| 00:00.15 | blackk | Dr-Linux|home: or you can try to put it in DMZ for now to see if that's solving the problem |
| 00:00.24 | Dr-Linux|home | bulle: my provider listens on different port than 5060, therefore i even changed my asterisk sip port |
| 00:00.51 | blackk | Dr-Linux|home: what router is that? |
| 00:00.54 | notoriousrab1982 | bulle: thanks for that, will have a look at my contexts - and do what you suggest |
| 00:01.13 | Dr-Linux|home | bulle: actually this router doesn't forward the ports, i tried alot, even my internet provider, this model never worked for them as well |
| 00:01.23 | bulle | Dr-Linux|home: what model is it ? |
| 00:01.30 | Dr-Linux|home | blackk: Eztec modem |
| 00:02.11 | flenders | wow, that's a brand I've never heard of |
| 00:02.16 | bulle | ok, never heard of that brand |
| 00:02.17 | blackk | Dr-Linux|home: does it have a DMZ setting? |
| 00:03.03 | Dr-Linux|home | blackk: actually it has all settings, but don't work, but defulat stuff works |
| 00:03.08 | blackk | Dr-Linux|home: also can we see your NAT settings from the asterisk box? |
| 00:04.29 | Dr-Linux|home | blackk: sir, nat is "nat=yes" |
| 00:04.31 | Defraz | I have this Grandstream 4104 and I am hearing some serious static on it. |
| 00:04.47 | flenders | ~gs |
| 00:04.49 | jbot | well, gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 00:05.01 | Dr-Linux|home | blackk: also i tried, all nat options |
| 00:05.35 | Dr-Linux|home | i can understand i've 2 options to make it work: |
| 00:05.45 | Dr-Linux|home | 1. assign live IP to the * box |
| 00:06.09 | Dr-Linux|home | 2. forward all sip and rtp ports range to the asterisk box |
| 00:06.32 | Dr-Linux|home | blackk: the modem is Aztec |
| 00:06.48 | *** join/#asterisk mrpillowhead (n=pillowhe@ROAMFG-AUTHCLIENTS-01.NATPOOL.NYU.EDU) |
| 00:06.58 | Defraz | I have 4 analog lines going into the 4104 |
| 00:07.11 | Defraz | has anyone else done that? |
| 00:07.13 | mrpillowhead | question installing 1.4 vs 1.2 |
| 00:07.30 | Dr-Linux|home | blackk: how can i setup DMZ ? |
| 00:07.35 | mrpillowhead | says i need a configuration file before I can make |
| 00:08.00 | mrpillowhead | where as 1.2.17 installs perfectly fine |
| 00:08.07 | mrpillowhead | with make clean; make install |
| 00:08.22 | blackk | Dr-Linux|home: there is only one case when Asterisk wouldn't work behind NAT, if you're modem is changing the src port when it's doing the masquerading and in that case you need to do forward on the router; it looks like you have bad luck |
| 00:08.26 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 00:08.52 | haroldp | blackk: thanks for the pointer. got it working. |
| 00:09.12 | Dr-Linux|home | hhm.. yeah |
| 00:09.14 | blackk | Dr-Linux|home: DMZ is usually an option on the router that forwards all ports (the ones that are not used by NAT) to a certain host inside the network |
| 00:09.18 | Dr-Linux|home | blackk: but on the same LAN , if my softphone registered to the same provider with same sip user and works |
| 00:10.21 | flenders | Defraz: how do the analogue lines go into the GS? |
| 00:10.37 | Dr-Linux|home | blackk: but softphones are on same network, how come they work with same user/sip/provider :S |
| 00:10.38 | irule | how can I convert a sound to make the voice run faster so the noise sounds like the chipmunks movie? I want to differentiate language files before I start translating what I need to my language :) |
| 00:10.50 | mrpillowhead | can someone tell me why it says " the configure script must be executed before running 'make'? |
| 00:10.58 | mrpillowhead | while installing 1.4 |
| 00:11.07 | mrpillowhead | doesn't do that with 1.2 |
| 00:11.13 | FuriousGeorge | mrpillowhead: run ./configure |
| 00:11.17 | blackk | Dr-Linux|home: can you do a sip debug and see that the IP sent in SDP is the correct public IP of the router? |
| 00:11.21 | irule | mrpillowhead before make, execute ./configure |
| 00:11.44 | FuriousGeorge | irule: one app that i can think of that does what you want is windows sound recorder |
| 00:12.03 | FuriousGeorge | im sure there is a way to do it with sox or something in the cli, but i dont know it |
| 00:12.07 | flenders | mrpillowhead: that's how you install anything from source on linux, by the way |
| 00:12.14 | Dr-Linux|home | blackk: it's sends the correct public IP, that's what debug shows me |
| 00:12.20 | flenders | mrpillowhead: ./configure ; make ; make install |
| 00:12.24 | mrpillowhead | thank you |
| 00:12.32 | mrpillowhead | ah ok |
| 00:12.41 | mrpillowhead | didn't have to do that for 1.2.17 |
| 00:12.47 | mrpillowhead | just make clean, make install |
| 00:12.54 | clyrrad | what is the default codec used for asterisk sounds? |
| 00:13.06 | FuriousGeorge | mrpillowhead: he meant to say "most things" |
| 00:13.09 | newsmafia | gsm |
| 00:13.10 | irule | great idea, but I want to do it with sox so that I dont have to spped up files manually :s |
| 00:13.16 | mrpillowhead | hehe :) |
| 00:13.45 | flenders | FuriousGeorge: thanks |
| 00:13.51 | clyrrad | newsmafia: im new to the recording stuff... but can you have a codec gsm - but have the sound file as a .wav? |
| 00:13.53 | Dr-Linux|home | blackk: actually this issue is with one of my client, i just came back home after trying to fix for 4 hours |
| 00:14.10 | Dr-Linux|home | blackk: i've debug info in the email, lemme open it for you |
| 00:14.19 | irule | clyrrad yes |
| 00:14.27 | clyrrad | irule:; thanks :) |
| 00:14.29 | blackk | Dr-Linux|home: pastebin |
| 00:14.41 | irule | clyrrad you record to wav, send a copy to the asterisk server and convert it ith sox |
| 00:14.57 | *** join/#asterisk Bazy (n=bazy@89.137.178.124) |
| 00:15.09 | clyrrad | irule: the recording studio wanted to know what codec and file format to use.... |
| 00:15.13 | irule | http://osdir.com/ml/audio.sox/2004-11/msg00020.html |
| 00:15.19 | Dr-Linux|home | blackk: i pasted |
| 00:15.22 | irule | just wav |
| 00:15.26 | irule | youll convert later |
| 00:15.29 | clyrrad | irule: so if I tell them ULA and WAV im good to go? |
| 00:15.44 | Bazy | Hi guys. Is there any way that I can make asterisk connect as a client to an MGCP gatekeeper? |
| 00:16.09 | irule | just WAV, of even MP3 for you to easyly recieve everything in a single CD |
| 00:16.26 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:16.55 | irule | you will then convert mp3 -> wav -> gsm |
| 00:16.57 | irule | or wav -> gsm |
| 00:17.05 | clyrrad | irule: is it mandatory to be converted to gsm? |
| 00:17.13 | irule | I am looking atm for ways to BATCH convert sound files |
| 00:17.19 | clyrrad | I can just leave as wav cant I? |
| 00:17.20 | irule | I dont know |
| 00:17.45 | clyrrad | irule: you could use a script with a FOR loop? |
| 00:17.47 | irule | I can only see gsm files in sounds directory |
| 00:17.53 | irule | exactly |
| 00:18.08 | Mercestes | irule: I batch convert files using sox/soxmix/lame and a cronjob. What are you trying to do? |
| 00:18.40 | irule | but I want to add sound speed up while mine are compressed, I dont know what language asterisk is speaking because I just copied the originals to es |
| 00:19.00 | irule | Mercestes that ;) |
| 00:19.35 | irule | Mercestes can I have your scrip pleas? |
| 00:19.55 | Mercestes | irule, yea, just google 2wav2mp3 and use that template from voipinfo.org and write a cronjob around it. |
| 00:20.25 | irule | Mercestes thanks, so why exactly do you use cron? |
| 00:20.58 | Mercestes | irule, to make it run automaticallyand process all the files in /var/lib/asterisk/monitor. |
| 00:21.49 | Mercestes | I don't wan tto do it on the fly so I do it overnight. |
| 00:21.49 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 00:21.49 | Mercestes | that google i gave you will show you how to do it on the fly too though |
| 00:23.50 | irule | is monitor for recording calls? |
| 00:24.06 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 00:24.06 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:24.41 | Mercestes | irule, Yea. monitor and record both work |
| 00:24.45 | irule | Mercestes btw fti there is a test for shells cript for existing files, that way you can skip files that are already converted |
| 00:25.08 | Mercestes | irule, test -r works |
| 00:25.15 | irule | cool |
| 00:25.34 | irule | I used that a couple weeks ago :) |
| 00:26.07 | irule | I found these, http://www.loligo.com/asterisk/current/extensions.conf are there any other good examples? |
| 00:28.35 | *** part/#asterisk SkramX (n=mark@HERCULES.sentiensystems.net) |
| 00:28.49 | joobie | thekidrio, still there? |
| 00:30.33 | joobie | guys im building a server for this asterisk and thekidrio recommended to make the server have some pci2.2 slots (for cards to interface with the telco).. |
| 00:30.52 | joobie | i spoke to the hardware vendor and they said they need to know if it's the 32bit or 64bit |
| 00:31.05 | joobie | not sure about this? anyone know if the cards are usually 32bit or 64bit? |
| 00:33.41 | joobie | also the mhz if possible..... |
| 00:34.27 | flenders | joobie: what cards do you want? |
| 00:35.20 | joobie | hey flenders |
| 00:35.39 | joobie | not sure flenders .. |
| 00:35.48 | joobie | the guy hasnt got back to me yet if it's ISDN or not |
| 00:36.01 | joobie | but say i offered him a solution and said you have to go ISDN, which would your ecommend? |
| 00:36.07 | joobie | also i can offer him PRI or BRI |
| 00:36.21 | joobie | im in the process of getting a quote from dell for the server..... |
| 00:36.57 | flenders | mate, I think the cards are pretty much 32bit |
| 00:37.03 | joobie | ahh k |
| 00:37.17 | flenders | joobie: have a look at digium and sangoma |
| 00:37.45 | flenders | techtopia.com.au has a decent price on those cards |
| 00:37.48 | joobie | flenders, is there one you recommend that is good quality (with this echo cancel hardware that you mentioned) ? |
| 00:38.18 | flenders | the problem with EC is that they only make dual port cards with EC |
| 00:38.29 | joobie | EC? |
| 00:38.40 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 00:38.47 | flenders | a sangoma dual port PRI with EC will be around 2100 at techtopia |
| 00:38.47 | *** join/#asterisk kje (n=kje@62-99-209-38.c-vzollerg.xdsl-line.inode.at) |
| 00:38.53 | flenders | Echo Cancel |
| 00:38.55 | joobie | ahh |
| 00:39.04 | joobie | damn |
| 00:39.06 | joobie | expensive |
| 00:39.15 | joobie | there's a huge cost difference between the analogue and digial |
| 00:39.17 | flenders | a single port PRI with no EC, will be around 700 |
| 00:39.27 | joobie | is there a huge notical different between the analogue quality and digital? |
| 00:39.45 | Qwell | joobie: in perfect conditions, no |
| 00:39.50 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 00:40.02 | joobie | in realistic conditions? |
| 00:40.08 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-188-8.nycmny.east.verizon.net) |
| 00:40.43 | flenders | I think digital are better |
| 00:41.08 | joobie | flenders, is it the TE207p you're looking at for the 2 port with EC ? |
| 00:41.12 | flenders | but of course, you pay for that |
| 00:41.23 | joobie | yea |
| 00:41.24 | flenders | no, sangoma a102d |
| 00:41.42 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-188-8.nycmny.east.verizon.net) |
| 00:41.42 | flenders | that was on the e-mail I sent you the other day... :o) |
| 00:41.57 | flenders | or maybe I didn't mention prices on that |
| 00:42.03 | joobie | flenders, it doesnt say it has EC on that webpage http://www.techtopia.com.au/index.php/cPath/36_62) |
| 00:42.18 | joobie | i dont think there were prices there |
| 00:42.56 | joobie | damn damn damn.. that's going to be too costly |
| 00:43.08 | flenders | TE207P "offer on-board DSP-based echo cancellation module" |
| 00:43.33 | flenders | "Sangoma A102D - dual E1 PCI with hardware echo cancellation" |
| 00:43.44 | joobie | <PROTECTED> |
| 00:43.45 | joobie | <PROTECTED> |
| 00:43.45 | joobie | <PROTECTED> |
| 00:43.45 | joobie | <PROTECTED> |
| 00:43.56 | joobie | hmm.. maybe it's just that page |
| 00:44.04 | joobie | ahh when you go into it |
| 00:44.06 | joobie | my bad, sorry. |
| 00:44.16 | *** join/#asterisk JoseBravo (n=jbravo@190.24.176.74) |
| 00:44.17 | flenders | :D |
| 00:44.27 | DeeJayTwo | has somebody used openser as a proxy to asterisk? |
| 00:44.30 | joobie | flenders, what about a 4 port equivilent? |
| 00:44.40 | flenders | I just bought a sangoma a102, with no echo cancellation |
| 00:44.43 | joobie | i think 2 wont be enough.. they will have 2 lines for voice and then a fax coming in |
| 00:44.51 | joobie | or can fax be a standard analogue? |
| 00:45.00 | flenders | joobie: a PRI has 30 something channels |
| 00:45.07 | joobie | ahh |
| 00:45.08 | flenders | E1 |
| 00:45.10 | mihinomenest | 24 |
| 00:45.11 | joobie | and a BRI ? |
| 00:45.22 | mihinomenest | 2 |
| 00:45.26 | joobie | ahh |
| 00:45.31 | joobie | so they are on the only forms of ISDN? |
| 00:45.33 | flenders | over here they sell BRIs as ISDN2 (telstra) |
| 00:45.36 | joobie | BRI and PRI ? |
| 00:45.47 | joobie | I see |
| 00:45.56 | joobie | i havent played with it much.. ISDN That is |
| 00:46.04 | mihinomenest | I wouldn't. |
| 00:46.09 | joobie | yea? why? |
| 00:46.19 | joobie | flenders, mentioned the costs, and they are not that much more expensive for voice |
| 00:46.21 | mihinomenest | cost. |
| 00:46.52 | mihinomenest | cheaper to go with VoIP and let the LEC eat the costs of the network. |
| 00:47.24 | joobie | LEC? |
| 00:47.29 | Cybertoy | i guess it depends where you're at... |
| 00:47.34 | mihinomenest | Local Exchange Carrier. |
| 00:47.37 | joobie | ahh |
| 00:47.41 | Cybertoy | I used to have ISDN over in Europe... was a lot cheaper there |
| 00:47.45 | flenders | pots lines here cost about 30 bucks a month, and a BRI is $79.50 |
| 00:47.45 | joobie | I dont want to roll out VOIP past the premisis though |
| 00:47.49 | joobie | not in the first stage at least |
| 00:48.08 | joobie | flenders, 79.50 for use of both the channel? |
| 00:48.12 | flenders | yeah |
| 00:48.18 | joobie | that's not bad at all |
| 00:48.27 | flenders | and that's telstra! |
| 00:48.29 | flenders | :D |
| 00:48.32 | joobie | heheeh |
| 00:48.33 | Cybertoy | isdn in Switzerland was $35 per month |
| 00:48.43 | joobie | does the tail telstra run (BRI) plug straight into that card you recommende flenders ? |
| 00:48.44 | flenders | wow! that's cheap |
| 00:49.01 | flenders | yeah |
| 00:49.01 | Cybertoy | about the same price as the analog line |
| 00:49.19 | mihinomenest | ISDN around here is something like $150/mo. |
| 00:50.35 | joobie | flenders, is 2 x BRI cheaper than PRI ? |
| 00:50.41 | joobie | i mean say you wanted 4 lines coming in.... |
| 00:50.58 | flenders | minimum on PRI is 10 |
| 00:51.04 | joobie | doh |
| 00:51.32 | flenders | optus had a deal that was $20 per line on the PRI |
| 00:51.32 | joobie | so if i get a 4 port card, and 2 x BRI for 4 lines.. would all 4 ports be full on the card? or does BRI bring in 1 cable for 2 lines? |
| 00:51.40 | flenders | so, minimum was 200 |
| 00:51.42 | joobie | nice |
| 00:51.55 | flenders | 1 cable, 2 lines |
| 00:52.01 | joobie | ahh k |
| 00:52.07 | flenders | so one BRI per port |
| 00:52.13 | joobie | so a 2 port card like you mentioned would support 4 lines on BRI ? |
| 00:52.27 | flenders | the cards I mentioned were PRIs |
| 00:52.31 | flenders | different hardware |
| 00:52.32 | joobie | oh |
| 00:52.42 | joobie | sorry flenders, what about BRIs? |
| 00:52.45 | joobie | i dont think they will go for 10 lines |
| 00:52.48 | joobie | 4 lines is more likely |
| 00:52.52 | joobie | 3 voice and 1 fax maybe |
| 00:53.20 | *** join/#asterisk komradebob (n=komradeb@164.55.254.106) |
| 00:53.50 | joobie | damn ill be wrapped once i get all this hardware crap out of the way and get stuck into asterisk |
| 00:54.30 | komradebob | so I create a meetme conference with a pin. I dial into the conference and no matter what I do, I cannot get logged in. almost as if the dtmf isn't coming through. ideas? |
| 00:54.48 | flenders | I think the only BRI card from digium is that 4 port, can someone correct me if I'm wrong? |
| 00:55.04 | flenders | B410P |
| 00:57.03 | flenders | I know JT has an octal BRI, but not sure what the manufacturer is |
| 00:57.06 | joobie | does that have EC? |
| 00:57.24 | joobie | The B410P features on-board hardware echo cancellation performing 64ms or 512 taps per channel for each of the eight voice channels. |
| 00:57.39 | flenders | :D |
| 00:57.43 | joobie | 64ms or 512taps per channel.. is that decent EC? |
| 00:57.51 | joobie | damn i have a lot of learning ahead. |
| 00:58.08 | joobie | the price is a lot better tho for that card |
| 00:58.24 | flenders | techie: thanks mate |
| 00:58.32 | flenders | http://www.junghanns.net/en/octoBRI_produkt.html |
| 00:58.37 | flenders | I think that's JT's card |
| 00:59.11 | flenders | http://www.junghanns.net/en/duoBRI2_produkt.html |
| 00:59.53 | joobie | which do you recommend flenders ? the B410P or that? |
| 01:00.13 | joobie | ive heard good things about digum |
| 01:00.16 | flenders | mate, I don't have experience with BRI |
| 01:00.24 | joobie | ahh |
| 01:00.26 | flenders | I guess their both decent cards |
| 01:00.57 | *** join/#asterisk Kumbang (n=macan@router-wmi.paume.ITB.ac.id) |
| 01:01.04 | joobie | so flenders if i go for that setup.. it will mean i get ISDN BRI / PRI and an associated card.. then run it into the box direct |
| 01:01.06 | flenders | :d |
| 01:01.24 | flenders | yes |
| 01:01.25 | joobie | from there it goes through the aasterisk box and shoots out a standard ethernet port? |
| 01:01.35 | flenders | yes |
| 01:01.44 | flenders | to the sip handsets |
| 01:02.00 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 01:03.45 | flenders | joobie: going to a meeting, back in a bit |
| 01:04.13 | BZBW | Anyone aware such a bug: a SIP 200 OK with SDP specifying a private IP, and * just went ahead send the RTP(in fact it's UDPTL) to the private IP, this SIP 200 OK is in respond to Re-Invite |
| 01:04.39 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:06.10 | bulle | BZBW: i have limited experience with asterisk and sip, but i have seen that type of situation, eg, remote sip phone places call, and in the SDP the remote end specifies a private ip, then asterisk indeed tries to contact that private ip, atleast in my small setup |
| 01:06.41 | blackk | BZBW: is nay=yes specified for the user/peer that's reinviting with private ip? |
| 01:06.54 | joobie | cheers flenders |
| 01:08.04 | komradebob | ok. definitly not getting dtmf through the upstream SIP provider. :- |
| 01:08.07 | komradebob | :-( |
| 01:10.54 | BZBW | Here is what I was trying to do: send FAX via my ATA to a remote ATA, going through *, my * version is 1.4.2, and when remote ATA receive a FAX tone and detected it is a FAX, it sends a reInvite with T.38 specification in the SDP, * recognize it and respond to the remote ATA, but when it pass the Re-Invite to the local ATA and received the 200 OK, it started to send UDPTL(T.38 packets) base on the private IP from 200ok fr |
| 01:11.53 | BZBW | I set NAT=Yes, because my ATA is behind the NAT |
| 01:12.40 | bulle | BZBW: im not sure i understand your problem, but have you set canreinvite=no on your sip.conf entry for your local ata ? |
| 01:13.18 | bulle | or am i mixing up what box is sending the private ip |
| 01:13.27 | BZBW | yes, becasue it's behind the NAT, it's is set to NO for canreinvite |
| 01:13.31 | blackk | BZBW: at least in RTP, Asterisk is sending to the private IP until it gets RTP packets from another IP, this way detecting that the destination is behind NAT; do you get any UDPTL from the device itself? |
| 01:14.43 | BZBW | no, that's the problem, my ATA is waiting for the far end to send UDPTL back to it. |
| 01:15.31 | BZBW | but how can * detect that the UDPTL packets is intended for the right T.38 (or any RTP) session? |
| 01:15.33 | blackk | BZBW: I'm not sure that a solution exists in this case; how would Asterisk know the port to send to? (let's see that the IP can be taken from the SIP messages) |
| 01:15.49 | blackk | BZW: from the port number |
| 01:16.10 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:16.25 | blackk | BZBW: it gets RTP on a certain local port number so it deduces that it's from that particular NATed device |
| 01:16.32 | BZBW | How does * recognize the device behind the NAT for RTP session in the first place? |
| 01:17.11 | blackk | BZBW: it recognizes it when the device start sending RTP to the Asterisk |
| 01:17.28 | blackk | BZBW: if there is no RTP coming in from the NATed device, there is no way to know |
| 01:18.00 | BZBW | it has to map the public IP with the SIP session and then conclude it is the session for that particular call, be it a T.38 or not |
| 01:18.48 | blackk | BZBW: you also need the port; don't forget that the NAT routers can change the port number when masq |
| 01:19.21 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:19.39 | BZBW | yes, but the device behind the NAT normally has no idea what port it is assigned to for it's outgoing RTP |
| 01:20.08 | blackk | BZBW: exactly, that's why you need incoming stream on the Asterisk to determin where to send |
| 01:20.12 | BZBW | if it's not using STUN/ICE:( |
| 01:21.43 | BZBW | thus, if I may to draw a conclusion, so long the ATA behind the NAT can send a few packets, in this case, UDPTL packets for T.38 fax, * should recognized the port for that T.38 session |
| 01:22.11 | blackk | yes, teoretically yes, if it's not a bug :) |
| 01:22.41 | blackk | but make sure that you get packets on the Asterisk box on the port specified by Asterisk when reinviting |
| 01:25.25 | BZBW | I just reviewed the trace file again, it seems that the local ATA is using the same local RTP port for sending T.38 fax. Thus initially, the call went through correctly, thus, not sure why * can not use the same port for passing through the UDPTL packets |
| 01:26.24 | blackk | BZBW: there were a few bugs with T38 and NAT but lately there were no complaints, maybe there is indeed a bug |
| 01:27.57 | BZBW | blackk: thx:) |
| 01:28.49 | blackk | BZBW: just wanted to make sure that it's not a config problem, you can start digging for the T38 bugs, there were 2 if I remember correctly |
| 01:33.10 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 01:33.50 | anonymouz666 | anybode in here running ast 1.4.2 in a call center system? |
| 01:33.54 | anonymouz666 | body |
| 01:34.39 | *** part/#asterisk jjshoe (n=jjshoe@cpe-76-175-157-242.socal.res.rr.com) |
| 01:34.50 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:41.19 | *** join/#asterisk Az_au (n=az@216.127.73.119) |
| 01:41.30 | *** join/#asterisk shaun33 (i=PPCUzer@cslnk01.ispdr.net.au) |
| 01:42.52 | BZBW | blackk: sorry, just got sidetracked:( |
| 01:43.06 | BZBW | what 2 things that you referring to? |
| 01:43.36 | BZBW | I know there is a udptl.conf and in sip.conf, I turn on t.38 |
| 01:44.26 | blackk | BZBW: two different T38 bugs related to NAT, I don't remember the numbers now, if you cannot find them I'll look for them |
| 01:47.58 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 01:48.00 | *** part/#asterisk shaun33 (i=PPCUzer@cslnk01.ispdr.net.au) |
| 01:50.25 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:53.43 | Mercestes | I just installed a sangoma card. When I call my main line on pri1, I hear a loud tone but asterisk CLI shows my playback IVR stuff like normal. Everything is green. Any ideas? |
| 01:54.26 | blackk | BZBW: bug 7679 |
| 01:54.40 | blackk | BZBW: 9222 |
| 01:55.13 | blackk | BZBW: and you find some more in the related to section |
| 01:58.49 | *** join/#asterisk Mportnoy (n=test@200.122.158.88) |
| 01:59.07 | *** join/#asterisk stridernzl (n=neville@125-237-110-53.jetstream.xtra.co.nz) |
| 01:59.36 | Mportnoy | anybody knows what is the best and free software for Outlook dialer? |
| 02:06.09 | BZBW | blackk: thx! |
| 02:06.32 | BZBW | later:) |
| 02:06.44 | blackk | BZBW: they were mostly closed because no negative feedback, if you find a problem just reopen them |
| 02:11.39 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-58-152.socal.res.rr.com) |
| 02:13.57 | mihinomenest | are there any examples of outbound SIP trunks? |
| 02:21.01 | Mercestes | No help for loud tone on Sangoma A104d? |
| 02:21.01 | Mercestes | on all ports. |
| 02:26.12 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-144-195.ny325.east.verizon.net) |
| 02:27.27 | flenders | what are SIP trunks? |
| 02:28.02 | flenders | is that a trixbox thing? |
| 02:30.23 | *** join/#asterisk shaun33_away (i=PPCUzer@cslnk01.ispdr.net.au) |
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| 02:30.33 | *** part/#asterisk shaun33_away (i=PPCUzer@cslnk01.ispdr.net.au) |
| 02:31.54 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 02:34.47 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 02:34.47 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:43.37 | *** join/#asterisk propagandhi (n=propagan@144.131.132.107) |
| 02:43.44 | propagandhi | howdy people |
| 02:43.52 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:44.17 | *** join/#asterisk shodan (n=shodan@ip154.99-113-216.pppoe4.joliette.intermonde.net) |
| 02:44.17 | *** join/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 02:45.13 | propagandhi | I have an AGI script which is implemented in PHP, when a call comes in, it gets the display name we want based on the DIN/DID and then plays the caller a greeting. However the CDR logs then show the time from the minute the call hits the system all the way until the end of the converstation, so its showing all the talk time and hold time together |
| 02:45.30 | propagandhi | is this a case when I should use ResetCDR |
| 02:46.46 | bpiper | Guys, I have been trying to install asterisk, but for the life of me, I cant figure it out. I downloaded 1.2.17, unzip'd, then make clean, make, make install... when I connect via asterisk -cvvvvvvv it doesn't... am I doing something wrong with the install? |
| 02:47.09 | bpiper | FYI, no error during the compiling |
| 02:47.32 | propagandhi | bpiper: is the asterisk process running? |
| 02:47.34 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 02:47.42 | Qwell | mitcheloc: !!! |
| 02:47.43 | Damin | What hppens when you start it w/ asterisk -vvvvvfg from the command line? |
| 02:47.58 | Qwell | Damin: You use -f? |
| 02:48.00 | mitcheloc | whats up j |
| 02:48.09 | Qwell | mitcheloc: not a lot |
| 02:48.13 | Damin | Qwell: Always have.. |
| 02:48.25 | Damin | Qwell: When I'm testing my configs for a first install.. yeah.. |
| 02:48.33 | Damin | Qwell: When I run it regularly, no.. |
| 02:48.41 | Qwell | why not -c? |
| 02:49.07 | bpiper | it goes through manager registered action, then parsing manager.conf & manager_custom.conf then dies |
| 02:49.17 | Qwell | _custom? |
| 02:49.20 | Qwell | ~freepbx |
| 02:49.30 | jbot | i guess freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:49.32 | bpiper | no, it's fresh from asterisk's site |
| 02:49.46 | bpiper | 1.2.17 |
| 02:49.46 | Damin | Qwell: Cause I don't think it was actually implemented when I started using -vvvfg :) |
| 02:49.53 | Qwell | Damin: fair enough |
| 02:50.05 | Qwell | bpiper: Your configs are from freepbx. Remove them and start fresh. |
| 02:50.13 | bpiper | ok |
| 02:51.11 | Damin | bpiper: rm -f /etc/asterisk ; cd /usr/src/asterisk ; make samples |
| 02:51.19 | Damin | At least I think it's make samples.. |
| 02:51.22 | Damin | It might be make config |
| 02:51.25 | Qwell | samples |
| 02:51.32 | Qwell | and -f won't kill the dir |
| 02:51.41 | Damin | -fr ;) |
| 02:51.54 | Qwell | -c! :P |
| 02:51.55 | bpiper | I moved the .conf files from another server, I'd rather not overwrite them |
| 02:51.58 | Qwell | (kidding...) |
| 02:52.01 | Mercestes | <PROTECTED> |
| 02:52.09 | Qwell | -d? |
| 02:52.11 | Mercestes | Qwell: are you any good with PRIs? |
| 02:52.16 | Qwell | nope |
| 02:52.18 | Mercestes | <PROTECTED> |
| 02:52.24 | Mercestes | --verbose --force --recursive |
| 02:52.25 | CrashHD | what do you need with a pri? |
| 02:52.26 | *** join/#asterisk kje (n=kje@lime-gw16.one.at) |
| 02:52.28 | Qwell | woah |
| 02:52.32 | Qwell | -d is evil |
| 02:52.38 | Mercestes | -d is evil |
| 02:52.48 | Mercestes | I tried two digium cards and one sangoma card in a PowerEdge 2850. |
| 02:53.09 | Mercestes | Digum cards failed patlooptest on anything but /dev/zap/1 Just gave "device busy." So I tried a sangoma card... |
| 02:53.16 | CrashHD | if it's a dell server chances are you'll need sangoma |
| 02:53.19 | *** part/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 02:53.24 | Qwell | not true at all |
| 02:53.27 | CrashHD | because they use the E1000 driver |
| 02:53.32 | Mercestes | Sangoma got errors on /dev/zap/1 and couldn't run on /dev/zap/2 (couldn't access or something) and I get a solid moderately deep tone on all my ports, in bound and outbound |
| 02:53.42 | Mercestes | with no audio passthrough |
| 02:53.46 | Mercestes | I followed the wiki |
| 02:54.06 | CrashHD | which version of firmware/software? |
| 02:54.14 | Mercestes | and I still got the annoying "D=chan on span 3 is up" message |
| 02:54.46 | Mercestes | 2.3.4-7 |
| 02:54.54 | Mercestes | and zaptel 1.2.15 |
| 02:55.00 | CrashHD | I just went through that install yesterday |
| 02:55.04 | CrashHD | upgrade anyway |
| 02:55.10 | CrashHD | you make sure you have the right firmware? |
| 02:55.33 | Mercestes | Yea. This PE has one of those riser "all in one" cards that has one PCIx and 2 PCIe on one big card that all hammers down into on eslot on the mobo |
| 02:55.47 | Mercestes | I got it from Sangoma a week ago but....no I didn' tmake sure. |
| 02:56.22 | CrashHD | double check that, go to the wiki firmware update page, you will find instructions on checking the current version |
| 02:56.35 | Mercestes | would that generate a loud tone? |
| 02:56.39 | Mercestes | or is that up in the air? |
| 02:56.48 | *** join/#asterisk hansin321 (n=hansin32@c-67-174-180-72.hsd1.co.comcast.net) |
| 02:57.04 | CrashHD | no clue, but the new software version requires a certain firmware version minimum |
| 02:57.43 | Mercestes | Yea I'll give that a look tomorrow. i have different PE at a Colo I want to test this Digum and sangoma card on with patlooptest to see if it passes |
| 02:57.51 | Mercestes | while I'm at it I'll give the firmware a look |
| 02:58.05 | *** join/#asterisk shaun33_away (i=PPCUzer@cslnk01.ispdr.net.au) |
| 02:58.43 | CrashHD | good luck |
| 02:58.51 | CrashHD | sangoma is definitely the card I would use |
| 02:58.58 | CrashHD | great card |
| 02:59.36 | Mercestes | I agree. I've had excellen tluck with them in the past and....that's why I'm starting to suspect this server. |
| 03:00.02 | Mercestes | either that or I totally suck |
| 03:00.11 | Mercestes | That's always a possibility |
| 03:00.27 | CrashHD | heh |
| 03:00.55 | sumasuma | Looks like Digium was poor in making the best telephony cards |
| 03:01.11 | sumasuma | was and is |
| 03:01.30 | CrashHD | great thing about this thing we call time |
| 03:01.34 | CrashHD | it's in constant motion |
| 03:01.39 | CrashHD | gives them the chance to change |
| 03:01.41 | CrashHD | :) |
| 03:02.04 | CrashHD | many people use digium cards with great success |
| 03:02.09 | *** part/#asterisk shaun33_away (i=PPCUzer@cslnk01.ispdr.net.au) |
| 03:02.34 | Mercestes | oh well, I'm going to head home now. TTYL. Thanks for the help, CrashHD |
| 03:02.37 | sumasuma | CrashHD: i did not mentioned success here, u said only many people, but not 90 or 100% satisfied |
| 03:02.39 | Mercestes | I'll let you know how the other srever tests go |
| 03:02.44 | CrashHD | no problem Mercestes, good luck |
| 03:03.15 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 03:08.39 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:17.30 | *** join/#asterisk znoG (n=gs@24.232.95.132) |
| 03:17.33 | znoG | howdy |
| 03:17.55 | znoG | somewhat OT: anyone recommend a VoIP provider with good international call rates? |
| 03:20.09 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 03:26.21 | CrashHD | I use vitelity |
| 03:26.29 | CrashHD | I find they have decent international rates |
| 03:27.23 | znoG | not bad, they have the best rate I've found so far |
| 03:27.32 | znoG | (in like... 20-30 i researched) |
| 03:27.37 | znoG | hows the call quality? |
| 03:27.45 | znoG | whats the IP of their server? (to test latency) |
| 03:28.20 | CrashHD | good call quality |
| 03:28.49 | znoG | got the IP handy? |
| 03:29.09 | CrashHD | outbound1.vitelity.com |
| 03:30.49 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-144-57.ny325.east.verizon.net) |
| 03:31.14 | znoG | mmm i can't resolve the host |
| 03:33.31 | CrashHD | outbound1.vitelity.net |
| 03:33.33 | CrashHD | sorry |
| 03:36.59 | znoG | :) thanks |
| 03:37.08 | znoG | hmmm 220ms average |
| 03:37.21 | znoG | i'd have to see what the call quality is like at that latency |
| 03:38.41 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 03:38.48 | yidiyuehan | hi, everyone. |
| 03:39.44 | yidiyuehan | if my server is not using intel processor, can i install the free G729 codec on it based on intel free evaluated coding? |
| 03:41.20 | JunK-Y | which architecture? x86? |
| 03:42.06 | yidiyuehan | ASUS, P5L-MX motherboard |
| 03:42.56 | JunK-Y | which architecture for the OS? |
| 03:43.24 | JunK-Y | ya probably can if its x86. |
| 03:44.19 | yidiyuehan | centos 4.4 |
| 03:44.29 | yidiyuehan | and it's EM64T |
| 03:48.31 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:49.42 | JunK-Y | dunno, giving a try should be the best option i think. |
| 03:53.15 | *** join/#asterisk bmg505 (n=leon@196.209.248.245) |
| 03:54.09 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 03:54.46 | yidiyuehan | JunK-Y, what i can try to see? |
| 03:55.31 | JunK-Y | try the g729 codecÉ |
| 03:55.32 | JunK-Y | ? |
| 03:56.22 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:56.44 | yidiyuehan | ok, i will try to use G729 and see how. if i can make the calls then which means it's fine right? |
| 03:57.29 | JunK-Y | if u do core show translation and ya see 729 in there, you're in business. |
| 03:58.00 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 03:58.57 | yidiyuehan | yes, in fact without calling trial, i can see the codecs are there |
| 03:59.29 | JunK-Y | so where ur codec_g729 comes from? |
| 03:59.29 | yidiyuehan | <PROTECTED> |
| 03:59.44 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:01.40 | yidiyuehan | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 04:01.50 | yidiyuehan | i followed this link exactly and installed them |
| 04:01.59 | JunK-Y | so it should be okay |
| 04:02.16 | JunK-Y | just define a peer as g729 and call meetme and see. |
| 04:02.57 | yidiyuehan | okie, |
| 04:03.14 | *** part/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 04:03.35 | JunK-Y | it should be able to create a path from 729 to slin |
| 04:04.13 | *** join/#asterisk sharp (i=sharp@gateway/tor/x-abb6265a0cb054c8) |
| 04:05.14 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:05.54 | yidiyuehan | http://www.asus.com/products4.aspx?modelmenu=2&model=1320&l1=3&l2=11&l3=194 |
| 04:06.13 | yidiyuehan | this is the motherboard i am using currently. it's not EM64T... |
| 04:07.11 | propagandhi | is there a way to play a message only at the time that the caller is joining the queue? |
| 04:09.42 | CrashHD | right before the Queue() use Playback() |
| 04:09.57 | [TK]D-Fender | propagandhi, Sure... just play the file BEFORE calling Queue... |
| 04:10.04 | CrashHD | or put a super lengthy time on the repeat timing |
| 04:10.16 | CrashHD | tk great minds do think alike :) |
| 04:10.24 | *** join/#asterisk kje (n=kje@62-99-209-38.c-vzollerg.xdsl-line.inode.at) |
| 04:10.50 | SplasPood | If I have a wav file I'd like to make a voicemail greeting, do I need to do anything special other than convert it to the 3 formats and place it in the proper place? |
| 04:11.07 | propagandhi | the only problem i have with that is that it delays the time the call can be answered, because technically it hasn't taken a position in the queue yet |
| 04:12.31 | [TK]D-Fender | CrashHD, Fools seldom differ ;) |
| 04:12.38 | CrashHD | hah |
| 04:13.05 | [TK]D-Fender | SplasPood, You don't ahve to convert it. Thats what transcoding is for. |
| 04:13.35 | [TK]D-Fender | propagandhi, Then its time to start coding, because what you want doesn't exist yet. |
| 04:14.04 | propagandhi | hmm, i thought as much |
| 04:14.11 | propagandhi | just clutching at straws is all |
| 04:15.32 | fetcher | a customer with remote Polycom 301's (g.729) reports a "squeal" sound when transferring calls to an outside number. Is there any common mistake or bug that might cause that? |
| 04:15.47 | fetcher | or maybe just a Polycom 301 quirk? |
| 04:16.41 | [TK]D-Fender | fetcher, Does it happen under G711? |
| 04:17.13 | [TK]D-Fender | fetcher, And should I raise a brow at you use of an "outside number"? |
| 04:17.26 | fetcher | [TK]D-Fender: haven't tried G.711 yet... that's on the agenda for testing tomorrow |
| 04:18.10 | SplasPood | [TK]D-Fender: ahh I just assumed since voicemail() itself seems to store them in 3 different formats |
| 04:18.50 | fetcher | [TK]D-Fender: outside PSTN numbers, as opposed to another SIP phone. PRI connection is through an AS5300, but * stays in the media path (reinvites disabled) |
| 04:19.10 | [TK]D-Fender | SplasPood, Maybe because you specify it in voicemail.conf perhaps? :) You should really pay attention to the BIG PRINT.... |
| 04:19.26 | SplasPood | [TK]D-Fender: heh, its the default, no? |
| 04:19.49 | [TK]D-Fender | fetcher, So non AS5300 calls = fine? Guess you know what to blame now don't you? |
| 04:20.07 | [TK]D-Fender | SplasPood, Get off your ass and actualyl ready your config file! ;) |
| 04:20.26 | SplasPood | [TK]D-Fender: heh, its been in my config file for so long I believed it to be the default |
| 04:20.31 | [TK]D-Fender | <- Zen master of the blatantly obvious |
| 04:20.59 | fetcher | [TK]D-Fender: except that ordinary incoming and outgoing calls (SIP phones <-> PSTN) also traverse the same AS5300, with no troubles. It's only the transfer case that gives the (temporary) squeal, which seems like a strange failure mode |
| 04:21.30 | [TK]D-Fender | fetcher, Sounds like a device isn't handing off RTP fast enough. |
| 04:21.45 | [TK]D-Fender | fetcher, first try another codec, then move on. |
| 04:21.49 | SplasPood | [TK]D-Fender: just double checked.. that *IS* the default! ;) |
| 04:22.01 | [TK]D-Fender | fetcher, I've never heard any peculiarities like tha before from them. |
| 04:22.24 | [TK]D-Fender | SplasPood, What exactly defines "defualt"? |
| 04:23.04 | fetcher | if G.711 makes no difference, I'll try swapping one phone for a Polycom 501 |
| 04:31.46 | SplasPood | [TK]D-Fender: well, in this case.. in the shipping example config file |
| 04:33.06 | [TK]D-Fender | SplasPood, well then thats not "default", its just a SAMPLE. And what are we supposed to do? Read the samples and make our OWN. |
| 04:33.24 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:33.39 | SplasPood | why would you want to have it stored in multiple formats? To save the CPU from the transcoding? |
| 04:34.18 | [TK]D-Fender | SplasPood, And means you don't need to convert when e-mailing* them since one format is popularly supported.... |
| 04:34.47 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 04:35.13 | jql | hold.ulaw = hold music, brahms in g minor or something. hold.g729 = me saying "you're still on hold" into my headset, looped |
| 04:35.51 | jql | to suit the audio bandwidth of the consumer, of course |
| 04:37.59 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 04:40.53 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:42.01 | [TK]D-Fender | jql, ...huh? What does the content have to do with the codec? |
| 04:49.17 | SplasPood | i'd imagine certain codecs work better with speech vs music etc |
| 04:52.41 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 04:52.54 | [TK]D-Fender | SplasPood, don't let your imagination wander... its too little to be let out alone ;) |
| 04:54.43 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:54.51 | [TK]D-Fender | ok, late and tired.... back later... |
| 05:02.18 | *** join/#asterisk juest (n=juest@c-67-162-30-71.hsd1.il.comcast.net) |
| 05:03.08 | juest | hello |
| 05:07.45 | *** join/#asterisk CuriosCat (n=stian@mack.bigrig.org) |
| 05:07.48 | CuriosCat | Howdy all |
| 05:09.20 | haroldp | I have a DID form teliax that works if I plug it into a sip phone. How do I add it to asterisk? |
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| 05:17.09 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-144-57.ny325.east.verizon.net) |
| 05:17.27 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-144-57.ny325.east.verizon.net) |
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| 06:00.50 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 06:03.32 | *** join/#asterisk nsphang88 (i=nsphang@203.118.15.20) |
| 06:03.56 | nsphang88 | Hi, is there any Asterisk consultant or users who does freelance work on implementing Asterisk servers? |
| 06:05.46 | hads | Yes |
| 06:11.09 | *** join/#asterisk voltagex (n=voltagex@124-254-124-5-dsl.ispone.net.au) |
| 06:11.37 | voltagex | how do I get music on hold working with MP3? one tutorial says I need zaptel, the other doesn't |
| 06:15.11 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 06:18.54 | tzafrir | voltagex, do you need to stream mp3 from a remote source? |
| 06:19.12 | voltagex | tzafrir: no, local mp3 |
| 06:19.32 | tzafrir | If not, it would probably help you much to downsample it and convert it to wav |
| 06:19.44 | voltagex | :? |
| 06:19.45 | voltagex | :/ |
| 06:20.12 | tzafrir | (of the type that Asterisk likes: 8000Hz, mono, 16 bits/sample |
| 06:20.17 | voltagex | yeah |
| 06:20.40 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:20.56 | voltagex | but I was hoping to have plain mp3s |
| 06:21.02 | tzafrir | And this off-line transcoding will save you doing the same transfcoding at runtime |
| 06:21.18 | voltagex | true, but this box will probably play hold music once a day |
| 06:22.05 | tzafrir | And chances are you'll get a file of approximately the same size or even smaller (due to downsampling) |
| 06:22.22 | voltagex | file size is not a problem |
| 06:22.27 | voltagex | there's still 17gb free on the drive |
| 06:22.47 | voltagex | I'm just compiling asterisk-addons now |
| 06:23.16 | nsphang88 | Hi, is there any Asterisk consultant or users who does freelance work on implementing Asterisk servers? |
| 06:24.43 | hads | Yes, I'm sure there are plenty |
| 06:24.55 | *** join/#asterisk Caplain (n=matt@adsl-75-45-249-77.dsl.sfldmi.sbcglobal.net) |
| 06:25.09 | Caplain | do i need the g729 codec for nufone? |
| 06:25.09 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 06:29.39 | voltagex | <PROTECTED> |
| 06:29.52 | voltagex | does not exist in any format |
| 06:33.41 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 06:34.16 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
| 06:34.47 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:35.16 | ManxPower | I'm trying to figure out what the best version of the HPEC would be for my system. "Celeron" is not listed. Pentium 4M, Pentium 3M and Prescot are listed. Anyone have any suggestions? |
| 06:39.03 | nsphang88 | hads: any recommendation that you can give me? |
| 06:40.33 | voltagex | how do I install the format_mp3 for Asterisk? |
| 06:41.38 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 06:44.51 | *** part/#asterisk billytwowilly (n=chris@S01060016b649355d.ed.shawcable.net) |
| 06:50.14 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 06:59.31 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 07:00.08 | CunningPike | voltagex: Download and untar asterisk-addons. Then enter the formatmp3 folder and 'make && make install' |
| 07:00.46 | voltagex | CunningPike: ok |
| 07:00.48 | voltagex | thanks |
| 07:00.53 | CunningPike | voltagex: np |
| 07:00.55 | voltagex | ah, my mp3s have to be 8000hz anyway |
| 07:01.35 | CunningPike | voltagex: They should be 8000hz mono |
| 07:01.52 | voltagex | aha, that would be my problem |
| 07:05.34 | *** join/#asterisk haiger (n=piper@ip51.haverhill-highland.sfo.ygnition.net) |
| 07:07.03 | *** join/#asterisk viler (n=mv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 07:07.33 | haiger | i have a problem, i read asterisktfot and the instruction was talking about install zaptel, but for now i only want to use a headset |
| 07:08.02 | haiger | can someone help me please |
| 07:09.34 | ManxPower | There is mouse droppings in my server!!!! |
| 07:10.38 | ManxPower | haiger: Asterisk is designed to be a PBX, not a telephone. It might be easier for you to install something like X-Lite on a Windows PC for calls. |
| 07:12.42 | haiger | ManxPower: please explain to me i am so confused |
| 07:16.19 | mitcheloc | Asterisk isn't a telephone! :( |
| 07:18.10 | haiger | mitcheloc: i know |
| 07:19.23 | *** join/#asterisk cholzi (n=root@70-162-28.dial.terra.cl) |
| 07:19.57 | haiger | ~PBX |
| 07:20.19 | jbot | well, pbx is a Private Branch eXchange |
| 07:20.19 | cholzi | quit |
| 07:21.00 | mitcheloc | i want to use my Asterisk telephone though =/ |
| 07:21.00 | haiger | jbot: i know it a private branch exchange, i thought i can get an explaniation |
| 07:21.02 | jbot | You know it a private branch exchange, i thought i can get an explaniation? |
| 07:21.34 | haiger | what pbx |
| 07:22.51 | *** part/#asterisk viler (n=mv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 07:22.53 | *** join/#asterisk cholzi (n=chris@70-162-28.dial.terra.cl) |
| 07:23.22 | fetcher | haiger: PBX == automated telephone switchboard, like a smaller version of what the phone companies use |
| 07:24.19 | haiger | fetcher: so if i want to make phone calls outside usa , i don't need asterisk |
| 07:25.07 | fetcher | haiger: if it's just you, and you have a service provider to connect to, probably not |
| 07:25.40 | fetcher | haiger: you probably want a "softphone" program instead, e.g. SJPhone |
| 07:29.20 | haiger | fetcher: but i really want to use asterisk |
| 07:29.39 | haiger | i want to learn about it |
| 07:31.58 | fetcher | haiger: you could use it as a conduit to route calls through (softphone -> asterisk -> VoIP provider), which can have some benefits. Or to provide voicemail, menu prompts etc. |
| 07:33.42 | *** join/#asterisk cholzi (n=chris@70-162-28.dial.terra.cl) |
| 07:33.47 | haiger | fetcher: yes , thats what i want to do |
| 07:35.30 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 07:38.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:39.59 | *** join/#asterisk cholzi (n=chris@70-162-28.dial.terra.cl) |
| 07:40.15 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 07:41.52 | *** join/#asterisk cholzi (n=chris@70-162-28.dial.terra.cl) |
| 07:42.43 | *** join/#asterisk zeeesh (i=GrTB0T@202.38.55.125) |
| 07:42.44 | zeeesh | hi |
| 07:43.16 | tcseke | Hello, can anybody advice me an application for fax detection, i found only app_nv_faxdetect, is it stable, or are there any other solution for fax detection? |
| 07:43.17 | haiger | fetcher: how can i swtup a voip with astreisk |
| 07:43.20 | *** join/#asterisk cholzi (n=chris@70-162-28.dial.terra.cl) |
| 07:49.42 | nsphang88 | Hi, is there any Asterisk consultant or users who does freelance work on implementing Asterisk servers? |
| 07:52.49 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 08:04.44 | *** join/#asterisk friedrich| (n=friedric@e177246155.adsl.alicedsl.de) |
| 08:05.19 | voltagex | any ideas why Asterisk would ignore random in musiconhold.conf ? |
| 08:11.23 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 08:16.16 | *** join/#asterisk saftsack (n=oliver@p54A7E9B2.dip.t-dialin.net) |
| 08:21.17 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 08:21.33 | yidiyuehan | hi, guys. with incoming call, dialparties.agi: Caller ID name is 'unknown' number is 'unknown' |
| 08:21.42 | yidiyuehan | how i can set to get the pstn caller id with TDM400p card? |
| 08:21.43 | *** join/#asterisk shay|work (i=shay@unaffiliated/shay) |
| 08:21.44 | shay|work | hello |
| 08:21.52 | yidiyuehan | this is my zapata.conf .http://www.pastebin.ca/425107 |
| 08:22.00 | yidiyuehan | and this is the CLI for incoming call.http://www.pastebin.ca/425112 |
| 08:23.03 | CrazyTux | Anyone here messed with asterisknow gui? |
| 08:23.30 | hads | yidiyuehan: /topic |
| 08:24.03 | yidiyuehan | hads, what do u mean,? sorry i could not get your point?/ |
| 08:24.17 | hads | Read the topic |
| 08:27.12 | yidiyuehan | yes,i do asked in freepbx, maybe they are all busy now. sorry for any inconvience caused as i have searched everything i could but don't know the reason. |
| 08:32.48 | *** join/#asterisk pkempgen (n=pkempgen@ACB3AA1D.ipt.aol.com) |
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| 08:58.18 | *** join/#asterisk voltagex (n=voltagex@124-254-124-5-dsl.ispone.net.au) |
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| 09:12.28 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:26.59 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 09:26.59 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:31.05 | *** join/#asterisk saftsack (n=oliver@p54A7E25C.dip.t-dialin.net) |
| 09:38.14 | hermuli | i got yet another simple question; when i record a call it automatically ends after one minute exactly. I haven't set a maxduration for the Record(). what could be wrong? |
| 09:39.48 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 09:39.53 | FreezeS | hey guys |
| 09:40.17 | FreezeS | anyone knows what is the "setvar" field from sip_buddies do ? |
| 09:40.53 | hads | No idea what sip_buddies is but I imagine it sets a variable |
| 09:41.13 | FreezeS | sip_buddies is the table used by asterisk realtime to store the sip users |
| 09:41.29 | hads | Ah, never used realtime |
| 09:42.18 | hads | There's a setvar option in sip.conf I believe. I've never used it but saw some info on voip-info the other day. |
| 09:42.30 | FreezeS | aha, let's check |
| 09:44.44 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:44.48 | hads | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 09:47.09 | FreezeS | thanks |
| 09:47.29 | FreezeS | it worked :) |
| 09:47.35 | hads | np :) |
| 09:47.46 | *** join/#asterisk ecam (n=ecam@bb121-6-58-13.singnet.com.sg) |
| 09:49.20 | ecam | anybody around to answer a newbie's question? |
| 09:49.58 | tcseke | Hello, can anybody advice me an application for fax detection, i found only app_nv_faxdetect, is it stable, or are there any other solution for fax detection? |
| 09:50.00 | voltagex | any ideas why Asterisk would ignore random in musiconhold.conf ? |
| 09:57.24 | *** join/#asterisk saftsack (n=oliver@p54a7e20f.dip.t-dialin.net) |
| 10:00.47 | ecam | okay, nobody's around, so I shall just leave a message. I have a voip account with my isp and I want to be able to attach more than one phone to it, but only one account can login to my isp at any time, no multiple logins from the same account allowed. I was wondering if I can run a SIP gateway that connects to my isp's account as a main line, then connect all my phones to the gateway, and configure the gateway in such a way that when a call is |
| 10:01.11 | ecam | does asterisk fit the bill? |
| 10:01.48 | hads | Yes, along with other projects. |
| 10:02.06 | ecam | how can i do it? this is my first time dealing with voip |
| 10:02.34 | hads | If you want to do it with asterisk then reading the book would be a good place to start. |
| 10:02.39 | hads | ~thebook |
| 10:02.41 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:02.51 | ecam | okay, i'm reading it now, seems managable... |
| 10:03.31 | ecam | so you mean i can use asterisk to pass the calls on from the main line to the phones connected to it |
| 10:03.47 | hads | yes |
| 10:03.53 | ecam | would the information from the main line like caller id and stuff be forwarded as well? |
| 10:03.59 | hads | yes |
| 10:04.30 | *** join/#asterisk kje (n=kje@62-99-209-38.c-vzollerg.xdsl-line.inode.at) |
| 10:04.38 | ecam | and when i make use of the phones to make outgoing calls, would the person on the receiving end see my internal number of my main line's number? |
| 10:07.06 | ecam | and also, if for example, one phone is engaged in an outgoing call, would the other phones be blocked from making outgoing calls or would it be possible to actually automatically add the user into the current outgoing conversation? As in conference calling? |
| 10:16.39 | *** join/#asterisk geoaxis (n=geoaxis@unaffiliated/geoaxis) |
| 10:17.07 | geoaxis | hello : can any one recomend me a ready to consume solution for CRM , which completemts asterisk |
| 10:17.21 | geoaxis | some thing that can be easily used in call centers |
| 10:17.52 | geoaxis | i found these two to be interesting : http://sourceforge.net/projects/covide/ http://sourceforge.net/projects/covide/ |
| 10:18.02 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:18.33 | geoaxis | oops : i meant http://sourceforge.net/projects/bizcom/ as second one |
| 10:28.16 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:46.27 | denon | hmm ... bizcom .. |
| 10:46.34 | denon | one developer, beta, but has a logo |
| 10:46.41 | denon | wonder if that means it doesnt work yet |
| 10:59.56 | *** join/#asterisk saftsack (n=oliver@p54a7f0f0.dip.t-dialin.net) |
| 11:06.52 | *** join/#asterisk friedrich| (n=friedric@85.177.246.155) |
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| 11:13.11 | *** join/#asterisk IPconfig (n=dd@mail.trueafrican.com) |
| 11:17.09 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 11:17.36 | nettie | Hi guys, I'm using asterisk 1.2.17 bristuffed |
| 11:17.50 | nettie | and I'm getting this error message continously Primary D-Channel on span 1 down |
| 11:17.58 | nettie | anyone know what could be wrong please? |
| 11:19.11 | *** join/#asterisk safira (n=serenity@202.5.145.13) |
| 11:19.14 | safira | app_meetme.c:1556 conf_run: Unable to write frame to channel: Success |
| 11:19.25 | safira | how do i get rid of it |
| 11:31.08 | *** join/#asterisk saftsack (n=oliver@p54a7d533.dip.t-dialin.net) |
| 11:35.23 | *** join/#asterisk yenno (n=yenno@152.96.193.209) |
| 11:36.44 | yenno | hi, when i try to call from xlite to xlite i'm getting a 403 error... why's that? |
| 11:43.16 | *** join/#asterisk friedrich| (n=friedric@85.177.246.155) |
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| 11:47.59 | mathai | Hi, is there anybody who could help me with Xorcom TS-1? |
| 11:48.14 | tzafrir | mathai, yes, naturally |
| 11:48.19 | *** join/#asterisk chris_asterisk` (n=test@217.205.167.170) |
| 11:48.24 | chris_asterisk` | afternoon :) |
| 11:50.29 | *** join/#asterisk friedrich| (n=friedric@85.177.246.155) |
| 11:52.33 | zeeesh | i have 2 registered peer at 2 different server... how can i make peer 2 peer call .. ??? |
| 11:54.58 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 11:55.16 | zeeesh | i have already 2 registered peer at the same server and i have been made peer calling ... now i have 2 registered peer at 2 different server how can i make peer 2 peer call??? |
| 11:55.44 | *** join/#asterisk friedrich| (n=friedric@85.177.246.155) |
| 12:02.05 | *** join/#asterisk saftsack (n=oliver@p54a7f985.dip.t-dialin.net) |
| 12:04.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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| 12:13.51 | *** join/#asterisk zeeesh (i=GrTB0T@202.38.55.125) |
| 12:13.53 | zeeesh | hi |
| 12:13.55 | zeeesh | i have already 2 registered peer at the same server and i have been made peer calling ... now i have 2 registered peer at 2 different server how can i make peer 2 peer call??? |
| 12:20.27 | menfin | by configuring iax link ? |
| 12:20.50 | *** join/#asterisk Jynger (n=rip@tti.tt.ee) |
| 12:20.57 | Jynger | hi |
| 12:21.55 | Jynger | is it possible to set up 2 or more asterisk, that share same user/etensions and dialplan config ? |
| 12:22.09 | Jynger | with realtime and sql ? |
| 12:23.33 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:24.04 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:27.07 | Jynger | load balancing somekind maybe? |
| 12:27.30 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 12:31.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:33.59 | tzafrir | Jynger, just sharing configuration: rsync |
| 12:36.13 | *** join/#asterisk saftsack (n=oliver@p54A7D402.dip.t-dialin.net) |
| 12:36.13 | Jynger | but if there are 2 asterisk with identical userconfig and dialplan, how can user1@asterisk1 call user2@asterisk2 ? |
| 12:36.14 | zeeesh | <menfin> : i tried by using exten => xxxx,1,Dial(SIP/${EXTEN}@servername) ??? could not get result .. receiving error at both asterisk "-- SIP/3219472112-08eb6418 is circuit-busy |
| 12:36.14 | zeeesh | <PROTECTED> |
| 12:36.14 | zeeesh | <PROTECTED> |
| 12:36.46 | Jynger | i mean how does asterisk1 know that user2 is registered to asterisk2 |
| 12:37.32 | *** join/#asterisk MrWup (n=neil@80-46-38-124.static.dsl.as9105.com) |
| 12:37.36 | MrWup | hey guys |
| 12:37.41 | zeeesh | i made asterisk1 peer at asterisk2 and as well i made asterisk2 peer at asterisk1 .. |
| 12:38.04 | *** join/#asterisk fo0b (n=rick@81-86-30-78.dsl.pipex.com) |
| 12:38.53 | fo0b | Is there a way to get asterisk to save voicemail files with a unique id/filename? The problem is that it changes the numbering of the voicemails as they move between mail boxes. |
| 12:38.57 | zeeesh | like i have been dialed out through a service provider .. |
| 12:42.40 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 12:43.54 | jtexter3 | I'm having a strange problem that I'm hoping someone can give me pointers with. I have a user in a remote location, with a VPN connection to my office. His phone has been working fine until today. Now, I can see the traffic coming into Asterisk, but Asterisk just sits re-transmitting until it destroys the call. Funny thing is, all other communication with the phone is just fine |
| 12:47.32 | *** part/#asterisk boojit (n=boojit@gw.carter.to) |
| 12:49.22 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
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| 13:01.59 | *** join/#asterisk Fieldy (i=vZ4WE5ry@gentoo/contributor/Fieldy) |
| 13:03.21 | MrWup | anyone know the aastra 9133i? |
| 13:03.32 | MrWup | how can i send a msg to the phone telling it to light an LED? |
| 13:05.31 | errr | is there a way to get the IP of all your SIP clients using perl? |
| 13:08.07 | *** join/#asterisk littleball (n=littleba@bb220-255-154-211.singnet.com.sg) |
| 13:08.09 | *** join/#asterisk etfonhome (n=etfonhom@64.191.188.199) |
| 13:08.44 | littleball | hello, how to specify the caller ID in asterisk A when i use IAX channel call asterisk box B |
| 13:08.45 | littleball | ? |
| 13:09.09 | *** join/#asterisk Kumbang (n=macan@router-wmi.paume.ITB.ac.id) |
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| 13:15.32 | slvrdrgn | hi does anybody has a hint how to disable callwaiting for a SIP-Client |
| 13:16.34 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
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| 13:16.52 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:20.31 | littleball | hi, how to pass channel variables from BOX A to BOX B over IAX ? |
| 13:21.26 | giasai68 | hello |
| 13:21.42 | giasai68 | any suggest regarding zaptel 1.4 with pci bri card? |
| 13:22.25 | giasai68 | I have installed it, but if I digit: modprobe zaptel - appear fatal error message |
| 13:23.39 | geoaxis | i am asking this thing again today : can any one recomend a good CRM solution (opensource ofcourse) to complement Asterisk |
| 13:24.30 | mocker | geoaxis: I thought sugarCRM had asterisk integration... |
| 13:24.59 | geoaxis | well some thing which would be highly useable for a call center |
| 13:28.04 | MrWup | i just cant get BLF to work |
| 13:28.06 | MrWup | ive set a hint |
| 13:28.11 | MrWup | ive set the subscribecontext |
| 13:28.25 | MrWup | but it just wont light up my aastra 9133i when someone gets called |
| 13:29.44 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:29.44 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 13:46.59 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 13:50.30 | tcseke | Hello, could anybody help me? i have a strange problem with app_queue, the members are paused after a while, i found autopause parameter in the source code, that i think could cause this, but i don't set this parameter. and i don't found its default value..., |
| 13:51.17 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
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| 13:54.58 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 14:01.09 | giasai68 | hello |
| 14:01.11 | giasai68 | I have installed it, but if I digit: modprobe zaptel - appear fatal error message |
| 14:01.12 | giasai68 | any suggest regarding zaptel 1.4 with pci bri card? |
| 14:02.07 | littleball | hello, how to forward caller ID from asterisk box A to asterisk Box B? |
| 14:02.22 | blitzrage | littleball: with dialplan logic |
| 14:02.36 | blitzrage | Set(CALLERID(name)=Foo Bar) |
| 14:02.41 | *** join/#asterisk mindCrime_ (n=chatzill@216.187.233.42) |
| 14:02.46 | blitzrage | Set(CALLERID(number)=8005551212) |
| 14:03.03 | blitzrage | you might also have to set trustrpid=yes and sendrpid=yes in sip.conf |
| 14:03.21 | littleball | blitzrage, i am using IAX |
| 14:03.24 | *** join/#asterisk [shodan] (n=shodan@ip010.96-113-216.pppoe1.joliette.intermonde.net) |
| 14:03.47 | littleball | mobile phone -->PSTN-->Asterisk BOX A-->IAX -->Asterisk Box B |
| 14:04.10 | blitzrage | when you dial the other Dial(), use the 'o' flag |
| 14:04.15 | littleball | i tried, |
| 14:04.22 | littleball | o flag. it doesn't work |
| 14:04.24 | blitzrage | then capture the incoming CID and reset it |
| 14:04.33 | blitzrage | with CALLERID() function |
| 14:06.44 | littleball | Set(CALLERID(all)=1234) in box A |
| 14:06.46 | littleball | ? |
| 14:08.05 | geoaxis | has any one tried beryl or compiz here |
| 14:08.15 | geoaxis | oops ..sorry wrong channel |
| 14:10.52 | *** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 14:11.05 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 14:11.17 | blitzrage | littleball: no -- you have to do like I told you above |
| 14:11.26 | littleball | it doesn't work |
| 14:11.41 | littleball | box A get the caller ID of mobile phone |
| 14:12.01 | littleball | and i want box A to pass the caller ID to asterisk Box B over IAX. |
| 14:12.19 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 14:12.26 | SwK | <PROTECTED> |
| 14:12.26 | SwK | <PROTECTED> |
| 14:12.26 | SwK | <PROTECTED> |
| 14:12.28 | SwK | hahah |
| 14:12.31 | SwK | thats what it was |
| 14:12.34 | SwK | sofia asserted |
| 14:12.45 | |Rain| | soooo... is the CUT() function supposed to modify its input string? 'cause, aside from being completely documented behavior, I probably don't have to tell anyone that it's kind of dumb... |
| 14:12.55 | SwK | damnit |
| 14:12.59 | SwK | wrong window sorry |
| 14:12.59 | littleball | blitzrage, anyway, i can work around by exchange caller ID and exten number when using iax |
| 14:15.00 | blitzrage | |Rain|: it doesn't modify the original string at all unless you assign it back to the same string |
| 14:15.51 | |Rain| | that's what I would expect to happen, but that's not what's happening (1.4.2) |
| 14:16.09 | blitzrage | |Rain|: I'm using 1.4 SVN and that's exactly how it works |
| 14:16.29 | |Rain| | I have: |
| 14:16.48 | blitzrage | don't paste here -- use pastebin |
| 14:16.49 | |Rain| | Set(wtf=a-b-c-d); NoOP(wtf pre : ${wtf}); NoOP(wtf: ${CUT(wtf,,1)}); NoOP(wtf post: ${wtf}); |
| 14:16.52 | |Rain| | fine |
| 14:16.59 | blitzrage | well, one line is ok :) |
| 14:17.02 | Qwell | heh |
| 14:17.04 | blitzrage | I was waiting for 30 |
| 14:17.13 | |Rain| | I cut out the newlines |
| 14:17.16 | |Rain| | I know better than that :P |
| 14:17.29 | Qwell | the 2nd and 4th wtf will be the same |
| 14:17.31 | |Rain| | the debug output will be a hell of a lot easier to read if I just use pastebin, so I'll use that |
| 14:17.40 | Qwell | CUT doesn't change the variable |
| 14:18.19 | |Rain| | again, that's what I expect |
| 14:18.22 | |Rain| | http://pastebin.ca/425432 is what happens |
| 14:18.49 | |Rain| | wait a second |
| 14:18.51 | |Rain| | it's fine now |
| 14:18.56 | |Rain| | dammit, hold on |
| 14:18.57 | |Rain| | :E |
| 14:20.12 | *** join/#asterisk ppyy (n=ppyy@222.185.18.127) |
| 14:21.08 | blitzrage | |Rain|: I just did it -- it works as expected |
| 14:21.08 | blitzrage | it does not modify the original string |
| 14:21.30 | |Rain| | right, my test case didn't reproduce the problem... I only thought it did because I misread it |
| 14:21.41 | blitzrage | yep -- layer 8 problem |
| 14:21.45 | |Rain| | it looks like the actual code still does, so I'm trying to reproduce in some simpler code again |
| 14:22.01 | blitzrage | how much simpler can you get? |
| 14:22.15 | |Rain| | that WAS the simplified code |
| 14:22.18 | blitzrage | although I suppose it is possible there is a bug in CUT(), but I make HEAVY use of it and have never run into that |
| 14:24.17 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 14:24.22 | JoelSolanki | hi all |
| 14:24.24 | |Rain| | awww, holy crap. the real problem was that I was doing Set(foo=${bar}); and 'bar' had |s in it |
| 14:24.28 | JoelSolanki | i m getting below error in asterisk console |
| 14:24.30 | JoelSolanki | Apr 5 09:06:55 WARNING[22205]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission 2825869828@61.17.177.28 for seqno 200 (Critical Response) |
| 14:24.30 | JoelSolanki | Apr 5 09:06:55 WARNING[22205]: chan_sip.c:1243 retrans_pkt: Hanging up call 2825869828@61.17.177.28 - no reply to our critical packet. |
| 14:24.35 | |Rain| | *sigh* nevermind me |
| 14:24.39 | JoelSolanki | call gets hungup |
| 14:24.46 | JoelSolanki | any ideas ? |
| 14:25.54 | ManxPower | JoelSolanki: Max retries means "device stopped responding" |
| 14:25.56 | *** join/#asterisk shodan (n=shodan@ip084.99-113-216.pppoe4.joliette.intermonde.net) |
| 14:26.50 | *** join/#asterisk b11d|bbl (n=no@234-200-29-134.hcc.mnscu.edu) |
| 14:26.58 | JoelSolanki | so this is normal ? |
| 14:27.11 | b11d | hi all |
| 14:28.14 | *** join/#asterisk ming_zym (n=ming_zym@124.254.54.39) |
| 14:28.20 | blitzrage | JoelSolanki: I see it a bunch -- you can pretty much just ignore it |
| 14:28.34 | blitzrage | depends on the device and the SIP msg exchange |
| 14:28.40 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 14:29.04 | JoelSolanki | ok. i trying to debug it. coz one of my client is getting disconnected after every 650 seconds |
| 14:29.05 | JoelSolanki | :) |
| 14:29.12 | JoelSolanki | so trying to find what is happening |
| 14:29.22 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:32.49 | *** join/#asterisk saftsack (n=oliver@p54A7F4A6.dip.t-dialin.net) |
| 14:33.51 | b11d | disconnected, as in how? |
| 14:35.04 | JoelSolanki | b11d: i will be back after 2 hrs. |
| 14:35.14 | JoelSolanki | i have to solve this issue. talk to u after 2 hrs |
| 14:35.16 | JoelSolanki | :) |
| 14:35.35 | b11d | yeah ok :) |
| 14:35.39 | b11d | cause i'll totally be here. |
| 14:39.18 | *** join/#asterisk slayer192 (n=slayer19@pirus.securax.be) |
| 14:41.02 | nettie | Hi guys, anyone brtistuff zaphfc here please? |
| 14:42.41 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:44.16 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:44.35 | *** part/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 14:59.50 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:00.10 | *** join/#asterisk friedrich| (n=friedric@e177246155.adsl.alicedsl.de) |
| 15:02.20 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 15:03.08 | [[blah]asfd | I am wondering if I may have something misconfigured. I periodically get this error: http://pastebin.ca/425502 |
| 15:06.59 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:09.27 | *** join/#asterisk Uatec (i=er00@bob.poddle.net) |
| 15:09.50 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 15:10.04 | hohum | is there a way I can get asterisk to reinvite a session every so often just to check to see if its alive? |
| 15:10.42 | hohum | I'm having a problem with a softphone that chronically crashes, and when that happens my asterisk box gets stuck in a loop because the channel will never hang up |
| 15:10.52 | [[blah]asfd | you must be using xlite |
| 15:10.57 | hohum | haha |
| 15:11.03 | hohum | yeah I am :P |
| 15:11.07 | hohum | good call |
| 15:11.12 | [[blah]asfd | try sjphone |
| 15:11.19 | [[blah]asfd | your problem will go away |
| 15:11.19 | hohum | that's not the solution, sorry |
| 15:11.35 | hohum | the solution is for Asterisk to periodically send re-INVITEs to check tosee if a session is still alive |
| 15:12.16 | [[blah]asfd | true... but since leaving xlite, has not been a concern |
| 15:12.21 | ManxPower | no, the solution is to get a stable phone |
| 15:12.38 | hohum | no, that's not the solutiion |
| 15:12.41 | ManxPower | but there are several rtp options in sip.conf and/or rtp.conf that might help |
| 15:12.45 | hohum | because people turn their ATAs off |
| 15:12.49 | hohum | and run shitty soft phones |
| 15:13.05 | hohum | and do all types of shit that could ptentially cause me to never see a bye |
| 15:13.15 | ManxPower | hohum: And when they turn them off asterisk will eventually give up |
| 15:13.29 | ManxPower | usually after x retries. |
| 15:13.39 | hohum | ManxPower: no, it really doesn't.. it'll sit there and play prompts over and over again forever |
| 15:13.42 | ManxPower | of course, if the devices are reinviting asterisk won't even be sending audio |
| 15:13.52 | ManxPower | hohum: then you have some OTHER issue.. |
| 15:13.54 | hohum | ManxPower: this is a call that's already been 200 OKed by asterisk |
| 15:14.04 | hohum | no I don't I know exactly what my issue is |
| 15:14.21 | hohum | there's got to be a way for Asterisk to do re-invites |
| 15:14.35 | [[blah]asfd | I am running 150 phones for 18 hours per day. over 800,000 minutes of long distance, and the only time i have had the trouble you are experiencing is when either using xlite or when the power goes out. then maybe one or two phones have to be soft hung up from the console. |
| 15:15.05 | ManxPower | hohum: so the rtptimeout option did not help? |
| 15:15.29 | hohum | asfd: you're running an office PBX it sounds like |
| 15:15.32 | ManxPower | I've never had to softhangup a SIP phone from the console |
| 15:15.40 | hohum | I'm doing voice mail hosting for 200,000 residential customers |
| 15:15.52 | hohum | and I run billions of minutes a month |
| 15:15.59 | hohum | so having to do a soft hang up is not an option |
| 15:16.29 | hohum | the correct answer to my question is "hohum, asterisk doesn't support re-invites so find another solution" or "hohum, this is how yo do what you want it to do" |
| 15:16.29 | ManxPower | oddly we still don't know the result of rtptimeout |
| 15:16.56 | ManxPower | hohum: asterisk supports reinvites |
| 15:17.07 | ManxPower | assuming it does not have to transcode or monitor audio |
| 15:17.09 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 15:17.10 | hohum | ManxPower: okay, so how do I turn them on? |
| 15:17.21 | ManxPower | and, of course, assuming you are not doing NAT. |
| 15:17.28 | hohum | I'm not doing nat |
| 15:17.36 | hohum | the asterisk box sits behind an SBC |
| 15:18.09 | ManxPower | hohum: Asterisk will reinvite by default unless you turn it off. If it is not doing reinvites (you can't really know without something like tcopdump, as SIGNALING stays with Asterisk) then you need to figure out why it cannot do reinvites |
| 15:18.12 | hohum | the SBC does the NAT traversal |
| 15:18.30 | ManxPower | hohum: perhaps it is your SBC that is causing a problem? |
| 15:18.35 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 15:18.37 | hohum | maybe |
| 15:19.00 | hohum | I can probably force the SBC to do the re-invites for me but I didn't want to do that, IMHO its cleaner from the far-end (asterisk) |
| 15:19.04 | ManxPower | hohum: so a "sip show channels" and make sure there is no codec translation being done. Make sure you don't have any options on the dial line as well. |
| 15:19.25 | ManxPower | hohum: none of your SIP clients are behind NAT? |
| 15:19.36 | hohum | ManxPower: no codec translation, but like I said the only calls that hit asterisk are voicemail calls, so it is doing media... |
| 15:19.51 | ManxPower | hohum: what codec? |
| 15:19.55 | hohum | ManxPower: they're all nated, but like I said there's an SBC in the middle |
| 15:20.06 | hohum | a mix of G.729 and G.711 |
| 15:20.19 | ManxPower | Dude, You can't reinvite a call off of asterisk for voicemail |
| 15:20.22 | *** join/#asterisk tigmmi (n=admin@adsl-245-184-192-81.adsl2.iam.net.ma) |
| 15:20.25 | tigmmi | hello |
| 15:20.41 | hohum | Manx: okay, then I'll find another solution |
| 15:20.46 | hohum | thanks |
| 15:20.46 | ManxPower | hohum: I really can't help you further as you have not tried rtptimeout |
| 15:20.57 | hohum | much appriciated |
| 15:21.11 | ManxPower | if you have audio being written to disk then the audio has to go to that host. |
| 15:21.32 | *** join/#asterisk sharp (i=sharp@gateway/tor/x-3b251897aa43b75f) |
| 15:21.37 | hohum | I'll do the reinvite from the SBC, no problem |
| 15:21.51 | tigmmi | Is there a way to fix the zaptel x100 card id on reboot, using udev for example. No valuable info found using udevinfo to destinguish between the two cards! |
| 15:21.59 | ManxPower | hohum: that won't work as the reinvite will still have to send the audio to Asterisk |
| 15:22.07 | hohum | if I get a 481 back from the far end or if the T2 timer expires, the SBC can BYE asterisk |
| 15:22.16 | hohum | ManxPower: It will |
| 15:22.32 | hohum | dude, you can send a re-invite with the same audio paramaters and keep the RTP stream flowing |
| 15:22.42 | hohum | the same SDP |
| 15:22.47 | ManxPower | how can you reinvite (send audio between the two endpoints) if you only have 1 endpoint. |
| 15:23.17 | [[blah]asfd | so if i add rtptimeout=60 to my sip.conf entries and the phone for whatever reason locks up or disconnects from the pbx, but the channel stays active, after 60 seconds it will kill the connection? |
| 15:23.21 | [[blah]asfd | is that correct? |
| 15:23.33 | hohum | ManxPower: the sdp negotiation stays the same through the calls |
| 15:23.52 | ManxPower | [[blah]asfd: if the audio is going thru asterisk yes |
| 15:23.52 | hohum | the SBC is an rtp ptoxy |
| 15:23.57 | hohum | it keeps the stream alive, to the asterisk box |
| 15:24.00 | [[blah]asfd | cool |
| 15:24.00 | *** join/#asterisk sharp (i=sharp@gateway/tor/x-061445281edf71c5) |
| 15:24.04 | [[blah]asfd | thats helpful |
| 15:25.14 | ManxPower | call: SIP Customer A <-> Asterisk <-> SIP Customer B. If a REINVITE happens then the signalling will be the same, but audio will be SIP Customer A <-> SIP Customer B. |
| 15:25.42 | hohum | ManxPower: that's not what my network looks like |
| 15:25.54 | hohum | trust me, it will work just fine the way I want to do it |
| 15:26.03 | ManxPower | call: SIP Customer A <-> Asterisk <-> SIP Customer B. The callee does not answer so the call goes to Asterisk's voicemail and would be SIP Customer A <-> Asterisk Voicemail. |
| 15:26.28 | ManxPower | as you can see the audio must go to asterisk be recorded in voicemail |
| 15:26.45 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 15:26.56 | ManxPower | therefore the audio must go to asterisk, threrefore a reinvite did not happen, therefore rtptimeout should kill the call. |
| 15:28.28 | tigmmi | I do have two x100p cards and I want to distinguish between them at reboot! is ther a way to do that. Note: udevinfo have no valuable info to do so |
| 15:28.57 | tigmmi | can anybody help? |
| 15:29.26 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 15:29.27 | Mahmoud | hello |
| 15:29.47 | Mahmoud | any way to increase the length of the dial tone in Sipura 3102? |
| 15:30.11 | Mahmoud | when i heard the dial tone "drrrrrr", it sends me the busy signal very soon, i want to increase it |
| 15:30.33 | ManxPower | Mahmoud: the docs for that device should be helpful |
| 15:30.43 | Mahmoud | it wasn't helpful |
| 15:31.18 | Mahmoud | any one here using Sipura? |
| 15:31.37 | ManxPower | Mahmoud: the FIRST hit in my google search should be the answer to your question. |
| 15:31.47 | Mahmoud | heh |
| 15:32.16 | ManxPower | and I basically copied and pasted your question into google |
| 15:32.17 | Mahmoud | ManxPower, you are right |
| 15:34.15 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:36.04 | *** join/#asterisk Falle (n=falle@87.ftth2.cust.cen2.sksk.se.borderlight.net) |
| 15:37.04 | Vec | How does asterisk handle multiple simultanious registrations ? |
| 15:37.47 | b11d | last to register gets it |
| 15:39.24 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:40.27 | Vec | b11d : that sounds tricky, is it possible for the phone that registered first to detect that it is no longer registered and then automatically re-register ? i.e. is the last to register rule reliable ? |
| 15:40.40 | b11d | if it re-registers, then it'll knock off the one one.. |
| 15:40.41 | ManxPower | Vec: no. |
| 15:40.56 | ManxPower | You should never have more than one device register as the same ID to Asterisk |
| 15:41.06 | b11d | yeah, with the same SIP account, that is. |
| 15:41.08 | ManxPower | I solve that problem pretty simply. |
| 15:41.21 | b11d | same.. |
| 15:41.27 | ManxPower | use the MAC of the device as the ID |
| 15:41.36 | b11d | "extension_phone1" and "extension_phone2" sip accounts, one for each.. both get the same number. |
| 15:41.39 | Vec | ManxPower : how do u solve it, if u want someone to have the same "number" on more then 1 location ? |
| 15:42.07 | ManxPower | Vec: DEVICES ARE NOT EXTENSIONS. The sooner you realize that the better your Asterisk Life will be |
| 15:42.42 | b11d | yeah its tricky when first moving from the old style terminology to this new fangled stuff |
| 15:42.47 | ManxPower | exten => 888,1,Dial(SIP/satan_home&SIP/satan_work) |
| 15:42.54 | Vec | oh yeh, so I just to a DIAL(SIP/acc1&SIP/acc2). |
| 15:42.57 | Vec | yeh thanks |
| 15:43.00 | b11d | two accounts, two phones, same exten. |
| 15:43.02 | Vec | u type faster |
| 15:43.33 | Vec | that also would work fine with DIAL(SIP/home&IAX/work) I assume |
| 15:43.52 | *** join/#asterisk saftsack (n=oliver@p54A7CB3B.dip.t-dialin.net) |
| 15:44.08 | b11d | sure |
| 15:44.13 | b11d | as long as those are valid |
| 15:44.25 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 15:45.40 | ManxPower | Our Dial lines are more along the lines of: exten => 3400,1,Dial(SIP/0004f201862c-b&SIP/0004f201fb0e-b&SIP/0004f21282a1-c&SIP/0004f204b4a7-b,30) |
| 15:46.15 | ManxPower | that dials the 2nd line appearance (the -b) and the 3rd line appearance (-c) on each of those 4 devices |
| 15:46.16 | b11d | whatever works |
| 15:46.46 | ManxPower | bbiaw |
| 15:47.26 | *** join/#asterisk Marty-OTT (n=marty@host-209-50-87-3.dyn.295.ca) |
| 15:47.40 | b11d | whats up Marty |
| 15:47.43 | b11d | how's ottawa these days |
| 15:47.51 | Marty-OTT | Hey ... not bad... EXCELLENT memory |
| 15:47.56 | Marty-OTT | or it is the OTT |
| 15:48.05 | b11d | OTT means ottawa? |
| 15:48.11 | Marty-OTT | yeah... |
| 15:48.13 | b11d | nice :) |
| 15:48.16 | b11d | well that'll remind me now.. |
| 15:48.22 | Marty-OTT | may yow would be better |
| 15:48.34 | Marty-OTT | YOW - itnl airport |
| 15:48.38 | b11d | so whats new.. get everything done? |
| 15:48.46 | b11d | yeah.. I remember Thunder Bay was YQT. |
| 15:49.17 | Marty-OTT | Well, I was on a contract for 3 months for a government client and am just getting back into it. My voip calls are fine.. my incoming calls from the outside are fine... but my outgoing calls.. not so fine. |
| 15:49.47 | b11d | whats the issue |
| 15:50.01 | Marty-OTT | It may simply be a rtfm issue.. but I'm trying to tell Asterisk... for all destination phone numbers you don't know about.. send the calls to this IP address (another voip server) |
| 15:50.24 | b11d | whats so hard aboot that? |
| 15:50.24 | Marty-OTT | exten => s,1,Dial(SIP/OUTGOING) |
| 15:50.34 | Marty-OTT | Well, I call out, I see no logs at all |
| 15:50.43 | b11d | nothing at all? |
| 15:51.18 | Marty-OTT | OUTGOING Is..... wait... |
| 15:51.34 | Marty-OTT | ah shit... probably case sensitive... in my sip.conf ..it's [outgoing] |
| 16:11.19 | [[blah]asfd | quick question regarding sip peers and users. |
| 16:11.41 | [[blah]asfd | if i am setting up a "gateway" pbx to connect to say 4 other pbxs so that I only had one pbx with t1s in it... |
| 16:11.59 | [[blah]asfd | I would do a sip peer on the gateway for each pbx connecting to it... |
| 16:12.12 | [[blah]asfd | but just one user for each of the other servers to connect to, is that correct? |
| 16:16.18 | *** join/#asterisk shodan (n=shodan@ip084.99-113-216.pppoe4.joliette.intermonde.net) |
| 16:17.24 | Marty-OTT | b11d you there? - that wasn't my problem |
| 16:24.10 | *** join/#asterisk infernix (n=nix@unaffiliated/infernix) |
| 16:24.19 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 16:24.22 | diclophis-work | hello all |
| 16:24.26 | diclophis-work | can asterisk play aiff files? |
| 16:24.38 | diclophis-work | also, whats the latest and greatest for TTS? |
| 16:25.12 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 16:27.31 | b11d | you guys know anything about using pin codes to allow Long Distance dialing within Asterisk? |
| 16:28.26 | diclophis-work | b11d: like a calling card system? |
| 16:29.07 | b11d | kind of.. i just have a phone where both a professor and students have access to it, and the professor needs LD access, but doesnt want students to be able to dial LD from his phone. |
| 16:29.22 | b11d | so.. I was thinking about some kind of pin.. |
| 16:30.46 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-137.usadatanet.com) |
| 16:31.22 | *** join/#asterisk mrpanik (n=loveu@88.210.113.231.rev.optimus.pt) |
| 16:32.00 | *** join/#asterisk zimdog (n=zimdog@63-227-112-12.hlrn.qwest.net) |
| 16:32.49 | mrpanik | hi |
| 16:32.49 | Marty-OTT | hmm... neat. |
| 16:33.31 | mrpanik | any1 for an channel congestion problem ? |
| 16:33.38 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-fbc4bc42456df1e5) |
| 16:33.49 | Marty-OTT | Man, how do I tell Asterisk... for ANY calls not in extensions.conf.. send to the outside line...???? |
| 16:33.55 | diclophis-work | b11d: look at the "Disa" application |
| 16:34.02 | b11d | I will, Thanks. |
| 16:34.57 | tigmmi | did any of the linmodem support zaptel? static struct pci_device_id wcfxo_pci_tbl[] can include the id of the linmodem card? |
| 16:35.32 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:39.29 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 16:42.07 | *** part/#asterisk ctaloi (n=ctaloi@nat-66-218-1-137.usadatanet.com) |
| 16:42.33 | *** join/#asterisk vtatian (n=vtatian@ns1.druzhba.lviv.ua) |
| 16:42.44 | vtatian | hello |
| 16:42.49 | b11d | hey hey |
| 16:42.50 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 16:42.55 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-137.usadatanet.com) |
| 16:43.33 | mrpanik | what can cause channel congestion cause 34, from asterisk to pbx |
| 16:43.35 | mrpanik | ? |
| 16:43.53 | mrpanik | irq or conf |
| 16:44.01 | vtatian | how register asterisk BE |
| 16:45.12 | b11d | mrpanik.. if all your circuits are in use, you get cause 34. |
| 16:45.14 | b11d | thats that. |
| 16:45.14 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:45.57 | b11d | oh no way, you live in Vansterdam? |
| 16:46.02 | b11d | im coming over |
| 16:46.55 | mrpanik | b11d: tnx, any other possible explanation? like cables or something? |
| 16:47.14 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 16:47.14 | b11d | not that i've ever seen.. you have a PRI I take it? |
| 16:47.20 | Marty-OTT | exten => s,1,Dial(SIP/OUTGOING) |
| 16:47.43 | mrpanik | im trying to configure asterisk transparent |
| 16:48.03 | b11d | looks fine Marty-OTT, as long as OUTGOING is a registred SIP peer. |
| 16:48.04 | Marty-OTT | See that statement above? It's in my [incoming] context whichshould say for any calls coming in for no sepecific destination, dial SIP/OUTOGING but it's not working... |
| 16:48.09 | mrpanik | from pstn all goes ok |
| 16:48.35 | Marty-OTT | b11d... hmm my SIP peer told me that I didn't have to specify a username and password to register - that I was permitted by IP. |
| 16:49.17 | thekidrio | ahh just finished some serious testing, testing sure does make me hungry :) |
| 16:49.39 | mrpanik | testing makes me mad :D |
| 16:49.41 | mrpanik | lol |
| 16:49.49 | thekidrio | hehe |
| 16:50.28 | mrpanik | b11d: i have a sangoma without hw echo cancelation, can this be a problem? |
| 16:50.38 | Marty-OTT | I thikn I know my problem... |
| 16:50.47 | Marty-OTT | OUTGOING (Unspecified) 5060 Unmonitored |
| 16:50.50 | thekidrio | b11d has spent too much time testing as well |
| 16:51.00 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-97-234.ph.ph.cox.net) |
| 16:51.05 | b11d | fuck, i tested ALL night last night |
| 16:51.14 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:51.19 | b11d | then I twisted up this 6-paper 'test' that knocked me out |
| 16:51.21 | b11d | :) |
| 16:51.37 | b11d | mrpanik.. no.. echo has nothing to do with congestion. |
| 16:51.43 | b11d | what does "zap show channels" report? |
| 16:51.50 | b11d | and have you enabled PRI debugging ? |
| 16:51.50 | mrpanik | all ok |
| 16:51.57 | b11d | well whats "all ok" mean? |
| 16:52.01 | mrpanik | but my span 2 is down |
| 16:52.24 | Marty-OTT | How do you hardcode an IP under a SIP context? I thought it was defaultip = .. |
| 16:52.25 | mrpanik | all alarms off |
| 16:52.47 | mrpanik | i have a lot on line violations :/ |
| 16:52.50 | mrpanik | of* |
| 16:52.57 | mrpanik | to the pbx |
| 16:53.07 | vtatian | what tell zttool |
| 16:53.08 | b11d | yeah thats strange.. |
| 16:53.11 | b11d | host = IP ADDRESS |
| 16:53.18 | b11d | thats it |
| 16:53.19 | Marty-OTT | crap... |
| 16:53.30 | Marty-OTT | thx.. something as silly as that was kicking my assl.. |
| 16:53.34 | b11d | :) |
| 16:53.39 | b11d | well maybe.. try it and see first :) |
| 16:54.09 | mrpanik | zttool says nothing that helps me , loop test, says nothing, no irq errors.. |
| 16:54.33 | *** join/#asterisk shinux__ (n=shinux@196.220.27.151) |
| 16:54.41 | b11d | yeah i dunno what you're trying to do there mrpanik.. |
| 16:54.51 | b11d | with asterisk being transparent and you running some other pbx in there.. |
| 16:55.05 | Marty-OTT | Well, it was one problem anways - I'm still not seeing any logs |
| 16:55.08 | *** join/#asterisk Rhab (n=Rhab@p3ee0fb57.dip.t-dialin.net) |
| 16:55.27 | Marty-OTT | Maybe I have to register? OUTGOING 208.65.240.133 5060 Unmonitored |
| 16:55.42 | Marty-OTT | But my sip peer told me I was allowed by ip |
| 16:55.44 | Rhab | HI |
| 16:55.47 | mrpanik | im going to give if another go, thanx anyway :) |
| 16:56.21 | vtatian | what linux you use |
| 16:56.29 | Rhab | whats the prolem when i recieve faxes via asterisk->iaxmodem->hylafax and the tif-file is broken?? |
| 16:56.41 | mrpanik | im am using debian variant |
| 16:56.52 | Rhab | problem with misdn? |
| 16:57.02 | mrpanik | now with smp because of irq sharing.. |
| 16:57.18 | vtatian | i have problem with te110p it work at slackware |
| 16:57.52 | vtatian | zttool is ok but pri show span is down |
| 16:58.11 | mrpanik | my zttool is ok too |
| 16:58.40 | Rhab | Anyone familiar with fax recieving via mISDN? |
| 16:58.52 | vtatian | you may see debug at pbx |
| 16:59.10 | mrpanik | but i use sangoma utils, they give lots of info |
| 16:59.12 | vtatian | layer 1 is active ? |
| 17:00.37 | mrpanik | how can u see that? |
| 17:00.37 | vtatian | debug at pbx |
| 17:00.37 | mrpanik | the pbx is a siemens card |
| 17:00.37 | vtatian | yes |
| 17:00.48 | vtatian | i use hicom pbx |
| 17:00.57 | mrpanik | yup, its hicom |
| 17:01.39 | Rhab | Someone knows an irc channel for hylafax? |
| 17:02.03 | mrpanik | Rhab: sorry, i dont |
| 17:02.24 | Rhab | sone1 uses it? |
| 17:02.55 | mrpanik | with asterisk no.. |
| 17:03.22 | Rhab | directly? |
| 17:03.26 | mrpanik | yes |
| 17:03.47 | Rhab | u ever got a fax and the tif file was broken? |
| 17:04.07 | mrpanik | i use pdf |
| 17:04.15 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:04.15 | mrpanik | but check your version |
| 17:04.18 | Rhab | that didnt work too |
| 17:04.42 | Rhab | is the fax first generated in ps? |
| 17:04.43 | mrpanik | the suport in latest hylafax is ok |
| 17:04.53 | Rhab | and then transformed? |
| 17:05.04 | mrpanik | hylafax does tif ps and txt |
| 17:05.19 | mrpanik | directly |
| 17:05.25 | Rhab | hm k |
| 17:05.46 | Rhab | where can i configure the paper size? |
| 17:05.58 | mrpanik | i never needed |
| 17:05.59 | _VoicemeUpDotCom | in the script tiff2ps |
| 17:06.15 | Rhab | i did it in the setup already but now it says american letter but i did A4 |
| 17:06.18 | _VoicemeUpDotCom | when you call that it takes tiff and makes it a post script with the arguents you passed.. |
| 17:06.25 | _VoicemeUpDotCom | like.. -w 11 -h 8 |
| 17:06.26 | _VoicemeUpDotCom | etc |
| 17:06.32 | _VoicemeUpDotCom | check the arguments im not sure |
| 17:06.43 | Rhab | hm |
| 17:07.02 | _VoicemeUpDotCom | tiff2ps -2eaz $FAXFILE | ps2pdf - | |
| 17:07.18 | _VoicemeUpDotCom | <PROTECTED> |
| 17:07.18 | _VoicemeUpDotCom | <PROTECTED> |
| 17:07.41 | vtatian | who use asterisk bussiness editions |
| 17:07.49 | _VoicemeUpDotCom | actually , if i dont call it it works fine.. adapt it self however if i do specify then it messes thing up |
| 17:07.56 | Rhab | well in recvq are only .tif files |
| 17:08.01 | _VoicemeUpDotCom | also check the -l for left margin |
| 17:08.10 | Rhab | wait |
| 17:08.20 | mrpanik | Rhab: 0 size? |
| 17:08.32 | Rhab | what 0 size? |
| 17:08.39 | mrpanik | empty |
| 17:09.10 | _VoicemeUpDotCom | also check that asterisk can ready or the process can read it.. unless you run everything as cough cough root ? |
| 17:09.11 | Rhab | I guess the problem is not the sice since didnt try to print it, but tried to open the file on a windows pc and it didnt work |
| 17:09.20 | _VoicemeUpDotCom | ah |
| 17:09.30 | _VoicemeUpDotCom | try another app.. what application you using ? |
| 17:09.42 | Rhab | for displaying? |
| 17:09.43 | _VoicemeUpDotCom | i had problems once with nero image viewer taking over |
| 17:09.57 | _VoicemeUpDotCom | yeah drop it in photocop , mspaint etc try difierent apps |
| 17:10.17 | Rhab | i use the standart windows xp picture viewer |
| 17:10.22 | Rhab | dont know the exact name |
| 17:10.24 | _VoicemeUpDotCom | if on unix.. cat file.tiff | head -30 |strings |
| 17:10.28 | aydiosmio | welcome to the jungle baby |
| 17:10.31 | _VoicemeUpDotCom | check ig the 4-5 digit code is tiff |
| 17:10.36 | b11d | strings rocks so much |
| 17:10.49 | Rhab | wait |
| 17:10.49 | _VoicemeUpDotCom | cat file.tiff | head -1 |strings actually |
| 17:10.53 | Rhab | ok |
| 17:12.05 | *** join/#asterisk sharp (i=sharp@gateway/tor/x-17d0e9516bda6465) |
| 17:12.15 | Rhab | 1 moment |
| 17:12.24 | vtatian | may change abcd bites at cas signaling at tormenta e1 cards |
| 17:12.47 | Rhab | just "cat file.tiff | head -1 |string" ? |
| 17:12.51 | vtatian | need change idle codes |
| 17:13.35 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 17:14.26 | Rhab | _VoicemeUpDotCom thats the output: Dj8)Qf" |
| 17:14.27 | Rhab | a8L7 |
| 17:14.53 | *** join/#asterisk ys1797 (n=yuri@80.70.236.69) |
| 17:15.12 | Rhab | Dj8)Qf" |
| 17:15.12 | Rhab | a8L7 |
| 17:15.22 | _VoicemeUpDotCom | hold |
| 17:15.24 | _VoicemeUpDotCom | finding a tif |
| 17:17.39 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 17:17.57 | fetcher | can a Polycom IP phone be set to call-forward only one of its line appearances? |
| 17:18.34 | fetcher | perhaps through some setting in sip.cfg? The normal user interface doesn't seem to allow for this |
| 17:18.50 | *** join/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net) |
| 17:19.18 | tuan_modulis | has anyone tried http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage |
| 17:19.49 | tuan_modulis | wanted to know if it doesnt introduce new problems |
| 17:19.57 | _VoicemeUpDotCom | hmm no idea bro |
| 17:20.15 | Rhab | k |
| 17:20.16 | b11d | http://www.google.com/search?hl=en&q=asterisk+voicemail+odbc+problem |
| 17:20.21 | *** join/#asterisk shinux_ (n=shinux@196.220.29.153) |
| 17:20.25 | wunderkin | fetcher, yes you can manually... |
| 17:20.34 | wunderkin | i don't know about a static config |
| 17:22.47 | fetcher | wunderkin: ah, I see it in the user doc PDF file now. thanks |
| 17:22.50 | *** part/#asterisk Rhab (n=Rhab@p3ee0fb57.dip.t-dialin.net) |
| 17:23.52 | nettie | Hi guys, anyone brtistuff zaphfc here please? |
| 17:23.59 | nettie | sorry |
| 17:24.09 | nettie | anyone using bristuff / zaphfc here please :) |
| 17:28.04 | *** join/#asterisk hedge77 (n=ambray@209.42.192.214) |
| 17:28.14 | *** join/#asterisk sysreq (n=sysreq@193.203-ppp.3menatwork.com) |
| 17:32.07 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:32.16 | paolob | Hi guys! |
| 17:32.19 | b11d | HEY!!!!! |
| 17:32.33 | b11d | where is .do ? |
| 17:33.03 | paolob | I want to build an * server with a very small computer, possibly without HD. What hardware can I use? thank you |
| 17:33.10 | hedge77 | if i send a dial command to multiple peers (ex. Dial(SIP/101&SIP/102|20|t) does ${DIALSTATUS} only return for the first extension? |
| 17:33.16 | paolob | b11d, dominican republic |
| 17:33.30 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:33.37 | b11d | sweet.. whats it like to live there? |
| 17:33.49 | _VoicemeUpDotCom | dialstatus will send the first answered one |
| 17:33.55 | paolob | b11d, for me it's a missionary service |
| 17:34.00 | b11d | right on, thats cool. |
| 17:34.04 | _VoicemeUpDotCom | dominc repl.. i went 5 times.. lots of corruption after 90's |
| 17:34.11 | b11d | a good friend of mine does that over in Haiti every year |
| 17:34.21 | _VoicemeUpDotCom | last time customs airport guy stole my cell and trvls checks |
| 17:34.27 | paolob | _VoicemeUpDotCom, before 90's too |
| 17:34.28 | _VoicemeUpDotCom | which i had canceld upon retourn lol |
| 17:34.32 | b11d | lol |
| 17:35.00 | _VoicemeUpDotCom | yeah well .. i had long hair at the time it was funny saying .. MARIA.. when they tried to grab me.. i guess shes the chick there |
| 17:35.00 | _VoicemeUpDotCom | lol |
| 17:35.01 | b11d | what are "the people" saying about whats going on around the world down there? Whats the general opinion like? |
| 17:35.05 | _VoicemeUpDotCom | hair mafia |
| 17:35.21 | _VoicemeUpDotCom | was in cabarete and sosua |
| 17:35.23 | paolob | anyone can help me with the question about what "small" hardware I can install * on? |
| 17:35.30 | b11d | stay on topic paolob :) |
| 17:35.31 | _VoicemeUpDotCom | cabarete is nicer.. less punks.. and nice surfing |
| 17:35.32 | b11d | hahah |
| 17:35.44 | b11d | is that on the south coast? |
| 17:35.45 | _VoicemeUpDotCom | sosua is the bronx |
| 17:35.53 | paolob | b11d, "Topic for #asterisk is Asterisk" |
| 17:35.59 | b11d | yeah I know :) it was a joke. |
| 17:36.03 | paolob | sosua is in the north |
| 17:36.21 | _VoicemeUpDotCom | yeah and cabaeter nort more |
| 17:36.35 | _VoicemeUpDotCom | actualy got my roomate stabbed wen we where there |
| 17:36.35 | b11d | ah.. i figured the south would have better surfing conditions.. |
| 17:36.41 | b11d | sweet |
| 17:37.00 | _VoicemeUpDotCom | yeah promised never to go back. then i got ran by 15 ni.. with machetas.. in the streets |
| 17:37.13 | _VoicemeUpDotCom | the rentacops hotel use are in with the players and steal in rooms |
| 17:37.16 | _VoicemeUpDotCom | not a nice place to go |
| 17:37.37 | b11d | yeah it sounds like ti |
| 17:37.37 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 17:37.38 | b11d | it |
| 17:37.43 | b11d | mercestessssssssss |
| 17:37.54 | *** join/#asterisk c_rosales (i=c_rlitos@189.173.13.204) |
| 17:37.55 | Mercestes | b11d!!!!!!!! |
| 17:38.00 | b11d | how art thou? |
| 17:38.33 | Mercestes | craptasticness. |
| 17:38.39 | Mercestes | how are you? |
| 17:38.49 | b11d | just fine, thanks. Just got over a wicked ass cold.. dropped me for a week! |
| 17:39.16 | c_rosales | hi, everyone |
| 17:39.24 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 17:39.29 | *** join/#asterisk zefrench (n=zefrench@192.219.29.50) |
| 17:40.39 | c_rosales | any1 speaks spanish ? |
| 17:40.49 | b11d | i dont speak spanish |
| 17:40.49 | Mercestes | lol |
| 17:40.56 | Mercestes | I do but....I'm evil |
| 17:41.20 | c_rosales | well, don't be today |
| 17:41.34 | zefrench | How to pass CALLERID(dnid) via IAX trunk? I am trunking calls between two asterisks boxes, I can verify via NoOp that the PSTN connected Asterisk that the dnid is received, but when I look at the remote asterisk server (again via NoOp,) that variable is now blank |
| 17:41.54 | coppice | what if they can speak spanish, but can't type it? :-\ |
| 17:42.07 | c_rosales | i'll try to ask in english |
| 17:42.12 | c_rosales | so please be patient |
| 17:42.24 | b11d | your english seems just fine |
| 17:42.35 | c_rosales | I am new to asterisk and just finished installing my first box |
| 17:42.50 | c_rosales | i setup a tdm2400 with 4fxs and 4fxo |
| 17:43.02 | c_rosales | i can make calls and receive calls |
| 17:43.28 | c_rosales | but on outside calls i have this hissing noise |
| 17:43.50 | Mercestes | did you do a fxotune? |
| 17:43.54 | c_rosales | it gets worse when there is a silence on the conversation |
| 17:44.15 | c_rosales | no, how to do it ? |
| 17:45.36 | Mercestes | google fxotune |
| 17:45.40 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 17:45.49 | c_rosales | could you please help me pointing me in the right direction ??? |
| 17:45.50 | Mercestes | and...when you plug a phone into FXS do you hear static there without maknig a call? |
| 17:45.53 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 17:46.00 | Mercestes | Sure! |
| 17:46.05 | _VoicemeUpDotCom | waht i hear is a big clunk on asnwer |
| 17:46.10 | hrmphh | anyone know if you can export the current config of a sipura/linksys 941 phone as their xml format? want to set up on phone, then save+modify on the tftp server |
| 17:46.13 | Mercestes | you open up Internet Explorer or firefox or links http://www.google.com |
| 17:46.15 | _VoicemeUpDotCom | but these boxes have been up for ages |
| 17:46.20 | Mercestes | and then you type in "fxotune" and hit enter. |
| 17:46.48 | c_rosales | actually this noise only appears when using pstn lines |
| 17:46.50 | *** join/#asterisk jarg (n=jarg@189.157.103.143) |
| 17:46.59 | c_rosales | on internal calls quality is perfect |
| 17:47.23 | hrmphh | how does a headset sound |
| 17:47.23 | hrmphh | err |
| 17:47.24 | hrmphh | handset |
| 17:47.26 | Mercestes | What happens when you run a phone straight to yoru PSTN and make a call?? |
| 17:47.30 | hrmphh | connected to the lines |
| 17:47.38 | hrmphh | yeh what mercestes said ;) |
| 17:47.44 | Mercestes | :) |
| 17:47.46 | *** part/#asterisk ming_zym (n=ming_zym@124.254.54.39) |
| 17:47.50 | Mercestes | hrmphh< it's the thought that counts. ;) |
| 17:47.54 | Marty-OTT | Is there a way to set registration to false in a sip config file when registering to another SIP server? My IP is permitted so they don't required my username of password... |
| 17:48.48 | _VoicemeUpDotCom | hmm.. you dont have to use register line |
| 17:48.48 | hedge77 | hrmphh: did you ever find out what ${DIALSTATUS} returned when dialing multiple extensions? |
| 17:48.51 | _VoicemeUpDotCom | the register => |
| 17:48.54 | _VoicemeUpDotCom | comment it out |
| 17:49.00 | _VoicemeUpDotCom | leave the rest but comment the secret |
| 17:49.16 | Marty-OTT | comment out the register? |
| 17:49.24 | hrmphh | hedge; yeah well if i ring say 5 extensions, so long as theyre not ALL busy, it wont return BUSY |
| 17:49.25 | Marty-OTT | ok.. that's what I had before... |
| 17:49.26 | Marty-OTT | hmmm... |
| 17:49.32 | _VoicemeUpDotCom | yeah in sip.cof you have register => blah:blue@host |
| 17:50.58 | _VoicemeUpDotCom | comment it out and the peer definition also comment the secret= |
| 17:50.58 | _VoicemeUpDotCom | sip reload and test |
| 17:50.58 | Marty-OTT | yes, that was commented out before but my calls still didn't go out.. i'll try again.. |
| 17:50.59 | _VoicemeUpDotCom | sip debug peer PEERNAMEHERE |
| 17:50.59 | _VoicemeUpDotCom | waht the answer .. 407 ? |
| 17:50.59 | Marty-OTT | I hvave this though: |
| 17:50.59 | Marty-OTT | [OUTGOING] |
| 17:50.59 | Marty-OTT | nat=no |
| 17:50.59 | Marty-OTT | type=peer |
| 17:50.59 | Marty-OTT | host=208.65.240.133 |
| 17:50.59 | Marty-OTT | I just get a fast busy and no log... it's odd |
| 17:50.59 | _VoicemeUpDotCom | that ok add a user .. |
| 17:51.02 | _VoicemeUpDotCom | as in user=theITSPuserassigned |
| 17:51.09 | _VoicemeUpDotCom | or fromuser or username whatever |
| 17:51.13 | Marty-OTT | oh, because in my sip, I say exten => s,1,Dial(SIP/OUTGOING) |
| 17:51.14 | hrmphh | hmm |
| 17:51.20 | hrmphh | anyone use sipura 941/942s? |
| 17:51.29 | _VoicemeUpDotCom | yeah you need an exten.. |
| 17:51.42 | Marty-OTT | So, I would think that if a call comes in... then it's sent to hose 208.65.240.133 |
| 17:51.44 | _VoicemeUpDotCom | how about SIP/OUTGOING/${EXTEN{ |
| 17:51.49 | _VoicemeUpDotCom | } |
| 17:52.17 | _VoicemeUpDotCom | i would seperate the context.. inbound aoutbound as peer/user for your thing |
| 17:52.21 | _VoicemeUpDotCom | else NEVER use S |
| 17:52.25 | Marty-OTT | I can try... but if I don't configure my peer, where do I put the IP of 208.65.240.133? |
| 17:52.29 | _VoicemeUpDotCom | you will loop back out to itsp. |
| 17:52.56 | _VoicemeUpDotCom | kep it as you have.. you need the 208 as host.. ..where did you see any of us saying to remove it ;) |
| 17:53.23 | _VoicemeUpDotCom | [outgoing] add context=outbound |
| 17:53.29 | Marty-OTT | ok... exten => s,1,Dial(SIP/OUTGOING/{$exten}) |
| 17:53.40 | _VoicemeUpDotCom | TRY UPPERCASE ${EXTEN} |
| 17:53.42 | _VoicemeUpDotCom | to be sure |
| 17:54.02 | Marty-OTT | right... ok.. and my $ sign.. sorry speeding to fast here.. |
| 17:54.08 | _VoicemeUpDotCom | make sure that [OPUTGOING]peer name doesnt use the same context as inbound would |
| 17:54.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:54.38 | Marty-OTT | in sip.conf you mean? |
| 17:54.46 | _VoicemeUpDotCom | hold im making you a pastebin |
| 17:54.53 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:54.53 | Marty-OTT | ok |
| 17:54.58 | flujan | hi guys... |
| 17:55.04 | Marty-OTT | thx btw... |
| 17:55.44 | flujan | I want to record my queues... |
| 17:55.44 | flujan | Where I can grab information about it? |
| 17:55.46 | *** join/#asterisk radikall (n=devnull@189.168.38.29) |
| 17:55.52 | Marty-OTT | yeah.. in my extensions.conf, my exten statement for outgoing calls is under [incoming] context |
| 17:57.18 | _VoicemeUpDotCom | http://pastebin.ca/425731 |
| 17:57.21 | Marty-OTT | thx |
| 17:57.21 | _VoicemeUpDotCom | try something like this |
| 17:57.50 | _VoicemeUpDotCom | no prob |
| 17:57.57 | _VoicemeUpDotCom | we offer support you know ;) |
| 17:58.06 | Marty-OTT | oh... actually, I paid someone before... |
| 17:58.34 | b11d | haha you soliticited sex from a prostitute |
| 17:58.36 | Marty-OTT | I paid to have the zaptel driver package updated |
| 17:58.39 | b11d | oh |
| 17:58.40 | b11d | :) |
| 17:58.44 | Marty-OTT | that too... |
| 17:59.02 | radikall | hi |
| 17:59.36 | radikall | somebody speak spanish ? |
| 17:59.36 | *** join/#asterisk vader-- (n=me@204.183.88.101) |
| 17:59.37 | Marty-OTT | I'm just not sure how when a call comes in ... how Asterisk differentiates from [outgoing] and [incoming |
| 17:59.37 | Marty-OTT | in extensions.conf |
| 17:59.37 | _VoicemeUpDotCom | type=peer |
| 17:59.37 | _VoicemeUpDotCom | type=user |
| 17:59.44 | b11d | you need to learn how "call flow" works within Asterisk then Marty-OTT |
| 17:59.52 | _VoicemeUpDotCom | asterisk think.. peer .. then its calls going TO.. user.. = calls coming FROM... |
| 18:00.27 | Mercestes | randikall c_rsoales does! |
| 18:00.30 | _VoicemeUpDotCom | so.. calls in matching ip 208.x.x.x. will match a user.. [INCOMING] and then will use context inbound |
| 18:00.43 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 18:00.53 | Marty-OTT | Ok, I reload my config files... going to try |
| 18:00.58 | radikall | c_rosales estas??? |
| 18:02.09 | flujan | guys, is there a way to record call from a call queue in asterisk? |
| 18:02.33 | aydiosmio | flujan: yes |
| 18:02.40 | _VoicemeUpDotCom | use.. hmm |
| 18:02.44 | _VoicemeUpDotCom | monitor |
| 18:02.45 | _VoicemeUpDotCom | ah wait |
| 18:02.46 | aydiosmio | vaya con dios.\ |
| 18:02.52 | aydiosmio | nah not even |
| 18:03.11 | Marty-OTT | gnna try again... |
| 18:03.19 | _VoicemeUpDotCom | monitor_format=wav |
| 18:03.22 | _VoicemeUpDotCom | if that there it will record |
| 18:03.23 | _VoicemeUpDotCom | next ! |
| 18:03.43 | flujan | aydiosmio, where can I acquire information of howto do it? |
| 18:03.51 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 18:03.53 | aydiosmio | internet |
| 18:04.44 | Marty-OTT | yeah... I don't see any logs.. |
| 18:05.21 | Marty-OTT | I didn't put the: user=USERFROMITSP |
| 18:06.03 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:06.09 | radikall | well, someone can help me???, i need to do, some like this: i have a db, with some data, a field "credit" with "credit" :P, well... all in my pbx is free, but if someone outs from specific channel, then, should check if have credit ( more than zero), then convert that credit on minutes $1.00 per minute for example, and then, limit the call to that time, then hangup, and update how mucho credit rest, (zero), but if the call ends before, only rest the cre |
| 18:06.51 | _VoicemeUpDotCom | lol |
| 18:06.55 | _VoicemeUpDotCom | agi |
| 18:07.05 | mvanbaak | radikall: tipical callingcard application |
| 18:07.09 | mvanbaak | right ? |
| 18:07.14 | Marty-OTT | VoicemeUpDotCom - would you mind checking out my pastebin? |
| 18:07.17 | aydiosmio | sounds like you need to hire soemone to write a callign card app |
| 18:07.18 | aydiosmio | lolz |
| 18:07.29 | mvanbaak | there are some |
| 18:07.37 | mvanbaak | check www.voip-info.org |
| 18:07.49 | _VoicemeUpDotCom | first run asterisk as non deamon and ful debug..with logger.conf havinf the full enabled |
| 18:07.58 | _VoicemeUpDotCom | then start asterisk -vgcf |
| 18:08.05 | _VoicemeUpDotCom | oups -vvvvvvvvvggggcdf |
| 18:08.56 | Marty-OTT | ok, I didn't do ggggcdf |
| 18:09.03 | Marty-OTT | I onluy did vvvvvvv |
| 18:09.04 | Marty-OTT | http://rafb.net/p/Z1m3hH20.html |
| 18:09.05 | flujan | aydiosmio, I tryed to get information about it and just find the automon command... |
| 18:09.19 | b11d | screw using multiple -v's and -d's just use "set verbose 100" and "set debug 100" when at the CLI. |
| 18:09.22 | b11d | way easier & cleaner |
| 18:09.37 | Marty-OTT | ok |
| 18:10.06 | b11d | imho :) |
| 18:10.22 | _VoicemeUpDotCom | ok...firs *98 move that to outbound context in extensions |
| 18:10.40 | aydiosmio | flujan: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
| 18:10.53 | _VoicemeUpDotCom | your 16xxx sip user should be friend or needs a user entry |
| 18:11.04 | aydiosmio | To enable monitoring, simply specify "monitor-format"; |
| 18:11.08 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:11.10 | aydiosmio | like _VoicemeUpDotCom said |
| 18:11.19 | _VoicemeUpDotCom | monitor_format=wav |
| 18:11.24 | _VoicemeUpDotCom | said that at 14:03 man lol |
| 18:11.25 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 18:11.29 | mvanbaak | monitor_format=wmv |
| 18:11.32 | mvanbaak | ;) |
| 18:11.59 | Marty-OTT | damn... I can't get my calls to go out... |
| 18:12.18 | Marty-OTT | no logs either... |
| 18:12.35 | _VoicemeUpDotCom | you have full enabled in logger.conf |
| 18:12.47 | _VoicemeUpDotCom | ls -l /var/log/asterisk |
| 18:12.51 | *** join/#asterisk notoriousrab1982 (n=notoriou@207.47.34.74.static.nextweb.net) |
| 18:13.17 | b11d | nothing from "sip debug peer X" |
| 18:13.18 | b11d | ? |
| 18:13.30 | Marty-OTT | drwxr-xr-x 2 root wheel 512 Jan 6 14:45 cdr-csv |
| 18:13.30 | Marty-OTT | drwxr-xr-x 2 root wheel 512 Jan 6 14:45 cdr-custom |
| 18:13.30 | Marty-OTT | -rw-r--r-- 1 root wheel 0 Jan 5 16:58 event_log |
| 18:13.30 | Marty-OTT | -rw-r--r-- 1 root wheel 179326230 Apr 5 14:27 messages |
| 18:13.30 | Marty-OTT | -rw-r--r-- 1 root wheel 608 Feb 13 17:26 queue_log |
| 18:13.44 | _VoicemeUpDotCom | no full in there |
| 18:13.51 | _VoicemeUpDotCom | so /etc/asterisk/logger.conf |
| 18:13.56 | _VoicemeUpDotCom | enable the line full = |
| 18:14.06 | Marty-OTT | 8197729232/8197729232 216.106.109.177 D N 5060 OK (107 ms) |
| 18:14.06 | Marty-OTT | 16477248834/16477248834 209.50.87.26 D 5060 Unmonitored |
| 18:14.06 | Marty-OTT | OUTGOING 208.65.240.133 5060 Unmonitored |
| 18:14.07 | _VoicemeUpDotCom | restart all .. pray |
| 18:14.16 | _VoicemeUpDotCom | smeels like a kcik |
| 18:14.16 | mvanbaak | enable it in /etc/asterisk/logger.conf and do a 'restart now' |
| 18:14.34 | Marty-OTT | ok.. see my 'OUTGOING' there... it doesn't say OK... |
| 18:14.42 | mvanbaak | please dont paste in channel |
| 18:14.46 | mvanbaak | use pastebin for that |
| 18:14.53 | mvanbaak | ~pb |
| 18:15.07 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:15.08 | _VoicemeUpDotCom | send us login /pass and 25$ ill fix |
| 18:15.19 | _VoicemeUpDotCom | make it 10$ |
| 18:15.21 | _VoicemeUpDotCom | ;) |
| 18:15.27 | b11d | can you even reach the host at 209.50.whatever ? |
| 18:15.39 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 18:15.54 | Marty-OTT | See my message VoicemeUpDotcom |
| 18:15.59 | Marty-OTT | b11d - yeah - that's no prob |
| 18:16.14 | b11d | do you have access to that host? |
| 18:16.34 | brodiem | in sip.conf when defining a peer, can host= be a block, i.e. host=123.123.123.0/24? I know I can use host=dynamic but I want to allow from only a specific block |
| 18:16.35 | Marty-OTT | Yeah, the host I don't have access to is 208.65 |
| 18:16.42 | b11d | DOH |
| 18:16.45 | b11d | i was misreading that |
| 18:16.46 | b11d | sorry |
| 18:16.49 | _VoicemeUpDotCom | no |
| 18:16.50 | _VoicemeUpDotCom | ohost=ip |
| 18:16.57 | hrmphh | does anyone have the full xml syntax of the sipura 941/942 phones?? |
| 18:16.57 | _VoicemeUpDotCom | but permit/deny can be masks |
| 18:17.01 | Marty-OTT | VoicemeUpDotCom - top of your IRC ... chat.. |
| 18:17.05 | b11d | use your firewall "_ |
| 18:17.22 | Marty-OTT | Chat with _VoicemeUpDotCom |
| 18:17.22 | Marty-OTT | Waiting for acknowledgement... |
| 18:17.33 | _VoicemeUpDotCom | ? i cant do that sorry |
| 18:17.35 | _VoicemeUpDotCom | pm me |
| 18:17.39 | Marty-OTT | pm? |
| 18:17.49 | hrmphh | he means /msg |
| 18:18.04 | brodiem | _VoicemeUpDotCom that's what i was looking for ty |
| 18:19.15 | tuan_modulis | Is there a standard way of customising the voicemail IVR (like to filter out selections) when you dial a mailbox? Or do I have to look into the code itself? |
| 18:19.35 | radikall | yes that |
| 18:19.38 | radikall | calling card |
| 18:19.41 | radikall | :P |
| 18:19.57 | *** join/#asterisk Fieldy (i=5G9YLw4c@gentoo/contributor/Fieldy) |
| 18:20.19 | radikall | but, the idea is a distribuited network |
| 18:20.31 | [[blah]asfd | I am having trouble understanding the concept and structure of sip peers and users. What I am trying to do is set up a server that will place calls to another server where a T1 is installed. Here is what I have: |
| 18:20.31 | [[blah]asfd | http://pastebin.ca/index.php |
| 18:20.31 | [[blah]asfd | can anyone help me to better understand this? |
| 18:20.50 | flujan | aydiosmio, so, I just need to set monitor-format = wav on the queues.conf and all calls on my queues will be recorded? |
| 18:20.59 | flujan | aydiosmio, only this configurations is needed? |
| 18:21.06 | radikall | with a root server, where people uses that cards, and that web server, update each other asterisk server with the credit |
| 18:21.25 | radikall | i don't known if i can do that with a2billing |
| 18:22.11 | radikall | aydiosmio tu nick suena a que hablas español, cierto? |
| 18:24.23 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 18:26.22 | hedge77 | is codefreeze around? |
| 18:26.35 | mvanbaak | try /whois |
| 18:26.52 | mvanbaak | 20:26 -!- idle : 0 days 15 hours 44 mins 11 secs [signon: Wed Apr 4 19:08:20 2007] |
| 18:27.13 | mvanbaak | I think that is a 'no, not around' |
| 18:27.16 | hedge77 | it didn't report idle time to me for some reason |
| 18:27.38 | mvanbaak | hhmm |
| 18:27.52 | mvanbaak | try: /whois codefreeze codefreeze |
| 18:28.06 | hedge77 | ah that did it |
| 18:28.27 | mvanbaak | irssi has this default alias /wii for that |
| 18:28.29 | mvanbaak | :) |
| 18:28.30 | b11d | neat.. didnt know about that :) |
| 18:28.49 | mvanbaak | <--- ircop |
| 18:29.07 | hedge77 | <--- ircnoob |
| 18:29.13 | codefreeze | hedge77: codefreeze present! |
| 18:29.20 | mvanbaak | wb codefreeze ! |
| 18:29.24 | b11d | g-line me please :) |
| 18:29.25 | hedge77 | awesome |
| 18:29.29 | b11d | ok.. dont !! |
| 18:29.31 | mvanbaak | lol b11d |
| 18:29.39 | b11d | hehe |
| 18:29.44 | mvanbaak | <PROTECTED> |
| 18:29.47 | b11d | lol |
| 18:30.12 | mvanbaak | not ircop on freenode though |
| 18:30.34 | hedge77 | codefreeze: is there a good way to emulate n+101 jumps in ael? |
| 18:30.51 | Qwell[] | hedge77: priority jumping is discouraged |
| 18:30.56 | mvanbaak | WHY ???? |
| 18:30.57 | mvanbaak | omg |
| 18:31.13 | hedge77 | right i know but I need similar functionality |
| 18:31.21 | mvanbaak | jump to a label |
| 18:31.32 | hedge77 | I want to do a ChanIsAvail() |
| 18:31.34 | codefreeze | hedge77: Well, AEL assumes you are using result codes... |
| 18:32.04 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 18:32.08 | codefreeze | hedge77: and therefore, you can do all the same things, easier/better. |
| 18:32.27 | mvanbaak | GotoIf() |
| 18:32.29 | hedge77 | codefreeze: you're saying switch on ${DIALSTATUS} then |
| 18:32.35 | mvanbaak | yeah |
| 18:32.38 | Qwell[] | hedge77: exactly |
| 18:32.50 | mvanbaak | ael is nice |
| 18:33.08 | olinux | anyone have a recommended telephone? |
| 18:33.20 | Qwell[] | ~phones |
| 18:33.27 | jbot | somebody said phones was http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 18:33.27 | hedge77 | does ChanIsAvail() return a status code I can switch on though? |
| 18:33.31 | hedge77 | ~gs |
| 18:33.33 | jbot | i guess gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 18:33.33 | Qwell[] | hedge77: show application chanisavail will tell you that |
| 18:33.47 | codefreeze | hedge77: A few days back, I wrote someone a letter about +101 jumping, and how it could be dangerous, using the term a bit loosely. |
| 18:34.10 | radikall | hum |
| 18:34.24 | b11d | that's jbot's best answer ever. |
| 18:34.44 | codefreeze | hedge77: Even in extensions.conf, +101 jumping can introduce problems you don't expect. Best to stay away and use result codes instead. |
| 18:34.45 | Mercestes | ~mercestes |
| 18:34.46 | jbot | methinks mercestes is almost a total nub |
| 18:34.49 | radikall | is possible to use one a2billing system for various asterisk servers ? |
| 18:35.03 | hedge77 | Qwell[]: and so it does: {AVAILSTATUS} |
| 18:35.11 | *** join/#asterisk s0lid (n=jlq@210.213.198.205) |
| 18:35.17 | Qwell[] | there you go then |
| 18:35.19 | b11d | ~b11d |
| 18:35.21 | jbot | b11d is a constant source of misinformation... |
| 18:35.23 | b11d | come on... coem on |
| 18:35.24 | b11d | NICE |
| 18:35.28 | b11d | thats totally accurate |
| 18:35.31 | mvanbaak | n+101 must be implemented while the developer was high on LSD or something |
| 18:35.34 | hedge77 | codefreeze: yeah i was trying to avoid it I just didn't realize there were return codes for that I could test on. |
| 18:35.47 | Qwell[] | mvanbaak: channel variables haven't always existed |
| 18:35.52 | hedge77 | n+101 has caused me all kinds of headaches converting this old dialplan |
| 18:35.54 | mvanbaak | I know :) |
| 18:36.01 | Qwell[] | mvanbaak: as disturbing as that sounds - that's probably the reason |
| 18:36.09 | mvanbaak | must be |
| 18:36.15 | b11d | dont knock LSD now.. |
| 18:36.23 | mvanbaak | lol b11d |
| 18:36.31 | b11d | :) |
| 18:36.50 | mvanbaak | I prefer the less heavy stuff |
| 18:36.59 | b11d | yeah, same here really.. |
| 18:37.04 | b11d | i'd do LSD one last time though.. |
| 18:37.33 | mvanbaak | it's great to live in .nl |
| 18:37.42 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:37.44 | mvanbaak | brb, food time |
| 18:37.44 | b11d | im coming over.. |
| 18:38.18 | MrWup | hey guys need a bit of noob-help |
| 18:38.19 | olinux | thanks i was looking at the linksys spa962 |
| 18:38.34 | Mercestes | MrWup: Insult and deride them until they leave you alone. |
| 18:38.37 | b11d | no, we dont need any newb help. Thanks |
| 18:38.37 | b11d | :) |
| 18:39.08 | MrWup | every time someone dials, i need asterisk to connect to an SQL database, and retreive a callerid string based on the extension number which is sent as the callerid |
| 18:39.24 | MrWup | basically this allows me to alter peoples callerids from the web as staff are hired or fired |
| 18:39.31 | MrWup | but how do i do it? |
| 18:39.33 | MrWup | call a macro each time? |
| 18:39.55 | MrWup | i know a macro can take args, but can it spit out a result (i.e. the callerid taken from mysql) ? |
| 18:40.47 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 18:40.52 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 18:41.17 | Mercestes | MrWup: look into RTA. |
| 18:41.21 | toerkeium | hello guys |
| 18:41.26 | Mercestes | and just cahnge the CallerIDname in mysql |
| 18:41.35 | MrWup | RTA? |
| 18:41.46 | MrWup | yeah i know mysql is all fine, i know how to do it |
| 18:41.49 | hedge77 | realtime asterisk i assume |
| 18:41.57 | MrWup | i got php scripts to put and pull stuff from a mysql database |
| 18:42.14 | toerkeium | is there any way to make *'s MOH get audio from a live internet radio from an url like: http://ip.add.re.ss:port or file.m3u ? |
| 18:42.15 | MrWup | but obviously, every time someone dials, its a bit stupid doing the connect to mysql, and fetch etc |
| 18:42.32 | tuan_modulis | MrWup: sure, call the macro |
| 18:42.53 | MrWup | far better to have like a function inside asterisk (or a macro or whatever its called) so that when someone dials an ext, it calls a macro, the macro connects to mysql, gets a callerid out of it, and then returns the callerid |
| 18:42.55 | hedge77 | so how often are people hired and fired that you couldn't just edit sip.conf and change the callerid manually? |
| 18:43.15 | flujan | guys, I am having problems making asterisk record calls on my queues |
| 18:43.30 | MrWup | tuan_modulis, but i know a macro can take args and look up form mysql, but how does a macro return the callerid to my current priority? |
| 18:43.43 | flujan | I set the variable MONITOR_FILENAME = "/home/flujan/monitor/teste.wav" before run the queue |
| 18:43.47 | flujan | queue(test) |
| 18:43.54 | hedge77 | you would use Set(CALLERID(name)=new name) |
| 18:43.54 | Mercestes | MrWup just phpagi it then. |
| 18:44.01 | toerkeium | :P |
| 18:44.04 | MrWup | like i could do exten => 1234,1,Macro(macro-getID,1234) |
| 18:44.04 | flujan | I set the variable monitor-format = wav on my queues.conf |
| 18:44.11 | MrWup | but the macro isnt returning anything |
| 18:44.22 | flujan | But it is not recording the call. |
| 18:44.25 | Mercestes | mrWup: use phpagi |
| 18:44.27 | tuan_modulis | MrWup: in macro, call agi perhaps... |
| 18:44.34 | hedge77 | yeah agi ftw |
| 18:45.23 | Mercestes | agi > macro |
| 18:45.34 | MrWup | php agi... |
| 18:45.37 | MrWup | not a clue |
| 18:45.39 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:45.44 | hedge77 | ~agi |
| 18:46.04 | jbot | well, agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 18:46.27 | MrWup | so i have to download a PHP AGI wrapper? |
| 18:47.14 | tuan_modulis | i suppose u can execute a dial using an AGI script OR using Dial, execute a macro that calls an AGI script |
| 18:48.02 | tuan_modulis | you dont need to download anything... except if u want php, install php |
| 18:48.51 | MrWup | it seems so crazy to me that asterisk doesnt just have some function like PHP(functionname,args):result |
| 18:49.02 | MrWup | or somethign to just get a result |
| 18:49.05 | Qwell[] | MrWup: eh? |
| 18:49.09 | MrWup | rather than only seeming to be able to tell php to do something |
| 18:49.12 | MrWup | without getting anything back |
| 18:49.14 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:49.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:51.11 | Mercestes | MrWup: That would be AGI() |
| 18:56.24 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 18:56.24 | MrWup | oh |
| 18:56.29 | MrWup | so u put PHP inside an AGI script? |
| 18:56.43 | tuan_modulis | php script = agi script in your case |
| 18:56.51 | tuan_modulis | it can be other languages as well |
| 18:56.56 | tuan_modulis | like perl |
| 18:57.13 | tuan_modulis | then u use the AGI application to call the php script |
| 18:57.19 | MrWup | so an AGI script is executed by the dialplan and can contain many types of languages? |
| 18:57.34 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 18:57.39 | irule | hi |
| 18:57.56 | b11d | no, you dont. |
| 18:57.56 | tuan_modulis | yea, it amounts to doing something like exten => s,n,AGI(asdasd.php|${arg1}|${arg2}) |
| 18:58.12 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 18:58.23 | tuan_modulis | errr, CAN use many types of languages |
| 18:59.18 | MrWup | i dont understand |
| 18:59.30 | MrWup | exten => s,n,AGI(asdasd.php|${arg1}|${arg2}) <-- where is the result from the php script? |
| 18:59.49 | *** join/#asterisk ctaloi (n=ctaloi@rrcs-24-97-110-187.nys.biz.rr.com) |
| 18:59.51 | irule | is there a fine tutorial for the COMPLETE newbie explaining the very basics? it all looks like stereo instructions :s |
| 18:59.52 | tuan_modulis | well, that depends on the exit code |
| 18:59.55 | tuan_modulis | BUT |
| 19:00.11 | tuan_modulis | u can set variables and stuff in the script itself |
| 19:00.26 | tuan_modulis | so that's like a returned result |
| 19:00.46 | tuan_modulis | agi command: SET VARIABLE |
| 19:01.01 | MrWup | but are all these on different threads? |
| 19:01.09 | MrWup | cause what if several calls are happening at the same time |
| 19:01.14 | MrWup | variables wont get mixed up right? |
| 19:01.26 | codazoda | AGI variables are unique to the channel. |
| 19:01.33 | MrWup | ok |
| 19:01.47 | tuan_modulis | yup |
| 19:02.09 | codazoda | I have a busy IVR and I want to look at the output from the CLI but only on a particular channel. Is there a command that I can use to limit the CLI output to that particular channel? Or, am I better turning on some type of logging? |
| 19:02.20 | bkw_ | codazoda, nope |
| 19:03.00 | codazoda | bkw_: Nope I can't look at just one channel? |
| 19:03.07 | bkw_ | codazoda, nope you can't |
| 19:03.08 | MrWup | so i need to load something into php.ini to allow me to set an AGI var from within my php script? |
| 19:03.25 | tuan_modulis | nono |
| 19:03.27 | bkw_ | MrWup, shouldn't have too |
| 19:03.35 | codazoda | Can I log the output to a file instead, in which case I could grep through it? |
| 19:03.37 | tuan_modulis | u need to execute AGI commands (see the wiki) |
| 19:06.17 | *** join/#asterisk mhammerton (n=mhammert@a24-244-140-15.maxil.com) |
| 19:06.20 | mhammerton | hey |
| 19:06.44 | MrWup | oh |
| 19:07.04 | MrWup | from within the PHP script i can write or echo "SET VARIABLE..." and than can access that var from the dialplan? |
| 19:07.14 | tuan_modulis | yea |
| 19:07.22 | MrWup | aha im getting it |
| 19:07.29 | tuan_modulis | u can do it within the script itself too |
| 19:07.33 | tuan_modulis | GET VARIABLE |
| 19:07.54 | MrWup | does my script.php have to be in a certain dir to be run with AGI( |
| 19:07.56 | MrWup | ? |
| 19:08.07 | tuan_modulis | var/lib/asterisk/agi-bin usually |
| 19:08.13 | MrWup | ah thanks |
| 19:08.18 | tuan_modulis | make subdirectories from there if u want |
| 19:08.21 | tuan_modulis | or sym links |
| 19:08.39 | *** join/#asterisk infernix (n=nix@unaffiliated/infernix) |
| 19:09.33 | mhammerton | hey i have 4 DID trunk lines and i want to know if i can use an tdm400 to do voice and fax on it |
| 19:09.37 | mhammerton | ? |
| 19:09.57 | tuan_modulis | now as for my problem... the client wants a customized voicemail system... *slaps forehead* |
| 19:11.28 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:11.58 | flujan | guys, I am trying to record my call queues calls... |
| 19:12.03 | flujan | without success. |
| 19:12.12 | flujan | someone can point me a tutorial of how to do it? |
| 19:12.19 | Mercestes | tuan_modulis, Good luck with all that |
| 19:12.55 | Mercestes | tuan_modulis, It's basically going to be a crap load of backgrounds, records, playbacks, etc with massively spaghettied dial plan matching. |
| 19:13.17 | thekidrio | anyone here use the aastra 9133i phone? |
| 19:13.19 | Mercestes | tuan_modulis, Eventually * is going to mvoe away frmo app_voicemail and go to an extensions.conf based solution. not sure how that's gonna look but you might wanna see if you can hold out for that |
| 19:13.48 | tuan_modulis | at the moment though, I'll try to stomach those 8000+ lines of code |
| 19:14.43 | MrWup | ah damn |
| 19:14.44 | MrWup | <PROTECTED> |
| 19:15.11 | MrWup | my php script is <?PHP echo "SET VARIABLE ID \"SIP/102\""; |
| 19:15.43 | bkruse | try to escape that / |
| 19:15.43 | bkruse | eg |
| 19:15.54 | bkruse | or, better yet, use string encap |
| 19:16.09 | bkruse | <? echo 'SET VARIABLE ID "SIP/102"'; ?> |
| 19:16.09 | bkruse | ;] |
| 19:17.04 | MrWup | same... exec format error |
| 19:17.47 | tuan_modulis | see php + agi on the wiki |
| 19:17.57 | tuan_modulis | there's a bunch of stuff you'll need to do as well |
| 19:18.04 | MrWup | oh |
| 19:18.11 | tuan_modulis | like to not output the headers |
| 19:18.12 | *** join/#asterisk harleya (n=harleya@001-782-782.area1.spcsdns.net) |
| 19:18.33 | *** join/#asterisk VaRMiT (n=varmit@CPE-76-177-75-124.natcky.res.rr.com) |
| 19:19.17 | VaRMiT | is there a way to setup where when i make a call, the other phones will know that Line1 is in use? on asterisk? |
| 19:19.52 | tuan_modulis | ischanavail |
| 19:20.06 | VaRMiT | tuan_modulis where would i pop that in at? |
| 19:20.21 | b11d | VaRMiT.. i think you want "presence" ? |
| 19:20.22 | tuan_modulis | in the dialplan |
| 19:20.48 | b11d | or "buddy list" stuff? |
| 19:21.22 | VaRMiT | tuan_modulis your not talking about FreePbx are you? |
| 19:21.34 | tuan_modulis | there |
| 19:21.41 | tuan_modulis | there's BristuffDevstate |
| 19:21.50 | VaRMiT | please excuse my ignorance |
| 19:22.18 | tuan_modulis | no, I'm talking about asterisk commands/applications |
| 19:22.23 | VaRMiT | ok |
| 19:22.37 | VaRMiT | im runnign trixbox.... |
| 19:22.45 | VaRMiT | so this wouldnt apply to what im running.... |
| 19:22.47 | b11d | go to #freepbx then |
| 19:22.48 | VaRMiT | correct |
| 19:22.53 | VaRMiT | b11d thank you |
| 19:22.55 | b11d | np |
| 19:23.02 | *** part/#asterisk VaRMiT (n=varmit@CPE-76-177-75-124.natcky.res.rr.com) |
| 19:25.45 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 19:26.10 | MrWup | how odd |
| 19:26.18 | MrWup | there is no php in my /usr/bin even though i got php installed |
| 19:26.58 | b11d | /usr/local/bin ? |
| 19:27.09 | tuan_modulis | which php |
| 19:28.01 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:28.13 | MrWup | not in usr/local/bin either |
| 19:28.21 | MrWup | php is working fine too, i been using it with apache |
| 19:28.25 | MrWup | php5 |
| 19:28.36 | tuan_modulis | i mean type "which php" |
| 19:29.12 | MrWup | into console? |
| 19:29.14 | MrWup | nothing happens |
| 19:29.22 | b11d | no the CLI, but the command prompt. |
| 19:29.22 | thekidrio | anyone know a good supplier for hawaiian DID's? 808 NPA |
| 19:29.34 | MrWup | i know. the linux command prompt |
| 19:29.36 | MrWup | nothing happens |
| 19:29.42 | thekidrio | $ whereis php |
| 19:29.50 | thekidrio | $ whereis php5 |
| 19:30.13 | MrWup | ah |
| 19:30.14 | MrWup | php5: /etc/php5 /usr/lib/php5 /usr/share/php5 |
| 19:30.26 | thekidrio | what distro? |
| 19:30.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:30.41 | MrWup | debian etch |
| 19:31.22 | *** join/#asterisk r0d3nt (n=RatMan@punk.valuetel.net) |
| 19:31.37 | thekidrio | hehe should have asked that first :) |
| 19:32.20 | MrWup | still cant find php though |
| 19:32.22 | MrWup | the actual binary |
| 19:32.39 | thekidrio | what are you trying to do with it? |
| 19:32.54 | MrWup | <PROTECTED> |
| 19:32.54 | MrWup | <PROTECTED> |
| 19:33.02 | MrWup | so i can exec a php script with AGI |
| 19:33.23 | thekidrio | you can run php from the command line if you have php5-cli installed |
| 19:33.56 | thekidrio | and you can view working php scripts via http right? |
| 19:34.00 | MrWup | yeah |
| 19:34.02 | MrWup | works fine |
| 19:35.56 | hedge77 | is the right way to add a hint for an extension in AEL: hint(SIP/4101) 4101 => NoOp(); ? |
| 19:36.09 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:36.26 | hedge77 | MrWup: does phpinfo() tell you where the binary is? |
| 19:37.20 | *** join/#asterisk kannan (n=kannan@210.211.179.42) |
| 19:38.53 | MrWup | for environment it says PATH /usr/local/bin:/usr/bin:/bin |
| 19:41.56 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 19:42.30 | MrWup | but it aint there! |
| 19:43.38 | hedge77 | might be time for an updatedb&; locate php | grep "/php" |
| 19:44.31 | MrWup | oh |
| 19:44.35 | MrWup | apparently i need php5-cli |
| 19:44.46 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 19:45.03 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:45.08 | thekidrio | hehe |
| 19:45.28 | xo8ox | guys why is it that when we dial 112 the extention rings busy after dialing 11.. but if we dial 122 the extention works just fine ? |
| 19:45.43 | xo8ox | its like the system doesn't like us dialin 1 after another 1 |
| 19:45.50 | thekidrio | is 11 an function code? |
| 19:45.52 | thekidrio | i can't recall |
| 19:45.59 | xo8ox | I don't know |
| 19:46.02 | thekidrio | s/an/a/n |
| 19:46.04 | thekidrio | hrmm |
| 19:46.07 | xo8ox | where do I see all these codes ? |
| 19:46.19 | thekidrio | extensions usually |
| 19:46.28 | thekidrio | mine are in extensions_feature_codes.conf |
| 19:46.33 | xo8ox | all of our extentions work except the ones that start with 1 and follows by either 1 or 0 |
| 19:47.37 | thekidrio | I wonder if those are reserved leaving [1-9][2-9]X your limit for ext |
| 19:47.39 | CunningPike | Quick pattern matching check - can I match 8170, 8172, 8273 and 8178 with a single exten? I've tried _817[0238] and _817[0|2|3|8] - neither does the trick |
| 19:48.04 | xo8ox | that could be |
| 19:48.24 | thekidrio | I am not sure though xo8ox I am very interested in a solution if you find one hehe |
| 19:48.28 | xo8ox | but get this.. if we dial the complete extention on the phone then hit the dial botton then it works |
| 19:48.48 | thekidrio | any phone or just one type? |
| 19:48.54 | xo8ox | any phone |
| 19:49.12 | xo8ox | and from outside dialing in the 11x extentions work just fine |
| 19:49.28 | CunningPike | nm - can't type properly |
| 19:49.29 | xo8ox | its just that internally if we pick up the phone and hear the dial tone and dial 11x then it rings busy |
| 19:49.33 | thekidrio | so inbound from outside works, but inbound from inside is failing? |
| 19:50.08 | thekidrio | CunningPike, hehe |
| 19:50.08 | xo8ox | yup |
| 19:51.04 | thekidrio | _817[0238] should work eh |
| 19:51.06 | MrWup | gah |
| 19:51.12 | MrWup | ive done all the stuff the wiki asks me to do |
| 19:51.17 | MrWup | and still i get the exec format error |
| 19:51.27 | thekidrio | I would take a quick break Wup |
| 19:51.33 | thekidrio | go get a drink or something and come back to it |
| 19:52.17 | thekidrio | then if that does not enlighten you, i guess start over at the beginning and take a long time with each step to make sure nothing was missed |
| 19:52.24 | MrWup | ok |
| 19:52.25 | thekidrio | I hate errors like that though hehe |
| 19:52.55 | thekidrio | what's worse is having another engineer walk up behind you and say, thats your problem right there, just do this... ugh hate tha |
| 19:53.16 | tuan_modulis | just do.... naah |
| 19:54.03 | tuan_modulis | the way I do it is by sleeping and hoping the problem solves itself the next day |
| 19:54.16 | tuan_modulis | it works sometimes |
| 19:56.12 | mhammerton | hey does asterisk support fax services, i know i can set up voicemail with it but i am not sure about fax |
| 19:56.36 | tuan_modulis | sure, see the wiki for different solutions |
| 20:00.52 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:00.56 | b11d | i like to hide under a pile of coats until I hear it's all clear. |
| 20:01.11 | MrTelephone | is there a device that will direct a t1 to a different port if there is a physical layer failure? |
| 20:02.15 | b11d | yeah just type "t1 failover" in google |
| 20:02.18 | thekidrio | b11d watch out, people like to shit on the coats :( |
| 20:02.31 | b11d | uh oh.. i better do some testing first then |
| 20:02.49 | thekidrio | heh |
| 20:03.10 | thekidrio | speaking of testing, it is lunch time |
| 20:03.20 | b11d | hahaha |
| 20:04.04 | MrTelephone | hmmm |
| 20:04.11 | MrTelephone | yeah i don't find too many products by googling |
| 20:04.39 | MrTelephone | t1 mux? |
| 20:04.49 | MrTelephone | something that will share your b channels with 2 pri cards? |
| 20:05.08 | tuan_modulis | hmm, maybe openvpn can accomodate that type of failover |
| 20:05.30 | b11d | i know on the wiki there is a "t1 failover" section |
| 20:05.38 | MrTelephone | using tdmoe |
| 20:05.43 | MrTelephone | ? |
| 20:07.21 | b11d | ohh man, thank god.. the new season of Trailer Park Boys has begun |
| 20:07.38 | Qwell[] | b11d: sweet |
| 20:08.00 | b11d | yeah, i've been waiting for this for awhile now.. |
| 20:10.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:11.56 | MrWup | sigh |
| 20:11.56 | MrWup | well ive got the script to execute now |
| 20:11.56 | MrWup | turns out i had whitespace at the top of my script |
| 20:11.56 | MrTelephone | http://www.gl.com/mltiport.html |
| 20:11.56 | MrWup | but even using the example php script |
| 20:11.56 | MrTelephone | what about something like this |
| 20:12.03 | MrWup | <PROTECTED> |
| 20:12.03 | MrWup | is all i get, and no output |
| 20:12.04 | MrTelephone | I think this is what I was looking for |
| 20:13.42 | hedge77 | ok i have this hint(SIP/4102) 4102 => NoOp(); in an context included in my internal one (where exten for VM etc. is). and I get "Got SUBSCRIBE for extension 4102@BLF-group from 192.168.1.211, but there is no hint for that extension." when registering |
| 20:15.06 | hedge77 | am I putting the hint() in the wrong place/way? |
| 20:15.51 | MrTelephone | Thats sweet.. you can have an snmp script to check if the http://dataprobe.com/products/switch/aps/t-aps/index.html is in failover mode.. if so, run the script to switch your IP to the main |
| 20:15.53 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 20:18.58 | tuan_modulis | pretty nice |
| 20:18.59 | Mahmoud | how to disable Sipura's voice menu? |
| 20:20.24 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-7b5136890c000103) |
| 20:20.24 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 20:22.50 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 20:25.52 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-222a502e7808aade) |
| 20:31.22 | hedge77 | hmm this might be an ael parsing problem. when it gets translated it shows up as exten => 4102,-1,SIP/4102 instead of exten => 4102,hint,SIP/4102 |
| 20:31.46 | Qwell[] | hedge77: hint is -1 |
| 20:31.54 | hedge77 | oh :( |
| 20:32.01 | Qwell[] | we only call it 'hint' for users sake |
| 20:32.36 | hedge77 | wait that makes even less sense. why would SUBSCRIBE not find the hint then? |
| 20:32.57 | Qwell[] | is the hint in BLF-group? |
| 20:33.02 | hedge77 | yes |
| 20:33.14 | Qwell[] | in, or included in? |
| 20:33.34 | hedge77 | in context BLF-group, included in internal, and the sip.conf entry has subscribecontext=BLF-group. |
| 20:35.15 | zimdog | Hello All. Does anyone know if it is possible to set the ftp server adddress for a polycom ip500 in the phone config file. I have phones setup and want to change the ip address for the ftp server and then have the suers reboot the phones instead of having them login and change the ftp ip address. |
| 20:36.46 | MrWup | yeeehaw its almost working |
| 20:37.05 | MrWup | with AGI vars... agi_callerid gives Name <num> whats the best way to just get the number of the caller? |
| 20:38.01 | Mahmoud | hedge77, i try putting a separate line with hint only, without SIP/foo and works.. dunno if it helps you |
| 20:38.51 | Mahmoud | hedge77, oh sorry, i put SIP/foo too |
| 20:39.03 | Mahmoud | hedge77, never mind |
| 20:40.42 | Mahmoud | hedge77, are you using macros? |
| 20:40.49 | hedge77 | i think i got it |
| 20:43.12 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 20:46.17 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 20:51.58 | tuan_modulis | is asterisk realtime configuration stable? |
| 20:52.43 | tuan_modulis | that's another step to the future right there |
| 20:54.56 | Dovid | yes. very stable. |
| 20:55.01 | Dovid | i have been using it since it came out |
| 20:55.05 | tuan_modulis | excellent |
| 20:56.55 | irule | how can I find the explanation of 's' withing a dialplan in voip wiki? |
| 20:57.15 | frigidzephyr | of the s, extension? |
| 20:57.18 | b11d | go to voip-info.org |
| 20:57.23 | b11d | read up on extensions.conf |
| 20:57.40 | b11d | there is a little white box on the left called a "search" box. |
| 20:57.48 | b11d | it allows you to 'search' the contents of the website. |
| 20:57.49 | b11d | its neat. |
| 20:57.55 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 20:58.02 | thekidrio | haha shaddup b11d |
| 20:58.05 | b11d | :) |
| 20:58.08 | irule | blld search for 's' is not as accurate as I hoped for |
| 20:58.14 | thekidrio | too broad |
| 20:58.15 | b11d | yeah dont search for "s" |
| 20:58.19 | b11d | search for "extensions.conf" |
| 20:58.21 | b11d | and then look at "s" |
| 20:58.23 | irule | ok thanks |
| 20:58.25 | b11d | np |
| 20:58.32 | frigidzephyr | if you google for "s extension asterisk" it comes upw ith voip-info page on it |
| 20:58.47 | b11d | wow it's like "the internet" just came out or something |
| 20:59.01 | frigidzephyr | it didnt? |
| 20:59.04 | b11d | :) |
| 20:59.11 | frigidzephyr | i just downloaded the internet a few days ago |
| 20:59.14 | frigidzephyr | =D |
| 20:59.16 | b11d | i've got a hard copy at home.. |
| 20:59.19 | b11d | :) |
| 20:59.35 | frigidzephyr | oh sweet, is it on 5 1/2 disk? My disk drive only accepts those |
| 20:59.47 | thekidrio | Al Gore invented the internet in 1995 |
| 20:59.48 | irule | can I create an extension that will route me into s to simulate calling myself from POTS line?\ |
| 20:59.49 | thekidrio | just ask him |
| 21:00.01 | thekidrio | irule, s is the default start extension |
| 21:00.16 | frigidzephyr | exten => 100,1,Goto(context,s,1) |
| 21:00.18 | b11d | I remember back in 2003 he came out and said "I didnt invent it, i just steered the direction towards the creation of it" |
| 21:00.19 | irule | just like that? dial s? |
| 21:00.19 | thekidrio | its like a variable, if the call has no where specific to go it will go to the first s exten |
| 21:00.22 | irule | great thanks! |
| 21:00.39 | thekidrio | or exten => s,1,application() and so on |
| 21:00.43 | *** join/#asterisk BitBandit (n=PX2@65-103-229-223.slkc.qwest.net) |
| 21:01.33 | b11d | quick, hide your bits.. the bitbandit has arrived |
| 21:01.36 | b11d | wow, that was bad. |
| 21:02.30 | zimdog | Hello All. Does anyone know if it is possible to set the ftp server adddress for a polycom ip500 in the phone config file. I have phones setup and want to change the ip address for the ftp server and then have the suers reboot the phones instead of having them login and change the ftp ip address. |
| 21:05.29 | MrTelephone | zimdog use tftp |
| 21:07.07 | zimdog | MrTelephone: I am not sure I understand how that resolves the situation of being able to point the phones to a different IP address once they are configured? |
| 21:12.01 | thekidrio | ok noob .call file question here. if I want to dial a number say 15552771212 ext 8888 how do I put that in a .call file ie. Exten: 15552771212888\r\n or is there a pause or something? |
| 21:13.05 | bkruse | thekidrio: your going to want some kind of call flow |
| 21:13.13 | bkruse | dial the number, and drop it into a local context |
| 21:13.37 | bkruse | that will wait like 3 seconds, or whatever, then sendtmf(8888) then have it call you and bridge the call |
| 21:13.39 | bkruse | if thats what u want |
| 21:13.40 | thekidrio | then dial exten? |
| 21:13.44 | thekidrio | gotcha |
| 21:13.56 | thekidrio | so I can't do it right in the original .call file? |
| 21:14.46 | bkruse | well |
| 21:14.47 | bkruse | you can |
| 21:14.50 | bkruse | but, the problem is |
| 21:15.04 | thekidrio | would be great if i could do Exten: 15552771212,,,8888 with the , indicating a seconds pause |
| 21:15.06 | bkruse | you can specify application: senddtmf |
| 21:15.07 | bkruse | argument(or whatever it is): 8888 |
| 21:15.14 | bkruse | but that would be it.......it would hangup when sendtmf is done |
| 21:15.39 | bkruse | write a patch :P |
| 21:15.58 | thekidrio | good idea, now only if I knew more :P |
| 21:22.56 | Dovid | any one here using 1.4.X for a stable enviroment ? |
| 21:23.11 | Dovid | i am getting a new box and I am thinking of making the switch. |
| 21:24.46 | bkruse | thekidrio: i think thats a pretty cool feature though, i might hack at that this week |
| 21:27.09 | hedge77 | hints aaaaaaaaag |
| 21:27.24 | bkruse | ? |
| 21:28.01 | hedge77 | subscribes keep saying there is no hint for an extension when there certainly is and it shows in the dialplan |
| 21:28.47 | bkruse | i love hints :D |
| 21:29.24 | hedge77 | :( |
| 21:31.03 | thekidrio | bkruse, yeah would be nice I will look at it and see how invasive the change would be |
| 21:31.12 | thekidrio | seems small, but those usually end up biting me in the ass |
| 21:32.02 | bkruse | thekidrio: ha, ya, just drop it into a context and go from there |
| 21:32.20 | bkruse | i want to, one day, do flow control for manager so that you can name off applications and make temporary extensions, if you will, on the fly |
| 21:32.37 | thekidrio | yeah that would be very useful! haha |
| 21:33.04 | thekidrio | hrmm now my question is can I set a VAR in the .call file that the context can grab the extension from |
| 21:33.36 | thekidrio | currently the .call file is being generated via click to call script |
| 21:35.27 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 21:37.26 | Dovid | too many call failes will send the server off. u may want to use the AMI |
| 21:39.24 | irule | exten => 700,1,Goto(context,s,1) *CLI> -- Executing [700@local:1] Goto("SIP/sip653-0819f8b8", "context|s|1") in new stack-- Goto (context,s,1) [Apr 5 10:26:55] WARNING[20626]: pbx.c:2432 __ast_pbx_run: Channel 'SIP/sip653-0819f8b8' sent into invalid extension 's' in context 'context', but no invalid handler |
| 21:39.58 | *** join/#asterisk bmg505 (n=leon@196.209.178.107) |
| 21:40.21 | thekidrio | call failes or files? (just want to make sure you are talking to me hehe) |
| 21:41.32 | tuan_modulis | irule: that context doesnt exist most likely |
| 21:41.38 | tuan_modulis | try Goto(s,1) |
| 21:44.23 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 21:48.42 | *** part/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 21:48.48 | irule | Goto(s,1) works grerat, awesome! thanks! ... |
| 21:50.37 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 21:50.37 | *** mode/#asterisk [+o file] by ChanServ |
| 21:50.51 | *** join/#asterisk garreel (n=garreel@host48-2-dynamic.16-87-r.retail.telecomitalia.it) |
| 21:51.10 | irule | how can I dial an operator with 0? |
| 21:51.20 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 21:51.32 | *** join/#asterisk [hC] (n=hardcore@host-185-58.dsl-sea.seanet.com) |
| 21:51.33 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 21:52.47 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:52.55 | diclophis-work | so whats the going rate for a cisco sip gateway? |
| 21:53.33 | hedge77 | what context do I need my hint rules in? The one the sip user is put in, an "internal" one (where the #'s for voicemail, meetme, etc. are), or somewhere else? |
| 21:54.17 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net) |
| 21:55.29 | *** join/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net) |
| 21:55.54 | knarfly | can anyone offer advice on parsing the * log file? |
| 21:56.21 | *** join/#asterisk shodan (n=shodan@ip084.99-113-216.pppoe4.joliette.intermonde.net) |
| 22:00.06 | mvanbaak | vim |
| 22:01.25 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 22:01.25 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 22:02.06 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 22:07.44 | irule | is this correct? Im dialing 603, thats the softphone that works atm exten => S0,n,Goto(context,603) |
| 22:08.32 | tzafrir_laptop | knarfly, regarding some specific errors, or a generic parser? |
| 22:12.31 | irule | what does Playback do? |
| 22:13.20 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 22:15.30 | Mercestes | irule: If you are dialing 603 then I'm pretty sure you want exten => 603,1,Goto(context,603) |
| 22:15.38 | Mercestes | irule: You always start with 1. |
| 22:15.46 | Mercestes | and I don't thing s0 == dialing 603 |
| 22:16.55 | knarfly | tzafrir_laptop: I think I found it...thanks anyway....cdr-custom works better |
| 22:18.27 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 22:19.23 | irule | thanks Mercestes, I got this new error what does it mena_[Apr 5 11:15:25] WARNING[20613]: pbx.c:4644 add_pri: Unable to register extension 's', priority 2 in 'default', already in use |
| 22:25.22 | irule | [Apr 5 11:22:32] WARNING[21271]: pbx.c:2476 __ast_pbx_run: Invalid extension '0', but no rule 'i' in context 'local' |
| 22:27.23 | Mercestes | irule it means you have two exten => s,2 in the same context |
| 22:27.55 | Mercestes | error 2 means you dialed "0" which is invalid in whatever context you are in, but you don't have an exten => i,1 to handle the invalid event. |
| 22:28.26 | Mercestes | if you add either exten => 0 or exten => i to your phone's context error #2 will go away, but you should understand the difference first. |
| 22:32.00 | demlak | hmm.. whats that? this was during a call between sip client and isdn caller http://rafb.net/p/iLkjoY67.html |
| 22:33.25 | irule | thanks a lot guys this is fun! |
| 22:33.42 | Mercestes | np |
| 22:33.46 | Mercestes | welcome to asterisk! :) |
| 22:33.57 | irule | exten => t,1,Macro(stdexten,603,SIP/sip603) call my chose operator line, is this the correct way to do things or is there a more propper way to set up an operator? |
| 22:34.14 | irule | awesome |
| 22:34.56 | Mercestes | irule, t is a timeout. you don't want to use it unless you really mean a timeout condition |
| 22:35.02 | irule | can I asign a different ring tone? I want to everyone to know when a real call is coming and when I am playing with the phones lol |
| 22:35.57 | irule | ok thanks |
| 22:37.03 | Mercestes | <PROTECTED> |
| 22:37.32 | irule | ok thanks |
| 22:37.41 | demlak | whats the cli syntax to get the chan_capi version? |
| 22:38.45 | jql | the header is Alert-Info: |
| 22:40.31 | e-milio | Hello all |
| 22:42.14 | Mercestes | 'elllo |
| 22:42.23 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 22:45.41 | irule | how can I send callers to an invalid message if they type a wrong number and too many numbers? |
| 22:45.48 | Mercestes | Yes |
| 22:45.55 | Mercestes | with exten => i |
| 22:46.22 | irule | are all its opntions in voip-info? |
| 22:46.45 | *** join/#asterisk type0 (n=type0@216-67-27-54-cdsl-rb2.cwc.acsalaska.net) |
| 22:46.48 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 22:47.12 | type0 | using a polycom 310 with asterisk doing a sip or iax session, whats the bandwidth needed? |
| 22:47.16 | type0 | am I stupid to estimate around 70kbps |
| 22:47.17 | type0 | ? |
| 22:48.51 | hedge77 | if you use the ulaw codec that's about right |
| 22:48.58 | type0 | score |
| 22:49.18 | type0 | what kind of bandwidth is taken when the phone isnt in use? |
| 22:49.48 | hedge77 | why is * not marking my hint'ed exts as busy when they're in use? They stay at "State:Idle" |
| 22:50.17 | hedge77 | type0: probably not very much at all |
| 22:50.18 | irule | that is more propper and why? exten = i,2,Goto(incoming,s,1) or exten => i,n,Goto(s,restart) |
| 22:50.25 | mcab | type0: just periodic REGISTER messgaes, so not a lot |
| 22:50.38 | type0 | I'm REALLY limited in my bandwidth |
| 22:50.56 | type0 | So far, its looking good.. all my shit is at 1.1mb |
| 22:51.04 | type0 | I only have a T-1 I can use |
| 22:51.17 | mcab | type0: just keep in mind Polycoms don't support IAX |
| 22:51.27 | hedge77 | uhhh a T1 should be plenty :confused: |
| 22:51.28 | type0 | all good |
| 22:51.31 | type0 | ill use sip |
| 22:51.34 | type0 | hedge77 |
| 22:51.39 | type0 | I have A LOT of other shit |
| 22:52.02 | type0 | I have an air to ground comm system that uses 256k |
| 22:52.14 | type0 | I have a SCADA system that uses 128k per site |
| 22:52.16 | hedge77 | oh you're utilizing 1.1mb worth of your T1 currently. |
| 22:52.20 | type0 | yeah |
| 22:52.32 | type0 | I really have 4 T-1's |
| 22:52.33 | mcab | type0: might be worth looking at G.729 |
| 22:52.35 | type0 | but im trying not to use them |
| 22:52.50 | hedge77 | use gsm it's what all the cool kids do |
| 22:53.06 | type0 | Cause at the end of the road, I'd have to mux the 4 T-1's worth of shit into 2 for the backhaul |
| 22:55.37 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:56.46 | type0 | Originally this morning a guy told me all of my radar gear was going to have to travel on INMARSAT |
| 22:56.50 | type0 | i was like.. are you fucking KIDDING ME |
| 22:57.00 | type0 | 11 dollars a minute, 24 hours a day, 365 a year |
| 22:58.05 | gordonjcp | type0: errr |
| 22:58.16 | gordonjcp | yeah inmarsat is not cheap |
| 22:58.27 | jql | aren't the new ones down to $3? |
| 22:59.03 | type0 | inmarsat is 11 dollars a minute |
| 22:59.27 | type0 | 6.1 million a year |
| 22:59.35 | jql | inmarsat a, perhaps. there's like 6 inmarsat networks |
| 22:59.43 | gordonjcp | naff all if you're an oil company of course |
| 22:59.51 | irule | is there a std sound that will say something similar to -or dial to listen to your options? |
| 23:00.04 | irule | 2 lazy 2 read 200 files names lol |
| 23:01.21 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:02.11 | type0 | this is for the united states airforce fighter training complex here in alaska |
| 23:02.19 | type0 | so in reality, money is nothing |
| 23:02.32 | type0 | however, they would NOT go for a 6.1 million dollar CONNECTION a year |
| 23:02.56 | gordonjcp | type0: how far to the nearest fibre? |
| 23:03.01 | type0 | HAHAHAHAHHA |
| 23:03.05 | type0 | 500 miles |
| 23:03.09 | gordonjcp | ah, pish easy |
| 23:03.14 | type0 | im doing all microwave shots |
| 23:03.17 | iCEBrkr | FSX Port Input Gain on a Sipura.. That'll change the gain on 'outgoing' audio from my phone to Asterisk, right? |
| 23:03.19 | jql | (of ice-covered tundra) |
| 23:03.21 | gordonjcp | type0: yeah |
| 23:03.37 | irule | has anybody ecualized a males voice to sound female? >:D |
| 23:03.50 | jql | irule: there's an app for that |
| 23:03.53 | gordonjcp | I was going to say, 8M GBP would probably do you 500 miles if at least some of the towers are in place |
| 23:04.04 | gordonjcp | fuck knows how much you'd need to spend in USD |
| 23:04.41 | type0 | we're running all adtran and alcatel 4 x T1 radios |
| 23:04.44 | type0 | 5.8ghz |
| 23:04.46 | type0 | unlicensed ism |
| 23:04.50 | gordonjcp | o_O |
| 23:04.57 | gordonjcp | for your entire backhaul? |
| 23:05.10 | type0 | nah, to get to the telcom's microwave site |
| 23:05.16 | gordonjcp | ah righty, how far's that then? |
| 23:05.23 | type0 | after it gets there, its on OC-3 radios, but they only have 2 available t-1's left |
| 23:06.11 | type0 | then it gets to another site, where its interconnected and routed to the airforce base |
| 23:06.11 | gordonjcp | what sort of distance are you covering on 5.8? |
| 23:06.25 | type0 | each hop, or total? |
| 23:06.25 | gordonjcp | I'm guessing you won't get a lot of interference |
| 23:06.25 | gordonjcp | total |
| 23:06.25 | type0 | total one range is over 250 miles |
| 23:06.33 | type0 | the other range is just short of 350 |
| 23:06.37 | *** join/#asterisk [hC] (n=hardcore@12-46-54-89.seatac.seattwa.wayport.net) |
| 23:07.06 | gordonjcp | that's a lot on 5.8, biggest I set up was 40 miles, across low-lying ground with trees at one end |
| 23:07.12 | gordonjcp | Orthogon Geminis |
| 23:07.12 | type0 | the longest hop is around 90miles |
| 23:07.22 | type0 | im at 5000+ft on each site |
| 23:07.29 | gordonjcp | aw you've got it made... |
| 23:07.42 | type0 | the only obstruction is like.. a bird. |
| 23:08.00 | hedge77 | wooooooo got hints/blf working now I can go home |
| 23:08.00 | gordonjcp | type0: this was from 10' ASL to 20' ASL with trees at the 20' end... |
| 23:08.06 | type0 | fuck that |
| 23:08.12 | gordonjcp | type0: that said both ends were on 100' towers |
| 23:08.26 | *** part/#asterisk hedge77 (n=ambray@209.42.192.214) |
| 23:08.26 | type0 | ahh so you were a ways off agl |
| 23:08.30 | gordonjcp | yeah |
| 23:08.36 | type0 | thats not SO bad then |
| 23:08.41 | type0 | i thought you were SHOOTING at 20' |
| 23:08.53 | gordonjcp | still a bit of incursion into the path like |
| 23:08.57 | type0 | I would have suggested a bulldozer at that point |
| 23:09.10 | gordonjcp | heh, I have a 7km link on 2.4GHz across town ;-) |
| 23:09.31 | type0 | I know a guy who has some personal 14ghz links |
| 23:09.35 | gordonjcp | the key is *biiiiig* antennas and a cavalier attitude to co-channel interference |
| 23:09.58 | gordonjcp | yeah, I nearly bought a 24GHz licenced-band pair, but didn't have the cash handy at the right time |
| 23:10.02 | gordonjcp | usual story |
| 23:10.14 | Sweeper | :D |
| 23:10.18 | type0 | anyone use T1 TDM tunnels before? |
| 23:10.27 | type0 | this is going to be the first time ive ever messed with them |
| 23:10.29 | gordonjcp | Sweeper: the Motorola gear isn't funny |
| 23:10.33 | type0 | I have one device which REQUIRES a t-1 |
| 23:10.39 | Sweeper | gordonjcp: it's aweful shit |
| 23:10.41 | gordonjcp | Sweeper: I set one up on a block of flats |
| 23:10.41 | type0 | to im going to plug it into a T-1 to ethernet 'converter' |
| 23:10.43 | Sweeper | horrible interference |
| 23:10.50 | Sweeper | I mean |
| 23:10.52 | Sweeper | it works with itsel |
| 23:10.55 | Sweeper | *itself |
| 23:10.56 | gordonjcp | pointed it *roughly* at my flat, a couple of miles across town |
| 23:11.01 | gordonjcp | and it got a signal |
| 23:11.14 | Sweeper | but the channel hopping it does wreaks havoc with anyone else on the spectrum |
| 23:11.17 | gordonjcp | fsck knows how much it cooks you when you're working near it |
| 23:11.21 | gordonjcp | Sweeper: yeah, it's well rude |
| 23:11.50 | Sweeper | it had just gone poof, all of the sudden |
| 23:12.01 | Sweeper | saw canopies next door |
| 23:12.02 | type0 | T1 to ethernet is pretty much the coolest thing ive ever heard |
| 23:12.15 | type0 | but im not so sure about the latency its going to create setting up and tearing down |
| 23:12.19 | gordonjcp | Sweeper: yeah, a lot of where I live is bad for interference on 5.8 too |
| 23:12.24 | Sweeper | the company with the 11b link was private security.... |
| 23:12.41 | gordonjcp | there was one particular site, there just wasn't a clear bit of band at all |
| 23:12.49 | Sweeper | I told them "if a stray bullet *happened* to hit that little grey thing over there, your internet will work again" |
| 23:12.58 | gordonjcp | heh |
| 23:13.20 | Sweeper | problem "solved itself" during the night <,< |
| 23:13.22 | gordonjcp | Sweeper: in some of the less pleasant parts of Glasgow, we've had Alvarion SUs shot off the roofs before |
| 23:13.40 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:13.46 | [hC] | are any of you guys using polycom 601's with sidecars on em and doing hints with asterisk? I have a couple installs that use 2 or 3 sidecars with hints and for whatever reason, the polycom will entirely reboot a couple times a day, and I cannot figure out why. |
| 23:14.10 | Sweeper | [hC]: so you turn hints off, and they stop rebooting? |
| 23:14.40 | mcab | [hC] what version of f/w are you running? |
| 23:15.06 | [hC] | Sweeper: I havent tried yet. |
| 23:15.16 | [hC] | mcab: one is 1.6.7 one is 2.0.8 i think. |
| 23:15.16 | type0 | http://www.engageinc.com/Products2/iptube_DLT1.htm |
| 23:15.18 | type0 | check that shit out |
| 23:15.21 | [hC] | i'll check. |
| 23:15.22 | Sweeper | [hC]: might try, you know, some troubleshooting |
| 23:15.35 | mcab | [hC]: 2.0.3? |
| 23:15.47 | [hC] | It seems to happen with only 2 or more sidecars |
| 23:15.50 | d00gster | guys, I have a sip client (eyebeam) overseas connecting to my asterisk. since they have high latency, I dud a qualify=5000 in one instance and =no in another. I also forced the client to register every 300 seconds (eyebeam option). the client can pickup the line and call me anytime of the day. when I call the client, they don't see the call come in and I go to vm. I asked the client to dial 7777 and dial his extension and tha |
| 23:15.51 | [hC] | on both firmwares |
| 23:15.54 | [hC] | let me check which 2.x i have |
| 23:15.57 | *** join/#asterisk Dirk- (n=a@82-33-155-212.cable.ubr04.wiga.blueyonder.co.uk) |
| 23:15.58 | Qwell[] | type0: it "exploits the efficiency of ethernet"? |
| 23:16.25 | Sweeper | type0: http://www.rad.com/Home/0,6583,2,00.html <-- these guys do it better |
| 23:16.27 | Sweeper | and bigger |
| 23:16.27 | type0 | well, for people who cant run a t-1 to each site |
| 23:16.53 | [hC] | mcab: 2.0.3 yes. |
| 23:17.11 | type0 | i saw that shit and was like |
| 23:17.12 | type0 | OH YES |
| 23:17.26 | type0 | im just wondering what kind of latency its going to create |
| 23:17.33 | mcab | [hC]: have a <mac>-app.log you can pastebin? |
| 23:17.45 | [hC] | This problem seems totally random, and sometimes happens multiple times a day, other times doesnt do it for a few days. |
| 23:17.53 | [hC] | mcab: of course. give me a sec. |
| 23:18.02 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 23:18.13 | Sweeper | type0: again, RAD does it better :3 |
| 23:18.17 | Sweeper | and no latency \o\ |
| 23:18.22 | type0 | really? |
| 23:18.28 | type0 | it does not create ANY latency? |
| 23:18.28 | Sweeper | at least, nothing more than the regular network latency |
| 23:18.29 | CrashHD | anyone having problems with vitelity lately? |
| 23:18.35 | Sweeper | nah, doesn't create any |
| 23:18.38 | type0 | wow |
| 23:18.44 | type0 | whats better about rad? |
| 23:18.47 | Sweeper | we use it a *lot* |
| 23:19.16 | [hC] | mcab: http://pastebin.ca/426187 |
| 23:19.24 | type0 | so you think it would be the same thing as hooking the device up to an actual T-1? |
| 23:19.49 | Sweeper | type0: yep |
| 23:20.00 | Sweeper | I mean, with asterisk, it's mostly pointless |
| 23:20.07 | type0 | im trying to do data here |
| 23:20.11 | [hC] | mcab: that was the entire contents of the file. |
| 23:20.11 | Sweeper | ..... |
| 23:20.17 | Sweeper | data over t1 over ip? |
| 23:20.24 | type0 | yeah |
| 23:20.28 | Sweeper | wtf dude |
| 23:20.34 | type0 | ok check it out |
| 23:20.38 | type0 | I have a thing called a TIS |
| 23:20.43 | type0 | its for tracking fighter jets |
| 23:20.50 | type0 | it requires a T-1 |
| 23:20.55 | Sweeper | really? |
| 23:20.59 | garreel | "AMP is intended first and foremost to be a proxy to an asterisk PBX server. Due to some inherent restrictions in Asterisk, multiple sockets connected to Asterisk's manager interface have been reported to make it crash."... is it true? |
| 23:20.59 | type0 | I can give it the T-1.. |
| 23:21.00 | Sweeper | doesn't have a v.35 or anything? |
| 23:21.03 | mcab | [hC]: do any of the date stamps cover the period where they were seeing this? unfortunately, there's not much here... |
| 23:21.03 | type0 | no |
| 23:21.06 | type0 | ONLY t-1 |
| 23:21.07 | Sweeper | mkay |
| 23:21.09 | Sweeper | so |
| 23:21.09 | type0 | so |
| 23:21.11 | Sweeper | get a cisco |
| 23:21.14 | type0 | I can give it the T-1 |
| 23:21.14 | Sweeper | and stick a t1 card in it |
| 23:21.21 | type0 | hold up |
| 23:21.30 | type0 | I can give it the T-1 it needs, but when it gets to the telcom |
| 23:21.35 | type0 | there's only 2 t-1's availabe |
| 23:21.39 | type0 | on their OC-3 microwave |
| 23:22.01 | bulle | garreel: i have no first hand experience with it, so check date when that was written, and search mailingslists |
| 23:22.03 | Sweeper | and...? |
| 23:22.14 | [hC] | mcab: yeah, i dont really know how to increase the logging level, and i believe this file is clobbered every time the phone reboots, i think this may be the log SINCE the last time it crashed and rebooted. |
| 23:22.16 | Sweeper | again, router \o\ |
| 23:23.04 | mcab | [hC]: if you're using FTP the log shouldn't get clobbered; however it could roll over the size limit |
| 23:23.20 | type0 | I cant have multiple routers at the sites |
| 23:23.22 | type0 | because of power |
| 23:23.28 | Sweeper | type0: add more cards.... |
| 23:23.31 | type0 | how much are the rad ethernet tdm? |
| 23:23.45 | Sweeper | and how does one of these things consume less power than a router? |
| 23:23.53 | [hC] | mcab: using FTP yes. It wouldnt have rolled over a size, unless the size limit is stupidly small... ive noticed that it does this all the time |
| 23:23.58 | Sweeper | uh, I think the t1's are like $1200 |
| 23:24.04 | [hC] | mcab: overwriting the file from scratch i mean. |
| 23:24.04 | Sweeper | maybe more, |
| 23:24.08 | Sweeper | I don't recall~ |
| 23:24.24 | type0 | yeah.. me too |
| 23:24.25 | type0 | haha |
| 23:24.34 | mcab | [hC]: could be your ftp server doesn't support APPE... |
| 23:24.38 | Sweeper | but srsly |
| 23:24.40 | Sweeper | if you're doing data |
| 23:24.41 | type0 | the problem with the router is |
| 23:24.44 | Sweeper | you don't want this |
| 23:24.46 | [hC] | mcab: should, Im using proftpd. |
| 23:24.56 | type0 | if im doing data |
| 23:24.56 | Sweeper | or that basestation thing |
| 23:24.59 | type0 | i dont want a tdm over ip? |
| 23:25.02 | [hC] | I'll look though. |
| 23:25.02 | hrmphh | anyone using "tone on hold"? |
| 23:25.19 | mcab | [hC]: I think you might have to configure proftpd to enable APPE, but I'm not sure |
| 23:25.28 | Sweeper | type0: well, you /could/ use it |
| 23:25.35 | Sweeper | I mean, it's transparent T1 |
| 23:25.43 | type0 | thats what I need |
| 23:25.47 | Sweeper | but I'm sure there's got to be about 10 billion better ways |
| 23:26.14 | type0 | well im using these in place of multiplexers |
| 23:27.43 | type0 | cause I dont want 4 T-1's coming down to a coastcom |
| 23:27.48 | type0 | then muxing all the channels into 2 T-1's |
| 23:27.57 | type0 | IF i can avoid spending 60k per mux |
| 23:27.59 | type0 | i'd be happy |
| 23:28.45 | d00gster | anyone here can help me with the sip client problem I have? |
| 23:29.00 | [hC] | mcab: kay, i just added "AllowStoreRestart on" -- maybe that'll take care of it. I guess ill wait for it to bomb again. |
| 23:29.05 | [hC] | mcab: have you run into this before? |
| 23:29.46 | mcab | [hC]: no, but I don't use hints. newer polycom loads are a little more informative when they crash though |
| 23:30.20 | [hC] | mcab: whats the latest stable build now? do you know? |
| 23:30.32 | mcab | 2.1.0 |
| 23:30.46 | mcab | 2.0.3 is the latest 2.0.x load though |
| 23:31.51 | [hC] | mcab: are you running 2.1.0? Im always weary of upgrading to milestone releases for fear of something else breaking |
| 23:33.13 | mcab | [hC]: not with asterisk, no. Not sure what the concencus has been on it with asterisk... |
| 23:33.57 | [hC] | mcab: ah. any new features/major bugfixes that you know of? What are you connecting it to? |
| 23:35.54 | wunderkin | i use 2.1.0... it is just 2.0 with 'fixes'... its nice... |
| 23:36.48 | wunderkin | some crackhead said that there are problems with 1.6.7+ and asterisk.. not sure wtf with .. sla or something? shrug.. i dont use that |
| 23:36.58 | CrashHD | with in transcoding would there be any echo caused? |
| 23:42.12 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 23:44.14 | *** join/#asterisk brussel_ (n=brussel@cpe-24-165-7-252.san.res.rr.com) |
| 23:47.31 | hrmphh | how can you check the syntax of extensions.ael without restarting? |
| 23:58.11 | knarfly | Unable to load config for mysql CDR's: cdr_mysql.conf |
| 23:58.18 | knarfly | how do I avoid this? |