| 00:00.51 | DocHolliday | the only thing that has not worked thus far is DND (not a huge issue) |
| 00:04.05 | DocHolliday | argh, so many phones on my desk i cant tell which one is ringing |
| 00:04.15 | Grnd-Wire | haha |
| 00:04.16 | jql | for DND, try it through the menu: http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_user_guide_chapter09186a0080087003.html#wp1028558 |
| 00:04.39 | jql | I have my cisco using the ringtone from 24. makes it easy to tell if a *real* call is coming in. :) |
| 00:05.05 | DocHolliday | jql, same here :P |
| 00:05.27 | DocHolliday | problem is i have 4 phones on my desk, and one of them is programmed with 4 different rings (line 1 / line 2 / line 3 / intercom) |
| 00:05.30 | jql | Heh. No point in having a voip phone if you don't set it to the 24 ringtone |
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| 00:05.51 | DocHolliday | so true |
| 00:06.20 | DocHolliday | thanks for that link, problem is i want a constant soft key, not one that disappears when you disable it |
| 00:06.40 | jql | distubingly, Cisco provides a softkey api |
| 00:06.46 | jql | I recommend you don't even look at it |
| 00:07.24 | DocHolliday | yeah that frightens me |
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| 00:08.18 | DocHolliday | jql, i spent 45 minutes today making the background image i want fit properly |
| 00:08.56 | jql | took me 2 days to get my logo on there right |
| 00:09.09 | jql | what a phone... |
| 00:09.31 | DocHolliday | to be honest i thought i was an idiot.. i was thinking.. this has to be easier, but once i figured out the pixel dimensions of the screen it was fairly easy |
| 00:10.47 | jql | an xml file, though... |
| 00:11.37 | DocHolliday | well the screen pixel dimensions for the background is 320x196, so in MS paint i made a white space that large and adjusted the image accordingly |
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| 00:20.51 | KuJaX | Anyone alive? :) |
| 00:23.44 | QueTwo | nope |
| 00:26.39 | JerJer | nobody here but us chickens |
| 00:26.44 | KuJaX | :o |
| 00:27.06 | KuJaX | Is there a standalone device which will convert a PTSN POTS analog line to digigal format (FXO) for my Asterisk box that doesn't connect directly into the Asterisk box? |
| 00:27.16 | JerJer | sure |
| 00:27.17 | KuJaX | Looking for two or possibly four analog phone lines for my ASterisk box. |
| 00:27.20 | JerJer | spa-300X |
| 00:27.32 | KuJaX | I haven't heard very many good things about the Digium PCI cards. too much echo (from what i've heard) |
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| 00:27.43 | JerJer | you have heard wrong |
| 00:27.51 | JerJer | echo problems are not because of Digium cards |
| 00:27.57 | KuJaX | What are they caused from? |
| 00:27.58 | JerJer | they are because of poor copper |
| 00:28.03 | JerJer | or improper wiring |
| 00:28.13 | JerJer | or just a shitty telco |
| 00:28.36 | KuJaX | but would the echo be there with the SPA-300x? |
| 00:29.23 | KuJaX | Basically a quick rundown. I have tried two voip providers, two ISP's incase it was that, and I am getting about 50% of my calls as staticy, in and out or simply dropped. So we are looking to move analog. |
| 00:29.47 | jql | well, that's a disturbing scenario |
| 00:30.02 | KuJaX | I know. It isn't THAT bad, but it is a constant struggle, and we only have two people on the phones at a time. |
| 00:30.17 | KuJaX | Everyday I hear "are you on a cell phone" or "we have really bad feedback on this line" |
| 00:30.23 | KuJaX | from customers or worse, potential customers. |
| 00:30.31 | DocHolliday | KuJaX, i have a solution for you |
| 00:30.42 | DocHolliday | deliver your voice over a PRI terminate it on an Asterisk ISR and use that to feed asterisk |
| 00:30.46 | KuJaX | We love the functionality and versatility of Asterisk..... but I think we need the clarity and professional sound of analog. |
| 00:30.49 | DocHolliday | err Cisco ISR |
| 00:31.01 | KuJaX | DocHolliday: What do you mean? What would that do? |
| 00:31.18 | DocHolliday | i hate to break it to everyone but the DSPs on a Cisco voice gateway will always beat the sound quality of asterisk |
| 00:31.27 | KuJaX | It is a big enough struggle that we may end up purchasing a NEC or Avaya analog phone system and ditch VOIP completely. :( |
| 00:31.47 | DocHolliday | KuJaX, you can keep asterisk, simply deliver your trunks over a PRI instead of VoIP |
| 00:32.12 | DocHolliday | terminate the PRI on a Cisco ISR and have the the ISR feed calls to Asterisk |
| 00:32.39 | QueTwo | or feed a PRI to a dialogic card within your asterisk box |
| 00:32.40 | KuJaX | DocHolliday: what would I be looking to spend for that setup? and would I still use my same VOIP providers? |
| 00:33.15 | jql | what model of voip phone do you have atm? |
| 00:33.24 | fetcher | I use a Cisco AS5300 full of VFC cards (in place of modems) for the same purpose. Sound quality is as good as any other TDM switch |
| 00:33.26 | KuJaX | Linksys spa-941 |
| 00:33.53 | fetcher | Haven't tried using the Cisco's DSPs for compression, though |
| 00:34.01 | KuJaX | Using Voicepulse provider, Teliux provider, Asterisk box with two SPA-941 phones and a Cisco 7940 IP Phone. |
| 00:34.10 | DocHolliday | QueTwo, i am not impressed with PCI non-DSP based PRI cards |
| 00:34.17 | DocHolliday | KuJaX, naw get rid of the damned voip :) |
| 00:34.18 | DocHolliday | how many voice trunks? |
| 00:34.38 | KuJaX | Right now we aren't getting consistant results. Terrible sound quality. When we page each other or talk within the local network, it isn't bad. But it is when the outside world enters or exits..... bad voice quality. Terrible most of the time. Sometimes it is fine. |
| 00:34.59 | jql | KuJaX: Are you in a big city, or off in the stix? |
| 00:35.01 | KuJaX | If we went the analog route, we would start with two POT lines, but would like expandability for up to 4 for the future (next 6 months) |
| 00:35.15 | KuJaX | jql: neither, not big city nor stix. Rural city area. |
| 00:35.18 | DocHolliday | yeah sounds like a combination of a shitty voip provider and internet connection |
| 00:35.24 | jql | using DSL service? |
| 00:35.28 | KuJaX | correct. |
| 00:35.52 | jql | Does the problem increase with both phones in use, or is it a permenent feature? |
| 00:35.56 | KuJaX | I wouldn't mind ditching VOIP at the moment until 1gb per second lines are out, so I believe we want to make the switch to analog. |
| 00:35.59 | DocHolliday | KuJaX, whats your call volume like? |
| 00:36.34 | KuJaX | It is a consistant problem. Like I said, when me and my co-worker talk via the phones over the network, nothing is wrong. When we use it to go outside or incoming calls, it is a hit or miss. About 50% are absolutely terrible (we have to call them back) and the other 50% will go in and out. |
| 00:36.54 | DocHolliday | KuJaX, you barely need 1Mbps for VoIP.. |
| 00:36.55 | jql | well, that sucks |
| 00:36.57 | KuJaX | DocHolliday - About 3000 to 4000 minutes a month. |
| 00:37.15 | KuJaX | DocHolliday - Yeah, I have tried cable internet (a bit better, but not perfect) and DSL. |
| 00:37.16 | jql | I'd fire my provider(s) if I couldn't get better than that |
| 00:37.30 | DocHolliday | how many simultaneous calls / circuits? |
| 00:37.34 | KuJaX | 2 max. |
| 00:37.46 | KuJaX | Right now there are only two of us. |
| 00:38.18 | jql | well, since you've already tried two different VoIP services, seems like you're right to move on. very weird |
| 00:38.22 | DocHolliday | KuJaX, so why did you get rid of the analog lines in the first place? |
| 00:38.31 | QueTwo | KuJaX: what is your jitter/delay like between you and your voip service? |
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| 00:39.57 | KuJaX | about 80ms |
| 00:40.05 | KuJaX | DocHolliday : we never had analog phone lines. New startup company. |
| 00:40.22 | jql | and this is always reproducable? |
| 00:40.42 | KuJaX | Yes. Everyday I hear "are oyu on a cell phone" from the person on the other end. or "we have a bad connection, let me cal back" |
| 00:40.54 | DocHolliday | yeah i love analog even though its expensive |
| 00:41.12 | KuJaX | I got teliux service and routed all services through them instead of Voicepulse and didn't tell my co-worker. HE didn't even notice a difference (i asked him if the call quality has been getting better) |
| 00:41.50 | KuJaX | DocHolliday - Well, in our situation, analog wouldn't be that much more per month. $30 per line, $25 for unlimited long distance, a few bucks for features such as call hunting.. so we are at about $150 a month for two lines the way we need it |
| 00:42.12 | QueTwo | 80ms is pretty high |
| 00:42.19 | KuJaX | Right now we are spending about $60 to $70 a month on VOIP phone service, of which we are easily losing the $80 difference each month in sales by having bad voice quality. |
| 00:42.39 | QueTwo | 80ms with no jitter will account for "cell-phone" quality |
| 00:42.40 | DocHolliday | yeah, i usually try to go the VoIP route for 'a couple lines' anything more is analog all the way |
| 00:42.53 | QueTwo | 80ms with little jitter will become unusable quickly |
| 00:43.01 | fetcher | KuJaX: ISDN BRI might be worth considering, depending on how it's prived in your area. |
| 00:43.01 | KuJaX | So we are going to make the switch to analog. PRoblem is that we like the functionality of Asterisk. We have to decide to try to implement analog lines with Asterisk or simply go with a totally analog NEC or Avaya phone system. |
| 00:43.09 | fetcher | s/prived/priced/ |
| 00:43.54 | KuJaX | jbot- right, well is ISDN analog phone lines or will we still be using VOIP? |
| 00:43.55 | QueTwo | you can get away with an Avaya partner system for under 5k |
| 00:44.20 | KuJaX | yeah we are looking at their One-X system, or the NEC DSX-40. |
| 00:44.27 | QueTwo | or they have that new One-X system too |
| 00:44.29 | KuJaX | both under $2,000./ |
| 00:44.42 | QueTwo | One-X is pretty neat... i've implemented a few of those |
| 00:45.02 | QueTwo | they are pretty basic, but do what 80%of the population need |
| 00:45.07 | KuJaX | How is the call quality? Because with One-X, it does change analog signal to digital. That is whawt I am afraid of, changing analog signal to digital and getting echo or problems that way. |
| 00:45.30 | KuJaX | What about NEC DSX-40 or TalkSwitch systems? |
| 00:45.44 | fetcher | KuJaX: ISDN gives you dedicated bandwidth... not VoIP-related at all. It can sound better than analog. Hardware is more expensive though |
| 00:47.44 | KuJaX | fetcher- I don't want to invest in a bunch of ISDN hardware which may not be around in a couple of years due to the demand. |
| 00:47.45 | QueTwo | no call quality issues from the Analog -> SIP box |
| 00:48.07 | QueTwo | i did have a few customers who had call quality issues when they used switches that didn't support QoS |
| 00:48.25 | QueTwo | they had some linksys pieces of garbage that couldn't handle it |
| 00:48.25 | KuJaX | Right now we have absolutely no QOS within the network. On the router nor switch level. |
| 00:48.39 | QueTwo | that may be something to look at |
| 00:48.55 | QueTwo | QoS can make a world of difference, at both echo and call quality |
| 00:49.22 | KuJaX | QueTwo - Yeah, but I may purchase a new QOS Router and nothing changes and then I am out even more money towards this stupid thing. :( |
| 00:49.56 | QueTwo | well, if you want to do any IPtelephony, you need to invest into QoS equipment |
| 00:50.09 | QueTwo | i mean, even some small soho routers do qos |
| 00:50.24 | KuJaX | Using Linksys right now. |
| 00:50.29 | QueTwo | wich one? |
| 00:50.47 | KuJaX | their 4 port dsl/cable router NON-wireless. |
| 00:50.59 | KuJaX | befsr41 |
| 00:51.05 | jql | ahh, the classic |
| 00:51.15 | QueTwo | check under the application & gaming section... some firmware versions have a QOS tab |
| 00:51.23 | QueTwo | it's easy to setup on those guys |
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| 00:51.39 | KuJaX | Not on this one. Maybe firmware upgrade will do it? Would I point the QOS to the Asterisk IP address? |
| 00:51.55 | QueTwo | whatever is comsuming your VoIP trunks |
| 00:52.00 | QueTwo | so, yes |
| 00:52.11 | KuJaX | QOS isn't there on that ta. |
| 00:52.13 | KuJaX | *tab |
| 00:52.29 | QueTwo | ok |
| 00:52.41 | QueTwo | firmware upgrade will put it in there, if your model has enough RAM |
| 00:52.45 | JerJer | qos has nothing to do with echo |
| 00:52.52 | QueTwo | sure it does |
| 00:53.14 | QueTwo | if you get too much jitter, the echo canclers will depleat and you get echo |
| 00:53.19 | JerJer | you are talking about 'talk back' which is not echo |
| 00:53.37 | KuJaX | we aren't getting much "talk back" |
| 00:53.50 | KuJaX | we can't hear ourselves, but sometimes the other person on the other end says they can hear themselves a little bit. |
| 00:53.55 | JerJer | if you get too much jitter the call generally sounds bad - but its not echo |
| 00:54.11 | JerJer | i have tested this extensively |
| 00:54.55 | jql | if your callers hear echo, it's probably caused by your end |
| 00:55.17 | jql | but then, I also have had nasty echo on PSTN calls, but usually I tell them to turn off their damn speakerphone |
| 00:56.34 | suma | when i do a read with p,vm-enter-num-to-call,,,,60000 and NoOP(${p}) it says user entered "12345" but the variable is not showing anything |
| 00:56.41 | QueTwo | REF : http://support.avaya.com/elmodocs2/comm_mgr/r3/IP_GUIDE_3.0.pdf |
| 00:58.48 | KuJaX | So... end result. I can either get something to convert analog signal (normal phone lines) to digital format for my Asterisk box. Or dump Asterisk and go with a propriteary Avaya, NEC or TalkSwitch pure analog based small business phone system??????? |
| 00:59.00 | *** join/#asterisk Dovid (n=Dovid@85.159.160.199) |
| 00:59.05 | Dovid | ~centosbug |
| 00:59.11 | jbot | hmm... centosbug is a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 00:59.28 | jql | there's always another provider and different hardware and whatnot |
| 00:59.35 | jql | but it's hurting your business. :/ |
| 00:59.46 | KuJaX | yeah, it is. I can't take any more of these calls like this. |
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| 01:05.12 | vlt|home | Hello. I installed a QuadBRI card to my asterisk box. Now I get the following output: |
| 01:05.24 | vlt|home | *CLI> zap show status |
| 01:05.27 | ModocNet | anything change from 1.2.0 -> 1.4.0 regrading subscribecontext in sip.conf and hint in extensions....I cant get BLF to work in 1.4.0 |
| 01:05.28 | vlt|home | Description Alarms IRQ bpviol CRC4 |
| 01:05.28 | vlt|home | quadBRI PCI ISDN Card 1 Span 1 [TE] (ca� UNCONFIGUR 0 0 0 |
| 01:05.28 | vlt|home | quadBRI PCI ISDN Card 1 Span 2 [TE] (ca� UNCONFIGUR 0 0 0 |
| 01:05.28 | vlt|home | quadBRI PCI ISDN Card 1 Span 3 [TE] (ca� UNCONFIGUR 0 0 0 |
| 01:05.28 | vlt|home | quadBRI PCI ISDN Card 1 Span 4 [TE] (ca� UNCONFIGUR 0 0 0 |
| 01:05.29 | vlt|home | Can anyone tell me which conf file to edit to access these channels? |
| 01:07.54 | [TK]D-Fender | KuJaX : y0 |
| 01:08.11 | ModocNet | my quad PRI card gets configured in zaptel.conf and zapata.conf |
| 01:09.19 | vlt|home | ModocNet: Both files? |
| 01:11.13 | suma | vlt|home: can you paste the zaptel.conf in pastebin ? |
| 01:12.05 | vlt|home | suma: It should be the original one (empty) but I'll paste it ;-) |
| 01:13.50 | Qwell | public-: If you want to add support - http://www.bluetooth.org/foundry/adopters/document/8_Fax/ |
| 01:16.26 | vlt|home | suma: Oops, there is only zapata.conf, no zaptel.conf. But I got an example from a vendor page: http://rafb.net/p/7gmKmt96.html |
| 01:17.06 | vlt|home | suma: I'll reload * |
| 01:19.26 | vlt|home | suma: I even restarted, no difference. Does the zaptel.conf file make sense? What do I have to edit in zapata.conf? |
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| 01:24.06 | jql | do the bri drivers use ztcfg or someething? |
| 01:24.15 | jql | cause it's required for pri usage |
| 01:26.01 | suma | vlt|home: you need to have zaptel.conf |
| 01:26.27 | suma | zaptel drivers uses zaptel.conf |
| 01:26.32 | suma | asterisk uses zapata.conf |
| 01:27.06 | suma | if the card is not zaptel based, then no need of zaptel.conf |
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| 01:27.21 | suma | But make sure the respective module is loaded with modprobe or insmod |
| 01:33.12 | vlt|home | suma: zaptel module is loaded. Before that I couldn't see the card in asterisk. I just found another example: zapata.conf: http://rafb.net/p/S7PJfV59.html -- But when restarting I get "Feb 25 02:28:53 ERROR[31499]: chan_zap.c:7239 mkintf: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1" |
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| 01:35.37 | *** join/#asterisk Phel (n=chatzill@adsl-223-225-183.mia.bellsouth.net) |
| 01:37.00 | Phel | Question: If you aren't supposed to have to forward ports in order to get a SIP client to work, why do they insist on listening on port 5060? |
| 01:38.06 | [TK]D-Fender | Phel : Because most NAT aware gateways can send "Qualify" packets to keep the UDP port active on your NAT router so that the inbound signalling doesn't get shut down./ |
| 01:38.43 | [TK]D-Fender | Phel : and when it comes time to start RTP, the clietn will initate the contact so as to begin a new UDP map which will survive the length of the call. |
| 01:39.03 | [TK]D-Fender | Phel : Its all about making sure the UDP stays mapped. |
| 01:39.08 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 01:40.32 | Phel | Well if I'm currently listening on UDP Port 5060, can you see that with a port scan? |
| 01:41.46 | Phel | If I port scan fwd.pulver.com, should I be able to see a service running on 5060 if it's not being blocked? |
| 01:41.49 | [TK]D-Fender | Phel : Yes, if the server you are qualifying with is helping keep your port mapped. |
| 01:42.10 | Phel | Can you port scan my ip please? |
| 01:42.22 | vlt|home | suma: Aah, now I know why I have to configure zapata.con AND zaptel.conf. I found this line in a wiki: Asterisk <--> Zap Module <--> Zaptel Driver <--> Digium Interface Card <--> Phone/switch/PSTN |
| 01:42.36 | vlt|home | suma: But it still doesn't start :( |
| 01:45.04 | Phel | Well should I see port 5060 on fwd.pulver.com? |
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| 01:57.18 | Dovid | where are the voicemail's stored ? |
| 01:57.39 | Dovid | in /var/lib? |
| 01:59.37 | Dovid | nm |
| 01:59.49 | vlt|home | suma: I had to tell `ztcfg` where zaptel.conf is ... Now the card shows up in cli. |
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| 02:05.45 | vlt|home | "CRC error for HDLC frame on card 1 (cardID 7) S/T port 1" and similar fill my dmesg since I ran `ztcfg` ... Any idea? |
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| 02:15.16 | *** join/#asterisk _paulos_ (n=paulos@200-207-113-92.dsl.telesp.net.br) |
| 02:15.28 | _paulos_ | Hi people... |
| 02:16.32 | _paulos_ | coppice, is soft-switch.org offline? |
| 02:19.25 | Bobthehunter | yes it is |
| 02:24.47 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 02:28.32 | Dovid | hi guys |
| 02:28.48 | Dovid | i moved over asterisk from one box to another am i using real time |
| 02:29.21 | suma | Dovid: ask your question. OK |
| 02:30.27 | Dovid | sorry got side tracted |
| 02:30.55 | Dovid | i removed from my sip users table the entries for the sip from mysql since i want the DID still ring on the old box |
| 02:31.07 | Dovid | asterisk seems to try to look for it in mysql even thought it is no longer ther |
| 02:31.08 | Dovid | ethere* |
| 02:31.21 | suma | you configured in, |
| 02:31.27 | suma | extconfig.conf ? |
| 02:31.28 | Dovid | MySQL RealTime: Everything is fine. |
| 02:31.28 | Dovid | MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = 'sip.sipmedia.com' |
| 02:31.30 | Dovid | yes |
| 02:33.46 | suma | Dovid: i really don't know what could be other problem out there |
| 02:34.14 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 02:34.16 | Dovid | any way to clear the real time cache ? |
| 02:36.01 | Dovid | nm |
| 02:36.03 | Dovid | found it |
| 02:37.22 | _paulos_ | is there some mirror to soft-switch.org? |
| 02:37.53 | jql | I haven't found one |
| 02:37.59 | jql | and I regret that |
| 02:38.03 | _paulos_ | :-( |
| 02:38.11 | jql | I needed to download an older version, and couldn't |
| 02:38.20 | jql | urge to mirror... rising |
| 02:38.49 | _paulos_ | jql: I have spandsp-0.0.2pre26.tar.gz. |
| 02:39.29 | _paulos_ | jql: would it serve you? |
| 02:39.32 | bkruse_home | sql is SO easy |
| 02:39.55 | jql | no, I was looking for a particular revision in the0.0.3 branch |
| 02:40.36 | _paulos_ | I have spandsp-20061026.tar.gz and some other also. |
| 02:40.47 | RoyK | sql is quite easy unless you start doing complex joins |
| 02:40.49 | jql | now that's closer to what I need |
| 02:41.15 | bkruse_home | RoyK: agreed, once you have your database setup, then its easy |
| 02:41.49 | _paulos_ | let me see... spandsp-20070112.tar.gz and spandsp-20070222.tar.gz |
| 02:42.04 | bkruse_home | i need to do more sql stuff |
| 02:42.35 | RoyK | bkruse_home: from a single table it's easy, but take a hundred table setup with intricate joins and it becomes a headache |
| 02:43.09 | _paulos_ | also spandsp-20061105.tar.gz |
| 02:43.37 | jql | you have more versions than I do. put up a mirror. :) |
| 02:45.58 | joaovianna | Anyone using video with asterisk ? I'm having trouble with grandstream 3000 sending video. Anyone ? |
| 02:46.10 | jql | I haven't bought one yet. |
| 02:46.13 | jql | Rather, two |
| 02:46.19 | jql | one is somewhat pointless for testing |
| 02:46.33 | joaovianna | That is my problem ! |
| 02:46.36 | DocHolliday | jql, i have everything working except DND, fine i guess |
| 02:47.07 | _paulos_ | jql: http://pabx.xtend.com.br/spandsp/ |
| 02:48.10 | _paulos_ | jql: hope you find it useful. |
| 02:48.22 | jql | cool, thanks |
| 02:48.48 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:49.42 | _paulos_ | I'm having troubles with new TE110P. Lots of frame slips. |
| 02:51.09 | _paulos_ | Digium told me to upgrade zaptel. zaptel-1.4.0 worked better, but its far from perfect |
| 02:51.25 | aptura | I have somone who wants to test my system in india for a call center. The calls will come in from north america and piped to india. He is going there for another purpous so I want to sugest a low cost call center phone. Polycom is perhaps one but what else is a low cost alternative? I was thinking of Sipura also. |
| 02:51.44 | aptura | what do you mean by far from perfect _paulos_ |
| 02:52.03 | bkruse_home | hmm |
| 02:52.03 | bkruse_home | interesting |
| 02:52.03 | bkruse_home | i dont know sql enough to argue, I havent found the need to create such complex structures, Yet. |
| 02:52.24 | _paulos_ | I'm in Brazil, so I have to use libunicall |
| 02:52.40 | jql | with a decent headset, even a grandstream isn't all that bad |
| 02:53.12 | aptura | bkruse` your bkw right? |
| 02:53.12 | _paulos_ | aptura: I'm having sync problemas since the upgrade |
| 02:53.22 | aptura | I see |
| 02:53.52 | _paulos_ | aptura, every hour or so, all my channels get blocked for a few seconds. |
| 02:54.32 | _paulos_ | aptura, since the upgrade, I cant receive fax anymore (with app_rxfax) |
| 02:56.18 | aptura | That sucks. |
| 02:56.25 | aptura | I have not used the fax feature yet. |
| 02:56.28 | bkruse_home | will write dialplan for food |
| 02:56.46 | jql | How can I compete with that? |
| 02:56.51 | jql | I might as well quit now. :) |
| 02:56.56 | xheliox | I was just thinking the same thing. |
| 02:57.06 | aptura | btw can cli show a current channel performance? |
| 02:57.17 | jql | Will write manager-api-using-tools for $ |
| 02:57.30 | jql | oh, I know |
| 02:57.38 | xheliox | I'll write dialplans just for the satisfaction of knowing bkruse will starve. ;) |
| 02:57.41 | jql | will write ser routes for $$$ |
| 02:57.43 | jql | :) |
| 02:58.02 | _paulos_ | damm digium, they shoud support unicall or MFC/R2. |
| 02:58.55 | xheliox | how dare you even think of it! |
| 02:59.26 | aptura | A few months ago Nortel was responding to media reports about there low stock price and thay blamed it on the low cost of chinese imported phones. The report never did mention what was driving those phones :) |
| 03:01.53 | aptura | A year ago I was visiting a well known city hall driven by nortel ippbx system and there phone sysadmin said its a headach always having issues and going down. Tech always comes in. Its a big system at 1,500 phones. Everything goes though three boxes. |
| 03:01.59 | aptura | BRB |
| 03:02.28 | bkruse_home | aptura: you could totally do that with 3 boxes, if you do it right and maybe just a couple other pieces of equipment :] |
| 03:04.43 | xheliox | I asked this the other day, and no one replied -- anyone seen got-name.com? It's a caller ID name lookup service. |
| 03:04.58 | jql | oh really? |
| 03:05.04 | aptura | well in there case it was mission critical and a bane for the phone admin. I asked her if she ever heard of asterisk and said nope :) |
| 03:05.56 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 03:06.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:07.11 | _paulos_ | jql: take a look at http://zarzamora.com.mx/mirror/www.soft-switch.org/downloads/spandsp/ |
| 03:07.33 | jql | aha, the version I wanted! |
| 03:08.04 | jql | thanks, _paulos_ |
| 03:08.12 | _paulos_ | you are welcome. |
| 03:08.39 | coppice | zarzamora.com.mx is a very useful mirror :-) |
| 03:09.46 | _paulos_ | coppice: soft-switch.org needs a new home? |
| 03:10.47 | _paulos_ | coppice: I think I can get you free hosting. |
| 03:10.56 | coppice | I can't contact the person who hosts it for me to find what has happened. dedicated rental servers are a pain |
| 03:12.51 | JT | shared hosting might work? |
| 03:13.10 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 03:13.30 | coppice | it's shared on someone else's rental server :-) |
| 03:13.36 | JT | heh |
| 03:13.59 | jql | a dedicated Celeron < shared P4 |
| 03:14.13 | jql | but, I like my Celeron nonetheless |
| 03:14.55 | coppice | colos really don't seem to care about power consumption. they rent all these AMD K7 2000+, and P4 1.8GHz boxes because the hardware has already been amortised, but they consume a lot for what they do |
| 03:15.29 | JT | heh |
| 03:15.34 | DocHolliday | coppice, i beg to differ |
| 03:15.58 | DocHolliday | power is an expensive commodity now in datacenters (probably one of the most expensive costs), especially in metro areas |
| 03:16.01 | jql | it wouldn't surprise me if $10/month of my bill is going to power |
| 03:16.12 | DocHolliday | jql, try $10+ per amp |
| 03:16.14 | _paulos_ | coppice: most colo business uses some datacenter facility, and they charge by rack space, not for Watt. |
| 03:16.37 | DocHolliday | _paulos_, again.. they are charged for each Amp of power they consume |
| 03:16.45 | jql | well then, I'll start running SETI on this box |
| 03:16.49 | coppice | so why do they use these old boxes, when a cheap current AMD will consume much less, especially at idle |
| 03:16.51 | jql | after all, it's unmetered |
| 03:16.52 | jql | muahaha |
| 03:17.38 | coppice | DocHoliday I've never known a colo charge per amp. they don't even measure that |
| 03:17.44 | _paulos_ | DocHolliday, we have a cage at Miami NAP, and they charge a flat fee. |
| 03:17.56 | DocHolliday | for power? |
| 03:18.14 | coppice | they charge per U, but not per amp |
| 03:18.23 | _paulos_ | DocHolliday, for the cage, including bandwidth and power. |
| 03:18.24 | DocHolliday | coppice, sure but how much power is provided to you? |
| 03:18.24 | wunderkin | yes i've seen colos charge for power |
| 03:18.27 | coppice | some charge for data, and some are unmetered |
| 03:18.53 | DocHolliday | right but chances are if they include say 20A.. if you use the 20A you pay for more (its just builtin to your fee at the moment) |
| 03:19.04 | coppice | DocHoliday: as much as any 1U could draw without melting :-) |
| 03:19.18 | DocHolliday | ohh 1U.. makes sense then. |
| 03:19.20 | *** join/#asterisk omarc55 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net) |
| 03:19.38 | JT | _paulos_ has a full cage |
| 03:19.41 | JT | not 1RU |
| 03:19.53 | jql | my company's colo doesn't permit blade systems, just to limit power/cooling needs |
| 03:20.04 | coppice | for rental servers they run rack of old AMD K7 2000+ boxes, rather than a modern energy efficient design |
| 03:20.25 | omarc55 | Hi all, I am trying to setup asterisk and I keep getting Unable to open master device '/dev/zap/ctl' when running ztcfg, I know the device is being created in /lib/dev-state and not /dev. what could I be doing wrong? |
| 03:20.31 | DocHolliday | jql, chances are if you were willing to pay a reasonable fee for the power they wouldnt care |
| 03:20.38 | bkruse_home | omarc55: re-modprobe |
| 03:20.45 | bkruse_home | aka rmmod zaptel modprobe zaptel |
| 03:20.50 | bkruse_home | wait till you can ls /dev/zap |
| 03:20.53 | bkruse_home | ztcfg -vv |
| 03:20.59 | DocHolliday | most blade systems are 208V though |
| 03:21.26 | jql | DocHolliday: If they hadn't hit a cap on cooling in their existing facility, perhaps so. And the backup generator also serves as a good limiting factor... |
| 03:21.33 | coppice | 208 is a weird number |
| 03:21.56 | _paulos_ | DocHolliday, may be there are datacenter charging per Watt, mine is not so I dont care about power consumption or heat. |
| 03:21.59 | DocHolliday | jql, thats a whole other can of worms all together |
| 03:21.59 | omarc55 | I tried that and /dev/zap doesn't get created. |
| 03:22.19 | bkruse_home | udev problem |
| 03:22.24 | bkruse_home | you running rhel? |
| 03:22.27 | jql | no amount of monthly fees will make a bigger generator an easy purchase. The thing's already bigger than my apartment |
| 03:22.27 | bkruse_home | dmesg, is zaptel running? |
| 03:22.37 | omarc55 | gentoo |
| 03:22.39 | omarc55 | yes, zaptel is running |
| 03:22.42 | bkruse_home | oh.......... |
| 03:22.45 | DocHolliday | jql, yes it will.. the power cost builds it generator costs |
| 03:22.49 | bkruse_home | eww, check the scripts |
| 03:23.08 | DocHolliday | every amp of power you buy includes hvac, ups, generator etc |
| 03:23.12 | bkruse_home | not sure where its at in gentoo, /etc/modprobe.d/blah blah, doesnt sound like z aptel problem |
| 03:23.12 | jql | well, I said easy, not necessarily possible. :) |
| 03:23.18 | bkruse_home | if it is, report a bug, bugs.digium.com |
| 03:23.23 | DocHolliday | heh |
| 03:23.51 | omarc55 | ok, will do. |
| 03:23.55 | JT | how much fuel to most datacentres in the states have on hand? |
| 03:24.46 | DocHolliday | JT, usually they agreements for fuel trucks to replace consumed fuel |
| 03:25.00 | kuku5 | how do I get dhcpd installed |
| 03:25.00 | kuku5 | ? |
| 03:25.04 | JT | yeah, but they still need fuel onsite or there's issues |
| 03:25.05 | jql | JT: depends on how quickly they'd expect a resupply. Even New Orleans got fuel shipments the first week |
| 03:25.35 | kuku5 | As in, I have an asterisknow box, and there is not dhcpd (weird), so Now I need to install it, any suggestions ? |
| 03:25.37 | coppice | of course most power problems occur in winter, when the fuel trucks have trouble getting to them :-) |
| 03:25.48 | JT | just looking at some d/c specs, a rather large facility here in sydney, australia here keeps 900000L of fuel onsite with agreements for more |
| 03:25.58 | _paulos_ | Here in Brazil its worst: eletricity is somewhat cheap and clean (90% hydroelectric). |
| 03:26.26 | coppice | its not so clean. hydro has major pollution problems |
| 03:27.12 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 03:27.31 | _paulos_ | well... hidro is way cleaner than burning coal... |
| 03:27.54 | coppice | hydro produces greater global warming than coal |
| 03:28.26 | kuku5 | ...anyone? |
| 03:28.56 | Dovid | kuku5: what do u need to install ? |
| 03:28.57 | jql | kuku5: you're sure it's not there? ls /usr/sbin/*dhcp* |
| 03:29.15 | _paulos_ | coppice, you mean due to plant material in flooded areas decaying in an anaerobic environment, and forming methane? |
| 03:29.21 | coppice | but coal produces more radioactive pollution. you're screwed whatever you do |
| 03:29.32 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 03:29.46 | Aces1Up | just a quick question, whats the easiest way to set the root password on mysql? |
| 03:29.53 | coppice | _paulos_ yes. coal -> CO2, but hydro -> methane which is much worse than CO2 |
| 03:29.57 | jql | well, fusion also has some radioactive byproducts as well, but it's too cool not to do |
| 03:31.08 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:31.30 | coppice | jql: the biggest problem with fusion would probably be disposing of th irradiated plant at the end of its life. this happens with every accelerator, and they usually dump the old radioactive material somewhere into the next generation accelerator system as a target :-) |
| 03:33.43 | wunderkin | but... 3 eyed fish are cool! haven't you seen the simpsons? |
| 03:34.21 | coppice | they won't be cool with all this global warming |
| 03:35.08 | jql | well, the inedible fish which remain will end up suffering from warm, dead oceans |
| 03:35.15 | jql | serves them right for not being tasty |
| 03:38.27 | _paulos_ | better spare my dilithium crystals. |
| 03:38.38 | kuku5 | jql: its not there |
| 03:40.15 | coppice | _paulos_ maybe brazil just needs to oxygenate its reservoirs. London did this to revitalise the river Thames :-) |
| 03:41.57 | _paulos_ | Brazil is blessed with so much water that most people her doesnt realizes its a finite resource. |
| 03:42.37 | coppice | not just finite. if you don't care for it, it turns into a pollution nightmare |
| 03:42.40 | _paulos_ | s/her/here/ |
| 03:43.18 | jql | kuku5: I'm not sure how you install new packages on rpath, but you should look for a dhcp package somewhere. maybe google rPath dhcp? |
| 03:43.40 | wunderkin | rpath uses conary or something like that |
| 03:43.51 | aptura | talking green house gases? |
| 03:43.52 | aptura | :) |
| 03:45.42 | _paulos_ | what is this smell??? |
| 03:46.31 | kuku5 | so this is rpath, is this asterisk now worth using? |
| 03:46.59 | aptura | _paulos_ what part of Brazil do you live in? |
| 03:47.01 | kuku5 | somehow i think it will create problems for me |
| 03:47.13 | coppice | the dry part |
| 03:47.15 | kuku5 | ok. time to install fedore |
| 03:47.35 | _paulos_ | I live ast São Paulo |
| 03:47.54 | aptura | I see |
| 03:48.06 | coppice | Ah, São Paulo, a hotbed of Unicall users :-) |
| 03:48.07 | aptura | Winter is comming to Sao right? |
| 03:48.22 | _paulos_ | outumm |
| 03:49.11 | d00gster | guys, is it possible to create a incoming call rule based on the dialed number (to field of the sip invite)? I have a secondery number -alias- and I want that to ring a separate extension. |
| 03:49.14 | _paulos_ | Winter is by July |
| 03:49.37 | coppice | can't you ski all year round in the Andes? |
| 03:50.04 | coppice | you would get altitude training at the same time |
| 03:50.13 | _paulos_ | Brazil has almost no snow. |
| 03:50.15 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f663:0:20e:a6ff:fec4:4e5d) |
| 03:50.21 | luke-jr_ | how can I see what codec a channel is using? |
| 03:50.36 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:50.56 | coppice | we have no snow at all. my kids keep wanting to see some |
| 03:50.56 | Qwell | show channel blah |
| 03:51.59 | coppice | Error: channel blah does not exist. Hit any key |
| 03:52.37 | luke-jr_ | Qwell: don't see it there |
| 03:53.16 | *** join/#asterisk bmg505 (n=leon@196.209.248.226) |
| 03:53.20 | _paulos_ | coppice, where do you live? |
| 03:53.28 | coppice | HK |
| 03:53.55 | luke-jr_ | Qwell: what am I looking for? |
| 03:55.07 | bkruse_home | coppice is a bot fool! |
| 03:57.14 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 03:57.34 | bkw_ | coppice, isn't a bot you fool! |
| 03:57.54 | bkw_ | coppice, you see ityet? |
| 03:57.55 | hohum | hey |
| 03:58.04 | ez` | asterlink always down ;( |
| 03:58.17 | hohum | from a pure SIP prespective would there ever be a reason to write seperate parsers for the From: To: and Contact: headers? |
| 03:58.20 | coppice | bkw_: nope |
| 03:58.23 | bkw_ | ez`, must just be you or your ISP.. it works everywhere else |
| 03:58.29 | bkw_ | ez`, you on pppoe? |
| 03:58.32 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:58.52 | ez` | no; adsl |
| 03:59.01 | Qwell | bkw_: actually, it doesn't work here :p |
| 03:59.12 | Qwell | not resolving |
| 03:59.12 | ez` | hhe thanks Qwell |
| 03:59.14 | bkw_ | Qwell thats nice it works everywhere else. |
| 03:59.27 | aptura | bkw back in business? |
| 03:59.32 | bkw_ | never was out of business |
| 03:59.37 | ez` | actualy unreachabl since 2 day now ... |
| 03:59.39 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 03:59.46 | Qwell | bkw_: dig @4.2.2.2 asterlink.com |
| 03:59.48 | Qwell | not responding |
| 03:59.48 | bkw_ | ez`, i'm on sbc/att dsl and I can reach it |
| 03:59.54 | bkw_ | and you can't use 4.2.2.2 to tell |
| 03:59.54 | elriah | Hey guys, in sip.conf, how do I bind to multiple ips with bindaddr? |
| 04:00.11 | bkw_ | its th emost used nameserver on the planet |
| 04:00.13 | bkw_ | :P |
| 04:00.18 | Qwell | exactly :P |
| 04:00.20 | _paulos_ | coppice is the guy who taught Graham Bell a few tricks. |
| 04:00.25 | Qwell | and it doesn't work for them :P |
| 04:00.31 | ez` | bkw_, i tested it with 3 diff isp ; all same .. ; asterlink.com = game over ;( |
| 04:00.40 | coppice | _paulos_ I ain't quite that old :-\ |
| 04:01.08 | kuku5 | So how not stable is 1.4 ? |
| 04:01.11 | coppice | I can see asterlink.com, but its very slow |
| 04:01.13 | ez` | bkw_, of cours there is sumthing wron man |
| 04:01.54 | bkw_ | ez`, its hard to find since it works fine for everyone I have test it |
| 04:02.07 | luke-jr_ | how can I see what codec a channel is using? |
| 04:02.13 | AJaymn | is there someway to make sure Asterisk is running so if there is a system crash with the software it doesnt lock up the machine? like some kind of safemode? |
| 04:02.19 | aptura | coppice do a tracert to asteriskink web site or his sip server. |
| 04:02.21 | bkw_ | http://www.dnsreport.com/tools/dnsreport.ch?domain=www.asterlink.com |
| 04:02.22 | Qwell | bkw_: hows the route look from the nameservers to gtei? |
| 04:02.31 | coppice | ez`: try a traceroute. they sometimes still work |
| 04:02.32 | Qwell | that's likely to be the problem |
| 04:02.44 | Qwell | coppice: can't resolve it to traceroute ;) |
| 04:02.45 | bkw_ | yep looks like that is an issue |
| 04:02.55 | ez` | bkw_, some user here have the same issue ; talked to them todays .. |
| 04:02.58 | elriah | Or rather, can I bind to multiple addresses in sip.conf? |
| 04:03.12 | JT | AJaymn: safe_asterisk script |
| 04:03.15 | coppice | oh, the nameserving is the *really* slow part for me. took a minute or more |
| 04:03.19 | bkw_ | no that traces |
| 04:03.27 | bkw_ | finally |
| 04:03.37 | bkw_ | <PROTECTED> |
| 04:04.02 | _paulos_ | coppice, but you seems to know so much about telephony that sometimes it looks like you have 100 years in the field... |
| 04:04.11 | bkw_ | I see zero traffic coming from 4.2.2.2 into our network |
| 04:04.31 | Qwell | heh, I still have a hardcoded IP for cogent.arishost.com in my hosts file.. I should remove that |
| 04:04.44 | aptura | _paulos_ that would be my instructor. guy had 30 years and any question you can toss at him he would give you a answer in a second ;) |
| 04:05.06 | coppice | _paulos_: its amazing how slowly the old stuff dies. you learn about some real ancient stuff by needing to interface to or emulate it |
| 04:05.16 | ez` | Qwell, hard coded = manualy add in your /etc/hosts ? ; is that what you mean ? |
| 04:05.22 | Qwell | yeah |
| 04:05.28 | ez` | k |
| 04:05.48 | elriah | Qwell: Is it possible to bind to more than one address in sip.conf? |
| 04:05.55 | Qwell | elriah: 0.0.0.0 |
| 04:05.59 | aptura | My instructor was giving us where some of the fiber was laid in washington. He knew it all :) |
| 04:06.03 | ez` | to vrything ;) |
| 04:06.21 | coppice | fiber? how modern :-) |
| 04:06.22 | elriah | Qwell: Ok, thanks. Will it send sip packets back on the receiving interface or first interface? |
| 04:06.57 | _paulos_ | coppice, how long are you in china? |
| 04:07.04 | coppice | since 1991 |
| 04:07.15 | bkw_ | ez`, i'm looking at this again.. |
| 04:07.25 | aptura | coppice are you native to the mainland? |
| 04:07.49 | coppice | I 'm one of those rare tall blonde blue eyed asians |
| 04:08.05 | aptura | I see. :) |
| 04:08.08 | _paulos_ | its a lot of time... |
| 04:08.48 | ez` | bkw_, why are you not beleiving me ; because i am from quebec, canada ... hehhe |
| 04:09.21 | bkw_ | ez`, no |
| 04:09.34 | mitcheloc | hi coppice, can i have your sip address? i chat you? |
| 04:09.49 | coppice | SIP address? what's that? |
| 04:10.04 | _paulos_ | coppice, 1991 was before the transfer of sovereignty, wasnt it? |
| 04:10.51 | coppice | yep. didn't need any work permits. I could just come here and do as I liked then. I can now, but for different reasons |
| 04:11.18 | ez` | bkw_, 2 time now i told you about this issue ; ... well ; |
| 04:11.27 | aptura | ez, met some montrialians today. I did not speak there language but thay were in for a medical checkup. It was interesting thay were working for a petochemical company in the interior of bc laying geophones into the ground every kilometer or so and was paid 10 bucks a hour for 14 hours a day. Cheap petro companies! I gave them some sugestions on working in a better industry. |
| 04:11.45 | bkw_ | ez`, I have checked and rechecked this..works everywhere I try |
| 04:12.16 | ez` | bkw_, ... |
| 04:12.34 | coppice | bkw_==oistrich |
| 04:12.34 | bkw_ | what are your nameservers? |
| 04:12.37 | aptura | BTW what is the longest line of site test for wifi? |
| 04:12.55 | bkw_ | aptura, isn't it like 100 miles |
| 04:12.59 | aptura | k |
| 04:13.03 | ez` | aptura, montreal city got very nice woman by the way ;) |
| 04:13.03 | JT | it's line of siGHT, and at least 100 miles |
| 04:13.29 | ez` | bkw_, 142.169.1.16 |
| 04:14.09 | aptura | These kids would lay I think fiber above the ground and walk many kilometers though the interor then detonate explosives. Idea is the echo from deep undergroundwould echo back gas or oil deposits. I was think perhaps a wifi setup would be better in this case. |
| 04:14.41 | *** join/#asterisk avleen (n=avleen@pear.silverwraith.com) |
| 04:14.56 | coppice | aptura: I think they need strict timing |
| 04:15.00 | ez` | bkw_, this problem was not there last week |
| 04:15.11 | aptura | Come to think about it you are probebly right. |
| 04:15.26 | coppice | ez`: maybe it doesn't like the year of the pig :-) |
| 04:15.38 | avleen | hey folks, i was wondering.. what would the memory and CPU requirements be to run asterisk to handle (at most) 5 simultanious calls? |
| 04:15.44 | JT | damn superstition :/ |
| 04:16.03 | avleen | could i do it with 32Mb of mem? |
| 04:16.05 | ez` | coppice, hum; dunno my friend ... |
| 04:16.07 | ez` | ;) |
| 04:16.46 | ez` | coppice, i only know they pig years slow down so much my ebay order ; they do fiesta ! ;) party ... |
| 04:17.11 | aptura | ez, we have these new minning laws that states you can place a claim on private land as long as you mine it without the owners permission. Thanks to Premiere Gorden cambell anyone can do it. It cost 35 dollars and 17 cents a acre to have minning rights. |
| 04:18.06 | ez` | impressive |
| 04:18.36 | *** join/#asterisk ocgltd (n=support@CPE004063e0ee74-CM00159a010632.cpe.net.cable.rogers.com) |
| 04:18.40 | aptura | Very. |
| 04:19.09 | _paulos_ | i was born in a pig year |
| 04:19.28 | aptura | When it became law I think spring of last year it caused a huge amount of network traffic for companies to file claimes on BC govermnts web site. |
| 04:19.30 | JT | i don't know what "animal year" i was born in and don't care |
| 04:19.50 | JT | don't care about starsigns either |
| 04:19.52 | JT | all bullshit |
| 04:19.57 | ez` | bkw_, i asked to 5 user arround our small planet ; 4/5 told me ; you site is unreachable. .. |
| 04:19.58 | aptura | BC is doing very well in about every sector. |
| 04:20.20 | ocgltd | Can someone offer some debugging advice: I have an asterisk server acting as a gateway between SIP and H323 networks. Audio traffic originating on the H323 leg and terminating on the SIP legs has great audio, both ways. Traffic originating on the SIP leg and terminating on the H323 has incredibly choppy audio. Any ideas? |
| 04:20.21 | coppice | JT: it is *not* bullshit. it gets us 3 days holiday |
| 04:20.26 | aptura | ez what is your ip address |
| 04:20.30 | JT | hah |
| 04:20.37 | JT | i get no holidays from it |
| 04:20.41 | aptura | the site name that is having issues |
| 04:20.49 | CrashHD | good evening everyone |
| 04:20.53 | Qwell | JT: Then you aren't doing it right |
| 04:21.10 | ez` | asterlink.com |
| 04:21.10 | coppice | JT: move to asia |
| 04:21.24 | ez` | aptura, www.asterlink.com |
| 04:21.30 | JT | as if, that's be worse, people who actually care about this superstitious stuff |
| 04:21.34 | aptura | mmm 6.3 billion dollars in revenue in BC last year. |
| 04:21.36 | JT | that'd |
| 04:21.44 | aptura | in minning revenue. |
| 04:23.06 | _paulos_ | move to Brazil, this week was "carnaval", 4 days holiday, beautiful women naked all over. |
| 04:23.30 | JT | brazil is way too hot and humid for my liking |
| 04:23.48 | _paulos_ | JT: mine too... |
| 04:23.57 | *** join/#asterisk generalhan (n=Red_Drag@ip72-204-242-138.ph.ph.cox.net) |
| 04:24.02 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 04:24.07 | generalhan | whats going on all ? |
| 04:24.17 | ez` | _paulos_, i wanna see it ;) ; canadian women ; are not naked this time on year ; its 10 celcius below zero ... 4 feet of snow ;)) |
| 04:24.19 | coppice | but the hot women compensate for that, don't they? |
| 04:24.25 | bkw_ | ez`, dig @208.67.222.222 www.asterlink.com |
| 04:24.41 | DocHolliday | i was given a workaround for Asterisk 1.2 in order to get the Cisco 7941G message waiting indicator working, if i upgrade to asterisk 1.4 will that still work? |
| 04:24.47 | bkw_ | ez`, or dig @208.67.220.220 www.asterlink.com |
| 04:25.09 | aptura | 96 ms here bkw |
| 04:25.51 | bkw_ | ez`, I asked people all over also but I can't find a problem.. I have restarted.. reloaded.. bitch smacked.. traced .. everything |
| 04:26.17 | *** part/#asterisk avleen (n=avleen@pear.silverwraith.com) |
| 04:26.29 | _paulos_ | DocHolliday, * changed a lot... Most like you will have to fix something in the old fix... |
| 04:26.37 | generalhan | i need some help with setting up a remote Cisco 7960 to my asterisk box. im pretty sure that i forwarded all the ports i needed, but for some reason it just doesnt get to the * machine. anyone have a good site to help walk me through it ? |
| 04:27.35 | ez` | bkw_, i added it to my hosts -> 208.67.220.220 , this on werk now |
| 04:27.45 | DocHolliday | _paulos_, in order to upgrade from 1.2 to 1.4 can i just download to the new tar files and compile them? |
| 04:28.13 | ez` | coppice, yes they really do ; our quebecoise ( quebec woman are very very hot .... ) |
| 04:29.16 | coppice | asia has most of the really hot women |
| 04:29.34 | JT | lies |
| 04:29.54 | ez` | coppice, i agree man ; i taste it only 1 time ; but i can remember ... ;) |
| 04:29.54 | coppice | why else would I live here? |
| 04:30.30 | bkw_ | ez`, did this start about 7 days ago? |
| 04:30.35 | DocHolliday | do people recommend starting with a fresh box for asterisk 1.4? |
| 04:31.19 | bkruse_home | DocHolliday: why not? |
| 04:31.20 | ez` | bkw_, i cant tell you exactly ; but this is a real issue ... |
| 04:31.33 | bkw_ | ez`, I think I figured it out.. its an issue with the webservers only |
| 04:31.35 | DocHolliday | bkruse_home, is it only 1.4 that supports T.38? |
| 04:31.42 | bkw_ | they wereall rebooted 7 days ago |
| 04:31.49 | bkruse_home | DocHolliday: t.38 passthru |
| 04:32.05 | ez` | bkw_, ok i give a try with voip iax ... |
| 04:32.07 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:32.07 | DocHolliday | bkruse_home, correct (for faxing |
| 04:32.09 | _paulos_ | DocHolliday, I think you will have to correct the patches. |
| 04:32.12 | bkruse_home | right |
| 04:32.18 | bkw_ | ez`, that should have always beenworking |
| 04:32.20 | bkruse_home | i believe so, but im not sure how HARD it would be to backport |
| 04:32.55 | DocHolliday | bkruse_home, my problem is that i had to comment out one line in chan_sip to make the MWI work on my cisco phones, can i do the same with asterisk 1.4? |
| 04:33.38 | bkw_ | doesn't help that one of the three webservers has three default gw's on it |
| 04:33.41 | *** join/#asterisk rene- (n=rene-@200.34.66.137) |
| 04:33.47 | rene- | hi gnite |
| 04:34.31 | ez` | bkw_, ;P |
| 04:34.57 | rene- | guys do you know of any wlan gear that would allow me to create 'vlans' as in two different ssids ? and then to route one vwlan using one wired interface and to another using a second one? |
| 04:36.18 | bkw_ | odd it rebooted and add it 4 times |
| 04:36.21 | bkw_ | dumb thing |
| 04:36.38 | bkruse_home | DocHolliday: ha, lets hope its that easy |
| 04:36.49 | DocHolliday | bkruse_home, any suggestions? |
| 04:37.03 | bkruse_home | let me check |
| 04:37.39 | rene- | well maybe that was too much to ask maybe an access point that can do roaming so wisip phones dont drop the connections? |
| 04:37.43 | ez` | bkw_, do you recommend m to use sip or iax with asterlink ??? |
| 04:38.04 | rene- | vwlan that sounds stupid doesnt it |
| 04:38.05 | DocHolliday | bkruse_home, appreciated as usual |
| 04:38.05 | JT | rene-: wifi sip phones are super dodgy |
| 04:38.09 | bkruse_home | :] |
| 04:38.25 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:39.24 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 04:39.36 | bkw_ | ez`, we recommend SIP right now |
| 04:39.51 | bkw_ | ez`, i'll have some new stuff to test out soon if you wanna try that |
| 04:39.53 | DocHolliday | <PROTECTED> |
| 04:39.58 | DocHolliday | that was the line i commented |
| 04:40.07 | bkw_ | just remove (0/0) |
| 04:40.10 | Qwell | DocHolliday: that's gonna remove the message entirely |
| 04:40.13 | bkw_ | the rest is right |
| 04:40.14 | Qwell | you don't want that |
| 04:40.20 | bkw_ | just (0/0) is WRONG |
| 04:40.26 | bkruse_home | DocHolliday: |
| 04:40.27 | bkruse_home | http://bugs.digium.com/view.php?id=5090 |
| 04:41.12 | DocHolliday | Qwell so just remove from (0/0) ? |
| 04:41.44 | _paulos_ | "Voice-Message: %d/%d\r\n" |
| 04:42.01 | ez` | bkw_, i will try sip , so . brb |
| 04:42.05 | DocHolliday | thats how it should read? |
| 04:42.22 | DocHolliday | Qwell, if so can i do it just like that with Asterisk 1.4? |
| 04:42.56 | _paulos_ | try it, its faster than asking. |
| 04:43.39 | ez` | bkw_, do i actualy need to add all this 7 servers ?? [asterlink-switch-01] @ [asterlink-switch-07] ; what happen if i only add one .,,, |
| 04:43.45 | DocHolliday | _paulos_, well i want to install the system from scratch heh |
| 04:43.53 | bkw_ | ez`, you can use just one if you like |
| 04:44.01 | ez` | k |
| 04:44.08 | bkw_ | ez`, that will be changing soon but i'll email everyone the beta info |
| 04:44.16 | ez` | k |
| 04:44.21 | ez` | brb 2 min |
| 04:44.39 | bkw_ | ez`, I have been working on adding FreeSWITCH to our stuff :P |
| 04:44.49 | bkw_ | i'm going to be using it for the IAX stuff |
| 04:45.18 | *** part/#asterisk rene- (n=rene-@200.34.66.137) |
| 04:45.44 | bkw_ | you guys wanna test something out? 1-712-872-3350 |
| 04:46.02 | *** join/#asterisk tb0301s (n=sowa@brln-4db1188a.pool.einsundeins.de) |
| 04:46.04 | aptura | Who wants to be a billionare? |
| 04:46.19 | bkw_ | aptura, I wish :P |
| 04:46.20 | aptura | Joking of course :) |
| 04:46.47 | *** join/#asterisk jpalmer (n=scorpio@fl-209-26-20-205.sta.embarqhsd.net) |
| 04:46.58 | aptura | But working up north in BC or Alberta and the money is very good. Met a 21 year old making 80k plus a year working in Alberta. |
| 04:47.04 | coppice | well, I suppose if I have to be one I could cope with it |
| 04:47.15 | jql | I will accept the burden |
| 04:47.29 | DocHolliday | aptura, not bad at all |
| 04:47.46 | aptura | with some years expraince in heavy industry try 120k a year. |
| 04:47.55 | aptura | :) |
| 04:47.56 | DocHolliday | crazy |
| 04:48.04 | aptura | Thats insane I know. |
| 04:48.17 | DocHolliday | and Centos 4.4 |
| 04:48.18 | coppice | yeah, but it takes 118k a year for heating fuel :-) |
| 04:48.22 | aptura | BC is a very RICH resource province. |
| 04:49.07 | jql | the console debugging of 1.4 alone made it so I feel pain using 1.2 |
| 04:49.18 | bkruse_home | DocHolliday: centos? lame. |
| 04:49.20 | bkruse_home | debian! |
| 04:49.35 | DocHolliday | bkruse_home, i have only used debian a handful of times |
| 04:49.37 | jql | it really makes me cringe not having context/extension/priority on every single console line |
| 04:49.46 | DocHolliday | besides for a box going into production i dont like taking chances |
| 04:49.50 | bkw_ | bitching about what distro someone uses is pointless |
| 04:50.31 | bkw_ | Anyone ever get asterisk working in Xen? |
| 04:50.41 | DocHolliday | bkw_, agreed, some may be better than others but it really comes down to how experienced you are. |
| 04:50.49 | bkw_ | DocHolliday, yep |
| 04:51.04 | bkw_ | moving on.. I wonder if anyone has gotten asterisk working in Xen |
| 04:51.11 | Qwell | well, you wouldn't run a server on lindash |
| 04:51.12 | DocHolliday | heh not me. |
| 04:51.14 | jql | I've wanted to get one in Xen, but haven't had a chance |
| 04:51.29 | Qwell | or, whatever the hell it's called now |
| 04:51.33 | DocHolliday | Qwell, i guess i didn't have to mail you that 7941 after all ;) |
| 04:52.02 | bkw_ | jql, the only thing about it is meetme won't work in xen I suspect |
| 04:52.50 | CunningPike | bkw_: Just about to set one up on openvz |
| 04:53.03 | bkw_ | CunningPike, tell me if you get meetme working in there |
| 04:53.29 | CunningPike | bkw_: OK - we're not planning to use this server for that, but I can sure see if it works |
| 04:53.41 | aptura | Evening CunningPike |
| 04:53.47 | CunningPike | Hey, aptura |
| 04:54.00 | bkw_ | CunningPike, I suspect it won't unless you can get zaptel loaded |
| 04:54.01 | aptura | CunningPike you ever follow the minning news up nort? |
| 04:54.07 | aptura | north :) |
| 04:54.21 | bkw_ | raise your hand if you wish you could use conferencing without the zaptel requirement? |
| 04:54.21 | CunningPike | aptura: Drinking again, are we? |
| 04:54.54 | coppice | bkw_ been there. done that :-) |
| 04:55.00 | bkw_ | coppice, same here :P |
| 04:55.03 | omarc55 | I am trying to setup some phones on asterisk 1.4, but when I pick up the phone and dial an extension, asterisk just hangs up.. the console says: -- Starting simple switch on 'Zap/2-1' -- Hungup 'Zap/2-1'. any ideas what could be wrong? |
| 04:55.07 | DocHolliday | shame to pull down a box thats been up 182 days just to reinstall the OS :( |
| 04:55.21 | bkw_ | the truth is you don't have to have a perfect clock either for it to work :P |
| 04:55.28 | aptura | CunningPike :) I dont drink but like the idea of a province's economy that is very diverisfied. |
| 04:55.32 | jql | pentiums has clocks |
| 04:56.31 | *** join/#asterisk sharp (n=sharp@2001:470:1f01:ffff:0:0:0:1c23) |
| 04:58.44 | *** join/#asterisk ManxPower (n=manxpowe@110.sub-70-212-217.myvzw.com) |
| 05:01.23 | *** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net) |
| 05:08.06 | Aces1Up | anyone know how i access the flash operator panel in asterisk? |
| 05:08.20 | Aces1Up | i just want to know if its is installed in my system and running correctly. |
| 05:09.33 | ManxPower | Aces1Up: did you follow the instructions? |
| 05:09.56 | Aces1Up | manx just point me in to the right page and i will follow the instructions :) |
| 05:10.21 | ManxPower | Well where did you download it from? |
| 05:10.57 | ManxPower | Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
| 05:10.58 | Aces1Up | well i'm running trixbox, and just am not sure if it is already installed or not. |
| 05:11.09 | ManxPower | Aces1Up: we cannot help you with Trixbox |
| 05:11.24 | JT | bkw_: is that a public conference server, that number you posted? |
| 05:11.33 | bkw_ | JT yes |
| 05:11.47 | bkw_ | JT thats freeswitch running on the Amazon Elastic Compute Cloud |
| 05:11.59 | bkw_ | http://freeswitch.dyndns.org/ |
| 05:12.35 | JT | ah ok |
| 05:12.46 | coppice | I think i'd be too embarassed to run my code on something called the Amazon Elastic Compute Cloud :-) |
| 05:13.53 | SwK | did he post 3350? |
| 05:16.59 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 05:21.09 | bkw_ | woooohooo |
| 05:21.15 | bkw_ | swk you see the website? |
| 05:22.52 | DocHolliday | anyone here running grsecurity on their asterisk box? |
| 05:23.19 | kuku5 | how unstable is 1.4 ? |
| 05:23.20 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 05:24.07 | ManxPower | kuku5: If you like your job don't run it. |
| 05:25.28 | SwK | yeah I just looked at it hahah |
| 05:25.50 | coppice | Oh, brave new world, that hath such people in it :-) |
| 05:26.41 | ez` | bkw_, good news ; you website is back |
| 05:28.24 | DocHolliday | do you technically need zaptel or libpri if you are not using any cards? |
| 05:29.38 | aptura | I havnt had a issue. |
| 05:30.02 | *** join/#asterisk bengl (n=bengl@CPE001346f74fbd-CM0011aec8496e.cpe.net.cable.rogers.com) |
| 05:30.20 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 05:30.22 | bkw_ | ez`, good deal |
| 05:30.43 | DocHolliday | aptura, is there a howto for asterisk 1.4? |
| 05:30.59 | DocHolliday | got it |
| 05:31.15 | aptura | I suspect there is |
| 05:32.25 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 05:35.50 | Qwell | dseeb_: around? |
| 05:35.55 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 05:36.08 | dseeb_ | Qwell: yep |
| 05:36.14 | Qwell | dseeb_: fyi, I'm working on headset support |
| 05:36.24 | dseeb_ | oh, ok |
| 05:36.28 | Qwell | I can get one-way audio so far...and it's completely hacked in |
| 05:36.29 | dseeb_ | cool |
| 05:36.33 | Qwell | but it works :p |
| 05:36.35 | Qwell | ...sorta |
| 05:36.48 | dseeb_ | in chan_cellphone? or something different |
| 05:36.52 | Qwell | in chan_cellphone |
| 05:37.20 | dseeb_ | cool, i just posted a new patch, fixes some issues and has the start of SMS support |
| 05:37.24 | Qwell | I saw. :) |
| 05:37.37 | Qwell | s'why I figured you were here |
| 05:37.43 | dseeb_ | ah, ok |
| 05:37.59 | dseeb_ | who do i talk to to get a code-review happening? |
| 05:38.12 | Qwell | I've been reviewing it as I've been going |
| 05:38.25 | dseeb_ | ah, ok. good. |
| 05:39.12 | Qwell | I went and rewrote your rfcomm_read func too.. that static char bit me pretty hard when I connected multiple devices |
| 05:39.24 | Qwell | I'll post a patch with that tomorrow probably |
| 05:39.34 | dseeb_ | great |
| 05:40.03 | Qwell | it's good though so far, I like it |
| 05:40.49 | dseeb_ | think it will make mainline? |
| 05:41.04 | Qwell | I don't know. That depends on licensing (of libbluetooth) |
| 05:41.22 | dseeb_ | ah, ok |
| 05:41.40 | Qwell | if nothing else, it'll probably go into -addons |
| 05:42.10 | dseeb_ | what does the libbluetooth license need to be? |
| 05:42.18 | Qwell | lgpl would be great |
| 05:42.29 | dseeb_ | dunno what it is, might find out |
| 05:42.34 | Qwell | there is conflicting information about what it really is.. I haven't researched it much |
| 05:42.57 | Qwell | Did you become a SIG member? Tons of documentation there |
| 05:43.26 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 05:43.31 | dseeb_ | bluetooth sig? |
| 05:43.34 | Qwell | yeah |
| 05:43.42 | dseeb_ | no, i should. |
| 05:43.49 | Qwell | there is a free membership |
| 05:43.59 | Qwell | which gives you access to all the specifications |
| 05:44.15 | dseeb_ | cool, i got my spec docs from there. ill sign up |
| 05:44.20 | JT | i have a pile of the bluetooth specs all printed out here... somewhere |
| 05:45.37 | dseeb_ | just looking at rfcomm_read(), thats bad, cant believe i did that.. |
| 05:45.40 | Qwell | dseeb_: I didn't see where you implemented the dialing from the AG... where would that be? |
| 05:45.44 | Qwell | heh, yeah :) |
| 05:46.25 | dseeb_ | ah, it detects if theres a call setup and it did not send the dial, if so it will disconnect |
| 05:47.40 | Qwell | ahh |
| 05:47.41 | dseeb_ | if figured if the user is dialling from the cellphone itself, theres a fair bet they are about to walk off |
| 05:47.41 | *** join/#asterisk Piano_ (n=Piano@c-67-175-92-171.hsd1.il.comcast.net) |
| 05:48.56 | kuku5 | Do I still need to change the header file if I want to use ztdummy ? |
| 05:49.06 | kuku5 | or will * do it automatically if nothing is there |
| 05:49.57 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 05:49.58 | DocHolliday | qwell, preparing to kill a machine thats been up for half a year :( |
| 05:51.08 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:52.49 | *** join/#asterisk foobar778 (i=johhny@ip68-100-41-120.dc.dc.cox.net) |
| 05:54.23 | Aces1Up | anyone here have much experience with a2billing? |
| 05:54.32 | Aces1Up | i need some help. |
| 05:54.40 | foobar778 | Any one have any opinions on inexpensive fxs/fx0 atas or fxs/fx0 converters under 100 dollars with ease of use with asterisk |
| 05:56.20 | JT | fxo damnit |
| 05:56.25 | JT | spa-3102 is around that mark |
| 05:56.51 | foobar778 | yes |
| 05:57.00 | foobar778 | read about it good reviews |
| 05:57.16 | foobar778 | read bad stuff in grandstream 488 |
| 05:57.35 | foobar778 | VPC1000 FXS to FXO Port Converter Jt what about |
| 05:57.47 | foobar778 | good price |
| 05:58.34 | foobar778 | seems so easy really no need for asterisk setup if fxs us setup in ata |
| 05:58.46 | [TK]D-Fender | foobar778 : SPA-3102 is "not bad". Depending where you are and a few toher random circumstances, you might be quite happy with it. |
| 05:58.56 | [TK]D-Fender | foobar778 : it IS very flexible, thats for sure. |
| 05:59.14 | foobar778 | Fender u talking about the converter |
| 05:59.25 | [TK]D-Fender | foobar778 : yes, the one I mentioned |
| 05:59.48 | foobar778 | u mentioned the supura |
| 06:00.13 | foobar778 | VPC1000 FXS to FXO Port Converter what about that Fender |
| 06:00.25 | foobar778 | any opinion? |
| 06:00.51 | [TK]D-Fender | foobar778 : Thats a real "hack", and only help change a port you have from giving voltage, to accepting it. Unadvised. |
| 06:00.52 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 06:01.09 | Qwell | ugh |
| 06:01.15 | Qwell | what a useless toy |
| 06:01.19 | [TK]D-Fender | foobar778 : Considering the base port may not be capable of sending out DTMF in a natuarl way to dial, etc... jsut not "healty" |
| 06:01.19 | foobar778 | ok so not good with the converter |
| 06:01.36 | [TK]D-Fender | foobar778 : I would highly advise against it. |
| 06:01.36 | foobar778 | ok gotcha |
| 06:01.47 | foobar778 | supura then huh |
| 06:02.07 | [TK]D-Fender | foobar778 : Linksys technically. |
| 06:02.17 | foobar778 | everything else seems to get very [pricy |
| 06:02.26 | Qwell | ~ygwypf |
| 06:02.35 | jbot | i guess ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 06:02.37 | Qwell | jbot: You suck |
| 06:02.39 | jbot | no, *you* suck! |
| 06:02.44 | [TK]D-Fender | foobar778 : I suppose it depends on your idea of pricy and consider the number produced, vs prospective clients. |
| 06:02.51 | Qwell | oh sure, now he answers fast |
| 06:03.39 | foobar778 | Fender I mean when u start getting more than one fx0 ports the price goes to 3 4 5 hundred do;;ars |
| 06:04.13 | [TK]D-Fender | foobar778 : Ask yourself how much a real phone system would cost to support multiple lines & offer what * does. |
| 06:04.26 | [TK]D-Fender | foobar778 : YGWYPF.... |
| 06:04.47 | Qwell | I should add an exception to the YGWYPF addage |
| 06:04.58 | [TK]D-Fender | foobar778 : And depending on what models you choose, a 4-port card would be under $400 |
| 06:05.03 | Qwell | Qwells Exception to the YGWYPF Addage |
| 06:05.05 | Qwell | : |
| 06:05.20 | Qwell | Unless it costs more than 5x the rest of the competition |
| 06:05.34 | foobar778 | BuThe power of an fxo port gateway is valuable now I just have a pstn passthrough but again one phone line versus many soho versus enterprise is the question |
| 06:05.46 | [TK]D-Fender | Qwell : Which basically says "Don't you wish you DIDN'T buy Cisco?" :) |
| 06:05.59 | Qwell | [TK]D-Fender: I'm thinking more along the lines of Avaya, but okay |
| 06:06.09 | [TK]D-Fender | Qwell : Same shit, different smell :) |
| 06:06.39 | [TK]D-Fender | foobar778 : Well the cost scales differently depending on your application. |
| 06:06.48 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 06:07.01 | [TK]D-Fender | foobar778 : The SPA's FXO may be too hit/miss for many companies. |
| 06:07.26 | foobar778 | are u saying its not reliable? |
| 06:07.34 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:07.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 06:08.29 | foobar778 | or are u saying that the many simultaneous calls it cant handle |
| 06:09.18 | [TK]D-Fender | foobar778 : That the audio quality / echo cancellation capabilities aren't always good enough for corporate use. |
| 06:09.45 | [TK]D-Fender | foobar778 : There is no comparison to a proper quality HWEC enabled FXO card. |
| 06:10.13 | kuku5 | Do I still need to change the header file if I want to use ztdummy ? |
| 06:10.21 | foobar778 | Im looking for a single home user with pstn gateway for friends and family at a reasonable price traliable and ease of use |
| 06:10.41 | foobar778 | FXO cards are expensive |
| 06:11.32 | *** join/#asterisk yonahw-work (n=yonahw@genie03-173-74.inter.net.il) |
| 06:12.06 | foobar778 | doesnt sipura have echo cancvellation |
| 06:12.40 | [TK]D-Fender | foobar778 : Well single port you'll probably be happy withthe SPA. Multi-port corporate use... probably not. |
| 06:12.55 | foobar778 | Echo Cancellation (G.165/G.168) on sipura |
| 06:13.13 | [TK]D-Fender | foobar778 : Its not the best, but like I said, might do well for you. |
| 06:13.31 | foobar778 | yea |
| 06:14.37 | DocHolliday | damned slow cd burner |
| 06:15.07 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 06:15.31 | foobar778 | http://afterburn.no-ip.info:8050/160.jpg |
| 06:15.37 | foobar778 | tahst what I have now |
| 06:15.45 | foobar778 | thats |
| 06:16.52 | [TK]D-Fender | foobar778 : looks like "life-line" use only.... can't sue as a real dual-direction FXO |
| 06:16.55 | [TK]D-Fender | use* |
| 06:16.56 | foobar778 | Not a gateway just a passthrough I, pretty sure tell me Im wrong I would be so happy |
| 06:17.14 | foobar778 | yes I thjink so Fender I agree |
| 06:17.22 | [TK]D-Fender | foobar778 : Sorry, your first assessment is just about right |
| 06:17.49 | foobar778 | yes so thatsa why Im looking for an alternative |
| 06:18.25 | *** join/#asterisk Flauto (n=zhao@adsl-69-212-194-128.dsl.chcgil.ameritech.net) |
| 06:21.54 | Flauto | anyone has problems with 1.4 would have error and stop running? |
| 06:22.40 | Flauto | [Feb 25 00:20:02] NOTICE[20701]: chan_sip.c:16441 reload_config: Can't add wildcard IP address to domain list, please add IP address to domain manually. |
| 06:22.40 | Flauto | <PROTECTED> |
| 06:22.46 | Flauto | i get this often too |
| 06:22.49 | Flauto | what does it mean |
| 06:22.50 | foobar778 | I had isuues with 1.4 one way audio yet same ports open on 1.214 no problem |
| 06:23.21 | Flauto | it seems 1.4 is still buggy |
| 06:24.35 | foobar778 | hey add free 411 to ur voip add dialplan to 1800free411 |
| 06:24.50 | foobar778 | addbased but free411 |
| 06:25.13 | foobar778 | cellphones will be happy |
| 06:25.44 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-151-204-137-181.ny325.east.verizon.net) |
| 06:26.09 | Flauto | 18003733411 |
| 06:26.11 | Flauto | ? |
| 06:26.20 | foobar778 | he how did u do that?? |
| 06:28.43 | Flauto | just put 411,1,dial(sip/18003733411@voip.trxte.com,60) |
| 06:28.59 | Flauto | trxtel |
| 06:29.11 | foobar778 | ok that will do |
| 06:29.25 | Flauto | i have not really used it much thoug |
| 06:29.26 | Flauto | h |
| 06:29.26 | *** join/#asterisk jjhall (n=chatzill@67.60.61.7) |
| 06:30.06 | Flauto | using fedora core 6 with 1.4 |
| 06:30.16 | Flauto | some minor problems |
| 06:30.42 | jjhall | I'm looking for a sound file for a project, and I've done some googling with no luck. The sound I need is an alarm that starts low for a second then sweeps high. I've heard it used on movies as a nuclear warning. Anybody know where I may be able to find it? |
| 06:30.48 | Flauto | though, gtalk does not seem working |
| 06:31.26 | foobar778 | how much is trixtel?? |
| 06:31.35 | foobar778 | Fender got a question |
| 06:31.44 | Flauto | ah? |
| 06:31.55 | Flauto | trxtel is paying you to use its service |
| 06:32.16 | foobar778 | I noticed that my frree batamaz voip providers allow simultaneous calls from the same user |
| 06:32.28 | foobar778 | How Flauto |
| 06:32.49 | Flauto | check out www.trxtel.com |
| 06:32.53 | foobar778 | I did |
| 06:33.01 | foobar778 | but havent read all yet |
| 06:33.06 | foobar778 | <PROTECTED> |
| 06:33.28 | Flauto | i have not got paid so far |
| 06:33.33 | Flauto | so, i dont' really know |
| 06:33.35 | Flauto | how it works |
| 06:34.00 | foobar778 | how do u start credit card? |
| 06:34.01 | Flauto | one thing for sure, i won't get rich from using it |
| 06:35.03 | Flauto | only if you can use it for millions of minutes every months |
| 06:38.26 | kuku5 | is there a problem with 1.2 and 2.6 kernel ? |
| 06:43.15 | fetcher | kuku5: no. Running fine here at several sites for over a year (albeit mostly without Zaptel) |
| 06:45.14 | [TK]D-Fender | kuku5 : nothing I've heard of. |
| 06:46.01 | kuku5 | do i need to compile zaptel if im not using any cards ? |
| 06:46.19 | kuku5 | I need ztdummy for timing right ? |
| 06:46.24 | [TK]D-Fender | kuku5 : Yes if you intend on using IAX2 trunking, or MeetMe |
| 06:46.52 | kuku5 | i get errors compilng |
| 06:47.20 | kuku5 | http://pastebin.ca/371313 |
| 06:49.05 | kuku5 | Any help would be great about now |
| 06:50.57 | kuku5 | ... |
| 06:55.59 | JT | kuku5: maybe no-one knows the answer or is around? being demanding won't help |
| 06:56.20 | kuku5 | probably not :( |
| 07:07.10 | Aces1Up | can someone point me to a good guide on dial rules? |
| 07:07.26 | Aces1Up | on how i set up dial rules for outgoing calls? |
| 07:07.31 | Aces1Up | i'm confused on all of it. |
| 07:07.38 | Aces1Up | is there a good guide somewheere? |
| 07:08.34 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
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| 07:27.13 | CunningPike | ~thebook |
| 07:27.18 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:31.36 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 07:49.52 | suma | i got a problem with the addons |
| 07:52.59 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:53.44 | kuku5 | How can I test if Sip is accepting session ? |
| 07:54.02 | kuku5 | sip debug shows nothing, and I can't register with different clients... |
| 08:02.19 | *** join/#asterisk coppice (n=chatzill@106.206.17.210.dyn.pacific.net.hk) |
| 08:07.49 | kuku5 | nm, it didnt bind to an ip |
| 08:08.13 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 08:29.57 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
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| 08:41.06 | pardove | is there any way to control a call after dial command, for example dial xxxx and then wai for n seconds and then send DTMF then wait again and again send some dtmf |
| 08:46.57 | pardove | Is there any way to control a call after dial command? ex: Dial xxxx and then wait for n seconds and then send DTMF then wait again n secs and again send some dtmf and then hangup |
| 08:51.31 | *** join/#asterisk pardove (n=chatzill@80.191.117.98) |
| 08:54.11 | *** join/#asterisk nirz (n=nir@bzq-88-154-194-62.red.bezeqint.net) |
| 08:54.13 | pardove | is there anybody in? |
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| 08:57.43 | *** mode/#asterisk [+o mog] by ChanServ |
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| 09:27.31 | *** join/#asterisk CabDriver151 (n=cabdrive@ip68-228-214-151.ph.ph.cox.net) |
| 09:28.13 | CabDriver151 | morning |
| 09:28.29 | CabDriver151 | does anyone have any experience with Cisco 7960 phones? |
| 09:29.45 | *** part/#asterisk CabDriver151 (n=cabdrive@ip68-228-214-151.ph.ph.cox.net) |
| 09:29.52 | *** join/#asterisk CabDriver151 (n=cabdrive@ip68-228-214-151.ph.ph.cox.net) |
| 09:30.08 | CabDriver151 | anyone? |
| 09:33.25 | *** part/#asterisk CabDriver151 (n=cabdrive@ip68-228-214-151.ph.ph.cox.net) |
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| 09:37.53 | tzafrir | hmmm, maybe add to the /topic: /msg jbot ask ? |
| 09:45.24 | *** join/#asterisk flopp (n=noadress@ram94-5-82-237-196-239.fbx.proxad.net) |
| 09:53.54 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
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| 09:56.10 | *** join/#asterisk Mpls-Eric (n=ejo1@v-209-98-205-186.mn.visi.com) |
| 10:01.36 | kezza491 | why would all my attempts to dial asterisks get rejected? |
| 10:02.30 | kezza491 | all i get from my logs is just `[Feb 25 20:46:42] NOTICE[2286] chan_iax2.c: Rejected connect attempt from 192.168.0.2, who was trying to reach '1@'` |
| 10:05.07 | tzafrir | How exactly do you try to dial? from what client? |
| 10:05.20 | tzafrir | or is it a different Asterisk server? |
| 10:05.55 | kezza491 | clien |
| 10:06.12 | kezza491 | iaxComm i am just trying to fiddle around learn how asterisks works |
| 10:06.15 | Mpls-Eric | Help! I've just upgraded from cvs around Dec 2004 to 1.2.15 and my pri d-channel will not come up. Any ideas? |
| 10:06.58 | Mpls-Eric | Description Alarms IRQ bpviol CRC4 |
| 10:06.58 | Mpls-Eric | T4XXP (PCI) Card 0 Span 1 OK 0 0 0 |
| 10:06.58 | Mpls-Eric | T4XXP (PCI) Card 0 Span 2 OK 0 0 0 |
| 10:06.58 | Mpls-Eric | T4XXP (PCI) Card 0 Span 3 RED 0 0 0 |
| 10:06.58 | Mpls-Eric | T4XXP (PCI) Card 0 Span 4 RED 0 0 0 |
| 10:06.59 | Mpls-Eric | Sending Set Asynchronous Balanced Mode Extended |
| 10:07.01 | Mpls-Eric | > [ 00 01 7f ] |
| 10:07.03 | Mpls-Eric | > Unnumbered frame: |
| 10:07.05 | Mpls-Eric | > SAPI: 00 C/R: 0 EA: 0 |
| 10:07.07 | Mpls-Eric | > TEI: 000 EA: 1 |
| 10:07.09 | Mpls-Eric | > M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] |
| 10:07.11 | Mpls-Eric | > 0 bytes of data |
| 10:08.53 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 10:09.03 | *** join/#asterisk existx (n=existx@seriously.fuckshaw.ca) |
| 10:09.11 | *** join/#asterisk vanumo (n=sensury@80.122.72.250) |
| 10:09.21 | vanumo | hi :-) |
| 10:09.26 | *** part/#asterisk existx (n=existx@seriously.fuckshaw.ca) |
| 10:10.36 | vanumo | i have a working chan_cellphone, i can dial out but if i tell in i can't reach my sip phone ? |
| 10:11.27 | tzafrir | ~pb |
| 10:11.33 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 10:11.59 | *** join/#asterisk existx (n=existx@seriously.fuckshaw.ca) |
| 10:12.10 | tzafrir | Mpls-Eric, the above was for you |
| 10:12.28 | Mpls-Eric | Thanks, I'm stumped... |
| 10:12.38 | *** join/#asterisk X-Rob (n=rob-x@124.150.99.11) |
| 10:12.59 | tzafrir | kezza491, can you get the iaxcomm to register with Asterisk? |
| 10:13.02 | Mpls-Eric | Do you know from the > or < if the data I see is from the telco or from asterisk to the telco? |
| 10:13.18 | kezza491 | i dont know i am new very confused |
| 10:13.26 | tzafrir | Do you see the IP address in 'iax2 show peers'? |
| 10:13.27 | Mpls-Eric | I'm guessing the telco is polling for me, but asterisk it's replying? |
| 10:13.53 | tzafrir | Which ports are connected to the telco? |
| 10:13.58 | Mpls-Eric | I'm guessing the telco is polling for me, but asterisk isn't replying? |
| 10:14.03 | Mpls-Eric | Port 1 is telco |
| 10:14.07 | kezza491 | tzafrir ? |
| 10:14.07 | Mpls-Eric | span 1 |
| 10:14.25 | tzafrir | kaldemar, Do you see the IP address in 'iax2 show peers'? |
| 10:14.50 | tzafrir | kaldemar, in the asterisk cli: run asterisk -r |
| 10:15.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 10:15.27 | kezza491 | ne? |
| 10:15.30 | kezza491 | *me? |
| 10:15.38 | tzafrir | Mpls-Eric, that trace showed you sending a SABME |
| 10:16.01 | tzafrir | kezza491, right, you. kaldemar, oops, sorry |
| 10:16.20 | Mpls-Eric | Thanks, So what should I be sending and what should I expect back? Whats a SABME |
| 10:16.31 | kezza491 | i am using AsterisksNOW if that means anything |
| 10:17.19 | tzafrir | You should expect a UA (Unnumbered Acknowlegement , IIRC) |
| 10:18.04 | tzafrir | SABME: Set Asynchronous Balanced Mode Extended . IIRC every second or so each party sends a SABME and should be replied with a UA, to see that the connection is still up |
| 10:18.30 | kezza491 | tzafrir i dont have the command iax2 |
| 10:19.11 | tzafrir | I need to be going now |
| 10:19.52 | Mpls-Eric | I wonder if this could be a bad patch connection? Maybe the telco is seeing errors from me? Wouldn't we see a yellow alarm from them, or does Asterisk not see yellows all that well? |
| 10:27.33 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 10:27.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:34.50 | kezza491 | whats the point of asterisk now?! |
| 10:35.15 | vanumo | have anybody chan_cellphone in use ? |
| 10:36.29 | kezza491 | I canseem to find any begginers guides to it |
| 10:37.15 | *** join/#asterisk xF|DarkSC (n=dark@server.vinylkind.de) |
| 10:42.53 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 10:43.20 | kezza491 | \what are the alterniatives to asterisks? |
| 10:46.07 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:46.17 | *** join/#asterisk lorinc (n=ang@caracas-3853.adsl.interware.hu) |
| 10:50.13 | tzafrir | kezza491, for what exactly? |
| 10:51.38 | tzafrir | openpbx is basically in the same space. yate likewise. ser/openser is a good for something that is basically a sip gateway |
| 10:51.51 | kezza491 | hmmk |
| 10:52.17 | kezza491 | I am just finding it hard to get help on problems and i think the docomentation is a little anoying |
| 10:52.53 | tzafrir | kezza491, I find it strange that there's no command iax2 |
| 10:53.05 | tzafrir | try: load chan_iax2.so |
| 10:53.20 | tzafrir | have you disabled automatic modules loading? |
| 10:53.44 | kezza491 | I am using AsteriskNOW to learn all of this so i dont know what they have got setup |
| 10:55.35 | *** join/#asterisk friedrich| (n=friedric@e177247195.adsl.alicedsl.de) |
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| 11:04.28 | vanumo | how must i extensions.conf edit to get over chan_cellphone incoming call to my sip phone ? |
| 11:11.09 | *** join/#asterisk KnowWhat (n=KnowWhat@host210-2-165-136.isb.dancom.net.pk) |
| 11:34.11 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
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| 11:42.09 | *** part/#asterisk nirz (n=nir@bzq-88-154-194-62.red.bezeqint.net) |
| 11:46.53 | yonahw-work | can anyone advise me on the correct zaptel.conf settings for bezeq (israel)? |
| 11:48.00 | *** join/#asterisk saftsack (n=oliver@pD9E064EC.dip.t-dialin.net) |
| 11:56.49 | sweeper | yonahw-work: depends entirely on what your telco line is, and what hardware you're using :P{ |
| 11:57.50 | uNK_ | chan_cellphone? |
| 11:57.58 | tzafrir | yonahw-work, try genzaptelconf ... |
| 11:58.09 | sweeper | ugh, no don't touch that script |
| 11:58.11 | sweeper | is NASTY |
| 11:58.43 | tzafrir | If you set the country to il, it has a special case for E1 setting for Israel. I'll have to look into it to see what they are |
| 11:59.05 | tzafrir | sweeper, that's a nasty thing to say |
| 11:59.15 | sweeper | and yet, true \o\ |
| 11:59.23 | sweeper | doing it by hand isn't hard at all |
| 12:00.14 | tzafrir | only you have to know the exact settings. or ask here |
| 12:00.33 | tzafrir | or guess, or whatever. |
| 12:01.24 | tzafrir | There are things that are easy to automate. If zaptel doesn't do them on its own, I want to at least script them |
| 12:01.46 | d00gster | guys, is it possible to create a incoming call rule based on the dialed number (to field of the sip invite)? I have a secondery number -alias- and I want that to ring a separate extension. |
| 12:02.23 | tzafrir | And speaking of scripts: http://svn.digium.com/svn/zaptel/branches/1.2/xpp/utils/zconf/ |
| 12:02.25 | sweeper | the script is ALSO just guessing |
| 12:02.38 | sweeper | only people who KNOW what the settings need to be is his provider |
| 12:02.51 | uNK_ | d00gster the incoming call is via SIP or something? or zaptel pots? |
| 12:03.02 | sweeper | and srsly, there's what, 5 settings for a t1? |
| 12:03.17 | tzafrir | right: guessing that if your country is "il" and you have E1, your settings are such-and-such. You can always correct it by hand. But it's a better starting point |
| 12:03.28 | tzafrir | If you can provide a smarter guess, let me know |
| 12:04.07 | sweeper | eh, you could put standard provider configs in it, but that's about it, and still not a guarantee |
| 12:04.09 | tzafrir | example script that uses this perl module: http://svn.digium.com/svn/zaptel/branches/1.2/xpp/utils/lszaptel |
| 12:05.16 | sweeper | a bit of a hassle for 5 settings, when it would take less time to just call up the provider and say "yo, what framing are you using, and which channel is my d-chan"? |
| 12:06.02 | tzafrir | sweeper, what exactly do you need to ask the provider? |
| 12:06.21 | tzafrir | Not to mention that getting the replies from the provider is no so straight-forward |
| 12:07.12 | sweeper | tzafrir: haha, true on the latter |
| 12:17.14 | d00gster | uNK_> via SIP |
| 12:23.44 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:42.53 | tzafrir | d00gster, basically the dialed number is the extension number |
| 12:51.41 | d00gster | not sure I understand your question tzafrir. I have to house number. when someone dials my secondry number my secondry number apprears in the sip invite specifically in the "to" field (ie TO:2121234567@username.domain.com) |
| 12:52.58 | d00gster | since the secondry number is for my wifes home business, I wanted it to ring her own asterisk extension and maybe get a different answering service on asterisk. to make the business isolated from the house number |
| 12:54.21 | uNK_ | d00gster the # dialed will most likely be the extension in asterisk.. which means you can do what you want with it from there |
| 12:54.31 | uNK_ | this your first time using asterisk? |
| 12:55.19 | d00gster | yes |
| 12:55.47 | uNK_ | well i havent used the new gui stuff yet.. you using the gui or just editing config files? |
| 12:56.03 | d00gster | I installed asterisknow. and I know I show go cli to learn this better |
| 12:56.12 | d00gster | I am on cli now |
| 12:56.35 | uNK_ | well its not the cli but the config files themselves really |
| 12:56.39 | d00gster | first problem with gui is that I can check my sip truml status |
| 12:56.48 | d00gster | trunk I mean |
| 12:56.57 | uNK_ | but basically in the extensions (extensions.conf) they have contexts.. like seperate sections |
| 12:57.24 | d00gster | so can asterisk route calls based on the "to" field? |
| 12:57.24 | uNK_ | then incoming numbers like a sip connection or zaptel connection get set to specific contexts (sections) |
| 12:57.35 | uNK_ | them the #s dialed in go to extensions |
| 12:57.47 | *** join/#asterisk Ebola (n=Ebola@host86-142-179-38.range86-142.btcentralplus.com) |
| 12:57.47 | uNK_ | like 5145551234 is an extension because thats the # dialed |
| 12:57.48 | d00gster | see it is not the caller number I am forwarding to an extension |
| 12:57.57 | d00gster | it is the called number |
| 12:58.01 | uNK_ | yeah i know |
| 12:58.03 | d00gster | ok |
| 12:58.20 | uNK_ | you are probably thinking of extensions like 6000 for phone 1.. and 6001 for phone 2 |
| 12:58.26 | uNK_ | but the phone #s themselves are extensions as well |
| 12:58.36 | d00gster | I see |
| 12:58.59 | d00gster | uNK_ how can I tell if my sip connection with Bell is up? |
| 12:59.18 | uNK_ | so like 5145551234 is an extension in context 'incoming' or whatever so whenever someone calls the # 5145551234 and it goes to my sip provider.. it will go there |
| 12:59.45 | uNK_ | from cli? sip show channels |
| 13:00.01 | uNK_ | or sip show peers |
| 13:00.13 | uNK_ | or sip show users |
| 13:01.35 | d00gster | so peers seems to be the one |
| 13:01.58 | d00gster | so now if I dial my house number, would asterisk answer? |
| 13:02.11 | uNK_ | depends if you configured it properly hehe |
| 13:02.23 | uNK_ | but if the connection is up |
| 13:02.25 | d00gster | actually, let me rephrase. how can I test asterisk and make sure it is getting the call |
| 13:02.29 | uNK_ | then asterisk should answer |
| 13:02.39 | uNK_ | but now the question is what asterisk will do |
| 13:02.47 | uNK_ | call the # and view debug messages |
| 13:02.54 | uNK_ | and it should tell you whats wrong/whats happening |
| 13:03.29 | uNK_ | i didnt even realize bell did SIP voip |
| 13:04.31 | d00gster | uNK_ PM? |
| 13:04.39 | uNK_ | sure shoot |
| 13:04.43 | d00gster | what is the command to view the debug |
| 13:04.55 | uNK_ | well normally you start asterisk with asterisk -vvvvc |
| 13:04.58 | uNK_ | to have it verbose |
| 13:06.17 | d00gster | root 2963 0.2 6.6 42496 15896 ? Sl Feb24 2:28 /usr/sbin/asterisk -vvvg -c |
| 13:06.34 | d00gster | so it is in verbose (Asterisk now) |
| 13:06.51 | d00gster | but I am ssh'ing to the box and not on console |
| 13:07.22 | d00gster | does the debug so to var/log/messages or something? |
| 13:07.31 | flying_Luck | use asterisk -r |
| 13:07.46 | uNK_ | to reattach to the console |
| 13:07.53 | d00gster | that takes me to the cli |
| 13:08.01 | uNK_ | yea |
| 13:08.04 | uNK_ | and should be in verbose mode |
| 13:08.24 | uNK_ | now make a call to the # |
| 13:08.40 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 13:08.44 | d00gster | well I need a softclient first ... |
| 13:08.59 | uNK_ | ? |
| 13:09.02 | uNK_ | use an outside line |
| 13:09.05 | uNK_ | cellphone.. home line |
| 13:09.09 | d00gster | ok |
| 13:10.00 | d00gster | ok so how do I see the debug messages? would they pop on the session or do I have to issue a command? |
| 13:10.10 | uNK_ | they would show up |
| 13:10.20 | uNK_ | what happens when you dial the # from an outside line? |
| 13:11.40 | d00gster | I must have done something wrong... |
| 13:11.51 | d00gster | no messages and I get the provider vm |
| 13:12.03 | uNK_ | it goes straight to provider vm? |
| 13:12.10 | d00gster | yeah |
| 13:12.13 | uNK_ | perhaps you aren't connected with them |
| 13:12.13 | uNK_ | hehe |
| 13:12.21 | d00gster | hummm |
| 13:12.53 | uNK_ | is this machine public IP or nat? |
| 13:13.13 | d00gster | nat |
| 13:13.36 | uNK_ | sip and nat dont like eachother much hehe |
| 13:13.40 | d00gster | but my Sipura is also behind it and conneting to the provider without issues |
| 13:13.40 | uNK_ | but did you forward the sip ports? |
| 13:14.03 | uNK_ | hmm |
| 13:14.09 | uNK_ | whats the network layout? |
| 13:14.14 | uNK_ | is sipura at edge? |
| 13:14.20 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 13:14.39 | d00gster | netscreen FW and on it there is the asterisk server and also my sipura |
| 13:14.54 | uNK_ | so netscreen is at edge |
| 13:14.59 | d00gster | yup |
| 13:15.03 | uNK_ | you have one ip which is dynamic i assume |
| 13:15.13 | uNK_ | then what is after |
| 13:15.29 | d00gster | modem (not layer 3) |
| 13:15.31 | uNK_ | asterisk-> netscreen... and sipura->netscreen? or is it netscreen->sipura->asterisk |
| 13:16.03 | d00gster | no sipura and asterisk are on separate netscreen ports |
| 13:16.24 | d00gster | netscreen > sipura and netscreen> server |
| 13:16.26 | uNK_ | you getting rid of sipura for asterisk? |
| 13:16.49 | d00gster | well eventually the sipura will be taking to my server instead of provider |
| 13:17.06 | d00gster | and it will be my ata to get analog |
| 13:17.29 | uNK_ | hmm yea could do that if its not locked down |
| 13:17.37 | uNK_ | ok so the thing is.. you need to forward your sip ports |
| 13:17.40 | uNK_ | to the asterisk server |
| 13:17.46 | uNK_ | but itll prob break the sipura |
| 13:17.55 | uNK_ | since you prob have them forwarded to it now |
| 13:17.58 | d00gster | true |
| 13:18.46 | uNK_ | http://www.velocityreviews.com/forums/t235350-which-ports-to-open-for-voip-sip.html |
| 13:18.57 | uNK_ | there's a quick sip ports google for ya |
| 13:19.59 | d00gster | ok. so to recap. If I configure and extension with the same phone number as my outside DID, it should reing that extension everytime someone calls that DID? |
| 13:20.39 | uNK_ | in the right context yeah |
| 13:20.54 | d00gster | loaded answer :-) |
| 13:20.58 | uNK_ | then from that extension you can say goto 6000 context whateverwhich is your phone extension |
| 13:22.49 | d00gster | is everything configurable from cli or so I need to edit files to achieve all this? |
| 13:23.04 | uNK_ | cli is just to see statuses and stuff |
| 13:23.11 | uNK_ | its the config files you need to change for settings |
| 13:23.21 | uNK_ | altho the gui edits the configs for you |
| 13:23.29 | uNK_ | but i havent used it at all so i cant help ;) |
| 13:24.04 | *** join/#asterisk key2 (n=key2@81.52.138.22) |
| 13:24.57 | d00gster | any good guide outhere you would recommend? |
| 13:25.27 | uNK_ | well the gui came out recently and is still in beta so prob not many good guides.. but www.asterisknow.org would be for that |
| 13:25.44 | uNK_ | but for general asterisk info http://www.voip-info.org/wiki/ rocks |
| 13:25.50 | d00gster | I don't want to sre gui anymore :-) |
| 13:25.54 | d00gster | ok |
| 13:26.07 | uNK_ | oh |
| 13:26.16 | uNK_ | then the config file you want to look at is extensions.conf |
| 13:26.17 | uNK_ | [incoming] |
| 13:26.17 | uNK_ | exten => 5145551234,1,Goto(main,6000,1) |
| 13:26.17 | uNK_ | [main] |
| 13:26.17 | uNK_ | exten => 6000,1,Macro(stdexten,6000,SIP/office,vm2) |
| 13:26.28 | uNK_ | is basically your config depending on your settings |
| 13:27.18 | d00gster | ok |
| 13:27.30 | d00gster | thanks a million uNK_ |
| 13:27.33 | uNK_ | yea np |
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| 14:06.45 | uNK_ | hmm |
| 14:06.53 | uNK_ | chan_cellphone looks pretty sweet |
| 14:08.37 | uNK_ | was gonna get a dock n talk but now there's no need |
| 14:08.54 | uNK_ | and i dont need to use a port |
| 14:12.16 | public- | uNK_: it works pretty well... lacks the call waiting feature for now, but still pretty nice |
| 14:12.16 | public- | :) |
| 14:12.52 | uNK_ | nice |
| 14:12.56 | uNK_ | yeah prob not gonna run it just yet |
| 14:13.01 | uNK_ | wait till it hits a stable branch |
| 14:13.22 | uNK_ | but was surprised it was out there |
| 14:13.33 | uNK_ | was thinkin about either hardware device or chan_celiax |
| 14:13.41 | uNK_ | but this is a much cleaner way of doing it |
| 14:20.51 | saftsack | where to find asterisks log? |
| 14:21.33 | saftsack | because theres not all in /var/log/asterisk |
| 14:25.53 | *** join/#asterisk `ghost (n=fopsdjfi@203.19.127.16) |
| 14:25.57 | `ghost | hi all long time no type |
| 14:27.27 | `ghost | question I want to use our asterish server I got working about a month ago to use with our skype account (user dials 0 then the phone number and it gose thu the skype system) are there any docs out and about or am I wasting my time ? |
| 14:28.25 | mafkees | you are wasting your time |
| 14:29.04 | flying_Luck | there are some commercial linux solutions for skype |
| 14:29.27 | `ghost | go on flying_Luck |
| 14:30.07 | flying_Luck | `ghost, http://www.rsdevs.com/ |
| 14:30.45 | `ghost | yeah they have someting for $19.95 us (lookin at it now) |
| 14:30.49 | `ghost | with a trial version |
| 14:32.24 | `ghost | not bad I guess cause we have temp sites we sometimes hook up via fibre (and I hook our asterisk server into the ethernet hub so we can talk to each site without having to make a interstate phone call) |
| 14:33.00 | `ghost | sience when we do these things we may as well use skype (cause these events use a lot of international phone calls) |
| 14:34.26 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 14:35.51 | `ghost | hummm psgw requires a runnin x server more work |
| 14:36.45 | mafkees | all them skype things require X |
| 14:37.00 | `ghost | oh ok |
| 14:37.15 | `ghost | only used skype on the xp desktop here |
| 14:44.16 | *** join/#asterisk aste (n=kvirc@161-118-207-82.ip.ukrtel.net) |
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| 14:59.58 | `ghost | ?? oh I do have x installed on this machine |
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| 15:22.26 | ocgltd | I have an Asterisk 1.40 system doing protocol conversion, with SIP on one leg and H323 on the other, all using G729 (no transcoding). When calls originate on the H323 leg, audio is great. When calls originate on the SIP leg, audio is horribly choppy. Can anyone offer ideas? |
| 15:23.43 | `ghost | the only words that registered with me is asterisk and sip :P |
| 15:23.50 | `ghost | sorry thats to deep for me |
| 15:27.54 | flying_Luck | anybody has an asterisk <-> nec neax 2000 E1 connection ? i'm stuck with q.sig protocol errors nec sends me on call setup |
| 15:28.31 | *** join/#asterisk brea (n=B@c-24-10-145-239.hsd1.ut.comcast.net) |
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| 15:39.02 | `ghost | hey flying_Luck have you ever had a crack at running psgw ? |
| 15:40.02 | flying_Luck | `ghost, i have never used it. just heard somewhere about existance of such thing |
| 15:40.22 | `ghost | oh ok well I installed the trial and tryin to chew over it now |
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| 16:05.56 | `ghost | note to self when screaming why your program cannot talk to a ip address make sure you have the right IP address |
| 16:06.03 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 16:06.07 | mafkees | lol |
| 16:06.24 | markit | hi, other people having problems in compiling asterisk 1.4 svn today? |
| 16:06.31 | `ghost | it's 3 in the morning |
| 16:08.36 | `ghost | this pgsw is still dring me up the wall... I think I am missing something |
| 16:08.42 | `ghost | marbles more like it |
| 16:13.10 | ocgltd | I'm getting tons of these errors, anyone know what they mean? [Feb 25 01:46:13] DEBUG[13282] rtp.c: Forcing Marker bit, because SSRC has changed |
| 16:15.31 | *** join/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 16:18.14 | *** part/#asterisk omarc55 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net) |
| 16:18.49 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 16:19.26 | *** join/#asterisk KnowWhat (n=KnowWhat@host210-2-165-136.isb.dancom.net.pk) |
| 16:19.42 | KnowWhat | Hello |
| 16:19.43 | KnowWhat | Feb 25 11:17:17 WARNING[2664]: chan_iax2.c:7103 socket_read: Call rejected by 64.2.142.29: No such context/extension |
| 16:19.52 | KnowWhat | why i am getting this error, if some body can help me |
| 16:24.18 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 16:25.21 | tzafrir | KnowWhat, so the iax2 user was authenticated, codec negotiation was successful etc. However there was no dialplan extension for it |
| 16:25.38 | tzafrir | do you know which iax2 user / friend was used there? |
| 16:26.26 | KnowWhat | yup |
| 16:26.37 | KnowWhat | you mean to say in extension? |
| 16:27.22 | KnowWhat | well you know when i dial a number, like 8008004214 |
| 16:27.46 | KnowWhat | at asterisk cli it shows 918008004214 |
| 16:28.33 | KnowWhat | while i have extension define as exten => _1NXXNXXXXXX,2,Dial(IAX2/mxconn@vitel-outbound/${EXTEN}) |
| 16:29.28 | *** join/#asterisk tr2x (n=alvar@ctse16.ulb.ac.be) |
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| 16:40.39 | public- | anyone been successful in getting music on hold working over chan_cellphone? |
| 16:44.38 | elriah | There's a chan_cellphone? |
| 16:44.53 | Nivex | I've tried to find this elusive chan_cellphone to no avail |
| 16:45.33 | public- | yes |
| 16:45.38 | public- | there is a chan_cellphone |
| 16:45.41 | Corydon76-home | It's in the bugtracker |
| 16:45.43 | public- | that uses BT |
| 16:45.51 | elriah | BT? Bluetooth? |
| 16:45.53 | public- | yes |
| 16:46.01 | public- | and then you can dial out from a SIP phone on the cell |
| 16:46.06 | public- | and receive calls on it as well |
| 16:46.15 | Corydon76-home | http://bugs.digium.com/view.php?id=8919 |
| 16:46.32 | elriah | What would you use it for? |
| 16:47.04 | Corydon76-home | Connection to BT headsets and BT-enabled cellphones |
| 16:47.29 | elriah | With a 50 foot range, I'm trying to see the value? |
| 16:47.39 | JT | the value is obvious |
| 16:47.43 | public- | you can use a BT enabled phone for dialing out |
| 16:47.48 | public- | from any voip phone |
| 16:48.08 | elriah | Right, if you're withing range of your asterisk server? |
| 16:48.12 | JT | put your mobile near the asterisk box, calls to mobiles come into asterisk server |
| 16:48.17 | JT | how is that not super useful |
| 16:48.28 | public- | yes |
| 16:48.43 | elriah | Oh, well, all the asterisk servers we have are geographically unreachable from our userbase so I didn't connect the dots. |
| 16:48.46 | Nivex | I have no landline. That could become my FXO port while I'm home. This makes me happy. |
| 16:48.47 | Corydon76-home | BT range can also be extended |
| 16:48.58 | public- | elriah: more for @home setups |
| 16:48.58 | JT | asterisk can be used in many ways |
| 16:49.12 | public- | Nivex: exactly.. for people with no landlines |
| 16:49.16 | public- | and this is my FXO port |
| 16:49.34 | elriah | Ahh, interesting. |
| 16:49.37 | Nivex | there was a chan_bluetooth floating around awhile ago. This based on that, or a new codebase? |
| 16:49.44 | JT | new |
| 16:49.53 | public- | chan_bluetooth lost support |
| 16:50.04 | public- | and chan_cellphone is doing a good job picking up where it left off |
| 16:50.10 | Nivex | ok. are there any particular requirements for a BT dongle, or if it works with Linux it should Just Work (tm) ? |
| 16:50.18 | public- | in theory |
| 16:50.18 | public- | yes |
| 16:50.24 | public- | if you have the BT stack installed |
| 16:50.25 | public- | you should be fine |
| 16:50.35 | public- | good documentation, makes it quite easy to get everything working |
| 16:50.56 | Nivex | I think I need to order that BT dongle soon :) |
| 16:51.20 | public- | yah |
| 16:51.24 | public- | it's nice on a laptop |
| 16:51.31 | public- | works as a perfect lil asterisk machne |
| 16:51.31 | public- | :P |
| 16:51.45 | Nivex | my asterisk box at the moment is an HP e-Vectra |
| 16:51.48 | Nivex | nice tiny little thing |
| 16:52.12 | public- | anyone know how to dial 2 extensions when a call comes in from the outside? |
| 16:52.25 | public- | I want to dial the 2 voip phones in my house |
| 16:52.35 | Nivex | SIP/phone1&SIP/phone2 |
| 16:52.36 | public- | when I receive an incoming call |
| 16:52.41 | public- | just an &? |
| 16:52.44 | Nivex | First to answer gets the call |
| 16:52.51 | tzanger | has anyone here done any high-availability asterisk installs? What did you use? heartbeat or something more complex? |
| 16:52.57 | public- | awesome |
| 16:52.58 | public- | thanks |
| 16:53.05 | mafkees | tzanger: dundi |
| 16:53.34 | tzanger | mafkees: oh yeah? |
| 16:53.48 | mafkees | yeah |
| 16:53.56 | mafkees | roundrobin dns |
| 16:54.06 | mafkees | so clients would connect to one of the asterisk boxen |
| 16:54.08 | tzanger | mafkees: so you just have two machines which provide the same routes/destinations? |
| 16:54.14 | mafkees | and dundi to find them routes/stations |
| 16:54.16 | tzanger | how does that allow for failover of sip phones, etc. |
| 16:54.24 | tzanger | dundi would just be a routing destination |
| 16:54.39 | mafkees | set the registry timeout really low |
| 16:54.44 | mafkees | like 5 seconds |
| 16:54.49 | elriah | tzanger: We have 5 two tier installs, mysql on the backend, asterisk on two or more front end... |
| 16:55.12 | elriah | tzanger: And load balancers in front |
| 16:55.20 | *** join/#asterisk netlouis (n=netlouis@a213-22-64-9.cpe.netcabo.pt) |
| 16:55.38 | tzanger | hmm... I'll have to investigate this a little further |
| 16:55.40 | elriah | tzanger: We did have a similar config without mysql just using rsync for configs and voicemails, worked great actually. |
| 16:56.03 | mafkees | we use shared storage for the configs and voicemails |
| 16:56.23 | mafkees | gnbd with ocfs2 |
| 16:56.24 | elriah | mafkees: As w/a san? Just nfs? |
| 16:56.26 | tzanger | elriah: "five two-tier installs" is that five separate highly-available setups? |
| 16:56.30 | elriah | Ahhh.. |
| 16:56.41 | elriah | tzanger: Yep. In 5 different datacenters. |
| 16:56.47 | `ghost | ok I understand no man on this planet has ever got psgw working but would any1 have any idea why I am getting this error "could not open any listnener from udp$10.1.1.13" (I set that ip cause thats the ip to my asterisk server is on (same with pgsw and skype) |
| 16:56.54 | tzanger | elriah: but they're totally separate from each other... i.e. the 5 installs do not inter-communicate |
| 16:57.28 | mafkees | elriah: no san, gnbd and ocfs2 |
| 16:57.37 | mafkees | we had to built it with little bugdet |
| 16:57.37 | tzanger | mafkees: Other than the wiki, do you have any helpful sites or other documentation on dundu? |
| 16:57.38 | elriah | tzanger: No, they are the same disperate cluster. |
| 16:57.40 | tzanger | er dundi |
| 16:57.44 | tzanger | elriah: ahh okay |
| 16:58.16 | mafkees | tzanger: nope |
| 16:58.22 | *** part/#asterisk netlouis (n=netlouis@a213-22-64-9.cpe.netcabo.pt) |
| 16:58.28 | elriah | tzanger: We rarely have issues with it, most of our problems are phone related, lol |
| 16:58.54 | *** join/#asterisk netlouis (n=netlouis@a213-22-64-9.cpe.netcabo.pt) |
| 16:59.01 | tzanger | elriah: :-) |
| 16:59.07 | mafkees | same here elriah |
| 16:59.10 | mafkees | stupid phones |
| 17:00.00 | elriah | I haven't seen any in action, but I bet a SAN with a bunch of bridgehead nodes would work great with flat files... |
| 17:00.11 | `ghost | .. oh hang on |
| 17:00.39 | elriah | Has anyone tried to install asterisk in an NFS environment? i.e., one set of configs and voicemail dirs? Does asterisk handle the locked files gracefully? |
| 17:00.59 | tzanger | I'm looking at something for a business with two separate "main" offices (about 600km away from each other) -- basically there'll be a decent WAN between them and I was going to have the phones for office A have a secondary register to office B's asterisk and vice-versa |
| 17:01.20 | elriah | How big of a userbase? i.e., how many phones and calls per hour (guestimate) |
| 17:01.38 | mafkees | elriah: yeah, voicemail and configs on nfs works great |
| 17:01.41 | tzanger | basically if asterisk A died any calls would die with it bu tthe phones should fail over to Asteirsk B automatically |
| 17:01.45 | elriah | tzanger: Don't underestimate the power of a properly setup rsync environment... |
| 17:02.03 | tzanger | elriah: well asterisk A and B would not be configured identically |
| 17:02.22 | elriah | brb |
| 17:02.46 | tzanger | but I was also thinking of forgetting that idea and just taking two boxes at A and two at B and using heartbeat to failover everything locally |
| 17:03.12 | tzanger | e.g. A1=main, A2=hot backup, heartbeat between them. A1 dies A2 takes over locally |
| 17:03.34 | *** join/#asterisk luisjose (n=ljd@unaffiliated/luisjose) |
| 17:04.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:04.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:04.58 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 17:06.52 | mafkees | tzanger: for that to work they should be able to takeover eachothers ip |
| 17:07.00 | tzanger | mafkees: yes exactly |
| 17:07.21 | mafkees | or run asterisk on openbsd |
| 17:07.52 | mafkees | that way you can use CARP for the ip address takeover |
| 17:07.52 | mafkees | that's what I do at home and at some setups at customers |
| 17:07.52 | tzanger | mafkees: hmm |
| 17:07.57 | mafkees | of course you cannot support zap channels that way but who cares ? |
| 17:08.35 | mafkees | I never use zap anymore |
| 17:08.52 | `ghost | YYYEEEESSSS I got it working (note when using psgw if you are running it on the same machine as asterisk server... change your port on the sip.config) |
| 17:09.14 | *** join/#asterisk mranostay (n=cold_boo@pdpc/supporter/base/mranostay) |
| 17:09.18 | mranostay | hello |
| 17:09.27 | public- | anyone here with a cisco 7940? |
| 17:09.30 | public- | using SIP? |
| 17:10.22 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 17:10.24 | mranostay | my company does, i have littl insight on it though |
| 17:10.51 | mranostay | does anyone have a sample asterisk call file using festival? |
| 17:12.50 | mafkees | public-: not me sorry. I use SCCP on them |
| 17:13.11 | public- | Got a couple of questions about them... and they may not be SIP specific. |
| 17:13.30 | public- | I am looking for the option to display the clock on the screen. Mine currently has no time display |
| 17:13.44 | public- | And the next option I'd like is being able to dial a number before hitting new call |
| 17:13.56 | public- | so the number will show up, but won't dial until I hit newcall |
| 17:14.13 | mafkees | I dont think that's possible |
| 17:14.27 | *** join/#asterisk ruied (n=ruied@bl7-211-73.dsl.telepac.pt) |
| 17:14.29 | mafkees | because newcall will simply open a channel |
| 17:14.43 | mafkees | like lifting the handset |
| 17:14.56 | elriah | tzanger: You want cheap failover? |
| 17:14.57 | public- | it works when the phone is on a call manager network |
| 17:15.01 | elriah | Is "cheap" the key? |
| 17:15.19 | public- | was wondering if it is possible with the SIP version |
| 17:15.20 | mafkees | public-: that's because it's in the callmanager/sccp stuff |
| 17:16.08 | public- | mafkees: ok |
| 17:16.17 | public- | any idea about the display of the clock? |
| 17:16.24 | public- | this an option I can put in SIPDefault.cnf/ |
| 17:17.08 | mafkees | no idea |
| 17:17.19 | mafkees | you can try to include the time in the date related option |
| 17:18.35 | tzanger | elriah: cost is always a factor, but I'm just looking at different ways of doing it... asterisk is only one part of the equation, db is another thing that needs to be failed over as well... I've never done any high availability before, but I've used asterisk for about 4 years and linux for 12 or so |
| 17:19.13 | elriah | Cost-effective and cheap are generally two different solutions... |
| 17:19.17 | elriah | Which one are you looking for? |
| 17:19.38 | elriah | cheap = bash script for failover and heartbeat |
| 17:19.55 | tzanger | elriah: I don't think that cost-effective and cheap are necessarily at odds... cost effective just means that you get the best bang for buck |
| 17:20.07 | tzanger | cheap that doesn't work is neither cheap nor effective. :-) |
| 17:20.17 | elriah | ugh, brb |
| 17:21.58 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:21.58 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 17:26.36 | *** join/#asterisk apardo (n=apardo@87.217.145.84) |
| 17:27.41 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 17:32.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:32.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:34.01 | *** join/#asterisk pardove (n=chatzill@195.146.46.147) |
| 17:35.10 | pardove | Is there anyway to control a call after dial command? ex: dial xxx, then wait for n sec. the send xx DTMF and then wait for n sec. then send DTMF again and finally hangup |
| 17:36.42 | mafkees | pardove: show application dial |
| 17:36.48 | mafkees | it's possible to do that indeed |
| 17:40.45 | pardove | mafkees: if you mean D option for Dial command. it just can send DTMF. no wait and no more control. am i right? |
| 17:41.03 | mafkees | w == wait |
| 17:41.13 | mafkees | you can use the w in the D stuff |
| 17:41.25 | mafkees | tzanger: windows 2000 advanced server |
| 17:41.40 | tzanger | mafkees: damn you have done work on windows clustering as well? |
| 17:41.42 | mafkees | dont know what version of windows 2003 can do clustering |
| 17:41.49 | mafkees | yeah |
| 17:41.52 | *** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 17:42.07 | tzanger | mafkees: does that let exchange server and IIS and SQL Server to custer as well? |
| 17:42.21 | mafkees | yeah |
| 17:42.23 | tzanger | or do you need special licenses for each of those to cluster as well? |
| 17:42.36 | pardove | mafkees: "w" just waits before dialing and does nothing else while the is established |
| 17:42.38 | mafkees | IIS is there by default |
| 17:42.52 | ManxPower | pardove: the Dial(Zap/G2/5551515,,D(1234wwww5667) will send 1234 as soon as the call is answered, wait 4 x .5 second, then send 5667. |
| 17:43.06 | mafkees | tzanger: I dont know about exchange because we never used it |
| 17:43.15 | bkruse_home | ManxPower: thats pretty sweet, i did not know that |
| 17:43.19 | mafkees | and sql server 2000 has clustering with it as well |
| 17:43.26 | ManxPower | Remember analog FXO channels are considered "answered" as soon as dialing is finished |
| 17:43.32 | mishehu | exchange, from a technical standpoint, is teh suck. |
| 17:43.34 | Qwell | bkruse_home: incoming sms support was added last night |
| 17:43.44 | Qwell | bkruse_home: it only outputs to the CLI, but yeah, it's a start |
| 17:43.51 | tzanger | mafkees: hmm okay, good stuff. I'm up against Objectworld's Unified Communications Server product on this job :_) |
| 17:43.51 | mishehu | a nice idea but very poorly implemented. |
| 17:43.57 | mafkees | as is hints on skinny ;) |
| 17:44.20 | pardove | ManxPower: thanks for your comment. but is there anyway to have more control over an established call. |
| 17:44.31 | mafkees | I had to revert chan_skinny.c before svn up would be nice to me ;) |
| 17:44.36 | bkruse_home | Qwell: ya, thats DEF a start!!! |
| 17:44.40 | bkruse_home | thats awesome!!!!!! im excited |
| 17:44.41 | ManxPower | pardove: define "more control:. |
| 17:44.42 | bkruse_home | i wana hack at it |
| 17:44.48 | bkruse_home | make some configs for somethin |
| 17:45.18 | pardove | ManxPower: so there no more now! btw, how can i hangup the call after sending DTMF string? |
| 17:45.42 | ManxPower | pardove: use the L option to Dial |
| 17:45.44 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 17:46.19 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:46.19 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:46.23 | ManxPower | Or use the Set(TIMEOUT(absolute)=5) to limite the total call length to 5 seconds, you would set this before the Dial |
| 17:46.25 | Qwell | russellb: !!! |
| 17:46.32 | Qwell | russellb: I saw you on TV the other night :P |
| 17:46.35 | bkruse_home | pardove: uh? Hangup() ? |
| 17:46.37 | Qwell | ...your legs, anyhow |
| 17:46.40 | bkruse_home | Qwell: ohrly? |
| 17:46.50 | bkruse_home | concerning "the incident?" |
| 17:46.53 | ManxPower | bkruse that won't work. |
| 17:46.55 | Qwell | mmhmm |
| 17:46.57 | russellb | Qwell: nice! |
| 17:47.06 | pardove | ManxPower: thanks again! I just wondering to have a way to pass a call to an AGI when its established. |
| 17:47.12 | Qwell | russellb: apparently they arrested the guy in Atlanta, "on unrelated charges" |
| 17:47.30 | Qwell | Kentral Smith or some such |
| 17:47.35 | ManxPower | pardove: look at "show application external ivr" |
| 17:47.41 | ManxPower | I don't know if it will help. |
| 17:48.02 | ManxPower | pardove: perhaps you need to use .call files to place a call and send the other leg of the call to an AGI |
| 17:48.16 | ManxPower | look on the Wiki for example of using .call files |
| 17:48.23 | russellb | the bank robbery pic :-p |
| 17:48.31 | Qwell | russellb: yeah :D |
| 17:49.42 | russellb | um, yes? |
| 17:49.42 | russellb | Qwell: good. he was a nub |
| 17:49.42 | pardove | ManxPower: thank you very much. i will read more on your suggestions ;-) |
| 17:53.38 | pardove | ManxPower: I have wrote a dirty patch to make the callprogress work accurate. and now while dialing on FXO channels it will answer the fxo channel as the callee picks up the phone |
| 17:53.38 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:53.38 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 17:55.40 | *** join/#asterisk phetaudicili (n=phetaudi@c-71-58-155-8.hsd1.pa.comcast.net) |
| 17:56.02 | `ghost | ok question about psgw I am trying to make a incomming call. skype catches it, sends it to psgw then it is ment to pass it to my asterisk server, but I am getting this wierd meassage "root did not answer your call" where is that comming from |
| 17:58.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:58.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:00.13 | *** join/#asterisk sharp (n=sharp@2001:470:1f01:ffff:0:0:0:1c23) |
| 18:09.46 | *** join/#asterisk XVampireX (n=serge@gateway/web/cgi-irc/ircatwork.com/x-7deee14cde4bab95) |
| 18:09.55 | XVampireX | Heylo :) |
| 18:10.35 | XVampireX | Can you people tell me if it is possible to build a PBX that works with a microphone/headset or something like that instead of having to use a phone? |
| 18:10.48 | XVampireX | Asterisk PBX^ |
| 18:11.06 | ManxPower | XVampireX: That is callled a "softphone". You need to read The Book first. |
| 18:11.07 | ManxPower | ~book |
| 18:11.10 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:11.20 | XVampireX | Ah |
| 18:12.00 | XVampireX | Well, softphone would be an application that connects to Asterisk, yeah? |
| 18:12.15 | uNK_ | yessir |
| 18:12.57 | XVampireX | I know that Gizmo Project can work with Asterisk, but say I can't use SIP because I'm behind a router and it just won't work for me... can I use jingle with asterisk/gizmo project? |
| 18:13.18 | XVampireX | Or there is no other softphone that I know that connects to Asterisk |
| 18:13.49 | XVampireX | Or maybe I have no clue ;) |
| 18:13.51 | uNK_ | ok wait a second |
| 18:13.56 | uNK_ | you want to use gizmo |
| 18:13.59 | uNK_ | so why not just use gizmo? |
| 18:14.02 | *** part/#asterisk phetaudicili (n=phetaudi@c-71-58-155-8.hsd1.pa.comcast.net) |
| 18:14.27 | XVampireX | I don't want to use Gizmo, gizmo uses SIP and it doesn't work well for me. |
| 18:14.52 | uNK_ | oh ok |
| 18:15.03 | uNK_ | did you open the SIP ports on your firewall? |
| 18:15.08 | XVampireX | Yes |
| 18:15.18 | pardove | whats the best software-based fax solution for asterisk? |
| 18:15.29 | XVampireX | But any connection that I make it either lags, or stutters... can't have a good connection with SIP, although with Skype it works fine. |
| 18:15.32 | *** join/#asterisk EmleyMoor (n=phil@topdeck.tinsleyviaduct.com) |
| 18:15.50 | uNK_ | oh ok |
| 18:15.55 | uNK_ | what kinda net connection do you have? |
| 18:16.24 | mafkees | pigeon carrier |
| 18:16.35 | uNK_ | when's the last time you fed the pigeons? |
| 18:16.39 | XVampireX | 1.5mb/96kb |
| 18:16.49 | EmleyMoor | Is there any way, other than by using a specific detect and Goto, that I can include some extensions from a "lower" context in a higher one to take priority over a less specific extension? |
| 18:16.49 | uNK_ | 96?! |
| 18:16.50 | mafkees | ouch, that 96kb will kill you |
| 18:16.50 | uNK_ | ouch |
| 18:17.07 | mafkees | uNK_: try this in sip.conf: |
| 18:17.10 | mafkees | disallow=all |
| 18:17.13 | mafkees | allow=gsm |
| 18:17.19 | mafkees | and try again |
| 18:17.31 | XVampireX | Yes will kill me |
| 18:17.35 | XVampireX | but skype works |
| 18:17.37 | XVampireX | >_< |
| 18:17.40 | mafkees | ehm |
| 18:17.43 | mafkees | I mean XVampireX |
| 18:18.13 | mafkees | yeah, skype uses ilibc (or however it's called) |
| 18:18.23 | mafkees | for asterisk, try gsm |
| 18:18.38 | uNK_ | he doesnt have asterisk installed i dont think |
| 18:18.44 | uNK_ | he's thinkin about switching from skype to it |
| 18:19.06 | mafkees | good thinking ;) |
| 18:19.10 | XVampireX | Yeah, I don't have asterisk installed but I'd like to try it, just don't have phone for it |
| 18:19.13 | mafkees | ow wait |
| 18:19.17 | mafkees | gizmo |
| 18:19.29 | mafkees | cant you select the codecs in the gizmo client ? |
| 18:19.44 | EmleyMoor | What, if anything, do I need to do to have asterisk take incoming SIP and/or IAX2 calls from clients not registered on my box? |
| 18:19.48 | XVampireX | Don't think |
| 18:19.59 | XVampireX | But I tried twinkle, that wasn't all that great either. |
| 18:20.14 | mafkees | XVampireX: in twinkle you can select them |
| 18:20.24 | EmleyMoor | I use ekiga, X-Lite, kiax and moziax |
| 18:20.27 | mafkees | try to disable all audio codecs and only enable the gsm |
| 18:20.36 | ManxPower | EmleyMoor: Registrations has nothing to do with calls Client -> Asterisk |
| 18:20.37 | XVampireX | kiax? |
| 18:20.52 | EmleyMoor | IAX2 softphone for KDE |
| 18:20.56 | ManxPower | All registration does is inform Asterisk what the IP address is for a specific user/pass |
| 18:21.19 | EmleyMoor | ManxPower: So, could someone outside with a SIP phone dial one of my internal extensions directly? |
| 18:21.26 | XVampireX | Hmm |
| 18:21.32 | mafkees | EmleyMoor: depends on your configs |
| 18:21.39 | *** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.in.comcast.net) |
| 18:21.43 | XVampireX | Is IAX2 free for PC-to-PC? |
| 18:21.46 | pardove | whats the best software-based fax solution for asterisk? |
| 18:21.48 | EmleyMoor | That is that upon which I am seeking information |
| 18:21.50 | XVampireX | Or what is it? :P |
| 18:22.02 | mafkees | EmleyMoor: if you enbable guest calls in sip/iax they will most likely endup in the [default] context |
| 18:22.09 | EmleyMoor | XVampireX: I think so |
| 18:22.09 | ManxPower | EmleyMoor: That depends on many things, but does not depend on registration. |
| 18:22.12 | mafkees | if you dont put any logic there nothing well happen |
| 18:22.17 | XVampireX | Where can I call with twinkle to test gsm? |
| 18:22.31 | ManxPower | EmleyMoor: Generally unauthenticated calls will land in the context specified in [general] in sip/iax.conf |
| 18:22.50 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:22.59 | XVampireX | nevermind |
| 18:23.00 | EmleyMoor | OK - so I need to read up on enabling guest calls? |
| 18:23.08 | mafkees | yeah |
| 18:23.59 | EmleyMoor | allowguest=yes is default... hold on a moment |
| 18:25.28 | XVampireX | are there any good free iax providers? |
| 18:25.47 | XVampireX | kiax looks strangely like skype |
| 18:26.37 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:27.44 | EmleyMoor | Can someone try calling me? sip:phil@firthpark.tinsleyviaduct.com |
| 18:27.49 | XVampireX | yeah\ |
| 18:28.08 | J4k3 | what the fuck |
| 18:28.08 | XVampireX | I called you |
| 18:28.11 | XVampireX | Can you hear me? :P |
| 18:28.15 | EmleyMoor | No |
| 18:28.20 | XVampireX | Ah |
| 18:28.22 | EmleyMoor | Vaguely |
| 18:28.30 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 18:28.35 | mafkees | J4k3: ??? |
| 18:29.02 | J4k3 | denon's kline. |
| 18:29.04 | XVampireX | So you can still hear me? :P |
| 18:29.05 | EmleyMoor | You are breaking up a lot |
| 18:29.12 | XVampireX | Ah, ok.... |
| 18:29.23 | XVampireX | Yeah, that's the problem with me and sip :P |
| 18:29.32 | mafkees | J4k3: looks like it's their quit msg |
| 18:29.42 | ManxPower | Rememeber before 1.4 Asterisk did not gave an RTP (SIP audio) jitter buffer. |
| 18:29.46 | J4k3 | oh, that freaked me out |
| 18:29.51 | mafkees | lol |
| 18:29.52 | ManxPower | s/gave/have/g |
| 18:30.07 | mafkees | I need that s//g script jbot is running |
| 18:30.09 | mafkees | it's leet |
| 18:32.22 | Carp1 | Any ideas why voice is only working one way???I cant here a caller talking but they can hear me. |
| 18:33.15 | EmleyMoor | Carp1: Using SIP? |
| 18:33.35 | XVampireX | :P |
| 18:33.45 | XVampireX | SIP gives lotsa problem to people |
| 18:34.01 | EmleyMoor | SIP and NAT don't mix well |
| 18:34.08 | ManxPower | SIP only gives problems to people that don't understand SIP, RTP, NAT and firewalls. |
| 18:34.15 | ManxPower | SIP and NAT work just fine. |
| 18:35.18 | ManxPower | If SIP didn't work thru NAT then how does Vonage, Broadvoice and all the other SIP ITSPs work thru NAT? |
| 18:35.31 | `ghost | you know wot folks... I think the reason it is not working is because I am very tired and have missed something so I will call stumps and I will get PSGw working tomrrow thanks anyway kids |
| 18:36.33 | XVampireX | nah |
| 18:36.53 | `ghost | night |
| 18:37.16 | ManxPower | Reinvites and Asterisk server behind a dynamic IP NAT would not work, but that's about it. |
| 18:40.32 | *** join/#asterisk [Latre] (n=chatzill@189.153.85.243) |
| 18:40.33 | *** join/#asterisk s1gny|wrk (n=s1gny@p54917250.dip.t-dialin.net) |
| 18:40.43 | *** part/#asterisk s1gny|wrk (n=s1gny@p54917250.dip.t-dialin.net) |
| 18:47.10 | Bobthehunter | so reinvites are blind and attended trasnfers and thats al ? |
| 18:51.11 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 18:58.35 | *** join/#asterisk friedrich| (n=friedric@e177240071.adsl.alicedsl.de) |
| 18:59.40 | bkruse_home | whats the commnad to show file version? |
| 18:59.46 | bkruse_home | core show file version right? |
| 19:01.23 | mafkees | bkruse_home: it should be |
| 19:01.26 | mafkees | but mine is empty |
| 19:01.27 | mafkees | Ifrid*CLI> core show file version |
| 19:01.27 | mafkees | File Revision |
| 19:01.27 | mafkees | ---- -------- |
| 19:01.27 | mafkees | Ifrid*CLI> |
| 19:01.36 | bkruse_home | mafkees: same! |
| 19:01.38 | bkruse_home | i think it got broke |
| 19:01.43 | mafkees | yup |
| 19:01.43 | bkruse_home | i was going to use it for something similar |
| 19:02.11 | mafkees | I'm on the bugtracker already ;) |
| 19:02.16 | bkruse_home | mafkees: thanks :] |
| 19:02.26 | bkruse_home | its been broke for awhile |
| 19:02.30 | bkruse_home | i need it for somethin else |
| 19:04.35 | mafkees | http://bugs.digium.com/view.php?id=9135 |
| 19:05.06 | bkruse_home | ty ty ;] |
| 19:05.13 | bkruse_home | i was going to do it, but i can just yell about it monday |
| 19:05.43 | bkruse_home | good way to get karma :P |
| 19:06.30 | mafkees | uhhuh |
| 19:08.34 | Bobthehunter | wahts the normal sip header for a call from peer BLAH callerid BOB 555 333 4444 ? |
| 19:09.32 | mafkees | bkruse_home: you'll have to wait with your application then ;) |
| 19:09.46 | bkruse_home | mafkees: nah, its somthing i can skip for now |
| 19:09.53 | bkruse_home | just has to do with tab completion and filename completion |
| 19:10.10 | bkruse_home | no big |
| 19:10.10 | bkruse_home | thanks though :] |
| 19:10.10 | mafkees | tab completion _is_ working |
| 19:10.13 | Bobthehunter | From: "BOB" <5556665555> <sip:BLAH@209.169.245.127> |
| 19:10.14 | Bobthehunter | ? |
| 19:10.26 | Bobthehunter | or number in the "" |
| 19:10.41 | Bobthehunter | From: "BOB <5556665555>" <sip:BLAH@209.169.245.127> |
| 19:10.43 | Bobthehunter | like this ? |
| 19:10.47 | bkruse_home | Bobthehunter: thats right |
| 19:10.51 | Bobthehunter | first ? |
| 19:11.38 | mafkees | bkruse_home: any idea where I can find this file version thing ? |
| 19:11.38 | bkruse_home | the only one |
| 19:11.39 | bkruse_home | yes first |
| 19:11.39 | bkruse_home | look at |
| 19:11.39 | bkruse_home | sample.call in /usr/src/asterisk ( or wherever your source is) |
| 19:11.39 | bkruse_home | search for callid: |
| 19:11.39 | bkruse_home | callerid: * |
| 19:12.01 | Bobthehunter | yes well its for SER.. asterisk doesnt use right values |
| 19:12.45 | bkruse_home | the function? |
| 19:12.46 | bkruse_home | grep -r '"core", "show", "file", "version"' * |
| 19:12.46 | *** join/#asterisk rrocha (n=ruyrocha@201.10.93.216) |
| 19:15.24 | Bobthehunter | doing what you said asterisk gives... agi_calleridname] => Bob <123123123> |
| 19:15.33 | Bobthehunter | and callerid= 123123123123 |
| 19:16.07 | mafkees | bkruse_home: found it |
| 19:17.38 | *** join/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
| 19:18.37 | mafkees | hhmm, looks like that whole function is broken |
| 19:19.27 | mafkees | core show file version like <something here> |
| 19:19.51 | bkruse_home | its in main/asterisk.c i believe...... |
| 19:19.51 | bkruse_home | mafkees: it is |
| 19:19.51 | bkruse_home | it USED to work |
| 19:19.51 | mafkees | whatever I try, I keep getting the usage info |
| 19:19.51 | mafkees | yeah |
| 19:19.51 | mafkees | it's in main/asterisk.c line 550 and below |
| 19:19.56 | Bobthehunter | wow rpid is the solution |
| 19:20.35 | Bobthehunter | that why its there lol ok |
| 19:21.46 | steve___ | MSG tzanger booga? |
| 19:22.32 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 19:23.27 | mafkees | hhmm |
| 19:23.31 | mafkees | the file is weird |
| 19:23.52 | mafkees | #if !defined(LOW_MEMORY) |
| 19:23.54 | mafkees | some code |
| 19:23.55 | mafkees | ... |
| 19:24.04 | mafkees | #if !defined(LOW_MEMORY) |
| 19:24.09 | mafkees | ow |
| 19:24.10 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:24.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:32.42 | *** join/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
| 19:32.43 | *** part/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
| 19:34.34 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
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| 19:48.40 | *** join/#asterisk DaeJeo1 (n=singh@124.62.151.53) |
| 19:49.15 | Carp1 | Any ideas why voice is only working one way???I cant here a caller talking but they can hear me. |
| 19:49.16 | DaeJeo1 | <PROTECTED> |
| 19:55.18 | ManxPower | DaeJeo1: ask on the #asterisknow channel |
| 19:55.23 | ManxPower | Carp1: nat |
| 19:57.34 | ManxPower | one way voice is a classic indication that you do not have the NAT stuff set up correctly. |
| 20:00.16 | ManxPower | But it's pretty obvious that Carp1 doesn't actually want to fix the problem, he just wants to complain about it. |
| 20:01.42 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:01.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:03.05 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 20:09.02 | Carp1 | I didnt complain about it? |
| 20:09.22 | Carp1 | Maybe adding nat=yes to sip.conf? |
| 20:11.28 | *** join/#asterisk Skaag (n=skaag@80.178.76.62.adsl.012.net.il) |
| 20:11.49 | *** join/#asterisk clive- (n=pirch@dsl-242-143-215.telkomadsl.co.za) |
| 20:12.57 | clive- | hi, anyone here know of a reasonable source of ITFS numbers ? |
| 20:13.42 | *** part/#asterisk DaeJeo1 (n=singh@124.62.151.53) |
| 20:14.23 | *** part/#asterisk XVampireX (n=serge@gateway/web/cgi-irc/ircatwork.com/x-7deee14cde4bab95) |
| 20:17.51 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 20:18.38 | Carp1 | nat=yes under [general] didn't work. |
| 20:19.11 | *** join/#asterisk Stridernzl (n=neville@222-152-248-128.jetstream.xtra.co.nz) |
| 20:19.53 | ManxPower | Carp1: What is the diagram of the connection? i.e. Asterisk <-> NAT router <-> Internet <-> SIP device? |
| 20:20.31 | Carp1 | Asterisk - Linksys router with DD-WRT - Internet |
| 20:20.44 | Carp1 | asterisk and phone are plugged into the router. |
| 20:21.15 | Carp1 | And NuFone providing inbound and outbound calls. |
| 20:21.16 | ManxPower | So you are trying to do a call between a phone on the local network and an asterisk on the local network? |
| 20:21.29 | ManxPower | What is the specific diagram of a problem call? |
| 20:21.39 | Carp1 | phone - asterisk -NuFone |
| 20:21.53 | ManxPower | And you are connecting to Nufone via IAX or SIP? |
| 20:22.05 | Carp1 | when i call my NuFone DID, I have * rinf my phone |
| 20:22.06 | Carp1 | SIP. |
| 20:22.23 | ManxPower | and you are calling out from the phone to a PSTN number via NuFone? |
| 20:22.46 | ManxPower | Well via Asterisk via NuFone |
| 20:23.09 | Carp1 | Yes. |
| 20:23.28 | Carp1 | Both ways, it doesnt work....I think....I can check quick. |
| 20:23.47 | ManxPower | And you have set /etc/asterisk/rtp.conf to be the same port range as your are port forwarding on your router? You set localnet= and externip= in [general] in sip.conf? |
| 20:24.11 | Carp1 | No.... |
| 20:24.15 | Carp1 | I didnt do any of that. |
| 20:24.19 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:24.24 | ManxPower | Carp1: Then it's not going to work. |
| 20:24.31 | ManxPower | and you will get 1-way audio |
| 20:24.48 | *** join/#asterisk yonahw (n=yonahw@IGLD-83-130-182-235.inter.net.il) |
| 20:25.15 | Carp1 | Ok....I wasnt aware of rtp.conf. |
| 20:25.28 | ManxPower | nat=yes is for a PHONE behind a REMOTE NAT. |
| 20:25.59 | *** join/#asterisk edgecase (n=jjackson@CPE00a0c9841796-CM000f9fa6b7d6.cpe.net.cable.rogers.com) |
| 20:26.17 | Carp1 | So if I added an extension and hooked a phone up at another location, I would use nat=yes? |
| 20:26.41 | edgecase | ok chan_bluetooth, go! |
| 20:26.41 | ManxPower | if that remote location was behind a NAT router, yes |
| 20:26.51 | Carp1 | Ok. |
| 20:27.45 | ManxPower | rip.conf tells asterisk what ports to accept incoming audio on. You must port forward that range of ports to the asterisk server on your NAT router. externip and localnet tells asterisk what devices are local and do not need fixup of the SIP headers. |
| 20:28.23 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 20:29.12 | Carp1 | rip.conf? |
| 20:29.18 | Carp1 | I dont see anything on google about it. |
| 20:29.50 | ManxPower | rtp.conf |
| 20:30.06 | ManxPower | see /path/to/src/asterisk/configs/rtp.conf.sample |
| 20:30.19 | Carp1 | Ok... |
| 20:30.41 | Carp1 | So I need to foward rtpstart and rtpend in my router? |
| 20:30.45 | Carp1 | 10000 to 20000 |
| 20:31.42 | ManxPower | yes, if you leave the defaults |
| 20:32.19 | ManxPower | Ithe porwarding would be UDP, of course. You also need to forward 5060 UDP |
| 20:32.52 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 20:33.17 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:34.35 | Carp1 | hmm |
| 20:34.44 | Carp1 | I already have 5060 fowarded to my ATA |
| 20:34.48 | EmleyMoor | Is nat=yes for situations when the extension is behind NAT or simply when it might be? |
| 20:35.28 | Carp1 | I'm wrong, 5060 is fowarded to * |
| 20:35.37 | Carp1 | hmm, still one way audio though. |
| 20:35.55 | blitzrage | EmleyMoor: when it might be -- basically it is just to tell Asterisk to ignore the IP in the Contact field and use the IP of where the packet actually came from |
| 20:36.06 | blitzrage | Carp1: 5060 is just for signalling and has nothing to do with RTP (media) |
| 20:36.24 | Carp1 | Ok. |
| 20:36.33 | EmleyMoor | Thought so |
| 20:36.37 | ManxPower | Carp1: you must forward all the ports listed in rtp.conf to your asterisk server |
| 20:36.37 | EmleyMoor | (hence my setting) |
| 20:36.43 | Carp1 | Well, I fowarded the range I have set in ftp.conf and I'm still getting one way audio. |
| 20:36.43 | ManxPower | I'm not going to say it again. |
| 20:36.58 | EmleyMoor | If it really gives me trouble, I use an iax phone while I investigate |
| 20:37.05 | ManxPower | Carp1: forwarding will do no good without localnet and externip |
| 20:37.12 | Carp1 | ManxPower: I did. |
| 20:37.14 | nibbler_de | is there a way to destinguish further than via dialstatus CHANUNAVAIL if a sip-peer exists or is just not logged in? |
| 20:37.19 | Carp1 | I forwarded the range. |
| 20:37.45 | ManxPower | Carp1: and you stopped and started asterisk or did a reload |
| 20:38.12 | Carp1 | Just a reload, But I didnt edit the config file....I left the default values. |
| 20:38.37 | ManxPower | Carp1: You cannot "just leave the default values" you have to set the values for your netwokr |
| 20:38.59 | ManxPower | externip is the public IP address of your connection |
| 20:39.12 | ManxPower | localnet specifies what your local network is |
| 20:39.44 | Carp1 | The sample config didnt show any of that. |
| 20:39.58 | Carp1 | Just rtpstart=10000 and rtpend=20000 |
| 20:40.52 | bkruse_home | Carp1: that should be default, correct? |
| 20:41.23 | Carp1 | Yes, it is. |
| 20:41.36 | ManxPower | Carp1: rtpstart and rtpend are in rtp.conf, you need to add localnet and externip in sip.conf |
| 20:42.27 | ManxPower | Carp1: It won't work unless all settings are correct. |
| 20:42.28 | Carp1 | Ok....Under [general] I assume. |
| 20:42.45 | EmleyMoor | Is there a recipe for dialplan-controlled call diversion on the web anywhere? |
| 20:43.06 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 20:43.07 | ManxPower | Carp1: localnet and externip would be in [general] in sip.conf, yes. |
| 20:43.32 | *** join/#asterisk chema (n=root@138.Red-81-37-130.dynamicIP.rima-tde.net) |
| 20:43.36 | Carp1 | localnet=192.168.1.1.255.255.255.0 |
| 20:43.50 | Carp1 | externalip=24.x.x.x (my outside IP) |
| 20:43.59 | Carp1 | whoops. |
| 20:44.04 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 20:44.06 | ManxPower | Carp1: the sip.conf.sample shows a / between the network and the netmaks |
| 20:44.17 | Carp1 | yeah, I made a typo. |
| 20:44.22 | ManxPower | localnet would be localnet=192.168.1.0/255.255.255.0 |
| 20:44.33 | ManxPower | assuming your internal network is 192.168.1.* |
| 20:44.42 | Carp1 | Yes, it is. |
| 20:45.05 | chema | <PROTECTED> |
| 20:45.16 | ManxPower | Carp1: also put caneinvite=no in the section of sip.conf for your phone, just in cse |
| 20:47.43 | Carp1 | Ok, I set those up and did a full restart...Still one way audio....I cant hear any audio in my earpiece on my IP phone. |
| 20:50.13 | ManxPower | To review: You forwarded UDP ports 10000-20000 and 5060 from your NAT router to the IP of the Asterisk server? You set localnet= and externip= to the correct values for your local network in sip.conf [general]. You set canreinvite=no in the section of your sip.conf for your phone. Just to be sure, also put disallow=all and allow=ulaw and allow=gsm in [general] sip.conf. Do not ever put allow=all in sip.conf. |
| 20:50.34 | markit | hi, is there the possibility to have a test "call back" in iax protocol to test if I'm reachable from the outside? |
| 20:51.13 | Carp1 | Ok...let me make a couple changes in sip.conf |
| 20:53.47 | markit | found, fwd does, thanks |
| 20:53.52 | Carp1 | Ok. I did everything you said....Still one-way audio :( |
| 20:53.57 | markit | ops, with sip :( |
| 20:54.26 | ManxPower | Carp1: do a sip debug and out the CLI output if a failed call on pastebin.ca |
| 20:54.42 | ManxPower | also put your sip.conf and the Dial line on pastebin.ca |
| 20:55.03 | Carp1 | Ok. |
| 20:56.21 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:57.23 | *** join/#asterisk nirz (n=nir@bzq-88-154-194-62.red.bezeqint.net) |
| 20:57.47 | Carp1 | CLI dump: http://pastebin.ca/372107 |
| 20:58.30 | nirz | hello , iv installed trixbox, but there is a problom with the root user(bad password) |
| 20:58.38 | nirz | iv tried root/password |
| 20:58.44 | Carp1 | exten: http://pastebin.ca/372108 |
| 20:58.49 | bkruse_home | nirz: /join #tribox |
| 20:58.52 | bkruse_home | trixbox* |
| 20:59.40 | nirz | thanks |
| 21:00.53 | Carp1 | sip: http://pastebin.ca/372112 |
| 21:01.21 | mranostay | This book is scary |
| 21:01.28 | ManxPower | So your failure is on an inbound call, not an outbound call. |
| 21:02.05 | ManxPower | and your sip.conf? |
| 21:02.09 | mranostay | anyone else read The Art of Demotivation? |
| 21:02.38 | Carp1 | Yes inbound only. |
| 21:02.49 | ManxPower | Carp1: what NAT router do you have? |
| 21:03.02 | Carp1 | Linksys flashed with DD-WRT |
| 21:03.44 | ManxPower | do you have the firewall features enabled? |
| 21:04.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 21:04.18 | Carp1 | SPI Firewall |
| 21:04.21 | Carp1 | is enabled. |
| 21:04.32 | ManxPower | turn that off |
| 21:05.05 | Carp1 | ok. |
| 21:05.27 | ManxPower | If that fixes it then you know the firewall is breaking it and you need to tell the firewall to allow all outbound UDP |
| 21:06.06 | wunderkin | i have spi on with dd-wrt and it works ok for me, didn't realize i had it on |
| 21:07.17 | ManxPower | wunderkin: are your running Asterisk behind that box, talking to sip devices on the internet? |
| 21:07.26 | *** part/#asterisk chema (n=root@138.Red-81-37-130.dynamicIP.rima-tde.net) |
| 21:07.31 | wunderkin | oh... yeah i don't have asterisk behind it.. just phones |
| 21:07.45 | Carp1 | hmmm |
| 21:07.48 | ManxPower | then your config is not comparable to Carp1's config |
| 21:07.50 | Carp1 | That didnt fix it either. |
| 21:07.56 | wunderkin | i wasn't paying attention sorry |
| 21:08.11 | ManxPower | Carp1: This is my THIRD request for your sip.conf to be put on pastebin.ca |
| 21:08.43 | Carp1 | I did lol |
| 21:08.54 | ManxPower | and that URL would be? |
| 21:08.57 | Carp1 | scroll us a little but. |
| 21:08.59 | Carp1 | but* |
| 21:09.04 | Carp1 | bit** |
| 21:10.23 | ManxPower | Carp1: you misspelled externip. |
| 21:10.27 | ManxPower | you called it externalip |
| 21:10.42 | Carp1 | Damn....Good catch, |
| 21:10.49 | ManxPower | also remote the port= line |
| 21:11.24 | Carp1 | remove? ok. |
| 21:11.35 | ManxPower | remove the port=5060 line, that is. |
| 21:14.52 | Carp1 | hmm |
| 21:14.59 | Carp1 | I cant call in my box at all now. |
| 21:15.17 | Carp1 | I am pretty sure its a problem with my internet connection though. |
| 21:16.34 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 21:22.39 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 21:26.48 | *** join/#asterisk masterm1nd (n=worldent@83.111.92.26) |
| 21:27.14 | masterm1nd | guys .. is this the place to ask config-related questions or is there another channell ? |
| 21:28.33 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 21:28.45 | masterm1nd | ?? |
| 21:29.41 | JT | this is a channel where you ask your question and hope someone is able to respond, not where you ask to ask :) |
| 21:30.00 | masterm1nd | ok .. here's the question |
| 21:30.15 | masterm1nd | I have box with one PRI |
| 21:30.23 | masterm1nd | attached to a PSTN gateway |
| 21:30.42 | masterm1nd | I want the box to play a message while the call is being switched from the VoIP side to PRI side |
| 21:31.15 | masterm1nd | if the call is successfully setup on the PRI side, i.e. you get a ringing tone, then it's switched |
| 21:31.18 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 21:31.25 | JT | what do you mean box with pri is attached to a pstn gateway, are they not the same thing? |
| 21:31.43 | masterm1nd | otherwise, I want the script to wait 20 seconds, and then drop the call with an unsuccessful message |
| 21:32.16 | masterm1nd | no, they are not. the PSTN gwateway in this case is a GSM gateway |
| 21:32.25 | masterm1nd | we pickup calls from VoIP side |
| 21:32.29 | JT | well you could've been more specific :) |
| 21:32.34 | masterm1nd | then route over the PRI card to the GSM gateway |
| 21:32.35 | JT | a box with a pri is also a pstn gateway |
| 21:32.49 | masterm1nd | ok .. symantics aside .. |
| 21:33.00 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:33.10 | JT | just trying to understand what is going on |
| 21:33.17 | masterm1nd | here it is: |
| 21:34.07 | masterm1nd | [VOIP] -> [asterisk H323 channel -> PRI card] -> GSM gateway |
| 21:34.28 | JT | oh ok |
| 21:34.28 | masterm1nd | issue is that I don't want network announcements to be played back to caller |
| 21:34.49 | masterm1nd | so really, caller calls, gets played a message "please wait while we connect you" ..etc |
| 21:34.57 | masterm1nd | call setup attempt |
| 21:35.13 | masterm1nd | if the call gets connected, bridging happens |
| 21:35.22 | masterm1nd | otherwise, wait 20s and timeout with a failure message |
| 21:35.50 | JT | what's the 20seconds for, waiting for a response from gsm? |
| 21:35.58 | masterm1nd | x |
| 21:35.59 | masterm1nd | yes |
| 21:36.20 | masterm1nd | because if a network announcement is being played, the call terminates in 11 seconds |
| 21:36.24 | lenne_dk | Waiting for the phone to get picked up |
| 21:36.33 | masterm1nd | exactly |
| 21:36.50 | JT | well if the gateway suitably signals on the pri, there's no need for silly timeouts :) |
| 21:36.53 | JT | but it might not |
| 21:36.59 | JT | i guess you'll have to find out |
| 21:37.29 | masterm1nd | it's not a signalling issue |
| 21:37.34 | *** join/#asterisk vlt|home (n=daniel@dslb-088-073-177-039.pools.arcor-ip.net) |
| 21:37.46 | masterm1nd | I just don't want it to play out of credit announcements |
| 21:38.02 | JT | haha |
| 21:38.31 | JT | so how will asterisk now if it is to timeout with a failure message, or to bridge the call? |
| 21:38.34 | JT | s/now/know/ |
| 21:38.51 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 21:38.54 | masterm1nd | sorry, my mistake |
| 21:38.58 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 21:39.02 | masterm1nd | we wait 7 seconds for setup |
| 21:39.05 | masterm1nd | not 20 seconds |
| 21:39.16 | masterm1nd | announcements are 11 seconds end-end |
| 21:39.26 | masterm1nd | i.e. connect -> announce -> drop |
| 21:39.31 | masterm1nd | this is an 11 second affair |
| 21:39.41 | masterm1nd | 7 - 9 seconds is within our acceptable PDD |
| 21:39.46 | JT | at the pri level, both a bridged call and an out of credit announcement, do they look the same? |
| 21:39.53 | JT | pdd? |
| 21:39.58 | masterm1nd | post dial delay |
| 21:40.07 | masterm1nd | the only difference is the duration of the call |
| 21:40.13 | masterm1nd | out of credit is always 11 seconds long |
| 21:40.15 | masterm1nd | thn drop |
| 21:40.24 | masterm1nd | s/thn/then/g |
| 21:40.46 | JT | so how do you tell whether it's an out of credit announcement, or the called party being bridged? |
| 21:41.03 | masterm1nd | hence the delay before bridging |
| 21:41.11 | lenne_dk | does the gsm announce the credit even if it is high? My network only announces if it is below 25kr, approx Eur 3.5 |
| 21:41.28 | masterm1nd | this is not really the issue |
| 21:41.35 | masterm1nd | the route is heavily loaded |
| 21:41.53 | masterm1nd | and you can never gaurantee a particular credit level throughout the day |
| 21:42.01 | masterm1nd | so, as a failsafe mechanism |
| 21:42.06 | lenne_dk | If you delay the setup always, you might cut off the start of the call |
| 21:42.11 | masterm1nd | I want the gateway to surpress out of credit announcements |
| 21:42.36 | JT | i still think you must be able to tell the difference if you want to handle them differently |
| 21:42.49 | masterm1nd | JT/all .. |
| 21:42.55 | lenne_dk | I have a send/expect script which logs on to my gsm-suppliers webpage to read the balance |
| 21:43.11 | *** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net) |
| 21:43.12 | vlt|home | Hello. I need some help configuring zaptel.conf and zapata.conf to run a QuadBRI card. I loaded the qozap module successfully. But after running `ztconf` I get lots of "CRC error for HDLC frame" messages. Maybe that's no problem but I don't know how to access the ports from extensions.conf. When I do a call on one of the BRI lines nothing happens (in *CLI). The error log: http://rafb.net/p/NEaoWb85.html |
| 21:43.13 | masterm1nd | if I can setup the call within the allowed 7-9s delay, then the call gets switched |
| 21:43.17 | masterm1nd | any longer and it's dropped |
| 21:43.20 | vlt|home | Can anyone help? |
| 21:43.24 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 21:43.42 | DocHolliday | is there an asterisk 1.4 install manual somewhere? |
| 21:43.43 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:43.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:44.03 | masterm1nd | lenne_dk .. no such facility on T-Mob UK |
| 21:44.31 | JT | masterm1nd: doesn't that mean up to 9 seconds of the call could be dead air for the callED party? |
| 21:44.49 | lenne_dk | Can you "recharge" your account over the web? |
| 21:45.06 | JT | vlt|home: crc errors aren't good |
| 21:45.10 | masterm1nd | JT .. no |
| 21:45.23 | masterm1nd | so really, if the call can be setup within 9 seconds, we switch |
| 21:45.39 | masterm1nd | i.e up to 9 seconds to setup the call is acceptable |
| 21:45.46 | masterm1nd | any longer, and it's an announcement |
| 21:45.46 | vlt|home | JT: So I should first get rid of them before trying to do anything in extensions.conf, right? |
| 21:46.06 | JT | masterm1nd: if an out of credit announcement is played, does the call not go to SETUP, or not fast? |
| 21:46.18 | masterm1nd | aha .. not fast |
| 21:46.19 | JT | vlt|home: yes, how often do they happen? |
| 21:46.41 | masterm1nd | so really, if the call gets setup in less than 9 seconds, then it is not an announcement |
| 21:46.48 | JT | masterm1nd: i'm seriously not sure if asterisk can do that |
| 21:47.46 | masterm1nd | I thought there was a T variable somewhere, which allowed you to set a maximum limit on dials |
| 21:47.48 | vlt|home | JT: Saw the logfile I pasted? Every 4-5 seconds on each (of the connected) ports |
| 21:48.28 | lenne_dk | The demo on t-mobile.co.uk shows you can top up and see the credit balance. |
| 21:49.04 | lenne_dk | So it should be possible to script a balance check and either top up, or call somebody to do it automatically. |
| 21:49.25 | JT | masterm1nd: i think the timeout variable waits for an extension to be answered, not for pri SETUP |
| 21:49.45 | masterm1nd | well, PRI is just an extention. that bit works. |
| 21:49.57 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 21:50.30 | JT | you want a timeout for PRI SETUP, i'm saying i don't think asterisk can handle this case |
| 21:50.36 | masterm1nd | lenne_dk: can you perhaps give me your send/expect script and the associated config in asterisk to invoke it ? |
| 21:50.57 | JT | give, lol, i'd think i'd be worth money :) |
| 21:51.13 | masterm1nd | lol |
| 21:51.16 | masterm1nd | ok, not give. |
| 21:51.26 | masterm1nd | just how do you invoke a script pre-dial ? |
| 21:51.29 | lenne_dk | Wouldn't help you much, because I use the danish provider CBB. |
| 21:52.19 | masterm1nd | but how do you invoke it before dialing every call ? |
| 21:52.22 | JT | you might need to hack on zaptel/libpri to get what you want |
| 21:52.24 | masterm1nd | where do you call it from ? |
| 21:53.47 | masterm1nd | JT .. that's what I was afraid of |
| 21:53.47 | vlt|home | JT: When I first plugged in the BRI card into my asterisk machine (ubuntu) an older hisax driver (now blacklisted) loaded. I could see every incoming call in syslog so I assume it's not a hardware problem ... |
| 21:53.47 | masterm1nd | but .. if I can ballance check before dialing, then that's problem solved |
| 21:53.47 | JT | vlt|home: did you patch qozap? |
| 21:53.48 | lenne_dk | It's actually a shell-script which "logs in" with wget. Then parse the amount with perl to deduce the amount to top up, then use the output as post-variables to wget again. |
| 21:54.03 | lenne_dk | I don't use it for the gsm gateway. |
| 21:54.10 | vlt|home | JT: Yes, I applied a patch I found for using Beronet cards with Junghanns driver and patched that again for using it with my QuadBRI clone with different vendor ID ... |
| 21:54.17 | masterm1nd | that's fine, but what I am curious about is the asterisk integration |
| 21:54.37 | JT | vlt|home: so what's plugged into it? |
| 21:56.24 | vlt|home | JT: It's an HFC4S based "4xS0" card that looks like a Beronet but is mentioned in Junghanns's qozap source as "CologneChip HFC-4S evaluation board". |
| 21:56.32 | *** join/#asterisk vanumo (n=test@80.122.72.250) |
| 21:56.33 | lenne_dk | However, that might be possible. what I DO do prior to a call is have a agi-call log on to my voip-provider, which have a price-lookup page. I read the per-minute price of the call, (cache the result), then only allow the calls from certain phones, if the price is too high. |
| 21:56.38 | JT | ah ok, did we speak before? |
| 21:56.40 | vanumo | hi |
| 21:56.49 | vlt|home | JT: Yes it's me again :( |
| 21:56.51 | vanumo | i have a question about chan_cellphone |
| 21:57.27 | JT | vlt|home: what's plugged into the card? |
| 21:57.44 | vanumo | i have connect my cellphone to asterisk via bluetooth over chan_cellphone i can make outgoing calls but i can't get incominng calls from gsm to my sip phone ?` |
| 21:58.24 | lenne_dk | masterm1nd: But the simple solution is to just make sure the credit is high to cover a days worth of calls. :-) |
| 21:58.36 | vlt|home | JT: port 1 and 2 is connected to telco's NT, port 3 to one of my ISDN pbx's "internal S0" ports. |
| 21:58.43 | mafkees | vanumo: unload chan_cellphone.so, set debugging to 3, load chan_cellphone.so, call your gsm and catch the DEBUG logging lines |
| 21:58.51 | JT | vlt|home: port 4 to nothing? |
| 21:58.55 | masterm1nd | lenne_dk, trust me it's not always possible with a large number of gws |
| 21:58.55 | mafkees | attach that to the ticket on bugs.digium.com |
| 21:59.06 | vlt|home | JT: port 4 to nothing, right. |
| 21:59.37 | JT | vlt|home: port 3 is setup wrong, it needs to be in NT mode |
| 21:59.43 | lenne_dk | Wouldn't it be simpler and cheaper to have billed cards instead of prepaid cards? |
| 21:59.50 | JT | what sort of cables are you using, vlt|home ? |
| 21:59.53 | vlt|home | JT: No, I don't think so. |
| 22:00.18 | vlt|home | JT: It's plugged into a "client S0" where ISDN phones are connected |
| 22:00.27 | JT | also make sure you use the correct line signalling, PTMP vs PTP |
| 22:00.44 | JT | oh ok, might work then |
| 22:00.49 | vanumo | mafkees why ? i think this is only a part in extensions conf |
| 22:00.56 | vlt|home | JT: ordinary ISDN cables, not longer than 2 metres, provided by the card vendor, I remember. |
| 22:00.56 | lenne_dk | Or don't the GSM-company like gateways, and shut down the cards when they discover them? |
| 22:01.08 | masterm1nd | lenne_dk: exactly |
| 22:01.24 | masterm1nd | not only that, you also remain liable for the contract even tough they cut you off |
| 22:01.30 | mafkees | ah, if it's an extensions.conf problem you can see that when you do: core set verbose 255 |
| 22:01.35 | masterm1nd | so it's a reall pain in the ass |
| 22:02.01 | vanumo | mafkees have you an sample configuration for incoming calls from chan_cellphone? |
| 22:02.16 | mafkees | nope |
| 22:02.27 | mafkees | I want to test it, but I cant get it to work on my ibook |
| 22:03.30 | vanumo | exten => Cell/David,1,Dial,SIP/30|30|r this have i entry for incoming calls from chan_cellphone ? |
| 22:03.48 | vlt|home | JT: The card appears as "BeroNet BN4S0 card" in the qozap debug because I didn't change that string in the source when modifying it to match my vendor id ... |
| 22:03.58 | *** part/#asterisk masterm1nd (n=worldent@83.111.92.26) |
| 22:04.12 | JT | ok |
| 22:04.16 | mafkees | huh ? |
| 22:04.22 | JT | i think it was a reference design |
| 22:04.23 | mafkees | exten => Cell/David ? |
| 22:05.22 | vanumo | i think this false but i don`t know what i should be entry ? |
| 22:05.24 | JT | vlt|home: is termination switched on the ports? |
| 22:05.34 | JT | vlt|home: and are you using the correct topology setting? |
| 22:06.12 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 22:06.41 | mafkees | vanumo: try this: |
| 22:06.52 | mafkees | for the phone entry in cellphone.conf |
| 22:06.58 | mafkees | give it some context |
| 22:07.03 | mafkees | in that context: |
| 22:07.25 | mafkees | exten => _X.,1,Verbose(1,Call from cellphone to number ${EXTEN}) |
| 22:07.59 | mafkees | exten => _X.,n,Goto(some_context_where_your_sip_phone_is|extension_for_your_sip_phone) |
| 22:08.13 | mafkees | that way it will show you the exten you can use for that phone |
| 22:09.18 | vlt|home | JT: On all ports the card is one of two ISDN clients on the bus (client 1 on the telco's NTs is the pbx, client 1 on the pbx's bus is an ISDN phone). What does "topology setting" mean? |
| 22:09.33 | lenne_dk | masterm1nd: I might accept the challenge of writing a script to check the balance on a t-mobile card. For the price of a sim with £10. |
| 22:10.02 | vanumo | ok sip i call local as context |
| 22:10.13 | JT | vlt|home: there are 2 isdn modes, Point to Point (PTP) and Point to MultiPoint (PTMP) |
| 22:10.16 | vanumo | i can use für cell maybe cell as context ? |
| 22:10.22 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 22:10.26 | JT | vlt|home: check with your provider and pabx settings |
| 22:10.48 | JT | lenne_dk: he left |
| 22:11.03 | mafkees | vanumo: yeah |
| 22:11.25 | JerJer | mafkees: Verbose? i always use NoOP |
| 22:11.36 | lenne_dk | Damn. :-) Can I write offline like yahoo so he gets it if he logs back on? |
| 22:11.48 | mafkees | JerJer: yeah, but I like my pbx to run at: core set verbose 1 |
| 22:11.52 | mafkees | you wont see the noop |
| 22:12.06 | mafkees | lenne_dk: gheh, nope |
| 22:12.08 | JerJer | ok - good point |
| 22:12.10 | vlt|home | JT: zapata.conf is set to "p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)": signalling = bri_cpe_ptmp |
| 22:12.11 | mafkees | lenne_dk: they call it email |
| 22:12.24 | JT | JerJer: yeah Verbose allows you to set the level it appears at |
| 22:12.35 | JT | vlt|home: check with your telco |
| 22:12.45 | vlt|home | JT: But the errors occur already before loading chan_zap.so ... |
| 22:13.18 | JT | ok, umm, pastebin zaptel.conf and zapata.conf |
| 22:13.22 | vlt|home | JT: So maybe it's one of zaptel.conf's settings .. I'll paste... |
| 22:13.52 | JT | vlt|home: is termination enabled in the card and on the NT1? |
| 22:14.00 | lenne_dk | mafkees: email needs something called a domain. You know, the thing on the right of @ |
| 22:14.19 | lenne_dk | Do you know that? |
| 22:14.21 | vanumo | mafkees you mean it so : [lokal] |
| 22:14.21 | vanumo | ; Erreichbarkeit der Nebenstellen 30-39 |
| 22:14.21 | vanumo | ; untereinander herstellen |
| 22:14.21 | vanumo | exten => _3X,1,NoCDR() |
| 22:14.21 | vanumo | exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr |
| 22:14.21 | vanumo | exten => _9X.,1,Dial(CELL/David/${EXTEN:1},45,tT) |
| 22:14.23 | vanumo | exten => _9X.,n,Hangup |
| 22:14.25 | vanumo | [cell] |
| 22:14.27 | vanumo | exten => _X.,1,Verbose(1,Call from cellphone to number ${EXTEN}) |
| 22:14.29 | vanumo | exten => _X.,n,Goto(lokal|30) |
| 22:14.41 | mafkees | yeah exactly |
| 22:14.44 | JT | vanumo: oi! |
| 22:14.44 | mafkees | that should work |
| 22:14.50 | JT | vanumo: pastebin.ca |
| 22:15.01 | mafkees | if you put 'context => cell' in your cellphone.conf |
| 22:15.02 | vanumo | sorry JT |
| 22:15.11 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.140.56.dyn.user.ono.com) |
| 22:15.11 | mafkees | ~pb |
| 22:15.19 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 22:15.23 | JT | < mafkees> lenne_dk: gheh, nope |
| 22:15.38 | JT | that's not so, you CAN send messages to people for when they logon |
| 22:15.44 | JT | if they are registered |
| 22:15.46 | mafkees | for real ? |
| 22:15.55 | JT | and you cannot get in this channel without being registered |
| 22:15.58 | JT | <PROTECTED> |
| 22:16.11 | mafkees | ah, freenode has memoserv |
| 22:16.28 | vlt|home | JT: On the NT: yes. (NT <---> pbx and pbx <---> isdn phone work fine) On the card I don't know. Is it a qozap option? |
| 22:16.42 | JT | no it is usually a hardware jumper |
| 22:16.50 | vanumo | mafkees what do you mean with call from cellphone to number ? |
| 22:17.03 | JT | 2 metre cable will probably cause tonnes of reflections with insufficient termination |
| 22:17.16 | mafkees | vanumo: it will tell you what the 'exten => some_nr' should be |
| 22:17.24 | mafkees | it's just information |
| 22:17.32 | mafkees | to get the right exten for it |
| 22:17.37 | mafkees | you now have Cell/dave |
| 22:17.43 | mafkees | that is not working for incoming calls |
| 22:17.49 | mafkees | it needs to be the correct extension |
| 22:17.54 | mafkees | but because you dont know it |
| 22:17.59 | mafkees | this little trick will tell you |
| 22:18.16 | vanumo | the correct extensions will be the number of cellphone ? |
| 22:19.01 | vlt|home | JT: I think hardware is configured fine because it kinda worked with the automatically loaded hisax or whatever driver (forgot the name) .. "works" means I could see what was happening on the D channels (calls, caller IDs, dialled MSNs ...) |
| 22:19.02 | vanumo | the context in cellphone.conf is David ;-) |
| 22:19.23 | JT | vlt|home: do a zttest then |
| 22:20.13 | vanumo | i am to stupid for it *grml* |
| 22:20.48 | mafkees | vanumo: I have no idea |
| 22:20.50 | mafkees | try it |
| 22:21.57 | mafkees | vlt|home: what side is not working ? |
| 22:22.07 | mafkees | the NT => pbx or the pbx => phone ? |
| 22:22.32 | vlt|home | JT: "Opened pseudo zap interface, measuring accuracy..." Then "8192 samples in 16383 sample intervals 0.012207%" or "... 0.000000%", one line a second ... counting ... |
| 22:22.58 | JT | 0.01....?? |
| 22:23.06 | JT | you must ctrl c to stop |
| 22:23.12 | JT | what is average |
| 22:23.22 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 22:23.25 | JT | what comes up the most, and what are the LOWEST numbers you see? |
| 22:23.42 | vlt|home | JT: 98 passes: Best: 0.012207 -- Worst: -0.012207 -- Average: 0.003737 |
| 22:23.53 | JT | vlt|home: something is stuffed |
| 22:23.57 | JT | those numbers are all wrong |
| 22:24.07 | JT | it should be percentages, 99%+ |
| 22:24.11 | vlt|home | ooops |
| 22:24.35 | JT | i think maybe it's the wrong version of zttest or something, it's just not parsing the data right |
| 22:26.12 | vlt|home | mafkees: I can't access the ports of the QuadBRI card from asterisk. |
| 22:27.10 | mafkees | hhmm |
| 22:27.20 | mafkees | I use them on several locations |
| 22:27.31 | JT | vlt|home: it's misleading to say QuadBRI |
| 22:27.41 | JT | it's a 4 port BRI card, possibly reference design |
| 22:27.45 | vlt|home | mafkees: It's a QuadBRI clone, actually |
| 22:27.48 | mafkees | ah |
| 22:27.50 | mafkees | ok |
| 22:28.00 | mafkees | I only use normal quadbri cards |
| 22:28.55 | vlt|home | mafkees: It's HFC-4S based, too, and looks like a beronet card ;-) |
| 22:29.19 | mafkees | what did you try ? |
| 22:29.20 | vlt|home | .cn |
| 22:29.22 | mafkees | bristuff I guess |
| 22:29.28 | vlt|home | mafkees: Yes |
| 22:29.34 | mafkees | did you try misdn ? |
| 22:29.38 | *** join/#asterisk KnowWhat (n=KnowWhat@63.246.132.30) |
| 22:29.59 | JT | vlt|home: so you still have no idea where the card came from? |
| 22:29.59 | mafkees | if you load the qozap.so, what does it say in dmesg ? |
| 22:30.03 | vlt|home | mafkees: I wanted to but the folks here tried to stop me ;-) |
| 22:30.44 | mafkees | the folks here are good ;) |
| 22:31.08 | mafkees | maybe I missed part of the conversation |
| 22:31.21 | mafkees | but can you pastebin the dmesg output of 'modprobe qozap' |
| 22:31.34 | vlt|home | mafkees: That was two weeks ago ... |
| 22:31.43 | mafkees | ah, sorry |
| 22:31.56 | vlt|home | mafkees: Yes: http://rafb.net/p/NEaoWb85.html |
| 22:32.23 | vlt|home | mafkees: It says "Beronet" because I didn't change that string in the qozap source ... |
| 22:33.09 | mafkees | uhhuh |
| 22:33.11 | mafkees | no need to |
| 22:33.26 | vlt|home | JT: I didn't manage to look between the card in the lowest PCI slots and the atx case ;-) ... and the server never went down since then ... |
| 22:33.29 | mafkees | if it finds the ports I really dont care what name it gives the card ;) |
| 22:33.50 | vlt|home | mafkees: Yes, I know, just wanted to prevent confusion ... |
| 22:33.56 | mafkees | hhmm, the infamous hdlc crc errors |
| 22:34.03 | JT | vlt|home: so you still can't explain how this card just appeared in the server? i find that very strange |
| 22:36.24 | vlt|home | JT: Maybe it appeared a bit strange to you that I didn't know the card's vendor (you thought something like inheritated I remember) ... I bought this card from our VoIP store as an "Asterisk 4xS0 card" |
| 22:36.34 | vlt|home | ... DOT C N ;-) |
| 22:36.54 | jql | good that you buy from reputable sources |
| 22:37.01 | jql | and not something that fell off the truck. :) |
| 22:37.50 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 22:37.52 | DocHolliday | i cant seem to find the package bison-devel is it important for asterisk 1.4? |
| 22:38.19 | KuJaX | Does anyone have any comments or experience with Sangoma FXO PCI with Echo Cancellation? Worth the extra cost over Digium? |
| 22:38.34 | *** join/#asterisk friedrich| (n=friedric@e177249184.adsl.alicedsl.de) |
| 22:40.05 | mafkees | vlt|home: you mailed that company to ask for help ? |
| 22:40.16 | mafkees | dumb question, I know |
| 22:40.27 | vlt|home | mafkees, JT: I think I should call him again, tomorrow. |
| 22:40.50 | mafkees | hhmm |
| 22:40.56 | vlt|home | mafkees: He said he had a lot of customers using the card successfully with * |
| 22:41.04 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 22:41.06 | JT | vlt|home: check the termination jumpers, try using longer cables |
| 22:41.08 | mafkees | the hdlc crc errors mostly come from bad cables |
| 22:41.32 | JT | a normal ethernet straight through cable will be fine |
| 22:41.46 | vlt|home | mafkees: Why did the folks here advise not to use misdn? |
| 22:41.54 | JT | vlt|home: make sure the termination jumpers are on |
| 22:42.18 | mafkees | misdn is a lot of work to setup |
| 22:42.33 | JT | that and i't alpha-code level software |
| 22:42.50 | JT | it's |
| 22:43.44 | vlt|home | JT: Can you imagine the card working (at least reading D channel) with wrong termination jumpers or bad cables when using the other driver? |
| 22:44.04 | DocHolliday | does asterisk 1.4 have an extra sounds and addons package? |
| 22:44.21 | JT | vlt|home: hrm not sure |
| 22:44.45 | vanumo | i geht message that Cell/David sento in invalid extension `s` |
| 22:44.46 | JT | vlt|home: do you have zaptel and zapata pastebinned? |
| 22:45.04 | *** join/#asterisk Vec (n=Vec@dsl-241-206-56.telkomadsl.co.za) |
| 22:45.11 | mafkees | hhmm |
| 22:45.16 | mafkees | vanumo: can you pastebin it ? |
| 22:45.29 | vanumo | <PROTECTED> |
| 22:45.37 | Vec | Is development still occuring on asterisk 1.2, or only bug fixes etc ? |
| 22:45.57 | JT | vanumo: do you have an extension s? |
| 22:46.33 | vanumo | no i have |
| 22:46.45 | JT | in the relevant context? |
| 22:46.53 | JT | EmleyMoor: haha, what's the point? |
| 22:46.53 | vlt|home | JT: zaptel.conf (from the junghanns page): http://rafb.net/p/raEfFa78.html |
| 22:47.22 | EmleyMoor | JT: I have a rotary phone and might not always be able to get at something easier |
| 22:47.24 | mafkees | hhmm |
| 22:47.26 | mafkees | hang on |
| 22:47.37 | EmleyMoor | (I rarely need call diversion, but I just cancelled BT's) |
| 22:48.02 | JT | EmleyMoor: do you actually have a sentimental liking for it or something? |
| 22:48.16 | vlt|home | JT: zapata.conf (from junghanns, too): http://rafb.net/p/S7PJfV59.html |
| 22:48.21 | JT | vlt|home: change all the x,x,3 bits to x,x,0 |
| 22:48.34 | JT | vlt|home: so the third parameter, change them all to 0 |
| 22:48.41 | EmleyMoor | The rotary phone? I like its bell and bought it for its amplified handset |
| 22:48.49 | vanumo | i pastebin my extenions.conf |
| 22:48.51 | JT | 3 is plain wrong, although i don't know if it makes any difference with bri |
| 22:48.54 | mafkees | vlt|home: try this: |
| 22:48.56 | mafkees | http://rafb.net/p/1fqRRR10.html |
| 22:49.11 | EmleyMoor | Someone is trying to get me a Hull rotary phone - the one I've got is Post Officer |
| 22:49.17 | EmleyMoor | Post Office, even |
| 22:49.27 | mafkees | vlt|home: that one is taken from a system that has the quadbri in production |
| 22:49.37 | Carp1 | ManxPower: You still here? |
| 22:49.39 | mafkees | just ssh-ed in for it |
| 22:50.15 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 22:50.18 | vanumo | [general] |
| 22:50.19 | vanumo | static=yes |
| 22:50.19 | vanumo | writeprotect=no |
| 22:50.19 | vanumo | ; -------------------------------------------------------------------- |
| 22:50.19 | vanumo | ; Es hat sich als gute Praxis erwiesen, die Inhalte der Datei |
| 22:50.19 | vanumo | ; extensions.conf modular aufzubauen. Diese Praxis wollen |
| 22:50.21 | vanumo | ; wir auch hier anwenden |
| 22:50.23 | vanumo | ; |
| 22:50.25 | vanumo | [lokal] |
| 22:50.27 | vanumo | ; Erreichbarkeit der Nebenstellen 30-39 |
| 22:50.29 | vanumo | ; untereinander herstellen |
| 22:50.29 | mafkees | vanumo: NOT HERE ! |
| 22:50.31 | vanumo | exten => _3X,1,NoCDR() |
| 22:50.32 | Carp1 | wtf |
| 22:50.32 | mafkees | pastebin it ! |
| 22:50.33 | vanumo | exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr |
| 22:50.35 | vanumo | exten => _9X.,1,Dial(CELL/David/${EXTEN:1},45,tT) |
| 22:50.37 | vanumo | exten => _9X.,n,Hangup |
| 22:50.37 | Carp1 | NEVER DO THAT AGAIN |
| 22:50.39 | mafkees | seesh |
| 22:50.39 | vanumo | [David] |
| 22:50.40 | JT | vanumo: stop pasting to channel |
| 22:50.41 | vanumo | exten => _X.,1,Verbose(1,David${EXTEN}) |
| 22:50.43 | vanumo | exten => _X.,n,Goto(lokal/SIP|30) |
| 22:50.45 | vanumo | <PROTECTED> |
| 22:50.45 | JT | vanumo: you've been warned already |
| 22:50.47 | vanumo | ; |
| 22:50.49 | vanumo | ; hier kommt der default-Context, in dem alle Geraete in der |
| 22:50.51 | vanumo | ; Grundkonfiguration erstmal laufen. |
| 22:50.53 | vanumo | ; Alle Geraete koennen sich gegenseitig anrufen |
| 22:50.55 | vanumo | [default] |
| 22:50.55 | mafkees | ut-oh |
| 22:50.57 | vanumo | include => lokal |
| 22:50.59 | vanumo | include => David |
| 22:50.59 | vlt|home | please! |
| 22:51.01 | vanumo | sorry |
| 22:51.03 | vanumo | http://pastebin.ca/372272 |
| 22:51.05 | vanumo | excuse me i want paste the link |
| 22:51.16 | DocHolliday | i got an error when compiling libpri about SELinux status |
| 22:51.46 | EmleyMoor | I only discovered today that it is valid to Goto a context which includes the context you actually need |
| 22:51.49 | DocHolliday | lol |
| 22:51.51 | mafkees | DocHolliday: you should disable selinux |
| 22:52.00 | DocHolliday | can anyone help me with a libpri selinux error |
| 22:52.00 | Carp1 | Any idea's why I'm getting only 1-way audio on incoming calls? |
| 22:52.16 | [TK]D-Fender | Carp1 : First guess, NAT issue |
| 22:52.28 | Carp1 | I dont think it should be an issue |
| 22:52.29 | DocHolliday | mafkees, any idea how i can do that? :P |
| 22:52.39 | mafkees | ehm, no |
| 22:52.43 | vanumo | in 10 minutes in my timezone it is 00:00 |
| 22:52.43 | mafkees | I never install it |
| 22:52.47 | Carp1 | I forwarded the ports defined in rtp.conf |
| 22:52.54 | EmleyMoor | Is there a program that can convert a table on a web page into something like CSV? |
| 22:53.18 | mafkees | EmleyMoor: shouldn't be too hard to code |
| 22:53.21 | KnowWhat | wow, EmleyMoor, if you find one let me know |
| 22:53.24 | Carp1 | I've added localnet to sip.conf and I tried to add externip but then it wouldnt let me recieve any incomgin calls. |
| 22:53.29 | [TK]D-Fender | Carp1 : Lots more to do than jsut that... |
| 22:53.30 | *** join/#asterisk Ebola (n=Ebola@host86-142-179-38.range86-142.btcentralplus.com) |
| 22:53.38 | EmleyMoor | I want a SIPBroker code list in domain name order |
| 22:53.58 | JT | EmleyMoor: yes there is a program |
| 22:54.08 | JT | EmleyMoor: don't laugh, it's called microsoft excel |
| 22:54.09 | Carp1 | Maybe you could point me in the right direction? |
| 22:54.11 | KnowWhat | JT: which one is that |
| 22:54.20 | mafkees | perl |
| 22:54.26 | JT | i've pasted stuff from html tables into excel |
| 22:54.39 | EmleyMoor | Hmmm... |
| 22:54.40 | KnowWhat | JT: does that open html table into it |
| 22:54.50 | KnowWhat | JT: ahhh he wants to get rid of copy pasting i thought |
| 22:54.57 | Vec | DocHolliday : y not disable selinux ? |
| 22:54.59 | JT | no, i copy it with web browser, paste into excel |
| 22:55.03 | JT | that's a little harder :) |
| 22:55.07 | JT | but doable |
| 22:55.27 | DocHolliday | Vec, i want to :P |
| 22:56.00 | Vec | DocHolliday : it should not be too difficult : google disable selinux |
| 22:56.07 | vanumo | mafkees do you have an idea ? |
| 22:56.14 | vanumo | what have i wrong ? |
| 22:56.15 | [TK]D-Fender | Carp1 : pastebin the [general] section of your sip.conf |
| 22:56.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:56.26 | DocHolliday | Vec, i got it |
| 22:58.01 | Carp1 | ok...give me a minute...im oln phone |
| 22:58.07 | EmleyMoor | I just want it to work |
| 22:58.16 | EmleyMoor | Copy and paste won't work with that page |
| 22:58.27 | JT | what happens? |
| 22:58.33 | JT | is it actually a html table? |
| 22:58.46 | EmleyMoor | It pastes the whole thing into 1 cell, so I suspect not |
| 22:59.02 | vlt|home | JT, mafkees: I tried all suggestions: changed 3rd bit to 0, changed "span=2,2,..." to "span=2,0,..." ... no success ... |
| 22:59.13 | JT | vlt|home: damn |
| 22:59.29 | JT | EmleyMoor: what browser? |
| 22:59.52 | EmleyMoor | Hmmm... it is |
| 22:59.54 | EmleyMoor | Firefox |
| 23:00.45 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-148-242.ny325.east.verizon.net) |
| 23:00.49 | mafkees | Day changed to 26 Feb 2007 |
| 23:01.26 | Carp1 | [TK]D-Fender: http://pastebin.ca/372112 |
| 23:01.40 | Carp1 | I fixed the typo in externalip to enternip |
| 23:01.59 | Carp1 | but right now I have it commented out because if I dont, it wont allow incoming calls, |
| 23:02.03 | JT | EmleyMoor: haven't tried with firefox, i used IE when i was successful |
| 23:02.06 | vanumo | ok thx good night |
| 23:02.31 | JT | mafkees: 10am here on the 26th |
| 23:03.30 | *** join/#asterisk netsurfer (n=netsurfe@user-514d1e74.l3.c1.dsl.pol.co.uk) |
| 23:03.37 | mafkees | try opera |
| 23:04.18 | [TK]D-Fender | Carp1 : Very important an missing in [general] : "nat=yes", "canreinvite=no", and make sure that typo's fixed |
| 23:05.44 | Carp1 | I have canreinvite under [NuFone-in] now |
| 23:06.05 | Carp1 | let me add nat=yes |
| 23:06.24 | [TK]D-Fender | Carp1 : add it to GENERAL. |
| 23:07.03 | mafkees | that way his phones will use that setting as well |
| 23:07.10 | mafkees | maybe that's not what he wants |
| 23:07.10 | Carp1 | ok |
| 23:07.14 | Carp1 | let me try this. |
| 23:08.05 | Carp1 | Ok, I've made those changes and still getting one-way audio. |
| 23:08.07 | JacksLivr | What is generally considered to be the best OS to load asterisk on? |
| 23:08.16 | Carp1 | Linux :) |
| 23:08.29 | JacksLivr | i know everyone has their own distro fav |
| 23:08.42 | JacksLivr | is fedora allroght? |
| 23:08.53 | [TK]D-Fender | Carp1 : Repastebin. And what are you forwarding exactly? |
| 23:09.02 | JT | i think you mean distro, not os, JacksLivr |
| 23:09.11 | [TK]D-Fender | JacksLivr : Whatever youre most comfortable administering. |
| 23:09.24 | JT | Carp1: what device is doing NAT? |
| 23:09.26 | Carp1 | im forwarding 10000 to 20000 to * server. |
| 23:09.47 | Carp1 | I'm guessing my NuFone connection would be NAT right? |
| 23:09.49 | [TK]D-Fender | JT : Linux is not an OS per-se, only a kernel. But thats a retarded semantic war jsut waiting to hapen. |
| 23:09.55 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:09.58 | mafkees | 2.6.20 has a state helper for sip ;) |
| 23:09.59 | [TK]D-Fender | Carp1 : What about SIp itself? |
| 23:10.06 | [TK]D-Fender | Carp1 : And what protocol? |
| 23:10.07 | JT | [TK]D-Fender: heh |
| 23:10.26 | JT | [TK]D-Fender: sounds like something an ESR fan would debate about |
| 23:10.28 | JT | omg gnu |
| 23:10.38 | mafkees | mooooooooooooo |
| 23:10.58 | mafkees | the only workable thing from gnu is gcc |
| 23:11.01 | mafkees | ;) |
| 23:11.09 | mafkees | <--- not a linux fan |
| 23:11.14 | Carp1 | [TK]D-Fender: I dont kn ow what you mean.. |
| 23:11.15 | JT | mafkees: linux nat is pretty decent, i haven't actually needed the sip contracker so far, have you found it useful? |
| 23:11.20 | Carp1 | sip is the protocol from NuFone. |
| 23:11.31 | [TK]D-Fender | mafkees : And Scrren, and a TON of other shit. Never diss the people who got us all where we are. |
| 23:11.32 | mafkees | JT: I never tried it |
| 23:11.37 | JT | he means did you forward tcp or udp? |
| 23:11.56 | JacksLivr | Oh, I prolly have most experience with Fedora and FreeBSD |
| 23:12.03 | [TK]D-Fender | Carp1 : ...... I was silently hoping you'd think about the protocol for those PORTS <- |
| 23:12.07 | mafkees | [TK]D-Fender: I'm not dissing them, I simply say _I_ dont like linux and the gnu stuff |
| 23:12.14 | JacksLivr | if Fedora is allright, I'll just stick with learning a new app instead of a new distro |
| 23:12.25 | mafkees | JacksLivr: indeed |
| 23:12.38 | mafkees | FC can run asterisk, no problem |
| 23:12.39 | Carp1 | hmmm. |
| 23:12.55 | JT | zapp-branigan: the device you put the port forwards on...... what is it? |
| 23:12.56 | KnowWhat | yeah it can |
| 23:13.01 | *** join/#asterisk dseeb_ (n=dcb@58.169.73.237) |
| 23:13.02 | KnowWhat | i dont know if it can run vicidial |
| 23:13.05 | KnowWhat | but i hope it can |
| 23:13.06 | JT | gar |
| 23:13.09 | JT | Carp1: i mean |
| 23:13.10 | JacksLivr | thanks guys, I'll starting loading Fedora. I'm sure I'll be bugging you all again real soon ;-) |
| 23:13.17 | JT | Carp1: what device? |
| 23:13.19 | KnowWhat | but i am always afraid of compiling through sources in fc |
| 23:13.30 | mafkees | I run * on OpenBSD and I'm happy with it |
| 23:14.01 | JT | mafkees: run any hardware? |
| 23:14.12 | mafkees | JT: ehm, cpu and stuff |
| 23:14.16 | mafkees | but no zaptel no |
| 23:14.23 | JT | asterisk related... |
| 23:14.35 | mafkees | I dont need zaptel |
| 23:14.38 | JacksLivr | i have a 4 port fxs fxo card |
| 23:14.39 | JT | sure |
| 23:14.55 | mafkees | JT: why should I need zaptel ? |
| 23:15.08 | JT | if you're only doing voip, you don't |
| 23:15.15 | mafkees | indeed |
| 23:15.20 | JT | i do realise that |
| 23:15.31 | JT | openbsd may have issues if you need to run hardware |
| 23:15.35 | mafkees | there's a reason pstn is called pstn |
| 23:15.41 | JacksLivr | FC run hardware allright? |
| 23:15.48 | mafkees | JacksLivr: yes |
| 23:15.49 | JT | JacksLivr: yes, it's linux |
| 23:15.59 | JacksLivr | cool |
| 23:16.01 | JT | the drivers are all written for linux |
| 23:16.04 | *** join/#asterisk Lgvp (n=Lgvp@200.138.20.49) |
| 23:16.14 | JT | mafkees: eh, what about the pstn? |
| 23:16.22 | mafkees | if I need an analog or isdn landline I use an FXO |
| 23:16.30 | mafkees | ATA |
| 23:16.43 | JT | that's one option |
| 23:17.19 | mafkees | I have a couple of installs on linux because customers wanted their pstn connected directly to the box |
| 23:17.40 | mafkees | but my preferred setup is asterisk under systrace on openbsd with ATA |
| 23:18.32 | JacksLivr | allright, last question and if off to build.... with FC, should i compile all the stuff from source or look into yum for the installs? |
| 23:18.53 | mafkees | grab the sources |
| 23:18.54 | [TK]D-Fender | JacksLivr : Use the Source Luke! |
| 23:19.06 | mafkees | know the force, read the source |
| 23:19.07 | luke-jr_ | ... |
| 23:19.47 | JacksLivr | [TK]D-Fender: wow, that was white and nerdy ;-) |
| 23:20.29 | JT | mafkees: locally attached hardware has the potential to be more flexible than ATAs |
| 23:20.49 | JT | mafkees: but most people don't need that flexibility |
| 23:21.00 | Opperior | if you are only receiving calls via IAX and SIP, is there a difference between "s" and "_." extensions as your starting point in your starting context? |
| 23:21.06 | JT | also PRI SIP gateways are super expensive |
| 23:23.06 | JacksLivr | apparently i am not done asking advice ;-) 1.2 or 1.4? |
| 23:23.19 | JT | 1.2 |
| 23:23.57 | JacksLivr | thanks |
| 23:24.34 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 23:26.15 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 23:26.18 | Dovid | morning all |
| 23:27.15 | JT | Carp1: what device is doing the nat and port forwarding? |
| 23:27.21 | mafkees | JT: pri cards are too |
| 23:27.25 | *** join/#asterisk lencho (n=lencho@pool-72-78-116-222.phlapa.fios.verizon.net) |
| 23:27.38 | JT | mafkees: have you actually priced both? |
| 23:27.44 | JT | mafkees: the difference is ridiculous |
| 23:27.46 | mafkees | same price |
| 23:27.56 | JT | yeah maybe if you get the redfone |
| 23:28.04 | JT | but what abut SIP ones |
| 23:28.15 | DocHolliday | hi guys i disabled selinux however libpri wont install because it seems to think its enabled :) |
| 23:28.16 | mafkees | redfone works |
| 23:28.22 | JT | where can you get cheap SIP gateways with hardware ec? |
| 23:28.26 | JT | it has no hardware ec |
| 23:28.51 | mafkees | JT: the pri cards with hardware ec are freaking expensive as well |
| 23:28.52 | vlt|home | "y-y-y-o-o-u-r-r c-c-a-l-l-l-l i-i-s-s n-n-o-w-w-w f-f-i-r-r-r-s-t i-n-n l-l-l-i-n-e-e ..." after I upgraded asterisk to "v1.2.14-BRIstuffed-0.3.0-PRE-1x". Where to look first for the reason? |
| 23:29.13 | mafkees | get the y version |
| 23:29.22 | mafkees | that's your first action |
| 23:29.23 | mafkees | ;) |
| 23:29.49 | vlt|home | mafkees: really? |
| 23:30.03 | JT | vlt|home: maybe bad zaptel timing, but your zttest was giving erroneous results |
| 23:30.11 | JT | vlt|home: probably not necessary to upgrade |
| 23:30.21 | mafkees | I would upgrade anyways |
| 23:30.23 | mafkees | it's easy |
| 23:30.26 | mafkees | download |
| 23:30.28 | mafkees | tar zxvf |
| 23:30.31 | mafkees | ./install.sh |
| 23:30.34 | mafkees | done |
| 23:30.38 | JT | mafkees: umm, what's the cheapest pri sip gateways you've seen? |
| 23:30.44 | JT | mafkees: nope, he must patch qozap |
| 23:30.47 | JT | not easy |
| 23:30.47 | mafkees | 500 euro |
| 23:30.49 | DocHolliday | anyone know a workaround for libpri? |
| 23:30.55 | JT | mafkees: what brand? |
| 23:31.04 | JT | mafkees: what about quad pri with hardware ec? |
| 23:31.07 | mafkees | cant remember |
| 23:31.18 | DocHolliday | its disabled but it think its enabled :) |
| 23:31.34 | mafkees | never used the hardware ec |
| 23:31.43 | mafkees | mg2 is doing the job for me |
| 23:31.48 | JT | mafkees: the quintums run at USD$16k or something outrageous |
| 23:31.49 | vlt|home | JT: I disabled zap for now. I'm connected via chan_sip now. Btw: moh (native and gsm files) plays fine ... |
| 23:32.04 | edgecase | why would chan_bluetooth fail to link to str2ba() in libbluetooth.so ? |
| 23:32.14 | mafkees | edgecase: wrong versions |
| 23:32.19 | JT | chan_bluetooth is unmaintained i believe |
| 23:32.29 | edgecase | yeah i'm fixing it up for * 1.4.0 |
| 23:32.36 | JT | umm |
| 23:32.40 | JT | chan_cellphone |
| 23:32.43 | JT | is the new one |
| 23:32.47 | edgecase | i wonder if str2ba() moved to a different lib |
| 23:32.52 | JT | by all reports a lot better |
| 23:32.59 | edgecase | i thought it only did AG, not headsets? |
| 23:33.07 | mafkees | http://bugs.digium.com/view.php?id=8919 |
| 23:33.10 | JT | headsets is in the pipeline |
| 23:33.50 | mafkees | there's the new chan_cellphone |
| 23:34.02 | JT | mafkees: is mg2 the default? |
| 23:34.05 | Dovid | whats new about it ? |
| 23:34.21 | JT | mafkees: anyway, the pri cards are waaay cheaper than anything like a quintum |
| 23:34.31 | JT | Dovid: the fact it's a totally new channel driver |
| 23:34.57 | Dovid | JT: not from the coding aspect but what does it to better that the old one didnt do ? |
| 23:35.01 | mafkees | JT: in 1.4 it's the default yeah |
| 23:35.13 | mafkees | and yeah, the quintum is expensive |
| 23:35.18 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 23:35.22 | Dovid | JT: does it work for 1.2 ? |
| 23:35.29 | mafkees | I have to admit that for most pri setups we use the sangoma cards |
| 23:35.31 | JT | Dovid: i think so |
| 23:35.51 | Dovid | :) |
| 23:35.54 | mafkees | but we have only a couple of those |
| 23:36.02 | mafkees | most setups have IAX2 setups |
| 23:36.11 | JT | mafkees: you can buy 2 pri cards and computers and a layer 1 failover box for the price of a quintum |
| 23:36.34 | lencho | newbie question. I want an Asterisk server that I would like to connect to the PSTN line. I need some calls that Asterisk receives through IP to be routed to PSTN line. Asterisk will be connected to internal line. To access exnernal line I need dial "9". |
| 23:37.36 | DocHolliday | can someone tell me based on my libpri output if libpri successfully compiled? |
| 23:37.55 | mafkees | DocHolliday: pastebin it |
| 23:37.56 | JerJer | DocHolliday: does it say Error ? |
| 23:38.07 | DocHolliday | the install script isnt giving me much to work with :P |
| 23:38.53 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:39.05 | DocHolliday | anyone willing to look at my pastebin *please* |
| 23:39.44 | mafkees | link ? |
| 23:40.03 | mafkees | damn, I should smoke less |
| 23:40.09 | DocHolliday | http://www.pastebin.ca/372312 |
| 23:40.16 | DocHolliday | mafkees, ^^ Thansk! |
| 23:40.26 | DocHolliday | just pasted, *thanks |
| 23:40.44 | JerJer | that is a successful install |
| 23:40.48 | [TK]D-Fender | DocHolliday : Well I don't SEE any errors. How about getting off your ass and trying to compile Zaptel? ;) |
| 23:40.51 | mafkees | that is a nice install |
| 23:41.11 | DocHolliday | mafkees, heh the link is above.. lots will kill you before smoking |
| 23:41.41 | Dovid | smokin is great... it keeps the world movin ;) |
| 23:41.48 | DocHolliday | [TK]D-Fender, LOL i just did :) |
| 23:41.51 | mafkees | well, smoking cigarette 32 of today..... |
| 23:42.10 | Dovid | hehe. for me it depends on the stress level |
| 23:42.34 | edgecase | JT, i only saw a brief mention of headset support in that forum. so here i fixed all the warnings and errors, just 1 linker problem remains for chan_bluetooth |
| 23:42.41 | mafkees | Dovid: that's the freaky part. not stessed at all today |
| 23:42.53 | Dovid | depends on the mood i guess. |
| 23:43.04 | mafkees | I need a fax |
| 23:43.05 | mafkees | lol |
| 23:43.08 | Dovid | my ex C**** is gettin married tonight so i am up in smoke |
| 23:43.16 | JT | edgecase: the developer for chan_cellphone hangs around in here and -dev |
| 23:43.31 | JT | edgecase: Qwell is working on headset support too |
| 23:43.44 | mafkees | Qwell is working on a lot of stuff |
| 23:43.47 | mafkees | (I hope) |
| 23:43.48 | JT | mafkees: a sip fax? |
| 23:43.59 | mafkees | JT: any fax will do |
| 23:44.12 | JT | sip fax will help increase stress |
| 23:44.13 | Dovid | !t38 |
| 23:44.17 | Dovid | ~t38 |
| 23:44.27 | jbot | hmm... t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 23:44.27 | Dovid | lol |
| 23:44.27 | mafkees | I need to get my disclaimer to digium |
| 23:44.34 | mafkees | but since I dont have a landline |
| 23:44.41 | mafkees | and no mail->fax services |
| 23:44.50 | Dovid | makfees: i think u can email one it |
| 23:44.51 | Dovid | in* |
| 23:44.55 | JT | man having no landline would suck |
| 23:44.56 | mafkees | no |
| 23:45.05 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 23:45.16 | mafkees | Dovid: I asked mog, he told me: 'fax or plain-old-snail-mail' |
| 23:45.25 | DocHolliday | i seem to have a lot of errors when doing a make linux with mpg1213 |
| 23:45.53 | mafkees | DocHolliday: that's because mpg12313 is not the program you need ;) |
| 23:46.00 | mafkees | sorry, couldn't resist |
| 23:46.05 | DocHolliday | mafkees, 123 :P |
| 23:46.17 | mafkees | I said sorry |
| 23:46.18 | mafkees | ;) |
| 23:46.19 | DocHolliday | mafkees, dont blame ya |
| 23:46.29 | Dovid | Doc: use native |
| 23:46.38 | Dovid | dont use mpg123 anymore |
| 23:46.42 | mafkees | me neither |
| 23:46.45 | Dovid | lol |
| 23:46.46 | mafkees | I use native as well |
| 23:46.53 | Dovid | btw any admins here for the bot ? |
| 23:47.01 | Dovid | ~t38 is no longer an article |
| 23:47.03 | jbot | ...but t38 is already something else... |
| 23:47.03 | DocHolliday | is that a separate program or built-in to asterisk? |
| 23:47.16 | Dovid | it just goes here |
| 23:47.17 | Dovid | http://www.cantata.com/ |
| 23:47.35 | Dovid | some one realized that thier link is used a lot and they prob. took it down for a link to make $$$$ |
| 23:47.57 | JT | Dovid: i think anyone can modify it |
| 23:48.05 | DocHolliday | mafkees, is 'native' built in or a separate app? |
| 23:48.41 | mafkees | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 23:49.00 | Dovid | JT: how is that done ? |
| 23:49.12 | Dovid | DocHoliday: built in to the asterisk add-ons |
| 23:49.22 | JT | Dovid: no jbot, t38 is blah blah blah |
| 23:49.34 | JT | or maybe it's jbot no |
| 23:49.35 | JT | anyway |
| 23:49.47 | Dovid | u just dont install mpg123 and u compile asterisk and then asterisk add ons |
| 23:49.49 | Dovid | kk |
| 23:49.51 | Dovid | thnaks |
| 23:50.15 | DocHolliday | mafkees, thanks for all the help thus far.. ;) |
| 23:50.25 | mafkees | DocHolliday: no problem :) |
| 23:50.58 | vlt|home | mafkees: upgrade to 1y solved the p-p-p-r-r-r-o-b-b-l-l-l-e-m-m-m, thanks. |
| 23:51.17 | DocHolliday | the install is really quick on a dual 2.4 :P |
| 23:51.18 | mafkees | vlt|home :) it solved that for me as well, that's why I told you |
| 23:51.31 | JT | vlt|home: is that voip or qozap problem? |
| 23:51.40 | mafkees | JT: bristuff problem |
| 23:51.50 | JT | for voip or qozap |
| 23:51.54 | vlt|home | JT: I disabled (qo)zap completely for today ... |
| 23:51.54 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:51.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:51.55 | mafkees | voip |
| 23:51.56 | JT | is it only with x? |
| 23:52.06 | mafkees | yeah, only X |
| 23:52.10 | mafkees | as far as I tested |
| 23:52.11 | vlt|home | x? |
| 23:52.15 | DocHolliday | what software do people typically use for TFTP on linux? |
| 23:52.16 | mafkees | but I did not test all of them |
| 23:52.17 | vlt|home | nm |
| 23:52.27 | kuku5 | tftpd |
| 23:52.30 | Dovid | i only use ftp |
| 23:52.31 | Dovid | ;) |
| 23:52.31 | mafkees | DocHolliday: I use tftpd package form debian |
| 23:52.32 | vlt|home | DocHolliday: tftp |
| 23:52.39 | kuku5 | tftp |
| 23:53.00 | mafkees | but that's only when there's no bsd box present |
| 23:53.26 | *** join/#asterisk Phel (n=chatzill@adsl-2-226-222.mia.bellsouth.net) |
| 23:53.41 | DocHolliday | used to loading all the firmware off a windows box.. times change i guess |
| 23:54.07 | Dovid | hehe |
| 23:54.14 | Dovid | Doc: goto learn the linux |
| 23:54.20 | Dovid | 3 years ago i didnt know anything |
| 23:54.23 | Phel | [TK]D-Fender: FYI, specifying the router's real IP addr seems to help (I get further) but still no registration |
| 23:54.25 | DocHolliday | heh |
| 23:54.29 | Dovid | now i am almost married to linux ;) |
| 23:54.39 | Dovid | doc: what kinda phones r u using ? |
| 23:54.40 | mafkees | Dovid: been there as well |
| 23:54.42 | JT | they're sharing an apartment |
| 23:54.44 | Phel | What will you be doing on your honeymoon |
| 23:54.53 | mafkees | but I broke up with linux and moved in with OpenBSD |
| 23:54.53 | Phel | Eww! |
| 23:55.04 | DocHolliday | do you guys find asterisk-addons for asterisk 1.4 worthwhile? |
| 23:55.10 | mafkees | DocHolliday: yeah |
| 23:55.21 | Dovid | nah. centos is my gal |
| 23:55.22 | mafkees | format_mp3 and cdr_mysql |
| 23:55.27 | Dovid | i am still on 1.2 |
| 23:55.30 | DocHolliday | its taking longer to compile then asterisk |
| 23:55.37 | mafkees | all my production boxes are 1.2 as well |
| 23:55.44 | mafkees | I wont run 1.4 in production right now |
| 23:55.50 | Dovid | me neither |
| 23:56.07 | DocHolliday | mafkees, this box is going production (1.4) what do you do if you need T.38? |
| 23:56.13 | Dovid | bug tracker has been busy - i would play with it but i am not a coder and i wouldnt know whats wrong |
| 23:56.27 | mafkees | DocHolliday: I never played with t38 |
| 23:56.33 | mafkees | as far as I know it's not ready |
| 23:56.43 | *** join/#asterisk coppice (n=chatzill@106.206.17.210.dyn.pacific.net.hk) |
| 23:56.45 | mafkees | there is passthru support, but that's it |
| 23:57.02 | Phel | Is there a good forum for discussing general SIP clients? |
| 23:57.16 | mafkees | asterisk-users@ |
| 23:57.20 | Dovid | also i have not seen many providers that support T.38 well that i would use it |
| 23:57.29 | DocHolliday | mafkees, am i going to regret using 1.4 for production? (small setup) |
| 23:57.32 | DocHolliday | all i need is passthru :) |
| 23:57.46 | Dovid | Phel: try wiki but more info on phones |
| 23:57.52 | Dovid | doc: i would still stay wit 1.2 |
| 23:57.58 | mafkees | DocHolliday: I run svn trunk here at home and it's working fine, but I wont use it in customer production setups |
| 23:58.08 | *** join/#asterisk flying_Luck (n=melifaro@ppp85-141-153-47.pppoe.mtu-net.ru) |
| 23:58.24 | Phel | I basically need a place to ask ppl to help me get my SIP softfone to work |
| 23:58.37 | mafkees | Phel: did you google ? |
| 23:58.40 | Phel | Yes |
| 23:58.47 | mafkees | still no answer ? |
| 23:58.50 | DocHolliday | mafkees, mind if I pm you? |
| 23:58.52 | Dovid | Phel: what kinda phone ? |
| 23:58.59 | mafkees | DocHolliday: go ahead :) |
| 23:59.08 | vlt|home | mafkees: Didn't patch qozap this time (while installing 1y). Now I loaded qozap module, ran ztconf (with junghanns's zaptelconf) and got no CRC errors by now. I think I'm a little too anxious to load chan_zap now ;-) What doeas happen to other cards than genuine QuadBRI with the driver? |
| 23:59.09 | Phel | I mean, I'm set up properly according to 3 different places instructions |
| 23:59.11 | Phel | Ekiga |
| 23:59.29 | Dovid | lol |
| 23:59.36 | Dovid | so it can be an issue that u r lookin over |
| 23:59.42 | mafkees | vlt|home: what other cards do you have ? |
| 23:59.59 | Phel | Dovid: lol at me? y? |
