| 00:00.09 | potential1 | exten => 3212501333,1,GoTo(stations,1001) |
| 00:00.10 | potential1 | This call is not going through |
| 00:00.12 | potential1 | I just hear beeping |
| 00:00.18 | potential1 | anyone know what would be wrong with this line? |
| 00:02.14 | The_DoC^ | I am just curious what would the max number of fxs's asterisk supports |
| 00:02.32 | The_DoC^ | or extensions rather |
| 00:02.53 | JT | The_DoC^: in what way? |
| 00:03.39 | burnproof | exten => 3212501333,1,GoTo(stations,1001) Goto([[context|]extension|]priority): |
| 00:03.52 | Moobius | anyone been working with the sip.conf directive "call-limit"? |
| 00:04.17 | *** part/#asterisk nachoguy (i=boster@ivan.dreamhost.com) |
| 00:07.09 | *** join/#asterisk mavior (n=Miranda@88-149-162-157.f5.ngi.it) |
| 00:07.13 | mavior | hello people! |
| 00:09.19 | potential1 | yes? |
| 00:09.22 | potential1 | exten => 3212501333,1,GoTo(stations,1001) |
| 00:09.28 | potential1 | can someone help me figure this thing out? heh |
| 00:09.41 | mavior | is it possible to make a sip call between two different asterisk server ( user1 and user2 are two user regularly connected respectively one to server1 and to server2) knowing servers ip of course ??? |
| 00:09.45 | *** join/#asterisk ManxPower (n=manxpowe@70.sub-75-202-174.myvzw.com) |
| 00:11.22 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 00:11.32 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 00:11.57 | Lurchtoke | what folder usually holds the voicemails that are recieved? |
| 00:12.09 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 00:12.09 | mavior | don't know...something like Dial(user1@server1) ?? |
| 00:12.23 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 00:12.37 | Lurchtoke | are they saved as wav files? |
| 00:13.01 | ManxPower | Lurchtoke: /var/spool/asterisk |
| 00:14.08 | JT | Moobius: is it possible to make a sip call between 2 asterisk servers? yes |
| 00:15.17 | mavior | JT: can you explain how if user1 is logged on server1 by xlite and user2 is a zap channel on server2? |
| 00:15.44 | ManxPower | mavior: the user does not make the call, asteirsk makes the call |
| 00:17.07 | JunK-Y | some1 running * with uclibc? |
| 00:17.10 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 00:17.22 | mavior | yes I mean...what I have to do with my Dial application? I need to define something in my config file to call, say , from user1 to user2, how to use Dial() in this case ? |
| 00:18.05 | BigCanOfTuna | When the command documentation talks about the "asterisk database" are they talking about MySQL or an embedded DB? Specifically, I am looking at DB_EXISTS. |
| 00:19.26 | *** join/#asterisk toresbe (n=toresbe@89.10.27.96) |
| 00:19.28 | toresbe | hey guys! |
| 00:19.38 | toresbe | I'm trying to set up a BBS system (I know, I know...) |
| 00:19.56 | toresbe | and I heard there's a SIP softmodem out there. Do any of you know about that? |
| 00:20.10 | toresbe | I could write my own, I guess, but I *am* lazy.. |
| 00:21.19 | Moobius | toresbe: doesn't your bbs support telnet? |
| 00:21.19 | Nugget | telnet is eeeeeeevil! |
| 00:21.28 | robl^ | gc |
| 00:21.37 | toresbe | Moobius: yes it does, but my 300 baud acoustic coupler doesn't |
| 00:21.56 | toresbe | Moobius: Neither do phone booths... |
| 00:22.20 | J4k3 | toresbe: modem-over-voip requires either the voip provider doing the modem tones for you, or a lot of hassle |
| 00:22.46 | Moobius | but if you're aiming for 300 baud... |
| 00:23.27 | mavior | ManxPower: but you have to use Dial() for intiate the connection right? |
| 00:25.02 | *** join/#asterisk saftsack (n=saftsack@pD9E04C12.dip.t-dialin.net) |
| 00:25.04 | saftsack | how is the service called which does a lookup in a telephone book on every incoming call? |
| 00:26.01 | wltjr | are variables set in one context available in an included one? |
| 00:26.27 | wltjr | like http://rafb.net/p/ueu5Ai52.html |
| 00:26.41 | wltjr | for some reason CHAN is not being passed along with a value to common_out |
| 00:27.06 | k-man_ | what is the difference between a user and a peer? |
| 00:27.50 | toresbe | J4k3: that's the point, it's not modern-day. It's a little plain thing which I can use to toy around in. |
| 00:29.54 | saftsack | http://bugs.digium.com/view.php?id=8919 is every bluetooth handy able to handle those calls? |
| 00:31.34 | *** part/#asterisk oys (n=eoyslebo@84.208.78.84) |
| 00:32.41 | *** join/#asterisk budmang (i=budmang@12-210-54-193.client.mchsi.com) |
| 00:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:33.24 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:33.34 | budmang | hello |
| 00:37.09 | JT | saftsack: bluetooth handy? |
| 00:37.38 | saftsack | yes. a handy with bluetooth function |
| 00:38.16 | saftsack | didnt you read the link? |
| 00:38.55 | dseeb_ | saftsack: so long as the phone supports the HFP spec, it should work |
| 00:38.55 | toresbe | Handy is a German term for a cellphone. |
| 00:39.06 | saftsack | oh, ok sry i didnt say cp |
| 00:39.40 | wltjr | could someone familiar with pri's plz take a look at http://rafb.net/p/xgvfwD84.html |
| 00:39.54 | wltjr | I think I have my span off? not sure if I need to do 1,1 or 1,0 |
| 00:39.58 | JT | "mobile phone" sounds find to me :P |
| 00:40.00 | JT | fine |
| 00:40.07 | dseeb_ | me too |
| 00:40.08 | saftsack | so 4 cheap cellphones will throw those gsm cards away? |
| 00:40.32 | dseeb_ | saftsack: most likely. test with one first though |
| 00:40.36 | toresbe | J4k3: so what happens if I'm willing to go through the hassle? where shoudl I start? |
| 00:40.55 | saftsack | yes i have a motorola here and a macbook |
| 00:41.00 | saftsack | will test it in the evening |
| 00:41.32 | saftsack | because i have to install linux on my macbook |
| 00:42.38 | saftsack | i go to bed now, good night |
| 00:45.08 | budmang | anyone recommend some cheap providers? VOIP? |
| 00:45.22 | budmang | texas/cali servers iax/sip |
| 00:45.43 | rudholm | I like Teliax, but they're not the cheapest |
| 00:45.53 | rudholm | but the cheaper outfits tend to suck (more) |
| 00:46.13 | budmang | I use teliax |
| 00:46.18 | budmang | not a fan of the softcap. |
| 00:46.21 | budmang | i dont mind paying more |
| 00:46.33 | rudholm | teliax is 2c/min for most of the First World |
| 00:46.40 | rudholm | and they seem pretty reliable and quality is good |
| 00:46.40 | budmang | my server is less then 1ms form teliax california beta server |
| 00:46.44 | rudholm | nice |
| 00:47.35 | rudholm | 64 bytes from voip-ca1.teliax.com (207.174.111.12): icmp_seq=1 ttl=54 time=1.29 ms |
| 00:47.40 | rudholm | I'm just over 1ms it seems |
| 00:48.22 | budmang | no other providers |
| 00:48.27 | budmang | i was hoping for some other then teliax :-) |
| 00:48.28 | budmang | lol |
| 00:48.45 | *** join/#asterisk Tebi (n=rantis@gw.aller.fi) |
| 00:49.03 | rudholm | well, there are companies like BroadVoice and voipjet |
| 00:49.15 | rudholm | voipjet is cheap, but they suck |
| 00:49.31 | toresbe | Nobody here knows about Asterisk softmodems? |
| 00:49.52 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 00:50.09 | hardwire | asterisk has softmodems? |
| 00:50.28 | hardwire | rudholm: what chu up tu mang |
| 00:50.51 | *** join/#asterisk terlouw (n=Naquada@proxy.amsterdam.intruder.nl) |
| 00:51.01 | rudholm | hardwire: just working |
| 00:51.38 | terlouw | hello all! i just installed the hudliteserver on trixbox2 but it wont connect... anyone got a clue on where to start looking? |
| 00:52.07 | kgx | if i do something like "exten => 1,2,Dial(SIP/david&SIP/andrew&SIP/lisa,15,rt)", how can i execute an agi as soon as someone picks it (and I need to know which extension picked it up) |
| 00:52.08 | hardwire | rudholm: actually, whats the issue? |
| 00:52.37 | rudholm | hardwire: the issue with voipjet? |
| 00:52.47 | hardwire | softmodems |
| 00:52.47 | rudholm | geez, what's *not* the issue with voipjet? |
| 00:52.49 | rudholm | oh |
| 00:53.01 | rudholm | that's someone else's question |
| 00:54.07 | JT | i think the x100p is pretty much it, for softmodems in asterisk |
| 00:54.15 | budmang | voipjet is a no go then |
| 00:54.32 | JT | kgx: M() dialplan arg probably |
| 00:54.35 | toresbe | hardwire: that's my question |
| 00:54.50 | rudholm | budmang: yeah, I wouldn't recommend voipjet for anything other than experimental use |
| 00:55.00 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 00:55.39 | budmang | lol |
| 00:55.41 | budmang | k |
| 00:55.46 | budmang | i need full action VOIP |
| 00:56.43 | JT | actionman hero voip |
| 00:56.59 | hardwire | toresbe: whats your question? |
| 00:57.09 | hardwire | oh |
| 00:57.10 | hardwire | haha |
| 00:57.11 | hardwire | stupid me |
| 00:57.13 | toresbe | :) |
| 00:57.19 | kgx | JT: thanks |
| 00:57.20 | hardwire | toresbe: what do you need to kno? |
| 00:57.35 | toresbe | 01:46 < hardwire> asterisk has softmodems? |
| 00:57.37 | toresbe | :) |
| 00:57.43 | JT | toresbe: i can't see the real question |
| 00:57.45 | hardwire | toresbe: right.. what do you need to know? |
| 00:57.54 | toresbe | Is there a softmodem for asterisk? |
| 00:58.09 | toresbe | I have searched the fine web, but no information revealeth she |
| 00:58.24 | *** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au) |
| 00:58.44 | JT | toresbe: what do you want, a single phone line connection for voice? |
| 00:59.09 | toresbe | JT: what? I want a... softmodem! |
| 00:59.17 | toresbe | a V.21 softmodem |
| 00:59.34 | JT | what on earth does this have to do with asterisk? |
| 00:59.42 | JT | ask the damn question already |
| 00:59.43 | hardwire | http://www.voip-info.org/wiki/view/Asterisk+Modem+channels |
| 00:59.53 | toresbe | hardwire: thanks! |
| 01:00.00 | hardwire | you said you searched |
| 01:00.02 | JT | v.21 implies data |
| 01:00.04 | hardwire | I thought posting that would be redundant |
| 01:00.30 | toresbe | JT: exactly |
| 01:00.37 | hardwire | Can I use my modem to connect to the PSTN? |
| 01:00.38 | hardwire | <PROTECTED> |
| 01:00.42 | hardwire | form the FAQ |
| 01:00.44 | hardwire | from |
| 01:00.45 | hardwire | :) |
| 01:00.54 | toresbe | hardwire: yes, but that applies to very advanced modems |
| 01:00.54 | *** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no) |
| 01:01.03 | toresbe | oh no wait |
| 01:01.04 | The_DoC^ | has anyone used a linksys rt31p2 router as a sip fxs? |
| 01:01.11 | toresbe | that applies to something entirely different |
| 01:01.24 | toresbe | RoyK!! Long time no see!! |
| 01:01.34 | hardwire | toresbe: if the answer for very advanced modems is NO. what are you expecting from stupid modems? |
| 01:01.51 | toresbe | hardwire: I want a software modem for asterisk - how much simpler can I put it |
| 01:01.59 | JT | much simpler |
| 01:02.09 | JT | what DO YOU WANT TO DO WITH THE SOFT MODEM IN ASTERISK? |
| 01:02.10 | toresbe | I want... to dial into … my Asterisk machine... and hear a V.21 carrier. |
| 01:02.21 | toresbe | JT: Transfer text? |
| 01:02.22 | JT | asterisk is not a dialup server |
| 01:02.27 | JT | it's a telephony server |
| 01:02.32 | RoyK | toresbe: hei |
| 01:02.39 | J4k3 | JT: you could buy an FXS and feed a real modem with it. |
| 01:02.40 | J4k3 | maybe |
| 01:02.49 | J4k3 | kinda |
| 01:02.51 | J4k3 | on a good day. |
| 01:02.52 | hardwire | toresbe: you could use a real modem |
| 01:02.58 | hardwire | and dial into it |
| 01:03.04 | JT | toresbe: what you want is a real modem and normal dialup software |
| 01:03.05 | hardwire | tada |
| 01:03.12 | toresbe | JT: no, I do not. |
| 01:03.17 | hardwire | something tells me his modem sucks |
| 01:03.18 | JT | why not |
| 01:03.18 | toresbe | hardwire: I don't want to spend money on hardware... :\ |
| 01:03.23 | hardwire | exactly |
| 01:03.34 | toresbe | JT: because I want to *call* the machine |
| 01:03.41 | hardwire | toresbe: so you want a software based modem |
| 01:03.44 | hardwire | with no hardware attached |
| 01:03.59 | hardwire | you do realize for a phone call to happen, you are gonna need somewhere to plug a phone line into it |
| 01:04.00 | toresbe | hardwire: Yes! That's exactly what I've been saying! |
| 01:04.05 | hardwire | cause 56k via SIP is no good |
| 01:04.07 | toresbe | *sigh* I have an external SIP number. |
| 01:04.13 | hardwire | yeh? |
| 01:04.20 | hardwire | and you really expect this to work? |
| 01:04.28 | toresbe | Well, yes |
| 01:04.29 | toresbe | 01:55 < toresbe> a V.21 softmodem |
| 01:04.32 | hardwire | I don't want to be rude, sorry. |
| 01:04.33 | toresbe | V.21 is 300 baud |
| 01:04.43 | hardwire | toresbe: more power to you |
| 01:04.55 | hardwire | there is a software modem that some folks here developed |
| 01:04.59 | toresbe | I have literally scotch taped a cellphone to my couplers and had this work |
| 01:05.20 | hardwire | toresbe: what happened to your scotch tape? |
| 01:05.29 | The_DoC^ | hi ...(dead air) ho... (dead air) and so on |
| 01:05.57 | toresbe | hardwire: *sigh* I have a terminal with acoustic couplers which is very lightweight. |
| 01:06.11 | hardwire | I already gave you an answer |
| 01:06.14 | toresbe | hardwire: I want to be able to ring my home SIP number, and get a carrier, and be able to communicate with it |
| 01:06.28 | k-man_ | when i reload my sip.conf file, i get this error: Got 404 not found on sip register to service 2134@mysiprovider.com, giving up |
| 01:07.02 | hardwire | http://www.voip-info.org/wiki-Asterisk+fax |
| 01:07.05 | hardwire | thats a good starting point toresbe |
| 01:07.19 | hardwire | http://www.voip-info.org/wiki-Asterisk+fax#IAXmodem |
| 01:07.19 | The_DoC^ | is there such a thing as a "free" Sip provider? |
| 01:07.31 | hardwire | if he is paying for it, its free |
| 01:07.33 | toresbe | hardwire: yeah, but they all tend to be FAX modems. |
| 01:07.37 | hardwire | I am guessing he has no telephone line |
| 01:07.47 | hardwire | toresbe: IAXModem IAAXmodem |
| 01:07.50 | hardwire | IAXModem |
| 01:07.52 | hardwire | IAXModem |
| 01:07.59 | potential1 | can you recieve faxes through asterisk? |
| 01:08.03 | toresbe | hardwire: ...is a fax modem. |
| 01:08.08 | hardwire | apt-cache search iaxmodem |
| 01:08.08 | hardwire | iaxmodem - software modem with IAX2 connectivity |
| 01:08.11 | toresbe | potential1: if you allocate sufficient bandwidth, you can. |
| 01:08.33 | potential1 | how mch b/w is needed? |
| 01:08.57 | hardwire | 300 baud worth :) |
| 01:08.58 | toresbe | hardwire: hmm, the Debian description makes it sound as if there are more possibilities... |
| 01:09.32 | hardwire | toresbe: you will scotch tape cell phones together to bridge v.21, but you won't randomly install packages and test them? |
| 01:10.03 | toresbe | hardwire: I tend to read docs before I install software on my machine. |
| 01:10.15 | toresbe | And the docs I read made it sound as if it was a fax modem. |
| 01:11.12 | hardwire | I tend to read the source even in long shot situations |
| 01:11.57 | toresbe | Then you, Sir, have far too much time on your hands :) |
| 01:12.13 | Moobius | says they guy setting up a bbs... |
| 01:12.20 | Moobius | the* |
| 01:12.22 | toresbe | Touché :) |
| 01:17.44 | hardwire | AT |
| 01:17.45 | hardwire | OK |
| 01:18.37 | toresbe | hm, indeed |
| 01:18.44 | toresbe | now to actually make it work.. |
| 01:19.21 | hardwire | you connect to it via IAX :) |
| 01:19.44 | hardwire | so just route your sip inbound directly to it |
| 01:20.57 | *** join/#asterisk WAudette70 (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 01:21.35 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 01:21.38 | hardwire | toresbe: you let us know how IAXmodem via SIP works for ya |
| 01:22.05 | hardwire | My guess is unless you are 32+kbps ain't much gonna happen |
| 01:22.36 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-0-0-cust301.bagu.broadband.ntl.com) |
| 01:22.51 | toresbe | hardwire: mine too... |
| 01:23.06 | toresbe | hardwire: thanks for the tips, even if I did have to wring them out of you ;) |
| 01:23.16 | hardwire | wring? |
| 01:23.17 | hardwire | no |
| 01:23.19 | hardwire | I looked it up |
| 01:23.32 | hardwire | not like it was just sitting there all useful like |
| 01:23.48 | hardwire | I dunno, IRC performs well at 300baud, I am guessing google doesn't so much eh :) |
| 01:23.55 | J4k3 | hah |
| 01:24.20 | J4k3 | 7 people all blabbering at once at 100 wpm |
| 01:25.42 | JT | the reason it's hard to do, toresbe, is technology has advanced |
| 01:25.50 | JT | we have this thing called gprs |
| 01:25.54 | JT | requires no scotch tape |
| 01:25.57 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 01:26.11 | Zand3r | Hi all - is there are good resource that compares SIP phones? I'm looking at SNOM, Polycom and Aastara soall should be good quality- I am therefore looking for a comparison specific to their ompatibility / useability with Asterisk. Things like how calls are transfered, SLA, BLA, BLF, etc. (I know Asterisk does not support all of those features but I understand some phones allow sterisk to get close. |
| 01:26.36 | b11d|bbl | if you find that, be sure to let us know about it :) |
| 01:28.44 | Zand3r | ahh - it's like that is it - fair enough :0 - I'll continue gathering my info from all over the place. |
| 01:29.28 | JT | polycom seems to be the channel favourite |
| 01:29.58 | b11d|bbl | yeah, agreed. I love mine.. |
| 01:30.25 | b11d|bbl | Zand3r.. perhaps you should build that resource.. might be a nice contribution eh :) -- would suck to maintain though. Maybe on voip-info.org? |
| 01:31.28 | Zand3r | b11d|bb1 - I'm digging up lots of info so maybe so - although much of it has actually originated from voip-info.org in th efirst place :) |
| 01:32.29 | Zand3r | I'm looking at the Polycom 501 and 301 phones - Do you find their use makes sense in an Asterisk context - i.e their various buttons can be configured to use the appropriate Asterisk control codes? |
| 01:32.31 | b11d|bbl | what sucks is trying to cut through the marketing BS on all the websites.. |
| 01:32.32 | Carp1 | No one from Teliax comes on this channel do they? |
| 01:32.36 | b11d|bbl | the 301, you will regret. |
| 01:32.53 | rudholm | Carp1: not that I know of |
| 01:32.56 | Carp1 | Hmm. |
| 01:33.01 | rudholm | Carp1: why? |
| 01:33.29 | b11d|bbl | the 501.. absoultely.. but im not doing very much that could be considered "advanced" with those phones. |
| 01:33.29 | The_DoC^ | just wondering if the Intel 537 modem works well as a single port fxo? |
| 01:33.29 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:33.29 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:33.37 | Carp1 | I registered with them and got a DID, I worked on getting it to work for like 2 hours then they emailed me and said the DID should not have been in the list because they dont own it. |
| 01:33.41 | b11d|bbl | but with the polycom, you are the most likely to find a solution becuase so many of us here use them and have done many things with them |
| 01:33.52 | Carp1 | Then they credited my account to register a new number |
| 01:33.56 | b11d|bbl | The_DoC^..not without serious work to the modem, i'd think |
| 01:34.04 | Carp1 | so I did and someone else owns that number becuase when I call it, someone answers lol |
| 01:34.10 | The_DoC^ | something about removing resistors |
| 01:34.16 | b11d|bbl | yeah good luck with that :) |
| 01:34.27 | rudholm | Carp1: weird. I didn't have any problems with them when I got a DID from them a few months ago |
| 01:34.47 | Zand3r | b11d|bb1: Thanks - Out of interest, what have you found is wrong with the 301? I was thinking of using higher spec phones for a receptionist type scenario and lower spec phones for the rest of the office - no good? |
| 01:34.57 | rudholm | Carp1: but I realized I didn't really need a DID from them so I submitted a ticket and they removed it in about a day. now I just use them for call completion. |
| 01:35.25 | The_DoC^ | now I need to come up with a new option, probably time to order a x100p |
| 01:35.48 | b11d|bbl | the 301 doesnt have a speaker phone, the display is hard to read and blocky, there is really nothing of note about the 301.. i bought a few but have gone with the 430 in their stead recently. |
| 01:35.54 | Carp1 | Do you think they will get mad if I press 9 for imidiate assistant? |
| 01:35.55 | b11d|bbl | yeah i'd say so The_DoC^.. |
| 01:35.55 | Carp1 | lol |
| 01:35.56 | JT | The_DoC^: do you only have a tiny budget? |
| 01:36.08 | Carp1 | because their office is closed |
| 01:36.16 | b11d|bbl | go with a 601 for your receptionist, and 501's for the offices.. |
| 01:36.17 | The_DoC^ | JT: yes and no |
| 01:36.38 | b11d|bbl | your end user will complain if they have to use a 301 for very long.. |
| 01:36.46 | b11d|bbl | the 430 isnt much more expenseive and has a much better display |
| 01:36.52 | b11d|bbl | improved audio, etc.. |
| 01:37.14 | The_DoC^ | if I can justify spending the money then no otherwise I don't want to spend alot |
| 01:37.38 | b11d|bbl | The_DoC^.. is this for a hobby, or for somethign that would be in actual production? |
| 01:37.55 | JT | The_DoC^: a SIP ATA unit is probably the cheapest, that would actually work properly most of the time |
| 01:37.57 | The_DoC^ | hobby at the moment |
| 01:38.00 | b11d|bbl | ah cool |
| 01:38.18 | J4k3 | grandstream 101 #2 hasn't locked up since I plugged it in... Grandstream 101 #1 hasn't locked up since I got the settings correct on it. Yay |
| 01:38.31 | J4k3 | #2 has been up for about 21 hours |
| 01:38.34 | The_DoC^ | wife is getting pissed cause I keep alling all of the extensions in the house |
| 01:38.40 | b11d|bbl | haha |
| 01:38.47 | Zand3r | b11d|bb1: Thanks - I'll stay away from the 301 - I hadn't seen the 430 - nothing about it in th epolycom documentation i;d seen - I'll go check it out. It's interestesting at least that the issues with the 301 are sound quality and not Asterisk related though. |
| 01:38.49 | JT | call them all at once :) |
| 01:39.07 | The_DoC^ | haven't figured out how to call them all at once |
| 01:39.27 | JT | dial(blah&blah&blah) |
| 01:40.00 | b11d|bbl | yeah you will not have any real issues with Polycom and Asterisk... that im aware of anyway, and I use them both.. |
| 01:40.17 | b11d|bbl | The_Doc, might want to use .call files maybe |
| 01:40.35 | Carp1 | I just talked to the guy who owns the number i registered |
| 01:40.46 | b11d|bbl | haha i keep reading that nick as <crap> |
| 01:41.03 | b11d|bbl | so I read that as: crap, I just talked to the guy who owns the number i registered |
| 01:41.09 | b11d|bbl | it brought a smile to my face :) thanks |
| 01:41.15 | Carp1 | he has a few numbers from teliax and he is porting them to a new carrier and teliax is just putting them back on their list for sale lol |
| 01:41.32 | JT | hah |
| 01:43.51 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:44.39 | The_DoC^ | when I got the bright idea to start playing with asterisk I bought a 4 port fxo and a 16 port fxs made by packetizer for Tundo, but there is no way to configure them without Tundo's software that is no longer available due to the company going belly up |
| 01:45.01 | *** join/#asterisk zpertee (n=chatzill@cpe-65-25-121-5.neo.res.rr.com) |
| 01:45.11 | JT | hah |
| 01:45.15 | JT | no serial or telnet? |
| 01:46.09 | zpertee | Hi all! I'm working on using asterisk for nagios notifications. Has anyone out there ever done this before? |
| 01:46.18 | The_DoC^ | it has both and I can access them, they use vxworks as the os and loads a config file from a tftp server but I have no way of figuring out what the config file consists of |
| 01:46.37 | JT | The_DoC^: it can't send files the other way? |
| 01:47.24 | The_DoC^ | it can but it stores averything in a nvram then when it boots it loads it all into memory |
| 01:47.59 | The_DoC^ | I know by errors it passes that it uses a elf file and cpp configs |
| 01:48.40 | JT | it should be alright if you can send configs the other way |
| 01:48.58 | JT | can you use the telnet interface to configure various options interactively? |
| 01:49.08 | The_DoC^ | I haven't dove to far into the locations they are stored yet |
| 01:49.23 | The_DoC^ | yes, but options are limited |
| 01:49.31 | Carp1 | www.sellvoip.net |
| 01:49.42 | JT | not enough to get it working? |
| 01:49.57 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 01:49.59 | Carp1 | The guy I talked to said he switched here because their service is better and cheaper. |
| 01:50.40 | The_DoC^ | no, you can't set much other than ip, host, routes |
| 01:50.50 | JT | is there a reason you don't want to use sip? |
| 01:51.10 | JT | The_DoC^: weird, so i'm guessing the phones annoying your wife aren't connected to it |
| 01:51.25 | The_DoC^ | no, I am using unlocked pap2's |
| 01:51.40 | DocHolliday | annoying the wife? what a shame. |
| 01:52.28 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 01:53.59 | The_DoC^ | I am good at it |
| 01:55.20 | The_DoC^ | http://pastebin.com/871737 |
| 01:56.56 | The_DoC^ | you can't set the gatekeeper ip in the command line |
| 01:58.17 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 02:04.05 | The_DoC^ | oh well I will figure out that bookend one of these days |
| 02:04.22 | JT | gar use pastebin.ca not .com |
| 02:04.26 | JT | .com ZZZzzzzzZZzz |
| 02:05.08 | The_DoC^ | I noticed it was slow |
| 02:05.58 | JT | The_DoC^: does it do h.323? |
| 02:07.43 | The_DoC^ | yes and supposedly G711u and G711a |
| 02:08.43 | JT | yeah because gatekeeper is a term specific to H.323 |
| 02:08.52 | JT | does it even do sip? |
| 02:10.23 | The_DoC^ | from reading all of the articles out there (not many) it was supposed to be sip or H.323 |
| 02:10.39 | CrashHD | how can I set my asterisk up so that an inbound sip call (using DIAL(SIP/1.1.1.1/${EXTEN}) will be recognized that it is a valid peer and route the call? |
| 02:10.41 | JT | have fun connecting to it if h.323 :P |
| 02:11.04 | The_DoC^ | thats even if I figure it out |
| 02:11.05 | hardwire | h8r |
| 02:12.08 | ManxPower | CrashHD: you would want Dial(SIP/${EXTEN}@1.1.1.1) |
| 02:12.30 | CrashHD | why man? |
| 02:12.39 | ManxPower | And for the most part, I believe Asterisk will accept pretty much any incoming call |
| 02:12.53 | ManxPower | CrashHD: because that a correct Asterisk Dial() like for SIP. |
| 02:12.58 | JT | because that's the correct way to do it |
| 02:13.20 | ManxPower | CrashHD: you may have to set insecure=very in sip.conf [general]. I don't know. Try it without it,. |
| 02:13.26 | CrashHD | but correct sip uri is user:pass@ip/exten |
| 02:13.41 | ManxPower | also whatever the extension is must exist in the context specified in [general] |
| 02:13.50 | ManxPower | CrashHD: Asterisk does not use SIP URIs when dialing. |
| 02:14.24 | CrashHD | well dial(sip/ip/exten) works |
| 02:14.24 | ManxPower | now, if you are dialing from a SIP client, then that Dial line you quoted would not be valid anyway |
| 02:14.38 | ManxPower | CrashHD: you actually type that into your SIP client? |
| 02:14.51 | CrashHD | into my asterisk box |
| 02:15.04 | *** join/#asterisk elriah (n=johnny@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 02:15.09 | ManxPower | Since we don't know what your SIP client requires there is no point in quoting the dial strings you give it as they are useless to us and confuzing |
| 02:15.09 | CrashHD | exten => 1,1,DIAL() |
| 02:15.18 | JT | does 'ip' have an entry in sip.conf? |
| 02:15.21 | CrashHD | these are two asterisk boxes |
| 02:15.23 | elriah | Hi all. Does Cisco ship a 7961 with SIP firmware? i.e., easy asterisk install? |
| 02:15.40 | ManxPower | CrashHD: then use the correct Asterisk SIP Dial syntax to start with |
| 02:16.02 | *** join/#asterisk azidenth (n=aby_azid@60.49.99.207) |
| 02:16.07 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 02:16.13 | ManxPower | elriah: no Cisco phone comes with SIP firmware, you would have to buy it. I don't know if the 7961 has SIP firmware available or not. |
| 02:16.14 | azidenth | good day everyone.. |
| 02:16.40 | CrashHD | well correct is a relative term |
| 02:16.44 | CrashHD | if it works is it not correct? |
| 02:16.45 | azidenth | im having a problem receiving call from sip provider..but i'm able to make call |
| 02:16.47 | elriah | Do you use the Cisco 7960 with asterisk? Anyonw know how it compares to the Polycom offerings? |
| 02:17.04 | azidenth | anyone can help? |
| 02:17.05 | ManxPower | CrashHD: mugging a little old lady is possible, but not correct. |
| 02:17.12 | JT | CrashHD: there are often varying degrees of correctness for a lot of things |
| 02:17.39 | ManxPower | CrashHD: and by doing it incorrectly, who knows what oddities you may encounter that could be avoided by doing it correctly. |
| 02:17.50 | CrashHD | the problem still occurs when the other dial statement is used |
| 02:17.51 | ManxPower | azidenth: what is the error message on the Asterisk console |
| 02:18.11 | ManxPower | CrashHD: what CLI output do you have for a failed call on the destination server? |
| 02:18.25 | azidenth | no error.. |
| 02:18.45 | azidenth | but i can make outbound calls to another sip proxy.. |
| 02:18.53 | ManxPower | azidenth: you may have to enable sip debug in the asterisk cli, but that will generate alot of information to sift thru. |
| 02:19.00 | azidenth | but cant receive/incoming call |
| 02:19.07 | ManxPower | azidenth: inbound and outbound are TOTALLY different |
| 02:19.13 | azidenth | already enable the debug info |
| 02:19.34 | CrashHD | the call hits the system but the originating system is not sending the secret when the call is sent |
| 02:19.35 | ManxPower | azidenth: what does "sip show registry" show for that provider? |
| 02:20.05 | ManxPower | CrashHD: um, you don't use secrets with anonymous calls, and that is what you are really doing. |
| 02:20.06 | CrashHD | because the originating dial string uses the ip directly not the username of the sip entry |
| 02:20.21 | CrashHD | it recognizes the incoming user |
| 02:20.32 | ManxPower | if you want a non-anonymous call then you need to put fromuser=whatever in the dialing server and reference it by that sip.conf entry |
| 02:20.41 | ManxPower | CrashHD: what is your incoming user? |
| 02:20.52 | CrashHD | ManxPower: fromuser= is being used |
| 02:21.13 | ManxPower | CrashHD: don't expect it to work if you are dialing by IP. |
| 02:21.15 | azidenth | connected to sip provide |
| 02:21.17 | azidenth | connected to sip provider |
| 02:21.29 | ManxPower | azidenth: paste the 1 line |
| 02:21.30 | The_DoC^ | JT: maybe if I have time I will throw it online tomorrow and you can telnet into it |
| 02:21.51 | CrashHD | hmmm |
| 02:22.02 | CrashHD | so it won't match an ip to a sip entry on an outbound leg |
| 02:22.07 | CrashHD | interesting |
| 02:22.15 | azidenth | Name/username Host Dyn Nat ACL Port Status Realtime |
| 02:22.15 | azidenth | karim/karim 60.49.99.207 D N 47766 Unmonitored |
| 02:22.15 | azidenth | Aby 60.49.99.207 D N 40533 Unmonitored |
| 02:22.15 | azidenth | Name/username Host Dyn Nat ACL Port Status Realtime |
| 02:22.15 | azidenth | karim/karim 60.49.99.207 D N 47766 Unmonitored |
| 02:22.15 | azidenth | Aby 60.49.99.207 D N 40533 Unmonitored |
| 02:22.29 | azidenth | wait |
| 02:22.40 | ManxPower | azidenth: DO NOT PASTE MORE THAN 1 OR TWO LINES |
| 02:22.42 | azidenth | wellsip/666 203.223.132.200 N 5060 OK (61 ms) |
| 02:22.44 | ManxPower | use pastebin.ca for larger pastes |
| 02:23.17 | ManxPower | azidenth: That is from "sip show registry" not "sip show peers"? |
| 02:23.44 | azidenth | nope its for sip show peer |
| 02:23.56 | azidenth | *from |
| 02:23.58 | ManxPower | azidenth: I did not ask for sip show peer |
| 02:24.05 | azidenth | then? |
| 02:24.17 | ManxPower | I need the line from "sip show registry" |
| 02:24.22 | azidenth | ooh ok |
| 02:25.28 | azidenth | Host Username Refresh State Reg.Time |
| 02:25.29 | azidenth | 203.223.132.200:5060 666 105 Registered Wed, 31 Jan 2007 10:32:46 |
| 02:25.49 | ManxPower | good, now we have confirmed that you do not have a registration problem. |
| 02:26.09 | ManxPower | azidenth: put the sip debug of a failed incoming call on pastebin.ca |
| 02:26.44 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 02:27.02 | azidenth | this maybe kind a stupid question but where to get pastebin.ca :) |
| 02:27.38 | ManxPower | azidenth: web browser |
| 02:27.41 | ManxPower | ~pb |
| 02:27.51 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:29.08 | azidenth | i dont get any sip debug of fail incoming call.. |
| 02:29.24 | azidenth | is there any configuration in extensions.conf i need to check? |
| 02:29.50 | Moobius | what has "extensions reload" changed into in 1.4? |
| 02:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:35.04 | JT | The_DoC^: hah, alright |
| 02:35.23 | JT | The_DoC^: tomorrow night my time might be the only good time tomorrow |
| 02:35.28 | JT | it's 1335 here now |
| 02:36.42 | k-man | when i reload my sip.conf file, i get this error: Got 404 not found on sip register to service 2134@mysiprovider.com, giving up |
| 02:36.52 | k-man | any ideas why it wouldn't be finding my sip provider? |
| 02:36.59 | k-man | i can ping the sip provider |
| 02:37.55 | *** join/#asterisk bkw__ (n=brian@216.48.25.151) |
| 02:38.57 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 02:41.41 | Moobius | is there a way to configure the colors in the asterisk cli? |
| 02:44.28 | ManxPower | k-man: your provider is saying not found |
| 02:44.36 | ManxPower | perhaps you need a password in your register |
| 02:44.45 | k-man | ManxPower, strange.. |
| 02:44.52 | k-man | yes, i have a password in there |
| 02:44.55 | k-man | maybe i mistyped it |
| 02:44.59 | k-man | let me double check |
| 02:45.46 | *** join/#asterisk bitbandit (n=polx@68-116-238-170.dhcp.stgr.ut.charter.com) |
| 02:46.11 | k-man | hmm |
| 02:46.16 | k-man | it's definately correct |
| 02:46.25 | k-man | and my sip phone is able to connec to the provider |
| 02:46.46 | bitbandit | what are the main causes of choppy music on hold ? from my googling i see alot of SATA and P4 HT problems i am not running either |
| 02:47.01 | JT | k-man: pastebin.ca your sip.conf, make sure you remove the secret |
| 02:47.11 | k-man | jt, ok, thanks |
| 02:48.50 | *** join/#asterisk bitbandit (n=polx@68-116-238-170.dhcp.stgr.ut.charter.com) |
| 02:49.02 | bitbandit | howdy |
| 02:49.12 | Moobius | bitbandit: the real answer is that music takes more bandwith than voice and the codecs are optimized for voice. |
| 02:49.55 | bitbandit | ah, even over POTS ? |
| 02:50.08 | JT | what interface? |
| 02:50.14 | k-man | http://pastebin.ca/333753 |
| 02:50.29 | test34 | bitbandit, you could tried to use a compressed music format ? |
| 02:50.33 | bitbandit | uh, i have a x100p clone intel modem |
| 02:51.01 | Moobius | POTS uses dedicated circuits to deliver the sound. Those circuits provide more bandwidth in than voip. |
| 02:51.12 | JT | k-man: it looks like the secrets are there, unless they're bogus? |
| 02:51.19 | Moobius | the issue isn't compression. this is bandwidth in its more raw sense. |
| 02:51.22 | k-man | bogus |
| 02:51.30 | JT | ah good |
| 02:51.34 | Moobius | the range of sound i can make with my voice is smaller than i can hear from music |
| 02:51.35 | k-man | jt, but thanks for checking :) |
| 02:51.50 | test34 | Moobius, compressed music could use less bandwidth |
| 02:51.57 | k-man | spose i could have made them XXXXXXX or something less real looking |
| 02:52.16 | JT | k-man: with verbose set to at least 10, could you pastebin an unsuccessful sip call? may aswell turn on sip debug for the relevant ip, too |
| 02:52.17 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 02:52.38 | bitbandit | Moobius: ah ok, so if i ran a TDM400P do you think that would fix it up ? |
| 02:52.45 | Carp1 | When I get kicked off IRC, and sign back on, it says my username is alreadyin use and I cant switch to it |
| 02:52.45 | k-man | jt, oh... i haven't gotten as far as trying to connect my sip phone to asterisk |
| 02:52.45 | Moobius | no. |
| 02:52.48 | Carp1 | how do I fix that |
| 02:53.01 | JT | k-man: isn't it a connection to sip provider not working |
| 02:53.02 | k-man | jt, i was just trying to get asterisk to connect to nodephone |
| 02:53.08 | k-man | jt, well.. |
| 02:53.17 | k-man | it gives me the error when i type "sip reload" |
| 02:53.28 | JT | ah |
| 02:53.38 | bitbandit | hmm, what shouldi try to boost my bandwidth for the MOH ? |
| 02:53.45 | JT | try removing the /s bit on the register statement |
| 02:53.59 | k-man | Moobius, you could possibly pass the music through a band pass filter to cut out low and high frequencies and hence the music would require less bw? |
| 02:54.09 | JT | Moobius: the POTS network does not have more bandwidth than voip if voip is using g.711 |
| 02:54.10 | Moobius | k-man: correct. |
| 02:54.10 | k-man | jt, i did try, no difference |
| 02:54.17 | Moobius | though the music will sound kinda band. |
| 02:54.17 | JT | however it does have better delivery |
| 02:54.19 | Moobius | bad* |
| 02:54.22 | JT | like no jitter |
| 02:55.12 | JT | k-man: wouldn't make any difference, audio is already band pass filtered at the A/D conversion stage |
| 02:55.12 | test34 | Moobius, phone isnt really high def. anyways |
| 02:55.13 | Moobius | JT: how much voip traffice, do you suppose, is in g.711? |
| 02:55.33 | JT | Moobius: codec 64kbit/s, total including sip overhead, brings it to about 85kbit/s each way |
| 02:55.48 | k-man | jt... unless you filtered it evem more than the A/D conversion's band pass filter |
| 02:56.01 | JT | that would just make it sound worse |
| 02:56.16 | JT | unless the a/d band pass filter is faulty |
| 02:56.37 | JT | i'm sure it's probably an implementation issue that bitbandit is actually having |
| 02:56.42 | JT | the x100p is not a very good card |
| 02:56.48 | ManxPower | g.711u is EXACTLY the same audio the telco uses in the usa. |
| 02:56.54 | Moobius | JT: you can say that again. |
| 02:56.59 | bitbandit | its all i have for the time being |
| 02:57.04 | JT | and g.711a most elsewhere :) |
| 02:57.07 | k-man | jt, yeah, of course it would... but it might make it work better over the codec Moobius is suggesting... maybe? |
| 02:57.15 | J4k3 | the x100p is a half-ass softmodem :P |
| 02:57.16 | bitbandit | i work at a computer repair shop and i came accross it and put it to use |
| 02:57.17 | JT | bitbandit: what does it score in zttest? |
| 02:57.29 | JT | k-man: bitbandit has the problem, with a POTS FXO interface |
| 02:57.32 | JT | no voip |
| 02:57.38 | bitbandit | dunno, never ran it |
| 02:57.45 | k-man | oh... sorry, i missed the first half of the conversation.... |
| 02:57.46 | JT | please run it :) |
| 02:57.54 | ManxPower | JT: IRQ conflicts have been eliminated? |
| 02:58.00 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 02:58.00 | bitbandit | the box is at the shop right now, i wil lahve ot run it tomarrow |
| 02:58.04 | Moobius | does zttest work on x100p's? |
| 02:58.11 | JT | what irq conflicts? |
| 02:58.18 | k-man | jt, so i should get my sip phone to talk to asterisk, then try and make a call and get some debugging info from it |
| 02:58.39 | bitbandit | i dont "think" i have any conflicts because its the only card in the sys but that doesnt really matter |
| 02:58.49 | JT | k-man: well it'd be good to be able to register first :), you might be able to make calls, but not receive them, atm |
| 02:58.55 | bitbandit | snap wrong convo my bad |
| 02:59.03 | k-man | jt, oh.. i see |
| 02:59.17 | k-man | jt, so registering is so you can receive a call? |
| 02:59.46 | ManxPower | bitbandit: cat /proc/interrupts make sure NOTHING is sharing the IRQ the wcfxo is on |
| 02:59.52 | k-man | so how do i set up asterisk so i can connect my sip phone to it? |
| 03:00.03 | JT | k-man: yes, if your ip address is not fixed on the remote server |
| 03:00.18 | k-man | jt, right, i see |
| 03:00.21 | *** join/#asterisk andremi (i=181025c7@gateway/web/cgi-irc/ircatwork.com/x-e8c96985672a35aa) |
| 03:00.30 | k-man | sorry.. its a steep learning curve and I'm right at the bottom of the hill |
| 03:00.51 | andremi | hi |
| 03:01.35 | Moobius | k-man: check out your sip.conf file. |
| 03:01.52 | k-man | check it out? |
| 03:01.58 | Moobius | read it. |
| 03:02.00 | andremi | I've been searching for the last 2 days web with no success trying to understand if I can pass an argument from php (or perl) to the asterisk context |
| 03:02.08 | JT | k-man: actually, i'm still interested in seeing sip debug output while it tries to register |
| 03:02.27 | k-man | Moobius, um... ok |
| 03:02.44 | Moobius | andremi: i had PHP create .call files to jump asterisk to specific places within my dialplan. |
| 03:02.46 | andremi | i.e. I want to pass in str="hello word" and want to execute Flite(str) in asterisk, |
| 03:03.13 | Moobius | andremi: you can CURL() out from asterisk... |
| 03:03.14 | bitbandit | does zttest stop or do i need to stop it ? |
| 03:03.19 | andremi | Thanks mobius, I've seen this technique |
| 03:03.24 | JT | bitbandit: ctrl + c |
| 03:03.30 | bitbandit | ok i do need ot stop it |
| 03:03.38 | JT | yes |
| 03:03.38 | bitbandit | it is scoring 99 - 100 |
| 03:03.43 | JT | sorry |
| 03:03.49 | bitbandit | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.991344 |
| 03:03.50 | JT | you need to be WAAY MORE specific |
| 03:03.56 | bitbandit | hows that |
| 03:03.58 | JT | ok, that's a good score |
| 03:04.21 | JT | it must remain equal to or above 99.97% at all times or you will have problems |
| 03:04.31 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 03:04.50 | bitbandit | think i found teh culpret |
| 03:04.51 | bitbandit | 5: 25001439 XT-PIC VIA8233, ehci_hcd, wcfxo |
| 03:05.07 | JT | ouch, maybe |
| 03:05.13 | bitbandit | looksl ike its sharing |
| 03:05.24 | JT | i'd be interested if the zttest scores drop while you have MoH on |
| 03:05.25 | Moobius | bitbandit: using a butter knife, pop any chips that say VIA off your motherboard. |
| 03:05.32 | Moobius | wait. that might be a bad idea... |
| 03:05.42 | JT | Moobius: i think the x100p especially hates irq sharing |
| 03:05.48 | JT | bitbandit: even |
| 03:05.50 | bitbandit | haha, yeah i think that will make it work worse than it is |
| 03:06.09 | JT | well if you buy non-via to replace it, probably better :P |
| 03:06.31 | Moobius | bitbandit: put the card in a different slot. reboot. pray. |
| 03:06.59 | bitbandit | i might have another board laying around at the shop |
| 03:07.32 | andremi | I was just reading about CURL(), seems promising, but I need still to pass the URL in, is there a way to do this with something like AppData? field of originate ? |
| 03:07.32 | JT | card swapping often fixes issues |
| 03:07.46 | k-man | jt, http://pastebin.ca/333781 |
| 03:08.00 | Moobius | andremi: aside from making an AGI program... |
| 03:08.02 | bitbandit | this is a spare parts machine, mostly a testing ground for me, i will change its slot in the morn and see if that fixes it |
| 03:08.24 | JT | arg, freepbx |
| 03:08.27 | andremi | I am fine with AGI, I have plenty of experience in C/C++, but I am new to asterisk |
| 03:08.38 | Moobius | andremi: you might try using the call files to go to callfileextension,sessionid |
| 03:08.53 | Moobius | then CURLing out from there using the session id to grab the right info |
| 03:09.36 | JT | k-man: there's no sip debug in that |
| 03:09.41 | k-man | jt... |
| 03:09.44 | k-man | hmm |
| 03:09.52 | k-man | i don;t think i know how to turn sip debugging on |
| 03:09.52 | bitbandit | thank you guys, your champs |
| 03:10.11 | JT | 'sip debug' |
| 03:10.15 | JT | or 'sip debug <ip>' |
| 03:10.26 | JT | sip no debug to disable |
| 03:10.57 | *** join/#asterisk wglenncamp (n=wglennca@cblmdm72-240-183-202.buckeyecom.net) |
| 03:11.07 | wglenncamp | Is the digium ftp site down? |
| 03:11.27 | wglenncamp | I ordered a g729 codec, and I can't download it. |
| 03:11.47 | wglenncamp | Any ideas? |
| 03:12.24 | andremi | OK, Moobius, I'll try that -- I just wasn't sure if there's a simple way to pass arguments onto the stack through "Originate" and I'd be doing something unnatural. Thanks! |
| 03:12.37 | Moobius | good luck. |
| 03:12.59 | andremi | bye |
| 03:13.21 | JT | manager interface can originate calls |
| 03:13.23 | JT | not agi |
| 03:13.31 | JT | not directly anyhow |
| 03:15.23 | k-man | jt, http://pastebin.ca/333789 |
| 03:17.43 | k-man | jt, it says method not allowed |
| 03:18.50 | k-man | could it be a nat issue? |
| 03:18.58 | JT | k-man: you should set nat=yes |
| 03:19.02 | k-man | does one need some sort of NAT to receive calls? |
| 03:19.04 | k-man | ok |
| 03:19.07 | JT | and why do you only have a peer definition? |
| 03:19.29 | JT | you need a user definition if you want to receive calls |
| 03:19.31 | k-man | jt, because i thought i would get the peer working, and then add my sip phone? |
| 03:19.43 | JT | a friend definition is a shortcut for peer and user |
| 03:19.46 | k-man | and then i thought i would add that part to receive calls after that |
| 03:19.51 | k-man | one small step at a time |
| 03:20.04 | JT | registering is only for receiving calls, 90% of the time |
| 03:20.23 | k-man | i don't understand what the difference is between a peer and a user |
| 03:20.23 | JT | some ITSPs only allow calls from hosts that have registered, though |
| 03:20.39 | JT | a user logs on to a host remotely and receives calls |
| 03:20.39 | k-man | ok |
| 03:20.45 | JT | a peer sends calls |
| 03:20.51 | JT | a friend does both |
| 03:20.56 | k-man | ok.. so a user could be a sip phone making a call |
| 03:20.58 | k-man | ooh |
| 03:20.58 | k-man | sorry |
| 03:21.05 | k-man | i mean a sip phone receiving a call |
| 03:21.20 | JT | yes exactly, it's acting as a user |
| 03:21.36 | k-man | ok... and a peer is a sip phone making a call? |
| 03:22.52 | k-man | well, i set net=yes and i still got that erro |
| 03:22.53 | k-man | r |
| 03:22.57 | JT | it's a definition in asterisk that refers to the host listed within it as a peer, yes to send calls |
| 03:23.10 | JT | add user or change it to friend |
| 03:23.19 | JT | there is no point registering otherwise |
| 03:23.53 | CrashHD | does a dundi lookup take in to account the cid? |
| 03:24.13 | CrashHD | for instance exten => 1/1,1,NOOP() |
| 03:24.20 | [TK]D-Fender | No, the ONLY reason for registering is when you have a dynamic IP, and need to keep the ITSP up to date with where it should be sending calls to. it serves no other purpose. |
| 03:24.30 | k-man | jt, so change the type=peer in the [nodephone] section to friend? |
| 03:24.36 | JT | yes, that is pretty much what i already said, [TK]D-Fender |
| 03:24.43 | erickperez | Hi there... |
| 03:24.54 | [TK]D-Fender | If you have a fixed IP, then there is no need for registration. It has nothing to do with AUTHENTICATING actual calls. |
| 03:24.59 | JT | also, [TK]D-Fender, some ITSPs do not accept outbound calls from unregistered hosts |
| 03:25.14 | [TK]D-Fender | JT : I just though my version was a little more condensed an inescapable :) |
| 03:25.25 | [TK]D-Fender | JT : Nevere heard of one.... |
| 03:25.36 | k-man | ok, but for the moment, i don't want to receive calls |
| 03:25.45 | JT | they do exist :) |
| 03:25.49 | [TK]D-Fender | JT : I have on the other hand seen a few ITSPS that don't use REGISTER. |
| 03:25.52 | JT | k-man: then try making one |
| 03:25.53 | erickperez | Question: I have a 4FXS in asterisk, connected to an avaya system...extensions 3001,3002,3003,3004. Is there any way i can call other avaya extensions using those 4 fxs? |
| 03:25.56 | k-man | i just want to get asterisk to talk to nodephone and get my sip phone to talk to asterisk |
| 03:26.19 | JT | erickperez: Dial(Zap/1) ?? |
| 03:26.25 | k-man | jt, but how do i set up asterisk so my sip hone can talk to it? |
| 03:26.27 | [TK]D-Fender | erickperez : thats an AVAYA question, not an * one. |
| 03:26.44 | erickperez | well, my asterisk is connected to an avaya, that why i ask here. |
| 03:26.56 | [TK]D-Fender | k-man : what model? pastebin your sip.conf masking only passwords. |
| 03:26.57 | JT | erickperez: ah yes, it is an avaya question :) |
| 03:27.07 | erickperez | the 4fxs is a fxs-to-sip. |
| 03:27.09 | k-man | [TK]D-Fender, i did already |
| 03:27.13 | JT | [TK]D-Fender: http://pastebin.ca/333753 |
| 03:27.24 | k-man | oh, thanks jt |
| 03:27.31 | [TK]D-Fender | erickperez : yes, but your questio is about AVAYA functionality which you should know. |
| 03:27.31 | k-man | [TK]D-Fender, passwords are bogus |
| 03:27.42 | [TK]D-Fender | k, loooking |
| 03:27.53 | k-man | [TK]D-Fender, it is a linksys spa942 (recommended by you ;) |
| 03:27.55 | erickperez | D-Fender, im not the aya guy... :(( |
| 03:28.02 | erickperez | D-Fender, im not the avaya guy... :(( |
| 03:28.16 | JT | k-man: have you tried making a call? you can use a callfile |
| 03:28.26 | JT | if you dont want to setup your phone yet |
| 03:28.34 | k-man | what is a callfile? |
| 03:28.40 | k-man | i had no idea about callfiles |
| 03:28.41 | [TK]D-Fender | k-man : [jason] looks like a perfectly valid interal SIP account you'd auth a phone as... |
| 03:28.58 | [TK]D-Fender | k-man : Forget .call files... lets get the basics down... |
| 03:29.21 | JT | k-man:if you write a correctly formated .call file and move it into /var/spool/asterisk/outgoing/, asterisk will make a call |
| 03:29.26 | JT | heh |
| 03:29.46 | [TK]D-Fender | erickperez : No matter how many time you repeat that question its Avaya's functionality you are asking about. you should already know what you can do with that port. |
| 03:30.14 | [TK]D-Fender | k-man : Is your phone registering as [jason] ? |
| 03:30.31 | k-man | [TK]D-Fender, no, i have not configured my phone yet |
| 03:30.36 | k-man | ill do that now... |
| 03:30.36 | erickperez | D-Fender, well it's my first time asking...if someone else already did that..sorry. thanks anyway. |
| 03:30.56 | [TK]D-Fender | k-man : well * looks ready for it, if that context is the one you were planning on setting it up with. |
| 03:31.05 | k-man | [TK]D-Fender, yes, it was |
| 03:31.17 | JT | erickperez: it's probably rather difficult on analogue, the only easy way i can think of is with outbound callerid if you device supports that, and if the avaya can read the callerid |
| 03:31.32 | [TK]D-Fender | k-man : Ok, well start up your phone and get cracking! |
| 03:31.34 | k-man | so what do i do, just set my sip proxy to the ip of asterisk? |
| 03:31.47 | [TK]D-Fender | k-man : You mean in the phone config? |
| 03:31.51 | JT | callerid works much better on digital too, and you can set the msn |
| 03:31.56 | k-man | yeah |
| 03:31.57 | erickperez | JT: thanks, I'll talk to the avaya guy in the morning. |
| 03:32.11 | JT | actually, you can't set callED number on analogue, so it seems highly unlikely |
| 03:32.16 | JT | unless you did something dicky with dtmf |
| 03:33.27 | [TK]D-Fender | k-man : yeah, sounds about right. |
| 03:33.27 | k-man | oh |
| 03:33.27 | k-man | i think its working |
| 03:33.27 | k-man | ill just try making a call |
| 03:33.27 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:33.27 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:33.30 | JT | k-man: you should ask for help when there's *actually* a problem :P |
| 03:33.37 | [TK]D-Fender | k-man : You only really need to fill in like 4-5 blanks for the phone to be up and running from scratch. they are dead-easy. |
| 03:34.09 | [TK]D-Fender | k-man : For sanity's sake make it an internal test. I suggest an echo test first, followed by a visit to VoiceMail. |
| 03:34.34 | k-man | [TK]D-Fender, ok.. internal it is then |
| 03:34.42 | k-man | how do i do an echo test? |
| 03:34.44 | JT | 1,1,Echo |
| 03:35.03 | JT | don't know if it needs an Answer line first |
| 03:35.17 | JT | some things don't work properly if you don't Answer them first |
| 03:35.30 | JT | so make that |
| 03:35.33 | JT | 1,1,Answer |
| 03:35.36 | JT | 1,n,Echo |
| 03:35.45 | JT | so when you dial the number 1 |
| 03:35.46 | k-man | ok... what do i do with that? |
| 03:36.01 | JT | it should go to it, assuming your linksys's dialplan allows that |
| 03:36.06 | JT | extensions.conf |
| 03:36.08 | k-man | ooh |
| 03:36.09 | k-man | i see |
| 03:36.09 | k-man | right |
| 03:36.11 | JT | in the incoming context |
| 03:36.13 | JT | for the phone |
| 03:37.30 | JT | then you need to do an extensions reload |
| 03:38.01 | k-man | sorry to be dense... |
| 03:38.17 | k-man | i currently have the extensions.conf template asterisk installed |
| 03:38.31 | k-man | should i make some new section in there? or start a new blank file? |
| 03:38.32 | JT | yeah i saw piles of crap in the pastebin you did earlier |
| 03:38.42 | [TK]D-Fender | k-man : TRASH IT. |
| 03:38.44 | k-man | ok |
| 03:38.47 | JT | it'd be ideal to start from blank after renaming the existing one |
| 03:38.47 | k-man | trashed it is |
| 03:39.37 | *** join/#asterisk ManxPower (n=manxpowe@20.sub-70-219-87.myvzw.com) |
| 03:39.52 | k-man | so i can just stick that line in there... under general? |
| 03:39.53 | *** join/#asterisk SethWhit (n=seth@207-224-14-167.clsp.qwest.net) |
| 03:40.57 | CrashHD | anyone recommend a freeware newsreader? |
| 03:41.42 | k-man | thunderbird |
| 03:41.53 | CrashHD | isn't that mozilla? |
| 03:41.59 | JT | k-man: you should have a context for every different type of function/device |
| 03:42.09 | *** part/#asterisk SethWhit (n=seth@207-224-14-167.clsp.qwest.net) |
| 03:42.20 | k-man | oh... ok |
| 03:42.24 | JT | CrashHD: slrn |
| 03:42.51 | JT | k-man: major security hazard otherwise, also inflexible |
| 03:43.03 | [TK]D-Fender | k-man : never make a context named [general] make one like [myphones] for your phones, and in there INCLUDE other contexts like [internal-extensions], [outbound-voip], [outbound-analog], etc as your systems needs dictate |
| 03:43.17 | k-man | ok |
| 03:43.39 | [TK]D-Fender | k-man : and naturally something like [internal-features] would be a good place to hold things like your echo test, direct access to VoiceMailMain, etc... |
| 03:43.50 | [TK]D-Fender | k-man : make it a tiered approach. |
| 03:44.11 | k-man | right |
| 03:45.37 | *** join/#asterisk jesdyn (n=jesdynf@adsl-68-95-163-229.dsl.rcsntx.swbell.net) |
| 03:45.58 | k-man | i think there is still something missign |
| 03:46.16 | JT | what makes you think that |
| 03:46.52 | jesdyn | Are there any good, free Windows softphones out there? |
| 03:47.20 | bitbandit | x-lite |
| 03:47.26 | k-man | jt, well... for one, its not working, and second... well... maybe just the first point |
| 03:47.49 | JT | is the call hitting asterisk? |
| 03:47.50 | k-man | nothing happens when i dial 1,1 |
| 03:47.56 | k-man | nafaict |
| 03:48.03 | jesdyn | bitbandit: Thanks! |
| 03:48.09 | JT | you're only meant to dial 1, but it should work anyway |
| 03:48.27 | JT | i think you need to read up on the book a bit :) |
| 03:48.36 | bitbandit | jesdyn: anytime, bout all i will eb able to helpwith in this chan though hah |
| 03:48.58 | JT | x-lite isn't really good though |
| 03:49.15 | J4k3 | xlite "works" and thats all I can say for it |
| 03:49.27 | JT | single account |
| 03:49.27 | J4k3 | its quality is inferior to my cheap-o hardware-based phones |
| 03:49.27 | [TK]D-Fender | X-Lite means well, but has issues. I'd suggest the Snom soft-phone, or Idefisk |
| 03:49.29 | JT | 2 lines |
| 03:49.31 | JT | no g.729 |
| 03:50.40 | [TK]D-Fender | JT : screw that... no TRANSFER, or CONFERENCE. |
| 03:50.50 | JT | that too |
| 03:50.57 | JT | never used it long enough to notice |
| 03:52.05 | *** join/#asterisk bmg505 (n=leon@c1-237-16.rndf.isadsl.co.za) |
| 03:53.14 | *** join/#asterisk jesdyn (n=root@adsl-68-95-163-229.dsl.rcsntx.swbell.net) |
| 03:53.29 | jesdyn | Sorry to leave so precipitously -- X-Lite snuck a reboot in on me. |
| 03:54.01 | jesdyn | I heard someone say it barely works... is there a better one you could recommend? I've literally never touched this before, and only just now installed an Asterisk card. |
| 03:55.06 | james_ | x-lite works fine |
| 03:55.34 | JT | james_: except for all the deficiencies mentioned above |
| 03:56.06 | jesdyn | Which I was unfortunately absent for. Could someone do me the favor of a paste? |
| 03:56.52 | james_ | JT: which aren't really so much deficiencies as features that dont meet your expectations... last i checked it could make clear phonecalls |
| 03:56.53 | JT | it's poo |
| 03:56.55 | JT | get irssi :) |
| 03:57.04 | james_ | and that's because it's the free lite version |
| 03:57.18 | JT | james_: yes but the question was asking for a GOOD free softphone |
| 03:57.25 | JT | i'm afraid none exist |
| 03:57.34 | JT | even the ones you pay for, none seem that good |
| 03:57.44 | [TK]D-Fender | Like I said, Snom's softphone, or Idefisk. |
| 03:57.48 | james_ | sorry, missed the original question |
| 03:58.05 | [TK]D-Fender | JT : I've got ill to speak for eyeBeam.... |
| 03:58.19 | *** join/#asterisk letomuaddib (n=lesgsod@bas5-montrealak-1128552993.dsl.bell.ca) |
| 03:58.44 | JT | heh |
| 04:10.05 | k-man | what is the extensions.ael file for? |
| 04:10.30 | k-man | and what should i put in there? |
| 04:12.22 | Corydon76-home | It's a new language for writing a dialplan |
| 04:12.26 | JT | you should rename it |
| 04:12.27 | k-man | ooh |
| 04:12.34 | JT | and not use it unless you want to use ael |
| 04:12.39 | k-man | i have extensions.conf and extensions.ael |
| 04:12.52 | k-man | should i remove the .ael file? |
| 04:13.00 | Corydon76-home | Basically it compiles free-form code into the same code as could be loaded as extensions.conf |
| 04:13.05 | JT | rename/remove |
| 04:13.23 | Corydon76-home | If you remove it, you should also noload pbx_ael.so in modules.conf |
| 04:14.08 | [TK]D-Fender | just blank the file but leave it there. |
| 04:14.52 | Corydon76-home | Generally speaking, you should use extensions.conf or extensions.ael but not both |
| 04:19.12 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 04:21.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:22.39 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 04:23.32 | olsen | hi, a friend asked me if i could receive sms from some cellphone, to some web based system, or anything |
| 04:23.37 | olsen | is that possible? |
| 04:23.58 | olsen | though smpp or asterisk |
| 04:24.55 | k-man | i'm sort of feeling like make sample was a bad idea |
| 04:25.08 | Nugget | remove the .ael |
| 04:25.10 | k-man | and i should have started with a blank canvas for the config files |
| 04:25.11 | k-man | i did |
| 04:26.25 | k-man | im testing a dial plan of exten => 1,1,Answer() exten => 1,2,Echo() |
| 04:26.32 | k-man | but nothing happens when i dial 1 |
| 04:27.12 | JT | does the phone have a log, does the call make it to asterisk? |
| 04:28.24 | k-man | jt, i can't see a log in the phone, its a linksys spa 942 |
| 04:28.45 | JT | does anything happen in the console at all when you make the call? |
| 04:29.15 | k-man | no |
| 04:29.19 | k-man | oh.. |
| 04:29.24 | k-man | maybe the phone is blocking it? |
| 04:29.24 | JT | what does the phone do? |
| 04:29.35 | JT | yes it has an internal dialplan |
| 04:29.39 | [TK]D-Fender | k-man : You need to have it in a context that your phone has access to in its definition... |
| 04:29.39 | JT | you should look into it |
| 04:29.55 | k-man | pick up, i hear dial tone, press 1, then silence..... |
| 04:30.12 | [TK]D-Fender | k-man : Do youSEE * answering the call? |
| 04:30.14 | k-man | then it gives me a sort of could not connect tone after about 10 seconds |
| 04:30.17 | k-man | no |
| 04:30.36 | k-man | [TK]D-Fender, i don't understand that bit about the context |
| 04:30.39 | k-man | oh |
| 04:30.47 | k-man | ooh |
| 04:30.48 | k-man | i see |
| 04:30.52 | k-man | hang on |
| 04:31.01 | JT | ~thebook |
| 04:31.24 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:31.25 | k-man | yes yes |
| 04:31.25 | k-man | i have been reading the book |
| 04:31.25 | k-man | but the book is not that helpful |
| 04:31.25 | k-man | its way to verbose |
| 04:31.30 | JT | as if, it explains all this stuff |
| 04:31.34 | JT | just read the necessary bits |
| 04:31.39 | k-man | well... |
| 04:31.42 | JT | stuff on the dialplan |
| 04:31.43 | k-man | tell me which bits those are |
| 04:31.44 | JT | especially |
| 04:32.28 | JT | page 67 (pdf pg 85) |
| 04:32.51 | JT | pg 77 moreso |
| 04:33.14 | JT | the whole section starting at 77 :) |
| 04:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:34.08 | rudholm | that book is a decent primer, but I agree it has some issues. |
| 04:34.48 | k-man | i found it lacking in a sort of "do these simple steps to get started" kind of wya |
| 04:34.49 | k-man | wya |
| 04:34.50 | k-man | way |
| 04:35.03 | rudholm | yeah, and it presents stuff in a sub-optimal order |
| 04:35.08 | k-man | i like a bit by bit approach and it felt like a configure it all approach |
| 04:35.31 | rudholm | it does start out with a simple dialplan example, iirc |
| 04:35.38 | k-man | yeah |
| 04:35.39 | k-man | i think so |
| 04:35.58 | JT | asterisk is not simple |
| 04:36.09 | [TK]D-Fender | k-man : there is always another recourse.... |
| 04:36.13 | JT | i think the book only covers a small proportion of all commands and options available |
| 04:36.13 | [TK]D-Fender | ~osmosis |
| 04:36.18 | JT | well that maybe an exageration |
| 04:36.25 | JT | but it seems like it at times |
| 04:36.34 | JT | i wish the book was twice as big |
| 04:36.35 | k-man | also, the book is 1.2 and im using 1.4... so some things are different |
| 04:36.38 | rudholm | yeah, and it's out of date |
| 04:36.41 | JT | because i usually use it as a reference |
| 04:36.44 | azidenth | ManxPower: can help with incoming sip call setting? |
| 04:36.46 | [TK]D-Fender | ~osmosis |
| 04:36.48 | jbot | [osmosis] the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 04:36.59 | JT | i don't see a point in using 1.4.x in production yet, without a specific reason to do so |
| 04:37.29 | [TK]D-Fender | JT : print it yourself using 120lb card stock :) |
| 04:37.30 | Damin | JT: VLDTMF |
| 04:37.46 | Damin | JT: That's the major reason I've got it in pre-production now.. |
| 04:37.50 | Nugget | I run 1.4.x in production because file said he'd beat me up if I didn't. |
| 04:37.53 | JT | [TK]D-Fender: more pages, not heavier pages :) |
| 04:37.53 | ManxPower | JT me neither |
| 04:38.02 | rudholm | and it massacres quantization theory |
| 04:38.17 | Damin | JT: As well as the increase in SIP call setups / second 1.4 can handle before shitting itself.. |
| 04:38.24 | rudholm | what is it, Chapter 7? althought it's not really important for * |
| 04:38.28 | JT | i didn't think the bit on quantisation theory was that bad, yes they got some frequencies wrong |
| 04:38.32 | JT | but still |
| 04:38.40 | rudholm | no, it sucked |
| 04:38.47 | [TK]D-Fender | JT : Oh sure.. NOW you get picky! |
| 04:38.49 | ManxPower | What's the frequency, Kenneth? |
| 04:39.00 | JT | 300-3400Hz :) |
| 04:39.11 | JT | it may vary a little |
| 04:39.18 | JT | but i thought 300-3400 was the standard |
| 04:39.22 | J4k3 | 700-750 hz, 130+ dB |
| 04:39.34 | olsen | can i use asterisk as a sms gateway? |
| 04:39.36 | JT | heh |
| 04:39.45 | k-man | so what should i change my phones dialplan to? |
| 04:39.56 | rudholm | they show these graphs of analog audio that is reconstructed from a digital data stream that has angles in it. |
| 04:40.35 | JT | rudholm: true |
| 04:40.46 | JT | so it's the graphs you mainly have issue with? |
| 04:40.47 | rudholm | completely missed the point that angles = infinite frequency |
| 04:40.56 | rudholm | yeah, that was the main thing that bothered me |
| 04:40.59 | rudholm | as I recall |
| 04:41.03 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 04:41.36 | rudholm | that's pretty basic Nyquist, and anyone publishing a primer on digital audio should understand the basics. |
| 04:41.42 | BigCanOfTuna | I've created my own GSM file for a greeting, but I don't see how to set it for use with VoiceMail...is it possible to configure it to use specific sound files? |
| 04:41.51 | JT | only a couple of the graphs show them with straight line bits |
| 04:41.55 | JT | just for comprehension |
| 04:42.08 | JT | and it is sort of correct, if the D/A conversion had no filtering or smoothing |
| 04:42.20 | JT | obviously they couldn't *actually* have infinite bandwidth |
| 04:42.23 | [TK]D-Fender | BigCanOfTuna : You should be using VoiceMailMain to do your prompt recordings. |
| 04:42.23 | rudholm | yeah, they should have shown what the waveform actually looks like, since clearly, you can't have infinite frequency components in an output signal |
| 04:42.39 | [TK]D-Fender | BigCanOfTuna : All of that functionality is built-in |
| 04:42.39 | JT | dude there's at least 5 graphs there |
| 04:42.45 | JT | some are completely smooth |
| 04:42.48 | BigCanOfTuna | [TK]D-Fender: Yea, but I want to use more creative sound files that I created externally. |
| 04:43.05 | rudholm | I think the smooth ones are the input or "pre-digitization" graphs |
| 04:44.05 | rudholm | then they explain the benefit of increased quantization resolution and sample rate as, near as I can tell, "you have a jagged output waveform that more closely approximates the input waveform" |
| 04:44.48 | JT | which is correct, to put it simply |
| 04:44.52 | rudholm | no, it's not |
| 04:45.18 | rudholm | there are no angles in the output wave |
| 04:45.28 | rudholm | and that's not a function of quantization resolution or sample rate |
| 04:45.54 | rudholm | the only thing on the output side are (a finite number of ) sine waves |
| 04:45.54 | JT | an output from D/A will always been an approximation |
| 04:46.02 | JT | some approximations are btter than others |
| 04:46.29 | rudholm | always be an approximation of an input signal of infinite BW and S:N ratio, yes. |
| 04:46.44 | JT | yes |
| 04:46.56 | olsen | can i use asterisk as a sms gateway? |
| 04:46.56 | rudholm | but given finite BW and S:N ratios on the input side, it is possible to capture 100% of the information via quantization. |
| 04:47.24 | rudholm | there is no wave in the real world that isn't a set of sine waves. |
| 04:47.37 | JT | do you think phone networks capture 100% of the information? |
| 04:47.38 | rudholm | (a finite set) |
| 04:47.46 | JT | output filtering is often what makes them nice and smooth |
| 04:49.19 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 04:49.56 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 04:51.30 | [TK]D-Fender | BigCanOfTuna : look under /var/spool/asterisk/voicemail for the recording files that * makes and substitute it with your then. |
| 04:51.46 | BigCanOfTuna | [TK]D-Fender: thanks! |
| 04:53.49 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 04:55.12 | rudholm | JT: filtering your output channel to match the spectral contraints of your input signal is integral to the A-D-A process |
| 04:55.25 | JT | of course |
| 04:55.36 | rudholm | if you take that jagged output curve and filter out anything > 4kHz, you get no angles |
| 04:56.05 | JT | that's right, but there *is* a fairly jagged curve during one or two stages of the process |
| 04:56.15 | rudholm | yeah, and there's PCM in there too |
| 04:56.22 | rudholm | which is not relevant |
| 04:56.47 | rudholm | since the point of the chapter was (obvsiouly) not a treatise on Nyquist and Fourier |
| 04:57.20 | JT | heh |
| 04:58.21 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 04:59.19 | rudholm | there are better ways of illustrating the limitations of quantization |
| 04:59.45 | JT | sure, maybe you should speak to them about it, they're writing the second edition :) |
| 04:59.54 | rudholm | you could show an input signal with a 2kHz tone and a 5kHz tone, and then show the output signal would be just the 2kHz tone (sans jaggies) |
| 05:00.02 | rudholm | yeah, I may |
| 05:00.06 | k-man | [TK]D-Fender, so what dial plan do you put in your sip phones so they can talk to asterisk? |
| 05:00.27 | rudholm | I was talking to Strom about contributing to a new Asterisk book. |
| 05:00.59 | rudholm | you could even relatively easily show the effects of quantization noise |
| 05:01.13 | rudholm | showing how low-level signals suffer from more of it |
| 05:01.15 | JT | rudholm: blitzrage is one of the authors |
| 05:01.50 | rudholm | (which was the point behind the nonlinear µlaw and alaw digitization schemes) |
| 05:02.36 | rudholm | but like I said, it was sort of an "extra" chapter anyway |
| 05:02.41 | rudholm | nobody needs to understand quantization to use asterisk |
| 05:02.43 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 05:02.48 | rudholm | so it's not that big a deal |
| 05:03.00 | JT | only if they want to use it well :) |
| 05:03.07 | rudholm | it just happens to be part of my field, so I'm sensitive to errors :) |
| 05:03.52 | JT | you can use asterisk on so many different levels of knowledge |
| 05:04.10 | rudholm | true |
| 05:04.35 | JT | (with different levels of difficulty) |
| 05:04.59 | rudholm | I've only scratched the surface |
| 05:05.32 | rudholm | but there are information resources online |
| 05:05.46 | rudholm | and when that fails, I bug Strom :) |
| 05:06.47 | JT | yeah |
| 05:07.02 | *** join/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net) |
| 05:07.15 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 05:07.58 | Hmmhesays | anyone deal with webserver auth before? |
| 05:08.07 | Hmmhesays | how does a client send its auth information |
| 05:08.17 | rudholm | you mean like http-access-authorization? |
| 05:08.28 | Hmmhesays | yeah |
| 05:08.31 | JT | http digest? |
| 05:08.33 | danp | hmm, i have a polycom 501 that won't ack its offered DHCP lease when it's trying to run sip.ld |
| 05:08.41 | rudholm | the credentials are in the header |
| 05:08.48 | Hmmhesays | ahh ok |
| 05:09.09 | Hmmhesays | so on the client side it keeps sending those credentials in the header while the session is open |
| 05:09.22 | rudholm | yep |
| 05:09.27 | rudholm | over and over |
| 05:09.29 | Hmmhesays | ok that makes sense |
| 05:09.45 | Hmmhesays | i've got an app here that verifies age and i'm trying to figure out how to get it to log in a user |
| 05:10.16 | JT | http digest auth or cookies are the main ways |
| 05:10.23 | Carp1 | I just watched all the systm videos lol |
| 05:10.47 | Hmmhesays | i need my activex control to send the credentials when the webserver asks for them |
| 05:10.51 | Hmmhesays | that is the tricky part |
| 05:13.32 | Hmmhesays | maybe I can have the activex control write a cookie browser loads |
| 05:15.26 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 05:15.44 | CrazyTux | Does asterisk have a playback for 'number currently disconnected' or phone disconnected, or? |
| 05:15.58 | Hmmhesays | look at the sound files on the wiki |
| 05:19.48 | CrazyTux | Hmmhesays, 'Replacement Sound Files' ? |
| 05:20.19 | JT | look at the sound files on the filesystem already |
| 05:20.59 | Hmmhesays | sound files additional |
| 05:21.46 | Hmmhesays | aka asterisk extra sounds on the ftp |
| 05:30.13 | *** join/#asterisk chirino_mbp (n=chirino_@2416447hfc86.tampabay.res.rr.com) |
| 05:31.21 | k-man | I can't work out why my sip phone does not appear to be able to dial asterisk |
| 05:31.37 | k-man | i mean, when i dial 1, the call never makes it to asterisk afaict |
| 05:31.46 | JT | have you checked its dialplan? |
| 05:31.54 | k-man | jt, i have looked at it |
| 05:32.03 | JT | what's in it? |
| 05:32.08 | k-man | but i'm not sure what I should be trying to acheive with it |
| 05:32.15 | k-man | its this great big long thing |
| 05:32.17 | k-man | that came with the phone |
| 05:32.21 | JT | hrm |
| 05:32.28 | k-man | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 05:32.32 | k-man | i also tried (x) |
| 05:32.35 | k-man | and that didn't work |
| 05:32.55 | k-man | maybe it should be (1) |
| 05:32.55 | *** join/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 05:33.01 | rudholm | that looks like a Sipura |
| 05:33.06 | k-man | as all i want to do is dial 1 |
| 05:33.07 | JT | i wonder if it has timeout dialling too |
| 05:33.07 | [TK]D-Fender | k-man : (*x.|x.|#.) |
| 05:33.11 | k-man | yeah, spa942 |
| 05:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:33.23 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:33.30 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 05:33.30 | rudholm | but yeah, that dialplan won't sent a 1 |
| 05:33.45 | rudholm | add the expressions that [TK]D-Fender listed |
| 05:33.51 | k-man | ok |
| 05:33.51 | k-man | thanks |
| 05:34.18 | [TK]D-Fender | The proble with "smart" phones is just how STUPID they can be... |
| 05:35.21 | joelsolanki | anybody knows unlimited usa/canada plan |
| 05:35.56 | k-man | no, that didn't work |
| 05:36.03 | k-man | <PROTECTED> |
| 05:36.09 | k-man | and if i wait, it does the same thing |
| 05:36.20 | k-man | and asterisk does not see the call |
| 05:36.59 | [TK]D-Fender | k-man : Now comes the time to distruct your configs. pastebin sip.conf and extensions.conf |
| 05:37.03 | [TK]D-Fender | :) |
| 05:37.30 | [TK]D-Fender | distrust* |
| 05:37.31 | k-man | ok |
| 05:37.31 | rudholm | you might want to file off the passwords, if any |
| 05:37.32 | k-man | yeah |
| 05:37.32 | k-man | thanks |
| 05:39.36 | k-man | http://pastebin.ca/index.php |
| 05:39.42 | litage_ | when i check my voicemail, why does the generated CDR have the "dst" field set to "s" (as opposed to, say, the # i dialled for voicemail)? |
| 05:39.42 | k-man | oops |
| 05:39.44 | k-man | hehe |
| 05:39.48 | JT | nah i think it's the linksys config |
| 05:39.50 | k-man | http://pastebin.ca/333905 |
| 05:40.01 | JT | [TK]D-Fender: you sure those patterns will match a single digit? |
| 05:41.46 | [TK]D-Fender | k-man : [myphones is actually EMPTY. therese the problem |
| 05:41.54 | k-man | ooh |
| 05:41.59 | k-man | i thought they cascade |
| 05:42.16 | [TK]D-Fender | k-man : what you should do is add "include => internal-features" in there |
| 05:42.23 | *** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
| 05:42.24 | [TK]D-Fender | k-man : You missed my BIG PRINT :) |
| 05:42.27 | k-man | in myphones |
| 05:42.30 | k-man | i did? |
| 05:42.37 | [TK]D-Fender | k-man : You TELL it to include others in useful combinations. |
| 05:42.44 | k-man | ok, thanks... |
| 05:42.46 | [TK]D-Fender | k-man : Indeed you did. |
| 05:42.52 | k-man | sorry |
| 05:44.09 | k-man | god damn! it worked! |
| 05:44.15 | k-man | thanks [TK]D-Fender |
| 05:44.22 | k-man | and thanks jt |
| 05:44.25 | k-man | for all your help |
| 05:44.39 | putzz | anyone recomment a place to buy cheap sip phones? |
| 05:44.47 | JT | nice |
| 05:47.25 | [TK]D-Fender | k-man : Quite welcome |
| 05:47.39 | [TK]D-Fender | putzz : Depends where you are and what you're expecting. |
| 05:47.58 | JT | k-man: for future reference, enabling sip debug is a good way to see if anything is hitting asterisk or not |
| 05:48.22 | k-man | jt, it was enabled |
| 05:48.25 | k-man | oh |
| 05:48.26 | k-man | no |
| 05:48.28 | k-man | maybe |
| 05:48.38 | k-man | i can't remember if i restarted asterisk since i enabled it |
| 05:48.41 | JT | when you made the call, a bunch of fat messages would've scrolled |
| 05:48.42 | k-man | but thanks for the tip |
| 05:48.55 | JT | err it's an interactive command |
| 05:48.59 | JT | sip debug |
| 05:49.16 | k-man | yes, i was in that mode before |
| 05:49.24 | k-man | but i think i restarted asterisk since then |
| 05:49.28 | JT | right |
| 05:49.31 | *** join/#asterisk AJaymn (n=me@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 05:49.33 | JT | that'd do it |
| 05:49.37 | k-man | i have a headache now |
| 05:49.43 | k-man | all this concentrating... |
| 05:49.52 | JT | at least you can hear yourself with the headache |
| 05:49.56 | JT | Echo |
| 05:51.52 | k-man | jt, yeah |
| 05:52.09 | k-man | are you in australia? |
| 05:52.40 | JT | yes |
| 05:53.02 | AJaymn | Is there a way to get Full Duplex in the conference room? |
| 05:53.17 | JT | it should be fdx |
| 05:53.58 | AJaymn | ? |
| 05:54.47 | JT | fullduplex, it should already be |
| 05:56.57 | AJaymn | its not :( you cant talk over someone else.. |
| 05:58.23 | k-man | [TK]D-Fender, you'll be pleased to know i am printing out chapter five of The Book |
| 06:00.43 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 06:01.06 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:02.18 | litage_ | how do you prevent messages like this from being printed to the asterisk console?: DEBUG[7094]: rtp.c:1299 ast_rtp_raw_write: RTP Transmission error of packet 20576 to 202.168.41.214:8976: Operation not permitted |
| 06:02.40 | JT | switch off debug |
| 06:02.42 | litage_ | i tried "debug level 0" and "sip no debug" but the messages still occur |
| 06:02.56 | JT | it's not sip debug, sip debug looks totally different |
| 06:03.17 | litage_ | JT: how do you turn off debug if "debug level 0" doesn't do it? |
| 06:03.31 | JT | that should've done it |
| 06:03.42 | JT | maybe the message gets raised because it's an error? |
| 06:03.50 | litage_ | JT: anything else i can try? |
| 06:04.42 | JT | not sure |
| 06:07.50 | litage_ | JT: i removed "debug" from the "console" line in logger.conf, ran "logger reload", and that got rid of the debug messages |
| 06:08.00 | litage_ | very strange that "debug level 0" didn't help though |
| 06:10.30 | Bobthehunter | theres no asterisk.conf ? |
| 06:17.46 | litage_ | Bobthehunter: i have one... |
| 06:24.12 | danp | weird, i have two polycoms here that seem to be having a lot of trouble doing DHCP via this netgear wireless bridge |
| 06:24.22 | danp | but i have a grandstream and a pap2 that are fine |
| 06:25.27 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 06:26.12 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 06:26.15 | Bobthehunter | can you pastebin it ? |
| 06:26.30 | Bobthehunter | and can i change an application name ? so its like FGI instead of agi? |
| 06:31.32 | Bobthehunter | ? |
| 06:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:36.01 | *** join/#asterisk dongc (n=dongc@203.117.206.249) |
| 06:41.48 | *** join/#asterisk shinux__ (n=shinux@196.220.24.248) |
| 06:42.36 | danp | no problems when i manually give them IPs, though...very weird |
| 06:42.50 | k-man | does this look right? exten => 2,1,Dial(SIP/nodephone,10,r,0871271201) |
| 06:43.14 | k-man | if i dial 2, it will dial that number on nodephone? |
| 06:43.49 | Bobthehunter | wats nodefone |
| 06:43.50 | Bobthehunter | lol |
| 06:44.03 | Bobthehunter | i got a bi-symetric openpbx-asterisk running |
| 06:44.03 | JT | k-man: no |
| 06:44.11 | Bobthehunter | stop now , start other |
| 06:44.16 | Bobthehunter | all runs |
| 06:44.21 | Bobthehunter | same configs |
| 06:44.31 | k-man | jt, its not clear in the doco how you specify a SIP extension, at least not clear to me |
| 06:44.31 | JT | SIP/nodephone/0871271201 |
| 06:44.33 | Bobthehunter | a few mods using table views to replace ogi with agi and macrop and proc |
| 06:44.35 | Bobthehunter | ;) |
| 06:44.36 | JT | oh and the options too |
| 06:44.39 | JT | if you want them |
| 06:45.15 | k-man | jt, the 08 number is the number i want asterisk to call when i dial 2 |
| 06:45.34 | JT | i realise |
| 06:45.58 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 06:46.05 | JT | <TECHNOLOGY>/<ENTRY/CHANNEL>/<CALLEDNUMBER> |
| 06:46.30 | k-man | oh, i see |
| 06:47.41 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 06:47.41 | k-man | jt, i get this error [Jan 31 17:46:08] WARNING[6588]: chan_sip.c:2821 sip_call: No audio format found to offer. |
| 06:47.58 | k-man | when i try and make a call |
| 06:48.17 | JT | err |
| 06:48.47 | JT | what codecs are enabled on the linksys, and in sip.conf, for both the phone and the link to nodefone? |
| 06:50.41 | k-man | linksys = all of them... prefered = G729a, nodephone = g729, and for the phone nothing is set |
| 06:51.17 | JT | just allow all |
| 06:51.29 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:51.29 | JT | see if it works then |
| 06:52.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:56.05 | k-man | didn't work |
| 06:56.10 | k-man | but i have to go now |
| 06:56.14 | JT | ok |
| 06:56.18 | k-man | thanks |
| 06:56.19 | k-man | seeya |
| 07:02.03 | Bobthehunter | Digium has announced a big move in its executive leadership as founder and VoIP technology leading light, Mark Spencer, shifts over to CTO to make room for Danny J. Windham, former President, COO and Director of ADTRAN. |
| 07:02.47 | JT | to be president? |
| 07:03.05 | J4k3 | digium looking to go public, or get a large quantity of money otherwise? |
| 07:03.31 | Bobthehunter | they already did.. that why they hmm ok |
| 07:03.44 | Bobthehunter | http://www.voip-news.com/feature/digium-asterisk-shuffle-adtran-013007/ |
| 07:04.46 | Bobthehunter | once you get risk money involved you sell your soul |
| 07:04.53 | J4k3 | yep |
| 07:04.56 | Bobthehunter | hes getting jerked down to cto |
| 07:05.02 | J4k3 | make the investors warm and fuzzy |
| 07:05.08 | Bobthehunter | in next 6 months youll see |
| 07:05.11 | Bobthehunter | janitor |
| 07:05.17 | Bobthehunter | j/k |
| 07:05.42 | Bobthehunter | man bill gates is doing a tour for vista himself.. he got same things happened to him on msoft |
| 07:06.06 | Bobthehunter | loosing control on copanies is bad |
| 07:06.23 | Bobthehunter | Second, Digium needs to take its sales, marketing and general approach to business to another level if it is going to go up against industry giants like Avaya, Toshiba, Mitel, Nortel and especially Cisco. |
| 07:06.46 | Bobthehunter | so i guess soon they will sell the drivers for digium ;) this wont interfere with GPL but will get eveyrone to pay |
| 07:07.04 | rudholm | trying to hold control too tightly is also bad |
| 07:07.18 | rudholm | to grow, companies must bring in outside capital and outside expertise |
| 07:07.35 | Bobthehunter | yes but controlling a company |
| 07:07.41 | Bobthehunter | money shoudnt controla company |
| 07:07.44 | J4k3 | rudholm: boardroom guys don't really do that. |
| 07:07.49 | J4k3 | they make investors warm and cozy |
| 07:07.56 | Bobthehunter | when you coo or ceo you on decision making operation..every day.. |
| 07:08.25 | rudholm | yep, I've served on boards and been an officer of a C corp before. |
| 07:08.26 | Bobthehunter | when boards start moving staff like that its a restructure..meaning lots of change |
| 07:08.32 | rudholm | I know what the responsibilities are |
| 07:08.33 | Bobthehunter | and since the investores are doing that.. well |
| 07:08.42 | Bobthehunter | its gonna be changes to get more cash in theyr pockets |
| 07:09.07 | Bobthehunter | maybe pushing mark on the side |
| 07:09.10 | Bobthehunter | maybe not |
| 07:09.11 | JT | ounds like what bill gates would always have liked to have done, touring himself |
| 07:09.11 | Bobthehunter | well see |
| 07:09.19 | JT | sounds |
| 07:09.20 | rudholm | but it's not necessarily a bad thing for the company |
| 07:09.22 | Bobthehunter | yah ill never know |
| 07:09.22 | james_ | Bobthehunter: how many companies have you controlled? |
| 07:09.51 | JT | bill gates still eats at mcdonalds and flies economy |
| 07:10.02 | Bobthehunter | 3-4 |
| 07:10.08 | J4k3 | bill gates is silly |
| 07:10.13 | Bobthehunter | first one i was 21 i netted 3.4 mill a year |
| 07:10.17 | J4k3 | he's a total waste of money |
| 07:10.25 | Bobthehunter | 5 emplyees |
| 07:10.28 | Bobthehunter | 1 partner |
| 07:10.28 | J4k3 | he'd be perfectly content on $250k/year |
| 07:10.29 | james_ | nice |
| 07:10.31 | Bobthehunter | ;') |
| 07:10.32 | J4k3 | the rest should be sent to me |
| 07:10.46 | JT | J4k3: not for his house he wouldn't |
| 07:10.52 | Bobthehunter | and went it went downhill it was simply moving principles for cash |
| 07:10.59 | J4k3 | JT: his house is a microsoft technology demo. |
| 07:11.00 | Bobthehunter | that kills a co's clear vision |
| 07:11.12 | J4k3 | JT: I believe technically, he rents a few rooms for himself, from microosft. |
| 07:11.15 | Bobthehunter | money is not eveything in life.. especialy around an open source project |
| 07:11.26 | JT | yeah it took over 10 years to build |
| 07:12.06 | J4k3 | money is everything, and theres lots of money in open source |
| 07:12.25 | J4k3 | its just not made in the same way as the typical commercial software market |
| 07:12.47 | J4k3 | and its not really all that corporate-friendly, unless the corporation is at least slightly knowledgeable about how to make it work for them (digium is a good example) |
| 07:13.18 | james_ | JT: are you trying to make out that his 30 car garage is humble or something? |
| 07:13.25 | J4k3 | they've got the whole world toubleshooting their code, people who would *never* be in the market for their standard commercial pbx product. |
| 07:13.49 | james_ | oops |
| 07:13.50 | JT | J4k3: i'm not |
| 07:13.53 | james_ | J4k3 rather |
| 07:13.57 | JT | james_: even |
| 07:14.01 | james_ | haha |
| 07:14.05 | JT | get some different letters, guys! |
| 07:14.08 | J4k3 | haha |
| 07:14.51 | J4k3 | so digium loses nothing, or very little |
| 07:14.58 | J4k3 | and gains a worldwide programming department. |
| 07:15.17 | JT | digium sell hardware, so their open source equation is not as hard as some |
| 07:17.16 | J4k3 | I bet digium ends up selling more support than they do hardware |
| 07:17.31 | JT | doubt it |
| 07:18.49 | J4k3 | hard to say, you get these people who have been willing to pay thousands to lease some doorstop of a pbx for the last 10 years.... |
| 07:19.08 | J4k3 | a few hundred dollars/year for good vendor support is nothing |
| 07:21.00 | drray | my boss refuses to lease equipment, he'd rather buy it and junk it |
| 07:21.04 | drray | as CE |
| 07:22.48 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
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| 07:39.16 | CrazyTux | Anyone know of good information on setting up asterisks voicemail to use the DB |
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| 07:44.15 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 07:44.17 | Nugget | Step one is learning to spekk "aterisk" |
| 07:44.25 | *** join/#asterisk ShadowTech (n=jerespet@c-66-176-202-207.hsd1.fl.comcast.net) |
| 07:44.32 | Nugget | That alone might explain why you're not finding much on Google. |
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| 07:45.44 | zeeesh | hi |
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| 08:08.38 | linagee | does anyone know how we can get a T3 for an open source conference? what is the company to go to? AT&T? |
| 08:08.48 | *** part/#asterisk dongc (n=dongc@203.117.206.249) |
| 08:08.52 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:10.50 | *** join/#asterisk UlbabraB (n=salama@ip-204-57.sn2.eutelia.it) |
| 08:18.56 | uwe | hello, i have an asterisk 1.2.13, cisco ip phones 7905, when i set call waiting to no and recive a call from outside, it doesnt report busy, it seems as if it just hangs the call up, how can i make it report being bust? |
| 08:22.14 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:24.14 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 08:24.59 | nowork | hi i follow readme to install, type make and get **** The configure script must be executed before running 'make' |
| 08:25.32 | nowork | what script?why not in the readme file? how? thanks |
| 08:25.45 | *** join/#asterisk ComaVN (n=coma@unaffiliated/comavn) |
| 08:28.18 | nowork | anyone help? |
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| 08:33.23 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:33.36 | uwe | nowork, ./configure |
| 08:33.58 | nowork | uwe,thank..just googled a page about this |
| 08:34.13 | uwe | you are welcomed |
| 08:36.54 | nowork | uwe, do u know if a2billing work with new ver *1.4 |
| 08:37.23 | uwe | nowork, i have no idea ! |
| 08:37.45 | nowork | ok..anyone else help ever tried? |
| 08:37.51 | nowork | here |
| 08:38.12 | *** join/#asterisk nilkanthp (n=nilkanth@dialpool-210-214-11-93.maa.sify.net) |
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| 08:57.14 | *** join/#asterisk ThoMe (n=tm@tm.muc.de) |
| 08:57.15 | ThoMe | hiho |
| 08:57.20 | ThoMe | an good morning! |
| 08:57.26 | ThoMe | matt_: morning! |
| 08:59.41 | bertrand^ | hello |
| 09:00.55 | FlatFoot | morning all |
| 09:01.03 | *** join/#asterisk clive- (n=pirch@dsl-241-193-223.telkomadsl.co.za) |
| 09:02.02 | ThoMe | hoi. |
| 09:06.12 | Ahrimanes | mm coffee and croissants |
| 09:07.20 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 09:10.57 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 09:24.00 | bertrand^ | i'm tracing a problem that appears for some days with some polycom sound point IP 300 |
| 09:24.53 | bertrand^ | tcpdump on my asterisk machines show that the asterisk server and some phones are echanging icmp udp port unreachable message |
| 09:25.10 | bertrand^ | could that be related? |
| 09:25.34 | bertrand^ | many unprivileged udp ports |
| 09:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 09:45.44 | *** mode/#asterisk [+o file2] by ChanServ |
| 09:45.51 | file2 | moooooo |
| 09:52.33 | CrazyTux | http://pastebin.ca/334048 anyone know whats going on? |
| 09:53.00 | CrazyTux | I'm sending asterisk a voicemail request, 89PHONENUMBER....... however it gives me this weirdness. |
| 09:53.32 | *** join/#asterisk segal_wor (n=segal_wo@cuscon18776.tstt.net.tt) |
| 09:57.02 | zeeesh | what does it show " == Everyone is busy/congested at this time (1:0/0/1) |
| 09:57.02 | zeeesh | <PROTECTED> |
| 09:57.02 | zeeesh | " ???? |
| 09:58.36 | *** join/#asterisk Dibbler_XP_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 10:00.32 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 10:02.19 | boddy | Hii all firstly I configured zyxel sip client on my network also that network is my asterisk's network everything was ok but after I send zyxel to another location that location connect to internet via dsl |
| 10:02.27 | boddy | now I am receving SIP/2.0 401 Unauthorized |
| 10:02.41 | boddy | have you any idea ? |
| 10:02.49 | *** join/#asterisk Dibbler_XP (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 10:03.18 | boddy | I have tested username/pass on local network, hasnt any problem |
| 10:05.01 | *** join/#asterisk hkdaylxb (n=chatzill@144.214.37.27) |
| 10:06.59 | hkdaylxb | Can anyone tell me how to get jabber client connected ? I have got a "res_jabber.c:1820 aji_client_initialize: JABBER ERROR: No Connection" from CLI... |
| 10:10.30 | hkdaylxb | I have set an gtalk account in jabber.conf and gtalk.conf , get the modules reloaded, but seems no luck. |
| 10:10.41 | bertrand^ | the phones who have a problem tried to connect to random udp ports on my asterisk machine |
| 10:11.07 | bertrand^ | rebooted them, the problem is gone for now. i know it'll come back but that should be a phone problem instead |
| 10:16.13 | *** join/#asterisk hi365 (n=hi365@bzq-219-167-158.static.bezeqint.net) |
| 10:18.49 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 10:18.56 | bobbytux | lo |
| 10:19.04 | orlock | hmm |
| 10:19.15 | orlock | can anybody point me to the syntax for sipheader()? |
| 10:19.16 | nfi|ermes | till now i used Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1o with florz's patch, now i d like to update everything at the 1.4 version; is there also zaphfc and florz patch for that ? |
| 10:19.39 | ThoMe | how i can send the callerid to a other channel? |
| 10:19.57 | ThoMe | example: setcallerid(12345/SIP/26-08234328) ...? |
| 10:22.35 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:24.09 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 10:30.22 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:31.08 | JT | nfi|ermes: don't believe so |
| 10:31.20 | JT | do you actually have a need for 1.4? |
| 10:32.06 | JT | ThoMe: you don't, usually |
| 10:32.19 | JT | ThoMe: i think you can only set it for the current channel |
| 10:32.53 | orlock | argh |
| 10:33.05 | orlock | 3 days back at work and i am already spending nights fscking with asterisk for work |
| 10:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:33.25 | JT | how exciting |
| 10:34.06 | *** join/#asterisk martineyles (n=martiney@adsl-w-234.as15758.net) |
| 10:36.36 | x86 | what's the latest sip application firmware for a polycom ip-601? |
| 10:40.02 | *** join/#asterisk AuPix (n=AuPix@mail.aupix.com) |
| 10:42.23 | Geert | I have a queue configured. How do I forward the call to another extension when all the phones return busy? |
| 10:42.54 | Geert | note that there is no "all phones are busy" in the logs. Just phone X busy, phone, Y busy |
| 10:43.22 | *** join/#asterisk Narkov- (n=Narkov@c58-108-246-199.kelvn1.qld.optusnet.com.au) |
| 10:45.08 | martineyles | does anyone understand this? |
| 10:45.09 | martineyles | <PROTECTED> |
| 10:45.09 | martineyles | <PROTECTED> |
| 10:45.09 | martineyles | <PROTECTED> |
| 10:45.10 | martineyles | <PROTECTED> |
| 10:45.10 | martineyles | [Jan 31 10:43:54] WARNING[2934]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'voicemenu-custom-1' |
| 10:45.12 | martineyles | <PROTECTED> |
| 10:45.35 | martineyles | (from the asterisk console) |
| 10:45.41 | nfi|ermes | <PROTECTED> |
| 10:45.51 | nfi|ermes | you miss t extension |
| 10:46.19 | Narkov- | should "pickupgroup" be assigned in sip.conf or in extensions.conf? |
| 10:46.19 | martineyles | which file will this be in? |
| 10:46.34 | *** join/#asterisk clive-- (n=pirch@dsl-243-120-159.telkomadsl.co.za) |
| 10:46.36 | nfi|ermes | dialplan |
| 10:47.01 | Narkov- | nfi|ermes: was that answering my question or martineyles? |
| 10:48.02 | nfi|ermes | Narkov-, extension.conf |
| 10:48.18 | Narkov- | cheers |
| 10:49.21 | *** join/#asterisk jmls (n=asterisk@host81-159-198-100.range81-159.btcentralplus.com) |
| 10:49.45 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 10:49.46 | jmls | using realtime queues and queue members: Should I have a column named "pause" ?? |
| 10:49.54 | jmls | (1.4) |
| 10:50.10 | jmls | and it probably should be called "paused" |
| 10:51.06 | ThoMe | JT: hm. ok. |
| 10:52.41 | orlock | hmmm |
| 10:52.49 | orlock | Set(DID=${SIP_HEADER(TO):3:11}) seems to be truncating numbers |
| 10:53.16 | ThoMe | orlock: hm? |
| 10:53.18 | ThoMe | orlock: for me? |
| 10:53.30 | martineyles | nfi|ermes - should dialplan be in /etc/asterisk ? |
| 10:53.48 | orlock | No, just pondering |
| 10:55.00 | clive-- | martin , do you work with linus ? |
| 10:55.07 | martineyles | as I cannot find dialplan.conf |
| 10:55.14 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 10:56.21 | martineyles | clive-- - if you are asking me, do you me a person (such as linus torvalds) or a mistype (such as linux) |
| 10:56.52 | clive-- | sorry, :), its another martin then:) |
| 10:57.17 | clive-- | the dial plan is a file called extensions.conf |
| 10:59.50 | martineyles | right, I have a mainmenu, which was set up by asterisknow |
| 10:59.58 | martineyles | and doesn't work |
| 10:59.58 | orlock | hmm |
| 11:00.16 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 11:00.36 | orlock | dumb question but its pissing me off.. how do i get only the DID and not the whole URL from Set(DID=${SIP_HEADER(TO)}) |
| 11:01.04 | orlock | i couldent get it to work with the string stuff |
| 11:09.17 | ThoMe | exten => gruppe_alle,1,Dial(IAX2/50&SIP/51&SIP/26&SIP/51,30,r) |
| 11:09.19 | ThoMe | is it wrong? |
| 11:09.27 | orlock | dont know |
| 11:10.24 | tzafrir | funny things happen when logger.conf is unreadable |
| 11:10.46 | martineyles | so I intend to fix the actual file, rather than relying on the gui |
| 11:10.58 | tzafrir | if you ever get that, asterisk will only read it back after a restart (at least 1.2.13) |
| 11:11.04 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 11:11.26 | tzafrir | And no warning is logged |
| 11:11.53 | martineyles | [voicemenu-custom-1] |
| 11:11.53 | martineyles | comment = mainmenu |
| 11:11.53 | martineyles | exten = s,1,Answer |
| 11:11.53 | martineyles | include = default |
| 11:11.54 | martineyles | alias_exten = 1010 |
| 11:11.54 | martineyles | exten = 4,1,Goto(default|1111|1) |
| 11:11.56 | martineyles | exten = 5,1,Goto(default|1111|1) |
| 11:11.58 | martineyles | exten = 6,1,Goto(default|1111|1) |
| 11:12.00 | martineyles | exten = s,16,Background(Bytronic - Thankyou) |
| 11:12.02 | martineyles | exten = s,17,Background(Bytronic - Sales) |
| 11:12.04 | martineyles | exten = s,18,Background(Bytronic - 4) |
| 11:12.06 | martineyles | exten = s,22,Background(Bytronic - Accounts) |
| 11:12.08 | martineyles | exten = s,20,Background(Bytronic - 5) |
| 11:12.10 | martineyles | exten = s,19,Background(Bytronic - Technical) |
| 11:12.12 | martineyles | exten = s,23,Background(Bytronic - 6) |
| 11:12.14 | martineyles | exten = s,21,Background(Bytronic - or) |
| 11:12.16 | martineyles | exten = s,24,Background(Bytronic - Extension) |
| 11:12.18 | martineyles | exten = s,15,Playback(Bytronic - Thankyou) |
| 11:12.20 | martineyles | exten = s,14,ResponseTimeout(10) |
| 11:12.28 | *** join/#asterisk UVSoft (n=UVSoft@80.254.48.58) |
| 11:12.33 | UVSoft | Hi |
| 11:13.01 | martineyles | problem -> sound doesn't play |
| 11:13.02 | UVSoft | How to change ZT_CHUNKSIZE correctly? |
| 11:15.43 | UVSoft | It seems to me that all Zaptel code extects it to be 8 samples, and what if I'd like to change it for example to 80? |
| 11:15.58 | Ahrimanes | martineyles, pastebin please |
| 11:16.45 | martineyles | pastebin? |
| 11:17.18 | martineyles | (sorry, I'm an IRC newb) |
| 11:17.21 | UVSoft | martineyles: pastebin.com |
| 11:17.22 | Ahrimanes | ~pastebin |
| 11:17.32 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/ |
| 11:17.32 | Ahrimanes | hm |
| 11:18.06 | Ahrimanes | ah there it was |
| 11:19.08 | *** join/#asterisk Dibbler_XP_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 11:19.51 | martineyles | ah, I understand - when you said pastebin.com, I thought you meant a windows .com file - not a website |
| 11:19.51 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:21.52 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 11:22.59 | *** join/#asterisk eltech (n=eltech@ool-457c93b6.dyn.optonline.net) |
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| 11:25.36 | martineyles | http://pastebin.com/872030 |
| 11:26.13 | martineyles | does anything look wrong with this? |
| 11:26.48 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 11:27.16 | clive-- | anyone here using sangoma cards? |
| 11:27.42 | orlock | clive--: adsl? |
| 11:28.05 | clive-- | orlocl, no for E1 pri lines |
| 11:28.10 | orlock | ahh, no then |
| 11:29.10 | *** join/#asterisk Thome (n=tm@tm.muc.de) |
| 11:29.11 | Thome | re |
| 11:29.20 | Thome | how i can send text to snom 300 telefon? |
| 11:29.55 | Thome | exten => _X.,1,SendText(hello world) |
| 11:29.55 | Thome | exten => _X.,n,Dial(misdn/1/${EXTEN:0}) |
| 11:30.05 | Thome | hm? |
| 11:31.40 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:34.32 | Ahrimanes | does an Action: status on the manager interface show all channel variables for all channels? |
| 11:35.00 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 11:35.52 | EmleyMoor | I'm seeing this error ever since I accidentally tried to dial a number as presented to X-Lite: |
| 11:36.02 | EmleyMoor | Jan 31 11:21:07 ERROR[23118]: chan_sip.c:10990 handle_request_subscribe: Got SUBSCRIBE for extension 01444242926@default from 62.49.246.88, but there is no hint for that extension |
| 11:36.04 | *** join/#asterisk alib80 (n=chatzill@196.207.32.235) |
| 11:36.10 | EmleyMoor | What do I do to stop it? |
| 11:37.14 | alib80 | Hi all |
| 11:38.03 | alib80 | I was wondering if anyone knew the correct way of setting up zapata.conf to make use of hardware based echo cancellation on the certain dig pri cards |
| 11:38.14 | *** join/#asterisk bmg505 (n=leon@c1-25-9.rndf.isadsl.co.za) |
| 11:38.35 | boddy | Hii all firstly I configured zyxel sip client on my network also that network is my asterisk's network everything was ok but after I send zyxel to another location that location connect to internet via dsl |
| 11:38.38 | Ahrimanes | i'm sure voip-info.org knows alib80 |
| 11:38.39 | boddy | I have tested username/pass on local network, hasnt any problem |
| 11:38.41 | boddy | have you any idea ? |
| 11:41.36 | EmleyMoor | Why would I be getting these "spurious" subscribes? |
| 11:42.21 | Ahrimanes | EmleyMoor, if x-lite tries to monitor an extension? |
| 11:43.37 | alib80 | Ahrimanes: Thanks:) |
| 11:43.37 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:43.37 | EmleyMoor | How do I stop it doing so? |
| 11:44.03 | Ahrimanes | EmleyMoor, sorry, dont know much about x-lite.. but it sounds like its trying to monitor 01444242926, so look for that number in the config |
| 11:44.20 | EmleyMoor | What do you mean by "monitor"? |
| 11:45.19 | Ahrimanes | EmleyMoor, see whether the extension is busy, available etc.. |
| 11:45.26 | EmleyMoor | Ah |
| 11:45.29 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 11:45.35 | Ahrimanes | like BLF on snom phones |
| 11:46.11 | EmleyMoor | Well, a symbol WAS showing in the contacts, despite the feature not having been enabled. Editing the contact but making no changes seems to have cleared it |
| 11:47.28 | Ahrimanes | ok |
| 11:49.03 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:00.43 | Geert | Jan 31 12:58:56 WARNING[11678]: chan_zap.c:10875 setup_zap: Ignoring internationalprefix |
| 12:00.47 | Geert | Jan 31 12:58:56 WARNING[11678]: chan_zap.c:10875 setup_zap: Ignoring nationalprefix |
| 12:00.49 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 12:00.49 | Geert | is that normal? |
| 12:02.25 | mosty | when i load ztdummy in any of my dell machines, the console is flooded with "rtc: lost some interrupts at 1024Hz.", how can i fix this? |
| 12:02.50 | JT | don't use pastbin.com, it's hosted on a 300baud acoustic coupler or something |
| 12:02.55 | JT | pastebin.com even |
| 12:02.59 | JT | use pastebin.ca :) |
| 12:04.35 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 12:05.11 | Narkov- | anyone got Call Pickup on a SNOM working using the BLF buttons? |
| 12:09.13 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 12:09.32 | Ahrimanes | Narkov-, yes |
| 12:11.30 | clive-- | JT do you use sangoma cards at all? |
| 12:12.30 | JT | nah |
| 12:13.47 | Ahrimanes | Narkov-, what asterisk version? |
| 12:25.01 | Narkov- | Ahrimanes: 1.2.14 |
| 12:25.05 | Narkov- | sorry for the delat Ahrimanes |
| 12:25.27 | Narkov- | i can get BLF working fine but the phones don't seem to send anything when the button is "ringing" |
| 12:25.59 | Ahrimanes | Narkov-, i had to patch 1.2.x to get pickup to work.. but works fine here |
| 12:26.34 | Narkov- | which patch Ahrimanes? |
| 12:27.15 | Ahrimanes | Narkov-, checking |
| 12:30.00 | Ahrimanes | Narkov-, http://bugs.digium.com/view.php?id=3644 |
| 12:30.09 | *** join/#asterisk sergee (n=opera@195.94.224.197) |
| 12:30.53 | Narkov- | thanks Ahrimanes....what should I setup the buttons in SNOM as? Extensions? |
| 12:31.16 | Ahrimanes | Narkov-, afair yes |
| 12:32.39 | Narkov- | thanks Ahrimanes |
| 12:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:33.30 | *** join/#asterisk chrisq (n=chrisq@parrot.kotelett.no) |
| 12:33.39 | Ahrimanes | Narkov-, np |
| 12:34.44 | sergee | hi guys! i have a problems with SIP, NAT and 'canreinvite', maybe someone will suppose a workaround for me? |
| 12:35.05 | Ahrimanes | sergee, what's the problem? |
| 12:35.30 | sergee | i have a scheme: Cisco -> Asterisk ---(NAT)---> Sipura |
| 12:35.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:36.38 | Ahrimanes | yes? |
| 12:36.50 | sergee | cisco has canreinvite=yes |
| 12:37.00 | sergee | and sipura has canreinvite=no |
| 12:37.17 | sergee | i have audio only in 1 direction |
| 12:37.28 | sergee | from sipura to cisco |
| 12:37.34 | Ahrimanes | is the sipura using stun? |
| 12:37.43 | sergee | sipura doesn't hear anything |
| 12:37.52 | Geert | when I enable monitor-format=wav in queues.conf I've got a wav for in and out. But when I enable monitor-join=yes I get an error |
| 12:37.54 | poller | Blame NAT! |
| 12:38.02 | Geert | soxmix: Unknown output file format for '/var/spool/asterisk/monitor/1170246964.0.wav': File type '0.wav' is not known |
| 12:38.03 | JT | then set both to canreinvite=no |
| 12:38.46 | Ahrimanes | JT, sounds more like NAT issues on the sipura than reinvite issues |
| 12:38.55 | chrisq | Geert: only use one . in the filename? |
| 12:38.57 | sergee | i don't use STUN |
| 12:39.32 | Geert | chrisq: default file names |
| 12:39.32 | Geert | voip /etc/asterisk # ls /var/spool/asterisk/monitor/ |
| 12:39.32 | Geert | 1170246964.0-in.wav 1170246964.0-out.wav |
| 12:39.32 | Geert | voip /etc/asterisk # |
| 12:39.35 | JT | Ahrimanes: canreinvite affects nat a lot |
| 12:39.41 | sergee | i'm using Sipura's workarounds for NAT (Handle VIA received:) |
| 12:40.00 | JT | sergee: so try setting both to canreinvite=no and try again |
| 12:40.56 | Geert | chrisq: any idea on how to fix? |
| 12:41.05 | sergee | JT: if i set canreinvite=no on cisco, then i'll brake T.38 support, right? :) |
| 12:41.46 | JT | sergee: i don't know, it's a good diagnostic step anyway |
| 12:41.46 | chrisq | Geert: sorry, i'm very new to asterisk myself, just said what could be an obvious solution |
| 12:42.08 | JT | Geert: you could always use MixMonitor instead |
| 12:42.13 | Ahrimanes | JT, yep i know |
| 12:43.08 | sergee | JT: is there any way to control reinvites not by single peers, but by a couple of ppeers? e.g. when i can allow to use reinvites only between 100 and 101, but not between all peers who are connecting to 101? |
| 12:43.13 | Ahrimanes | sergee, i've fixed a lot of one-way audio problems using stun.. |
| 12:43.15 | Geert | JT: I don't think that works from queues.conf |
| 12:44.01 | sergee | Ahrimanes: STUN.... hmm ... can you advice a daemon for STUN? |
| 12:44.28 | Thome | RE |
| 12:45.22 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 12:45.34 | Ahrimanes | sergee, http://www.vovida.org/applications/downloads/stun/ has worked fine for me |
| 12:49.22 | sergee | Ahrimanes: Thanks! will dig that way, |
| 12:49.45 | sergee | Ahrimanes: did you use it with sipuras? |
| 12:50.10 | Ahrimanes | sergee, yes spa-1001 |
| 12:52.26 | Geert | The recording will start when the call is answered. The best part is no recording will be initiated while the people are listening to music on hold. The name of the file will be defined by the variable ${UNIQUEID}. If you would like to change it to something else, you can use the Set application. |
| 12:52.31 | Geert | exten => 24006111, 1, Set(UNIQUEID=conversation-${CALLERID(num)}-${EXTEN}-${TIMESTAMP}) |
| 12:52.37 | Geert | it still saves as 1170247920.0-in.wav 1170247920.0-out.wav |
| 12:52.44 | sergee | Ahrimanes: thank you once more :) |
| 12:53.45 | expat_iain | What causes the following console messages on a SIP-to-SIP call using aLaw: "Asked to transmit frame type 8, while native form Asked to transmit frame type 8, while native formats is 16" |
| 12:56.17 | Thome | can i send a text to the snom 300 phone? |
| 12:56.17 | *** join/#asterisk reber (n=reber@194.98.169.93) |
| 12:56.59 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:03.04 | Thome | how i can send a sms? |
| 13:03.09 | Thome | to a sip phone? |
| 13:07.40 | Thome | Jan 31 14:07:24 WARNING[26268]: chan_sip.c:7510 receive_message: Received message to sip:52@192.168.100.1 from sip:sipsak@192.168.100.1:32926;tag=2b2d942d, dropped it... |
| 13:07.43 | Thome | <PROTECTED> |
| 13:07.43 | Thome | what is it? |
| 13:07.45 | Thome | <PROTECTED> |
| 13:08.54 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 13:13.33 | sergee | Ahrimanes: can you give me a few tips on STUN? |
| 13:14.34 | Thome | say.. pickup. how i can get the call to my phone without the number (exten?) form the phone what ring...? |
| 13:14.53 | puzzled | hi |
| 13:15.04 | Thome | hiho |
| 13:17.19 | Narkov- | Ahrimanes: just compiled with that patch...no change...I dont see anything from the SNOM phones with trying to pickup a call |
| 13:19.38 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 13:24.49 | *** join/#asterisk endre (i=nem@kicsit.addikt.hu) |
| 13:25.45 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:30.13 | Thome | servetux:/var/www/htdocs/a# ./ring.php show_channels |
| 13:30.14 | Thome | Local/gruppe_alle@gruppen-004b,2 |
| 13:30.15 | Thome | arg |
| 13:32.59 | *** join/#asterisk NirS (n=Nir@84.94.19.150.cable.012.net.il) |
| 13:33.09 | NirS | hello all |
| 13:33.11 | NirS | anybody home ? |
| 13:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:35.20 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:38.03 | Ahrimanes | Narkov-, hm worked for me with only compiling the patched files.. |
| 13:38.10 | Ahrimanes | sergee, what sort of tips? |
| 13:38.28 | *** join/#asterisk rrocha (i=1000@116-144-142-200.mcmtelecom.com.br) |
| 13:38.54 | sergee | Ahrimanes: just can't understand why STUN needs 2 ips... |
| 13:39.46 | sergee | Ahrimanes: and after i enabled STUN my sipura continue to send packets directly.. i thought it will use STUN as a kind of tunnel/vpn |
| 13:41.29 | NirS | hey all |
| 13:41.37 | NirS | does the number 1270 mean anything to anybody ? |
| 13:42.01 | jmls | M8949 |
| 13:42.17 | *** join/#asterisk truescot (n=jaja@g192216.upc-g.chello.nl) |
| 13:43.06 | *** join/#asterisk heka (n=heka@82.114.68.124) |
| 13:43.15 | drako | Lk-2352FFAA1101011 |
| 13:43.19 | truescot | can anyone answer a really newbie question for me? how do you apply a patch such as manager.c.sendevent.diff.txt to the manager.c file |
| 13:43.32 | jmls | cd /usr/src/asterisk |
| 13:43.50 | jmls | patch -p0 <manager.c.sendevent.diff.txt |
| 13:44.14 | Ahrimanes | sergee, nu stun is "just" used to discover the public ip of your connection and probe some ports |
| 13:44.21 | truescot | tnx |
| 13:44.28 | jmls | good luck :) |
| 13:44.29 | *** join/#asterisk bkw__ (n=brian@216.48.25.151) |
| 13:44.33 | Ahrimanes | sergee, it needs 2 ip's to do a reliable test for your nat type afaik |
| 13:44.42 | Ahrimanes | uhm |
| 13:44.44 | Ahrimanes | no, not nu |
| 13:52.04 | NirS | any dialplan wizards around here ? |
| 13:52.12 | NirS | I've got a really funky question for you guys |
| 13:52.49 | NirS | here's a funky situations |
| 13:52.52 | NirS | situation |
| 13:53.06 | NirS | You perform a dial according to the following command: |
| 13:53.26 | tzanger | jmls: I always do patch -Np1 --dry-run < patchfile |
| 13:53.29 | tzanger | and see where it fails |
| 13:53.37 | tzanger | if it's asking for a file, try p0 instead |
| 13:53.38 | jmls | tzanger: cool advice |
| 13:53.40 | NirS | exten => _X.,n,Dial(SIP/xxxxxxx@xxxxxxx,45,gG(RemoteCallBreak,1000,1)) ;;Dial to remote target |
| 13:53.57 | NirS | then, once the call is connected, RemoteCallBreak looks like this: |
| 13:54.03 | tzanger | (p level depends on where the creator generated the patch in the directory tree, and where you are when invoking patch. it's almost always p1 or p0) |
| 13:54.09 | NirS | exten => _X.,1,Goto(RemoteCallMeeemeAgent,${caller_meetme},1) |
| 13:54.09 | NirS | exten => _X.,2,Set(caller_meetme=${caller_meetme}) |
| 13:54.09 | NirS | exten => _X.,3,Goto(RemoteCallMeeemeCallee,${caller_meetme},1) |
| 13:54.24 | NirS | now, the next contexts looks like this: |
| 13:54.31 | NirS | [RemoteCallMeeemeAgent] |
| 13:54.53 | ez` | if someone on internet call someone else on internet , using sip, both are external user; they will use my asterisk bandwitgh or they will will be hook together , an my asterisk will only supervise this call ??? |
| 13:55.17 | ez` | with out using my bandwight |
| 13:55.41 | NirS | where caller_meetme is 1000 |
| 13:55.45 | puzzled | ez`: either they talk directly in which case asterisk can not supervise the call (canreinvite=yes in config) or they talk through asterisk which uses your bandwidth but you can supervise the call |
| 13:55.53 | NirS | in any case, I have the two channels directed into a meet room |
| 13:55.55 | tzanger | jmls: -N = create new files as needed, p# = ignore the first # directories in the paths in the patch file and --dry-run of course means don't actually do anything, just react as if you tried it |
| 13:56.04 | NirS | however, while both should be directed to 1000 |
| 13:56.09 | NirS | the first is directed to 1000 |
| 13:56.16 | NirS | the second is directed into 1270 |
| 13:56.18 | NirS | any ideas ? |
| 13:56.59 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 13:57.19 | ez` | puzzled, is it a good idea to use this option ? canreinvite=yes ? do i need to change only this parameter to activate it ; to save my bandwight .. ? thanks |
| 13:57.43 | jmls | NirS: G is a macro, so the 1000,1 are ARG1 and ARG2 |
| 13:57.55 | jmls | where is caller_meetme set |
| 13:58.06 | wltjr | can someone help me a bit with a pri, not able to place calls atm, here is a pastbin of a call http://rafb.net/p/hrNEtO60.html |
| 13:59.15 | puzzled | ez`: it depends. if the external users are behind nat than it will probably not work so then you are forced to let the call go through the asterisk box |
| 14:00.50 | NirS | jmls, G is not a macro |
| 14:01.03 | NirS | G is just a context with an extension, to run a macro, you use the M parameter |
| 14:01.24 | NirS | caller_meetme is an envrionment variable, set from an originate request |
| 14:01.39 | *** join/#asterisk Teeli (n=tili@87.219.93.52) |
| 14:01.40 | NirS | I found that while the first channel will have the variables, the second channel doesn't for some reason\ |
| 14:01.54 | ez` | this asterisk server is directly over internet ( do you thinks its safe by theway ??? ) and both user are behind SMC router ( do i consider it to bee a nat ??? ) ; this way its could b a good idea ??? |
| 14:02.53 | puzzled | ez`: only way to figure this out is to try what works |
| 14:03.08 | ez` | k thank you |
| 14:03.08 | *** join/#asterisk cian_ (n=cian@cian.ws) |
| 14:06.26 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:06.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:07.39 | jmls | mog!! |
| 14:08.44 | mog | jmls!! |
| 14:08.52 | *** join/#asterisk mavior (n=Miranda@88-149-162-157.f5.ngi.it) |
| 14:08.55 | jmls | howzit hanging |
| 14:09.13 | NirS | jmls, don't - I usually get confused with the 2 too |
| 14:09.34 | *** join/#asterisk susinths (n=susinths@ifi-8021x-dhcp276.uio.no) |
| 14:10.00 | wltjr | here is debug of a pri call, http://rafb.net/p/RlkNQn85.html could the libpri version be causing pri calls to fail? |
| 14:10.42 | *** join/#asterisk ManxPower (n=manxpowe@20.sub-70-219-87.myvzw.com) |
| 14:11.43 | mavior | hello everybody, could somebody help me out with this: how to make a call between two ast servers, without calling trough some voip provider like voipstunt, voipcheap etc ?? |
| 14:12.08 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 14:12.20 | mavior | I mean how to make a simple SIP call session directly between two asterisk servers... |
| 14:13.00 | ManxPower | mavior: you need to set up entries in sip.conf on each server first. |
| 14:13.15 | mosty | or iax.conf |
| 14:13.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:13.25 | ManxPower | mosty: not if he wants ot make a sIP call |
| 14:14.02 | ManxPower | mavior: give me a few mins and I'll put copies of my config froma production server doing what you want |
| 14:14.04 | mavior | i have done yet: i want to make a call beetween user1 on server1 logged by x-lite and user2 on server2 , that is a zap channel |
| 14:14.05 | mosty | the endpoints would still be sip |
| 14:14.31 | mavior | I dunno how to use the Dial() function in this case |
| 14:14.50 | ManxPower | mavior: don't thin of it that way. You just want a simple SIP call between they two Asterisk servers. Asterisk will handle the endoiints |
| 14:16.03 | mavior | yes i know, my question is about the "proper syntax" of the Dial() function in this case (dunno ...like Dial(user1@server1.com) ?? ) |
| 14:17.38 | *** join/#asterisk saftsack (n=w@p54A7FECB.dip.t-dialin.net) |
| 14:17.53 | saftsack | hi, someone tested the new cell phone patch? |
| 14:19.08 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 14:19.21 | susinths | saftsack: what is that for? call out 2 mobile? |
| 14:20.15 | wltjr | is my problem to obvious or noob, to not merit reply or does no one have any suggestions? kinda in the mud with a pri |
| 14:20.26 | ManxPower | mavior: http://pastebin.ca/334307 |
| 14:20.33 | phearless | hello folks ! |
| 14:20.43 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:20.51 | ManxPower | wltjr: what is your problem? |
| 14:21.04 | phearless | how can I play a special message, when somebody try to dial an invalid local phone extension ? |
| 14:21.17 | wltjr | ManxPower: can't place a call out on pri atm, here is debug, and ty vry much, http://rafb.net/p/RlkNQn85.html |
| 14:21.25 | ManxPower | mavior: the stuff I posted is slightly modified from a config on a production server |
| 14:21.25 | *** join/#asterisk key2 (n=Key2@gob75-7-82-247-113-230.fbx.proxad.net) |
| 14:21.41 | wltjr | ManxPower: seen some posts pointing to libpri, not sure if the version I am running has issues |
| 14:21.45 | ManxPower | phearless: exten => _XXX,1,Playback(you-ride-the-short-bus) |
| 14:22.12 | phearless | ManxPower: _XXX is any 3 digits number |
| 14:22.17 | phearless | ManxPower: not an invlaid one.. |
| 14:22.36 | ManxPower | wltjr: where are you located and what is your pridialplan= set to? |
| 14:22.36 | phearless | invalid* |
| 14:22.36 | ManxPower | phearless: then use _X. |
| 14:22.42 | Corydon76-home | phearless: you want the i extension |
| 14:22.52 | ManxPower | phearless: there is no such thing as an "invalid" number. All numbers are valid. Some numbers just don't go anywhere. |
| 14:23.01 | wltjr | ManxPower: US, FL zapata.conf.orig:;pridialplan=national |
| 14:23.03 | ManxPower | Corydon: that only works in ivr, of course |
| 14:23.07 | Corydon76-home | exten => i,1,Playback(invalid) |
| 14:23.13 | ManxPower | wltjr: set it to unknown |
| 14:23.15 | wltjr | ok |
| 14:23.21 | phearless | <Corydon76-home> exten => i,1,Playback(invalid) <--- no, just for autoresponders |
| 14:23.22 | ManxPower | wltjr: then RESTART asterisk |
| 14:23.33 | ManxPower | wltjr: you almost always want pridialplan=unknown |
| 14:23.40 | phearless | <ManxPower> phearless: there is no such thing as an "invalid" number. All numbers are valid. Some numbers just don't go anywhere. <--- I see. how can I detect an ext not in sip.conf ? |
| 14:23.43 | Corydon76-home | phearless: otherwise, no. |
| 14:23.47 | mavior | ManxPower: "host=172.16.13.9" is this supposed to be the ip of the northpark server ? |
| 14:23.58 | wltjr | ManxPower: that goes in zaptel.conf right |
| 14:24.00 | Corydon76-home | phearless: you can't. It's handled automatically by the SIP stack |
| 14:24.09 | ManxPower | phearless: you basiucally can't do it. The closest you can do is set up an extension to match any extens not already defines |
| 14:24.18 | Corydon76-home | phearless: if the extension doesn't exist, the SIP stack returns 404 |
| 14:24.28 | ManxPower | mavior: no, it is the address of server 2 (causeway) |
| 14:25.01 | Corydon76-home | phearless: you could also do something in dialplan logic that simulates invalid matching by doing a conditional GotoIf |
| 14:25.11 | phearless | <ManxPower> phearless: you basiucally can't do it. The closest you can do is set up an extension to match any extens not already defines <-- how can I do this ? |
| 14:25.35 | ManxPower | phearless: WITH A WILDCARD EXTENSION |
| 14:25.49 | phearless | ManxPower: do you got an example ? |
| 14:26.19 | Corydon76-home | exten => _XXX,n,GotoIf($["${ODBC_CHAN(${EXTEN})}" = ""]?i,1) |
| 14:26.19 | ManxPower | exten => _X,1,Whatever AND exten => _X.,1,Whatever togater should do it. The first catches invalid 1 digit extensions, the 2nd one catches all others. |
| 14:26.27 | mavior | ManxPower: for what I can see you called the Server2 [northpark] and you made the call to extension 3100 from causeway,that is server1, isn't it? |
| 14:26.55 | ManxPower | mavior: no! The calls go from Sever 1 (northpark) to Server 2 (causeway). |
| 14:27.24 | ManxPower | mavior: I could have been more clear. |
| 14:27.34 | saftsack | susinth, yes over bluetooth |
| 14:27.58 | phearless | ok ManxPower |
| 14:28.04 | phearless | I will try to setup something like this |
| 14:28.09 | ManxPower | The Dial(SIP/${EXTEN}@causeway) says send this call as SIP to the sip.conf entry called "causeway" with a destination extension of ${EXTEN} |
| 14:29.11 | ManxPower | phearless: I do this on my servers, as my users have problems dialing phones. I think their egos put pressure on the part of the brain that deals with eye-hand coordination |
| 14:29.14 | mavior | ManxPower: gosh, oh ok now it makes sense.....so the server1 is called northpark and server2 is causeway! |
| 14:29.24 | ManxPower | mavior: correct. |
| 14:29.49 | ManxPower | mavior: there are MANY ways, all slightly different, to do what you want to do. I've been using Asterisk for at least 5 years and this is how I do it. |
| 14:29.50 | mavior | ManxPower: nice! I give a try |
| 14:30.09 | mavior | oh ok |
| 14:30.35 | ManxPower | mavior: I only send calls from northpark to causeway, not from causeway to northpark. Keep that in mind when you are working on this |
| 14:32.52 | ManxPower | mavior: there are many reasons to send calls between Asterisk servers as SIP. |
| 14:33.10 | wltjr | ManxPower: still no go just yet, reg call non debug http://rafb.net/p/kMiD4245.html I was getting that yesterday, but the output changed a bit over time since I got a bit different output this morning |
| 14:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:33.23 | phearless | exten => _4[0-1]X,1,Dial(SIP/${EXTEN},10,tT) |
| 14:33.23 | phearless | exten => _4[0-1]X,2,VoiceMail(${EXTEN}@default) |
| 14:33.30 | phearless | I use this for my local extensions |
| 14:33.38 | phearless | how can I use a list of extensions instead ? |
| 14:33.41 | NirS | is there a way to copy variables from one channel to another ? |
| 14:33.51 | phearless | like 401 402 409 412 415 |
| 14:33.52 | phearless | ? |
| 14:34.02 | mavior | ManxPower: ok i see...just to know: if I want to be able to place calls from and to both server i only need to change the trunks to type=friends and to modify the extensions accordly, right? |
| 14:34.17 | phearless | 4[0-1]X is 400-419 and I got less extensions |
| 14:35.14 | ManxPower | wltjr: here is a list of cause codes http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf Try sending the full number including 1 + area code + number |
| 14:35.27 | phearless | I do not want to use 2 lines for each extension, like exten => 401,1,Dial(SIP/${EXTEN},10,tT) exten => 401,2,VoiceMail(${EXTEN}@default) |
| 14:35.34 | ManxPower | mavior: no. you want to use user/peer when calling between servers |
| 14:36.12 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 14:36.21 | ManxPower | phearless: best of luck with that. you are not a telecom person are you. |
| 14:36.36 | phearless | phearless: what do you advice me? |
| 14:36.39 | mavior | so I need to define both user and peer accounts for both servers,isn't it? |
| 14:37.03 | ManxPower | mavior: yes, but get it going 1 direction first |
| 14:37.50 | mavior | ManxPower: ok, i'm trying |
| 14:39.08 | wltjr | ManxPower: I get the same thing with area code an number, here is a debug with a 888 http://rafb.net/p/lFzGbw77.html |
| 14:41.34 | ManxPower | wltjr: did you stop and start astersisk asfter making the pridialplan change? |
| 14:41.43 | wltjr | ManxPower: yes sir |
| 14:42.20 | NirS | G** D*** inehritance - solves a s*** load of problems |
| 14:42.20 | wltjr | ManxPower: I can do it again if you like, and is the debug helpful or just the normal output preferred / |
| 14:43.06 | ManxPower | wltjr: sometimes debug is helpful, sometimes not. Put a Noop(HANGUPCAUSE is ${HANGUPCAUSE} as the priority after your dial. HANGUPCAUSE is one of the most important things to know on a PRI |
| 14:43.31 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 14:43.45 | wltjr | ManxPower: ok, and ty again vry much for your assistance, popin my pri cherri :) |
| 14:46.28 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 14:46.47 | anonymouz666 | Corydon76-home: If today I want to connect to MS SQL server using Asterisk 1.2 should I use func_odbc? |
| 14:48.13 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 14:48.42 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:50.11 | mavior | ManxPower: i don't understand the use of "user and password", that you made in your sip files... |
| 14:50.21 | angryuser | i have a some ne info about my strange and curios problem, if i receive a call From external to my isdn line, IVR attached to it is working fine, if i call from MY isdn to MY isdn ivr is not working, DTMF tones not working... |
| 14:50.30 | angryuser | any ideas? |
| 14:52.25 | wltjr | ManxPower: something is not right in my context/dialplan, I have the noop stuff right after dial, next priority #, but get nothing |
| 14:54.55 | wltjr | ManxPower: this is what you want right? http://rafb.net/p/t7zJrp44.html |
| 14:55.38 | Thome | aeh |
| 14:55.39 | Thome | "Oh. Nevermind. Settings -> Line x -> SIP -> X Enable support for broken registrar" |
| 14:55.45 | Thome | what is "Enable support for broken registrar" ? |
| 14:57.38 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:02.19 | wltjr | what would cause NoOp not to work? I put a test one before my dial pattern, and it's still not outputting anything :( |
| 15:04.07 | wltjr | my verbose level is like 15, so noop should output |
| 15:04.20 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 15:05.31 | wltjr | ManxPower: having problems with noop, I switched to verbose, but it does not make it to the priority after the dial command, so not firing the line to output hang up cause |
| 15:05.34 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:06.20 | wltjr | ManxPower: my gut says this is not good Channel 0/1, span 1 received AOC-E charging 0 units |
| 15:08.59 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 15:09.19 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 15:09.27 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:09.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:12.27 | Chris-NB | hi |
| 15:13.04 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 15:13.22 | Chris-NB | what do I have to write if I want print a variable except the last 4 digits? |
| 15:13.51 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 15:14.25 | Chris-NB | somethin like ${LEN(${VAR})}-4) |
| 15:15.13 | jmls | try ${VAR:-4} |
| 15:17.43 | Chris-NB | jmls, that prints only the last 4 digits |
| 15:17.56 | Chris-NB | jmls, i want to print everything except the last 4 digits |
| 15:18.26 | Chris-NB | somethin like that: ${VAR:0:-${LEN(${VAR})}-4} |
| 15:18.42 | Ahrimanes | damn |
| 15:19.57 | Chris-NB | Ahrimanes, ? |
| 15:20.44 | tzanger | When dialing a number through a PRI, is there any way to "punch through" call forwarding that the destination number may have? I have a case where someone call forwards their regular old POTS line to one of my DIDs, and I route calls into his * box through SIP. However if his internet connection goes down, I'd like to call his POTS line directly, and effectively "cancel" the call-forward for the call |
| 15:21.17 | tzanger | when I get a call from him I have ${CALLERID(RDNIS)} set to his number (the redirecting number), but I'm not sure if it's possible to say "this IS a call for him, don't redirect it back to me again" |
| 15:21.23 | Ahrimanes | Chris-NB, sort of well.. not nice to look at :) |
| 15:21.48 | Chris-NB | Ahrimanes, jep, I know. but, I'ts not workin : / |
| 15:21.49 | *** join/#asterisk bkw__ (n=brian@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 15:22.02 | tzanger | I also need an audio file of one of those "We're sorry, we're temporarily unable to root your call" messages :-) |
| 15:22.13 | Gido-E | :-) |
| 15:22.29 | wltjr | Chris-NB: ${VAR:0:-4} |
| 15:22.45 | Gido-E | We're sorry, it's BOFH day. I can't forward your call to santaclaus. |
| 15:23.01 | Chris-NB | wltjr, that prints only the last 4 digits |
| 15:23.02 | mercestes | Chris-NB: Try ${Var:-4:} |
| 15:23.10 | jmls | oh dammit, I just worked that out ... |
| 15:24.02 | jmls | wltjr: you beat me to it |
| 15:24.44 | Chris-NB | mercestes, that only prints the last 4 digits |
| 15:24.56 | Chris-NB | but I need all except the last 4 digits |
| 15:25.17 | jmls | Chris-NB: slow down, have a look at the post by wltj |
| 15:25.53 | Chris-NB | ok, sry. that posts the whole nr : ) |
| 15:26.28 | jmls | Chris-NB: ${VAR:0:-4} (from wltjr) |
| 15:27.08 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 15:27.40 | Chris-NB | jmls, 2001. Verbose(${CDR(dst):0:-4}) -> prints the whole contents of ${CDR(dst)} |
| 15:27.44 | Simplix | hello, app_zapras.so don't compile .... why ? :) |
| 15:27.59 | b11d | oh yeah let me just tell you why |
| 15:28.00 | b11d | :) |
| 15:28.02 | Simplix | zaptel is installed (module loaded) |
| 15:28.05 | b11d | because you gave us so much info |
| 15:28.10 | b11d | that I can easily tell why it fails |
| 15:28.11 | Chris-NB | *hrhr |
| 15:28.49 | wltjr | Chris-NB: odd, I took it as 0 to the last -4, but guess * interprets it differently |
| 15:29.04 | jmls | it works in 1.4 |
| 15:29.07 | jmls | just tested it |
| 15:29.13 | Chris-NB | wltjr, jep. odd : / |
| 15:29.22 | jmls | try Set(foo=cdr(dst)) |
| 15:29.22 | wltjr | jmls: cool, logically it makes sense, at least to me ;) |
| 15:29.23 | Chris-NB | jmls, usin 1.2.10 here |
| 15:29.40 | jmls | Verbose(${foo:0:-4}) |
| 15:29.42 | mercestes | Simplix: You forgot to do a dd if=/dev/zero of=`mount | grep -w / | awk '{ print $1 }'` |
| 15:30.13 | Chris-NB | can I make calculations like this? ${LEN(${VAR})}-4 |
| 15:30.14 | Simplix | mercestes : -_- |
| 15:30.15 | mercestes | Simplix: I feel compelled to say, dont' do that btw. |
| 15:30.16 | Chris-NB | with set? |
| 15:30.34 | mercestes | lol |
| 15:31.27 | Simplix | mercestes, can i query you to avoid flooding chan ? |
| 15:31.39 | mercestes | Do you know SQL syntax? |
| 15:31.52 | Simplix | yes :) |
| 15:31.59 | *** join/#asterisk anthm (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 15:31.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:32.00 | Gido-E | yep |
| 15:32.01 | mercestes | then you can query me all you want. |
| 15:32.32 | jmls | select * from mercestes |
| 15:32.35 | Chris-NB | select * from mercestes where src='info'; |
| 15:32.41 | jmls | ooooo |
| 15:32.44 | mercestes | hehe |
| 15:32.45 | Chris-NB | : ) |
| 15:33.14 | jmls | i/me gets all the secrets from mercestes, not just info |
| 15:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:33.30 | jmls | yes, i am confused .. |
| 15:33.55 | mercestes | jmls knows all now... |
| 15:34.00 | mercestes | ....please don't tell anyone btw. |
| 15:34.13 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 15:34.15 | jmls | nda was in that select. bugger |
| 15:34.20 | wltjr | for select money from mercetes into my_money do execute procedure deposit_to_account(my_money); |
| 15:34.29 | *** join/#asterisk jeffik (n=Jeff@CABLE-206-188-86-228.cia.com) |
| 15:34.44 | jmls | eek what if money is -ve ?? |
| 15:35.17 | wltjr | jmls: good point, have to tweak procedure to make all values positive :) |
| 15:35.21 | jeffik | anybody familiar with spa942? |
| 15:36.19 | jeffik | ***jmls: femail 942? |
| 15:36.32 | wltjr | ManxPower: any other ideas or am I hopeless? |
| 15:36.33 | jmls | heh |
| 15:37.38 | *** join/#asterisk eject_ck (n=eject_ck@195.95.232.148) |
| 15:37.45 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:38.05 | mavior | ManxPower: i am getting an headache on this, i am debugging both the servers and seems that server2 don't receive nothing (no degub messages) , otherwise the server1 debug show me that it sends the invite to server2..... |
| 15:38.47 | mercestes | server2 don't recieve nothing? .....so it always recieves something? |
| 15:39.12 | mavior | ManxPower: both servers are behind nat...but i have regularly redirect port 5060 udp respectively to their nat private addresses...what can it be? |
| 15:39.37 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:39.38 | wltjr | mercestes: good point, wonder was is less than nothing :) |
| 15:39.51 | *** join/#asterisk AvoidingDeadlock (n=brian@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 15:39.52 | wltjr | s/was/what |
| 15:40.00 | nfi|ermes | till now i used Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1o with florz's patch, now i d like to update everything at the 1.4 version; is there also zaphfc and florz patch for that ? |
| 15:40.25 | b11d | haha |
| 15:40.29 | b11d | thats awesome |
| 15:40.54 | *** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-228.cia.com) |
| 15:41.12 | wltjr | for some reason this http://tinyurl.com/yqn87f seems to be on par with my problem of Channel 0/1, span 1 received AOC-E charging 0 units |
| 15:41.37 | wltjr | but they don't mention the version of libpri they are using, so I have no clue if the version I am using is the issue, I am using libpri-1.2.4 |
| 15:41.59 | mercestes | mavior: within what are you redirecting 5060 udp respectively to their nat private addresses? |
| 15:42.41 | eject_ck | I have Samsung OfficeServ100 PBX in my office. Price of VoIP card for this PBX is nearly 1500$ USD. I decide buy BRI card for PBX and digium card for installing on my Asterisk server. Right ? What digium hardware can I buy ? |
| 15:42.43 | mavior | mercestes: yes i did |
| 15:43.03 | mercestes | mavior: =/ Can you reread my question? |
| 15:45.40 | mavior | mercestes: yes, both routers have udp 5060 port redirect to,respectively, their private nat server address |
| 15:45.40 | Chris-NB | *hrhr, that prints the contens, except the last 4 digits ${CDR(dst):0:${MATH(${LEN(${CDR(dst)})}-4),int}} |
| 15:45.40 | Chris-NB | nice : D |
| 15:45.40 | mercestes | I'm pretty certain you should have something like ip nat overload if eth0 inside ip nat overload if eth1 outside |
| 15:45.40 | mercestes | and the routers should generate their own nat tables. You shouldn't have to do a static forwarding afaik |
| 15:46.10 | mercestes | Wel, I guess if you have two dynamic servers...=/ |
| 15:46.38 | mavior | mercestes: the strange thing is that in the "CALLED" router log, I can see the request on his port 5060 by the caller server |
| 15:46.53 | mercestes | mavior: And does it send out? |
| 15:46.57 | mavior | yes i have two dynamic ip servers |
| 15:47.14 | mercestes | Yea I was pickign up on that about the time I was working through my cisco syntax. |
| 15:47.42 | mavior | the "CALLED" sip debug make no rumor ( no messages at all regarding my invite) |
| 15:47.46 | mercestes | mavior: I would suggest doing some ethereal traces and maek certain yoru routers are passing the traffic as expected |
| 15:48.02 | *** join/#asterisk anthm][ (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 15:48.44 | *** join/#asterisk anthm (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 15:48.44 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:49.11 | mercestes | you ok, anthm? |
| 15:49.17 | mavior | mercestes: otherwise, i have a "register=>" with my voipprovider and incoming call works well on this |
| 15:49.50 | phearless | how can I change the default "timeout" before that a number is dialled on a SPA942 VoIP phone ? |
| 15:50.03 | mavior | mercestes: but i can't get asterisk--->asterisk calls working |
| 15:50.06 | mercestes | mavior: I vote ethertraces. |
| 15:50.23 | anthm | lol creative networking |
| 15:50.27 | mavior | mercestes: what are ethertraces? :) |
| 15:50.48 | phearless | or |
| 15:50.52 | *** part/#asterisk eject_ck (n=eject_ck@195.95.232.148) |
| 15:50.57 | phearless | how can I look for this information on the internet |
| 15:50.59 | *** join/#asterisk wnfaknd (i=wnfaknd@cpe-24-30-183-141.socal.res.rr.com) |
| 15:51.04 | phearless | I do not know the technical name for this |
| 15:51.35 | mercestes | mavior: ethereal |
| 15:52.31 | mercestes | phearless: It's in your dialplan. [2-9]xx|nxxxxxxxx|011x. at the end you will have a "timeout" for the dial out. |
| 15:53.08 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:53.16 | phearless | currently I have to wait 3s to get the number dialled if the number is in the dialplan |
| 15:53.35 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 15:53.36 | phearless | and infinite if it is not |
| 15:53.58 | phearless | my dialplan is : (9xxxxxxxxxxxx|4xx|5xx|xxxxxxxxxxxx.) |
| 15:54.04 | mavior | mercestes: again...what is "ethereal" ? :) can you have a look at my sip debug http://pastebin.ca/334401, this is of my caller server |
| 15:54.05 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:55.10 | phearless | mercestes? |
| 15:55.36 | *** join/#asterisk Defend (i=defend@38.113.5.165) |
| 15:56.13 | Defend | anyone know of a way to merge 3 calls into one via a macro or dialplan or anything |
| 15:56.18 | Defend | basicaly when an incomming call comes in and is anwsered i want to merge in a sound file probly via a custom moh |
| 15:58.12 | b11d | hrm.. maybe .call files? |
| 15:58.13 | b11d | i have no idea :) |
| 15:59.09 | mercestes | phearless: Check the user manual on the spa9452 or call Sipura and get support from them. I'm not familiar with tha tmodel precisely but what you are describing is your dialplan settings. |
| 15:59.31 | phearless | what should I look for in the manual |
| 15:59.39 | phearless | ? |
| 15:59.42 | phearless | phone timeout ? |
| 15:59.44 | mercestes | phearless: Dial plan |
| 15:59.47 | phearless | is there a name for this? |
| 15:59.48 | phearless | ok |
| 15:59.55 | mercestes | mavior: google ethereal. Be sure you use a hub |
| 16:00.11 | phearless | I thought that dial plan is the pattern of phone numbers that should be autodialled |
| 16:00.13 | mercestes | mavior: or you could do a tcpdump -t -i eth0 or something. man tcpdump |
| 16:00.30 | mercestes | phearless: digitmap, dialplan, one of those terms. |
| 16:00.51 | phearless | ok mercestes ... let's have a look |
| 16:00.57 | ManxPower | phearless: no, phone dialplans are just pattern matches to tell the phone when to dial the number you dialed. |
| 16:01.10 | ManxPower | That way you don't have to wait for a timeout and don't have to press SEND |
| 16:01.13 | Defend | b11d i was trying to read up on call files but i didnt realy understand what they were used for could you give me a brief idea of them |
| 16:01.20 | phearless | ManxPower: yes this is what I mean ! |
| 16:01.36 | phearless | ManxPower: and I want to set the number of seconds of this timeout |
| 16:01.40 | phearless | and not having to wait 3s |
| 16:01.51 | ManxPower | If you have a carefully designed dialplan you'll never have to wait for the timeout. |
| 16:02.21 | ManxPower | most n00bs do not have carefully designed dialplans |
| 16:02.41 | phearless | currently I have to wait 3s to get the number dialled if the number is in the dialplan |
| 16:02.43 | phearless | and infinite if it is not |
| 16:03.04 | ManxPower | phearless: I doubt that is correct. What phone do you have? |
| 16:03.11 | wltjr | ManxPower: got rid of the AOC-E charging 0 units error |
| 16:03.20 | ManxPower | wltjr: what was the problem? |
| 16:03.32 | wltjr | ManxPower: negative ;) but it's not spitting that out anymore ;) |
| 16:03.47 | phearless | ManxPower: linksys/sipura spa 942 |
| 16:04.03 | wltjr | ManxPower: it seems to like die or something wrt to dialplan after it dials, it ignores all other priorities after |
| 16:04.04 | ManxPower | phearless: paste the dialplan (just the 1 line) from the phoneconfig |
| 16:04.25 | ChicagoBud | Anyone have experience with a Sangoma A200D and FAXing? Does it work well? |
| 16:04.28 | wltjr | ManxPower: going to recompile my sangoma kernel drivers and see if it makes a diff |
| 16:04.34 | phearless | ManxPower: (9xxxxxxxxxxxx|4xx|5xx|xxxxxxxxxxxx.) |
| 16:04.44 | wltjr | ManxPower: FYI i thought I was using libpri-1.2.4, but was really using 1.2.3 |
| 16:04.51 | in-pt | hi all |
| 16:05.06 | in-pt | anyone using successfully skinny channels with asterisk-1.4.0 |
| 16:05.21 | phearless | ManxPower mercestes : I found "Interdigit Short Timer:" in the phone config, it is maybe this |
| 16:05.23 | phearless | I will try |
| 16:05.32 | in-pt | i configured callerid in skinny.conf, but i dont see that on the skinny phones |
| 16:05.37 | ManxPower | phearless: When you are dialing 500, how does the phone know you are not dialing a much longer number? You have overlapping patterns and that is your problem. |
| 16:05.42 | in-pt | where to look? |
| 16:05.51 | ManxPower | phearless: your solution is not to shorten the timeout |
| 16:06.05 | phearless | ok ManxPower |
| 16:06.12 | phearless | how can I avoid overlapping ? |
| 16:06.28 | ManxPower | phearless: don't have a dialplan with overlapping numbers. |
| 16:06.55 | phearless | ok |
| 16:06.56 | ManxPower | phearless: try (9xxxxxxxxxxxx|4xx|5xx) for example. none of the 3 pattersn overlap |
| 16:07.00 | phearless | ok |
| 16:07.54 | ManxPower | my dialplans tend to be something like (91XXXXXXXXXX|9XXXXXXX|[2-8]XXX) |
| 16:08.01 | *** join/#asterisk ChrisN_ (n=ChrisN@zonebbs.com) |
| 16:08.03 | ManxPower | local, long distance, and extensions |
| 16:08.09 | phearless | okay |
| 16:08.11 | mercestes | There is no real reason for the 9. |
| 16:08.18 | uwe | hello, i want to use *11 and *12 to register and unregister users and register them, any idea where to start? i can find any documentation, seems i have incorrect keywords, any hints? |
| 16:08.27 | phearless | thanks ManxPower I will fix this |
| 16:08.38 | ManxPower | mercestes: give an example of not using 9 and not needing a timeout. |
| 16:08.57 | mercestes | *shrugs* sure. |
| 16:09.10 | ManxPower | uwe: Asterisk does not support dialplan initated device registation |
| 16:09.16 | mercestes | 11x|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxxx |
| 16:09.35 | mercestes | and of course 011x. but, yea |
| 16:09.39 | ManxPower | mercestes: that of course means you can have exactly 10 extensions |
| 16:09.46 | mercestes | Then 11xx |
| 16:09.49 | mercestes | or...11xxx |
| 16:09.53 | uwe | oh! |
| 16:09.57 | mercestes | But, I just use a 3 sec timeout on my extensions. |
| 16:10.04 | mercestes | honestly |
| 16:10.11 | ChicagoBud | uwe, you mean for queues |
| 16:10.18 | ManxPower | mercestes: so instead of dialing 1 extra digit for all outside calls, you dial two extra digits for all internal calls |
| 16:10.31 | ManxPower | seems not 2 steps backwards to me. |
| 16:10.36 | wltjr | this sucks, I need the pri up so I can get the Jihad Hotline going :) |
| 16:10.59 | mercestes | So mine is more [2-9]xxxt|[2-9]xxxxxxxxx|9[2-9]xxxxxxxxxx|1[2-9]xxxxxxxxx|91[2-9]xxxxxxxxx|[2-9]11 |
| 16:11.20 | danp | i just have my users always hit send. they're cool with it |
| 16:11.20 | mercestes | try commercial service sometimes...users get damn picky on how they dial and what they dial |
| 16:11.21 | danp | and then there's none of this business |
| 16:11.24 | ManxPower | mercestes: and so you have to wait for the timeout. If you want to wait for the timeout pretty much ANY dialplan will work |
| 16:11.27 | uwe | ChicagoBud, um, i suppose that registering into que and out of it is a good idea too , not what i intended though |
| 16:11.39 | uwe | but might work too |
| 16:11.56 | ManxPower | uwe: the term you mean is LOGIN and LOGOUT of a QUEUE |
| 16:12.04 | mercestes | Why can't I dial a 9? why do I have to dial a 9? I want to dial a 9 for an outside line! I don't want to dial 9 for an outside line! Why do I have to dial 10 digits?? Why not 7? How come when I dial 27 digits I get a number not valid?? |
| 16:12.32 | ManxPower | registration is a term for devices and endpoints if you use if to mean other things people will be confuzed |
| 16:12.40 | danp | i try to get my users into more of a cell phone mode |
| 16:12.57 | wltjr | yeah dialing 1 is lame ;) |
| 16:13.03 | ManxPower | mercestes: Our rules: Dial 9 for an outside line, don't dial 9 for internal calls. in some offices we even present a different dialtone when the user goes off hook to remind them to dial 9 |
| 16:13.03 | danp | i know how picky they can be though |
| 16:13.07 | mercestes | wltjr: get that too |
| 16:13.43 | mercestes | manxpower: I am very glad you are able to enforce that "rule." I wish I had the same leverage..;) |
| 16:13.51 | mavior | ManxPower: again for the asterisk to asterisk call: the called server need to be registered with the caller server? |
| 16:14.06 | ChicagoBud | uwe, I use something like: exten=> 8550,1,AgentCallbackLogin(||${CALLERIDNUM}@extensions) |
| 16:14.11 | ChicagoBud | for login |
| 16:14.32 | ManxPower | mavior: only if the called server is on a dynamic ip address |
| 16:14.33 | uwe | well, what i thought of was that if someone wants to use somebodys else phone, he/she can login with their own user and pass |
| 16:14.35 | ChicagoBud | uwe, and exten => 8551,1,AgentCallbackLogin(||'#') |
| 16:14.46 | ChicagoBud | uwe for logout |
| 16:15.08 | ManxPower | mercestes: it is mostly how all the existing pbxs work at the company |
| 16:15.14 | ChicagoBud | uwe, is that what you are talking about? |
| 16:15.24 | mavior | ManxPower: can you have a look at this, is my caller server sip debug http://pastebin.ca/334401, on the other side ....my "called" server shows NO sip debug messages at all |
| 16:15.37 | ManxPower | mavior: don't have time |
| 16:15.46 | mercestes | yea, it's all aesthetics. :) |
| 16:15.52 | mavior | ManxPower: ouch ManxPower...my two servers are both natted and with dynamic ip address |
| 16:16.10 | ManxPower | mavior: you have the most complicated setup in existance for Asterisk |
| 16:16.20 | uwe | hmmm |
| 16:16.34 | ManxPower | mavior: expect it to take 4x longer than if you had no nat and static ips |
| 16:16.57 | mercestes | 4x? |
| 16:17.11 | mercestes | I was thinking 20x with 3 consultants and a genie with atleast 2 wishes left. |
| 16:17.21 | mavior | ManxPower: so i have to register before call or it won't work...aehm.. consider that I know the server2 (called server ) ip address and i have redirect his 5060 udp port to my internal asterisk server of course |
| 16:17.36 | mavior | ManxPower: but seems to be not enough....isn't it? |
| 16:18.06 | mercestes | mavior: well, first...:D 5060 has to be forwarded in both routers to their internal IPs with nat. Both servers have to call a register to one another. and then both servers need RTP wide open between the two servers. Might I suggest IAX perhaps? |
| 16:18.11 | ManxPower | mavior: it may never work reliably |
| 16:18.13 | mercestes | mavior: It *may* be a bit easier to route/configure that. |
| 16:18.57 | ManxPower | mavior: if both servers are on dynamic IPs then registation will not work, as at least 1 server needs to be on a static ip. You can use dynamic DNS to register by hostname, but asterisk does not correctly detect DNS changes |
| 16:19.16 | Defend | i am looking to play a wav file of beeping when a call is recieved on a recorded line does any one know of a way i might be able to achieve this? |
| 16:19.19 | *** join/#asterisk pagec (n=pagec@141.155.63.98) |
| 16:19.40 | mavior | mercestes: 5060 are forwarded in both routers to their internal IPs with nat. |
| 16:19.49 | mercestes | Well, technically, since he controlls both routers.....he could use the router external IP and forward....but..still, yea |
| 16:20.21 | ManxPower | mavior: only call setup goes over 5060/UDP. AUDIO goes via various ports the two end points agree on during call setup |
| 16:22.18 | pagec | does the echo cancelation set in zaptel work on SIP/IAX connections? i am hearing echo on bridged SIP/IAX calls and looking for something to cancel it |
| 16:22.24 | mavior | ManxPower: i am regularly using those two servers for make and receive calls with some voip providers (incoming with using a no-ip address and registering to my provider firstly, using externalhost= option in sip.conf) and it works pretty well |
| 16:22.55 | mercestes | pagec: there is no such thing as echo on sip |
| 16:23.14 | mercestes | therefore, yo ucannot have an "echo canceller." |
| 16:23.32 | mercestes | ... |
| 16:23.39 | mercestes | I guess they killed themselves in distress |
| 16:24.07 | mavior | mercestes: why do you think iax.conf would be simpler and effective than sip in my case? |
| 16:24.19 | *** join/#asterisk pagec (n=pagec@141.155.63.98) |
| 16:24.22 | mercestes | because it uses one port instead of 50 billion ports |
| 16:24.35 | phearless | is it invalid or dumb ? : |
| 16:24.35 | phearless | (90[1-9]xxxxxxxxxx|900.|4xx) |
| 16:24.40 | mercestes | pagec: welcome back. There are no sip echo cancellers. Check your networking. |
| 16:24.49 | b11d | the only echo i hear on my sip phones, is echo generated on the far end.. |
| 16:24.54 | phearless | 900. for intl calls |
| 16:25.03 | phearless | 90[1-9] for local UK calls |
| 16:25.11 | phearless | 4xx for local ext |
| 16:25.21 | mercestes | funny. |
| 16:25.28 | mercestes | 900 is a very different type of call here. |
| 16:25.31 | mavior | mercestes: but then i have to always use again register with iax? (sorry i haven't experience at all with iax) |
| 16:25.35 | mercestes | ....of course...now that I think about it...those all go to the UK too |
| 16:25.59 | pagec | mercestes: so if i hear echo at the end of a sentance I am speaking on a IAX called bridged to SIP, i should talk to the phone company that terminates the IAX to PSTN then? |
| 16:26.05 | coppice | many SIP phones generate echo, through coupling between the earpiece and mic. good phones echo cancel that, but most phones don't. if you have a handset volume control, try turning it down |
| 16:26.12 | mercestes | mavior: You have to register, but if you can get it to register at all it is guaranteed to work because IAX delivers audio over the same path it uses to register to begin with. |
| 16:26.25 | mercestes | mavior: So if you have SIP registered now, iax would be working right now. |
| 16:26.40 | pagec | coppice: so you know if Polycom is a "good" phone? |
| 16:27.00 | mercestes | Polycom is a great phone. |
| 16:27.34 | coppice | seems most polycoms don't cancel it. if you have a hard of hearing user who turns up the volume, the other end hears echo |
| 16:27.34 | mercestes | pagec: Check your networking. Your system will play your own voice back to you for comfort, if you have massive lag between you and the PBX or the PBX is overloaded, it will play your voice back to you with a long delay. |
| 16:28.07 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 16:30.00 | pagec | mercestes: so if my networking has a long lag time i will hear the tail end of what i am saying echoed back to me? |
| 16:30.21 | mercestes | pagec: that has been my experience |
| 16:30.28 | mercestes | What does mtr say? |
| 16:30.34 | pagec | mercestes: how long is "long"? |
| 16:30.48 | wltjr | what's the recommend gnome softphone these days, last I used one I think it was messing with linophone |
| 16:30.49 | mavior | mercestes: aehmmm....just one thing: i am using sip with my voip providers (with register option for incoming calls) without open any INBOUND udp ports for audio! i have only opened 5060 ( but i can use my voip provider both incoming/outcoming direction even if i do NOT open 5060 udp)! my OUTBOUND policy firewall is permissive by default insted (of course) |
| 16:31.08 | mercestes | what does mtr say? |
| 16:31.38 | mercestes | mavior: ok. |
| 16:31.42 | wunderkin | i have some ip430s where people are hearing themselves, the recording shows a very slight delayed echo, but they are on private t1s within the metro area, very low latency, although they do apparantly have some hardware problems with the phones... we are in the process of returning them.. (still) ugh |
| 16:32.11 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 16:32.25 | mercestes | wunderkin: If they are on T1's and using PRI and not SIP...then it is real echo or it can be |
| 16:33.02 | wunderkin | oh, yes they are on a pri, and using sip internally |
| 16:33.04 | mercestes | now if it's sip over a data t1, then your back to the "not really echo" ordeal again. |
| 16:33.04 | mavior | mercestes: ihih is it not strange? did you say that i have to open audio udo ports to get it works or not? |
| 16:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:33.40 | mercestes | mavior: I never said anything about audio udo ports. I said you should use IAX because it only deals with one port. |
| 16:33.57 | mercestes | and that your routing is going to be a nightmare. |
| 16:35.28 | mavior | mercestes: ok i have only one doubt now: why sip incoming calls won't work if i don't use register ? (even though i've redirected nicely my 5060 udp port - that is the pport where the invite arrives?) |
| 16:36.26 | phearless | <mercestes> funny. |
| 16:36.26 | phearless | <mercestes> 900 is a very different type of call here. |
| 16:36.38 | phearless | 9 is just use for our building, for outgoing calls |
| 16:36.42 | phearless | it is annoying |
| 16:36.43 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:36.53 | mercestes | phearless: I know. 1-900 in the US is usually a phone-sex number. |
| 16:36.59 | phearless | okay |
| 16:37.01 | wunderkin | ip430 -> point to point t1 -> colo -> pri, i don't think everyone has the problem but there are only certain people that make a high volume of calls, of those people i'm not sure if it happens on every call or intermittant... they stopped sending me reports, i tried asking about speakerphone, some of them use headsets, but i believe it happens w/o speakerphone or headset, on some of the recordings their output volume is high and i hear |
| 16:37.12 | wunderkin | their echo on the other side of the recording, faintly |
| 16:37.17 | mercestes | mavior: Because the UDP comes in anywhere between 1002-4000 and 10000-30000. |
| 16:37.35 | wunderkin | i thought if they hear themselves the echo is from the other side or a bad phone |
| 16:37.43 | *** join/#asterisk Virtugon (n=virtugon@beast.dierentuin.com) |
| 16:37.45 | mercestes | mavior: and each sip device attaches to it's own port. |
| 16:38.08 | mercestes | mavior: 5060 on SIP is only for register...nothing more. Just a "hi, I'm here." |
| 16:38.29 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 16:38.31 | mercestes | mavior: It's like walking into a whore-house with no money and no credit cards. Sure you can say "hi" and let them know you are there, but you ain't gettin' no play. |
| 16:38.46 | mercestes | mavior: What's what you have now, all the "hi" you can hand out and no $$$ for the fun stuff. |
| 16:39.06 | mercestes | mavior: $$$ being open and routeable ports. |
| 16:39.06 | JoNate | Telepathy...Thats the way to go... |
| 16:39.16 | mercestes | JoNate: Yes! Finally, someone gets it. |
| 16:39.49 | mercestes | IAX is different tho. IAX is based on Norway. All the chicks are free there. If you can say "hi" you can get laid. You dont' need $$$ or open and routeable ports |
| 16:39.54 | mercestes | IAX is world happiness. |
| 16:40.25 | mavior | mercestes: well so seen that IAX uses a single UDP port 4569, open and redirect it should be enough, isn't it( thank you for the explanation) |
| 16:40.27 | mercestes | other than the one port (your mouth) that you use to say "hi" with. |
| 16:40.38 | mercestes | Mavior: exactly the case. |
| 16:42.42 | mavior | mercestes: now a question that i HAVE to make: and what effectively register=> things do for permit the game ( i mean inc/out working) and to have the nat traversal ? |
| 16:43.30 | mercestes | register => delivers the routable IP address and accompanying port to the server wishing to contact it. |
| 16:44.05 | mercestes | so if youare on 192.168.1.2, and another server wants to contact you, it can't, because 192.168.1.2 isn't routeable. |
| 16:44.07 | mavior | is it too much for explanations, uhm ? |
| 16:44.23 | mavior | sorry i haven't read |
| 16:44.48 | mercestes | So your server goes out the router, and the router assigns it a public IP, 24.2.1.2 and a port, 543885. So your server goes, "I am here, 24.2.1.2:543885" to the other end. |
| 16:45.39 | mercestes | and the other servre goes "ok, 24.2.1.2:543885" and send it back to the router ,the router recognizes port 543885, and tells 192.168.2.2, "That other guy said ok." |
| 16:46.22 | mercestes | *or* You could Vlan. |
| 16:48.27 | mavior | ok so it only tells the "called and registered" that when it receives a call from the "register server" it has to reply to the caller(always the register server)....otherwise it made no sense |
| 16:49.23 | mercestes | Server B wishes to pass Server A a call. |
| 16:49.29 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 16:49.31 | mercestes | Server A registers with Server B. |
| 16:49.53 | mercestes | Registration gives Server B Server A's external IP address, and the port needed to NAT to Server A's internal address...nothing more. |
| 16:50.16 | mercestes | technically...since you will port forward 4569 in both routers and do the nattin gstatically, registration is unnecessary. |
| 16:50.55 | mercestes | otherwise, Server B would need to know what port the router assigned to Server A. |
| 16:51.02 | mavior | why server A don't read server's ip address and port of the request from the server B INVITE on his sip port ? |
| 16:51.32 | mavior | sorry for my qnglish , dunno if i am clear |
| 16:51.36 | mavior | *english |
| 16:52.10 | mavior | i mean : why server A don't read ip and port of server B when it receives the INVITE request ? |
| 16:53.51 | NirS | anyone seen this before on 1.2.14 ? |
| 16:53.51 | NirS | build_tools/make_version_h > include/asterisk/version.h.tmp |
| 16:53.52 | NirS | /bin/sh: build_tools/make_version_h: Permission denied |
| 16:54.07 | supjigatr | Anyone have a solution to get 411 listing submitted? |
| 16:55.33 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 16:55.43 | dhill | rtp debug .. is it okay for ts to be negative? |
| 16:56.38 | dhill | timestamp |
| 16:57.06 | mavior | mercestes: i mean : why server A don't read ip and port of server B when it receives the INVITE request ? ( even though not registered to server b) ? |
| 16:57.32 | *** join/#asterisk cian_ (n=cian@cian.ws) |
| 16:57.35 | wunderkin | supjigatr, the provider responsible for the number needs to list it |
| 16:57.58 | mercestes | mavior: Pick a number between 10,000 and 30,000. |
| 16:58.21 | mavior | ihih done |
| 16:58.32 | mavior | guess |
| 16:58.34 | mercestes | ....you have to tell me what it is... |
| 16:58.50 | mavior | 15000 |
| 16:58.58 | mercestes | wrong. pick a different one. |
| 16:59.04 | *** join/#asterisk ReD-MaN (n=redman@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 16:59.10 | wunderkin | 42 |
| 16:59.19 | mercestes | wunderkin: Omg, you got it! |
| 16:59.26 | mavior | 28525 |
| 16:59.28 | mavior | ahahaha |
| 16:59.28 | wunderkin | zomg |
| 16:59.40 | mercestes | mavior: still wrong. Now let's add a new element to this...hehe |
| 16:59.48 | mercestes | mavior: Try to guess my number..without talkign to me. |
| 17:00.01 | supjigatr | wunderkin: I am the provider but the bellsouth and verizon processes are so tedious and not automated. I have looked at LSSI. |
| 17:00.09 | mercestes | you have to go through some other person....in ...oh, #bondage-toys. |
| 17:00.16 | mercestes | and yo ucan't refer to me by anything other than "that guy." |
| 17:00.43 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:00.43 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:00.50 | mavior | mercestes: so register pass the server A the "good" port for making a connection back to B ? |
| 17:01.03 | mercestes | no. |
| 17:01.13 | mavior | ouch! |
| 17:01.21 | mercestes | you don't have a connectino between Server A and Server B. |
| 17:01.56 | mercestes | server a isn't evne allowed to talk to server b |
| 17:02.11 | *** part/#asterisk UVSoft (n=UVSoft@80.254.48.58) |
| 17:02.23 | mavior | oh ok is the vice-versa, right? |
| 17:02.24 | JoNate | TELEPATHY! |
| 17:02.29 | mercestes | you only have registration because you did a static nat for port 5060. |
| 17:02.37 | mercestes | now yo uhave to do static natting for 10,000 through 30,000 |
| 17:03.08 | mercestes | *or* |
| 17:03.09 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 17:03.17 | mercestes | ....you could do static natting for 4569 |
| 17:03.32 | *** join/#asterisk chiardon (n=jorge@200.71.58.39) |
| 17:03.36 | chiardon | Hello |
| 17:06.01 | wunderkin | supjigatr, verizon or bellsouth, whoever it is that is the resporg, needs to list it then, have fun |
| 17:07.39 | mavior | mercestes: register => delivers the routable IP address and accompanying port to the server wishing to contact it. ( So if i redirect all the ports from 10000 to 30000 to my internal server address , register would be not necessary anymore ? ) |
| 17:09.31 | b11d | telepathy? is that a voip company? |
| 17:09.33 | b11d | :) |
| 17:09.47 | mercestes | b11d: ya. |
| 17:09.54 | b11d | haha |
| 17:09.59 | supjigatr | wunderkin: I am the resporg. |
| 17:10.07 | mercestes | mavior: what part of static nat 1 port v/s static nat 30,001 ports are we not syncing up on here? |
| 17:10.18 | supjigatr | I have a solution now but its very hard to do and not an automated process. |
| 17:10.36 | mavior | mercestes: ?i don't understand |
| 17:10.40 | mercestes | mavior: I can see that. |
| 17:10.49 | mercestes | mavior: Can't you just do a VLAN in the routers? |
| 17:12.26 | mavior | mercestes: no no I will try with iax....it is only to understand why incoming calls without a register now work |
| 17:12.38 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 17:12.41 | mavior | s/now/not/ |
| 17:12.43 | mercestes | mavior: magic. |
| 17:13.03 | mavior | ihih |
| 17:13.28 | mercestes | mavior: you are not understanding a basic networking fundamental. |
| 17:13.33 | *** part/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 17:13.43 | mercestes | What is the IP of server A? |
| 17:14.04 | mavior | my outgoing calls with my voip provider work oly because it is not natted nad have a static ip address ? |
| 17:14.18 | mavior | s/oly/only/ |
| 17:14.42 | mercestes | mavior: If your voip provider is not natted and has a static IP address then ....yes. |
| 17:14.53 | mavior | i use voipcheap |
| 17:15.18 | mercestes | not *really* but...for the sake of clarity, yes |
| 17:17.49 | mercestes | Should I go into *why* it works that way? |
| 17:18.11 | mavior | mercestes: ok...the registered server tells the other server the ip and the port where it can receives incoming call from it |
| 17:18.34 | mavior | ? |
| 17:18.50 | mercestes | ....ok, let's try it this way. |
| 17:19.12 | mavior | ok.. i'm reading :) |
| 17:19.33 | mercestes | NO nat <=> No Nat Works. Nat => No Nat Works. No Nat <=> Nat Works. Nat <=> Nat does *NOT* work. |
| 17:20.15 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 17:20.30 | mercestes | but you can *force* it to work using static NAT tables....or by using a VLan |
| 17:21.19 | martineyles | bye |
| 17:21.21 | *** part/#asterisk martineyles (n=martiney@adsl-w-234.as15758.net) |
| 17:22.00 | mercestes | so your setup works with your voip provider, because you have NAT <=> No Nat. That works. |
| 17:22.02 | mavior | No Nat => Nat Works, even without register ? |
| 17:22.14 | mercestes | your Nat <=> Nat is *not* working because that does not work, and you are not forcing it properly. |
| 17:22.31 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 17:24.32 | mavior | mercestes: No Nat => Nat Works, even without register ? |
| 17:24.51 | wltjr | I need to test a remote * box, making calls, what's the least painful route? sip softphone to remote box? connect my * to the other * via iax? remote iax softphone? |
| 17:25.40 | mercestes | mavior: .....no, you have to register. |
| 17:25.43 | wltjr | sip seems like it might be a bit painful with nat and firewalling on both ends |
| 17:25.47 | mercestes | by "works" I mean, it's possible without major voodoo |
| 17:27.21 | mavior | mercestes: ok i left a piece :"Registration gives Server B Server A's external IP address, and the port needed to NAT to Server A's internal address...nothing more." so it' s a way to know the route to the natted server |
| 17:29.35 | mavior | mercestes: So this one "No Nat <=> Nat Works" is wrong because only Nat => No Nat Works. |
| 17:30.39 | mercestes | No nat <= > nat works, but the nated server would have to register to the no nat server first. |
| 17:30.52 | mavior | ihhi ok |
| 17:31.17 | mavior | so,register=>, it' s a way to know the route to the natted server,isn't it? |
| 17:31.51 | mercestes | yes. |
| 17:31.58 | mercestes | but you can't contact a nat'd server without a registration. |
| 17:32.19 | mercestes | so therefore, you cannot register to a nat'd server because that's what the registration does to begin wtih, establish a communication path. |
| 17:32.25 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:32.27 | mercestes | so two nat'd servers can never register with each other. |
| 17:32.36 | mercestes | again...not without major voodoo (ie: static routing/vlan) |
| 17:33.15 | *** join/#asterisk knobenheimer (n=knoben@c-71-205-184-144.hsd1.mi.comcast.net) |
| 17:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:33.45 | mavior | port redirecting of the port 5060 (the port where register=> works) to the nat'd server should do the trick? |
| 17:34.49 | mercestes | mavior: no. |
| 17:35.02 | mavior | why not? |
| 17:35.20 | mercestes | For a small consulting fee...I can tell you exactly why not. |
| 17:35.31 | mavior | how much ? :P |
| 17:35.35 | mercestes | actually, I've already told you why not but I don't wish to tell you again for free. |
| 17:35.56 | mavior | let me think |
| 17:36.25 | mercestes | and....I dun wanna spam * any more with this networking stuff. |
| 17:36.34 | mercestes | Manx said it before, yoru setup is *BAD*. |
| 17:36.53 | mercestes | You need static routing...Vlans, or a static IP. Please pick one that suits your technical expertise/consultant budget. |
| 17:36.58 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 17:37.15 | *** join/#asterisk drone1 (n=drone1@tech.quentris.be) |
| 17:37.23 | drone1 | evening everyone |
| 17:39.45 | drone1 | anybody knows rtsp? |
| 17:42.35 | *** join/#asterisk shinux__ (n=shinux@196.220.25.73) |
| 17:43.55 | sevard | mercestes: are you saying there are free whores in Norway? |
| 17:45.50 | *** part/#asterisk sergee (n=opera@195.94.224.197) |
| 17:48.50 | *** join/#asterisk Calisto (n=sod@82-47-200-204.stb.ubr04.shef.blueyonder.co.uk) |
| 17:50.46 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:51.22 | Calisto | can anyone advise on a chan_cellphone issue i'm having |
| 17:54.32 | docelmo | probably not since I dont know of any chan_cellphone |
| 17:54.54 | *** join/#asterisk letomuaddib (n=lesgsod@bas5-montrealak-1128552993.dsl.bell.ca) |
| 17:56.55 | *** join/#asterisk jart (n=user@ool-43551046.dyn.optonline.net) |
| 18:06.34 | Dr-Linux | damnit |
| 18:06.55 | Dr-Linux | anybody please get a chance at >> http://phpfi.com/199232 |
| 18:07.21 | Dr-Linux | why these warnings appear |
| 18:07.33 | Dr-Linux | Jan 31 10:04:43 WARNING[25129]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 44 scheduled tasks all at once |
| 18:07.34 | Dr-Linux | Jan 31 10:04:45 WARNING[25129]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 82 scheduled tasks all at once |
| 18:12.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:13.15 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 18:14.06 | *** join/#asterisk qdk (n=qdk@0xc213c3df.inet.dsl.telianet.dk) |
| 18:14.58 | Dr-Linux | any clue? |
| 18:20.33 | *** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net) |
| 18:20.43 | Bobthehunter | whats callerd(name_) length rfc ? |
| 18:21.54 | SplasPood | whats the easiest way to verify a callerid name mapping as far as the general public is concerned without just calling a POTS line with CID+Name service? |
| 18:22.45 | Bobthehunter | yes |
| 18:22.59 | Bobthehunter | but im passing something and the zap not passing name it seems |
| 18:23.30 | Bobthehunter | Display (len=14) Charset: 31 [ Domaine Honda ] |
| 18:23.30 | Bobthehunter | > [6c 0c 21 81 35 31 34 36 34 35 36 37 30 30] |
| 18:23.37 | Bobthehunter | but.. its not passing |
| 18:24.11 | Bobthehunter | Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) |
| 18:24.19 | Bobthehunter | <PROTECTED> |
| 18:24.23 | Bobthehunter | but shows only number |
| 18:31.35 | SplasPood | anyone with a standard POTS line +CID Name service that I can perform a test call to? |
| 18:31.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:31.37 | Bobthehunter | yes |
| 18:31.40 | SplasPood | Bobthehunter: yea? |
| 18:32.38 | *** join/#asterisk hardwire (n=hardwire@rdbck-4746.wasilla.mtaonline.net) |
| 18:32.42 | hardwire | mudafuka |
| 18:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:34.13 | Dr-Linux | any clue? |
| 18:34.14 | *** join/#asterisk Tako-san (n=sysadmin@24.108.162.254) |
| 18:34.20 | *** part/#asterisk Calisto (n=sod@82-47-200-204.stb.ubr04.shef.blueyonder.co.uk) |
| 18:37.01 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 18:37.25 | *** join/#asterisk Tako-san (n=sysadmin@24.108.162.254) |
| 18:38.07 | b11d | telepathy? is that a voip company? |
| 18:38.25 | JoNate | Yes it is... |
| 18:38.30 | b11d | :) |
| 18:38.33 | b11d | wow.. neat |
| 18:38.34 | b11d | haha |
| 18:38.51 | Dr-Linux | anybody please get a chance at >> http://phpfi.com/199232 |
| 18:38.56 | Dr-Linux | what's wrong |
| 18:39.03 | Dr-Linux | zaptel timing issue? |
| 18:39.09 | b11d | i dont know.. I havent used iax yet.. |
| 18:39.58 | hypnox | got any other symptoms? |
| 18:39.58 | *** join/#asterisk x-ip (n=x@host33.201-253-1.telecom.net.ar) |
| 18:40.00 | Dr-Linux | b11d: same answered from many guys, |
| 18:40.08 | Dr-Linux | b11d: looks like it's a bug or something |
| 18:40.17 | b11d | possibly.. could be anything |
| 18:40.38 | Dr-Linux | maybe i should upgrade |
| 18:40.53 | b11d | someone mentions it may be caused by slow DNS lookups.. |
| 18:41.02 | b11d | i assume you've already googled that error and looked at the top results already |
| 18:41.08 | x-ip | hi, i cant get down the red alarm from a motorolla wildcard x100p, any sugestion ? |
| 18:41.32 | b11d | the author also states that it may be due to an excessively high workload on the PBX system |
| 18:41.36 | b11d | what kind of load are you seeing/ |
| 18:41.37 | b11d | ? |
| 18:41.48 | b11d | x-ip.. check the cables? |
| 18:41.54 | x-ip | done |
| 18:41.58 | b11d | replace them? :) |
| 18:42.03 | Dr-Linux | b11d: i can't see any load |
| 18:42.06 | b11d | uhh |
| 18:42.07 | x-ip | but they are working :| |
| 18:42.09 | b11d | there must be *some* load |
| 18:42.25 | Dr-Linux | b11d: what's you way to check the load status? |
| 18:43.30 | b11d | pastebin the results of the command 'uptime' |
| 18:43.31 | b11d | at the command line.. not the asterisk CLI |
| 18:43.31 | Bobthehunter | oh |
| 18:43.31 | Bobthehunter | so you cant pass name to ZAP ? |
| 18:43.33 | b11d | x-ip.. how can they be working when you have a red alarm? |
| 18:43.56 | x-ip | they are not working |
| 18:43.57 | hypnox | Dr-Linux it sometimes indicates network problems |
| 18:44.08 | b11d | [12:43] <x-ip> but they are working :| |
| 18:44.09 | b11d | ok |
| 18:44.22 | Dr-Linux | hypnox: yes, but it's only with IAX trunk :S |
| 18:44.27 | hypnox | it just means that something is slowing down iax2 |
| 18:44.29 | x-ip | well , the cables are working with a normal analog phone |
| 18:44.31 | Dr-Linux | hypnox: what things should i check |
| 18:44.33 | hypnox | yeah only iax2 has that message |
| 18:44.44 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 18:44.45 | x-ip | but when i put theirs in the motorolla card, i stiil get a red alarm |
| 18:44.46 | b11d | wheres your load averages? |
| 18:44.47 | hypnox | have you any problems with call quality etc? |
| 18:44.52 | b11d | do you have interrupt issues? |
| 18:44.54 | Dr-Linux | hypnox: i was using domain name for other asterisk server, but i changed it to IP address, but still same warnings |
| 18:45.07 | b11d | WHAT IS YOUR LOAD AVERAGE |
| 18:45.10 | b11d | maybe you're not seeing that |
| 18:45.21 | b11d | meh |
| 18:45.31 | Bobthehunter | ?? |
| 18:46.23 | Dr-Linux | hypnox: well, basically i was using SIP trunk between my 2 asterisk server and every thing was fine, due to low bandwidth i setup IAX trunk between my servers, and i'm getting those warnings, but didn't recieve any complaint about voice quality yet |
| 18:46.26 | Dr-Linux | b11d: |
| 18:46.26 | x-ip | it could be that the tone that the local central telephony is useless ? i try to explain: [pstn] ==> [local central telephony] --> telephony lines --> [motorolla wildcard x100p at trixbox ] |
| 18:46.27 | Dr-Linux | uptime |
| 18:46.27 | Dr-Linux | <PROTECTED> |
| 18:46.35 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 18:46.49 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 18:46.50 | b11d | trixbox support is in #freepbx |
| 18:46.56 | x-ip | ouch |
| 18:47.05 | x-ip | sorry |
| 18:47.07 | b11d | cya x-ip :) |
| 18:47.13 | *** part/#asterisk x-ip (n=x@host33.201-253-1.telecom.net.ar) |
| 18:47.14 | Dr-Linux | b11d: i think that's not a load |
| 18:47.26 | Dr-Linux | load average: 0.01, 0.07, 0.04 |
| 18:47.26 | b11d | yeah well i dont know what your problem is.. i dont use iax |
| 18:47.28 | Bobthehunter | im just asking..can you passout Callerid(NAME) on a zap to PRI T1 |
| 18:47.30 | Bobthehunter | unlocked T1 |
| 18:47.36 | Dr-Linux | ok |
| 18:47.38 | b11d | Bob.. yes.. I do. |
| 18:47.41 | b11d | it just happens.. |
| 18:47.48 | b11d | theres nothing special to do |
| 18:47.51 | Bobthehunter | hmm then why |
| 18:47.52 | Dr-Linux | hypnox: i guess it's zaptel timing issue :S |
| 18:48.07 | b11d | your telco is likely stripping it off |
| 18:48.44 | b11d | Bob.. do you do a "pri intense debug span 1" |
| 18:48.54 | b11d | and then, do you see your CID info being passed out on the PRI? |
| 18:48.56 | Bobthehunter | Display (len=17) Charset: 31 [BOB ]> [6c 0c 21 81 34 35 30 34 33 37 38 30 30 30]> Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)> Presentation: Presentation permitted, user number passed network screening (1) '4504371234' ] |
| 18:49.10 | Bobthehunter | modded the values of the acutal number and name but its same stuff |
| 18:49.12 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 18:49.15 | b11d | sure |
| 18:49.34 | b11d | yeah your telco is probably stripping it all off |
| 18:49.38 | b11d | mine does.. grrr |
| 18:50.44 | Bobthehunter | does grr ? |
| 18:50.54 | mavior | mercestes: just to let you know...damn voodoo: actually i'mregistering to a nat'd server from a nat'd server and sip to sip is working :X |
| 18:51.36 | mavior | register=> is more powerful than you expect |
| 18:52.58 | b11d | in what way? |
| 18:54.15 | sevard | It can register to your mom |
| 18:54.21 | b11d | yeah, who cant? |
| 18:54.26 | sevard | what other piece of software can do that? |
| 18:54.28 | sevard | oh excellent |
| 18:54.29 | b11d | most.. |
| 18:54.30 | b11d | :) |
| 18:54.44 | b11d | Microsoft Live! has a "b11ds mom scheduler" |
| 18:54.57 | b11d | dont cut her.. geeze.. |
| 18:55.01 | b11d | last time that happened.. ugh |
| 18:55.06 | sevard | blarh. |
| 18:55.13 | b11d | yeah you're going to get nowhere :) |
| 18:55.54 | b11d | sweet. 8 more days until this Vista box gets locked up.. |
| 18:56.06 | b11d | cant seem to activate/register whatever the fuck MS wants of us these days |
| 18:56.17 | J4k3 | no great loss. |
| 18:56.21 | b11d | yeah, really. |
| 18:56.22 | Qwell[] | b11d: blood |
| 18:56.30 | b11d | hehe |
| 18:56.30 | Qwell[] | you're supposed to send it via certified letter |
| 18:56.33 | sevard | + soul |
| 18:56.42 | Qwell[] | sevard: soul is taken during the install process |
| 18:56.46 | Qwell[] | it's automatic |
| 18:56.46 | b11d | i already sent my first three children to them.. |
| 18:56.57 | Qwell[] | b11d: for XP... You need to pay the upgrade fee |
| 18:57.12 | sevard | two of those children were mine! :'( |
| 18:57.18 | b11d | haha.. otherwise they'll send some hired goons to come break my thumbs |
| 18:57.35 | Qwell[] | b11d: What, did you upgrade to home basic or something? |
| 18:57.38 | Qwell[] | thumbs...pfft |
| 18:57.46 | b11d | Business Edition.. |
| 18:57.50 | b11d | its ghey |
| 18:58.02 | b11d | its going to be a LONG time before i start rolling this out across the campus.. |
| 18:58.08 | sevard | you only need thumbs for your spacebar, you can use your palms for that. silly hired goons. |
| 18:58.25 | b11d | i remapped the spacebar to the caps lock key.. i use space as an enter now. |
| 18:58.33 | mercestes | b11d: I have my copy of Vista Business coming to me now. |
| 18:58.37 | mercestes | :) |
| 18:58.44 | b11d | yeah.. that :) will be :| and then :( before you know it |
| 18:59.03 | mercestes | Oh! Oh! Oh! I bought Secretary last night!! |
| 18:59.07 | b11d | nice! |
| 18:59.08 | mercestes | ...err...wrong channel |
| 18:59.11 | b11d | where'd you find that |
| 18:59.24 | mercestes | Best Buy |
| 18:59.28 | mercestes | go figure. |
| 19:01.18 | mercestes | mavior: Congratz |
| 19:01.46 | Bobthehunter | secretary ? |
| 19:01.51 | Bobthehunter | you got a russian bride ? |
| 19:02.19 | b11d | Russian women are hot as hell until they get married.. |
| 19:02.20 | mercestes | Bobthehunter: asian. Why? |
| 19:02.27 | b11d | at least, thats what my grampa says |
| 19:02.47 | J4k3 | yeah, then you meet nina reiser. |
| 19:03.35 | b11d | i dont know who that is |
| 19:03.44 | mercestes | Hans Reiser's dead wife. |
| 19:03.45 | J4k3 | hans reiser's wife |
| 19:03.46 | J4k3 | ex-wife |
| 19:04.00 | mercestes | dead ex-wife |
| 19:04.02 | b11d | oh |
| 19:04.04 | J4k3 | http://en.wikipedia.org/wiki/Hans_reiser |
| 19:04.08 | J4k3 | mercestes: low chance of that |
| 19:04.09 | b11d | yeah i know of him |
| 19:04.14 | b11d | ReiserFS and all that |
| 19:04.30 | SomeOne1 | Qwell: whats your story |
| 19:04.31 | mercestes | J4k3: low chance of what?? |
| 19:04.38 | mercestes | that she's dead? |
| 19:04.40 | SomeOne1 | how long have you been an op at this channel? |
| 19:04.42 | b11d | that guy looks like he should only work on commodore 64's in bad russian hacker movies.. |
| 19:04.47 | J4k3 | mercestes: him killing his wife. Theres a random lack of a body and there wasn't much blood recovered |
| 19:04.48 | Qwell[] | SomeOne1: dunno, 6 months? |
| 19:04.52 | SomeOne1 | heh |
| 19:05.00 | SomeOne1 | howd you rise through the ranks? |
| 19:05.01 | mercestes | j4k3: .......yea, you think OJ is innocent too? |
| 19:05.16 | Qwell[] | SomeOne1: /whois Qwell[] |
| 19:05.16 | b11d | he gives great head |
| 19:05.16 | b11d | oh |
| 19:05.16 | b11d | :P |
| 19:05.16 | mercestes | He does. |
| 19:05.26 | SomeOne1 | heh |
| 19:05.30 | SomeOne1 | thats a lot of channels |
| 19:05.39 | Qwell[] | SomeOne1: plus 2 on efnet ;/ |
| 19:05.47 | mercestes | Ok, gee, wife's gone, only a *little* of her blood was recovered from his home, his clothes, his car....yea, definately innocent. |
| 19:05.48 | SomeOne1 | do you feel powerful? |
| 19:05.49 | J4k3 | mercestes: no, but there was a LOT of evidence against OJ... the evidence against reiser is pretty damned worthless, which is why the judge is pissed off because now his kids won't bother to show up to testify |
| 19:05.49 | mercestes | =/ |
| 19:05.50 | b11d | i figured it had more to do with: qwell@pdpc/sponsor/digium/ |
| 19:05.52 | Qwell[] | I only have so many alt-keys |
| 19:06.07 | J4k3 | from this point it sounds like unless something changes reiser has a good chance of getting released. |
| 19:06.08 | SomeOne1 | heh |
| 19:06.22 | SomeOne1 | Qwell: share the power and prestige man |
| 19:06.24 | SomeOne1 | jk |
| 19:06.49 | b11d | there are the perfect number of ops in here.. |
| 19:06.51 | J4k3 | plus... if nina reiser was on the up-n-up she wouldn't have been trying to get her kids russian citizenship without his knowledge. |
| 19:06.55 | J4k3 | the whole thing is... sketchy |
| 19:07.50 | J4k3 | personally I think if there was a federal law against dramatic women, nina reiser would be in prison... at least from *everything* I've read on the net from people that knew/knows them both. |
| 19:08.49 | mercestes | Yea, I believe everything I read on the Internet too |
| 19:09.36 | J4k3 | well, I don't believe people are automatically guilty of things like murder. |
| 19:09.40 | J4k3 | especially when it involves dramatic russians. |
| 19:10.14 | Bobthehunter | russian brides dot com lol you can marry on for 10k$ |
| 19:10.23 | Bobthehunter | or brides.ru i thnk |
| 19:10.31 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 19:10.40 | hardwire | there are stricter laws nowadays for russian bridges |
| 19:10.40 | hardwire | heh |
| 19:10.43 | J4k3 | pft... if you want a prostitute, try a street corner. |
| 19:10.55 | hardwire | including background checks on the buyer to cut down on human trafficing |
| 19:10.56 | mercestes | j4k3: I am not entirely clear on how her being a dramatic russian makes it *less* likely that her ex-husband murdered her. |
| 19:11.17 | J4k3 | mercestes: she could easily just have went back to russia. |
| 19:11.32 | mercestes | j4k3: That explains alot. |
| 19:11.40 | SomeOne1 | how much does business insurance cost |
| 19:11.47 | hardwire | mercestes: dramatic ones build up to it, there was no plot. |
| 19:11.48 | hardwire | :) |
| 19:12.22 | mercestes | hardwire: Yea, the dramatic ones tend to just vanish.....with the kids still easily locatable.... |
| 19:12.22 | J4k3 | its the body and the evidence. Theres a general lack of both. |
| 19:12.27 | J4k3 | mercestes: easily locatable... IN RUSSIA. |
| 19:12.30 | hardwire | I have no idea what any of you are talking about |
| 19:12.41 | hardwire | and quite frankly |
| 19:12.44 | hardwire | I don't give a damn |
| 19:12.47 | Jason99 | If I have quality=15000, the server sends a OPTIONS packet every 15 seconds. Is there a way to set the number of seconds before the peer is considented unreachable if no response is received? |
| 19:12.58 | Jason99 | qualify=15000 rather |
| 19:13.03 | J4k3 | hence why the judge is pissed off... one of the kids was supposed to be in court last week (or week before last?) and the judge was told the child couldn't show up because the russian grandparents wouldn't release the kid. |
| 19:13.03 | Corydon-w | The real question is, is the kid traumatized because his mother is dead or because he's now living without his father in Russia? |
| 19:13.11 | hardwire | Jason99: not independantly |
| 19:13.25 | hardwire | if qualify misses one 15 second check it shuts down the trunk |
| 19:13.31 | J4k3 | Corydon-w: or the kid's living with his mom and grandma in russia right now, and its all hooey. |
| 19:13.47 | Corydon-w | J4k3: possible |
| 19:13.52 | mercestes | j4k3: I agree..and taht whole walking on the moon thing? totally fake. |
| 19:13.57 | mercestes | space travel is a hoax. |
| 19:14.09 | J4k3 | mercestes: are you a murderer? I'm not and I assume most others aren't. |
| 19:14.14 | J4k3 | especially until theres evidence |
| 19:14.24 | Jason99 | hardwire: Are you sure thats the way it works? It seems like the server considers it unreachable within seconds after the OPTIONS packet is sent |
| 19:14.30 | J4k3 | and unless theres a LOT of evidence that the SF Meteor and wikipedia aren't listing... theres no case against hans reiser. |
| 19:14.43 | Corydon-w | There's evidence, just not sufficient to prove that anyone is dead |
| 19:14.49 | hardwire | Jason99: you should fix that |
| 19:14.50 | J4k3 | exactly |
| 19:15.00 | J4k3 | theres some blood, but not a big ol' puddle of it. |
| 19:15.06 | Jason99 | hardwire: I am trying to.. this is why I'm here |
| 19:15.25 | mercestes | http://www.ufos-aliens.co.uk/cosmicapollo.html |
| 19:15.28 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-54-119.red.bezeqint.net) |
| 19:15.38 | J4k3 | I mean shit, my own blood is all over my car... papercuts while doing the deposit... shreading my hand on some bad galvanization on towers I'm climbing and slinging blood around... that shit happens |
| 19:15.53 | hardwire | Jason99: you may have something overriding your qualify |
| 19:16.07 | hardwire | use sip/iax show ... ... whatever your trunk is and wherever.. |
| 19:16.24 | hardwire | it shuold show you the exact qualify variable for that peer/friend/user/trunk/grandma |
| 19:17.09 | hardwire | and of course make sure the other server you are connected with isn't using qualify at a lower time interval |
| 19:17.31 | hardwire | you know I haven't used asterisk in over a year? |
| 19:17.36 | Jason99 | hardwire: i will check it out.. however I believe its working correctly.. the problem I'm having is that I have about 400 SIP peers on the server and when I do a reload, half the peers say they are unreachable.. |
| 19:17.40 | hardwire | I kinda feel bad giving advice without knowing my shit |
| 19:18.01 | hardwire | well |
| 19:18.01 | hardwire | yeh |
| 19:18.08 | hardwire | sucks doesn't it |
| 19:18.23 | hardwire | you should have said that in the first place :) |
| 19:18.45 | hardwire | and I don't know the answer to that one either :) |
| 19:19.59 | jart | hardwire: are you having a conversation with yourself? |
| 19:21.12 | tzanger | jmls: you got it into 1.4, congrats :-) |
| 19:22.25 | hardwire | jart: you new here? of course I am talking to myself. |
| 19:22.52 | jart | hardwire: i am a newbie |
| 19:23.00 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 19:23.08 | Jason99 | Anyone else here running 400+ sip peers on one server? |
| 19:23.32 | jart | hardwire: are you a newbie? |
| 19:23.35 | hardwire | oh what.. just because I don't know the answer means I don't know the answer?! |
| 19:23.50 | hardwire | jart: I am neither a newbie or an oldbie |
| 19:24.12 | hardwire | just a bie |
| 19:25.07 | Qwell[] | damn midbies |
| 19:25.08 | Qwell[] | :p |
| 19:25.18 | *** join/#asterisk MikeB (n=chatzill@89.192.14.133) |
| 19:25.56 | Jason99 | is there a way to make asterisk send keep alives without actually qualifying? In other words, if I dont get a response, I dont want to mark the peer as down |
| 19:26.04 | MikeB | Hello, Anyone know about ENUMLOOKUP, its seems to have changed in 1.4 and is not I think incorrectlt returing IAX2:blah and not IAX2/blah |
| 19:27.36 | hardwire | Jason99: I know an answer |
| 19:27.44 | hardwire | but it requires the sip phones to check in more often |
| 19:27.55 | b11d | set registration to 30 seconds? |
| 19:28.02 | hardwire | bingo |
| 19:28.11 | mercestes | Hey, this space travel hoax site is actually pretty cool |
| 19:28.12 | mercestes | lol |
| 19:28.18 | b11d | the best site ever: www.timecube.net |
| 19:28.20 | b11d | go there NOW |
| 19:28.31 | hardwire | pushy! |
| 19:28.34 | b11d | oops |
| 19:28.36 | b11d | .com |
| 19:28.36 | Bobthehunter | anyway to park a call but for gorups ? |
| 19:28.36 | MikeB | It seems like the correct DNS entry is IAX2:user@host and that the old 1.0 ENUMLOOKUP applictation would return IAX2/user@host in ENUM thus changing the : for a / to go in the dial command but 1.4 does not change the : to / |
| 19:28.37 | b11d | www.timecube.com |
| 19:28.42 | b11d | this guy is on the ball |
| 19:28.43 | b11d | :) |
| 19:28.43 | Bobthehunter | so group1 doesnt have pickup for gorup 2 ? |
| 19:30.01 | jart | b11d: as a time cubist, i am offended by your comment |
| 19:30.46 | MikeB | no ideas then ? |
| 19:31.38 | Bobthehunter | like is valet parking in asteirsk 1.2.14 |
| 19:31.40 | b11d | :) |
| 19:31.52 | b11d | jart.. allow me to opposite apologize |
| 19:32.17 | MikeB | I have checked the RFC and it should be : in the dns |
| 19:33.08 | jart | b11d: apology accepted you dog brain singularity worshipping human :) |
| 19:33.13 | b11d | hahahaha |
| 19:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:33.38 | b11d | let us rise above singulatity.. with the clarity of cubist thought we can transend gender |
| 19:33.55 | b11d | is that site supposed to be a joke, or what |
| 19:34.14 | jart | b11d: did you hear about how the time cube guy lectured a body of MIT students? |
| 19:34.51 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 19:35.09 | Bobthehunter | ?? |
| 19:35.28 | mavior | mercestes: did you read? |
| 19:36.24 | jart | b11d: youtube "gene ray" |
| 19:36.35 | mercestes | mavior: Abuot the time cube? yea. I'm sold. space travel is a hoax too |
| 19:36.49 | mercestes | I'm a believer. |
| 19:36.53 | mercestes | bu tnot so much |
| 19:37.49 | Hmmhesays | um |
| 19:37.53 | *** join/#asterisk supers (i=supers@Sia.AnimeNfo.com) |
| 19:40.01 | *** join/#asterisk Gankhuu (n=gankhuu@72-166-51-162.dia.static.qwest.net) |
| 19:40.27 | *** join/#asterisk angom (n=angom@red-corp-201.143.59.181.telnor.net) |
| 19:40.48 | MikeB | so noone knows about ENUMLOOKUP |
| 19:41.41 | Bobthehunter | so ifear valetparking is not applied to current branches |
| 19:42.10 | mavior | mercestes: no, i'm talking about the 2 natted servers that register each other |
| 19:42.33 | b11d | what? |
| 19:42.33 | b11d | jart |
| 19:42.38 | b11d | he lectured at MIT? |
| 19:42.45 | b11d | gene ray eh.. im checking that out :0 |
| 19:43.27 | b11d | ohh man, i want to see his documentary "Above God' |
| 19:43.34 | b11d | it's probably as good as anythiing David Icke put out |
| 19:44.03 | *** join/#asterisk [shodan] (n=shodan@ip125.96-113-216.pppoe1.joliette.intermonde.net) |
| 19:47.15 | in-pt | please anyone help me about skinny channels of asterisk-1.4.0 |
| 19:47.44 | in-pt | i had registered the skinny phone correctly but all the tiime i am getting some weird error logs on asterisk cli |
| 19:47.45 | in-pt | <PROTECTED> |
| 19:47.46 | in-pt | [Jan 31 19:53:44] ERROR[3825]: chan_skinny.c:2875 handle_register_message: Rejecting Device SEP0004C1879775: Device not found |
| 19:47.46 | in-pt | [Jan 31 19:53:44] WARNING[3825]: chan_skinny.c:3993 handle_message: Client sent message #2 without first registering. |
| 19:47.46 | in-pt | [Jan 31 19:53:44] WARNING[3825]: chan_skinny.c:4172 get_input: Skinny Client sent less data than expected. Expected 4 but got 0. |
| 19:47.48 | in-pt | [Jan 31 19:53:44] NOTICE[3825]: chan_skinny.c:4260 skinny_session: Skinny Session returned: Success |
| 19:47.50 | in-pt | [ |
| 19:48.06 | in-pt | ohh its a mistake |
| 19:48.19 | in-pt | i was suppose to paste the pb |
| 19:48.29 | in-pt | http://pastebin.ca/334655 |
| 19:48.41 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 19:50.50 | jart | i think it's time for a career change |
| 19:51.04 | *** join/#asterisk s1gny|wrk (n=s1gny@p54917445.dip.t-dialin.net) |
| 19:51.25 | *** part/#asterisk s1gny|wrk (n=s1gny@p54917445.dip.t-dialin.net) |
| 19:51.27 | b11d | same here |
| 19:51.29 | b11d | lets CUBE THE WORLD |
| 19:51.44 | *** join/#asterisk RoyK (n=roy@ti211310a080-8125.bb.online.no) |
| 19:53.00 | jart | i'm tired of asterisk |
| 19:53.12 | jart | so tired... tired of listening to gossip |
| 19:53.42 | mercestes | mavior: oh, congratz |
| 19:53.59 | b11d | well get a shack in the woods.. |
| 19:54.24 | jart | i can walden it up |
| 19:54.37 | b11d | im down |
| 19:54.54 | b11d | you can be theareau (i know thats wrong) and I'll be kazinski (probably wrong too) |
| 19:55.30 | jart | oh my goddess |
| 19:56.27 | b11d | or we can swap.. if you got the unibomber glasses already, im half nuts.. |
| 19:56.28 | mavior | mercestes: i would just let you know that a natted server can register to a natted server if port is redirected |
| 19:56.49 | mavior | mercestes: read in yours a bit of sarcasm |
| 20:00.01 | Bobthehunter | so wehre can i get valetparking current ? |
| 20:00.39 | mercestes | mavior: really? I woul dhave never guessed. |
| 20:01.13 | mavior | mercestes: ah. ah. ah. |
| 20:03.26 | hardwire | jart: don't spy |
| 20:03.44 | hardwire | Jason99: did that work? |
| 20:04.07 | SomeOne1 | can asterisk receive faxes? |
| 20:04.35 | hardwire | Jason99: setting the registration on sip phones to say every 10-15 seconds |
| 20:04.43 | SomeOne1 | like fax over SIP |
| 20:04.47 | SomeOne1 | and save it in a PDF or something |
| 20:04.54 | hardwire | you aren't alone in this question |
| 20:04.58 | hardwire | check out iaxmodem |
| 20:05.21 | hardwire | you should have been here yesterday, its probably all in the IRC logs somewhere online |
| 20:05.22 | hardwire | heh |
| 20:05.24 | MikeB | SomeOne1: spand can receive from zap channels im not sure about sip. |
| 20:05.29 | MikeB | rxfax also |
| 20:05.41 | MikeB | www.softswitch.org I think |
| 20:05.56 | *** join/#asterisk mafkees (i=michiel@82.103.136.139) |
| 20:05.59 | mafkees | heya all |
| 20:06.08 | MikeB | hey |
| 20:06.31 | mafkees | tzafrir_laptop: I think this is more conveinient then mail ;) |
| 20:06.50 | mafkees | or tzafrir ;) |
| 20:06.55 | tzafrir_laptop | here |
| 20:07.01 | SomeOne1 | is asterisk better or openPBX? |
| 20:07.08 | hardwire | yes |
| 20:07.15 | mafkees | I was wondering if the from-hell.eu domain was ok |
| 20:07.17 | hardwire | by this much |
| 20:07.26 | mafkees | I like it, but not sure if it's ok for the users |
| 20:07.35 | tzafrir_laptop | sure it is |
| 20:07.41 | mafkees | cool cool |
| 20:07.54 | tzafrir_laptop | SomeOne1, this is #asterisk, so the answer is "Asterisk". |
| 20:08.10 | tzafrir_laptop | SomeOne1, on #openpbx the answer is "OpenPBX.org" |
| 20:08.21 | mafkees | OpenPBX, OpenNTPD, OpenSSH, OpenBSD |
| 20:08.25 | mafkees | gheh |
| 20:08.36 | tzafrir_laptop | openoffice |
| 20:08.40 | tzafrir_laptop | openview |
| 20:08.49 | mafkees | openbgpd |
| 20:08.56 | mafkees | opencvs |
| 20:10.28 | mafkees | hhmm |
| 20:10.38 | mafkees | anyone here what happened with sergio ? |
| 20:10.47 | mercestes | SomeOne1: It's ahrd to say. OpenPBX is just asterisk with the source modified a little bit to screw a bunch of shit up to make it largely unusable and busted, based upon source done oh, 13 revisions ago or something, mostly advertised through a web page dedicated to slandernig asterisk, not promoting whatever it is their broken crap is supposed to do |
| 20:10.56 | mafkees | he's as responsive as junghanns :) |
| 20:11.05 | mercestes | SomeOne1: but *you* decide what is better for you..:) |
| 20:11.18 | mafkees | and chan_sccp is not working with 1.4 :( |
| 20:11.54 | *** part/#asterisk jmls (n=asterisk@host81-159-198-100.range81-159.btcentralplus.com) |
| 20:12.57 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:13.12 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:13.16 | *** join/#asterisk E0x (n=moya@pri-133-b32.codetel.net.do) |
| 20:13.19 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 20:13.23 | ShadowHntr | Corydon-w: yo ! |
| 20:13.36 | E0x | hello |
| 20:13.42 | ShadowHntr | howdy |
| 20:14.03 | E0x | looking for guide/howto/whatever of asterk how ivr using mysql |
| 20:14.07 | Corydon-w | Afternoon |
| 20:14.22 | mercestes | E0x: huh? |
| 20:14.24 | mafkees | MikeB: yeah. but if it's not about the hardware.... well, lets not start that war indeed |
| 20:14.27 | mafkees | heya Corydon-w |
| 20:14.47 | mercestes | E0x: Exactly what role would mysql be playing in the IVR? |
| 20:14.59 | E0x | save the sound files |
| 20:15.13 | Nugget | the same role mysql adds anywhere -- adding randomness and unpredictability to the process. :) |
| 20:15.18 | mercestes | E0x: phpagi is going to be just about your only recourse. |
| 20:15.33 | E0x | Nugget, hahaha |
| 20:15.46 | mafkees | why would you store binary data in a database ? |
| 20:15.50 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 20:16.18 | mercestes | mafkees: It's popular for webpages and jpgs. It's called a "BLOB" Why he would want to wait for Mysql to coff up a sound file I dunno. |
| 20:16.32 | E0x | mafkees, i really dont , just looking info for a friend |
| 20:16.47 | elriah | Hi all. Is there a way to create trunk groups with a single sip peer? We are using a provider that allots us X amount of inbound/outbound calls, is there a way to further divide that up with asterisk on our side? i.e., this context can only make 3 outbound/inbound calls, this other context can do 10, etc.? |
| 20:17.09 | mafkees | look at the group functions elriah |
| 20:17.09 | MikeB | mafkees: Indeed, I dont know why these people like to break the standard astrisk stuff. |
| 20:17.21 | mercestes | E0x: You will have to use phpAGI or some other "web language" interface to facilitate drawing the data out of Mysql and feeding it to *. |
| 20:17.28 | mercestes | E0x: And even then......it probably own't work. |
| 20:17.30 | MikeB | the Quad GSM card is very cool though. |
| 20:17.52 | mafkees | MikeB: so I heard. We are still using the 2N voiceblue |
| 20:18.45 | MikeB | madkees you have to like debug messages all over the console though but its end to end GSM |
| 20:19.11 | mafkees | can it do sms as well ? |
| 20:19.11 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 20:19.45 | E0x | mercestes, well i will say to me friend that come here to ask because i dont exacly know what him want |
| 20:20.20 | mercestes | E0x: ok. |
| 20:20.27 | tzafrir_laptop | mafkees, BTW: can you send SMS through your telco? |
| 20:20.28 | mercestes | that actually sounds like a sound plan. |
| 20:20.37 | elriah | I read the voip-info groups faq, but it wasn't clear how to actually create groups. I can do it with zapata and physical lines, but I need to do it with sip peers. I would be greatful for any help you can provide getting me going in the right direction. |
| 20:20.40 | mafkees | tzafrir_laptop: yeah I can |
| 20:20.51 | mafkees | we used it before we found bayhamsystems |
| 20:21.06 | E0x | mercestes, i think him is build a system for check info via phone call , example result of school Test |
| 20:21.07 | E0x | etc |
| 20:21.13 | mafkees | it's rather expensive here in .nl to send sms using normal phonelines |
| 20:21.19 | mafkees | 0.22 euro/sms |
| 20:21.30 | *** join/#asterisk wylie (n=wswanson@ip68-231-80-171.ph.ph.cox.net) |
| 20:21.40 | mafkees | we still do read the sms from the line tho |
| 20:21.44 | tzafrir_laptop | What about gsm SMS? |
| 20:21.45 | Nugget | gekke nederlanders. |
| 20:22.08 | mafkees | tzafrir_laptop: no problem. but we cannot do that with the voiceblue |
| 20:22.12 | mafkees | Nugget ;) |
| 20:22.23 | mercestes | E0x: I think the only way to do that would be using creative naming conventions, not database access....but, it is possible. |
| 20:22.29 | J4k3 | my dog's name is Nugget |
| 20:23.03 | MikeB | tzafrir_laptop: Adrian Kennard wrote the app_sms thing for sending sms via BT dont know if it works elsewhere. |
| 20:23.10 | Jason99 | After doing a packet trace I notice that Asterisk sends an OPTIONS packet once every 60 seconds if the sip peer is up and every 10 seconds if the peer is down. Is there a way to change the timing of this through the configs? |
| 20:23.15 | E0x | maybe with a speech system that read the data |
| 20:23.20 | E0x | somewhere |
| 20:23.39 | tzafrir_laptop | MikeB, generally basically works in many othe places |
| 20:23.49 | Nugget | my cat's breath smells like cat food. |
| 20:23.57 | tzafrir_laptop | Luigi Rizzo fixed it a bit recently |
| 20:24.03 | mafkees | tzafrir_laptop: the app_sms MikeB is talking about is able to send sms using dutch gsm networks |
| 20:24.38 | mafkees | but still bayham is giving us better prices |
| 20:24.41 | mafkees | so we use that |
| 20:24.48 | J4k3 | pft... I dunno about in europe but a single SMS in the USA generally costs the same as 2-3 minutes of voice airtime. |
| 20:24.55 | tzafrir_laptop | not the one in Asterisk used originally to end SMS through British Telecom? |
| 20:24.55 | mafkees | it has the added benifit our colocated asterisk machines can use it as well |
| 20:25.09 | J4k3 | "always on" data service (gprs, edge, cdma 1xRTT, etc) is cheaper. |
| 20:25.13 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 20:25.22 | mafkees | and some of our webapps use bayham as well |
| 20:25.38 | mafkees | much easier for us. Have one account for everything |
| 20:25.43 | elriah | So this command: exten=> s,1,Set(GROUP(g9)=3) does what? Creates a group called G9 and set the limit to 3? (Guessing) |
| 20:26.11 | mafkees | it creates a group called g9 and sets the counter to 3 |
| 20:26.13 | mafkees | not the limit |
| 20:26.20 | mafkees | you have to check it yourself |
| 20:26.40 | elriah | mafkees, thanks. So I would have dial cmd if/then type scenerio to enforce groups? |
| 20:26.51 | mafkees | yeah |
| 20:27.16 | elriah | exten=> s,1,Set(GROUP(g9)=GROUP_COUNT(g8)+1) ... and -1 respectively? |
| 20:27.23 | elriah | rather, g9 for both group names |
| 20:27.51 | elriah | sudo -> if(GROUP_COUNT(g9) > 5) do nothing else place call, right? |
| 20:27.53 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 20:28.07 | mafkees | exten => s,n,GotoIf($[${GROUP_COUNT(kpn)} >= 5]?iax:kpn) |
| 20:28.07 | mafkees | exten => s,n(kpn),Set(GROUP()=kpn) |
| 20:28.07 | mafkees | exten => s,n,Dial(${TRUNK}/${ARG1},45,r) |
| 20:28.07 | mafkees | exten => s,n,Congestion(20) |
| 20:28.07 | mafkees | exten => s,n,Hangup() |
| 20:28.09 | mafkees | exten => s,n(iax),Dial(IAX2/bovendonk/${ARG1}) |
| 20:28.28 | *** join/#asterisk bmg505 (n=leon@196.209.249.86) |
| 20:28.28 | Bobthehunter | any way to check an array ? |
| 20:28.31 | elriah | Ok, cool I got it. Thanks again! |
| 20:28.38 | Bobthehunter | ${NPA}= 514,450,999 |
| 20:28.38 | mafkees | you're welcome |
| 20:28.47 | Bobthehunter | then check if in ${NPA{ |
| 20:28.50 | Bobthehunter | or somethign |
| 20:32.58 | toresbe | okay... so I have got an SIP link working, to my VoIP provider.. |
| 20:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:33.26 | *** join/#asterisk RevK-Laptop (i=RevK-Lap@27.0.169.217.in-addr.arpa) |
| 20:33.36 | *** join/#asterisk ping2921 (n=marc3234@206-248-129-91.dsl.teksavvy.com) |
| 20:33.52 | ping2921 | Hi,. |
| 20:34.11 | toresbe | How can I make a phone call from the Asterisk CLI using the SIP proxy? |
| 20:35.03 | MikeB | Evening RevK We have been talking about your app_sms. |
| 20:35.13 | app-sms-author | hence my arrival |
| 20:35.15 | MikeB | Aparently it needed fixing |
| 20:35.24 | *** join/#asterisk DocHolliday (i=tabmeist@gateway/gpg-tor/key-0x0E4F6D6C) |
| 20:35.29 | app-sms-author | Well, it works for me, what is now not working? |
| 20:35.47 | MikeB | charging ? ;) |
| 20:35.50 | voipman | hi |
| 20:35.54 | app-sms-author | We send thousands of texts on an ISDN30 line |
| 20:35.59 | app-sms-author | charging??? |
| 20:36.00 | ping2921 | I have calls coming from pstn. The problem I am having is that roughly the first 1sec of my greeting is cutoff. Anyone knows why? |
| 20:36.14 | voipman | any digium guys here? I have a g729 codec install problem |
| 20:36.16 | MikeB | BT are too expensive. |
| 20:36.29 | app-sms-author | The same protocol works with some premium rate text providers |
| 20:36.37 | app-sms-author | and incoming texts are free to receive on BT |
| 20:37.00 | voipman | I have an old link to the ftp://ftp.digium.com/pub/telephony/asterisk/g729/ directory that isnt working for downloading the binary and codec, does anyone know the updated url? |
| 20:37.13 | app-sms-author | We could add slugs to app_sms so it could be used on the smsc end of a premium rate number I guess... |
| 20:37.22 | app-sms-author | And make more money |
| 20:40.44 | anonymouz666 | what about an AMD 1.8GHZ 256 RAM handling 15 SIP calls g729... |
| 20:41.11 | anonymouz666 | do you think it's possible ? |
| 20:42.02 | mercestes | anonymouz666: Sure it's possible. |
| 20:42.09 | sevard | Quite possible |
| 20:43.26 | J4k3 | anonymouz666: pure 729 with no services (ie - no voicemail, moh, etc) would do a lot more than that, I believe. |
| 20:43.32 | JoNate | With Telepathy, anything is possible... |
| 20:43.37 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 20:44.08 | *** part/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 20:44.08 | JoNate | have I beat the horse to death yet? |
| 20:44.14 | sevard | voipman: iirc, you get the url to download the codec once you've paid for a license from digium |
| 20:44.16 | *** part/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:44.24 | mercestes | Jonate: LOl...no..it's still funny..:) |
| 20:44.33 | mercestes | they are a great telephony company, btw...the best in communications |
| 20:44.51 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 20:45.04 | JoNate | no connection fees I hear, and a per minute charge that would blow your mind... |
| 20:45.15 | J4k3 | yeah but... as a human race, why do we need a "telecommunications company" at all? |
| 20:45.29 | J4k3 | ;) |
| 20:45.46 | mercestes | J4k3: I don't get it. |
| 20:46.42 | J4k3 | mercestes: peer-based telephony is the future, IMHO. |
| 20:46.58 | JoNate | telepathy is the future... |
| 20:47.08 | mercestes | J4k3: Hm, your just full of new and novel ideas. Lots of time to think? you must spend lots of time on the potty. |
| 20:47.35 | voipman | sevard: i have the URL from the digium email which is old it's outdated and non-working |
| 20:47.48 | mafkees | I'll be back later |
| 20:47.49 | mafkees | bye |
| 20:47.51 | voipman | sevard: i already purchased the licenses. |
| 20:48.18 | *** part/#asterisk app-sms-author (i=RevK-Lap@27.0.169.217.in-addr.arpa) |
| 20:49.30 | sevard | Interesting, have you tried calling them? |
| 20:49.35 | sevard | they should be open for another couple of hours. |
| 20:50.00 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:50.52 | *** part/#asterisk jsandnes (i=jsandnes@sip.meet24.com) |
| 20:53.16 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 20:55.29 | sevard | voipman: http://ftp.digium.com/pub/asterisk/g729/ |
| 21:00.19 | *** join/#asterisk HushPe (n=HushPe@mail.kamar.co.nz) |
| 21:01.06 | HushPe | my asterisk system has been behaving for the last few days, but even after a quick reboot i'm getting Zap lines dropped for no reason with this message: == Everyone is busy/congested at this time (1:1/0/0) |
| 21:01.11 | in-pt | why i am getting this error "RTCP SR transmission error, rtcp halted" |
| 21:01.23 | in-pt | my music on hold is not working |
| 21:01.33 | in-pt | any one knows ? |
| 21:03.27 | HushPe | in-pt: what is the error for your moh? you have mpg123 installed (59r) |
| 21:04.12 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:04.14 | in-pt | no i havent installed mpg123 |
| 21:04.20 | in-pt | do i needs to install that ? |
| 21:05.04 | sevard | in-pt: are your MoH files in a native format? |
| 21:05.25 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:05.30 | HushPe | in-pt: if you want to play mp3s for on hold (which is probably easiest) |
| 21:05.46 | in-pt | sevard: well i am upgrading from asterisk-1.2 to 1.4 and they were in gsm format i copied them as it is |
| 21:05.50 | HushPe | if you could pb your debug too |
| 21:05.59 | in-pt | ya i can just a min |
| 21:06.12 | HushPe | hi sevard :) |
| 21:06.18 | sevard | Hello HushPe. |
| 21:06.22 | HushPe | cheers in-pt |
| 21:06.45 | HushPe | sevard: the only reason zap would drop a line would be irqs really eh (hardware related) |
| 21:06.46 | sevard | HushPe: do I know you? :s |
| 21:06.48 | *** join/#asterisk oej (i=olle@nat/digium/x-c64e5acbcb08dace) |
| 21:06.57 | HushPe | do i know me ;) |
| 21:07.00 | sevard | hah |
| 21:07.06 | sevard | do you have IRQ conflicts? |
| 21:07.48 | HushPe | sevard: story of my life really, my mobo doesn't allow forcing irq to my 2 pci busses, only a 'preferred irq' so if i change it, the onboard lan follows it along :( |
| 21:08.28 | HushPe | sevard: line is clear (no crackle etc...), but i get the odd dropped line, but today i've had about 6 in a row <30 after it's connected me (incoming or outgoing) |
| 21:08.30 | sevard | interesting, I'd suggest getting another motherboard. Digium hardware really, really doesn't like sharing interrupts. |
| 21:08.51 | HushPe | there is no way to force with the kernel? |
| 21:08.56 | in-pt | http://pastebin.ca/334754 |
| 21:09.00 | HushPe | as it seems to be the kernel that's doing it |
| 21:09.00 | in-pt | here is my pb |
| 21:09.05 | HushPe | cheers |
| 21:09.07 | sevard | I had lots of dropped calls on my PRI until I found out that the TDM cards do not play well with others. I changed to a motherboard that I could manually configure the IRQ and all problems vanished. |
| 21:09.31 | sevard | HushPe: there also might be a BIOS upgrade out for your specific motherboard, check your manufactor |
| 21:09.49 | sevard | IIRC, the BIOS assigns your IRQs, your kernel just uses them. |
| 21:09.50 | *** join/#asterisk Dandan (i=dandan@ip68-9-250-223.ri.ri.cox.net) |
| 21:09.51 | HushPe | that means i'll need to find a fdd and a disk LOL |
| 21:09.53 | Dandan | hey all :) |
| 21:10.07 | HushPe | in-pt: that's about skinny, i thought your problem was MOH ? |
| 21:10.08 | sevard | HushPe: are you PXE booting? |
| 21:10.17 | HushPe | no sevard |
| 21:10.21 | in-pt | ok keep skinny away for a while |
| 21:10.39 | in-pt | u see the starting of moh and stopping of moh lines |
| 21:10.39 | HushPe | ah i see it now LOL my bad |
| 21:10.44 | Dandan | trying to build zaptel 1.4.0 and i am getting error "The configure script was just executed, so 'make' needs to be restarted." anyone has anything to help me? |
| 21:10.58 | in-pt | and there is a error line for moh |
| 21:11.04 | HushPe | Dandan: do you have libpri installed? |
| 21:11.11 | Dandan | libpri... yeah :) |
| 21:11.12 | sevard | Dandan: no idea what you screwed up, but did you try a make clearn ; make |
| 21:11.18 | sevard | erm, make clean* |
| 21:11.26 | Dandan | sevard: tried that |
| 21:11.28 | HushPe | make rubadubdub |
| 21:11.31 | in-pt | do i needs to install mpg123 |
| 21:11.37 | Dandan | <spam> |
| 21:11.38 | Dandan | configure: *** Zaptel build successfully configured *** |
| 21:11.38 | Dandan | **** |
| 21:11.38 | Dandan | **** The configure script was just executed, so 'make' needs to be |
| 21:11.38 | Dandan | **** restarted. |
| 21:11.40 | mercestes | maybe a nice make distclean |
| 21:11.43 | Dandan | </spam> |
| 21:11.55 | Dandan | Makenshi: *** [config.status] Error 1 |
| 21:11.56 | HushPe | in-pt: if you're using gsm it shouldn't need it |
| 21:11.58 | Dandan | argh |
| 21:12.00 | Dandan | <PROTECTED> |
| 21:12.09 | sevard | Makenshi |
| 21:12.13 | HushPe | Dandan: make linux26 (if you're on 2.6 kernel) |
| 21:12.16 | sevard | what language is that |
| 21:12.21 | Dandan | hmmmmm |
| 21:12.26 | Jason99 | Is there a way to change the retransmit interval for SIP? |
| 21:12.35 | sevard | Ah yes, I forgot about make linux26 |
| 21:12.43 | Dandan | HushPe: I am... :/ |
| 21:12.51 | Dandan | (i just double checked) |
| 21:13.25 | *** join/#asterisk docelm0 (n=vircuser@c-68-85-97-222.hsd1.de.comcast.net) |
| 21:13.29 | sevard | sup do |
| 21:13.34 | sevard | ermBLARHG |
| 21:13.40 | sevard | sup docelm0 |
| 21:13.56 | Dandan | hm, strange... |
| 21:13.59 | Dandan | isn't it? |
| 21:14.09 | Dandan | i was trying to build a slackware package... |
| 21:14.26 | Dandan | with a script that i successfully used to build 1.2.* |
| 21:14.42 | Dandan | libpri worked, lib-newt too... |
| 21:14.44 | Dandan | zaptel didn't |
| 21:15.16 | EyeCue | fear zaptel upgrade not working :| |
| 21:15.34 | sevard | Dandan: do it by hand |
| 21:16.07 | Dandan | sevard: lemmie see... |
| 21:17.40 | Dandan | silly me... |
| 21:17.51 | Dandan | they introduced ./configure in 1.4.0... |
| 21:17.52 | Dandan | :) |
| 21:17.55 | Dandan | :X |
| 21:18.04 | HushPe | hehe |
| 21:18.15 | sevard | heh. |
| 21:18.40 | Dandan | HushPe: run for your life! |
| 21:19.20 | sevard | this is silly. |
| 21:19.32 | sevard | who needs a consultant? i need to pay for dinner |
| 21:19.33 | Dandan | of course, it is IRC really :) |
| 21:19.41 | Dandan | sevard: lol!!! |
| 21:19.41 | in-pt | sevard: can mpg123 play wav files also |
| 21:19.58 | b11d | ... |
| 21:20.04 | sevard | bawhahgha |
| 21:20.31 | sevard | anyone else want this one? |
| 21:20.46 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:21.11 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 21:21.34 | sevard | in-pt: Do you need a consultant? :) |
| 21:22.30 | Dandan | sevard: how good are you? :) |
| 21:22.32 | ctooley | I'm still looking for a Full time Asterisk admin in either Austin or Dallas. Lots of different types of uses. |
| 21:22.44 | sevard | Dandan: ask your mother |
| 21:22.50 | ctooley | I'm _not_ looking for a contractor |
| 21:22.52 | Dandan | if you could move our infostructure to 1.4.0 would be great... |
| 21:22.58 | *** join/#asterisk trixman (n=andy@rrcs-67-53-168-147.west.biz.rr.com) |
| 21:23.11 | trixman | what is the best voip service for asterisk |
| 21:23.16 | trixman | for reliability |
| 21:23.33 | sevard | Dandan: I know other people in here might suggest it, but at this stage I can't suggest moving an infrastructure from 1.2.x to 1.4.0 |
| 21:23.50 | Dandan | sevard: from 1.0.X :) |
| 21:23.55 | sevard | youza |
| 21:24.12 | sevard | is there a _need_ to upgrade? |
| 21:24.14 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:24.21 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 21:24.21 | sevard | ctooley: /msg |
| 21:24.27 | Dandan | sevard: nope :/ |
| 21:24.28 | Dandan | :) |
| 21:24.46 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 21:25.55 | sevard | Dandan: you're out of your mind |
| 21:26.22 | Dandan | well, the thing is that i grandfathered that installation |
| 21:26.22 | wunderkin | he's thinking outside of the box |
| 21:26.33 | Dandan | and i need to get a hold of what is going on with the system |
| 21:26.53 | Dandan | the biggest problem: the message waiting light goes on while the person is still leaving a VM |
| 21:26.57 | Dandan | a bit inconvenient |
| 21:29.36 | in-pt | sevard: no thnx :) |
| 21:31.32 | Dandan | ok, time to test this 1.4.0 :) |
| 21:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:38.41 | HushPe | i think i solve my call drop problem, it was just one person, the line said hangup |
| 21:39.11 | Dandan | IT"S ALIVE! |
| 21:39.19 | Dandan | zaptel-slackware.11.0 :) |
| 21:39.23 | Dandan | nice |
| 21:40.18 | HushPe | i'm running asterisk on slackware too, but not too much luck with zap on 1.4, so used 1.2 |
| 21:40.34 | sevard | Dandan: date it and tack -local on the end |
| 21:40.52 | Hmmhesays | sevard find me a wii |
| 21:40.54 | Dandan | sevard: I will submit it to linuxpackages.net after some testing |
| 21:41.21 | Dandan | and to pat, i had some correspondence with him re: asterisk in slack |
| 21:42.06 | sevard | Hmmhesays: http://www.nintendofinder.com |
| 21:47.01 | mercestes | http://www.humanclock.com |
| 21:47.06 | *** part/#asterisk Narkov- (n=Narkov@c58-108-246-199.kelvn1.qld.optusnet.com.au) |
| 21:49.16 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 21:53.41 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 21:54.45 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:55.06 | Dandan | ok, time to go :) |
| 21:55.15 | Dandan | hi and bye [tk]d- |
| 21:55.16 | Dandan | hi and bye [TK]D-Fender |
| 21:55.18 | Dandan | :D |
| 21:55.32 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 21:57.44 | [TK]D-Fender | HiBye |
| 21:58.16 | krondorl | bu-bye |
| 21:59.24 | b11d | www.timecube.com |
| 21:59.32 | b11d | its so informative.. its the new voip-info.org |
| 22:00.10 | sevard | b11d: I thought I'd never see timecube.com again |
| 22:00.22 | b11d | yeah its still kicking :) |
| 22:02.27 | b11d | Could MS just do one thing right and give me tabbed browsing within Windows Explorer.. |
| 22:02.34 | b11d | that'd be nice.. |
| 22:03.16 | perd | i have a 7960 with SIP firmware on it. i used to be able to dial a number on the phone, then press the 'dial' softkey to dial automatically with speakerphone when i had the SCCP firmware. anyone know how to enable this functionality in sip? |
| 22:03.28 | mercestes | tabbed browsing in ...oh , windows explorer..I thought you meant IE< I was gonna be like, "WTF? Can you say web flood?" |
| 22:03.48 | *** join/#asterisk J4k3- (i=jsuter@237.sub-70-216-154.myvzw.com) |
| 22:03.53 | sevard | b11d: IIRC explorer 7 has tabbed browsing |
| 22:03.59 | krondorl | mercestes, the new ie has tabbed browsing.. |
| 22:04.07 | hardwire | IE! |
| 22:04.09 | mercestes | for history results or for everythign? |
| 22:04.15 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 22:04.19 | b11d | yeah., i didnt say IE |
| 22:04.20 | krondorl | for browsing.. |
| 22:04.22 | b11d | i said WINDOWS explorer |
| 22:04.28 | sevard | Tabbed browsing, so you can get virii 8x as fast. |
| 22:04.32 | hardwire | oH! |
| 22:04.36 | krondorl | yup |
| 22:04.41 | krondorl | Hey Fender, wonder if I can pass something by you? |
| 22:04.44 | mercestes | krondorl: Your response did not answer my question.. =/ |
| 22:04.52 | b11d | I get to tuck all my IE browsing into one nice window, but I get like 8 Windows Explorer boxes |
| 22:04.54 | b11d | aigh |
| 22:04.58 | mercestes | Fender! Say no! It's herpes! |
| 22:05.09 | The_DoC^ | thats why I use firefox |
| 22:05.22 | krondorl | mercestes, neither of those only for opening sites.. different tab for each site. |
| 22:05.26 | b11d | firefox wouldnt give me tabbed Windows Explorer now would it |
| 22:05.46 | The_DoC^ | no, but anything ms browser wise sucks |
| 22:06.04 | krondorl | nercestes unless I didn't understand the initial question... |
| 22:06.08 | b11d | i actually dont have any issues with IE |
| 22:06.23 | mercestes | krondorl: I probably didn't undrestand the original statement. I'm thinking tab=completion line in bash |
| 22:06.29 | krondorl | I like the new IE 7.. but I still use firefox.. |
| 22:06.34 | mercestes | krondorl: I know what your talking about now. |
| 22:06.41 | sweeper | IE7 lacks extensions |
| 22:06.48 | krondorl | mercestes Ah I gotcha now.. doh!!! |
| 22:07.37 | mercestes | yea, doh. |
| 22:07.43 | mercestes | I blew that one up. lol |
| 22:08.37 | krondorl | I don't seem to be having any luck finding a site that can show me an example of how to add a number to a phone number if the first 3 digits match certain numbers. |
| 22:08.50 | [TK]D-Fender | krondorl : So long as it isn't infectious, I'm listening ;) |
| 22:08.53 | *** join/#asterisk docelmo (n=vircuser@c-68-85-97-222.hsd1.de.comcast.net) |
| 22:09.46 | krondorl | Fender: lol, actually just look at my last comment and that's waht I am looking for.. :) |
| 22:09.48 | [TK]D-Fender | krondorl : pastebin what you're doing now and I'll show you. |
| 22:09.53 | [TK]D-Fender | ~pb |
| 22:09.54 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 22:10.06 | b11d | yeah well you heard WRONG jbot.. |
| 22:10.16 | b11d | :P |
| 22:10.20 | krondorl | Actually I haven't even attempted it yet.. was looking for examples. |
| 22:10.56 | perd | if i'm getting weird sound from time to time (clicks, robot sounding voice for a second or two) what should i be looking at for troubleshooting? the jitter buffer? this is for SIP |
| 22:11.20 | mercestes | krondorl: http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching |
| 22:11.39 | mercestes | perd: Your network connectivity |
| 22:11.50 | mercestes | perd: You have bad jitter and probably a hardware debauchery with a firewall or switch. |
| 22:11.52 | perd | it's all local |
| 22:12.08 | mercestes | perd: Oh! In that case.... |
| 22:12.12 | b11d | I had that problem before, it was keystrokes on a PS/2 keyboard.. |
| 22:12.12 | mercestes | perd: your network connectivity. |
| 22:12.15 | b11d | going to USB fixed it |
| 22:12.18 | mercestes | <PROTECTED> |
| 22:12.30 | perd | asterisk server connected via gigabit ethernet, cisco phones connected directly to switch |
| 22:12.38 | mercestes | cisco switch? |
| 22:12.41 | perd | gigabit ethernet to thje switch i mean |
| 22:12.43 | perd | no, foundry |
| 22:12.47 | b11d | irq conflict? |
| 22:12.53 | mercestes | nah |
| 22:13.00 | mercestes | I still blame the infrastructure |
| 22:13.11 | b11d | like i said, i had clicks on my lines due to a keyboard.. |
| 22:13.26 | perd | hrm |
| 22:13.27 | *** join/#asterisk dasenjo (n=dasenjo@190.5.196.105) |
| 22:13.29 | mercestes | b11d: With a digium card? |
| 22:13.32 | b11d | yes |
| 22:13.42 | b11d | the clicks only happened when I was typing, of course. |
| 22:13.43 | wunderkin | "you're using voip, there's your problem" -integra |
| 22:13.48 | b11d | but I could hear them on the phone |
| 22:14.00 | perd | Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" B8ZS/ESF RED |
| 22:14.00 | perd | <PROTECTED> |
| 22:14.01 | mercestes | b11d: that's ......freaky |
| 22:14.09 | perd | that's the only shit i see in regards to irqs |
| 22:14.11 | perd | i dont see conflicts |
| 22:14.11 | b11d | yeah I thought so.. like I said, switching to USB fixed it. |
| 22:14.19 | mercestes | hehe |
| 22:14.26 | dasenjo | Hi! I need help with format_mp3 in asterisk 1.2 I'm getting errors about junk bits I have no idea about get rid of them |
| 22:14.41 | perd | thing is, i dont get them on my softphone that i have noticed |
| 22:14.50 | perd | maybe it's the cisco phone, i think they have jitter buffers in them hrmm |
| 22:14.57 | [TK]D-Fender | perd : Check "cat /proc/interrupts" and verify if the card is sharing its IRQ. |
| 22:15.23 | perd | yea it appears to be sharing it with the raid controller and usb controller |
| 22:15.25 | perd | 169: 255126080 257609853 IO-APIC-level 3w-9xxx, uhci_hcd, wctdm24xxp |
| 22:15.27 | b11d | ahhh |
| 22:15.27 | [TK]D-Fender | dasenjo : Have you verifiedt that the MP# is not VBR, and has no ID3 tags? |
| 22:15.30 | b11d | raid eh |
| 22:15.31 | perd | at least the 24 port card |
| 22:15.34 | b11d | that can be intensive |
| 22:15.36 | [TK]D-Fender | perd : the is HORRIFIC |
| 22:15.37 | perd | the t1 card is on its own irq |
| 22:15.45 | [TK]D-Fender | that* |
| 22:15.48 | perd | haha |
| 22:15.54 | perd | i suppose i should move the 24 port |
| 22:16.01 | [TK]D-Fender | perd : "duh" <- |
| 22:16.04 | b11d | :) |
| 22:16.12 | perd | JERKS! /me runs off crying |
| 22:16.20 | perd | ok i'll try that and see if i still suck |
| 22:16.22 | perd | thanks dudes :) |
| 22:16.27 | perd | now for pizza |
| 22:16.32 | [TK]D-Fender | :O |
| 22:16.43 | dasenjo | [TK]D-Fender, the mp3 is not VBR, I think it has and id3v1 empty tag ... |
| 22:16.59 | [TK]D-Fender | dasenjo : Trash the tags. |
| 22:17.01 | b11d | goodnight chaps |
| 22:17.06 | [TK]D-Fender | b11d|bbl : later |
| 22:17.47 | dasenjo | [TK]D-Fender, how can I do that, I use audacity and easy tag, but can't find a way |
| 22:18.50 | dasenjo | [TK]D-Fender, give a minute apt-getting id3ed :) |
| 22:19.35 | Jason99 | I'm trying to find the best way of controlling call waiting with the server... At the moment i'm putting call-limit=1 for customers who don't want call waiting and its working.. the only problem is that when you do a reload, the server no longer knows how many calls were in progress on each peer... any ideas anyone? |
| 22:20.33 | mercestes | Jason99: What type of phones? |
| 22:20.56 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 22:21.12 | Jason99 | Using Linksys ATA.. I know I can disable/enable on the ATA but I would rather the server to handle it |
| 22:21.30 | mercestes | Jason99: You'll be better suited handlign it in the ATA> |
| 22:21.58 | mercestes | however, if a "reload" makes * dump callwaiting on a peer with a call in progress....and you can always reproduce that, you could bug report it if it's not already reported. |
| 22:22.03 | mercestes | What v. of asterisk?? |
| 22:22.27 | [TK]D-Fender | Jason99 : "show application chanisavail" |
| 22:22.48 | mercestes | ya, I so never got that thing to work..;) |
| 22:22.59 | mercestes | maybe it will work for you tho. |
| 22:23.49 | Jason99 | [TK]D-Fender: Thanks I will test it out |
| 22:24.00 | Jason99 | looks like it'll do what I need.. if it works :P |
| 22:24.12 | mercestes | Jason99: Let me know if it does work, plz. |
| 22:24.18 | mercestes | I've been tinkering with that thing for awhile |
| 22:25.25 | docelmo | Can you setup a linksys wrt54g router be setup as a wireless repeater? |
| 22:30.01 | *** join/#asterisk nahirean (i=nahirean@unaffiliated/nahirean) |
| 22:30.47 | mercestes | docelmo: I don't think that's quite on topic. |
| 22:30.54 | mercestes | Ask Cisco |
| 22:32.15 | Jason99 | [TK]D-Fender: ${AVAILSTATUS} returns 0 no matter if im on a call or not.. |
| 22:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:33.24 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:36.42 | docelmo | mercestes dont EVEN come to me about topics.. We are ALWAYS off topic.. |
| 22:37.02 | docelmo | its a basic question.. either yes or no.. if you dont know then dont say anything.. simple enough.. |
| 22:38.39 | *** join/#asterisk J4k3 (i=jsuter@42.sub-70-216-215.myvzw.com) |
| 22:38.52 | hardwire | can anybody give me the chinese chars for "F*ck You" |
| 22:38.54 | [TK]D-Fender | Jason99 : then you're clearly not doing something right. |
| 22:39.01 | hardwire | I need to respond to a person in china taking over my village online |
| 22:39.15 | toresbe | What is the default extension for the demo recording? |
| 22:39.23 | docelmo | 500 |
| 22:39.47 | toresbe | awesome, thanks!! |
| 22:39.51 | Jason99 | [TK]D-Fender: this is what i'm doing |
| 22:39.52 | Jason99 | exten => 1,1,ChanIsAvail(SIP/TEST1) |
| 22:39.52 | Jason99 | exten => 1,n,NoOp(AVAIL STATUS ${AVAILSTATUS}) |
| 22:39.55 | The_DoC^ | docelmo, you can with different firmware |
| 22:40.08 | [TK]D-Fender | Jason99 : read the instructions AGAIN. |
| 22:40.11 | *** join/#asterisk shodan- (n=shodan@ip101.99-113-216.pppoe4.joliette.intermonde.net) |
| 22:41.38 | perd | so what's a quick way to change the irq my digium board is taking |
| 22:42.13 | nahirean | sup folks. |
| 22:42.34 | Jason99 | [TK]D-Fender: I dont know what I'm missing.. |
| 22:42.36 | [TK]D-Fender | perd : Disable as much as you can in your BIOS, and check there for any hopes of dedicating one to it. |
| 22:42.40 | perd | fucking bios irq crap, such a pain in my ass |
| 22:42.42 | perd | yeah |
| 22:42.50 | perd | it's a supermicro server board, at least i can do that |
| 22:42.56 | perd | needs more softirq |
| 22:42.56 | [TK]D-Fender | Jason99 : Keep re-reading it until you get it or your eyes bleed. |
| 22:43.58 | mercestes | doclemo: As I recall, you were jumping my ass discussing polycom phones as being "off topic." so...I believe I shall come to you about topics all I like. |
| 22:44.31 | [TK]D-Fender | mercestes : Nobody wants to know ANYTHING about what goes on with your ass... ok? |
| 22:44.37 | [TK]D-Fender | ;) |
| 22:44.44 | Jason99 | hmm |
| 22:44.47 | mercestes | [TK]D-Fender: ....that's not....*entirely* true....>.> |
| 22:44.59 | nahirean | yes it is. >:) |
| 22:45.37 | Jason99 | [TK]D-Fender: so ChanIsAvail(SIP/TEST1) should tell set the value of ${AVAILSTATUS} correct? |
| 22:46.04 | mercestes | Jason99: that's the idea. |
| 22:46.23 | [TK]D-Fender | Jason99 : I don't see blood flowing, nor even 10 minutes since being told to re-read the instructions..... |
| 22:46.26 | mercestes | Jason99: use sip=friend and don't set call-limit btw. and make sure to use the |s I believe flag. |
| 22:46.42 | Jason99 | [TK]D-Fender: there is 1 line to read.. it doesnt take 10 minutes to read |
| 22:46.47 | Jason99 | <PROTECTED> |
| 22:46.47 | Jason99 | This application will check to see if any of the specified channels are |
| 22:46.47 | Jason99 | available. The following variables will be set by this application: |
| 22:46.48 | Jason99 | <PROTECTED> |
| 22:46.48 | Jason99 | <PROTECTED> |
| 22:46.48 | Jason99 | <PROTECTED> |
| 22:46.50 | Jason99 | <PROTECTED> |
| 22:46.52 | Jason99 | <PROTECTED> |
| 22:46.54 | Jason99 | <PROTECTED> |
| 22:46.54 | mercestes | ...zomg. Pastebin. |
| 22:46.57 | nahirean | ugh |
| 22:46.59 | nahirean | stop |
| 22:47.00 | mercestes | ~lart jason99 |
| 22:47.14 | Jason99 | lol sorry |
| 22:47.19 | mercestes | no your not |
| 22:47.26 | Jason99 | yes I regret it |
| 22:47.32 | mercestes | ok, I forgive you...*this* time. |
| 22:47.33 | [TK]D-Fender | Jason99 : Well now that you spammed the hell out of us with that... READ IT AGAIN!!!!! |
| 22:47.38 | Jason99 | it looked smaller in my ssh screen ;) |
| 22:47.48 | [TK]D-Fender | mercestes : that's "you're" <- |
| 22:47.59 | mercestes | yessir. |
| 22:48.18 | [TK]D-Fender | Grammar Rangers.... ATTACK!!!!!! |
| 22:48.29 | EyeCue | http://www.pastebin.ca/334860 |
| 22:48.47 | EyeCue | got a compile fail in zaptel 1.0 - 1.1 upgrade on freebsd |
| 22:48.48 | EyeCue | any ideas? |
| 22:48.55 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:49.33 | [TK]D-Fender | EyeCue : 1.0?!?! |
| 22:49.41 | EyeCue | it's what in the ports tree :) |
| 22:49.53 | Jason99 | [TK]D-Fender: maybe you dont know what I'm trying to accomplish.. I'm trying to figure out if a channel is in use or not so that I know if I should send a second call |
| 22:49.53 | EyeCue | zaptel-1.0_1 < needs updating (port has 1.1_1) |
| 22:50.54 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 22:51.00 | teknoprep | http://www.voip-info.org/wiki-Cisco+POE <---- is this true ? |
| 22:51.10 | teknoprep | about the switching of pairs ? |
| 22:51.29 | [TK]D-Fender | Jason99 : Yes, I know EXACTLY what you want, and you still are having problems focusing on the big print... |
| 22:52.34 | brodiem | anybody here use vitelity? |
| 22:52.40 | J4k3 | I do |
| 22:52.45 | [TK]D-Fender | Jason99 : Keep re-reading the instructions... and don't forget the occular exsanguination.... |
| 22:52.52 | EyeCue | hmm |
| 22:52.58 | EyeCue | # |
| 22:52.59 | EyeCue | ztcfg.c:865: error: `ZT_GET_PARAMS_RETURN_MASTER' undeclared (first use in this function) |
| 22:53.00 | EyeCue | :| |
| 22:53.05 | brodiem | J4k3 did your call quality degrade severely after they moved their outbound carrier? |
| 22:53.25 | J4k3 | brodiem: yeah, my call quality isn't what it originally was |
| 22:53.27 | J4k3 | :| |
| 22:53.50 | brodiem | J4k3 yet they will not admit to there being any sort of problem |
| 22:54.22 | Qwell[] | EyeCue: You aren't going to get any help with compiling 1.0 |
| 22:54.29 | EyeCue | 1.1 ? |
| 22:54.46 | Qwell[] | no such thing as 1.1 |
| 22:55.14 | EyeCue | perhaps its the zaptel-bsd version |
| 22:55.20 | EyeCue | should get the port renamed infact. |
| 22:55.51 | Qwell[] | just get it from the zaptel-bsd svn |
| 22:55.51 | brodiem | i'm thinking of trying teliax even though they're expensive..just want decent termination |
| 22:56.20 | [TK]D-Fender | brodiem : VoicePulse seems to be pretty decent.... |
| 22:56.32 | [TK]D-Fender | brodiem : and really inexpensive as well. |
| 22:57.56 | brodiem | [TK]D-Fender oh yeah? I'll check them out, I'm tired of hanging on with vitel and hearing everyone complain about it that makes any calls lol |
| 22:57.56 | mercestes | teknoprep: I dunno, try it and let us know |
| 22:57.56 | *** join/#asterisk kenaeda (n=bobert20@CPE-76-178-146-94.natnow.res.rr.com) |
| 22:58.15 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:58.53 | brodiem | [TK]D-Fender err, no per-minute pricing? |
| 22:58.57 | [TK]D-Fender | brodiem : Well I'd never heard of Vitel until your mention of them just now. |
| 22:59.17 | [TK]D-Fender | brodiem : Sure they do. < .01c mostly, and IAX |
| 22:59.30 | brodiem | [TK]D-Fender ah, they used to be sixtel, and then merged with exgn and called themselves vitelity |
| 22:59.33 | [TK]D-Fender | brodiem : Look for their "VoicePulse Connect" service |
| 22:59.57 | brodiem | cool, ty |
| 23:00.36 | kenaeda | how can u record to mp3 with phone? im confused sorry :( |
| 23:01.02 | mercestes | kenaeda: google asterisk monitor and asterisk 2wav2mp3 |
| 23:01.08 | *** join/#asterisk cian_ (n=cian@cian.ws) |
| 23:01.20 | kenaeda | ty |
| 23:01.46 | Jason99 | No go.. but thanks.. I'll find another way to get this done |
| 23:02.05 | Jason99 | According to some pages I read, it doesnt work with SIP properly |
| 23:02.29 | mercestes | kenaeda: Tring to see if Record() supports mp3. I don't think so tho. |
| 23:02.36 | Qwell[] | mercestes: no |
| 23:02.44 | Qwell[] | it only reads...can't write |
| 23:03.19 | mercestes | Qwell[]: Just saw that. Fig'd with the format_mp3 It might have picked up write by now. :) guess not. |
| 23:03.38 | mercestes | kenaeda: 2wav2mp3 and monitor is your best bet to "automatically" make mp3's. Otherwise you can choose another format and convert. |
| 23:03.44 | Qwell[] | mercestes: we can't legally add write support (just like all other open source projects - but some just ignore that fact) |
| 23:04.23 | mercestes | yea, I heard that come up before. I remember now |
| 23:04.51 | kenaeda | mercestest: thanks. my main confusion is what hardware do i need to best accomplish this |
| 23:05.43 | [TK]D-Fender | Jason99 : Yes it works perfectly fine. I've done it with all sorts of SIP devices, Polycom Phones, Sipura ATA's, etc. |
| 23:05.54 | mercestes | kenaeda: harddrive, ram, motherboard, a phone. |
| 23:06.18 | [TK]D-Fender | Jason99 - <Jason99> s - Consider the channel unavailable if the channel is in use at all <------ wake up and smell the toast burning |
| 23:06.53 | Jason99 | [TK]D-Fender: I have tested that as well and it didnt work either |
| 23:06.53 | [TK]D-Fender | Jason99 : Dear God there were only 2 things to pass to that app. the channel and OPTIONS. |
| 23:07.18 | mercestes | [TK]D-Fender: Yea, I never got it to work either. |
| 23:07.36 | [TK]D-Fender | Jason99 : Check with "j" as well. I've done it plenty of times. shove them in, doa "show channels", then place your test. if it fails, this I wanna see... |
| 23:07.41 | mercestes | Jason99: Try to NoOp ${AVAILSTATUS} in different scenarios and see what it returns. |
| 23:07.45 | The_DoC^ | ok, I need a few more opinions before I try this. I don't have a fxo yet and I happen to have a intel 537 modem. I have read that if you remove r13 and r19 it disables the vendor code thus resulting in a x100p. what I am asking is does anyone suggest doing this? |
| 23:07.49 | kenaeda | mercestes: my goals is to try to let multiple users call a phone number/extension and enter a special number and record a message which saves to mp3 format |
| 23:08.25 | kenaeda | this is a job for asterisk , right? |
| 23:08.28 | [TK]D-Fender | kenaeda : MP3 is not a format you can record to IIRC. You'd have to record into another format and then convert it externally. |
| 23:08.52 | [TK]D-Fender | kenaeda : But yes, the overall task is suited to *, plus a few common audio tools. |
| 23:08.58 | kenaeda | okay cool |
| 23:09.12 | kenaeda | do you have to buy digium software? |
| 23:09.20 | kenaeda | hardware i mean |
| 23:09.40 | [TK]D-Fender | kenaeda : * does not require any special hardware. |
| 23:09.54 | Qwell[] | unless you want to connect analog phones or T1s or something |
| 23:09.57 | Qwell[] | but yeah |
| 23:10.18 | kenaeda | have you guys seen www.snapvine.com? they let users record by cell phone and it puts a voice message on a webpage |
| 23:10.24 | kenaeda | i want to try something similar to that |
| 23:11.14 | Jason99 | [TK]D-Fender: "show channels" shows an outgoing call on the specified channel.. ${AVAILSTATUS} ${AVAILCHAN} returns 0 and channel name |
| 23:11.44 | Jason99 | [TK]D-Fender: I will try with j now |
| 23:11.51 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 23:12.54 | Jason99 | [TK]D-Fender: same results.. didn't jump to line 101 |
| 23:13.02 | [TK]D-Fender | Jason99 : make sure you have priorities to match and PASTBIN the attempts. |
| 23:13.17 | [TK]D-Fender | and show your dialplan.... |
| 23:13.32 | *** join/#asterisk progeek (n=andrewb@ip-66-235-230-20.sterlingnetwork.net) |
| 23:13.44 | [TK]D-Fender | Qwell : Not even, but we won't dwell on the laternatives ;) |
| 23:13.49 | [TK]D-Fender | alternatives* |
| 23:14.00 | mercestes | kenaeda: search for monitor and 2wav2mp3 on voip-info.org. They ahve working examples. |
| 23:14.26 | progeek | I have a network that has no default route to the internet, all communucations must use proxies.. is there a way to interact with asterisk through a proxy? (is there proxy software designed specifically for this?) |
| 23:14.59 | progeek | by interact, I mean make and recieve calls |
| 23:15.16 | progeek | or would the solution be to setup an asterisk instance as the proxy? |
| 23:15.24 | mercestes | progeek: Could try SER to route to *. As far as via proxies, your phones should be on the * network. The only "outbound" you should do is a register to your SIP provider telling them where to find you. |
| 23:15.58 | progeek | but i can't connect directly from my phone to the outside sip provider. |
| 23:16.07 | mercestes | progeek: Now if you are talking about having phones all over the world then I would put * on an external IP...or use Vlans. |
| 23:16.24 | progeek | i don't want to/can't change infrastructure |
| 23:16.29 | mercestes | progeek: You don't want to connect directly from your phone to yoru sip provider. |
| 23:16.41 | mercestes | You want your phone to connect to * and * to connect to yoru provider |
| 23:16.59 | progeek | so asterisk would be teh proxy. |
| 23:17.01 | progeek | great. |
| 23:17.02 | progeek | thanks :) |
| 23:17.08 | mercestes | any time..:) |
| 23:17.10 | nahirean | anyone here know of an ITSP that can port vegas numbers? |
| 23:21.25 | *** join/#asterisk catpants (n=catling@12-214-191-244.client.mchsi.com) |
| 23:27.31 | *** join/#asterisk PaulTech85 (i=PaulTech@72.29.76.254) |
| 23:28.06 | *** join/#asterisk Gankhuu (n=gankhuu@72-166-51-162.dia.static.qwest.net) |
| 23:28.25 | PaulTech85 | Simple question, I have a menu prompt that prompts for a 4 digit ID, WaitExten would be the expected function, I have _XXX that should populate {$EXTEN} with the extension desired |
| 23:28.32 | PaulTech85 | This is not happening, Any suggestions |
| 23:30.07 | mercestes | PaulTech85: show application authenticate |
| 23:30.57 | [TK]D-Fender | PaulTech85 : I suggest you pastebin the entire context so we can see what you're doing. |
| 23:30.58 | [TK]D-Fender | ~pb |
| 23:31.01 | 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 |
| 23:31.21 | PaulTech85 | It's not a static number to enter, They can enter any 3 digit code and are then prompted for a second code |
| 23:31.27 | PaulTech85 | I know what pastebin is ;) |
| 23:31.40 | [TK]D-Fender | PaulTech85 : Around here, you never know..... |
| 23:31.41 | mercestes | PaulTech85: You'd be the only one around here. |
| 23:31.44 | PaulTech85 | Let me grab the context |
| 23:32.14 | PaulTech85 | Well if I was to 'brag', I am a server admin on efnet, Been around IRC for about 5 years. I know my way around |
| 23:32.28 | PaulTech85 | Hehe, let me grab that context, and thank you guys |
| 23:32.47 | mercestes | good thing he doesn't brag....he might come off as pretentious. |
| 23:32.50 | mercestes | lol |
| 23:33.01 | *** join/#asterisk J4k3 (i=jsuter@61.sub-70-216-152.myvzw.com) |
| 23:33.05 | orlock | PaulTech85: and i muck out sewers! |
| 23:33.09 | shido6 | shouldnt the followme app use Background and not "playback" |
| 23:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:33.23 | orlock | efnet, right of way by the most bots. |
| 23:33.30 | orlock | :) |
| 23:33.31 | PaulTech85 | I was attempting to convey that I am not a complete noob and not sound like a dick at the same time |
| 23:33.36 | PaulTech85 | We've cleared alot of drones ;) |
| 23:34.21 | mercestes | PaulTech85: ;) D-Fender's point is, you can't tell a n00b from a luser by a username so better safe than sorry and specify what a PB is before they flood us with their entire extensions.conf |
| 23:34.43 | mercestes | n00bs will appreciate the info and lusers will understand that n00bs do exist and tend to flood. |
| 23:36.05 | PaulTech85 | http://pastebin.ca/334893 |
| 23:36.20 | PaulTech85 | mercestes, Completely understood :-) |
| 23:38.13 | [TK]D-Fender | PaulTech85 : Care to share some * CLI output for the attempt? |
| 23:38.46 | PaulTech85 | Just jumps to no timeout 't' |
| 23:39.00 | PaulTech85 | Let me get full output |
| 23:39.31 | mercestes | PaulTech85: Call an Answer() first. |
| 23:39.48 | PaulTech85 | Playback calls the answer for me |
| 23:40.47 | PaulTech85 | Shouldnt it? |
| 23:40.47 | mercestes | http://pastebin.ca/334896 |
| 23:40.47 | mercestes | Nah |
| 23:40.47 | mercestes | not that I am aware of |
| 23:40.47 | [TK]D-Fender | PaulTech85 : Technically, though I wouldn't rely on that personally. Explicit wins every time. |
| 23:40.48 | PaulTech85 | Ok |
| 23:41.10 | [TK]D-Fender | PaulTech85 : That PB doesn't seem to have * CLI output.... |
| 23:41.24 | PaulTech85 | I'm adding it on |
| 23:41.26 | mercestes | I pb'd that, Fender, sorry |
| 23:41.35 | [TK]D-Fender | oops.... |
| 23:41.42 | [TK]D-Fender | On the house! |
| 23:41.50 | mercestes | :) |
| 23:41.51 | mercestes | thanks! |
| 23:45.05 | mercestes | aww.... |
| 23:45.25 | PaulTech85 | Worked prefectly, not to prompt for a second one |
| 23:45.34 | PaulTech85 | That should be the interesting part |
| 23:45.41 | PaulTech85 | Should I send to a second context to prompt for it? |
| 23:46.16 | mercestes | Could throw in a DISA somewhere if that's what your trying to do. |
| 23:46.29 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 23:46.38 | cekc | yay my digium card has arrived! |
| 23:46.38 | mercestes | but yea, a secondary context should work. |
| 23:46.46 | mercestes | then you can continue with the _xxx thing. |
| 23:47.10 | cekc | whoa, it came in a box, the last one I ordered was OEM or something |
| 23:47.51 | cekc | mousepad! |
| 23:47.51 | PaulTech85 | Well, Hmm the digittimeout is a second, So if someone is typing in a outbound number NXXNXXXXXX it wouldnt match the _XXX correct? |
| 23:47.51 | PaulTech85 | So no need for a second context '/in theory/' |
| 23:47.51 | PaulTech85 | but playback answers a context in theory |
| 23:48.30 | [TK]D-Fender | PaulTech85 : Or you could just call DISA...... they'd get some nifty dialtone too :) |
| 23:48.33 | mercestes | not really in theory, in fact, it would not match _XXX |
| 23:48.41 | mercestes | Yea, I vote DISA |
| 23:50.54 | PaulTech85 | Yeah I dont want the second to match _XXX |
| 23:51.07 | PaulTech85 | Hmm also, Using set how do I set the value to a pre-existing var? |
| 23:51.13 | PaulTech85 | Set(Foo={$bar}) |
| 23:51.13 | mercestes | It won't, unless they dial 3 digits then wait for a really logn time..... |
| 23:51.15 | mercestes | which is common |
| 23:51.32 | PaulTech85 | yeah |
| 23:52.20 | PaulTech85 | Any idea on the set question? |
| 23:52.31 | wylie | Question; In zapata.conf logical groups can be assigned to allow outgoing rollover, (e.g. group=1). Can I also include IAX2 or SIP channels as part of a outbound rollover group? If so, how? |
| 23:52.58 | mercestes | PaulTech85: Check out Disa. I think it does what you want. |
| 23:53.00 | *** part/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 23:53.14 | PaulTech85 | mercestes, its the other direction :-) |
| 23:53.31 | PaulTech85 | People on the inside, Wanting to call outside line must enter a companyid and then number |
| 23:53.42 | mercestes | It should work either way really |
| 23:54.14 | PaulTech85 | Ah |
| 23:54.36 | PaulTech85 | Any idea on the set question? I am looking at Disa now |
| 23:55.16 | PaulTech85 | Oh |
| 23:55.17 | PaulTech85 | I see |
| 23:56.04 | [TK]D-Fender | wylie : Nope. its up to you to do it in your dialplan. |
| 23:56.35 | mercestes | PaulTech85: :) Have fun |
| 23:56.37 | mercestes | goodnigh tall |
| 23:57.08 | wylie | [TK]D-Fender; could you point to a reference or example that would show attempting to use first line, detecting in use, trying next outbound line? |
| 23:58.02 | [TK]D-Fender | wylie : Ending up on the next priority after a Zap call is pretty much an immediate reason to dial out the next tech. |