| 00:07.32 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 00:07.42 | rene- | hello |
| 00:09.56 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 00:10.04 | rene- | my calls are ZAP-SIP |
| 00:10.14 | rene- | actually SIP->ZAP |
| 00:10.26 | rene- | but then theya re transferred to a queue |
| 00:10.40 | rene- | via an attended transfer using the hardphone functions |
| 00:11.18 | rene- | not asterisk's, but after the transfer, the queue member cant hear the customer |
| 00:11.32 | rene- | LEG1 SIP-ZAP LEG2 SIP-SIP |
| 00:11.46 | rene- | no nat involved, (LAN environment) |
| 00:11.48 | rene- | why? |
| 00:19.39 | trogs | baliktad: thanks for that. looking at voicepulse. |
| 00:19.47 | rene- | how can i have one way audio if i am doing outbound to ZAP, and then intra-lan transfer to another SIP phone |
| 00:20.26 | *** part/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 00:23.16 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 00:23.33 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 00:24.48 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:25.01 | *** join/#asterisk SiberAIR (n=SibRphre@160.79.176.178) |
| 00:26.25 | SiberAIR | hey - got a ? - setup asterisk in my home, my softphone from another computer in my home can call the IVR and audio is fine. Outside my home, i can connect to the asterisk server, but when i call anything no audio is transmitted |
| 00:26.28 | SiberAIR | ideas? |
| 00:27.01 | joat | you'll need a sip proxy or port forwarding on your router |
| 00:27.13 | SiberAIR | i'm already port forwarding port 5060 |
| 00:27.29 | joat | what about ports 10000-whatever? |
| 00:27.35 | SiberAIR | hrm |
| 00:27.41 | SiberAIR | what ports 10000-whatever? |
| 00:28.00 | joat | the audio is carried over udp ports starting at 10000 |
| 00:28.05 | SiberAIR | oh |
| 00:28.08 | SiberAIR | did not know that |
| 00:28.09 | SiberAIR | ok |
| 00:28.12 | SiberAIR | will google that and find out |
| 00:28.14 | SiberAIR | thanks joat |
| 00:28.38 | joat | the upper end can be whatever you say it is (but you have to configure it that way in rtp.conf) |
| 00:28.48 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 00:28.57 | joat | you'll need at least 4 ports for each call (if I remember correctly) |
| 00:29.14 | SiberAIR | well the default is 10000-20000, so i can just open those yes? |
| 00:29.43 | joat | yeah, but if you have any other computers in the house that dynamically use those ports, they'll have "issues" |
| 00:29.58 | SiberAIR | i don't have anything else |
| 00:30.03 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 00:30.09 | joat | then it shouldn't matter.... |
| 00:30.09 | SiberAIR | hrm |
| 00:30.18 | joat | forward all 10000 of them |
| 00:30.20 | SiberAIR | need to figure out how to open a range of ports on an airport base station |
| 00:30.33 | joat | that's your firewall? |
| 00:30.48 | SiberAIR | it is for my home |
| 00:31.11 | joat | didn't know that they filtered traffic |
| 00:31.45 | joat | thought it was just an access point that you stuck _in_ your network rather than in front of |
| 00:31.52 | joat | my mistake then |
| 00:31.59 | SiberAIR | no it's a router |
| 00:32.01 | SiberAIR | does DHCP and nat |
| 00:32.04 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 00:32.08 | SiberAIR | i have a sonicwall i just haven't put it in yet |
| 00:32.16 | SiberAIR | ah looks like it restarted as my other SN just logged in |
| 00:32.31 | SiberAIR | w00hoo |
| 00:32.58 | joat | in any case, NAT is what usually causes the "no audio" issue |
| 00:33.10 | joat | SIP isn't tolerant of NAT |
| 00:33.17 | SiberAIR | yeah i know |
| 00:35.11 | SiberAIR | and i have in my sip.conf that it's behind a nat |
| 00:43.53 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:49.38 | *** join/#asterisk metfan2007 (n=jc@201.103.43.23) |
| 00:50.47 | metfan2007 | hi all! a funny question :) Is it possible to extract the sip,conf from Asterisk? Accidentaly I deleted the file, but Asterisk is still working with the sip.conf in the memory... |
| 00:51.54 | drmessano | ...... |
| 00:53.30 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0f074b73d7c171cb) |
| 00:53.58 | joat | you can get some of it from the sip show .... command |
| 00:54.10 | joat | however, passwords are probably unrecoverable |
| 00:54.38 | metfan2007 | mmmmm, really? |
| 00:54.46 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 00:57.02 | seanbright | unless you can read the memory from a remote process |
| 00:57.07 | pcrane | might be a save options |
| 00:57.12 | seanbright | there isn't |
| 00:57.14 | pcrane | I remember reading about it somewhere |
| 00:57.21 | pcrane | might be thinking of something else then |
| 00:57.26 | seanbright | there is a manager command |
| 00:57.36 | seanbright | UpdateConfig or something like that |
| 00:57.52 | seanbright | but the *Config commands just read and write the files, not the in memory structures. |
| 00:58.10 | pcrane | I've got an interesting problem... |
| 00:58.25 | pcrane | I've got a dialstring that's 193 characters long |
| 00:58.32 | pcrane | and can contain @ characters |
| 00:58.44 | seanbright | put it in a variable |
| 00:58.47 | pcrane | any (easy) way to get it to behave in asterisk? |
| 00:58.49 | seanbright | err |
| 00:58.51 | seanbright | ignore me |
| 00:59.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:59.34 | pcrane | cause it ends up like: ~7770100~Local/7770100@internal@inter-asterisk |
| 00:59.45 | pcrane | anything before the @inter-asterisk is part of the dial string |
| 01:00.05 | pcrane | so, I'm thinking that asterisk gets confused with the first @ and tries to send it to internal |
| 01:00.12 | pcrane | and not to the inter-asterisk server |
| 01:00.33 | joat | pcrane... dereference the "@"? |
| 01:00.44 | pcrane | like \@? |
| 01:00.51 | joat | yeah |
| 01:01.01 | pcrane | when it's part of another variable? |
| 01:01.07 | *** join/#asterisk n3hxs (n=IceChat7@pool-70-110-19-76.washdc.fios.verizon.net) |
| 01:01.18 | pcrane | first=CUT(var,@,1) |
| 01:01.26 | joat | oh... |
| 01:01.26 | pcrane | second=CUT(var,@,2) |
| 01:01.28 | *** join/#asterisk moeSizlak (n=kthxbai@static-69-95-250-34.buf.choiceone.net) |
| 01:01.37 | pcrane | var=first\@second |
| 01:01.43 | pcrane | something like that? |
| 01:02.16 | *** part/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 01:03.06 | moeSizlak | ok, when I call home from work, my caller ID shows up as unavailable. But when i call a pizza shop from work, they have my callID. hows this possible |
| 01:03.11 | joat | i was thing more along the lines of "bob@somewhere" vs. "bob\@somewhere". The seond would be used if it were going into another variable (say passed as an argument to another function" |
| 01:03.14 | tzanger | tzafrir_laptop: around? |
| 01:03.21 | joat | your example, I have no idea |
| 01:04.04 | pcrane | yeah |
| 01:04.13 | pcrane | I'm passing variables around in a dial string ;) |
| 01:04.48 | *** join/#asterisk invalidrecord (n=fares@92.40.13.26.sub.mbb.three.co.uk) |
| 01:05.14 | pcrane | so, it's variables used else where |
| 01:05.39 | joat | then it depends on when you want it referenced... don't deref it when you want the local program to convert it to a value |
| 01:05.40 | invalidrecord | anyone know what the fiels is with the number dialed on an incomming call from an outside line callerid (numeric) |
| 01:06.04 | *** join/#asterisk SiberAIR (n=SibRphre@160.79.176.178) |
| 01:06.21 | pcrane | so, is there a replace function? |
| 01:06.27 | pcrane | or am I being too hopeful |
| 01:06.28 | pcrane | no |
| 01:06.29 | pcrane | wait |
| 01:06.36 | pcrane | I'll look on the wiki;) |
| 01:06.36 | joat | you may have deref it more than once if it's passed to other programs (I've seen \\\\@value before) |
| 01:06.46 | pcrane | mm |
| 01:07.00 | pcrane | it's just passed from server a to server b |
| 01:07.00 | joat | what language? |
| 01:07.21 | pcrane | so, on server a, I escape it, on server b I unescape it |
| 01:07.45 | joat | it also depends on how you're passing it between the services |
| 01:08.00 | pcrane | Dial(SIP/1~stuff) |
| 01:08.31 | moeSizlak | how is it that my caller id shows up as unavailable on normal ppl's phones, but for some reason the pizza place can see it? |
| 01:08.39 | joat | ??? not a language then... just a dialplan entry |
| 01:08.54 | joat | moe: your home phone voip? |
| 01:09.02 | *** join/#asterisk duity (n=Yeah@pool-173-65-11-44.tampfl.fios.verizon.net) |
| 01:09.25 | pcrane | yep |
| 01:09.30 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 01:09.51 | joat | moe: do you pay for callerid? |
| 01:10.10 | joat | pcrane: are you dialing an extension on that box or just passing the string? |
| 01:10.28 | pcrane | the setup is this |
| 01:10.34 | pcrane | machine a is a 1.2 asterisk machine |
| 01:10.38 | pcrane | machine b is a 1.4 |
| 01:10.53 | pcrane | I want a call to originate on the 1.2 machine, go to the 1.4 to do answer machine detectin |
| 01:10.58 | pcrane | then end up back on the 1.2 machine |
| 01:11.03 | pcrane | I've got all this working |
| 01:11.10 | pcrane | I just need some variables passed around |
| 01:11.22 | phix | hey hey hey |
| 01:11.24 | pcrane | so, the only way I can think of doing it is with a massive dial string |
| 01:11.39 | pcrane | one (or two) of the variables have @ in them |
| 01:13.56 | joat | hmm... you're using redirect in there somewhere? |
| 01:14.19 | pcrane | so... |
| 01:14.22 | pcrane | 1.2 does: |
| 01:14.37 | pcrane | Dial(1~stuff~stuff~stuff...@inter-asterisk) |
| 01:14.40 | pcrane | then 1.4 does the amd |
| 01:14.55 | pcrane | and sends the call back (adjusting the 1 to a 2 if it's a machine) |
| 01:14.59 | pcrane | kinda cool |
| 01:15.05 | pcrane | apart from these variables.. |
| 01:15.22 | phix | looks complicated and error prone |
| 01:15.40 | pcrane | yep |
| 01:16.29 | pcrane | that's what the customer's stuck with |
| 01:16.36 | pcrane | (they're unable to change the 1.2 machine) |
| 01:16.40 | phix | awesome, that will stop the newbs using it so only the true geeks can use it! an awesome idea |
| 01:16.52 | pcrane | ;) |
| 01:16.52 | joat | trixbox, huh |
| 01:16.55 | pcrane | yeah |
| 01:16.56 | joat | heh |
| 01:17.06 | pcrane | hits head against brick wall |
| 01:17.13 | pcrane | every time I try to do a dialplan relow |
| 01:17.16 | pcrane | reload* |
| 01:17.18 | joat | yeah, if there's more than one |
| 01:17.18 | pcrane | it complains |
| 01:17.25 | *** part/#asterisk moeSizlak (n=kthxbai@static-69-95-250-34.buf.choiceone.net) |
| 01:17.27 | joat | @ in there, you'll want to dereference it |
| 01:17.31 | pcrane | yeah |
| 01:17.44 | joat | and you'll probably need some tolerant AGI script on the 1.4 |
| 01:17.45 | pcrane | so, really the only way to do it is to look for it before doing the dial |
| 01:17.53 | pcrane | and cut approrialtely |
| 01:18.05 | pcrane | (or replace it with something else |
| 01:18.06 | pcrane | ) |
| 01:18.12 | joat | yep |
| 01:18.41 | pcrane | heh |
| 01:18.42 | pcrane | ok |
| 01:18.48 | pcrane | I'll sort that out |
| 01:19.15 | *** join/#asterisk moeSizlak (n=kthxbai@static-69-95-250-34.buf.choiceone.net) |
| 01:19.31 | moeSizlak | do ytou need to have an 800# to get ANI? |
| 01:19.48 | moeSizlak | or can small businesses without tollfree #'s also subscribe to ANI? |
| 01:23.34 | pcrane | I've got another problem |
| 01:23.38 | pcrane | I have a PRI installed |
| 01:23.57 | pcrane | and I want to have the inbound caller id presented |
| 01:24.09 | pcrane | but it doesn't pick it up from CALLERID(all) |
| 01:24.29 | pcrane | I know that the caller id exists (I'm calling from my cell phone so I should see it appear) |
| 01:24.32 | pcrane | but it doesn't |
| 01:24.54 | BBHoss | moeSizlak: What do you want to do with ANI? |
| 01:27.06 | moeSizlak | get the number of ppl who call me |
| 01:27.12 | moeSizlak | what else? |
| 01:29.51 | *** join/#asterisk ZenBSDi (n=bsdi@unaffiliated/ZenBSDi) |
| 01:31.28 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 01:33.21 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 01:33.25 | *** join/#asterisk homeins6 (n=root@ip-208-109-154-197.ip.secureserver.net) |
| 01:33.34 | homeins6 | How are blast groups created? |
| 01:33.58 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:35.06 | ZenBSDi | blast groups? |
| 01:35.20 | ZenBSDi | and here I thought when it comes to asterisk I heard it all :p |
| 01:35.34 | homeins6 | sorry, that is probably not the correct term |
| 01:35.35 | BBHoss | moeSizlak: can you not get that from caller id? |
| 01:35.50 | homeins6 | I call an ext, and it rings like 50 phones at once, and whoever answers first gets the call |
| 01:35.53 | ZenBSDi | What are you trying to accomplish? |
| 01:36.21 | BBHoss | homeins6: just use a bunch of &s in the dial command |
| 01:36.25 | BBHoss | it will ring them all |
| 01:36.26 | lanning | homeins6, you Dial() like 50 phones at once. |
| 01:36.30 | homeins6 | Oh ok |
| 01:36.38 | ZenBSDi | Dial(SIP/EXT&SIP/EXT&SIP/EXT,30) |
| 01:37.36 | homeins6 | I wasn't sure. Our hosted provider is able to do it. Like we dial an ext, and 50 phones ring at once. So If Person A does the "blast group" and Person B answers, they are in a call together, but sometimes, Persons C and D will end up on a call together |
| 01:39.07 | jaytee | homeins6, it would be probably useful to create a global variable or a database entry that holds a string value that contains all the devices you wish to have ring at once so that you could simply put Dial(${BLASTGROUP},20,tT) or something like that where you need it in your dialplan. |
| 01:39.45 | homeins6 | That would be awesome. I am reading the PDF on line. Is it possible to tie asterisk to a mysql table? |
| 01:39.57 | ZenBSDi | asterisk realtime |
| 01:40.01 | ZenBSDi | thats fun to setup |
| 01:40.03 | jaytee | yes it is, I use mysql for CDR data |
| 01:40.29 | jaytee | if you're new to Asterisk I'd hold off on doing realtime stuff until you master the basics |
| 01:40.30 | ZenBSDi | homeins6, just google for asterisk realtime mysql and get ready for some fun :p |
| 01:40.40 | homeins6 | uh oh |
| 01:40.41 | homeins6 | :P |
| 01:41.19 | ZenBSDi | homeins6, If you do the sip users and extensions from mysql .. thats the fun. Then you should write a script or make a good php interface to the database to use it. |
| 01:41.26 | homeins6 | I am currently reading about the dialplan applications. CURL is pretty neat. I use it in php all the time. I can make it so that if someone dials an ext, it curls a page and causes that page to generate an email, etc. |
| 01:41.52 | ZenBSDi | homeins6, phpagi :) |
| 01:42.43 | homeins6 | Ah, neat. I had seen ASTERIS:: modules in perl earlier today. Hadn't had a chance to mess around with it. I built a predictive dialer though, using php's sockets and the AMI interface. How simple is that? * devs thought of everything. |
| 01:43.08 | homeins6 | We currently use a service called JahJa on our website for a "click to call" type scenario, this could be so much easier and cheaper. |
| 01:43.23 | ZenBSDi | you built your own PD? You should look at vicidial |
| 01:43.41 | jaytee | homeins6, make sure you read chapters 5 and 6 of the pdf before reading chapters 9 and 10 which focus on AGI and AMI |
| 01:44.23 | homeins6 | ZenBSDi: Its actually more of a presumptive dialer based on the rules of our CRM system. |
| 01:44.54 | homeins6 | jaytee: Thanks, I am just in the intro right now. I checked a Barnes and Noble, they can order the book but its $45. I have seen it online for in the mid $20s |
| 01:45.33 | jaytee | homeins6, yeah B&N charges full price. I think Amazon has it cheaper |
| 01:45.38 | hardwire | ... |
| 01:46.10 | homeins6 | Spent most of the time playing with the Festival/SayDigits/SayAlpha :P |
| 01:46.11 | *** join/#asterisk pcrane (n=pcrane@121.90.92.86) |
| 01:46.22 | homeins6 | Having the other guy in the IT department dial extensions and listen to silly messages |
| 01:46.47 | jaytee | tt-monty-knights is a funny sound file to play :-) |
| 01:47.09 | homeins6 | Is that part of the core sounds? |
| 01:47.16 | jaytee | no, it's in extras |
| 01:47.33 | *** join/#asterisk jmacz (n=jmacz@201.245.230.135) |
| 01:47.38 | homeins6 | Ah ok. I saw it in the package list in the repos, but only installed asterisk and asterisk-addons |
| 01:48.19 | jaytee | so you're running just straight SIP, no analog or PRI stuff then |
| 01:48.57 | homeins6 | Correct. We do not need to do any analog lines or anything. I may eventually dable with a IVR/PRI system. |
| 01:49.21 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 01:50.11 | jaytee | as a general rule I install libpri and either zaptel or DAHDI before Asterisk whether I'm going to use TDM or not because if you add them afterwards you have to recompile asterisk so I figure why not have it there if I need it. |
| 01:50.13 | homeins6 | How does AMD work? |
| 01:50.27 | homeins6 | I installed from RPM. |
| 01:50.31 | *** join/#asterisk chendy (n=chatzill@121.34.152.108) |
| 01:50.47 | jaytee | homeins6, type core show application AMD from the CLI |
| 01:52.17 | homeins6 | ah thanks |
| 01:52.35 | jaytee | but basically it's analyzing the rtp stream incoming for timing patterns and making an educated guess as to what's on the other end, human or machine |
| 01:52.58 | homeins6 | Thats really amazing. |
| 01:53.27 | homeins6 | If it detects an answering system, it could leave a pre-recorded message, and the agent would never have to even be on the line. |
| 01:53.29 | jaytee | it's cool but it's not always 100% accurate or they wouldn't have included a NOTSURE result :-) |
| 01:53.48 | homeins6 | hehe |
| 01:55.02 | homeins6 | I saw some voice actors/actresses online that do records for auto attendants |
| 01:56.14 | jaytee | homeins6, Allison Smith, the voice of Asterisk will do custom prompts and recordings. She has her own website. |
| 01:56.16 | homeins6 | I made the mistake of installing the win32 version yesterday at home. Lasted about 30 minutes :) |
| 01:56.38 | homeins6 | jaytee: I am going to have a hard time justifying that in the budget :P cheapskates. |
| 01:56.41 | homeins6 | ~cheap |
| 01:56.41 | jbot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 01:56.55 | jaytee | homeins6, that was actually an April Fool's joke that got out of hand and developed a life of it's own. it won't go away. |
| 01:57.23 | jaytee | homeins6, yeah I know all about cheapskates |
| 01:58.05 | homeins6 | Why are grandstream phones bad? We have vodavis, but I saw someone in here earlier do a command to the bot. |
| 01:59.03 | jaytee | homeins6, I've used both GS and Polycom. Polycom tends to cost a little more but the difference in price isn't that significant when you consider the gains in quality and the savings in time having to make things work properly. |
| 01:59.31 | homeins6 | We have a polycom analog device. I am not sure what the name for it is, but you use them in conferences, so that multiple can sit around and talk on it? |
| 02:00.04 | jaytee | I have 3 GSX-2000's I'd love to pitch in a dumpster. When I first setup Asterisk my MOH would cut in and out unless I was constantly blowing into the mic on the phone's handset. |
| 02:00.13 | jaytee | echo was terrible and lots of jitter |
| 02:00.14 | homeins6 | A buddy at work brought in a Grandstream SIP hardphone today and we were playing around w/ it. One of the lower priced models, like 1 line, $50 or so. It was pretty neat. |
| 02:00.55 | homeins6 | I am extremely new to voip. So I am trying to learn the mistakes from others before I make them. |
| 02:00.57 | jaytee | turns out I had to upgrade the firmware because of a bug in the version the phone came with that even if you turned silence suppression off in the web gui it wouldn't actually turn it off in the firmware. |
| 02:01.26 | jaytee | ~gs |
| 02:01.26 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 02:01.35 | jaytee | ~grandstream |
| 02:01.35 | jbot | extra, extra, read all about it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 02:01.47 | homeins6 | I saw a wireless IP phone that was kind of neat. It was a cell phone style. Not the coordless ones. |
| 02:02.05 | jaytee | Polycom bought Spectralink |
| 02:02.13 | homeins6 | http://www.telephonydepot.com/product_p/105-056-101.htm |
| 02:02.23 | homeins6 | What do you call those polycoms that are used for conferences? |
| 02:02.23 | Carlos_PHX | Grandstream....ugh...kill me |
| 02:02.32 | Carlos_PHX | homeins6: IP 4000 |
| 02:02.48 | jaytee | Soundstation |
| 02:03.03 | Carlos_PHX | homeins6: I got about 30 of those phones free. I tried a few, and then gave them all away. |
| 02:03.10 | homeins6 | :D |
| 02:03.12 | jaytee | homeins6, I have two of those as well. they work fair |
| 02:03.27 | Carlos_PHX | If you want cheap AND good, try the Linksys SPA 921 |
| 02:03.31 | homeins6 | Carlos! :) <--- James Freeman |
| 02:03.55 | Carlos_PHX | If you want super-cheap with no frills, Linksys has one with no display but still good sound and management. |
| 02:04.08 | Carlos_PHX | I kinda figured from the nick |
| 02:04.24 | Carlos_PHX | You guys are making big changes over there, huh? |
| 02:04.47 | homeins6 | I installed a random nick chooser when I was playing with perl modules, and it always chooses a different one. I never can tell the difference |
| 02:04.56 | homeins6 | Yeah, we are fed up with our current system. |
| 02:05.05 | Carlos_PHX | I dated a girl like that once...random personalities. |
| 02:05.17 | jaytee | you dated her too? damn |
| 02:05.32 | Carlos_PHX | Well, she had a twin sister, so mighta been the other one. |
| 02:05.36 | homeins6 | Anytime we need to add a seat, or reprovision a phone, its a 24 hours or more turn around. Also, we are in this situation where we can't have more than 50 users on 1 "blast group" |
| 02:05.41 | homeins6 | its getting to be a mess |
| 02:05.53 | Carlos_PHX | Heh, yeah, that makes no sense. |
| 02:05.58 | jaytee | Carlos_PHX, did yours have a personality named "Marci" that liked anal and bondage? |
| 02:06.14 | Carlos_PHX | How did you know? Why else would I have kept her around?? |
| 02:06.25 | jaytee | hehe |
| 02:06.40 | homeins6 | Also, they are blocking/filtering the most common SIP ports, so we can't even pass a device through. |
| 02:07.01 | Carlos_PHX | Bandwidth is...?? |
| 02:07.04 | homeins6 | Our setup is Fibre Internet -> Bandwidth.com Router -> Switches |
| 02:07.05 | homeins6 | Yes |
| 02:07.13 | Carlos_PHX | What a pain. |
| 02:07.32 | homeins6 | Our fiber optic junction box only allows 1 device to be attached to it, even though it has multiple LAN ports on the first one is enabled, and they wont enable another port. |
| 02:07.43 | homeins6 | Or else we could just throw another box along side of it and avoid the problem. |
| 02:07.58 | Carlos_PHX | Huh, well, so you just put in a router? Oh, does Bandwidth control your internet too? |
| 02:08.09 | homeins6 | No, only the router. |
| 02:08.42 | homeins6 | We asked them about putting a nice router in front of it, and they said that the way that the box is setup, that it would cause a lot of problems because of NAT, etc.. |
| 02:08.50 | Carlos_PHX | Sheesh |
| 02:09.26 | Carlos_PHX | Well, you could always colo a box with us. Not to try to sell, but it's an option. We do that for a few people. |
| 02:09.32 | homeins6 | Also, the phones they sold us are MGCP only. |
| 02:09.37 | Carlos_PHX | Argh |
| 02:09.39 | Carlos_PHX | Bastards |
| 02:09.42 | homeins6 | I think we are going to bring everything in house. |
| 02:10.05 | Carlos_PHX | Yeah, it makes sense to have control of your own server. |
| 02:10.14 | homeins6 | Yes! AND not only that, but , I am not sure how they are even registering. The phones listen on port 2727 (i think) and i nmap'd them on tcp and udp... and they are filtered. |
| 02:10.17 | Carlos_PHX | What phones do you have? |
| 02:10.29 | homeins6 | LG Nortels 6812 |
| 02:10.34 | ZenBSDi | speaking of colo.. |
| 02:10.34 | homeins6 | Nortel/Vodavi.. |
| 02:10.42 | ZenBSDi | who knows a good cheap one? |
| 02:10.47 | Carlos_PHX | Huh, well, one way to keep your customers, I mean other than great service, is to just make everything proprietary. |
| 02:11.05 | Carlos_PHX | !cheap |
| 02:11.07 | Carlos_PHX | Oops |
| 02:11.10 | Carlos_PHX | ~cheap |
| 02:11.10 | jbot | cheap is probably a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 02:11.20 | homeins6 | So, I run wireshark to figure out why * isn't seeing the connection from the phone , and i get a lot of errors where Asterisk can't connect to the phones, ICMP destination unreachable. |
| 02:11.37 | Carlos_PHX | We do colo for $50/u if you're a customer of our other services. |
| 02:12.00 | Carlos_PHX | homeins6: That's odd. |
| 02:12.26 | Carlos_PHX | Well if you're looking at new phones do look at the Linksys, we have hundreds deployed with 100% satisfaction. |
| 02:12.40 | Carlos_PHX | Where we have both Polycom and Linksys, the customers are asking for more Linksys. |
| 02:12.41 | *** join/#asterisk SiberAIR (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 02:12.45 | homeins6 | The only ports that are open are 8000 a web interface for configuring the phone , and port 6000 |
| 02:13.13 | homeins6 | Can't find any documentation on the web for making this phone work with asterisk :/ |
| 02:13.16 | Carlos_PHX | 6000? Isn't that Xwindows? |
| 02:13.25 | homeins6 | I think so. |
| 02:14.07 | homeins6 | I contacted nortel there is no way to upgrade them to SIP firmware. |
| 02:14.22 | Carlos_PHX | Nortel and "open" are rarely found in the same sentence. |
| 02:14.56 | Carlos_PHX | You can talk MGCP with Asterisk (I've never tried it), but if they are also locked down... |
| 02:15.18 | homeins6 | Well here is what is squirrely. We purchased a few nortels off of ebay, and are having the same issue. |
| 02:15.42 | homeins6 | Its says they are connecting to 2742.. or something for the asterisk box, and then expecting 2772, but, those ports are closed aswell |
| 02:15.48 | homeins6 | Making the ports up from memory ^ |
| 02:16.47 | giovani | homeins6: Nortel phones run UNIS afaik |
| 02:16.48 | Carlos_PHX | Well hey, looks like you can get $50 for them on eBay. |
| 02:17.02 | homeins6 | damn |
| 02:17.06 | homeins6 | We paid $70 :P |
| 02:17.20 | giovani | http://www.voip-info.org/wiki/view/Asterisk+UNISTIM+channels |
| 02:17.21 | Carlos_PHX | Huh, saw at least one completed sale for $50 |
| 02:18.17 | homeins6 | giovani: My model isn't listed in that site. It says it is a nortel 6812 mgcp on the bottom. |
| 02:18.37 | giovani | ah ok |
| 02:19.13 | homeins6 | thank you though. |
| 02:20.23 | giovani | they probably work |
| 02:20.31 | giovani | I doubt they're actually broken |
| 02:20.40 | *** part/#asterisk moeSizlak (n=kthxbai@static-69-95-250-34.buf.choiceone.net) |
| 02:20.49 | homeins6 | They work, we use them w/ bandwidth.com , we just cant get them to register to our asterisk box. |
| 02:20.54 | homeins6 | Perhaps, bandwidth.com is using another pbx? |
| 02:21.14 | giovani | I wouldn't presume bandwidth.com to be using asterisk |
| 02:21.38 | giovani | maybe you just didn't configure asterisk properly, I don't know |
| 02:21.39 | giovani | google around |
| 02:21.52 | homeins6 | I wasn't sure, because our Edgemark router is running asterisk as like pass through device |
| 02:22.02 | homeins6 | I ssh'd in and saw asterisk, but no conf files |
| 02:22.24 | giovani | well it's configured somewhere :) |
| 02:22.49 | homeins6 | Its a pretty strippe down version of linux. Most commands segfault. |
| 02:22.52 | Carlos_PHX | That's a strange one. |
| 02:22.58 | *** topic/#asterisk by lmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- SVN IS DOWN FOR MAINTENANCE TILL ~TOMORROW MORNING CST -=- Asterisk 1.6.0.1 (2008/10/09), 1.4.22 (2008/10/02), *-Addons 1.6.0 (2008/10/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 02:23.32 | Carlos_PHX | So either they have it nailed down, or someone has hax0r3d your router... |
| 02:23.45 | homeins6 | :D |
| 02:23.53 | *** join/#asterisk edp (n=ed@c-98-193-105-250.hsd1.il.comcast.net) |
| 02:25.08 | homeins6 | I think bringing it in house is the best solution for us. Except, who do you blame when it crashes or phone quality sucks? :/ |
| 02:25.40 | Carlos_PHX | Well, yeah, there's that. Ask me about the first couple years in running a call center on Asterisk 0.9... |
| 02:25.58 | Carlos_PHX | But mostly you will be in better shape on your own, assuming skills, which apparently your team has. |
| 02:26.24 | homeins6 | Ryan, who you spoke with earlier is very fluent in voip technologies, he does the pots, or whatever too... if that makes sense? lol |
| 02:26.42 | homeins6 | I am pretty good w/ linux and programming, but learning the voip portion so that i can back him up if he goes on vacation or is sick |
| 02:27.01 | Carlos_PHX | Yeah, I'm pretty sure you're in decent shape. |
| 02:27.10 | homeins6 | Reading about MeetMe() right now, that is amazing. |
| 02:27.11 | Carlos_PHX | I mean, once it works, really, I never hear from our call center customers. |
| 02:27.28 | Carlos_PHX | 99% of call center issues are from analog devices. |
| 02:27.36 | Carlos_PHX | Dialers and headsets on analog lines. |
| 02:27.39 | Carlos_PHX | Ug |
| 02:27.49 | homeins6 | A lot of the partner companies that we do business w/ have an 800 # where you dial in, enter an access code, and then you are all in a bridged conference call, that is awesome. |
| 02:28.12 | homeins6 | Does this exist? Have 1 SIP account, w/ unlimited outbound calls at one time? |
| 02:28.47 | Carlos_PHX | Yeah, you can make your own conferences at will, record them, whatever. |
| 02:28.56 | Carlos_PHX | Yes, that's what we do. |
| 02:29.22 | giovani | homeins6: uh "unlimited" meaning, no charge per minute? |
| 02:29.30 | giovani | or as in, no restrictions on number of concurrent calls? |
| 02:29.38 | homeins6 | Ok, because we were looking at some sip trunk providers online, and you had to have 1 line per user |
| 02:29.43 | homeins6 | unlimited concurrent calls |
| 02:29.53 | giovani | any decent termination provider offers that |
| 02:30.01 | giovani | you pay per minute |
| 02:30.06 | giovani | as many calls/minutes as you want to push |
| 02:30.18 | giovani | within reason, of course |
| 02:30.23 | homeins6 | Probably a dumb question, but if I call Person A, and then Call Person B, and conference them together, and I leave the call through a hangup, whose connection is that on then? |
| 02:30.25 | Carlos_PHX | See, here's the thing. We all have to make money, and every call costs somewhere. So we can either play games with "unlimited" or number of channels, or just tell you up front what costs us money and how to make it work. |
| 02:30.30 | giovani | I'm sure many would choke if you wanted to send a few thousand concurrent calls |
| 02:30.39 | Carlos_PHX | NOBODY has unlimited for real. Doesn't exist. |
| 02:30.56 | giovani | homeins6: that's tricky :) |
| 02:31.05 | giovani | homeins6: you're speaking internally in your office? |
| 02:31.08 | homeins6 | Maybe 100 concurrent at a time. Most are going to be no answer or busy signals. |
| 02:31.10 | Carlos_PHX | So for call centers we typically do unlimited concurrent with per-minute billing. |
| 02:31.14 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 02:31.23 | homeins6 | giovani: Well, say I call someone outside of my office, and bridge them to someone else outside of my office |
| 02:31.32 | Carlos_PHX | homeins6: FYI, that type of dialing is considered abusive by most carriers. |
| 02:31.41 | giovani | homeins6: you'll want to contact most providers first and make sure it's ok to be pushing telemarketing traffic out through them |
| 02:31.51 | homeins6 | Its not telemarketing. |
| 02:32.01 | Carlos_PHX | For example, Vitelity says you can't have more than 20% auto-dial calls, and a certain ratio of dials versus minutes. |
| 02:32.06 | giovani | homeins6: the call will be using up two channels then |
| 02:32.17 | homeins6 | Its people that have come to our website and are generally interested in our product. |
| 02:32.25 | giovani | homeins6: one for person A, one for person B, the bridging is happening at your PBX |
| 02:32.26 | Carlos_PHX | From a carrier perspective, dial/no answer is very costlyl. |
| 02:32.36 | homeins6 | Well, these arent auto dials.. like in the background, its preally just to make it so our agents dont have to dial the number manually |
| 02:32.59 | giovani | why are most of them no-answer then? |
| 02:32.59 | Carlos_PHX | Yeah, it's the dial/no answer ratio that puts load up. |
| 02:33.02 | giovani | if people are requesting the calls |
| 02:33.08 | homeins6 | good question :P |
| 02:33.13 | Carlos_PHX | We know about it, and choose to deal with it, but many of our back end providers won't. |
| 02:33.21 | giovani | sounds like you have a bad business practice there :) |
| 02:33.25 | homeins6 | Calling them at the wrong times, they do not want a call, but they submitted their information anyhow..etc |
| 02:33.50 | homeins6 | Or, they are already on the phone w/ one of our partner companies, etc. |
| 02:34.02 | giovani | then it sounds like you shouldn't be auto-calling |
| 02:34.04 | giovani | but, whatever |
| 02:34.07 | Carlos_PHX | I have a call center that does member calls for a major medical company--people WANT this call, and they still get something like 80% no answer. |
| 02:34.12 | homeins6 | So, within our crm system, once you pull a "lead", the system will call you, then immediately call the customer as soon as you pick up |
| 02:34.17 | *** join/#asterisk zchaos (n=none@CPE0013f7ef8b58-CM0013f7ef8b54.cpe.net.cable.rogers.com) |
| 02:34.24 | homeins6 | Carlos_PHX: exactly :) |
| 02:34.45 | homeins6 | Carlos_PHX: Just trying to eliviate(sp) some of the headache, and also increase productivity by forcing agents to dial |
| 02:34.59 | homeins6 | Then we can see on the back end if the agent is immediately hanging up, etc .. |
| 02:35.04 | Carlos_PHX | Oh yeah, I'm with you, we help people automate as much as possible. |
| 02:35.31 | Carlos_PHX | I'm just tossing out those factoids on the dialer so you know the carrier perspective on it also. |
| 02:35.54 | homeins6 | If the read the notes associated w/ a customer and they are like "oh.. this person has been called 5 million times in 2 days,im not calling them" and they just mark it as contacted anyhow |
| 02:37.14 | giovani | well that's more common sense than having an automated system attack-dial someone |
| 02:37.47 | homeins6 | Well, most of the time someone submits information on our website, but they are at work when we do it, and we call their home #... |
| 02:38.05 | homeins6 | So, maybe in the afternoon when they are home an waiting on a call, we never do, because the agents are making their own rules |
| 02:38.05 | giovani | if someone really wants to speak to a company, they call themselves |
| 02:38.11 | giovani | they don't wait for an automated system to call them |
| 02:38.27 | Carlos_PHX | giovani: You don't seem to grasp the stupidity of the average consumer. |
| 02:38.37 | homeins6 | lol |
| 02:38.45 | Carlos_PHX | Here's an example. |
| 02:38.46 | giovani | well ... I don't know any average consumer that requests "callbacks" |
| 02:38.52 | Carlos_PHX | Customer: My fax isn't working. |
| 02:38.57 | Carlos_PHX | Me: How do you know? |
| 02:39.09 | homeins6 | giovani: You are very intelligent and think logically.. most customers dont :P |
| 02:39.12 | Carlos_PHX | Her: Because I keep trying to send myself a test fax and it's always busy. |
| 02:39.14 | giovani | I know normal people, 99% of the consumer population looks up the phone number and calls it when they want to speak to a company |
| 02:39.40 | giovani | if that werent' the case, then consumers would never call companies, and there'd be no need for incoming call centers |
| 02:39.49 | Carlos_PHX | giovani: This morning I was shopping for CNAM database providers and filled out half a dozen forms for them to call me. |
| 02:39.53 | Carlos_PHX | It's easier that way. |
| 02:39.59 | homeins6 | Exactly. |
| 02:40.02 | giovani | how is that easier than dialing a number? |
| 02:40.10 | Carlos_PHX | I tried to call Verisign... |
| 02:40.11 | giovani | filling out a form takes minutes |
| 02:40.13 | Carlos_PHX | Phone tree... |
| 02:40.15 | homeins6 | Let them do all the quoting and processing on their end, so my call is less time |
| 02:40.20 | giovani | ok -- so that's the company's fault |
| 02:40.22 | Carlos_PHX | Then someone who didn't know what CNAM is. Then hold. |
| 02:40.24 | Carlos_PHX | Then... |
| 02:40.46 | giovani | ok, those are faulty phone systems, we're not comparing an average phone tree to a callback system |
| 02:40.46 | Carlos_PHX | In my browser I right-click and select "fill form" and submit. |
| 02:41.08 | cvnet | [Nov 20 21:40:25] WARNING[3262]: rtp.c:891 ast_rtcp_read: RTCP Read too short <-- what does this mean? |
| 02:41.20 | justdave | is there a way to set how frequently asterisk will re-register a sip or iax connection that it has to register for as a client? |
| 02:41.21 | giovani | yep, except that you end up writing out and answering questions so they can fill their database, and then have no clue what you want when they call, etc |
| 02:41.24 | giovani | what a waste of time |
| 02:41.32 | giovani | but, ok |
| 02:41.37 | justdave | The only related options I can find in the docs seem to apply to server-side of the connection rather than asterisk being the client |
| 02:42.29 | Carlos_PHX | giovani: You are just too much smarter than me. I just learned to use teh intarnets yesterday so I'm tryin' it out. |
| 02:43.12 | giovani | haha |
| 02:43.16 | giovani | ok |
| 02:43.38 | giovani | I just know that the reason companies have these callback systems is because it's cheaper for them, and they get more people that way |
| 02:43.48 | giovani | at the cost of being obnoxious |
| 02:52.08 | *** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 02:52.11 | kb3ien | anyone seen this before? http://pastebin.com/mf3441c2 |
| 02:52.45 | kb3ien | my iax peers keep desyncing. a quick `module reload` seems to fix everything. |
| 02:55.34 | BBHoss | kb3ien: looks like voipjet is changing its port to 3648 from 4569, the proper iax2 port |
| 02:56.20 | BBHoss | dunno why |
| 02:56.30 | BBHoss | maybe look at iax2 debug for tips |
| 02:56.36 | BBHoss | or just use SIP |
| 02:57.29 | cvnet | WARNING[3262]: rtp.c:891 ast_rtcp_read: RTCP Read too short <-- what does this mean? |
| 03:04.28 | justdave | kb3ien: that's what I was just trying to find out registration timeout information for |
| 03:04.32 | justdave | same problem I was having |
| 03:04.43 | justdave | iax2 reload fixed it for me |
| 03:06.26 | justdave | in my case it was with detele.dk |
| 03:08.08 | kb3ien | it fixes it, for about 20 minutes... |
| 03:18.23 | kb3ien | okay getting some good debug now. now to play the waiting game... |
| 03:21.00 | *** join/#asterisk ManxPower (n=manxpowe@247.sub-70-220-70.myvzw.com) |
| 03:24.06 | kb3ien | all turning on debugging did was: render the 'module reload' command impotent! |
| 03:24.32 | kb3ien | also the useful messages scrolled by whilst i was using the web browser! |
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| 03:31.46 | kb3ien | [Nov 20 22:43:58] NOTICE[8265]: dnsmgr.c:170 dnsmgr_refresh: dnssrv: host 'dialer2.voipjet.com' changed from 208.72.186.71:4569 to 208.72.186.71:3648 |
| 03:31.46 | kb3ien | white*CLI> [Nov 20 22:43:58] NOTICE[8265] dnsmgr.c: dnssrv: host 'dialer2.voipjet.com' changed from 208.72.186.71:4569 to 208.72.186.71:3648 |
| 03:31.46 | kb3ien | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE |
| 03:31.46 | kb3ien | <PROTECTED> |
| 03:32.15 | *** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 03:32.17 | kb3ien | sorry |
| 03:32.24 | kb3ien | bad puter. |
| 03:32.37 | kb3ien | http://pastebin.com/m5792a1dd |
| 03:33.01 | kb3ien | i'm using my wife's mac and x11 has some NASTY clipboard bugs. typical apple... |
| 03:34.20 | Carlos_PHX | Why the hell would anyone use X11? Typical Windows user. |
| 03:35.41 | kb3ien | hrm, you know how to boot darwin into consol mode if its installed as macos, i'm all game. |
| 03:35.58 | justdave | hold Command-S during the boot chime |
| 03:36.07 | justdave | and hold it until the text starts scrolling |
| 03:36.13 | kb3ien | i dont spend enough time in the windowed world to be proficient. |
| 03:36.31 | kb3ien | AH. danke. |
| 03:37.42 | kb3ien | still if anyone knows what's up with voipjet and dnsmgr i'm eager to fix this. |
| 03:39.21 | kb3ien | addendum: [Nov 20 22:45:02] NOTICE[8285] chan_iax2.c: Peer 'dialer.voipjet' is now UNREACHABLE! Time: 13 \n[Nov 20 22:45:59] WARNING[8313] chan_skinny.c: Skinny Client sent less data than expected. Expected 4 but got 3.\n[Nov 20 22:45:59] WARNING[8313] chan_skinny.c: Trying to delete nonexistent session 0x8770e0?\n (last line 2 times) then asterisk exits abruptly.... |
| 03:39.48 | kb3ien | man there are a lot of blinking red leds all of a sudden... |
| 03:39.49 | jtodd | OK. Perhaps I am missing something obvious in the year or two since it's been that I set up * behind a NAT... |
| 03:40.11 | jtodd | But is externip= broken? I've set it, but I still see my internal IP address in all my SIP dialogs. |
| 03:41.25 | [TK]D-Fender | jtodd: no |
| 03:41.27 | jtodd | This is with a bone stock (i.e.: no changes in sip.conf other than externip and context) asterisk 1.6.0.1 install. |
| 03:41.46 | ManxPower | you need a localnet= too |
| 03:41.50 | [TK]D-Fender | jtodd: trash everything commented, and pastebin the rest masking only passwords |
| 03:41.54 | ManxPower | externip does not take a hostname |
| 03:42.12 | jtodd | I'm using IP addresses. I got that much. :-) So localnet is a requirement for externip ? |
| 03:42.25 | *** join/#asterisk kerx (n=prepro@adsl-68-125-33-231.dsl.irvnca.pacbell.net) |
| 03:42.33 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-13-32.phlapa.east.verizon.net) |
| 03:42.33 | [TK]D-Fender | jtodd: Gotta know what counts as "external" |
| 03:42.49 | jtodd | I would have assumed if you had "externip" set, it would treat all traffic the same. |
| 03:43.17 | jtodd | In other words, why would it even assume there was a localnet that was different? Localnet seems to be an "exception" rule, and exceptions are typically not required. |
| 03:43.20 | jtodd | But I'll give it a try. |
| 03:43.31 | [TK]D-Fender | jtodd: I don't disagree with that methodoly, but it doesn't seem to be the case |
| 03:44.17 | [TK]D-Fender | jtodd: Think of it as Externip being the exception, and localnet being the exception to the exception :p |
| 03:44.40 | jtodd | Ah. Uh. Ah. |
| 03:45.05 | Carlos_PHX | Assuming that Asterisk uses logic is fraught with peril. |
| 03:45.23 | jtodd | Yes, that seems to do the trick, but that is indeed perilous logic. |
| 03:45.37 | *** join/#asterisk telnettech (n=telnette@12.236.122.2) |
| 03:45.49 | jtodd | I hate NATs. This is the first * I've set up behind a NAT in a long, long time. But EC2 makes it a requirement. |
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| 03:49.13 | *** mode/#asterisk [+o Corydon76-dig] by irc.freenode.net |
| 03:49.13 | [TK]D-Fender | *b00m* |
| 03:49.28 | jtodd | Holy segmentation, IRC-man! |
| 03:50.50 | jtodd | Anyone done any work with Asterisk on EC2, by any chance? I'm still trying to understand some of the EC2 fundamentals, but it seems like a big win with the new bridging code that (supposedly) allows MeetMe without zap timers. |
| 03:51.23 | jtodd | Don't know how much it would cost for a lot of RTP to/from the system, though. Could be pretty pricey with the cost per GB. |
| 03:53.58 | [TK]D-Fender | jtodd: I've heard everybody go crazay and when the reality settles in its "meh" |
| 03:54.20 | jtodd | Seems to be nice to offload stuff out of one's own server room and bandwidth, though. |
| 03:57.21 | [TK]D-Fender | jtodd: And then just watch the $ meter roll like a Jerry Lewis Telethon :p |
| 03:57.32 | jtodd | that does seem to be the problem, yes. |
| 03:57.44 | jtodd | But it depends on what your goals are and what you have to spend and the people you have on hand. |
| 03:58.32 | [TK]D-Fender | jtodd: I get the feeling that EC2 is a hard option to effectively leverage from a budgetary point of view |
| 03:58.57 | [TK]D-Fender | jtodd: To send enough its way for teh offload to be of value, but not so much as to break the bank |
| 03:59.58 | jtodd | "It depends." If I was building an application using Asterisk, for instance, that I needed to go from 10 channels of voice to potentially 10,000 I might consider EC2 after some further testing. |
| 04:00.16 | jtodd | Quick scale is possible, in a matter of hours, with an appropriately written app. |
| 04:00.32 | jtodd | and if you never use that scale, then you just don't pay for it. |
| 04:01.02 | jtodd | Keeping a bunch of servers around for flash crowds is a very expensive proposition, especially if you're paying for 220V at 20A times 2 per rack. |
| 04:01.17 | jtodd | (or 4, as I was.) |
| 04:01.52 | [TK]D-Fender | jtodd: Yes.. tricky to leverage. |
| 04:02.47 | jtodd | I typically have either app designers or sysadmins who are able to build the stuff themselves out of the parts on hand, and they'd be happy to get rid of the hardware for just a tiny bit more thought in designing the app. |
| 04:02.58 | hi365_m | svn is still down? |
| 04:03.03 | jtodd | Yes, SVN is still down. |
| 04:03.09 | jtodd | ETA: "This evening." |
| 04:03.12 | hi365_m | any eta? |
| 04:03.15 | jtodd | s/ETA/ETR/ |
| 04:03.21 | hi365_m | cool. thanks |
| 04:03.29 | hi365_m | R=? |
| 04:03.33 | jtodd | Repair. |
| 04:03.37 | hi365_m | :) |
| 04:04.11 | hi365_m | arent oyu supposed to do such things durring non business hours? |
| 04:04.22 | hi365_m | i geuss that the beauty of open source |
| 04:04.41 | jtodd | It's unclear how much of this event is scheduled versus unscheduled. |
| 04:05.09 | jtodd | But I know it's being worked on. |
| 04:08.04 | [TK]D-Fender | ETR : Eventually To Return. As in sit back and grab a drink |
| 04:08.34 | telnettech | if the *8 is setup in the features.conf and the extensions are in the same callgroup and pickup group, do you have to set anything up in the dialplan for the feature to work? |
| 04:14.01 | telnettech | anybody? |
| 04:14.32 | *** join/#asterisk jman24 (n=justinh@c-24-21-168-118.hsd1.or.comcast.net) |
| 04:15.01 | jman24 | Sorry if I sound like a nube but is the asterisk svn server down? |
| 04:15.57 | jman24 | duh.. sorry it's in the header.. gn |
| 04:16.07 | *** part/#asterisk jman24 (n=justinh@c-24-21-168-118.hsd1.or.comcast.net) |
| 04:17.00 | [TK]D-Fender | telnettech: Shouldn't |
| 04:18.57 | kb3ien | My new freezer has leveled out at -30 that's better than average? |
| 04:18.58 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 04:19.22 | telnettech | TK: any idea what needs to be uncommented in the feature.conf sample file to make the features work? I have the *8 line uncommented. Anything else? |
| 04:19.42 | [TK]D-Fender | telnettech: Did you restart * following that? |
| 04:19.50 | telnettech | yes |
| 04:20.09 | kb3ien | I'm more keen on getting a fault-tolerant dial now. figuring even if i think its fixed, it may not be. |
| 04:20.21 | [TK]D-Fender | telnettech: What happens when you hit *8? |
| 04:20.41 | telnettech | TK: i get a fast busy(error tone) |
| 04:20.48 | [TK]D-Fender | telnettech: What phone? |
| 04:21.04 | telnettech | as if it doesnt know what it is supposed to do |
| 04:22.50 | telnettech | i verified that the module is running (app_directed_pickup.so) |
| 04:23.01 | [TK]D-Fender | telnettech: What phone? <-------- |
| 04:23.07 | telnettech | TK: Grandstream GXP 2000 |
| 04:23.23 | [TK]D-Fender | telnettech: gO VERIFY IN sip DEBUG THAT * IS EVEN SEEING THE REQUEST |
| 04:23.37 | telnettech | ok |
| 04:23.40 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 04:24.16 | telnettech | TK: btw.....im using 1.2 version |
| 04:24.24 | telnettech | that doesnt make a difference right? |
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| 04:27.23 | [TK]D-Fender | telnettech: Go check what I told you. |
| 04:33.42 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 04:51.09 | *** join/#asterisk pcrane (n=pcrane@202.20.97.154) |
| 04:52.11 | pcrane | does anyone know about call waiting for Linksys spa 942s? |
| 04:53.45 | [TK]D-Fender | pcrane: what about it? |
| 04:53.59 | pcrane | if it exists? where to find settings for it? |
| 04:54.15 | pcrane | I've had a look through every page of the phone, and I can't find anything |
| 04:54.22 | pcrane | *sigh* |
| 04:57.58 | [TK]D-Fender | pcrane: tell it to use more that 1 key |
| 04:58.15 | pcrane | mmm |
| 04:58.32 | pcrane | the tireness must be setting in, I ment 922 |
| 04:58.32 | pcrane | not 942 |
| 04:59.58 | [TK]D-Fender | pcrane: Go look in the CLASS codes section to see if CW is enabled |
| 05:00.45 | pcrane | under Supplementart Services? |
| 05:00.49 | pcrane | CW Setting? |
| 05:02.43 | [TK]D-Fender | pcrane: big list |
| 05:03.39 | pcrane | that does it |
| 05:03.46 | pcrane | was expecting it to be call waiting |
| 05:03.48 | pcrane | or something |
| 05:03.50 | pcrane | not CW |
| 05:03.54 | pcrane | cheers [TK]D-Fender |
| 05:07.49 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
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| 05:29.27 | yidiyuehan | hi, anybody knows why it couldn't catch DTMF tone correctly during a call? and where I could change the dtmf minimum detection timing ? |
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| 05:35.57 | [TK]D-Fender | yidiyuehan: From where? |
| 05:39.11 | demonist | ok |
| 05:39.15 | demonist | time for me to get a new job |
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| 05:41.10 | adilwali | hi anyone knows a good softphone that will do wideband with g722? |
| 05:43.22 | yidiyuehan | Hi, D-Fender, what i want to do is to detect the dtmf tone correctly when two parties are in call |
| 05:44.04 | yidiyuehan | however I noticed that * does not detect it correctly as long as two parties are in a call, if I press digit 1, it may detect as 111, or 1111, or 11 |
| 05:44.49 | yidiyuehan | For IVR system it's working fine and it could detect it correctly as there is no conversation yet. |
| 05:48.38 | [TK]D-Fender | yidiyuehan: and I asked what YOU were calling on |
| 05:50.03 | yidiyuehan | Hi, D-Fener, I called from a internal SIP Phone ==> FXO of asterisk 1 ==> FXO of asterisk 2 ==> another SIP phone |
| 05:50.34 | yidiyuehan | I could do another testing use sip phone to ==> FXO ==> analog phone |
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| 06:39.25 | demonist | be quiet you |
| 06:39.33 | demonist | i know youre the one in control of my brain |
| 06:39.39 | demonist | i know youre the one speaking through her |
| 06:41.56 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 06:41.58 | Chris-NB | hi |
| 06:42.08 | Chris-NB | anyone tried mass deployment with snom phones? |
| 06:42.38 | Chris-NB | and have some experience how to upgrade firmware from 6.x to 7.x via auto provisioning and dhcp? |
| 06:43.02 | Chris-NB | my problem is firmware 6.x uses text files, 7.x uses xml files. |
| 06:43.27 | Chris-NB | I want my 6.x phones to upgrade firmware and then provision with xml files |
| 06:43.33 | Chris-NB | is this possible? |
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| 07:07.33 | Jenna | hey all, I just could find any info of this term ¨premise-based PABX¨. I know what asterisk is & what voip is but wtf is this pemise-based thing |
| 07:07.54 | Jenna | anyone ? |
| 07:10.27 | jtodd | "Premise-based" means that the equipment is located at your business or otherwise "on site" at your location. |
| 07:16.38 | *** part/#asterisk ramuk (n=ramuk@208-78-67-58-accessmedia3-inc-metrop.pt2.ord.sparkplugbb.net) |
| 07:20.39 | Jenna | jtodd, thanx. Ã was venture into preface/prose n stuff |
| 07:21.11 | Jenna | dont u just hate it when marketeers try to make mole out of hill of any thing basic. throwing buzz words to sell u products |
| 07:21.19 | Jenna | venturing/* |
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| 07:39.14 | puppet | wewt N95 connected to the * |
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| 08:03.45 | raasdnil | everyone see the new interface to Asterisk 3.0 ? http://oblong.com/ |
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| 08:13.01 | farah | I have a question: SIP works through TCP, and IAX2 through UDP. But can we change the implementation of IAX2 so that it can work with TCP? |
| 08:13.44 | hadronzoo | Hello, is there a way to mix a periodic prerecorded message in realtime over MoH? |
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| 08:18.45 | farah | can anyone answer my question please? |
| 08:21.54 | kaldemar | farah: sure you can. |
| 08:22.41 | farah | so IAX2 supports TCP? |
| 08:22.46 | kaldemar | no. |
| 08:23.02 | kaldemar | you asked if you can change the implementation, and i said yes. |
| 08:23.21 | farah | ah ok thank you |
| 08:23.25 | yang | farah: I think SIP uses UDP for VOIP traffic to |
| 08:23.38 | farah | but where should i change it? |
| 08:23.40 | hadronzoo | I guess I will have to generate the stream using another application and then tie the MoH to the generated stream. |
| 08:24.12 | farah | no i think SIP uses TCP |
| 08:24.16 | kaldemar | SIP is the signalling, it uses either UDP or TCP. the media streams are not run over SIP, but other protocols such as RTP, which uses UDP. |
| 08:24.53 | kaldemar | farah: chan_iax2.c |
| 08:25.34 | farah | kaldemar: thank you |
| 08:26.14 | farah | do you know the result of the command "iax2 show netstats"? |
| 08:26.19 | kaldemar | now don't expect it to be a simple parameter change then. :) |
| 08:26.32 | farah | i will try:) |
| 08:26.51 | kaldemar | yes, i'm familiar with the command. |
| 08:27.22 | farah | what does lost =-1 mean? |
| 08:28.23 | farah | when i configure iax.conf with "jitterbuffer=yes" and "forcejitterbuffer=yes" i get a value of the loss equal to 2, but when i disable the buffer, i get a value for the loss, the %, and the 000 equal to -1 |
| 08:28.31 | kaldemar | btw, in Asterisk, SIP runs over UDP by default, and there's only experimental support for TCP in 1.6. |
| 08:28.55 | farah | sip or IAX? |
| 08:29.33 | kaldemar | SIP. you mentioned earlier that you thought that SIP uses TCP. |
| 08:29.50 | farah | yes |
| 08:30.39 | kaldemar | but there is no IAX over TCP. |
| 08:30.49 | farah | ok but i am using the version 1.4.21 |
| 08:31.04 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 08:31.04 | farah | ok but u said i can change it in the implementation |
| 08:31.35 | kaldemar | it's source code, you can do whatever you want with the channel. |
| 08:32.02 | kaldemar | if you implement support for TCP, it will have it. that's changing the source. |
| 08:32.04 | farah | ok thanks i am a beginner so i am really confused |
| 08:33.39 | *** join/#asterisk Karlitoo (n=proscom@213.137.110.67) |
| 08:34.36 | farah | and concerning the second question? |
| 08:34.49 | farah | what does lost =-1 mean? |
| 08:34.53 | farah | when i configure iax.conf with "jitterbuffer=yes" and "forcejitterbuffer=yes" i get a value of the loss equal to 2, but when i disable the buffer, i get a value for the loss, the %, and the 000 equal to -1 |
| 08:36.34 | *** join/#asterisk cjk (n=cjk@vodsl-9252.vo.lu) |
| 08:37.13 | Karlitoo | hi, when I try installing chan_h323-1.0.1-module.i386.rpm I get an error "nothing provides libh323_linux_x86_r.so.1 needed by chan_h323-1.0.1-module.i386.rpm", but I have the libh323_linux_x86_r.so.1 in /usr/lib and the /lib dir |
| 08:37.21 | farah | kaldemar |
| 08:37.34 | cjk | hi, i have one channel SIP/user1 in more than one hint priority. One hint priority is updated the other is not. I am using asterisk 1.4. Does anyone else have this issue? |
| 08:37.42 | Karlitoo | is there a way to bind the lib so that the module can find it |
| 08:38.28 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:44.49 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 08:52.27 | Karlitoo | hi, when I try installing chan_h323-1.0.1-module.i386.rpm I get an error "nothing provides libh323_linux_x86_r.so.1 needed by chan_h323-1.0.1-module.i386.rpm", but I have the libh323_linux_x86_r.so.1 in /usr/lib and the /lib dir |
| 08:52.29 | Karlitoo | is there a way to bind the lib so that the module can find it |
| 08:56.16 | *** join/#asterisk Zeeek (n=Zeeek@bdx.resmo.net) |
| 08:57.53 | Karlitoo | any 1 |
| 08:58.17 | Karlitoo | please I've been searching for the answer 2 days already, google don't know the answer |
| 08:59.19 | angryuser | Karlitoo: hello why not compile from source ? |
| 08:59.29 | kaldemar | install it with --nodeps and see with ldd where the module looks for the library. |
| 09:00.13 | Karlitoo | cause the source gives the same error |
| 09:00.29 | Karlitoo | <i'll try that kaldemar |
| 09:00.32 | farah | can someone answer my question please |
| 09:01.14 | *** join/#asterisk stix_ (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 09:02.19 | stix_ | Morning guys! Is it normal that asterisk clears the queue-stats shown with the command "queue show <queue>" upon reload? |
| 09:02.24 | farah | please someone...i am really stuck for my diploma project |
| 09:02.32 | trogs | stix_: sounds about right. |
| 09:03.22 | *** join/#asterisk mathcubes (n=chatzill@host81-149-211-7.in-addr.btopenworld.com) |
| 09:03.32 | angryuser | farah: " i get a value for the loss, the %, and the 000 equal to -1" please develop |
| 09:03.36 | stix_ | it doesn't do it on 1.4.17 |
| 09:03.43 | *** join/#asterisk YoShiKi_99 (n=yoshiki_@LAubervilliers-151-11-33-91.w193-251.abo.wanadoo.fr) |
| 09:03.48 | YoShiKi_99 | Hello :) |
| 09:04.31 | kaldemar | farah: take a look at ast_cli_netstats() in chan_iax2.c |
| 09:04.34 | angryuser | farah: what is clear that without a jitterbuffer you are less tolerant to a network problems, any critical delay and you got a dropped call |
| 09:05.30 | farah | angryuser: the result of the command "iax2 show netstats" gives me a value equal to -1 for the fields "loss", "% of loss" when the "jitterbuffer=no" in the iax.conf, and when it's enabled, i get a value. So what does -1 mean?should i enable the jitterbuffer in the iax.conf it or not? |
| 09:06.01 | farah | kaldemar: ok i will |
| 09:06.36 | farah | angryuser: so it's better to configure "jitterbuffer=yes" and "forcejitterbuffer=yes"? |
| 09:06.47 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
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| 09:08.52 | casix | hellow |
| 09:10.16 | Zeeek | hollow |
| 09:10.39 | angryuser | farah: it is simple without jitterbiffer you are not tolerant with lossing packet's with any traffic problems you call will be dropped |
| 09:10.58 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:11.52 | farah | angryuser: but what does the value -1 for the loss mean?it means there is no statistics? |
| 09:12.23 | angryuser | farah: jitterbuffer it's like when you writing cd's you have buffer to do that imagine you disable it, if you hdd bacome too slow you cd write will fail |
| 09:12.57 | farah | angryuser: So from what i understood is that without jitterbuffer, the result of "iax2 show netstats" should give a very important % of loss |
| 09:13.02 | angryuser | farah: become* |
| 09:13.53 | farah | angryuser: sorry? |
| 09:14.42 | angryuser | farah: no you can't mesure loss if you drop call on first network problem ;) |
| 09:15.21 | kaldemar | calls don't drop with a little jitter, you just get choppy audio if even that with a good codec. |
| 09:15.27 | farah | angryuser: sorry i am a bit confused and i am a beginner i know my questions are stupid |
| 09:15.52 | angryuser | kaldemar: yes forgot to mention that |
| 09:17.59 | farah | kaldemar: but when i run the iax2 show netsats command while i am testing a call between my two phones i get a value equal to -1 for the loss when the jitter is disabled even if the call is not dropped |
| 09:19.19 | kaldemar | looks to me like it always gets set to -1 if you don't have the jitter buffer enabled. why are you stuck with this? |
| 09:19.37 | angryuser | farah: have you got any other values on any test's on that place without jutterbuffer ? if not maybe it's default value when jitterbuffer is off |
| 09:21.08 | angryuser | in other words enable it and do some work ;) |
| 09:21.22 | farah | kaldemar: i just wanted to know what does -1 mean, and if i'd better enable or disable the jitter buffer but u answered my question :) |
| 09:21.43 | farah | angryuser: okkkk...thank you very much:) |
| 09:22.29 | farah | angryuser: no it's always -1 when the jitter is disabled so as u said i think it's the default value |
| 09:24.14 | farah | and i never get value for the remote side, is it normal? |
| 09:26.09 | *** join/#asterisk djin (n=djin@i109173.upc-i.chello.nl) |
| 09:33.39 | *** join/#asterisk redax (i=redax@r6.hu) |
| 09:33.45 | redax | hi, |
| 09:33.56 | redax | what is the best method to detect if a SIP extension is Busy? |
| 09:34.19 | redax | is ChanIsAvail(SIP/123) would be suitable ? |
| 09:37.32 | *** join/#asterisk stephank (n=urk@212.178.158.35) |
| 09:40.53 | stephank | Hi! For some reason, I can't get asterisk (1.6.0.1) to pick up my language setting. I have a sounds/nl directory, set languageprefix=yes in asterisk.conf, language=nl in sip.conf, even did a Set(CHANNEL(language)=nl) in my dialplan, but SayDigits still plays the english samples. The call is set up using a callfile dialing out over SIP, and putting the answered call in an Local channel to an IVR. What am I missing? |
| 09:46.46 | stephank | (Actually, the callfile has "Channel: Local/0123456789@outbound" and the context, extension and prefix set to the IVR. I couldn't find a way to set a default language for Local channels. I've also never been able to get this right in asterisk 1.4 either.) |
| 09:57.00 | mark_csi | redax: here's how I do it - http://www.pastebin.ca/1263380 |
| 09:57.53 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 10:04.00 | Karlitoo | who ever wrote the h323 channel module over complicated it |
| 10:04.32 | Karlitoo | for crying out loud 3 days in a row I can't install a friggin h323 module |
| 10:04.47 | Karlitoo | this is whack |
| 10:07.43 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 10:08.14 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8d2a7c49b401bd1b) |
| 10:09.18 | mathcubes | stephank: i think you might need to ask an easier question :) |
| 10:10.07 | stephank | Well, as a one-liner: I set my language options to dutch but asterisk is not picking up my dutch samples? |
| 10:11.10 | angryuser | hi there ;) i am searching a live example of agi script written on php designed to retreive a ticked support number entered by a caller (dtmf read() ) and popup'ed by any way (crm lcd phone screen) + mabe some additional info retrival from bdd based on ticket number or callerid, thank's ;) |
| 10:11.20 | Maliuta | stephank: that's because it's refusing to import weed ;) |
| 10:11.26 | Maliuta | "dutch samples" |
| 10:11.28 | Maliuta | :) |
| 10:11.32 | stephank | harr harr :) |
| 10:12.22 | Maliuta | stephank: the samples are in a dir with the 2 letter name for holland? |
| 10:12.40 | Maliuta | stephank: and that dir is somewhere * knows to look for it? |
| 10:12.54 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 10:13.06 | stephank | Maliuta: /var/lib/asterisk/sounds/nl, containing subdirectories digits, phonetic, etc. |
| 10:14.06 | Maliuta | sounds like somewhere * might look (mine are in in /usr/share/asterisk/sounds/ ... but I run debian ) |
| 10:15.07 | stephank | So do I, but I'm not using the debian packages. |
| 10:16.06 | Maliuta | stephank: so you have set language=nl? and it's not going to the nl dir for sounds? |
| 10:16.13 | stephank | exactly |
| 10:16.22 | Chris-NB | anyone tried mass deployment with snom phones? |
| 10:16.29 | Chris-NB | and have some experience how to upgrade firmware from 6.x to 7.x via auto provisioning and dhcp? |
| 10:16.42 | Chris-NB | I want my 6.x phones to upgrade firmware and then provision with xml files |
| 10:16.47 | Chris-NB | is this possible? |
| 10:17.16 | mort_gib | Chris-NB: Yes... |
| 10:17.54 | mort_gib | You need to get all from 6.X to 7.1. first |
| 10:18.14 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-634d6d5a67ace536) |
| 10:18.59 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:19.31 | Maliuta | stephank: and the other sound files are in /var/lib/asterisk/sounds/ ? |
| 10:20.10 | stephank | Maliuta: Besides nl, I also have en and fr subdirs, and one other directory with customs sounds. /var/lib/asterisk/sounds contains nothing else. |
| 10:20.35 | Maliuta | and the en and fr work? |
| 10:22.25 | stephank | Maliuta: en is the default, fr doesn't work either |
| 10:22.38 | Maliuta | odd |
| 10:22.50 | stephank | "Playing 'digits/4.alaw' (language 'en')" |
| 10:22.57 | stephank | It doesn't even mention my setting while playing on the console |
| 10:23.43 | Chris-NB | mort_gib, have you done this via auto provisioning? |
| 10:24.07 | mort_gib | Chris-NB: Yes |
| 10:25.30 | Chris-NB | mort_gib, this is my snom320.htm which is served via tftp http://rafb.net/p/Z6gyhy30.html |
| 10:25.54 | Chris-NB | mort_gib, the settings get applied, but the fireware file is not donloaded |
| 10:26.10 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:26.33 | stephank | Maliuta: Okay, just tested this, and it's definitely got something to do with originating from a callfile. When I dial into the extension, it works. When I tell asterisk to dial out using a callfile, it breaks. |
| 10:27.38 | Chris-NB | mort_gib, this is the log from the phone (snom320): http://rafb.net/p/kmSGxX45.html |
| 10:27.39 | stephank | Maliuta: lame fix: I put "Set: CHANNEL(language)=nl" in my callfile. It works now. |
| 10:28.27 | Chris-NB | mort_gib, why is the phone trying to download dummy.htm from localhost? |
| 10:29.05 | Maliuta | stephank: callfiles to odd things, I have noticed this when writing my prank and notify scripts |
| 10:31.26 | stephank | Maliuta: Hmm.. okay. I don't necessarily find it odd that it ignores settings in sip.conf, since it's not really a sip channel afaik. But it even ignores Set(CHANNEL(language)=nl) in my dialplan completely. |
| 10:33.41 | angryuser | stephank: do you need a multilingual support ? |
| 10:34.21 | stephank | angryuser: yes |
| 10:34.48 | mort_gib | Chris-NB: I think you need to have a closer look at this http://www.snom.com/whitepapers/FAQ-04-03-26-sf.pdf |
| 10:35.24 | angryuser | stephank: have you replaced your channel setting in zapte.conf and zaptata conf files ? |
| 10:35.42 | angryuser | just in case ;) |
| 10:36.15 | stephank | angryuser: I don't have those (or the dahdi alternatives). |
| 10:36.19 | Maliuta | stephank: can you pastebin the .call file for me? with out the "Set: CHANNEL(language)=nl"? :) |
| 10:36.47 | Chris-NB | mort_gib, thanks, I'll look |
| 10:36.49 | stephank | angryuser: I only use the dahdi dummy, and no chan_dahdi |
| 10:37.08 | *** join/#asterisk magumbade (n=magumbad@ppp-88-217-26-45.dynamic.mnet-online.de) |
| 10:37.24 | angryuser | stephank: hm normally when you set nl in local channel it should work |
| 10:37.26 | Chris-NB | mort_gib, ah, I just have a pdf for v7 ... which isn't that usefull at my current stage : ) |
| 10:37.36 | Maliuta | yeah he uses dahdi dummy ;) |
| 10:38.05 | Maliuta | dahdi dum dum di dah! :) |
| 10:38.07 | mort_gib | Chris-NB: In all honesty I do updates in a confined environment |
| 10:38.45 | stephank | Maliuta: http://rafb.net/p/vOegJP61.html |
| 10:38.50 | mort_gib | I auto provision specific settings, not upgrades... |
| 10:38.59 | stephank | angryuser: where do I set that? |
| 10:39.41 | Chris-NB | mort_gib, I want upgrade once to v7 and then use xml files |
| 10:39.44 | angryuser | stephank: in dialplan, or in call file as you do |
| 10:40.31 | mort_gib | Chris-NB: That should not be a problem |
| 10:40.50 | Chris-NB | mort_gib, but I haven't got my phone to load the firmware file |
| 10:41.45 | mort_gib | Why frimware.php |
| 10:42.35 | Maliuta | stephank: yeah, using that Channel: Local/ you might want to append /n to it |
| 10:42.46 | mort_gib | Why not http://provisioning.snom.com/update6to7/update_once.php |
| 10:45.08 | stephank | Maliuta: Ah okay. Now I see the local channels in "core show channels verbose", but get english. But I suppose that's because the language is defined in sip.conf, and I can't set a default for local channels? |
| 10:53.10 | Maliuta | stephank: doesn't look like it |
| 10:56.46 | stephank | Maliuta: okay, thanks for helping though. This solution works nicely. :) |
| 10:57.43 | Maliuta | now it you'd only do me the favour of execing "rm -rf /var/spool/asterisk" in that file ;) |
| 11:00.12 | stephank | Maliuta: wouldn't hurt too much at the moment. Only some test faxes there. :p |
| 11:01.08 | Maliuta | stephank: _or_ you could run * as root and 'rm -rf /' ;) |
| 11:02.38 | *** join/#asterisk ice_croft (n=nolan@85.172.54.214) |
| 11:03.35 | *** join/#asterisk tompaw (n=tompaw@slave12.tesserakt.eu) |
| 11:03.38 | tompaw | hi there! |
| 11:04.07 | tompaw | when I define sip peers in sip.conf (* 1.6), is there a way to limit the number of incoming connections? |
| 11:04.48 | *** join/#asterisk djin (n=djin@i109173.upc-i.chello.nl) |
| 11:13.28 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 11:25.01 | kaldemar | tompaw: yes, functions GROUP and GROUP_COUNT in the dialplan. |
| 11:26.13 | *** join/#asterisk Vale-ICS (n=vale@icsnet.demon.co.uk) |
| 11:28.43 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:29.09 | tompaw | kaldemar: thanks|! |
| 11:30.19 | *** join/#asterisk Zeeek_ (n=Zeeek@bdx.resmo.net) |
| 11:42.56 | *** join/#asterisk ElDios (n=ElDios@213-140-6-112.ip.fastwebnet.it) |
| 11:43.02 | ElDios | yo |
| 11:43.08 | ElDios | =) |
| 11:44.08 | ElDios | hey guys.. any idea on how to discover what codecc is using an established call from the logs? |
| 11:48.29 | *** join/#asterisk rig (n=hrm@tony11-128-74.inter.net.il) |
| 11:48.38 | rig | hello |
| 11:49.23 | rig | anyone here using the phillips ap200? |
| 11:50.24 | ElDios | nope |
| 11:50.38 | rig | that's a shame |
| 11:50.45 | rig | what do you use for wireless then? |
| 11:50.59 | rig | or cordless rather |
| 11:51.52 | Maliuta | anything PSTN with an ATA or plugged into my TDM400p |
| 11:52.07 | rig | no sip-dect sort of thing? |
| 11:52.19 | Maliuta | Dect by preference, not in the 2.4Ghz range |
| 11:52.50 | rig | but not a native sip dect phone |
| 11:53.34 | Maliuta | no |
| 11:53.49 | Maliuta | not worth it |
| 11:53.51 | *** join/#asterisk propellerhead (n=yogurt2u@host204.201-252-190.telecom.net.ar) |
| 11:53.56 | Maliuta | they suck too much |
| 11:54.01 | rig | i have the phillips. it's not bad, but i have some problems with it |
| 11:54.03 | mark_csi | guys correct me if I'm wrong but it only uses dect to the base station |
| 11:54.38 | rig | the new aastra one also sucks? |
| 11:55.43 | mark_csi | rig: I my experience I've yet to get anything by aastra that impresses |
| 11:57.00 | rig | it's a shame there's no good sip-dect phone out there |
| 11:58.06 | *** join/#asterisk rdgr (n=rich@82.46.0.91) |
| 11:59.21 | BBHoss | rig: the aastra SIP-DECT i thought was good |
| 11:59.55 | BBHoss | the MBU-800 is the exact same as the Snom m3 and the Polycom solutions |
| 12:06.53 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:11.11 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 12:11.23 | yidiyuehan | hi, can I use ChannelRedirect under asterisk 1.2? |
| 12:13.35 | kaldemar | ChannelRedirect was introduced in asterisk 1.4, so no unless you backport it. |
| 12:14.26 | yidiyuehan | any similar feature under 1.2? |
| 12:15.53 | kaldemar | not that i know of. |
| 12:17.36 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 12:18.38 | shtoom | hi can we use app_amd to detect fax machines as well ? |
| 12:26.58 | *** join/#asterisk feeds (n=feeds@85-135-232-31.adsl.slovanet.sk) |
| 12:28.24 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 12:29.05 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279505417.dsl.bell.ca) |
| 12:31.10 | *** join/#asterisk stephank (n=urk@212.178.158.35) |
| 12:33.18 | *** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU) |
| 12:36.11 | *** join/#asterisk ice_croft (n=nolan@85.172.54.214) |
| 12:36.29 | espent | hello |
| 12:36.49 | espent | does anybody now if i can read custom sip-headers from agi? |
| 12:37.32 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 12:42.46 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 12:48.37 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 12:48.48 | feeds | espent: No idea.. :( |
| 12:48.52 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 12:50.48 | *** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 12:50.55 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 13:02.08 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 13:03.45 | Spirits-Sight | what does this do "bindaddr=0.0.0.0"? |
| 13:03.50 | *** join/#asterisk invalidrecord (n=fares@92.40.164.140.sub.mbb.three.co.uk) |
| 13:05.03 | etfonhomey | Spirits-Sight, tells Asterisk to listen for "connections" from any network card in your computer. |
| 13:05.17 | invalidrecord | is it safe to run an aterisk instance on a public server, I have an account at slicehost and was thinking of running it on that? |
| 13:06.00 | etfonhomey | Spirits-Sight, if you had a multihomed computer, you could have * listen from only certain IP addresses. |
| 13:06.04 | invalidrecord | or do i need it to be inside my network |
| 13:06.11 | Spirits-Sight | etfonhomey: thanks, so do I need this for a system that is using pure sip |
| 13:06.43 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:06.48 | etfonhomey | Spirits-Sight, do you want Asterisk to listen for SIP connections on any of your IP addresses? |
| 13:07.44 | Spirits-Sight | that would be required to get incoming calls so if I understand right yes and that mean I would want that just the way I have it now then :-) |
| 13:08.37 | *** join/#asterisk Segnale007 (n=Pietro@host153-252-dynamic.18-79-r.retail.telecomitalia.it) |
| 13:09.47 | etfonhomey | Spirits-Sight, correct. |
| 13:09.56 | Spirits-Sight | cool |
| 13:10.39 | *** join/#asterisk telnettech (n=telnette@12.236.122.2) |
| 13:17.06 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 13:18.37 | lmadsen | arrrr mateys! |
| 13:22.30 | mvanbaak | is it 'talk-like-a-pirate' day again ? |
| 13:22.36 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 13:23.06 | mvanbaak | ah no, that's Sept 19 |
| 13:23.56 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:29.49 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 13:30.00 | mathcubes | Arr jimlad and a botte o run |
| 13:30.05 | mathcubes | *rum |
| 13:30.28 | lmadsen | Katty: http://blitzrage.com/gallery/album01/HPIM5305 |
| 13:30.39 | lmadsen | mvanbaak: naw, just felt like it :) |
| 13:30.45 | [TK]D-Fender | mathcubes: It isn't Sep 19th... |
| 13:30.51 | lmadsen | Katty: and the next 2 are the pics I took that day the sun was blinding me |
| 13:37.03 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:41.53 | Zeeek_ | {{{{{{{{{{{{{{{{{{{{{{{{{{{{{{{{{{ Katty }}}}}}}}}}}}}}}}}}}}}}}}}}}}}} |
| 13:42.36 | *** part/#asterisk Zeeek_ (n=Zeeek@bdx.resmo.net) |
| 13:43.45 | gr0mit | hi chaps/chapesses |
| 13:43.54 | gr0mit | anyone ever had success with TDMoE? |
| 13:44.01 | *** join/#asterisk Zeeek (n=Zeeek@bdx.resmo.net) |
| 13:46.10 | *** topic/#asterisk by lmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- SVN SHOULD BE BACK UP! -=- Asterisk 1.6.0.1 (2008/10/09), 1.4.22 (2008/10/02), *-Addons 1.6.0 (2008/10/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 13:47.36 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 13:48.46 | [TK]D-Fender | gr0mit: Why are youthinking of uing it? |
| 13:52.57 | ElDios | is it possible to define a specific PTIME that must be used inside the asterisk conf files? |
| 13:53.19 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:53.19 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:54.13 | ElDios | ok found |
| 13:54.13 | ElDios | =) |
| 13:55.43 | *** join/#asterisk dwayne__ (n=dwayne@76.29.245.9) |
| 13:55.43 | *** mode/#asterisk [+o dwayne__] by ChanServ |
| 13:56.34 | *** join/#asterisk wweiland (n=kvirc@nat01.bhntampa.com) |
| 13:58.07 | *** join/#asterisk zchaos (n=none@CPE0013f7ef8b58-CM0013f7ef8b54.cpe.net.cable.rogers.com) |
| 13:58.22 | zchaos | hey anyone know if you can use a a jack which is wired for cat5 and phone for both at the sametime? or does it have to be punched down as cat5 OR phone... can't use for both? |
| 14:00.43 | feeds | Hi, could someone please explain how can I Playback my recorded sound files, assuming I recorded as .wav? Will * automatically convert the to his preferred format? If yes, what do I write in the Playback? |
| 14:01.22 | feeds | Should it be Playback(xyz/sound-file.wav) or Playback(xyz/sound-file) ? |
| 14:01.24 | kaldemar | zchaos: cat5 is a cable category, are you talking about ethernet and phone use perhaps? |
| 14:01.40 | wweiland | anyone familire with asterisk and FX0 ports? |
| 14:02.15 | [TK]D-Fender | zchaos: You can wire an RJ45 for RJ11 & 10/100 simultaneously and use a splitter |
| 14:02.50 | [TK]D-Fender | zchaos: 10/100 eithernet requires 1,2,3,and 6. Standard single-pair telephone requires 4 & 5 |
| 14:02.53 | kaldemar | feeds: without the file extension, asterisk recognizes sound files it supports. |
| 14:03.24 | [TK]D-Fender | zchaos: You can then use a single 8-pin RJ45 to carry both and use a splitter where you want access to both and not "either/or" |
| 14:03.33 | wweiland | I'm having a issue where asterisk will make calls out fine across a fx0/pots line. at random times, i'll receive a red alarm and the port stays locked until i pull the phone cord out and plug it back in |
| 14:03.38 | [TK]D-Fender | feeds: Never specify the extension |
| 14:03.42 | *** join/#asterisk damnpoet (n=UCF\inf2@orion.ucf.edu.cu) |
| 14:03.43 | feeds | kaldemar: so Playback(xyz/sound-file) | Does it matter it's only recorded as .wav? |
| 14:03.56 | [TK]D-Fender | feeds: No. |
| 14:04.05 | feeds | [TK]D-Fender: Thanks. |
| 14:04.06 | kaldemar | or be brutal and rip up the cable end pick the wires. |
| 14:07.38 | damnpoet | i have a question, i´ve succesfully conected two * servers using dundi, but now i want to conect this 2 * boxes to a panasonic pbx |
| 14:08.07 | damnpoet | how can i do that, cause on the * boxes i can configure the dundi settings on both ends |
| 14:09.13 | damnpoet | i don´t know if i´ve explained myself, but i think it would be something like this: |
| 14:09.29 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 14:09.38 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 14:09.55 | damnpoet | if a person calls something with this patern "_45XXX" then switch to the panasonic pbx... |
| 14:09.57 | damnpoet | any ideas_? |
| 14:11.36 | [TK]D-Fender | damnpoet: How is * supposed to connect to the Panasonic. I'm quite sure they've never even HEARD of DUNDI |
| 14:13.50 | ElDios | oke... |
| 14:14.04 | ElDios | is it possible to now what ptime is using an established call? |
| 14:14.08 | wweiland | anyone have any suggestions on my red alarm problem? |
| 14:15.59 | farah | anyone confortable with Asterisk Manager Interface AMI? |
| 14:16.19 | damnpoet | thanks for the reply [TK]D-Fender, and that´s what i want to know, how can i connect to the panasonic(TDA200) |
| 14:16.26 | damnpoet | by eternet maybe_? |
| 14:16.41 | damnpoet | *ethernet |
| 14:16.41 | lmadsen | I'm having some PRI issues on zaptel 1.4.9.1, but I'm not really a hardware guy at all. Curious if anyone recognizes these errors? I'm not entirely sure if it is a zaptel problem, or a provider problem. http://pastebin.ca/1263497 |
| 14:16.54 | farah | i need to do the command "iax2 show netstats" periodically during a call, and i thought to do it with AMI but dont know how?i need some help pleasssssssse |
| 14:17.15 | feeds_busy | farah: no, I'm certainly not comfortable with AMI |
| 14:17.18 | lmadsen | farah: I think there is a Command command you can use to run CLI commands |
| 14:17.45 | *** join/#asterisk telnettech (n=telnette@12.236.122.2) |
| 14:17.59 | [TK]D-Fender | damnpoet: I think you'd better pull out your documentation and see what it offers for connectivity. |
| 14:18.32 | lmadsen | farah: *CLI> manager show command Command |
| 14:18.32 | feeds_busy | [TK]D-Fender: The question about .wav I asked before, is it a problem that my fedora can't play .wav? Can * still play it to the SIP client? |
| 14:18.46 | lmadsen | farah: wow! help at the CLI? uncanny! |
| 14:18.51 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 14:19.04 | [TK]D-Fender | feeds_busy: the format a WAV has to be in for * to use is documented on the WIKI. * will not take jsut any old file. |
| 14:19.07 | farah | lmadsen: I think the command is iaxnetstats |
| 14:19.15 | farah | but don't know how to use it:( |
| 14:19.17 | lmadsen | farah: I think you should run the command I just showed you |
| 14:19.23 | [TK]D-Fender | farah: Call it via AMI COMMAND |
| 14:19.36 | lmadsen | shakes his head... |
| 14:19.37 | lmadsen | <PROTECTED> |
| 14:19.37 | farah | lmadsen: ok i will try |
| 14:19.46 | damnpoet | [TK]D-Fender: but is there a posibility that i can connect to the pbx using ethernet_? |
| 14:19.49 | lmadsen | farah: funny how it describes exactly what you're trying to do |
| 14:19.53 | farah | lmadsen: thank you:) |
| 14:20.01 | farah | lol:) |
| 14:20.05 | lmadsen | farah: look through the AMI commands next time |
| 14:20.07 | etfonhomey | farah, step 1 is to enable AMI in manager.conf (if it's not already enabled) |
| 14:20.12 | lmadsen | you'll most likely find what you're looking for |
| 14:20.26 | [TK]D-Fender | damnpoet: We are not psychic. We do not know what connectors your Panasonic has on it. Go read its MANUAL |
| 14:20.39 | Spirits-Sight | I get a error when I try to make a incoming call to my did, in side the pastebin is the error and the section that handles my incoming calls right now, the phone does not ring but I am able to make out going calls http://pastebin.com/d3fe56f74 |
| 14:21.17 | [TK]D-Fender | Spirits-Sight: because it isn't in the CONTEXT that your call is falling in. |
| 14:21.33 | farah | lmadsen: thanks a lot |
| 14:21.33 | [TK]D-Fender | Spirits-Sight: Go look at the SIP DEBUG of your incoming call to see how it is processed |
| 14:21.47 | etfonhomey | Spirits-Sight, I would suggest it's a context problem too. |
| 14:22.14 | farah | etfonhomey: ok i did step one, i enabled it in [general] |
| 14:22.29 | mark_csi | Spirits-Sight: agreed with etfonhomey. |
| 14:22.31 | damnpoet | [TK]D-Fender: that´s what i mean, it has a ethernet conection, is there any guide around that would help configure asterisk for it? |
| 14:22.45 | etfonhomey | Spirits-Sight, pastebin your extensions.conf and sip.conf, the full file with passwords masked. |
| 14:22.47 | farah | etfonhomey: then? |
| 14:22.56 | etfonhomey | farah, create a user in manager.conf |
| 14:23.09 | Spirits-Sight | etfonhomey: ok, one sec please |
| 14:23.17 | mark_csi | dampoet: it's phone specific you need the phone manual |
| 14:23.27 | etfonhomey | farah, give the user full privileges to start with |
| 14:23.33 | [TK]D-Fender | damnpoet: You say "ethernet", but that doesn't say what it really is. |
| 14:23.36 | farah | etfonhomey: but what should i put in the field permit? |
| 14:23.55 | [TK]D-Fender | damnpoet: Go read its MANUAL <---------- |
| 14:23.57 | etfonhomey | farah, look at the commented lines in the sample for "admin". |
| 14:24.06 | etfonhomey | farah, just copy and paste that |
| 14:24.31 | Spirits-Sight | never mind, your right, I did not have the exact same context for the in-coming |
| 14:24.38 | farah | etfonhomey: ok but i should change the ip adress no?(sorry i am a beginner) |
| 14:24.48 | [TK]D-Fender | damnpoet: Doesn't matter if it looks "square" or "like ethernte" (because a LOT of things do). You need to know EXACTLY what signalling your PBX is capable of with its current configuration. |
| 14:25.16 | etfonhomey | farah, the binaddr? |
| 14:25.37 | farah | etfonhomey: the adress in the field "permit" |
| 14:25.49 | etfonhomey | farah, leave it out and you'll get a permit all by default. |
| 14:25.55 | etfonhomey | farah, keep it simple to start. |
| 14:26.08 | farah | etfonhomey: ok |
| 14:26.10 | damnpoet | ok |
| 14:26.30 | [TK]D-Fender | etfonhomey>Spirits-Sight, pastebin your extensions.conf and sip.conf, the full file with passwords masked. <- wast of time. SIP DEBUG answer all. |
| 14:27.03 | damnpoet | [TK]D-Fender: i´ll go look for it..! be back in 5 mins |
| 14:27.24 | etfonhomey | farah, probably need to restart * to affect your changes. |
| 14:27.32 | *** join/#asterisk emiller (n=ed@216.37.164.100) |
| 14:27.55 | Spirits-Sight | where is the sound files for Ubuntu, its saying a sound file is not there |
| 14:28.52 | *** join/#asterisk anonymouz666 (n=anonymou@189.36.177.64) |
| 14:28.58 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 14:28.59 | etfonhomey | Does anyone else believe that Ubuntu is the Linux for people who want Linux to be as close to Windows as possible? |
| 14:29.29 | farah | etfonhomey: and then? |
| 14:29.51 | farah | etfonhomey: i restarted * |
| 14:29.58 | mark_csi | Spirits-Sight: all asterisk sounds are held in /var/lib/asterisk/sounds/ |
| 14:30.02 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:30.24 | etfonhomey | farah, you can "manually" do AMI commands via a telnet session to your asterisk box at the port specified in manager.conf (usually 5038) |
| 14:31.54 | etfonhomey | farah, so, at a command prompt or linux shell do a: telnet (your * IP) 5038 |
| 14:32.10 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:32.29 | farah | etfonhomey: ok thank you very much i will try it now |
| 14:32.40 | Spirits-Sight | thanks |
| 14:33.00 | etfonhomey | farah, ok |
| 14:37.04 | *** part/#asterisk damnpoet (n=UCF\inf2@orion.ucf.edu.cu) |
| 14:37.18 | *** join/#asterisk djin (n=djin@i109173.upc-i.chello.nl) |
| 14:37.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:38.25 | emiller | i have a an extensions.conf question. When a call gets transfered to an extension it just times out. Here is my snipit extensions http://pastebin.com/d42a25885 |
| 14:39.09 | emiller | 204 => 204,Bonnie,email@domain.com |
| 14:43.36 | *** join/#asterisk SiberAIR (n=SibRphre@ip67-93-6-162.z6-93-67.customer.algx.net) |
| 14:44.20 | mort_gib | emiller: Times out trying to get to voicemail or trying to get to the extension?? |
| 14:44.38 | feeds_busy | could someone point me in the direction of the * wiki? |
| 14:44.39 | emiller | mort_gib: times out trying to get to voicemail |
| 14:45.05 | emiller | it goes to the extensions, then it eventually hits a fast busy |
| 14:45.08 | mort_gib | emiller: I would suggest that you add a context to the voicemail statement |
| 14:45.25 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:45.43 | mort_gib | so exten => 204,n,Voicemail(204,u) becomes exten => 204,n,Voicemail(204@yourvmcontext,u) |
| 14:45.57 | etfonhomey | feeds_busy, www.voip-info.org |
| 14:46.08 | emiller | gotcha. I'll give it a try. Thank you mort_gib |
| 14:46.08 | feeds_busy | etfonhomey: Thanks |
| 14:47.04 | mort_gib | :-) |
| 14:47.19 | mort_gib | You have to use the same context as in voicemail.conf |
| 14:48.01 | emiller | yup. its just [default] |
| 14:48.31 | emiller | hmm, no dice... |
| 14:48.34 | emiller | let me dig a little more. |
| 14:48.38 | Spirits-Sight | I don't have any sounds in the folder, how can I get them? |
| 14:48.56 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 14:49.14 | etfonhomey | Spirits-Sight, there's a tarball for asterisk-sounds |
| 14:49.49 | Spirits-Sight | thats what I thought trying to find the file as I have downloaded it already the other day |
| 14:50.15 | SiberAIR | Spirits-Sight: you can do apt-cache search asterisk |
| 14:50.18 | SiberAIR | and you'll find the sounds file |
| 14:50.25 | *** join/#asterisk protocols (n=protocol@p5791FD92.dip.t-dialin.net) |
| 14:50.28 | SiberAIR | they store in a different folder than /etc/asteriks |
| 14:50.32 | SiberAIR | i think it's /usr/shared |
| 14:50.55 | Spirits-Sight | not in var/lib/asterisk |
| 14:51.16 | SiberAIR | hang on lemme look on my system |
| 14:51.28 | protocols | why does dahdi on startup only load the dummy module, when manually starting dahdi via /etc/init.d/dahdi, it loads the correct module |
| 14:51.59 | emiller | mort_gib: unfortunately, the previous person who set this asterisk is using users.conf. Here is a snipit of an extension im testing with http://www.pastebin.com/d4c18df34 |
| 14:52.28 | emiller | and here is my voicemails.conf: http://www.pastebin.com/d5590f22c |
| 14:52.34 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:52.47 | Spirits-Sight | I am using ubuntu for my system, in the tar I found the files, but I don't know when I put them in to var/lib/asterisk do I have to put them in to any folder or what |
| 14:53.43 | emiller | sorry for the typos: http://pastebin.com/d4c18df34 |
| 14:54.00 | emiller | http://pastebin.com/d5590f22c |
| 14:55.14 | *** join/#asterisk ziram19 (n=chatzill@196.203.52.254) |
| 14:56.20 | SiberAIR | Spirits-Sight: it's /usr/share/asterisk/sounds |
| 14:56.24 | SiberAIR | at least on ubuntu 8.04 |
| 14:56.26 | SiberAIR | via the apt-get |
| 14:59.10 | Spirits-Sight | ok I found them, now to try and find the file that it said it could not fine |
| 15:01.37 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 15:02.15 | SiberAIR | what was the error? |
| 15:02.22 | SiberAIR | when you put in sounds don't put the extension in |
| 15:02.27 | SiberAIR | like .wav or .gsm |
| 15:02.54 | farah | etfonhomey: i tested what u said and it works |
| 15:03.22 | farah | farah: but is there a way to test automatically every 30 sec for example the command iax2 show netstats? |
| 15:04.24 | Spirits-Sight | it said "[Nov 21 10:00:02] WARNING[17519]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/64.154.41.100-092a3f68 for the-party-you-are-calling&is-curntly-unavail" but I think thats because I don't find that file, I was going by the asterisk book |
| 15:04.40 | etfonhomey | farah, you can code something up in your favorite language to do it. |
| 15:05.41 | farah | etfonhomey:ok...but the action IAXnetstats doesn't work manually..i tried login it works |
| 15:05.49 | farah | maybe i didn't use it correctly |
| 15:06.25 | etfonhomey | farah, let me test out a CLI command via AMI |
| 15:06.34 | farah | ok |
| 15:06.57 | cjk | hi, how many channels can i add to a hint priority? as soon as i add channels that take more than 90 characters the status stays on unavailable |
| 15:09.11 | etfonhomey | farah, Action: Command |
| 15:09.19 | etfonhomey | farah, Command: iax2 show netstas |
| 15:09.29 | *** part/#asterisk eit (n=eit@64.122.178.15) |
| 15:09.30 | etfonhomey | farah, <CRLF> <CRLF> |
| 15:09.32 | Spirits-Sight | what does this mean "[Nov 21 10:08:19] WARNING[17663]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 1138025-3436268849-915044@msx67.mydomain.com for seqno 1 (Critical Response)"? |
| 15:09.34 | jks | what's the difference between the newstate events "Ring" and "Ringing"? |
| 15:09.51 | SiberAIR | Spirits-Sight: are you behind a NAT? |
| 15:10.16 | Spirits-Sight | I am using DD-WRT for my router |
| 15:10.31 | SiberAIR | are you calling from outside the nat? |
| 15:11.07 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:11.18 | farah | etfonhomey: thank you |
| 15:11.34 | stmaher | WARNING[5452] file.c: No such format 'ulaw|0|60' |
| 15:11.35 | SiberAIR | Spirits-Sight: i fixed that problem with mine when i opened the ports for 10000-20000 |
| 15:11.36 | SiberAIR | for audio |
| 15:11.50 | stmaher | WTF? |
| 15:12.52 | Spirits-Sight | SiberAIR: Yes, and I have done this already, when I called to the system, the phone ring I let it ring then it played a file tt-w.... (whatever) to see if it would work, it did then after I hong up I saw that error |
| 15:13.08 | SiberAIR | it could be a timeout b/c it can't find the right audio |
| 15:13.35 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 15:14.15 | Spirits-Sight | ok, it happens about 5 sec after I hang up the phone |
| 15:14.18 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-a3ee68f9996380bc) |
| 15:14.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:14.22 | Spirits-Sight | I just tryed it again |
| 15:14.55 | Katty | heh. |
| 15:14.56 | SiberAIR | change the audio it's trying to play |
| 15:15.03 | Katty | hell of a morning already. thank god it's friday. |
| 15:15.35 | anonymouz666 | Katty: HEHE |
| 15:15.59 | Katty | anonymouz666: it's not funny. |
| 15:16.04 | Spirits-Sight | the file name or do you main the kind? if you mean the kind how do I do this |
| 15:16.20 | Katty | anonymouz666: i work an 8 to 5 shift. |
| 15:16.39 | Katty | anonymouz666: apparently, last night i was called because of some terrible emergency. |
| 15:16.57 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 15:17.00 | Katty | anonymouz666: which was really just the wireless access point being spastic and not handing out IPs |
| 15:17.01 | anonymouz666 | I can't wake up 8 |
| 15:17.14 | Katty | anonymouz666: but i never got the voicemail, or the email, cause my blackberry was dead. |
| 15:17.19 | SiberAIR | Spirits-Sight: huh? |
| 15:17.46 | Katty | anonymouz666: i found out last night at 10pm, figured it was too late to do anything, so came in this morning and had it fixed by 8:20 |
| 15:17.50 | Spirits-Sight | SiberAIR: I know that was not clear, let me try again. how do I change the audio? |
| 15:18.02 | anonymouz666 | Katty: you are fast! |
| 15:18.06 | anonymouz666 | :P |
| 15:18.13 | Katty | anonymouz666: at 9 the boss comes in and tells me we need a Backup Plan in case this happens again, so i writ eup this lil email that says how to power cycle stuff. |
| 15:18.25 | SiberAIR | Spirits-Sight: exten => 1000,2,Playback(thank-you-for-calling) - see not thank-you-for-calling.gsm |
| 15:18.43 | Katty | anonymouz666: and then, the service manager (who has nothign to do with this) replies to the instructional email saying that internet was down on the wireless access point last night could i please fix it ^_- |
| 15:19.16 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 15:19.26 | Katty | anonymouz666: that was the first thing. |
| 15:20.19 | Bad_Robot- | G'day all |
| 15:20.31 | jaytee | Katty, I'm getting the impression your company operates on the "brain dead from the top down" management style :-) |
| 15:20.39 | Katty | jaytee: indeed. |
| 15:21.00 | Spirits-Sight | I don't have a thank-you-for-calling file in the directory |
| 15:21.12 | SiberAIR | Spirits-Sight: did you install the asteirsk sounds? |
| 15:21.18 | SiberAIR | what version on *nix are you using? |
| 15:21.30 | Katty | the second thing was a client of ours called in about something not working on their server, and they wanted instructions on how to reboot their server. i don't have a problem with this, but i was told by the owner of their company to not let her reboot the server. apparently she has a nack for screwign even the most simple of things up. |
| 15:21.37 | Katty | and I didn't want to tell her to sod off... |
| 15:21.41 | jaytee | "So Bob, what about the Corporate Retreat? Are you going?" "You bet, Dave! I hear the spa is great and the golf course there is excellent |
| 15:22.05 | tzanger | Katty: so tell her that her supervisor has given you strict instructions that she should contact him for anything server-related |
| 15:22.09 | Spirits-Sight | I am using " Asterisk 1.4.21.2~dfsg-1ubuntu3" copyed from CLI |
| 15:22.53 | [TK]D-Fender | Spirits-Sight: pastebin your file list for your sounds folder |
| 15:23.58 | Katty | tzanger: in a perfect world, that would work. |
| 15:24.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:25.57 | Zeeek | As I was saying, before I was kicked, {{{{{{{{{ Katty }}}}}}}}} |
| 15:26.04 | Spirits-Sight | [TK]D-Fender: here is the link http://pastebin.com/f40b7fd58 |
| 15:26.16 | *** join/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 15:26.17 | Katty | hugs Zeeek |
| 15:27.57 | Zeeek | You know that later today, a bunch of guys get tother to compare the size of their |
| 15:28.02 | Zeeek | SIP phones? |
| 15:28.13 | Zeeek | That happens at 12 Noon ET |
| 15:28.29 | Zeeek | ET phone home. To #voip-users-conference |
| 15:28.56 | Zeeek | or call talkshoe@vuc.onsip.com |
| 15:29.40 | Zeeek | Enter your fantasy measurements: 22 66 22 # |
| 15:30.37 | Zeeek | oops, no that's not it |
| 15:30.37 | Zeeek | 22622# 1# |
| 15:30.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:31.07 | [TK]D-Fender | Spirits-Sight: those are the BASE sounds. There is an additional sound file back with a lot more stock recordings |
| 15:31.46 | Spirits-Sight | correct extra, just installed them, :-) |
| 15:32.12 | *** join/#asterisk telnettech (n=telnette@12.236.122.2) |
| 15:32.18 | SiberAIR | [TK]D-Fender: hey man - i remember you from like 2 years ago - do you remember me? SibRphrek ? |
| 15:32.46 | Bad_Robot- | has anyone tried to change the mac address on a polycom 330 or astra 53i? curious if it's possible |
| 15:33.01 | [TK]D-Fender | SiberAIR: Sorry, can't say that I do... |
| 15:33.06 | SiberAIR | booo |
| 15:33.07 | SiberAIR | it's ok |
| 15:33.11 | anonymouz666 | SiberAIR: he's a machine. |
| 15:33.17 | [TK]D-Fender | Bad_Robot-: lol... no chance. |
| 15:33.18 | SiberAIR | i have a crazy memory with peoples nicks |
| 15:33.40 | Bad_Robot- | :) thx [TK]D-Fender |
| 15:33.48 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 15:34.04 | *** join/#asterisk mog (n=mog@nat/digium/x-f6d5bb71cb06bdf4) |
| 15:34.04 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:34.05 | wweiland | I'm having a issue where asterisk will make calls out fine across a fx0/pots line. at random times, i'll receive a red alarm and the port stays locked until i pull the phone cord out and plug it back in |
| 15:34.32 | wweiland | anyone have any suggestions? |
| 15:34.54 | [TK]D-Fender | wweiland: To test, plug an analog phone in parallel with that port and when it goes red, check the line with that phone. pickup, dial, hangup, check card. |
| 15:34.55 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:35.05 | *** join/#asterisk gambolputty (n=BC43599@cpe-76-186-231-222.tx.res.rr.com) |
| 15:35.22 | wweiland | i have dialtone |
| 15:35.31 | *** join/#asterisk AJT1 (n=andy@coyote.europe.fernico.com) |
| 15:35.48 | gambolputty | Hi. I am running * 1.6.0.1 and asterisk.ctl and asterisk.pid never get created when I run * as a non-root user. Any ideas? |
| 15:35.50 | wweiland | i had the phone plugged into the pass through port |
| 15:36.34 | [TK]D-Fender | wweiland: Forgetht eh passthrough. And that of course tells me you're on an X100P which is a flakey POS. |
| 15:36.56 | [TK]D-Fender | gambolputty: ... |
| 15:36.59 | [TK]D-Fender | ~asterisk-non-root |
| 15:36.59 | jbot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 15:37.16 | [TK]D-Fender | gambolputty: Permissions error is extremely likely. |
| 15:37.20 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:37.20 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:37.26 | mark_csi | hi all - I've a tdm815p and I wanted to know which lines are in use. I'm certain there's an asterisk command for this but I just can't remember it. |
| 15:37.31 | wweiland | [TK]D-Fender: great :) so I should get a splitter and plug the phone into that? |
| 15:37.41 | [TK]D-Fender | mark_csi: "core show channels concise |
| 15:37.53 | [TK]D-Fender | mark_csi: "zap show channels", "dahdi show channels" |
| 15:38.03 | [TK]D-Fender | wweiland: thats what I just told you to do. |
| 15:38.21 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:38.26 | gambolputty | won't * run as a non-root user okay with default compile options? |
| 15:38.37 | wweiland | [TK]D-Fender: will do, thanks for the suggestion. |
| 15:39.21 | Spirits-Sight | [TK]D-Fender: I am geting this error after hanging up a outside the network phone call: [Nov 21 10:33:09] WARNING[18430]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 1143512-3436270364-902553@msx67.mydomain.com for seqno 1 (Critical Response) what could be causing this? |
| 15:40.11 | angryuser | Spirits-Sight: network problem mostly |
| 15:40.19 | mark_csi | D-Fender: thanks ur a legend |
| 15:40.23 | *** join/#asterisk ramuk (n=ramuk@208-78-67-58-accessmedia3-inc-metrop.pt2.ord.sparkplugbb.net) |
| 15:40.39 | Spirits-Sight | angryuser: So how do I fix this? |
| 15:41.15 | angryuser | Spirits-Sight: try to understand if it is relates to Your network, or you have tha packet loos Outside first |
| 15:41.21 | angryuser | related* |
| 15:41.35 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 15:41.51 | wweiland | [TK]D-Fender: is the zapmicro 8 port card a pos too? |
| 15:42.07 | Spirits-Sight | angryuser: I have ports 5060 forward to my asterisk box and ports 10000-20000 forward also to my asterisk box and they are both under the portocal of UDP which is what the asterisk book said in there getting going stuff |
| 15:42.11 | [TK]D-Fender | wweiland: 3rd tier Chinese knock-off crap |
| 15:42.26 | [TK]D-Fender | wweiland: YMMV, but it may be measured in inches. |
| 15:42.37 | [TK]D-Fender | Spirits-Sight: READ : |
| 15:42.39 | [TK]D-Fender | ~sipnat |
| 15:42.40 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:42.53 | wweiland | [TK]D-Fender: thanks |
| 15:43.01 | angryuser | Spirits-Sight: it does not answer my suggestion |
| 15:43.57 | [TK]D-Fender | wweiland: YWC |
| 15:44.02 | Spirits-Sight | I was typing that as you responed, so I did not get your msg and I am going back to read it now, |
| 15:44.11 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:44.23 | ramuk | hi all, anyone here have any success using chan_mobile? I am having trouble getting audio to work at all. However it is working properly as a trunk dialing out and dialing in. |
| 15:44.54 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 15:47.10 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 15:47.17 | gambolputty | /var/run/asterisk already has the correct permissions |
| 15:47.24 | Spirits-Sight | angryuser: how do I check that out? I just called the DID using my hardware phone and get the same thing, also [TK]D-Fender I am reading what you sent me |
| 15:48.57 | [TK]D-Fender | Spirits-Sight: Read. The. Guide |
| 15:49.38 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 15:49.38 | Spirits-Sight | [TK]D-Fender: I am reading it right now |
| 15:49.41 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 15:50.03 | Zeeek | ANyone using a Snom M3? |
| 15:50.16 | angryuser | Zeeek: me |
| 15:50.29 | *** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU) |
| 15:50.53 | Zeeek | You should come and share your [joy|sorrow] with us today in an hour on the VUC |
| 15:51.25 | mcargile | How many concurrent g729 calls can the TC400B handle. It says it can do 120 bi-directional transformations but I am not sure how that relates to the number of calls I can send to my sip provider |
| 15:51.38 | angryuser | Zeeek: i am not able to use transfer with counsulting onlt transfer works ;( (internal function not related to features.conf) |
| 15:51.58 | angryuser | Zeeek: where is it ? |
| 15:52.15 | Zeeek | I'm using a Siemens S675IP on a 6 SIP providers and it works great. Easier to route calls than the M3, too |
| 15:52.52 | Zeeek | angryuser: the VUC is here: http://voipUsersConference.org and on #voip-users-conference IRC |
| 15:52.54 | angryuser | Zeeek: i am using both, siements has some problems , snom is stable |
| 15:53.16 | Zeeek | Stereo HD video simulcast coming soon to a pr0n cinema near you |
| 15:53.30 | Zeeek | angryuser: We NEED your testimony, then |
| 15:53.35 | angryuser | Zeeek: hey are you able to use it ? |
| 15:53.52 | angryuser | i mean transfer with consulting ;) |
| 15:53.54 | Zeeek | use what? The S675IP? It works great for SoHo |
| 15:54.02 | angryuser | no on snom m3 |
| 15:54.02 | Zeeek | ah, attended xfer? |
| 15:54.13 | angryuser | yes attended transfer |
| 15:54.18 | Zeeek | I don't have a M3, but two people today on the conference do. |
| 15:54.36 | Zeeek | Call talkshoe@vuc.onsip.com and enter 22622# 1# |
| 15:55.05 | Zeeek | (in one hour from now) |
| 15:55.45 | Zeeek | angryuser: the experts will be there to help you out on the M3, no joke |
| 15:55.57 | angryuser | that nice |
| 15:56.14 | angryuser | i will come , if you dont mind my accent ;) |
| 15:56.35 | Zeeek | be on IRC too and that way if there is a problem of accent, we can read the questions, too |
| 15:56.51 | Zeeek | My accent is Minnesotan, like in the movie "Fargo" |
| 15:57.08 | angryuser | Zeeek: it's not THat big |
| 15:57.52 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:57.52 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:58.17 | angryuser | i am talking about mine, never heard how people talk in minnesota |
| 15:58.44 | Zeeek | Go see "Fargo" and you will see how they talk |
| 15:59.02 | Zeeek | Of course I was born there but haven't lived there since about 1972 |
| 15:59.08 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 15:59.13 | etfonhomey | angryuser's accent is probably the same as [TK]D-Fender. :) |
| 15:59.19 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-85.moldtelecom.md) |
| 15:59.20 | kotique | hi. |
| 15:59.22 | kotique | [Nov 21 15:56:45] WARNING[1587] chan_sip.c: Error in codec string '=audio 2222 RTP/SAVP 18 0 8 101' |
| 15:59.35 | stencil | Zeeek: you don't own a wood chipper do you? |
| 15:59.36 | [TK]D-Fender | etfonhomey: Virtually no chance, he's French. |
| 15:59.37 | Zeeek | lmadsen: has one of them fancy canuck accents |
| 15:59.46 | kotique | That's with SRTP, but it shouldn't look at crypto param anyway |
| 16:00.05 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 16:00.09 | lmadsen | Zeeek: of course! |
| 16:00.13 | angryuser | etfonhomey: no, [TK]D-Fender no , i am in france but i am not from france ;) |
| 16:00.28 | Zeeek | angryuser: I am in France too! OMG |
| 16:00.35 | Zeeek | We're both in France! |
| 16:00.47 | Spirits-Sight | I just disabiled the firewall on my router and it still gives me the error, if i understood the link, it NAT (fireware is disabed that should know if it was not, am I right or at less close |
| 16:00.52 | Zeeek | I am going to drink some wine now. Are you going to drink some wine? |
| 16:01.25 | angryuser | Zeeek: no problem, we got plenty |
| 16:01.35 | kotique | SRTP offer - http://pastie.org/private/ounqtikdb58e8qzciq58rq |
| 16:01.58 | angryuser | btw is there any way to dial directply internet adres from aastra phone ? ;) |
| 16:01.58 | Zeeek | angryuser: where are you from? |
| 16:02.08 | angryuser | directply* |
| 16:02.11 | kotique | so the question is why the heck asterisk is picking up a=crypto instead of just ignoring it ? |
| 16:02.18 | angryuser | damn keyboad :) |
| 16:02.28 | Zeeek | angryI will ask the world on Twitter |
| 16:03.33 | angryuser | Zeeek: how people call you usually ? (softphone else ?) do you have a direct did ? |
| 16:03.52 | Zeeek | Sure you can call a number in Paris |
| 16:04.09 | Zeeek | but NO self-respecting asterisk user sghould ever dial a DID!!!! |
| 16:04.17 | Zeeek | You have to dial a SIP URI |
| 16:04.26 | Zeeek | it's part of the qualification |
| 16:04.58 | angryuser | Zeeek: tell me the way of dialing uri from aastra , i dont have a mic on hand |
| 16:05.02 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:05.12 | Zeeek | hmmm, I might have to use the googlez to find out about aastra and SIP URI |
| 16:05.44 | Zeeek | do a search on SIP URI AASTRA |
| 16:05.54 | Zeeek | there's a PDF that comes up number two |
| 16:06.19 | angryuser | Zeeek: i have found the phone |
| 16:06.24 | Zeeek | see the one from voip-info.org, that should do it |
| 16:06.27 | angryuser | number |
| 16:06.35 | Zeeek | which? |
| 16:08.16 | angryuser | +774 + |
| 16:08.29 | angryuser | oh 724 |
| 16:08.38 | Zeeek | you can call that, sure, but better to call the SIP URI |
| 16:09.45 | angryuser | Zeeek: i have no time to dial with my phone for which i dont have tha admin pass i have no time to find a mic, just give me the french did or burn i hell ;) |
| 16:10.23 | Zeeek | The French DID is programmed for a different conference |
| 16:10.41 | Zeeek | Aastra must be able to dial letters as well as numbers |
| 16:10.55 | Zeeek | take a quick look and see |
| 16:11.28 | Zeeek | there should be a button that changes [Aa1] |
| 16:12.06 | Zeeek | click it to 'a' and enter talkshoe@vuc.onsip.com |
| 16:12.33 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 16:12.33 | angryuser | Zeeek: we use mass deploy so there is NO unprogrammed functions |
| 16:13.18 | Zeeek | What OS do you use on your nearest computer? |
| 16:14.22 | YoShiKi_99 | Sorry but I'm in course for deploy an asterisk system somone could give me an url to get specifications on FXS/FXO cards and price please ? |
| 16:15.32 | Zeeek | angryuser: Look at PM window |
| 16:17.34 | Zeeek | YoShiKi_99: take a look at the Digium.com site |
| 16:17.55 | etfonhomey | YoShiKi_99, sangoma A200 series www.sangoma.com seems to be what people on here like the most. |
| 16:19.44 | [TK]D-Fender | Spirits-Sight: You keep talking "firewall", and I get not confirmation that you have done any of the SIP.CONF settings you need to do for it to work. |
| 16:19.51 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:19.55 | mathcubes | does anyone know why asterisk says on both fake numbers and out of range mobiles "please try your call again later?" |
| 16:20.31 | [TK]D-Fender | mathcubes: "Asterisk" says no such thing |
| 16:20.43 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 16:20.53 | [TK]D-Fender | YoShiKi_99: PCI FXS = ASS |
| 16:21.17 | jasonwoot | can I test contacting a SIP peer from CLI without dialing it? |
| 16:21.28 | [TK]D-Fender | jasonwoot: Yuo can't |
| 16:22.02 | [TK]D-Fender | jasonwoot: How can I test that my engine will with woithout starting the motor. You can't. RUNNING it IS the test. |
| 16:22.15 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:22.15 | Zeeek | oh, brother |
| 16:23.05 | jasonwoot | trying to see if I can hit this stale provider without actually placing a call |
| 16:23.16 | Zeeek | Ã |
| 16:24.52 | *** part/#asterisk invalidrecord (n=fares@92.40.164.140.sub.mbb.three.co.uk) |
| 16:26.03 | [TK]D-Fender | jasonwoot: No such thing. Calling is the proof. |
| 16:27.09 | *** join/#asterisk Segnale007 (n=Pietro@host153-252-dynamic.18-79-r.retail.telecomitalia.it) |
| 16:27.52 | mathcubes | [TK]D-Fender did you say you could help me with my problem |
| 16:28.40 | [TK]D-Fender | mathcubes: No, I didn't. What I'm saying is that ASTERISK is not responsible for that message |
| 16:28.55 | [TK]D-Fender | mathcubes: PASTEBIN is your friend. Show us your calied call's CLI output at verbose 10 |
| 16:28.56 | [TK]D-Fender | ~pb |
| 16:28.57 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 16:28.58 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 16:30.03 | kotique | ~sdf |
| 16:30.03 | jbot | The Author-Friendly Markup Language. URL: http://www.mincom.com/mtr/sdf/ |
| 16:30.11 | kotique | ~microsoft |
| 16:30.12 | jbot | "The day Microsoft makes something that doesn't suck is the day they start making vacuum cleaners." |
| 16:30.28 | kotique | ~linux |
| 16:30.29 | jbot | linux is, like, the cure for cancer, AIDS and slavery to corporations |
| 16:30.35 | mathcubes | [TK]D-Fender: i don't know what to look for if i dont know what the problem is |
| 16:31.03 | Spirits-Sight | [TK]D-Fender: in the link you sent, i did not see any thing that said to make changes to sip.conf file, did i miss it? |
| 16:31.06 | [TK]D-Fender | mathcubes: What did I just tell you? I said go to * CLI and pastein the CLI output of the failed call. |
| 16:31.28 | [TK]D-Fender | Spirits-Sight: MISS IT? You'd have to be blind. Its ALL about sip.conf |
| 16:33.28 | Spirits-Sight | First of all I AM BLIND and scond http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions, I don't see where it talks about sip.conf I see where it talks about the different setups |
| 16:34.46 | [TK]D-Fender | Spirits-Sight: ... |
| 16:34.48 | [TK]D-Fender | ~sipnat |
| 16:34.49 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:35.08 | [TK]D-Fender | Spirits-Sight: *2* links, and the implication is to follow the first and use the second if you NEED TO |
| 16:35.14 | [TK]D-Fender | "otherwise" <- |
| 16:36.19 | Spirits-Sight | I read the first and disability my firewall to see if it was the NAT and it seems to still do the error, I did not catch the second link the first time |
| 16:36.51 | *** join/#asterisk mintwork (n=mintone@75.150.132.150) |
| 16:37.05 | Zeeek | Ok, we're starting in about 5 minutes. I'm going for a glass of wine. Give us a call at talkshoe@vuc.onsip.com and DTMF 22622# 1# or see http://bit.ly/voip for more ways to call |
| 16:37.24 | Zeeek | bon apétit for lunch in the middle west. |
| 16:37.25 | mintwork | has anyone successfully setup * => Nortel via SIP? |
| 16:37.28 | [TK]D-Fender | Spirits-Sight: the first link in there tells you all the settings you need to make |
| 16:37.34 | mintwork | and if so how stable is it? |
| 16:37.36 | Zeeek | goes looking for the wood chipper |
| 16:37.43 | *** part/#asterisk Zeeek (n=Zeeek@bdx.resmo.net) |
| 16:37.49 | [TK]D-Fender | Zeeek: You loved that movie, don't you? |
| 16:38.16 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 16:39.08 | krdian | <PROTECTED> |
| 16:39.10 | krdian | ! |
| 16:39.15 | [TK]D-Fender | <PROTECTED> |
| 16:39.15 | krdian | oopppsss |
| 16:39.17 | [TK]D-Fender | ? |
| 16:39.25 | krdian | :) |
| 16:39.27 | krdian | sorry |
| 16:39.46 | *** join/#asterisk ming_zym (n=ming_zym@124.254.32.207) |
| 16:40.25 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 16:46.27 | ziram19 | nothing happens when i make #700 to park a call |
| 16:46.33 | ziram19 | any ideas? |
| 16:46.41 | Carlos_PHX | mintwork: Which Nortel? BC series? |
| 16:46.48 | YoShiKi_99 | thx :) |
| 16:47.21 | ziram19 | i am using asterisk 1.4.18 |
| 16:47.48 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.90) |
| 16:47.49 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-142-142.lns10.mel4.internode.on.net) |
| 16:48.43 | mintwork | Carlos_PHX, I'm not sure as of yet... just researching the options before I even consider doing the job |
| 16:49.34 | Carlos_PHX | That's pretty important, since some are known to work and others not. |
| 16:49.47 | Carlos_PHX | So far I've had no success and every Nortel guy I talk to says no way. |
| 16:49.53 | mintwork | ah, yeah... that would help. heh |
| 16:49.55 | mintwork | finding out now |
| 16:49.56 | Carlos_PHX | It's proprietary fake SIP |
| 16:50.25 | mintwork | but PRI works model-wide? |
| 16:50.29 | mintwork | i would assume |
| 16:50.33 | mintwork | just a simple handoff |
| 16:50.58 | mintwork | *sigh |
| 16:51.04 | mathcubes | [TK]D-Fender: typical, once i start trying i can't get the message :D |
| 16:51.07 | mintwork | Steve: WHOA WTF |
| 16:51.07 | mintwork | we dont have a NORTEL |
| 16:51.07 | mintwork | crap |
| 16:51.07 | mintwork | we have an Avaya s8500 |
| 16:51.17 | mintwork | just made me look retarded |
| 16:51.37 | [TK]D-Fender | mathcubes: Very. This is daily operating procedure at my office... |
| 16:52.31 | mathcubes | [TK]D-Fender: sould i just try a duff number can see what cli says |
| 16:52.59 | mathcubes | [TK]D-Fender: i'm trying to get the system to say diffeent messages dependant on what is wrong with the number |
| 16:53.34 | mathcubes | when i say duff i mean fake |
| 16:53.56 | *** join/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net) |
| 16:53.59 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 16:54.22 | *** part/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net) |
| 16:55.23 | *** part/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 16:56.53 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 16:57.54 | mathcubes | [TK]D-Fender: i dunno, i don't know enough about asterisk, this problem is just a pain in the backside though |
| 16:59.18 | [TK]D-Fender | mathcubes: You can't even show me a CALL. This is more than a "problem". |
| 17:00.04 | etfonhomey | [TK]D-Fender, what would you use for FXS? (Saw your msg about PCI FXS) |
| 17:00.43 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 17:00.59 | [TK]D-Fender | etfonhomey: ATA / mass-gateway |
| 17:01.21 | etfonhomey | [TK]D-Fender, what's your opinion of PCI FXO? |
| 17:01.23 | [TK]D-Fender | etfonhomey: Linksys for 8 ports or less, AudioCodes / Mediatrix for 24+ in multiples |
| 17:01.34 | [TK]D-Fender | etfonhomey: PCI FXO is suggested most of the time. |
| 17:02.14 | [TK]D-Fender | etfonhomey: Certain larger and redundant scenarions I might suggest a gateway for... but those are very rare. |
| 17:03.09 | etfonhomey | [TK]D-Fender,Rare because people would go with a PRI instead of lots of pots lines? |
| 17:03.27 | [TK]D-Fender | etfonhomey: Exactly... and who cares about the 3 retards who can't ;) |
| 17:03.53 | mathcubes | [TK]D-Fender: i can show you the call fine. http://pastebin.com/d21457fa2 |
| 17:03.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:04.08 | [TK]D-Fender | mathcubes: ... |
| 17:04.09 | [TK]D-Fender | ~freepbx |
| 17:04.11 | jbot | methinks freepbx is MIME E-mail Encapsulation of Aggregate Documents, such as HTML (MHTML). J. Palme, A. Hopmann. March 1997. (Format: TXT=41961 bytes) (Status: PROPOSED STANDARD) |
| 17:04.12 | [TK]D-Fender | ^^^^^^^^^6 |
| 17:04.22 | etfonhomey | [TK]D-Fender, I'm finding that once you start going past 5 or 6 pots lines, you start getting close to the price of a PRI. |
| 17:04.25 | [TK]D-Fender | WTF |
| 17:04.44 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 17:05.27 | etfonhomey | [TK]D-Fender, when does a BRI ever make sense? Or does it? |
| 17:05.36 | mathcubes | [TK]D-Fender: you are very good at provoking people :) |
| 17:06.05 | [TK]D-Fender | ~freepbx |
| 17:06.05 | jbot | [freepbx] MIME E-mail Encapsulation of Aggregate Documents, such as HTML (MHTML). J. Palme, A. Hopmann. March 1997. (Format: TXT=41961 bytes) (Status: PROPOSED STANDARD) |
| 17:06.14 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 17:06.41 | Qwell | ... |
| 17:06.42 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:06.43 | mathcubes | so i should go to them for help then i guess |
| 17:07.04 | *** part/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:07.07 | etfonhomey | mathcubes, affirmative |
| 17:08.35 | *** join/#asterisk udigits (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:08.38 | [TK]D-Fender | Qwell: Somebody ^&%#$ed with my bitch... |
| 17:09.13 | [TK]D-Fender | Qwell: IMMA GONNA KILL |
| 17:10.36 | mathcubes | whats wrong with freepbx? :P |
| 17:10.42 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 17:10.53 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:12.11 | mathcubes | i dunno you guys are too geeky for me |
| 17:12.21 | mathcubes | thanks for telling me to sod off |
| 17:12.25 | mathcubes | bai |
| 17:12.29 | *** part/#asterisk mathcubes (n=chatzill@host81-149-211-7.in-addr.btopenworld.com) |
| 17:13.29 | [TK]D-Fender | What we really mean to say is "FreePBX owns your sorry ass, and if you can't tell it what to do, we can't support it". Too bad he isn't here to hear it |
| 17:13.52 | Spirits-Sight | [TK]D-Fender: I am not sure what to put for this "externip=222.222.222.222 |
| 17:13.52 | Spirits-Sight | ; this is our routerâs WAN IP." I know its a dum question? |
| 17:14.08 | SiberAIR | Spirits-Sight: your external ip |
| 17:14.12 | SiberAIR | Spirits-Sight: www.ipchicken.com |
| 17:14.16 | [TK]D-Fender | Spirits-Sight: You put your router's WAN IP, just like it says. |
| 17:15.26 | etfonhomey | Spirits-Sight, do you know what NAT is? |
| 17:15.34 | udigits | Hey guys, check out our new Asterisk project: http://udigits.com |
| 17:15.51 | *** join/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
| 17:15.55 | scoates | hi |
| 17:16.19 | scoates | anyone know if there's a way to set up asterisk with Fido's UNO (UMA) service? |
| 17:16.30 | Qwell | scoates: There is not. |
| 17:16.30 | scoates | (I'd like my phone to talk to asterisk, not have to use Fido to provide service over wifi) |
| 17:16.41 | Spirits-Sight | network address translation, it translates ip address numbers into they longer counterpart |
| 17:17.17 | scoates | Qwell: )-: too bad. Is there someone I can read up on it? I don't really care about number/call portability. I'd just like to use it as a wifi phone when home (I'm already set up for SIP) |
| 17:17.25 | scoates | s/someone/somewhere/ |
| 17:17.28 | Qwell | scoates: Does the phone do SIP? |
| 17:17.43 | scoates | ooh.. is jbot Phergie? |
| 17:18.05 | Spirits-Sight | and it allows you to have one ip address that is map to a number of things using the router |
| 17:18.13 | scoates | oh, no.. infobot. |
| 17:18.20 | jasonwoot | IP masquerading still sounds cooler |
| 17:18.30 | scoates | Qwell: I don't know. Nokia 6301. |
| 17:18.50 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 17:19.10 | Qwell | lots of Nokia's can |
| 17:19.11 | stmaher | hi guys.. im trying to get meet me working.. |
| 17:19.16 | stmaher | can you please take a look at chan_iax2.c: Unable to open IAX timing interface: No such file or directory |
| 17:19.17 | jtodd-zzz | 6301 can do SIP, according to Google. |
| 17:19.34 | etfonhomey | Spirits-Sight, how does it work, though? |
| 17:19.38 | Qwell | steals jtodd's zzz's |
| 17:19.40 | Spirits-Sight | what happens if your WAN is dynimic |
| 17:19.51 | Spirits-Sight | I don't understand that |
| 17:19.55 | *** part/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 17:20.00 | etfonhomey | Spirits-Sight, it doesn't matter if you have a static or dynamic IP address, it still works the same. |
| 17:20.00 | stmaher | I have the ztdummy driver installed and working.. |
| 17:20.44 | Spirits-Sight | no, what I mean is if my WAN is dynamic what do I write in my sip.conf then |
| 17:21.03 | [TK]D-Fender | Spirits-Sight: When your WAN is dynamic you need to use a DynDNS type service along with "externhost=mydyndnshost" and "externrefresh=60" (seconds... can play around if needed) |
| 17:21.25 | Spirits-Sight | great, I already have this :-) |
| 17:23.54 | Spirits-Sight | ok, I am still reading |
| 17:24.03 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 17:24.17 | gsiener | Anyone here using a Voismart vGSM card? I can't get mine working. |
| 17:25.15 | Spirits-Sight | what does the /24 after localnet=192.168.1.0/24 mean? |
| 17:25.40 | Spirits-Sight | thats not a port right? |
| 17:26.09 | IsUp | class |
| 17:26.15 | Spirits-Sight | also the ip address here is my routers right? |
| 17:26.17 | [TK]D-Fender | Spirits-Sight: SUBNET |
| 17:26.29 | Spirits-Sight | have to go read that to now |
| 17:26.37 | [TK]D-Fender | Spirits-Sight: You probably should take a course on basic netowrking... |
| 17:26.43 | scoates | thanks. |
| 17:26.43 | *** part/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
| 17:26.57 | [TK]D-Fender | Spirits-Sight: and no, that is not an IP address. |
| 17:27.05 | [TK]D-Fender | Spirits-Sight: that is a subnet & mask |
| 17:29.49 | *** join/#asterisk wastrel (n=wastrel@nylug/member/wastrel) |
| 17:30.17 | Spirits-Sight | I know after the / is the subnet & mask, its before the / is the networks main ip address (in this case the router?) |
| 17:30.49 | wastrel | hi hi. we're experiencing a large delay before playback of sound files in the menus |
| 17:30.56 | wastrel | where would i look to start to troubleshoot this? |
| 17:31.03 | jasonwoot | I'd like to thank Spirits-Sight for making all my questions look really, really complicated in comparison |
| 17:32.18 | stmaher | ~nortel |
| 17:32.18 | jbot | rumour has it, nortel is Equivalences between 1988 X.400 and RFC-822 Message Bodies. H. Alvestrand & S. Thompson. August 1993. (Format: TXT=37273 bytes) (Status: PROPOSED STANDARD) |
| 17:33.16 | [TK]D-Fender | Spirits-Sight: No, it is not. notice the last byte is 0 and the "/24". |
| 17:33.30 | [TK]D-Fender | Spirits-Sight: that is not an IP ADDRESS |
| 17:33.42 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 17:33.47 | [TK]D-Fender | Spirits-Sight: it is a subnet & mask |
| 17:33.47 | Spirits-Sight | jasonwoot: your welcome, I am at less trying to understand this stuff, I don't do this for a living, I don't plan on to and I am just doing this as much as I have to get up a phone system that will allow a couple little things and thats all, make & get calls (done) try to get rid of error, be able to have a little menu |
| 17:34.18 | [TK]D-Fender | stmaher: jbot is FUBAR'd |
| 17:34.37 | demonist | . |
| 17:34.43 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 17:35.15 | Spirits-Sight | localnet = net.ip.addr/subnet.mask, so because it has a 0 at the end and the router would not have a 0 it would be likly a 1 |
| 17:35.31 | Spirits-Sight | or any thing else beside just a 0 |
| 17:37.21 | [TK]D-Fender | Spirits-Sight: this is not an IP address. |
| 17:38.11 | Spirits-Sight | .I understand that now, see it looks like one so thats why I call it one, so 192.168.1.0 would be a part of a subnet & mask |
| 17:38.21 | [TK]D-Fender | Spirits-Sight: this is jsut like the misconception people make when I tell them to point to my computer and they point their finger at the SCREEN. |
| 17:39.20 | Spirits-Sight | ok, at less I am not that crazy (don't know how to word that any other way) |
| 17:39.21 | [TK]D-Fender | Spirits-Sight: 192.168.1.0 <- subnet "/24" ,- mask the 1st 24 bits, the last *8* is the IP range on that subnet. 0 is network, 255 is broadcast, the REST are usable. |
| 17:40.50 | *** join/#asterisk profxavier (n=fook@unaffiliated/neverblue) |
| 17:41.09 | Spirits-Sight | so does the router its self make what the subnet is? or do I contral that? |
| 17:41.48 | hardwire | boss won't let me use his useless vegastream as a keyboard riser anymore :( |
| 17:41.50 | hardwire | it was so nice |
| 17:41.55 | hardwire | it hat a contoured front bevel |
| 17:42.02 | [TK]D-Fender | Spirits-Sight: It is possible to have multiple subnets on the same ethernet segment, but multiple DHCP servers = no-no |
| 17:42.54 | [TK]D-Fender | Spirits-Sight: You're typical network only has 1 subnet per ethernet network, and your typicaly Joe Blow cump router is a DHCP server that hands out IP addresses for the subnet you define. |
| 17:43.14 | [TK]D-Fender | hardwire: Yeah... but he needed a door-stop :p |
| 17:44.04 | rene- | does a lot of u guys use the wan capabilities of zaptel tdm cards? |
| 17:44.04 | hardwire | I can't believe these sell for $300/ebay |
| 17:44.41 | profxavier | on my new Polycom 330, when I dial a number and the headset is picked up, i cannot dial a number longer than 11 digits long, is this something in the config of the phone, as we used to use Grandstream phones, and could dial anything |
| 17:45.10 | rene- | yes there are a dialplan configuration you need to change |
| 17:45.13 | [TK]D-Fender | rene-: As in? |
| 17:45.25 | rene- | as in T1, Dual T1 or E1 data routers |
| 17:45.29 | profxavier | me rene- ? |
| 17:45.38 | rene- | profxavier: yes |
| 17:45.40 | [TK]D-Fender | rene-: You want it to act as an IP device, not TDM voice? |
| 17:45.43 | profxavier | ah |
| 17:45.49 | hardwire | I got my polycoms in.. |
| 17:45.52 | [TK]D-Fender | profxavier: Change your phone's dialplan |
| 17:45.53 | rene- | [TK]D-Fender |
| 17:45.54 | hardwire | but I don't have any space to put them |
| 17:45.56 | rene- | i have |
| 17:46.45 | rene- | i have used a digium card for both Data and Voice T |
| 17:46.45 | rene- | s |
| 17:46.45 | Spirits-Sight | Ok so my router (DD-WRT) is my DHCP (understand that) so what it has in it is my subnet and the computer that are connected to it all get IP addresses, right? so my subnet is indeed 192.168.1.0 then and I want /24 after it to do what? (I hope you don't mind explaining this to me) |
| 17:48.00 | rene- | it was stable for like six months, then i began experiencing PCI card compatibility errors on the Dell 2950 together with line faults on the Data T1, i ended up getting a second hand T1 data Router |
| 17:48.07 | rene- | a cisco 17xx |
| 17:48.25 | rene- | i wonder if somebody has used digium or sangoma as a long term wan solution |
| 17:48.30 | rene- | for linux |
| 17:48.46 | [TK]D-Fender | Spirits-Sight: 192.168.1.0/24 says that your local network is comprised of addresses from 192.168.1.0 to 192.168.8.255. the first is the NETWORK address, the last is the BROADCAST address for the subnet. |
| 17:49.11 | [TK]D-Fender | Spirits-Sight: Your router TAKES one of the range for itself (commonly given the FIRST in the range) |
| 17:49.29 | [TK]D-Fender | Spirits-Sight: And its DHCP server gives out addresses in the rage you tell it to. |
| 17:49.47 | Spirits-Sight | oo ok, I get it more now, thanks |
| 17:49.59 | [TK]D-Fender | rene-: Sangoma has been doing T1 WAN for 20 years |
| 17:50.04 | [TK]D-Fender | rene-: rock solid stuff |
| 17:50.13 | [TK]D-Fender | rene-: And never had issues with Dell :) |
| 17:51.03 | [TK]D-Fender | Spirits-Sight: You are stongly advised to go do some serious reading on networking basics. how NAT & routing works, DNS, etc. |
| 17:52.21 | Spirits-Sight | I need the information given to me in as clear and simple way as can as if its not its gets hard for my mind / brain to process it |
| 17:52.40 | [TK]D-Fender | Spirits-Sight: Just go find a way. |
| 17:53.23 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.10.119) |
| 17:56.35 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 17:57.12 | Spirits-Sight | [TK]D-Fender: when I am connecting a phone to the router directly, should I use the ip address of the asterisk box or my dynDNS address (whatever.homedns.org) I know when connecting from out side I want to do this but not sure for inside phone or even a softphone that may be in / out side of the network |
| 17:57.30 | Spirits-Sight | for the proxy |
| 17:57.46 | [TK]D-Fender | Spirits-Sight: Local SIP devices should be given the local subnet IP address of your * server. |
| 17:58.03 | *** join/#asterisk szallol (n=szallol@86.105.195.113) |
| 17:58.20 | *** join/#asterisk damnpoet (n=UCF\inf2@orion.ucf.edu.cu) |
| 17:58.22 | Spirits-Sight | ok so in this case I would do the 192.168.1.100 |
| 17:59.06 | [TK]D-Fender | Spirits-Sight: if that's your server's address, yes |
| 17:59.39 | Spirits-Sight | beatful, and on the device I would say no for nat because I am behind the router |
| 17:59.59 | Spirits-Sight | so for that extion I would put nat=no |
| 18:00.53 | Spirits-Sight | any thing that happens that is plug into my router is not effected by the NAT right? and stuff out side the router is effected right? |
| 18:04.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:05.01 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 18:05.28 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 18:07.24 | [TK]D-Fender | Spirits-Sight: Your local lan is local and communication to IP's inside have nothing to do with your router or NAT |
| 18:08.44 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 18:08.52 | Spirits-Sight | I ask this because in the example it gave in the link you gave it says for A user it would have nat=yet because its behind its own NAT (router) |
| 18:10.16 | *** part/#asterisk damnpoet (n=UCF\inf2@orion.ucf.edu.cu) |
| 18:11.45 | Spirits-Sight | Sorry I ment to say B user |
| 18:12.22 | *** join/#asterisk feeds (n=feeds@85-135-232-31.adsl.slovanet.sk) |
| 18:13.26 | etfonhomey | Spirits-Sight, I've had time to go to Best Buy and buy a Mac Book Pro and you still haven't figured out NAT? |
| 18:16.25 | wastrel | getting a weird delay between sound files playing in the voice mail menus. "press one for" (2-3 second pause) "Inbox" etc. |
| 18:16.47 | wastrel | they're the default sound files. what would i look at to start toubleshooting this? |
| 18:17.25 | wastrel | hrm asterisk is taking up 100% cpu |
| 18:17.34 | szallol | how can I originate a call with asterisk through oh323 trunk? |
| 18:17.42 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-87-202.w86-215.abo.wanadoo.fr) |
| 18:19.14 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-87-202.w86-215.abo.wanadoo.fr) |
| 18:19.15 | etfonhomey | wastrel, what kind of CPU is this running on? |
| 18:21.56 | etfonhomey | wastrel, I've read about some bugs in the zaptel stuff that cause the CPU to go crazy. |
| 18:22.49 | wastrel | core2duo |
| 18:30.29 | Qwell | etfonhomey: Rule #1 of #asterisk. If you buy yourself an MBP, you have to buy one for all the channel ops. :) |
| 18:30.32 | gambolputty | * still isn't writing a asterisk.ctl or asterisk.pid file, even when run as root. Any ideas? |
| 18:31.00 | etfonhomey | Qwell, Phew, good thing I didn't buy it for myself. |
| 18:31.04 | Qwell | foiled |
| 18:32.12 | etfonhomey | Qwell, does the soundcard you have come into play at all when playing audio files? |
| 18:32.23 | Qwell | only if you're using a console channel driver |
| 18:32.27 | *** join/#asterisk tengulre (n=tengulre@124.173.186.139) |
| 18:35.43 | etfonhomey | Qwell, hence all the OSS errors I get when doing a dial fromt he CLI with no soundcard in the machine. :) |
| 18:36.16 | Spirits-Sight | I thank you very very [TK]D-Fender it appears to be working right, it does not give me the error any more. |
| 18:37.08 | feeds | hi all, what is the dialplan application to redirect a call to another exten, or if there is no app how can I do it then? |
| 18:37.43 | Spirits-Sight | I do how every have another issue with out any errors on the CLI, when I got a call coming throw I answer the phone and don't hear any thing, but the person on the other end can hear me |
| 18:42.29 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:43.51 | Spirits-Sight | [TK]D-Fender: is there any thing else I should do to my sip file for NAT stuff ? I believe I have followed the isturtion as stated, and it appears to be working http://pastebin.com/m785878c2 |
| 18:44.33 | etfonhomey | Spirits-Sight, that means the SIP is working throught NAT but the RTP is not. |
| 18:47.17 | Spirits-Sight | ok I have looked in the rtp.conf file to see what numbers are in there and its 10000 - 20000 and my router is setup to forward ports 10k to 20k to my asterisk box is there any thing else I had to do for that |
| 18:50.38 | *** join/#asterisk rhousand (n=rhousand@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:51.12 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 18:52.45 | *** join/#asterisk Alan_Hicks (i=alan@cardinal.lizella.net) |
| 18:53.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:53.52 | jblack | 5060 for sip |
| 18:54.07 | Alan_Hicks | Howdy. I've got a strange echo/static condition on my phones. I'm using Polycom IP 320 phones and a Digium TDM410P with 2 FXO modules. When I turn "echocancel" off in zapata.conf, my phone calls are free of static but the echo is strong. If I turn "echocancel" on, the echo goes awa completely, but there is a lot of line noise on the channel. |
| 18:54.15 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:54.43 | Alan_Hicks | This line noise can only be heard by the local caller/callee. In other words, the remote person in the phone conversation hears no echo or static. |
| 18:55.06 | Alan_Hicks | I've tried this with and without a hardware echo cancelling module on the card without change. |
| 18:55.17 | Alan_Hicks | Does anyone have any ideas what I could check next? |
| 18:56.11 | Spirits-Sight | jblack: Yes port 5060 is forwarded also |
| 18:59.10 | *** join/#asterisk telnettech (n=telnette@12.236.122.2) |
| 18:59.28 | *** join/#asterisk VoipForces (n=courchea@office.privalodc.com) |
| 18:59.51 | VoipForces | Hi all, Anyone has clues for on on outgoing calls fax detection? |
| 19:00.17 | VoipForces | Basically sending fax via a call file, but I need to know if it's a person or a fax that answered. |
| 19:06.49 | telnettech | here is a good question for all |
| 19:08.16 | telnettech | I have a PRI for outbound calls. In the CDR we are passing the Caller ID which is set thru my extension.conf file. I need to send to the cdr.conf file to capture the extension that is making the call. what is the identifier that I need to put in my mapping |
| 19:08.34 | etfonhomey | Spirits-Sight, where is the other endpoint that you are calling? |
| 19:08.47 | etfonhomey | Or that is calling you? |
| 19:09.09 | Spirits-Sight | etfonhomey: it was a house phone and a cell phone |
| 19:09.47 | Spirits-Sight | house phone = comcast digital phone and cell phone = AT&T if that makes any difference |
| 19:09.55 | *** join/#asterisk feeds (n=feeds@85-135-229-21.adsl.slovanet.sk) |
| 19:10.22 | etfonhomey | Spirits-Sight, do you have your * box registering to an ITSP? |
| 19:10.39 | etfonhomey | Somewhere your call gets changed from the PSTN to SIP? |
| 19:11.05 | Spirits-Sight | Yes, its only when person seem to call me on my DiD (*) |
| 19:12.29 | feeds | hi all, what is the dialplan application to redirect a call to another exten, or if there is no app how can I do it then? |
| 19:12.38 | etfonhomey | feeds, GoTo |
| 19:12.51 | feeds | so GoTo(1000) ? |
| 19:12.54 | feeds | for example |
| 19:13.18 | kaldemar | core show application goto will tell you how to use it |
| 19:14.10 | feeds | kaldemar and etfonhomey: Thanks |
| 19:15.12 | kaldemar | Spirits-Sight: do you have UDP ports 10000-20000 open in your asterisk box's firewall? |
| 19:16.38 | Spirits-Sight | the only firewall that I know that I have is the router one, I did not install any on my Ubuntu system |
| 19:16.54 | Spirits-Sight | is there one by default I don't know |
| 19:17.05 | Spirits-Sight | i know not a surprise that I don't know |
| 19:19.06 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:20.03 | etfonhomey | Spirits-Sight, what's the pathway of the incoming SIP connection? (I.E. ITSP (Vitelity.net) ---> PIX firewall (dynamic IP) ---> * box) |
| 19:21.45 | Spirits-Sight | ipcomm => cable modem -> Router (DD-WRT) -> asterisk Box |
| 19:21.45 | Spirits-Sight | <PROTECTED> |
| 19:22.10 | kaldemar | theres iptables firewall in ubuntu also. |
| 19:22.37 | Spirits-Sight | thanks, so let me look to change that |
| 19:23.39 | Spirits-Sight | so I want to open the ports on the Ubuntu system do I forward them to the same ip address or just open them |
| 19:24.00 | *** join/#asterisk plasmid (n=noway@c-71-225-10-10.hsd1.pa.comcast.net) |
| 19:24.12 | kaldemar | open them. your ubuntu system is "the same ip address". |
| 19:24.26 | Alan_Hicks | No one have any ideas of things to try to eliminate this echo/static problem I'm having? |
| 19:25.07 | *** join/#asterisk digitalScream200 (n=outkast@office.telifon.com) |
| 19:25.54 | plasmid | Today, I installed asterisk for the first time, everything is ok but the incoming calls sound like robotic, almost like a human is speaking with a voice-aided device near their throats. I have a PAPT-2NT outside. What can I do to improve the quality of the voice? change codecs? or buy an internal Digium card? change hardware inside computer? This installation is strictly for home use. |
| 19:26.02 | etfonhomey | Spirits-Sight, turn the firewall off on everything. Make it easy on yourself. |
| 19:26.16 | etfonhomey | Then when you have * working, secure things again. |
| 19:27.30 | kaldemar | "sudo iptables -I INPUT -p udp --dport 10000:20000 -j ACCEPT" will open the ports for you, but you better learn what that command does. |
| 19:27.32 | Alan_Hicks | plasmid: Change codecs. I'll bet you're using gsm. |
| 19:27.52 | Alan_Hicks | Try changing to ulaw and all those problems should go away if that's the case. |
| 19:27.55 | *** join/#asterisk wolffear (n=wolffear@97.89.125.69) |
| 19:28.05 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 19:28.13 | wolffear | anyone here willing to assist me with a grandstream BudgeTone 100 configuration that has been requested |
| 19:28.16 | plasmid | yes I am using ulaw according to my asterisk.conf file. |
| 19:28.44 | Alan_Hicks | plasmid: This is on incoming and outgoing calls on the PSTN? |
| 19:28.46 | telnettech | let me post again |
| 19:28.53 | telnettech | here is a good question for all |
| 19:28.54 | telnettech | <telnettech> I have a PRI for outbound calls. In the CDR we are passing the Caller ID which is set thru my extension.conf file. I need to send to the cdr.conf file to capture the extension that is making the call. what is the identifier that I need to put in my mapping |
| 19:29.14 | plasmid | Alan_Hicks No, strictly over RJ45. NO pstn. I have not used a PSTN for over 6 years. |
| 19:29.14 | *** join/#asterisk omaha- (n=squall@ns2.squallnetwork.net) |
| 19:29.48 | wolffear | plasmid, how much do you know about grandstream phones other than there cheap :P |
| 19:30.49 | plasmid | wolffear, do not use one. |
| 19:30.49 | *** join/#asterisk fede2 (n=alvaro@201.192.28.246) |
| 19:30.49 | wolffear | lol, I would agree, but the previous tech before me ruined the company with them, now thats all I have to work with |
| 19:31.36 | wolffear | the owner of the company has requested for me to setup the msg button so that it bypasses the login for VM and goes straight to the msgs |
| 19:31.51 | wolffear | we are not using 3rd party VM |
| 19:31.56 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 19:32.09 | telnettech | wolffear: is it the grandstream GXP 2000 model |
| 19:32.17 | wolffear | I wish, more a budgetone 100 :( |
| 19:32.37 | telnettech | wolffear: it has a web interface right? |
| 19:32.42 | wolffear | yes :) |
| 19:32.51 | wolffear | it has a field for Voice Mail UserID |
| 19:32.58 | wolffear | and it doesnt give much info on the manual on configuring |
| 19:33.24 | wolffear | all it says is to enter the ext... but all it does is forward me to there unavail msg |
| 19:33.29 | *** join/#asterisk luke-jr (n=luke-jr@2002:48c4:141a:0:20e:a6ff:fec4:4e5d) |
| 19:33.38 | telnettech | put the voicemail retrieval number in that field and it should work.....i just set up 8 GXP 2000 that way and it is working here at this customer site |
| 19:33.48 | wolffear | like |
| 19:33.53 | wolffear | *97ext |
| 19:34.02 | *** join/#asterisk km2 (n=x@32.178.18.234) |
| 19:34.02 | wolffear | *971337 |
| 19:34.11 | wolffear | is this the proper format? |
| 19:34.17 | telnettech | whatever you dial to get into the voicemail....example is 770 |
| 19:34.23 | wolffear | ok |
| 19:34.31 | [TK]D-Fender | wolffear: no such thing as "proper". Depends on your dialplan <- |
| 19:34.39 | wolffear | is it possible to set a string to automatically enter the pw for the box as well? |
| 19:34.58 | [TK]D-Fender | wolffear: No, again that is just you and your dialplan. |
| 19:34.59 | wolffear | [TK]D-Fender: understood |
| 19:35.03 | wolffear | ok |
| 19:35.18 | [TK]D-Fender | wolffear: "show application voicemail" |
| 19:35.58 | wolffear | ok, so if I set it to the dialplan, it will forward me to the vm system |
| 19:36.00 | wolffear | I understand |
| 19:36.14 | wolffear | but is it possible to set it to just automatically connect directly to the box |
| 19:36.26 | wolffear | w/o them having to enter any info |
| 19:37.00 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 19:37.41 | Alan_Hicks | What could cause static on a PSTN line on incoming/outgoing calls, but not cause static for SIP or IAX2 calls? I've verified with a regular analogue phone that the lines themselves are not somehow faulty. |
| 19:38.08 | wolffear | what does that static sound like? |
| 19:38.13 | wolffear | the matrix? |
| 19:38.16 | plasmid | Alan_Hicks static is caused by foreign voltage. You may need a filter. |
| 19:38.17 | wolffear | or like water in your lines? |
| 19:38.46 | Alan_Hicks | I wouldn't know what either the matrix or water in the lines sounds like. :^) |
| 19:39.16 | Alan_Hicks | But like I said, if I hook a butt-set or analogue phone direct to the line and place/make calls, there is no static noise. |
| 19:39.39 | etfonhomey | Alan_Hicks, what are the POTS lines connecting to on your * system? |
| 19:39.53 | etfonhomey | Alan_Hicks, PCI card or some external ATA? |
| 19:39.55 | *** join/#asterisk Un1x (n=Shaanay@CPE001d7e1e2ddb-CM001692fa22fa.cpe.net.cable.rogers.com) |
| 19:39.55 | Alan_Hicks | etfonhomey: Digium TDM410 |
| 19:39.58 | Alan_Hicks | PCI |
| 19:40.36 | Un1x | Hey, i'm having a problem i had a power outage my server was working fine and i reboot now and i start asterisk, and i dont hear a tone, alongside the fact the lights on the card are not lighting up but when i do lspci it sees the card there what could be the problem? |
| 19:41.22 | wolffear | Un1x: did u check your asterisk info to ensure your provider is in an OK status? |
| 19:41.43 | [TK]D-Fender | dialplan = EXTENSIONS.CONF |
| 19:41.49 | wolffear | ty fender |
| 19:41.51 | wolffear | thats what I needed! |
| 19:41.53 | etfonhomey | Alan_Hicks, I tried the Digium cards once and did not have good luck with them. I will qualify that by saying that I did not contact Digium support to fix it. |
| 19:41.58 | [TK]D-Fender | Un1x: Lack of a zaptel init script |
| 19:42.07 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 19:42.09 | Un1x | [TK]D-Fender how may i fix that? |
| 19:42.18 | [TK]D-Fender | Un1x: Set one to start up. |
| 19:42.23 | Alan_Hicks | Well, this didn't start to happen until I replaced the old TDM400 card that was in this box. |
| 19:42.30 | Un1x | [TK]D-Fender do you have a link to a readme/guide? |
| 19:42.42 | Alan_Hicks | After a power outage, the TDM400 card stopped working, so I had to replace it with this card. |
| 19:42.47 | *** part/#asterisk udigits (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 19:43.03 | Alan_Hicks | I'd used this card in a development box without problems, so I doubt there's anything wrong with the card itself. |
| 19:43.11 | [TK]D-Fender | Un1x: the docs w/ your source. |
| 19:43.30 | [TK]D-Fender | Un1x: normally "make config" will do it for several common distros |
| 19:43.42 | etfonhomey | Alan_Hicks, I would contact Digium support and see if they can work with you to get it going. |
| 19:44.15 | Un1x | hrmp i did ... /etc/init.d/dahdi start |
| 19:44.18 | Un1x | and it worked.. |
| 19:44.22 | Un1x | so i guess its not start dahdi |
| 19:46.17 | Alan_Hicks | etfonhomey: Thanks. I'll see about contacting them Monday if I don't get this resolved today. |
| 19:48.41 | *** join/#asterisk feeds (n=chatzill@85-135-234-118.adsl.slovanet.sk) |
| 19:48.49 | feeds | ~book |
| 19:48.50 | jbot | book is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:50.29 | wolffear | [TK]D-Fender, if I setup voicemail.conf to point directly to the box, would any update to extension.conf be needed? |
| 19:50.54 | [TK]D-Fender | wolffear: voicemail.conf doesn't point to ANYTHING |
| 19:51.12 | wolffear | ok, ty |
| 19:51.13 | [TK]D-Fender | wolffear: "show application voicemail" |
| 19:51.33 | [TK]D-Fender | wolffear: Dialplan is 99% of * |
| 19:52.02 | wolffear | [TK]D-Fender I'm not to versed in the format of this file |
| 19:52.21 | [TK]D-Fender | wolffear: Considering the above ratio, you need to. |
| 19:52.45 | wolffear | [TK]D-Fender ok, I will do my research |
| 19:52.51 | wolffear | [TK]D-Fender ty for the info :) |
| 19:53.18 | [TK]D-Fender | wolffear: Kinda like saying "I know how to drive with the exception of the steering, parking, signalling, changing gears, and accelerating" |
| 19:53.47 | wolffear | [TK]D-Fender hah, the phone system here isnt my job, I'm new to it and attempting to cleanup a previous admins problems |
| 19:53.54 | Spirits-Sight | etfonhomey: ok I don't have any firewall on my system any more, its still not allowing a person to call me and hear them, I am going to disable my router fireware to see if that make any difference |
| 19:53.56 | jasonwoot | Fender with the car analagies... |
| 19:54.05 | [TK]D-Fender | wolffear: thats what consultants are for. |
| 19:54.06 | [TK]D-Fender | :) |
| 19:54.16 | [TK]D-Fender | Spirits-Sight: Nope... |
| 19:54.24 | wolffear | only if you could convince my boss that, I would love u! |
| 19:54.27 | etfonhomey | Spirits-Sight, disable all firewalls. |
| 19:54.36 | [TK]D-Fender | Spirits-Sight: Go read the guide again CAREFULLY. and then read it another *10* times and see what you missed |
| 19:54.51 | [TK]D-Fender | Spirits-Sight: You missed the big print on this... |
| 19:55.15 | Spirits-Sight | aright going to read it again |
| 19:55.31 | wolffear | [TK]D-Fender: I found two websites on the matter |
| 19:55.32 | Alan_Hicks | Thanks for the advice guys. I gotta run. |
| 19:55.32 | wolffear | http://www.voip-info.org/wiki/index.php?page=Asterisk+-+documentation+of+application+commands |
| 19:55.47 | wolffear | http://www.asteriskguru.com/tutorials/extensions_conf.html |
| 19:56.00 | wolffear | do you have any further recommended material on this subject matter? |
| 19:56.11 | [TK]D-Fender | wolffear: Be warned that the version of * any WIKI info may pertain to might not apply to yours. |
| 19:56.14 | [TK]D-Fender | ~book |
| 19:56.14 | jbot | somebody said book was probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:56.22 | [TK]D-Fender | wolffear: and the docs in your source tarball. |
| 19:57.02 | *** join/#asterisk mike-ekim (n=mike@apoc.digiport.com) |
| 19:57.08 | wolffear | [TK]D-Fender: which docs are you reffering to? |
| 19:57.26 | [TK]D-Fender | wolffear: and the docs in your source tarball. <--- go find your source tarball and LOOK |
| 19:57.41 | wolffear | [TK]D-Fender: ok |
| 19:58.22 | plasmid | Today, I installed asterisk for the first time, everything is ok but the incoming calls sound like robotic, almost like a human is speaking with a voice-aided device near their throats. I have a PAPT-2NT outside. What can I do to improve the quality of the voice? change codecs? or buy an internal Digium card? change hardware inside computer? This installation is strictly for home use. I currently use ulaw |
| 19:58.33 | *** join/#asterisk homeins6 (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 19:59.06 | [TK]D-Fender | plasmid: 1st guess : your bandwidth is too low and are getting cut-off. |
| 19:59.31 | plasmid | [TK]D-Fender, so it's QoS at the router level for VOIP? |
| 19:59.49 | [TK]D-Fender | plasmid: Test this by modding your ITSP peer & phone to G.729 ensuring that * needs to play audio into the stream and see if the issue improves. if so, its definitely bandwidth |
| 20:00.14 | [TK]D-Fender | plasmid: QoS does not exist on the internet, and only helps what heads OUT of the WoS'd side |
| 20:00.22 | [TK]D-Fender | QoS* |
| 20:00.55 | Spirits-Sight | [TK]D-Fender: your talking about the link "http://www.aocomputing.net/?p=3" right? |
| 20:01.14 | [TK]D-Fender | Spirits-Sight: Yes |
| 20:01.44 | plasmid | [TK]D-Fender checking for G.729 codec .... and QoS was for the router (assuming it is a bandwith problem and my router is not dedicating enough of it to my asterisk-debian build) |
| 20:02.39 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 20:02.57 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 20:03.15 | Spirits-Sight | I don't see what I am doing wrong, my router has port 5060 forwarded to my * box and ports 10000 - 20000 forwarded to my * box, I don't see any other thing in there that is BIG print and I don't see what I have missed |
| 20:03.18 | Spirits-Sight | sorry |
| 20:04.29 | *** join/#asterisk ziram19 (n=chatzill@41.226.86.78) |
| 20:07.05 | Spirits-Sight | the only problem that seem to be is I can not hear a call talking to me, they can hear me (this is with them calling me) |
| 20:14.26 | *** join/#asterisk kj5t (n=steve@65-120-138-35.dia.static.qwest.net) |
| 20:15.13 | kj5t | Can someone assist me with Asterisk paging? My boss wants us to be able to page back in forth but doesn't want to be interupted when someone makes a page to the whole company if he is in a call with someone |
| 20:17.51 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 20:18.16 | kj5t | is there an argument for the page command that will stop the page from eaching a phone if there is a call in progress? |
| 20:18.29 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 20:18.40 | Akiyuki | What can I use to play back .raw files generated by ExtenSPY? |
| 20:19.03 | WimpMan | kj5t: I'd expect any decent phone to do so by default. |
| 20:19.15 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 20:21.12 | Akiyuki | I tried using playback, but it said infalid format. |
| 20:21.46 | mikealeonetti | What's the best way to increase the performance of offsite phones? |
| 20:22.03 | mikealeonetti | so they don't sound "garbled" |
| 20:23.07 | [TK]D-Fender | Spirits-Sight: Yes, i'm well aware. You left a SETTING out of your configs |
| 20:23.30 | [TK]D-Fender | Spirits-Sight: this is not a port forwarding issue, this was "I'm not doing the settings I was told to" problem |
| 20:24.09 | Spirits-Sight | ok, let me read again, my eyes and brain are going to be dead :-) |
| 20:24.19 | Akiyuki | [TK]D-Fender: What can I use in linux to play the .raw files created by r() in extenspy? |
| 20:24.26 | [TK]D-Fender | kj5t: there is non, this is for you to determine in the dialplan first |
| 20:24.33 | [TK]D-Fender | Akiyuki: No idea |
| 20:24.52 | [TK]D-Fender | Akiyuki: I'd naturally try VLC as it plays just about anything... |
| 20:24.57 | *** join/#asterisk szallol (n=szallol@86.105.195.113) |
| 20:25.47 | Spirits-Sight | [TK]D-Fender: its not the extern= ..... right? |
| 20:25.59 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.197.131) |
| 20:26.20 | [TK]D-Fender | Spirits-Sight: I said an option you MISSED. plpease pay attention. No co COMPARE the two and read the guide CAREFULLY. |
| 20:26.25 | Akiyuki | I tried catting it to /dev/audio but it sounded like shit |
| 20:27.05 | [TK]D-Fender | Spirits-Sight: I said an option you MISSED. please pay attention. Now go COMPARE the two and read the guide CAREFULLY. |
| 20:27.14 | [TK]D-Fender | dang my typing is heading down-hill fast |
| 20:28.43 | *** join/#asterisk ziram19 (n=chatzill@196.203.221.213) |
| 20:30.37 | Spirits-Sight | [TK]D-Fender: this is what I see missing that they have under each user A and B, is this the right option your talking about -> canreinvite=no |
| 20:32.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:36.15 | *** join/#asterisk Kvant (i=hh@unaffiliated/kvant) |
| 20:36.21 | Spirits-Sight | the only two option I see are: context=miscsipcalls & canreinvite=no under the [general] the only one that I know is improtant is the canrreinvite so my guess would be this is cause the issue??? |
| 20:40.00 | *** part/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 20:40.37 | *** join/#asterisk AndyML (n=quassel@pool-96-227-91-204.phlapa.fios.verizon.net) |
| 20:41.58 | *** join/#asterisk AndyML (n=quassel@pool-96-227-91-204.phlapa.fios.verizon.net) |
| 20:45.52 | *** join/#asterisk root52 (n=F745082a@ip70-191-120-39.cl.ri.cox.net) |
| 20:48.50 | etfonhomey | Spirits-Sight, you definitely want canreinvite=no |
| 20:48.56 | Akiyuki | [TK]D-Fender: Unfortunately VLC can't play it either. |
| 20:48.59 | Spirits-Sight | it works it works |
| 20:49.14 | etfonhomey | Spirits-Sight, well, what was it? |
| 20:49.24 | Spirits-Sight | canreinvite |
| 20:49.46 | Spirits-Sight | I had it in one area but not in general |
| 20:50.21 | Spirits-Sight | now is there any good read just for creating menu, I am still reading asterisk book also |
| 20:50.56 | mike-ekim | anyone know some common issues of dropped calls, and unreachable states ? |
| 20:51.47 | mike-ekim | as far as network setup? |
| 20:52.00 | mike-ekim | affected only on inbound calls also |
| 20:54.54 | *** join/#asterisk jayrod422 (n=jayrod42@pool-72-77-0-145.pitbpa.fios.verizon.net) |
| 20:55.54 | *** join/#asterisk AndyML (n=quassel@pool-96-227-91-204.phlapa.fios.verizon.net) |
| 20:56.09 | jayrod422 | i have an issue that whenever asterisk receives a reinvite it replies back with the ip of the terminating sip device and tries to remove itself from the media path |
| 20:56.20 | jayrod422 | any idea on how to setup asterisk to not do this |
| 20:56.31 | [TK]D-Fender | Spirits-Sight: I guess you noticed that UBER important warnng in the docs... |
| 20:56.40 | VoipForces | Anyone has experiences with txfax over a PRI? Fax are sending but txfax does not hangup... |
| 20:57.17 | Spirits-Sight | I seen it all that time, it just did not get processed in the brain, I then look at something else and it click |
| 20:58.05 | jaytee | would having my asterisk server constantly connected to a share on a Windows server have any negative impact? |
| 20:59.21 | [TK]D-Fender | Spirits-Sight: it got 2 lines of warning attached to it with exclamation points. Let this be a lesson! |
| 21:00.03 | [TK]D-Fender | jaytee: Yes, M$ will have one last persistant foothold in your organization! |
| 21:00.39 | Spirits-Sight | but how does that effect sound only one way, that is so that a phone can not connect directly right? that was what I read |
| 21:00.44 | jaytee | [TK]D-Fender, so basically no performance impact then. thanks |
| 21:01.06 | [TK]D-Fender | jaytee: For a largly idle connection, who cares? |
| 21:01.45 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 21:02.04 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:02.12 | SkramX | in queues.conf, can I just say *all* agents are a member of the queue.. so I dnt have to define each of them? |
| 21:02.19 | jaytee | I've got it setup as a mountpoint and wasn't sure what kind of overhead that entailed. Doesn't appear to be significant looking at memory and cpu utilization. barely registers at all. |
| 21:02.32 | jaytee | so I'll leave it up. |
| 21:02.39 | [TK]D-Fender | Spirits-Sight: they can hear you because * passed the far side's RTP port info to the phone during the reinvite, however all the INBOUND RTP still gets through to * because of FORWARDING <- |
| 21:02.58 | [TK]D-Fender | Spirits-Sight: and not the phone. So the phone transmits, but doesn't receive. |
| 21:03.09 | [TK]D-Fender | Spirits-Sight: ALL traffic to the outside must pass through * |
| 21:03.21 | *** join/#asterisk feeds (n=feeds@85-135-226-145.adsl.slovanet.sk) |
| 21:03.36 | Spirits-Sight | oh I see |
| 21:03.50 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM001ade84db36.cpe.net.cable.rogers.com) |
| 21:03.56 | jaytee | Asterisk is my "gateway drug" of choice |
| 21:03.56 | Spirits-Sight | now I am reading about extension file, any recommend ones for this |
| 21:04.01 | [TK]D-Fender | jaytee: idle is idea. My home server runs Gnome (on demand only), HTTP, FTP, routing, DHCP, *, SMB, etc |
| 21:04.02 | szallol | ast_request: No translator path exists for channel type OH323 : what does this mean? |
| 21:04.21 | [TK]D-Fender | jayOn an AMD XP2000+ 256meg RAM |
| 21:04.49 | [TK]D-Fender | Spirits-Sight: ... |
| 21:04.51 | [TK]D-Fender | ~book |
| 21:04.51 | jbot | i guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 21:04.53 | [TK]D-Fender | ~wikis |
| 21:04.53 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 21:04.58 | zchaos | anyone know how to get my Cent0S server to boot up without a keyboard connected? i am trying to run the tower in the closet with no monitor, keyboard, or mouse ... but during boot up it freezes saying no keyboard is connected... i'm assuming this is a bios problem? |
| 21:05.24 | [TK]D-Fender | zchaos: is it a BIOS warning, or an OS warning? |
| 21:05.40 | zchaos | bios i believe but i will reconfirm |
| 21:06.30 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 21:06.58 | SkramX | OK.. .how do I say which agents are in which groups? |
| 21:07.40 | SkramX | nvm damn |
| 21:08.17 | feeds | Is it possible to send ekiga "messages" to a SIP hardware phone? If yes what components does the phone need? |
| 21:08.38 | etfonhomey | zchaos, BIOS |
| 21:08.51 | [TK]D-Fender | feeds: * does not pass on SIP messaging |
| 21:09.05 | etfonhomey | zchaos, hook up a keyboard and monitor, go into the BIOS setup, change it to NOT halt on keyboard (or mouse) errors. |
| 21:09.08 | feeds | [TK]D-Fender: Thanks |
| 21:09.23 | feeds | What about the jabber module? |
| 21:09.57 | kj5t | WimpMan: We have the SPA-962, it has its built in paging feature "where I dial *96" and that just rings the given extension (kind of silly if you ask me). I set-up *98 and it works to page and auto-answer and such but if someone is in a call it interrupts |
| 21:09.59 | feeds | for *? Does it support Jabber messaging? |
| 21:10.26 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 21:12.04 | [TK]D-Fender | feeds: Same thing |
| 21:16.58 | wfaulk | I have installed trixbox-ce-2.6.1.13 on a machine with an X100P card and am trying to get it to accept h.323 calls and forward them to an analog line. can anyone help? |
| 21:17.11 | wfaulk | I have enabled the h.323 module, but it might be configured wrong |
| 21:17.32 | wfaulk | when I try to make a call ,there is data spit out on the asterisk console |
| 21:18.09 | [TK]D-Fender | wfaulk: Trixbox is not supported here |
| 21:18.50 | wfaulk | okay, I'm happy to deal directly with asterisk config underneath the trixbox admin |
| 21:19.50 | [TK]D-Fender | wfaulk: Won't help you here |
| 21:19.51 | wfaulk | or I could install a different distribution if you have a suggestion |
| 21:20.08 | [TK]D-Fender | wfaulk: Anything that YOU are in control of, not some GUI |
| 21:20.26 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:21.59 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 21:22.06 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 21:22.20 | *** join/#asterisk gammacoder (n=chatzill@64.132.192.29) |
| 21:22.32 | wfaulk | okay. I'm not really sure how ignoring the trixbox webadmin is different than installing asterisk on a stock OS, but okay |
| 21:22.55 | *** join/#asterisk Tako-san (n=Tako-san@24.68.129.29) |
| 21:23.00 | [TK]D-Fender | wfaulk: Becasue the second you hit the "Apply changes" button for any reason all of your work gets trashed |
| 21:23.26 | [TK]D-Fender | wfaulk: FreePBX OWNS your ass. |
| 21:24.36 | gambler1 | Hi, I am a little confused by the meaning of the "sip trunk". Is there any good doc about this? Google didnt find out nothing and I am very close to opinion that it's nothing more than just a number of simultaneous calls. Right? |
| 21:24.37 | wfaulk | I understand. I can disable the web server altogether, but if it's a dealbreaker for you, I'll install ubuntu and an asterisk package |
| 21:25.09 | [TK]D-Fender | wfaulk: And as for debugging you'll need to look at CLI output for the calls coming in/out, configs, etc |
| 21:25.22 | [TK]D-Fender | wfaulk: And packages are not recommended normally either. |
| 21:25.33 | [TK]D-Fender | ~siptrunk |
| 21:25.33 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 21:25.36 | [TK]D-Fender | ^^^ |
| 21:25.38 | wfaulk | okay. I can install from source. |
| 21:25.46 | wfaulk | any other caveats? |
| 21:25.56 | [TK]D-Fender | gambMost people abuse that term to mean using an ITSP's service |
| 21:25.58 | [TK]D-Fender | ~itsp |
| 21:25.58 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:26.21 | [TK]D-Fender | wfaulk: Packages are often maintained by ass-hats and you never know what you're going to get. |
| 21:26.28 | gammacoder | please help... asterisk box with digium PRI card, zaptel running with OSLEC. All of a sudden no audio passes either way on any SIP to PRI calls (but is recorded perfectly if I use the monitoring). SIP to SIP works fine. SIP to FXS/FXO works fine. Any ideas? |
| 21:26.29 | wfaulk | I can understand that |
| 21:26.57 | wfaulk | I'm just asking so I don't throw another 3 hours away |
| 21:27.01 | [TK]D-Fender | wfaulk: Most of us compile from source and thats what we are able to debug. When Distro-X doesn't keep up and things are broken should we clean up HIS mess? |
| 21:27.29 | [TK]D-Fender | wfaulk: backup your configs, deactivate your GUI and you can try, starting from your physical install |
| 21:27.52 | wfaulk | okay, fine. install OS, install asterisk from source. is ther anything else I might do that would prevent you from being able to help? |
| 21:29.22 | [TK]D-Fender | wfaulk: thats jsut what's suggested for running a system. For assistance you simply need to be running basic configs, not 10 tons of GUI crap |
| 21:29.43 | [TK]D-Fender | wfaulk: Now if your actual * version has issues, then the other stuff comes into play |
| 21:30.09 | [TK]D-Fender | wfaulk: Debugging your configuration jsut means you need sane clean configs. Debugging your code-base is that separate matter. |
| 21:30.21 | [TK]D-Fender | wfaulk: just depends on what your needs requires |
| 21:30.35 | [TK]D-Fender | Anyways, checkout time here... packing up to head home. BBIAB |
| 21:31.14 | wfaulk | really, all I want to do is set up an outgoing "trunk" from an h.323 device to POTS |
| 21:32.23 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
| 21:32.54 | *** join/#asterisk feeds (n=chatzill@85-135-226-145.adsl.slovanet.sk) |
| 21:33.08 | feeds | why do I keep getting this? http://asterisk.pastebin.com/f8de2bde |
| 21:33.24 | feeds | when I use reload and try to call 10001? |
| 21:34.07 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 21:36.07 | thedonvaughn | feeds: what's 1001's default context? extension 10001 isn't defined in it. |
| 21:36.16 | thedonvaughn | it doesn't know how to reach 10001 |
| 21:38.09 | feeds | why? what should I change? my extensions.conf: http://asterisk.pastebin.com/f1b4accf2 |
| 21:38.51 | feeds | ahh, so I have to put 10001 into Internal? |
| 21:39.14 | thedonvaughn | feeds: ok so 10001 is only defined in employees context. if the SIP phone 1001 is not in employee context or doesn't have it defined, it can't dial 10001. |
| 21:39.38 | feeds | or how can I write that the same priviliges as [internal] also have [employees] ? |
| 21:39.42 | thedonvaughn | feeds: so you need to include => employees inside your context or move your phone to employee context. either one would fix it |
| 21:39.52 | feeds | is it include=employees? |
| 21:39.57 | feeds | :D |
| 21:39.59 | feeds | thanks |
| 21:40.00 | thedonvaughn | feeds: include => employees |
| 21:40.10 | feeds | now I get it |
| 21:40.12 | feeds | :) |
| 21:40.12 | thedonvaughn | then asterisk -rx 'dialplan reload' and you should be good |
| 21:40.26 | feeds | That I already know ;) |
| 21:40.53 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 21:41.19 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 21:42.08 | thedonvaughn | feeds: btw when i say 1001's context i mena whatever you have defined in /etc/asterisk/sip.conf or users.conf (not sure what you're using) to define user 1001, you should have a context= directive there. Just fyi |
| 21:42.42 | feeds | thedonvaughn: wait I'll look. |
| 21:42.56 | *** part/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 21:43.28 | feeds | what did you mean by "what you're using" - how can I use one or the other, haven't looked at users.conf too much yet |
| 21:44.04 | thedonvaughn | well i see that you're dialing from '1001'. how did you add? Did you use asterisk-gui or by hand? |
| 21:44.08 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 21:44.13 | feeds | I have [1000] in sip.conf in context [general] ; by hand |
| 21:44.20 | thedonvaughn | ok |
| 21:44.21 | feeds | I don't use the gui |
| 21:44.35 | SkramX | what's the best free Asterisk-integratable Text To Speech system out there that's free? I need something temporary for a proof of concept project |
| 21:44.53 | thedonvaughn | yah that's fine; i don't either ;0 So in sip.conf for your [1000] peer, do you have a context defined in there? |
| 21:45.13 | StephenF | SkramX I think there is something called Festival... Have you seen that? |
| 21:45.22 | SkramX | ah yes, i used that a long time ago |
| 21:45.25 | SkramX | thanks for reminding me |
| 21:45.26 | thedonvaughn | feeds: if not then your default context will be [default] which means you will need 10001 defined or included in your [default] context in extensions.conf |
| 21:46.22 | feeds | btw: thedonvaughn: should I create some contexts in there? like: [employees] and put the 1000, 10001, etc. in there? |
| 21:46.32 | thedonvaughn | feeds: no |
| 21:46.46 | Spirits-Sight | I want when a person calls phone number I want it to play a sound, so do I have to put the phone number as the extension like this : s,1,Playback(thank-you-for-calling) or like :xxxxxxxxxx,1,Playback(thank-you-for-calling)? |
| 21:47.20 | thedonvaughn | feeds: this just defines which context defines your dial plan in extensions.conf. so you could do context = employees if you want [employees] context in extensions.conf to define 1001's dialplan |
| 21:47.44 | gammacoder | >please help... asterisk box with digium PRI card, zaptel running with OSLEC. All of a sudden no audio passes either way on any SIP to PRI or PRI to FXO/FXS calls (but they recorded perfectly if I use the monitoring). SIP to SIP works fine. SIP to FXS/FXO works fine. Any ideas? |
| 21:48.29 | feeds | ahh, so if 10001 is in extensions.conf in conetxt [employees] the in sip.conf: [10001] and under that context=employees ??? |
| 21:48.41 | feeds | * conetxt |
| 21:48.47 | feeds | * uhh context |
| 21:49.13 | thedonvaughn | feeds: heh no. For example if [1001] (you) has context = employees then when _you_ dial 10001 it'll check [employees] in extensions.conf to see how to get to 10001. |
| 21:49.25 | root52 | Spirits-Sight: That depends on how you the call is comeing in. if it is the only extenstion in the context where your DID lands then "s" or "_X." will work |
| 21:50.19 | root52 | Spirits-Sight: Howver if you have multiple DID landing in the same context then you will need to define them better. |
| 21:50.41 | thedonvaughn | feeds: so it appears right now, that since extension 1001 doesn't have a context defined, it's going to use the default context in extensions.conf to define it's dial plan. In [default] context there is no 10001 defined. |
| 21:50.42 | feeds | thedonvaughn: lets say I have context xyz in extensions.conf and he has context employees than I have to include include => employees in context xyz? |
| 21:51.06 | feeds | to be able to call him |
| 21:51.14 | feeds | and all employees |
| 21:51.18 | Spirits-Sight | Going to have one main number, that number I want to be a menu, there is going to be no other numbers at this time |
| 21:52.24 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:52.46 | root52 | Then "s" will work |
| 21:53.02 | root52 | @Spirits-Sight:^^^^ |
| 21:53.07 | thedonvaughn | feeds: kind of. Just think of the context you define in sip.conf as the context that your phone uses to dial out. When you define a context on a say a zapata.conf PRI channel or SIP trunk, than that context is used to define inbound calls. Either way you can define what contexts you want to use to define your dialplan. So when you say dial 10001 it knows to use which context to look it up. Right now it doesn't know to use [employees] cont |
| 21:53.41 | thedonvaughn | feeds: you are in the [default] context since you haven't told asterisk other wise. |
| 21:53.44 | jasonwoot | Team, why does stuff work on development but not on production? |
| 21:53.56 | thedonvaughn | feeds: so you could include => employees inside default if you wanted. or change 1001's default context. |
| 21:54.06 | feeds | in sip.conf? |
| 21:54.13 | feeds | or exten conf? |
| 21:54.18 | thedonvaughn | feeds: sip.conf to change 1001's context |
| 21:54.19 | [TK]D-Fender | [default] is a BAD choice for context name and should never be done |
| 21:54.35 | *** join/#asterisk [Nikon] (n=_Nikon@201.76.135.136) |
| 21:54.36 | thedonvaughn | yah i agree. just telling him his options so he understands how it works |
| 21:54.40 | thedonvaughn | or her :) |
| 21:54.48 | feeds | thanks, I'll try now. BTW: him ;) |
| 21:56.35 | *** join/#asterisk goodjoke (i=1827a8fa@gateway/web/ajax/mibbit.com/x-e243ec5444f67a5d) |
| 21:59.59 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 22:00.02 | feeds | so, I put 1001, 1002, 1000 into context employees, and the exten 10001 is in exten conf in context, employees, so now if I have include => employees in context internal, then employees can call all extensions in internal AND employees? |
| 22:01.13 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 22:01.51 | [TK]D-Fender | feeds: No |
| 22:01.54 | thedonvaughn | feeds: well, since 1001, 1002, and 100 use the employees context, you'd want to "include => internal" inside the [employees] context to make sure 100[0-2] have access to those definitions |
| 22:02.28 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 22:02.54 | [TK]D-Fender | feeds: You SIP devices have access to [employees]. [employees] would need to "include = internal", not the other way around. |
| 22:03.04 | Spirits-Sight | root52: so in my sip file I have the context for ipcomm (ITSP) and it points to mainmenu which is in my extision file, under that I have three simple lines " http://pastebin.com/m6a829f19 " and I get the error msg " http://pastebin.com/d3b183c13 " should this work the way it is |
| 22:03.10 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 22:04.02 | [TK]D-Fender | Spirits-Sight: and I answered this past night, enable SIP debug and go look at WHAT CONTEXT the call is falling, because it sure isn't the context that your exten there is in. |
| 22:04.33 | feeds | so for employees to access extens in internal, I have to write include => internal in employees? Not the other way around? |
| 22:05.10 | [TK]D-Fender | Spirits-Sight: And it does assume that is even the exten your provider is targiing when they send you the call |
| 22:05.37 | thedonvaughn | feeds: correct. your SIP devices only have access to [employees] they don't know about [internal]. if you "include = internal" inside of [employee] they do. |
| 22:05.52 | feeds | YAY! |
| 22:06.13 | [TK]D-Fender | feeds: think of it as "Joe knows Mary". So does Mary know Joe? NO. |
| 22:06.31 | thedonvaughn | feeds: if you have any sip devices who's context is [internal] and you want them access to [employees] THEN you'd need "include = employees" inside of [internal]. make sense now? |
| 22:06.33 | feeds | now I can go to sleep! Thanks a lot thedonvaughn and [TK]D-Fender . Know I understand! |
| 22:06.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:06.49 | thedonvaughn | cool |
| 22:07.51 | [TK]D-Fender | targetting* |
| 22:08.13 | feeds | [TK]D-Fender: who? :D |
| 22:08.29 | [TK]D-Fender | feeds: correcting previous typo |
| 22:08.39 | feeds | ahh :D |
| 22:09.00 | feeds | then bye |
| 22:09.06 | *** join/#asterisk ManxPower (n=manxpowe@14.sub-75-249-239.myvzw.com) |
| 22:09.20 | [TK]D-Fender | Spirits-Sight: and please note [Nov 21 16:57:40] NOTICE[26798]: chan_sip.c:14035 handle_request_invite: Call from 'ipcomm' to extension '4017534951' rejected because extension not found. <--- this is not looking for "s" <- |
| 22:09.40 | [TK]D-Fender | Spirits-Sight: Doesn't matter where you put that exten, this call is looking for an exten to mtach that NUMBER |
| 22:11.20 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 22:11.38 | ManxPower | [TK]D-Fender: Sometimes I really think the people that claim "s" is some sort of catchall be taken out back and shot. |
| 22:11.49 | ManxPower | Same for using "r" |
| 22:14.29 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 22:15.41 | thedonvaughn | yup, 'start' != 'catch all'. you still have to specifically call extension 's' within a context, or atleast call that context period. |
| 22:15.47 | Spirits-Sight | here is the sip area and the whole ext file, the context looks the same to me and I don't see where ipcomm would be looking for ext xxxxxxxxxx |
| 22:16.14 | [TK]D-Fender | thedonvaughn: No such thing as "call that context period" |
| 22:16.22 | thedonvaughn | [TK]D-Fender: well Goto |
| 22:16.25 | thedonvaughn | bad choice of words |
| 22:16.51 | [TK]D-Fender | Spirits-Sight: and please note [Nov 21 16:57:40] NOTICE[26798]: chan_sip.c:14035 handle_request_invite: Call from 'ipcomm' to extension '4017534951' rejected because extension not found. <--- You don't see them asking for htat NUMBER? I do. Its right here. |
| 22:16.54 | thedonvaughn | i.e. if you go to a context you don't need to specify s, it'll look for s by default is what i meant. |
| 22:17.21 | Spirits-Sight | So you do need to tell it a extion for the phone number DID to point it to |
| 22:17.39 | [TK]D-Fender | Spirits-Sight: isn't this # the DID you are paying them for? |
| 22:17.44 | thedonvaughn | Spirits-Sight: you want to match 4017534951 and so something with it yes. |
| 22:17.51 | thedonvaughn | Spirits-Sight: many ways to do this |
| 22:18.02 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 22:18.38 | Spirits-Sight | I want a nice and easy way to undersand, so I may learn from it and not get confused :-) I want a good way to do it |
| 22:19.24 | [TK]D-Fender | Spirits-Sight: They are dialing a number. You are showing me an exten with "s" in it. You do not have a match for what they are dialing IN to you. |
| 22:20.27 | Spirits-Sight | [TK]D-Fender: no its a ipcomm free account, using for playing with incoming calls |
| 22:20.49 | [TK]D-Fender | Spirits-Sight: FINE, forget "paying" is that not a NUMEBR they provide to you?> |
| 22:21.43 | Spirits-Sight | yes, so I would want to get rid of the s and put the number there like I had before changing it to s |
| 22:21.52 | *** join/#asterisk ManxPower (n=manxpowe@198.sub-75-201-243.myvzw.com) |
| 22:21.55 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 22:22.07 | thedonvaughn | Spirits-Sight: yup, change s to 4017534951 |
| 22:23.04 | [TK]D-Fender | Spirits-Sight: to accept the call you need a match. |
| 22:23.17 | [TK]D-Fender | Spirits-Sight: "s" certainly doesn't |
| 22:23.24 | Spirits-Sight | this is the only way to do this, I ask this as once I have switched to the real number then I would have to go in change it, now do I have to put the number on all the menu lines and submenu? |
| 22:23.33 | ManxPower | Is Spirits-Sight just dense? You guys have been giving him the solution for 5 mins. |
| 22:23.45 | Spirits-Sight | [TK]D-Fender: I understand that and should of know better with out changing what I had already |
| 22:24.17 | plasmid | what can I do to increase bandwith to my dedicated asterisk box at home for VOIP? so that voices don't sound robotic? codec changed to G.729 |
| 22:24.30 | thedonvaughn | Spirits-Sight: not necessarily. You could just do something like exten = 4017534951,1,Goto(menu|s|1) then it'd go through your [menu] context and you can use exten s to play your menu |
| 22:24.36 | thedonvaughn | Spirits-Sight: you just aren't doing it right |
| 22:24.52 | ManxPower | Most people have calls from untrusted sources (PSTN, ITSP, etc) land in a separate context. The extension lines in that context would use a Goto to send the call to the right context. |
| 22:25.02 | Spirits-Sight | ManxPower: Sorry I don't process as fast as you! |
| 22:25.06 | ManxPower | plasmid: upgrade your internet connection |
| 22:25.29 | plasmid | ManxPower 6 megs down 2 megs up... More?? |
| 22:25.40 | plasmid | thinks that ought to be enough. |
| 22:25.41 | [TK]D-Fender | plasmid: Did it improve on your G.729 passthrough test? |
| 22:25.51 | plasmid | [TK]D-Fender oh yes it did. :-) |
| 22:25.52 | ManxPower | plasmid: Sounds like plenty to me, but you asked how to get more bandwidth. That is the answer |
| 22:25.53 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 22:26.02 | [TK]D-Fender | plasmid: there you have it then. |
| 22:26.08 | ManxPower | I suspect some OTHER issue is at work here. |
| 22:26.12 | [TK]D-Fender | plasmid: bandwidth issue |
| 22:26.25 | plasmid | [TK]D-Fender sounds like it... hmm.. comcast it is. |
| 22:26.38 | [TK]D-Fender | ManxPower: ULAW was choppy, G.729 was not. Signs off as BW issue |
| 22:26.52 | Spirits-Sight | thedonvaughn: thanks, I understand what [TK]D-Fender about the when a incoming call comes in it has to match some thing, so now I know this for sure |
| 22:27.09 | plasmid | [TK]D-Fender BTW, thanks for your help. Greatly appreciated. WIll look for upload speed alternatives... |
| 22:27.12 | ManxPower | [TK]D-Fender: or an RTP packet size of 30ms, which would sound horrible unless there was passthru at work |
| 22:27.19 | thedonvaughn | Spirits-Sight: yup match it first, then send it to a menu with s extensions defined. |
| 22:27.22 | Spirits-Sight | what does the 1 in the Goto do? I understand the s as its the proity I believe |
| 22:27.34 | thedonvaughn | Spirits-Sight: s is extension 1 is the step |
| 22:27.40 | thedonvaughn | so s,1 |
| 22:27.40 | ManxPower | Goto(context,extension,priority) |
| 22:27.45 | thedonvaughn | or priority yah |
| 22:27.47 | thedonvaughn | i'm tired |
| 22:27.51 | [TK]D-Fender | ManxPower: * wouldn't accept it anyways IIRC... * only works on a fixed packet size |
| 22:28.00 | ManxPower | [TK]D-Fender: that is incorrect. |
| 22:28.09 | [TK]D-Fender | ManxPower: Last I checked it was so... |
| 22:28.17 | [TK]D-Fender | maxDid this change somewhere? |
| 22:28.17 | ManxPower | [TK]D-Fender: Some SPA units default to a 30ms RTP packet size. |
| 22:28.56 | beek | If anyone is interested, Level 3 communications has just risen to the top of my "World's shittiest Telco" |
| 22:29.23 | [TK]D-Fender | ManxPower: I'll have to look into that again.. I had serious problems with a Multitech gateway at my head office for RTP packet size mismatch (they set to 30 or 40 and * didn't like that |
| 22:31.23 | SkramX | can I make an extension that is just a dialplan for a trunk? |
| 22:31.35 | SkramX | *dialTONE |
| 22:32.06 | [TK]D-Fender | SkramX: HUH? |
| 22:33.21 | SkramX | i want it to just make the sound of a dialtone and let the user dial a number |
| 22:33.29 | SkramX | simulate like they just picked up the phone.. |
| 22:33.56 | Spirits-Sight | are the , converted to | in the exten => xx,x,xxxxx to xx|x|xxxxx just wondering |
| 22:36.07 | [TK]D-Fender | SkramX: You want the user to dial this exten just to get another dialtone? |
| 22:36.16 | [TK]D-Fender | Spirits-Sight: no commas |
| 22:36.25 | [TK]D-Fender | Spirits-Sight: (in your pattern) |
| 22:36.38 | [TK]D-Fender | Spirits-Sight: in CLI yes, they are |
| 22:36.53 | SkramX | that was the point but it's not critical |
| 22:36.56 | SkramX | (obviously) |
| 22:37.29 | *** join/#asterisk rdgr (n=rich@82-46-0-91.cable.ubr01.aztw.blueyonder.co.uk) |
| 22:39.39 | [TK]D-Fender | SkramX: "core show application disa" |
| 22:41.27 | SkramX | okay |
| 22:41.53 | SkramX | WARNING[3504]: app_festival.c:519 festival_exec: Festival returned ER :( |
| 22:47.18 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 22:48.29 | gammacoder | has anyone ever seen a PRI's audio totally fail, but the signaling work perfectly? I am guessing the B channels are screwed, but the D channel is working as expected. |
| 22:50.02 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 22:50.37 | Akiyuki | How can I make multiple contexts available to a sip peer? IE context=foo in sip.conf, but I would also like to make bar available. |
| 22:52.07 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 22:52.49 | [TK]D-Fender | Akiyuki: You make another context that includes foo & bar and point your device to that one |
| 22:53.24 | Akiyuki | Thanks [TK]D-Fender |
| 22:53.39 | Akiyuki | I was wondering if you could do like context=foo,bar,baz |
| 22:53.46 | Akiyuki | BUt your way makes more sense |
| 22:54.23 | [TK]D-Fender | Akiyuki: No, you cannot do it that way. |
| 22:55.14 | Akiyuki | Read a good article about dialplan sort order today :) |
| 22:55.29 | Akiyuki | After findingout that asterisk doesnt care what order you specify..hehe |
| 22:55.34 | lmadsen | aye |
| 22:57.02 | [TK]D-Fender | Akiyuki: a device has ONE context. all other call processing is based on what you put in that context. |
| 22:57.24 | Spirits-Sight | here is a question for you [TK]D-Fender, it it possible just using asterisk and ITSP to have when a person that is deaf calls in on a TDD to have asterisk reconize the sound and give a msg |
| 22:57.36 | Spirits-Sight | would this be hard to do? |
| 22:58.26 | [TK]D-Fender | Spirits-Sight: Yes. I'm sure TDD is a specific tone and one * won't know how to interpret. |
| 22:59.20 | Akiyuki | :D |
| 23:00.28 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 23:00.44 | shazaum | hi all guys |
| 23:00.55 | gambolputty | Hi Captain Marvel |
| 23:01.00 | Spirits-Sight | I am a little confused at the response, you say Yes. then at the end you say asterisk won't know how to interpret, so are you saying yes it can be done but your need to build a interpreter for it |
| 23:01.06 | shazaum | heuaheua |
| 23:01.53 | [TK]D-Fender | Spirits-Sight: You would need to opn up the source code and know intricate details about the tones & timings and then code how * would precess it (like how fax tones trigger the "fax" extension) |
| 23:03.03 | [TK]D-Fender | Spirits-Sight: but this requires real coding, though this is something very worthy of being adapted into trunk. |
| 23:03.08 | [TK]D-Fender | (SVN) |
| 23:03.51 | Spirits-Sight | well at less you think its worthy, I wonder if there is a add-on for this already |
| 23:04.14 | [TK]D-Fender | Spirits-Sight: No. It souldn't be an add-on.. this requires some core coding |
| 23:04.19 | [TK]D-Fender | wouldn't* |
| 23:05.09 | Spirits-Sight | I did a quick google and it says zap channel has a mode for TDD |
| 23:06.02 | Spirits-Sight | http://www.voip-info.org/wiki/view/tdd+mode |
| 23:06.20 | Spirits-Sight | so does this mean that it can do it already |
| 23:08.00 | [TK]D-Fender | Spirits-Sight: Interesting, but rather incomplete, and I'm not sure where that originates from. |
| 23:08.17 | [TK]D-Fender | Spirits-Sight: Could be relevant, but its poorly documented.... |
| 23:08.20 | Spirits-Sight | I don't know either but I would like to know more about this |
| 23:08.58 | [TK]D-Fender | Spirits-Sight: Continue your research.... |
| 23:09.04 | *** part/#asterisk Kvant (i=hh@unaffiliated/kvant) |
| 23:09.09 | Spirits-Sight | I will do so |
| 23:09.23 | Akiyuki | Can someone point me at documentation for working w/ mysql + asterisk? |
| 23:09.48 | Akiyuki | Not for sip.conf/extensions.conf dialplans/peers but for just querying and using the result as perhaps a name or telephone number |
| 23:11.02 | Akiyuki | So I can pipe in like 1,2,Festival(mysql select) |
| 23:14.32 | Spirits-Sight | do you know, if I call a toll free number on the voicepulse system does that take away from my account |
| 23:14.43 | [TK]D-Fender | Akiyuki: func_odbc in the BOOK. |
| 23:14.54 | [TK]D-Fender | Akiyuki: or MYSQL from asterisk -addons |
| 23:15.18 | [TK]D-Fender | Spirits-Sight: Ask them. |
| 23:15.27 | Spirits-Sight | they are now close |
| 23:15.37 | [TK]D-Fender | Spirits-Sight: Ask them LATER |
| 23:15.44 | Akiyuki | thanks |
| 23:17.15 | Spirits-Sight | I wanted to call a toll free number now but did not want to use it if I use about the money on that account, I don't wnat to use it intill its all setup |
| 23:19.45 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 23:19.45 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- SVN SHOULD BE BACK UP! -=- Asterisk 1.6.0.1 (2008/10/09), 1.4.22 (2008/10/02), *-Addons 1.6.0 (2008/10/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 23:20.35 | [TK]D-Fender | Akiyuki: When in doubt : AGI |
| 23:21.20 | Akiyuki | Ah, you have to escape the piss out of it. |
| 23:21.26 | Akiyuki | Ask Google Interface? :D |
| 23:21.36 | [TK]D-Fender | Akiyuki: or as I just pointed you towards... AGI |
| 23:21.41 | [TK]D-Fender | ~agi |
| 23:21.42 | jbot | i heard agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 23:21.51 | [TK]D-Fender | Akiyuki: Go read the book <- |
| 23:22.02 | Akiyuki | Its on back order. |
| 23:22.12 | [TK]D-Fender | Akiyuki: HTTP serves right NOW |
| 23:22.19 | [TK]D-Fender | ~book |
| 23:22.20 | jbot | rumour has it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:22.33 | Akiyuki | I dont want to read a 600 page pdf online.. I should print it out :P |
| 23:22.36 | [TK]D-Fender | Akiyuki: So don't give me any whiny excuses :p |
| 23:22.41 | Akiyuki | hehe |
| 23:22.48 | [TK]D-Fender | Akiyuki: And you don't have to, HTML is right there |
| 23:22.58 | Akiyuki | $48 at Barnes N Nobles, $23 at Amazon |
| 23:23.02 | Akiyuki | Ah, I didnt see the HTML one before. |
| 23:23.59 | Akiyuki | Does Festival have multiple voices? |
| 23:24.19 | Akiyuki | I want the female voice when you use SayDigits(2,f) to work all the time... the default guy in Festival sounds scary as shit |
| 23:27.31 | LeddyHM | is it possible to get a list of vm's in a mailbox with the manager api or do you have to rely on the filesystem (1.4) |
| 23:28.25 | *** join/#asterisk ManxPower (n=manxpowe@10.sub-75-249-138.myvzw.com) |
| 23:29.12 | [TK]D-Fender | LeddyHM: what level of detail for "list"? |
| 23:29.21 | *** join/#asterisk km2 (n=x@mobile-166-217-048-147.mycingular.net) |
| 23:30.51 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.108) |
| 23:31.22 | CrashHD | Hello |
| 23:31.27 | CrashHD | when using mixmonitor |
| 23:31.39 | CrashHD | what happens with the rtp stream with regard to jitterbuffer, etc? |
| 23:31.39 | LeddyHM | ultimately list all items, CID length, and listen to message |
| 23:34.32 | [TK]D-Fender | LeddyHM: look at the vmail.cgi script that comes with * for some "inspiration. |
| 23:34.34 | ManxPower | CrashHD: I believe it will be dejittered coming into Asterisk,. |
| 23:34.38 | [TK]D-Fender | LeddyHM: and ARI |
| 23:34.51 | ManxPower | Since Asterisk basically has to transcode for Monitor |
| 23:35.14 | CrashHD | so when the jb is forced, it is done priorer to splitting the streams and forwarding the traffic? |
| 23:35.34 | CrashHD | my issue is I have reports of garbled (electronic sounding speech) |
| 23:35.37 | ManxPower | CrashHD: I assume so, but I suspect the only way to know for sure is to look at the code. |
| 23:35.49 | CrashHD | the recording seems clean |
| 23:35.55 | LeddyHM | I think vmail.cgi uses filesystem for stuff I'll take a look though |
| 23:35.59 | LeddyHM | what's ari short for? |
| 23:36.06 | CrashHD | tests from the server to the phone are clean |
| 23:36.39 | [TK]D-Fender | LeddyHM: "Asterisk Recording Interface". I believe there was some VM integration concerning this |
| 23:36.41 | ManxPower | I doubt it has anything to do with Monitor. What happens if you put "t" or "T" on the Dial line. If that reproduces the issuse then chances are it's a packet size issue. |
| 23:36.49 | [TK]D-Fender | LeddyHM: 3rd party stuff |
| 23:36.57 | ManxPower | ARI is Asterisk REALTIME Interface |
| 23:37.06 | ManxPower | wait, maybe not. |
| 23:37.09 | [TK]D-Fender | ManxPower: multple acronyms... |
| 23:37.21 | ManxPower | perhaps some more coffee is in order |
| 23:37.26 | CrashHD | I believe one of the T/t's is being used on the line |
| 23:37.26 | [TK]D-Fender | ManxPower: nothing stops people from creating overlap you know :) |
| 23:37.35 | CrashHD | how would that relate to packet size? |
| 23:37.46 | [TK]D-Fender | ManxPower: But thanks for the coffee idea anyway :) |
| 23:38.21 | ManxPower | CrashHD: If one or more of the endpoints are using 30ms packet sizes nothing bad will happen if they are reinviting. Monitor and T/t and a few other options disable reinvites. |
| 23:38.21 | [TK]D-Fender | ManxPower: I'm more "conscious" asleep than most people are when they are awake :p |
| 23:38.47 | ManxPower | CrashHD: Are any of the endpoints SIPura boxes? |
| 23:38.59 | CrashHD | no, aastra 480i's. 20ms ulaw being used |
| 23:39.20 | *** join/#asterisk sasargen (n=chatzill@68.245.131.215) |
| 23:40.38 | ManxPower | CrashHD: still... trying t/T might give you one more datapoint |
| 23:40.39 | CrashHD | the audio that is digital/garbled is from the asterisk server to the end point |
| 23:40.49 | CrashHD | t/T is being used |
| 23:41.29 | ManxPower | Then you have a problem I've never seen before. i.e. garbled audio when MixMonitor is being used, but not if just T/t are used. |
| 23:41.46 | ManxPower | you realize that using T/t can open up your PBX to hacking, right? |
| 23:42.22 | ManxPower | You do not want random people that you call or call you to be able to transfer themselves out of an IVR or even a call. |
| 23:42.34 | CrashHD | yes |
| 23:42.49 | denon | but before you change it, call me :) |
| 23:43.00 | CrashHD | :) |
| 23:43.09 | CrashHD | what is the alternative to T/t |
| 23:43.12 | CrashHD | to allow transfers? |
| 23:43.52 | *** join/#asterisk ManxPower (n=manxpowe@97.sub-70-221-123.myvzw.com) |
| 23:44.23 | ManxPower | These random Verizon disconnects are starting to piss me off. |
| 23:46.56 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 23:47.37 | *** join/#asterisk Fairman (n=Fairman@c-76-105-10-247.hsd1.ca.comcast.net) |
| 23:47.52 | Fairman | hey guys... |
| 23:49.01 | [TK]D-Fender | CrashHD: Use REAL phones :) |
| 23:49.20 | Fairman | anybody have any luck w/ 1.6.0.1, meetme and dahdi_dummy? |
| 23:49.32 | CrashHD | hah tk |
| 23:49.40 | [TK]D-Fender | CrashHD: Whic... apparently you are. Go read your phone's manuals... you should never have been using DTMF transfers on those... |
| 23:49.47 | ManxPower | [TK]D-Fender: you're not on the mailinglists are you? |
| 23:50.05 | CrashHD | [TK]D-Fender: we use the transfers for some call parking functions right now |
| 23:50.06 | [TK]D-Fender | ManxPower: I'm sure they slander me constantly ;) |
| 23:50.25 | ManxPower | CrashHD: Use the NATIVE transfer features of your phone |
| 23:50.28 | [TK]D-Fender | CrashHD: Never needed EITHER |
| 23:50.58 | CrashHD | we have a dtmf inband application |
| 23:50.58 | ManxPower | [TK]D-Fender: not at all, but there were several people saying very unpleasant things about 1.4 and Digium with regards to bug reports being ignored, etc. |
| 23:51.07 | CrashHD | for one button call park to specific parking lot |
| 23:51.09 | CrashHD | *spot |
| 23:51.38 | ManxPower | [TK]D-Fender: CrashHD is never going to listen, why bother even talking to him. |
| 23:51.45 | CrashHD | I listen |
| 23:51.46 | [TK]D-Fender | CrashHD: nice way to chain your flaws together in super-flaws :p |
| 23:51.55 | CrashHD | but you can't do what we are doing with native transfers |
| 23:52.08 | CrashHD | so it is what it is |
| 23:52.17 | [TK]D-Fender | CrashHD: Keep investing down that route.... you're almost pasinted right into a corner... |
| 23:52.20 | ManxPower | CrashHD: My customers use native transfers ALL the time to park calls |
| 23:52.39 | [TK]D-Fender | ManxPower: He's doing 1-touch DTMF parks, etc. |
| 23:52.51 | [TK]D-Fender | ManxPower: His users turned him into a DTMF-whore :p |
| 23:52.52 | denon | you know, parking has been goofy for me lately |
| 23:52.54 | CrashHD | yes man but not one touch to specific spots, etc |
| 23:52.59 | denon | when I xfer a call to the lot, it doesnt play back the sounds |
| 23:52.59 | CrashHD | yes a whore I've become |
| 23:53.00 | ManxPower | [TK]D-Fender: the poor sod. |
| 23:53.02 | denon | nor does it whine on the console |
| 23:53.16 | denon | it just disconnects |
| 23:53.23 | [TK]D-Fender | denon: "whine=yes" |
| 23:53.28 | ManxPower | denon: use attended transfers 9-) |
| 23:53.29 | denon | heh heh |
| 23:53.38 | denon | ManxPower: of course, you dweeb |
| 23:53.46 | ManxPower | denon: hence the 9-) |
| 23:54.04 | denon | hehe |
| 23:54.06 | denon | never knwo |
| 23:54.10 | ManxPower | denon: how many years have you been using Asterisk anyway? |
| 23:54.27 | denon | um |
| 23:54.30 | denon | lots |
| 23:54.45 | denon | remember when it really sucked and was barely usable |
| 23:54.49 | Spirits-Sight | what can I do to have a menu play over and over till the person press a option, this is with a few sec delay between playings, I think this would be a waitexten() and a Goto() is this right and if not how or is there a better way to handle it |
| 23:54.53 | denon | well, ok, that was last week .. but .. before that |
| 23:54.54 | ManxPower | I've been using it since late 2001, using it in production since mid 2002 |
| 23:54.56 | denon | grins |
| 23:55.33 | denon | I remember all the headaches of having to switch everything from IAX1 to 2 |
| 23:55.40 | ManxPower | Spirits-Sight: there were no examples of that in extensions.conf.sample |
| 23:55.58 | denon | that was a pain |
| 23:56.05 | ManxPower | denon: I remember all the great things when we switched from IAX2 to SIP. 8-) |
| 23:56.26 | denon | hehe |
| 23:56.36 | denon | so that's once sip started to suck less, you mean |
| 23:57.12 | ManxPower | denon: All I know is I stopped getting dropped call reports 10 times a day. |
| 23:57.19 | denon | I remember wishing MGCP existed .. and now wishing it didnt |