| 00:00.00 | piper69 | EmleyMoor: FX port |
| 00:00.04 | [TK]D-Fender | riddlebox : Ondemand + the ability to multi-task.... priceless |
| 00:00.14 | EmleyMoor | FXS port - for analog phones |
| 00:00.39 | dendrite | piper69: FXS ports are how telephone systems provide dial tones. It's what asterisk would need to attempt to support your DVR. |
| 00:00.56 | perd | or you could just get a pots line |
| 00:01.19 | perd | if faxing sucks over VOIP i cant see how a modem would be better |
| 00:01.22 | [TK]D-Fender | EmleyMoor : I love how a .co.uk site is showing *US* load & tone zones, and HOPES of getting CID working :D |
| 00:01.24 | riddlebox | [TK]D-Fender, I thought about it too, my apartment building (4 apts) could use one cable tv connection and my mythbackend and wireless and we could all have tv-but that would require them to all have linux |
| 00:01.42 | *** part/#asterisk piper69 (n=piper69@unaffiliated/piper69) |
| 00:01.46 | dendrite | piper69: Borrow dial tone from your neighbors? |
| 00:01.50 | [TK]D-Fender | riddlebox : VLC streaming <------ |
| 00:01.59 | *** join/#asterisk Aces1Up (n=really@ip68-96-224-23.lv.lv.cox.net) |
| 00:01.59 | perd | mmm vlc |
| 00:02.03 | EmleyMoor | [TK]D-Fender: The information is verbatim from Digium - they do advise different settings if you are actually in the UK |
| 00:02.05 | perd | that is some hot shit |
| 00:02.08 | [TK]D-Fender | VLC = the BESTEST |
| 00:02.12 | perd | it really is, fender |
| 00:02.14 | perd | i love it |
| 00:02.27 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 00:02.31 | riddlebox | [TK]D-Fender, thats a good idea |
| 00:02.31 | perd | i kiss it, i hug it, i molest it, i .... |
| 00:02.35 | Aces1Up | hey all, whats the normal reason for a remote sip softphone do be able to dial numbers but hear no voice? |
| 00:02.35 | [TK]D-Fender | EmleyMoor : and the presented verbatim to WHOM.. the UK! whee! |
| 00:02.48 | perd | aces1up: RTP |
| 00:02.55 | Aces1Up | perd i see. |
| 00:03.01 | Aces1Up | sooo...... |
| 00:03.06 | Aces1Up | is that a port issue then? |
| 00:03.10 | EmleyMoor | myphonecall.co.uk are Digium's official UK agent - I suggest it's "fair enough" in this case |
| 00:03.12 | perd | yeah most likely |
| 00:03.21 | Aces1Up | that usually on the client or server????? |
| 00:03.23 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 00:03.24 | EmleyMoor | At least they replaced my FXO module at minimal quibble |
| 00:03.30 | Aces1Up | server works fine locally... |
| 00:03.35 | perd | depends who cant hear the audio |
| 00:03.38 | [TK]D-Fender | Aces1Up : Could be. How about you describe some CIRCUMSTANCES, show us your CONFIGS, before expecting us to turn psychic in order to help you? :D |
| 00:03.42 | perd | client cant hear audio, but server hears audio from client? |
| 00:03.49 | Aces1Up | the client can't hear audio. |
| 00:04.03 | Aces1Up | tkd sorry :) |
| 00:04.10 | Aces1Up | i'm impatient sometimes. |
| 00:04.13 | Aces1Up | well |
| 00:04.31 | [TK]D-Fender | there! |
| 00:04.39 | [TK]D-Fender | NEXT!@!@!@ (c) BKW |
| 00:04.39 | riddlebox | does anyone have a phone line through the phone company? |
| 00:04.41 | perd | angry trolls |
| 00:04.51 | perd | fender is their leader |
| 00:04.57 | [TK]D-Fender | riddlebox : Any? |
| 00:05.04 | robin_sz | asa rule, imaptience and solving software problems is rarely succesful |
| 00:05.04 | riddlebox | oops |
| 00:05.18 | EmleyMoor | riddlebox: I have one through BT - the incumbent here |
| 00:05.22 | [TK]D-Fender | riddlebox : What kind of a line? Its more of a closed loop really... does that count? |
| 00:05.22 | Aces1Up | robin yeh but its great at making them 10 times worse :) |
| 00:05.40 | riddlebox | [TK]D-Fender, on the phone and typing and it didnt come out right, does anyone have a phone line through their cable company |
| 00:05.42 | perd | aces1up have you tried a tcpdump of udp traffic |
| 00:05.51 | robin_sz | Aces1Up, just start randlomly deleting stuff, thats always a sure fire winner |
| 00:06.11 | perd | do a tcpdump on the server and the client for all udp traffic and see what's not getting through |
| 00:06.18 | [TK]D-Fender | riddlebox : Several of my friends. Is this a stats question, or are you goign to get to a point soon? ;0 |
| 00:06.27 | robin_sz | Aces1Up, is ther a nat firewall involved? |
| 00:06.44 | Aces1Up | robin, no just NAT, but no firewall. |
| 00:06.48 | Aces1Up | on either end. |
| 00:06.53 | perd | oh boy! |
| 00:06.57 | riddlebox | [TK]D-Fender, do faxes and modems work ok on them, is it a voip line from the cable company or is it an actual line? |
| 00:06.58 | perd | nat! what a hassle |
| 00:07.03 | robin_sz | yeah |
| 00:07.08 | [TK]D-Fender | perd : Don't give specific hints when he hasn't given specific details. Thats like fish trying to "fish" for fishermen :) |
| 00:07.26 | perd | haha |
| 00:07.31 | perd | good point |
| 00:07.37 | robin_sz | Aces1Up, well, nat will need setting up |
| 00:07.58 | yassine | while starting asterisk i get this error : http://rafb.net/p/rR1D5m27.html anyidea what is exactly missing or what the error means ? |
| 00:08.01 | robin_sz | Aces1Up, is this two locations of the same company offices or soem such? |
| 00:08.03 | Aces1Up | server location is behind basic nat router.. remote location is behind basic router.. at the remote location, should i have certain ports forwarded to the client box? |
| 00:08.06 | EmleyMoor | Just out of interest, would giving my * box interfaces both on public IP and private get round any NAT-related issues? |
| 00:08.11 | [TK]D-Fender | riddlebox : Cable-based phone is EFFECTIVELY voip in as much as its packet based through a residential loop like the rest of their data. the on difference is its their private medium, and not technically over the internet. |
| 00:08.20 | robin_sz | Aces1Up, no no no ... |
| 00:08.32 | [TK]D-Fender | riddlebox : and the typical answer is no modems, faxes, alarm systems, etc supported |
| 00:08.43 | Aces1Up | robin 2 locations. they are not part of any type of corporate network. |
| 00:08.53 | robin_sz | Aces1Up, easiest answer: set up an openVPN tunnel so the remote end can see all of the local end |
| 00:08.56 | [TK]D-Fender | Aces1Up : So NAT on both ends? |
| 00:09.08 | [TK]D-Fender | robin_sz : NOT! hold up on that... |
| 00:09.18 | Aces1Up | tkd yes nat on both ends. |
| 00:09.22 | robin_sz | [TK]D-Fender, works fine for me |
| 00:09.30 | Dr-Linux|home | [TK]D-Fender: one of my user is having some problem while installing digium card on 1.4, since i never used 1.4 yet, so maybe you could get the issue: |
| 00:09.44 | [TK]D-Fender | Aces1Up : Does your server at least have a FIXED IP (on the router's WAN side)? |
| 00:09.44 | Dr-Linux|home | [TK]D-Fender: here is the pb : http://phpfi.com/195254 |
| 00:09.48 | robin_sz | [TK]D-Fender, there is an encryption overhead, sure, but its worth the lack of NAT hassle |
| 00:09.58 | Aces1Up | tkd yes its fixed. |
| 00:10.10 | [TK]D-Fender | robin_sz : Works for you, sure, best or at all necessary, almost never. |
| 00:10.35 | EmleyMoor | How do I open a backgrounded ekiga?? |
| 00:10.38 | robin_sz | [TK]D-Fender, well, the benefits extend beyond just * |
| 00:10.51 | [TK]D-Fender | robin_sz : Yeah, but thats not why he's HERE :) |
| 00:11.07 | dendrite | yassine: lsof may help you find processes that have opened or locked devices. strace may help you get more detailed error output. |
| 00:11.10 | Aces1Up | tkd hey, i don't mind researching, just point me on something to google so i can see what causes the holdups between SIP and Nat traversal. |
| 00:11.12 | robin_sz | [TK]D-Fender, fair point ... but NAT is a real hassle, even at the best of time |
| 00:11.14 | robin_sz | s |
| 00:11.16 | [TK]D-Fender | Aces1Up : ok, what is on the remote side? |
| 00:11.24 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:11.30 | Aces1Up | remote side, just desktop hooked to router... |
| 00:11.32 | Aces1Up | nat on router. |
| 00:11.33 | [TK]D-Fender | robin_sz : I run double NAT'd setups all the time just fine.... |
| 00:11.45 | [TK]D-Fender | Aces1Up : What kind of phone, etc.... |
| 00:11.47 | robin_sz | [TK]D-Fender, but you are an expert :) |
| 00:11.57 | Aces1Up | softphone is x-lite |
| 00:11.59 | [TK]D-Fender | robin_sz : And thats why it WORKS :p |
| 00:12.38 | *** join/#asterisk BB|AtWork (n=karl@38.99.18.98) |
| 00:12.56 | [TK]D-Fender | Aces1Up : Thank you. for your server you'll need to add "canreinvite=no", "nat=yes",'localnet=[your subnet&maskhere]", "externip=[your wan ip here]". |
| 00:13.04 | robin_sz | [TK]D-Fender, I just find having my home network as 192.168.1 and the office as 192.168.3 and both of them visible from each other, well, its great |
| 00:13.09 | BB|AtWork | how important is echo cancelation on cards that would be connected to lines comming in from our phone provider? (dont want to spend the extra cash if i don't have to) |
| 00:13.28 | Dr-Linux|home | [TK]D-Fender: any suggestions? |
| 00:13.45 | Aces1Up | thats all in the extension for the softphone correct? |
| 00:13.48 | [TK]D-Fender | Aces1Up : for your CLIENT, you'll need to specifi in his device config (in your *), "nat=yes", "qualify=yes". You will need to forward 5060, 10000-20000 from your router to *. thats about it. |
| 00:14.19 | [TK]D-Fender | Dr-Linux : funny, I don't see "cat /proc/interrupts" or "dmesg" output to prove a module ever loaded for your card... |
| 00:14.44 | [TK]D-Fender | Aces1Up : the first part was for your [general] esection of sip.conf, sorry. |
| 00:15.01 | Aces1Up | tkd ok, i have that all except the careinvite part... |
| 00:15.04 | Aces1Up | lemme fix that. |
| 00:16.56 | Dr-Linux|home | [TK]D-Fender: okey thanks |
| 00:17.20 | [TK]D-Fender | Aces1Up : if you don't, "bad things" will happen. once you've done all of this, then well need to proof-read your configs. once that passes then we move on to sip debug. All one step at a time. |
| 00:18.46 | Aces1Up | tkd cool just read up on canreinvite, so that keeps asterisk as a kind of translator for both parties.. |
| 00:19.13 | Aces1Up | one question, if i have local softphones, will this mean asterisk is going to stay in the middle of those phones as well? |
| 00:19.25 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 00:19.27 | Aces1Up | i don't want to unnessecarily load down my box. |
| 00:19.53 | *** join/#asterisk battini (n=inittab@cpe-24-209-36-174.neo.res.rr.com) |
| 00:20.16 | [TK]D-Fender | Aces1Up : yes, and this is typically a good thing anyways. your internal phones trying to reinvite to anything exterior will fail without it passinng through and its best just to leave it that way. |
| 00:20.26 | [TK]D-Fender | Aces1Up : For the record, what router on the * side? |
| 00:21.12 | [TK]D-Fender | Dr-Linux : and you know this message disturbs me... "line 4: Unable to read Zaptel version information." <- don't you think that's a good sign that may Zaptel didn't compile nice, or that you need to do a better job of clearing out 1.2? |
| 00:22.03 | Aces1Up | * i take it is remote side? its a linksys BEFSR41 V3 |
| 00:22.31 | [TK]D-Fender | Aces1Up : No, * = Asterisk |
| 00:23.21 | Aces1Up | :( d-link di-604 |
| 00:23.48 | [TK]D-Fender | Aces1Up : Hrm.. not sure on the 604 specifically, but I have heard on some D-Links being uncooperative NAT wise |
| 00:23.58 | [TK]D-Fender | aces, ok, well you changed everything, and.... ? |
| 00:24.04 | perd | haha fender |
| 00:26.32 | *** join/#asterisk Asterman (n=newkinet@shell2.sea5.speakeasy.net) |
| 00:26.54 | Asterman | hey peeps |
| 00:27.18 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 00:27.49 | Asterman | I asked this question the other day, but I think people were asleep or something... what's the main advantages of using the DB for holding * config rather than the /etc/asterisk files? (apart from it's in a DB and not text files) |
| 00:27.50 | Aces1Up | tkd acually trying to figure out how to set up those tps ports... |
| 00:27.56 | Aces1Up | rtp ports on the router. |
| 00:28.11 | Aces1Up | it only lets me forward 1 port at a time on this router, not a range. |
| 00:28.48 | [TK]D-Fender | Asterman : that IS the point you know. Being in a DB its more open to dynamic change without having to tell * to reload configs. |
| 00:29.19 | [TK]D-Fender | Aces1Up : Oh yes, because I wasn't explicit before, 5060,10000-20000 are all UDP <- |
| 00:29.27 | Asterman | TK : that's it though? no magical added surprises? |
| 00:29.46 | [TK]D-Fender | Asterman : What were you expecting? |
| 00:29.53 | wunderkin | not like you need all of those rtp ports |
| 00:30.02 | [TK]D-Fender | Asterman : Also easier to do dynamic SIP users, etc... |
| 00:30.25 | Asterman | TK : I don't mind just issuing a reload if it means things are kept super simple in text files. I dunno, I was expecting something with a lil more wow for the effort of setting up the DB :) |
| 00:30.30 | [TK]D-Fender | wunderkin : TECHNICALLY no, but geez, jsut work with the common standards first. |
| 00:30.42 | wunderkin | :P |
| 00:31.11 | [TK]D-Fender | Asterman : What would you imagine for it? its just a SOURCE of config ino. you think you could somehow specify MORE in there that isn't offered in the nomal text files? |
| 00:31.29 | perd | AEL is more WOW |
| 00:31.36 | perd | and O.O |
| 00:31.44 | perd | and probably even a little OMG |
| 00:31.55 | Asterman | TK : I figured there might be something like, I dunno, something like new features turned on because some module only worked with a DB |
| 00:32.18 | [TK]D-Fender | AEL is a complete waste of time. |
| 00:32.30 | danp | why is that? |
| 00:32.30 | [TK]D-Fender | Asterman : LOL. *no* |
| 00:32.36 | Aces1Up | tkd can i im you? |
| 00:33.28 | Asterman | TK : btw, thanks for your help earlier this week :) We've just moved offices here at work, and I convinced the company to ditch their old phone system and I just put in a new 50 phone * system with all the trims and SIP term/orig with everything working just fine. |
| 00:33.36 | [TK]D-Fender | danp : Historically unstable, allows you to create even MORE unreadable configs, and offers nothing that standard logic can't do. Consumes MEMORY just for its existance too, and stole time away from important things like REWRITING THE SIP STACK <- |
| 00:33.43 | Asterman | TK : we saved a boatload of $ :) |
| 00:33.55 | [TK]D-Fender | Asterman : I'm quite sure you would. |
| 00:34.21 | [TK]D-Fender | Aces1Up : if you're looking to PM a boatload of spam, then NO. use www.pastebin.ca |
| 00:34.41 | Asterman | TK : so what's your association with *, long time user, code contributor, digium guy? |
| 00:34.57 | [TK]D-Fender | MGCP = yeah nice for * to offer it... again, how many people truely care? |
| 00:35.34 | [TK]D-Fender | Asterman : Just a guy who likes its, uses it, and promotes technological freedom. I do MINOR scripting, and normals configs. |
| 00:35.49 | [TK]D-Fender | Asterman : I am also do consulted installs, etc. |
| 00:36.01 | Asterman | TK : cool...what region are you based? |
| 00:36.02 | [TK]D-Fender | wow... that sounds really broken :) |
| 00:36.09 | Asterman | lol |
| 00:36.12 | wunderkin | .. and hangs out here all of the time and helps everyone... and promotes polycom... and sometimes aastra ;p |
| 00:36.19 | [TK]D-Fender | Grammar challenged tonight... I must be preoccupied. |
| 00:36.37 | Asterman | hehehe...nothing wrong with promoting polycom, I love our new soundstation 4000 :) |
| 00:36.39 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-170-186.socal.res.rr.com) |
| 00:36.45 | [TK]D-Fender | wunderkin : Amongst many other products :) |
| 00:37.27 | [TK]D-Fender | Asterman : Personally being wired I wouldn't have done that personally. i run a SoundStation 2W (wireless analog) on an ATA. means great speakerphone wherev its needed |
| 00:37.58 | yassine | hi everyone i have a Motorola Wildcard X100P and for some reason i can not get it to work any idea if im missing some modules ? these are modules loaded in my kernel : http://rafb.net/p/Uk2W1183.html |
| 00:38.20 | Asterman | TK : interesting, the 4000's working great for speakerphone for us (it's in our main big conference room), and we could reuse the remote mics from our old polycom |
| 00:38.55 | [TK]D-Fender | Asterman : Oh I wasn't doubting its QUALITY. Just the MOBILITY factor, including the ability to bring it elsewhere than your own PBX. |
| 00:39.42 | Asterman | TK : ahhhh....yeah, I can see that, luckily the only time that phone will be moved is when we move offices again and seeing as we just did that last weekend I don't think we'll be doing it again for a few years ;) |
| 00:39.49 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 00:40.28 | Asterman | TK : so you never answered, whereabouts are you from btw? Cali here but originally from the UK |
| 00:41.00 | [TK]D-Fender | Asterman : Fully understand, and I also got mine BEFORE I got * in anticipation of succeding in selling the idea. We had a Norstar for which we could ahve gotten a proprietary one. I went the smarter way :) |
| 00:41.18 | *** join/#asterisk ManxPower (n=manxpowe@173.sub-75-200-248.myvzw.com) |
| 00:41.20 | [TK]D-Fender | Asterman : Sorry, that'd be Montreal, QC |
| 00:42.58 | Asterman | TK : nice, haven't made it out that way yet, Vancouver (Whistler) and Toronto have been my limited experiences of Canada, but loved both places |
| 00:43.38 | [TK]D-Fender | BC is pleasent, but Toronto is a cold flat place. Feels "dead" to me. |
| 00:44.49 | [TK]D-Fender | Montreal is "sometimes troubled" to those living here, but generally a multi-cultural haven to most visitors |
| 00:44.49 | perd | nova scotia is the place to be |
| 00:44.49 | Asterman | I was there in the summer time, pretty humid, I liked the cosmopolitan feel of the city centre, but after living in London for quite a while I get bored of cities |
| 00:45.12 | [TK]D-Fender | Nova Scotia is like watching astro-turf grow or paint dry. Sure you can sit and watch the time go by.... you'll just discover it had been WASTED :) |
| 00:45.27 | perd | camping there was great |
| 00:45.41 | Asterman | alrighty then....time for me to wrap up things here in the office and head out for the weekend, you guys have a safe and happy weekend, catch you later |
| 00:45.44 | perd | and there are a lot of geologically interesting areas |
| 00:45.45 | perd | :)_ |
| 00:46.01 | perd | and of course, the minus basin. need i say more!? |
| 00:46.25 | [TK]D-Fender | If Quebec were to actualyl succeed it its nationalistic attempts at sovereignty, the entire eastern seabord would become a 3rd world county (or would it be more appropriate to say FOURTH?) |
| 00:47.03 | perd | or minas basin |
| 00:47.43 | perd | na, just the parts that are in canada |
| 00:47.52 | perd | usa would be fine, we wouldnt let the smelly frogs in |
| 00:47.53 | perd | :P |
| 00:50.06 | [TK]D-Fender | USA? between the MCA, and the destruction of all the protections of the Constitution, the Bill Of Right (no longer friggen PLURAL), and so on I'm thinking I will do all in my poewr to avoid that side of the border. |
| 00:51.29 | perd | haha so true |
| 00:51.49 | perd | USA and UK are both pretty much f'd as far as the rights of non-elite go |
| 00:52.09 | [TK]D-Fender | UK... you mean Brave New World, right? |
| 00:52.22 | perd | pardon me, brave new world. |
| 00:52.27 | [TK]D-Fender | "Remember, remember the fifth of November..." |
| 00:52.32 | perd | i got my eye on youuuu, i got my eye on the bubblee |
| 00:53.26 | sevard | They've been talking about Quebec's seperation for years. They should just bomb it. |
| 00:53.45 | [TK]D-Fender | sevard : Been done :) |
| 00:55.04 | [TK]D-Fender | So far thats one good thing for the US. Succession = treason = federal death penalty. that would have nipped this in the bud... |
| 00:55.13 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:56.07 | perd | you fucking canuks are all fascists, if quebec wants to do their own thing you have no right to stop them |
| 00:56.11 | perd | :P |
| 00:56.26 | sevard | [TK]D-Fender: I need to buy that movie |
| 00:56.38 | [TK]D-Fender | perd : I'm from Quebec and I want my own way! |
| 00:56.45 | *** join/#asterisk ManxPower (n=manxpowe@68.113.119.198) |
| 00:56.49 | Juggie | sigh |
| 00:56.55 | Juggie | this isnt #asterisk-politics |
| 00:56.59 | sevard | He's SICK OF THE SWISS! |
| 00:57.10 | perd | ok so does quebec use T1 or E1 |
| 00:57.13 | file | Juggie: Swiss Chalet! |
| 00:57.14 | Qwell | Just to prove Juggie wrong :P |
| 00:57.17 | Juggie | T1 |
| 00:57.26 | Juggie | file, i'm on a diet ;) |
| 00:57.31 | Juggie | and i joined the gym. |
| 00:57.33 | perd | does the T1 communicate in french, canadian or english |
| 00:57.49 | Juggie | perd, you clearly havnt ever visited quebec. |
| 00:57.50 | [TK]D-Fender | sevard : The wiss? Nobody messes with the swiss because of conscripture. More crime is commited there by VISITORS than natives. |
| 00:57.53 | perd | lot of overhead for t1 canadian communications, eh |
| 00:57.54 | Juggie | they dont speak french |
| 00:57.58 | perd | haha no juggie i havent |
| 00:58.10 | perd | i just use what little knowledge i have to be an ass |
| 00:58.12 | Juggie | nothing in quebec is close to french. |
| 00:58.18 | Juggie | its more like Franglais |
| 00:58.18 | perd | poutine? |
| 00:58.28 | sevard | [TK]D-Fender: It's a KITH skit. You're not a true canadian if you don't know it. |
| 00:58.38 | [TK]D-Fender | Juggie : Je parle bilingue pour me sauver du temp, ostie! |
| 00:58.54 | Juggie | [TK]D-Fender, je parle francais aussi :) |
| 00:59.14 | perd | kith.. those guys were a little light in the pants, if you catch my drift |
| 00:59.14 | [TK]D-Fender | Juggie : Alors, mange un char de merde ;) |
| 00:59.26 | Juggie | i did french for 13 years, then i came to ontario and i had no idea what any of the frenchies were saying |
| 00:59.32 | Juggie | because they all talk in slang. |
| 00:59.50 | sevard | perd: Sounds like wishful thinking on your part |
| 00:59.53 | Juggie | and a french sentence here is spoken like a 53 sylilable word. |
| 00:59.59 | [TK]D-Fender | "I don't speak french or Hangligh.. I speak Quebecois et jouale!" |
| 01:01.15 | [TK]D-Fender | Juggie : http://www.youtube.com/watch?v=9STULzMLvbc |
| 01:02.21 | Juggie | i've seen this before |
| 01:02.24 | Juggie | and club super sex is a fantastic place. |
| 01:03.21 | [TK]D-Fender | Juggie : Depends what you're looking for I guess... |
| 01:03.28 | Juggie | i suppose :) |
| 01:03.56 | Juggie | i love montreal though, awesome city. |
| 01:04.42 | perd | club super sex? is that a brothel? |
| 01:05.26 | [TK]D-Fender | perd : Officially? ;) |
| 01:06.17 | perd | hah |
| 01:06.49 | [TK]D-Fender | ~trixbox |
| 01:06.51 | jbot | [trixbox] unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 01:07.31 | Aces1Up | hrmm, ok so whats the difference between trixbox and asterisk just curious > |
| 01:07.31 | Qwell | newbb? Are they trying to say something? |
| 01:07.45 | Qwell | Aces1Up: trixbox is a linux distribution, which just happens to include asterisk |
| 01:08.21 | nick125_lappy | Qwell: Haha |
| 01:08.29 | [TK]D-Fender | Aces1Up : Trixbox runs Freepbx which OWNS YOUR ASS. Don't dream of escaping the configs it makes for you. The second you go to commit a change all your maul work turns to ASH. |
| 01:08.58 | nick125_lappy | I remember when freepbx used to be called AMP and when i used to use AMP.. |
| 01:09.10 | nick125_lappy | the best day of my life is when I got rid of AMP and went pure-asterisk |
| 01:09.12 | Qwell | not to bash X-Rob_ or anything, but if freepbx were to not use a DB for the configs, it would be so much better, IMO |
| 01:09.30 | Grnd-Wire | [TK]D-Fender: I create my own config files, with a simple include.. Seems to work for me.. ? |
| 01:10.59 | [TK]D-Fender | Grnd-Wire : Congrats. You're still tied ot its real way of operating. Everything else might as well be considered a hack. |
| 01:11.04 | *** join/#asterisk SomethingISODD (n=dan@142-217-4-15.telebecinternet.net) |
| 01:11.34 | SomethingISODD | hello all question how do increase the time out when you drop a call in /var/spool/asterisk/outgoing? |
| 01:11.37 | sevard | nick125_lappy: I wouldn't call it the best day of my life, but I had a similar day, and, 'Yay It was Good' |
| 01:11.39 | SomethingISODD | so it doesn`t time out so fast |
| 01:11.53 | [TK]D-Fender | SomethingISODD : post-date the timestamp first |
| 01:12.30 | SomethingISODD | i don`t currently put a timestamp in it |
| 01:12.33 | Juggie | hah awesome |
| 01:12.38 | Grnd-Wire | [TK]D-Fender: hmm - That's true.. So where is the repository of all of the dialplan code that is freely useable? I heard once about something like that.. |
| 01:12.40 | SomethingISODD | could you possibly show me the format |
| 01:12.42 | sevard | you didn't, linux did. |
| 01:12.43 | Juggie | i submitted a bug in fedora core in september 2005 |
| 01:12.46 | Juggie | and someone just answered it |
| 01:13.09 | [TK]D-Fender | Grnd-Wire : ummm... what? |
| 01:13.24 | sevard | [TK]D-Fender: you mean asterickrecipies.com or voip-info.org ? |
| 01:13.42 | [TK]D-Fender | sevard : ? |
| 01:13.56 | sevard | aimed at Grnd-Wire, sorry I just woke up. |
| 01:14.11 | [TK]D-Fender | sevard : go caffeinate! |
| 01:14.13 | Grnd-Wire | sevard: oh! Is that what I'm asking about? I'll need to go look.. Sounds about right though.. :D asteriskrecipes.com |
| 01:14.49 | sevard | [TK]D-Fender: nothing near! I was supposed to drive four hours north today but took a loooooong nap instead. idiotic. |
| 01:15.28 | ManxPower | SomethingISODD: Which of the several timeouts are you referring to? |
| 01:16.03 | Grnd-Wire | sevard: hmm. It's astrecipes.net, just so you know.. :P |
| 01:16.04 | [TK]D-Fender | ok, time to head out for pool... later all |
| 01:16.32 | sevard | Grnd-Wire: sorry. |
| 01:16.48 | sevard | good thing google exists. |
| 01:16.59 | perd | i'm a fan of lemonparty |
| 01:17.55 | Grnd-Wire | sevard: Yeah! That's how I found it.. :D |
| 01:17.56 | *** join/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 01:18.01 | littleball | hello |
| 01:18.05 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 01:18.17 | *** join/#asterisk darius_ (n=darius@integrity.bourg.net) |
| 01:18.24 | littleball | who can help me configure the communication between two *. |
| 01:18.33 | *** join/#asterisk piper69 (n=piper@unaffiliated/piper69) |
| 01:19.15 | littleball | I have sip phone -->* 1 (answered the call, and then Dial(SIP/pstnno@asterisk2)--->* 2? -->PSTN. The problem here is that *2 answers the call immediately. What i expected is that the PSTN phone answer the call instead of *2 answer the call |
| 01:19.38 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:19.38 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:20.24 | ManxPower | littleball: Asterisk considers all calls to be answered as soon as dialing is finished when sending the call out an analog FXO port |
| 01:21.27 | piper69 | my friend used to have a voip with a company called lingo, he gave me his adaptor is it possable to use it with lingo |
| 01:22.37 | littleball | ManxPower, sip phone -->asterisk 1 (some dialplans , which Dial(SIP/pstnno@asterisk2) --->asterisk 2 --->PSTN. How to configure asterisk 2 to work as PSTN gateway? |
| 01:22.47 | piper69 | i am new here so i know i have alot to learn about and alot lot to reading is waitting me, can someone please guide me for a good start |
| 01:23.06 | ManxPower | littleball: Asterisk acts as a PSTN gateway automatically |
| 01:23.39 | ManxPower | witout know what things are executed in the dialplan before the dial, I cannot help further. How are you connecting to the PSTN on Asterisk 2 |
| 01:24.43 | ManxPower | Playback() and Background() will automatically answer the line unless you tell it not to. |
| 01:24.51 | littleball | E1/Zap channel. in asterisk 1, the dial plan is try to connect to PSTN. But there is no zap channel on asterisk 1. so, i try to use sip connect pass call to asterisk 2 |
| 01:25.20 | ManxPower | little E-1 or E-1 PRI? |
| 01:25.33 | littleball | ManxPower, E1 PRI. |
| 01:25.46 | ManxPower | littleball: then the problem is with the dialplan on Asterisk 2 |
| 01:26.04 | littleball | this is not the problem yet. The problem is the configuration or dialplan. |
| 01:26.26 | ManxPower | littleball: copy the CLI output on Asterisk 2 to pastebin.ca |
| 01:26.43 | littleball | on asterisk 1 dialplan, i got the pstn number to call and also, i know the asterisk 2 has zap channel |
| 01:26.54 | littleball | ok |
| 01:27.10 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:27.54 | Grnd-Wire | Can anyone explain to me why the fee for G.729, when all of the phones support it natively? Is it still a viable patent, and so Digium has to pay royalties? and all of the phone vendors are paying the royalties as well when you buy your phone? |
| 01:28.37 | ManxPower | Grnd-Wire: the phone vendor pays the patent licensing fees. More correctly, the companies that makes the codec chips pay the fee and pass the cost onto the phone makers, which pass it on to you |
| 01:28.57 | yassine | my card is reconized and its modules are loaded but its not working i really need your ideas guys its an x100p from motorola |
| 01:29.23 | ManxPower | yassine: what error message do you get? |
| 01:29.59 | yassine | ManxPower, ZT_CHANCONFIG failed on channel 5: No such device or address (6) |
| 01:30.14 | ManxPower | yassine: your /etc/zaptel.conf is wrong |
| 01:30.29 | Grnd-Wire | ManxPower: Just as I suspected.. and since I don't have to pay to use Asterisk.. there is no money changing hands, so therefore it doesn't work with Asterisk without the fee.. That makes alot of sense. :D |
| 01:31.06 | yassine | ManxPower, this all i have in zaptel.conf : 16 fxsks=5 # X100P 17 defaultzone=us 18 loadzone=us |
| 01:31.18 | blitzrage | Hrmm... when calling a static member from a Queue, is there any way to set a variable for the SIP channel that you are calling? Looks like setvar= in sip.conf is only called when that channel originates the call |
| 01:31.27 | ManxPower | yassine: your card is on channel 1 |
| 01:31.39 | blitzrage | other than calling a Local channel and setting the variables before calling the device |
| 01:32.06 | ManxPower | blitzrage: the only way I can think of is to use Local/ |
| 01:32.16 | blitzrage | yah, I think you're right ManxPower |
| 01:32.30 | blitzrage | although, I just realized the username I need is probably in the CHANNEL variable |
| 01:32.36 | blitzrage | so I think I'm going to get lucky this time |
| 01:32.59 | yassine | ManxPower, but cat /proc/interrupts -->5: 421958827 XT-PIC Intel 82801BA-ICH2, wcfxo |
| 01:33.10 | ManxPower | yassine: that is the IRQ, not the channel |
| 01:33.33 | yassine | ManxPower, i changed it to 1 now and i still get the same error |
| 01:34.06 | ManxPower | yassine: your card is sharing interrupt with Intel 82801BA-ICH2 which will eventually cause problems. |
| 01:34.14 | ManxPower | yassine: did you rerun ztcfg again? |
| 01:34.21 | yassine | yes |
| 01:34.27 | littleball | hi, ManxPower, what is the link of pastin? |
| 01:34.33 | ManxPower | yassine: then the card is not being detected correctly |
| 01:34.39 | ManxPower | littleball: pastebin.ca |
| 01:34.41 | ManxPower | ~pb |
| 01:34.51 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:35.08 | littleball | http://www.pastebin.ca/322115 |
| 01:35.13 | yassine | ManxPower, coeur2lion:/etc# dmesg | grep Wildcard |
| 01:35.14 | yassine | Found a Wildcard FXO: Wildcard X100P |
| 01:35.56 | littleball | ManxPower, i pasted |
| 01:36.10 | ManxPower | checking |
| 01:37.33 | ManxPower | littleball: I see no indication that the call was answered |
| 01:38.19 | ManxPower | littleball: if you want to know WHY the call failed put a Noop(HANGUPCAUSE is ${HANGUPCAUSE}) as the priority after Dial |
| 01:38.57 | littleball | ok. let me do now |
| 01:39.07 | littleball | put in asterisk 2? |
| 01:39.10 | ManxPower | the value of hangupcause will tell you what the failure is |
| 01:39.15 | ManxPower | littleball: yes |
| 01:39.25 | ManxPower | littleball: the call was not answered. It failed |
| 01:39.33 | yassine | ManxPower, you can see here that the card is being loaded and reconized correctly : http://rafb.net/p/h4zmY185.html |
| 01:40.05 | ManxPower | yassine: put the contents of /etc/zaptel.conf on pastebin.ca |
| 01:40.17 | yassine | ManxPower, okay |
| 01:40.56 | yassine | ManxPower, http://rafb.net/p/d9PoBq90.html |
| 01:41.50 | littleball | ManxPower, http://www.pastebin.ca/322119 |
| 01:42.18 | ManxPower | yassine: it should be working, the only thing can suggest is to fix the IRQ problem |
| 01:42.42 | yassine | mhh and how ? :s |
| 01:43.27 | ManxPower | littleball: http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
| 01:43.46 | ManxPower | littleball: I suspect that the switchtype the telco has configured is not the same as you have configured. |
| 01:44.09 | littleball | but i can call from sip phones which connect to asterisk 2 directly |
| 01:44.40 | ManxPower | yassine: you either assign the IRQ to the slot the card is in via the BIOS or you must move the card to a different slot if the BIOS does not allow you to assign the IRQs in the BIOS |
| 01:45.28 | yassine | ManxPower, i will try to put the card in an other slot since the box is open now |
| 01:46.10 | ManxPower | littleball: it still could be a switchtype issue. |
| 01:46.32 | littleball | ManxPower, would u mind explain what is the difference here? |
| 01:47.03 | littleball | and how to solve this problem by only changing either *1 or *2 configuratin? |
| 01:47.06 | ManxPower | littleball: Asterisk will try to translate the SIP information to PRI information. If the switchtype is wrong, some things will work, but some things will not work. |
| 01:47.21 | ManxPower | littleball: change the switchtype= line in Asterisk 2 |
| 01:47.40 | littleball | you mean the zap config file? |
| 01:47.45 | ManxPower | correct. |
| 01:47.55 | littleball | ok. thanks let me try now |
| 01:48.15 | perd | pequinotesticles |
| 01:48.20 | littleball | my current value is euroisdn |
| 01:49.05 | ManxPower | littleball: I have never in my 5 years of working with PRI and Asterisk seen a HANGUPCAUSE of 100 |
| 01:49.32 | littleball | the same result. shoue i restart the asterisk server after changing switchtype? |
| 01:49.35 | littleball | or just reload |
| 01:49.43 | ManxPower | do a restart |
| 01:49.43 | yassine | ManxPower, its fine now i can run ztcfg with no errors :) merci |
| 01:50.13 | ManxPower | yassine: I have been doing Asterisk for a very long time. |
| 01:51.00 | yassine | i dont doubt that at all :) |
| 01:52.10 | yassine | can i dial from asterisk command line ? |
| 01:52.39 | ManxPower | yassine: in you have a sound card and if you configure it. |
| 01:53.06 | yassine | i have a sound card let me configure it |
| 01:54.19 | ManxPower | More proof that the end of the world is near: The New Orleans Saints made it to the NFL playoffs. |
| 01:54.32 | littleball | ManxPower, if i set to switchtype to line, i cannot start the asterisk. and it could not be found line type of switchtype |
| 01:54.51 | ManxPower | littleball: what country are you in? |
| 01:54.55 | littleball | Jan 20 10:10:00 ERROR[11216] chan_zap.c: Unknown switchtype 'line' |
| 01:54.56 | littleball | Singapore |
| 01:55.10 | ManxPower | littleball: put it back to euroisdn |
| 01:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:55.14 | littleball | yes |
| 01:55.21 | ManxPower | littleball: I can't think of anything else to suggest. |
| 01:55.40 | littleball | already. i am wondering whether using IAX to communicate between two asterisks wlll solve this problem or not |
| 01:55.47 | *** join/#asterisk inv_Arp (n=junya@c-75-74-183-191.hsd1.fl.comcast.net) |
| 01:56.13 | Grnd-Wire | littleball: Are you using a PRI card to link two Asterisk machines together? |
| 01:56.23 | littleball | no |
| 01:56.36 | littleball | i am using SIP over IP |
| 01:57.27 | littleball | Grnd_Wire, i have sip phone to *1 . but *1 has no ZAP channel to terminate. So, i try to connect *1 to *2 . *2 has ZAP channels |
| 01:58.24 | Grnd-Wire | littleball: oh, well the only input I have for that is if you're linking to Asterisk boxes together, you definately SHOULD be using IAX.. but it sounds like you're having PRI issues, at least that's what ManxPower thinks, and I'm not in a position to doubt that. :D |
| 01:58.32 | yassine | ManxPower, now sound card is configured |
| 01:58.43 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 01:58.45 | Grnd-Wire | I'll be buying myself a T-1 board next week, but it'll be so I can link up with Merlin Magix and test the integration.. :D |
| 01:59.06 | littleball | Grnd-Wire, yes. i will try IAX. But i thought SIP should do this also, right? |
| 01:59.17 | ManxPower | SIP should work. |
| 01:59.27 | ManxPower | In fact I use the EXACT same setup with T-1 PRI |
| 01:59.55 | littleball | ManxPower, would you mind post you sip configuraion to me? in both *1 and *2 |
| 02:00.07 | littleball | i am a bit confused by the peer/user/friend |
| 02:00.35 | ManxPower | littleball: on Asterisk 1 I use type=peer on Asterisk 2 I use type=user |
| 02:00.43 | littleball | ok. NAT? |
| 02:00.50 | littleball | set both to YES? |
| 02:00.52 | ManxPower | no NAT |
| 02:00.56 | littleball | ok. |
| 02:01.06 | littleball | actually doesnt matter because of public ip |
| 02:01.38 | *** join/#asterisk osas (n=nnnnnnnn@CABLE-72-53-75-252.cia.com) |
| 02:01.49 | littleball | ManxPower, should asterisk 1 register as an user in asterisk 2? |
| 02:01.59 | littleball | actually, i did this |
| 02:01.59 | yassine | ManxPower, and idea how i could try to dial from asterisk cmd line ? |
| 02:02.21 | littleball | yassine, help Dial |
| 02:03.44 | yassine | is this correct : Dial(LOCAL/004916292239000) ?? |
| 02:03.52 | littleball | ManxPower, i have set both to peer. is this could be the reason? |
| 02:04.04 | ManxPower | littleball: I guess it could be a problem |
| 02:04.25 | littleball | yassine, i only use it to dial my zap channel. ManxPower, let me try now |
| 02:05.09 | littleball | ManxPower, do you have a register => in asterisk 1? |
| 02:05.23 | littleball | ManxPower, do you have a register => xxxxx in sip.conf of asterisk 1? |
| 02:06.17 | ManxPower | no. each box is on a static IP and so registration is not required |
| 02:07.34 | littleball | ok |
| 02:08.01 | littleball | ManxPower, if i put type to user on *2, then the registration from *1 will fail |
| 02:08.14 | littleball | <PROTECTED> |
| 02:08.25 | littleball | nevermind. let me remove registrer first |
| 02:08.32 | ManxPower | a USER MAKES calls, a PEER RECEIVES calls |
| 02:13.16 | littleball | ManxPower, both USER and PEER refer to CHANNEL? right? |
| 02:14.20 | littleball | still not work. let me try IAX |
| 02:14.37 | littleball | it is not codec issue. i allow all codec already |
| 02:19.52 | ManxPower | That could be the problem. |
| 02:20.10 | ManxPower | allowing all codecs could cause a codec to be used that cannot be converted to the pstn |
| 02:20.17 | ManxPower | disallow=all and allow=alaw |
| 02:20.59 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:22.36 | littleball | ok. let me try. |
| 02:23.00 | *** join/#asterisk Kerry_G (n=Kerry_G@ip68-5-206-230.oc.oc.cox.net) |
| 02:23.07 | *** part/#asterisk Kerry_G (n=Kerry_G@ip68-5-206-230.oc.oc.cox.net) |
| 02:24.27 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 02:24.31 | littleball | ManxPower, the same. no chang |
| 02:24.33 | *** join/#asterisk javauser (n=rdtreefo@pool-71-244-56-112.dllstx.fios.verizon.net) |
| 02:26.24 | *** join/#asterisk ez` (n=Ez@c66.203.210-59.clta.globetrotter.net) |
| 02:27.37 | javauser | is there a good full features gui for 1.4 yet? i want to use the new features, but get a headache when i think of teaching my users to maintain the config files |
| 02:27.40 | *** join/#asterisk karmatronic (n=karmatro@84.77.137.35) |
| 02:32.12 | *** join/#asterisk connect321231654 (i=PJirc@24-197-105-141.dhcp.buft.sc.charter.com) |
| 02:33.17 | connect321231654 | Can someone help me with a couple of lines on a custom script |
| 02:33.21 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 02:34.21 | connect321231654 | I need help with sending variables with the goto command |
| 02:35.27 | connect321231654 | Can anyone see me typing? |
| 02:35.54 | battini | i cant see past your annoying nickname, sorry :/ |
| 02:37.36 | connect321231654 | I didn't want this stupid name but that was the only I could get in here |
| 02:53.05 | *** join/#asterisk Urgo (i=Feh45@ool-4571bccf.dyn.optonline.net) |
| 02:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:56.57 | *** part/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 02:57.08 | *** join/#asterisk brussel_ (n=brussel@cpe-75-80-175-84.san.res.rr.com) |
| 02:57.25 | javauser | come one, they gave me java user -- theres gota be a lot of good names left. ircs like a ghost town these days |
| 03:03.40 | *** join/#asterisk enk4yptd1 (n=cshore@c-75-69-191-42.hsd1.vt.comcast.net) |
| 03:06.48 | coppice | a ghost town? so all the spooky names will have been taken, then? |
| 03:09.36 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:13.31 | javauser | nah, just casper through casper9999999 |
| 03:13.39 | *** join/#asterisk ahattar (n=aha@ool-4354b3d3.dyn.optonline.net) |
| 03:16.02 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 03:16.19 | ahattar | hi, is there any docs about configuring zoom voipata on asterisk? |
| 03:21.28 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 03:32.29 | *** join/#asterisk brussel_ (n=brussel@cpe-75-80-175-84.san.res.rr.com) |
| 03:36.55 | *** join/#asterisk brussel (n=brussel@cpe-75-80-175-84.san.res.rr.com) |
| 03:38.08 | *** join/#asterisk enk4yptd1 (n=cshore@c-75-69-191-42.hsd1.vt.comcast.net) |
| 03:39.08 | yassine | gn8 everyone |
| 03:41.36 | *** join/#asterisk marty-ott-athome (n=me@host-209-50-87-86.dyn.295.ca) |
| 03:43.05 | marty-ott-athome | hey, I'm expanding my testing here. My Asterisk server is behing a router doing NAT. I tried setting nat=yes and externip = x.x.x.x - is ther anything else I need to do? I have only one phone that manage to register.. |
| 03:43.20 | marty-ott-athome | a shitty phone that probably doesn't take account of who's returning hte packet |
| 03:44.49 | sivana[work] | what are you trying to do? |
| 03:45.57 | marty-ott-athome | just get 2 phones and voip gateway to register ot the ASterisk box. |
| 03:46.31 | marty-ott-athome | EXCELLENT - I just got the voip gateway to work |
| 03:46.44 | marty-ott-athome | HAve 1 phone left... Grandstream is being a pain.. |
| 03:46.45 | sivana[work] | what kind of phone is not registering |
| 03:47.05 | sivana[work] | are both phones Grandstreams? |
| 03:47.18 | marty-ott-athome | .. Grandstream. I've got a Mediatrix that registered. I've got a noname chinesephone that registered. Here's my log: |
| 03:47.38 | marty-ott-athome | <PROTECTED> |
| 03:48.19 | sivana[work] | did you enable nat in the grandstream? |
| 03:48.41 | marty-ott-athome | Sure did.. I tried Nat with / without STUN server. I even set the NAt IP |
| 03:48.42 | sivana[work] | I haven't used them in a long time, but I seem to recall some kind of checkbox or something |
| 03:49.13 | sivana[work] | it would definately be without STUN unless you have a STUN server :) |
| 03:49.28 | sivana[work] | and your sip peer has nat=yes? |
| 03:49.47 | marty-ott-athome | well, just a public stun server like stun.fwdnet.net ... yep. Even my Mediatrix Gateway works now |
| 03:50.22 | marty-ott-athome | yeah.. I'm not too sure what to make of the Grandstream.. |
| 03:50.34 | sivana[work] | I'll never order another one :) |
| 03:50.55 | [TK]D-Fender | Marty-OTT : Plenty more you need. pastebin your sip.conf [general] section |
| 03:50.56 | [TK]D-Fender | ~pb |
| 03:50.59 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:51.30 | marty-ott-athome | sure... but remember, my Mediatrix Gateway AND my No Name chinese phone are registered and they're all beside me behind my router |
| 03:51.49 | sivana[work] | ~[TK]D-Fender |
| 03:51.50 | jbot | rumour has it, [tk]d-fender is rockin' the casbah !!! |
| 03:52.19 | sivana[work] | ~sivana |
| 03:52.21 | jbot | [sivana] not exactly the sharpest tool in the shed |
| 03:52.28 | [TK]D-Fender | huzzah! |
| 03:52.30 | [TK]D-Fender | lol |
| 03:52.32 | sivana[work] | :) |
| 03:53.44 | marty-ott-athome | Well.. here you go.. hope you have all you need there.. |
| 03:53.46 | marty-ott-athome | http://rafb.net/p/Esiw4q50.html |
| 03:54.32 | marty-ott-athome | I keep getting these: - Got SIP response 400 "Bad Request" back from 209.50.87.86 |
| 03:54.44 | marty-ott-athome | I unplugged my Grandstream to see if it's that generating those |
| 03:54.46 | sivana[work] | ugh |
| 03:55.02 | sivana[work] | sip debug |
| 03:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:55.13 | marty-ott-athome | DOH... still getting those.. |
| 03:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:55.22 | [TK]D-Fender | marty-ott-athome : you are missing a LOT. |
| 03:55.46 | [TK]D-Fender | marty-ott-athome : very importantly your localnet, canreinvite=no, and so on. |
| 03:56.06 | marty-ott-athome | in each user? |
| 03:56.14 | sivana[work] | in general |
| 03:56.15 | [TK]D-Fender | marty-ott-athome : Also a lot more of the common stuff you'll need. What ports have your forwarded to your * box? |
| 03:56.23 | [TK]D-Fender | Yes, in [general] |
| 03:56.33 | marty-ott-athome | one -to-one nat |
| 03:56.37 | marty-ott-athome | on the Asterisk box |
| 03:56.59 | marty-ott-athome | so 192.168.1.201 is mapped to 209.x.x.x |
| 03:58.03 | marty-ott-athome | Yeah, stuff isn't right.. I can get the phones to ring between the devices that did register but no sound is getting through |
| 03:58.09 | sivana[work] | ~seen |
| 03:58.30 | nick125_lappy | marty-ott-athome: you'll want to set localnet=192.168.1.0/255.255.255.0 and externhost to the 209.x.x.x address |
| 03:59.19 | marty-ott-athome | cool.. I'll try right now |
| 03:59.32 | *** join/#asterisk brussel (n=brussel@cpe-75-80-175-84.san.res.rr.com) |
| 03:59.42 | nick125_lappy | that should fix any nat issues |
| 03:59.47 | nick125_lappy | (well, most) |
| 04:00.03 | [TK]D-Fender | marty-ott-athome : Just because you're so far out... http://www.pastebin.ca/322190 |
| 04:00.45 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 04:01.09 | blitzrage | fo |
| 04:01.16 | sivana[work] | jo |
| 04:01.23 | marty-ott-athome | thanks TK |
| 04:01.24 | [TK]D-Fender | fum? |
| 04:02.03 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:02.39 | nick125_lappy | I've been waiting for 3 days for a DID from vitelity :/ |
| 04:03.13 | *** join/#asterisk brussel_ (n=brussel@cpe-75-80-175-84.san.res.rr.com) |
| 04:03.38 | asdx | nick125_lappy: ugh |
| 04:04.05 | *** join/#asterisk brussel_ (n=brussel@cpe-75-80-175-84.san.res.rr.com) |
| 04:05.55 | Grnd-Wire | Is there a way in Asterisk to concatenate additional text onto the end of a variable that already exists?? |
| 04:06.27 | [TK]D-Fender | Grnd-Wire : Set(myvar=${marvar}plusextracrap) |
| 04:06.55 | marty-ott-athome | hmmm.. is it possible that I may have voice issues before the phones are on a local private ip network here off the same switch and that Aterisk is on anohte private IP network somewhere else? |
| 04:07.45 | [TK]D-Fender | marty-ott-athome : That sounded like english... but you are making no sense at all! |
| 04:08.04 | marty-ott-athome | yeah.. I figured as much |
| 04:08.37 | marty-ott-athome | Well, put it this way. I've got the server at my workplace on a private IP mapped one-to-one to a public ip |
| 04:08.52 | Grnd-Wire | [TK]D-Fender: hmm.. ok, I was on the right track then, I think it may have just been my execution.. Let me try this one more time. :) |
| 04:08.54 | marty-ott-athome | I've got 3 voip devices at my home: 2 phones and 1 analog voip gateway |
| 04:08.58 | [TK]D-Fender | marty-ott-athome : You port forwarded it? |
| 04:09.22 | marty-ott-athome | shouldn't have to - it's one to one Nat |
| 04:09.53 | marty-ott-athome | One phone and the voip gateway will register and ring each other but the Grandstream won't even register |
| 04:10.06 | Grnd-Wire | [TK]D-Fender: yay! Thanks.. I was using too much whitespace in the code - apparently Asterisk really hates that.. :P |
| 04:10.08 | [TK]D-Fender | marty-ott-athome : fine, so work is NAT'd, and you want to have multiple SIP devices behind your NAT at home? |
| 04:10.28 | marty-ott-athome | exactly.. |
| 04:10.34 | [TK]D-Fender | Grnd-Wire : NO spaces around the "=" if you know whats ood for you.. |
| 04:10.57 | Grnd-Wire | [TK]D-Fender: heh - Well, I'm just now learning what's good for me.. :P |
| 04:10.57 | [TK]D-Fender | Marty-OTT : then each of your SIP devices at hom have to be told to use a different port. (not all on 5060) |
| 04:11.03 | marty-ott-athome | When I deploy this for real, I'm going to have the voip gateway and the asterisk server on public IPs. But right now, I'm testing and everything is behind nat. |
| 04:11.51 | [TK]D-Fender | nd of course *'s peers must be set up accordingly. |
| 04:12.17 | marty-ott-athome | ohhh... so, you mean, change the sip source port on the devices so that when Asterisk replies.... |
| 04:13.29 | marty-ott-athome | ok, so, for example, the local sip port on this Grandstream is 5060 - say I change it to 5061 |
| 04:13.42 | marty-ott-athome | .. I'd have to set that in sip.conf as well for that device? |
| 04:14.38 | sivana[work] | hrm.. most phone will automatically pick a free port if 5060 is taken |
| 04:15.48 | marty-ott-athome | Well Grandstream (looking at it's config) defaults to 5060.. but what's the paramater to match in sip.conf? |
| 04:16.31 | marty-ott-athome | (if I'm understanding this properly) |
| 04:17.18 | sivana[work] | change your gs to 5061 |
| 04:17.25 | marty-ott-athome | ok.. |
| 04:17.38 | marty-ott-athome | but do I have to change anything in the sip.conf file to match for the gs? |
| 04:17.44 | [TK]D-Fender | Marty-OTT : "port=5066" |
| 04:17.53 | marty-ott-athome | .... |
| 04:18.02 | marty-ott-athome | should of guessed |
| 04:18.08 | sivana[work] | I have port=5060 in sip.conf and phones that range in 5060-5061 |
| 04:18.09 | [TK]D-Fender | yes.. THAT obvious a parameter |
| 04:18.17 | sivana[work] | and they both register |
| 04:18.54 | [TK]D-Fender | sivana[work] : if all phones expect an answer on 5060, when your router gets a packet, how is it supposed to know which phone to send it to? |
| 04:18.54 | marty-ott-athome | gs rebooting now |
| 04:19.24 | [TK]D-Fender | Please note : REGISTERING DOESN'T MEAN SHIT |
| 04:19.25 | sivana[work] | I only have one port= and that's in general |
| 04:19.30 | sivana[work] | and it's 5060 |
| 04:19.43 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 04:19.45 | marty-ott-athome | OH!... yeah, that Chinese phone registered with 5020 |
| 04:19.48 | sivana[work] | 192.168.0.1:5060 |
| 04:19.49 | sivana[work] | 192.168.0.1:5061 |
| 04:19.55 | [TK]D-Fender | sivana[work] : that's *'s expected port, as well it SHOULD be. |
| 04:20.36 | sivana[work] | registering tells *, if you need to find me, i'm here |
| 04:20.42 | sivana[work] | at ip:port |
| 04:21.13 | *** join/#asterisk lorinc (n=ang@caracas-1955.adsl.interware.hu) |
| 04:21.26 | [TK]D-Fender | sivana[work] : Except where *5* devices all think they want that port :) However your setup is referring to having * listen, for which it only needs the one port |
| 04:21.31 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 04:21.44 | [TK]D-Fender | sivana[work] : Works for *, not the SIP clients |
| 04:21.48 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 04:22.05 | sivana[work] | maybe I'm spoiled with smart sip clients :) |
| 04:22.07 | marty-ott-athome | What a pain.. maybe I'll leave Grandstream alone and concentrate on making the voice work with the gateway and the chinese phone |
| 04:22.17 | sivana[work] | it didn't register? |
| 04:22.25 | marty-ott-athome | nope |
| 04:22.31 | sivana[work] | using port 5061? |
| 04:22.36 | marty-ott-athome | yep |
| 04:22.47 | sivana[work] | nothing else is on 5061? |
| 04:23.03 | [TK]D-Fender | marty-ott-athome : Using "smart", "grandstream", and "chinese phone" in the same sentence like that is contradictory enough :) |
| 04:23.09 | sivana[work] | lol |
| 04:23.21 | [TK]D-Fender | GrandSuck |
| 04:23.32 | sivana[work] | yea... I'll never buy one again |
| 04:23.45 | sivana[work] | had the old gs101 and gs102 |
| 04:23.48 | marty-ott-athome | no.... lol... The Mediatrix is the gateay and I've got an analog phone punched down on the amphenol |
| 04:23.48 | sivana[work] | joke |
| 04:24.39 | marty-ott-athome | ok, so screw the GS. Next.. trying to get the voice to work from the Mediatrix gateway and the chinese phone. What could keep my voice from getting through? Boht phones ring (mediatrix analog and chinese phone) |
| 04:24.57 | sivana[work] | qualify=yes |
| 04:25.17 | sivana[work] | I bet the firewall is closing up after so long |
| 04:25.34 | marty-ott-athome | I do have qualify=yes |
| 04:25.36 | sivana[work] | the phones ring? |
| 04:25.39 | marty-ott-athome | Going to call myself right now - yep |
| 04:25.48 | marty-ott-athome | ring |
| 04:26.14 | marty-ott-athome | Crap - yeah voice not getting through at all |
| 04:26.24 | sivana[work] | both ways or one way |
| 04:26.29 | marty-ott-athome | both ways.. |
| 04:26.34 | sivana[work] | what kind of firewall? |
| 04:27.01 | marty-ott-athome | just a linksys here at my house... ... OH.. waitaminute...I'm an idiot.. |
| 04:27.15 | marty-ott-athome | basic stuff... I have to get the RTP port through |
| 04:27.26 | sivana[work] | if only I had a nickel everytime I said that |
| 04:27.53 | sivana[work] | :) |
| 04:27.54 | [TK]D-Fender | Don't spend it all in one place, ok! |
| 04:28.05 | marty-ott-athome | What's the RTP port range real quick.. |
| 04:28.12 | sivana[work] | rtp.conf |
| 04:28.20 | sivana[work] | 10k - 20k I think |
| 04:29.05 | marty-ott-athome | k.. man trying to find the spot in the linksys to let those through.. |
| 04:29.05 | *** join/#asterisk Growly (n=himself@125-236-140-42.broadband-telecom.global-gateway.net.nz) |
| 04:30.27 | sivana[work] | do you have a firewall on the server? |
| 04:30.39 | sivana[work] | I've never had to mess with the firewall on the client side |
| 04:31.28 | marty-ott-athome | ok... no no firwall on the server side |
| 04:31.31 | marty-ott-athome | one to one nat |
| 04:31.39 | sivana[work] | nat != firewall |
| 04:31.49 | marty-ott-athome | gonna try again |
| 04:32.20 | marty-ott-athome | no work.. let me check router config at office |
| 04:33.13 | marty-ott-athome | yeah, damn, that wasn't the problem |
| 04:33.54 | marty-ott-athome | damn... this bites |
| 04:34.28 | sivana[work] | what kind of router at office? |
| 04:34.53 | marty-ott-athome | cisco box |
| 04:35.40 | marty-ott-athome | have you ever been eable to communicate between 2 voip phones on the same private lan but where the server is on another private lan somehwere else? |
| 04:35.53 | *** join/#asterisk adker_ (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 04:36.03 | sivana[work] | you have no access-group ? |
| 04:36.15 | sivana[work] | marty-ott-athome: all the time |
| 04:36.33 | sivana[work] | we use NAT as well... not the same as firewall |
| 04:37.05 | [TK]D-Fender | marty-ott-athome .... tell me thats not a PIX.... |
| 04:37.05 | sivana[work] | sorry... do you have an access-list defined on the server router? |
| 04:37.18 | marty-ott-athome | yes.. but is't definitely not blocking anything ... not it's a 2601 |
| 04:37.32 | marty-ott-athome | no ACLs on the server either |
| 04:37.37 | marty-ott-athome | hmmm.. |
| 04:37.39 | sivana[work] | so you have something like |
| 04:37.47 | sivana[work] | permit udp any host 209.91.155.155 |
| 04:37.58 | sivana[work] | of course your ip is different |
| 04:38.02 | [TK]D-Fender | PIX has been notorious for SIP/RTP. it is on the blacklist... |
| 04:38.18 | marty-ott-athome | first it's an ACL blocking part of the office from getting access to another part of the office but permit ip any any at the end ... all good |
| 04:38.25 | marty-ott-athome | good to know... |
| 04:39.04 | sivana[work] | I doubt it's the client side |
| 04:39.12 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 04:39.23 | marty-ott-athome | think something is missing in my asterisk config eh.. |
| 04:39.42 | marty-ott-athome | well.. I'm sure if one of these phones was somewhere else.. behind ANOTHER public ip - it would probably be fine |
| 04:40.01 | xpot | marty: are you able to test such a senerio? |
| 04:40.01 | sivana[work] | no.. we use linksys routers all the time |
| 04:40.23 | marty-ott-athome | no.. unfornately |
| 04:41.04 | marty-ott-athome | the 2 phones ring.. they establish a path .. jsut no voice.. sounds like an RTP issue |
| 04:41.05 | xpot | what ports are you using for RTP? |
| 04:41.27 | marty-ott-athome | on the aserisk servers... default 10000 - 20000 |
| 04:42.09 | xpot | ok, are you using SIP or IAX? (I might have missed this if you already stated earlier) |
| 04:42.16 | xpot | or other |
| 04:42.26 | marty-ott-athome | sip |
| 04:43.02 | marty-ott-athome | it's basically: ASTERIK SERER AT WORK ------ Internet ----------------- home linksys router and 3 sip devices (2 phones and 1 sip gateway) |
| 04:43.36 | marty-ott-athome | The Asterisk server has a private IP but a one-to-one nat.. it's private ip of 192.168.1.201 is mappted to 209.50.x.x |
| 04:44.15 | xpot | ok, on asterisk server console try 'sip show peers' and see what port is the phone is registering on |
| 04:44.51 | marty-ott-athome | you know what... woud the following be possible? (5020 and 5060)... would it be possible that.. |
| 04:46.01 | xpot | well 5060 should be the port that connects to the AST server, and where you get your ring... but 'sip show peers' should display the RTP port you are using |
| 04:46.08 | marty-ott-athome | the 2 phones register and when I place a call from one to the other.. it has to go through Aterisk. Once the path is established.. the RTP is attempted between the 2 devices. the problem is the source IP in the sip message is the public ip so when the other phone receives the rtp packet, it sends it back out the router, as opposed to back to the other phone... |
| 04:46.32 | marty-ott-athome | xpot: I'll paste the sip show peers |
| 04:46.36 | marty-ott-athome | in rafb |
| 04:46.58 | sivana[work] | marty-ott-athome: do you have canreinvite=no in both peers? |
| 04:47.16 | marty-ott-athome | http://rafb.net/p/9Va2aR64.html |
| 04:47.30 | marty-ott-athome | let me have a look I don't thikn it's set - what's the default? |
| 04:47.31 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:47.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:47.49 | sivana[work] | not sure the default |
| 04:48.00 | marty-ott-athome | what do you think of my theory though - could very well make sense |
| 04:48.30 | marty-ott-athome | yeah - it's set to no |
| 04:48.40 | sivana[work] | you need to nat them all |
| 04:48.50 | sivana[work] | I only see #8 as having nat=yes |
| 04:49.04 | xpot | I have actually had the same problem before: it turned out my linksys router was not allowing the RTP ports through |
| 04:49.32 | xpot | I am thinking this my be related to the linksys router; do you have another router you can test with? |
| 04:49.33 | marty-ott-athome | odd... 224 definitely has NAT - the others I don't care. |
| 04:49.48 | marty-ott-athome | xpot: No.. I forwarded ... ohhhhhhhhhhhhhhhhh |
| 04:49.55 | marty-ott-athome | yeah. |
| 04:49.57 | marty-ott-athome | hmmm |
| 04:50.11 | sivana[work] | are those other ones online? |
| 04:50.51 | marty-ott-athome | damn... 224 was missing the nat=yes... no - the only ones I care about are the 7729232, 824615whatever and 224 |
| 04:51.24 | marty-ott-athome | yeah, I thin kit it an rtp issue |
| 04:51.32 | marty-ott-athome | I'm going back to the linksys for a minute |
| 04:51.42 | sivana[work] | did you fix the nat=yes issue |
| 04:52.11 | marty-ott-athome | but wait... *yes I did* .... is RTP exchanged between the server and the phones or just between te phones once the path is established? |
| 04:52.23 | xpot | the linksys router I was using was an N router (piece of crap); ended up switching to a dlink and all worked successfully. |
| 04:52.48 | sivana[work] | depends... canreinvite will move the rtp |
| 04:53.13 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 04:54.50 | sivana[work] | with canreinvite=no, * stays in the path |
| 04:55.07 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:55.13 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:55.49 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 04:56.17 | marty-ott-athome | sorry - back... |
| 04:56.39 | marty-ott-athome | ok, well, I have a feeling it's linksys issue.. trying one last time and then it's sex with the girlfriend time |
| 04:57.54 | marty-ott-athome | yeah.. screw it.. can't get RTP through.. man alive.. this was a lot of work for nothing. I'll have to hand the phone to someone else who's behind a different public ip and has a better friewall |
| 04:58.40 | sivana[work] | try pluggin the phone directly into the internet |
| 04:59.22 | marty-ott-athome | Yeah.. I'm sure it would work but I'd have no one to call ... lol!! I'd just see hte Register on Asterisk. I had all these devices in my office earlier (no nat) ... everything was cool |
| 04:59.25 | sivana[work] | or maybe assign the dmz to it |
| 04:59.30 | marty-ott-athome | no nat, no firewall. |
| 04:59.38 | marty-ott-athome | dmz.. yeah.. |
| 04:59.43 | marty-ott-athome | let me check that again on thelinksy |
| 05:00.16 | sivana[work] | I don't know.. personally, I've never had issues with nat |
| 05:00.47 | xpot | you may want to verify that the DMZ setting is actualy working on the linksys, my experience is linksys was ignoring my settings |
| 05:00.56 | marty-ott-athome | yeah, i don't think it's a nat issue - I think it's a firewall issue with the linksys box. the DMZ had just one option: enable/disable and ip of host... ?? |
| 05:01.13 | sivana[work] | yea |
| 05:01.22 | sivana[work] | give it the private IP of the phone |
| 05:01.32 | marty-ott-athome | but what about the other phones? |
| 05:01.41 | sivana[work] | you can only dmz 1 ip |
| 05:02.06 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 05:02.10 | sivana[work] | what's the model of the linksys? |
| 05:02.41 | marty-ott-athome | wireless-g... hmm... |
| 05:02.45 | sivana[work] | dsl or cable? |
| 05:03.13 | marty-ott-athome | wrt54g - dsl and I'm the isp... :) |
| 05:03.22 | sivana[work] | using speedstream? |
| 05:03.34 | marty-ott-athome | yeah actually |
| 05:03.38 | sivana[work] | lol |
| 05:03.45 | sivana[work] | those have firewalls in them, no? |
| 05:03.56 | marty-ott-athome | don't think so in modem mode |
| 05:04.09 | asdx | can i use ekiga with asterisk? |
| 05:04.11 | asdx | for making calls |
| 05:04.12 | sivana[work] | Bell hands those out... we stopped supporting Bell... hehe |
| 05:04.20 | marty-ott-athome | I really think it's the linksys |
| 05:04.38 | xpot | me too! (damn linksys) |
| 05:05.05 | marty-ott-athome | I have some Genteks at the office ... but yeah, I mean, I even read that the DMZ option is there to enable all ports to ... ONE host. |
| 05:05.21 | sivana[work] | does it work when you do that? |
| 05:05.35 | marty-ott-athome | There are better routers or maybe even Linksys router today that recognize the voice issues and you can configure that specifically to be permitted through. |
| 05:05.56 | marty-ott-athome | Didnt' test - I can't test... who would I call? |
| 05:06.06 | sivana[work] | your home # |
| 05:06.10 | marty-ott-athome | I'd have ot do it with at least 2 phones. |
| 05:06.14 | sivana[work] | or cell |
| 05:06.23 | marty-ott-athome | oh! the server is not hooked up to a PRI or outside line |
| 05:06.42 | sivana[work] | ya, that sucks :) |
| 05:06.45 | xpot | voipjet has a free test server you can use... upto 25 cents |
| 05:06.52 | xpot | www.voipjet.com |
| 05:06.56 | sivana[work] | try with canreinvite=yes then |
| 05:07.06 | sivana[work] | on both peers |
| 05:07.12 | marty-ott-athome | :) - in the end, I'm testing the mediatrix unit.. works well so far - I'm impressed |
| 05:07.14 | sivana[work] | with nat=yes on both peers |
| 05:07.29 | marty-ott-athome | Ah, I closed my ssh windows.. I'll attack this tomorrow. |
| 05:07.40 | marty-ott-athome | I |
| 05:08.02 | marty-ott-athome | ll probably simply bring a phone over to a co-worker's house and tell him to test with me |
| 05:08.32 | marty-ott-athome | or.. shit.. I have a cisco 2601 here.. maybe I'll drop a dslconfig on it tomorrow |
| 05:08.56 | marty-ott-athome | anyways, thanks for the help guys.. I'm going to bed.. |
| 05:09.03 | xpot | night marty |
| 05:09.08 | marty-ott-athome | night |
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| 05:46.01 | *** join/#asterisk _brc (n=brc_@smtp.mxsecure.com) |
| 05:46.45 | _brc | having a problem compiling zaptel and my brain is fried....anybody got any ideas? http://pastebin.ca/322247 |
| 05:47.19 | _brc | zaptel.c:426: error: syntax error before "zone_lock" |
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| 06:17.09 | hardwire | anybody want to buy an AudioCodes MP-114 4FXO for a delightfull price? |
| 06:17.37 | Qwell | $2.50 |
| 06:18.10 | hardwire | add two zeros |
| 06:18.14 | hardwire | to the end |
| 06:18.15 | Qwell | $2.5000 |
| 06:18.22 | hardwire | move the dot a bit |
| 06:18.28 | Qwell | $.25000 |
| 06:18.28 | hardwire | shit.. not a literal bit |
| 06:18.42 | Qwell | sold |
| 06:18.47 | hardwire | $250.00 Qwell! |
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| 07:08.49 | _brc | can anybody give me a hint on how to add an app to compile into the menuselect system? |
| 07:21.17 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 07:21.48 | mosty | i have a problem with an IVR that doesn't respond to my keypresses, what could be wrong? |
| 07:25.17 | mosty | also, when i hangup this particular sip phone after dialling (snom 320) the call continues through the dialplan until somebody picks up the other end, any ideas about that? |
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| 07:44.33 | mosty | it works fine when i dial via another PBX, but not when I dial direct to the PBX in question |
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| 08:33.19 | j0 | does anyone here have success using a pocketpc phone with a sip client over wifi as their main phone? |
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| 08:35.49 | piper69 | j0 what are you trying to do |
| 08:36.08 | j0 | i need some sort of cordless headset for my desk phone, but i'm wondering if i should just get a pocket pc that does wifi and bluetooth with my current headsets |
| 08:36.33 | j0 | piper69: that and i like the idea of all-in-one, however dumb that may be |
| 08:37.12 | j0 | i'm just wondering if anyone has real world experience with ppc sip clients |
| 08:37.39 | piper69 | j0 i have did that long time ago, i had a ppc and i installed a software in it and then i had to sign up "free" somewhere and it worked |
| 08:39.27 | j0 | hmm |
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| 09:15.51 | chat_jokey | hi steve r u there |
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| 10:01.20 | zoa | yo yo oej |
| 10:01.53 | oej | Yo |
| 10:02.11 | zoa | its alive! |
| 10:02.21 | zoa | olle did you try zoiper yet ? :) |
| 10:05.28 | oej | Wasn't able, did not have windows. Installed a virtual machine for it last week, so I will do it soon. |
| 10:05.59 | zoa | cool |
| 10:06.08 | zoa | the mac version is getting there |
| 10:07.19 | oej | That's even more cool |
| 10:07.23 | oej | See you later... |
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| 12:45.00 | bradoaks | zoa: is the mac version of zoiper available for download yet? I wouldn't mind trying out a beta or alpha for that platform. |
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| 12:46.29 | zoa | not yet, we are struggling with a nasty deadlock in the audio driver on mac |
| 12:46.44 | zoa | but it will soon be there |
| 12:47.11 | zoa | i also want to work more on the mac touch 'n feel |
| 12:49.15 | bradoaks | good luck with the deadlock. looking forward to it. |
| 12:50.25 | bradoaks | i'll fire up the laptop and try out the windows version soon. |
| 12:53.49 | zoa | super |
| 12:53.56 | zoa | it will go out of beta next week ik think |
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| 13:04.12 | cygar | Hello |
| 13:04.25 | poller | Hi |
| 13:04.54 | cygar | One question, I am having a trouble confiuring an E1 with ISDN... I can receive calls but i can not send outgoing calls. The hangupcause i get is cause 31 of Q.931 |
| 13:04.57 | cygar | Can anyone give me a hint ? |
| 13:16.31 | xheliox | look at pridialplan |
| 13:16.33 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 13:16.47 | xheliox | Just a hunch. I really don't know what the fuck I'm talking about, fair warning. |
| 13:17.06 | xheliox | (in your zapata.conf) |
| 13:18.34 | chat_jokey | can any one tell me if libss7 is out to use in production ? |
| 13:21.31 | chat_jokey | hello anyone here to tell me bout libss7 ? |
| 13:21.40 | chat_jokey | ~libss7 |
| 13:21.48 | chat_jokey | pb~ |
| 13:21.49 | chat_jokey | ~pb |
| 13:21.51 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 13:22.01 | chat_jokey | ~ss7 |
| 13:22.03 | jbot | i heard ss7 is can be used in conjunction with ss7box.com - see the website. possible, perhaps, to use with chan_ss7 |
| 13:22.03 | cygar | xheliox: xheliox: in my zapata i just has the "context" that i am using to get inbound calls, not outbound calls, Is it possible to be that ? |
| 13:22.20 | chat_jokey | ~chan_ss7 |
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| 13:22.57 | zoa | chat_jokey: yes it is |
| 13:23.29 | xheliox | cygar: No. The context is for incoming calls. Your outgoing calls can be in any context, as long as it's available to the device(s) you're trying to dial out over. |
| 13:23.54 | xheliox | cygar: If you're getting a cause code, I suspect it's not a context issue -- it's going out over the E1, presumably. |
| 13:24.01 | chat_jokey | zoa chan_ss7 aint working for me .. i get MTP2 CRC errors on my cli |
| 13:24.18 | chat_jokey | and there is no doc for libss7 |
| 13:24.20 | chat_jokey | :( |
| 13:24.43 | chat_jokey | i believe libss7 is digium's ss7 stack ? |
| 13:25.12 | cygar | xheliox: mh.. in the context for dialout i have the Zap with the Dial like this : DIAL(Zap/g1/${EXTEN:0},90,T) |
| 13:25.31 | cygar | xheliox: if i do a noop to read the HANGUPCAUSE i just see cause 31 [ normal ]... |
| 13:25.32 | chat_jokey | if anyone can help me, I am willing to help .. i want to try libss7 on my box .. |
| 13:25.48 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 13:25.56 | chat_jokey | oops i meant if anyone willing to help .. i want to try out libss7 .. |
| 13:26.07 | xheliox | the pridialplan=national is what's default... but often wrong. |
| 13:26.51 | cygar | xheliox: let me check... is in brazil |
| 13:26.54 | xheliox | or you could always debug it by sending it out zap/1 instead of g1 -- make sure you don't have your groups mucked up. |
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| 13:29.06 | cygar | xheliox: tried zap/1 and the same error... what do you mean by groups mucked up? |
| 13:30.10 | xheliox | I mean.. you could have had groups misconfigured. |
| 13:30.15 | xheliox | It doesn't matter. |
| 13:33.04 | zoa | hello olle |
| 13:33.07 | zoa | damn too late |
| 13:36.29 | cygar | mh... if i am receiving calls the signalling is ok. What else could it be? |
| 13:36.47 | cygar | xheliox: i dont have a pridialplan=national in my zapata |
| 13:37.27 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 13:37.35 | xheliox | cygar: it's the default option, whether it's there or not. |
| 13:55.45 | cygar | xheliox: I think there could be a problem with the group... |
| 14:00.17 | *** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 14:00.52 | ZX81 | hi all, anyone know how you're supposed to use the console command "dialplan add extension"? |
| 14:01.04 | ZX81 | copy the example from the help text doesn't work |
| 14:02.11 | ZX81 | http://pastebin.ca/322529 |
| 14:02.12 | ZX81 | :) |
| 14:02.26 | ManxPower | cygar: most of the time pridialplan=unknown is what you want |
| 14:08.57 | ManxPower | pridialplan=national means that Asterisk will tell the telco that all calls are national calls (not local, not international) |
| 14:09.26 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 14:09.52 | zoa | Helloooooooooooooo there |
| 14:15.47 | ManxPower | Hiya, zpa |
| 14:15.50 | ManxPower | and zoa too |
| 14:17.30 | zoa | hey ho |
| 14:17.37 | zoa | where do you live now ? |
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| 14:27.19 | ManxPower | Here and there. My cabin arrived at the mountian last week, so things are slloooowwwlllyyyy moving forward. |
| 14:28.38 | zoa | is it easy to work there ? or you do everything remotely ? |
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| 14:29.00 | cygar | ManxPower: I have already tried it.... |
| 14:29.32 | cygar | ManxPower: i tested unknown... but it doesnt even get's the channel in the asterisk... i just get the chan unavailable at the moment |
| 14:29.47 | *** join/#asterisk RamsesII (n=rha@core1-gw.net.cubemedia.it) |
| 14:29.51 | RamsesII | hi to all |
| 14:30.01 | RamsesII | I've a little question, can i Ask? |
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| 14:33.59 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:34.34 | puzzled | hi |
| 14:35.50 | RamsesII | hi |
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| 14:43.12 | ManxPower | RamsesII: ask your question |
| 14:43.38 | RamsesII | tnx |
| 14:44.30 | RamsesII | when i call from external phone to my asterisk box i don't listen the "tuuuuuuuu tuuuuuuu " ring, but only the ivr can respond |
| 14:44.34 | *** join/#asterisk chat_jokey (n=chat_jok@59.181.111.61) |
| 14:45.17 | leejohn | RamsesII: you mean you heard only the IVR thing but not the ring?? |
| 14:45.31 | RamsesII | ex |
| 14:45.35 | RamsesII | exactly |
| 14:45.40 | RamsesII | don't ring but respond |
| 14:46.06 | leejohn | RamsesII: exten => s,1,Answer |
| 14:46.14 | RamsesII | exist |
| 14:46.15 | leejohn | RamsesII: exten => s,n,Ringing |
| 14:46.16 | RamsesII | :) |
| 14:46.18 | RamsesII | ahhhhh |
| 14:46.39 | RamsesII | trere are in my config |
| 14:47.14 | leejohn | RamsesII: what channel involve in the incoming call ? Sip/Zap/IAX? |
| 14:47.18 | RamsesII | sip |
| 14:47.26 | RamsesII | excusemy :) i'm a newbie :P |
| 14:47.55 | leejohn | RamsesII: can you pastebin your config? |
| 14:48.08 | RamsesII | I see in extension --> from-sip-external |
| 14:48.23 | RamsesII | of course said me the part can u like to see ^_^ |
| 14:48.43 | leejohn | RamsesII: just plain dialplan on incoming call from sip |
| 14:48.52 | RamsesII | from-sip-external? |
| 14:49.13 | leejohn | RamsesII: if that's the context for incoming sip call then that's it :p |
| 14:49.14 | RamsesII | there are more plain |
| 14:49.19 | RamsesII | ah ok |
| 14:49.42 | RamsesII | [from-sip-external] |
| 14:49.42 | RamsesII | ;give external sip users congestion and hangup |
| 14:49.42 | RamsesII | ; Yes. This is _really_ meant to be _. - I know asterisk whinges about it, but |
| 14:49.42 | RamsesII | ; I do know what I'm doing. This is correct. |
| 14:49.42 | RamsesII | exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN}) |
| 14:49.43 | RamsesII | exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})}) |
| 14:49.45 | RamsesII | exten => _.,n,Goto(s,1) |
| 14:49.47 | RamsesII | exten => s,1,Ringing |
| 14:49.49 | RamsesII | exten => s,n,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1) |
| 14:49.51 | RamsesII | exten => s,n,Set(TIMEOUT(absolute)=15) |
| 14:49.53 | RamsesII | exten => s,n,Answer |
| 14:49.57 | RamsesII | exten => s,n,Wait(2) |
| 14:49.58 | leejohn | RamsesII: ouch don't paste it here you might get kick :p |
| 14:49.59 | RamsesII | exten => s,n,Playback(ss-noservice) |
| 14:50.01 | RamsesII | exten => s,n,Playtones(congestion) |
| 14:50.03 | RamsesII | exten => s,n,Congestion(5) |
| 14:50.05 | RamsesII | exten => h,1,NoOp(Hangup) |
| 14:50.07 | RamsesII | exten => i,1,NoOp(Invalid) |
| 14:50.09 | RamsesII | exten => t,1,NoOp(Timeout) |
| 14:50.20 | leejohn | RamsesII: please goto http://www.pastebin.ca then post it there |
| 14:50.21 | RamsesII | ops |
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| 14:50.27 | RamsesII | ahh ok excusemy! |
| 14:50.30 | Gido-E | RamsesII newbee? |
| 14:51.07 | RamsesII | http://www.pastebin.ca/322582 |
| 14:51.14 | RamsesII | Gido-E of course :D |
| 14:51.55 | RamsesII | but i've masquerade :P |
| 14:54.51 | leejohn | RamsesII: base on the config snippet that you gave me if you don't know the incoming peer then Playback(ss-noservice) right? but i don't see the IVR stuff on it |
| 14:55.10 | RamsesII | ah wait |
| 14:55.12 | RamsesII | http://www.pastebin.ca/322588 |
| 14:55.23 | RamsesII | complete extension.conf |
| 14:56.49 | RamsesII | but my problem it's the same withivr or announcement |
| 15:02.21 | leejohn | RamsesII: please give me another minute your extensions.conf was huge to parse with my eye :p wait i'll get my reading glass :) |
| 15:03.21 | RamsesII | hahaha ok |
| 15:03.26 | yassine | any idea what i doo need to avoid this : Jan 21 00:29:45 WARNING[11502]: res_musiconhold.c:493 monmp3thread: Unable to spawn mp3player |
| 15:03.53 | cygar | I have pasted the ISDN problem I have [ I can get incoming calls but i can NOT make outgoing calls ]. I have changed from R2 signalling to ISDN and i am having that problem. if anyone can give me a hint the complete post is at http://www.pastebin.ca/322595 |
| 15:03.54 | leejohn | yassine: do you have mp3player installed ? |
| 15:04.12 | RamsesII | ùif u need i have extension-additional where is [app-announcement-2] can I use now |
| 15:04.14 | RamsesII | :) |
| 15:04.28 | yassine | leejohn, which one exactly i could not find any package on my distro that match mp3player im running on debian |
| 15:05.25 | leejohn | RamsesII: yes please because i don't see the context where if the DID match from yours then call routed to IVR |
| 15:05.40 | RamsesII | mmm i have solveeeee |
| 15:05.54 | leejohn | RamsesII: ? what do you do ? |
| 15:06.16 | leejohn | RamsesII: just add ringing on the IVR part then it should work |
| 15:06.29 | leejohn | yassine: could you pastebin your musiconhold.conf ? |
| 15:06.51 | leejohn | yassine: i don't know what external program you are using to play your mp3 stream |
| 15:06.57 | RamsesII | i've resolv the question |
| 15:07.15 | RamsesII | <PROTECTED> |
| 15:07.40 | nicklinn | Anyone know of a function where I can pipe in the sip peer name and it return the mailbox= field? |
| 15:07.44 | yassine | leejohn, i did not touch that file yet so i assume its set to default |
| 15:08.10 | leejohn | yassine: if your config is default that should be mpg123 |
| 15:08.38 | cygar | Can anyone take a look at my paste bin since I am having that trouble to make an outbound call with ISDN meanwhile that i can get inbound calls easily. Getting hangupcause number 31. www.pastebin.ca/322595 |
| 15:10.13 | yassine | leejohn, okay so i may need to install it |
| 15:11.12 | leejohn | yassine: if you are using a .deb package then you should fine mpg123 player compatible with asterisk if you are installing from source then make mpg123 will do the job |
| 15:11.20 | leejohn | find* |
| 15:11.53 | yassine | leejohn, im using a debi package and i juts installed the mpeg123 too |
| 15:12.10 | RamsesII | leejohn tnx tnx tnx tnx tnx tnx very much :D |
| 15:13.05 | leejohn | RamsesII: are you sure your problem has been solve?? |
| 15:13.30 | RamsesII | of course!!!! |
| 15:14.05 | RamsesII | :D in the conf frepbx don't add exten => s,n,Ringing, i don't know why... |
| 15:14.24 | RamsesII | but leejohn thankyou very much for ur help! |
| 15:14.24 | RamsesII | ù |
| 15:14.26 | RamsesII | :D |
| 15:15.32 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 15:15.32 | leejohn | RamsesII: ok your the boss :) |
| 15:15.37 | RamsesII | nono you the boss :D |
| 15:15.58 | leejohn | yassine: i'm not very familiar with debian but if mpeg123 = mpg123 then that's it |
| 15:16.24 | leejohn | yassine: last note mpg123 should match the path on your installation in order for this to work |
| 15:16.47 | *** join/#asterisk duckz (n=duckz@141.85.3.18) |
| 15:17.00 | leejohn | yassine: well i suggest just use rawplayer instead of mpg123 if you are using asterisk 1.2 branch |
| 15:17.32 | leejohn | yassine: there's a lot of issue on mpg123 like resource hogging |
| 15:17.48 | yassine | ahh okay thanks leejohn |
| 15:18.10 | yassine | should i add it in the config file you mentioned above? |
| 15:18.44 | *** join/#asterisk benno2 (n=benno2@host98-17-dynamic.4-87-r.retail.telecomitalia.it) |
| 15:18.44 | cygar | xheliox: Can you take a look at the pastebin www.pastebin.ca/322595 please ? |
| 15:18.54 | leejohn | yassine: the path should match on musichonhold.conf there are sample config there too |
| 15:19.47 | yassine | leejohn, this is what i have there: 7 mode=files 8 directory=/var/lib/asterisk/mohmp3 9 #include musiconhold_additional.conf |
| 15:20.08 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 15:20.35 | benno2 | hi, I have a problem when wanting asterisk registering with a SIP provider. it provides only an IP (not realm) and a alphanumeric user: for example: IP 1.2.3.4 user: benno pass: 1234 if I enter this data in X-Lite then I can get both outbond and inbound calls (the provider offers a real pstn number) but when configuring the same in asterisk, I can do outbound calls, but inbound calls does not work because |
| 15:20.42 | zapp-branigan | hi, the codec speex work fine in asterisk 1.4 ? |
| 15:20.56 | benno2 | the asterisk console gives me: Got SIP response 405 "Method not allowed" back from 1.2.3.4 |
| 15:21.39 | benno2 | I use something like this in sip.conf: register => benno:1234@1.2.3.4/12345678 |
| 15:21.44 | benno2 | what could be wrong ? |
| 15:21.57 | leejohn | yassine: you are using freepbx too? could you pastebin musiconhold_additional.conf? |
| 15:21.58 | benno2 | the strange thing is that this same register line works with other SIP providers |
| 15:22.26 | benno2 | do you think X-Lite registers differently ? |
| 15:23.33 | leejohn | benno2: can you pastebin your extensions.conf related to incoming DID ? |
| 15:24.05 | zapp-branigan | speex codec give me and error in load module of the asterisk : undefined symbol: speex_nb_mode plese someone know what is the problem ? |
| 15:24.07 | leejohn | benno2: i presume you are having problem from an inbound call to your SIP GW provider right?? |
| 15:24.43 | *** join/#asterisk mrempire (n=trefpunt@mrempire.demon.nl) |
| 15:25.09 | leejohn | cygar: I'm not very familiar with ISDN but you could try google your problem |
| 15:25.11 | yassine | leejohn, i have freepbx but there is no musiconhold_additional.conf in /etc/asterisk/* |
| 15:25.11 | benno2 | leejohn: thanks, yes I have this problem that I cannot get inbound calls working. notice that all other SIP providers I have provide a domainname, for example sipgate.de , while this only an IP |
| 15:27.29 | benno2 | leejohn: and if I call the real phonenumber of that sip provider while asterisk is active then it's like no one was registered to the SIP account. the SIP provider sends me to their voicemailbox |
| 15:27.47 | benno2 | leejohn: so I assume the registerin fails completely as the asterisk console tells me |
| 15:28.11 | leejohn | benno2: sip show registry ? what the output should look like? |
| 15:28.36 | leejohn | benno2: to be honest my sip provider is just like your's no realms or domain or anything :) |
| 15:29.00 | benno2 | 1.2.3.4:5060 benno 120 Request Sent |
| 15:29.14 | leejohn | benno2: it seems like you are not going anywhere :) |
| 15:29.20 | benno2 | I could try to ethereeal both x-lite and asterisk |
| 15:29.23 | benno2 | while registering |
| 15:29.26 | benno2 | and then see what happens |
| 15:29.37 | yassine | leejohn, http://rafb.net/p/5WjmsI44.html my musiconhold.conf |
| 15:29.38 | benno2 | leejohn: but the strange thing is that outbound calls work well |
| 15:29.52 | benno2 | are those 2 independent things ? |
| 15:30.49 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 15:30.49 | leejohn | benno2: some sip providers doesn't have authentication at all, they just controlled the customer by firewall |
| 15:31.12 | leejohn | benno2: and that's true just like with my case |
| 15:31.49 | benno2 | leejohn: should I try to pass some special params to the sip.conf register cmd ? |
| 15:31.51 | leejohn | benno2: but you are also correct if they provide authentication if you can't register at all then they should not route the call to pstn |
| 15:32.09 | leejohn | benno2: your parameters is correct |
| 15:32.31 | leejohn | benno2: how about if you are going to register your softphone directly to sip provider? |
| 15:32.50 | benno2 | leejohn: yes my softphone xten x-lite can register without problems and inbound calls work there |
| 15:32.53 | [TK]D-Fender | leejohn : No, registering has nothing to do that authing calls TO the PSTN. |
| 15:32.59 | benno2 | leejohn: this is why I am puzzled |
| 15:33.31 | blitzrage | registration only tells the provider where you are |
| 15:33.39 | [TK]D-Fender | leejohn : The ONLY purpose of registering is to inform the ITSP what IP address to send incoming calls TO. |
| 15:33.58 | benno2 | [TK]D-Fender: ok, but why does it then say method not allowed ? |
| 15:34.20 | benno2 | does asterisk use different sip commands than Xten X-lite ? |
| 15:34.25 | yassine | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) any idea what could be the issue here ?? |
| 15:34.37 | blitzrage | yassine: asterisk isn't running |
| 15:34.39 | leejohn | yassine: your asterisk doesn't start properly |
| 15:34.43 | yassine | blitzrage, its running |
| 15:34.48 | leejohn | ?? |
| 15:34.50 | [TK]D-Fender | yassine : Asterisk is not RUNNING. Or your don't have rights to it. |
| 15:34.59 | [TK]D-Fender | yassine : and what tells you its running? |
| 15:35.10 | blitzrage | ps ax | grep asterisk |
| 15:35.39 | [TK]D-Fender | blitzrage : Prepare for the worst :) |
| 15:35.48 | leejohn | [TK]D-Fender]: ok, i'm not pretending to be expert here, i just want to narrow the cause :) |
| 15:36.02 | leejohn | :) |
| 15:36.04 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 15:36.10 | yassine | blitzrage, you are right |
| 15:36.14 | yassine | its not running |
| 15:36.16 | [TK]D-Fender | benno2 : Time to apstebin your SIP debug info, and your configs... |
| 15:36.17 | [TK]D-Fender | ~pb |
| 15:36.27 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:36.38 | benno2 | [TK]D-Fender: does the successful registration of the register command in your opinion depend from the params you specify in sipprovider-out context ? |
| 15:36.46 | zoa | hey blitzy |
| 15:37.07 | benno2 | [TK]D-Fender: thanks, any idea how do I log SIP debug messages in asterisk ? |
| 15:37.11 | zoa | and mr fender |
| 15:37.18 | [TK]D-Fender | benno2 : No. everything in the register line itself is all that exists. |
| 15:37.25 | leejohn | benno2: no it has nothing to do with context |
| 15:37.33 | [TK]D-Fender | benno2 : "sip debug" |
| 15:37.35 | benno2 | [TK]D-Fender: thanks |
| 15:37.45 | [TK]D-Fender | zoa : y0 |
| 15:38.43 | *** part/#asterisk leejohn (n=jsharryp@58.69.18.20) |
| 15:39.09 | yassine | leejohn, now asterisk is no more running and when i run asterisk -vvvgc i gets this : http://rafb.net/p/wdnhaw54.html |
| 15:40.23 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 15:40.48 | [TK]D-Fender | yassine : we need more... we don't see you being crashed out back to a linux CLI or anything there. |
| 15:40.48 | [TK]D-Fender | yassine : Also... all that terminal emu junk... ick |
| 15:41.31 | yassine | [TK]D-Fender, can you point me to a place to get more debug data ? |
| 15:42.21 | danp | yassine, run 'asterisk -vvvvvvc' and paste the last 20 or so lines before it returns you to the command prompt |
| 15:42.41 | [TK]D-Fender | yassine : wht do you mean a place? you simply didn't paste up to the END where your problem could be detected |
| 15:42.42 | tzafrir | asterisk -rvvvvvvvv |
| 15:42.50 | [TK]D-Fender | yassine : use www.apstebin.ca please |
| 15:42.57 | [TK]D-Fender | errr .. www.pastebin.ca |
| 15:43.47 | yassine | [TK]D-Fender, i pasted everything i got displyed |
| 15:44.27 | danp | yassine: it shows something about ExecIfTime and just hangs there? |
| 15:44.35 | benno2 | [TK]D-Fender: leejohn: here is my sip debug out, it seems the SIP provider does not allow the REGISTER cmd ? http://pastebin.ca/322627 |
| 15:45.09 | yassine | asterisk -vvvvvvc gives me this back : http://rafb.net/p/Z7Q03983.html |
| 15:45.45 | danp | hmm |
| 15:46.08 | *** join/#asterisk |Vulture| (n=_Vulture@101.222.121.70.cfl.res.rr.com) |
| 15:46.09 | yassine | its suspecious since there are no error i could see :s |
| 15:46.22 | [TK]D-Fender | yassine : try like : asterisk -gvvvvvvvvvc |
| 15:46.41 | yassine | okay one sec |
| 15:46.42 | |Vulture| | anyone know if ${TIMESTAMP} is broken in 1.4? I cant seem to get it to function and there is no documentation about it changing on the wiki |
| 15:47.14 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 15:47.17 | danp | could also try adding '; echo' to the command line to make sure the last thing it shows isn't mangled by the prompt |
| 15:47.17 | tzafrir | more than 4 v-s don't help, I believe |
| 15:47.24 | tzafrir | try some d-s |
| 15:47.43 | [TK]D-Fender | tzafrir : Sure it does... andpressing the floor buttons on an elevator makes it go faster too! |
| 15:47.51 | zoa | yes i have the same impression |
| 15:48.00 | benno2 | [TK]D-Fender: any idea if asterisk can be persuaded to work like x-lite ? |
| 15:48.00 | zoa | you should walk around in circles around the server |
| 15:48.02 | zoa | counter clockwise |
| 15:48.09 | tzafrir | alternatively: strace -f asterisk |
| 15:48.09 | zoa | benno2 it could |
| 15:48.16 | zoa | but |
| 15:48.25 | tzafrir | (or gdb) |
| 15:48.28 | zoa | it will involve a lot of work to do it exactly the same |
| 15:48.54 | [TK]D-Fender | benno2 : it works with X-Lite jsut fine... |
| 15:48.54 | zoa | just changing the device name could work also |
| 15:49.18 | benno2 | zoa: ah you think this is the reason ? |
| 15:49.20 | [TK]D-Fender | benno2 : Did you pastebin up SIP debug and your config files for us to look at to help you? |
| 15:49.21 | yassine | [TK]D-Fender, does this seem to be the issue : http://rafb.net/p/uPI2bY23.html |
| 15:49.22 | zoa | no |
| 15:49.24 | zoa | it shouldnt |
| 15:49.26 | benno2 | [TK]D-Fender: ok |
| 15:49.31 | zoa | unless they block it on purpose |
| 15:49.45 | benno2 | zoa: can the device name be changed in a config file or is it a compile time option ? |
| 15:49.46 | zoa | do a sip debug |
| 15:49.50 | danp | yassine: i would say so |
| 15:49.52 | zoa | compile time i think |
| 15:49.56 | tzafrir | yassine, maybe. Please run: ldd /usr/lib/asterisk/modules/format_mp3.so |
| 15:49.56 | benno2 | :( |
| 15:50.01 | [TK]D-Fender | yassine : Recently upgrade your * box? I'm betting theres a version mismatch... |
| 15:50.13 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 15:50.17 | skirmisha | hello |
| 15:50.31 | yassine | [TK]D-Fender, no i did not i just installed freepbx on a debian package for asterisk |
| 15:50.35 | benno2 | zoa: I mean a SIP provider should be happy for any calls made through their networks independently if it's asterisk or a softphone |
| 15:50.41 | danp | dunh dunh dunhhhh |
| 15:50.45 | tzafrir | yassine, the freepbx deb? |
| 15:50.55 | zoa | benno2: depends |
| 15:50.58 | yassine | tzafrir, no |
| 15:51.04 | *** join/#asterisk saftsack (n=oliver@p54A7E224.dip.t-dialin.net) |
| 15:51.04 | yassine | asterisk deb |
| 15:51.10 | zoa | if they bundle it with something like vonage does they might not want it |
| 15:51.16 | zoa | did you try a different phone ? |
| 15:51.17 | |Vulture| | ah depricated for ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} |
| 15:51.20 | [TK]D-Fender | yassine : you need to recompile asterisk-addons and update your defective format_mp3.so |
| 15:51.30 | tzafrir | yassine, anyway, format_mp3 is from asterisk_addons, right? |
| 15:51.34 | zoa | if different phones work but asterisk does not, then its a different problem |
| 15:51.42 | zoa | try sjphone or zoiper |
| 15:51.54 | yassine | mhh [TK]D-Fender tzafrir i will see if i can find some on deb packages |
| 15:52.13 | tzafrir | yassine, what is your distro? what asterisk deb? |
| 15:52.20 | [TK]D-Fender | yassine : we advise strongly AGAINST packaging here. Use source tarballs if you know whats good for you.... |
| 15:52.43 | tzafrir | I, however, am very fond of debs |
| 15:52.57 | skirmisha | can someone tell me which file is concerned about sip registration |
| 15:53.01 | skirmisha | in asterisk? |
| 15:53.25 | tzafrir | sip.conf |
| 15:53.34 | skirmisha | no no |
| 15:53.40 | skirmisha | i am talking about C file |
| 15:53.50 | tzafrir | channels/chan_sip.conf |
| 15:53.57 | tzafrir | channels/chan_sip.c |
| 15:54.17 | tzafrir | Sadly, everything is in there |
| 15:54.21 | |Vulture| | hahaha thats no good... when rxfax executes * restarts |
| 15:55.07 | [TK]D-Fender | tzafrir : Fondness should be reserved for ancient memories :) |
| 15:55.30 | tzafrir | [TK]D-Fender, they're good. I should know |
| 15:55.36 | skirmisha | tzafrir is this the file which update asterisk db about user registrations |
| 15:55.46 | [TK]D-Fender | tzafrir : For common apps BESIDES *, sure... |
| 15:56.00 | [TK]D-Fender | skirmisha : /etc/asterisk/sip.conf |
| 15:56.39 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 15:56.40 | tzafrir | skirmisha, the SIP registrations are not maintained in *the* asterisk DB (/var/lib/asterisk/astdb) |
| 15:56.55 | yassine | after i installed mpeg123 this happens does it happen to have a problem with it ? |
| 15:56.57 | skirmisha | tzafrir they are in memory only? |
| 15:57.00 | yassine | because it was working before |
| 15:57.11 | [TK]D-Fender | yassine : Your probelm has nothing to do with mpg123. |
| 15:57.14 | tzafrir | skirmisha, yes, in memory data structures |
| 15:57.32 | [TK]D-Fender | yassine : format_mp3.so is a seperate module that has nothing to do with mpg123 |
| 15:57.36 | yassine | [TK]D-Fender, i only asked since that was the last thing i did before restarting asterisk |
| 15:57.59 | tzafrir | yassine, again, what is the output of that ldd command? |
| 15:58.01 | [TK]D-Fender | yassine : Yet we've just told you REPEATEDLY what you have to do. so go DO IT. |
| 15:58.08 | tzafrir | yassine, maybe. Please run: ldd /usr/lib/asterisk/modules/format_mp3.so |
| 15:58.10 | yassine | tzafrir, the file is not found |
| 15:58.30 | tzafrir | format_mp3.so is not found? |
| 15:58.57 | yassine | yes its not there |
| 15:59.10 | yassine | tzafrir, coeur2lion:/usr/lib/asterisk/modules# ls *mp3* |
| 15:59.10 | yassine | app_mp3.so |
| 15:59.15 | yassine | thats all |
| 15:59.22 | [TK]D-Fender | APP!? |
| 15:59.29 | [TK]D-Fender | wtf.... |
| 15:59.31 | tzafrir | so why does asterisk attempt to load it? any explicit load command in modules.conf? |
| 15:59.36 | yassine | [TK]D-Fender, i would tip its a debian thing |
| 15:59.42 | tzafrir | grep mp3 /etc/asterisk/modules.conf |
| 15:59.56 | yassine | mhh good point tzafrir let me see there |
| 15:59.56 | tzafrir | yassine, it's not |
| 16:00.21 | tzafrir | If Debnian has some extra apps, they are ones from bristuff. |
| 16:00.22 | yassine | <PROTECTED> |
| 16:00.33 | skirmisha | tzafrir is it possible at the time of registration to make separate copy of registration entry to external db by modifying cahn_sip.c |
| 16:00.45 | yassine | tzafrir, i mean the prefix "APP" |
| 16:00.45 | tzafrir | so you have an explicit request to load format_mp3.so . remove it |
| 16:00.55 | [TK]D-Fender | skirmisha : If you're modding the source you can do whatever the hell you want... |
| 16:00.58 | danp | skirmisha: that's basically what realtime does |
| 16:01.09 | benno2 | [TK]D-Fender: http://pastebin.ca/322639 |
| 16:01.20 | skirmisha | danp where is this patch? |
| 16:01.22 | tzafrir | skirmisha, hmmm... technically, yes. Not sure what is the overhead of this |
| 16:01.23 | yassine | ahh its up again |
| 16:01.26 | skirmisha | where can i download it |
| 16:01.29 | danp | skirmisha: it's built in |
| 16:01.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 16:01.42 | danp | http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime |
| 16:01.43 | yassine | tzafrir, i assume its being added via freepbx there |
| 16:01.45 | tzafrir | skirmisha, I recall someone talking about this in some developers talk. |
| 16:01.46 | skirmisha | no there was something about res_data |
| 16:01.50 | skirmisha | or something like that |
| 16:02.06 | [TK]D-Fender | benno2 : You're missing the other HALF of the settings you need for * to work behind NAT. |
| 16:02.08 | tzafrir | yassine, suit yourself. This is the immediate cause to the crash of your Asterisk system. |
| 16:02.41 | [TK]D-Fender | benno2 : add "canreinvite=no", "nat=yes", "externip=[yourWANiphere]" |
| 16:02.45 | yassine | thanks tzafrir [TK]D-Fender |
| 16:05.18 | skirmisha | where is res_data project |
| 16:05.36 | tzafrir | skirmisha, updating an external database is probably the simpler part. Consalting that external source is probably the more complicated part. |
| 16:05.48 | danp | skirmisha, i just did some quick searching. looks like it was renamed ast_data which is also MIA |
| 16:07.32 | [TK]D-Fender | danp : Better gt Bush looking for it... at the very least he'll trash every site in his wake :) |
| 16:07.51 | danp | heh |
| 16:07.55 | [TK]D-Fender | Apps of Mass Productivity! terrism! |
| 16:08.02 | skirmisha | tzafrir do u think so |
| 16:08.06 | [TK]D-Fender | MG.. AMP! |
| 16:08.43 | tzafrir | skirmisha, I haven't really given this any thiought. I only try to think of potential problems (my usual habbit) |
| 16:08.57 | yassine | [TK]D-Fender, are you aware of any interface i could use via tcp/ip to talk(configure/manage) to asterisk ? |
| 16:09.11 | tzafrir | yassine, AMI |
| 16:09.19 | tzafrir | ~ami |
| 16:09.31 | tzafrir | ~manager |
| 16:09.32 | jbot | i heard manager is a thing that should be killed |
| 16:09.45 | yassine | :) |
| 16:09.58 | yassine | tzafrir, i that a socket ? AMI ? |
| 16:10.10 | yassine | is it documented somewhere ? |
| 16:10.15 | benno2 | [TK]D-Fender: I am not behind nat, do you think canreinvite=no can help ? |
| 16:10.31 | [TK]D-Fender | yassine : Putty :) |
| 16:10.42 | tzafrir | http://voip-info.org/wiki/view/Asterisk+manager+API |
| 16:11.01 | benno2 | [TK]D-Fender: BTW I found there is an useragent=.... command in sip.conf so I set it to grandstream h286 but in the sip headers I still see from sip:asterisk@myip |
| 16:11.07 | [TK]D-Fender | tzafrir : c'mon... that question deserved n equally specific answer1 |
| 16:11.08 | [TK]D-Fender | ! |
| 16:11.12 | benno2 | can I suppress that asterisk@ string too ? |
| 16:11.32 | yassine | tzafrir, i would like to use a small java application to interact with it or at leat test that scenario |
| 16:11.34 | [TK]D-Fender | benno2 : YOU DIDN'T SET YOUR EXTERNIP. PAY ATTENTION. i ALREADY GAVE YOU THE ANSWER TO THIS... |
| 16:12.04 | skirmisha | tzafrir i just want to make a copy to sql db |
| 16:12.33 | tzafrir | jbot, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 16:12.35 | jbot | okay, tzafrir |
| 16:12.37 | skirmisha | so upon registration asterisk will fill that db and when user unregister it will delete that entry from db |
| 16:13.06 | benno2 | [TK]D-Fender: thanks, but the problem is I have a dynamic IP and other SIP providers already work. I try to set my current IP and see what happens |
| 16:13.32 | yassine | tzafrir, thanks thats look to be what i want, now i only need to make more reading about asterisk it self to know what i want to manage remotly |
| 16:13.52 | tzafrir | skirmisha, what's the point in having a copy of that data if you can't use it? |
| 16:14.12 | tzafrir | skirmisha, you can sample it periodically |
| 16:14.20 | *** join/#asterisk oQPa (n=roque@189.Red-81-39-148.dynamicIP.rima-tde.net) |
| 16:14.30 | [TK]D-Fender | benno2 : You need to specify your IP or * behind NAT will fail. |
| 16:14.33 | tzafrir | and then update the database |
| 16:14.57 | [TK]D-Fender | benno2 : Get a dynDNS type service and use "externhost" and "externrefresh" in that case |
| 16:15.54 | benno2 | [TK]D-Fender: I now specified externip too, canreinvite=no (btw asterisk is no a public,although dynamic IP) ... still no success, method not allowed |
| 16:19.33 | [TK]D-Fender | benno2 : Perhaps your provider does not upport registration. |
| 16:19.40 | benno2 | [TK]D-Fender: the question is: I see "Allow: INVITE,CANCEL,BYE,MESSAGE,ACK,OPTIONS" in the return SIP packets, does this mean that registration is not allowed ? |
| 16:19.51 | [TK]D-Fender | benno2 : and some SIP debug info from CLI would help... |
| 16:20.19 | benno2 | [TK]D-Fender: yes I had the same doubt, but the question is how can X-lite softphone get inbound calls ? Wait .. I pastebin the sip debug messages I get now |
| 16:21.10 | skirmisha | tzafrir i want to know each user on which asterisk is registered |
| 16:21.13 | skirmisha | that's all |
| 16:21.53 | tzafrir | skirmisha, you can also update a database from the dialplan or snapshot periodically from the manager interface |
| 16:22.07 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:22.51 | skirmisha | tzafrir give me an example of how to dump registrations |
| 16:22.51 | tzafrir | skirmisha, do you get any event to the manager when someone registers/unregisters? |
| 16:22.51 | benno2 | [TK]D-Fender: here is the SIP debug output (including CLI output) http://pastebin.ca/322650 |
| 16:22.57 | skirmisha | yes i do |
| 16:23.19 | tzafrir | ok, so your application should probably be a manager interface watcher |
| 16:24.07 | tzafrir | And no need to patch chan_sip.c |
| 16:24.20 | skirmisha | i just log as asterisk -r |
| 16:25.41 | benno2 | AFAIK SIP INVITE causes the sip client phone to ring |
| 16:25.56 | [TK]D-Fender | benno... |
| 16:25.57 | [TK]D-Fender | # |
| 16:25.57 | [TK]D-Fender | From: <sip:sbenno@159.148.8.105>;tag=as59d75a24 |
| 16:25.57 | [TK]D-Fender | # |
| 16:25.57 | [TK]D-Fender | To: <sip:sbenno@159.148.8.105> |
| 16:26.04 | [TK]D-Fender | you are registering to YOURSELF!? |
| 16:26.07 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 16:26.12 | skirmisha | tzafrir and then |
| 16:26.18 | skirmisha | how can i capture |
| 16:26.25 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 16:26.35 | benno2 | [TK]D-Fender: :( so perhaps the problem is in the register line ? |
| 16:26.40 | tzafrir | skirmisha, you have the manager action SIPpeers |
| 16:26.50 | tzafrir | show manager command SIPpeers |
| 16:26.57 | skirmisha | yes i have it |
| 16:27.07 | [TK]D-Fender | benno2 : REALLY!?! |
| 16:27.20 | tzafrir | or you can send an arbitrary CLI command (e.g; sip show regiatry) |
| 16:27.21 | [TK]D-Fender | benno2 : Go re-read it again, and check your providers settings |
| 16:27.30 | skirmisha | let me see |
| 16:27.36 | benno2 | [TK]D-Fender: ok let me check, thanks |
| 16:31.16 | skirmisha | anyone familar with ser/openser |
| 16:35.36 | [TK]D-Fender | skirmisha : How about asking some SPECIFIC questions.... |
| 16:35.51 | [TK]D-Fender | skirmisha : Yes, those are 2 apps for SIP routing proxying, etc! |
| 16:35.54 | skirmisha | [TK]D-Fender u mean about ser |
| 16:36.02 | [TK]D-Fender | skirmisha : About EVERYTHING. |
| 16:36.30 | sweeper | voice over ip is on its way out |
| 16:36.43 | sweeper | voice over longcat is the new black \o |
| 16:36.55 | skirmisha | i need to know if ser is capable to send one call to many destination and first picked up will take the call |
| 16:37.16 | [TK]D-Fender | skirmisha : * can already do that..... |
| 16:37.47 | skirmisha | yes i know but it is a bit slower |
| 16:37.55 | skirmisha | i just want pure proxy/routing |
| 16:38.18 | skirmisha | is dispatcher module created for that |
| 16:38.28 | yassine | does this warning means somthing i should take care of ? CallerID returned with error on channel 'Zap/1-1' |
| 16:38.30 | tzafrir | skirmisha, this is #asterisk . ask about ser/openser in #ser or #openser , Iguess |
| 16:38.43 | skirmisha | well this is what i thought |
| 16:39.36 | tzafrir | yassine, hmm... caller ID detection routine failed mid-way? Where are you? What telco? |
| 16:40.00 | yassine | tzafrir, netcologne (germany) |
| 16:41.07 | tzafrir | yassine, I'm not sure what they use. maybe dtmf? |
| 16:41.07 | yassine | but its using the infrstructure of the German Telekom |
| 16:41.07 | tzafrir | how do you have callerid detection set up? Is this an FXO ? |
| 16:41.53 | yassine | <PROTECTED> |
| 16:42.14 | yassine | callerid=asrecieved |
| 16:44.05 | tzafrir | maybe try: cidsignalling=v23 or: cidsignalling=dtmf |
| 16:44.49 | yassine | in zapata.conf ? |
| 16:45.46 | tzafrir | yes, right next to where you set the other callerid settings |
| 16:45.55 | yassine | okay |
| 16:46.44 | tzafrir | 'reload' should update those settings |
| 16:47.13 | yassine | ok |
| 16:47.23 | yassine | let me try |
| 16:50.48 | benno2 | normally I set fromdomain=name_of_sip_proxy |
| 16:51.22 | benno2 | for example with sipgate.de you set fromdomain=sipgate.de and register user:secret@sipgate.de/inboundextension |
| 16:51.32 | benno2 | what if the provides does not have a domainname ? |
| 16:52.44 | benno2 | provides=provider |
| 16:52.53 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:52.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:52.57 | [TK]D-Fender | benno2 : Don't set anything. just double check your IP for your register. |
| 16:54.32 | benno2 | [TK]D-Fender: and type=friend ? |
| 16:54.44 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 16:56.00 | *** join/#asterisk yatesy (i=yatesy@unaffiliated/yatesy) |
| 16:56.36 | benno2 | [TK]D-Fender: the problem is I still get From: <sip:sbenno@159.148.8.105>;tag=as5e332261 in my sip register message, which is the IP of my SIP provider |
| 16:57.27 | *** join/#asterisk rene- (n=rene-@200.34.66.137) |
| 16:57.45 | rene- | hey there dudes |
| 16:58.02 | rene- | wonder if anyone has some spare time to shoot me a g729 call over sip |
| 16:58.36 | rene- | 123456@200.34.66.132 |
| 17:01.05 | [TK]D-Fender | rene- : *ring* |
| 17:01.39 | [TK]D-Fender | rene- : Hrm... is Ulaw so far, but since its not answering, no cedec neg's I suspect |
| 17:01.54 | rene- | sorry |
| 17:01.56 | rene- | can u hear me |
| 17:02.03 | [TK]D-Fender | I can hear you. |
| 17:02.07 | rene- | cant hear you |
| 17:02.09 | [TK]D-Fender | maybe bad server settings |
| 17:02.10 | rene- | are u on ulaw? |
| 17:02.15 | [TK]D-Fender | I am, * is transcoding |
| 17:02.34 | [TK]D-Fender | <PROTECTED> |
| 17:02.39 | yatesy | if i've got a phone thats capable of g729 do i still need a license on my asterisk server? (assuming the endpoint is g729 capable so no transcoding is needed) |
| 17:02.40 | [TK]D-Fender | 1aebace-7ed 00101/00002 ulaw No Rx: ACK |
| 17:02.43 | [TK]D-Fender | thats my side |
| 17:02.49 | adorah | well transcoding to g729 is not always very good.. |
| 17:02.52 | rene- | Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message |
| 17:02.52 | rene- | 172.101.1.214 6403 28c4d69f6b9 00102/00000 g729 No Tx: ACK |
| 17:02.53 | rene- | 64.235.216.2 21 60fb5c306fc 00101/00102 g729 No Rx: ACK |
| 17:02.57 | [TK]D-Fender | rene- : So I can hear you just fine... (well actually.. garbled a bit... |
| 17:03.21 | [TK]D-Fender | rene- : dunno whats wrong |
| 17:03.32 | [TK]D-Fender | rene- : Not sure WHY it's not working.. |
| 17:03.36 | [TK]D-Fender | pastebin up your sip.conf |
| 17:03.48 | [TK]D-Fender | k |
| 17:05.01 | *** join/#asterisk evisu (i=hIRC@bzq-88-155-128-155.red.bezeqint.net) |
| 17:05.07 | rene- | http://www.pastebin.ca/322667 |
| 17:05.24 | *** join/#asterisk yassine (n=yassine@xdsl-87-78-100-2.netcologne.de) |
| 17:05.45 | yassine | how can i change the audio files from en to de ? |
| 17:05.54 | [TK]D-Fender | rene- : * has a public IP on a direct interface? |
| 17:06.09 | rene- | [TK]D-Fender: yes |
| 17:06.27 | [TK]D-Fender | yassine : http://www.voip-info.org/wiki/view/Asterisk+multi-language |
| 17:06.59 | rene- | yassine: changing your sip.conf language to de from en might do the trick for you if you have the 'de' sound files |
| 17:07.04 | [TK]D-Fender | rene- : 6403 is your phone? |
| 17:07.07 | rene- | yes |
| 17:07.11 | rene- | i am routing in the dialplan |
| 17:07.19 | rene- | 12345 dial sip 6403 |
| 17:07.23 | yassine | [TK]D-Fender, rene- thanks |
| 17:08.13 | [TK]D-Fender | rene- : Not sure if you can hear me... |
| 17:08.21 | rene- | i cant |
| 17:08.28 | [TK]D-Fender | rene- : ok, not sure why again, pastebin |
| 17:08.31 | [TK]D-Fender | rene- : Yeah I hear you |
| 17:08.50 | [TK]D-Fender | rene- : PB some more of the setup you're using to specifically route this call (dialplan and all) |
| 17:09.17 | [TK]D-Fender | rene- : adn while you're at it add "canreinvite=no" into [general] as well. |
| 17:10.06 | [TK]D-Fender | rene- : try it... |
| 17:10.36 | rene- | ok |
| 17:11.11 | [TK]D-Fender | rene- : you'll need to do a "sip reload" and I'll call back. |
| 17:11.16 | *** join/#asterisk BitBandit (n=polx@68-116-238-170.dhcp.stgr.ut.charter.com) |
| 17:11.30 | [TK]D-Fender | rene- : but do continue to PB the rest of the bits being processed by this call. |
| 17:11.41 | [TK]D-Fender | rene- : Will call again when you're ready |
| 17:11.46 | rene- | http://www.pastebin.ca/322675 |
| 17:12.25 | rene- | it is very odd, i am able to call the pots using g729 on the phone |
| 17:12.51 | [TK]D-Fender | rene- : That is the bare minimum you need for a working call... and it SHOULD work... :/ |
| 17:13.12 | [TK]D-Fender | rene- : Anything possibly being filtered on your internet connection? |
| 17:13.22 | *** join/#asterisk shinux__ (n=shinux@80.89.187.82) |
| 17:13.27 | *** join/#asterisk verylowsodium (n=verylows@adsl-074-244-143-225.sip.mco.bellsouth.net) |
| 17:14.11 | verylowsodium | hi, I have the following problem |
| 17:14.20 | verylowsodium | I have two * servers on different locations |
| 17:14.21 | Qwell | low sodium - yeah, we know |
| 17:14.32 | Qwell | take some vitamins |
| 17:14.32 | evisu | heh |
| 17:14.36 | [TK]D-Fender | NaCl FTW! |
| 17:14.37 | verylowsodium | server A takes an iax2 call and routes it to server B |
| 17:14.40 | verylowsodium | through iax2 |
| 17:15.32 | verylowsodium | server b then makes a pstn call using a tdm02b |
| 17:15.32 | verylowsodium | everything works except |
| 17:15.34 | [TK]D-Fender | NaI3.NaI = fun! |
| 17:15.35 | verylowsodium | that when server b tries to call the pstn |
| 17:15.47 | verylowsodium | server log says calling Zap/g1/extension |
| 17:16.04 | rene- | D-Fender: thats what i what thinking |
| 17:16.13 | verylowsodium | but the extension never gets called |
| 17:16.39 | rene- | D-Fender: can you shoot me another call? |
| 17:16.44 | rene- | i have disabled g729 |
| 17:16.51 | Qwell | [TK]D-Fender: where's my copy? |
| 17:17.07 | [TK]D-Fender | rene- : getting closer.... |
| 17:17.15 | rene- | Packet2Packet bridging SIP/64.235.216.2-08b0c7a0 and SIP/6403-08b28880 |
| 17:17.41 | [TK]D-Fender | rene did you add "canreinvite=no" to [general]? |
| 17:17.45 | rene- | i did |
| 17:17.49 | [TK]D-Fender | Qwell : Int he mail :) |
| 17:18.01 | [TK]D-Fender | Qwell : Take heed of the anthrax though ;) |
| 17:18.09 | benno2 | OPTIONS sip:159.148.8.105 SIP/2.0 .... From: "asterisk" <sip:asterisk@87.4.17.98>;tag=as38573b03 |
| 17:18.16 | verylowsodium | any idea what could cause the tdm to not actually make the call? |
| 17:18.21 | benno2 | any idea how I can change the "asterisk" string ? |
| 17:18.28 | [TK]D-Fender | rene- : and did a reload? |
| 17:18.32 | benno2 | I still have my doubts that the SIP provider blocks asterisk |
| 17:18.39 | rene- | [TK]D-Fender: sure |
| 17:18.41 | [TK]D-Fender | rene- : Try switching to ULAW, jsut as a sanity check. |
| 17:19.02 | *** join/#asterisk tRSS (n=tRSS@124.29.255.220) |
| 17:19.16 | rene- | i switched to ulaw in general but not in the phone, should i do it in both? |
| 17:19.30 | verylowsodium | benno2, tried changing useragent on sip.conf? |
| 17:19.42 | [TK]D-Fender | rene- : on * first, then another for the phone afterwards. |
| 17:19.49 | [TK]D-Fender | rene- : Lets break this down step by step |
| 17:19.59 | *** join/#asterisk |Vulture| (n=_Vulture@101.222.121.70.cfl.res.rr.com) |
| 17:19.59 | [TK]D-Fender | rene- : PM |
| 17:20.02 | rene- | ok |
| 17:20.05 | benno2 | verylowsodium: yes set it to "Grandstream H286", now trying "X-Lite" |
| 17:20.13 | |Vulture| | Anyone here get spandsp working on 1.4.0? |
| 17:20.27 | tRSS | quick question: my asterisk was registering with FWD just fine up until a few days back. now it is spitting this msg out, without registering: chan_iax2.c:7900 iax2_poke_noanswer: Peer 'fwd-gw' is now UNREACHABLE! Time: 0. Any help would be much appreciated. |
| 17:20.35 | |Vulture| | I got it to compile but everytime rxfax is executed it crashes * without an error |
| 17:20.38 | [TK]D-Fender | benno2 : You shouldn't have to be touching the UA at all... |
| 17:20.43 | [TK]D-Fender | benno2 : REMOVE that entirely |
| 17:21.24 | benno2 | [TK]D-Fender: my first tests were without useragent=... therefore it seems to make no difference |
| 17:21.25 | tRSS | FWD is reachable. I am getting a ping reply and my account settings are in place. |
| 17:25.22 | anthm | rm the qualify= line |
| 17:25.24 | wunderkin | because you can ping it means nothing |
| 17:26.52 | tRSS | wunderkin: but it was working just fine roughly 3 days ago. now it won't even register with FWD |
| 17:26.56 | tRSS | does that mean anything? |
| 17:27.19 | tRSS | no configuration changes at all. the machines runs by itself for weeks, until I log onto it myself. |
| 17:28.50 | wunderkin | well, it is a free service, it is probably flaky, i haven't used it for years |
| 17:29.29 | wunderkin | i'm guessing you are saying it never came back |
| 17:29.57 | *** join/#asterisk topping (n=topping@207.47.6.185.static.nextweb.net) |
| 17:32.41 | |Vulture| | Anyone here get spandsp working on 1.4.0? |
| 17:32.48 | *** join/#asterisk connecta (n=Administ@175.6.188.72.cfl.res.rr.com) |
| 17:33.00 | connecta | is anyone here in florida? |
| 17:33.09 | verylowsodium | orlando |
| 17:33.18 | connecta | wow really |
| 17:33.23 | connecta | http://www.google.com/calendar/events?q=central+florida+asterisk&ql=&qt=&qtd=&sa=N&page=vl&afp=4b562d9ae9813bbc |
| 17:33.31 | tRSS | yup... it is still not back.... well, FWD has never betrayed me. and 3 days is a long time. If there was a issue on their end, they would have posted about it. |
| 17:33.55 | connecta | if you care to join, we're having a little conference in about a half an hour |
| 17:35.26 | |Vulture| | orlando here too |
| 17:35.27 | verylowsodium | cant today, but will next saturday |
| 17:35.57 | connecta | well what do you do with asterisk? personal or professional? |
| 17:36.18 | |Vulture| | professional here |
| 17:37.40 | connecta | well theres a usergroup on meetup.com for orlando users, not very organized now, but im trying to work toward that |
| 17:37.46 | connecta | it would be great if you joined. |
| 17:38.00 | |Vulture| | what area of orlando are you guys located? |
| 17:38.11 | connecta | im in kissimmee |
| 17:38.16 | verylowsodium | kirkman rd |
| 17:38.50 | |Vulture| | I am over by Waterford area |
| 17:41.03 | *** join/#asterisk karmatronic (n=karmatro@84.77.137.35) |
| 17:59.49 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:59.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:02.18 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 18:06.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:06.23 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:06.35 | *** join/#asterisk Martin_Lundstrom (n=martin@ip-145.net-82-216-73.rev.numericable.fr) |
| 18:08.44 | Martin_Lundstrom | Hello |
| 18:08.52 | Martin_Lundstrom | anyone awake? |
| 18:09.05 | connecta | yes |
| 18:09.10 | Martin_Lundstrom | :) |
| 18:10.35 | sweeper | someone plz discuss the relative advantages of * on multi-core systems, vs multiple servers |
| 18:10.47 | Qwell | cost |
| 18:10.47 | Martin_Lundstrom | anyone have a good ip telephony provider in france that work with asterisk? |
| 18:11.14 | nick125_lappy | sweeper: redundancy |
| 18:11.33 | sweeper | Qwell: well, I keep reading that * is better off with more servers than with more procs |
| 18:11.47 | sweeper | is this an issue of total systems resources? |
| 18:12.02 | sweeper | or is it something about the architecture? |
| 18:12.13 | sweeper | I mean, I assume it's multi-threaded |
| 18:13.06 | sweeper | is that a mistaken assumption? |
| 18:13.19 | russellb | it's heavily threaded :) |
| 18:14.11 | Martin_Lundstrom | anyone know any good source of information on active load balancing? |
| 18:14.42 | Martin_Lundstrom | I guess im serching for a good firewall!? |
| 18:16.22 | connecta | actually thats what our conference call is partially about |
| 18:16.36 | tzafrir | Martin_Lundstrom, those are two separate things |
| 18:17.13 | *** join/#asterisk l2cache (n=Administ@102.133.202.68.cfl.res.rr.com) |
| 18:17.35 | connecta | vulture and verylowsodium |
| 18:17.40 | connecta | do you have xlite? |
| 18:17.50 | connecta | and martin, do you as well? |
| 18:18.08 | tzafrir | xlite? why? |
| 18:18.15 | connecta | im sorry, or idefisk |
| 18:18.23 | tzafrir | connecta, ask specific questions |
| 18:18.25 | connecta | because i'd like to send a link to join |
| 18:18.37 | connecta | a conference call |
| 18:19.58 | nick125_lappy | eek, xlite |
| 18:20.26 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:20.38 | verylowsodium | connecta, no, still using my cell |
| 18:20.59 | l2cache | whats wrong with xlite |
| 18:21.05 | evisu | has anyone been able to acheive numbers like 300 concurrent calls on a single server? |
| 18:21.13 | nick125_lappy | l2cache: all softphone sucks |
| 18:21.17 | Qwell | evisu: sure |
| 18:21.20 | nick125_lappy | *softphones suck |
| 18:21.24 | l2cache | wow, good generalization |
| 18:21.52 | l2cache | any facts? or just i think it sucks end of story? |
| 18:21.56 | evisu | Qwell, really.... what hardware would you recommend to reach the highest amount of concurrent calls ? |
| 18:22.00 | nick125_lappy | Though, the general unresponsiveness of xlite kind of pisses me off |
| 18:22.37 | connecta | yah really, it's important to speak in such generalizations to show your level of a lack of real knowledge about things |
| 18:22.52 | *** join/#asterisk gniretar (i=gniretar@wcc3-169.wccnet.org) |
| 18:22.56 | gniretar | hi all |
| 18:23.17 | gniretar | i ahve 2 * servers |
| 18:23.22 | gniretar | i want an IAX trunk between then |
| 18:23.31 | nick125_lappy | For example, I hung up a call, and xlite still kept the call connected for about 30 seconds after I hit the disconnect button |
| 18:23.44 | gniretar | so do i need one entry on each side? |
| 18:23.46 | gniretar | with type=friend? |
| 18:23.51 | connecta | affirmative |
| 18:24.12 | connecta | verylow, give a call if you can spare 10 minutes |
| 18:25.05 | gniretar | connecta: your saying yes to me? |
| 18:25.22 | connecta | yah sorry |
| 18:26.06 | connecta | each has to have an entry in the conf file so that when the neighbor tries to place a call, it will authenticate and have a context to route to |
| 18:27.58 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:28.36 | EmleyMoor | Anyone here using VoIPtalk iaxtalk? Can you get your VoIPtalk ID to work? Can you call other VoIP networks over it? |
| 18:29.37 | *** join/#asterisk oQPa (n=roque@189.Red-81-39-148.dynamicIP.rima-tde.net) |
| 18:31.22 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 18:36.13 | EmleyMoor | Is there a way of defining codes from rotary phones that replace * and #? |
| 18:44.31 | *** join/#asterisk write_erase (n=olivier@bon13-2-82-237-125-220.fbx.proxad.net) |
| 18:45.36 | write_erase | hi, How can I create en extend which play a mp3 ? |
| 18:46.14 | sweeper | write_erase: MusicOnHold |
| 18:46.33 | sweeper | google it, use it, rejoice \o |
| 18:46.54 | EmleyMoor | write_erase: Trying to create your own Dial-a-Disc service? <g> |
| 18:46.56 | write_erase | moh works well, but I need to map a phone number to a mp3. |
| 18:47.25 | write_erase | I'd like to listen to news stream when I compose 1234 for exampke |
| 18:47.38 | sweeper | write_erase: yea |
| 18:47.51 | sweeper | just make an extension that set moh as soon as it gets picked up |
| 18:47.59 | *** join/#asterisk switch (n=switch@saya.attrition.jp) |
| 18:48.18 | sweeper | you can set up different moh contexts, ya know |
| 18:48.28 | sweeper | so like MoH(news) |
| 18:48.29 | write_erase | oh, so it is possible to have multiple moh :-) that's it thx |
| 18:48.35 | write_erase | great |
| 18:49.10 | EmleyMoor | show application MusicOnHold from the CLI seems useful |
| 18:49.29 | write_erase | Great |
| 19:04.14 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-170-186.socal.res.rr.com) |
| 19:06.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:06.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:06.33 | *** join/#asterisk evisu (i=hIRC@bzq-88-155-128-155.red.bezeqint.net) |
| 19:08.40 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 19:10.56 | connecta | vis there anyone here who does asterisk development |
| 19:11.37 | sweeper | connecta: if you have to ask that, you probably don't need to be talking to them \o |
| 19:12.25 | connecta | nvm, i found em |
| 19:18.46 | *** join/#asterisk karmatronic (n=karmatro@84.77.137.35) |
| 19:19.43 | *** part/#asterisk oQPa (n=roque@189.Red-81-39-148.dynamicIP.rima-tde.net) |
| 19:20.21 | *** join/#asterisk Tebi (n=rantis@gw.aller.fi) |
| 19:28.48 | *** join/#asterisk {tasker} (n=dsgsdfga@cpe-24-90-149-2.nyc.res.rr.com) |
| 19:28.58 | furibondox | hi all, someone use skypho? |
| 19:29.00 | {tasker} | hi |
| 19:29.10 | {tasker} | how can I set HANGUPCAUSE when dropping a SIP call? |
| 19:29.25 | {tasker} | I need to drop calls and send back a release code 34 |
| 19:33.55 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 19:45.38 | *** join/#asterisk jm|work (n=jamie@zen.jamiem.com) |
| 19:51.53 | {tasker} | anyone? |
| 19:55.11 | tzanger | hmm |
| 19:55.20 | tzanger | how do I *disable* simple cdr logging? I want to use odbc only |
| 19:55.29 | tzanger | cdr.conf does not appear to have a way to turn it off |
| 19:57.32 | bkruse | tzanger: noload => cdr |
| 19:57.39 | bkruse | noload your cdr modules that you do not want |
| 20:01.02 | tzanger | bkruse: I've noloaded all cdr_* except for cdr_odbc |
| 20:01.09 | tzanger | noload => cdr? |
| 20:01.38 | bkruse | tzanger: you could do that |
| 20:01.43 | bkruse | do module unload cdr(tab)(tab) |
| 20:01.44 | tzanger | bkruse: still says "simple cdr logging enabled" |
| 20:01.47 | bkruse | and look at the cdr modules |
| 20:01.52 | bkruse | theres more than just 1 |
| 20:02.04 | tzanger | I see cdr, cdr_manager, cdr_odbc and cdr_pgsql |
| 20:02.54 | bkruse | yep |
| 20:03.13 | bkruse | cdr_custom cdr_sqlite cdr_tds cdr_csv |
| 20:03.16 | tzanger | hmm |
| 20:03.21 | tzanger | cdr_odbc isn't showing up |
| 20:04.22 | bkruse | i see it |
| 20:04.53 | tzanger | oh it's loaded |
| 20:04.59 | tzanger | module load cdr_odbc.so says it's already loaded |
| 20:05.07 | {tasker} | is there a way to set HANGUPCAUSE before dropping a call on SIP? |
| 20:05.08 | tzanger | but cdr status doesn't show it |
| 20:05.23 | tzanger | {tasker}: I *thought* heard soemthing about that ages ago |
| 20:05.40 | {tasker} | me too but I can't find anything |
| 20:06.11 | {tasker} | i've tried everything but nothing works |
| 20:06.20 | {tasker} | PRI_CAUSE=xxx works on ZAP channels |
| 20:06.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:06.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:06.34 | {tasker} | neither that nor HANGUPCAUSE=x works on SIP |
| 20:07.04 | {tasker} | if i drop a call without answering or passing it through, it always returns code 21 |
| 20:08.49 | *** join/#asterisk arkadi (n=arkadi@87.246.143.140) |
| 20:13.14 | {tasker} | hello |
| 20:13.17 | {tasker} | is anyone here? |
| 20:14.34 | [TK]D-Fender | {tasker} : there is no code to execute before terminating a call |
| 20:15.22 | {tasker} | so we're stuck with Asterisk sending back a 21 |
| 20:15.55 | {tasker} | or 603 Declined, as it may be |
| 20:16.34 | [TK]D-Fender | what are you actually trying to do? |
| 20:17.00 | {tasker} | very simple |
| 20:17.02 | {tasker} | just test this |
| 20:17.03 | {tasker} | exten => 9218,1,Set(HANGUPCAUSE=1) |
| 20:17.03 | {tasker} | exten => 9218,2,Hangup |
| 20:17.14 | {tasker} | that code will always send back 603 declined |
| 20:17.40 | {tasker} | unless i answer the call or switch it to the terminating end, where it passes through the cause code received on a rejected call |
| 20:18.00 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-170-186.socal.res.rr.com) |
| 20:18.10 | [TK]D-Fender | I don't see the point yet... all you are doing is making an exten that just hangs up... |
| 20:18.19 | {tasker} | for test purposes, yes |
| 20:18.30 | {tasker} | if i reject a call based on criteria, i need to send back a cause |
| 20:18.36 | [TK]D-Fender | please show what you're REALLY trying to do... |
| 20:18.59 | {tasker} | customer -----> asterisk ----> carrier |
| 20:19.18 | {tasker} | when i get the call from customer, i check certain criteria in a database |
| 20:19.31 | {tasker} | then i either switch the call to CARRIER |
| 20:19.34 | {tasker} | or i reject the call |
| 20:19.54 | {tasker} | the point is i need to set the release code |
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| 20:26.29 | {tasker} | anyone? |
| 20:26.33 | L|NUX | can some one tell me how can i setup Asterisk in this way so that if clients use rfc8223/info/auto/inband as dtmf then i do not need to modify * for clients |
| 20:27.48 | tzanger | {tasker}: just for shits and giggles, what happens if you say Hangup(1) |
| 20:28.08 | {tasker} | let me try |
| 20:28.17 | L|NUX | tzanger : hey |
| 20:28.28 | L|NUX | tzanger : if you got some time can you help me with my question bro |
| 20:29.18 | tzanger | L|NUX: don't think it can be done |
| 20:29.18 | gniretar | l2cache: hey, you still on? |
| 20:29.25 | tzanger | you're looking for autodetection of DTMF type |
| 20:29.33 | L|NUX | yupz |
| 20:30.23 | {tasker} | still get 21 |
| 20:30.31 | {tasker} | or 603 Declined |
| 20:30.40 | tzanger | {tasker}: what version of asterisk |
| 20:30.41 | {tasker} | asterisk 1.12.13 and 1.12.14 |
| 20:30.45 | tzanger | ah |
| 20:30.47 | tzanger | in trunk |
| 20:30.52 | tzanger | app_hangup takes a causecode |
| 20:30.57 | {tasker} | hmm |
| 20:31.00 | tzanger | [Description] |
| 20:31.00 | tzanger | <PROTECTED> |
| 20:31.00 | tzanger | If a causecode is given the channel's hangup cause will be set to the given |
| 20:31.01 | tzanger | value. |
| 20:31.08 | {tasker} | but is 1.14 stable? |
| 20:31.13 | tzanger | 1.4 you mean? |
| 20:31.16 | {tasker} | sorry |
| 20:31.20 | {tasker} | yeah |
| 20:31.28 | tzanger | it's been released as such, but I don't know |
| 20:31.30 | {tasker} | and mine was 1.2.13 and 1.2.14 :( |
| 20:31.32 | tzanger | I have always used svn trunk |
| 20:31.43 | {tasker} | so did I until 1.4 started crapping out on SIP calls |
| 20:33.53 | {tasker} | 1.4 beta, anyway |
| 20:34.04 | *** join/#asterisk arkadi (n=arkadi@87.246.143.140) |
| 20:35.56 | danp | {tasker}: by crapping out do you mean using 100% CPU? |
| 20:37.27 | {tasker} | asterisk exits |
| 20:37.36 | {tasker} | segfault |
| 20:37.42 | danp | you might try a recent svn checkout |
| 20:37.52 | {tasker} | more recent than 1.4.0 release? |
| 20:37.56 | danp | yeah |
| 20:38.06 | {tasker} | ok, i'll try that |
| 20:38.13 | {tasker} | but can you use svn in production? |
| 20:38.15 | danp | check out a copy from http://svn.digium.com/svn/asterisk/branches/1.4/ and give it a go |
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| 20:43.09 | L|NUX | tzanger : what about dtmfmode=auto ? |
| 20:43.27 | tzanger | L|NUX: don't know about any such mode |
| 20:43.46 | L|NUX | http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20dtmfmode |
| 20:43.49 | L|NUX | check this |
| 20:43.51 | joe | in http://www.freedomphones.net/polycom/files/ what are the SounPointIP_SIP* and the spip_ssip_sip* used for the first is for upgrading the phones I understand but what is the second for? |
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| 20:46.43 | [TK]D-Fender | joe : jsut different filenameing conventions. pay attention to the full name. You'll need a compatable set of SIP & BootROM. |
| 20:46.52 | [TK]D-Fender | joe : its in 2 pieces |
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| 20:51.20 | joe | [TK]D-Fender: Thanks. My issue is that I have a bunch of phone running bootroom 2.6.2 sip 1.5.2 but we got a bunch of new phone running 3.1 and sip 1.6 iirc. so they have issues w/ the version in my ftp server and keep rebooting. Due to the warning about the booroom upgrade I'm hesitant to upgrade them all and now sure what to do now |
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| 20:51.57 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 20:56.14 | *** join/#asterisk connecta (n=Administ@175.6.188.72.cfl.res.rr.com) |
| 20:57.30 | joe | well now it's not rebooting but just doesn't take the sip setting w/ the server and all that, is my only option to upgrade them all so they'll work together basically? |
| 20:59.46 | [TK]D-Fender | joe : You can't mix the configs between the two. You should be provisioning them from seperate folders |
| 21:00.02 | [TK]D-Fender | SIP 1.6 configs will screw up 1.5 firmwares as will the reverse |
| 21:03.22 | joe | [TK]D-Fender: that's what I figured, which is why I setup a test setup but how does one split them up or is it worth it just to upgrade them all? |
| 21:03.38 | joe | can't seem to find docs on this |
| 21:04.13 | joe | just printed the sip docs from polycom and about to rtfm but is there a better source to learn about how to do this? |
| 21:04.39 | connecta | welll, i can tell you how to mass upgrade them |
| 21:04.46 | connecta | i did and had 0 problems |
| 21:05.01 | connecta | but to redo all your configs for a newer version |
| 21:05.09 | *** part/#asterisk l2cache (n=Administ@102.133.202.68.cfl.res.rr.com) |
| 21:05.10 | connecta | thats a little harder |
| 21:05.15 | connecta | that would require a script |
| 21:05.45 | joe | connecta: I have the tftpboot/ftp setup all ready to go, I'm just hesitant to do it w/o more docs. I have very limited configs in the sip.cfg and phone1.cfg |
| 21:06.07 | *** part/#asterisk {tasker} (n=dsgsdfga@cpe-24-90-149-2.nyc.res.rr.com) |
| 21:06.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:06.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:06.47 | joe | about to do a diff and see how different they would really be.. |
| 21:07.07 | connecta | gimme a few |
| 21:07.17 | joe | connecta: thanks |
| 21:08.58 | joe | oh and one last thing how does one tell what version the bootroom.ld is via the cli? the person who set this up has a bunch of bootroom versions in the ftp dir. I know which is getting loaded on the phones via the status of the phone but would like to know how to check that manually.. |
| 21:09.11 | [TK]D-Fender | joe : Make 2 different provisioning accounts. thats all. |
| 21:09.42 | [TK]D-Fender | joe : And a "better source"? That what, the official documents? No.... |
| 21:09.54 | joe | [TK]D-Fender: where do I go read about how to do that? |
| 21:10.08 | joe | [TK]D-Fender: so that is the best source, perfect, thanks |
| 21:10.25 | [TK]D-Fender | joe : You shouldn't technically be suing only a single file like phone1.cfg. its a sample name and shuold be personal to each phone. |
| 21:10.41 | [TK]D-Fender | joe : What models do you have? |
| 21:10.56 | joe | [TK]D-Fender: 301, 501 and 4000 |
| 21:11.40 | [TK]D-Fender | joe : Ok, well I might suggest you get the latest firmwares from your vendor. Freedomphones is out of date as is normal. |
| 21:11.45 | joe | [TK]D-Fender: basically I want to reconfigure them all cleanly and standardize them all w/o breaking shit... |
| 21:13.19 | joe | k, they got them from voipsupply.com and cdw from what I can tell. but no one seems to want to get back to me about this. So I was going to try the latest from freedomphones |
| 21:13.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 21:14.18 | [TK]D-Fender | How many? |
| 21:14.21 | joe | ie I asked how I could get the latest version... |
| 21:16.27 | [TK]D-Fender | Better way to ask "Hi I'm a VoIP Supply customer and I've got a numbr of Polycom phones for which I need the latest SIP & BootROM firwmwares. What link or FTP server do I have to follow to get them?". |
| 21:16.50 | [TK]D-Fender | be direct and say it like they owe you this service. |
| 21:17.37 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 21:19.11 | connecta | first of all, what method do your phones currently use to get their configs when they boot |
| 21:20.09 | joe | connecta: |
| 21:20.10 | *** join/#asterisk nicklinn (n=maximag@mrtc-580035.mis.net) |
| 21:20.29 | joe | [TK]D-Fender: yeah, I'll be calling them shortly |
| 21:20.32 | joe | or on monday |
| 21:23.12 | connecta | well i think yourp problem can be solved now if you like |
| 21:23.41 | connecta | no nvm |
| 21:23.46 | connecta | im having a shitty day |
| 21:24.09 | evisu | can anyone recommend some hardware to acheive a maximum amount of concurrent calls? |
| 21:24.53 | nick125_lappy | evisu: 4xQuad Core with 100TB of RAM ;) |
| 21:25.19 | evisu | 100TB of ram ey .. :P |
| 21:25.28 | nick125_lappy | What codecs are you planning on using? Are you planning on doing transcoding? |
| 21:25.35 | evisu | and how many concurrent calls would you say i can do on that? |
| 21:25.44 | evisu | no zaptel, g.711 |
| 21:25.54 | evisu | iax trunks |
| 21:26.00 | evisu | no sip clients registered |
| 21:26.42 | joe | connecta: nvm? sorry you are having a shitty day :/ |
| 21:27.35 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 21:30.01 | [TK]D-Fender | evisu : "maximum" is a dangerous term, and can mean almost anything we want it to mean. What do you NEED to support? |
| 21:30.05 | *** join/#asterisk voipme (n=voipme@86.41.169.223) |
| 21:30.20 | evisu | i need to calculate my costs based on an extra 1000 calls |
| 21:30.32 | [TK]D-Fender | evisu : Otherwise I'll start suggesting gear that will destroy your unstated budged and give you convulsions :) |
| 21:30.36 | voipme | evening all.. |
| 21:31.13 | voipme | can someone point me in teh right direction of the right branch to checkout in the svn that will support ss7 |
| 21:31.14 | [TK]D-Fender | evisu : "extra 1000 calls" can you try to be a little clearer on what functions * will be performing, as well as confirming concurrency. |
| 21:31.55 | voipme | sorry libss7 that should say |
| 21:32.45 | evisu | fender: i'm preforming callbacks ie each leg is a call via an iax trunk. billing & database is handled on seperate servers. using g.711 |
| 21:33.11 | evisu | no ivr's, no qeues, no voicemails, no zaptel :) |
| 21:33.48 | [TK]D-Fender | evisu : just playback over G.711? |
| 21:34.41 | evisu | pretty much, just bridging two calls over g.711 |
| 21:34.53 | evisu | two legs to form a call actually |
| 21:35.12 | evisu | so none of the sides are a sip/iax extension |
| 21:35.39 | [TK]D-Fender | evisu : I'm not totally clear on whats on the other side. Can you describe a call (beginning to end) |
| 21:37.31 | evisu | i set up a call using the manager interface, ie, channel would look something like iax2/provider/1646xxxxx and exten is set up as the number to call, and so a call is formed once the channel leg answers |
| 21:37.54 | evisu | so there's noone 'dialing' from an extension, its all calls formed by manager |
| 21:38.57 | evisu | hmm perhaps i just made this more confusing :p |
| 21:39.14 | [TK]D-Fender | evisu : Ok, so who (as a person) is triggering this call? leg 1 of the call, leg 2, or some 3rd party? |
| 21:39.22 | evisu | leg 1 |
| 21:39.37 | [TK]D-Fender | evisu : how many concurrent calls? |
| 21:39.47 | evisu | its all triggered via http which is also on a sperate server |
| 21:40.17 | evisu | is what i'm trying to figure out :) |
| 21:41.31 | [TK]D-Fender | You don't know what kind of lod you'll have? thats sortof key to this. Because the calls themselves are largely irrelevent.... |
| 21:42.35 | evisu | its really quite difficult to predict, and i suppose 'lots of load' isnt really... helpful |
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| 21:43.14 | [TK]D-Fender | evisu : You are catching on... |
| 21:44.05 | evisu | but it is possible to come to some sort of estimate in order to calculate costs based on an extra 1000 calls |
| 21:44.14 | evisu | concurrent |
| 21:44.54 | bkruse | any php semi-guru's in here? |
| 21:46.25 | evisu | so, any ballpark figures fender? |
| 21:47.04 | evisu | thought you were going to recommend something that would destroy the budget.... :P |
| 21:47.07 | [TK]D-Fender | evisu : aahh.. 100 more CONCURRENT. OUCH. |
| 21:47.12 | [TK]D-Fender | 1000* |
| 21:47.36 | [TK]D-Fender | evisu : well...... How many are you puching off a single box successfully? |
| 21:48.16 | evisu | i havent tested the server i have now as I know for a fact its not what i'm going to use in production.. |
| 21:49.04 | bkruse | [TK]D-Fender: ive successfully done 600 sim calls on a 2950 |
| 21:49.13 | bkruse | with lots of room to move |
| 21:49.26 | evisu | whats the highest amount of concurrent calls you've come across so far? and on what hardware? :) |
| 21:50.20 | [TK]D-Fender | evisu : Not that much experience... I'm using you as a guidepost for your own need :) |
| 21:50.40 | evisu | hehe, true that does work sometimes :) |
| 21:51.23 | [TK]D-Fender | evisu : What do you have now, and how many calls are you successfully pushing through at a time on it? How is the load? |
| 21:51.59 | evisu | its all in a test enviornment, havnt done stress tests on this box |
| 21:52.05 | bkruse | evisu: ive done over 800 just doing playbacks on a 2950 4 gigs of ram with avg 30% load average |
| 21:52.19 | *** join/#asterisk essaredee (i=srd@24-182-113-208.dhcp.sprn.tx.charter.com) |
| 21:52.51 | essaredee | what variable would you use to figure out what the extension of the person calling is? |
| 21:53.03 | evisu | many thanks bkruse... that does sound like a lot for one box, i really didnt imagine being able to get that high... |
| 21:53.08 | evisu | but thats good news obviously |
| 21:54.06 | bkruse | evisu: just playback, keep in mind :] |
| 21:54.24 | evisu | hmm |
| 21:54.25 | bkruse | and you have to watch your network constraints.......800 ulaw' calls :X |
| 21:54.37 | essaredee | anyone? |
| 21:54.40 | Qwell | bkruse: ! |
| 21:54.46 | bkruse | essaredee: ${CALLERID(num)} |
| 21:54.49 | bkruse | Qwell: :X |
| 21:54.52 | essaredee | thanks |
| 21:54.53 | evisu | it is going to be quite a large production system... |
| 21:55.11 | bkruse | evisu: awesome, i would suggest maybe some load balancing if you really wana get into it |
| 21:55.11 | bkruse | Qwell wuts up, you over here at the atrium?? |
| 21:55.14 | Qwell | nah |
| 21:55.26 | Qwell | I stay as far away from the Atrium as possible on the weekend :D |
| 21:55.27 | evisu | oh definitly |
| 21:58.40 | bkruse | Qwell: good idea |
| 21:58.54 | bkruse | evisu: ya, its fun stuff, and you can build very nice load balance/failover |
| 22:00.33 | evisu | by the way, what cpu do you have on your 2950? |
| 22:00.44 | Qwell | bkruse: Those are the quad core xeons, right? |
| 22:01.31 | Qwell | I should try to requisition one for my new desktop |
| 22:01.32 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 22:05.43 | bkruse | Qwell: nah, dual core, dual xeon |
| 22:05.50 | Qwell | oh |
| 22:05.55 | bkruse | still very fast :D |
| 22:06.06 | Qwell | I still want one for my desktop :P |
| 22:06.40 | bkruse | me too :D |
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| 22:38.36 | bkw__ | You wanna hear something really funny... I tried to make an IAX call from here... behind a free-hotspot.com access point.. it wouldn't do it.. SIP worked fine. |
| 22:39.12 | Tebi | :D |
| 22:39.27 | Qwell | user error :P |
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| 22:49.37 | bkw__ | no it wasn't |
| 22:49.43 | bkw__ | I suspect a firewall rule |
| 22:49.45 | bkw__ | :P |
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| 22:54.46 | lirakis | im trying to figure out a simple simple dialplan for my extensions.conf .. i have ll sip phones.. all the same context. I just want to be able to dial any extension .. then have it connect me to the extension i dailed... i think there must be an easy way to do this with out hardcoding all my extensions into extensions.conf... but im having trouble figureing it out |
| 22:56.34 | *** join/#asterisk obnauticus (i=admin@c-24-21-116-29.hsd1.mn.comcast.net) |
| 22:56.49 | lirakis | aha! i think i got it... exten=> _XXXX,1,Dial(SIP/${EXTEN},,r) |
| 22:57.25 | fetcher | lirakis: you might want to look into the 'regexten=' parameter in sip.conf also |
| 22:57.47 | fetcher | I've never tried it, though... always either hardcode, or generate parts of extensions.conf from a script |
| 22:57.57 | lirakis | fetcher: well that line i just posted does work |
| 22:58.09 | lirakis | for any 4 digit extension |
| 22:58.56 | karmatronic | i cant hear any sound from chan_bluetooth with sip clients |
| 22:59.08 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 22:59.23 | *** join/#asterisk Axelatino (n=Axel@200.12.40.89) |
| 22:59.42 | *** join/#asterisk battini (n=inittab@cpe-24-209-36-174.neo.res.rr.com) |
| 22:59.44 | fetcher | lirakis: and any exceptions (special 4-digit extensions if needed) should still override it |
| 23:00.06 | lirakis | fetcher: ?? |
| 23:01.09 | bkruse | battini: hey :] |
| 23:01.57 | fetcher | lirakis: say you wanted 9999 to go to VoicemailMain instead of matching the SIP pattern. exten => 9999,1,... should still take precedence |
| 23:02.09 | fetcher | which is helpful :) |
| 23:02.12 | lirakis | fetcher: ah .. i got you now |
| 23:02.23 | Axelatino | hi guys, i have problem with a ooh323 setup anyone knows a good tutorial? |
| 23:04.34 | Qwell | ~wikis |
| 23:04.44 | jbot | methinks wikis is http://www.voip-info.org |
| 23:04.44 | Qwell | Axelatino: have a look there |
| 23:05.22 | [TK]D-Fender | fetcher : regextenis of no help in reducing the size of your dialplan. |
| 23:05.23 | Axelatino | i already did that but it restart when i send a h323 to the asterisk |
| 23:05.30 | Axelatino | i don;t know why |
| 23:05.42 | [TK]D-Fender | and for 11 extensions, who cares about hard-coding. you're better off doing it that way |
| 23:06.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:06.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:06.39 | obnauticus | erm |
| 23:06.44 | obnauticus | I installed zaptel and apperentally |
| 23:06.46 | obnauticus | oh nevermind |
| 23:06.49 | obnauticus | root@aaopwner# /usr/local/etc/rc.d/zaptel.sh start |
| 23:06.49 | obnauticus | kldload: can't load /usr/local/lib/zaptel/zaptel.ko: File exists |
| 23:06.50 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 23:06.54 | obnauticus | that means it's loaded doesn't it.. |
| 23:07.15 | Qwell | obnauticus: yeah.. |
| 23:07.15 | fetcher | [TK]D-Fender: so, what's regexten useful for? |
| 23:07.15 | obnauticus | .lol. |
| 23:07.15 | Qwell | try a kldunload, then kldload |
| 23:07.16 | obnauticus | ya |
| 23:07.17 | Qwell | You should get output from both commands |
| 23:07.23 | obnauticus | not start.. |
| 23:07.32 | obnauticus | but it's irrelevant it shows up in ps aux |
| 23:07.42 | Qwell | kernel modules show up in ps? |
| 23:07.44 | obnauticus | wait.. no it doesn't.. |
| 23:07.45 | obnauticus | lol. |
| 23:08.07 | Qwell | freebsd seriously annoys me :p |
| 23:08.17 | obnauticus | sometimes to me too |
| 23:08.21 | obnauticus | only because of the lack of support |
| 23:08.34 | obnauticus | http://www.voip-info.org/wiki-Asterisk+FreeBSD <-- I'm following that guide and.. |
| 23:08.37 | obnauticus | # pkg_add -rv asterisk |
| 23:08.37 | obnauticus | # pkg_add -rv zaptel |
| 23:08.43 | obnauticus | those packages don't even exist on the server |
| 23:08.57 | obnauticus | so im installing asterisk and zaptel from the ports, which i think i already did |
| 23:09.22 | [TK]D-Fender | fetcher : not friggen much |
| 23:10.39 | fetcher | at least wcfxo was, when I last tried it a year or so back |
| 23:11.08 | obnauticus | ok umm |
| 23:11.10 | obnauticus | this is a wtf thing |
| 23:11.11 | obnauticus | ===> openldap-client-2.3.33 conflicts with installed package(s): |
| 23:11.11 | obnauticus | openldap-client-2.2.30 |
| 23:11.13 | obnauticus | asterisk installed |
| 23:11.14 | obnauticus | rofl |
| 23:11.15 | obnauticus | install*( |
| 23:11.15 | fetcher | various caller-ID problems, plus unplugging the POTS circuit would sometimes reboot the whole box |
| 23:11.18 | *** join/#asterisk sasch (n=sasch@82.51.56.246) |
| 23:11.21 | obnauticus | wait.. |
| 23:11.22 | obnauticus | nevermind |
| 23:11.24 | sasch | hi all |
| 23:11.35 | qdk | fetcher: the drivers are good enough with OpenPBX. |
| 23:11.41 | [TK]D-Fender | fetcher : in otherwords... jsut as stable a card as with Linux :) |
| 23:12.28 | tzafrir | anybody here well familiar with zaptel/freebsd? |
| 23:12.36 | Qwell | tzafrir: somewhat |
| 23:13.09 | Qwell | well, probably more than most people, actually :p |
| 23:13.25 | fetcher | [TK]D-Fender: could have been motherboard issues, perhaps, but wcfxo has never given be trouble under Linux, at a handful of sites using them |
| 23:13.35 | sasch | i have one question with asterisk and VoiceMail |
| 23:13.38 | sasch | can help me |
| 23:14.22 | fetcher | audio's a bit poorer than with better cards, of course |
| 23:16.32 | obnauticus | tzafrir well.. |
| 23:16.40 | obnauticus | im installing right now |
| 23:16.40 | obnauticus | lol. |
| 23:16.45 | [TK]D-Fender | sasch : just ask. Don't ask to ask. |
| 23:16.47 | sasch | i have a dial plan that have 3 istruction ... first answer, after dial(SIP/12) and after VoiceMail |
| 23:17.02 | sasch | why when a person call me and hangup in dial |
| 23:17.14 | sasch | after voicemail register the call every time ?? |
| 23:17.25 | sasch | excusme for my english ... i'm italian :-P |
| 23:17.44 | [TK]D-Fender | sasch : who hun up? |
| 23:18.52 | sasch | a person that call me |
| 23:18.53 | [TK]D-Fender | sasch :and what is the call from the outside coming in on? |
| 23:19.12 | sasch | one moment i post my dial plan in pastebin |
| 23:22.03 | EmleyMoor | ... though how that would cope with message notification, I don't know <g> |
| 23:22.14 | tzafrir | EmleyMoor, basically edit zonedata.c in the zaptel source |
| 23:22.29 | sasch | this is my dial plan |
| 23:22.30 | sasch | http://pastebin.ca/323019 |
| 23:22.52 | tzafrir | I believe it is from there and not from asterisk. But it may actually be indications.conf |
| 23:23.27 | [TK]D-Fender | sasch : your problem is you aren't getting a clear call disconnect supervision trigger from your telco so * doesn't know they've hungupfor several seconds. |
| 23:24.22 | sasch | <[TK]D-Fender> in whic mode i can resolv my problem ?? |
| 23:25.00 | [TK]D-Fender | sasch : you might not be able to. check with your telco. |
| 23:25.15 | [TK]D-Fender | they need to enable "call disconnect supervision". |
| 23:25.35 | sevard | http://www.tigerdirect.com/applications/searchtools/item-Details.asp?EdpNo=2573758&sku=GEN-52048&CMP=EMC-TIGEREMAIL&SRCCODE=WEM1282AF |
| 23:25.44 | sevard | http://tinyurl.com/22ha4d |
| 23:27.06 | Qwell | rebate? screw that |
| 23:27.32 | Qwell | TWO rebates even |
| 23:27.37 | sevard | i wish it wasn't with a rebate |
| 23:27.47 | sasch | http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision |
| 23:30.27 | [TK]D-Fender | sevard : Nice idea, but be weary... MIR's tend to be scamed a lot. I bought a projector based on the MIR added-value, and got the box intentionally missing the UPC (cut out mechanically) |
| 23:30.29 | *** join/#asterisk Gankhuu (n=gankhuu@ns2.digis.net) |
| 23:30.42 | sevard | ouch |
| 23:30.49 | sevard | usually tiger direct tends to be pretty good about those |
| 23:31.37 | [TK]D-Fender | they do bank on your not getting it filled in right. uniden scammed me once claiming I didn't provide the UPC. total BS... |
| 23:31.59 | Qwell | and the government loves them, because they get the extra sales tax |
| 23:32.09 | Qwell | it's win-win-win |
| 23:32.14 | Qwell | ...oh, except you, you're screwed |
| 23:33.20 | [TK]D-Fender | sevard : Oh, and if you didn't know, TigerDirect ding you hard on shipping, even the small stuff. I'm relatively certain they split shipment everythingon purpose, not because of true B/O, but rather because they charge your full PPD&CHG shipping each time. they aren't a computer parts reseller... they are a freight re-charging company... |
| 23:33.35 | Qwell | http://www.newegg.com/Product/Product.asp?Item=N82E16820227145 |
| 23:34.42 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 23:34.50 | sevard | yes they do. i read somewhere that it helps if you order multiple things, like go together to buy something with a group of people on td, apparently the shipping price drops dramatically |
| 23:35.03 | Qwell | newegg > tigerdirect |
| 23:35.08 | Qwell | except for you silly canadians |
| 23:35.20 | EmleyMoor | Any of you lot using voiptalk |
| 23:35.21 | EmleyMoor | ? |
| 23:35.57 | yatesy | i am, well i've got a few accounts :P don't use it that much |
| 23:36.20 | EmleyMoor | yatesy: Have you got your voiptalk ID working for incoming calls using iax? |
| 23:36.22 | sasch | i go to sleep |
| 23:36.25 | sasch | bye bye |
| 23:36.25 | yatesy | yup |
| 23:36.31 | EmleyMoor | How do you do it? |
| 23:36.40 | yatesy | followed the guide on their website pretty much |
| 23:37.10 | EmleyMoor | I didn't spot anything in their notes on iax setup to do with that |
| 23:37.11 | yassine | asterisk is playing goodbye sound after a call is being sent to a voicmail and the hungup any idea whats wrong please ? |
| 23:37.25 | EmleyMoor | Can get my 0871 and 020 numbers in but not the voiptalk ID |
| 23:37.38 | yatesy | oh right i see what you mean |
| 23:37.44 | EmleyMoor | Also, should I be able to use sipbroker or direct peers through iax with it? |
| 23:37.58 | EmleyMoor | (have asked them but they are taking their time responding) |
| 23:38.14 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 23:38.22 | *** join/#asterisk gankhuu (n=IceChat7@ns2.digis.net) |
| 23:38.43 | yatesy | i just added another entry in iax.conf which has my ID in it rather than the phone number |
| 23:38.55 | yatesy | haven't tested it tho so i've no idea if it works |
| 23:38.56 | kgx | does anyone if i can specify a different interval for cdr_mysql to update the db? |
| 23:39.48 | yatesy | nah its ok, thanks |
| 23:40.31 | yatesy | one thing i have noticed tho is that i can't call a friend who uses SIP version of voiptalk by his ID |
| 23:40.34 | yatesy | it just fails |
| 23:40.47 | yatesy | i mean i can call his assigned number, but then that charges me so thats no good |
| 23:40.57 | EmleyMoor | Should I hear from them with a workable solution, I will make it public :-) |
| 23:41.05 | yatesy | cool |
| 23:42.08 | *** part/#asterisk gankhuu (n=IceChat7@ns2.digis.net) |
| 23:42.11 | EmleyMoor | I got a fax call to one of my "assigned but not yet done anything with" numbers the other day (!) |
| 23:42.33 | yatesy | heh |
| 23:42.40 | *** join/#asterisk gankhuu (n=IceChat7@ns2.digis.net) |
| 23:42.56 | *** part/#asterisk gankhuu (n=IceChat7@ns2.digis.net) |
| 23:43.46 | EmleyMoor | At the outside, I could even relinquish the current function of one of them for a further use |
| 23:44.30 | yatesy | kinda sucks you have to pay for an incoming geo number, some other providers give those numbers away for free |
| 23:44.47 | yatesy | like sipgate, except thats routed through .de so not as good as .uk based voiptalk! |
| 23:46.34 | EmleyMoor | It took a while for them to sort it out though |
| 23:46.47 | yatesy | how much you paying for all of those then? |
| 23:47.23 | EmleyMoor | About 15 a month - paid 20 for 2 months but they gave me 10 call credit when they got it sorted |
| 23:47.51 | yatesy | pretty good |
| 23:49.01 | yatesy | my internet connection's latency is all over the place so i've never really had the chance to use VoIP on a regular basis :/ |
| 23:49.30 | EmleyMoor | I only use it when it's cheaper, or for incoming calls, or when the BT line is either busy or dead |
| 23:49.56 | EmleyMoor | (though if it's dead, it would have to be dead in a way that didn't affect ADSL) |
| 23:50.07 | yatesy | yea in the future when i've got a decent net connection i'll probably use it all the time for incoming calls and possibly outgoing |
| 23:50.53 | EmleyMoor | At the moment, it takes priority for international calls, and for mobile calls during peak hours. BT takes priority for most other calls |
| 23:51.05 | EmleyMoor | (FWD for US, DE, NO, NL toll-free of course) |
| 23:51.22 | yatesy | have you got your BT line hooked up to your asterisk system then? |
| 23:51.26 | EmleyMoor | Yes |
| 23:51.36 | yatesy | using one of the cards or something else? |
| 23:51.49 | EmleyMoor | One of the cards (got a TDM31B) |
| 23:52.25 | yatesy | cool, i wanna be able to do that, trouble is i use OpenBSD on my server which doesn't support those cards as the drivers are linux only |
| 23:53.22 | EmleyMoor | I used to run an OpenBSD box - my need for it was less than for a Linux box to run asterisk, though |
| 23:53.42 | *** join/#asterisk JamesDotCom (i=jamesdot@creep.bur.st) |
| 23:53.44 | yatesy | fair enough |
| 23:53.56 | EmleyMoor | I used Linux before I used any of the BSDs - but I tried most of them eventually |
| 23:54.03 | yatesy | same |
| 23:54.38 | EmleyMoor | I prefer Debian GNU/Linux |
| 23:54.44 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 23:55.14 | yatesy | yea debian still is my distribution of choice when using linux |