| 00:00.51 | Mez | boo |
| 00:02.39 | Mez | I just want something (semi) decent to use as an example for work, and prove that SIP/VoiP is viable. |
| 00:02.51 | Mez | I *dont* want to hook it up in magrathea ... |
| 00:03.16 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:03.26 | citywok | ah, in that case sign up for a couple carriers and just test them out |
| 00:03.39 | citywok | it will work fine, the question is how good your internet connection is |
| 00:04.19 | Mez | citywok: 10Mb leased line... |
| 00:04.38 | citywok | the qustion is the line quality, not the bandwidth |
| 00:04.54 | citywok | low jitter, good routing, low ping |
| 00:05.34 | *** join/#asterisk plut0 (~cory@cpe-74-76-182-29.nycap.res.rr.com) |
| 00:05.35 | citywok | I'd suggest googling, pick a couple and try it out with a few phone calls to see how they work. |
| 00:05.48 | plut0 | gtalk break today for anyone else? |
| 00:05.56 | citywok | it's been broken for a couple days |
| 00:06.04 | plut0 | citywok: there a bug report? |
| 00:06.06 | Mez | citywok: I've only got a softphone on this line. While incoming audio is perfect... outgoing is shocking. |
| 00:06.16 | citywok | plut0: yes |
| 00:06.23 | plut0 | citywok: got the # handy? |
| 00:06.24 | citywok | supposedly it's on the gtalk side |
| 00:06.28 | citywok | nope, sorry |
| 00:06.30 | Mez | citywok: and I thought VoiP jitter was caused by the encoding/decoding? |
| 00:06.41 | citywok | plut0: 18412 |
| 00:06.49 | *** join/#asterisk simplydrew (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 00:06.56 | plut0 | citywok: thanks bud |
| 00:07.14 | drmessano | https://issues.asterisk.org/view.php?id=18412 |
| 00:07.14 | citywok | Mez: if there is transit jitter... you will have problems. i tihnk the encoding/decoding is only an issue if your box is slammed. |
| 00:07.56 | citywok | you may also run in to issues if somebody starts downloading something and your connection peaks. |
| 00:08.03 | Mez | citywok: aw, that means I'm gonna have to take in my phone to work and test it out. |
| 00:08.19 | citywok | test it with softphone+headset |
| 00:08.24 | citywok | or take your phone in |
| 00:08.44 | citywok | it will most likely work just fine, but if you have to prove it to your bosses then hvaing a working demo is much easier. |
| 00:09.05 | Mez | citywok: yeah, will have to take my phone in to test on jitter. |
| 00:09.53 | Mez | citywok: But then I need to setup a demo showing how we can display the "name" of the line it's coming in (So people can answer the calls with the right "sub" company) |
| 00:11.39 | citywok | set callerid(name) = CO1 |
| 00:12.05 | Mez | citywok: I know that bit... |
| 00:13.40 | Mez | though, I thought it was set(CALLERID(name)) |
| 00:13.48 | citywok | yes\ |
| 00:17.05 | *** part/#asterisk plut0 (~cory@cpe-74-76-182-29.nycap.res.rr.com) |
| 00:21.13 | *** join/#asterisk greezmunkey (~greezmunk@173.241.161.62) |
| 00:21.29 | mlsmith99999 | Is anybody here using vucinc.com (Velocity) as thier provider? |
| 00:30.37 | carrar | does the happy Friday dance |
| 00:36.51 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-233-75-73.mia.bellsouth.net) |
| 00:47.30 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 00:50.31 | *** part/#asterisk d-fens (~jscalia@su-nat.int.smq.datapipe.net) |
| 00:54.24 | *** join/#asterisk BenC[UK] (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 00:59.14 | phix | carrar: heh |
| 00:59.56 | phix | i did that 20 hrs ago :) |
| 01:00.04 | phix | your lagged |
| 01:06.14 | carrar | heh |
| 01:06.52 | *** join/#asterisk b14ck (~rdegges@cpe-72-129-70-245.socal.res.rr.com) |
| 01:06.56 | b14ck | Sup guys. |
| 01:06.58 | b14ck | Been a while :o |
| 01:08.30 | b14ck | So hey, random question, how stable is the AEL stuff in 1.8.x? |
| 01:08.33 | b14ck | Is it better than in 1.6? |
| 01:09.19 | b14ck | Been wanting to port my code to AEL, but wasn't sure how far along it was, and if it was going slowly replacing dialplan or what. |
| 01:14.47 | phix | werd up b14ck! |
| 01:14.52 | b14ck | sup! |
| 01:14.58 | phix | nadda |
| 01:15.02 | phix | I dont know what AEL is |
| 01:15.07 | phix | and I am still using 1.4.x |
| 01:15.07 | b14ck | Ah. |
| 01:15.14 | b14ck | I'm using 1.6.x atm. |
| 01:15.15 | phix | any reason to update? |
| 01:15.15 | b14ck | shrug |
| 01:15.21 | b14ck | Depends, what do you use it for? |
| 01:15.31 | phix | making and recieving phone calls :) |
| 01:15.38 | b14ck | Probably not then. |
| 01:15.39 | b14ck | heh |
| 01:15.54 | phix | to SIP and / or landline via zaptel stuff |
| 01:16.04 | b14ck | Yah, you can stay on 1.4, no reason to update. |
| 01:16.25 | b14ck | uses Asterisk for some complex projects. |
| 01:16.40 | b14ck | So I need the bleeding edge stuff =p |
| 01:16.57 | phix | define complex |
| 01:17.18 | b14ck | Various asterisk applications (telco-level). |
| 01:17.34 | phix | ok |
| 01:17.49 | phix | telcos use asterisk? |
| 01:17.51 | b14ck | We do a variety of termination stuff, conferencing, party lines, phone-games. |
| 01:18.11 | b14ck | Not primarily, but for anything termination related. |
| 01:18.18 | b14ck | We use opensips for most SIP stuff. |
| 01:18.20 | phix | I thought they all used giant propriatary systems so they can jusitify charging you a fuckload |
| 01:18.31 | b14ck | Dunno about the other guys. |
| 01:20.29 | b14ck | pokes @Qwell |
| 01:20.47 | b14ck | Got any opinions on AEL? |
| 01:22.17 | phix | ok |
| 01:28.53 | drmessano | b14ck, there are no plans whatsoever to replace the standard dialplan with AEL |
| 01:29.07 | b14ck | drmessano: Gotcha. |
| 01:29.11 | drmessano | b14ck, that question comes up time to time and it's always answered the same |
| 01:29.45 | b14ck | I'd really love to port my stuff to AEL, but if it's not going to replace dialplan as the default thing, I don't know if it's worth a big investment. |
| 01:29.52 | b14ck | =/ |
| 01:29.55 | drmessano | It wouldn't be |
| 01:30.02 | b14ck | Damn. |
| 01:30.05 | b14ck | Oh well, thanks man. |
| 01:30.10 | b14ck | Just thought I'd check. |
| 01:30.20 | drmessano | AEL seems to exist to offer an option to dialplan.. But it's never had the full feature set, and I doubt it ever will |
| 01:30.29 | b14ck | Yah. |
| 01:30.40 | b14ck | The syntax is so much nicer. |
| 01:30.48 | b14ck | That's the attraction =p |
| 01:32.44 | greezmunkey | hmm, i get this when dialing out, I'm concerned with the "relm" statement: Authorization:Digest response="04e7fdcea86b08894e1f86c3415ea69f",username="1001",realm="pbx12.vitell.co.uk",nonce="5995dae5",algorithm=MD5,uri="sip:33@192.168.1 |
| 01:34.11 | greezmunkey | This is a relatively new "test" system. I have disabled extensions.ael, and lua, and my extensions.conf makes no reference to vitell.co.uk - where is this coming from? |
| 01:37.49 | p3nguin | I guess that peer must be on vitell.co.uk. |
| 01:38.18 | greezmunkey | p3nguin: the peer is a audiocodes gateway sitting here on my bench! |
| 01:38.20 | b14ck | ^ some /quit message |
| 01:38.39 | p3nguin | (1938.30) [freenode] -!- There is no such nick ravnos |
| 01:39.01 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 01:41.31 | k-man | how do i turn on voice mail debugging messages in the console? |
| 01:42.20 | p3nguin | Never heard of it. |
| 01:42.42 | k-man | oh |
| 01:43.27 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 01:46.13 | k-man | how can i diagnose why unanswered calls are not being sent to voicemail? |
| 01:46.50 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 01:47.32 | p3nguin | core set verbose 4 |
| 01:47.45 | p3nguin | Make a call, watch call flow. |
| 01:52.54 | k-man | ok, when i call my system, after a 30 seconds of ringing, i seems tp pick up, then i get some seconds of silence, then it hangs up |
| 01:52.57 | k-man | http://pastebin.ca/2010154 |
| 01:53.08 | k-man | s/i/it |
| 01:53.45 | *** join/#asterisk n3hxs (~HAMming@pool-71-246-19-76.phlapa.fios.verizon.net) |
| 01:55.58 | hydester | hi. i can use asterisk fine from my intranet via my android/sipdroid, but when using it from outside of my network the calls connect but there is no sound. i have 5060 TCP/UDP forwading via my firewall and since i successfully connect to my server from outside and connect a call, i figure it is something besides a firewall thing. any ideas? |
| 02:05.02 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 02:05.10 | p3nguin | hydester: You need UDP 5060 for SIP and UDP 10000-20000 for RTP. |
| 02:06.09 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 02:06.14 | hydester | p3nguin: didn't know anything about RTP. so TCP 10000-20000 too, if i use TCP? |
| 02:06.36 | p3nguin | hydester: You don't use TCP. |
| 02:06.45 | p3nguin | hydester: You need UDP 5060 for SIP and UDP 10000-20000 for RTP. <--- this |
| 02:07.07 | p3nguin | hydester: Signaling and media are separate, so call "connection" has very little to do with the sound. That's why you could connect the call but hear nothing. |
| 02:07.46 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 02:13.51 | Beirdo | I know it's a wee off-topic, but would anyone happen to know if there's a default password for a Cisco 7940 (running SIP)? |
| 02:14.06 | p3nguin | Password for what? |
| 02:14.26 | Beirdo | the telnet and unlocking config |
| 02:14.26 | Nugget | telnet is eeeeeeevil! |
| 02:14.48 | chuckf | Nugget: only if you use it |
| 02:15.01 | Beirdo | agreed, but we have these two phones that were sent from another office... and the people there were all laid off :) |
| 02:15.22 | p3nguin | The password is set in SIPDefault.cnf and/or SIP<MAC>.cnf. |
| 02:15.41 | Beirdo | that's what I figured. sigh. |
| 02:16.14 | p3nguin | And telnet has to be manually enabled in SIPDefault.cnf, too. telnet_level: 1 (or 2) |
| 02:16.31 | p3nguin | Why is that a problem? |
| 02:17.19 | Beirdo | because the people who would know where they stashed those configs... are all gone ;) |
| 02:17.33 | p3nguin | You can't find the tftpd? |
| 02:17.41 | Beirdo | not yet |
| 02:17.56 | p3nguin | The default password is cisco. |
| 02:18.15 | Beirdo | heh, that's one I did NOT try. too obvious. let's try it |
| 02:18.38 | Beirdo | thanks ;) |
| 02:18.40 | Beirdo | yay! |
| 02:19.22 | Beirdo | way too obvious. I tried 1234, et al. but cisco unlocked the config. Now I can reconfig it the way we need. |
| 02:19.25 | p3nguin | They enabled telnet but didn't change the default password. Interesting. |
| 02:20.44 | Beirdo | no wonder they got let go :) |
| 02:20.58 | hydester | p3nguin: thanks. still not working, but i'll play around with it some more |
| 02:22.58 | p3nguin | hydester: Did you configure any necessary NAT settings? |
| 02:23.51 | hydester | p3nguin: in asterisk? i was just doing port forwarding |
| 02:24.02 | p3nguin | hydester: That's not sufficient. |
| 02:24.48 | p3nguin | hydester: In addition to forwarding UDP 5060,10000-20000 you must configure a few things in Asterisk to be compatible with NAT. |
| 02:25.32 | p3nguin | hydester: In sip.conf, you need to adjust the values of nat, localnet, and externhost or externip. |
| 02:27.05 | p3nguin | hydester: Oh, also canreinvite. Can't forget that one. |
| 02:27.39 | p3nguin | nat, localnet, externhost or externip, and canreinvite |
| 02:28.02 | p3nguin | in the [general] section of sip.conf. |
| 02:38.00 | ecrane | What dialplan strategy can you recommend for use in a dial plan to have asterisk call up my voicemail, wait a few seconds for a prompt, dial the password, then finally bridge up the call with me? |
| 02:40.04 | p3nguin | Use Dial()'s option D along with two w's for each one second of waiting you desire, followed by your passcode. |
| 02:44.59 | ecrane | Thanks! I am trying this/reading up on Dial(). |
| 02:45.36 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 02:47.13 | p3nguin | example: exten => 999,1,Dial(SIP/itsp/${myVoicemailNumber},30,D(wwwwwwwwww12345)) |
| 02:47.59 | p3nguin | You call 999, it uses peer named "itsp" to call your voicemail number, wait 5 seconds before sending DTMF 12345, and then bridge the call to you. |
| 02:48.19 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 02:48.54 | ecrane | p3nguin: I love you. Thanks. |
| 02:49.06 | ecrane | Works perfectly. |
| 02:49.17 | p3nguin | Cool. |
| 02:49.46 | ecrane | (wwwwww1234#ww1) (they want me to press #, press 1 to confirm, etc...) |
| 02:50.54 | p3nguin | While you are waiting for the 1234 # 1 to be sent, do you get complete silence on the phone? |
| 02:52.00 | p3nguin | I've never had an occasion to use this format of dial command, so I'm just curious. |
| 02:52.41 | ecrane | Yes, I get complete silence on the phone. |
| 02:53.57 | p3nguin | That would bug me and I'd end up playing with Dial(M()) and a local channel to provide some ringing sound or moh. |
| 02:55.07 | hydester | p3nguin: thanks man. that worked! |
| 02:55.30 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 02:55.35 | p3nguin | Cool. |
| 03:02.46 | ecrane | Got time for one more noob quest? I'm trying to use 'WaitExten(8)'. That is supposed to wait for 8 seconds, and during those 8 seconds, if someone dials an extension that is in the dialplan, it should switch to the 'new' extension they dialed right? I set it up to do this, however it seems like asterisk does not 'hear' the new extension being dialed. Any debugs to help tell if Asterisk is hearing the DTMF? |
| 03:03.17 | p3nguin | What's before the WaitExten(8) ? |
| 03:04.47 | ecrane | p3nguin: An AGI call. Here's the whole dialplan... http://ideone.com/cBVSF |
| 03:06.26 | DeHackEd | someone have a sample sip.conf config block for an SIP phone to register with asterisk? I keep getting "No matching peer found" or "Not a local domain" and I don't quite understand what I'm doing wrong... |
| 03:06.51 | p3nguin | ecrane: Try this: http://ideone.com/57tQT |
| 03:07.33 | p3nguin | Actually, I see something else I want to change... |
| 03:08.01 | p3nguin | ecrane: http://ideone.com/7yoJS |
| 03:08.36 | ecrane | heh, 'pimp my dialplan' |
| 03:10.32 | p3nguin | I also decided I should have used BackGround instead of Playback. |
| 03:10.55 | p3nguin | Playback() would allow 1 second where the caller's keypress would be ignored. |
| 03:11.01 | p3nguin | http://ideone.com/Gn9RQ |
| 03:13.57 | ecrane | It looks a lot cleaner now, thanks. Unfortunately, it does not appear to be collecting the digits I am dialing, or at least.. it isn't sending me to extension '1' even when I dial a '1' during the 'BackGround' or 'WaitExten' steps. |
| 03:14.56 | ecrane | So that's why I was wondering about a way to debug dtmf detection or something'. I already tried to do a SIP debug and the digits aren't being sent over as SIP messages. |
| 03:14.58 | p3nguin | dehacked: http://pastebin.com/qf8ejp1G |
| 03:15.20 | ecrane | I guess I can google it too... |
| 03:15.22 | p3nguin | ecrane: Check that you have configured dtmfmode appropriately in sip.conf. |
| 03:19.26 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 03:20.10 | DeHackEd | p3nguin: no love. keeps saying "not a local domain". I'm using linphone right now and the reg line (using 'sip set debug on') says "REGISTER sip:192.168.0.45" and "From: <sip:username@192.168.0.45>;tag=blah" |
| 03:21.40 | p3nguin | I'm not sue what "not a local domain" indicates is wrong. Can you paste your _entire_ sip.conf in the pastebin? |
| 03:21.48 | p3nguin | s/sue/sure/ |
| 03:24.01 | DeHackEd | http://pastebin.com/fZDR1Bny |
| 03:24.26 | DeHackEd | obviously stripped out a lot of private stuff |
| 03:24.43 | b14ck | .oart |
| 03:24.45 | *** part/#asterisk b14ck (~rdegges@cpe-72-129-70-245.socal.res.rr.com) |
| 03:26.08 | DeHackEd | http://pastebin.com/18X7RuCK with the SIP debuging output as well |
| 03:26.29 | *** join/#asterisk smeet2002 (~smeet2002@dsl-173-248-230-237.acanac.net) |
| 03:27.43 | smeet2002 | Hello people! |
| 03:30.12 | smeet2002 | Does anybody know how to set up a call from when you just lift the handset. Like an emergency line, you lift the handset and it's calling predifined number...you don't need to press buttons... |
| 03:32.02 | smeet2002 | ? |
| 03:32.15 | cmnky | smeet2002, ooo ... a red phone batphone hotline ;) |
| 03:32.52 | ecrane | p3nguin: Still having trouble... I tried forcing the sip peer to use ulaw, and checked dtmfmode. I ran asterisk in -vvvvvvvv -dddddd mode and I didn't see anything about 'DTMF' flying by when I dialed digits. Any other ideas? I'm thinking I'll get with my carrier to check their config..... |
| 03:32.55 | cmnky | "batman ... commissioner gordon here ... we have a problem .. the penguin has highjacked our pbx" |
| 03:33.43 | cmnky | smeet2002, i dont know myself, im sure someone here does tho ... im a total noob ... just good for jokes and abuse |
| 03:33.48 | *** join/#asterisk tish (~dna6@unaffiliated/tish) |
| 03:34.17 | p3nguin | If you have a phone that uses SCCP, chan_sccp-b v3 has hotline configuration that does exactly what you want. |
| 03:34.26 | ecrane | p3nguin: Bleh, yah.. I just tried another/outside line and the dialplan you helped me with works great. It's just DTMF collection with a particular carrier. Thanks for the help! I'm taking off. |
| 03:35.03 | p3nguin | Good luck straightening out that other carrier. |
| 03:36.11 | ecrane | p3nguin: thanks. I'm still learning about voip, and from what I've seen this whole issue of DTMF can be tricky ^^. |
| 03:36.28 | p3nguin | They are probably using the wrong mode. |
| 03:36.35 | smeet2002 | I am a newbee as well |
| 03:37.24 | p3nguin | smeet2002: What kind of phone do you have that you want to use for a hotline? |
| 03:37.42 | smeet2002 | I have Grandstream and Aastra |
| 03:38.14 | smeet2002 | red phone it's a good name cmnky |
| 03:38.38 | smeet2002 | But my question is how to catch the moment when handset is picked up? |
| 03:38.48 | p3nguin | Too bad you don't have a Cisco phone. |
| 03:38.50 | cmnky | smeet2002, id get a real red phone for it too ;) |
| 03:38.59 | smeet2002 | It doesn't relate to the phone...am I right? |
| 03:39.06 | cmnky | or paint an old beige one |
| 03:39.20 | smeet2002 | they all have great grey color... |
| 03:39.23 | cmnky | w/ some high quality automotive paint that would last |
| 03:40.26 | smeet2002 | I know you can program some buttons... |
| 03:40.47 | smeet2002 | But I would like to make it with handser pick-up... |
| 03:41.34 | p3nguin | I've never personally seen a SIP phone that directly supports hotline, but it's easy to make it so that pressing any key dials the hotline. |
| 03:42.20 | p3nguin | Or grab an old Cisco 7910 or 7912 for that purpose. |
| 03:42.42 | smeet2002 | So, P3nguin..that means when you pick up the handset in ordinary phones no signals go to Asterisk? |
| 03:42.58 | smeet2002 | I will try to see |
| 03:43.14 | p3nguin | With SIP, picking up the handset does not send any phone call to Asterisk. |
| 03:43.58 | p3nguin | You might be able to configure a very short timeout on the phone to dial a specific number if no keys have been pressed. |
| 03:44.16 | p3nguin | Or you can configure dialplan so that dialing any number calls the hotline. |
| 03:44.38 | p3nguin | But with an SCCP phone, the channel driver knows when the handset is picked up. |
| 03:44.43 | smeet2002 | Yea..I though the same way..but it's still pressing something |
| 03:45.06 | p3nguin | And you can make it dial the hotline by picking up the handset without dialing anything. |
| 03:45.39 | smeet2002 | As I understand in SIP device it's more phone feature then pbx box |
| 03:46.37 | p3nguin | I don't have personal experience with every SIP phone out there... maybe there is a phone which allows hotline configuration on the phone. |
| 03:48.07 | smeet2002 | I found that it's called ""Offhook Auto-Dial" |
| 03:48.51 | p3nguin | Does your phone support it? |
| 03:50.13 | smeet2002 | I am trying to find right now |
| 03:52.04 | *** join/#asterisk coppice (~chatzilla@96.157.17.210.dyn.pacific.net.hk) |
| 03:52.42 | smeet2002 | It's funny, but I have it:))) |
| 03:53.01 | smeet2002 | At least in one Grandstream |
| 03:53.40 | smeet2002 | The other one has it as well |
| 03:53.49 | smeet2002 | I am going to test it right now |
| 03:54.04 | *** join/#asterisk stope (~nobody@sud-cable-cmts3-69-60-242-213.vianet.ca) |
| 03:57.21 | DeHackEd | I figured it out. Yay! |
| 03:57.56 | smeet2002 | Works really great! Dials really fast...You can't even bring the handset to you head the other phone is already dialing |
| 03:59.27 | p3nguin | smeet2002: That's really great. Both the Grandstream and the Aastra both have hotline built in? |
| 03:59.58 | smeet2002 | Actually, I am able to check only Grandstream right now |
| 04:00.09 | smeet2002 | BOth Grandstrem have this feature |
| 04:00.14 | smeet2002 | Grandstreams |
| 04:00.50 | p3nguin | Wow, Grandstream phones might actually be good for something after all! |
| 04:02.09 | smeet2002 | One is GXP2000, another is GXP2020..is somebody needs this info..I was told it's a cheepest brand.. |
| 04:02.19 | smeet2002 | if |
| 04:02.22 | p3nguin | ~gs |
| 04:02.22 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 04:02.27 | p3nguin | :) |
| 04:02.36 | smeet2002 | OK.. |
| 04:02.52 | smeet2002 | We will see if Aastra 480i has this feature... |
| 04:04.51 | smeet2002 | Is it a bot? |
| 04:05.16 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 04:05.40 | p3nguin | infobot: Are you a bot? |
| 04:06.44 | smeet2002 | TimRiker's infobot |
| 04:09.47 | smeet2002 | OK..as long as I set up an emergency line, I can sleep tight... |
| 04:10.02 | smeet2002 | thank you everybidy |
| 04:10.11 | smeet2002 | see you all tomorrow |
| 04:23.41 | greezmunkey | p3nguin: seriously on the grandstream phones? I was considering getting a couple... |
| 04:24.48 | p3nguin | ~grandstream |
| 04:24.48 | infobot | hmm... grandstream is the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 04:24.58 | greezmunkey | nice! |
| 04:25.26 | p3nguin | I've never used any, so I have no personal opinion about them. |
| 04:25.32 | greezmunkey | yikes, what then - I guess polycom, but they'er $$$ |
| 04:25.45 | p3nguin | I use Cisco. |
| 04:25.55 | p3nguin | Snom is aparently good, too. |
| 04:26.16 | p3nguin | Aastra has some good models. |
| 04:26.21 | greezmunkey | Cisco, I thought they were proprietary. |
| 04:26.34 | greezmunkey | learn something new all the time here. |
| 04:26.36 | p3nguin | They are (kind of). |
| 04:27.28 | greezmunkey | p3nguin: I have a cisco ata knocking around in a drawer at work, any chance I can experiment with that successfully? I mean is there any chance it may work? |
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| 04:27.56 | p3nguin | I don't see any reason it wouldn't work. |
| 04:28.05 | p3nguin | It's most likely a regular SIP device. |
| 04:28.15 | stope | you could always try yealink, it's a cisco knock off |
| 04:28.21 | greezmunkey | p3nguin: I'll check it out then. ty |
| 04:28.49 | p3nguin | You think it's like an ATA 186? |
| 04:29.03 | greezmunkey | I'm looking at some aastra phones, have you used them at all? |
| 04:29.23 | greezmunkey | p3nguin: I really can't say right now. I do know it's pretty old. |
| 04:29.25 | p3nguin | Nope, I'm Cisco only. |
| 04:29.44 | stope | I find aastra good, easy to provision |
| 04:31.07 | greezmunkey | stope: man they sure have a lot to choose from. |
| 04:31.55 | stope | I like the 9143i's as they're like the old nortel's |
| 04:32.06 | stope | but the newer ones we have all over the office |
| 04:32.19 | stope | even the touch screen, but it's more just for appearance if you ask me |
| 04:32.53 | stope | snom not bad either |
| 04:33.03 | stope | simple is good |
| 04:33.53 | greezmunkey | I looked at their page a few days ago, they don't seem to have as many to choose from, but their reviews are good. |
| 04:35.12 | greezmunkey | the astra 9480 kind of looks like a shoretel phone. |
| 04:35.29 | stope | but in the end, grandstream is not the best, although I saw on of their video phones recently, it was nice |
| 04:36.17 | stope | stay away from the old aastra's, newer hardware is out now |
| 04:36.35 | stope | 53, 55,57i's etc... |
| 04:36.37 | greezmunkey | stope: did they have problems |
| 04:36.51 | stope | we have some 480's, no issues |
| 04:37.57 | stope | although everything has issues at some point or another, we haven't had problems with aastra, snom or yealink's yet |
| 04:38.17 | stope | generally it comes down to what you want and price |
| 04:38.26 | stope | as does everything.... |
| 04:40.09 | greezmunkey | I just want a few moderately capable phone for my home lab. |
| 04:41.21 | greezmunkey | p3nguin: I fiured out that realm= thing (that I mentioned earlier) |
| 04:41.27 | stope | aastra or yealink I guess, polycom is a pain to configure(at least for me) |
| 04:41.59 | stope | maybe the aastra 9143, it's a good all round cost effective device that has a lot of features |
| 04:43.29 | greezmunkey | stope: I'll order one, and see how it goes. |
| 04:48.50 | p3nguin | greezmunkey: You'll have to refresh my memory. |
| 04:55.36 | greezmunkey | p3nguin: oh, I reconfigured my mp-118, and was test dialing to a kts. I was getting a realm statement in an error message that said Pbx12.vinet.co.uk, or some such. It was actually in my sip.conf file! |
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| 04:56.14 | *** part/#asterisk ManxPower (~ewieling@user-24-214-153-32.knology.net) |
| 05:04.46 | p3nguin | Hmm, I thought that was someone else with that problem. I remember that realm. |
| 05:05.32 | p3nguin | Well, vitel.co.uk, anyway. I assume you just typed it wrong. |
| 05:06.51 | p3nguin | Did I ask to see sip.conf? I thought I did and got it, and looked at it even. |
| 05:07.50 | *** join/#asterisk zpertee (~zpertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 05:08.52 | greezmunkey | p3nguin: no, you ignored me as usual...hehe |
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| 05:13.30 | p3nguin | I doubt that's the case. |
| 05:14.39 | p3nguin | I've got to scroll back and see what happened. I know I'm medicated, but I'm not THAT medicated. |
| 05:17.21 | p3nguin | I did respond. We just didn't look into it any further. |
| 05:18.09 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:39.05 | greezmunkey | p3nguin: it's all good! i was poking you with a stick, that's all! |
| 06:04.06 | ChannelZ | Is there any way on the mac to keep people from using / and * and crap like that in filenames? |
| 06:04.14 | ChannelZ | I swear I'm going to cut some people's hands off |
| 06:15.55 | drmessano | Your problem in the first place is using a Mac |
| 06:16.48 | ChannelZ | yeah it's one of those necessary evils |
| 06:16.59 | joeyjones | hhmm... |
| 06:17.09 | joeyjones | time ti try to get asterisk and freepbs running |
| 06:17.21 | joeyjones | *to |
| 06:17.36 | joeyjones | i will know shorlty if this asterisk install script for debian works |
| 06:19.47 | greezmunkey | pbs isn't free, they're always asking for money... |
| 06:20.21 | ChannelZ | It's free if you don't give them any (and don't pay your taxes) |
| 06:20.31 | joeyjones | *freepbx |
| 06:21.07 | joeyjones | greezmunkey: lol, their damn telethons get in the way of me watching their crappy shows |
| 06:21.31 | joeyjones | and btw, it seems to help to be running as root to install and compile things |
| 06:21.44 | joeyjones | the damn script should have forced an sudo or su |
| 06:22.20 | greezmunkey | when you modify your scripts, please be sure to post them. |
| 06:23.07 | joeyjones | it's the install script from http://www.corenetworks.com.au/wiki/doku.php?id=debian_asterisk_freepbx_script |
| 06:23.28 | joeyjones | i should look over my mplayer compile script, it forces root |
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| 06:49.27 | joeyjones | hhmm... |
| 06:49.33 | joeyjones | the install may have failed |
| 06:49.48 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 06:53.34 | *** join/#asterisk sam_affable (sam@202.53.10.123) |
| 06:53.36 | sam_affable | hi |
| 06:55.04 | sam_affable | im trying to configure and install asterisk 1.8 im not able to select chan_alsa it says depends on ALSA, i have alsa installed im on debian lenny 2.6.36.1 ck sources |
| 06:55.13 | joeyjones | would the wrong locale cause issues when compiling? |
| 07:03.29 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 07:35.13 | greezmunkey | thanks [TK]D-Fender for the debugging pointers, it'll help a lot. |
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| 08:33.17 | adelas | Hey, is it possible to translate a UDP port? |
| 08:33.38 | adelas | like open up port 5330 UDP on router, forward to port 5060 on server |
| 08:33.50 | adelas | my sonicwall allows it, but it doens't seem to work for UDP |
| 08:33.56 | adelas | if i change all protocol to TCP it works :( |
| 08:44.30 | shamelessn00b | 5060 is SIP? |
| 08:45.35 | jazzy-J | hello to All! |
| 08:46.05 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 08:46.30 | adelas | yea |
| 08:46.41 | jazzy-J | I need to know HW configuration to handle 500 sim calls; task is - protocol convertion SIP to H.323 |
| 08:46.51 | adelas | i want to setup remote phones to connect to a different port on router |
| 08:47.02 | jazzy-J | can asterisk help me with that ? |
| 08:47.28 | adelas | well basically i'm testing with a tftp server for the port now |
| 08:47.48 | adelas | tftp looking at local port 69, while on router i'm trying to forward different ports to 69 |
| 08:51.53 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 08:56.37 | joeyjones | can anyone suggest a good place to start reading on asterisk and the dialpan? |
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| 09:08.08 | WIMPy | ~book |
| 09:08.08 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 09:08.25 | WIMPy | ~newbook |
| 09:08.25 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/ |
| 09:08.46 | adelas | if i can't PAT a UDP port, is it possible to let asterisk sip to listen on more then then 5060 port? |
| 09:09.28 | k-man | is there an example somewhere of a simple incoming call dialplan, that sends the call to voicemail after no answer? |
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| 09:10.33 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 09:11.15 | shamelessn00b | jazzy-J: you can use SIPp |
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| 09:32.33 | adelas | eff yaaa |
| 09:32.41 | adelas | screw pat i can do multiple port binding :D |
| 09:34.38 | k-man | how do i reload the dialplan from the console? |
| 09:35.24 | WIMPy | dialplan reload |
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| 10:24.24 | k-man | i'm having trouble getting voicemail to work, i suspect its because i had to install the config files from a backup, but i lost the voicemail greeting files |
| 10:24.52 | k-man | where does asterisk look for those file? and how can i re-record them? |
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| 10:34.55 | Godfather_ | k-man, what problem do you have? |
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| 10:41.12 | k-man | Godfather_: well, incoming calls dont; got to voice voicemail, after 10 rings or so, i just get silence - here is my dialplan http://pastebin.com/qCmKUFYZ |
| 10:42.36 | k-man | i have to go for a bit, bbs |
| 10:43.11 | Godfather_ | ok |
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| 10:46.33 | ChannelZ | You have to actually send them to voicemail.. it doesn't look like you are |
| 10:47.45 | Godfather_ | ChannelZ, i solve my problem detecting the hangup in pstn with answerpolarityswitch=yes |
| 10:47.45 | ChannelZ | or.. I guess you're talking about your 'internal' context? |
| 10:52.42 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 10:53.41 | v1s | has any one used digiums g729 and any of the free ones? and can tell me if its worth buying vs using the free one ? |
| 10:53.59 | ChannelZ | There is no such thing as a free one |
| 10:54.34 | ChannelZ | g729 is a licensed algorithm, the "free" ones are old ripped off code |
| 10:55.25 | v1s | so is the licensed digium one better then the ripped off ones ? :) |
| 10:55.53 | ChannelZ | In so much as they are legal and supported... yes |
| 10:56.16 | v1s | and quality? |
| 10:57.28 | ChannelZ | In pure voice quality? I don't know, I imagine they are equivilent as the algorithm is the algorithm |
| 10:58.56 | v1s | can u recommend any other good low bandwidth codec? |
| 11:00.46 | v1s | i tried gsm but it sounded kinda hollow |
| 11:00.57 | ChannelZ | I guess 'good' is a relative term. |
| 11:04.01 | v1s | I have did provider to server one using g711 then server 1 to server 2 via iax using g729 server 2 to gsm modem |
| 11:04.40 | v1s | server 1 no bandwidth issues plenty of bandwidht |
| 11:04.54 | v1s | server 2 bandwidth not always so great |
| 11:10.01 | joeyjones | v1s: if it always flows from 1 to 2, why not use the same ompression method and pass it along w/o compressing a second time? |
| 11:11.04 | EmleyMoor | has just taken some action that he hopes will cut down on the SPITting attempts |
| 11:12.18 | v1s | joeyjones: not all my did providers support 729 |
| 11:13.02 | joeyjones | v1s: but, if it always goes from 1-2-gsm it will be a waste of resources |
| 11:13.49 | v1s | if I use gsm from 1-2 it sounds hollow like lots of echo or something |
| 11:14.24 | v1s | and theres no bandwidht needed from 2-gsm |
| 11:15.11 | joeyjones | v1s: think about this logically for a moment |
| 11:15.36 | joeyjones | it will need to convert it from g711 to g729 at the second server |
| 11:16.01 | joeyjones | now, if there was a compatible codec you could skip that conversion |
| 11:16.10 | v1s | i am converting to 729 at the first server |
| 11:16.20 | joeyjones | w/e |
| 11:16.25 | joeyjones | it skips having to convert |
| 11:16.37 | joeyjones | and compression is lossy |
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| 11:46.54 | joeyjones | is asterisk receives a registration attempt but it fails will CLI still show it when verbosity at least 3? |
| 11:46.57 | joeyjones | *if |
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| 12:02.42 | joeyjones | it seems to not like it when i use secret... |
| 12:02.52 | joeyjones | oh, but it still fails evben when it shows it being registered |
| 12:03.11 | joeyjones | and sip show peers shows the host as the softphone's LAN IP |
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| 12:29.48 | Godfather_ | if a have a call in 729 between 2 phones (asterisk in passthrough) and i have no g729 codec, can i MixMonitor it in g729 ? |
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| 13:52.50 | *** join/#asterisk kaylee (~mable@unaffiliated/kaylee) |
| 13:52.56 | kaylee | howdy |
| 13:53.26 | kaylee | anyone alive here? |
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| 14:03.59 | kaylee | hi |
| 14:04.26 | *** part/#asterisk sdward (d04a8d70@gateway/web/freenode/ip.208.74.141.112) |
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| 14:29.54 | brainiac | I am alive here |
| 14:34.42 | cmnky | +1 for being alive |
| 14:35.11 | cmnky | def a good thing |
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| 15:14.02 | kaylee | woo |
| 15:20.51 | cmnky | hoo ? |
| 15:21.20 | Jasnejac | ooh |
| 15:22.33 | kaylee | you're alive |
| 15:23.09 | cmnky | as alive as a monkey who types at a keyboard all day can be ;) |
| 15:23.32 | kaylee | as long as you've washed the poo off your hands first |
| 15:24.02 | cmnky | of course ... i dont want butt worms ... again ;) |
| 15:24.16 | kaylee | hahaha butt worms |
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| 15:27.06 | cmnky | i think i need more coffee ... hard starting engine this morning |
| 15:30.42 | cmnky | on the + side ... me user space packet filter/logger seems to be working |
| 15:32.25 | cmnky | which means i get to do fun stuff next ... GIS mapping |
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| 16:12.17 | *** join/#asterisk Roconda (~roconda@blackmind.roconda.nl) |
| 16:12.59 | Roconda | hi when creating IAX2 trunks, is it ok if I define; type=friend ? |
| 16:14.23 | p3nguin | I guess if you NEED the type to be friend, then it's okay. |
| 16:14.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 16:14.58 | Roconda | ok, thx |
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| 16:52.21 | *** join/#asterisk Tayeb (~VoIpGuy@41.200.4.161) |
| 16:52.23 | Tayeb | hi |
| 16:52.44 | Tayeb | lol, so many users;) |
| 16:53.25 | *** join/#asterisk sbszulu (~dundubala@41.14.37.212) |
| 16:56.14 | Tayeb | any 1.8.0 change log to see? |
| 16:56.16 | Tayeb | or feature page? |
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| 17:31.31 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 17:33.58 | ruben23 | hi guys any idea on this error but im confuse how to install source__________ http://pastebin.com/UUU3c1TV |
| 17:34.20 | ruben23 | my kernel is this ------> 2.6.18-194.26.1.el5PAE |
| 17:36.27 | pabelanger | Tayeb: Look in the source directory |
| 17:36.50 | Tayeb | pabelanger: i allready did, thanks |
| 17:42.53 | *** join/#asterisk root52 (~root52@wsip-98-175-224-146.cl.ri.cox.net) |
| 17:45.32 | root52 | Good Day, So I need to set a custom channel var. I then want to check that var from another channel to see it it is set or not. I am kind of at a loss for how to do this. Any thoughts? I thought about setting a global var to something and then checking to see if that was set. Problem with that is that another channel will overwrite that global when it starts up. Sorry if that sounds confusing. |
| 17:49.18 | root52 | To clarify, I want to make sure the same USER ID entered by the caller is not currently being used by any other channel. I should have just said that. ;-) |
| 17:49.54 | pabelanger | *CLI> core show function SHARED |
| 17:51.47 | root52 | pabelanger: ahh thank you!! that give me a good starting point! |
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| 19:13.28 | kt2lu | bows deeply |
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| 19:13.48 | Tayeb | dears |
| 19:13.52 | Tayeb | anyone have ipv6 runing? |
| 19:13.54 | Tayeb | sip with ipv6 |
| 19:14.53 | Tayeb | [TK]D-Fender: you have ipv6? |
| 19:24.53 | kt2lu | is they any sort of tool that will examine a dialplan that is deprecated, parse it and make it function with latest build of asterisk? |
| 19:25.29 | kt2lu | so everything has to be edited by hand? |
| 19:25.36 | kt2lu | *doesnt |
| 19:26.04 | Tayeb | kt2lu: ;lol |
| 19:29.29 | kt2lu | lol... yeah my dialplan here http://pastebin.ca/2010624 i guess i'll have to manually update. :( |
| 19:33.52 | kt2lu | thanks |
| 19:36.06 | kt2lu | yup... i've got 1.6 installed and love it... just migrating stuff over slowly. |
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| 20:54.26 | d1rtyd | hello |
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| 21:05.56 | neurosys | can improper ram (timings) on a board cause sound quality issues? |
| 21:05.59 | neurosys | sproatically? |
| 21:07.04 | cmnky | i doubt it ... normally you'd get memory corruption followed by freezes and crazyness |
| 21:07.38 | neurosys | what about older realtek drivers on CentOS. Heard anything about that? |
| 21:08.11 | cmnky | dont know anything about centos ... realtek drivers have been flakey in the past |
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| 21:13.32 | EmleyMoor | My Asterisk box has two network cards. How do I persuade my client phones to connect over the 192.168 rather than the public side? |
| 21:14.10 | EmleyMoor | (they are all on the 192.168 but one of the computers running a softphone is also on the public side) |
| 21:14.44 | EmleyMoor | All client phones seem to prefer to NAT via the router |
| 21:15.17 | EmleyMoor | (all relevant phones are SIP) |
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| 21:24.48 | *** join/#asterisk smeet2002 (~smeet2002@dsl-173-248-230-237.acanac.net) |
| 21:25.12 | smeet2002 | Hi there! |
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| 21:39.44 | ChannelZ | EmleyMoor: I guess that would depend on how they physically connect to the network, if they get DHCP from somewhere, and what IP they are told to contact * at |
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| 22:09.24 | joeyjones | ChannelZ: lol, i was going through the sip.conf setup in the book and it didn't say to use nat=yes, so my connections failed |
| 22:13.01 | smeet2002 | I still can't understand how this "nat=yes" works.. |
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| 22:13.22 | smeet2002 | If you just mention external IP isn't it enought? |
| 22:13.39 | joeyjones | smeet2002: nat=yes handles the clients being behind NAT |
| 22:14.04 | ChannelZ | Asterisk has to know it's external IP so it doesn't lie to the other side. Asterisk has to be told a peer is behind NAT so it ignores the IP the peer tells Asterisk. |
| 22:14.09 | joeyjones | ie. my clients were passing local IPs, and with nat=yes the server used the IP header to idtentify the external IP |
| 22:14.18 | ChannelZ | They control two different aspects. |
| 22:14.57 | smeet2002 | I had an issue with the SIP phone under the NAt until I put "directmedia=no" |
| 22:14.59 | joeyjones | it seems though like it would make sense to have nat=yes be a default |
| 22:15.08 | ChannelZ | Why? |
| 22:15.29 | ChannelZ | Some people use peers behind NAT, some don't. |
| 22:15.40 | joeyjones | i see no issues if it's local, and if it's external it will cause shenannigans |
| 22:15.58 | ChannelZ | Some people might have wierd routing issues and want to use the IP the device says and not where the packet came from. Who knows. |
| 22:16.06 | smeet2002 | CHannelZ, if client behind theh NAT his packet will have external IP of the firewall |
| 22:16.28 | joeyjones | if there's no NAT the IP in the header will be the local IP |
| 22:16.36 | joeyjones | *right IP |
| 22:17.12 | ChannelZ | What if the call comes from a proxy? The SIP packet isn't coming from the same place as the device. |
| 22:19.22 | joeyjones | wouldn't the proxy be used for the connection anyway? |
| 22:19.46 | smeet2002 | In case of proxy you are using INSECURE=port,invite...it 's still confusing to me... |
| 22:19.48 | ChannelZ | Maybe, maybe not. |
| 22:21.28 | joeyjones | anyway, i'd imagine thatas a default nat=yes would be a lot more helpful |
| 22:21.34 | ChannelZ | smeet2002: even if you use that, if Asterisk thinks the remote end is lying (nat=yes) it sends its media to the wrong place. |
| 22:24.39 | smeet2002 | ChannelZ,I think you just contradicts to joeyjones |
| 22:26.23 | ChannelZ | No but I'm just saying that defaults are defaults.. they will work for some, and not others. Change the defaults and you'll just be hosing up a _different_ group of people. |
| 22:26.28 | ChannelZ | put "nat=yes |
| 22:26.40 | ChannelZ | err put "nat=yes" at the top of your sip.conf and be done with it if that's what you want |
| 22:27.43 | ChannelZ | but turning on a function as a default that hotwires how the protocol normally works is going to confuse as many people as it might help, IMHO. |
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| 22:39.16 | *** join/#asterisk stram (alex@71-83-183-4.dhcp.psdn.ca.charter.com) |
| 22:39.32 | stram | are there any DiD providers which will allow me to transfer in my POTS phone #? |
| 22:43.29 | trollasaurus | Yes |
| 22:45.06 | ChannelZ | I'd say most |
| 22:49.44 | stram | sipgate isn't one of them-- can someone give me a few examples of ones that will? |
| 22:51.52 | trollasaurus | ~itsplist-us |
| 22:51.52 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 22:53.12 | trollasaurus | Try those |
| 22:54.21 | ChannelZ | I know vitelity and flowroute do at least |
| 22:58.23 | stram | oh, just remembered, our # isn't actually a POTS anymore, its actually VOIP through our ISP (charter communications). would that cause any problems in transferring? |
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| 23:01.02 | joeyjones | stram: afaik all service providers by law have to allow transfers |
| 23:01.32 | joeyjones | although some may not request the number be transferred |
| 23:02.09 | stram | gotcha |
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