |
| 03:20.18 | timholum1 | ok, I think that will solve my issue, :) Thank you |
| 03:20.27 | Katty | http://photos-f.ak.fbcdn.net/hphotos-ak-snc3/hs151.snc3/17877_643959293627_37617946_36398128_3437893_n.jpg <- i saw this when i went to target tonight. |
| 03:20.47 | Katty | ^- it seems somehow fitting. |
| 03:22.56 | eppigy | harmacy ^_______________^ |
| 03:23.14 | Katty | (= |
| 03:24.01 | Katty | eppigy: would you like some lip gloss? |
| 03:24.06 | Katty | eppigy: i don't think i'm going to use this stuff again |
| 03:24.14 | eppigy | well uh |
| 03:24.18 | *** join/#asterisk [Outcast] (n=anonymou@ipn36372-d82722.cidr.lightship.net) |
| 03:24.19 | eppigy | i only really like it |
| 03:24.21 | Katty | don't be like that |
| 03:24.28 | eppigy | if a girl is applying it ith her lips |
| 03:24.28 | Katty | i saw a guy at target who had better make up than me |
| 03:24.43 | Katty | he even had double wing eye linger. |
| 03:25.18 | dmast | did he have emo hair? |
| 03:25.25 | jaytee | metrosexuals are creepy |
| 03:25.27 | [Outcast] | most likley |
| 03:25.39 | Katty | yep, he had emo hair. |
| 03:25.45 | Katty | side part. very shreddy |
| 03:25.48 | [Outcast] | dmast called it |
| 03:26.02 | Katty | he sure did! |
| 03:26.05 | dmast | :) |
| 03:26.41 | Katty | eppigy: tell me you at least wear lip balm/chapstick |
| 03:26.49 | eppigy | I do not |
| 03:26.51 | Katty | eppigy: lipmedics, carmex... |
| 03:26.55 | Katty | oh dear :< |
| 03:26.56 | eppigy | negative |
| 03:27.02 | Katty | do your lips chap? |
| 03:27.03 | eppigy | I do not kno why i would |
| 03:27.06 | eppigy | negative |
| 03:27.06 | Katty | and crack, and bleed. |
| 03:27.09 | Katty | k |
| 03:27.10 | eppigy | oh o no |
| 03:27.18 | Katty | mine do :< |
| 03:27.20 | Katty | it's /awful/ |
| 03:27.37 | eppigy | :< |
| 03:27.42 | Katty | and then little bits of skin come lose, and i end up biting them off |
| 03:27.44 | Katty | >.< |
| 03:28.38 | dmast | was just compelled to apply chapstick |
| 03:29.17 | Katty | dmast: what kind do you use? |
| 03:29.45 | dmast | Chapstick...Ultra? Does that sound right? |
| 03:29.49 | dmast | And Bert's Bees |
| 03:29.59 | [Outcast] | i hate hotel wifi |
| 03:30.08 | p3nguin | Bert has some nice products. |
| 03:30.19 | dmast | Bert's Bees = luxury chapstick |
| 03:30.19 | Katty | ultra chapstick does have spf 30 i believe... |
| 03:30.21 | Katty | and that's very good. |
| 03:30.34 | [Outcast] | chapstick medicated for me |
| 03:30.35 | Katty | bert's bees...has a beeswax lipbalm. i have some, but i've not tried it yet |
| 03:30.46 | Katty | comes in a little tin. |
| 03:31.18 | dmast | Never tried that... we just have in a stick |
| 03:31.39 | Katty | http://www.comparestoreprices.co.uk/images/bu/burts-bees-burtand39-s-bees-beeswax-lip-balm-tin-8-5g.jpg |
| 03:31.52 | Katty | so far, my favorite is plain ole lipmedics |
| 03:32.58 | Katty | or carmex. |
| 03:33.08 | Katty | carmex is pretty darn good. |
| 03:33.18 | dmast | yup |
| 03:33.18 | *** join/#asterisk chendy (n=chatzill@113.91.38.116) |
| 03:33.30 | dmast | I have to switch over to cold sore cream about twice a winter... that sucks |
| 03:33.51 | p3nguin | Herpacin L |
| 03:33.55 | Katty | really? |
| 03:34.05 | Katty | i've never had a cold sore, but i've heard they're just plain awful. |
| 03:34.28 | dmast | It's pretty much like having a friction blister on your lip |
| 03:34.28 | Katty | http://www.cybelesays.com/my_weblog/images/2008/09/22/bootsbalm_2.jpg <- target had this stuff |
| 03:34.40 | Katty | ouch )= |
| 03:35.43 | Katty | http://modaily-cosmetic.exteen.com/images/photo208.JPG <- better photo. |
| 03:36.00 | Katty | they had another one...olive oil and sage i think, but it smelled attrocious. |
| 03:36.15 | dmast | p3nguin: I've been using Campho-Phenique ... could give Herpacin a try though. |
| 03:36.27 | Katty | how long does it take a cold sore to go away? |
| 03:36.58 | p3nguin | Back 20 years ago, Campho was THE shit to use for everything. I wonder how degraded they have become over the years. |
| 03:37.11 | dmast | katty: 2-3 days if you're diligent with the meds |
| 03:37.19 | Katty | well that's not too long, i guess. |
| 03:37.33 | p3nguin | Every product I know of has evolved into nearly a piece of crap. |
| 03:38.17 | Katty | p3nguin: well... |
| 03:38.28 | Katty | p3nguin: yes, but there are always new products coming out that are good. |
| 03:38.40 | Katty | p3nguin: and you can't expect to get much from a drugstore for a few bucks, really. |
| 03:38.40 | p3nguin | But none as good as the old products once were. |
| 03:39.09 | Katty | that 1.5ml grape lip butter i bought was close to 10 bucks |
| 03:39.23 | Katty | but i'm okay with it, cause i'm trying to find a better lip balm. |
| 03:39.30 | Katty | and i know that 2 bucks isn't going to buy anything good. |
| 03:39.45 | dmast | 10 bucks? it better clean your teeth too |
| 03:40.00 | Katty | no it doesn't clean your teeth :P |
| 03:40.16 | Katty | i /do/ carry the little trial size of scope in my purse tho |
| 03:40.25 | p3nguin | minty fresh |
| 03:41.21 | Katty | cinnamon actually |
| 03:41.44 | dmast | is it pretty decent? |
| 03:41.44 | Katty | "cinnamon ice" the bottle says |
| 03:42.01 | Katty | well.... |
| 03:42.06 | Katty | i like it. |
| 03:42.10 | dmast | has only ever used the classic mintohol flavor |
| 03:42.13 | Katty | but the minty stuff usually makes me gag. |
| 03:42.29 | Katty | mint anything usually makes me gag |
| 03:42.57 | Katty | peppermint candys, mint mojitos... |
| 03:43.10 | dmast | doesn't like peppermint |
| 03:43.34 | Katty | blue bunny has a seasonal peppermint icecream bar out, but i've not tried it. |
| 03:45.31 | Katty | their eggnog ones were good :> |
| 03:45.50 | dmast | Didn't get much nog this year :-\ |
| 03:46.26 | p3nguin | I didn't get any, and I'm not pleased about it. |
| 03:46.56 | p3nguin | I like to pick up the Southern Comfort eggnog from Walmart when they clearance it. |
| 03:47.09 | jaytee | I love the Hagen-Daz Peppermint Bark ice cream that they have during the Holiday season |
| 03:51.55 | Katty | p3nguin: is that premixed? |
| 03:52.09 | Katty | jaytee: i try to avoid that stuff due to the extremely high calorie content >.< |
| 03:52.15 | Katty | jaytee: but their stuff is veryyyyyyyyyyyyyyyyyyyyyyyyyyyyy good. |
| 03:53.03 | p3nguin | katty: It's pretty much like Prarie Farms eggnog or holiday nog, but made by SoCo. |
| 03:53.36 | jaytee | they used to have a flavor called Double Chocolate that had lots of belgian cocoa in it. it was heaven, tasted like chocolate ice cream tasted back when I was a kid and the world was real and genuine instead of artificial and phony. |
| 03:54.09 | Katty | p3nguin: so...it's not alchoholic? |
| 03:54.25 | p3nguin | correct, no alcohol when you buy it. |
| 03:54.33 | Katty | :< |
| 03:54.37 | Katty | boooo |
| 03:56.14 | p3nguin | I had never used Southern Comfort in eggnog until I say Southern Comfort brand eggnog. It's rather delicious. |
| 03:56.22 | p3nguin | s/say/saw/ |
| 03:56.37 | [TK]D-Fender | jaytee: .... and the greatest threat to man was swooping pteradactyls? ;) |
| 03:56.39 | Katty | i like the land o lakes brand |
| 03:57.31 | [TK]D-Fender | jaytee: Oh... and HAPPY NEW YEAR! |
| 03:57.34 | jaytee | [TK]D-Fender, no but America had the largest GDP and owed no one and most countries owed us money. Beef actually tasted good back then |
| 03:57.45 | jaytee | Happy New Year to you too! |
| 04:15.29 | jaytee | nite everyone |
| 04:23.19 | *** join/#asterisk moy (n=moy@74.12.129.52) |
| 04:23.26 | Katty | hi |
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| 05:10.59 | psiforce | hi all |
| 05:12.43 | psiforce | can someone confirm that the digium servers for registering g729 licenses is down? tried running register on 4 different servers and all do not work |
| 05:16.14 | *** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) |
| 05:20.17 | psiforce | can someone confirm that the digium servers for registering g729 licenses is down? tried running register on 4 different servers and all do not work |
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| 05:24.58 | *** join/#asterisk ruben23 (n=RPL@203.87.178.23) |
| 05:29.32 | ruben23 | hi |
| 05:32.37 | ruben23 | i have existing asterisk server on production and it working well, but i got new project which will handle a hosted asterisk server, since my existing local asterisk server is using public IP for wan connection. how would i add up the new setup to my existing network..any suggestion.. |
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| 05:41.02 | *** part/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
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| 05:46.37 | p3nguin | ruben23: You have your phones and * server in one location, but you want to add another * which is in a remote datacenter? |
| 05:46.46 | p3nguin | What will the additional * server do? |
| 05:47.20 | [TK]D-Fender | None of those pieces add up |
| 05:49.29 | ruben23 | p3nguin: actually its another project separet frommy setup but users will be on the same network i have on my local asterisk server, using sofphones to access the hosted asterisk service.. |
| 05:50.23 | lost_sou1 | wouldn't it be easier to just make a specific set of extensions for the other project |
| 05:52.02 | p3nguin | Okay, so you have users who wish to access a remote * server. So what? That's normal. |
| 05:52.18 | lost_sou1 | p3nguin: he's gone now |
| 05:54.08 | lost_sou1 | Have any of you tried the Perelli SIP/GSM/WIFI phone? |
| 05:54.45 | [TK]D-Fender | And his description broken... |
| 05:56.31 | *** join/#asterisk ruben23 (n=RPL@203.87.178.17) |
| 05:57.00 | p3nguin | ruben23: (2352.01) <p3nguin> Okay, so you have users who wish to access a remote * server. So what? That's normal. |
| 05:57.10 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 05:57.21 | ruben23 | <PROTECTED> |
| 05:58.07 | ruben23 | p3nguin: yeah, problem is how do i separte its traffic to my existing local asterisk server.. |
| 05:58.36 | p3nguin | Luckily, computers and phone use IP addresses. |
| 05:58.47 | p3nguin | Each of which must be unique. |
| 05:59.00 | p3nguin | Problem solved. |
| 05:59.33 | ruben23 | p3nguin: i mean do i need a voice gateway for my hosted asterisk server..? |
| 05:59.51 | ruben23 | or just a router..separte to my local asterisk server |
| 06:00.28 | p3nguin | Well, you'll need a router to connect your local network with the internet. |
| 06:00.42 | p3nguin | The internet is where the second * server will reside. |
| 06:01.39 | p3nguin | Use either SIP or IAX2 for communications between your phones and the * server. What's the problem? |
| 06:01.51 | p3nguin | I fail to see an issue with anything at this point. |
| 06:03.02 | [TK]D-Fender | ruben23: Voice gateway? What the hell are you talking about? |
| 06:03.10 | ruben23 | p3nguin:ok i guess my hosted asterisk will be routed to a linux router box then my local asterisk will have it wan ip for it own traffic.. |
| 06:03.10 | [TK]D-Fender | ruben23: Nothing you are saying is making any sense. |
| 06:03.45 | p3nguin | I'm with [tk]d-fender on this, i.e. what the hell are you talking about? |
| 06:04.09 | p3nguin | Are you networking-challenged? |
| 06:04.24 | ruben23 | sorry guys... |
| 06:05.57 | p3nguin | Soft phones talk SIP, * talks SIP. Both are IP-enabled. There is no problem to solve. |
| 06:06.32 | jermey_g | p3nguin: dont be wid [TK]D-Fender , he is actually a bot |
| 06:06.43 | jermey_g | p3nguin: written in Python |
| 06:07.06 | p3nguin | Hmm. |
| 06:07.14 | p3nguin | Is "wid" even a word? |
| 06:07.18 | [TK]D-Fender | is not 3-laws-safe |
| 06:08.58 | *** join/#asterisk joako_ (n=ston3d@opensuse/member/joak0) |
| 06:09.06 | jermey_g | substitutes 3 with e in p3nguin |
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| 06:42.11 | soman | tzafrir_laptop: TSM: Hi, thanks for the support.. I have added the override option in /etc/modprobe.d/dahdi and restarted.. and now is working fine... The file is getting recorded now.. no errors ... thanks a lot for your support |
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| 06:55.56 | n0cturnal | what would cause asterisk to send a BYE when an outgoing call is answered? |
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| 08:37.37 | pentanol | hey anybody alive? |
| 08:39.30 | ChannelZ | not you |
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| 08:43.46 | ChannelZ | wb |
| 08:44.19 | pentanol | ChannelZ hi, did you use or some one else web-meetme interface? |
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| 08:46.31 | ChannelZ | Wasn't me |
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| 09:54.41 | Get_The_Fish | can anyone give me any suggestions or tips on their favorite naming conventions for global and channel variables in their dialplan? |
| 09:56.43 | *** join/#asterisk jabka (n=jabka@DSL212-235-40-207.bb.netvision.net.il) |
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| 09:59.18 | jabka | anyone remember the name of modem emulator (fax) (it just writes the data to file and work like the end of tty is connected to line and on the side there is a fax) |
| 10:01.49 | jabka | just got hint thanks iaxmodem |
| 10:11.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:19.34 | *** join/#asterisk Hexie (n=shane@dsl-244-128-70.telkomadsl.co.za) |
| 10:20.13 | Hexie | Hi, is there anyone here that could help me with some technical (development) information regarding asterisk ? |
| 10:21.17 | gr0mit | Hexie, go on... |
| 10:21.37 | Hexie | i would like to know if there are .net components avaliable for asterisk? |
| 10:21.48 | Hexie | as in a .dll or object based for .net flatform |
| 10:22.06 | gr0mit | asterisk is linux |
| 10:22.17 | Hexie | ie - all the functions and methods are avaliable in a .net library or object |
| 10:22.29 | Hexie | correct and i saw that there is some windows based versions |
| 10:22.29 | gr0mit | no idea |
| 10:22.46 | gr0mit | there is a version for windows but why would you want to do that?! |
| 10:23.52 | Hexie | see i am not looking for the software - i am looking for the controls that the software uses for me to implement in other software (ie - use the methods avaliable for calling etc that are already in place) - no point in re-inventing the wheel |
| 10:23.58 | TommyBotten | Hehe.. There is a windows port, but I know of no native DLLs. There exists an abstraction layer for the AMI though. |
| 10:24.58 | Hexie | would this layer have the functionality of providing methods used (for a .net platform - i.e VB / C# / C++ etc) |
| 10:25.40 | TommyBotten | But ... are you going to run Asterisk on windows in a prod environment |
| 10:25.57 | gr0mit | just dont go there! |
| 10:26.13 | Hexie | i would not need to run Asterisk in any platform if the controld / objects are avaliable in an object based form (.dll etc) |
| 10:26.33 | Hexie | why not "just go there" that is where the software is most powerful. |
| 10:26.36 | tzafrir_laptop | Hexie, there are such interfaces for the AMI (Asterisk manager interface) and such |
| 10:26.36 | TommyBotten | Well, the library I mentioned is on http://sourceforge.net/projects/asterisk-dotnet/ |
| 10:26.49 | TommyBotten | It does AMI and FastAGI |
| 10:27.00 | Hexie | are those UI's |
| 10:27.01 | Hexie | ? |
| 10:27.02 | tzafrir_laptop | Hexie, can you give an example of something you want to do? |
| 10:27.10 | Hexie | "AMI and FastGUI" |
| 10:27.19 | Hexie | ok i will give an example 1 sec |
| 10:27.28 | tzafrir_laptop | Both don't need UI |
| 10:28.07 | Hexie | basic example: create a simple application that implements the use of some web based controls or (even better) a .dll that will allow this software to make calls and have some other functionality |
| 10:28.14 | TommyBotten | Hexie: FastAGI - Asterisk gateway interface - let's your dialplan execute external scripts |
| 10:28.18 | Faustov | I'm getting 2 lines spammed in my CLI: |
| 10:28.19 | Faustov | [Jan 4 11:27:52] -- Remote UNIX connection |
| 10:28.19 | Faustov | [Jan 4 11:27:52] -- Remote UNIX connection disconnected |
| 10:28.29 | Faustov | any idea what could be causing this? |
| 10:28.32 | tzafrir_laptop | Hexie, making calls is trivial with the manager interface |
| 10:28.37 | jabka | Enormus wow - it seems that smartbox aka openRG is using asterisk , i hope we can demand the source code :D |
| 10:28.39 | TommyBotten | Faustov: Probably flash operator panel, or some other GUI |
| 10:29.02 | tzafrir_laptop | (you should not, however, expose this functionality to just any client. If you care about your phone bill) |
| 10:29.17 | Faustov | TommyBotten: no gui here, but i think I got an idea |
| 10:29.43 | tzafrir_laptop | Hexie, there are various such existing programs ("dialers", whatever) |
| 10:30.25 | Hexie | correct - that is exactly what i was / am looking for "dialer" but it seemed that Asterisk had the SIP already added as well as all the components avaliable? |
| 10:31.19 | Hexie | see i would not like to rely on other software providers, id rather pay for some small .dll that does the basics and then add my own functionality (i could write such a .dll but i dont have much experiance with SIP servers) |
| 10:31.53 | jabka | Is there a cummertical Asterisk version ? |
| 10:32.09 | TommyBotten | jabka: Yes. www.asterisk.org ;) |
| 10:32.16 | Hexie | lol |
| 10:32.16 | Faustov | Is there a way I could filter these "Remote UNIX connection" messages out? The thing producing them is needed, however they are flooding the logs |
| 10:32.29 | jabka | bummer i thought it is only GPL :-( |
| 10:33.01 | Chainsaw | Faustov: Using the management interface instead of "asterisk -rx" calls would "cure" it. |
| 10:33.06 | TommyBotten | jabka: It's dual licensed... and the commercial version has some features that the free does not. |
| 10:33.35 | jabka | TommyBotten , thank you |
| 10:34.27 | Faustov | Chainsaw: do you mean that something is polling the info by doing "asterisk -rx" while it should do something else to get the information it needs? |
| 10:34.40 | tzafrir_laptop | TommyBotten, such as? |
| 10:35.48 | TommyBotten | tzafrir_laptop: res_fax_digium for instance |
| 10:36.25 | tzafrir_laptop | TommyBotten, is also available separately |
| 10:37.10 | TommyBotten | tzafrir_laptop: Ok. Where can I find it? I don't see it in the source tree. |
| 10:41.20 | *** join/#asterisk mattbUK (n=mattbrid@92.27.125.154) |
| 10:41.32 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 10:44.02 | mattbUK | Hi I've got 2 identical asterisk boxes if different locations. Both with the same config, one is working fine with inbound IAX the other fails. Failing debug output at: http://pastebin.com/m4413a5ae - anyone got any ideas - I'm totally stuck |
| 10:48.08 | Tim_Toady | some prob with ur iax peer authentication? |
| 10:49.11 | fenrus | well, it says that it acceppts the call |
| 10:50.37 | Tim_Toady | iax.conf on pastebin might help |
| 10:55.11 | mattbUK | @Tim_Today: http://pastebin.com/m54f9bae1 |
| 10:57.52 | mattbUK | Tim_Toady: the strange thing is it works perfectly on a mirrored box on another lan |
| 11:00.14 | mattbUK | This is working debug: http://pastebin.com/m76901f34 |
| 11:03.01 | Tim_Toady | no idea, are you sure its not a dialplan prob? |
| 11:03.34 | mattbUK | the dialplans are identical on both boxes |
| 11:04.14 | Tim_Toady | dialplan show 08455570481@default in both boxes to make sure |
| 11:05.38 | mattbUK | both are: '08455570481' => 1. Answer() [pbx_config] 2. goto(555|1) [pbx_config] |
| 11:06.12 | mattbUK | <PROTECTED> |
| 11:06.13 | mattbUK | <PROTECTED> |
| 11:06.13 | mattbUK | <PROTECTED> |
| 11:15.22 | pentanol | hey, anybodt knows how I can make callback, i.e. when I made meeting room I want meet there someone, i.e. i need call then and when he hangup his phone he would be in this meeting room |
| 11:24.58 | manxpower | remove your AGI test and see if it works |
| 11:27.26 | mattbUK | manxpower: nope exactly the same |
| 11:28.34 | pentanol | any one? |
| 11:30.14 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 11:33.23 | mattbUK | if it do dail 08455570481 from the console it runs through fine |
| 11:33.41 | mattbUK | if I dial from a phone it just fails |
| 11:37.32 | mattbUK | aha! |
| 11:37.37 | mattbUK | Bloody bindaddr |
| 11:37.47 | mattbUK | Machine that isn't working has multiple ips |
| 11:37.56 | ManxPower-work | That's why we usually recommend not setting a bindaddr. |
| 11:38.15 | ManxPower-work | That is not a reason |
| 11:38.41 | ManxPower-work | Without binadddr, Asterisk relies on the OS ROUTING TABLE. |
| 11:38.53 | ManxPower-work | Which is where routing stuff should happen -- in the OS. |
| 11:39.23 | mattbUK | I can check udp is listening on each of the multiple ip's - and that's fine |
| 11:39.35 | pentanol | anybody knows I can meet someone just from phone, avoid web-meetme? |
| 11:39.37 | mattbUK | but without the bindaddr it's failing |
| 11:39.39 | ManxPower-work | UDP does not listen on anything. |
| 11:40.15 | ManxPower-work | Remember SIP uses port 5060/UDP (and by default) ports 10,000/UDP - 20000/UDP. bindaddr does nothing for RTP. (audio) |
| 11:40.15 | mattbUK | Ok well perhaps I'm being thick but the solution was to change the bindaddr to the IP the IAX provider is sending the request too |
| 11:40.23 | mattbUK | I'm using IAX |
| 11:48.01 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 11:48.01 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.0 (2009/12/18), Asterisk 1.6.1.12 (2009/12/18), 1.6.0.20 (2009/12/18), 1.4.28 (2009/12/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #asterisk-gui |
| 11:48.09 | ManxPower-work | I doubt you'll have good luck solving this as you don't know why or how everything is set up already |
| 11:48.32 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 11:49.05 | *** part/#asterisk mattbUK (n=mattbrid@92.27.125.154) |
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| 12:05.11 | ManxPower-work | I hate winter |
| 12:06.05 | benngard | me 2, cold as hell in gothenburg |
| 12:06.23 | ManxPower-work | Yes, but it's not supposed to be cold as hell where I am (Southern USA) |
| 12:07.02 | *** join/#asterisk dandre (n=daniel@161.251.203-77.rev.gaoland.net) |
| 12:07.03 | benngard | any1 worked with avaya phones and asterisk? |
| 12:10.57 | benngard | README.callingpres <- where do i find that file, google gives me like 100 refernces to it but not the file itself :( |
| 12:11.15 | ManxPower-work | Should be in the Asterisk source tree |
| 12:11.25 | ManxPower-work | (same place all official docs live) |
| 12:11.34 | benngard | find . -name README.callingpres -print |
| 12:11.45 | benngard | i did look for it there |
| 12:12.17 | ManxPower-work | [pbx.nyigc.net asterisk-1.4.25.1]# find . -name "*pres*" -print |
| 12:12.17 | ManxPower-work | ./doc/callingpres.txt |
| 12:12.37 | ManxPower-work | perhaps you should BROWSE that directory? |
| 12:15.51 | benngard | doc# ls -al call* |
| 12:15.51 | benngard | -rw-r--r-- 1 root src 4843 2009-12-08 19:35 callfiles.txt |
| 12:16.00 | benngard | i dont have that file :( |
| 12:16.10 | ManxPower-work | try doc# ls -al *call* |
| 12:16.17 | ManxPower-work | what version of Asterisk do you have? |
| 12:16.43 | benngard | SVN-trunk-r237098 |
| 12:16.51 | ManxPower-work | That's not a version. |
| 12:17.03 | ManxPower-work | Why are you using an unreleased/development version of Asterisk? |
| 12:17.21 | benngard | trying some ooh323 stuff together with May213 |
| 12:17.46 | ManxPower-work | Released versions of Asterisk don't have ooh323? |
| 12:18.11 | benngard | not a 1 that was working with avaya cm :( |
| 12:18.13 | ManxPower-work | In any case, I wish you the BEST of luck. |
| 12:18.42 | ManxPower-work | I think I'd rather quit than be forced to use H323 with Asterisk. |
| 12:20.32 | benngard | we are probably gonna "throw out" our avaya and then i can use sip trunks to our pstn provider |
| 12:20.46 | ManxPower-work | Avaya can't do SIP? |
| 12:21.36 | benngard | avaya can do sip, but require a pretty big upgrade for us, and i am not sure i am ready to put that money there |
| 12:22.57 | benngard | there i a lot of things that i lack in the avaya, agents and stuff like that, sure if u open up and pay a lot of bucks u can have it |
| 12:23.13 | ManxPower-work | That is what happens when you go with a closed PBX. |
| 12:23.35 | benngard | it was before my time, rhey vought the avaya |
| 12:23.44 | benngard | they bought* |
| 12:25.06 | benngard | so right know i am like testing if we can replace our avaya with an asterisk |
| 12:26.12 | ManxPower-work | It looked to me like you were trying to use Asterisk to work around the poor decisions of someone else. Asterisk is seldom very good at those sorts of projects. |
| 12:26.31 | ManxPower-work | There's a reason almost nobody uses H323 with Asterisk. |
| 12:28.05 | benngard | like reusing all avaya 9650 phones, converting them from h323 to sip, getting "connected party" to work and ofc a lot of more stuff |
| 12:28.41 | ManxPower-work | you won't get "connected party" with Asterisk. |
| 12:29.19 | *** join/#asterisk Wildy (n=simba@83.149.41.95) |
| 12:30.16 | ManxPower-work | Convert one of the phones to SIP first. Otherwise you're just wasting your time with H323 |
| 12:30.31 | benngard | my 9650 is converted to sip |
| 12:30.53 | benngard | 9650 - sip - asterisk - h323 - avaya |
| 12:31.01 | benngard | that is working |
| 12:31.07 | Wildy | wanted to ask: anyone had good results with KIRK/Polycom DECT hardware? |
| 12:31.19 | Wildy | we'll test a KIRK 300 IP DECT system soon, so need input on the subj |
| 12:32.01 | benngard | why will not connected party be in asterisk? |
| 12:32.23 | ManxPower-work | Asterisk does not supported called party presentation |
| 12:32.31 | *** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk) |
| 12:32.33 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 12:33.50 | benngard | but i guess it will be supported |
| 12:34.11 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 12:34.18 | ManxPower-work | I don't know. I don't participate in the development process. |
| 12:34.34 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:35.01 | benngard | most of the stuff is working, we sip phones, sip dect phone, ata's connected to asterisk |
| 12:35.15 | benngard | some queues, voicemail |
| 12:35.37 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
| 12:35.46 | FlaPer87 | hey guys |
| 12:35.47 | benngard | simple php manager script that let people log in and out of queues |
| 12:36.35 | benngard | i even let one of our suppiort girsl record some of the most common messages |
| 12:37.01 | FlaPer87 | question, I've 2 inbound numbers but for some reason in the inbound context one number wants to be matched with s,1,..... and the other one with NUMBER,1,....., did it happen to any of you? |
| 12:37.44 | ManxPower-work | FlaPer87: one carrier is not sending a destination number (calls with no destination match "s"). One carrier is sending the destination number. "s" IS NOT A WILDCARD. |
| 12:38.50 | FlaPer87 | ManxPower-work: and is there a way to match both with the same rule? |
| 12:39.04 | FlaPer87 | with destionation number you mean the inbound number, right? |
| 12:40.19 | ManxPower-work | destination number == dialed number |
| 12:40.49 | ManxPower-work | FlaPer87: the only way to match both is so discouraged there is special code in Asterisk to print a warning when it's used. |
| 12:41.19 | ManxPower-work | Just use a goto. Concentrate on real problem, not a silly issue like this. |
| 12:42.13 | ManxPower-work | They are two different numbers. Don't get upset when Asterisk treats them differently. |
| 12:42.29 | FlaPer87 | ManxPower-work: ok, thanks =D |
| 12:48.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
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| 13:04.53 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 13:12.19 | Hexie | anyone know of a software (im looking for a dialler) that comes as a component for the .NET platform? (ie - comes in a .dll format, where all the functions and methods are open to the developer to customise as he/she pleases?) |
| 13:13.38 | tzafrir_laptop | Hexie, look into "originate" in the manager interface |
| 13:13.43 | tzafrir_laptop | That's really all you need |
| 13:14.01 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:14.18 | Hexie | i will do thanks (originate from the asterisk software?) |
| 13:16.05 | tzafrir_laptop | from the manager API |
| 13:16.24 | tzafrir_laptop | someone already pointed you previously to a dotnet wrapper for it |
| 13:16.46 | Hexie | k got the API - see the originate functions - u have any docs on how to use this? |
| 13:16.54 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 13:17.22 | ManxPower-work | Smells like a telemarketer to me |
| 13:18.23 | Hexie | me? telemarketer |
| 13:23.36 | *** join/#asterisk DND (n=arabia@94.200.7.26) |
| 13:23.52 | DND | hi guys how can i connect more than 3 asterisk servers using iax? |
| 13:26.26 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
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| 13:29.30 | Hexie | is there any documentation on using the Asterisk API functions (http://sourceforge.net/projects/asterisk-dotnet/) ?? |
| 13:30.09 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 13:30.16 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:30.23 | littleball | hello, I encount iLBC problem. My program send iLBC to asterisk, which send to mobile. I heard the mobile voice very bad. But I can hear very clear from PC. Sure the iLBC encoding is correct because I test with the sample bit files |
| 13:30.27 | littleball | any idea/ |
| 13:30.28 | littleball | ? |
| 13:31.57 | *** join/#asterisk boch (n=fran@200.61.191.9) |
| 13:31.58 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:33.26 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 13:42.36 | [TK]D-Fender | Hexie: .... |
| 13:42.39 | [TK]D-Fender | ~book |
| 13:42.40 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:42.42 | [TK]D-Fender | ~wiki |
| 13:42.47 | [TK]D-Fender | ~wikis |
| 13:42.48 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:42.51 | tzafrir_laptop | Hexie, basic documentation of Asterisk manager commands can be found in the on-line help: |
| 13:42.59 | tzafrir_laptop | manager show commands |
| 13:43.05 | tzafrir_laptop | manager show command Login |
| 13:43.06 | Naikrovek | rofl check out this malarkey i got over the weekend: http://imgur.com/pQFph |
| 13:43.35 | Naikrovek | a bunch of my stupid friends fell for that |
| 13:43.54 | ManxPower-work | What virus was attached? |
| 13:43.59 | Naikrovek | none |
| 13:44.00 | Naikrovek | thankfully |
| 13:44.10 | [TK]D-Fender | littleball: Your mobile (or tis bandwidth in that direction) sucks |
| 13:44.57 | DND | hi guys how can i connect more than 3 asterisk servers using iax? |
| 13:45.11 | Naikrovek | same way you connect less than 3 |
| 13:45.16 | Naikrovek | you trunk them all together |
| 13:45.16 | ManxPower-work | DND: Exactly the same way you connect 3 or fewer servers to Asterisk using IAX |
| 13:45.43 | ManxPower-work | ROFL! Naikrovek you are quite funny for being this early in the morning. |
| 13:45.52 | fenrus | =) |
| 13:45.54 | Naikrovek | thanks |
| 13:46.22 | boch | how can i do parallel forking for sending the same media to two peers ? |
| 13:46.42 | ManxPower-work | boch: you can't. You would have to "hack" it by using Meetme |
| 13:47.02 | TommyBotten | littleball: You're sending iLBC to the phone via the GSM networ? ... |
| 13:47.23 | [TK]D-Fender | boch: Go implement some other proxy. Or chanspy on it if you only want to listen in. |
| 13:47.36 | boch | manxpower, even if one of the two peers wants receive only ? |
| 13:48.12 | ManxPower-work | boch: Perhaps you could tells what you want to accomplish rather than only trying one method to accomplish your goal. |
| 13:48.29 | [TK]D-Fender | boch: I jsut gave you the alternative for that. |
| 13:48.35 | [TK]D-Fender | boch: "core show application chanspy" |
| 13:49.07 | boch | [TK]D-Fender, thank i ll try that app |
| 13:52.14 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 13:52.14 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 13:53.54 | *** join/#asterisk stix (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 13:55.15 | stix | Anyone here who has tried to send text-messages(sms) with the AT-command? My system sends too long sms'es and needs to be divided into two - I don't know how to do it or where to ask |
| 13:55.18 | littleball | yes |
| 13:55.23 | littleball | to GSM network |
| 13:55.47 | ManxPower-work | I see it is a Monday on #Asterisk |
| 13:55.48 | littleball | I think I got the reason. I think I need to use "jitter buffer" in sending side also |
| 13:56.21 | ManxPower-work | littleball: jitter buffers only buffer incoming audio, since you can't buffer outgoing packets |
| 13:57.00 | littleball | I know. I mean I need to buffer the recording |
| 13:57.01 | littleball | data |
| 13:57.23 | littleball | I will test it out . I am coding my own iax client |
| 13:57.33 | TommyBotten | littleball: Why not use the GSM codec for the GSM network? |
| 13:57.58 | ManxPower-work | TommyBotten: Because the telco will convert it back to ulaw/alaw anyway? |
| 13:58.00 | littleball | no |
| 13:58.04 | littleball | GSM is very bad |
| 13:58.15 | Naikrovek | very |
| 13:58.27 | littleball | I send traffic to voice provider |
| 13:58.31 | littleball | through my asterisk |
| 13:58.50 | [TK]D-Fender | And no mention of the carrier protocol... |
| 13:58.52 | ManxPower-work | littleball: Yes. That provider will convert the voice to ulaw or alaw and then send the call to the cell provider. |
| 13:58.59 | [TK]D-Fender | Of course not... why would we dos omething like that... |
| 13:59.33 | littleball | never mind. I almost finish. |
| 14:00.58 | littleball | 120 to 300 ms delay is reasonable? |
| 14:01.06 | ManxPower-work | littleball: no. |
| 14:01.27 | ManxPower-work | At about 150ms humans start to notice the lag. |
| 14:01.37 | littleball | ok |
| 14:01.44 | Naikrovek | well how far are the two end points |
| 14:01.58 | Naikrovek | i have an asterisk system in india, and some extensions in india connected to my system in illinois |
| 14:02.06 | Naikrovek | 400ms is not uncommon, and is not a problem |
| 14:02.12 | Naikrovek | but the lag is indeed noticable |
| 14:03.17 | coppice | 400ms is awful. it badly breaks up conversations |
| 14:03.46 | ManxPower-work | coppice: I think Naikrovek simply has low standards. |
| 14:04.05 | Naikrovek | well i'm a slave of the speed of telecommunications |
| 14:04.17 | coppice | or he never listens to the other guy :-) |
| 14:04.30 | Naikrovek | it's 400ms or no phone calls, i choose 400ms |
| 14:04.39 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-37-228.acanac.net) |
| 14:08.07 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 14:08.30 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:08.31 | fenrus | stix, i have done some sms-sending with AT commands, but i've not investigated how i get linked sms to work |
| 14:08.55 | ManxPower-work | Asterisk does not support "AT commands" |
| 14:09.06 | Naikrovek | my boss is freaking out because he can't dial 407. he gets a busy signal, system shows phone is in use, and that's my fault somehow |
| 14:09.06 | stix | fenrus, oh so it is called linked sms? |
| 14:09.24 | stix | ManxPower-work, didn't know where else to ask |
| 14:09.26 | *** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk) |
| 14:10.17 | fenrus | stix, i'll se if i can get some info about it |
| 14:10.51 | stix | fenrus, thank you very much :) |
| 14:12.19 | voipmonk | Naikrovek: At least you can find out what's happening with a little sip debug |
| 14:12.38 | Naikrovek | voipmonk: the phone is in a box at my desk waiting to be shipped to india, actually |
| 14:12.55 | Naikrovek | someone gave him that extension number who is fiddling with a soft phone |
| 14:13.01 | Naikrovek | and they cant' make it work |
| 14:13.20 | Naikrovek | and he's yelling at me |
| 14:13.24 | Naikrovek | wtf |
| 14:14.22 | Naikrovek | i'm about to institute a policy up in this little organization |
| 14:14.37 | voipmonk | Naikrovek: Where is this softphone user in relation to the asterisk system |
| 14:14.55 | Naikrovek | halfway around the world |
| 14:15.23 | Naikrovek | their rights have been revoked on this system now. they have their own system they should be screwing up |
| 14:15.28 | Naikrovek | not mine |
| 14:17.20 | *** join/#asterisk oktay (n=oktay@81.215.202.193) |
| 14:17.40 | oktay | hello. anybody have the SPA3102 ? Need help with VOIP-to-PSTN Gateway. |
| 14:17.42 | jaytee | came in this morning and had 2 computers infected with the friggin "Security Tool" malware. |
| 14:18.25 | voipmonk | What do you want to do with it oktay? |
| 14:18.46 | oktay | voipmonk: just dial in from the internet and get a local PSTN line |
| 14:18.48 | fenrus | stix, there is something called "multipart bits" |
| 14:18.51 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 14:19.23 | voipmonk | oktay: What have you setup already? |
| 14:19.49 | oktay | another question first. |
| 14:19.59 | oktay | can this be done without using the functionality on the spa ? |
| 14:20.11 | oktay | just as a bare ATA |
| 14:20.36 | ManxPower-work | oktay: your question makes no sense. |
| 14:20.38 | oktay | voip to pstn is actually the only thing I really need. |
| 14:20.52 | [TK]D-Fender | oktay: the 2 prots are completely independant of each other |
| 14:21.04 | oktay | ok. i will try to rephrase. |
| 14:21.28 | oktay | All I need is to somehow provide PSTN access to voip callers |
| 14:21.45 | oktay | and spa3102 is the device I could locally source |
| 14:22.10 | oktay | since I have asterisk, i'm thinking maybe it can handle the logic of things too |
| 14:23.05 | ManxPower-work | It's pretty easy to have phones registered to Asterisk call out via the SPA. |
| 14:23.46 | Chainsaw | Does 1.6.2.0 drop ODBC support, or is no longer a module? |
| 14:23.50 | Chainsaw | +it |
| 14:23.51 | ManxPower-work | There are many documents on the web, all somewhat confusing, about setting up the SPA to allow Asterisk to dial out the PSTN port. |
| 14:24.01 | *** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) [NETSPLIT VICTIM] |
| 14:24.03 | [TK]D-Fender | Chainsaw: Its something you should be paying attention to make menuconfig for |
| 14:24.15 | oktay | ManxPower-work: you mean easy to to voip -> pstn ? |
| 14:24.19 | *** join/#asterisk t- (i=tom@freenode/staff/tomaw) |
| 14:24.28 | Chainsaw | [TK]D-Fender: --without-odbc is no longer respected, just wondering why. |
| 14:24.28 | ManxPower-work | Dropping something like ODBC would be in the UPGRADE*.txt |
| 14:24.37 | ManxPower-work | oktay: stop saying "voip". Be SPECIFIC. |
| 14:24.50 | ManxPower-work | It is easy to have asterisk -> SPA -> PSTN if that is what you want. |
| 14:25.01 | Chainsaw | ManxPower-work: I'd hope so, yes. |
| 14:25.02 | [TK]D-Fender | Chainsaw: Perhaps they expect you to use menuconfig for this <- |
| 14:25.14 | Katty | hi |
| 14:25.14 | oktay | ManxPower-work: this is all spa terminology. sorry about the confusion. |
| 14:25.19 | Chainsaw | [TK]D-Fender: Which is new, yes. Why wasn't that mentioned in CHANGES? |
| 14:25.27 | Chainsaw | Morning Katty :) |
| 14:25.42 | [TK]D-Fender | Chainsaw: What were you running prior? |
| 14:25.44 | oktay | oktay: let's say.. someone using a softphone sip client.. should have access to the PSTN line the spa is connected to |
| 14:25.46 | ManxPower-work | oktay: Are yo using Asterisk? |
| 14:25.47 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 14:26.08 | oktay | ManxPower-work: yes |
| 14:26.14 | Chainsaw | [TK]D-Fender: I never see menuselect, it's driven automatically. I'm a packager. |
| 14:26.15 | ManxPower-work | the start putting asterisk in your diagrams |
| 14:26.29 | ManxPower-work | " oktay: oktay: let's say.. someone using a softphone sip client.. should have access to the PSTN line the spa is connected to" <-- I see no mention of Asterisk |
| 14:26.40 | [TK]D-Fender | Chainsaw: Now try answering the question I asked... |
| 14:26.52 | _cgc | oktay: u can just setup a sip softphone to connect to the asterisk box, then its just down to the extensions.conf and dahdi configuration i think |
| 14:27.03 | Chainsaw | [TK]D-Fender: You've never answered mine. I'll have a look myself. |
| 14:27.06 | ManxPower-work | _cgc: he's not using DAHDI |
| 14:27.13 | Katty | Chainsaw: hello. |
| 14:27.14 | *** join/#asterisk tris (i=tristan@207.241.238.17) [NETSPLIT VICTIM] |
| 14:27.20 | [TK]D-Fender | Chainsaw: Your answer is a pre-req for mine. |
| 14:27.30 | _cgc | ManxPower-work: ahhh sorry |
| 14:27.45 | Chainsaw | [TK]D-Fender: One day it will be revealed that you're just a hacked-up alice script. |
| 14:27.46 | [TK]D-Fender | oktay: As I said, the ports are INDEPENDENT. You don't need to configure or use the FXS on it |
| 14:28.08 | [TK]D-Fender | Chainsaw: I do not understand. Please rephrase your question. |
| 14:28.50 | oktay | [TK]D-Fender: which one is FXS again? I mix up FXS and FXO |
| 14:29.03 | ManxPower-work | ~fxoFXS |
| 14:29.04 | infobot | [fxofxs] An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 14:29.04 | [TK]D-Fender | ~fxofxs |
| 14:29.05 | infobot | hmm... fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 14:29.05 | oktay | i know the ports are independend by the way |
| 14:29.43 | oktay | so what I want is SIP Client -> Asterisk -> SPA3102 -> FXS -> PSTN -> Happiness |
| 14:29.57 | [TK]D-Fender | oktay: And thats fine. Now go do it |
| 14:30.08 | [TK]D-Fender | oktay: Plenty of google-able guides on setting it up |
| 14:30.13 | oktay | oh how I never miss coming back to this channel :) |
| 14:30.22 | [TK]D-Fender | oktay>so what I want is SIP Client -> Asterisk -> SPA3102 -> FXS -> PSTN -> Happiness <-- and that should be FXO, not FXS |
| 14:30.25 | ManxPower-work | oktay: You did not read the [fxofxs] correctly |
| 14:30.41 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:30.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:30.57 | oktay | true that |
| 14:31.10 | oktay | [TK]D-Fender: spa guides are like guitar tabs |
| 14:31.17 | oktay | they are all a little bit broken |
| 14:31.21 | [TK]D-Fender | oktay: Both work for me. |
| 14:32.43 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 14:34.46 | tzafrir_laptop | Chainsaw, the equivalent of using menuconfig to enable/disable a feature is to add a patch that sets/removes the 'defaultenabled' property |
| 14:36.07 | oktay | thanks guys. i know you are trying to help in some way. |
| 14:40.07 | [TK]D-Fender | orkWe ahve, by telling you you can do exactly what you were hoping to do with it. |
| 14:40.19 | [TK]D-Fender | oktay: We have, by telling you you can do exactly what you were hoping to do with it. |
| 14:40.35 | ManxPower-work | Well, at least as much as we understand by your random and not well thought out questions and statements. |
| 14:50.14 | *** join/#asterisk moy (n=moy@74.12.129.52) |
| 14:51.01 | littleball | hello, what is the reason of 'gulugulu' sound (like flowing water ) ? |
| 14:51.12 | littleball | background noise? |
| 14:51.41 | oktay | they won't help you unless asterisk is making the gulugulu sound |
| 14:51.56 | ManxPower-work | littleball: Why not try testing with a simple setup |
| 14:52.32 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 14:52.50 | littleball | My program works already. It is the same as other voip client (like zoiper). both have gulugulu sound |
| 14:53.18 | littleball | maybe because of my build in MIC |
| 14:53.23 | ManxPower-work | littleball: I wish you the BEST of luck. |
| 14:54.39 | littleball | I think it is due to my aircon noise. :-). |
| 14:55.04 | littleball | I found that the gulugulu is exactly the same as my aircon noise ;( |
| 14:55.13 | oktay | littleball: doesn't sound that bed to me.. it could be a sledgehammer sound too |
| 14:55.21 | oktay | it's somewhat zen even |
| 14:55.39 | littleball | Ok. finally, i have make my own IAX client work ... |
| 14:56.23 | *** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net) |
| 15:02.24 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
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| 15:07.38 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:07.59 | *** join/#asterisk smooth_penguin (n=smoove@59.95.20.135) |
| 15:09.59 | *** join/#asterisk insatiable_2 (n=insat@212.36.209.22) |
| 15:10.42 | insatiable_2 | Hello, i have a problem with extensions changing status to UNREACHABLE, anyone can help ? |
| 15:13.10 | [TK]D-Fender | insatiable_2: Try providing some useful details and maybe we can. |
| 15:13.41 | insatiable_2 | <PROTECTED> |
| 15:13.43 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 15:14.00 | insatiable_2 | when remote users behind the nat try to register, they get registered and they can make calls |
| 15:14.17 | insatiable_2 | but after like 1minutes, they get unregistered the status changes to "UNREACHABLE" |
| 15:14.23 | insatiable_2 | hwo can i solve this ? |
| 15:15.38 | _cgc | insatiable_2: are the clients using sip? |
| 15:15.45 | insatiable_2 | yes |
| 15:15.51 | insatiable_2 | SIP client - SJPhone |
| 15:16.18 | voipmonk | might be a device thinking its getting hax0red... then it closes up or limits the traffic |
| 15:16.44 | _cgc | insatiable_2: sip has some serious problems with natting, I haven't found a solution but would be interested in any solution you find, from what I know x-lite uses stun to fix it, but I haven't tested it |
| 15:17.05 | insatiable_2 | i did change the "qualify" to 4000 ms and it didnt work too |
| 15:18.57 | insatiable_2 | any solution ? |
| 15:19.16 | [TK]D-Fender | insatiable_2: Go join #freepbx and do "?? nat" , and ""?? ports" in channel |
| 15:19.16 | _cgc | have you tried x-lite or another softphone? |
| 15:19.27 | *** join/#asterisk Gopal (n=Miranda@117.193.110.2) |
| 15:19.41 | insatiable_2 | yes i did try 3 softphones... same thing |
| 15:19.43 | Gopal | have anybody integrated jbiling with asterisk? |
| 15:20.44 | voipmonk | did you buy the documentation for jbilling? |
| 15:20.46 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 15:21.23 | Skeeter- | happy new year everyone |
| 15:21.30 | yang | hey [TK]D-Fender can I make register string out of userid only ? I tried register => userID@PBX-domain:5060 and register => userID:password@PBX-domain:5060, which usually works for others, but not with this provider... |
| 15:21.37 | Gopal | voipmonk: no |
| 15:21.56 | oktay | Skeeter-: happy new year |
| 15:22.11 | voipmonk | Good luck, Gopal |
| 15:22.16 | voipmonk | let me know how it works out |
| 15:22.26 | [TK]D-Fender | jaytee: user:pass@host:port/exten |
| 15:22.55 | yang | [TK]D-Fender: exten should be the phone number starting with 00... ? |
| 15:22.58 | voipmonk | there are some wholesalers that jump up and down and do a dance about jbilling & asterisk - but I never had the time to dick with it |
| 15:23.05 | [TK]D-Fender | yang: register=> user:pass@host:port/exten |
| 15:23.07 | Gopal | voipmonk: I am looking for a integration but not able to |
| 15:23.14 | ManxPower-work | SIP does not have serious problems with MOST types of NAT setups. |
| 15:23.32 | [TK]D-Fender | yang: It should be whatever you want it to be |
| 15:25.14 | yang | [TK]D-Fender: are you able to see about the error from SIP debug ? |
| 15:26.16 | yang | [TK]D-Fender: I get only this state - 120 Auth. Sent |
| 15:26.54 | ManxPower-work | sounds like you are either authing to the wrong server or you have an unresolved NAT issue |
| 15:28.21 | yang | I could display my SIP info in a query, don't want to make the password public (actually with their info provided password is not visible) |
| 15:28.24 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:28.37 | [TK]D-Fender | yang: Means no answer is making it BACk and I don't see your SIP DEBUG |
| 15:29.49 | yang | ok lets try SIP debug first |
| 15:33.34 | Katty | brr. |
| 15:33.54 | Katty | shivers a bit |
| 15:34.30 | Katty | checks on the squirrels. |
| 15:34.46 | *** part/#asterisk pentanol (n=pentanol@77.35.52.14) |
| 15:35.27 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:36.01 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 15:36.09 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:39.35 | Katty | http://i.imgur.com/5C8Qn.jpg <- previous decade changes. |
| 15:41.49 | yang | [TK]D-Fender: i am sending pastebin link to your query, you can respond here later |
| 15:42.40 | smooth_penguin | doesnt look like much of a progress Katty :p |
| 15:43.02 | yang | [TK]D-Fender: look for bluesip.net error, thanks |
| 15:46.48 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 15:47.16 | [TK]D-Fender | yang: Auth errors everywhere |
| 15:48.06 | yang | Katty: I don't like the statement about gas prices - doubled, they have been tripled/quad here in Europe |
| 15:48.31 | *** join/#asterisk vally (n=fu@hermes.weelya.com) |
| 15:48.34 | leifmadsen | yang: that was a US centric list though |
| 15:48.44 | Chainsaw | yang: Yes, European gas is taxed where as US gas is subsidised. |
| 15:48.53 | Chainsaw | yang: The difference is very noticeable, even after currency conversion they're a lot better off. |
| 15:48.56 | leifmadsen | yang: it wasn't comparing Europe to the world, it was comparing the US to the world (mostly) |
| 15:50.04 | yang | yes, anyway with an European car someone could drive for very cheap around the Us, but yeah most have SUVs there |
| 15:50.10 | Katty | a new baby lemur was born :> |
| 15:50.27 | Katty | that makes 18 baby lemurs living in captivity. |
| 15:50.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:51.00 | Katty | http://denverpost.slideshowpro.com/albums/001/496/album-84965/cache/YEARENDER_2009-SCIENCE-ZOOL.sJPG_920_590_0_95_1_50_50.sJPG |
| 15:51.20 | yang | [TK]D-Fender: well, how weird, becouse other VoIP uplinks are working |
| 15:52.29 | yang | [TK]D-Fender: are you able to spot bluesip.net specific error ? |
| 15:52.30 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:52.31 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:53.29 | [TK]D-Fender | yang, no |
| 15:53.35 | *** part/#asterisk benngard (n=benngard@213.88.138.230) |
| 15:55.11 | yang | Katty: hehe |
| 15:55.20 | *** join/#asterisk Gopal (n=Miranda@117.193.110.2) |
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| 15:56.19 | Gopal | voipmonk: did you ever tried jbilling + asterisk? |
| 15:56.28 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:58.19 | voipmonk | beginning of last year , yes |
| 15:58.48 | yang | [TK]D-Fender: would you know how to make the correct register string, if I tell you the data, I tried several combinations, which don't work. |
| 15:59.09 | ManxPower-work | yang: try srvlookup=no in your sip.conf [general] |
| 15:59.22 | yang | ok |
| 16:00.59 | *** join/#asterisk sassyn (n=sassyn@93-173-106-38.bb.netvision.net.il) |
| 16:01.03 | sassyn | hi all |
| 16:01.29 | sassyn | Can someone please tell me if he using asterisk 1.6? |
| 16:01.34 | sassyn | I have version 1.4 runing |
| 16:01.38 | sassyn | for a long time |
| 16:01.48 | sassyn | And I want to know if it is safe to upgrade |
| 16:01.52 | yang | I received from VoIP uplink - SIP address, sip username, sip domain, and DID number |
| 16:01.53 | sassyn | I'm using FreePBX |
| 16:01.59 | [TK]D-Fender | sassyn: Go ask them then |
| 16:02.00 | yang | ~freepbx |
| 16:02.01 | infobot | [~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, or fpbx |
| 16:02.55 | sassyn | [TK]D-Fender, All I want to know is what is the status of version 1,6 |
| 16:02.55 | yang | ManxPower-work: it didn't solve my case |
| 16:03.03 | ManxPower-work | yang: it was worth a try. |
| 16:03.05 | sassyn | [TK]D-Fender, is it stable |
| 16:03.13 | [TK]D-Fender | sassyn: Yes |
| 16:03.25 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:03.33 | sassyn | [TK]D-Fender, So I can upgrade from my 1.4.17 version? |
| 16:03.55 | sassyn | [TK]D-Fender, What does 1.6 give me insted of version 1.4? |
| 16:04.13 | Naikrovek | if you haven't done that research then why are you asking about upgrading |
| 16:04.28 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 16:04.47 | sassyn | Naikrovek, I had a system crash on the disk |
| 16:05.03 | ManxPower-work | sassyn: you are still asking on the wrong channel |
| 16:05.04 | sassyn | And I'm not sure if I want to recover from back or do a new install |
| 16:05.16 | Naikrovek | 1.6.0 is perfectly stable and all that. freepbx is a different matter entirely |
| 16:05.16 | Katty | personally, i'd do a new install. |
| 16:05.22 | sassyn | ManxPower-work: Why the worng channel? |
| 16:05.26 | [TK]D-Fender | sassyn: go read the CAHNGELOGS, etc |
| 16:05.40 | ManxPower-work | sassyn: you are using FreePBX and you are not asking on the FreePBX channel |
| 16:06.08 | sassyn | ManxPower-work, All I'm asking is the status of version 1.6 |
| 16:06.15 | sassyn | Leave the FreePBX out. |
| 16:06.16 | Naikrovek | 1.6 is stable |
| 16:06.18 | ManxPower-work | sassyn: Which 1.6 version? |
| 16:06.26 | Naikrovek | there is 1.6.0 and 1.6.2 |
| 16:06.28 | [TK]D-Fender | sassyn: Yes, 1.6 is stable |
| 16:06.31 | [TK]D-Fender | and 1.6.1 |
| 16:06.35 | Naikrovek | true |
| 16:06.42 | ManxPower-work | 1.6.0.x and 1.6.1.x seem reasonably stable. I would not use 1.6.2.x because it's just been released. |
| 16:07.13 | sassyn | 1.6.2 |
| 16:07.17 | sassyn | the lastest |
| 16:07.24 | ManxPower-work | Obviously we don't know if 1.6.x is compatible with FreePBX or not. |
| 16:07.28 | ManxPower-work | sassyn: NO! |
| 16:07.42 | ManxPower-work | 1.6.2 is a BRANCH not a RELEASE. |
| 16:07.54 | Katty | a branch. |
| 16:07.56 | Katty | blowing in the wind? |
| 16:08.09 | Katty | i would type wind noises, but i'm not sure how to do that on irc. |
| 16:08.16 | Katty | woooooooshhhh? |
| 16:08.18 | sassyn | asterisk-1.6.2.0.tar.gz |
| 16:08.31 | tzafrir_laptop | Katty, you should be well aware that those branches are more stable than trunk |
| 16:08.42 | tzafrir_laptop | ? |
| 16:08.48 | sassyn | ManxPower-work, asterisk-1.6.2.0.tar.gz is not a RELEASE? |
| 16:08.51 | ManxPower-work | sassyn: 1.6.2 is a generic term for all 1.6.2.x releases in the 1.6.2 branch. |
| 16:08.54 | [TK]D-Fender | sassyn: 1.6.2 not so recommended jsut yet |
| 16:09.01 | ManxPower-work | sassyn: yes, 1.6.2.0 is a release. 1.6.2 is not. |
| 16:09.36 | sassyn | I will install the latest one |
| 16:09.54 | sassyn | Will build a new and fresh RPM plg |
| 16:09.56 | sassyn | pkg |
| 16:09.59 | x86 | I'm trying to setup my sip peers to be realtime (by peers I mean SIP VSP providers) |
| 16:10.25 | x86 | I've already got my local SIP users to be realtime via MySQL, but I can't get my peers to be realtime also |
| 16:10.29 | ManxPower-work | in order of major releases: 0.65, 1.0.x, 1.2.x, 1.4.x, 1.6.0.x, 1.6.1.x, 1.6.2.x As you can see 1.6.0.x, 1.6.1.x and 1.6.2.x are all major releases. |
| 16:10.58 | x86 | chan_sip.so will no longer load after removing my peer from sip.conf, while keeping the 'register' line for the peers in sip.conf |
| 16:11.00 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 16:11.02 | bcrisp | mornin |
| 16:11.12 | Katty | hi crispy. |
| 16:11.12 | x86 | do I have to make the registrations realtime for this to work properly? |
| 16:11.26 | bcrisp | garage door broke today :< |
| 16:11.46 | Katty | oh man |
| 16:11.51 | Katty | is your house going to be freezing cold? |
| 16:11.51 | Talkradio | that sucks |
| 16:11.59 | bcrisp | nah just the opener itself |
| 16:12.01 | chuckf | not while you were under it I hope |
| 16:12.05 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:2843:15a0:a077:acde) |
| 16:12.06 | cusco | hi |
| 16:12.15 | bcrisp | the spring is all good.. its the screw-driven opener |
| 16:12.20 | Talkradio | that manual labor will help warm you up early in the morning |
| 16:12.31 | cusco | im trying to telnet into asterisk's manager... but seems that Im not getting the wanted results |
| 16:12.31 | Nugget | telnet is eeeeeeevil! |
| 16:12.33 | bcrisp | i think the wiring is bad |
| 16:12.59 | Katty | cusco: yeah i want some tea too, but it seems like my mind powers just aren't good neough to make it appear. |
| 16:13.08 | cusco | [Jan 4 16:12:46] VERBOSE[15244] manager.c: == Connect attempt from '192.168.2.228' unable to authenticate |
| 16:13.32 | ManxPower-work | cusco: did you authenticate |
| 16:13.52 | cusco | first thing as telnet opened: Action: Login |
| 16:13.56 | cusco | Username: blah |
| 16:14.00 | cusco | Secret: bleh |
| 16:14.10 | cusco | but telnet did not respond back |
| 16:14.48 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:15.27 | ManxPower-work | do you have the user and secret set up in /etc/asterisk/manager.conf? |
| 16:15.35 | ManxPower-work | and reload manager after making changes to that file? |
| 16:15.52 | cusco | yes I do |
| 16:15.56 | cusco | yes |
| 16:16.12 | cusco | http://paste.debian.net/55581/ |
| 16:16.12 | ManxPower-work | I think there's a "manager show users" in the CLI you can use to verify. |
| 16:16.23 | cusco | telnet just does not reply |
| 16:16.27 | *** join/#asterisk bmoraca (n=bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:16.50 | ManxPower-work | I believe you need TWO enters after the last line of your login |
| 16:17.06 | leifmadsen | yes you do |
| 16:17.11 | ManxPower-work | you should read manager.txt or whatever obvious manager related files are in doc/ |
| 16:17.41 | cusco | ahh |
| 16:17.53 | cusco | ok |
| 16:17.56 | cusco | thanks |
| 16:18.28 | bcrisp | hmm can you configure voicemail.conf to send vm-attached emails to multiple recipients? |
| 16:18.50 | ManxPower-work | bcrisp: I don't think so. |
| 16:19.12 | ManxPower-work | bcrisp: You can create an e-mail alias in whatever MTA you are using. |
| 16:19.18 | bcrisp | there we go |
| 16:19.19 | bcrisp | ty |
| 16:31.02 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:32.14 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.15.229) |
| 16:33.11 | *** join/#asterisk ruyo (n=psantos@195.23.253.223) |
| 16:34.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:36.27 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 16:36.41 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 16:36.59 | *** join/#asterisk KaneHau (n=KaneHau@133.40.166.155) |
| 16:39.09 | KaneHau | ok... help! I have ASTERISK dialing an outside phone number via DAHDI to deliver an automated voice message. The phone rings, but * is not waiting for the phone to be answered (it is simply plowing right ahead in the script). What do I need to do to WAIT until the phone is actually answered? |
| 16:39.47 | bcrisp | KaneHau, same issue from last week ? |
| 16:40.00 | KaneHau | yes, hoping more people are around after the holidays ;) |
| 16:40.05 | bcrisp | good plan |
| 16:40.58 | p3nguin | You didn't take the advice that was given to you before and develop a working solution? |
| 16:41.21 | KaneHau | "develop a working solution" - cute... but nearly impossible without some guidance |
| 16:41.26 | KaneHau | I tried delays, but that doesn't work |
| 16:41.31 | [TK]D-Fender | you didn't find a clue and think to come with something to SHOW US? |
| 16:41.56 | KaneHau | No, I expected that PHONE HARDWARE would at the VERY LEAST - recognize when a phone is picked up |
| 16:42.12 | beek | Mornin' [TK]D-Fender |
| 16:42.59 | [TK]D-Fender | KaneHau: Maybe if you configured it right. |
| 16:43.03 | p3nguin | DAHDI channels apparently are in ANSWER status as soon as they are done dialing. |
| 16:43.13 | [TK]D-Fender | KaneHau: Now go show us what you've done and what happens. |
| 16:43.27 | KaneHau | p3: yes |
| 16:43.32 | [TK]D-Fender | KaneHau: And you haven't told us what you're using either |
| 16:43.47 | KaneHau | hold on |
| 16:43.55 | KaneHau | what is that url to the posting thing you like to use? |
| 16:44.10 | beek | ~pastebin |
| 16:44.11 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:44.12 | KaneHau | thanks |
| 16:44.21 | carrar | same stuff, different year! |
| 16:44.27 | ManxPower-work | p3nguin: only on FXO signalled ports. |
| 16:45.18 | [TK]D-Fender | .... |
| 16:46.50 | KaneHau | ok.... http://pastebin.ca/1737544 |
| 16:47.00 | ManxPower-work | KaneHau: Your issue is specific to FXO ports. |
| 16:47.09 | KaneHau | yes |
| 16:47.18 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:47.32 | KaneHau | my application dials scientists at remote numbers and speaks audio alarms (we monitor observatory sensors) |
| 16:47.52 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:47.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:48.04 | [TK]D-Fender | KaneHau: Lets try this again..... |
| 16:48.15 | ManxPower-work | KaneHau: then you will either have to stop using FXO ports or develop the rather complicated system to work around the issue. |
| 16:48.20 | [TK]D-Fender | KaneHau: WHAT &^#$ING HARDWARE ARE YOU USING? |
| 16:48.44 | KaneHau | Digium 410 board with one FXO and one FXS port (I got the FXS just to play with) |
| 16:48.58 | KaneHau | and hardware echo cancel |
| 16:49.10 | [TK]D-Fender | KaneHau: Its answered automatcailly because you did not enable CALLPROgreSS for that port |
| 16:49.21 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 16:49.22 | KaneHau | ok... thanks. looking into that |
| 16:49.48 | *** join/#asterisk etnos (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 16:50.03 | ManxPower-work | You are recommending callprogress? What did he do to piss you off? |
| 16:50.44 | ManxPower-work | callprogress= should be renamed randomlydisconnectmycalls= |
| 16:51.08 | [TK]D-Fender | ManxPower-work: I'll burn that pridge when i come to it. |
| 16:51.11 | [TK]D-Fender | bridge* |
| 16:51.18 | KaneHau | hmmm, searching the ASTERISK PDF for "callpro" produces zero results |
| 16:51.24 | KaneHau | turns to google |
| 16:51.29 | ManxPower-work | KaneHau: STOP! |
| 16:51.34 | voipmonk | lol! |
| 16:51.37 | KaneHau | screaches to a halt |
| 16:51.56 | ManxPower-work | KaneHau: look in /path/to/src/asterisk/configs/zapata.conf or chan_dahdi.conf |
| 16:52.06 | [TK]D-Fender | KaneHau: CALLPROGRESS=YES |
| 16:52.08 | ManxPower-work | the .sample files of course. |
| 16:52.18 | [TK]D-Fender | KaneHau: Shift-slip. Clues are in a bin to your left |
| 16:53.06 | KaneHau | ok, I see it in chan_dahdi_additional.conf |
| 16:53.43 | ManxPower-work | KaneHau: Asterisk does not come with a chan_dahdi_additional.conf. Perhaps you are confused and are using FreePBX? |
| 16:53.51 | ManxPower-work | ~guis |
| 16:53.52 | infobot | [guis] "FreePBX/Trixbox is to Asterisk as Windows 95 is to DOS" |
| 16:54.02 | x86 | has anyone done realtime sipregs with 1.6.1 or 1.6.2 yet? |
| 16:54.04 | KaneHau | yes, I installed FreePBX to help with the initial setup... the rest I'm doing via the configuration files |
| 16:54.10 | x86 | I know it's kind of brand new, so just curious |
| 16:54.24 | ManxPower-work | KaneHau: Using the FreePBX config files is exactly the same as using FreePBX. |
| 16:54.30 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 16:55.43 | ManxPower-work | In fact all the GUI does is modify those really complicated, confusing config files that come with FreePBX |
| 16:56.29 | KaneHau | I had originally setup Asterisk on SUSE myself, but could only get it to answer calls, never make a phone ring. Got frustrated and trashed the entire system and did the FreePBX to see if it would make it "easier" - it didn't really, but did at least help me to get the phone to ring |
| 16:57.41 | ManxPower-work | I'm not interested in your story. |
| 16:58.05 | KaneHau | too late |
| 16:58.10 | [TK]D-Fender | goes to find a publisher |
| 16:58.13 | ManxPower-work | You'll soon come to learn how much of a mistake trying to use FreePBX is. No need for me to try to convince you. |
| 16:58.22 | KaneHau | I've already reached that conclusion |
| 16:58.23 | Kobaz | heh |
| 16:58.46 | ManxPower-work | Almost every piece of advice you get will have to be "translated" into FreePBX terms. |
| 16:59.01 | KaneHau | I don't see why. At the config level, it's all asterisk and dahdi |
| 16:59.14 | KaneHau | I'm doing exactlyt he same thing I was doind without freepbx |
| 16:59.16 | ManxPower-work | KaneHau: You'll understand soon enough. |
| 16:59.24 | Kobaz | yeah but if you're gonna just monkey with the asterisk configs, then freepbx is a waste of time |
| 16:59.41 | ManxPower-work | you had a chan_dahdi_additional.conf when you were not running FreePBX? |
| 16:59.58 | KaneHau | no, but I had a chan_dahdi.conf |
| 17:00.02 | *** join/#asterisk dandre (n=daniel@49.193.203-77.rev.gaoland.net) |
| 17:00.36 | ManxPower-work | exactly. So now everytime someone says "chan_dahdi.conf" you'll have to figure out which of the several chan_dahdi*.conf files you will actually have to edit. The same goes for most anything else in Asterisk. |
| 17:01.06 | KaneHau | manx: I no longer use the gui - I just did it to get started. I can simply merve chan_dahdi*.conf into chan_dahdi.conf - that isn't a stretch fo rme |
| 17:01.11 | KaneHau | merge |
| 17:01.30 | ManxPower-work | KaneHau: and all the other non-standard config files too, of course. |
| 17:03.56 | KaneHau | doing "dahdi show channnel 1" does not show if callprogress is on or not. Is there some way to confirm that it is set to yes? |
| 17:05.12 | [TK]D-Fender | KaneHau: Look at your configs, and ensure you've restarted * |
| 17:05.33 | KaneHau | I put it in the config, and restarted dahdi and reloaded the dialplan |
| 17:06.09 | [TK]D-Fender | KaneHau: Then start guessing, because that's all we can. |
| 17:06.26 | KaneHau | is there a DAHDI manual somewhere? |
| 17:06.28 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 17:07.02 | *** part/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 17:07.06 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 17:07.12 | KaneHau | wouldn't mind reading what exactly "callprogress=yes" does |
| 17:12.13 | tzafrir_laptop | KaneHau, sort of. The README files (of both -linux and -tools) |
| 17:12.31 | KaneHau | rebooted... well, just putting the callprogress=yes didn't make any changes. Googling isn't finding a dahdi manual nor much info on how to use callprogress |
| 17:13.07 | ManxPower-work | KaneHau: too bad you don't have the sample config files handy, all the options are documented in that file. |
| 17:13.48 | KaneHau | maybe I"ll just trash the freePBX, reinstall SUSE or UBUNTU and reinstall asterisk and dahdi |
| 17:14.06 | voipmonk | from source? |
| 17:14.08 | voipmonk | :) |
| 17:14.09 | KaneHau | yes |
| 17:14.29 | voipmonk | well mayBE :) |
| 17:14.29 | KaneHau | makes a tarball of the current config |
| 17:14.40 | ManxPower-work | just remember to blow away the config files or all your work will be for nothing |
| 17:14.57 | KaneHau | umm, installing a new OS should blow away the config files :) |
| 17:16.12 | *** join/#asterisk jsolis (n=Jimmy@200.121.176.59) |
| 17:17.04 | jsolis | hi guys |
| 17:17.27 | voipmonk | hello jsolis , I sense..... a question.... |
| 17:17.37 | jsolis | why i get this warning res_monitor.c: Execute of ( nice -n 19 sox -m "/var/spool/asterisk/monitor/agent-1037-1262624867-8560-in.gsm" "/var/spool/asterisk/monitor/agent-1037-1262624867-8560-out.gsm" "/var/spool/asterisk/monitor/agent-1037-1262624867-8560.gsm" && rm -f "/var/spool/asterisk/monitor/agent-1037-1262624867-8560-"* ) & failed. |
| 17:18.20 | ManxPower-work | jsolis: because something didn't work |
| 17:18.38 | jsolis | :o |
| 17:20.26 | *** join/#asterisk galeras (n=galeras@186.81.106.212) |
| 17:27.30 | *** join/#asterisk [Outcast] (n=anonymou@ipn36372-d82722.cidr.lightship.net) |
| 17:27.35 | Katty | mmm, lunch |
| 17:29.28 | Katty | http://www.voilawednesdays.com/include/images/varieties_garlicChicken_lg.jpg <- lunch. |
| 17:30.12 | *** join/#asterisk Heretic (n=fallen@dsl-246-92-139.telkomadsl.co.za) |
| 17:32.27 | *** join/#asterisk zippytech (n=ron@71.155.129.241) |
| 17:32.43 | zippytech | i just installed asterisk and have 6 new polycom 650 phones, I have a rebot like sound every few words any idea.s? |
| 17:32.53 | [TK]D-Fender | [12:12]<KaneHau>rebooted... well, just putting the callprogress=yes didn't make any changes. Googling isn't finding a dahdi manual nor much info on how to use callprogress <- what makes you think I'd take it at face value that you did it right? |
| 17:34.16 | Katty | zippytech: your question does not parse, please try again. |
| 17:34.23 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:36.13 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 17:36.15 | bcrisp | whats a rebot? |
| 17:36.33 | zippytech | robot |
| 17:36.45 | Naikrovek | made of rebar |
| 17:36.49 | *** join/#asterisk joesuffceren (n=chatzill@ip68-104-167-226.ph.ph.cox.net) |
| 17:37.02 | KaneHau | tkd: I'm trashing the system, installing UBUNTU and reinstalling asterisk/dahdi source and will try again. freePBX was just too wacky |
| 17:37.19 | joesuffceren | does anyone use Skype for Asterisk? I'm trying to find out if it's possible to display my telco's phone number as the callerid when placing a call out via skypeout/Skype for Asterisk |
| 17:37.20 | zippytech | just like brrrrp |
| 17:37.29 | *** join/#asterisk smooth_penguin (n=smoove@59.95.20.135) |
| 17:38.02 | bcrisp | burp? |
| 17:38.37 | jaytee | I've never heard a robot go "brrrrp". Mostly just "whirrrr, click, click, beep-boop, beep-boop" |
| 17:38.55 | jaytee | or "DANGER, WILL ROBINSON!!!" |
| 17:39.00 | tzafrir_laptop | KaneHau, you will attempt using freePBX again? |
| 17:39.05 | Naikrovek | well they can synthesize any sound they want, but only the evil versions |
| 17:39.25 | KaneHau | I've head 'em say "EXTERMINATE... EXTERMINATE..." |
| 17:39.32 | KaneHau | tza: no |
| 17:39.41 | bmoraca | Crush...Kill...Destroy! |
| 17:40.08 | KaneHau | freePBX at least made me aware of what was wrong with my original asterisk configuration... so I learned a bit. But I find it just to confusing in what they did with the configuration files |
| 17:40.29 | bcrisp | KaneHau, found a publisher yet? |
| 17:40.33 | bcrisp | :/ |
| 17:40.40 | Naikrovek | KaneHau: yeah, but once you get it set up you don't use the config files anymore. freepbx overwrites 'em |
| 17:40.46 | *** join/#asterisk GreGrenada (i=4549f3c2@gateway/web/freenode/x-ayzcmupabvyrcood) |
| 17:41.07 | KaneHau | I rather use vi :) |
| 17:41.44 | KaneHau | tza: thank you |
| 17:42.29 | jaytee | who the hell is tza? |
| 17:42.49 | bmoraca | why would you use vi to edit asterisk files? |
| 17:42.55 | bmoraca | winscp is 100x more efficient |
| 17:43.07 | KaneHau | got a private message from them with the DAHDI manual... I think he's logged off now |
| 17:43.25 | tzafrir_laptop | jaytee, I guess there are basically two options. And tzanger is not here right now |
| 17:43.29 | KaneHau | because I've used vi for the last 30 years and am very good at it :) |
| 17:44.00 | tzafrir_laptop | bmoraca, because vim has syntax hilighting for its syntax? |
| 17:44.00 | bmoraca | doesn't make it any more efficient. winscp is still far, far more efficient. |
| 17:44.01 | Naikrovek | vi is very nice |
| 17:44.11 | Naikrovek | bmoraca: still gotta edit the files somewhere |
| 17:44.16 | KaneHau | bmor: only if you know how to use it :) |
| 17:44.21 | Naikrovek | and efficiency isn't always priority #1 |
| 17:44.26 | KaneHau | and 'vi' is on EVERY unix system by default |
| 17:44.33 | tzafrir_laptop | Why would you need winscp? vim can edit files over scp/sftp |
| 17:45.15 | tzafrir_laptop | bmoraca, you actually work on an OS that does not include a vi? |
| 17:45.37 | KaneHau | tza: to be fair, I wouldn't recommend vi/vim to anyone who isn't already familiar with it |
| 17:45.38 | bmoraca | i work on windows because i crave efficiency in my productivity...i don't like to tinker. |
| 17:45.52 | Naikrovek | vi is WAY faster than notepad + winscp |
| 17:45.53 | ManxPower-work | bmoraca: same reason I use linux |
| 17:45.54 | KaneHau | hmmm, 'windows' and 'efficiency' and 'productivity' don't go together |
| 17:46.09 | Linuturk | I'm trying to edit some prompts on my computer, but none of my audio players can work with alaw, gsm, ilbc, sln, or ulaw |
| 17:46.12 | Naikrovek | KaneHau: they do if you don't want to fiddle all the time |
| 17:46.26 | tzafrir_laptop | Linuturk, use sox |
| 17:46.36 | bmoraca | i'm not having this argument. i don't argue religion in online forums. linux is nowhere near as efficient as windows as far as productivity goes. |
| 17:46.42 | tzafrir_laptop | to convert them to/from wav |
| 17:47.00 | [TK]D-Fender | Sure they do... when paired with the matching 'crashing' 'deficient' and 'loss' |
| 17:47.01 | Corydon76-dig | bmoraca: you just contradicted yourself |
| 17:47.03 | tzafrir_laptop | well, sox doesn't support ilbc. But should support all others |
| 17:47.23 | Linuturk | which format should I convert to the wav? |
| 17:47.28 | tzafrir_laptop | Linuturk, and there's naturally 'file convert' in the asterisk CLI |
| 17:47.30 | Corydon76-dig | bmoraca: if you're not going to be party to an argument, then don't state a position |
| 17:47.35 | KaneHau | of course, it depends on what you are producting |
| 17:47.37 | KaneHau | producing |
| 17:47.46 | Naikrovek | bmoraca: we agree, and yes, the argument is endless because (it seems) linux users don't know what windows users want productivity-wise. look at gnome or kde or any other window manager and tell me any of those people understand productivity. har. |
| 17:47.51 | tzafrir_laptop | Linuturk, also: what type of editing do you want to do? |
| 17:48.11 | Linuturk | I'm wanting to change a portion of the file |
| 17:48.18 | Linuturk | replace a sentance or two |
| 17:48.36 | Corydon76-dig | Naikrovek: You're just making a fool of yourself, if you're going to compare productivity in a vacuum |
| 17:49.30 | Corydon76-dig | Naikrovek, bmoraca: productivity is a statement of efficiency. You're ignoring the training necessary to get someone up to speed on a platform. |
| 17:49.33 | Naikrovek | Corydon76-dig: i don't want to talk about it either. i'm a linux user of over a decade and a windows user of over 20 years, so i'm not really talking out of my ass here. for me, and many others, windows is better, end of story. for others the opposite is true |
| 17:50.05 | tzafrir_laptop | Linuturk, see 'help file convert' in the asterisk CLI |
| 17:50.09 | Corydon76-dig | Naikrovek: FOR YOU. This is the key phrase. |
| 17:50.17 | Naikrovek | Corydon76-dig: i never said it was better for anyone else |
| 17:50.30 | Corydon76-dig | Naikrovek: bmoraca said something to the contrary |
| 17:50.42 | chuckf | wonders if we could have asterisk without Linux |
| 17:50.43 | Naikrovek | Corydon76-dig: i said that linux users and developers don't seem to understand what makes windows users efficient on windows |
| 17:51.03 | Naikrovek | chuckf: it runs on *bsd and there's a probably crappy port for windows |
| 17:51.17 | voipmonk | chuckf: what do you mean? |
| 17:51.24 | Linuturk | you guys need to discuss the linux/windows thing elsewhere |
| 17:51.40 | Naikrovek | says the linux turk |
| 17:51.46 | Naikrovek | i'm done with the topic anyway |
| 17:52.29 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 17:53.48 | voipmonk | chuckf: you can try to port it to the Killer network card ( http://www.pcper.com/article.php?type=expert&aid=379 ) |
| 17:54.28 | Chainsaw | voipmonk: They still sell that thing? |
| 17:54.33 | Chainsaw | voipmonk: I thought it fizzled out years ago. |
| 17:54.38 | voipmonk | hehe |
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| 18:22.18 | oktay | voipmonk: you won't believe what the problem was |
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| 18:23.28 | jaytee | dental floss stuck in the flux capacitor? |
| 18:23.59 | oktay | ewww |
| 18:24.08 | oktay | my flux capacitor is all organic |
| 18:25.27 | jaytee | impossible! only polybendum reinforced unobtanium can withstand the stresses of temporal distortion field generation |
| 18:25.54 | oktay | in that case we need a locomotive |
| 18:25.58 | oktay | or a watch tower |
| 18:27.16 | jaytee | I scrolled back aways but I can't remember what problem you were having. |
| 18:27.50 | oktay | i was asking nonsensical questions about a setup that didn't make sense |
| 18:27.54 | oktay | that's what i was told anyway |
| 18:27.57 | oktay | ( : |
| 18:28.45 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 18:29.45 | [TK]D-Fender | oktay: By one person anyway. |
| 18:29.51 | oktay | :) |
| 18:31.02 | jaytee | oh, was he the one with the SPA-3102? |
| 18:31.29 | oktay | yeah. |
| 18:32.14 | Katty | HELLO LOVIES |
| 18:32.17 | Katty | I HAS CAFFEINATED |
| 18:33.52 | voipmonk | lol!!!! |
| 18:33.59 | carrar | I HAS ALSO |
| 18:34.05 | jaytee | me too! |
| 18:35.16 | jaytee | was just readin about how the Burj Dubai was built by low-wage migrant workers from the Indian sub-continent. funny, back when they built the pyramids they used to call them "slaves". |
| 18:35.41 | carrar | Who wants to work in that building |
| 18:36.02 | jaytee | I wouldn't |
| 18:36.05 | Naikrovek | well they found evidence that they weren't even slaves; they worked for their god of their own free will, some evidence suggests. |
| 18:36.10 | Naikrovek | though i don't have a reference |
| 18:36.11 | jaytee | but then I wouldn't want to work anywhere in the middle east |
| 18:36.47 | Naikrovek | how about eastern east or western east. northern east? southern west east? |
| 18:37.45 | jaytee | I'd like to work in the northernmost section of the southern part of the west coast of Australia |
| 18:38.12 | Naikrovek | i used to work on the south east coast of the east part of australia for 2 years and loved it |
| 18:38.29 | jaytee | Melbourne or Syndey? |
| 18:38.44 | Naikrovek | ah that is a bit ambiguous huh. Sydney. You want Perth it sounds |
| 18:38.58 | Naikrovek | like |
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| 18:39.24 | jaytee | anyplace far away from the middle east although I've got nothing against Israel the rest of the countries over there suck ass |
| 18:39.33 | [TK]D-Fender | turns up the juice on his South-by-Southwest Homing Pigeon Disruptor and warms up the stove.... |
| 18:43.40 | jaytee | News bulletin!!: A large flock of disoriented and lost homing pigeons have taken up temporary residence on the Capitol dome in Washington, D.C. While scientists are struggling to find the cause of this White House officials are working around the clock to arrange a bailout package for them. |
| 18:43.43 | Katty | hovers over said stove. |
| 18:44.00 | Katty | ahah ahhahaaaa |
| 18:44.05 | Katty | <3 jaytee |
| 18:44.10 | Katty | jaytee: i love you. |
| 18:44.11 | jaytee | :-) |
| 18:44.17 | jaytee | i love you too! |
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| 18:46.37 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 18:46.57 | jaytee | Oprah Winfrey as said that it's "OK" to give your server a 10% tip because of these tough economic times. Someone should tell that fat assed rich bitch to shut the hell up or make up the difference. |
| 18:47.45 | oktay | hah. i'm so glad I don't have to pay a %18 base tip for lousy service and shitty food anymore :) |
| 18:49.41 | Naikrovek | i never pay a tip unless they earn it |
| 18:49.50 | Naikrovek | they usually do, but sometimes they do not |
| 18:50.07 | kfife | Anybody else getting this errror compilign 1.6.2.0 - incorrectly believes that libxml2 is not installed. |
| 18:50.17 | oktay | Naikrovek: you don't live in the US? |
| 18:50.19 | mykhyggz | Naikrovek: you stiffed a waitstaffer? |
| 18:50.23 | Naikrovek | like when i have to tell them 3 times to refill my empty soda or that we need extra napkins |
| 18:50.33 | kfife | Installed libxml2 using centos yum repositories. |
| 18:50.35 | Naikrovek | there's no law in the US that one MUST tip |
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| 18:50.46 | [TK]D-Fender | kfife: check for the devel |
| 18:50.52 | Naikrovek | libxml2-dev |
| 18:51.03 | oktay | Naikrovek: people have been sued for not tipping |
| 18:51.08 | bmoraca | <PROTECTED> |
| 18:51.11 | Naikrovek | let them try and sue me for not tipping |
| 18:51.12 | mykhyggz | Naikrovek: there's no guarantee you don't get spit in your food either. |
| 18:51.20 | kfife | duhh. So obvious in retrospect. SOrry. Brainfart. |
| 18:51.22 | Naikrovek | mykhyggz: actually there is |
| 18:52.19 | Naikrovek | the only ones i don't tip are the ones that do not deserve a tip. everyone else, from mildly acceptable to great get a tip depending on how well they did |
| 18:52.40 | oktay | how about when gratuity is included? |
| 18:53.10 | Katty | wow, 10% tip? |
| 18:53.15 | Katty | not cool. |
| 18:53.26 | Naikrovek | i've never eaten at a restaurant or eaten with a large enough group to get that BS thrown at me |
| 18:53.45 | Katty | oktay: they will take it off at request. |
| 18:53.51 | oktay | you've never eaten at a restaurant? :) |
| 18:53.58 | Katty | oktay: usally under the asumption you're going to tip more. |
| 18:54.10 | oktay | you can tip more if you want |
| 18:54.14 | oktay | why have it taken out? :) |
| 18:54.18 | Katty | ryan and i went out thursday night for new years. |
| 18:54.27 | Katty | we spent.. 30 on food, 30 on drinks, and tipped 20 |
| 18:54.50 | Naikrovek | if i get a great waiter/ress i will tip 50% of the bill |
| 18:55.02 | Naikrovek | if i get one that can't keep my soda full after three trips by my table then they get nothing |
| 18:55.18 | Katty | i never run out of soda. |
| 18:55.23 | Katty | but i don't drink a lot when i eat. |
| 18:55.24 | Naikrovek | i rarely do |
| 18:56.02 | oktay | that's great. |
| 18:56.09 | oktay | money can be better spent IMO |
| 18:56.14 | Naikrovek | but when i do i am not happy about it. it's not hard to fill a glass when you're not busy. and every time i've had an empty glass i've heard the staff in the back joking around and wasting time while i sit there thirsty |
| 18:56.20 | oktay | depends on where you live too i guess. |
| 18:57.19 | Katty | oktay: we don't go out often. |
| 18:57.27 | Katty | oktay: maybe once a month. |
| 18:57.40 | Katty | oktay: well, at least to dinner. we do go out often. |
| 18:57.41 | oktay | cool |
| 18:57.47 | Naikrovek | one time, my waitress sat at the table right behind me and caught up with a college friend, i sat there without any silverware or napkins while my wife and daughter ate. i spoke with the manager after that one and not only did I not tip I didn't pay for any of the food |
| 18:57.49 | Katty | mostly to the theater. |
| 18:57.57 | bmoraca | biggest tip i ever gave was $65...course, it was a $250 meal for my wife and me...and the waiter was very attentive |
| 18:58.02 | oktay | Naikrovek: i think i can top that |
| 18:58.08 | Naikrovek | not hard really |
| 18:58.11 | oktay | :) |
| 18:58.15 | Katty | bmoraca: where was that? |
| 18:58.18 | Naikrovek | but i never went back so i never will get a chance to top it |
| 18:58.26 | bmoraca | Katty, Morton's Steakhouse |
| 18:58.29 | Katty | bmoraca: the most i've spent on the two of us was 150ish at a japanese steakhouse. |
| 18:58.39 | oktay | had some desert at a cafe with two friends. we are what you might call "alien" so we were speaking our own alien language. |
| 18:58.40 | Katty | bmoraca: hmm. never heard of it. |
| 18:58.48 | oktay | so he takes us for tourists (which I wasn't) |
| 18:58.55 | bmoraca | I think they're a west-coast thing...but they're damn good |
| 18:58.55 | Naikrovek | :) |
| 18:59.10 | bmoraca | Mama's Fish House on Maui was almost as much, but not quite :) |
| 18:59.10 | oktay | brings the check. folds it. puts more than 20% on it as tip and writes down the total. |
| 18:59.17 | Naikrovek | grr. |
| 18:59.22 | oktay | went to the manager. the waiter got 0 tip. |
| 18:59.27 | Naikrovek | yes! |
| 18:59.39 | Katty | the best steak i've ever had was at a little lodge in van buren missouri. the type of place where you rent a cabin for a week and go floating down the river. |
| 18:59.41 | Naikrovek | they pulled that on me in hong kong a few times |
| 18:59.47 | oktay | he was a foreigner himself by the way. weird. |
| 18:59.48 | oktay | :) |
| 18:59.52 | Katty | well the lodge had a resturant, with some amazing ny strip steak. |
| 19:00.08 | Katty | there were like... 6 tables in the whole building. |
| 19:00.16 | Katty | teeny tiny little place out in the middle of /nowhere/ |
| 19:01.02 | *** join/#asterisk axelilly (n=jfenner@66.181.75.69) |
| 19:01.08 | Naikrovek | i was roomies with some waiters/resses when i lived in australia, they told me that if they don't make $50/hr or more its because they don't need it. they can get that with their eyes closed and half asleep apparently. i don't feel bad when i don't leave a tip, so long as they don't deserve it |
| 19:01.26 | axelilly | Does anyone know what the W: indicates in the output of this command: queue show |
| 19:01.39 | axelilly | I checked google, but I can't seem to locate that information. |
| 19:02.12 | oktay | Naikrovek: around NYC they rarely deserve the tip. They know that they will at least get 15%.. and they are only waiting tables until they become famous anyway.. |
| 19:02.15 | oktay | so they don't care :) |
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| 19:02.48 | oktay | hereabouts waiters don't even expect a tip at most places |
| 19:02.56 | Naikrovek | lol! couple of germans walked in |
| 19:03.00 | oktay | and if you tip generously they make you feel like a king |
| 19:03.00 | Naikrovek | love hearing those dudes talk |
| 19:03.05 | Naikrovek | they're so energetic |
| 19:03.14 | kfife | Sample config files: (Ugrading 1.6.0 to 1.6.2). I like to review the latest version-generated sample files. Naturally "Make samples" blows away my configs unless I copy them, generate samples, then restore my configs Is there a make option to place the sample config files elsewere so as to eliminate those "copy, generate, restore" steps? I assume the sample files are generated dynamically (& therefore I can't just pull them out of the source t |
| 19:03.49 | [TK]D-Fender | kfife: there is a blatant folder you should kick yourself for not looking at |
| 19:04.25 | Naikrovek | i will tip generously if they keep our soda filled and if i get everything i ask for in short order (extra napkins, whatever my wife and daughter want) and they don't check on me if everything is okay. (if everything weren't okay I'll find you) |
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| 19:04.55 | oktay | Naikrovek: so i take it you don't like it when they try to take away your half full plate :) |
| 19:05.07 | Naikrovek | no one has tried that yet |
| 19:05.10 | Katty | wait, what? |
| 19:05.15 | Naikrovek | i'm a big dude, they assume i'll finish it usually |
| 19:05.16 | Katty | people never take my plate away ^_- |
| 19:05.43 | oktay | i guess that a local thing then :) |
| 19:05.54 | Katty | sounds like it |
| 19:05.56 | oktay | here in barbarian land |
| 19:06.03 | Naikrovek | they just want you out so they can sucker a tip out of the next guy |
| 19:06.03 | Katty | where's barbarian land? |
| 19:06.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:06.07 | oktay | istanbul |
| 19:06.12 | Katty | oh. |
| 19:06.13 | Katty | hmm. k |
| 19:06.19 | Katty | wouldn't know about out there. |
| 19:06.28 | Naikrovek | mmm turkey sandwiches |
| 19:06.35 | jaytee | gobble gobble |
| 19:06.43 | oktay | there's a plate fetish here. nice restaurants. they change your plates like 4-5 times during a meal. |
| 19:07.01 | KaneHau | what if your not done eating |
| 19:07.32 | oktay | oh this plate changing thing is when you're done with one dish |
| 19:07.48 | oktay | they take away everything including the serving plate, fork, knife |
| 19:07.52 | oktay | you get clean ones |
| 19:07.59 | KaneHau | hate to be the dishwasher |
| 19:08.04 | *** part/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 19:08.09 | oktay | labor is cheap |
| 19:10.04 | axelilly | I answered my own question by reading the source code for app_queue.c. The answer is that the W: indicates the weight of the queue. |
| 19:10.07 | Katty | i'm sure they have high end dish washers at a place like that |
| 19:10.11 | Katty | and i don't mean people. |
| 19:10.25 | oktay | perhaps |
| 19:10.45 | *** join/#asterisk naxxfish (n=fish@barney.naxxfish.eu) |
| 19:11.11 | naxxfish | does users.conf have any place in a system without Asterisk GUI installeD? |
| 19:12.22 | oktay | best iphone sip client ? |
| 19:12.26 | [TK]D-Fender | naxxfish: Not really. |
| 19:12.35 | [TK]D-Fender | naxxfish: Nothing yuo can't do better individually. |
| 19:12.39 | kfife | [TK]D-Fender: Got it. I had searched for it calling it out the same way the makefile did "samples' rather than ...conf.sample (singular). I was so close. Thanks! |
| 19:12.54 | kfife | [TK]D-Fender: not generated dynamicaly. Thanks |
| 19:13.02 | [TK]D-Fender | kfife: You may now proceed to kick yourself :) |
| 19:13.53 | kfife | I already did ('booted myself in the a$$'), Shall I re-boot? :-) |
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| 19:16.24 | oktay | ok siphone works |
| 19:17.04 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 19:18.14 | blitzrage | kfife: enable loop mode! |
| 19:18.46 | oktay | later boys & girls. puppy wants out. |
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| 19:21.06 | Naikrovek | blitzrage: when is edition 3 gonna be done? just bought 2nd ed then i read on your blog that you're about to finish 3rd. |
| 19:21.22 | blitzrage | I don't think I said we were about to finish the 3rd :) |
| 19:21.24 | blitzrage | I said we had started it. |
| 19:21.35 | blitzrage | Expect in the time frame of several months |
| 19:21.38 | Naikrovek | ah well you gave a timeline of a few months or something |
| 19:21.46 | Naikrovek | i assume that's near completion |
| 19:21.50 | blitzrage | nope |
| 19:21.50 | Naikrovek | fair enough |
| 19:22.25 | Katty | mister madsen. |
| 19:23.13 | kfife | blitzrage: ORA-01031 host kfife loop mode command rejected: insufficent priviliges :-) |
| 19:23.36 | naxxfish | [TK]D-Fender: thing is we configured this thing with asterisk-gui (not that I wanted to) and now we're going manual - is there any way to make users.conf work? |
| 19:24.14 | Katty | so now that you're doing it the proper way |
| 19:24.18 | Katty | you don't want to do it properly? |
| 19:24.24 | Katty | that doesn't make much sense, naxxfish |
| 19:24.41 | bcrisp | naxxfish, dont use users.conf |
| 19:24.55 | naxxfish | i'd rather not but all our config is in it |
| 19:25.13 | naxxfish | asterisk-gui put all the sip details into it |
| 19:25.21 | bcrisp | put them into sip.conf |
| 19:25.27 | [TK]D-Fender | naxxfish: It doesn't just stop working... |
| 19:25.46 | naxxfish | it didn't seem to |
| 19:25.46 | [TK]D-Fender | naxxfish: Port the appropriate bits to their respective configs |
| 19:26.28 | naxxfish | the thing is i think i've removed something vital in extensions.conf and now it won't dial the extensions that are defined in that file |
| 19:26.54 | naxxfish | just getting extension not found |
| 19:27.01 | [TK]D-Fender | naxxfish: then go make some more |
| 19:27.15 | naxxfish | more what/ |
| 19:28.29 | Katty | 13:33 < naxxfish> just getting extension not found |
| 19:28.34 | Katty | 13:33 < [TK]D-Fender> naxxfish: then go make some more |
| 19:28.45 | Katty | find the noun. |
| 19:28.59 | [TK]D-Fender | GRAMMAR RANGERS UNITE!!! |
| 19:29.12 | voipmonk | what about grampar? :) |
| 19:29.18 | Katty | your mom. |
| 19:29.20 | voipmonk | no one helps the old man across the street |
| 19:30.09 | Katty | that's because the old man is cranky and likes to hit people with his cane when they touch the mailbox. |
| 19:30.32 | naxxfish | the extensions existed before, and i'm pretty sure they were created from users.conf |
| 19:30.48 | naxxfish | (rather than defined explicitally in extensions.conf) |
| 19:31.04 | [TK]D-Fender | naxxfish: PERHAPS THINGS AREN'T POINTING WHERE THEY USED TO... |
| 19:31.14 | Katty | adjust [TK]D-Fender's volume setting. |
| 19:31.23 | Nivex | Katty: you can do that? |
| 19:31.35 | [TK]D-Fender | 's is set to 11. That's 1 more.... |
| 19:31.36 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 19:34.00 | naxxfish | hmm, seem to have got it working now, possibly |
| 19:36.49 | *** part/#asterisk wam (i=wam@unaffiliated/wam) |
| 19:47.00 | zippytech | asterisk best codecs |
| 19:47.09 | zippytech | what is |
| 19:47.21 | zippytech | free or paid |
| 19:47.22 | x86 | anyone ever use realtime for 'sipregs'? |
| 19:47.33 | x86 | zippytech: depends on your application... |
| 19:47.41 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 19:47.45 | x86 | local users, unlimited bandwidth --> g722 |
| 19:49.38 | yang | Hi ! Is there another way to avoid syntax error if my username is "bluesip/janprunk" ? |
| 19:49.48 | tzafrir_laptop | zippytech, ulaw/alaw, if you have the bandwidth (ignoring wideband) |
| 19:51.02 | voipmonk | http://forums.digium.com/viewtopic.php?p=45124&sid=5093342ae8d696fd5c211746aaa6dfa8 |
| 19:52.04 | zippytech | we have 6 phones olycom 650 , that make a robot burrp sound every few words, no network traffic for say 3.0 gig ram system |
| 19:52.51 | [TK]D-Fender | zippytech: And you've told us SO much more than last time... |
| 19:53.38 | zippytech | sorry |
| 19:54.02 | kfife | anyone had reliable results using 1.6.x and t.38 fax with a t.38 ITSP? |
| 19:54.19 | kfife | There are some copper loops I would like to drop. |
| 19:54.39 | Qwell | ~faxforasterisk |
| 19:54.40 | infobot | well, faxforasterisk is Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 19:54.44 | Qwell | kfife: ^^ |
| 19:57.17 | kfife | Qwell: Yes indeed. I'm a user, and have purchased licenses etc. I'm wondering if anyone has had reliable results with an ITSP who will relay a fax via t.38 to an asterisk machine, who can then send it to a "familiar-to-end-users" fax machine/appliance. |
| 20:08.58 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:09.21 | *** join/#asterisk RobH (n=RobH@cpe-173-169-30-118.tampabay.res.rr.com) |
| 20:10.48 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 20:12.46 | *** join/#asterisk bahjons (n=robert@140.99.23.26) |
| 20:14.24 | bahjons | http://lists.digium.com/pipermail/asterisk-users/2010-January/242838.html - Can someone provide some help with this one? |
| 20:18.55 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 20:20.34 | Katty | munches popcorn |
| 20:23.07 | beek | Hello Katty ! |
| 20:25.16 | [TK]D-Fender | bahjons: Go try |
| 20:27.47 | bahjons | [TK]D-Fender: yea, just finished up trying ton 1.6.2. Nothing good seems to come from realtime. Though it has huge potential |
| 20:27.51 | ManxPower-work | kfife: Since this is a purchased product I recommend contacting Digium support. |
| 20:33.46 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
| 20:34.37 | Skeeter- | Anyone got a bunch of ringtones for the polycoms phone?? |
| 20:34.53 | raden_work | have not got |
| 20:36.53 | raden_work | how can i stream audio to an extension ? |
| 20:36.57 | *** join/#asterisk DrGeek (n=geek@c-24-21-244-173.hsd1.or.comcast.net) |
| 20:36.59 | DrGeek | g'day all! |
| 20:37.04 | raden_work | like extension 999 internet radio |
| 20:37.09 | raden_work | g'day |
| 20:37.35 | DrGeek | I'm scratching my head about accessing voicemail: When I dial *97 or *98 it just sits there waiting, and then times out. |
| 20:37.58 | DrGeek | its like it is waiting for additional digits |
| 20:38.17 | beek | DrGeek: What is the "it" that just sits there waiting? |
| 20:38.38 | DrGeek | The sipphone. No tone, and eventually it times out and gives an error sound. |
| 20:38.50 | beek | Have you tried hitting "send"? |
| 20:39.07 | *** join/#asterisk _Raptor_ (i=raptorbl@131.188.30.242) |
| 20:39.10 | DrGeek | heh, yeah. I an dial other extensions fine |
| 20:39.11 | ManxPower-work | What model of SIPPhone? I don't know if the SIPPhone company even has more than once model. In any case, I suspect it's the dialplan of the phone. |
| 20:39.28 | ManxPower-work | Why did you not go with a well known company for your phones? |
| 20:39.50 | DrGeek | ManxPower-work, avaya 4410 |
| 20:39.54 | ManxPower-work | DrGeek: did you set up a *97 or *98 extension? |
| 20:40.03 | ManxPower-work | DrGeek: Then why did you not say that when asked? |
| 20:40.15 | DrGeek | ManxPower-work, there are exten => *97... entries |
| 20:40.20 | DrGeek | It's the FreePBX 2.5 config |
| 20:40.40 | ManxPower-work | DrGeek: It sucks to be you. |
| 20:40.57 | ManxPower-work | DrGeek: Is *97 allowed in the Avaya phone dialplan? |
| 20:41.10 | ManxPower-work | ~freepbx |
| 20:41.11 | infobot | [~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, or fpbx |
| 20:43.34 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 20:46.06 | DrGeek | ManxPower-work, as far as I can tell, the avaya phones do not have a dialplan internally. the tftp config is pretty raw. |
| 20:46.33 | ManxPower-work | DrGeek: All IP phones have internal dialplans |
| 20:46.54 | ManxPower-work | Sorry, all SIP phones have internal dialplans. MGCP phones don't. |
| 20:47.11 | DrGeek | ManxPower-work, well that could be it then. |
| 20:47.12 | beek | DrGeek: You could always use a softphone to test your theory: is it the phone or is it FreePBX? |
| 20:47.28 | DrGeek | beek, yeah I'm not sure... I think thats the next test. |
| 20:49.26 | ManxPower-work | If it was freepbx you'd see stuff on the console. |
| 20:49.36 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.12.111) |
| 20:50.13 | DrGeek | ManxPower-work, thats what I thought too... we do see asterisk report *98 being dialed from the sipphone. |
| 20:50.38 | *** join/#asterisk brezular (n=brezular@adsl-dyn219.95-103-201.t-com.sk) |
| 20:51.22 | ManxPower-work | DrGeek: Then it's time to go to #FreePBX |
| 20:51.42 | DrGeek | ManxPower-work, thanks. |
| 20:53.32 | *** join/#asterisk wawl (n=wawl@85-120.79-83.cust.bluewin.ch) |
| 20:54.18 | ManxPower-work | DrGeek: you might want to pick a better phone in the future |
| 20:54.56 | DrGeek | ty, ManxPower-work . It was the sip internal dialplan. |
| 20:55.06 | DrGeek | it works with a softphone. |
| 20:55.06 | *** part/#asterisk bahjons (n=robert@140.99.23.26) |
| 20:57.43 | Akiraa | Has anyone integrated number portability into Asterisk for call cost minimization? |
| 20:59.22 | ManxPower-work | Akiraa: What protocol would be used to look up the carrier for a number |
| 20:59.35 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 21:01.10 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 21:01.55 | *** join/#asterisk davix (n=reachout@89-138-192-51.bb.netvision.net.il) |
| 21:02.48 | Akiraa | ManxPower-work: I have no idea, the best I can come up with is: 1.Assume no numbers are ported 2.When making a call on the same PSTN carrier, if there is a distinct tone sequence at the beginning (first 3-5 second), assume ported number. 3. Access a web interface to identify the carrier of the ported number. 4.Next time a call is made to the identified ported number, use the appropriate carrier line. |
| 21:03.45 | Akiraa | or 5.Try carriers in sequence until you have found the identity of the ported number (this works because the first few seconds of the tone are not taxed) |
| 21:06.42 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 21:07.05 | ManxPower-work | I was not aware carriers played "tones" for ported numbers. |
| 21:09.19 | Akiraa | ManxPower-work: there is an audible warning when leaving the local carrier which can be potentially exploited |
| 21:09.49 | Akiraa | do yo have some general thoughts about capturing and redirecting a call based on that? |
| 21:10.02 | Akiraa | or hints etc |
| 21:10.55 | Katty | i has popcorn in my toofs. |
| 21:11.49 | ManxPower-work | Akiraa: you must not be in the USA |
| 21:12.00 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 21:12.04 | Akiraa | ManxPower-work: EU based |
| 21:13.18 | Akiraa | there are some public portability web services available, but they're not robust enough to use for each and every call |
| 21:13.37 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 21:16.05 | Akiraa | MansPower-work: this is the actual tone sequence: http://www.portabilitate.ro/content/sound/beep.wav |
| 21:16.15 | bmoraca | Akiraa, i would check with your wholesaler first. I'd imagine that if anyone had an API for you to use, it would be them |
| 21:18.25 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 21:18.52 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:19.13 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 21:19.29 | *** join/#asterisk jtodd (i=hkiy4yet@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 21:19.29 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:26.05 | Katty | hmm. no one's in the front yard. how unusual. |
| 21:26.15 | Katty | oh wait, nevermind. |
| 21:27.08 | Katty | omnomnomnomnom |
| 21:31.25 | bpgoldsb | Trying to compile both Asterisk 1.6.1.12 and 1.6.2 failed on compiling chan_agent with a bunch of undefined references. Anyone seen this kind of behavior? |
| 21:31.55 | ManxPower-work | bpgoldsb: what references? |
| 21:32.43 | bpgoldsb | ManxPower-work: mostly ast_* functions |
| 21:33.54 | bpgoldsb | Ya, I don't see anything that looks like it can't find an external library. |
| 21:35.21 | ChannelZ | but are you sure the failures you're seeing are the actual cause? (IE something failed to compile earlier, but you're seeing linking issues at the end?) |
| 21:36.07 | ManxPower-work | i'd pastebin the whole output |
| 21:36.49 | ManxPower-work | ~pb |
| 21:36.50 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 21:37.39 | bpgoldsb | ManxPower-work: http://pastebin.org/71149 |
| 21:39.32 | ChannelZ | that's not the whole thing - I suspect something else failed earlier |
| 21:39.50 | bpgoldsb | Sorry, I'll get the entire thing. |
| 21:40.05 | ChannelZ | like whatever module has all of the ast_ stuff in it |
| 21:40.20 | ManxPower-work | bpgoldsb: I doubt *I* will be able to help, but someone might |
| 21:41.08 | bpgoldsb | http://pastebin.org/71150 (thats got everything). |
| 21:43.30 | ChannelZ | hmm wierd |
| 21:44.16 | ChannelZ | ARGH this pastebin.org has a fucking onclick pop-up |
| 21:46.04 | ChannelZ | are you building this from a packaged source? |
| 21:46.27 | bpgoldsb | No, I'm attempting to create a package. |
| 21:46.48 | bpgoldsb | But it was happening under non-packaging too |
| 21:47.11 | Qwell | bpgoldsb: mandrake? |
| 21:47.14 | bpgoldsb | Debian |
| 21:48.22 | Qwell | you're overriding compiler flags.. |
| 21:49.12 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 21:49.14 | *** join/#asterisk tgrman (n=jcmoore@unaffiliated/tgrman) |
| 21:50.09 | bpgoldsb | I imagine the problems on my end. I'll muck around until I can figure it out, then. |
| 21:51.10 | ChannelZ | yah there's some fudging happening |
| 21:51.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:51.46 | ChannelZ | Hey Qwell I have a ? if you have a sec |
| 21:51.52 | Qwell | ? |
| 21:52.49 | *** join/#asterisk lenne_dk (n=leif@0x573cc07b.odnxx13.dynamic.dsl.tele.dk) |
| 21:53.05 | ChannelZ | I don't know if you remember I was having issues with gsm->ulaw transcoding which was a compiler optimization bug |
| 21:53.59 | Qwell | and? |
| 21:54.08 | ChannelZ | sorry was trying to find the right bug number |
| 21:54.58 | ChannelZ | In regards to https://issues.asterisk.org/view.php?id=16516 |
| 21:55.48 | ManxPower-work | ~gsmbug |
| 21:55.48 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 21:55.58 | ChannelZ | There was a code change that was supposed to fix it but it doesn't under gcc 4.2 - so I guess my question is, should I open another issue? I'm not really sure whose bug it is considered, asterisks' or GCC's at the end of the day |
| 21:56.16 | Qwell | gcc |
| 21:56.53 | Qwell | the patch was a guess, based on a fix for a similar issue, for another codec |
| 21:57.02 | lenne_dk | Howdy. is page() available in 1.6? there is an app_page.c, but no other *page* is found after compiling under freebsd |
| 21:57.03 | Qwell | all the patch did was work around the issue |
| 21:57.42 | *** join/#asterisk lynxsys (n=thelynx@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 21:57.48 | lynxsys | Hey all |
| 21:57.53 | ChannelZ | Although it now doesn't :) (the 'maybe-asm' one I'm talking of) |
| 21:58.13 | Qwell | ChannelZ: it was never confirmed that it did |
| 21:58.49 | lynxsys | has anyone had a problem with freepbx and cisco 7941 attended transfer not completing? |
| 21:59.24 | bcrisp | ~freepbx |
| 21:59.25 | infobot | [~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, or fpbx |
| 21:59.32 | *** join/#asterisk chazzm (n=chazz@173-24-238-25.client.mchsi.com) |
| 22:00.00 | ChannelZ | OK - my question really is, should I talk to tilghman directly about it, as he seems to know a little about what is going on and GCC's optimization? I don't know that I could report it to the GCC maintainers myself as I have no idea about the asm code in the first place :) |
| 22:00.02 | bmoraca | lynxsys, Cisco 7941 phones are VERY, VERY finicky with Asterisk. i wouldn't recommend trying to use them. |
| 22:00.30 | ChannelZ | or what would be the preferred way to proceed |
| 22:00.33 | Qwell | ChannelZ: 4.2 is dead to gcc. they don't care anymore. it's fixed |
| 22:00.33 | lynxsys | I have the phone working well just this one niggle |
| 22:01.00 | bmoraca | lynxsys, my point exactly |
| 22:01.25 | ChannelZ | oh in a later rev then.. okey dokey |
| 22:01.43 | lynxsys | I see |
| 22:02.05 | lynxsys | what phone would you recommend? |
| 22:02.09 | ChannelZ | (I'm just using the makefile patch to turn down optimization on gsm.c which works fine) |
| 22:03.12 | bmoraca | lynxsys, i'm partial to Polycoms. IP330s and IP550s are great phones. if you're on more of a budget, you can't go wrong with IP501s, either. or, if you're dead-set on Cisco (as my boss is), you can always use the 7940s |
| 22:03.57 | lynxsys | I have a polycom IP430 and tbh I find the cisco's easier to setup |
| 22:04.02 | ChannelZ | lenne_dk: I have Page() ... ? |
| 22:04.32 | bmoraca | lynxsys, Polycom's phones are very easy to set up once you've got the initial config file out of the way |
| 22:05.18 | lynxsys | I will keep playing with it, is the speaker phone as good as the cisco phones? |
| 22:05.39 | lenne_dk | core show application page |
| 22:05.59 | lenne_dk | Not installed. |
| 22:07.10 | ChannelZ | module load app_page |
| 22:07.51 | bmoraca | lynxsys, better. |
| 22:09.56 | lynxsys | whats the main advantage of running asterisk without a freepbx/trixbox web gui?\ |
| 22:10.20 | Katty | birdy birdy everywhere |
| 22:10.20 | bmoraca | lynxsys, flexibility and the fact that everything isn't obfuscated...WAY easier to troubleshoot |
| 22:10.56 | lenne_dk | ChannelZ: It seems I have disabled it in menuselect or something. make config doesn't show this in FreeBSD. |
| 22:11.05 | bpgoldsb | lynxsys: It greatly depends on what you plan on doing with your Asterisk. Trixbox can be nice because it makes it easy, or aweful because managers want crazy features that can't be fit in well with Trixbox. |
| 22:12.27 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 22:12.33 | *** join/#asterisk rizwank (n=rizwank@76.89.131.47) |
| 22:12.43 | lynxsys | is Trixbox generally seen as a better soloution than freepbx? |
| 22:12.50 | Qwell | ~trixbox |
| 22:12.51 | infobot | well, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 22:13.00 | bmoraca | trixbox uses freepbx and a lot more other crap |
| 22:13.50 | rizwank | We've got an AsteriskNow box, a homegrown centos box, and I think a freepbx box - we're going to consolidate onto a few virtual servers. Is there a distro that's 1.6 that is recommended and well maintained - and that means I have to spend less time making sure settings are similar between the different machines? Is that AsteriskNow? |
| 22:13.56 | lynxsys | is there a steep learning curve from freepbx to a clean asterisk distro |
| 22:14.24 | Qwell | rizwank: You can install Asterisk 1.6 on AsteriskNOW |
| 22:14.36 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 22:14.50 | Qwell | rizwank: See the channel topic in #asterisknow - there's a link that gives instructions. |
| 22:15.11 | rizwank | would that be your recommended VM? or just homegrown CentOS? |
| 22:15.21 | Qwell | I have a slight bias. |
| 22:15.40 | ChannelZ | lenne_dk: look in your menuconfig and see if it's XXX'd out for a dependency |
| 22:16.00 | VxJasonxV | How often should Asterisk be doing SRV lookups for external SIP registrations? |
| 22:16.06 | Qwell | rizwank: if you want the packages on a "standard" CentOS install, without the FreePBX stuff, that's also possible. |
| 22:16.21 | lenne_dk | menuconfig say app_page depends on dahdi. |
| 22:16.23 | VxJasonxV | external = my asterisk instance doing a lookup to another asterisk (or any SIP) server. |
| 22:16.33 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 22:16.48 | ChannelZ | lenne_dk: yah as it's MeetMe-based and that requires a timing source |
| 22:18.24 | lenne_dk | ChannelZ so where do I find such a timing source? |
| 22:19.29 | ChannelZ | in dahdi |
| 22:19.53 | ChannelZ | but for freebsd.. I think you are SOL? |
| 22:20.22 | rizwank | any advantage/disadvantage of using three CentOS boxen versus three AsteriskNow boxen? |
| 22:20.24 | lenne_dk | Darn... |
| 22:20.39 | voipmonk | why do you believe you need 3? |
| 22:20.41 | ChannelZ | dahdi_dummy is a kernel driver |
| 22:22.13 | Qwell | rizwank: Do you need/want FreePBX? |
| 22:22.34 | rizwank | the GUI? Um - I'd rather not, it'd keep my telecom engineer honest, but I wouldn't like to close the door to it. |
| 22:23.13 | Qwell | rizwank: then yes :p |
| 22:23.39 | Qwell | but, like I said, you can install "AsteriskNOW" without all that stuff. Just install the repo file, and install only the Asterisk packages |
| 22:23.57 | Qwell | http://packages.asterisk.org/centos/5/current/x86_64/RPMS/asterisknow-version-1.5.0-1_centos5.noarch.rpm |
| 22:24.01 | Qwell | (or i386. whichever) |
| 22:24.18 | *** join/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 22:25.37 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 22:25.58 | rizwank | hmm. |
| 22:26.16 | rizwank | AsteriskNow repos keep up with each point release of asterisk - 1.61 and the like, or is it significantly lagging. |
| 22:26.33 | Katty | wow. rizwank. |
| 22:26.35 | Qwell | 1.6.0.latest is available |
| 22:26.39 | Katty | that sounds like a name from a soft porn movie. |
| 22:26.48 | rizwank | *chortle* |
| 22:26.57 | ChannelZ | do you know Kumbang? |
| 22:27.02 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:27.06 | *** part/#asterisk axelilly (n=jfenner@66.181.75.69) |
| 22:27.36 | Katty | ahahaha, who was Kumbang |
| 22:27.37 | Katty | i forget. |
| 22:27.43 | Katty | was that jaytee's name? |
| 22:27.56 | jaytee | no |
| 22:27.56 | ChannelZ | ?? just another nick I see in here from time to time that always gives me pause |
| 22:28.26 | Katty | jaytee is gunnar quickie |
| 22:28.35 | jaytee | huh? |
| 22:28.43 | Katty | http://gangstaname.com/porn_name.php |
| 22:28.56 | Katty | just something stupid i ran across. |
| 22:29.08 | lynxsys | anyone know i there is an IRC channel for Cisco handsets? |
| 22:29.16 | jaytee | yeah, jaytee comes out as gunnar quickie |
| 22:29.25 | Katty | wow, that guy at the bottom sure could afford a lot of tattoos in 4 weeks. |
| 22:29.26 | jaytee | my real name comes out as Butt Stroker |
| 22:29.31 | Katty | hahaha |
| 22:29.42 | Katty | mine's Kitty jam |
| 22:30.03 | jaytee | If I add my middle initial I get Butt Hornball |
| 22:30.11 | Katty | LOL |
| 22:30.24 | Katty | oh wow |
| 22:30.30 | Katty | my full name is: Madam the Really Famous Porn Star |
| 22:30.36 | jaytee | if I use my family nickname and last name I get Ricky Asstronut |
| 22:30.38 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.32.94) |
| 22:31.42 | jaytee | Katty, your full name? is that with middle initial? |
| 22:31.46 | lenne_dk | I'm Daddy Spankadocious. |
| 22:31.54 | Katty | full middle name |
| 22:32.02 | Katty | my full name, as shown on the birth certificate. |
| 22:32.11 | lenne_dk | Wife agrees. |
| 22:32.13 | jaytee | ok, cuz just your first and last name give me Slappy Spreadum |
| 22:32.29 | *** part/#asterisk lenne_dk (n=leif@0x573cc07b.odnxx13.dynamic.dsl.tele.dk) |
| 22:32.40 | Katty | yep |
| 22:32.44 | Katty | stick jayne in there |
| 22:33.21 | jaytee | this is lame, my mom's name comes back as Daddy Jiggles |
| 22:33.47 | Katty | creepy |
| 22:35.14 | Katty | hmm. it must be dinner time. |
| 22:35.15 | Deeewayne | <-- Harley Dangle |
| 22:35.19 | jaytee | oh, I forgot to click the radio button for female when I did my mom's name |
| 22:35.47 | Katty | what is it now? |
| 22:36.07 | jaytee | I think they were talking about and playing with this website on the Bob and Tom radio show one morning |
| 22:36.19 | Katty | yeah maybe. i forget. |
| 22:36.46 | jaytee | when I change the sex on the radio button my mom's name becomes Cara Jiggles |
| 22:37.35 | jaytee | my gangsta name is Threepac Hob-nobba |
| 22:38.16 | Katty | what's mine |
| 22:39.32 | jaytee | Whipped Chimpanzee |
| 22:40.04 | Katty | ^_- |
| 22:40.09 | Katty | mmmmkay then. |
| 22:40.27 | jaytee | strange app |
| 22:41.16 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 22:41.18 | _abc_ | hello |
| 22:41.24 | Katty | hi |
| 22:41.27 | _abc_ | is the skinny channel not reloadable? |
| 22:41.28 | *** part/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 22:41.33 | _abc_ | i can't reload it in 1.6.1.11 |
| 22:41.45 | _abc_ | module reload chan_skinny.so also fails |
| 22:41.49 | Katty | the skinny channel seems to be featuring ugg boots this week. |
| 22:41.51 | _abc_ | the channel works |
| 22:41.57 | _abc_ | Katty: huh? |
| 22:42.17 | Katty | nevermind. |
| 22:42.19 | _abc_ | hopes for an acronym translation |
| 22:42.53 | mmlj4 | ser, openser, opensips... which to use, which to ignore? |
| 22:43.02 | blitzrage | _abc_: that module very well may not support reloading. You may need to 'module unload chan_skinny.so' and then 'module load chan_skinny.so' |
| 22:43.22 | _abc_ | blitzrage: that also fails, it says Skinny could not be initialized |
| 22:43.27 | _abc_ | yet the channel works |
| 22:43.35 | _abc_ | if i restart asterisk it also loads right |
| 22:43.48 | Katty | _abc_: http://monichika2.files.wordpress.com/2009/11/skinny-jeans.jpg <- skinny |
| 22:43.49 | ChannelZ | I am Dirk Phukzalot. |
| 22:43.50 | _abc_ | i understand that chan_sccp supersedes skinny |
| 22:44.09 | blitzrage | _abc_: it may be possible the module can't be unloaded and changes can only take effect on a restart -- not sure, since I've never used that module. |
| 22:44.17 | _abc_ | aargh |
| 22:44.39 | jaytee | "Welcome to Hell, here's your Cisco phone!" |
| 22:44.40 | _abc_ | the command skinny reload certainly does not exist |
| 22:44.44 | Katty | _abc_: http://www.mydaydayblog.com/wp-content/uploads/2009/11/ugg_boots.gif <- ugg boots |
| 22:44.50 | _abc_ | jaytee: you can say that again |
| 22:44.56 | Katty | _abc_: the skinny channel seems to be featuring ugg boots this week. |
| 22:44.59 | _abc_ | Katty: i got the point |
| 22:45.03 | Katty | excellent. |
| 22:45.04 | rizwank | any easy way to detect if a machine is from AsteriskNow (and what version?) |
| 22:45.20 | _abc_ | rizwank: connect to it and then core show version |
| 22:45.33 | rizwank | that'll show asterisknow versus just asterisk? |
| 22:45.39 | _abc_ | would asterisk reload also reload skinny? |
| 22:45.55 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 22:45.56 | _abc_ | rizwank: no but it will show what it was built on (unix machine id etc) |
| 22:46.03 | _abc_ | you can deduce things from that |
| 22:46.10 | rizwank | built by root. |
| 22:46.15 | rizwank | I think i may have done a source install here. |
| 22:46.25 | _abc_ | rizwank: root @ what ? |
| 22:46.33 | _abc_ | and date |
| 22:46.34 | rizwank | localhost.localdomain |
| 22:46.39 | _abc_ | ohh clevvver |
| 22:46.43 | rizwank | 2009-12-21 |
| 22:46.52 | _abc_ | well it's recent |
| 22:47.11 | _abc_ | so you can nail it down by file timestamps |
| 22:47.44 | _abc_ | get a package that you suspect it the origin and open the tar and look at the file dates (e.g. stat `which bin/asterisk`) |
| 22:48.07 | Deeewayne | ~nowwhat |
| 22:48.08 | infobot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=FJ3oHpup-pk |
| 22:48.13 | _abc_ | seriously is there no hope in (cisco) h@11 that i can change the skinny conf without restart? |
| 22:48.24 | rizwank | hmm, okay. I'll install another asterisknow instance, had planned to anyhow, and go from there to compare. thanks. |
| 22:55.57 | *** join/#asterisk jblack (n=jblack@71.181.248.16) |
| 22:56.43 | bmoraca | woot...D&H started selling PAP2Ts and SPA-8000s |
| 23:00.07 | ChannelZ | wets self |
| 23:00.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 23:04.53 | bmoraca | is chan_skinny fairly useful in 1.6.2? more so than in 1.4.x? by which, i mean: does meetme() work with skinny now? |
| 23:06.24 | Kobaz | hmm |
| 23:06.31 | Kobaz | i just crashes asterisk with moh show files |
| 23:07.44 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 23:17.01 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 23:21.35 | *** join/#asterisk etfonhomey (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 23:22.11 | *** part/#asterisk jsolis (n=Jimmy@200.121.176.59) |
| 23:22.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:28.23 | ChannelZ | no music for you! |
| 23:30.35 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 23:30.43 | *** join/#asterisk romb (i=Romb@89.28.249.108) |
| 23:31.42 | romb | hello |
| 23:31.47 | generalhan | hey all ... is there anything like the linear strategy for 1.6 in 1.4 ? i need a queue to attempt a member everytime, and then move to the next member, if the 1st is 'in use' or doesnt answer. |
| 23:31.51 | romb | i have a problem with festival |
| 23:31.56 | romb | http://pastebin.ca/1738079 |
| 23:32.02 | generalhan | roudnrobin doesnt do it, and rrmemory might start with the 2nd member instead of the 1st |
| 23:32.08 | romb | i'm getting app_festival.c: Festival WV command |
| 23:32.25 | romb | but no app_festival.c: Last frame |
| 23:33.00 | Katty | this mask feels gooood. |
| 23:33.09 | Katty | cept it's drying, and it's hard to move my nose |
| 23:33.49 | *** part/#asterisk rizwank (n=rizwank@76.89.131.47) |
| 23:34.55 | romb | generalhan, by one way you can use Goto on your own |
| 23:37.35 | ChannelZ | generalhan: what version of asterisk |
| 23:37.57 | generalhan | ChannelZ: 1.4.18 |
| 23:38.11 | Katty | eppigy: I CAN"T MOVE MYF ACE |
| 23:38.54 | ChannelZ | generalhan: How is roundrobin behaving? |
| 23:39.07 | generalhan | ChannelZ: same as rrmemory |
| 23:43.24 | *** join/#asterisk pfn (i=pfnguyen@socal.hanhuy.com) |
| 23:46.22 | *** join/#asterisk tzafrir (n=tzafrir@212.179.75.202) |
| 23:46.51 | ChannelZ | well you can pastebin your queues.conf so we can see if anything jumps out |
| 23:48.19 | eppigy | Katty: :< |
| 23:48.27 | Katty | eppigy: it's for a good cause. |
| 23:48.31 | Katty | eppigy: i got another 5 minutos. |
| 23:51.18 | eppigy | nice |
| 23:52.03 | Katty | :> |
