| 00:00.41 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 00:00.49 | Andeya | any reason why an extenal x-lite client can get registered, make a call to an extension but then there is no audio after that? It works fine if the call is place inside the network |
| 00:01.40 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 00:01.54 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 00:03.01 | Brandon_W | tzanger - well well well, I think that hit the nail on the head |
| 00:03.02 | Brandon_W | thanks man |
| 00:03.43 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
| 00:07.20 | *** join/#asterisk jsolares (n=jsolares@216.106.165.63) |
| 00:08.25 | jsolares | hi |
| 00:08.46 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 00:09.06 | jsolares | i'm having a bit of a weird problem, i have 2 asterisk boxes connected via iax2 and trunking, over time audio degrades and begins to kind of stutter |
| 00:09.55 | lancey | sounds like a timing issue, but no ideas how to help :/ |
| 00:10.00 | jsolares | it's a dual xeon 5150 :S, ubuntu on kernel 2.6.19-smp |
| 00:10.22 | jsolares | if we reboot the machine it starts ok anew even with ~40 calls with g729, but over time the problem comes back |
| 00:10.32 | lancey | jsolares: does disabling trunking fix the issue? |
| 00:10.44 | ariel_ | how many connections between them? |
| 00:10.49 | jsolares | no idea, it's bandwidth limited so we cant disable trunking |
| 00:11.11 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:11.16 | jsolares | by connections you mean new calls coming in? or average calls up? |
| 00:11.18 | *** part/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:11.29 | ariel_ | iax2 is only good for about 8 calls once you start going over it starts to get jitter |
| 00:11.42 | jsolares | 30~40 calls |
| 00:11.46 | ariel_ | sip will work better due to it's using different ports for sound |
| 00:12.14 | ariel_ | that is far too many for iax2 and only one port for rtp and registration |
| 00:12.18 | jsolares | we would use sip if it supported VAD since the bw savings would be around the same |
| 00:12.29 | lancey | i'm having problems getting zaptel to work - [May 5 03:03:38] WARNING[61253]: loader.c:415 load_dynamic_module: Error loading module 'chan_zap.so': /usr/local/lib/asterisk/modules/chan_zap.so: Undefined symbol "pri_keypad_facility" |
| 00:12.33 | JerJer | http://www.fcw.com/article102645-05-04-07-Web |
| 00:12.34 | lancey | anyone seen such error? |
| 00:13.07 | jsolares | seems like libpri wasnt installed |
| 00:13.20 | lancey | xmm |
| 00:13.27 | lancey | do i need libpri for just ztdummy? |
| 00:13.54 | jsolares | shouldnt, but shouldnt != no :S |
| 00:14.05 | lancey | :P |
| 00:14.40 | jsolares | ariel_: any reason why the jitter would show up after hours of correct behaviour? and not "instantly"? |
| 00:14.43 | lancey | should i recompile zaptel / chan_zap after installing lib_pri ? |
| 00:15.24 | ariel_ | jsolares, no it's just something that has happened for as long as I been using asterisk |
| 00:15.28 | jsolares | install libpri and see if it just "works" |
| 00:15.45 | lancey | codec_zap.so => (Generic Zaptel Transcoder Codec Translator) |
| 00:15.54 | lancey | jsolares: you deserve a beer or two |
| 00:16.01 | lancey | very very big thanks |
| 00:16.19 | jsolares | i deserve a working iax2 for ~100 calls between two servers XD |
| 00:16.55 | lancey | why not try switching to SIP, as ariel_ suggested |
| 00:17.25 | jsolares | because of bandwidth |
| 00:17.46 | jsolares | unless there's a beta with VAD floating around |
| 00:17.49 | ariel_ | JKADJJJJJKJ:LDADF |
| 00:18.06 | ariel_ | YIKES my keyboard is working backwards |
| 00:18.12 | lancey | :))) |
| 00:18.41 | lancey | jsolares thought VAD is part of the codec, not the protocol? |
| 00:19.28 | lancey | anyways, is bandwidth so expensive out there? it couldn't be worse than here. |
| 00:19.33 | jsolares | asterisk itself doesnt support VAD AFAIK, so it cant take E1 (PRI) and send it with VAD enabled g729 either via IAX2 or SIP |
| 00:19.49 | ariel_ | I have about 15 asterisk working together delivering calls between them and there all on sip. |
| 00:19.52 | jsolares | 1000$ for 1mbit aprox |
| 00:19.59 | lancey | wow |
| 00:20.04 | lancey | jsolares where is that? |
| 00:20.05 | ariel_ | but I have an MPLS network 45mb running. |
| 00:20.15 | jsolares | guatemala, down in central america |
| 00:20.24 | lancey | nice |
| 00:20.28 | jsolares | you can get cheaper with volume but that's the going rate |
| 00:20.36 | lancey | thought we've been the most fucked up place |
| 00:20.48 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 00:21.11 | lancey | guys, that silence_suppression = yes in asterisk.conf |
| 00:21.22 | lancey | is it available in 1.4.x or just in trunk ? |
| 00:22.30 | lancey | it seems like it is ... i no longer see those silenceSupp:off in the sip invites. |
| 00:22.30 | jsolares | well i cant find it in the 1.4.4 sample config files |
| 00:23.38 | lancey | nor do i, but it works |
| 00:23.54 | kuku5 | I have 1 mb for $10/month |
| 00:23.56 | lancey | but i still can't connect to that shitty softswitch |
| 00:24.26 | lancey | kuku5: our national telco charges 50E / Mbit just for interconnection over ~130 km |
| 00:24.27 | ariel_ | wow 1mb for $ 10 per month is low |
| 00:24.42 | kuku5 | Cogents |
| 00:24.45 | kuku5 | Cogent |
| 00:24.46 | lancey | yup, nice |
| 00:24.55 | kuku5 | They get cheaper if you go over 1GB |
| 00:25.02 | jsolares | :P |
| 00:25.31 | lancey | :)) |
| 00:25.57 | ariel_ | Cogent that is strange there trying to charge me for a t1 at over $ 499.00 per month |
| 00:26.20 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:26.21 | jsolares | meh why does it only start to be crappy after hours of working, sounds like a bug somewhere :S |
| 00:26.47 | lancey | jsolares does stopping and starting asterisk fix the issue |
| 00:26.47 | ariel_ | jsolares, what do you do to clear it out |
| 00:26.56 | lancey | or only when you restart the machines? |
| 00:27.00 | jsolares | only on restart |
| 00:27.09 | lancey | did you try loading/unloading zaptel modules? |
| 00:27.11 | ariel_ | your have a memory leak some place |
| 00:27.25 | ariel_ | are you using the free g729 from intel? |
| 00:27.27 | jsolares | which seems like a linux bug compounded by the large use of a single connection perhaps *shrugs* |
| 00:27.48 | jsolares | ariel_: for the time being yes, no use of buying a codec if it wont work :( |
| 00:28.00 | lancey | same goes here |
| 00:28.09 | lancey | digium dropped support for freebsd :/ |
| 00:28.23 | lancey | i have 20 licenses, if anyone needs them... |
| 00:28.26 | ariel_ | the g729 from intel had a bug in it |
| 00:28.34 | jsolares | lancey: no did not try loading/unloading will try that next time |
| 00:28.36 | ariel_ | that would leak memory |
| 00:28.57 | lancey | ariel_: if it was the g729, it should be fixed by stopping and starting asterisk again, no? |
| 00:29.05 | jsolares | yeah i wonder if i have the leaky one or the good one, heh time to go check |
| 00:29.07 | ariel_ | yes |
| 00:29.20 | lancey | jsolares says it doesn't |
| 00:31.44 | jsolares | i would try going to 1.4.x but i already did many changes to 1.2.x that would need to be replicated, hurray for R2 and added signalling needed by customer >_> |
| 00:32.36 | lancey | R2D sux |
| 00:32.47 | lancey | no ISDN out there? |
| 00:32.58 | jsolares | well one side is ISDN |
| 00:33.03 | jsolares | the other side, the remote one is R2 |
| 00:33.30 | ariel_ | I still use 1.2.17 on allot of my boxes |
| 00:33.34 | ariel_ | if it works don't change |
| 00:33.37 | jsolares | it's kinda like a cellular node, wireless end points connected to each cell wich is R2 |
| 00:33.52 | lancey | ~iidbdfi |
| 00:34.01 | dc3aes | would anybody happen to know why a Linksys PAP2 extension would receive a ring, but no rings afterwards until another registration? new to this but can't seem to corrolate documentation to the problem I am having.. |
| 00:34.13 | lancey | ~iiabdfi |
| 00:34.16 | jbot | from memory, iiabdfi is If it ain't broke, don't fix it! |
| 00:34.16 | lancey | gr |
| 00:34.32 | jsolares | it's broken, horribly broken XD :( |
| 00:34.47 | ariel_ | ring only after registration sounds line a you need a stun server or correct nat settings |
| 00:35.11 | lancey | dc3aes what firmware you're running on the PAP2 |
| 00:35.13 | jsolares | yeah perhaps a qualify=yes in your sip.conf for that device |
| 00:35.34 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 00:36.14 | jsolares | heh 34 channels and problems.. *yet* |
| 00:36.34 | jsolares | calls actually heh |
| 00:39.37 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 00:42.01 | dc3aes | lancey. let me check.. im inside a lan |
| 00:42.06 | dc3aes | I will check qualify as well |
| 00:42.36 | dc3aes | 3.1.3LS |
| 00:43.56 | lancey | i had some similar problems with 3.1.7 |
| 00:44.05 | lancey | which went away after upgrading |
| 00:44.22 | lancey | lemme check what i'm running |
| 00:45.10 | lancey | well, it IS 3.1.7, so the problems were with a previous version |
| 00:45.50 | Waverly360 | sangoma vs. digium..who wins? |
| 00:49.03 | ariel_ | well seening that your on a digium supported site. it's fairly easy to say most here are on digium's side. |
| 00:49.27 | Waverly360 | Hmm...guess I didn't know that. |
| 00:49.51 | Waverly360 | I'm just trying to narrow down what might be causing my boxes to crash. |
| 00:50.23 | ariel_ | what crash? |
| 00:50.32 | Waverly360 | kernel panics |
| 00:50.32 | ariel_ | which boards are you using? and how? |
| 00:51.03 | ariel_ | are you using x86_64 linux setup? |
| 00:51.06 | Waverly360 | we have about 30 different servers using miscellaneous sangoma cards |
| 00:51.23 | Waverly360 | we have one server, that was built before all of them, using a digium card |
| 00:51.31 | Waverly360 | that server has been up for 468 days |
| 00:51.45 | lancey | Waverly360: you now got the answer :) |
| 00:52.01 | Waverly360 | the others barely make it a month before they crash...typically there are messages on the screen saying kernel panic |
| 00:52.18 | Waverly360 | lancey: well..the problem is...the pstn cards aren't the only thing different in those boxes |
| 00:52.28 | ariel_ | in the debug do you get anything with memlock |
| 00:52.30 | LeddyHM | what settings do you guys recommend for polycom phones on a sip trunk, rfc2833, and reinvite no? |
| 00:52.47 | ariel_ | and agin are you using x86_64 or normal 32bit linux? |
| 00:52.55 | Waverly360 | ariel_: normal 32 bit linux. |
| 00:53.01 | ariel_ | LeddyHM, depends |
| 00:53.15 | ariel_ | use use queues? or channel_spy |
| 00:53.44 | wushin | so anyone ever had an issue were a normally functional card starts going bad and ports start to fail? |
| 00:53.44 | LeddyHM | queues are setup, but not in use currently |
| 00:54.21 | LeddyHM | I'm having issues with dtmf codes being transmitted/understood by some remote systems (intermittently) |
| 00:54.44 | Waverly360 | ariel_: I don't remember seeing memlock, but I can't always get to the boxes. I've only seen a couple of them, and neither mentioned anything about memlock. |
| 00:55.09 | Waverly360 | ariel_: I am using queues, but not the agent login functions |
| 00:55.13 | ariel_ | I had some kernel panic with digiums TE412b as well. But it was more of memory and channel_spy issue with too many people listening to same call on our call centers |
| 00:55.43 | Waverly360 | ariel_: I'm wondering if it could be a problem with the zaptel kernel drivers..that's also different on the boxes |
| 00:56.10 | ariel_ | could be do you have any setups you can put up as test boxes? |
| 00:56.30 | Waverly360 | Yes, but I need to figure out how to force a crash. |
| 00:57.11 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 00:57.21 | ariel_ | test, calls , test, calls, work it, work it. |
| 00:58.27 | jsolares | heh silence_suppression=yes on asterisk.conf is for bug 5374 generating audio so that a sip phone with VAD would still work correctly with meetme/chan_spy/etc if you have a timing source. oh well |
| 00:58.32 | Waverly360 | ariel_: heh. Yeah...I guess. Gotta figure this out soon..customers are getting pissed. |
| 00:59.03 | ariel_ | you have any of the logs when they core? |
| 00:59.09 | lancey | jsolares not only |
| 00:59.15 | ariel_ | it should tell you what made it crash |
| 00:59.21 | Waverly360 | ariel_: /var/log/messages never shows me anything. Just a gap in time. |
| 00:59.22 | lancey | it does make astering send out frames by itself |
| 00:59.26 | lancey | not only when it receives one |
| 00:59.36 | lancey | => better quality on jittery links |
| 00:59.44 | Waverly360 | ariel_: I think there might be a way to turn out more kernel logging..I'm just not quite sure how. |
| 00:59.52 | jsolares | if using sip |
| 01:00.56 | Waverly360 | ariel_: I don't suppose you know of any burn-in software I could use to load test an asterisk box do you? |
| 01:01.50 | jsolares | sipp |
| 01:01.57 | ariel_ | yes put it into a call center that does out bound dialings at around 360 channels per minute. |
| 01:01.59 | lancey | yup, i suppose |
| 01:02.08 | Waverly360 | ariel_: I was hoping for something a little less public :) |
| 01:02.22 | ariel_ | lol |
| 01:02.55 | ariel_ | there are nights we blow up our uplink providers 2 and 3 ds3's he has.... makes him kinda mad at us sometimes... but he loves our biz. |
| 01:03.27 | Waverly360 | jsolares: Oh..sipp awesome. :) |
| 01:03.35 | Waverly360 | ariel_: I'll bet :) |
| 01:03.50 | ariel_ | Waverly360, we have already told him about sip.... |
| 01:04.15 | Waverly360 | ariel_: huh? |
| 01:04.21 | ariel_ | there are some load emulation software on the voip-info for sip |
| 01:04.23 | jsolares | sipp not sip :P |
| 01:05.12 | jsolares | another way to test would be .call files between two asterisks |
| 01:05.45 | Waverly360 | .call files? |
| 01:05.48 | jsolares | call the other end, other end answers and plays back some audio, and you connect your local also to another playback |
| 01:05.51 | ariel_ | Waverly360, he was saying this: http://sipp.sourceforge.net/ |
| 01:05.54 | *** join/#asterisk JuRaDo (n=asdubal@201-1-109-2.dsl.telesp.net.br) |
| 01:06.21 | jsolares | .call files => http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 01:06.32 | ariel_ | call files can lead to loops in dial plans and really mess up your box... and b/w |
| 01:06.43 | ariel_ | you need to be careful with them. |
| 01:07.09 | jsolares | good way to recreate his load and kernel panic heh |
| 01:07.16 | Waverly360 | ariel_: I think I'm going to build a couple of boxes specifically for that purpose. I want to see if I can force one of these boxes to crash reliably |
| 01:07.41 | Waverly360 | ariel_: Then I can start trying things to fix it..otherwise I'll just be guessing. |
| 01:08.10 | ariel_ | sounds like a plan |
| 01:08.24 | Waverly360 | ariel_: Let's hope it's a fruitful one. :P |
| 01:08.30 | Waverly360 | thanks for the help. |
| 01:10.36 | jsolares | Waverly360: FWIW i have 8 e1's from 2 sangoma cards in two machines without any kernel panics, i have audio problems which seemingly come from using iax2, but no kernel panics, they are xeon 5150 with 2gb of ram, supermicro servers |
| 01:11.13 | lancey | supermicro might be the keyword |
| 01:11.22 | Waverly360 | jsolares: What version of wanrouter are you using? |
| 01:12.03 | ariel_ | I have a few Dell 2850's with sagoma boards working fine. Dual Xeons 4 gig ram 3.04ghz.... |
| 01:12.03 | *** join/#asterisk uppal (n=uppal@2001:618:400:888c:218:deff:fe9f:a77f) |
| 01:12.07 | jsolares | on those 2.3.4-7 |
| 01:12.31 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 01:13.00 | Waverly360 | jsolares: Do you use rpms to install those drivers, or did you compile them by hand? I had those one some of my servers too, but I couldn't get the PRI cards I had to work with them. |
| 01:13.08 | ariel_ | ManxPower, long time no see. hope your doing well. |
| 01:13.09 | jsolares | ariel_: with the same customer there are 3 other servers, dual xeon 2.8ghz 1gb ram with 1 sangoma 4 E1 card with no problems, also not nearly the same load, but with close to no iax2 |
| 01:13.23 | ManxPower | Getting a message on your voicemail that says "the harddrive in the mail server failed" means you'll have a REALLY SUCKY DAY. |
| 01:13.24 | jsolares | Waverly360: ubuntu and compiled |
| 01:13.37 | jsolares | ouch |
| 01:13.45 | Qwell | ManxPower: at least you weren't using imap storage |
| 01:13.49 | ariel_ | I see |
| 01:13.59 | LeddyHM | damn intermittent dtmf |
| 01:14.02 | Waverly360 | ManxPower: yuck. Sorry to hear that. |
| 01:14.18 | LeddyHM | any idea what would cause issues? i.e. pbx, phone, provider? |
| 01:14.51 | ariel_ | I have been fighting my Inet 50mb provider all day. The keep giving me 5 to 10 sec drops. Which makes it hell on a call center.... |
| 01:14.53 | jsolares | ariel_: we also had a client that had the audio problem but only when using 2 E1's with digium g729 and iax2/trunking, the problem went away tho ( stopped working with the customer XD ) |
| 01:15.13 | ManxPower | Oh, and the latest 1.2.x asterisk we installed crashes about once per day. |
| 01:15.14 | jsolares | LeddyHM: all of the above |
| 01:16.07 | LeddyHM | damn that's no help |
| 01:17.28 | ariel_ | I am trying to see how I can get off telcov ever since they where taken over by L3 out inet connect keeps dropping off. |
| 01:17.28 | ariel_ | out/our |
| 01:17.28 | jsolares | LeddyHM: the pbx and phone should be configured so that they work with each other preferably with DTMF out of band, either sip info or rfc2833 |
| 01:17.28 | LeddyHM | yeah I am using rfc2833 |
| 01:17.36 | LeddyHM | tried reinvite yes, and no |
| 01:18.21 | jsolares | well reinvite only is for letting the two devices talk to each other directly or having asterisk always in the middle |
| 01:18.21 | jsolares | tried with info? |
| 01:18.34 | LeddyHM | info? |
| 01:18.49 | jsolares | dtmfmode=info |
| 01:18.59 | LeddyHM | I'll try that now |
| 01:19.19 | jsolares | your phone needs to support it as well |
| 01:19.26 | jsolares | what phone are you using |
| 01:19.33 | ariel_ | what phones |
| 01:19.59 | LeddyHM | polycom 501 |
| 01:20.13 | ariel_ | polycoms use rfc2833 |
| 01:21.18 | ariel_ | bbl need to go deal with my child and why she is not wanting to go to bed........ |
| 01:21.31 | LeddyHM | info hangs up on "*" |
| 01:22.11 | LeddyHM | my voip provider keeps wanting us to use reinvite=yes with the phones on a subnet provided by the iad |
| 01:22.31 | jsolares | http://bugs.digium.com/view.php?id=9538 look at that :S |
| 01:23.39 | LeddyHM | still reading |
| 01:23.47 | LeddyHM | but noticed "cebyond" |
| 01:23.52 | LeddyHM | who is our voip provider |
| 01:24.16 | jsolares | hrm, you could see if your asterisk is detecting all the dtmf tones from your phone |
| 01:24.45 | jsolares | the problem with that bug is that asterisk does in fact get all the dtmf correctly from your phone, but only sends the first one to the voip provider |
| 01:24.58 | jsolares | how do you have your dtmfmode set with your provider? what do they support? |
| 01:25.30 | jsolares | you could create a test account with voipjet, and try it with iax2 |
| 01:25.45 | LeddyHM | yeah we get all tones |
| 01:25.53 | LeddyHM | using wireshark |
| 01:25.56 | GoRapsGo | ManxPower: bad weeks suck |
| 01:26.06 | GoRapsGo | I've done very few hours this week worth of work.... |
| 01:26.31 | LeddyHM | that isn't necessarily our problem |
| 01:26.46 | jsolares | that means your phone sends all tones, just not that your asterisk recognizes the dtmf |
| 01:26.54 | LeddyHM | we can't always get the first digit |
| 01:27.08 | LeddyHM | but it's a * in this case |
| 01:27.17 | LeddyHM | i.e. conference bridge |
| 01:27.24 | jsolares | i'm not sure if the cli shows the dtmf's read with a verbosity or debug set |
| 01:27.38 | LeddyHM | I have it set to 10, is that highest? |
| 01:28.38 | jsolares | which debug or verbose? |
| 01:29.24 | LeddyHM | verbose |
| 01:30.15 | LeddyHM | and no keystrokes detected :( |
| 01:30.17 | jsolares | try setting debug to 3 or higher and see if it shows the sip dtmf detected by asterisk |
| 01:30.28 | jsolares | or even sip debug |
| 01:32.10 | ManxPower | if the DTMF is inband you won't see it in SIP debug |
| 01:32.27 | ManxPower | you may not even see OOB DTMF, as it is really an RTP packet, not SIP packet |
| 01:32.59 | LeddyHM | yeah that's what I was thinking |
| 01:33.12 | LeddyHM | since we are having problems I brought a dev box online |
| 01:33.30 | LeddyHM | I'm going to install wireshark to make sure I'm still getting tones |
| 01:34.09 | jsolares | try using dtmf in asterisk itself, like logging into voicemail then, if that works correctly 100% of the time all would point to your voip provider |
| 01:34.48 | LeddyHM | for sure |
| 01:34.51 | LeddyHM | that works 100% |
| 01:35.10 | LeddyHM | it's just a few of our conference bridges we have issues with |
| 01:36.22 | LeddyHM | they say they can't troubleshoot unless they can get a packet capture at the remote end |
| 01:36.27 | jsolares | so sending the dtmf to your provider is the problem :S, did your provider tell you what dtmfmode to use? info or rfc2833? |
| 01:36.49 | LeddyHM | they just want us to use reinvite=yes, and rfc2833 |
| 01:37.11 | LeddyHM | but our phones have to be on thei subnet so they can nat properly to the phone |
| 01:37.49 | jsolares | yeah canreinvite=yes takes asterisk out of the way and tries to connect your phone directly to your provider |
| 01:38.12 | LeddyHM | yeah but I can't imagine our phone is doing anything differently then * would |
| 01:38.53 | jsolares | capturing the packets leaving your * box would tell you |
| 01:39.19 | LeddyHM | yeah I'm compiling wireshark now to see |
| 01:40.54 | *** join/#asterisk VJFROMGT (n=vjfromgt@pool-71-125-247-84.nycmny.east.verizon.net) |
| 01:41.38 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 01:42.21 | *** join/#asterisk bawb2 (n=bawb2@ip50210.estcmp.ku.edu) |
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| 01:55.07 | *** part/#asterisk Andeya (n=paviles@dsl-7-36.cofs.net) |
| 02:04.58 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 02:05.35 | LeddyHM | hmm |
| 02:05.57 | LeddyHM | I see the code coming in from my external ip, then getting passed on to voip provider |
| 02:09.19 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
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| 02:19.38 | plasmid | Scenario: is it possible to use an 1-866# that's tied to one DID# provider (A) to ring a DID# from provider (b)? |
| 02:21.21 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 02:22.24 | jsolares | i guess you could create a new call to DID#B from the * box that received DID#A |
| 02:22.40 | jsolares | also some providers have fallback to pots for DID's you could try that as well |
| 02:24.46 | LeddyHM | jsol: when * sends a dtmf evt out, the provider receives it and translates it to an actual tone through a pots line? |
| 02:25.52 | jsolares | LeddyHM: it should |
| 02:26.31 | jsolares | try a different voip provider, voipjet has a free iax2 account with 0.25$ enough for testing, nufone is great also |
| 02:26.59 | LeddyHM | free is better :) |
| 02:27.31 | LeddyHM | at one point they said we needed to upgrade to 1.4 tree to change the delay or something (for tones) |
| 02:27.41 | jsolares | you get what you pay for as well, but yeah free is better |
| 02:28.19 | LeddyHM | voipjet.com ? |
| 02:28.42 | jsolares | yep |
| 02:29.30 | jsolares | it's not even great, but good enough for you to test |
| 02:29.46 | LeddyHM | looks like a pay as you go credit based |
| 02:29.57 | Waverly360 | Welp, I have sipp running on one of those boxes. I'm gonna let it run all weekend and see if the damned thing crashes. |
| 02:30.02 | LeddyHM | seems like something useful |
| 02:30.12 | LeddyHM | wonder if they do auto creation of accounts |
| 02:30.19 | jsolares | good luck Waverly360 |
| 02:30.35 | jsolares | LeddyHM: they do, and paypal as well, just only verified paypal |
| 02:31.01 | Waverly360 | jsolares: Thanks. night everyone. |
| 02:31.02 | LeddyHM | damn, never verified our paypal |
| 02:31.17 | jsolares | nufone is also pay as you go, better service, did's but more pricey and only US termination is "offered" altho i think A-Z works just as well just no official pricing |
| 02:31.51 | LeddyHM | I don't need any "services" |
| 02:31.55 | LeddyHM | just an external sipconnect |
| 02:32.15 | jsolares | voipjet is iax2 only, nufone i think works with both |
| 02:32.37 | LeddyHM | be better to do apples to apples, I think |
| 02:32.53 | jsolares | apples to apples? sip to sip? |
| 02:33.01 | LeddyHM | non iax |
| 02:33.19 | jsolares | a friend has a "call center" with 4 booths sip phones, only uses IAX2 for outbound calls, works great |
| 02:33.23 | LeddyHM | since we aren't connecting to cbeyond via iax |
| 02:33.29 | threat | ok back to my original question I asked yesterday that nobody answered. I am having DTMF problems, where I ring up a company that has a menu but when I press a button on my analog phone the menu doesn't recognised that i pressed it. |
| 02:34.00 | jsolares | threat: it could be anything, your phone not configured correctly, asterisk not configured correctly, your pots provider not configured correctly |
| 02:34.16 | LeddyHM | or god farted |
| 02:34.17 | dc3aes | so I uuhh... may or may not have set one of my other PAP2 devices to the same userid causing the problem with internal extensions not ringing on the second try :/ |
| 02:34.24 | LeddyHM | ;) |
| 02:34.34 | threat | jsolares, ok, do you know any way to eliminate any of those variables? |
| 02:34.40 | LeddyHM | dtmf is the bane of voip existence, for me at least |
| 02:34.53 | LeddyHM | threat: contact your voip provider |
| 02:34.54 | threat | LeddyHM, :( |
| 02:35.00 | threat | LeddyHM, What is not reassuring |
| 02:35.09 | jsolares | analog phone is connected how to your asterisk box? |
| 02:35.24 | jsolares | sip ata? iax2 ata? fxs card? |
| 02:35.26 | threat | LeddyHM, I did, they said I need dmtfmode yes to rfc something or other, which I have already defined. |
| 02:35.47 | jsolares | who's your provider? |
| 02:35.51 | threat | I have a 4 port TDM card, 1 FXO 1 FXS |
| 02:35.58 | threat | GoTalk (AU) |
| 02:36.14 | threat | I use SIP to connect to them from my asterisk box |
| 02:36.28 | jsolares | heh i knew the guy responsible for the veridas collapse |
| 02:36.47 | threat | the what? |
| 02:36.49 | jsolares | well you could try and see if dtmf works within asterisk, AGI's, IVR, voicemail, etc. |
| 02:37.05 | jsolares | there was a small isp called veridas that crashed and burned sep of last year |
| 02:37.07 | threat | jsolares, ok |
| 02:37.31 | jsolares | related to the visp wild that also crashed and burn not long ago and got bought off by gotalk |
| 02:37.43 | threat | heh |
| 02:37.45 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 02:38.11 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 02:38.27 | *** part/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
| 02:38.36 | jsolares | you could try the whirlpool forums they have a voip section as well |
| 02:39.04 | threat | hmmm, is this an issue? ---> May 5 12:11:39 WARNING[26858]: chan_zap.c:3958 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. |
| 02:39.08 | threat | jsolares, ok |
| 02:39.13 | threat | jsolares, anything else? :) |
| 02:39.39 | jsolares | i dont think that's an issue per se, just seems your caller-id isn't getting detected and with it being a fxs port there shouldnt be one |
| 02:39.50 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 02:40.11 | threat | ok good :) |
| 02:40.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:40.58 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 02:42.00 | LeddyHM | 10 bucks for 500 minutes |
| 02:42.12 | jsolares | you could also try a softphone on your computer to your * box then onto gotalk |
| 02:42.13 | LeddyHM | 9 hours of testing should be plenty |
| 02:42.19 | threat | LeddyHM, is that for your tech support? :) |
| 02:42.30 | LeddyHM | no that's for a test |
| 02:42.35 | LeddyHM | using a different provider |
| 02:42.45 | jsolares | who did you decide upon? |
| 02:42.50 | LeddyHM | nufone |
| 02:42.53 | LeddyHM | looks like they do sip |
| 02:43.11 | jsolares | i think shido6 is with nufone, also jerjer ;) |
| 02:43.24 | LeddyHM | I don't think configuring iax is a fair shake |
| 02:43.35 | jsolares | iax2 just works with dtmf ;) |
| 02:43.36 | LeddyHM | god forbid it be a sip issue |
| 02:44.10 | jsolares | hehe |
| 02:44.39 | jsolares | threat: http://forums.whirlpool.net.au/index.cfm?a=wiki&tag=VOIP_Gotalk dtmf should work just fine :S |
| 02:44.59 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 02:45.09 | threat | yay |
| 02:45.50 | threat | What is DID again ./ ? |
| 02:46.16 | jsolares | DID = Dial Inward Digits, IE the number they give you so that everyone else can call you, with gotalk that would be a 09XXXXXXXXX number |
| 02:47.30 | threat | ok |
| 02:47.53 | threat | G711a, G711u, that is alaw and ulaw? |
| 02:47.59 | jsolares | correct |
| 02:48.16 | threat | :D |
| 02:49.20 | LeddyHM | Direct Inward Dial |
| 02:49.35 | CBU[^_^]M`` | anyone knows where to buy a cheap GSM gateway eqp for home use? |
| 02:49.41 | threat | hmmm I could never get this to work, I needed to drop the /09XXXXXX bit --> Asterisk Registration String: register => 09xxxxxx:PASSWD@sip.gotalk.com/09xxxxxx |
| 02:49.43 | LeddyHM | toMAYto, toMAHto |
| 02:49.46 | jsolares | no idea CBU |
| 02:49.53 | CBU[^_^]M`` | :) |
| 02:50.20 | jsolares | the /09XXXXXX is what tells your asterisk which extension to call when a call comes in, otherwise it sends it to the 's' extension in the context you have it in |
| 02:50.54 | threat | ok that is good :) |
| 02:51.29 | LeddyHM | hah, wire transfer, business check, google checkout |
| 02:51.30 | LeddyHM | GAY |
| 02:51.52 | jsolares | lol |
| 02:52.51 | jsolares | i used google checkout |
| 02:54.33 | jsolares | oh well time to go, gnite/gday everyone |
| 02:57.27 | LeddyHM | dang |
| 02:59.57 | plasmid | Scenario: is it possible to use an 1-866# that's tied to one DID# provider (A) to ring a DID# from provider (b)? |
| 03:01.33 | *** join/#asterisk key2 (n=key2@gob75-7-82-247-113-230.fbx.proxad.net) |
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| 03:10.54 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 03:56.48 | coppice | tzafrir: are you around? |
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| 05:12.41 | JT | haha i just got a pile of polycom 501s for AUD$75ea (USD$50) |
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| 05:22.35 | ShadowHntr | ~thebook |
| 05:22.47 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:23.22 | *** join/#asterisk joaovianna (i=joaovian@ool-4354d1a8.dyn.optonline.net) |
| 05:24.01 | joaovianna | I'm trying to compile version 1.4.4 but I have errors. How to apply a patch ? Any manual ? |
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| 05:58.58 | rue_mohr | hello |
| 05:59.12 | rue_mohr | anyone using a newbridge 3624? |
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| 06:45.21 | trugaa | hola |
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| 07:19.03 | truggas | termination |
| 07:19.54 | KeltusEx | I'm using exten => 85,n,Monitor(wav,/home/admin/test.wav) to record my calls. however, it only records the output sound, and not the input sound... any idea why? |
| 07:21.27 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
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| 07:27.53 | joaovianna | KeltusEx: The sound is recorded in two files in and out, and in the end is mixed. |
| 07:28.09 | KeltusEx | I think I just figured it out |
| 07:28.20 | KeltusEx | the call's dialplan falls through |
| 07:28.25 | KeltusEx | so it finishes recording before I say anything |
| 07:28.33 | KeltusEx | would that cause what I'm hearing? |
| 07:28.35 | joaovianna | KeltusEx: Try to see is sox is not completing... |
| 07:28.56 | KeltusEx | I don't have mixing on |
| 07:28.59 | KeltusEx | here is my script: |
| 07:29.05 | KeltusEx | exten => 85,1,Zapateller(nocallerid) |
| 07:29.05 | KeltusEx | exten => 85,n,Wait(1) |
| 07:29.05 | KeltusEx | exten => 85,n,Monitor(wav,/home/admin/test.wav) |
| 07:29.05 | KeltusEx | exten => 85,n,Background(hello-world) |
| 07:29.05 | KeltusEx | exten => 85,n,StopMonitor() |
| 07:29.05 | KeltusEx | exten => 85,n,Hangup() |
| 07:29.47 | joaovianna | Monitor(wav,/home/admin/test.wav,m) |
| 07:29.50 | KeltusEx | so does it Hangup before it can record what I'm saying? |
| 07:30.02 | joaovianna | and test if you have write access to /home/admin |
| 07:30.12 | KeltusEx | I do |
| 07:30.16 | KeltusEx | the output records fine |
| 07:30.53 | KeltusEx | yeah, it just records the output :( |
| 07:31.01 | joaovianna | in the consele, any errors ? |
| 07:31.01 | KeltusEx | only "hello world" and not what I say on the phone |
| 07:31.25 | joaovianna | in the console, any errors ? |
| 07:31.37 | KeltusEx | lemm see |
| 07:32.21 | joaovianna | I never used stopmonitor, when the channels is done, the record is done too. |
| 07:32.37 | KeltusEx | http://rafb.net/p/Imntwl96.html |
| 07:32.44 | KeltusEx | that's my log |
| 07:32.51 | KeltusEx | it hangsup before I can finish talking... |
| 07:33.55 | joaovianna | try without stopmonitor |
| 07:34.54 | KeltusEx | k |
| 07:35.05 | joaovianna | I never tried with zapateller, I have working with monitor without problems to record a conversation in a bridge call |
| 07:35.09 | KeltusEx | I also get this - |
| 07:35.09 | KeltusEx | [May 5 00:32:24] WARNING[25133]: chan_sip.c:1900 retrans_pkt: Maximum retries exceeded on transmission 44c9c4101cc7b6ef23634f877a90709e@66.54.140.46 for seqno 102 (Critical Response) |
| 07:35.27 | joaovianna | Nat problems |
| 07:36.06 | KeltusEx | still only output is recorded :/ |
| 07:39.38 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 07:39.47 | joaovianna | I think the problem is: You have no sound anyway. Your call is mude. The system could not retransmit the packets back to your system |
| 07:39.51 | joaovianna | Nat problem |
| 07:39.52 | Sargun | Anyone have the BootROM, and SIP software for Polycom IP phones? |
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| 07:51.20 | JT | Sargun: should be on the web |
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| 09:07.07 | phix | hey |
| 09:07.14 | phix | I found out my issue with DTMF |
| 09:07.19 | phix | it was the RTP payload type |
| 09:07.25 | phix | how do I change this? |
| 09:16.16 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 09:16.24 | phix | hmmm |
| 09:16.35 | phix | can this be changed without a recompilation? |
| 09:16.59 | phix | I am using a 1.2.7.1 |
| 09:21.40 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 09:24.53 | phix | hmmmm, ok well get prv msg me with the answer, thanx |
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| 09:50.06 | manopulus | hello folks ;) I want to make iax -- iax over SSH tunnel, I do ssh root@remotehost -L 4570:localhost:4569. in my iax.conf I have register => user:secret@localhost:4570 and... it does not work :( any ideas? |
| 09:52.00 | phix | ummmm, you have to use SSH tunnel? |
| 09:52.06 | phix | can't you setup openvpn or something? |
| 09:53.01 | phix | I hope you have a lot of bandwidth and don't mind laggy conversations, SSH tunnel uses TCP/IP. Setting up openvpn will give you encryption plus it uses UDP. |
| 09:53.18 | phix | better still I think IAX supports encryption, then again I may be wrong. |
| 10:06.41 | manopulus | phix: sorry. was on phone. why I use tunnel, I want to manage our customers, let say, I have to look and manage ~30 asterisk boxes. due of infrastructure, I cannot connect to them from "inside" as they already under VPN etc, I can ssh in through our central |
| 10:07.09 | manopulus | but it is necessary to make tests as I am at their office, and only one way is ssh tunnel |
| 10:11.04 | manopulus | think, found my mistake ;) |
| 10:11.05 | *** part/#asterisk manopulus (n=manopulu@213.197.181.154) |
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| 11:01.15 | key2 | lol |
| 11:01.15 | key2 | it's the second time calvino crashes |
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| 11:19.04 | christo | how do I bind a callback to a 'hangup' eveng in AGI? |
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| 11:58.09 | robin_sz | so, currently I have chan_mISDN and lots of echo ... what other options are there? |
| 11:58.52 | robin_sz | this is with a basic 'cologne' chipset ISDN card |
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| 12:17.44 | JT | bristuff |
| 12:35.01 | phix | ok so back to my question |
| 12:36.23 | phix | is there a way to set the rfc2833 payload type to 96 without recompiling asterisk? (I am using asterisk 1.2.7.1) |
| 12:38.32 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 12:47.10 | robin_sz | sigh ... |
| 12:47.20 | robin_sz | why oh why did I buy a grandstream? |
| 12:47.45 | robin_sz | i think I shall finally trash it ... |
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| 12:49.05 | Strom_M | robin_sz: you were lured in by low low prices |
| 12:49.17 | robin_sz | sadly, yes |
| 12:49.27 | phix | hello |
| 12:49.33 | robin_sz | there are better phones for the same money now |
| 12:49.40 | robin_sz | eg elmeg 290s |
| 12:54.46 | phix | jt |
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| 13:00.08 | Strom_M | catsex |
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| 13:04.12 | Assid | grandstream no good? |
| 13:04.18 | Qwell | Assid: not at all |
| 13:04.23 | Qwell | ~grandstream |
| 13:04.24 | jbot | hmm... grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 13:04.31 | Assid | yugo ? |
| 13:04.40 | jer | i'm having a wierd problem. after upgrading asterisk (i didn't perform the upgrade, but the guy who did is no longer here); voicemail stopped working. i think it's a permissions problem, but not exactly sure which dir to check? |
| 13:04.44 | Qwell | Assid: a car made in the 80s (and probably earlier) |
| 13:04.53 | Assid | hmm |
| 13:04.55 | Qwell | yugo...nowhere |
| 13:05.07 | Strom_M | yugo was a very inexpensive yugoslavian automobile |
| 13:05.14 | Assid | i see |
| 13:05.15 | jer | and by "stopped working" i mean of course, it says that the mailbox is full after playing the initial greeting |
| 13:05.22 | Strom_M | infamous for its low low price and notorious unreliability |
| 13:05.29 | Qwell | jer: is he not there because of the upgrade? |
| 13:05.43 | Qwell | jer: but, turn up verbosity, and watch the console |
| 13:06.42 | jer | interesting |
| 13:06.42 | jer | May 5 09:10:13 ERROR[11904]: app_voicemail.c:2612 leave_voicemail: Unable to create message file: No such file or directory |
| 13:06.59 | Qwell | jer: what version are you on now? |
| 13:07.07 | Qwell | there was a bug that got fixed for that some time back |
| 13:07.13 | jer | Qwell, 1.2.17 |
| 13:07.16 | Qwell | quite some time back |
| 13:07.24 | jer | the guy who upgraded said 1.4 wasn't that stable so wanted to avoid it |
| 13:07.27 | Qwell | it was definitely earlier than that |
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| 13:07.36 | Qwell | jer: well, he's obviously not there for a reason now, eh? |
| 13:07.40 | jer | =] |
| 13:07.45 | jer | yes, but let's not go into that hehe |
| 13:07.47 | Qwell | I wouldn't take the advice of somebody who got fired shortly after making a statement like that |
| 13:07.55 | Assid | this channel has people talking today |
| 13:07.56 | Assid | awesome |
| 13:08.22 | jer | Qwell, well it'son my long term plans to learn asterisk better than i know right now, but i got nominated to fix this asap =/ |
| 13:08.35 | Qwell | ~hafc |
| 13:08.37 | jbot | methinks hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 13:08.53 | Qwell | of course, your problem is probably pretty simple |
| 13:09.07 | Qwell | jer: log in to that users mailbox, and then try leaving a message again |
| 13:09.34 | Qwell | that, or it's a permissions problem... |
| 13:10.16 | *** join/#asterisk kala1 (n=kala@84-50-22-3-dsl.kjj.estpak.ee) |
| 13:10.35 | jer | hrmm forgive the really stupid question, and feel free to tell me to rtfm; but how do i go about doing that when i don't have access to a phone connected behind his gear (or his sip password) ? |
| 13:11.00 | jer | (noting too that it's not just one user, it's all users) |
| 13:11.02 | zcw100 | You can try checking what user and group asterisk is running as and then check the folders under /var/spool/asterisk |
| 13:11.31 | jer | zcw100, those perms are correct, i checked those before i came here |
| 13:12.53 | jer | nm, found the error; was in the users tmp/ dir; somehow it was being owned by the wrong user |
| 13:13.02 | jer | thanks zcw100 & Qwell |
| 13:14.33 | robin_sz | jer: if you have access to the server, you have access to the paswords, see sip.conf for details |
| 13:15.28 | jer | robin_sz, passwords are stored off the server in another db which i have login information for, but not permission to access (id' rather not get myself nix'd too =]) |
| 13:15.44 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 13:16.18 | robin_sz | jer, heh, well, thats what tcpdump is for ;) |
| 13:16.26 | jer | true =] |
| 13:16.48 | jer | well its working now, which is good =] |
| 13:16.48 | robin_sz | and typing sip show peers on the * console? |
| 13:19.51 | robin_sz | so ,,, im trying to get the lights on thses Snoms to show when people are on the phone |
| 13:22.25 | robin_sz | ive done the subscribe and hints thing |
| 13:22.38 | robin_sz | all I have acheived is: |
| 13:22.51 | kala1 | hi does IAX service provider decides how your numbers will be shown to other providers? I have set up asterisk and when I make call to my cell phone I'll get wierd number like 8450223 or 84502223 etc. |
| 13:22.59 | robin_sz | phone 1 in call, phoen 2 no light |
| 13:23.12 | robin_sz | phoen 1 on hook, phone 2 no light |
| 13:23.26 | robin_sz | phoen 1 cable out, phone 2 has light |
| 13:23.50 | robin_sz | not exactly what was wanted, but at least shows some correlation |
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| 13:47.09 | phix | Qwell: g'day |
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| 13:55.19 | deeperror | if i have a function that returns -1 in the dialplan how do I handle this instead of getting extension exited non-zero? |
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| 13:59.42 | blitzrage | deeperror: you don't -- those return values are for Asterisk to handle internally |
| 14:00.10 | deeperror | so if it returns -1 the dialplan just hangsup the call and i lose control? |
| 14:00.45 | deeperror | must be a way to move into the next priority? |
| 14:03.47 | shido6 | 103 ? |
| 14:04.30 | deeperror | ? |
| 14:05.17 | phix | hmmmm |
| 14:08.10 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 14:13.02 | blitzrage | deeperror: many programs return a STATUS variable, like ${DIALSTATUS} |
| 14:13.09 | deeperror | i know |
| 14:13.14 | deeperror | this doesn't for some reason |
| 14:13.21 | deeperror | it just says exited non-zero status |
| 14:13.21 | blitzrage | which application? |
| 14:13.25 | deeperror | then hangs up |
| 14:13.28 | deeperror | nvfaxdetect |
| 14:13.37 | blitzrage | I see |
| 14:14.00 | mvanbaak | me too |
| 14:14.17 | deeperror | everything else works fine, but if the called number answers and does nothing or hangs up it will return -1 and this is killing the call and i lose control |
| 14:14.33 | deeperror | everything else evaluates on dialstatus |
| 14:14.48 | deeperror | this really has nothing to do with faxing its more logic on the extensions.conf |
| 14:15.12 | deeperror | and how to continue on in the dialplan if one of the funcitons returns -1 |
| 14:15.42 | deeperror | i tried priority jumping to extension 104 ? didn't seem to do anything |
| 14:17.16 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 14:18.56 | phix | ok I am ready for an answer now |
| 14:19.12 | Strom_M | the answer is "cheese" |
| 14:19.43 | phix | so the answer to can I set the rfc2833 payload type without recompiling asterisk is cheesE? |
| 14:20.03 | Strom_M | indeed |
| 14:21.01 | phix | I asked a boolean question though, is cheese yes or no? |
| 14:21.32 | Strom_M | cheese is cheese |
| 14:22.07 | phix | hmmm |
| 14:22.08 | phix | you suck :) |
| 14:24.24 | blitzrage | cheese is both yes and no |
| 14:24.50 | blitzrage | phix: and yes, of course you can -- just set it in the <channel>.conf file (i.e. sip.conf, iax.conf) |
| 14:25.36 | phix | blitzrage: great, 1.2.x.x supports that? |
| 14:25.45 | phix | and what is the directive? |
| 14:25.51 | blitzrage | phix: they all support that -- not sure why you thought you'd need to recompile... |
| 14:26.08 | blitzrage | dtmfmode=[info|inband|rfc2833] |
| 14:26.29 | phix | yes but I want to set the rfc2833 payload type to 96 |
| 14:26.49 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:27.22 | blitzrage | oh I see what you mean now |
| 14:27.27 | xheliox | What's Digium's standard warranty on hardware? |
| 14:27.30 | blitzrage | you probably need to do code modifications, yes :) |
| 14:28.11 | phix | blitzrage: :) |
| 14:28.14 | phix | gay |
| 14:40.27 | lesouvage | . |
| 14:40.27 | Corydon76-home | phix: he's not, but I am |
| 14:41.42 | phix | I wasn't calling him gay, I was calling the process of recompile asterisk as being gay. |
| 14:42.25 | lesouvage | When I park a call and th message comes the message comes that it is parked on 701. When I dial 701 in the cli I see the message Rejected connect attempt from 82.xx.xx.xxx, request '701@intern' does not exist. What is going wrong here? |
| 14:43.29 | lesouvage | Is the parking lot part of the default context? |
| 14:43.32 | hads | eggs and hashbrowns, mmm. |
| 14:43.50 | mosty | i find that people who use the word gay as a derogatory term tend to be so far in the closet that the have a coatrack up their arse |
| 14:43.50 | key2 | http://www.youtube.com/watch?v=o1O7ZvDFP_8&NR=1 |
| 14:43.56 | [TK]D-Fender | lesouvage, its wheres you INCLUDE it |
| 14:44.51 | lesouvage | You mean the line: include parkedcalls |
| 14:46.28 | *** part/#asterisk deeperror (n=deeperro@d14-69-154-125.try.wideopenwest.com) |
| 14:47.06 | phix | heh |
| 14:51.42 | lesouvage | I have Executing ParkedCall("IAX2/erik-14", "701") in new stack |
| 14:51.43 | lesouvage | <PROTECTED> |
| 14:51.45 | lesouvage | <PROTECTED> |
| 14:53.15 | *** join/#asterisk _ViperNetworks (n=Nitesh@c-68-41-22-218.hsd1.mi.comcast.net) |
| 14:53.55 | *** join/#asterisk Brijn (n=bas@S010600e0b601c51e.vn.shawcable.net) |
| 14:54.08 | Brijn | Goodmorning all |
| 14:55.17 | Brijn | Just moved to * 1.4 and noticed that VoicemailMain(s${CALLERIDNUM}) no longer dumps me directly in my mailbox. Google / voip-info didn't show any clues to what has changed? |
| 14:59.23 | Strom_M | ${CALLERIDNUM} was deprecated as of 1.2 release |
| 14:59.29 | Strom_M | didn't you read upgrade.txt? :) |
| 14:59.58 | hads | hint: CALLERID(num) |
| 15:00.55 | Brijn | upgrade.txt hmmm, euhhh, let me look :-) I always used voip-info.. But it will probably take a while before it's updated to 1.4 :) |
| 15:01.46 | Brijn | huhh, upgrade.txt is not in the 1.4.4 tarball?? |
| 15:02.11 | Brijn | ls ../doc/u* |
| 15:02.12 | Brijn | ls: ../doc/u*: No such file or directory |
| 15:02.18 | blitzrage | try UPGRADE |
| 15:02.42 | Strom_M | that too |
| 15:02.47 | Brijn | Ahh, not doc, in the root |
| 15:03.51 | lesouvage | I have an error message at the bottom of idefisk (iax2 softphone) "Resyncing the jb. last_delay 0, this delay when I call into a parked call. Any idea? |
| 15:04.07 | *** join/#asterisk Fieldy (i=FGkjrlrq@gentoo/contributor/Fieldy) |
| 15:11.49 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
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| 15:19.58 | Brijn | Another newbie question.. Voicemail will no longer support prefixes for option. Show app says I should : VoiceMail(mailbox[@context][&mailbox[@context]][...][|options] |
| 15:20.22 | Brijn | How do I add the option: Voicemail(${ARG1}|b) ? |
| 15:20.55 | blitzrage | ${ARG1} should contain mailbox@context |
| 15:22.27 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:22.46 | _ViperNetworks | Hello All.. |
| 15:23.23 | Brijn | blitzrage: I had b${ARG1}.. But it warned me I shoudl get rid of the b in front.. Will it be ${ARG1}|b now, or ${ARG1},1, or .. |
| 15:23.27 | _ViperNetworks | Anyone know how I can make an ISO image of my working Asterisk System and install on other system without any problems..? |
| 15:24.04 | blitzrage | Brijn: you just had it -- Voicemail(${ARG1}|b) is correct form |
| 15:24.13 | blitzrage | its an option just like other applications |
| 15:24.21 | blitzrage | it was moved to be more consistant with the other dialplan applications |
| 15:24.25 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 15:24.58 | Brijn | Has been a while since i did my initial extentions.conf.. It's so realiable you never have to touch it :) |
| 15:25.23 | blitzrage | heh |
| 15:25.27 | Brijn | But moved to a new house, so will be adding an ATA or two.. Do wireless ATA's exist at all? |
| 15:25.31 | blitzrage | I built our entire new platform on 1.4 |
| 15:25.47 | blitzrage | about 6-8 mths in the making |
| 15:25.55 | remmo | hmmm |
| 15:25.55 | blitzrage | which I gotta go work on now... |
| 15:29.18 | phix | blitzrage: nice |
| 15:29.39 | mosty | _ViperNetworks: it's not really possible unless you have identical hardware on both machines |
| 15:29.50 | blitzrage | found a lot of bugs, got them fixed, and now its much more reliable due to my testing :) |
| 15:30.27 | blitzrage | _ViperNetworks: like mosty said, the hardware needs to be identical for the most part -- you're probably looking for something like Ghost |
| 15:31.17 | mosty | ghost requires windows though. dd will create a raw disk image, but won't compress it or anything fancy |
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| 15:36.04 | _ViperNetworks | hmm ok... well hardware is abit different |
| 15:36.37 | _ViperNetworks | I think may be if i 'tar' the whole system and extract it onto other.. |
| 15:37.16 | Strom_M | why not just build the new system and copy over only the relevant asterisk configs? |
| 15:38.28 | _ViperNetworks | I want to keep the installation to minimal... I dont want to keep on recompiling the Asterisk and Zaptel drivers... I have to configure 5 Systems... |
| 15:38.58 | mosty | _ViperNetworks: just use a distribution with binary packages, no need to compile anything |
| 15:39.10 | Strom_M | you're going to spend so much time trying to save time that in the end you'll take longer than you would have otherwise :) |
| 15:39.45 | _ViperNetworks | hmm OK |
| 15:41.42 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 15:42.14 | coolbeans | Greets. Anyone use the Polycom IP 320 yet? Does it use the same firmware as the 301's, 501's, 601's? |
| 15:45.23 | VJFROMGT | i need to pass call to my a2billing, does anyone know how to do such? |
| 15:45.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:47.56 | *** join/#asterisk Waverly360 (n=mirc@209.149.58.214) |
| 15:48.12 | *** join/#asterisk brea (n=brea@c-76-23-9-101.hsd1.ut.comcast.net) |
| 15:49.08 | brea | Anyone have a recommendation for IAX origination providers... one that'll do a T1 worth of concurrent calls? |
| 15:49.21 | coolbeans | Use a SIP provider that supports re-invites. |
| 15:49.57 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 15:50.19 | brea | ok |
| 15:51.13 | Brijn | is there a way to show the bandwidth used by a call? |
| 15:51.57 | *** join/#asterisk lwh (n=lwh192@rdsl-0313.tor.pathcom.com) |
| 15:52.56 | *** join/#asterisk [Mr_X] (n=mrx@88.118.89.208) |
| 15:53.55 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-175-68-176.socal.res.rr.com) |
| 15:55.53 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 15:58.50 | *** part/#asterisk bkw__ (i=brian@ppp-70-128-119-106.dsl.tulsok.swbell.net) |
| 16:06.32 | *** join/#asterisk wolferine (n=profx@unaffiliated/wolferine) |
| 16:19.52 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:20.47 | rue_mohr | so asterisk dosn't understand the hook switch signaling from my mainstreet 3624 channelbank... |
| 16:21.46 | *** join/#asterisk ctaloi (n=ctaloi@pool-71-176-69-82.syrcny.fios.verizon.net) |
| 16:21.57 | *** join/#asterisk X-Gen (n=x-gen@dsl-242-1-254.telkomadsl.co.za) |
| 16:22.19 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 16:22.23 | christo | how do I bind a callback to a 'hangup' eveng in AGI? It seems so obvious in the docs, but with the latest version of asterisk, the Asterisk::AGI just doesn't seem to capture a hangup, so I can't cleanup at the end of the process |
| 16:22.54 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 16:24.41 | rue_mohr | <PROTECTED> |
| 16:25.59 | christo | it should be something along the lines of $agi->setcallback( sub{ <do something here> } ); but I find the call just gets hung up and that's it |
| 16:27.59 | rue_mohr | what I know is that I have to find a modify a state machine |
| 16:30.11 | Juggie | christo. you have to tell your perl to ignore the SIGHUP |
| 16:30.16 | Juggie | or it will never see the hangup |
| 16:30.44 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 16:31.42 | *** join/#asterisk Humblgrumpf (n=henkelmi@p54B0BCD5.dip0.t-ipconnect.de) |
| 16:31.53 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 16:34.27 | *** join/#asterisk Humblgrumpf (n=henkelmi@p54B0BCD5.dip0.t-ipconnect.de) |
| 16:34.54 | *** join/#asterisk Humblgrumpf (n=henkelmi@p54B0BCD5.dip0.t-ipconnect.de) |
| 16:36.05 | *** join/#asterisk menfin (n=cray@AMontpellier-152-1-24-179.w81-251.abo.wanadoo.fr) |
| 16:38.15 | *** join/#asterisk Humblgrumpf (n=humblgru@p54B0BCD5.dip0.t-ipconnect.de) |
| 16:38.28 | rue_mohr | <PROTECTED> |
| 16:38.38 | rue_mohr | ok, so I'm looking at the function |
| 16:38.59 | rue_mohr | ast = asterisk state table? |
| 16:39.08 | christo | Juggie - do you mean like this?: $SIG{HUP} = sub {}; |
| 16:41.24 | dlynes_laptop | [TK]D-Fender: btw...that fix for peers working in both directions was applied to 1.2.16 and later |
| 16:41.37 | dlynes_laptop | [TK]D-Fender: anything previous to 1.2.16 didn't have that behaviour |
| 16:41.42 | dlynes_laptop | [TK]D-Fender: just thought you'd like to know |
| 16:42.06 | dlynes_laptop | [TK]D-Fender: olle added that fix in feb 13th |
| 16:42.34 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:44.38 | *** join/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 16:44.41 | piper69 | hi all |
| 16:45.10 | rue_mohr | <PROTECTED> |
| 16:45.55 | rue_mohr | Ring/Off-hook in strange state AST_STATE_UP on channel 2 |
| 16:46.01 | rue_mohr | erm |
| 16:46.26 | piper69 | i have linksys RT31P2 and i have a free account with sipnumbers and i am able to reciver incoming call, i want to try asterisk |
| 16:46.31 | rue_mohr | hmm |
| 16:47.04 | rue_mohr | so its seeing what it thinks is either ringing or an offhook when it knows the phone is already off hook |
| 16:47.04 | *** join/#asterisk Humblgrumpf (n=humblgru@p54B0BCD5.dip0.t-ipconnect.de) |
| 16:47.33 | christo | so it seems the callback handler in AGI just doesn't work any more - Juggie, defining a custom SIG{HUP} handler works as an alternative. thanks! |
| 16:48.00 | piper69 | just want to test asterisk for the hell of i t |
| 16:48.02 | piper69 | lol |
| 16:48.07 | *** part/#asterisk Humblgrumpf (n=humblgru@p54B0BCD5.dip0.t-ipconnect.de) |
| 16:48.29 | rue_mohr | piper69, is that like having a channelbank in hte house for the hell of it ? :) |
| 16:48.41 | rue_mohr | if anything it should make a good answering machine |
| 16:49.10 | rue_mohr | pending I get past teaching asterisk the new states from my channelbank |
| 16:49.12 | piper69 | rue_mohr: what do you think |
| 16:49.14 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:49.24 | rue_mohr | go for it |
| 16:49.47 | piper69 | i think it will be a good experniess |
| 16:49.59 | rue_mohr | if anything its a learning experiance, and thats worth a lot of money (or schools seem to give that impression) |
| 16:50.16 | rue_mohr | I just need to keep my brain busy |
| 16:50.29 | rue_mohr | and I want everyone in the house to have their own extension |
| 16:50.46 | piper69 | rue_mohr: hahahhaahah |
| 16:50.58 | piper69 | rue_mohr: but i don't know where to staet |
| 16:51.01 | piper69 | *start |
| 16:51.12 | piper69 | well let me take that back, |
| 16:51.32 | piper69 | i have my beer infrot of me , so i guess this is step one |
| 16:52.28 | piper69 | sevard: ?? |
| 16:54.05 | piper69 | anyone have asterisk running in debian 4.0 |
| 16:56.48 | piper69 | what is the diffrenet between Asterisk 1.2.18 and 1.4.4 |
| 16:57.42 | christo | piper69 - the key differences will be in the upgrade.txt file |
| 16:58.09 | christo | as far as I can see there's a whole bunch of broken stuff in 1.4.4 |
| 16:59.40 | jer | is it possible to mix g.729(pass-thru) and g.711 ulaw voicemail clients? documentation i'm reading doesn't seem to support that theory |
| 16:59.44 | *** join/#asterisk alexpe (n=alex@cev75-1-81-57-14-91.fbx.proxad.net) |
| 17:01.37 | *** part/#asterisk X-Gen (n=x-gen@dsl-242-1-254.telkomadsl.co.za) |
| 17:03.06 | dc3aes | btw.. canadian LES.NET is a pile of crap.. I have been calling and emailing them for WEEKS and they only responded to one e-mail when I wrote it all in caps lock, and the response was a question that I had addressed in a previous e-mail |
| 17:05.05 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 17:08.18 | blitzrage | jer: as soon as you change audio between codecs, you have to transcode -- that requires a license |
| 17:08.45 | jer | right, and we're willing to purchase the required number of licenses, but i want to make sure things will work |
| 17:08.55 | rue_mohr | I wonder if there is a debug level that will tell me all the changes in channel states |
| 17:09.06 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:09.17 | jer | (pass-thru would just be for testing) |
| 17:09.22 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:10.44 | blitzrage | jer: you can't "passthrough" from one codec to another |
| 17:10.51 | jer | ok |
| 17:10.56 | blitzrage | that is called transcoding |
| 17:11.01 | jer | so you basically have to buy a license before you can test? |
| 17:11.09 | blitzrage | anytime you change the codec from one to another, you're transcoding, not passthrough |
| 17:11.25 | blitzrage | to change from ulaw to g.729, you have to have a license |
| 17:11.44 | blitzrage | passthrough means g.729->g.729, or ulaw->ulaw |
| 17:11.49 | jer | ah |
| 17:11.59 | blitzrage | ulaw->g729 or g729->ulaw is called transcoding |
| 17:12.10 | jer | yes i got that the first time you said it |
| 17:12.15 | blitzrage | it didn't seem like it |
| 17:13.14 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 17:20.02 | *** join/#asterisk brussel_ (n=brussel@cpe-72-130-172-213.san.res.rr.com) |
| 17:21.45 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581822.dsl.bell.ca) |
| 17:25.14 | LeddyHM | damn nufone |
| 17:25.22 | LeddyHM | still waiting for my "credit" to clear |
| 17:25.26 | Assid | err.. anyone know any of those unltd US/Europe providers? for monthly credit ? |
| 17:26.08 | rue_mohr | what eh highest asterisk debug level? |
| 17:26.51 | LeddyHM | how many 0's can you type? |
| 17:26.55 | LeddyHM | ;) |
| 17:27.20 | rue_mohr | :) |
| 17:27.36 | blitzrage | nothing above 4 is useful |
| 17:27.43 | LeddyHM | no idea, I'm just here for comic relief |
| 17:27.52 | rue_mohr | interesting, I connected two new circuits to the t1 in the channelbank and ... someone was rinign them |
| 17:28.02 | SwK | I <3 when people set debug to 9999 |
| 17:28.09 | blitzrage | ya |
| 17:28.12 | blitzrage | totally useless |
| 17:28.17 | blitzrage | set debug 4 == set debug 100000000000000000000000 |
| 17:28.35 | rue_mohr | need to know what states asterisk is recieving from this channelbank |
| 17:28.46 | LeddyHM | speaking of, how do you set the default verbose? is it in a config file? |
| 17:29.00 | blitzrage | you set it by how you start Asterisk |
| 17:29.05 | SwK | you can set it when you start asterisk |
| 17:29.19 | rue_mohr | :??? why is this phone ringing |
| 17:29.26 | SwK | hah |
| 17:29.26 | rue_mohr | I see nothing on the terminal |
| 17:29.28 | SwK | NACK |
| 17:29.31 | LeddyHM | the -vvvvv = v * x? |
| 17:29.43 | SwK | -vvv = set verbose 3 |
| 17:29.50 | LeddyHM | that's what I was thinking |
| 17:29.58 | SwK | -ddd == set debug 3 |
| 17:30.00 | blitzrage | SwK: any idea how to pass variable information across an IAX2 connection in 1.4? (might not be possible) |
| 17:30.02 | LeddyHM | cool |
| 17:30.19 | blitzrage | I see IAXVARS() function, but looks to be only in trunk |
| 17:30.59 | SwK | blitz: hmmmm you could dp something like (psuedo) dial(iax/foo/18005551212-var_val-var_cal |
| 17:31.01 | SwK | heh |
| 17:31.17 | blitzrage | hrmmmm |
| 17:31.20 | *** join/#asterisk Blackthorn (n=support@w-l4.smyth.net) |
| 17:31.25 | blitzrage | yes I could.... |
| 17:31.27 | Blackthorn | Hi, I have a Ubuntu 6.06 LTS system that I want to install asterisk 1.4.4. I have attempted to use several documents on voipinfo.org but I still can not get it installed. When issuing make install I get "The configure script must be executed before running 'make'." |
| 17:31.30 | blitzrage | I hadn't thought of that |
| 17:31.34 | SwK | or just use SIP and add additional headers |
| 17:31.45 | blitzrage | yah, thats what I was thinking of doing -- but I'd rather keep it all IAX |
| 17:31.52 | blitzrage | I think what you propose might just work |
| 17:32.28 | pipwerk | Blackthorn: the commands are: ./configure; make; make install |
| 17:32.52 | blitzrage | ./configure ; make menuselect ; make install is what I use |
| 17:33.27 | Blackthorn | :) compliing |
| 17:33.30 | pipwerk | blitzrage: I guess make install will do |
| 17:33.46 | pipwerk | make menuselect is optional I'd say :) |
| 17:34.41 | Blackthorn | what does menuselect do? |
| 17:34.42 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 17:34.48 | MrWup | hey guys |
| 17:34.57 | MrWup | how do i check if a variable is set, and only set it if it is not set? |
| 17:35.06 | pipwerk | gives you the opperunity to select certain options |
| 17:35.31 | LeddyHM | what's the "g" and "c" mean for args? |
| 17:36.06 | blitzrage | MrWup: ${ISNULL(${MY_VARIABLE})} |
| 17:36.58 | Blackthorn | where docs that say to use ./configure btw? I had been looking for several days to figure out how to do this... just curious as to where i missed that instruction? |
| 17:37.00 | *** join/#asterisk saftsack (n=saftsack@pD9E0612C.dip.t-dialin.net) |
| 17:37.03 | blitzrage | Exec(${IF($[${ISNULL(${SOME_VALUE})}]?Set(MY_VARIABLE=${SOME_VALUE}):NoOp())}) |
| 17:37.56 | blitzrage | that'll Set() a variable (MY_VARIABLE) if ${SOME_VALUE} is NULL (not set), else, Execute NoOp() |
| 17:38.23 | *** join/#asterisk asdx (n=Desarrol@200.26.177.4) |
| 17:38.30 | blitzrage | You could replace ${SOME_VALUE} with the same name as the variable you wanna check if its set or not |
| 17:39.03 | pipwerk | Blackthorn: in the UPGRADE.txt |
| 17:39.18 | pipwerk | maybe it should be in the readme as well |
| 17:39.42 | sevard | that's the ugliest bit of code I've ever seen |
| 17:40.01 | MrWup | blitzrage, thanks. but how do i do the sort of if isnul then setit? |
| 17:40.04 | blitzrage | sevard: its very functional -- I use stuff like that all the time, and its not bad to read if you understand it |
| 17:40.17 | blitzrage | MrWup: I don't know what you just asked -- try again |
| 17:40.22 | sevard | blitzrage: it just makes my eyes bleed to read, I understand it just fine. |
| 17:40.33 | blitzrage | sevard: then you don't wanna see my dialplan :) |
| 17:40.38 | sevard | heh |
| 17:40.39 | *** join/#asterisk alrs_ (i=nobody@cpe-76-81-15-170.socal.res.rr.com) |
| 17:40.53 | MrWup | blitzrage, sorry. what i mean is ${ISNUL()} will return true or false right? but whats the code to check what it returned and then do something based on it? |
| 17:41.08 | MrWup | like in c youd write if (isnull() == true) {} |
| 17:41.12 | blitzrage | MrWup: look at my 2nd example -- ${ISNULL()} returns 0 or 1 |
| 17:41.14 | MrWup | BUT OBVIOUSLY IN THE DIALPLAN... |
| 17:41.22 | MrWup | oops |
| 17:41.24 | MrWup | sorry caps |
| 17:41.31 | MrWup | ohhh |
| 17:41.32 | MrWup | thanks |
| 17:41.35 | MrWup | sorry didnt see second example |
| 17:42.02 | blitzrage | Exec(${IF($[${ISNULL(...)}]?return_this_text_if_true:return_this_text_if_false)}) |
| 17:42.17 | MrWup | thanks :) |
| 17:42.30 | blitzrage | then replace the appropriate text with what you want to execute |
| 17:42.39 | MrWup | right |
| 17:42.48 | blitzrage | there are various ways of doing it on multiple lines -- but that works all on a single line |
| 17:43.00 | MrWup | thanks |
| 17:43.46 | MrWup | Exec(${IF($[${ISNULL(${__TEST})}]?Set(__TEST=${UNIQUEID}):NoOp())}) |
| 17:43.49 | blitzrage | I've written enough /ugly/ lines of dialplan code with multiple embedded dialplan functions that tracking the brackets isn't really that big a deal anymore |
| 17:44.02 | MrWup | so this will check if __test is set, and if not... will set TEST for this channel and every child channel? |
| 17:44.07 | blitzrage | you shouldn't need the __ in the test |
| 17:44.17 | blitzrage | all that says is that its transitive to other channels |
| 17:44.31 | blitzrage | but you got the idea |
| 17:44.32 | MrWup | id only need the __ when i set TEST right? |
| 17:44.35 | blitzrage | yes |
| 17:44.37 | MrWup | ok |
| 17:44.51 | MrWup | thanks |
| 17:46.10 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 17:46.45 | *** join/#asterisk connecta (n=Administ@76.23.188.72.cfl.res.rr.com) |
| 17:47.18 | connecta | when asterisk is playing back an IVR, Ii notice the call drops if a sound file is not found . Is there a way to make asterisk jump to the next step without failing the call? |
| 17:49.01 | MrWup | blitzrage, what i'm trying to do is: SIP1 calls SIP2, SIP2 calls SIP3 and transfers SIP1 to SIP3... and i need to link all those events in the database- so that later on i can see, SIP1 called SIP2 who then transfered to SIP3 |
| 17:49.19 | MrWup | so obviously i need some sort of variable which is common throughout the call path |
| 17:49.19 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 17:49.31 | blitzrage | I don't do CDR :) |
| 17:49.33 | MrWup | common to the SIP1->SIP2 call, then SIP2->SIP3 call and the SIP1->SIP3 call |
| 17:49.38 | MrWup | ah |
| 17:49.43 | blitzrage | for that very reason |
| 17:49.50 | connecta | MrWup, I don't know how to do it, and I don't know your question, but i do know this can be done and is common . so stick with it |
| 17:50.18 | MrWup | and i thought that setting __TEST might set the uniqueID of SIP1->SIP2 to all the SIP2->SIP3 and SIP1->SIP3 calls as well |
| 17:50.49 | blitzrage | connecta: that was handy.... |
| 17:50.51 | MrWup | so when i finally reach that point in the dialplan where im dealing with the SIP1->SIP3 call... i can see, ah, this call came from x |
| 17:51.32 | blitzrage | jbot: cookbook is The Asterisk Cookbook, the next offering from O'Reilly Media that you can help write! The Asterisk Cookbook is truly a community effort. Get involved today at http://asteriskcookbook.com/wiki/index.php/Main_Page |
| 17:51.37 | jbot | ...but cookbook is already something else... |
| 17:51.39 | connecta | MrWup, i'm postiive asterisk makes a unique id for each call, i wish i had messed uwith it. to be honest it shouldnt be too hard |
| 17:51.41 | blitzrage | ~cookbook |
| 17:51.44 | jbot | somebody said cookbook was "Over 1,500 time-saving recipes and hints for busy modern computer users" at http://dsl.org/cookbook/ |
| 17:52.04 | blitzrage | jbot: ast_cookbook is The Asterisk Cookbook, the next offering from O'Reilly Media that you can help write! The Asterisk Cookbook is truly a community effort. Get involved today at http://asteriskcookbook.com/wiki/index.php/Main_Page |
| 17:52.22 | jbot | okay, blitzrage |
| 17:52.23 | MrWup | connecta, yeah it does create a uniqueID for each call. thing is, when SIP1->SIP2 and SIP2 transfers SIP1 to SIP3... three calls have happen |
| 17:52.32 | blitzrage | when you figure it out -- write a recipe! :) |
| 17:52.40 | MrWup | first when SIP1 calls SIP2, then when SIP2 calls SIP3 to do the attended transfer and finally when SIP1 is talking to SIP3 |
| 17:52.42 | MrWup | so thats three calls |
| 17:52.47 | MrWup | all with 3 different unique ids |
| 17:52.58 | MrWup | so in a database how on earth would you tell they were all part of a transfer |
| 17:53.15 | MrWup | unless there is some variable common to all three calls |
| 17:53.28 | MrWup | so u can identify that they all formed part of the same call path from SIP1->SIP3 |
| 17:54.58 | Blackthorn | Thanks for the help i'm now got asterisk up and running the ubento box. One more question, I followed some instructions on voipinfo had me setup kernal headers and udev.rules.. was that nessary to get this isntall working? |
| 17:54.59 | connecta | MrWup, hrmn, that kinda sucks. Have you looked into setting your own variable for the call? |
| 17:55.08 | Blackthorn | http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu |
| 17:55.36 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 17:56.05 | MrWup | connecta, indeed. thats what i was attempting to do above. so when SIP1 first calls SIP2, i do set __TEST, and then hopefully using the __ will make test apply not only to the SIP1->SIP2 call, but also to the SIP2->SIP3 call and also to the finally resulting SIP1->SIP3 call when the transfer is complete |
| 17:56.22 | MrWup | unfortunately setting __TEST in the first call doesnt seem to mean that __TEST appears as set in the second two calls |
| 17:56.55 | MrWup | so im back to square one of trying to think how those three calls (which basically from the users perspective all form 1 call path) can be linked via some variable |
| 17:56.55 | connecta | hrmn, tricky... nice try though |
| 17:56.58 | MrWup | yeah |
| 17:57.12 | connecta | how about incorporating the callerid? |
| 17:57.35 | dc3aes | I have been fighting with LES.NET over phone and email for weeks.. I needed a DID removed that costs $15 per month... I requested it to be deleted multiple times over 2 months and they kept saying it would be done.. now my account id deactivated because of balance.. my last reply from them is "your system doesnt work because your balance is negative"... duuuuuhhhh grrr canadians take heed, we need another alternative :) |
| 17:58.54 | andrew` | link2voip.com has worked well for me so far dc3aes...I have a (604) with them |
| 17:59.01 | MrWup | connecta, that would not work in a database, as i could have millions of records with the same callerID where SIP1 has been transfered to SIP5 SIP6 SIP7 etc |
| 17:59.04 | dc3aes | i will check them out thanks :) |
| 18:00.13 | LeddyHM | Anyone use nufone? I keep getting "declined" even though I am properly registered (according to their page) |
| 18:02.34 | dc3aes | me too |
| 18:02.45 | dc3aes | i was getting "congested" all last ngiht |
| 18:02.49 | *** join/#asterisk alrs_ (i=nobody@cpe-76-81-15-170.socal.res.rr.com) |
| 18:02.50 | dc3aes | and their CID works 30% of the time |
| 18:02.54 | dc3aes | at least in my area |
| 18:03.54 | LeddyHM | yeah I get that after declined |
| 18:03.58 | dc3aes | [May 5 11:02:49] NOTICE[16885]: chan_iax2.c:2690 __auto_congest: Auto-congesting call due to slow response |
| 18:03.59 | dc3aes | <PROTECTED> |
| 18:04.20 | dc3aes | thats what ive been getting since yesterday... i mean ive been configuring a lot its possible I screwed something up.. |
| 18:04.47 | blitzrage | s/possible/probable |
| 18:05.03 | LeddyHM | <PROTECTED> |
| 18:05.03 | LeddyHM | <PROTECTED> |
| 18:05.03 | LeddyHM | <PROTECTED> |
| 18:05.43 | LeddyHM | I may have improperly configured it, I'm just using them to test |
| 18:05.50 | dc3aes | ok LES.NET finally fixed my account.. ONLY after I freaked out in an email and threatened to post the entire 'saga' on my blog + in #asterisk |
| 18:06.04 | LeddyHM | I just replaced my regular provider name with the nufone entry |
| 18:06.26 | dc3aes | I also made sure I emailed through my PI firms e-mail system to add the signature lol |
| 18:06.48 | dc3aes | well im getting identical error ... hmmm |
| 18:08.35 | LeddyHM | I wonder who else uses it |
| 18:08.43 | LeddyHM | dc3: you feel like callin? ;) |
| 18:08.58 | LeddyHM | I had to just call to get my payment pushed through so I had credit |
| 18:09.42 | dc3aes | heeh i dont know.. im only using nufone [May 5 11:02:49] NOTICE[16885]: chan_iax2.c:2690 __auto_congest: Auto-congesting call due to slow response |
| 18:09.42 | dc3aes | <PROTECTED> |
| 18:09.46 | dc3aes | soorrrryyyy |
| 18:09.55 | LeddyHM | me too |
| 18:10.03 | dc3aes | apparently the insert key is in a different spot on this keyboard |
| 18:10.11 | dc3aes | I have $10 credit, but its not going down :) |
| 18:10.14 | dc3aes | I shouldnt brag about that |
| 18:10.22 | dc3aes | but ive made a toooon of calls and the $10 is still $10.00 |
| 18:10.27 | LeddyHM | nice |
| 18:10.51 | dc3aes | hmmmm |
| 18:10.59 | LeddyHM | I'm just trying to debug dtmf problems with our current provider |
| 18:11.14 | dc3aes | ive basically found through experience + online postings .. to not use nufone for anything but CIDSPOOFing |
| 18:11.30 | blitzrage | *coughtrollcoughcough* |
| 18:11.35 | dc3aes | hahaahah |
| 18:11.40 | LeddyHM | heh |
| 18:11.56 | dc3aes | i guess that did sound trolly |
| 18:12.12 | dc3aes | whatever.. im not happy with it lol |
| 18:13.28 | LeddyHM | jsolares recommended them as a pay as you go sip provider |
| 18:13.33 | LeddyHM | I wanna GO! ;) |
| 18:14.30 | dc3aes | hehe.. I will tinker on this some more and make sure my prob isnt configuration but it really isnt working.. if anyone else has a nufone trunk they could confirm is working.. that would be much appreciated |
| 18:15.05 | LeddyHM | I'm switching back to the working pbx |
| 18:15.12 | LeddyHM | gonna give them a call |
| 18:15.25 | LeddyHM | working pbx != nufone ;) |
| 18:16.33 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 18:16.55 | ghenry | how often should you stop and start * , even if you've not changed/added anything? |
| 18:17.24 | blitzrage | never |
| 18:17.32 | blitzrage | you shouldn't have to restart |
| 18:17.50 | ghenry | ok, so a lot of old docs around then. |
| 18:17.53 | ghenry | that's what I thought |
| 18:17.53 | blitzrage | yes |
| 18:17.56 | blitzrage | VERY old docs |
| 18:18.00 | ghenry | must be for 1.0.xxx |
| 18:18.07 | blitzrage | more like 0.9 |
| 18:18.22 | ghenry | phew ;-) |
| 18:18.23 | ghenry | I was thinking how to explain that ;-) |
| 18:18.28 | LeddyHM | hrm, doesn't say tech staff is 24x7 |
| 18:18.32 | blitzrage | I run around 500+ simultaneous calls on one box and never restart :) |
| 18:18.39 | ghenry | cool |
| 18:18.52 | ghenry | just going to be setting up a system for 50 users |
| 18:18.57 | blitzrage | w00t! IAX2 registration point built and working for origination -- time to go biking! |
| 18:18.59 | ghenry | then one next month for 100 |
| 18:19.03 | blitzrage | ghenry: you should have no problems |
| 18:19.09 | blitzrage | (well... with the restarting part :)) |
| 18:19.15 | ghenry | not many concurrent calls |
| 18:19.19 | ghenry | ;-) |
| 18:19.40 | ghenry | concurrent calls are really the key for specing ram/cpu etc. as I understand |
| 18:19.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 18:19.50 | ghenry | and codet conversions etc. |
| 18:19.53 | ghenry | codec |
| 18:20.17 | blitzrage | ghenry: correct |
| 18:20.34 | blitzrage | how many simultaneous calls do you think? 50 people... probably like 10 avg, 30-40 spike |
| 18:20.40 | ghenry | we got a 2gb ram, with one decent cpu for the 50 users |
| 18:20.47 | ghenry | they have 10 lines out |
| 18:20.55 | blitzrage | the RAM isn't really that big of an issue in Asterisk |
| 18:20.57 | ghenry | and calss interal |
| 18:21.00 | blitzrage | its more a CPU intensive application |
| 18:21.01 | ghenry | calls |
| 18:21.06 | ghenry | yup |
| 18:21.09 | ghenry | brb |
| 18:21.15 | blitzrage | gotcha... you should probably be good with any newer P4 or a Xeon |
| 18:21.21 | blitzrage | I'm going biking for a couple hours -- peas! |
| 18:21.41 | dc3aes | yay now i can troll :P |
| 18:22.22 | robin_sz | sigh ... chan_mISDN ... how hard is DTMF detection ffs |
| 18:22.32 | ghenry | bye |
| 18:22.56 | asdx | is the asterisk appliance already available? |
| 18:23.28 | ghenry | aye |
| 18:23.36 | asdx | nice |
| 18:23.45 | ghenry | ages now |
| 18:23.51 | ghenry | digium one, yeah? |
| 18:23.56 | asdx | yeah |
| 18:24.23 | asdx | how much cost? |
| 18:24.42 | ghenry | a good chuck of cash |
| 18:24.44 | ghenry | check the site |
| 18:24.48 | ghenry | might have come down |
| 18:24.56 | asdx | k |
| 18:25.55 | ghenry | blitzrage: getting 40 9112i aastras and 10 9133i |
| 18:26.10 | etfonhomey | ghenry, why not Polycoms? |
| 18:26.29 | ghenry | cheapness, and bang for buck |
| 18:26.43 | ghenry | good feature sent and good quality |
| 18:26.49 | ghenry | not "great", but good |
| 18:26.51 | LeddyHM | mmm polycom |
| 18:26.59 | ghenry | we like them and know them |
| 18:27.07 | ghenry | not had a complaint yet.... |
| 18:27.09 | etfonhomey | How much are the two asstra models? |
| 18:27.18 | SwK | polycom > aastra |
| 18:27.23 | ghenry | £59 and £87~ |
| 18:27.28 | ghenry | yeah |
| 18:27.43 | ghenry | not tried they just released range yet |
| 18:27.48 | etfonhomey | The we're at $2 / british pound now, right? |
| 18:27.50 | ghenry | didn't want to deploy them |
| 18:28.01 | ghenry | about that yeah |
| 18:28.14 | etfonhomey | Polycoms are about the same price. |
| 18:28.22 | ghenry | we get cheaper with reseller accounts etc |
| 18:28.26 | ghenry | are they? |
| 18:28.27 | SwK | the thing I dont like about the aastra's is I never could get them to work properly thru nat |
| 18:28.29 | etfonhomey | IP 330 2 line phone $129 |
| 18:28.45 | ghenry | will have to try some etfonhomey |
| 18:28.58 | ghenry | do some more r&d soon |
| 18:29.16 | ghenry | but like I said, we have experience with the aastras, so a bit late for these 2 projects to switch ;-) |
| 18:29.29 | ghenry | I hear nothing but good about them |
| 18:29.36 | ghenry | you can do blf on them right? |
| 18:29.37 | etfonhomey | ghenry, I got lucky and just tried the Polycom's first. I'm extremely happy with all of them, except the 301, but it's OLD. |
| 18:29.44 | ghenry | aye |
| 18:29.52 | ghenry | they got a nice web gui too ? |
| 18:29.53 | etfonhomey | ghenry, yes, you can do BLF |
| 18:30.04 | SwK | the 301 has its uses |
| 18:30.08 | etfonhomey | ghenry, the preferred method is centralized provisioning. |
| 18:30.08 | ghenry | I think they are probably the best supported? |
| 18:30.17 | SwK | but the 430 and higher models are much better |
| 18:30.23 | SwK | I'm wiaitng on my 550 to get here |
| 18:30.26 | ghenry | since digium partnered |
| 18:30.39 | etfonhomey | 430 is great. The only negative I have about the 301 is that you can't do presence on a line key. |
| 18:31.03 | etfonhomey | I have experience with the 301, 430 and 501. |
| 18:31.32 | ghenry | etfonhomey: centralised? |
| 18:32.00 | ghenry | you mean tftp and dhcp configs etc |
| 18:32.00 | etfonhomey | keep your configs on an ftp server, and configure the phones using DHCP. |
| 18:32.30 | SwK | screw tftp or ftp w/ polycoms... you use https and be done with it :) |
| 18:32.37 | SwK | much safer and easier to deal with |
| 18:32.40 | etfonhomey | It's very clean. However, you can easily configure a regular SIP extension via the web interface. |
| 18:32.48 | ghenry | aye |
| 18:32.53 | ghenry | we do it via tftp |
| 18:33.00 | ghenry | need to move over to https/ftp etc |
| 18:33.14 | SwK | i still u se tftp on lans all the time |
| 18:33.16 | etfonhomey | I'm still on tftp as well. |
| 18:33.21 | ghenry | yeah |
| 18:33.58 | etfonhomey | On here, I've heard that the preferred method for the polycoms is ftp, but different people here have different opinions. |
| 18:34.00 | connecta | yah i'm also excited to try th 330 |
| 18:34.02 | ghenry | anyone know how to use postgres for realtime on 1.2.xx |
| 18:34.15 | ghenry | not with unixodbc |
| 18:34.43 | ghenry | I was looking at which version from svn got into 1.4 and use that version from svn |
| 18:34.56 | connecta | etfonhomey, i'll tell you the nicest thing about the polycoms (for mass provisioning) is that they're preconfigured to pull from ftp. All you have to do is create the polycom account on an ftp server and put the files in the home directory |
| 18:35.09 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 18:35.19 | sasch | anyone use grandstream gpx 2000 ?? |
| 18:35.25 | sasch | i have a problem with hint ... |
| 18:35.26 | connecta | from that point forward, you can pull a phone out of the box, get the mac, and create the configs in that folder |
| 18:35.33 | connecta | sorry sasch, not me |
| 18:35.33 | etfonhomey | connecta, yeah, that way you don't have to touch each one you deploy. I've never done a very large installation. So, it hasn't been a problem to touch each one to change to tftp. |
| 18:35.37 | ghenry | connecta: same as aastra ;-) |
| 18:35.52 | ghenry | MAC.cfg in tftp server |
| 18:35.57 | ghenry | then add tftp to dhcp |
| 18:35.59 | ghenry | done. |
| 18:36.00 | connecta | yah but... why set it up to use tftp and then change all the phones, when it's already set to use ftp? i never understood that |
| 18:36.16 | ghenry | with polycom aye |
| 18:36.23 | etfonhomey | ghenry, if I were in the UK, I'd sell you a Polycom... |
| 18:36.30 | connecta | the dhcp option 66 and 67 can hand out any server, tftp, ftp, http, etc |
| 18:36.48 | ghenry | yeah, and? ;-) |
| 18:36.51 | etfonhomey | connecta, the only reason is that I already had a tftp server setup on my * box and I'm lazy. |
| 18:36.57 | connecta | and changing back from tftp to ftp is easy, just reset all the phones to default |
| 18:37.01 | ghenry | etfonhomey++ |
| 18:37.06 | ghenry | lazy rules ;-) |
| 18:37.13 | connecta | hrmn.... not cool deed |
| 18:37.25 | ghenry | I was joking |
| 18:37.35 | ghenry | etfonhomey: I just got your nick, doh |
| 18:37.38 | etfonhomey | my college CS prof always said, "computer scientists are lazy." |
| 18:37.54 | ghenry | yeah, do it right first time lazy |
| 18:38.05 | connecta | ghenry, Yah mang. lol |
| 18:38.11 | sasch | can help mi with asterisk 1.4 and hint |
| 18:38.27 | ghenry | connecta: fit? |
| 18:38.40 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 18:38.41 | *** part/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 18:38.43 | etfonhomey | sasch, pastebin your sip.conf and your extensions.conf |
| 18:39.11 | *** join/#asterisk grndslm (n=grndslm@24-119-81-201.cpe.cableone.net) |
| 18:40.14 | sasch | <etfonhomey> ok |
| 18:41.23 | ghenry | in what way is a hint different from blf? |
| 18:41.47 | sasch | http://pastebin.ca/472613 |
| 18:42.33 | sasch | <etfonhomey> and in my gpx 2000 i set first button with BLF and line 10 |
| 18:42.36 | andrew` | good place for UK DIDs? |
| 18:43.31 | ghenry | anderiv: gradwell |
| 18:43.47 | ghenry | normal dids or sip? |
| 18:44.08 | etfonhomey | sasch, I don't see line 10 in your sip.confg? |
| 18:44.12 | etfonhomey | conf |
| 18:44.44 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 18:44.53 | ZaVoid | morning |
| 18:45.07 | ZaVoid | can anyone decipher iax debugs well? |
| 18:45.28 | sasch | http://pastebin.ca/472621 |
| 18:45.38 | sasch | excusme i don't paste all my sip.conf :-P |
| 18:45.48 | sasch | beacause is very big :-P |
| 18:46.06 | ZaVoid | Tx-Frame Retry[-01] -- OSeqno: 254 ISeqno: 000 Type: IAX Subclass: LAGRQ |
| 18:46.07 | ZaVoid | <PROTECTED> |
| 18:46.07 | ZaVoid | Rx-Frame Retry[ No] -- OSeqno: 148 ISeqno: 148 Type: IAX Subclass: VNAK |
| 18:46.07 | ZaVoid | <PROTECTED> |
| 18:46.11 | ZaVoid | trying to figure out what that means |
| 18:46.18 | ghenry | ZaVoid: pastebin |
| 18:46.24 | ZaVoid | 4 lines henry :) |
| 18:46.38 | etfonhomey | sasch, what version of *? |
| 18:47.28 | sasch | in CLI what is the command for find my version ?? |
| 18:47.30 | ghenry | ZaVoid: yeah 2 is the norm ;-) |
| 18:47.33 | ghenry | anyway..... |
| 18:47.41 | etfonhomey | sasch, show version |
| 18:47.50 | ZaVoid | ah |
| 18:47.53 | ZaVoid | my bad :) |
| 18:48.00 | ZaVoid | so any idea what it means? |
| 18:48.04 | sasch | Asterisk 1.4.4 built by root @ centres on a i686 running Linux on 2007-05-02 20:40:31 UTC |
| 18:48.06 | ZaVoid | it kept repeating over and over and over |
| 18:48.23 | ZaVoid | had to restart asterisk to get iax connections working again.. this is a new feature in 1.4.2 i guess lol |
| 18:48.27 | ZaVoid | 1.2xxx worked fine |
| 18:48.35 | etfonhomey | sasch, pastebin the output of "core show hints" and "sip show subscriptions" |
| 18:49.19 | dlynes_laptop | sevard: you're still here? |
| 18:49.43 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 18:50.06 | sasch | <etfonhomey> now i turn off sip/10 and sip/11 now I use for hint sip/12 and sip/13 |
| 18:50.28 | etfonhomey | sasch, I think you need to change type=friend to type=peer |
| 18:50.40 | sasch | http://pastebin.ca/472626 |
| 18:50.47 | sasch | ok i try |
| 18:51.02 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 18:51.33 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:51.50 | sasch | no i change type=peer but my led of my blf channel don't start |
| 18:52.13 | etfonhomey | sasch, did you restart *? |
| 18:52.23 | etfonhomey | sasch, or at least sip reload? |
| 18:53.02 | sasch | yes |
| 18:53.08 | etfonhomey | sasch, I have presence (aka blf) working in 1.4.4 and I have type=peer for all my phones |
| 18:53.43 | etfonhomey | sasch, another thing you might try is to put the subscribecontext=blf in each of your sip entries. |
| 18:54.00 | sasch | http://pastebin.ca/472629 .... ok I try |
| 18:54.18 | sasch | but in my CLI don't appear any line that write hint ... |
| 18:54.38 | sasch | excusme for my english I'm Italian and I don't speak very Well english .... |
| 18:54.41 | sasch | i try it :-P |
| 18:55.07 | LeddyHM | dc3aes: ? |
| 18:55.54 | etfonhomey | sasch, if you see nothing in sip show subscriptions, then BLF is not working. also, if core show hints gives "Watchers 0", that's not good either. Someone correct me if I'm giving bad info here. |
| 18:56.09 | etfonhomey | sasch, I saw the language=it in your config. |
| 18:56.32 | etfonhomey | sasch, we'll get through it |
| 18:56.58 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 18:57.05 | dc3aes | LeddyHM? |
| 18:58.30 | dc3aes | les.net finally fixed my account, then accused me of not using proper e-mail etiquette... even though I had called 5 times, left 3 voicemails, 2 personal messages, and about 5 emails.. apparently i didnt attach previous emails to my later emails.. arrgghh see you later les.net |
| 18:59.06 | LeddyHM | I'm on phone with them new |
| 18:59.11 | LeddyHM | he was able to duplicate my error |
| 18:59.17 | LeddyHM | he's looking into it now |
| 18:59.19 | dc3aes | you mean nufone? |
| 18:59.21 | LeddyHM | yes |
| 18:59.25 | dc3aes | ok, im talking about les.net |
| 18:59.33 | dc3aes | :) but this is also good news |
| 18:59.43 | LeddyHM | oops |
| 18:59.50 | etfonhomey | sasch, here's what a working BLF looks like when you call "sip show subscriptions" and "core show hints": http://pastebin.ca/472640 |
| 19:00.18 | etfonhomey | sasch, obviously you can tell that station2 is turned off. It's a softphone that I have on this laptop and I don't have the program running. |
| 19:00.20 | dc3aes | Les from LES.NET also said to make sure I let everyone in this channel know that he credited my account the amount that they erroneously charged me haah |
| 19:00.45 | etfonhomey | dc3aes, is that the $10 that never got deducted from? |
| 19:01.03 | sasch | [May 5 20:43:42] NOTICE[5712]: chan_sip.c:14533 handle_request_subscribe: Got SUBSCRIBE for extension 12@intranet from 192.168.0.180, but there is no hint for that extension. |
| 19:01.56 | etfonhomey | sasch, That's because your hint is 12@blf |
| 19:01.56 | dc3aes | nope, thats nufone.. les.net was chaos.. as stated above.. it tooks HOURS of effort to get someone to listen to me.. then when they did, after i got angry and threatened to spread the truth about them.. he wrote me a huge novel threatening me back and claiming that it was all my fault, etc.. and that my last email wasnt proper.. ignoring the many voicemails, calls, etc. from the last 30 days |
| 19:02.38 | *** part/#asterisk connecta (n=Administ@76.23.188.72.cfl.res.rr.com) |
| 19:02.44 | etfonhomey | sasch, is this a production system that you can't really take down right now? |
| 19:03.31 | sasch | 13@blf : SIP/13 State:Idle Watchers 1 |
| 19:03.32 | etfonhomey | dc3aes, I use vitelity.net and they've been good so far. I've been on them about 3 weeks now. |
| 19:03.39 | etfonhomey | sasch, that's good! |
| 19:03.46 | sasch | but led don't start |
| 19:03.56 | etfonhomey | sasch, sounds like phone config now. |
| 19:04.05 | etfonhomey | sasch, but you have * right for at least this extension. |
| 19:04.43 | etfonhomey | sasch, unfortunately, I only have experience with Polycoms and X-Lite as far as presence config. |
| 19:05.05 | dc3aes | im willing to try anything.. that was really my plan.. i put $10-20 into a bunch of voip providers.. I have found les.net to have the worst sense of customer support, nufone is awesome for support but im finding their cid to be patchy but the call quality is great.. i will continue my trials :P |
| 19:05.24 | etfonhomey | dc3aes, that's exactly what I'm doing. |
| 19:05.44 | dc3aes | LeddyHM: what was outcome of the nufone congestion problem? still on the phone? |
| 19:05.59 | dc3aes | its nice to run your own pbx, because you can mix and match trunks/sip/etc.. |
| 19:06.11 | etfonhomey | dc3aes, I'd like to try some SIP (or IAX) ITSP only setups with one analog backup, but I want to find the best ITSP for my area. |
| 19:06.16 | sasch | <etfonhomey> thank for all ... In italy is 21:00 now i must go out with my love :-P |
| 19:06.24 | sasch | thanks for all !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 19:06.29 | etfonhomey | sasch, where in Italy? |
| 19:06.47 | sasch | Buonconvento near Montalcino (Brunello di Montalcino) |
| 19:06.55 | etfonhomey | sasch, I was in Florence and Rome last summer. |
| 19:07.35 | sasch | Florence is at 100 km .... |
| 19:07.55 | dc3aes | i used vonage in my office for about 3 years.. but it went really bad in my town for 8 months.. nobody at vonage would listen to me or like 15 other people who were 100% convinced there was a "local" problem... turns out they did.. then they offered us all credit but nobody went back.. a friend of mine is having good success here again.. im almost considering using them as a SIP account in my pbx :P |
| 19:09.44 | dc3aes | from what I understand, vonage is also the only company that has nanaimo (my town) numbers.. one of the phone support guys @ les.net said they would get Nanaimo numbers within 2 weeks.. that was over a month ago.. not going to wait |
| 19:11.14 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 19:11.38 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 19:11.50 | LeddyHM | dc3: he was able to replicate my problem and was going to take a look at the proxy |
| 19:12.08 | LeddyHM | he said he'll let me know when it's resolved, hopefully within the hour |
| 19:12.43 | dc3aes | ok whats your geography leddy? |
| 19:12.52 | dc3aes | probably shouldnt matter but hey who knows |
| 19:13.32 | LeddyHM | houston |
| 19:14.08 | dc3aes | western canada here.. dialing western canada numbers |
| 19:14.18 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:14.32 | etfonhomey | dc3aes, you can't just get a SIP account with Vonage can you? They have the consumer stuff, and then there's the Vonage Business Plus plans that start at $150 a month. |
| 19:14.56 | LeddyHM | I'd like to try sunrocket |
| 19:14.58 | etfonhomey | dc3aes, the VBPP give you the sip accunt. |
| 19:15.04 | LeddyHM | they always have those smoking 2 year deals |
| 19:15.20 | etfonhomey | LeddyHM, I talked to another person near houston and they liked vitelity.net there. |
| 19:15.33 | dc3aes | vbpp? |
| 19:15.45 | etfonhomey | dc3aes, Vonage Business Plus plan |
| 19:15.59 | LeddyHM | we are using cbeyond with great success, except the dtmf issue we're troubleshooting |
| 19:16.17 | dc3aes | ok. |
| 19:17.21 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
| 19:17.29 | MrWup | anyone here a CDR guru? |
| 19:18.12 | etfonhomey | dc3aes or LeddyHM, are you using these SIP lines for person use just testing? Or do you have any SIP only production installations at a business? |
| 19:18.40 | LeddyHM | this is for business |
| 19:19.20 | dc3aes | etfonhomey: both really.. I have a home office with a server rack + 4 sip extensions.. that needs to link to my other office.. |
| 19:19.38 | LeddyHM | cebyond wants to blame our intermittent dtmf issue on * |
| 19:19.41 | dc3aes | so testing until it is stable I guess |
| 19:19.48 | *** join/#asterisk dacter (n=dlittrel@207.200.33.213) |
| 19:19.55 | LeddyHM | we are trying to get setup with nufone to prove them wrong |
| 19:20.10 | etfonhomey | LeddyHM, take * out of the picture and test with a softphone. |
| 19:20.46 | LeddyHM | if we do reinvite yes it seems to work well |
| 19:20.52 | LeddyHM | so that took * out of the picture |
| 19:20.56 | etfonhomey | LeddyHM, AHHH! Yep. |
| 19:21.12 | LeddyHM | but our phones have to be on a subnet that their IAD uses |
| 19:21.58 | etfonhomey | LeddyHM, gotcha |
| 19:22.28 | LeddyHM | we're at a loss :( |
| 19:23.20 | *** join/#asterisk asdx (n=diego@200.26.177.4) |
| 19:25.46 | LeddyHM | is reinvite yes a bad practice? |
| 19:27.25 | MrWup | if PSTN Caller calls SIP1, in my dialplan i do Dial(SIP/1). is there any way to tell asterisk to jump into a particular context in the event that SIP1 tries to transfer to another SIPuser |
| 19:27.28 | MrWup | ??? |
| 19:27.42 | *** join/#asterisk Humblgrumpf (n=humblgru@p54B0BCD5.dip0.t-ipconnect.de) |
| 19:28.49 | MrWup | like... PSTN Caller calls SIP1, SIP1 transfers to SIP2 (successfully, i.e. SIP2 answers the transfer) and then asterisk immediately jumps into a context and allows me to update my database to say "PSTN caller is now talking to SIP2 instead of SIP1" |
| 19:30.18 | dc3aes | LeddyHM: nufone just "started working" ... and i didnt change anything so they definitely had a problem they fixed |
| 19:30.38 | LeddyHM | so you were using them |
| 19:30.41 | LeddyHM | <-- confused |
| 19:31.12 | dc3aes | I have multiple accounts in my pbx |
| 19:31.29 | LeddyHM | I thought so |
| 19:31.37 | LeddyHM | mine still doesn't work |
| 19:31.41 | dc3aes | weird. |
| 19:32.04 | LeddyHM | at least it shows they are working on something :) |
| 19:44.57 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 20:12.35 | *** join/#asterisk Simon- (i=simon@proxima.lp0.eu) |
| 20:13.37 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 20:13.39 | kink0 | hello |
| 20:13.42 | kink0 | hi , I have an urgent problem, I have 4 E1 connected to TE412, if i set in zapata.conf 1-15,17-31,32(...)124 I get a lot of cause code 47 |
| 20:14.10 | kink0 | but if I set zapata.conf for just 3 E1, then goes fine. Any sugestion ? |
| 20:26.35 | JT | pastebin.za zaptel.conf and zapata.conf |
| 20:26.36 | JT | pastebin.ca |
| 20:36.25 | *** part/#asterisk Simon- (i=simon@proxima.lp0.eu) |
| 20:38.34 | JT | kink0: guessing it's not that urgent then |
| 20:41.03 | pipwerk | JT: I guess kink0 doesn't know pastebin |
| 20:41.52 | JT | that's no excuse |
| 20:42.47 | pipwerk | and maybe didn't catch that you were actually adressing him? |
| 20:43.02 | JT | that's be pretty daft |
| 20:43.10 | pipwerk | true |
| 20:43.13 | JT | anyway he hasn't said anything since before |
| 20:43.21 | JT | he musn't need help |
| 20:43.57 | pipwerk | maybe got distracted in the 12 minutes that passed trying to sove his problem himselve |
| 20:44.01 | pipwerk | don't know |
| 20:44.19 | pipwerk | 12 minutes is al lot when you're in shit ;-) |
| 20:47.12 | kink0 | pipwerk, I know pastebin, but for just to paste two lines there |
| 20:47.59 | kink0 | pasting... sorry for delay , effectively I was on the problem trying to fix it |
| 20:48.03 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 20:49.09 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 20:49.32 | kink0 | http://pastebin.ca/472833 |
| 20:50.19 | kink0 | I also try to do differents groups for every one span in zapata.conf, assigning every 30 channel to one group, but the same. |
| 20:50.40 | kink0 | total channels does not pass over 60 up never |
| 20:52.12 | JT | looks |
| 20:52.15 | JT | looks fine |
| 20:52.22 | JT | are they all from the same telco? |
| 20:53.05 | kink0 | yes, and the same happens in most than one box |
| 20:53.31 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:53.35 | JT | are the boxes the same, what spec? |
| 20:53.37 | kink0 | the problem occurs in five simmilar boxes, every one with 4 E1 |
| 20:53.41 | kink0 | yes, all the same. |
| 20:53.58 | kink0 | never pass over 60 channels , even must be 120 |
| 20:54.21 | JT | what type of boxes? |
| 20:54.30 | kink0 | supermicro + TE412 cards |
| 20:54.43 | JT | supermicro tells me nothing |
| 20:54.55 | kink0 | zttest is ok, normally over 99.98% |
| 20:55.06 | JT | hmm |
| 20:55.09 | kink0 | dual xeon 3.2/2M 2GRAM |
| 20:55.34 | JT | so all the E1s are independant? |
| 20:56.29 | kink0 | I did now as independent, but the same. No, are join |
| 20:56.46 | JT | what? |
| 20:56.49 | kink0 | even I do all channel in one group, or I create a group for every 30 |
| 20:56.57 | JT | no, telco side |
| 20:57.05 | kink0 | are grouped in telco side |
| 20:57.10 | JT | are you sure you're not meant to be using NFAS signalling? |
| 20:57.15 | JT | NFAS is not setup |
| 20:57.30 | kink0 | I am not sure, but under 60 channels runs fine |
| 20:57.47 | kink0 | the problem starts when the channel #61 is used |
| 20:57.48 | MrWup | ive come across a weird bug |
| 20:57.52 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:57.53 | MrWup | maybe someone can help me |
| 20:57.58 | JT | sort of thing you should check with the telco |
| 20:58.23 | JT | if you need NFAS or not... |
| 20:58.29 | lesouvage | I have a strange problem. Inbound calls are routed to a queue and one of the agents that has logged into the queue will receive this call. If he/she parked the call and take it back by dialing 701 sometimes it works and sometimes it doesn't although everything in the cli indicates that the channels are bridged. (Channel IAX2/moniek-13 connected to parked call 701) Any clue, idea or advice? |
| 20:58.38 | kink0 | JT: other question, when I put as separate, I place 4 Dial() , every one with 101+n |
| 20:59.03 | kink0 | the second attemp is done, but if priority 103 fails, then do not pass to 204, is that normal ? |
| 20:59.15 | MrWup | SIP1 calls SIP2. SIP1 calls SIP3. SIP1 transfers SIP2 to SIP3. SIP3 answers and SIP2 and SIP3 speak. SIP2 hangs up. after this, SIP2 cannot be dialed, and when SIP2 tries to dial another SIP user, the aastra 9133i phone just hangs and needs to be powered off and on again to work |
| 20:59.29 | JT | dunno, i don't use priority jumping, it's deprecated |
| 20:59.50 | *** join/#asterisk sysko (n=sysko@modemcable113.9-203-24.mc.videotron.ca) |
| 20:59.52 | kink0 | DIALSTATUS then ? |
| 21:00.14 | kink0 | or just place the four attemps in consecutive priorities ? |
| 21:01.52 | JT | i'm heading to bed |
| 21:02.07 | JT | you should check out NFAS or if swapping the PRIs plugged in helps |
| 21:03.20 | kink0 | ok, will try |
| 21:03.49 | JT | seems silly if they're not offered in NFAS by the telco if you have so many PRIs |
| 21:03.52 | JT | inefficient |
| 21:10.24 | MrWup | anyone know why im getting NOTICE[4524]: src/chan_h323.c:1804 reload_config: Unable to load config ooh323.conf, OOH323 disabled |
| 21:10.26 | MrWup | ? |
| 21:11.20 | pipwerk | MrWup: because you don't have an ooh323.conf file in your asterisk config directory? |
| 21:11.31 | MrWup | is that bad? |
| 21:11.40 | MrWup | should there be one? |
| 21:11.52 | pipwerk | no, you just can't make h323 calls, which is not that bad |
| 21:11.53 | blitzrage | if you want the module to have a configuration, ya |
| 21:12.13 | MrWup | h323 calls... |
| 21:12.19 | MrWup | whats that? |
| 21:12.35 | pipwerk | if you don't know, you will probably not care |
| 21:12.53 | MrWup | ok |
| 21:13.13 | pipwerk | just put a 'noload chan_ooh323.so' in your modules.conf |
| 21:13.31 | pipwerk | h323 is a voip signaling protocol |
| 21:13.38 | MrWup | oh |
| 21:13.47 | MrWup | as opposed to SIP? |
| 21:13.53 | pipwerk | yes |
| 21:13.58 | MrWup | is h323 better? |
| 21:14.07 | pipwerk | different |
| 21:14.27 | pipwerk | H.323 is an ITU standard, SIP is an internet standard |
| 21:14.38 | pipwerk | both are widly deployed |
| 21:14.45 | blitzrage | IETF to be more precise |
| 21:15.03 | pipwerk | of course |
| 21:15.29 | pipwerk | H.323 is quite a bit more complex |
| 21:15.39 | pipwerk | imnsho |
| 21:15.48 | blitzrage | ya, it is :) |
| 21:17.53 | blitzrage | time to order za |
| 21:22.39 | ghenry | why would i get when using: --with-postgres=/usr/local/pgsql/8.2.4 "checking for PQexec in -lpq... no" with * 1.4.4. Configure has found pg_config: checking for pg_config... /usr/local/pgsql/8.2.4//bin/pg_config |
| 21:24.39 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:36.58 | tzafrir | ghenry, take a look at config.log |
| 21:37.13 | ghenry | ah, yes. forgot |
| 21:37.14 | ghenry | cheers |
| 21:37.19 | tzafrir | search for '-lpq', I guess |
| 21:37.28 | tzafrir | I never said it is easy |
| 21:39.00 | *** join/#asterisk xcarioca (i=xcarioca@200.103.128.24) |
| 21:40.21 | ghenry | I'm getting /usr/local/pgsql/8.2.4/lib/libpq.so: undefined reference to `SSL_CTX_set_client_cert_cb |
| 21:40.24 | ghenry | my ssl setup |
| 21:40.27 | ghenry | cool, no worries |
| 21:40.29 | ghenry | thanks |
| 21:40.40 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 21:41.19 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 21:42.53 | lesouvage | I noticed that that there is no value for "requested prefs" when connecting Idefisk (iax2) with a parked call. The connection is there but there is no sound. Could there be a relation between the requested prefs value and the silence in the connection. (> requested format = alaw, > requested prefs = (), > actual format = alaw,) |
| 21:43.10 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 21:45.14 | MrWup | where are the asterisk doc files? |
| 21:45.18 | MrWup | the normal path? |
| 21:47.41 | kiscokid | there are asterisk doc files? |
| 21:48.37 | pipwerk | <PROTECTED> |
| 21:48.40 | kiscokid | try man asterisk |
| 21:49.40 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:52.25 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 21:53.12 | tzafrir | the asterisk man page is rather useless |
| 21:53.23 | tzafrir | the doc/ directory is rather handy |
| 21:53.37 | tzafrir | also: config/ , with the sample config files |
| 21:53.48 | pipwerk | and of course, there is the book |
| 21:54.16 | kiscokid | did you look at the new version of the book? |
| 21:54.26 | kiscokid | rough cuts |
| 21:54.54 | pipwerk | nope, url? |
| 21:55.07 | kiscokid | not available free |
| 21:55.15 | pipwerk | ow, pitty |
| 21:55.43 | pipwerk | the 'current' version is getting a bit outdated |
| 21:55.55 | kiscokid | someone has to make money on free software :-) |
| 21:56.02 | pipwerk | true |
| 21:56.33 | pipwerk | I've just received the german book 'Voice over IP mit Asterisk und SER' |
| 21:56.49 | pipwerk | very curious about that |
| 21:56.51 | kiscokid | what is SER? |
| 21:57.09 | pipwerk | SIP Express Router, a real sip proxy |
| 21:57.55 | tzafrir | though there is also an active fork called openser (actually seems to be more active than the original) |
| 21:58.31 | pipwerk | the original is more geared toward stability than features |
| 21:58.37 | pipwerk | as I understand it |
| 21:59.32 | pipwerk | anyway... I'm very curious to learn a bit about SER/OpenSER |
| 22:01.55 | blitzrage | 2nd edition of the book will be out soon |
| 22:02.20 | pipwerk | cool |
| 22:02.35 | kiscokid | July |
| 22:03.25 | blitzrage | oh ya? Is that definitive? |
| 22:04.48 | kiscokid | http://www.oreilly.com/catalog/9780596510480/index.html |
| 22:05.26 | blitzrage | well it's really up to when the authors finish the book... that'll just be an estimate |
| 22:05.28 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-380211b3959fcb14) |
| 22:05.28 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 22:06.21 | MrWup | gah this is a nightmare |
| 22:06.30 | *** join/#asterisk saftsack (n=saftsack@pD9E0612C.dip.t-dialin.net) |
| 22:06.33 | tzafrir | ghenry, found anything interesting in config.log ? |
| 22:06.35 | MrWup | i cant find an example anywhere of how to use asterisks's features.conf's blind or attended transfers |
| 22:06.41 | MrWup | the wiki is about as clear as mud |
| 22:07.00 | ghenry | tzafrir: yeah, my custom ldap and ssl |
| 22:07.46 | ghenry | /usr/local/pgsql/8.2.4/lib/libpq.so: undefined reference to `SSL_CTX_set_client_cert_cb' |
| 22:07.55 | ghenry | /usr/local/lib/libldap-2.3.so.0: undefined reference to `SSL_CTX_set_info_callback' |
| 22:08.22 | ghenry | any ideas tzafrir ? |
| 22:08.37 | ghenry | I've got both lib dirs in /etc/ld.so.conf |
| 22:08.44 | ghenry | and ran ldconfig |
| 22:09.07 | pipwerk | what does ldd say? |
| 22:09.14 | tzafrir | you have the build command slightly above that. What happens when you re-run it? |
| 22:09.39 | ghenry | yeah, will run ldd, 2 secs forgot |
| 22:10.05 | ghenry | ldd on libldap is goof |
| 22:10.08 | ghenry | good |
| 22:10.27 | ghenry | and good on /usr/local/pgsql/8.2.4/lib/libpq.so |
| 22:10.29 | ghenry | hmm# |
| 22:10.39 | ghenry | will run configtest compile by hand |
| 22:10.41 | ghenry | 2 secs |
| 22:11.12 | pipwerk | rm your config.cache and config.status and try again? |
| 22:12.54 | ghenry | same |
| 22:13.10 | ghenry | where is conftest.c ? |
| 22:14.07 | MrWup | ah ok ive done it |
| 22:14.36 | ghenry | I think I need -lssl in that test |
| 22:14.41 | MrWup | but say i define the xfer sequence as *1... |
| 22:14.47 | MrWup | i have to press the 1 very quickly after the * |
| 22:14.52 | MrWup | is there any way to change this? |
| 22:14.56 | MrWup | so i can type them more slowly? |
| 22:18.13 | robin_sz | bah ... chan_mISDN no longer seems to recongise DTMF :( |
| 22:18.47 | robin_sz | and yes, I did a /etc/init.d/misdn-init restart |
| 22:18.59 | robin_sz | pooh monkeys. |
| 22:22.23 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:23.01 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 22:25.06 | MrWup | ok heres another one |
| 22:26.03 | MrWup | i use the asterisk attended transfer... SIP1 calls SIP2, SIP1 xfers SIP2 to SIP3. when SIP2 speaks into the mic, SIP3 gets the voice with a 1 second lag. when SIP3 replies, the voice comes through immediately |
| 22:26.49 | robin_sz | bah :( |
| 22:27.05 | robin_sz | surely dtmf detection is not hard? |
| 22:28.42 | MrWup | ive decided... getting any sort of help with asterisk is pointless |
| 22:28.54 | MrWup | its full of fucking bugs and nobody knows solutions |
| 22:29.03 | robin_sz | no |
| 22:29.14 | ghenry | why would you get /usr/local/ssl/lib/libssl.so: undefined reference to `BIO_method_type' |
| 22:29.25 | ghenry | ssl libs not linking to themselves |
| 22:29.43 | robin_sz | MrWup, if we coud understand your questions, you would do better |
| 22:29.49 | ghenry | ah, -lcrypto |
| 22:30.10 | MrWup | what dont you understand |
| 22:30.19 | MrWup | is english a foreign language to you? |
| 22:30.44 | MrWup | after an asterisk attended transfer, there is one-way voice lag between the two people who speak |
| 22:30.45 | ghenry | yeah |
| 22:30.51 | ghenry | 1.4.4 configured |
| 22:31.11 | ghenry | needed -lssl and -lcrypto in configure script test for PQexec |
| 22:31.13 | MrWup | SIP2 speaks, SIP3 gets SIP2's speech after over a second's lag |
| 22:31.26 | MrWup | SIP3 speaks, SIP2 gets SIP3's voice immediately |
| 22:31.26 | ghenry | should that be classed as a configure bug? |
| 22:31.57 | robin_sz | phones in use, codecs? |
| 22:32.22 | robin_sz | is there transcoding going on here? |
| 22:32.22 | MrWup | nothings wrong with the codecs |
| 22:32.33 | MrWup | it works fine in all circumstances except after an attended transfer |
| 22:32.40 | pipwerk | ghenry: since I've succesfully build 1.4.4 without any hacks I guess it's something wrong in your config |
| 22:32.43 | robin_sz | phones in use, codecs? |
| 22:33.10 | ghenry | pipwerk: yeah, my busted ssl |
| 22:33.11 | MrWup | aastra 9133i, alaw |
| 22:33.12 | ghenry | ;-) |
| 22:33.20 | robin_sz | ok, |
| 22:33.22 | MrWup | its not the damned phones or the codecs |
| 22:33.32 | pipwerk | ghenry: I guess the ssl test failed first? |
| 22:33.35 | robin_sz | ooh, tetchy |
| 22:33.48 | MrWup | they work perfectly. but as soon as stupid asterisk organises an attended or blind transfer, it creates one-way voice lag |
| 22:34.00 | ghenry | pipwerk: nope, ssl test is after postgres, which is dumb? |
| 22:34.14 | pipwerk | possibly? |
| 22:34.23 | pipwerk | I don't know |
| 22:34.31 | ghenry | checking for ssl2_connect in -lssl... yes comes after pg test |
| 22:42.28 | MrWup | does anyone know if there's a way to program any of the aastra 9133i's softkeys to dial *1 (and therefore access the features.conf features of asterisk)? |
| 22:53.46 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 23:02.41 | ghenry | what's the quickest way to simulate a sip call on the cli? |
| 23:02.52 | ghenry | haven't got a gui here on a remote box |
| 23:03.03 | ghenry | what to do sip:4234@342343243 etc |
| 23:06.00 | blitzrage | use the 'dial' command |
| 23:06.01 | pipwerk | you could do something with the manager interface |
| 23:06.09 | blitzrage | assuming you've got chan_oss or chan_alsa loaded |
| 23:07.15 | blitzrage | idefisk <--IAX2--> Asterisk <--IAX2--> Asterisk <--SIP--> softswitch.... call sets up, but audio drops after a few seconds... teh sux |
| 23:07.28 | blitzrage | has to be a reinvite issue or something.... |
| 23:09.01 | ghenry | where do you get the res_pgsql sql files from? |
| 23:09.07 | ghenry | or even a sample res_pgsql.conf |
| 23:09.17 | ghenry | nothing is bundled with 1.4.4 |
| 23:09.26 | ghenry | or in trunk |
| 23:10.33 | robin_sz | ghenry, just do a dial foo@context, it shoudl dial from the cli |
| 23:10.41 | ghenry | thanks |
| 23:12.02 | robin_sz | you need to have a suitable audio stack loaded though, or the dial command aint there |
| 23:12.52 | tzafrir | 'originate' a call from an Local/ echo-test extension? |
| 23:13.09 | ghenry | sp res pgsql docs? |
| 23:13.13 | ghenry | so, even. |
| 23:13.39 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:14.17 | tzafrir | ghenry, nothing under the doc/ and nothing under the configs/ ? |
| 23:14.25 | ghenry | nope |
| 23:14.29 | ghenry | not even in trunk! |
| 23:14.40 | ghenry | no res_pgsql.conf.sample either, wtf |
| 23:15.47 | ghenry | should I just use the same tables and unixodbc settings |
| 23:15.48 | ghenry | http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 23:16.27 | *** join/#asterisk segal_wor (n=segal_wo@cuscon13938.tstt.net.tt) |
| 23:16.38 | segal_wor | hi, Is it true that if you connect telephone lines to the fxs module of a tdm4000 it will destory it? |
| 23:17.06 | hads | It's not a good idea anyway. |
| 23:17.17 | lesouvage | This is driving me crazy. I park a call, call 701 after a short while, the connection is made but no sound and cli output looks ok. I updated zaptel and asterisk to the latest 1.2 version and tried different iax2 softphones. It has worked once after the updates but then it failed again. Any suggestions to fix this would make me very happy. |
| 23:17.30 | segal_wor | hads? |
| 23:17.33 | [TK]D-Fender | ghenry, that guide says right from the start that they'll use ODBC to do this, not PGSQL direct. |
| 23:17.48 | ghenry | [TK]D-Fender: I know |
| 23:17.57 | hads | segal_wor? |
| 23:18.03 | ghenry | but I'm trying to find a sample conf and table layout |
| 23:18.06 | segal_wor | hads: Is it true that if you connect telephone lines to the fxs module of a tdm4000 it will destory it? |
| 23:18.06 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 23:18.11 | hads | It's not a good idea anyway. |
| 23:18.25 | ghenry | I can't believe there's a res_config_pgsql with no docs or sample conf |
| 23:18.30 | ghenry | how did it get in? |
| 23:18.31 | [TK]D-Fender | ghenry, And * dropped PG support and now focus' on ODBC. |
| 23:18.52 | ghenry | really? |
| 23:19.02 | segal_wor | hads: i am not getting a dail tone on the fxs module. |
| 23:19.10 | ghenry | and a nother db [TK]D-Fender ? |
| 23:19.23 | segal_wor | hads: even if it reports that it is properly configured. |
| 23:19.32 | segal_wor | hads: show zap channels |
| 23:19.45 | [TK]D-Fender | ghenry, Where fdo you see res_config_pgsql mentioned? |
| 23:19.49 | blitzrage | ghenry: you need the postgresql libraries and such installed for Asterisk to be able to build res_pgsql.so |
| 23:20.02 | ghenry | it's all built and in |
| 23:20.17 | [TK]D-Fender | segal_wor, More like if the line RINGS you have a serious risk. |
| 23:20.29 | ghenry | [TK]D-Fender: --with-postgres in ./configure |
| 23:20.31 | ghenry | in 1.4.4 |
| 23:20.58 | blitzrage | hrmmmmmmm... |
| 23:20.59 | *** join/#asterisk someninjamaster (n=jamesric@c-68-53-159-99.hsd1.tn.comcast.net) |
| 23:20.59 | ghenry | <PROTECTED> |
| 23:21.07 | someninjamaster | ~the book |
| 23:21.20 | ghenry | I've got it built |
| 23:21.20 | blitzrage | why the heck does Asterisk keep transferring my media off? |
| 23:21.20 | blitzrage | ~book |
| 23:21.23 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:21.23 | ghenry | but no docs or sample conf |
| 23:21.24 | segal_wor | TKD: how can i know if it is fried. |
| 23:21.24 | ghenry | why include it then |
| 23:21.34 | segal_wor | TKD: current sign is no dial tone. |
| 23:22.00 | [TK]D-Fender | segal_wor, try USING it of course. |
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| 23:29.58 | joaovianna | Any way to disable H323 in configure before asterisk installation ? Version 1.4.4 not working ./configure |
| 23:31.19 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 23:31.44 | mitcheloc | blitzrage: ping |
| 23:32.19 | blitzrage | mitcheloc: pong... |
| 23:32.36 | blitzrage | joaovianna: make menuselect |
| 23:32.49 | blitzrage | you do it after ./configure |
| 23:34.53 | [TK]D-Fender | blitzrage, You give them a giant menu that screams "cherry-pick your options here" and they still look elswhere... you can guide a horse to the water..... |
| 23:35.21 | mitcheloc | blitzrage: did your e61i come in? i've been tryiing for the last 24 hours to get mine to connect to asterisk |
| 23:36.11 | tzafrir | [TK]D-Fender, maybe because there is no such option for the configure script |
| 23:36.45 | blitzrage | mitcheloc: ya, it came in, but I only have mine via truphone right now -- you can't use a secret with it if I remember correctly |
| 23:36.45 | blitzrage | it won't authenticate with it |
| 23:36.48 | tzafrir | whereas there are some --with options |
| 23:37.05 | tzafrir | (though --without-openh323 should probably do tyhe trick as well) |
| 23:37.31 | mitcheloc | blitzrage: i'm not even getting it to attempt to authenticate, i read somewhere that the e61, a year back, had no nat support... i'm assuming they didn't f ix it and that's what i'm having trouble with... |
| 23:37.57 | mitcheloc | blitzrage: will truphone work off of asterisk? or do i have to use their service |
| 23:37.58 | blitzrage | can't remember... I did get it to work at some point... but I forget what I did |
| 23:38.07 | blitzrage | truphone is an independant service |
| 23:38.15 | blitzrage | works well, other than the latency |
| 23:39.12 | mitcheloc | blitzrage: that sort of defeats the purpose for the phone then as an extension to the office system :( |
| 23:39.50 | blitzrage | mitcheloc: I did get it to work, I just forget what I did |
| 23:40.37 | mitcheloc | blitzrage: okay, i'll try without a password then in a bit |
| 23:40.41 | blitzrage | I just couldn't use it with the service I built because it won't accept a # symbol in the name |
| 23:41.58 | mitcheloc | blitzrage: so just leave the secret line in sip.conf commented out as well? |
| 23:43.16 | blitzrage | like I said, I forget -- you'll have to try some stuff |
| 23:44.20 | joaovianna | blitzrage: The problem is I can't run ./configure ! When I tried run make /menuselect I got "Unable to open 'build_tools/menuselect-deps' for reading! Did you run ./configure ? |
| 23:44.20 | joaovianna | " |
| 23:44.20 | mitcheloc | no problem, the fact that you had it working is enough encouragement, thanks |
| 23:44.40 | blitzrage | joaovianna: why can't you run ./configure? |
| 23:44.45 | blitzrage | mitcheloc: *thumbs up* |
| 23:45.35 | joaovianna | When I tried to run ./configure I got "checking if PWLib version is compatible with chan_h323... ./configure: line 27683: test: -lt: unary operator expected |
| 23:45.35 | joaovianna | " |
| 23:46.02 | joaovianna | blitzrage: Is there any way to run ./configure with no H323 ? |
| 23:46.19 | blitzrage | are you only running ./configure? |
| 23:46.30 | blitzrage | sounds like either you're doing something wrong, or there is a bug in the configure script |
| 23:46.33 | joaovianna | blitzrage: Yes |
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| 23:49.22 | joaovianna | I can see it was correct in Bugfix http://svn.digium.com/view/asterisk?rev=43863&view=rev but I don't know how to fix it. |
| 23:49.48 | joaovianna | blitzrage: How do I patch it ? |
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