| 00:00.18 | *** join/#asterisk mofu (~eric@pcp04232064pcs.plyntv01.mi.comcast.net) |
| 00:00.25 | Marlow | jhb : you just let your asterisk box register with sipgate .. |
| 00:00.38 | paulc | Shaq: Hmm.. is NAT involved? how's your network connection set up (PC/asterisk/IP etc?) |
| 00:00.46 | *** join/#asterisk shadebob (~shadebob@ll81-134-144-192-81.ll81.iam.net.ma) |
| 00:00.48 | shadebob | hi |
| 00:00.55 | Marlow | jhb : and send the call into the conference .. that's actually doable with those examples .. slightly modified .. |
| 00:01.07 | jhb | marlow: register =>user:pass@sipgate.de/mylocalextension ? |
| 00:01.16 | *** part/#asterisk eKo1 (~abc@63.245.57.70) |
| 00:01.33 | Shaq | paulc: Internet --> Router (forwarded ports) --> Asterisk |
| 00:01.42 | shadebob | I have a probleme when i try to "modprobe wcfxo" for my digium x100p. I have an error : "FATAL: Error inserting wcfxo (/lib/modules/2.6.7-1-386/misc/wcfxo.ko): Invalid module format" |
| 00:01.54 | Shaq | paulc: The PC is on the same level as asterisk. |
| 00:02.11 | paulc | so PC <--> Asterisk works ok, echo test, voicemail etc, 2 way audio? |
| 00:02.17 | Marlow | jhb : yes .. something like that .. |
| 00:02.42 | Shaq | paulc: Yes, everything works there. I can call 999 and things works =) |
| 00:03.01 | Shaq | paulc: And I can call to my PC from my mobilephone |
| 00:03.11 | Shaq | So there is a connection. |
| 00:03.39 | jhb | marlow: I am looking forward to that stuff. |
| 00:04.02 | paulc | so it's just PC --> Asterisk --> Outbound that's the problem? |
| 00:04.09 | paulc | how are the ports forwarded on your router? |
| 00:04.36 | Kb1_Kanob | speaking of echos - anyone got any ideas why I seem to get different tail lengths off of different parties on the same serving central office? Some calls are fine, others are handled by the mark2 can ok, others never converge... |
| 00:04.44 | Shaq | paulc: Yes.. and vioce från Phone --> Asterisk --> PC |
| 00:05.16 | Shaq | paulc: 5060 to asterisk udp/tcp, and 10000-10003 udp to asterisk. |
| 00:05.34 | jhb | marlow: brb |
| 00:05.35 | paulc | have you tried opening udp wider? 10000-20000? |
| 00:05.36 | Marlow | jhb : ok .. this will take some time .. the Makefile needs to be tweaked ... |
| 00:05.51 | *** join/#asterisk florz (nobody@p508A6A32.dip.t-dialin.net) |
| 00:06.05 | zeldar | hello, has anyone used Freebsd 4.10 w/ the port version of asterisk? |
| 00:06.24 | Shaq | paulc: No I havent, my Swedish provider says 10000-10003, and i worked without asterisk. Do you think that could couse any problems? |
| 00:06.57 | paulc | hmm.. so if you use a softphone on your PC it works fine, but if you call via asterisk, you get one way audio? inbound or outbound? or both? |
| 00:07.27 | jhb | marlow: back. I got an already tweaked Makefile - I applied the changes mentioned on the mailing list |
| 00:07.54 | jhb | marlow: need it? |
| 00:08.49 | xkev | where is voip-info.org hosted? We'd be happy to donate some if needed |
| 00:09.22 | Shaq | paulc: Softphone works fine without asterisk. I don't have any hardware. I can describe everything. If I call from my mobilphone I reaches the pc, I can talk to the phone from the pc but i can't talk to the pc from the phone. |
| 00:09.23 | xkev | we were one of (if not the only) the electoral-vote mirrors that stayed up |
| 00:09.39 | xkev | ..if that says anything :) |
| 00:09.42 | Shaq | paulc: And I can't dial out from the PC at all, just internal numbers to asterisk. |
| 00:10.07 | paulc | Shaq: Can you call to voicemail in Asterisk from the PC, hear the greeting, leave a message, then retrieve it? |
| 00:10.47 | *** join/#asterisk Shaneful (~shane@d154-20-37-11.bchsia.telus.net) |
| 00:10.55 | Shaq | I haven't configured voicemail, I'm new to this. What's the default number to the voicemail? |
| 00:11.06 | paulc | 8500 |
| 00:11.13 | paulc | setup a mailbox in voicemail.conf |
| 00:11.25 | Shaneful | will the polycom 500 h.232 phone work with asterisk |
| 00:11.37 | paulc | then add an extension to go to voicemail - something like: 8501,1,Voicemail(1234) |
| 00:11.41 | Marlow | jhb : naeh .. i'm integrating it into the package .. |
| 00:11.51 | Marlow | jhb : that's easier done from scratch .. |
| 00:12.02 | jhb | marlow: ok. |
| 00:12.32 | Umaro | No one here knows the magic voodoo code that is the sipura provisioning config format? |
| 00:12.42 | *** join/#asterisk bonbon-home (~happy@81-86-185-223.dsl.pipex.com) |
| 00:13.30 | *** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
| 00:14.00 | *** join/#asterisk sphyrna (~mstocco@adsl-2-155-119.mia.bellsouth.net) |
| 00:14.05 | Marlow | Umaro : if nobody has answered, then nobody knows .. |
| 00:14.17 | Marlow | Umaro : try to ask sipura support .. |
| 00:14.22 | Shaq | paulc: All I get is Call failed: 404 not found |
| 00:14.39 | paulc | you have to reload after changing extensions.conf - do "extensions reload" |
| 00:14.58 | Umaro | Marlow, tried asking them.. they said no :( |
| 00:14.59 | xkev | shaneful I suggest the sip firmware |
| 00:15.19 | xkev | all the phones are the same hardware and will take whichever codebase |
| 00:16.09 | xkev | shaneful, http://www.freedomphones.net/polycom/files/ |
| 00:16.52 | Moc | polycom rock .. |
| 00:17.00 | Moc | I should make a autotext ;) |
| 00:17.17 | ipso | Anyone know how I can merge two wav files in to one, so the second wav file plays after the first in sequence? |
| 00:17.45 | Shaneful | thanks xkev |
| 00:17.46 | Shaneful | brb |
| 00:17.48 | [TK]D-Fender | ipso : is there a reason not to just play them in order seperately? |
| 00:18.13 | [TK]D-Fender | <PROTECTED> |
| 00:18.14 | Shaq | paulc: Hehe.. 990 was my number =) Trying to log in now! |
| 00:18.17 | *** join/#asterisk WilliamK (~wkeller@24.0.130.60) |
| 00:18.32 | sphyrna | ipso: you can 'cat' the files together |
| 00:18.50 | Rez | sphyrna: not wav files. the header will sound funny |
| 00:19.40 | Umaro | ipso: you can merge raw files.. |
| 00:19.46 | ipso | [TK]D-Fender: I want to use the resulting wav with GET DATA, playing them one after another doesn't give me the desired result. And GET DATA doesn't seem to allow you to pass two files to play. |
| 00:19.56 | Umaro | ipso: so convert the wav files to raw first, then cat them, then convert the raw file to wav |
| 00:19.57 | sphyrna | Rez: yeah I guess your right, I was think about raw |
| 00:19.58 | ipso | sphyrna: hrmm, I tried catting them together, didn't work. :( |
| 00:20.19 | ipso | sphyrna: hrmm, sox should convert them to raw I assume? |
| 00:20.32 | sphyrna | ipso: yep |
| 00:20.35 | Kb1_Kanob | Seeking an echo cancellation guru - or a DSP guru. Is there anyone here, or can someone suggest a channel? I'm trying to understand why an overly long Transversal Filter is a problem. |
| 00:21.30 | [TK]D-Fender | ipso : ok, there are tons of ausio mixing programs for whatever OS you you are probably running. In WinxP I'd use the one that came with my Audigy. |
| 00:21.42 | Shaq | paulc: Configuring, seems to work this far. What's the pound key? |
| 00:22.06 | [TK]D-Fender | Most Linux distro's likely come with one or have one packaged and easily installable |
| 00:22.24 | paulc | Shaq: Pound key = # |
| 00:22.32 | Shaq | paulc: Ok =) Thx |
| 00:22.34 | [TK]D-Fender | SOX for Linux is always a viable option. |
| 00:23.15 | Shaq | paulc: I have now recorded my name succesfully |
| 00:23.31 | paulc | so audio works ok both ways * <--> PC ? |
| 00:23.41 | Shaq | paulc: Yes |
| 00:24.04 | ipso | [TK]D-Fender: yeah, I'm trying to use sox... |
| 00:25.06 | ipso | Does sox tell what the sampling rate of a .wav file is by chance? It seems to require that when working with raw files. |
| 00:26.03 | sphyrna | ipso: are you using the -v switch with sox? |
| 00:26.20 | Shaq | It feels like the firewalls fault but I can't understand why |
| 00:26.44 | *** join/#asterisk Ehsan (ehsan@216.236.116.251) |
| 00:26.51 | ipso | sphyrna: -v = volume |
| 00:26.54 | Shaq | err Router |
| 00:27.33 | sphyrna | ipso: -V (verbose) |
| 00:28.08 | ipso | sphyrna: Doesn't seem to change the fact that when using sox and raw data, you need to specify things like sample rate and data size. |
| 00:28.43 | *** join/#asterisk BurnedOutGeek (~BurnedOut@res-66-191-172-161.spa.sc.charter.com) |
| 00:29.05 | ChulJin | Anyone have any examples using the M() [macro] option with Dial? |
| 00:29.57 | ChulJin | I mean...I understand how to use it...I just wonder why to use it...a practical application |
| 00:30.16 | Shaq | paulc: Should I set any defaultip for the * -> PC connection? |
| 00:30.17 | *** join/#asterisk file (~joshnet@mctn1-142166195059.nb.aliant.net) |
| 00:31.19 | sphyrna | ipso: sox -V your_wav_file.wav -r 8000 -c 1 -w your_raw_file.raw ??? |
| 00:32.03 | ipso | sphyrna: I get really strange static.. |
| 00:32.25 | ipso | Is there a way to play a raw file by chance? |
| 00:32.31 | paulc | shaq: how do you mean? set it where? |
| 00:32.34 | ipso | so I can confirm my merge raw file is actually "correct" |
| 00:32.57 | Shaq | paulc: In sip.conf, but nevermind, should matter |
| 00:33.11 | paulc | you using NAT? |
| 00:33.39 | elproducto | wiki is back |
| 00:33.42 | elproducto | wiki is back |
| 00:33.43 | elproducto | wiki |
| 00:33.45 | elproducto | wiki |
| 00:33.51 | Rez | elproducto: thanks, I think we get it |
| 00:33.54 | Shaq | paulc: Honestly I don't know, I have a D-Link router. Nothing else.. |
| 00:35.29 | Shaq | paulc: Don't think so no. |
| 00:35.29 | sphyrna | ipso: I would import it into audacity |
| 00:35.29 | Shaq | paulc: When I dial out i get this: |
| 00:35.29 | Shaq | Nov 5 01:28:19 NOTICE[15174]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum |
| 00:35.32 | Shaq | Nov 5 01:28:19 WARNING[15174]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'outgoing' |
| 00:35.35 | paulc | Shaq: Ok.. so PC <--> Asterisk works fine, full duplex audio.. it's only PC --> Asterisk --> external that there's a problem? |
| 00:35.45 | *** join/#asterisk drray (~drray@dsl081-162-121.sea1.dsl.speakeasy.net) |
| 00:35.46 | Shaq | paulc: Yes |
| 00:36.11 | paulc | Shaq: What are you dialling on the PC, and where should it go Asterisk/VoIP? Put your extensions.conf in pastebin.ca maybe? |
| 00:37.20 | Shaq | paulc: http://213.64.123.216/extensions.conf |
| 00:37.20 | ipso | sphyrna: Okay, I'll give that a shot... I must be getting the switches wrong with sox, cause the output is pretty nasty |
| 00:37.43 | Shaq | paulc: What i'm calling? I'm usin X-Lite to call to my mobilephone. |
| 00:37.45 | paulc | Shaq: I'll go look.. what do you dial on the PC |
| 00:38.23 | Shaq | paulc: I dial my phonenumber.. I don't understand the question. |
| 00:38.47 | paulc | what are the exact digits you dial on the PC, and what are the exact digits you should be sending to your VoIP provider? |
| 00:39.38 | Shaq | paulc: I dial 070 788 49 48, and that's what I should send to my provider, maybe 707884948 would be enough. |
| 00:40.07 | paulc | You're not sending the 0 to your provider though - you have ${EXTEN:1} that strips the 0 |
| 00:40.12 | Shaneful | xkev: how hard is it to flash these phones? |
| 00:40.14 | paulc | does the console show the whole call going out? |
| 00:40.51 | paulc | Shaq: Make a test extension.. 8001,1,Dial(SIP/digithing/070xxxx) then another 8002,1,Dial(SIP/dig../70xxx) |
| 00:40.56 | Shaq | paulc: Yeah, but it worket without asterisk, without the 0 i mean, so I don't think that's the problem. |
| 00:41.00 | paulc | then call 8001 and 8002 and see how it works |
| 00:41.07 | Shaq | paulc: Ok |
| 00:41.10 | *** join/#asterisk PlainWhiteTrash (~matt@sbi-24-177-189-7.mtv.al.charter.com) |
| 00:41.17 | paulc | Do you see on the console the whole digit string going out? |
| 00:41.37 | paulc | there's overlapping matches in your dial plan, _0. versus 001, 002 etc.. that's ok, but it complicates matters maybe.. |
| 00:41.46 | Shaq | paulc: Yes, hold on lets talk in private |
| 00:41.54 | brc_ | ~seen critch |
| 00:41.55 | jbot | critch <~critch@pcp08694367pcs.500ash01.tn.comcast.net> was last seen on IRC in channel #asterisk, 92d 8h 10m 44s ago, saying: 'I'm getting an exception 20 then alarm(4)'. |
| 00:42.10 | *** join/#asterisk RH_athome (~xdslguy@user-12l2p2p.cable.mindspring.com) |
| 00:42.11 | Kb1_Kanob | VBR Speex vs. g792a, which has superior audio quality for the least bandwidth? |
| 00:43.56 | brc_ | depends on what speex settings you use |
| 00:44.08 | ipso | sphyrna: Yeah, audacity makes life much easier, I didn't even realize this existed. Thanks. ;) |
| 00:44.09 | PlainWhiteTrash | Anyone know if a potential problem with high volume sip usage in outbound queue calls in which rtp resources get allocated & not released is know / fixed since 1.0.1? |
| 00:44.27 | *** join/#asterisk jbaach (~jbaach@62.206.114.179) |
| 00:44.27 | sphyrna | ipso: no prob :) |
| 00:44.33 | *** join/#asterisk shoofly (~supybot@brc.base.supporter.pdpc) |
| 00:44.37 | Kb1_Kanob | brc_: makes sense I suppose - speex has many knobs. |
| 00:45.09 | jbaach | marlow: I (jhb) am back - I got accidently disconnected. The network cord and I decided to go different ways... :-) |
| 00:45.31 | jbaach | marlow: not on wlan. Freedom! |
| 00:46.25 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
| 00:46.30 | *** part/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 00:48.02 | *** join/#asterisk darkthrn (~darkthorn@producto-valvo.com) |
| 00:48.22 | Marlow | jbaach : let's see what it says .. |
| 00:48.31 | Marlow | jbaach : i'm compiling now .. |
| 00:48.54 | jbaach | marlow: good luck ;-) |
| 00:50.59 | sung | life in the fast lane |
| 00:51.02 | sung | snorting lots of cocaine |
| 00:51.05 | sung | dirnking some beer |
| 00:51.10 | sung | doot. |
| 00:52.02 | darkthrn | sung you crackhead. |
| 00:53.59 | Marlow | jhb : i don't expect it to compile the first time ... usually some tweaking has to be done, when integrating into other sourcecode |
| 00:55.53 | jhb | marlow: yeah, I guess that you expectations are a bit more real life - I would sometimes be happy with stuff just working... |
| 00:56.52 | darkthrn | ~seen kram |
| 00:56.53 | jbot | kram is currently on #asterisk (1d 9h 52m 38s) |
| 00:56.56 | ChulJin | brc: from his list posts, I'm surprised the last thing he said was quite that nice. |
| 00:58.13 | Marlow | jhb: this code won't work against 1.0.2 ... it's back to 1.0, if not before .. |
| 00:58.18 | Marlow | jhb : just cleaning up ... |
| 00:58.49 | jhb | marlow: over here 'just' usually marks the hard part ;-) |
| 00:59.00 | Marlow | jhb : the problem is, that asterisk just is 1.0 .. it's new .. it was cvs only for a very long time |
| 00:59.39 | Marlow | jhb : i'm no developer either .. (at least i use to say that ... would bore me to death to develop all day) :) |
| 00:59.53 | jhb | marlow: from what I have seen so far I am still quite happy with the 'just 1.0' |
| 01:00.52 | jhb | marlow: same with me. Even though I code, I tend to not call myself a programmer. |
| 01:02.55 | jhb | marlow: I like to translate between tech- and customer-world |
| 01:03.53 | Marlow | jhb : i hate that ... it's because it's exactly what i do in my daily work .. |
| 01:03.57 | Marlow | jhb : :( |
| 01:05.14 | jhb | marlow: I spent most of my working time doing php and python stuff. So for me it would be good to get back to having more contact with customers. |
| 01:05.54 | herag | what kind of entry do I need to put in my sip.conf to get the * server to act as an sip device?...like do sip:<hostname of * server> and then that would get routed to a context |
| 01:05.59 | Marlow | jhb: i did that half a year for a german company .. php development .. was hired as a sysadm though .. |
| 01:06.10 | jhb | marlow: but if you don't like customer talk, and you would be bored by developing, what are you aiming at? |
| 01:06.15 | Marlow | jhb : told them to go f*ck themselfes |
| 01:06.25 | jhb | marlow: :-) |
| 01:06.28 | nestAr | well.. my MD3200 shipped |
| 01:06.31 | Marlow | jhb : i don't have any problems with customers .. |
| 01:06.39 | Roger123 | marlow: looks like i just had to recompile to get * working on deb 3.1 |
| 01:06.41 | Marlow | jhb : just currently .. too much routine .. |
| 01:07.03 | Marlow | Roger123 : sometimes that's enough .. yess |
| 01:07.08 | Roger123 | after a reboot and a new kernel |
| 01:07.09 | Marlow | Roger123 : -s |
| 01:07.49 | jhb | marlow: I try to escape the php mess as well. |
| 01:07.52 | Marlow | Roger123 : as long as it works :) |
| 01:08.06 | Marlow | jhb: it get's boring in time .. |
| 01:08.15 | Roger123 | so true. at least i only have to rebuild 1 instead of 12 |
| 01:08.16 | Marlow | jhb: i'm a hardware guy .. |
| 01:08.34 | Marlow | jhb: i like working with customers, but i want to do creative stuff sometimes .. |
| 01:08.47 | Marlow | jhb : right now my job is just "break and fix" |
| 01:09.10 | Marlow | Roger123 : is you do it from packages .. sure .. that's easy .. |
| 01:09.30 | jhb | marlow: I like to translate unclear customer ideas into a working design, but do not really like to implement |
| 01:09.34 | Marlow | Roger123 : one reason why i roll my own packages .. i can do them once, test them, works, deploy |
| 01:10.37 | jhb | marlow: what kind of hardware do you do? |
| 01:10.47 | *** join/#asterisk Legend (~legend@24.244.176.111) |
| 01:11.00 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 01:11.26 | Marlow | jhb : anything .. from embedded systems to san solutions .. |
| 01:11.42 | zeldar | will a p3 700 mhz with 256 meg of ram ben enuf for a 6 - 10 user VoIP system? w/ 3 incoming analog line? |
| 01:11.53 | Marlow | zeldar : should do .. |
| 01:12.04 | jhb | marlow: I take it you are in Ireland right now? |
| 01:12.12 | Marlow | zeldar : depending on the amount of transcoding .. |
| 01:12.18 | zeldar | transcoding? |
| 01:12.23 | twisted[work] | bah |
| 01:12.24 | zeldar | sorry i'm a little new to this phone styf |
| 01:12.26 | zeldar | stuf |
| 01:12.27 | Marlow | jhb: i am indeed .. but i'm not originally from here .. |
| 01:12.28 | twisted[work] | digium's ftp is not responding |
| 01:12.34 | twisted[work] | again. |
| 01:12.42 | Marlow | zeldar : transcoding is translating between two codecs .. |
| 01:12.50 | zeldar | ok |
| 01:12.52 | Marlow | twisted[work] : what do you need ? |
| 01:12.58 | zeldar | what about hard drive space? |
| 01:13.05 | twisted[work] | Marlow: g729 |
| 01:13.13 | zeldar | how much for 1 minute of recorded audio through the pbx? |
| 01:13.13 | twisted[work] | for 2.3 |
| 01:13.23 | Marlow | twisted[work] : use my mirror .. ftp://debian.marlow.dk/mirrors/digium |
| 01:13.31 | twisted[work] | sweet |
| 01:13.32 | twisted[work] | thx |
| 01:14.07 | *** join/#asterisk foster (~NNSCRIPT@217.68.143.7) |
| 01:14.16 | zeldar | marlow: and the Digium TDM40B should beable to take my standard analog lines into the Asterisk PBX... |
| 01:14.50 | Marlow | zeldar : indeed .. if equiped with FXO modules |
| 01:15.05 | zeldar | Marlow: thats a compile setting i take it? |
| 01:15.09 | zeldar | the FXO module? |
| 01:16.21 | zeldar | and how stable was the asterisk-0.7.2 release? |
| 01:16.28 | Marlow | zeldar : no .. FXO modules are for landline |
| 01:16.35 | Marlow | zeldar : FXS modules for phones .. |
| 01:16.44 | zeldar | so its another piece of hardware? |
| 01:16.54 | Marlow | zeldar : the TDM400 can be equiped with either .. depending on how you buy it .. |
| 01:16.56 | zeldar | that sits between the walljack and the PBX box? |
| 01:17.03 | zeldar | Ok that makes sense |
| 01:17.16 | zeldar | so its just something with the configuration of the actual card |
| 01:17.19 | Marlow | zeldar : no .. it's the daughterboards on that card ... i think what you want is a TDM04B .. |
| 01:17.31 | Marlow | zeldar : but i would have to look that up |
| 01:17.35 | zeldar | 04 or 40? |
| 01:17.51 | Marlow | zeldar : i think 04 is 4xFXO, 40 is 4xFXS |
| 01:18.03 | zeldar | ok |
| 01:18.03 | Marlow | zeldar : check it on the website .. |
| 01:18.15 | *** join/#asterisk DrRighteous (SystemLoad@ool-435710b6.dyn.optonline.net) |
| 01:18.21 | Marlow | zeldar : if you want to connect 4 landlines, then you need 4 FXO boards |
| 01:18.23 | zeldar | yeah the 40B is a FXS bundle |
| 01:18.38 | Marlow | zeldar : and you would need a FXO bundle for landlines |
| 01:19.30 | zeldar | ok, also any feed back on how the Cisco 7910 phones work? |
| 01:20.05 | zeldar | http://www.digium.com/index.php?menu=wildcard_x100p ?? that the correct card? |
| 01:20.28 | Marlow | zeldar : that's a single line card .. |
| 01:20.36 | Marlow | zeldar : you don't want that .. |
| 01:20.46 | Marlow | zeldar : TDM04B .. look for that .. |
| 01:20.48 | zeldar | yeah thats what i was thinking |
| 01:20.54 | zeldar | i am >.> |
| 01:21.06 | Marlow | zeldar : in the shop .. should be there .. |
| 01:21.13 | zeldar | how about as far of that freebsd ported version of asterisk? |
| 01:21.25 | zeldar | [justin@dev: /usr/ports]> make search name=asterisk |
| 01:21.25 | zeldar | Port: asterisk-0.7.2 |
| 01:21.26 | croaker | http://www.digium.com/index.php?menu=wildcard_tdm400p2 |
| 01:22.01 | *** join/#asterisk ToyMan (~konversat@user-12lcqq2.cable.mindspring.com) |
| 01:22.02 | croaker | TDM04B - 4-port FXO |
| 01:22.13 | zeldar | thanks much croaker |
| 01:22.22 | croaker | np |
| 01:22.27 | croaker | wish I had one of them |
| 01:23.11 | croaker | I have an x100p and a T400P |
| 01:23.27 | Marlow | same here .. but don't use the x100p ... |
| 01:23.51 | zeldar | well i'm trying to upgrade our office's phone system |
| 01:23.54 | Marlow | i used to use zaphfc instead .. now i don't even have a phoneline anymore |
| 01:23.56 | zeldar | cuz the stuff we have now is garbage |
| 01:24.12 | croaker | well I haven't touched the x100p in about a year but an currently hacking on the Tor2 |
| 01:24.58 | croaker | should know if MGMT approves of it soon and a can buy a few more servers and quad T1 cards of the newer variety |
| 01:25.23 | zeldar | is the version 0.7.2 of asterisk pretty stable? and what type of functionality would be missing from the 1.0 described on the site? |
| 01:25.41 | croaker | I do plan to put a writeup of what I learn about CAS e&m wink start signalling for the wiki |
| 01:26.04 | Marlow | croaker : i have a TE410p .. sweet thing :) |
| 01:26.12 | croaker | zeldar, I'd stick with the 1.0 |
| 01:26.29 | zeldar | hmmmm hate using non-ported software |
| 01:26.32 | Marlow | zeldar : i would definatly go for 1.0 |
| 01:26.40 | Marlow | zeldar : ported to what ? |
| 01:26.42 | zeldar | i'm not very linus savy...know freebsd a lot better |
| 01:26.51 | Marlow | ah .. |
| 01:26.53 | zeldar | linux |
| 01:27.08 | croaker | Marlow, what's the difference vs T400P, newer and different power/slot configs but anything else? |
| 01:27.15 | zeldar | i started with linux and moved to freebsd about 4 years ago and never looked back :) |
| 01:27.29 | zeldar | is there a changelog somewhere that will outline the changes? |
| 01:28.07 | croaker | basically 0.7.2 and 1.0 differ by only a few bug fixes from what I know but 1.0 is a "stable release" |
| 01:28.18 | croaker | zeldar, the wiki should have it |
| 01:28.46 | zeldar | ok, i'll look on wiki, i'll prolly email the port maintainer and see if there are any plans for updating the port tree to the newest version |
| 01:29.07 | zeldar | i see there is a basd list also...prolly jump on that when i get closer to putting all this together |
| 01:29.26 | zeldar | haven't done all this PBX stuff to much yet |
| 01:29.33 | Marlow | croaker : for one .. it does both do T and E interfaces .. |
| 01:29.51 | Marlow | croaker : and a lot of the stuff has been moved from the driver and into hardware .. |
| 01:30.07 | zeldar | just watched as one of our other companies upgraded to a NEC NEAX PBX about a year ago...but didn't really move twoard the VoIP side, this office will be strictly VoIP internally |
| 01:30.59 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166195059.nb.aliant.net) |
| 01:31.17 | Marlow | uhhh ... |
| 01:31.28 | Marlow | jhb : this looks good .. |
| 01:31.51 | jhb | marlow: yes? |
| 01:32.38 | Marlow | jhb : where the heck was the config file for app_conference ... |
| 01:32.55 | Marlow | ahh ... there is none :) |
| 01:34.23 | Marlow | jhb : it builds .. i'm just doing a clean package .. |
| 01:34.38 | Marlow | jhb : and install that ... see if it accepts app_conference .. |
| 01:36.44 | Marlow | +help |
| 01:37.28 | jhb | marlow: you are rebuilding your whole asterisk package? Or updating the app_conference only? |
| 01:38.04 | *** join/#asterisk docelm0 (~me@116-39.202-68.tampabay.rr.com) |
| 01:38.36 | Marlow | jhb: the whole shebang including app_conference :) |
| 01:39.08 | Marlow | jhb : i didn't just compile it, i added it into the package .. |
| 01:39.27 | *** join/#asterisk jhb (~jbaach@62.206.114.179) |
| 01:40.18 | Wi_Fi | yo |
| 01:40.26 | jhb | marlow: was kicked out again (too much fiddling around). Could not read the last couple of minutes. Now I behave :-) |
| 01:40.27 | Wi_Fi | anyone good with Apache? |
| 01:40.40 | paulc | depends what you mean by "good" ;-) |
| 01:40.47 | [TK]D-Fender | Wi_Fi : whats your goal? |
| 01:40.58 | Marlow | jhb : i said: the whole shebang including app_conference :) |
| 01:41.00 | Wi_Fi | i need to get perl to work |
| 01:41.06 | Wi_Fi | settin up AM |
| 01:41.17 | Marlow | jhb: it's on it's way up on the repository .. |
| 01:41.19 | docelm0 | anyone here good with SIP? Im having a error and I for the life of me cant figure it out |
| 01:41.38 | jhb | marlow: you are going to make 5 people happy ... |
| 01:41.48 | *** join/#asterisk sudoer2 (~toy@pool-151-203-91-218.bos.east.verizon.net) |
| 01:42.03 | Marlow | jhb : if it works .. it does compile cleanly, and load .. |
| 01:42.13 | sudoer2 | hello, does anyone here have experience with click to dial on asterisk? |
| 01:42.20 | Marlow | jhb : we need to test the conference part first, though |
| 01:42.38 | docelm0 | Im getting a 403 Forbidden error.. I have checked and double checked all my config's but I still get the error. The guy on the other end says he is fine so I dunno.. any ideas? |
| 01:42.49 | jhb | marlow: I volunteer ;-) |
| 01:42.57 | Marlow | jhb : thought so :) |
| 01:43.10 | Wi_Fi | paulc |
| 01:43.26 | sudoer2 | i am using some php to do click to dial with this: Channel:$channel\nMaxRetries:1\nRetryTime:60\nWaitTime:30\nContext:voipjet\nExtension:$num\nPriority:1 and it works fine, but the problem is after 60 seconds, while i am already talking to a person, * will ring my phone again and if i switch lines, it will call out to the outgoing party again |
| 01:43.31 | sudoer2 | does that make any sense? |
| 01:43.33 | paulc | Wi_Fi: so mod_perl? what's AM? |
| 01:43.53 | Wi_Fi | asterisk manager |
| 01:43.58 | paulc | sudoer2: did they double click? like 2 files created? |
| 01:44.02 | Wi_Fi | perl based |
| 01:44.10 | paulc | Wi_Fi: Don't know about that, sorry |
| 01:44.11 | jhb | marlow: is it going to be head or stable? |
| 01:44.17 | Wi_Fi | crap |
| 01:44.18 | Wi_Fi | k |
| 01:44.21 | sudoer2 | no, only 1 file, and i didnt double clicked, i tested it several times |
| 01:44.52 | Marlow | jhb : if you pull an upgrade from the stable repository .. it has app_conference now .. .. need to run apt-get update first |
| 01:45.08 | sudoer2 | and i did ls on the outgoing folder, the file would not diseapear |
| 01:45.14 | sudoer2 | until i hung up the call |
| 01:45.35 | paulc | sudoer2: hmm.. weird.. it's as though it doesn't see the outbound call successfully answered |
| 01:45.40 | docelm0 | Anyone? |
| 01:46.23 | jhb | marlow: just a sec |
| 01:46.31 | sudoer2 | testing now |
| 01:46.36 | sudoer2 | im on the phone |
| 01:46.46 | sudoer2 | waiting for 60 seconds for 2nd call to comein |
| 01:46.47 | sudoer2 | here it is |
| 01:47.59 | sudoer2 | im not sure whats wrong either :( but i cant leave it like this |
| 01:48.00 | docelm0 | Anyone familiar w/ a 403 forbidden error w/ asterisk? |
| 01:49.28 | jhb | marlow: apt-get upgrade is running... |
| 01:49.29 | *** join/#asterisk _RaYmAn_ (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 01:49.40 | Marlow | jhb : can you call a uk number or iax ? |
| 01:51.01 | jhb | marlow: I try to get my grandstream thingi working - than i can use sipgate |
| 01:54.15 | Marlow | jhb : it's disconnecting me after some time .. |
| 01:54.31 | Marlow | jhb : i might need to try to see, if debugging helps . |
| 01:54.56 | sudoer2 | anyone? |
| 01:56.14 | Marlow | jhb : give me a note, once your're ready .. i'm trying to recompile with debugging .. |
| 01:56.38 | jhb | marlow: having some problems with the connection here (www.c-base.de). It will not take long, hopefully |
| 02:00.08 | docelm0 | Does anyone know anything about SIP 403 errors? Anyone w/ a clue? Come on |
| 02:01.59 | *** part/#asterisk Kokey (~jramirez@201.129.164.192) |
| 02:02.57 | *** part/#asterisk DrRighteous (SystemLoad@ool-435710b6.dyn.optonline.net) |
| 02:05.56 | Marlow | wsuff i allways have a backup plan :) |
| 02:16.32 | file[laptop] | it's quiet... too quiet... |
| 02:16.42 | *** join/#asterisk brettnem (~brettnem@user-0ccsr4b.cable.mindspring.com) |
| 02:18.56 | Marlow | file[laptop] : easy :) |
| 02:19.01 | redder86 | boo! |
| 02:19.37 | brettnem | eek |
| 02:20.08 | brettnem | so quiet in here tonight |
| 02:21.04 | file[laptop] | echo.... echo.... |
| 02:21.13 | brettnem | fantastic |
| 02:21.21 | redder86 | ....ohce ....ohec |
| 02:21.43 | brettnem | hmmm |
| 02:23.22 | redder86 | slePP: around? |
| 02:23.29 | redder86 | fearnor-: around? |
| 02:23.37 | redder86 | coppice: around? |
| 02:23.47 | Marlow | redder86 : you've got no luck tonight :) |
| 02:23.54 | redder86 | Marlow: around? |
| 02:23.57 | brettnem | yeah, you're just stuck with us |
| 02:23.57 | redder86 | :-) |
| 02:24.18 | redder86 | oh no. I'm trapped. |
| 02:24.41 | Marlow | redder86 : ok .. i am :) |
| 02:24.51 | brettnem | we are all trapped |
| 02:25.25 | Marlow | stuck in space :) |
| 02:25.47 | redder86 | wish I could get stuck in time |
| 02:25.55 | redder86 | or at least slow it down |
| 02:26.07 | Marlow | not really ... you wouldn't learn anything, while being stuck .. |
| 02:26.23 | redder86 | Marlow: sometimes I don't need to learn anything. I just need to do lots of stuff. |
| 02:26.25 | Marlow | redder86 : groundhot day ? |
| 02:26.33 | Marlow | s/hot/hog/ |
| 02:27.02 | Marlow | with bill murray :) |
| 02:27.37 | Marlow | where he wakes up on the same day .. over and over and over again .. |
| 02:28.05 | Marlow | and starts "exploiting" what he can get away with, since it wont have happened anyway the next morning .. |
| 02:29.29 | Marlow | file got deleted :) |
| 02:29.49 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166195059.nb.aliant.net) |
| 02:32.02 | Marlow | file got undeleted :) |
| 02:32.45 | file[laptop] | indeed |
| 02:34.11 | pfn | play |
| 02:34.12 | pfn | oops |
| 02:34.43 | Marlow | pfn : you are right ... |
| 02:34.51 | Marlow | pfn : play (as in play movie) .. |
| 02:37.03 | *** join/#asterisk balaji (~balajigop@dsl254-064-094.nyc1.dsl.speakeasy.net) |
| 02:37.38 | balaji | I recently upgraded to mac osx server and am having problems with Asterisk |
| 02:37.47 | balaji | does any one have experience with osx ? |
| 02:38.31 | JamesDotCom | i have experience |
| 02:38.35 | JamesDotCom | may not be of much help though |
| 02:38.38 | JamesDotCom | give it a shot :D |
| 02:42.39 | BurnedOutGeek | When I dial an entention over an IAX connection, why is my callerid number showing as "Asteirsk" |
| 02:46.30 | Wi_Fi | guys |
| 02:46.37 | Wi_Fi | this is flooding my console |
| 02:46.42 | Wi_Fi | Nov 4 18:32:46 WARNING[1158883520]: chan_iax2.c:5971 socket_read: Received mini frame before first full voice frame |
| 02:53.56 | tzanger | Wi_Fi: who to? |
| 02:55.01 | *** join/#asterisk GizmoTech (~nb@s205-206-60-208.ab.hsia.telus.net) |
| 02:55.40 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:55.49 | *** join/#asterisk kcah (ircap751@71.Red-80-32-32.pooles.rima-tde.net) |
| 02:55.54 | kcah | hy |
| 02:56.39 | kcah | excuse my question... can i use a creative voice modem to mak a simple interactive answer machine with asterisk? |
| 02:56.59 | tzanger | kcah: short answer: no |
| 02:56.59 | *** join/#asterisk jdg (~jdg@CA03F81A.adsl.mana.pf) |
| 02:57.09 | kcah | :( |
| 02:57.13 | kcah | ok thanks |
| 02:57.45 | *** join/#asterisk Mike (~mike@201.135.48.52) |
| 02:58.22 | kcah | excuse again... i was working with audiotex of APEX and dialogic boards... maybe i can found some soft for linux like that... to work with simple voice modems? |
| 02:58.43 | tzanger | kcah: short answer again: no |
| 02:58.55 | kcah | it's a joke |
| 02:59.00 | kcah | ? |
| 02:59.10 | tzanger | kcah: longer answer: the x100P driver works with a very specific intel driver. If you have a softmodem with that chipset, maybe |
| 02:59.15 | tzanger | kcah: but you'll get no support here |
| 02:59.31 | Wi_Fi | kcah to another iax client on wirless ap |
| 03:00.10 | kcah | ok really thanks... i'm crying but... then i need to use windows |
| 03:00.30 | kcah | thanks for all!!!! |
| 03:00.57 | tzanger | Wi_Fi: iax client as in what |
| 03:00.59 | BurnedOutGeek | can someone help me get an IAX connection working properly? |
| 03:01.14 | tzanger | BurnedOutGeek: have you checked the wiki |
| 03:01.23 | BurnedOutGeek | tzanger: yeah.. been here all night |
| 03:01.24 | BurnedOutGeek | :) |
| 03:01.38 | BurnedOutGeek | Ive just confused myself to death now |
| 03:02.00 | BurnedOutGeek | tzanger: I have it working, but it shows the incoming call from "asterisk" instead of the phone number |
| 03:02.03 | BurnedOutGeek | server to server |
| 03:02.29 | tzanger | BurnedOutGeek: you need to set (or not clear) callerid |
| 03:03.02 | BurnedOutGeek | Hmmm. Tried that. All I am doig when the call come is is saying Dial(IAX2/server2/${EXTEN}) |
| 03:03.09 | BurnedOutGeek | Im not oing anything with the callerid at all |
| 03:04.02 | kcah | [tzanger] ... it's crazy to try to make some scripts to receive calls (without callerid) on voice modems with linux... with interactive menus... i can't belive that |
| 03:04.15 | kcah | ops excuse for the private |
| 03:04.26 | kcah | ... it's crazy to try to make some scripts to receive calls (without callerid) on voice modems with linux... with interactive menus... i can't belive that |
| 03:04.54 | Marlow | why is that crazy ? |
| 03:05.16 | tzanger | BurnedOutGeek: are you SetCallIdNum() before that? |
| 03:05.21 | BurnedOutGeek | no |
| 03:05.36 | kcah | cause... you can do too many things with linux... and thats really basic... there is too many soft in windows like Voiceguide |
| 03:05.36 | balaji | I get this message from aserisk : Spawn extension (autocontext, 32, 1) exited non-zero on 'SIP/45-9935' |
| 03:05.50 | tzanger | kcah: vgetty will do that IIRC |
| 03:05.58 | Marlow | kcah : but if you don't want to use windows .. |
| 03:06.18 | Marlow | kcah : loads of reasons for not to use it .. |
| 03:06.26 | kcah | yep but i think that vegetty is really limited to some little thinks, it's true? |
| 03:07.02 | balaji | anyone here compiled * on osx ? |
| 03:07.02 | tzanger | kcah: well if you want to do something with *, use hardware * supports... |
| 03:07.16 | Marlow | kcah : what has vgetty with softmodems to do ? |
| 03:07.26 | kcah | yep i don't want to use windows cause i want to operate with apache and Mysql... and make that the calls be registered on the database and can send mails depend of the menu option on the call...(excuse my english) |
| 03:07.47 | tzanger | ... both apache and mysql will run on win32 |
| 03:08.11 | Wi_Fi | guys |
| 03:08.13 | Marlow | exactly ... don't need to use linux for that .. |
| 03:08.14 | Wi_Fi | i needs help |
| 03:08.15 | Marlow | :) |
| 03:08.20 | Wi_Fi | Attempting native bridge of IAX2/10@10/14 and IAX2/50/15 |
| 03:08.22 | kcah | yep... but i don't like windows to left the server started all the day :) |
| 03:08.23 | *** join/#asterisk krw (~krw@209.242.52.25) |
| 03:08.28 | Wi_Fi | should that be done |
| 03:08.35 | kcah | okok |
| 03:08.46 | Marlow | Wi_Fi : sure . |
| 03:08.50 | kcah | excuse if it's the only want solution i'll do it |
| 03:09.01 | Wi_Fi | it hangs up as fast as its built |
| 03:09.04 | kcah | want=one |
| 03:09.14 | Marlow | kcah : what you do and what you need is up to you .. |
| 03:09.19 | tzanger | Wi_Fi: eliminate teh wireless connection,d oes it still do that |
| 03:09.25 | kcah | thanks for all again... i have no money to buy new hardware compatible |
| 03:09.29 | Marlow | kcah : has nothing to do with complaining about the usage of a softmodem |
| 03:09.43 | tzanger | kcah: I understand but what you're trying to do is gonna cost you more in the end with hassle and lost time |
| 03:09.48 | Wi_Fi | other person is on the AP at his company |
| 03:09.49 | Marlow | kcah : then write some drivers that support your hardware .. |
| 03:10.00 | tzanger | Wi_Fi: take your box to his network tomorrow |
| 03:10.01 | kcah | hehehehe.... |
| 03:10.16 | kcah | really thanks... for the help bye... |
| 03:10.20 | tzanger | eliminate the weird stuff and see if it works over straight ethernet on the same LAN and logical subnet |
| 03:10.26 | tzanger | if that works, start seeing where else it could be |
| 03:10.32 | tzanger | I am guessing wireless weirdness |
| 03:10.40 | Marlow | sounds like it .. |
| 03:11.02 | Marlow | what a wanker .. |
| 03:11.02 | tzanger | Wi_Fi: this is basic troubleshooting 101... reduce it to as few variables as possible and start testing |
| 03:12.04 | Wi_Fi | hes gonna try goin thru his lan |
| 03:12.27 | Wi_Fi | iax was flawless till now |
| 03:12.28 | Wi_Fi | hhehe |
| 03:14.30 | Marlow | hmm ... anybody up for testing something ? |
| 03:17.20 | scromp | whoooot |
| 03:17.25 | scromp | ahah. mischan. |
| 03:19.18 | *** part/#asterisk krw (~krw@209.242.52.25) |
| 03:22.32 | *** join/#asterisk ipdeman (~Gary@cpe-maint-port0.netpathway.com) |
| 03:23.00 | libpcp | anyone can help me with astcc |
| 03:23.48 | Wi_Fi | this sucks |
| 03:24.00 | libpcp | im the brands menu. what info should i enter for published number|DID|inc|Markup ? |
| 03:24.43 | *** join/#asterisk drumkilla (~russell@user-24-214-77-148.knology.net) |
| 03:24.43 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 03:32.34 | *** join/#asterisk jbaach (~jbaach@62.206.114.179) |
| 03:34.00 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 03:34.40 | balaji | anyone using asterisk on mac osx |
| 03:37.26 | tessier_ | Can anyone here get me 310 area codes? |
| 03:38.27 | *** join/#asterisk jaiger (~jaiger@65.86.55.130) |
| 03:38.53 | Roger123 | marlow: you about? |
| 03:39.43 | Marlow | sure .. |
| 03:39.52 | Roger123 | marlow: is there a performance difference between debain 3.0 and 3.1 |
| 03:40.15 | brc_ | I *HIGHLY* suggest you use debian sarge |
| 03:40.17 | brc_ | aka testing |
| 03:40.24 | Roger123 | thanks brc |
| 03:40.24 | brc_ | it is almost ready for offical release |
| 03:40.41 | brc_ | Roger123, also, note: DO *N*O*T* use asterisk or zaptel packages |
| 03:40.43 | Marlow | Roger123 : sure .. every software package is up to date in 3.1 |
| 03:40.58 | Marlow | Roger123 : 3.0 is more than two years old .. and at that state |
| 03:41.02 | brc_ | yep |
| 03:41.11 | brc_ | and about to be depricated by sarge |
| 03:41.20 | Roger123 | sweet |
| 03:41.30 | brc_ | Roger123, I suggest you use the brand new debian installer cd's...specifically the "business card" cd installer |
| 03:41.39 | brc_ | !google debian installer |
| 03:41.46 | shoofly | brc_: Search took 0.459039 seconds: Debian -- Debian - Installer: <http://www.debian.org/devel/debian-installer/>; Debian -- Debian - Installer Ports Status: <http://www.debian.org/devel/debian-installer/ports-status>; Debian Installer Test Candidate 1 Released - Debian Planet: <http://www.debianplanet.org/node.php?id=1114>; Debian Installer Beta 4 Relased - Debian Planet: (2 more messages) |
| 03:42.16 | brc_ | http://cdimage.debian.org/pub/cdimage-testing/sid_d-i/i386/pre-rc2/sarge-i386-businesscard.iso |
| 03:42.29 | brc_ | Roger123, you'll probably want asterisk stable |
| 03:42.31 | brc_ | ~asterisk stable |
| 03:42.32 | jbot | well, asterisk stable is export CVSROOT=:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot && cvs checkout -r v1-0 asterisk |
| 03:43.03 | brc_ | Roger123, also if you google you'll find a couple of pages that are specific to installing asterisk on debian with exact package names for needed deps |
| 03:43.05 | Wi_Fi | how is 1.0.2 guys |
| 03:43.17 | brc_ | however, beware. many of the aforementioned pages are very out of date |
| 03:43.29 | brc_ | Wi_Fi, imho it's great |
| 03:43.31 | Roger123 | well i just re-installed the office server with debian 3.0 |
| 03:43.35 | brc_ | I'm so happy to have a stable branch |
| 03:43.38 | brc_ | EWW |
| 03:43.38 | Roger123 | going to dist-upgrade it |
| 03:43.43 | Roger123 | and run CVS-HEAD |
| 03:43.56 | brc_ | Roger123, yeah, dist-upgrade should work to upgrade from debian woody to debian sarge |
| 03:44.07 | jaiger | brc_: but there are debian packages for asterisk |
| 03:44.13 | Roger123 | yeah it did work last time. just had to recompile everything |
| 03:44.32 | brc_ | jaiger, yes, which are very very poorly made last time I checked (admitedly awhile ago) |
| 03:44.33 | Wi_Fi | brc 1.0.2 is stable? |
| 03:44.42 | brc_ | Wi_Fi, again, IME yes |
| 03:45.04 | brc_ | Wi_Fi, oh, you mean is the 1.0.2 tarball the same as cvs stable (1-0)? |
| 03:45.10 | jaiger | brc_: what are the issues? I haven't had too many problems with them so far |
| 03:45.18 | jaiger | in my test system that is |
| 03:45.20 | Wi_Fi | how do go from cvs head to 1.0.2 |
| 03:45.35 | brc_ | Wi_Fi, uhhh.....download, compile, make install |
| 03:45.38 | jaiger | I'm gonna upgrade to debian on my production pbx soon |
| 03:45.54 | Wi_Fi | cant get it thru cvs? |
| 03:46.05 | brc_ | jaiger, well let's see...for one thing there is a log rotate script which sends asterisk a "STOP GRACEFULLY" every so often |
| 03:46.08 | brc_ | ! |
| 03:46.18 | brc_ | Wi_Fi, huh? |
| 03:46.22 | *** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net) |
| 03:46.36 | kram | where did twisted go? |
| 03:46.39 | Wi_Fi | cvs 1.0.2 |
| 03:47.03 | jaiger | brc_: ouch |
| 03:47.06 | brc_ | ~tell Wi_Fi about asterisk stable |
| 03:47.08 | brc_ | jaiger, yeah |
| 03:47.30 | Wi_Fi | tnx b |
| 03:48.21 | brc_ | kram, no idea |
| 03:48.44 | *** part/#asterisk Qubis (~Qubis@65.77.78.3) |
| 03:48.47 | Wi_Fi | checkout -r v1-0-2 asterisk? |
| 03:48.52 | drumkilla | kram is my hero |
| 03:48.53 | brc_ | Wi_Fi, no |
| 03:48.56 | brc_ | read please |
| 03:48.59 | brc_ | ~asterisk stable |
| 03:49.00 | jbot | from memory, asterisk stable is export CVSROOT=:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot && cvs checkout -r v1-0 asterisk |
| 03:49.02 | Wi_Fi | i did |
| 03:49.03 | Wi_Fi | sorry |
| 03:49.21 | Wi_Fi | my question is will that give me 1.0.2 |
| 03:49.42 | brc_ | yes |
| 03:49.55 | Wi_Fi | one more |
| 03:50.02 | drumkilla | with whatever fixes have been put in since then ... |
| 03:50.02 | Wi_Fi | shutdown asterisk befor i do it |
| 03:50.17 | brc_ | exactly |
| 03:50.48 | Wi_Fi | thnx |
| 03:51.30 | balaji | my asterisk server sends a BYE message to SIP phones about 3 secs into the conversation |
| 03:51.36 | balaji | any idea on this ? |
| 03:51.42 | Wi_Fi | crap |
| 03:51.47 | Wi_Fi | cvs checkout: move away asterisk/sounds/fpm-sunshine.mp3; it is in the way |
| 03:52.13 | Wi_Fi | one of many errors |
| 03:52.32 | Wi_Fi | im an idiot |
| 03:53.27 | moonwick | redundant, too |
| 03:57.13 | Wi_Fi | collect2: ld returned 1 exit status |
| 03:57.13 | Wi_Fi | Makenshi: *** [asterisk] Error 1 |
| 03:58.08 | Wi_Fi | ./usr/bin/ld: cannot open output file asterisk: Is a directory |
| 03:59.25 | *** join/#asterisk jsharp (~jsharp@122-119.34-65.swfla.rr.com) |
| 04:01.06 | Wi_Fi | rf ? asterisk? |
| 04:01.50 | brc_ | drumkilla! |
| 04:01.53 | brc_ | whatup man |
| 04:02.49 | Wi_Fi | how do i remove old asterisk install dir |
| 04:03.26 | Wi_Fi | rm -f asterisk? |
| 04:03.56 | *** join/#asterisk DenyMe (~dengdeng@202.73.118.206) |
| 04:04.18 | DenyMe | good morning |
| 04:05.19 | DenyMe | how do i create dsn in resconfig? |
| 04:06.02 | *** join/#asterisk Peanut486 (~dustin@net24-164-112-183.neo.rr.com) |
| 04:06.19 | JerJer | who broke chan_sip? |
| 04:06.42 | brc_ | you say broke like that's a new development |
| 04:06.46 | Peanut486 | Anyone have advise for configuring zaptel /w 2.6 kernels? Anyone know of good documentation. |
| 04:06.47 | JerJer | cvs -head's chan_sip is giving me some funky messages... 400 Bad Request, malformed From header |
| 04:07.12 | jaiger | README.26 |
| 04:07.14 | *** join/#asterisk ChulJin (chuljin@24-205-55-37.gln-eres.charterpipeline.net) |
| 04:07.21 | brc_ | Peanut486, all you've gotta do is symlink /usr/src/linux-2.6 to your kernel source |
| 04:07.31 | Peanut486 | humm |
| 04:07.32 | Moc | I guess I wont update |
| 04:08.07 | DenyMe | Hi JerJer |
| 04:08.48 | brc_ | drumkilla++ |