| 00:00.44 | Sephen | CunningPike: But how would I catch a pattern that didn't exist in the Goto-Context? |
| 00:01.00 | distortion | you build all the routes as NXXNXXXXXX |
| 00:01.34 | distortion | so you have the goto catch anything wiht "1" and "9" and strips it, then sends it to the NXXNXXXXXX list |
| 00:03.12 | Zodiacal | anyone know how asterisk can connect to a fire alarm, or can it at all? |
| 00:03.16 | Sephen | distortion: I'm sorry if I'm making this more difficult than it is (maybe), but, I'd want to match a 10 digit dial regardless if it had a 1, a 9, or a 91 prefixed. |
| 00:03.55 | mds2 | has anyone managed to get a Linksys SIP9000 to register with their Asterisk server? I can see inbound SIP REGISTER requests in a tcpdump but asterisk is ignoring them, nothing showing up in a 'sip debug' either. any ideas? |
| 00:04.42 | distortion | Stephen: 3 goto extensions would be setup to strip "1" and "9" and "91" they would send the call then to the list of exten => _NXXNXX.,1,Dial entries |
| 00:05.15 | distortion | that way you have 1 list of 590 exten => _NXXNXX.,1,Dial(blah) and 3 goto extensions that point to this list that strip off teh leading digs |
| 00:05.36 | Sephen | distortion: What about fallback though? If I strip those digits, and then send it to this new context, and it doesn't match? How do I handle that? |
| 00:06.07 | distortion | add a final catchall context: exten => _1NXXNXXNXXXX,1,Dial(Long distance) |
| 00:06.09 | Sephen | do I just add an extension of 'i' with a Goto of Default? |
| 00:06.14 | distortion | err not context, extension |
| 00:06.56 | distortion | sorry, in your case it would be: exten => _NXXNXXXXXX,1,Dial(Long distance) |
| 00:07.47 | Sephen | distortion: Another twister: We use the Switch/Realtime feature in Asterisk.. Won't that be a problem, since its not like a flat config file, read from beginning to end until it finds a match? |
| 00:08.06 | distortion | it works- i actually just re-wrote my dialplan like this yesterday so that I ALWAYS get 1+10 digits stored in the DST field of cdrs regardless of if my customers send calls with a tech prefix on the front of the number |
| 00:08.18 | distortion | (for billing purposes) |
| 00:09.39 | TripleFFFFFFFFFF | can one please sned me a test fax |
| 00:09.57 | TripleFFFFFFFFFF | 8008548957 |
| 00:10.00 | distortion | not 100% sure, but the concepts should be the same with realtime |
| 00:10.01 | TripleFFFFFFFFFF | okease.. form USA |
| 00:11.04 | *** join/#asterisk stevedl (n=steve_dl@eth87.tas.adsl.internode.on.net) |
| 00:11.59 | Sephen | distortion: I'll give it a try and see if it'll match a more specific match before it matches a more generic one. Thanks for your time. |
| 00:12.28 | distortion | np. |
| 00:14.54 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:15.36 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:15.38 | TripleFFFFFFFFFF | i guess not |
| 00:16.33 | Zodiacal | can asterisk handle alarm systems? |
| 00:16.41 | Zodiacal | i.e. i guess just route the calls |
| 00:16.50 | Zodiacal | would i put the alarm system on a Fxo? |
| 00:16.53 | Zodiacal | err fxs |
| 00:17.09 | Zodiacal | do they need to call in? |
| 00:22.08 | TripleFFFFFFFFFF | okease |
| 00:22.09 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
| 00:22.13 | *** join/#asterisk oadaeh (n=jason@las-cust-208.57.199.83.mpowercom.net) |
| 00:23.03 | CunningPike | Zodiacal: It might work, it might not - some systems are very sensitive to tones |
| 00:23.19 | Manipura | how do I start asterisk in the background? Even if I leave the -c out I still can't Exit the ssh without asterisk shutting down |
| 00:23.32 | *** part/#asterisk oadaeh (n=jason@las-cust-208.57.199.83.mpowercom.net) |
| 00:25.08 | Manipura | asterisk -vvvvvv brings me into the console |
| 00:28.07 | harryvv | Other then ManxPower who else knows the ip500 in a out? |
| 00:29.39 | harryvv | somehow my ip500 is perm stuck in call forwarding |
| 00:31.00 | distortion | manipura, use the safe_asterisk script or issue a "make config" to install asterisk as a startup service |
| 00:31.22 | distortion | then you can start it (in red hat) with "service asterisk start" |
| 00:31.54 | *** join/#asterisk [tasty]freeze (n=simba@pm3.5-20.wctc.net) |
| 00:31.54 | babyju | I know the voip wiki list a couple of good 3rd party web gui's for asterisk...there is no ratings to go by so can anyone give me a recommendation? I wish not to install and try all the available products. |
| 00:31.55 | *** part/#asterisk stevedl (n=steve_dl@eth87.tas.adsl.internode.on.net) |
| 00:32.15 | [tasty]freeze | Does asterisk support an address book sort of feature that is accessable thru a SIP phone? |
| 00:32.36 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 00:32.49 | [tasty]freeze | for like a list of extensions... I'm sorry I feel like I shouldnt ask this question here, but I am unable to find it in the voip-info... |
| 00:34.44 | Qwell[] | babyju: they all rate about 3 of 10, tops |
| 00:36.50 | distortion | qwell: you familiar with codec negotiation between endpoints through *? |
| 00:37.40 | distortion | ie: sipura (Codec 1. g729, 2. g711) -> asterisk (allow=g729,ulaw) -> endpoint b (allow=ulaw) forces asterisk to try and transcode (not good) |
| 00:37.43 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 00:37.55 | babyju | Qwell[], are you saying there isn't one good gui out there? |
| 00:37.58 | harryvv | dammit, fixed my call forwarding problem. |
| 00:38.58 | harryvv | :) |
| 00:39.46 | Qwell[] | babyju: basically, yes |
| 00:39.52 | Qwell[] | GUI config tools are inherently bad |
| 00:40.11 | CunningPike | [tasty]freeze: Directory application |
| 00:40.56 | [tasty]freeze | thanks a lot CunningPike |
| 00:55.20 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 00:57.27 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 01:00.16 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:00.58 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-38-79.dsl.tul2ok.sbcglobal.net) |
| 01:01.24 | ManxPower | hello, hardwire |
| 01:01.26 | ManxPower | ..er.. harryvv |
| 01:01.49 | paolob | Guys, I have to sip phones connected to the same asteriks, in sip.conf their configuration is perfectly equivalent, in particular the context is the same. Nevertheless, calling from the two phones gets different result. How is it possible? |
| 01:02.02 | ManxPower | harryvv, what was the cause of the problem>? |
| 01:02.38 | paolob | well, one is a phone connected to a pap2, the other is ekiga |
| 01:02.55 | ManxPower | paolob, in [general] in sip.conf put context=INVALID Make each device config has a context=whateveryouwant in it. |
| 01:02.59 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 01:03.08 | ManxPower | if one phone can no longer call then it's userid/password does not match what is in sip.conf |
| 01:04.33 | paolob | ManxPower, I had context=default in sip.conf, [general], and context=default in the definition of the sip phones... |
| 01:04.51 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 01:05.03 | ManxPower | paolob, then put context=INVALID in [general] |
| 01:05.33 | harryvv | manx, some how the xml configuration was changes on this phone. I found a link that sugested to make changes in reg.htm and that did it. Now, this is a old issue and its something that irks the other person who uses the phone. A incomming caller will ring the other extention when she is already on the phone and she does not know who it is since the cid display does not show who the second calling party is. This is setup on a sipura ata |
| 01:05.39 | ManxPower | In many configurations Asterisk will accept calls from unauthenticated devices, this will send those calls to a context that is not valid and make asterisk reject the calls from unauthenticated devices |
| 01:05.41 | paolob | ManxPower, I put it, but it gives:" pbx_extension_helper: Cannot find extension context 'INVALID'" |
| 01:06.13 | ManxPower | paolob, that means that device is NOT matching the sip.conf entry for that device. |
| 01:06.16 | ManxPower | and there is your problem |
| 01:06.35 | harryvv | Manx, there is no reason this change should have occured. |
| 01:06.43 | harryvv | But tis fixed anyway |
| 01:06.45 | *** join/#asterisk Ixthod (n=Ixthod@198.174.206.41) |
| 01:06.45 | paolob | ManxPower, no, I put context=INVALID in the [general] section |
| 01:07.26 | ManxPower | paolob, correct. The ONLY time you will get " pbx_extension_helper: Cannot find extension context 'INVALID'" is when a call comes in from a device that does NOT authenticate with an entry in sip.conf |
| 01:07.44 | harryvv | madrid released one of those train bombers by mistake but wont put him back in jail..that makes a hell of a lot of sence. |
| 01:07.52 | ManxPower | so either your device configuration is wrong or the sip.conf entry for that device is wrong. |
| 01:07.59 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:08.44 | docelm0 | Anyone know what the redial key is on the GXP2000? |
| 01:09.38 | justinu | press send |
| 01:09.48 | harryvv | Manx, do you know if a ipphone thats not so nat sensitive that it can pass sip traffic though a firwall? |
| 01:10.00 | docelm0 | really? cool |
| 01:10.11 | docelm0 | works thanks J |
| 01:10.26 | ManxPower | harryvv, Um, any IP phone I've used has gone thru firewalls just fine. |
| 01:10.55 | ManxPower | Those would be Cisco, SIPura, and Polycom |
| 01:12.05 | paolob | Maxxed, that's very strange, because sip show peers tells me that ekiga has registered... |
| 01:12.19 | ManxPower | Heck, my SIPura ATA was able to move between the internal network (Asterisk behind NAT), to a public IP to another NAT network with no problems at all. |
| 01:12.26 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 01:13.05 | ManxPower | Frequently a phone will use different userid/pass for registration and for calls |
| 01:14.52 | harryvv | ManxPower Im talking not with the assistance of a asterisk box. Just plug it into some office network and if the phones setup right it should pass though to the outside world. |
| 01:17.55 | paolob | ManxPower, it's very strange that I get context INVALID, because sip show peers tells me that ekiga has registered... |
| 01:20.03 | ManxPower | paolob, registration is ONLY for calls Asterisk -> SIP device. Registration has NOTHING to do with SIP Device -> Asterisk calls |
| 01:21.56 | paolob | ManxPower, but, what's the reason why ekiga registers, but call from ekiga aren't recognized? ekiga registers as dirbasica-e, and I do have a [dirbasica-e] section in sip.conf, with context=default |
| 01:22.39 | paolob | ManxPower, what else do I need in order to get incoming calls from ekiga be considered? |
| 01:30.16 | paolob | Guys, how do I set the language for the messages played back for incoming calls? |
| 01:33.00 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-38-79.dsl.tul2ok.sbcglobal.net) |
| 01:34.13 | paolob | Guys, I have a pap2 and a ekiga softphone, either are defined the same way in sip.conf, but when I call a menu extension, from the pap2 it gets the messages (language=es), but from ekiga it doesn't. Any hint? |
| 01:41.51 | De_Mon | SET(${ARRAY(var1,var2)=${CUT(var|\,|)}) |
| 01:41.54 | De_Mon | is that right? |
| 01:44.58 | paolob | Guys, I have a pap2 and a ekiga softphone, either are defined the same way in sip.conf, but when I call a menu extension, from the pap2 it gets the messages (language=es), but from ekiga it doesn't. Any hint? |
| 01:49.00 | paolob | Guys, what a reason why asterisk treats differently the calls coming from a pap2 and those coming from ekiga? Calls coming from a pap2 are answered in a different context than those coming from ekiga. Why? |
| 01:56.44 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 01:57.20 | *** join/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 02:00.41 | *** part/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 02:01.12 | De_Mon | <PROTECTED> |
| 02:01.12 | De_Mon | <PROTECTED> |
| 02:01.33 | De_Mon | it's not setting the 2nd variable :( |
| 02:04.22 | De_Mon | exten => 3137,n,SET(ARRAY(index,ticketN)="0,12%") |
| 02:04.22 | De_Mon | exten => 3137,n,NoOp(${index}-${ticketN}) |
| 02:05.22 | *** join/#asterisk astermick (n=mtur2848@CPE-60-231-112-137.qld.bigpond.net.au) |
| 02:05.47 | astermick | hiya's |
| 02:05.50 | De_Mon | hoi |
| 02:06.06 | astermick | question if you dont mind |
| 02:07.23 | astermick | setting for number of rings? |
| 02:08.12 | De_Mon | number of rings? you can set a timeout for Dial(), a ring happens every... 5 seconds? |
| 02:08.53 | astermick | about 5 secs yes. |
| 02:09.02 | astermick | so its a timeout value |
| 02:09.19 | astermick | extensions or sip conf ? |
| 02:09.31 | De_Mon | extensions |
| 02:09.46 | astermick | your a gentleman and a scholar, thankyou! |
| 02:10.09 | De_Mon | have a good evening and thankyou for trying asterisk(tm) |
| 02:10.13 | De_Mon | :D |
| 02:10.23 | pjchilds | astermick: do you mean inbound or outbound calls? |
| 02:10.30 | astermick | outbound |
| 02:10.51 | astermick | still same answer? |
| 02:11.11 | pjchilds | yes .. http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 02:11.23 | astermick | good stuff *thumbs up* |
| 02:11.25 | pjchilds | Dial(type/identifier, timeout, options, URL) |
| 02:12.32 | astermick | ah, so its the 20 value in the exten => values |
| 02:12.39 | astermick | rings x 5 secs = 20 |
| 02:12.43 | astermick | 4 rings* |
| 02:13.51 | De_Mon | I need an example of func_odbc using CUT and ARRAY this is being obnoxous |
| 02:14.38 | tainted- | anyone know where 483 'Too Many Hops' stems from? |
| 02:15.43 | *** join/#asterisk watchy (n=watchy@h236.176.255.206.cable.cmdn.cablelynx.com) |
| 02:15.43 | pjchilds | from your service provider? |
| 02:15.48 | tainted- | yea |
| 02:15.59 | watchy | zap show status shows all working lines or what? |
| 02:16.02 | De_Mon | hrm hang on, found a different bug |
| 02:16.22 | watchy | any command to show that all have dialtone? |
| 02:16.55 | astermick | exten => _8888X.,4,Dial(SIP/${EXTEN},60,tT) |
| 02:17.01 | astermick | the 60 is the timeout? |
| 02:17.03 | pjchilds | tainted-: typically a default SER type config will return a 483 if the number of forward headers >= 10... |
| 02:17.09 | gandhijee | anyone know if there is away to make the polycoms default to numeric input on a text field? |
| 02:17.39 | tainted- | pjchilds how can i prevent that from happening? where are the fwd headers coming from? |
| 02:18.31 | pjchilds | tainted-: who knows... 'sip debug' should show where they come from .... |
| 02:18.51 | pjchilds | astermick: yup... |
| 02:18.56 | astermick | thx =) |
| 02:23.54 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 02:23.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:24.11 | astermick | finally, lunch break. Thx pjchilds and De_Mon, worked fine. |
| 02:24.14 | astermick | afk |
| 02:25.23 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:30.37 | De_Mon | uhhh cut is a built-in function in 1.2 isn't it? |
| 02:30.39 | gandhijee | anyone know if there is away to make the polycoms default to numeric input on a text field? |
| 02:30.44 | De_Mon | show functions doen't list it? |
| 02:32.13 | De_Mon | erm, they 'funcion' cut and sort are provided by 'app_cut.so' ? |
| 02:34.09 | *** join/#asterisk Altair256 (n=Altair25@mail.clccorp.com) |
| 02:34.29 | De_Mon | ok, app_cut loads funcions and applications... nevermind |
| 02:36.49 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:37.50 | *** join/#asterisk The_Isle_of_Mark (n=mark@c-68-85-63-96.hsd1.ga.comcast.net) |
| 02:37.54 | The_Isle_of_Mark | lo all |
| 02:39.50 | The_Isle_of_Mark | quick question if you don't mind: I have my extensions.conf setup with a very simple exten => s,1,Dial(SIP/200) |
| 02:39.50 | The_Isle_of_Mark | <PROTECTED> |
| 02:42.37 | The_Isle_of_Mark | during testing I tried as the asterisk book I downloaded suggested and answer() and playback(hello-world) I would just hear the end of the hello-world something like "rld" on the incoming line |
| 02:43.04 | The_Isle_of_Mark | it seemed to answer and playback before the connection was fully bridged |
| 02:43.46 | The_Isle_of_Mark | I think these are somehow related. any input? |
| 02:44.03 | harryvv | Anyone ANYONE have a idea how i can get a telus incomming call cwcid to goto voicemail and show WHO is calling when I am on the phone talking to another party? I just perhaps missed a important call and cannot have this happen anymore. I am running asterisk with polycom ip500. Same happens for the sipura ata so I suspect this is a astrisk config issue? |
| 02:44.22 | De_Mon | so you have s,1,Answer() then s,2,Playback(hello-world) |
| 02:45.19 | The_Isle_of_Mark | De_Mon, yes |
| 02:45.26 | The_Isle_of_Mark | De_Mon, that was for testing |
| 02:45.40 | The_Isle_of_Mark | now it is exten => s,1,Dial(SIP/200) |
| 02:49.18 | *** join/#asterisk kimosabe (n=kimosabe@dsl-200-78-71-61.prod-infinitum.com.mx) |
| 02:49.49 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 02:50.35 | kimosabe | i need an administrator to post this i have solved Qos isues easy effective and cheap |
| 02:51.11 | tainted- | kimosabe is it called LAN? |
| 02:51.15 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:51.34 | The_Isle_of_Mark | oc-192 heheh |
| 02:51.40 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 02:51.53 | De_Mon | well, that fits the cheap and easy part |
| 02:52.05 | kimosabe | no it is choppy voice isues over dsl lines with user thgat like 2 download music and stuff |
| 02:52.22 | kimosabe | there a new switch that just came out |
| 02:52.32 | The_Isle_of_Mark | De_Mon, any info on that time issue? |
| 02:52.46 | kimosabe | encore enh908-nwy voip priority |
| 02:52.55 | De_Mon | The_Isle_of_Mark add a Wait(1) and see if that helps |
| 02:53.07 | Agrajag- | gday. i can't figure out how to configure SIP users so that if someone calls them and they're not registered, instead of getting 'user rejected the call', i get 'user not online'? at the moment when i dial a SIP user that isn't registered, asterisk tells me "Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 02:53.07 | kimosabe | if there is some real pros here that run many lans this can help u i know it helps me |
| 02:53.28 | De_Mon | kimosabe is that some how different from all the other qos routers already available? |
| 02:54.08 | kimosabe | yes becuase its 40 dlls cheaper and needs no config |
| 02:54.15 | kimosabe | the switch cost 13 dlls |
| 02:54.25 | kimosabe | it has one voice ppriority port |
| 02:54.56 | kimosabe | how much more difrent can u ask for |
| 02:55.25 | The_Isle_of_Mark | De_Mon, ok that fully solves the playback or background issue |
| 02:55.40 | The_Isle_of_Mark | De_Mon, how about the incoming call time to ring the extension? |
| 02:55.40 | tainted- | kimosabe link? |
| 02:56.01 | kimosabe | im purchasing it here in mexico in a shop but search for it by name |
| 02:56.17 | kimosabe | enh908-nwy voip |
| 02:56.19 | De_Mon | lastlog voip-info 5 |
| 02:56.27 | De_Mon | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 02:56.35 | De_Mon | The_Isle_of_Mark: set a timeout |
| 02:56.37 | The_Isle_of_Mark | De_Mon, yeah checked that out |
| 02:56.49 | The_Isle_of_Mark | I'll check it again..thanks |
| 02:58.09 | The_Isle_of_Mark | De_Mon, I am not haveing a timeout problem. My problem is that the extension takes about 5 seconds to ring...meanwhile the calling party hears 2-3 rings and might hangup |
| 02:59.08 | De_Mon | oh.. is the phone over dialup or wireless by chance? |
| 02:59.18 | De_Mon | maybe on the other side of the world? |
| 02:59.28 | The_Isle_of_Mark | De_Mon, negative. Sip ata |
| 02:59.38 | The_Isle_of_Mark | De_Mon, right in the same room :) |
| 02:59.51 | De_Mon | try a different ata? |
| 03:00.03 | The_Isle_of_Mark | De_Mon, yep, tried 3 |
| 03:00.32 | De_Mon | hrm, shrug |
| 03:00.45 | *** join/#asterisk camelon (n=chiardon@201.228.4.85) |
| 03:01.11 | camelon | Hi everyone |
| 03:01.43 | camelon | I have a recent problem with my *Box |
| 03:02.26 | camelon | my instalation:2E1s . . . 2channel bank |
| 03:03.27 | camelon | 30% of the calls to my PBXs give to the caller the busy signal . . some idea what happen? |
| 03:03.58 | X-Rob | The_Isle_of_Mark, I guess caller ID detection timeout. |
| 03:04.10 | camelon | the same when the people isdoing calling out . . .20-30% times de busy signal |
| 03:04.34 | The_Isle_of_Mark | X-Rob, ok...makes sense. any way to shut it off? |
| 03:04.39 | X-Rob | depends on the ATA |
| 03:04.42 | dlynes_ | Does anyone know if there's a keyword for zapata channels to hold off on answering the line for a certain period of time? |
| 03:04.44 | X-Rob | I suggest RTFM 8) |
| 03:05.05 | The_Isle_of_Mark | X-Rob, ok...I didn't realize it was at the ATA level not the * level |
| 03:05.06 | dlynes_ | i.e. before it gets to the dial plan ? |
| 03:05.21 | The_Isle_of_Mark | X-Rob, thanks, I'll see if I can disable it in the ATA |
| 03:05.58 | X-Rob | dlynes try 'start=10000' on the channel |
| 03:06.03 | X-Rob | I'm not sure if that's it, but it might be 8) |
| 03:06.30 | dlynes_ | is there such an option? |
| 03:06.46 | X-Rob | dlynes_, have you even _read_ zapata.conf? |
| 03:07.07 | camelon | x-rob . . .could you help me? |
| 03:07.12 | X-Rob | camelon, nope. |
| 03:07.17 | dlynes_ | X-Rob: yes, i have |
| 03:07.25 | dlynes_ | X-Rob: there's no start option in the sample one |
| 03:07.43 | X-Rob | Dlynes_ - funny. There is in mine. |
| 03:07.52 | X-Rob | lin 201 |
| 03:07.54 | dlynes_ | X-Rob: nor is there a start option on the zapata.conf wiki |
| 03:07.58 | harryvv | dlyne call forwarding problem solved |
| 03:08.02 | dlynes_ | X-Rob: are you using 1.2.7.1? |
| 03:08.09 | dlynes_ | harryvv: so what was your problem, then? |
| 03:08.18 | X-Rob | branches/1.2, which is close enough to 1.2.7.1 |
| 03:08.28 | dlynes_ | weird...maybe something different in there, then |
| 03:09.07 | dlynes_ | does it explain what the value after the equals sign is then? |
| 03:09.07 | X-Rob | grep start: /usr/src/asterisk/configs/zapata.conf.sample |
| 03:09.07 | X-Rob | ; start: Start time (default 1500ms) |
| 03:09.07 | X-Rob | [root@wpm4l-gw ~]# |
| 03:09.07 | dlynes_ | i.e. seconds, milliseconds, ...? |
| 03:09.44 | harryvv | dlynes, well i NEVEr made the changes but some how call forwarding in the configuration was made. |
| 03:09.53 | dlynes_ | oh...in the timing parameters for t1's |
| 03:09.55 | harryvv | in the phones configuration. |
| 03:10.01 | dlynes_ | but i'm using an fxo port, not a t1 |
| 03:10.25 | dlynes_ | I was searching for start=, not start:...that's why i didn't find it |
| 03:10.57 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 03:11.11 | harryvv | Now anyone here using a polycom ip500 and use pstn incomming line with cwcid and it does/does not work? I can be on the line and a second call with cw would beep the hand set. I dont have the ability to flash over to second caller or have them goto voice mail. Is there a way to do that with this setup? |
| 03:11.13 | dlynes_ | I'll try it, and see if it works anyways |
| 03:11.17 | dlynes_ | maybe i might get lucky |
| 03:11.47 | camelon | someone giving a hand? |
| 03:12.32 | The_Isle_of_Mark | X-Rob, The ATA timeout is not the problem. I have verbosity at 9 and when a call comes in * says: Starting simple switch on 'Zap/1-1' |
| 03:12.32 | The_Isle_of_Mark | <PROTECTED> |
| 03:12.49 | camelon | a weird problem with my *Box |
| 03:13.06 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 03:13.27 | X-Rob | The_Isle_of_Mark, sorry, I misunderstood. |
| 03:13.35 | X-Rob | but that definately _is_ caller id |
| 03:14.09 | X-Rob | put 'usecallerid=no' in zapata.conf |
| 03:14.42 | The_Isle_of_Mark | X-Rob, did that |
| 03:14.47 | The_Isle_of_Mark | X-Rob, not dice |
| 03:15.54 | The_Isle_of_Mark | X-Rob, s/not/no |
| 03:16.19 | harryvv | Why is my cwcid not working when talking with somone on one call and it does not show on second incomming call? |
| 03:17.15 | X-Rob | The_Isle_of_Mark, *shrug*. |
| 03:18.03 | The_Isle_of_Mark | X-Rob, would hardware speed be a problem? |
| 03:18.10 | *** join/#asterisk \etc\bin (n=root@58.71.13.194) |
| 03:18.40 | X-Rob | The_Isle_of_Mark, possibly... Could be many things (eg, a Wait(3) in the dialplan? 8) |
| 03:18.58 | dlynes_ | harryvv: is your callwaitingcallerid in your zapata.conf set to yes? |
| 03:19.18 | The_Isle_of_Mark | X-Rob, nah...very basic dialplan...ah well tomorrow is another day...thanks all for the help |
| 03:19.21 | gandhijee | anyone know if there is away to make the polycoms default to numeric input on a text field? |
| 03:20.08 | kimosabe | has nyone got cisco to register with asterisk ciasco router |
| 03:20.13 | kimosabe | cisco |
| 03:20.24 | Hmmhesays | unless you die tonight |
| 03:20.56 | astermick | or get him by a bus |
| 03:20.59 | harryvv | yes, callwaitingcallerid=yes |
| 03:21.00 | astermick | hit |
| 03:21.10 | Hmmhesays | i've never been "him'd" before |
| 03:21.12 | Hmmhesays | does it hurt? |
| 03:21.18 | astermick | lol sigh |
| 03:21.46 | kimosabe | is that for me becuase i got it to send me messages but it registered only couldnt use it |
| 03:22.21 | harryvv | dlynes The phone reciver will do the typical tone stating there is another caller calling in when im on the phone with somone else. It does not allow the ability to flash the hand set so...what do i do in this case? |
| 03:22.49 | harryvv | It also does not show the callerid when im on the phone with somone else. |
| 03:23.07 | harryvv | same with the sipura ata and other phone. |
| 03:23.15 | dlynes_ | harryvv: oh...no idea...i always disable callwaiting with asterisk in the mix because i dont' feel like trying to deal with it :) |
| 03:23.28 | dlynes_ | harryvv: all of my customers have multiple lines...no point to having call waiting |
| 03:23.30 | harryvv | I see |
| 03:23.36 | harryvv | true |
| 03:23.53 | dlynes_ | X-Rob: just for future reference |
| 03:24.04 | harryvv | anyway i need to split. |
| 03:24.20 | dlynes_ | X-Rob: that start parameter definitely doesn't work on pstn fxo |
| 03:24.27 | X-Rob | dlynes_, bugger. |
| 03:24.33 | dlynes_ | yeah, no kidding |
| 03:24.40 | dlynes_ | Thanks for the thought, though |
| 03:25.06 | dlynes_ | I just don't want asterisk answering the phone when the lines are call forwarded |
| 03:25.13 | dlynes_ | I've got it figured out for sipura 3000 units |
| 03:25.25 | dlynes_ | but not for x100p's, tdm400p's, and sangoma a200's |
| 03:25.55 | dlynes_ | anyways...thanks again for trying, but i've gotta run now |
| 03:27.09 | *** join/#asterisk d0wn3r (i=downer@tollfreelines.com) |
| 03:27.35 | d0wn3r | holy crap |
| 03:27.37 | *** join/#asterisk MrDigital (n=wildside@pool-72-81-11-65.phlapa.east.verizon.net) |
| 03:27.44 | MrDigital | anyone here famliar wit WRT54G |
| 03:27.56 | d0wn3r | sure |
| 03:27.57 | gandhijee | what about them |
| 03:28.16 | d0wn3r | they make internet |
| 03:28.31 | MrDigital | i have a WRT54G Linksys and a Linksys B Router, can the wrt54G conenct to the network via the B's wifi signal? |
| 03:28.36 | MrDigital | i know it can do it but exactly how |
| 03:29.02 | d0wn3r | by what means? |
| 03:29.27 | d0wn3r | to repeat the signal? |
| 03:29.52 | MrDigital | ok The B router is in the other room, |
| 03:29.59 | MrDigital | the g router is in this room |
| 03:30.10 | MrDigital | i want the G router to connect to the network via the B router's Wifi |
| 03:30.25 | MrDigital | so i dont have to run a cable from the B router to the G |
| 03:30.35 | gandhijee | isn't this the wrong place to be asking that MrDigital? |
| 03:30.51 | d0wn3r | i would say so |
| 03:31.09 | Altair256 | the answer is yes, MrDigital |
| 03:31.17 | MrDigital | Altair256: have you doen it? |
| 03:31.22 | Altair256 | you set the WRT54G in "client" mode |
| 03:31.22 | MrDigital | well this is part of my asterisk system |
| 03:31.29 | MrDigital | can you pm me the info/ |
| 03:31.37 | Altair256 | no DNS at the moment |
| 03:31.41 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:31.44 | Altair256 | just powered down my servers |
| 03:31.50 | Altair256 | but still connected to IRC >.> |
| 03:32.02 | Altair256 | I've never done client mode on a new Linksys router... |
| 03:32.06 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:32.13 | MrDigital | what else do i do? |
| 03:32.15 | Altair256 | but I remember doing it with some of the older WAP11B's |
| 03:32.40 | Altair256 | if you are running MAC-ACL, you need to add the MAC address of the WRT54G onto the B Router |
| 03:32.54 | Altair256 | but first, read about setting the router in client mode |
| 03:33.10 | MrDigital | mac-acl? |
| 03:33.21 | Altair256 | MAC = Media Access Control |
| 03:33.32 | Altair256 | the unique ID on each network device (ie, network card, etc) |
| 03:33.37 | Altair256 | ACL = Access Control List |
| 03:33.54 | Altair256 | so a MAC-ACL is a list of allowed MAC addresses to access your wireless network |
| 03:34.11 | MrDigital | no i dont use it |
| 03:34.12 | Altair256 | it is a rudimentary method that generally keeps honest people from accidently connecting to your network |
| 03:34.54 | justinu|laptop | heh |
| 03:34.54 | *** part/#asterisk d0wn3r (i=downer@tollfreelines.com) |
| 03:35.05 | Altair256 | well... at the end of the day, you are going to treat the WRT54G like it's a CLIENT on the other's network |
| 03:35.20 | Altair256 | so you would set it to CONNECT to the other networks SSID, WEP/WPA etc |
| 03:35.35 | Altair256 | like I said, I have not personally tested this with a WRT54G on the client end |
| 03:35.59 | Altair256 | but I have used cheap WAP11B systems to do this |
| 03:37.21 | Altair256 | one second.. I'll set my DNS to point to some random DNS server on the net |
| 03:37.29 | Altair256 | and I'll look it up for you and see if you can do this |
| 03:38.58 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 03:40.23 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 03:40.46 | Jaxxan | hey guys |
| 03:41.05 | Altair256 | sup Jaxxan |
| 03:41.32 | Altair256 | MrDigital, you'd be better off to just buy one of these http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1115416826619&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 03:41.37 | Jaxxan | so i'm using exten => ####,107,Voicemailmain(${CALLERIDNUM}) in my dialplan |
| 03:41.58 | Jaxxan | now when it matches the callerid to a number in the voicemail.conf everything works the way i want it too |
| 03:42.32 | Jaxxan | but when a number hits that context that is *not* in my voicemail.conf i wanna redirect it to a different application... and i'm stumped as to what i should do |
| 03:43.15 | kimosabe | has any one unlocked the pap 2 device |
| 03:43.32 | Agrajag- | Jaxxan: not sure if MailboxExists is what you want? |
| 03:43.47 | Jaxxan | lemme check |
| 03:43.55 | [TK]D-Fender | <PROTECTED> |
| 03:44.15 | Altair256 | MrDigital, based on what I can gather from the UserGuide on Linksys's website, this feature (wireless bridging) is not supported |
| 03:44.27 | Jaxxan | nice |
| 03:44.31 | Jaxxan | i think that's what i want |
| 03:44.39 | Jaxxan | thx guys |
| 03:44.43 | Altair256 | MrDigital, they say to buy a Wireless Ethernet Bridges (WET54G, WET11). |
| 03:46.09 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 03:47.16 | *** join/#asterisk Abydos313 (n=abydos31@adsl-71-129-61-88.dsl.irvnca.pacbell.net) |
| 03:47.28 | kimosabe | i can help with wireless |
| 03:47.43 | kimosabe | i run 25 mile links with cheap equipment |
| 03:48.39 | camelon | If in my *Box.. .when the people call to my PBXs use to have busy signal . . .it is a Telco problem or mine? |
| 03:49.21 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) |
| 03:49.37 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 03:51.10 | ManxPower | kimosabe, Nifty. Wish I could. |
| 03:51.36 | kimosabe | i use cb3 pluss delux 200mw devices |
| 03:51.40 | *** join/#asterisk bmg505 (n=leon@c1-63-15.rndf.isadsl.co.za) |
| 03:51.42 | ManxPower | camelon, you need to tell Asterisk to provide a busy. See the macro-std-exten in extensions.conf.sample |
| 03:51.53 | ManxPower | kimosabe, Ah. I have those too. |
| 03:52.02 | ManxPower | I have two of them actually. |
| 03:52.23 | kimosabe | there cool huh i use 2 use wrap boards but these are cheaper 2 set up |
| 03:52.35 | ManxPower | kimosabe, What is your jitter and latency on a clear day, on a foggy day, and during a downpour |
| 03:52.45 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 03:53.08 | ManxPower | Heck. my DirecTV with a 20" dish goes out during a bad storm |
| 03:53.16 | ManxPower | Can you run VoIP over the wireless link? |
| 03:53.28 | h3x | thats 22,000 miles of atmosphere |
| 03:53.28 | h3x | heh |
| 03:53.35 | ManxPower | kimosabe, I'm 11 miles from the CO, no DSL, no Cable |
| 03:53.42 | *** join/#asterisk mrdigital (n=wildside@pool-72-81-11-65.phlapa.east.verizon.net) |
| 03:53.46 | mrdigital | hmm whatever i did |
| 03:53.47 | kimosabe | manx power it hardlly rains here its rather how yesterday it was 119 |
| 03:53.49 | mrdigital | worked |
| 03:53.53 | mrdigital | what did i do tho lol |
| 03:53.59 | camelon | ManxPower . . .but the proble is that 30%of the times when the people call to my +Box . . receive a busy signal . . is it related with the macro or the telco?. . . TIA |
| 03:53.59 | kimosabe | my links really never have problems |
| 03:54.02 | ManxPower | h3x, Yes, but less than a mile of rain, and they transmit a lot more than 200mw |
| 03:54.15 | ManxPower | kimosabe, Ah, that explains why it works for you |
| 03:54.38 | ManxPower | camelon, Ah. people calling INTO your Asterisk get a busy signal? |
| 03:55.00 | camelon | yepp . .30 % of the times |
| 03:55.12 | h3x | notthat much more |
| 03:55.14 | h3x | a couple watts? |
| 03:55.18 | h3x | they are powered by solar panels |
| 03:55.27 | kimosabe | manxpower im runni vo/ipon wireless link 9 miles long and i have a 4 megabit link i run data and voice its been up for 1 yr 9 month with out a flaw only once it raineds and water got in the cable but i put tar on it and its been up sonce |
| 03:55.38 | kimosabe | no i run milliwats |
| 03:55.39 | X-Rob | h3x, you realise you said 22,000 miles of atmosphere, right? |
| 03:55.43 | ManxPower | "Oh, I do 23 mile wireless links, using cheap equipment." They forget to mention "It rains here once every 50 years and we never get above 20% humidity." |
| 03:55.46 | kimosabe | 200 miliwat |
| 03:55.49 | h3x | its 11,000 each way |
| 03:55.52 | kimosabe | yes i have several links |
| 03:55.59 | h3x | satellite |
| 03:56.06 | kimosabe | in fact about 18 links 2 be exact |
| 03:56.07 | znoG | kimosabe: complete line of sight, right? |
| 03:56.08 | X-Rob | h3x, the ammosphere finishes about 6 miles up. |
| 03:56.13 | h3x | true |
| 03:56.13 | h3x | heh |
| 03:56.18 | kimosabe | yes i live in the highest area in the city |
| 03:56.33 | h3x | but its still a 22k mile trip |
| 03:56.34 | kimosabe | i can see the entire business part of the city |
| 03:56.36 | SwK | anyone know (off the top of your head) how many DS3's per OC-12? |
| 03:56.39 | X-Rob | oh yeah. |
| 03:56.41 | h3x | SwK: 12 |
| 03:56.49 | X-Rob | but not through 22000m of atmosphere tho8) |
| 03:56.49 | kimosabe | the 25 mile link is towards colombia |
| 03:56.57 | h3x | OC-n where n is a ds3 channel |
| 03:57.49 | kimosabe | i had isues at first but i put the front part of my antenna on the fron of direct tv dishes and they shoot a clean signal right through a noisy area |
| 03:57.55 | ManxPower | As far as I can figure, my only option is a Voice and Data T-1, which kinda sucks |
| 03:58.16 | h3x | ManxPower: there are cheap ds3 wireless gear on ebay sometimes |
| 03:58.23 | SwK | h3x: are you sure its just 12? |
| 03:58.26 | h3x | SwK: yes |
| 03:58.29 | SwK | its like a 600meg link |
| 03:58.33 | ManxPower | h3x, Define "cheap" |
| 03:58.33 | h3x | 622 |
| 03:58.34 | SwK | thats like 13 DS3s |
| 03:58.36 | h3x | about a grand |
| 03:58.40 | kimosabe | manx power run the new 900 mghz equipment its great |
| 03:58.43 | h3x | SwK: Overhead |
| 03:58.43 | SwK | 621.84 with verhead |
| 03:58.48 | ManxPower | h3x, That's 2 months of T-1 service |
| 03:58.52 | kimosabe | run it on a wrap board |
| 03:58.54 | h3x | ManxPower: well |
| 03:58.59 | h3x | how far apart is your locations |
| 03:59.06 | ManxPower | h3x, and suggestions on what to search for? |
| 03:59.14 | SwK | yeah but 601 mega after over head 45meg for a DS3 for 601/45 = 13 and some change |
| 03:59.20 | h3x | ManxPower: whats your npa/nxx and addresses of each side |
| 03:59.23 | SwK | I'm just trying to figure out where the other bits go heh |
| 03:59.32 | ManxPower | h3x, not sure. The CO I'm connected to doesn't even have ANY DSL. I guess 20 miles or so |
| 03:59.36 | SwK | unless its eaten by additional DS3 overhead |
| 03:59.40 | h3x | SwK: you want to use packet over sonet (POS) |
| 03:59.48 | camelon | ManxPwer . . .the busy signal happens around 30% times |
| 03:59.48 | h3x | if you arent using channelized circuits on it |
| 03:59.50 | ManxPower | h3x, I don't have an address for the remote site yet. |
| 03:59.51 | denon | ManxPower is just a cheapskate, I already offered him an uber-cheap ds1 |
| 04:00.00 | *** join/#asterisk mrdigital (n=wildside@pool-72-81-11-65.phlapa.east.verizon.net) |
| 04:00.05 | denon | hehe |
| 04:00.08 | h3x | is xo in your city? |
| 04:00.08 | mrdigital | wheres that guy that told me how to use my router as a client? |
| 04:00.11 | Altair256 | wb MrChimpy |
| 04:00.15 | Altair256 | err.. mrdigital |
| 04:00.16 | h3x | I get free inter-co mileage on xo |
| 04:00.22 | Altair256 | any luck? |
| 04:00.22 | h3x | if the two co's are lit |
| 04:00.32 | mrdigital | hey Altair256, yeah i dont know how its working lol |
| 04:00.34 | mrdigital | question, |
| 04:00.41 | mrdigital | will this affect my internet speed? |
| 04:00.41 | Altair256 | congratulations ^^ |
| 04:00.55 | SwK | h3x: channelized to DS0s is what i'm trying to figure out or DS1s running PRI |
| 04:00.56 | Altair256 | well... you would have been better off to put the G on as your household router... |
| 04:01.01 | Altair256 | and use the B as the client |
| 04:01.03 | ManxPower | My NPA-NXX -s 256-538 |
| 04:01.18 | mrdigital | why |
| 04:01.19 | h3x | an oc-12 encapsulates 12 DS3s which encapsulate 28 T1s |
| 04:01.20 | SwK | manxpowr red neck |
| 04:01.26 | h3x | if you run it in M23 mode |
| 04:01.31 | SwK | ok |
| 04:01.33 | SwK | close enuff |
| 04:01.34 | SwK | thanks |
| 04:01.35 | Altair256 | so if you have any G devices, they would connect to the G network and connect to each other faster |
| 04:01.39 | Altair256 | but your internet connection should be fine |
| 04:01.41 | SwK | I didnt think 13 was right |
| 04:01.41 | SwK | heh |
| 04:01.50 | SwK | all tho the math works for 13 |
| 04:02.00 | h3x | there is a lot of overhead in TDM |
| 04:02.16 | SwK | well yeah |
| 04:02.25 | mrdigital | Altair256: you cant set the B router to Client |
| 04:02.29 | Damin | ManxPower: If it is any consolation to you, I've managed to remove codec_g723 from my system w/ bweschke's patch! ;) |
| 04:02.33 | ManxPower | SwK, No, the rednecks are the ones that keep shooting up our mailbox |
| 04:02.39 | SwK | but TDM vs Clear Channel a clear channel DS3 is still 45 megabit |
| 04:02.41 | ManxPower | Damin, don't speak to me about htat |
| 04:02.43 | SwK | hah |
| 04:02.52 | Altair256 | then seems you're in your best scenario right now, without having to buy any additional hardware |
| 04:02.53 | Damin | ManxPower: Why? You were the one on the soapbox.. |
| 04:03.09 | h3x | a channelized DS3 only carries a 43 megabit payload |
| 04:03.16 | ManxPower | Damin, because you know you are right and I'm wrong and I know that I'm right and you are wrong. |
| 04:03.30 | SwK | a non channelized DS3 is 45 megabit tho |
| 04:03.39 | Damin | ManxPower: Hehehe.. OK.. if I run into you at a conference, I'll still buy you a beer.. ;) |
| 04:03.44 | ManxPower | h3x, so, can you get me a voice/data t-1 for under $550/month? |
| 04:03.47 | h3x | minus ppp or hdlc overhead :) |
| 04:03.53 | ManxPower | Damin, And I, you. |
| 04:03.55 | SwK | well yeah |
| 04:04.06 | SwK | the clock rate is 45meg |
| 04:04.20 | Damin | ManxPower: Where do you need a DS1 circuit? |
| 04:04.29 | SwK | I need a DS1 at my house |
| 04:04.33 | ManxPower | Damin, My NPA-NXX -s 256-538 |
| 04:04.37 | SwK | just out side of HSV |
| 04:04.47 | h3x | i cant look up anything without addresses |
| 04:04.47 | h3x | heh |
| 04:04.51 | ManxPower | SwK, Actually, just outside Gadsden / Birmingham |
| 04:05.05 | SwK | I need one just outside of HSV |
| 04:05.14 | ManxPower | h3x, address in /msg |
| 04:05.17 | Damin | ManxPower: Does it need to have channels split between Voice and data? I.E. via a DACS? |
| 04:05.19 | h3x | are you doing mostly ld or local? |
| 04:05.27 | SwK | I really just want an OC192 @ my house |
| 04:05.34 | Sedorox | SwK: don't we all |
| 04:05.46 | SwK | I just want it for surfing Pr0n |
| 04:05.48 | ManxPower | Damin, I'm not running VoIP as my only PSTN access. |
| 04:05.55 | Damin | ManxPower: Or will straight data w/ IAX/SIP Term and QOS do? |
| 04:06.06 | docelm0 | a single computer will not run OC192.. |
| 04:06.16 | h3x | oc192 isnt even a single pipe |
| 04:06.19 | ManxPower | Damin, I would prefer 512K - 768K Internet Data with 7 - 8 Bchannels and a D channel |
| 04:06.19 | h3x | its a group of oc48 waves |
| 04:06.23 | Sedorox | docelm0: who says it would be a single computer :p |
| 04:06.32 | kimosabe | manpower can u help me one thing why does my x-lite say registered but when i dial it says call not aproved |
| 04:06.47 | h3x | ManxPower: the only clecs you have out there is level3 which only does voip dids |
| 04:06.53 | SwK | docelm0 no shit... besides I have computes that can saturate bonded quad GigE interfaces |
| 04:06.55 | h3x | pacwest which in east coast markets is voip only |
| 04:07.02 | h3x | tcg which is probably somebody else now |
| 04:07.05 | h3x | and your bellsouth |
| 04:07.05 | Damin | ManxPower: I can't help you then... I can do really aggressive data pricing w/ long-haul loops, but I don't have the ability to drop and insert channels on a DS1 bases.. |
| 04:07.08 | h3x | oh ITC deltacom |
| 04:07.15 | docelm0 | ohh nice.. :) |
| 04:07.18 | SwK | ITC Deltacom is the ass |
| 04:07.26 | docelm0 | me with my measly 2 DS3's.. :( |
| 04:07.31 | ManxPower | Damin, under $550 long haul loop? |
| 04:07.35 | Damin | ManxPower: As an example, I got pricing on a 500 mile loop for a client of mine, and it was $220. |
| 04:07.55 | ManxPower | The problem is I want my own TDM with local DIDs |
| 04:07.59 | SwK | docelm0 I have a GigE port 2 hops off L3s core |
| 04:08.10 | SwK | (just not at home) |
| 04:08.14 | Sedorox | lol |
| 04:08.17 | docelm0 | Damin what kinda 500 mile loop? |
| 04:08.23 | Damin | ManxPower: Yeah.. I can't do anything except VoIP.. just don't have the DACS for it.. |
| 04:08.26 | Jaxxan | awesome |
| 04:08.28 | ManxPower | Damin, Only good pricing on InterLATA, I assume./ |
| 04:09.05 | Damin | ManxPower: Well, we get pretty agressive pricing on IntraLata as well, but the InterLata stuff is just ridiculously cheap.. |
| 04:09.08 | ManxPower | I have had NOTHING but trouble with VoIPoInternet |
| 04:09.09 | SwK | manxpower you do a lotta intralata calls? |
| 04:09.11 | denon | just jack this in please :) |
| 04:09.38 | Sedorox | lol |
| 04:09.47 | Damin | docelm0: ESF/B8ZS, DS1 500 mile loop. ;) |
| 04:09.50 | Sedorox | denon: if it could only be that easy |
| 04:09.54 | docelm0 | eww |
| 04:09.57 | ManxPower | SwK, only for small values of "lotta" |
| 04:10.06 | denon | SwK: good thing I'm booting off a soldered prom :) |
| 04:10.37 | ManxPower | If I can get a PRI cheap enough, I don't care where the end of the Data T-1 is. |
| 04:10.38 | SwK | .me needs another shot of tequila |
| 04:11.02 | denon | ..and a visit from mavis beacon |
| 04:11.05 | SwK | manxpower is centurytel there? |
| 04:11.07 | Damin | ManxPower: We could terminate that as PRI, but the minutes will cost you all the same.. |
| 04:11.19 | ManxPower | SwK, BellSouth is the ILEC |
| 04:11.25 | Damin | ManxPower: You need to find someone locally that can offer you flat rate local in/out and port your numbers to their PRI.. |
| 04:11.27 | denon | SwK: you have deals with centurytel? |
| 04:11.37 | SwK | i have centurytel in my office |
| 04:11.45 | Damin | CenturyTel must DIE!!!!!!!! |
| 04:11.45 | SwK | they play CLEC in some markets |
| 04:11.46 | Damin | CenturyTel must DIE!!!!!!!! |
| 04:11.49 | SwK | haqhahaahahahah |
| 04:11.57 | CunningPike | camelon: Does anything show in the CLI when your callers get busy tone? |
| 04:12.02 | SwK | settle down damin |
| 04:12.03 | ManxPower | Damin, I don't have numbers at the moment. |
| 04:12.06 | denon | SwK: you mean they have a POP in your office? or you use them for DIA? |
| 04:12.20 | Damin | SwK: SBC shut them out of the metro markets locally, so they are on the outskirts.. :) |
| 04:12.21 | SwK | we had another smaller clec and then centurytel bought them out about 6 months ago |
| 04:12.30 | SwK | DIA |
| 04:12.33 | denon | ah |
| 04:12.42 | denon | Ive gotta find someone who can do good deals on Frontier T1s/etc |
| 04:13.04 | *** join/#asterisk mrgoby (n=mrgoby@c-68-42-71-60.hsd1.mi.comcast.net) |
| 04:13.05 | ManxPower | Damin, the problem is that the potential revenue is fairly small for this project. |
| 04:13.09 | denon | and that I can sell SBC through |
| 04:13.57 | ManxPower | I've gotten quotes for about $550/month for Voice/Data T-1 |
| 04:13.57 | SwK | they are facilities based here with their purchase... they buy dry copper from bellsuck and back haul it to their switch over HDSL |
| 04:14.20 | Sedorox | ManxPower: from what I've heard.. thats average |
| 04:14.40 | Jaxxan | Qwell you on ? |
| 04:14.40 | mrgoby | hey, trying to build * on suse10 ... getting errors building zaptel, no make target 'modules'. |
| 04:14.56 | ManxPower | So if someone can offer Voice PRI / Data with unlimited local in/out and decent rates on toll calls I'll put you on the list of potential carriers. |
| 04:15.15 | mrgoby | from this : make -C /lib/modules/2.6.13-15.8-default/build SUBDIRS=/usr/src/zaptel-1.2.5 XPPMOD= modules |
| 04:15.18 | SwK | mrgoby there is no modules target in the zaptel makefile that I recall... just cd zaptel && make install |
| 04:15.27 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 04:15.29 | mrgoby | ah... no make ? |
| 04:15.42 | denon | mrgoby: probably worth reading the install instructions on the site :) |
| 04:15.51 | mrgoby | same thing |
| 04:15.52 | denon | cd /usr/src/whatever; make clean; make install |
| 04:15.52 | Jaxxan | or the README |
| 04:15.53 | *** join/#asterisk EastWolf (n=eastwolf@221.217.217.243) |
| 04:16.08 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 04:16.10 | Damin | ManxPower: All I have to say is that "function SIP_CALLERPREFCODECS" fucking rocks.. |
| 04:16.32 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 04:16.39 | mrgoby | i've built asterisk several times... sorry, let me clarify... it looks like it is looking for make targets in the kernel tree itself ? ... it builds a few things and then fails |
| 04:17.00 | mrgoby | let me pastebin |
| 04:17.01 | Jaxxan | anyone here using queuemetrics ? |
| 04:20.18 | mrgoby | http://pastebin.ca/55298 |
| 04:20.31 | mrgoby | i included the kernel info |
| 04:21.35 | mrgoby | any ideas ? |
| 04:21.41 | SwK | you do know make linux26 isnt required any more |
| 04:22.02 | mrgoby | i get the same behavior out of 'make' and 'make install' |
| 04:22.02 | Jaxxan | why are you using linux26 ? |
| 04:22.06 | SwK | and you did install the kernel sources and headers right? |
| 04:22.28 | *** join/#asterisk tuxd00d (n=tuxinato@adsl-75-7-235-117.dsl.irvnca.sbcglobal.net) |
| 04:22.40 | mrgoby | i'm new to suse, so i am only pretty sure... i installed kernel-sources and all gcc-dev etc |
| 04:22.55 | Jaxxan | is this a fresh install ? |
| 04:22.57 | Jaxxan | or an upgrade ? |
| 04:23.06 | mrgoby | pretty fresh... |
| 04:23.18 | mrgoby | it is a client's box... but that is my understanding |
| 04:23.20 | mrgoby | suse 10 |
| 04:23.37 | mrgoby | or, you mean an upgrade from 9.3 or something ? |
| 04:23.40 | Jaxxan | and the current zaptel modules aren't running right ? |
| 04:24.13 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-170-44.dsl.chcgil.sbcglobal.net) |
| 04:24.15 | Flauto | hi all |
| 04:24.24 | Flauto | i am testing a service provider |
| 04:24.33 | Jaxxan | i think i ran into problems compiling zaptel on my current redhat box when i was upgrading to latest SVN versions, and i had forgot to unload the old modules |
| 04:24.46 | mrgoby | there are some built already in the modules tree, but they arent loaded |
| 04:24.49 | Jaxxan | then i couldn't unload them and had to reboot the box |
| 04:24.51 | Flauto | when i call a pstn line |
| 04:25.04 | Flauto | i don't hear the other person and the other person can not hear me |
| 04:25.21 | Flauto | but when i call thrie voip user, i can hear, and the other party can hear too |
| 04:25.37 | Jaxxan | pstn is a POTS line ? |
| 04:26.12 | Jaxxan | sorry, i dont spend much time with POTS lines, only PRI's |
| 04:26.17 | Flauto | regular landline |
| 04:26.19 | Jaxxan | just curious (= |
| 04:26.28 | *** join/#asterisk Eggplant (n=none@dsl-216-155-215-185.cascadeaccess.com) |
| 04:26.33 | mrgoby | i don't understand why having the modules loaded would affect your ability to build them |
| 04:26.49 | Jaxxan | mrgoby: me neither, and that's not the issue at hand apparently (= |
| 04:27.01 | Jaxxan | mrgoby: but i tell you, it drove me nuts for like 10 minutes lol |
| 04:27.23 | mrgoby | yeah... i'm figuring i am doing something wrong with the linking somehow |
| 04:27.26 | Jaxxan | Flauto: FXS ? |
| 04:27.27 | mrgoby | but the error throws me off |
| 04:27.34 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:28.14 | Jaxxan | mrgoby: you try the svn version of zaptel ? |
| 04:28.29 | Jaxxan | someone told me that 1.2.5 had issues, so i never used it |
| 04:29.05 | mrgoby | fair enough |
| 04:32.21 | Jaxxan | i should probably spend more time with my fx0 lines |
| 04:32.26 | Jaxxan | i dont think they hang up correctly |
| 04:33.06 | Jaxxan | maybe i should put a hangup command in there |
| 04:33.38 | CunningPike | ~pots |
| 04:33.39 | jbot | rumour has it, pots is Plain Old Telephone Service as in "Old Analogue Crap" |
| 04:34.02 | Jaxxan | ~pstn |
| 04:34.04 | jbot | it has been said that pstn is Pubic Switched Telephone Network, or "please stop the nonsense" |
| 04:34.21 | Jaxxan | so a pstn line is a ?? |
| 04:34.32 | CunningPike | Ambiguous |
| 04:34.35 | Sedorox | normal telephone |
| 04:34.37 | Sedorox | line |
| 04:34.42 | Jaxxan | ok |
| 04:34.49 | Jaxxan | normal telephones suck |
| 04:35.08 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 04:35.52 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:36.24 | mrgoby | same error with svn version |
| 04:37.20 | mrgoby | make -C /lib/modules/2.6.13-15.8-default/build SUBDIRS=/usr/src/zaptel modules this is the culprit |
| 04:37.54 | mrgoby | why does it cd to that directory ? |
| 04:38.56 | *** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 04:39.08 | Tier_1 | ok whats up with SayUnixTime |
| 04:39.14 | Tier_1 | its not working |
| 04:40.16 | Jaxxan | works for me |
| 04:40.33 | *** part/#asterisk mrgoby (n=mrgoby@c-68-42-71-60.hsd1.mi.comcast.net) |
| 04:40.35 | Jaxxan | exten => XXXX,1,sayunixtime |
| 04:45.15 | Jaxxan | there's a weather feature right ? |
| 04:45.32 | Jaxxan | that connects to weather.com or something and tells you the current weather ? |
| 04:47.36 | ManxPower | Jaxxan, not as part of Asterisk |
| 04:47.50 | *** join/#asterisk terrapen_ (n=cjs@chrissnell.dsl.xmission.com) |
| 04:54.30 | Jaxxan | what's asterisk@home ? |
| 04:54.39 | Jaxxan | that like a ... consumer edition or something ? |
| 04:55.53 | ManxPower | Jaxxan, Asterisk@Home / AMP / FreePBX are all Asterisk + something (usually a GUI, but could be an ISO distro, or whatever) for people too lazy or not smart enough to learn Asterisk's text config files. |
| 04:56.11 | De_Mon | Once a call is exstablished from the console, how do I dial numbers? |
| 04:57.00 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 04:57.10 | ManxPower | De_Mon, did you try typing numbers? |
| 04:57.30 | De_Mon | ManxPower ya no such command |
| 04:58.02 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 04:58.11 | ManxPower | huh? |
| 04:58.22 | ManxPower | Ah. |
| 04:58.30 | ManxPower | I don't think Console/ channel supports that |
| 04:59.14 | De_Mon | I'm trying different ways of 'sendDTMF' without success |
| 04:59.28 | De_Mon | I can execute the dial command from the console, but that's it? |
| 04:59.52 | *** join/#asterisk lately (n=doug@ppp167-252-31.static.internode.on.net) |
| 05:01.09 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:01.09 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:01.35 | lately | Hi. Is there a way to limit the number of channels? For example, the number of channels used when going through the Internet for VoIP |
| 05:01.45 | lately | not the internal channels |
| 05:01.54 | lately | within a LAN |
| 05:02.01 | ManxPower | De_Mon, pretty much. It's onlt used for testing AFIK |
| 05:02.06 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 05:02.18 | ManxPower | lately, read The Book |
| 05:02.20 | ManxPower | ~thebook |
| 05:02.22 | jbot | thebook is probably Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 05:02.29 | lately | Reading... |
| 05:02.36 | ManxPower | Love the book, buy the book, send the book flowers |
| 05:02.43 | lately | I started reading last night |
| 05:02.46 | De_Mon | well testing dialplans usually invokes, pressing menu numbers |
| 05:02.48 | lately | Fell asleep :-/ |
| 05:03.13 | De_Mon | oh well, thats what softphones are for |
| 05:03.32 | ManxPower | see also: setgroup, checkgroup and README.variables in the source tree |
| 05:17.45 | astermick | does asterisk have problems with calling phones that have been diverted to another number (gsm or pstn) |
| 05:17.48 | astermick | ? |
| 05:21.07 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 05:24.41 | ManxPower | astermick, how would askerisk know a call was diverted? |
| 05:28.30 | *** join/#asterisk Jestie (n=Jestie@dsl-146-30-85.telkomadsl.co.za) |
| 05:28.39 | Jestie | good morning y'eal all |
| 05:29.05 | *** join/#asterisk austinnichols101 (i=austinni@dsl-10-169.cofs.net) |
| 05:29.43 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 05:36.27 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:38.30 | Jestie | anyone up to give some help on a voicemail issue I have ... ? |
| 05:39.01 | CunningPike | Jestie: Soot |
| 05:39.06 | CunningPike | Shoot, I mean |
| 05:40.05 | *** part/#asterisk catch23 (n=catch23@c-67-191-252-3.hsd1.ga.comcast.net) |
| 05:40.53 | Jestie | Fanks man .. |
| 05:41.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:41.40 | Jestie | I want users to have the option, to go back to receptions desk, when they do not want to leave voicemail for the user ... |
| 05:41.52 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:42.12 | Jestie | So typically .. voicemail must start, but also prompt the user that if they want to speak to reception, they must press 9 ...or something ... that will ring receptions desk ... |
| 05:42.43 | CunningPike | Press 0 - place an 'o' extension in the context from which VoiceMail was called |
| 05:43.55 | CunningPike | exten => o,1,Dial(foo) |
| 05:44.07 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 05:45.01 | Jestie | not sure i'm following ... |
| 05:45.26 | Jestie | this is what I have at the moment |
| 05:45.34 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 05:45.35 | Jestie | exten => 1234,1,Dial(SIP/2300, 30) |
| 05:45.36 | Jestie | exten => 1234,2,VoiceMail(2300@other) |
| 05:45.38 | Jestie | exten => 1234,3,Background(9_foroperator) |
| 05:45.39 | Jestie | exten => 1234,4,Hangup() |
| 05:47.36 | CunningPike | Jestie: The person's vm message should mention "to speak with someone else, dial 0" and then, in the same context as 1234 is in, have another 'o' extension as my example |
| 05:48.11 | Jestie | ohh ... |
| 05:48.13 | Jestie | Fanks man ... |
| 05:49.20 | Jestie | Can I just append the 'o' extension ? or should it be right at the start ? |
| 05:49.32 | *** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net) |
| 05:50.55 | *** join/#asterisk clive- (n=pirch@dsl-165-188-238.telkomadsl.co.za) |
| 05:51.40 | CunningPike | Jestie: It shouldn't matter - it's just another extension in that context |
| 05:52.43 | Jestie | Kewl bananananans ! |
| 05:52.58 | CunningPike | I take it that it worked, then |
| 05:53.01 | hardwire | heh |
| 05:54.26 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 05:55.08 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) [NETSPLIT VICTIM] |
| 05:55.08 | *** join/#asterisk riksta (n=rick@62.6.163.87) [NETSPLIT VICTIM] |
| 05:55.08 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
| 05:55.28 | raidenz | Hello. |
| 05:56.03 | raidenz | Does anyone know where to get an updated test g.729 ipp codec patch that works with the latest asterisk svn? |
| 05:59.19 | *** join/#asterisk catch23 (n=catch23@c-67-191-252-3.hsd1.ga.comcast.net) |
| 06:00.02 | clive- | raidenz isnt thre that website in lituania or somewhere |
| 06:00.21 | *** join/#asterisk Creperum (n=ilya@tex.tsua.net) |
| 06:01.19 | raidenz | clive-: That is the old patch (from 2005) that won't work with the latest asterisk svn. It has to be updated to work with the new asterisk loader changes. |
| 06:02.25 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 06:03.38 | clive- | raidenz, I am behind the times then...:), I am sure someone has figured it out though |
| 06:04.22 | raidenz | I hope so. I was wondering if anyone here has updated it to send it to me if possible. :) |
| 06:04.27 | X-Rob | raidenz, no. |
| 06:04.29 | *** join/#asterisk Creperum (n=ilya@tex.tsua.net) |
| 06:04.32 | X-Rob | no-one's re-written it yet |
| 06:04.37 | raidenz | No? |
| 06:04.40 | raidenz | Hmm |
| 06:05.49 | raidenz | I tried adapting it with no luck. |
| 06:10.33 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 06:19.23 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
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| 06:35.10 | *** join/#asterisk kay2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 06:44.39 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 06:46.10 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 06:47.40 | littleball | hi, i am try to use xlite to connect to the asterisk which is running on the localhost . the xlite always try to detect firewall and then cannot connect to my asterisk testing server. who can help? |
| 07:04.47 | *** join/#asterisk koenvi (n=root@tech.ascom.be) |
| 07:05.12 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 07:06.55 | koenvi | ready help |
| 07:07.28 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 07:07.59 | Jestie | littleball .. shoot .. i'm listineng |
| 07:07.59 | koenvi | anyone working with zaptel hardware in E1 connections? |
| 07:09.40 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.154.185.telnor.net) |
| 07:10.21 | littleball | <PROTECTED> |
| 07:10.35 | littleball | my xlite cannot connect to my local asterisk . can help? |
| 07:12.30 | Jestie | yep |
| 07:13.12 | littleball | Jestie, i click on the configuration of the xlite software |
| 07:13.13 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 07:14.23 | littleball | for the system setting, what is function of Network and SIP Proxy? |
| 07:21.39 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 07:24.41 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 07:27.20 | *** join/#asterisk MikeB (n=chatzill@175.54.187.81.in-addr.arpa) |
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| 07:43.37 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
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| 07:48.44 | littleball | hi Jestie |
| 07:49.18 | *** join/#asterisk Uzzi (n=Andrea@host18-239.pool875.interbusiness.it) |
| 07:49.27 | Uzzi | hi |
| 07:49.59 | Uzzi | Woh know if it's possibly to use asterisk with hcf modem conexant chipset? |
| 07:50.17 | *** join/#asterisk The_Isle_of_Mark (n=mark@c-68-85-63-96.hsd1.ga.comcast.net) |
| 07:50.25 | clive- | uzzi bristuff works with hfc-s |
| 07:53.58 | Jestie | Uzzi .. |
| 07:54.00 | Jestie | here is a link http://bach-online.de/blog/?p=50 |
| 07:54.16 | littleball | Jestie, it works. i just change the domain to my localhost name |
| 07:54.33 | Jestie | =;0) |
| 07:56.05 | Uzzi | then I've to istall bristuff-0.3.0-PRE-1? |
| 07:56.13 | Jestie | yep |
| 07:56.18 | skeffling | Hello, Has anyone seen problems with Asterisk 1.2.7 and Eyebeam 1.5? eg Pressing Hold puts the call on hold, but unholding them does not bring them back. In asterisk I don't get the ' Stopped music on hold...' message. But works fine with a SNOM. |
| 07:58.35 | Uzzi | asterisk-bristuff it's the same? |
| 07:59.24 | Jestie | don;t think so ... |
| 07:59.25 | Jestie | junghans wrote it specifically for HFC chipsets |
| 08:00.40 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 08:05.27 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 08:08.00 | x86 | are there any sounds in the asterisk-sounds package that have one of those nostalgic number disconnected error tones? |
| 08:11.01 | *** part/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 08:14.02 | Jestie | HELP !!! |
| 08:23.13 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
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| 08:31.12 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 08:31.58 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 08:32.09 | Ikarus | Right, I am about to toss out Asterisk and simply get my boss to pay 2000 euro for a ready made PBX as I don't seem to be able to find a way to get the echo down, I tried everything I could find. I am using a normal HFC card (currently with zaphfc driver, but that can be altered) and I am on KPN's ISDN network, switching to a native VoIP solution for the backend is not an option, and I am testing a variety of SIP phones |
| 08:32.51 | stoffell | Ikarus, what * version you're using? bristuff? what vers? |
| 08:33.20 | Ikarus | stoffell: Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1k |
| 08:33.36 | stoffell | Ikarus, and what echo cancellation? (in zaptel config) |
| 08:34.11 | Ikarus | echocancel=32 |
| 08:34.11 | Ikarus | echotraining = yes |
| 08:34.12 | Ikarus | echocancelwhenbridged=yes |
| 08:34.26 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 08:35.22 | stoffell | Ikarus, and did you change anything in zconfig.h (in bristuff/zaptel dir) |
| 08:35.30 | Ikarus | no |
| 08:35.46 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:35.54 | stoffell | Ikarus, seems to be something worth trying.. |
| 08:37.47 | stoffell | Ikarus, i pasted some info here: http://pastebin.ca/55358 |
| 08:42.17 | *** join/#asterisk cced1 (n=dev2003@222.33.36.205) |
| 08:43.55 | Ikarus | Right |
| 08:43.58 | Ikarus | let's hope that works |
| 08:44.42 | Ikarus | Any suggestions for the echocancel= setting after I enable it ? |
| 08:47.06 | *** join/#asterisk heka (n=heka@82.114.68.123) |
| 08:47.32 | heka | Hello, does the h323 channel comming with asterisk (asterisk/channels/h323) support jitter buffer? |
| 08:48.56 | heka | or do I need to build the openh323 comming with asterisk-addons |
| 08:48.57 | heka | ? |
| 08:50.08 | Ikarus | stoffell: ? |
| 08:51.13 | stoffell | Ikarus, my settings in zapata.conf are: echocancel=yes, echocancelwhenbridged=yes, echotraining=100 |
| 08:54.02 | Jestie | Was wondering if someone can tell me how to redirect Voicemail to reception if I do NOT WANT TO LEAVE Voicemail for a user ? |
| 08:55.00 | *** join/#asterisk bartpbx (n=bartpbx@p54B00978.dip0.t-ipconnect.de) |
| 08:55.05 | bartpbx | hello |
| 08:55.41 | bartpbx | I'm looking for a list of all existing PRI_CAUSEs. In libpri.h are only some of them defined |
| 08:56.16 | heka | any idea about h323 jitter buffer? |
| 08:56.34 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 08:57.05 | bartpbx | for example what is 8 ,20, 77, 119, or 386? |
| 09:02.46 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:05.18 | clive- | heka , yes, cisco |
| 09:05.25 | clive- | lol |
| 09:10.18 | heka | clive-: my qestion was about openh323 |
| 09:11.30 | *** part/#asterisk bartpbx (n=bartpbx@p54B00978.dip0.t-ipconnect.de) |
| 09:11.31 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 09:11.35 | dyn | hi |
| 09:12.36 | clive- | truthfully, afaik, h323 and asterisk need lots of tweaking to work, you may find chan_woomera a good option, although I am not sure about jitter buffering. Your best bet for a large scale h323 installation imho is cisco |
| 09:15.19 | heka | what kind of router do I need to handle a network with about 10 concurrent calls! |
| 09:15.35 | heka | any idea clive- ? |
| 09:16.08 | clive- | 10 calls is not a lot, asterisk could handle it, but then again yuor jitter buffer question remains a mystery..:) |
| 09:16.54 | heka | I heard that openh323 support jitter buffering, but not sure if the channel that comes with asterisk does it |
| 09:17.30 | clive- | chan_woomera uses openh323 afaik |
| 09:18.11 | heka | let me check it! |
| 09:19.38 | dyn | anyone can give me a hint what's that: May 12 11:18:48 NOTICE[18319]: chan_iax2.c:2447 iax2_read: I should never be called! |
| 09:20.00 | dyn | (i'm playing with forwarding calls with 2 asterisk servers using IAX2) |
| 09:23.17 | heka | clive-: from voip-info.org: oh323 driver uses the RTP/RTCP stack and the adaptive jitter buffer implementation of OpenH323. |
| 09:23.38 | clive- | so your sorted |
| 09:23.39 | heka | so I think openh323 supports the jitter buffering |
| 09:23.46 | heka | :) |
| 09:26.21 | clive- | g'luck |
| 09:26.33 | *** join/#asterisk Uzzi (n=Andrea@host229-236.pool872.interbusiness.it) |
| 09:26.54 | Uzzi | now i've installed bristuff,now how i can test it? |
| 09:27.07 | heka | thx! |
| 09:27.59 | dyn | anyone knows if I can use macros or something to shorten sip.conf entries for all the softphones used in our office? |
| 09:28.22 | dyn | each softphone entry having about 15 lines in sip.conf makes it a bit error-prone to maintain |
| 09:33.58 | stoffell | Ikarus, making progress? |
| 09:36.23 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 09:36.26 | Delvar | dyn: you can put most of the common stuff in the [default] but that doesnt help with vociemail lines etc.. you could also write a script to generate an external file and include that file. |
| 09:37.47 | Ikarus | stoffell: not yet, dealing with some other crap right now |
| 09:38.15 | stoffell | Ikarus, okay, keep me posted, i'm curious on the progress.. |
| 09:39.30 | dyn | Delvar: i considered scripting it but i'd prefer a cleaner built-in solution |
| 09:39.43 | dyn | Delvar: if i add common options in [default], it will get applied to all softphones? |
| 09:39.53 | Delvar | dyn: yes |
| 09:39.57 | dyn | Delvar: cos basically the default IP is what's changing for all users |
| 09:40.11 | dyn | that's great, i'll try it out, thanks! |
| 09:40.14 | Delvar | dyn: ah that wont help then |
| 09:40.18 | dyn | oh |
| 09:40.20 | dyn | hm |
| 09:41.06 | dyn | i'll write a simple bash script then which generates the config into a sip-users.conf which will get included from sip.conf |
| 09:41.19 | Delvar | dyn: all optiosn like DTMFMODE, CONTEXT etc.. can be set in default.. then for each extension you would have to set username,secret,type,defaultip... |
| 09:41.34 | dyn | :( |
| 09:41.39 | dyn | yeah, that's the problem |
| 09:41.58 | dyn | it's just strange that noone found that tiresome enough to develop some configuration helper right into asterisk for this |
| 09:42.29 | Ikarus | Gah, loser sysadmin here has filled his PC with warez |
| 09:42.32 | Delvar | a script is the quickest way to it, just store an array at the top of exten and defaultip |
| 09:42.55 | Delvar | tehn every update ust run your script and reload |
| 09:43.32 | dyn | Delvar: yeah that will work fine |
| 09:43.36 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 09:45.33 | ghenry | anyone recommend a good sip hardware phone? We're in the UK, if that matters (don't think so) |
| 09:46.06 | dyn | ghenry: i'd be interested in that too |
| 09:46.15 | Delvar | ghenry: snom 320 are prety good |
| 09:46.33 | ghenry | I was looking on http://www.voipon.co.uk |
| 09:47.13 | ghenry | http://www.voipon.co.uk/index.php?category=VoIP_IP_Telephones&cPath=1 |
| 09:47.33 | ghenry | <PROTECTED> |
| 09:47.43 | ghenry | quite a lot of cash for that |
| 09:48.07 | ghenry | does asterisk do sips or rtps yet? |
| 09:49.20 | RoyK | anyone that knows a good way to do billing with mail->fax with asterisk? |
| 09:50.26 | Delvar | ghenry: look a tthe grandstream GXP then |
| 09:50.57 | Delvar | http://www.voipon.co.uk/product_info.php?cPath=1_48&products_id=117 |
| 09:51.48 | Delvar | if you want cheaper then a budgetone.. but i dont much like them but they work prety well |
| 09:54.43 | ghenry | thanks |
| 09:54.56 | ghenry | oh, I was reading last night about why this guy doesn't like asterisk |
| 09:55.13 | ghenry | saying that the only reason digium open sourced it was to make money on their hardware |
| 09:55.30 | ghenry | but as I understand it from other sources, the book etc. the hardware came after that |
| 09:55.49 | ghenry | after it was oss, that is. |
| 10:00.20 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 10:02.21 | x86 | are there any sounds in the asterisk-sounds package that have one of those nostalgic number disconnected error tones? |
| 10:02.57 | RoyK | what are those? |
| 10:03.03 | RoyK | nostalgic tones? |
| 10:04.34 | RoyK | x86: do you mean the ones available from playtones? |
| 10:05.57 | x86 | playtones? |
| 10:06.09 | *** join/#asterisk Malthus (n=admin@uslec-66-255-41-2.cust.uslec.net) |
| 10:06.27 | x86 | when i pick up my analog POTS phone connected to the PSTN, and dial a number no longer in service (in the US) |
| 10:06.34 | x86 | it gives a annoying tone |
| 10:06.36 | x86 | i want that ;) |
| 10:07.16 | Malthus | -- Playing 'ss-noservice' (language 'en') |
| 10:07.36 | Malthus | thats the file right there :) |
| 10:08.38 | x86 | ah cool |
| 10:08.41 | x86 | thanks :) |
| 10:08.57 | x86 | Malthus: hmm actually, i just need the tone itself |
| 10:09.07 | *** join/#asterisk xbit` (n=xbit@frugalware.elte.hu) |
| 10:09.09 | xbit` | hi |
| 10:09.10 | Malthus | bah |
| 10:09.14 | Malthus | use a sound editor |
| 10:09.17 | x86 | i want to use it for noservice as well as when someone's account is disabled |
| 10:09.33 | Malthus | edit and take what you want |
| 10:10.03 | Malthus | you might want to download the wav/raw sound files instead of the gsm ones that come by default |
| 10:10.17 | x86 | those are in the asterisk-sounds package? |
| 10:10.22 | Malthus | no |
| 10:10.33 | x86 | where do i get them then? :) |
| 10:10.37 | Malthus | you have to search, I don't remember where I saw them |
| 10:10.47 | Malthus | maybe astlinux's website |
| 10:11.07 | Malthus | I have a funky problem with a e&m wink T1 |
| 10:11.26 | Malthus | it drops part of the DID info |
| 10:11.26 | *** join/#asterisk Jestie (n=Jestie@dsl-165-149-83.telkomadsl.co.za) |
| 10:11.43 | Malthus | so I tried featd instead of em_w |
| 10:12.08 | Malthus | asterisk tells me the line is not featd so its assuming em_w |
| 10:12.19 | Malthus | and when it does that the DID info works fine |
| 10:12.38 | x86 | heh |
| 10:12.41 | Malthus | when I seet it back to em_w in zapata.conf it continues to drop parts! |
| 10:12.57 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 10:15.20 | Malthus | and then, both the em_w AND an fxo_ls link starts to ignore DTMF after asterisk has been running for a while |
| 10:15.42 | Malthus | but they don't both start ignoring at the same time |
| 10:15.53 | Malthus | so one might be working while the other isn't |
| 10:19.41 | x86 | what version of asterisk? |
| 10:20.47 | Malthus | 1.2.4 |
| 10:21.33 | x86 | upgrade |
| 10:21.40 | Malthus | I tried the other em signalling types but they didn't work |
| 10:22.02 | Malthus | downloaded the new packages already |
| 10:22.31 | Malthus | don't see anything in the changelogs that seem relevant |
| 10:26.45 | stoffell | anyone else having the same bristuff problem with hangup problems? (zap not detecting end of call) |
| 10:32.11 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 10:34.55 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 10:36.07 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.79.40) |
| 10:37.35 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 10:48.29 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:50.12 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 10:52.51 | *** join/#asterisk Red5 (n=Red5@CPE-203-56-119-133.ethertech.com.au) |
| 10:57.15 | *** join/#asterisk eds0n (n=edson@pc36-73.vlab.iu.hio.no) |
| 10:57.32 | eds0n | hey, has anyone here been running any tests with astertest? |
| 10:58.46 | eds0n | I've set up two servers, but I have problems connection. The windows gui says "connection refused" and I can't figure out why :-/ |
| 10:59.04 | *** join/#asterisk tdi (n=tdi@80.48.205.2) |
| 10:59.13 | tdi | hi |
| 10:59.44 | tdi | howto switch off silence supression for chansip ? |
| 10:59.55 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 11:00.01 | tdi | between two asterisks |
| 11:02.07 | Red5 | astertest relies on iax interconnects.. 'connection refused' sounds like an iax.conf misconfiguration |
| 11:02.14 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 11:02.51 | Red5 | have you been through the tutorials on asteriskguru ? |
| 11:02.54 | Red5 | http://www.asteriskguru.com/tutorials/astertest.html |
| 11:07.46 | *** join/#asterisk heart (n=zippetto@lugbari/people/heart) |
| 11:10.09 | eds0n | Red5: I've followed the guide at the tutorial |
| 11:20.55 | dyn | can you guys recommend a free and easy-to-use SIP softphone client for windows? |
| 11:21.09 | dyn | for linux, Ekiga (formerly gnomemeeting) is beautiful and working perfectly |
| 11:21.16 | dyn | are there any similar apps for windows? |
| 11:27.30 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:28.06 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-34-22.neo.res.rr.com) |
| 11:28.15 | sleepy_one | greetings everyone :-) |
| 11:28.45 | sleepy_one | anyone have Polycom IP 501s? |
| 11:32.36 | dyn | ok actually i see lots of windows softphones, but are there any which are using a native windows-looking user interface? |
| 11:34.53 | sleepy_one | X-Lite, sjphone |
| 11:35.21 | dyn | X-lite had some extreme UI as I remember |
| 11:36.18 | *** join/#asterisk Boter (n=mitja@local.BSDroot.org) |
| 11:36.24 | Boter | hello |
| 11:36.27 | Boter | anybody here? |
| 11:37.19 | sleepy_one | hi |
| 11:37.34 | Boter | hey :) |
| 11:37.44 | sleepy_one | dyn, "I think" sjphone has a reasonable interface |
| 11:37.47 | Boter | can i take some your time? |
| 11:37.56 | dyn | sleepy_one: i'll check it out |
| 11:38.01 | Boter | today it's first time i installed asterisk |
| 11:38.15 | sleepy_one | Boter, ok, what's your question? |
| 11:38.17 | Boter | as far as i heard and read... there is a way to dial to phone numbers? |
| 11:38.22 | dyn | sleepy_one: ofc the best choice would still be a gaim-plugin as everyone around is using gaim here |
| 11:38.36 | sleepy_one | dyn, ya |
| 11:38.57 | dyn | sleepy_one: i assume no such exists? not even in the upcoming gaim2? |
| 11:39.10 | dyn | Boter: there is |
| 11:39.19 | sleepy_one | Boter, you can use a VoIP provider to make calls to the PSTN over the internet |
| 11:39.34 | sleepy_one | dyn, I'm not sure |
| 11:40.03 | Boter | so i cant be provider itself with this program? |
| 11:41.19 | sleepy_one | Boter, you can be a provider yourself you just need some hardware to connect to the PSTN |
| 11:41.39 | Boter | hm |
| 11:41.51 | Boter | where can i find links about that hardware? |
| 11:42.06 | sleepy_one | digium.com |
| 11:42.13 | oinkmld | digium.com, voip-info.org |
| 11:42.37 | Boter | is it expensive? |
| 11:42.40 | Boter | and do you use it? |
| 11:43.34 | sleepy_one | http://www.digium.com/en/products/hardware/digitalcards.php http://www.digium.com/en/products/hardware/analogcards.php http://sangoma.com/main/products/cards/voice |
| 11:44.46 | sleepy_one | Boter, it depends what are you trying to do exactly? You can pickup cheap X100p clones for $15 each on ebay or you can spend $200 or more on some TDM400p's or $500 and up for T1 cards |
| 11:45.10 | Boter | well |
| 11:45.13 | sleepy_one | Boter, it all depends on what you need |
| 11:45.28 | Boter | i want to make free phone calls with this voip :) |
| 11:45.29 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-37-dsl.end.estpak.ee) |
| 11:45.37 | oinkmld | I have 3 T0 lines coming in my office, what card should I use ? |
| 11:45.58 | Boter | so i can call others free |
| 11:46.13 | sleepy_one | oinkmld, T0? you mean ISDN? |
| 11:47.25 | Boter | so what equipment would i need for that? |
| 11:47.27 | sleepy_one | Boter, in that case you might want to look at FWD ( free world dialup ) http://www.freeworlddialup.com/ but not everything is free |
| 11:47.46 | sleepy_one | Boter, are you running asterisk now |
| 11:47.57 | Boter | i just stopped |
| 11:47.59 | Boter | why? |
| 11:48.06 | sleepy_one | what kind of computer do you have? |
| 11:48.20 | oinkmld | Yes, it's called T0 in France |
| 11:48.28 | Boter | sleepy_one hm |
| 11:48.32 | Boter | router is 133mhz |
| 11:48.34 | Boter | linux |
| 11:48.43 | Boter | but i can create better router if needed |
| 11:48.45 | sleepy_one | Boter, you do not need any hardware to make VoIP calls, just a Linux computer and asterisk |
| 11:49.16 | Boter | hm |
| 11:49.21 | Boter | thats good than :) |
| 11:49.42 | Boter | what do i need to set now? |
| 11:49.59 | Boter | i dont find any man pages which it shows me how to make calls |
| 11:50.07 | sleepy_one | oinkmld, look at the ISDN cards here http://www.asterisk.org/hardware |
| 11:50.43 | sleepy_one | Boter, what kind of internet connection do you have? |
| 11:50.57 | Boter | adsl 2mbit/384kbit |
| 11:51.01 | Boter | i have isdn too |
| 11:52.08 | sleepy_one | oinkmld, "ISDN4Linux Any ISDN terminal adapter supported by isdn4linux should provide connectivity." theoretically any Linux supported ISDN card should work |
| 11:52.57 | Boter | soon will have t2 10/2 |
| 11:52.58 | oinkmld | Ok. Thanks ;-) |
| 11:53.02 | sleepy_one | Boter, you have to configure asterisk by editing the config files in /etc/asterisk or by using a web interface like FreePBX ( formerly known as AMP ) or astguiclient or something |
| 11:54.55 | Boter | ok i'm going to install freepbx from sourceforge |
| 11:55.23 | Boter | i looked config file of asterisk and there i dont see very much useful stuff |
| 11:55.35 | sleepy_one | asterisk comes with configuration examples in asterisk-*/configs asterisk-1.2.7.1/configs for the current version |
| 11:55.58 | Boter | would this work from cygwin? |
| 11:56.05 | Boter | just Q i'm not gonna try it ;:) |
| 11:56.16 | sleepy_one | Boter, maybe |
| 11:56.28 | sleepy_one | the kernel modules wouldn't work |
| 11:56.41 | Boter | k :) |
| 11:56.53 | Boter | so i must than make calls directly from router (linux box) |
| 11:57.35 | sleepy_one | Boter, there's documentation with examples in asterisk-1.2.7.1/configs also asterisk-1.2.7.1/doc and online @ http://asterisk.org and http://voip-info.org/wiki/ |
| 11:57.51 | dyn | does asterisk support the SIMPLE protocol? |
| 11:58.02 | Boter | i must set it before freepbx? |
| 11:58.06 | sleepy_one | dyn, I dunno |
| 11:58.17 | sleepy_one | dyn, tried the wiki? |
| 11:58.36 | sleepy_one | Boter, do you have anything better than a 133MHz machine? |
| 11:58.39 | dyn | sleepy_one: the problem is that you can hardly find correct matches on a search query like 'simple' :) |
| 11:58.52 | sleepy_one | dyn, true |
| 11:58.52 | Boter | sleepy_one not yet |
| 11:59.04 | dyn | sleepy_one: btw, gaim2 seems to support the Sip/SIMPLE protocol (whatever that is) |
| 11:59.06 | Boter | but if it will be needed i will set up bettet machine |
| 11:59.17 | dyn | sleepy_one: so currently i'm trying to get both sides configured for a quick test |
| 11:59.40 | sleepy_one | Boter, asterisk will run on older hardware but your results will vary quite a bit, I would recommend a 500MHz CPU and at least 128MB RAM the more the better |
| 12:00.18 | dyn | damn, SIMPLE is an IM-only protocol |
| 12:00.21 | dyn | no Sip calls there |
| 12:00.28 | Boter | ok i have enought RAM :) |
| 12:00.46 | Boter | sleepy_one i'll just test this and than see if it's worth of new hardware |
| 12:01.04 | sleepy_one | Boter, it's worth it |
| 12:01.08 | Boter | hm ok now i go set .conf files in /etc/asterisk |
| 12:01.37 | Boter | so i go from 1 conf to another? |
| 12:02.11 | sleepy_one | Boter, you can do make samples and make progdocs |
| 12:02.19 | dyn | Boter: i suggest reading the asterisk handbook. it's a draft version but it's a great way to get into the basics |
| 12:02.24 | sleepy_one | which will install the default config files |
| 12:02.26 | dyn | Boter: (i'm right on doing that myself :) |
| 12:02.41 | MrChimpy | boter: better still, buy the asterisk book |
| 12:02.44 | Boter | sleepy_one i installed from .deb archive |
| 12:03.04 | PiPiPi | man voip-info's wiki is plain UGLY |
| 12:03.09 | MrChimpy | only it lacks stuff about the E1 cards |
| 12:03.10 | PiPiPi | somebody has to add some CSS to it with class |
| 12:03.24 | sleepy_one | Boter, you should only have to mess with iax.conf sip.conf and extensions.conf |
| 12:03.49 | sleepy_one | Boter, you can read all about it at the wiki |
| 12:03.50 | sleepy_one | ~wiki |
| 12:04.05 | sleepy_one | ~wiki |
| 12:04.08 | sleepy_one | ~docs |
| 12:04.09 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:04.28 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 12:05.06 | tdi | what is the best way to send fax between two asterisks using SIP channel ? |
| 12:08.00 | sleepy_one | tdi, http://www.voip-info.org/wiki-Asterisk+fax |
| 12:08.08 | sleepy_one | hey file :-D |
| 12:08.23 | file | I'm the last one left for AstriDevCon Europe |
| 12:10.09 | sleepy_one | hehe |
| 12:10.37 | Boter | ok i go now checking the handbook |
| 12:10.42 | file | my flight doesn't leave till tonight at 8:30, I've already been all over Pisa with oej... |
| 12:10.49 | *** join/#asterisk cced1 (n=dev2003@222.33.36.205) |
| 12:10.52 | file | so I'm sitting here at the hotel... paid 5 Euros for 5 hours of internet... |
| 12:11.13 | tdi | sleepy_one: i know this doc, there is no info howto send fax thru sip channel betwoeen two asterisks directly and howto switch off silence supression, which is a problem in my case |
| 12:12.02 | sleepy_one | tdi, did you see the "Sending a fax to a SIP device" section? |
| 12:12.10 | tdi | yes |
| 12:12.34 | file | why am I suddenly getting such decent speeds here... where all of this week it's been previously crappy |
| 12:12.55 | oinkmld | where's the astricon ? |
| 12:13.26 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:13.35 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 12:13.35 | tdi | asterisks send silence supression in SIP packets, but they do not appear to take it under consideration |
| 12:13.43 | tdi | just ignore them |
| 12:13.53 | file | oinkmld: it was a developer get together essentially... |
| 12:14.00 | [TK]D-Fender | file : Lasik day today! |
| 12:14.14 | file | [TK]D-Fender: ooh |
| 12:16.01 | file | I wonder how long my flight to Paris is |
| 12:16.17 | MrChimpy | where from? |
| 12:16.19 | sleepy_one | file, where are you flying out of? |
| 12:16.25 | file | Pisa, Italy |
| 12:16.33 | MrChimpy | oh, about an hour then |
| 12:16.45 | file | ooh k |
| 12:16.47 | sleepy_one | should be couple of hours depending on traffic |
| 12:17.33 | sleepy_one | about one hour transit time + a few hours in a holding pattern over CDG |
| 12:17.44 | file | this hotel has better internet then I have at home :\ |
| 12:17.56 | file | well |
| 12:17.59 | wasim | better maid service too, i'm sure |
| 12:18.14 | file | I fly out of here to Paris, then I'm staying overnight in a hotel, then flying to Montreal and finally home |
| 12:18.21 | MrChimpy | i keep forgetting what I'm doing. today it's mostly trying not to buy a pentium 805 |
| 12:18.45 | MrChimpy | look out for the mo-turd-bikes in paris |
| 12:18.46 | oinkmld | file: Any plans in France ? ;-) |
| 12:18.47 | sleepy_one | MrChimpy, oh my gosh an 805????? |
| 12:18.58 | MrChimpy | they invented the motorbike dogshit vaccum cleaner |
| 12:19.04 | file | sleep, sleep, oh and sleep |
| 12:19.11 | shiznatix | what settings can i play around with to try to make my faxing more reliable. right now i have like 70% success with spandsp |
| 12:19.39 | oinkmld | file: You should come over and have a beer ;-) |
| 12:19.44 | sleepy_one | MrChimpy, the 805 will cost you a FORTUNE in cooling and power costs if it doesn't explode on you |
| 12:19.50 | file | meh |
| 12:20.06 | MrChimpy | yep. i was thinking of less than the 4GHz :) |
| 12:20.40 | sleepy_one | MrChimpy, @ 4.1GHz is consumes over 260W while IDLE :-[ |
| 12:20.42 | file | my cellphone bill grows ever higher |
| 12:20.46 | MrChimpy | :) |
| 12:20.55 | MrChimpy | aye, didn't escape my notice :) |
| 12:21.07 | MrChimpy | 3 would do nicely |
| 12:21.12 | sleepy_one | shiznatix, are you faxing over IP or PSTN ? |
| 12:21.29 | MrChimpy | though I think i'll hold on and get a desktop core due |
| 12:21.30 | MrChimpy | duo |
| 12:21.35 | [TK]D-Fender | sleepy_one : For your earlier question, I have an IP 501 (and every other desk model) whats your question on it? |
| 12:21.52 | MrChimpy | just the hacker in me wants whizzy things that ain't supposed to go so fast |
| 12:22.18 | ghenry | is a Cisco 7960G Good with Asterisk? |
| 12:22.29 | sleepy_one | ghenry, yes it can be very good |
| 12:22.41 | sleepy_one | ghenry, with the SIP firmware |
| 12:22.52 | ghenry | which is extra? |
| 12:22.54 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:22.57 | Boter | btw sleepy_one is possible to have own "phone number"? |
| 12:22.59 | shiznatix | sleepy_one, PSTN |
| 12:23.15 | ghenry | and that does the directory search feature? |
| 12:23.25 | ghenry | any other phones do the directory search? |
| 12:23.45 | sleepy_one | shiznatix, it should be more reliable than that.... what kind of analog card ( s ) are you using? PSTN faxing should be 95% ok |
| 12:24.17 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 12:24.45 | sleepy_one | ghenry, you should be able to download the firmware from cisco if you have a cisco.com account or find it around the net |
| 12:24.58 | ghenry | thanks |
| 12:25.16 | sleepy_one | ghenry, the 7960G supports html directories served from a web server |
| 12:25.30 | ghenry | cool |
| 12:26.18 | ghenry | 7902G looks ok. a bit cheaper though |
| 12:27.19 | ghenry | ZyXEL Prestige 2000W Wireless IP Phone looks quite cool too |
| 12:27.35 | ghenry | brb |
| 12:28.51 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 12:30.34 | sleepy_one | ghenry, I don't know how well the lower end Cisco's work, the 7960s work fine with SIP 7.x |
| 12:31.25 | shiznatix | sleepy_one, Digium TIGER 320 is the card |
| 12:31.48 | sleepy_one | shiznatix, you mean TDM400p ? |
| 12:32.14 | *** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net) |
| 12:33.30 | sleepy_one | shiznatix, many cards have a Tiger 3xx chipset including the X100p clones and the real Digium TDM400p's http://www.digium.com/en/products/hardware/tdm400p.php is this what you have? |
| 12:34.49 | sleepy_one | shiznatix, I have at least two of them: 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 12:35.22 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:35.31 | cced1 | :) |
| 12:36.49 | *** join/#asterisk buzzdee (n=buzz@174.56.204.212.ediscom.de) |
| 12:40.29 | *** join/#asterisk Uzzi (n=Andrea@host229-236.pool872.interbusiness.it) |
| 12:40.53 | *** join/#asterisk lorinc (n=ang@caracas-4235.adsl.interware.hu) |
| 12:41.01 | ghenry | thanks sleepy_one |
| 12:41.17 | sleepy_one | ghenry, yw |
| 12:41.37 | ghenry | will go for a slightly cheaper one |
| 12:42.09 | ghenry | probably a Grandstream GXP 2000 IP Telephone |
| 12:42.23 | sleepy_one | ghenry, what about Linksys? |
| 12:42.36 | ghenry | not sure |
| 12:42.40 | sleepy_one | ghenry, I think they are better than Grandstream / Crapstream |
| 12:42.46 | ghenry | Ha |
| 12:42.55 | ghenry | something £70<>£90 |
| 12:42.58 | ghenry | x2 |
| 12:43.30 | ghenry | can't seem to find those on voipon.co.uk |
| 12:43.49 | ghenry | what about sipura |
| 12:44.00 | ghenry | snom 300 |
| 12:44.04 | ghenry | or snom |
| 12:44.30 | shiznatix | sleepy_one, Wildcard TDM400P is my card to be a bit more specific |
| 12:46.05 | buzzdee | hi |
| 12:46.10 | buzzdee | i have a problem dialling directly into my second conference room from an external number |
| 12:46.18 | buzzdee | have the configuration and logs there: http://pastebin.com/713405 |
| 12:46.30 | ghenry | this looks quite good sleepy_one http://www.voipon.co.uk/product_info.php?cPath=1_79&products_id=260 |
| 12:48.25 | coppice | that's quite expensive |
| 12:49.11 | [TK]D-Fender | and FUGLY. |
| 12:49.44 | ghenry | ha |
| 12:49.49 | ghenry | so on this page: http://www.voipon.co.uk/index.php?category=VoIP_IP_Telephones&cPath=1 |
| 12:49.52 | ph|ber | if my PRI lines are working for incoming, will the outgoing automatically work? |
| 12:49.53 | ghenry | what are good? |
| 12:50.03 | ghenry | !>=£90 |
| 12:50.25 | ghenry | unless there's a better site somewhere else in the UK? |
| 12:50.44 | coppice | phlber: not necessarily |
| 12:51.00 | [TK]D-Fender | ghenry : JUST over 90 is the Polycom IP 301 |
| 12:51.48 | [TK]D-Fender | ghenry : Hate to say I'd also rather suggest an Aastra 9133 at 91 |
| 12:52.07 | [TK]D-Fender | SPA-941 @ 100 is pretty good too... |
| 12:52.17 | austinnichols10 | 9133's not bad |
| 12:52.20 | [TK]D-Fender | possibily the best value on that site |
| 12:52.54 | austinnichols10 | solid handset - can do some damage with it |
| 12:54.30 | [TK]D-Fender | austinnichols10 : Yeah, Aastra is a solid product typically, but I find their interface a little too "old analog" for my tastes. Not bad for "dumb business" who may be LUCKY to even pick up their phones ;) |
| 12:55.12 | *** join/#asterisk TheBigSpark (n=TheBigSp@71.39.194.197) |
| 12:55.38 | austinnichols10 | [TK]D-Fender: agree completely. I didn't like the 941 because it kept tipping over (topheavy) |
| 12:56.12 | austinnichols10 | [TK]D-Fender: whereas for around the same money the 9133 was stable and I can knock someone unconcious with the handset |
| 12:56.30 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) |
| 12:56.39 | [TK]D-Fender | austinnichols10 : I felt it was too light overall, the feet were more rubber than foam and slips a lot, and the prop was uber cheap |
| 12:56.57 | [TK]D-Fender | austinnichols10 : Yes, the Aastra Tactical Defense Phone strikes again! |
| 12:57.29 | austinnichols10 | [TK]D-Fender: definitely. However, it didn't just scream 'cheap' sitting on a desk like some of the other sets. I have one in the cabinet at the data center and then ciscos at the office (gotta present a certain image) |
| 12:57.41 | *** join/#asterisk aetius (n=aetius@cpe-069-134-208-043.nc.res.rr.com) |
| 12:58.38 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) |
| 12:58.48 | [TK]D-Fender | austinnichols10 : Cisco makes great products.... overpriced, and over controlled, but still VERY nice products |
| 12:59.04 | austinnichols10 | [TK]D-Fender: Might be good to have a test to see how many strikes a specific phone can handle. I'm thinking the 941's are gone in 1-2. |
| 12:59.08 | [TK]D-Fender | austinnichols10 : I still feel that Aastra is a cheap-out alternative to Polycom however. |
| 12:59.41 | [TK]D-Fender | austinnichols10 : Asuming no LCD-direct its still solid, if light.... |
| 12:59.46 | austinnichols10 | [TK]D-Fender: I have a definitely love/hate with those ciscos. |
| 13:00.45 | [TK]D-Fender | austinnichols10 : Top 3 "hate" factors for you are? |
| 13:00.47 | austinnichols10 | [TK]D-Fender: I keep meaning to try polycom w/Asterisk. Their PSTN stuff is great. |
| 13:01.34 | [TK]D-Fender | austinnichols10 : Let me say that its worth every penny. |
| 13:01.40 | austinnichols10 | 1. Put all of the features I need in the firmware and document it well |
| 13:01.57 | [TK]D-Fender | austinnichols10 : And now with SIP 1.6.6 BLF support makes the 601 + Att Modules godly. |
| 13:02.17 | austinnichols10 | 2. notify |
| 13:02.23 | *** join/#asterisk Becky75 (n=pirch@dsl-165-220-194.telkomadsl.co.za) |
| 13:02.39 | austinnichols10 | 3. Backlight (I just spent a bunch of freaking money. If the linksys has one I should too) |
| 13:02.45 | austinnichols10 | But still worth the money |
| 13:03.15 | Becky75 | hi all... I am getting dead air in calls from sip to my zap trunk from sip to sip its fine how can i get rid of it its some times up to 4 seconds |
| 13:03.19 | austinnichols10 | [TK]D-Fender: Plus there's that occasional voodoo ritual I have to do to get one flashed... |
| 13:03.43 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:04.20 | [TK]D-Fender | austinnichols10 : Back-light is VERY rare on any phone really... I don't hold it against any of them, only as a nice PLUS for one I'm evaluating. |
| 13:04.37 | austinnichols10 | [TK]D-Fender: all nice-to-haves, IMO |
| 13:05.30 | [TK]D-Fender | austinnichols10 : Based on rarity I's a plus, not a minus for not having. |
| 13:05.34 | Becky75 | yeah if i push the buttons and it dials then its purfect.. back-light is a nice to have... unless u like working in the dark :P |
| 13:05.52 | [TK]D-Fender | austinnichols10 : just my POV. try to compare on what IS normally there. |
| 13:07.03 | ghenry | austinnichols10, [TK]D-Fender I think I'll get 2 9133 then |
| 13:07.23 | MrChimpy | does asterisk support rotary dialling? |
| 13:07.26 | [TK]D-Fender | ghenry : Rather than the SIP TONE, yeah.... though its a toss-up with the 941.... |
| 13:07.29 | MrChimpy | yes, i'm that bored. |
| 13:07.54 | [TK]D-Fender | MrChimpy : I just want a hand-crack to ring central ;) |
| 13:07.59 | [TK]D-Fender | crank* |
| 13:08.10 | Becky75 | I am getting dead air in calls from sip to my zap trunk from sip to sip its fine how can i get rid of it its some times up to 4 seconds |
| 13:08.14 | ghenry | [TK]D-Fender: who makes the 914? |
| 13:08.25 | [TK]D-Fender | MrChimpy : Actually "immediate=yes" would probably do :) |
| 13:08.35 | [TK]D-Fender | ghenry : Linksys |
| 13:08.57 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 13:08.57 | ghenry | where can you get that from in the UK? |
| 13:09.02 | ghenry | This one vs the 914: http://www.voipon.co.uk/product_info.php?cPath=1_50&products_id=124 |
| 13:09.30 | [TK]D-Fender | ghenry : http://www.voipon.co.uk/product_info.php?cPath=1_57&products_id=159 |
| 13:09.40 | elg | does anyone here use ARA (realtime) over odbc for voicemail settings? |
| 13:09.44 | ghenry | The 9133 does the XML Directory thing, yeah? |
| 13:10.21 | ghenry | [TK]D-Fender: I'll get the slightly cheaper one ;-) |
| 13:11.02 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 13:11.06 | Becky75 | am i speaking to the walls to day boys |
| 13:11.08 | [TK]D-Fender | ghenry : depends on your call usage. If its really low I'd say go with the Aastra |
| 13:11.23 | ghenry | yeah, low |
| 13:11.38 | clive- | becky, its called dead air :) |
| 13:11.41 | [TK]D-Fender | Becky75 : Is this delay on incoming, outgoing or both? |
| 13:12.00 | austinnichols10 | ghenry: porsche |
| 13:12.01 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 13:12.16 | Becky75 | its only on out going audio on zap channels |
| 13:12.28 | ghenry | austinnichols10: eh? |
| 13:13.04 | austinnichols10 | 914 |
| 13:13.04 | Becky75 | from sip to zap i just get up to 4 seconds of dead air |
| 13:13.26 | austinnichols10 | actually vw did the assembley |
| 13:14.05 | ghenry | austinnichols10: http://www.voipon.co.uk/product_info.php?cPath=1_57&products_id=159 ??? |
| 13:14.08 | [TK]D-Fender | Becky75 : pastebin your zaptel, zapata, and relevant sections of extensions.conf |
| 13:15.06 | *** join/#asterisk ClubBarf (n=spamme@host-212-158-228-2.bulldogdsl.com) |
| 13:15.42 | *** part/#asterisk ClubBarf (n=spamme@host-212-158-228-2.bulldogdsl.com) |
| 13:16.39 | Becky75 | ok busy in pastebin |
| 13:17.03 | Becky75 | my extentions.conf is a mess just the way freepbx likes to make it :| |
| 13:18.38 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 13:18.46 | elg | i see ARA connect to the db, and i have this in extconfig.conf: "voicemail => odbc,asterisk.voicemail_users" |
| 13:18.50 | MrChimpy | club barf sounds like the follow up to my night club |
| 13:18.55 | MrChimpy | http://www.clubbeer.co.uk |
| 13:20.10 | *** join/#asterisk S4w (n=saw@adsl-3-65-52.mia.bellsouth.net) |
| 13:20.57 | shiznatix | when i convert a .pdf to a .tiff to send over a fax it works fine if the .pdf only have letters and maybe some small. The problem comes when i try to fax a complex pdf with some crazy stuff on it. it just faxes a blank page and thats it |
| 13:20.59 | S4w | hello guys, I have a question. If I am using a T1 with a digium card will I have the same answer detection problems that I run into with a normal FXO card like the x100p? |
| 13:22.51 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 13:23.16 | *** part/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 13:23.45 | [TK]D-Fender | Becky75 : oK, NEVERMIND... NEVER MENTIONED fREEpbx TILL NOW.... |
| 13:23.49 | sevard | Does anyone have access to that database which shows you who owns / which carrier is providing a toll free 8XX ? |
| 13:24.15 | wasim | S4w: no |
| 13:24.53 | S4w | wasim: ohh, it uses a special technique to detect answer or what? |
| 13:25.23 | wasim | S4w: yes, its called signalling |
| 13:25.43 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.161.Dial1.SanJose1.Level3.net) |
| 13:26.14 | wasim | oh answer detection ... sorry i read that as hangup detect |
| 13:26.20 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 13:26.23 | wasim | answer by a person, or just in general? |
| 13:26.27 | S4w | wasim: :-O |
| 13:26.32 | S4w | wasim: by the person |
| 13:26.36 | ghenry | [TK]D-Fender: So 2 of these it is then: http://www.voipon.co.uk/product_info.php?cPath=1_50&products_id=124 |
| 13:26.52 | ghenry | A good phone with plenty of features for the future? |
| 13:27.15 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 13:27.20 | coppice | ah, the mystical human answering a phone detector :-) |
| 13:27.32 | S4w | hmm? :| |
| 13:28.23 | mercestes | I think he's referring to a device that can "detect" artificial voices, can't remember what it's called. |
| 13:29.00 | S4w | no no no, the t1 card can detect when a zap channel is answered? |
| 13:29.57 | S4w | because if you want to bill you need answer detect of some sort so that you dont start charging your customer when the channel starts ringing |
| 13:30.50 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:31.02 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:31.13 | mercestes | Hell, charge them when they start dialing...pathetic mortals...wanting their free dial tones. |
| 13:31.26 | S4w | hmmm... |
| 13:33.22 | jake1932 | S4w: you're not getting reliable answer detection on a T1 card? |
| 13:33.43 | jake1932 | a T1 card connected to the CO? |
| 13:33.44 | shiznatix | when i convert a .pdf to a .tiff to send over a fax it works fine if the .pdf only have letters and maybe some small. The problem comes when i try to fax a complex pdf with some crazy stuff on it. it just faxes a blank page and thats it |
| 13:34.04 | S4w | jake1932: I am just asking if I would run into the same problems that I run with an analog FXO adaptor like the x100p with a T1 line |
| 13:34.12 | jake1932 | no |
| 13:34.27 | sleepy_one | shiznatix, are you sure the PDF to TIFF conversion is being done correctly? |
| 13:34.37 | S4w | jake1932: so it will detect always successfuly answers? |
| 13:34.55 | jake1932 | S4w: you should get a message over the D channel that says the call is connected |
| 13:35.12 | jake1932 | whether or not a human answers is a seperate issue |
| 13:35.21 | S4w | jake1932: the thing is taht I dont want to invest on a T1 like if it have such problems |
| 13:35.30 | clive- | jake1932, sometimes one expereinces that when calling countries like india etc, but normally a T1 is accurate with answer detection |
| 13:36.02 | *** join/#asterisk Kyler (n=chatzill@12.208.60.224) |
| 13:36.18 | S4w | what is the process involved in that? or is it the driver itself that detects answers? |
| 13:37.08 | jake1932 | asterisk can interpret those messages from the d channel |
| 13:37.18 | Kyler | I'm experimenting with different ways of compressing .wav files resulting from Dictate(). Any suggestions? I noticed that -i and -A cause Asterisk to report "check_header: Not a wav file 17" if I try to play back the resulting files. |
| 13:37.54 | jake1932 | no special drivers are necessary |
| 13:38.44 | Becky75 | ok i get any idea why i will have dead air on a phone call some times up to 4 seconds |
| 13:38.49 | *** part/#asterisk clive- (n=pirch@dsl-165-188-238.telkomadsl.co.za) |
| 13:39.16 | jake1932 | Becky75: in what circumstances - what config? |
| 13:39.46 | MrChimpy | you can detect that easily |
| 13:39.50 | Becky75 | jake1932 isdn junghanns octo bri card its only on zap calls |
| 13:39.58 | MrChimpy | scrollback dammit |
| 13:39.59 | Becky75 | only on out going or in comming calls |
| 13:40.46 | Becky75 | its some times up to 4 seconds of dead air then it comes back |
| 13:40.49 | [TK]D-Fender | ghenry : Oh you want to know what a GOOD phone is? Polycom/Cisco, but you are aiming to low for either. |
| 13:41.05 | *** part/#asterisk tdi (n=tdi@80.48.205.2) |
| 13:41.07 | [TK]D-Fender | jake1932 : FreePBX |
| 13:41.14 | Kyler | How do I use a comma (not as a delimiter) in a command? Is there a way to escape it? |
| 13:41.54 | Becky75 | polycom is good phones |
| 13:41.57 | jake1932 | <PROTECTED> |
| 13:42.25 | [TK]D-Fender | jake1932 : What Becky75 is using for configs |
| 13:42.31 | jake1932 | aargh |
| 13:42.49 | [TK]D-Fender | Kyler : Can you clarify that a bit? |
| 13:43.06 | Becky75 | yeah well i asked for help on standard asterisk here b4 and no one helped so what am i saposed to do?... download aah 2.8 and load it heh |
| 13:43.24 | Kyler | Looks like I just solved it. I can escape a comma with a backslash. Then it's not interpreted as a delimiter in a command. |
| 13:43.36 | mercestes | Becky75: Well that just makes it worse....now instead of ignoring you we point you to channel #freepbx..:P |
| 13:43.37 | [TK]D-Fender | Becky75 : Bad timing and didn't ask the right people the right questions... |
| 13:43.46 | [TK]D-Fender | mercestes : lol |
| 13:43.57 | Becky75 | lol yeah well i asked and re asked |
| 13:44.01 | shiznatix | sleepy_one, yes the conversion is done just fine. i open up the converted tiff and its correct then it won't send |
| 13:44.08 | Becky75 | and all i got was dont flood and dont repeat yourseldf |
| 13:44.09 | mercestes | I'm kidding Becky..:P I'm sorry I wasn't here or I would've helped you. |
| 13:44.11 | Becky75 | and all i got was dont flood and dont repeat yourself |
| 13:44.24 | mercestes | don't flood and don't repeat yourself, Becky75 |
| 13:44.28 | Becky75 | yeah right mercestes and i just saw a pig fly past my window |
| 13:44.39 | jake1932 | Becky75: just read up a little and install and get asterisk installed without all those nasty configs - i spent about 30 minutes on a simple issue yesterday because those damn things were so convoluted. |
| 13:44.42 | sleepy_one | shiznatix, very strange |
| 13:44.49 | [TK]D-Fender | Becky75 : Maybe it was a frequency thing. Describe the harware and connectivity of your PBX. |
| 13:45.13 | jake1932 | include - include include |
| 13:45.18 | Becky75 | ISDN BRI phone lines connected to a junhanns octo card |
| 13:45.40 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 13:45.43 | [TK]D-Fender | Becky75 : Yeah that is harder to get help on for sure.... |
| 13:46.03 | [TK]D-Fender | Becky75 : Need to be on at the right time to find people experienced with BRI |
| 13:46.08 | Becky75 | yeah well asterisk in general is hard to get help on |
| 13:46.24 | Becky75 | ppl either wanna charge you or they just ignore you |
| 13:46.25 | shiznatix | sleepy_one, yes strange but do you have any ideas |
| 13:46.28 | [TK]D-Fender | Becky75 : I beg to differ given my excessive presence here :) |
| 13:46.37 | kaz0358 | does anyone have a few moments to help me out with some testing? i just want to confirm that i do have an echo problem with our new linksys WIP300 |
| 13:46.40 | Becky75 | the days of linux ppl helping each other just seem to be long gone |
| 13:46.46 | jake1932 | or just e-mail jugghanns |
| 13:46.46 | kaz0358 | i'm looking for someone not on another voip phone. :) |
| 13:47.13 | [TK]D-Fender | kaz0358 : So that leaves what? calling someone on PSTN? |
| 13:47.13 | Becky75 | tried to email junghanns he says look at your zapata fine |
| 13:47.22 | Becky75 | just managed to get rid of echo on this damn thing |
| 13:47.50 | Becky75 | boss's is at least looking at a proper avaya voip solution maybe it will work and the support will be better than asterisk and on here |
| 13:47.54 | kaz0358 | fender, exactly.. but i don't have any too readily.. i need someone long distance and everyone i personally know is at work and not in a position to chat for a few moments |
| 13:48.05 | kaz0358 | err anyone too readily |
| 13:48.05 | jake1932 | oh god - avaya support |
| 13:48.17 | mercestes | Feel better, Becky75? |
| 13:48.23 | jake1932 | i've been waiting 4 days for an answer to a simple query with them |
| 13:48.38 | jake1932 | and i already paid for it!!! |
| 13:48.59 | Becky75 | jake1932 well i waited for months here for a simple answer and never got it |
| 13:49.03 | Becky75 | thats why i loaded aah |
| 13:49.03 | ph|ber | where do you configure the outgoing mappings for a PRI line? |
| 13:49.10 | mercestes | She reminds me of my ex-girlfriend. Where do you live Becky? |
| 13:49.11 | Becky75 | rather wait for days then for months |
| 13:49.26 | jake1932 | i still haven't got an answer Becky75 |
| 13:49.26 | pythos | anyone else using "TDM400P REV E/F (4 modules)" ?? I have some of the config files right, but having trouble with zapata.conf <I believe> because I don't get dial-tone on handset <fxs ports> |
| 13:50.39 | Becky75 | mercestes there is a reason i moved far away form you :P |
| 13:50.51 | Becky75 | err from |
| 13:50.57 | jake1932 | that plot thickens |
| 13:51.22 | jake1932 | Becky75: is it on the beginning of calls? |
| 13:51.43 | [TK]D-Fender | ph|ber : What have you done so far? |
| 13:51.51 | Becky75 | jake1932 nope i understand begining of calls but in the middle or like 2 minutes it really at random |
| 13:51.54 | ph|ber | ok.. i have the inbound working. |
| 13:52.00 | ph|ber | but the outbound is not going. |
| 13:52.27 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:52.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:52.31 | [TK]D-Fender | ph|ber : And clarify outbound mappings? |
| 13:52.31 | [TK]D-Fender | pythos : Does * start? And did you plug the molex connector on your card to your PSU? |
| 13:52.43 | ph|ber | well, i cant make an outbound call. |
| 13:52.55 | [TK]D-Fender | ph|ber : Sho me what you'd tried for setting up outbound. |
| 13:53.20 | ph|ber | exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,1,${EXTEN},,) |
| 13:53.20 | ph|ber | exten => _1NXXNXXXXXX,n,Macro(outisbusy,) |
| 13:53.21 | [TK]D-Fender | ph|ber : And it'd help if you showed output detailing the errors you get. |
| 13:53.22 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 13:53.26 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 13:53.30 | [TK]D-Fender | ph|ber : PASTEBIN please |
| 13:53.31 | [TK]D-Fender | ~pb |
| 13:53.34 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:53.55 | jake1932 | Becky75: any output from the CLI when it happens? |
| 13:53.57 | [TK]D-Fender | ph|ber : And show the macro's you are calling. why should I trust they are right jsut because you are calling them? |
| 13:53.59 | ph|ber | its trying to go through a zap? |
| 13:54.21 | *** join/#asterisk hgaillac (n=Harry@151.15.119-80.rev.gaoland.net) |
| 13:54.23 | Becky75 | jake1932 nothing |
| 13:54.34 | hgaillac | hello, |
| 13:54.52 | *** join/#asterisk \etc\bin (n=root@210.5.103.28) |
| 13:55.28 | hgaillac | Is it possible to set up call park and moh between sip and fxo line |
| 13:55.29 | *** part/#asterisk S4w (n=saw@adsl-3-65-52.mia.bellsouth.net) |
| 13:55.34 | jake1932 | Becky75: the thing with those card is that Jugghans is probably the best one to support them. but you might want to at least get regular asterisk installed just to rule out any funky aah configs |
| 13:56.19 | Becky75 | yeah but when i asked for help on normal astaerisk here on simple stuff a noob saposed to ask i get no help |
| 13:56.34 | ghenry | what do you guys think of sipX? |
| 13:56.37 | ghenry | just reading about it |
| 13:56.51 | mercestes | *raises hand* I'm here for the next 8 hours. |
| 13:56.58 | jake1932 | Becky75: what did you ask that you got know help on? |
| 13:56.58 | mercestes | :) |
| 13:57.02 | jake1932 | no |
| 13:57.08 | [TK]D-Fender | ghenry : nifty in its own way, but no native hardware capability, and not as flexible as *, but more scalable for more "basic" use |
| 13:57.41 | ghenry | ok |
| 13:58.00 | jake1932 | (working on mind/hand relationship) |
| 13:58.00 | Becky75 | jake1932 simple things like how to make a queue and how to make agents simple stuff |
| 13:58.05 | jake1932 | oh |
| 13:58.16 | jake1932 | yeah - i can see getting bashed for that |
| 13:59.24 | jake1932 | the deal here is that people will help with a problem if they can - but you're not going to usually get a step by step - key is usuing the available info first and then coming here after googling a bit |
| 14:00.34 | jake1932 | for instance - you should be able to find info on how to set up a queue on voip-info.org |
| 14:00.46 | ghenry | look forward to testing those new phones ;-) |
| 14:00.49 | ghenry | http://www.voipon.co.uk/product_info.php?cPath=1_50&products_id=124 |
| 14:05.09 | *** join/#asterisk DeeJayTwo (n=deejay2@37-179.sh.cgocable.ca) |
| 14:05.37 | hgaillac | IS IT POSSIBLE TO PARK and PUT ON MOH A CALLER BETWEEN PSTN AND SIP AGENTS ???????????? |
| 14:05.53 | *** join/#asterisk mugawump (n=bbentley@rrcs-24-172-3-11.midsouth.biz.rr.com) |
| 14:05.58 | nahirean | whOA toggle the caps l0ck :) |
| 14:06.04 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 14:06.56 | DeeJay[2] | hi... |
| 14:06.57 | *** join/#asterisk keyhack (n=keyhack@68.236.93.219) |
| 14:07.15 | *** join/#asterisk Tili (i=Tili@202.133.67.86) |
| 14:07.16 | DeeJay[2] | does anybody knows how to disable the polycom's feature that consists in automatically adjusting the gain? |
| 14:07.25 | DeeJay[2] | It causes a lot of echo problems.. |
| 14:08.00 | coppice | polycoms don't EC the handset. this is a very low end thing to do |
| 14:08.21 | DeeJay[2] | ? |
| 14:08.29 | Becky75 | DeeJay[2] weird i have no problem with polycom on my * box's |
| 14:08.33 | [TK]D-Fender | coppice : You sure on that? Could have sworn I saw it written up somewhere. |
| 14:08.54 | coppice | how clear do you want it? polycoms are crap :-) |
| 14:08.57 | DeeJay[2] | Becky75: it only happens when talking to other people over ip communication... (inducing latency) |
| 14:09.06 | DeeJay[2] | when I'm starting the conversation there's no echo |
| 14:09.08 | [TK]D-Fender | DeeJay[2] : Descibe the scenarios where you get echo on them, and ones where you don't |
| 14:09.18 | DeeJay[2] | suddenly, the gain is changing to cope with low volume or high volume.. |
| 14:09.18 | coppice | people complain a lot about polycoms and snoms when the handset volume is turned up |
| 14:09.22 | DeeJay[2] | and the echo appears.. |
| 14:09.35 | DeeJay[2] | if I speak louder.... the echo dissappears because the gain adjust back to its initial level.. |
| 14:09.37 | DeeJay[2] | I would like to fix it.. |
| 14:10.09 | Becky75 | well DeeJay[2] no offence but i think u asking the wrong ppl here they cant even help me with a simple dead air problem |
| 14:10.12 | DeeJay[2] | [TK]D-Fender: I have an IP phone at home... when I call at office on their polycom it happens after some time... |
| 14:10.20 | DeeJay[2] | from a PSTN phone line, it doesn't appear.. |
| 14:10.52 | ghenry | what does "Insufficient information for SDP" mean? |
| 14:12.00 | Becky75 | ok so lemme ask again i am getting dead air not just on BRI but on analoque as well i have just been told |
| 14:12.40 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 14:13.20 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:14.02 | cytrak | hey guys I got a wierd problem.... I have a TE205P and TE110P card on my linux box |
| 14:14.34 | cytrak | the TE205P has span1 connected to pstn and span2 conneted to a siemens hicom pbx |
| 14:15.07 | cytrak | I have noticed that every so often asterisk keeps restarting the spans and calls are droped in the middle of conversations |
| 14:15.35 | cytrak | is that a bug or something wrong I have setup on my spans ? |
| 14:15.47 | Becky75 | cytrak i had that on BRI as well there is a way to prolong the checking |
| 14:16.28 | cytrak | what checking are you talking about ? some settings on zapata.conf ? |
| 14:17.01 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 14:17.55 | littleball | hello, who can help me to configure x-lite to asterisk over sip? i get the error "Call failed: 404 Not Found" |
| 14:18.06 | littleball | which is displayed on x-lite screen |
| 14:18.14 | Tier_1 | what dould cause sayunixtime date and saydigits not work |
| 14:18.26 | Becky75 | cytrak i wish i can help but i am yet ot be helped in this channel so i am just wiating for a answer on my question |
| 14:18.27 | Tier_1 | app_sayunixtime loads |
| 14:18.49 | Tier_1 | but it fails when I use the sayunixtime in a app |
| 14:19.00 | cytrak | Becky75: sorry |
| 14:19.03 | Damin | file: How was the food? |
| 14:19.06 | ghenry | found it |
| 14:19.16 | ghenry | softphone is requesting SIP/2.0 |
| 14:19.24 | Becky75 | all i can say is thank gawd i did not recommend asterisk to any one or done any rollouts on it yet |
| 14:19.32 | ghenry | SIP/2.0 488 Not acceptable here via sip debug |
| 14:19.35 | Becky75 | the support is almost on parr with microsoft hehehe |
| 14:20.14 | [TK]D-Fender | ghenry : thats a codec refusal error |
| 14:20.17 | ghenry | Becky75: Try e-mailing the list. Many more people there. |
| 14:20.28 | ghenry | SIP/2.0 488 Not acceptable here |
| 14:20.31 | ghenry | Aye |
| 14:21.04 | [TK]D-Fender | ghenry : Verify your codec list on the phone and in * |
| 14:21.23 | ghenry | doing so now |
| 14:21.25 | Becky75 | ghenry heh and almost as usefull as wearing lead boots on a frozen lake |
| 14:21.47 | ghenry | Becky75: Go buy support from Digium if you are not happy then. |
| 14:22.09 | cytrak | see this is waht I keep getting B-channel 0/16 successfully restarted on span 2 |
| 14:22.22 | cytrak | and goes on to through span 1 - 3 |
| 14:22.44 | ghenry | [TK]D-Fender: In sip.conf? |
| 14:23.05 | [TK]D-Fender | ghenry : yes |
| 14:23.39 | [TK]D-Fender | cytrak : intermittently? |
| 14:24.40 | *** join/#asterisk Abydos313 (n=abydos31@adsl-71-129-61-88.dsl.irvnca.pacbell.net) |
| 14:24.41 | ManxPower | cytrak, That is normal. Asterisk restarts idle channels to work around a bug in some PRI switches |
| 14:24.45 | Uzzi | Can I use Asterisk with hcf pci modem? |
| 14:24.47 | Becky75 | ghenry whahahah see thats why we going back to mitel or maybe buying avaya equipment... ppl just dont wanna help open source grow they just wanna charge you |
| 14:25.26 | mercestes | How old are you, Becky75? Don't they have child labor laws in South AMerica? |
| 14:25.30 | tzanger | [TK]D-Fender: ran into an interesting problem yesterday |
| 14:25.37 | ghenry | ppl can only offer the help that they have knowledge of Becky75 |
| 14:25.39 | cytrak | ManxPower: well is there a way to stop or increase that ? cause it restarts the span on every single channel and even during a conversation |
| 14:25.42 | tzanger | ip501 transfers broke somewhere in the last week or so of svn trunk |
| 14:25.50 | docelmo | Becky75 what's wrong with asterisk in a production environment? I have 6 asterisk servers for just one platform |
| 14:25.51 | tzanger | had to turn OFF reinvite/canreinvite |
| 14:25.56 | ghenry | Becky75: If knowone knows here, seek help on the list or call the guys who know |
| 14:26.11 | ghenry | Simple. |
| 14:26.21 | docelmo | What does she wanna know? |
| 14:26.23 | Becky75 | ghenry i bet you 50 pounds some one here knows |
| 14:26.35 | docelmo | WHAT? |
| 14:26.37 | docelmo | :) |
| 14:27.12 | ghenry | Becky75: Have you read: http://www.catb.org/~esr/faqs/smart-questions.html |
| 14:27.26 | Becky75 | they just to "proud" of there knowledge to share it |
| 14:27.42 | Becky75 | ghenry not arsed about smart questions |
| 14:28.06 | ghenry | well that == no help |
| 14:28.11 | docelmo | Becky75 Im gonna say it quit being a bitch and acting like this. YOU are the reason I charge out the ass on a commercial install |
| 14:28.39 | docelmo | We are here to help each other. You said your looking for an answer.. Well ask again. Im sure I could figure it out |
| 14:28.49 | ghenry | someone kick Becky75 Please!!! |
| 14:28.55 | docelmo | And no I am NOT gonna go back in my history |
| 14:29.02 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 14:29.10 | docelmo | ghenry it takes ALOT for anyone to get booted from here |
| 14:29.15 | tzanger | heh |
| 14:29.15 | Becky75 | docelmo last time i asked the same thing the 4th time i managed to get flamed and kicked |
| 14:29.19 | tzanger | what's Becky75 complaining about? |
| 14:29.20 | ghenry | docelmo: Just messing. |
| 14:29.35 | shiznatix | can anyone help me out? i can not fax a image but all text i try sending works fine |
| 14:29.40 | docelmo | becky WOULD YOU JUST ASK THE FUCKING QUESTION! |
| 14:29.44 | ghenry | [TK]D-Fender: I can't see anything about SIP versions in the sip.conf.sample |
| 14:29.51 | Becky75 | tzanger easy i just need help on getting rid of dead air in the middle of a call |
| 14:30.01 | docelmo | cause now I am getting annoyed which by other than Rehan doesnt happen very often |
| 14:30.04 | tzanger | Becky75: what protocol? |
| 14:30.09 | pythos | hah! I got dial-tone on my fxs modules! Kewlies. Ok, now I am wondering about the strange sounds Im hearing... one of the 2 fxs ports has much static or perhaps clicking/noisy condition. The other is clear dial-tone. Anyone seen such a thing? |
| 14:30.09 | Becky75 | its some times up to 4 seconds and only happens on ZAP calls |
| 14:30.17 | Becky75 | from sip to sip its fine |
| 14:30.24 | tzanger | Becky75: interesting |
| 14:30.34 | coppice | people rarely get booted, cos they just come back. the painful ones get ignored |
| 14:30.37 | docelmo | Becky75 what version Zap did you install |
| 14:30.39 | Becky75 | both in comming and out going ncalls |
| 14:30.47 | docelmo | coppice YES! |
| 14:30.53 | Becky75 | docelmo the latest version |
| 14:30.53 | tzanger | which zap interface? FXS or FXO? |
| 14:30.58 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 14:31.02 | Becky75 | i even thouight upgrading might help |
| 14:31.03 | *** join/#asterisk Lino` (n=Lino@87.123.243.255) |
| 14:31.06 | docelmo | Who is the manufacture of the card? |
| 14:31.36 | Becky75 | docelmo i am getting it on a tdm400 and a junghanns octo bri |
| 14:31.43 | C4T3l | hello world |
| 14:31.45 | Tier_1 | ok if a module loads but does not work what could be the issue |
| 14:31.45 | Becky75 | both systems same symptoms |
| 14:31.47 | tzanger | Becky75: both in the same system? |
| 14:31.55 | mercestes | C4T3l: your program is successful |
| 14:32.02 | C4T3l | sweet |
| 14:32.04 | Becky75 | tzanger 2 diff systems running the same software |
| 14:32.05 | docelmo | tzanger since there are 2 cards Im thinking possible memory or IRQ |
| 14:32.08 | docelmo | ohh |
| 14:32.15 | docelmo | that kills my theory |
| 14:32.22 | C4T3l | i think that mandi is going to time warner |
| 14:32.27 | C4T3l | oops |
| 14:32.34 | Becky75 | both genuine intel mother boards with p4 3.8's on them |
| 14:32.50 | tzanger | Becky75: that doesn't mean much |
| 14:32.52 | tzanger | but ok |
| 14:33.05 | *** join/#asterisk MoRpHeUz (n=artur@tux14.ltc.ic.unicamp.br) |
| 14:33.12 | Tier_1 | anyone ? |
| 14:33.19 | tzanger | Becky75: can you do a packet capture to see if there is RTP during this 4s silence? |
| 14:33.21 | brettnem | who's mandi? |
| 14:33.23 | *** part/#asterisk MoRpHeUz (n=artur@tux14.ltc.ic.unicamp.br) |
| 14:33.27 | mercestes | Tier_1: Configs. |
| 14:33.33 | mercestes | Hot chick that likes me. |
| 14:33.40 | Becky75 | tzanger i did a tcpdump on that also nothing funny there |
| 14:33.50 | Tier_1 | I have not changed the configs |
| 14:33.54 | coppice | Hot chick == has avian flu |
| 14:33.55 | Becky75 | but i see my cpu goes up to 80% duting the silance if i watch top |
| 14:34.08 | tzanger | coppice: hahaha |
| 14:34.10 | C4T3l | hehe |
| 14:34.11 | mercestes | coppice: *nod nod* very hot chick. |
| 14:34.12 | tzanger | Becky75: so you have RTP every 20ms as expected? you checked the timestamps? |
| 14:35.01 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:35.20 | Becky75 | tzanger i even started ot debug the sip calls and iax trunks |
| 14:35.47 | docelmo | coppice hahah |
| 14:36.20 | docelmo | have you tried to trade out the hardware? are these the echo can boards? |
| 14:36.31 | docelmo | nevermind TDM400's dont have echo can |
| 14:37.01 | tzanger | Becky75: answer my question please |
| 14:37.03 | Becky75 | docelmo i swoped servers still the same |
| 14:37.50 | [TK]D-Fender | docelmo : the TDM400's does echo like its in a tin can! ;) |
| 14:38.17 | Splat | anyone happen to be able to point me to a good book or tutorial for configuring asterisk from sratch? |
| 14:38.36 | [TK]D-Fender | ~thebook |
| 14:38.38 | jbot | i heard thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 14:38.40 | [TK]D-Fender | Splat : there |
| 14:39.01 | Becky75 | yeah got the book and printed it out |
| 14:39.20 | Becky75 | there must be a way to get rid of that little 4 second dead air |
| 14:39.32 | Becky75 | what gets me is it does not happen on every call |
| 14:40.27 | mercestes | Becky75: I believe Tazanger asked for a packet capture during the silence. |
| 14:40.59 | Becky75 | mercestes i am trying to get one but every one has left for the weekend |
| 14:41.33 | Becky75 | problem with that book is it shows you a non real life dial plan not a dial plan you can get going and grow on in a real working environment |
| 14:41.51 | C4T3l | i agree |
| 14:43.57 | *** join/#asterisk fjean (n=fjean@201.29.140.206) |
| 14:44.10 | Becky75 | well docelmo and [TK]D-Fender thank u for your hep thus far |
| 14:44.17 | fjean | hello guys |
| 14:44.46 | ManxPower | cytrak, See zapata.conf.sample in the Asterisk source tree. |
| 14:44.59 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-188.z143-154-67.customer.algx.net) |
| 14:45.09 | ghenry | [TK]D-Fender: I can't see anything abou SIP 2.0 in sip.conf |
| 14:45.17 | fjean | tell me, where do we get the channel id to use whenever we want to use soft hangup, I do sip show channels but only see a call ID |
| 14:45.57 | ManxPower | fjean, you use the call id as shows in sip show channels for soft hangup |
| 14:47.43 | pythos | Whew! I pulled the modules off, and re-inserted them, and the static problem went away! |
| 14:48.24 | fjean | manxpower: thanks, i ll try that |
| 14:48.34 | ManxPower | pythos, "cat /proc/interrupts" Make SURE there are no devices on the same IRQ as the Digium cards |
| 14:49.10 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 14:49.16 | sleepy_one | pythos, like ManxPower said that's very important, btw procinfo is more readable ( yum install procinfo ) |
| 14:50.23 | cytrak | ManxPower: should I try the resetinterval to never ? |
| 14:50.45 | cytrak | ManxPower: you mentioned something about a bug on asterisk , will that cause any asterisk crashes ? |
| 14:50.48 | ManxPower | cytrak, never or 0 I don't recall which |
| 14:50.57 | cytrak | ok |
| 14:51.06 | ManxPower | cytrak, no, the bug is in OTHER PRI equipment. |
| 14:51.12 | cytrak | ok thanks |
| 14:51.27 | ManxPower | I think some Adtran PRI stuff will remove channels from service if it doesn't see them used for a while. |
| 14:51.50 | ManxPower | And some (Toshiba?) PRI devices will kill all calls if any channel is restarted. |
| 14:51.54 | nettie | Hey guys, how's going? I'm still having issues with MOH, when my call goes trough the voip carrier it's very jumpy, on local calls it sounds great. Other than asking the voip carrier if he's doing silence suppression (he claims he's not) and be 100% sure that my phones have silence supression disabled. What else should I check? |
| 14:51.58 | Tier_1 | it seems sayunixtiime saydigits and date in svn head is broken |
| 14:52.02 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 14:52.02 | pythos | k, reviewed, no overlaped IRQs |
| 14:52.17 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 14:52.18 | Tier_1 | I backed steo to 8am yesterday and it worked fine |
| 14:52.34 | Tier_1 | but current svn does not |
| 14:52.45 | ManxPower | nettie, Any codec except for ulaw and alaw can cause poor quality sound with music |
| 14:52.59 | sleepy_one | nettie, what codecs are you using? MOH doesn't sound as good with GSM, etc |
| 14:53.07 | jake1932 | Becky75: fyi - they just got back to me - wonder if someone is monitoring this channel |
| 14:54.36 | nettie | ManxPower I'm using ulaw, I check with g729 and I can hear a small degradation of music quality but the problems I'm having oesnt seems to be codec related. In fact if I make noise the MOH sounds smooth, if I stop talking or making noise MOH stops as well and make short jumps |
| 14:55.41 | ManxPower | nettie, That is a CLASSIC symptom of VAD/silence supression being enabled. |
| 14:55.54 | *** part/#asterisk fjean (n=fjean@201.29.140.206) |
| 14:55.55 | *** join/#asterisk carlos-the-man (n=carlos@201.155.235.25) |
| 14:56.08 | ManxPower | Diagram your setup. example Asterisk -> SIP -> VoIP Company -> PSTN |
| 14:56.22 | nettie | ManxPower that's exactly what I said them.. the funny thing is that with other voip carriers it's fine. |
| 14:56.25 | nettie | it's |
| 14:56.27 | nettie | like that |
| 14:56.45 | ManxPower | nettie, think about it. the call direction is not the same as your setup. |
| 14:57.06 | ManxPower | Perhaps you mean Caller -> PSTN -> SIP ITSP -> Asterisk |
| 14:57.43 | nettie | polycoms->cisco877w->myasterisk@colo->voipcarrierasteriskusingSIP->some3rdpartyrealcarrier->PSTN |
| 14:57.47 | ManxPower | nettie, doing a SIP debug will give you the call setup packet dumps and you'll see if the carrier is sendin that it can do VAD or not. |
| 14:58.22 | ManxPower | nettie, and who is placing the call on hold? The polycom or someone on the PSTN? |
| 14:58.23 | nettie | interesting |
| 14:58.28 | Tier_1 | 1.2-svn is forked |
| 14:58.29 | nettie | polycom |
| 14:58.46 | nettie | the MOH starts on myasterisk@colo |
| 14:59.11 | ManxPower | nettie, The person on the PSTN heard bad MoH? |
| 14:59.19 | nettie | ManxPower exaclty |
| 14:59.37 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 14:59.37 | ManxPower | Yup, your carrier is doing VAD |
| 14:59.40 | stack_ | morning everyone |
| 14:59.42 | nettie | ManxPower the person on the PSTN might be just me using the mobile phone |
| 14:59.48 | ManxPower | do sip debug |
| 15:00.02 | nettie | ManxPower the question is .. who's doing it.. MY voip carrier OR his VOIP carrier |
| 15:00.08 | nettie | ok |
| 15:00.11 | nettie | I'll sip debug now |
| 15:00.13 | Dr-Linux | Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 15:00.13 | Dr-Linux | Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 15:00.20 | Dr-Linux | what this line mean? |
| 15:00.36 | ManxPower | nettie, does your carrier support IAX? |
| 15:00.54 | *** join/#asterisk froguz (n=alvaro@pc-95-155-104-200.cm.vtr.net) |
| 15:01.35 | file | Dr-Linux: your peer wanted g723, you did not have that in your Asterisk capabilities list... so it has no usable audio codec |
| 15:01.36 | *** join/#asterisk trumpetinc (n=irc_kevi@1Cust45.VR1.PHX1.broadband.uu.net) |
| 15:01.48 | nettie | I wish, it used to support it but I'm not sure it still supports it.. I could ask them. Do you think that using iax2 would be better? other then be more NAT agnostic? |
| 15:01.58 | nettie | ok I'm logging |
| 15:02.39 | ManxPower | nettie, No implimentation of IAX2 supports VAD |
| 15:02.44 | cytrak | you know one other wierd thing is my D-channel , even though I set them to be the 24th channel asterisk complains that it's not |
| 15:03.23 | ManxPower | So if you talk IAX2 to your carrier you KNOW that leg of the call is not using VAD |
| 15:03.33 | nettie | ManxPower ok, done what keyword should I look for in the log? VAD ? |
| 15:03.38 | nettie | ah |
| 15:03.40 | nettie | nice |
| 15:03.55 | ManxPower | nettie, silenceSupression or something like that in the call setup |
| 15:04.02 | nettie | checking.. |
| 15:04.08 | ManxPower | I think it would be in the same packet as the codec info |
| 15:04.15 | trumpetinc | hi folks - wondering if anyone is interested in trying out a new echo coefficient tuning algorithm I've added to fxotune? |
| 15:04.36 | *** part/#asterisk froguz (n=alvaro@pc-95-155-104-200.cm.vtr.net) |
| 15:05.02 | nettie | all SilenceSupp are off |
| 15:05.22 | stack_ | About once or twice a day, I get a hang up on our PRI. I just get a "Hungup 'Zap/5-1'" on the console. Any ideas? |
| 15:05.46 | ManxPower | stack_, callprogress=no busydetect=no |
| 15:05.48 | carlos-the-man | hi all, I installed mandriva and asterisk with in it, and want to add something to control bandwith to make sure the compulsive downloaders will never interfere with voip, what is the best option? thanks |
| 15:05.55 | ManxPower | trumpetinc, is it easy to test? |
| 15:06.02 | stack_ | ManxPower, those were never setup before... |
| 15:06.09 | twisted[asteria] | z0mg Becky75 |
| 15:06.10 | stack_ | ManxPower, are they on by default? |
| 15:06.22 | ManxPower | stack_, they are off by default |
| 15:06.23 | nettie | I can see I issue the call and after a couple of lines I get a=silenceSupp:off - - - - |
| 15:06.33 | stack_ | ManxPower, then they are already off |
| 15:06.52 | ManxPower | nettie, I don't have any more ideas |
| 15:06.56 | nettie | ManxPower the problem imho could be from their asterisk to their carrier? |
| 15:07.13 | ManxPower | nettie, could be. Can't you just drop the problem carrier? |
| 15:07.37 | nettie | ManxPower unfortuantely not the number is not portable and it's already everywhere eheh |
| 15:07.53 | nettie | ManxPower they're supposed to be one of the best ones .. |
| 15:08.19 | ManxPower | nettie, spank the person that did that without fully testing everything before telling everyone about the number |
| 15:09.31 | CoffeeIV_ | when I have a list of priorities, why can't I number them 5 - 10 - 15 - 20 styled like basic code so I can add stuff without renumbering ? It fails if they are not sequential starting with 1 -- I have a recent version of asterisk from svn |
| 15:10.03 | sleepy_one | CoffeeIV_, I think you can use s,n,whatever now |
| 15:10.10 | [TK]D-Fender | CoffeeIV_ : Because thats how it works. PERIOD. |
| 15:10.20 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) |
| 15:10.21 | CoffeeIV_ | yes, I thought so too, but that didn't work either |
| 15:10.35 | [TK]D-Fender | CoffeeIV_ : and because priorits JUMP on occasiona dn the numbering matters |
| 15:10.41 | nettie | ManxPower I should spank myself then.. we were so in hurry that we didnt test it deeply. Anyway thanx again for your help. I'll call the carrier and try to figure out with them what coul dbe wrong.. maybe I'll ask them to do a sip debug to see if they have silent suppression off versus their carrier |
| 15:11.12 | Dr-Linux | file: i just loaed g723 codec, but same error |
| 15:11.12 | Dr-Linux | Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 15:11.13 | Dr-Linux | Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 15:11.13 | Dr-Linux | May 12 07:58:15 NOTICE[7034]: chan_sip.c:3588 process_sdp: No compatible codecs! |
| 15:11.29 | MrChimpy | is asterisk dialplan based on something else? some kind of bastardisation of dialplans on older telephone switches? cos to me it's frikkin weird |
| 15:12.29 | file | Dr-Linux: your sip.conf is not configured correctly |
| 15:12.49 | CoffeeIV_ | I'm cool with bastardisations of older systems -- heck my my main occupation is C programming in Unix |
| 15:12.54 | sleepy_one | CoffeeIV_, but as [TK]D-Fender said the priority is there for a reason |
| 15:13.41 | MrChimpy | coffee: if dialplans were c like they'd make way more sense to me... |
| 15:13.52 | Dr-Linux | file: what's wrong with my sip.conf, what thing i should look in? |
| 15:14.07 | ManxPower | MrChimpy, AEL is the attempt to address many of the limitations of the current dialplan config |
| 15:14.53 | KranZ | Dr-Linux: your sip.conf is probably fine |
| 15:15.06 | KranZ | its the device that's trying to use * that needs reconfiguring |
| 15:15.18 | CoffeeIV_ | I took a dialplan from an older asterisk and put it on a recent svn checkout, and the s,n, thing didn't work when I tested it in a context. Is there a flag that has to be set somewhere in the dialplan to activate it ? |
| 15:15.26 | KranZ | ..its only using g723, which * isnt using |
| 15:15.36 | KranZ | reconfigure it for ulaw/alaw |
| 15:16.23 | Dr-Linux | file: that problem is solved, but another error now appearing: |
| 15:16.23 | Dr-Linux | May 12 08:03:04 WARNING[7034]: chan_sip.c:3414 process_sdp: Insufficient information for SDP (m = '', c = '') |
| 15:16.23 | Dr-Linux | Destroying call '10381917502604@202.125.141.2' |
| 15:16.24 | Dr-Linux | sorry "warning" |
| 15:17.20 | [TK]D-Fender | MrChimpy : its a waste of time, the format is likely to change again while still adding little of relevence and no-one here is likely to bother touching it. |
| 15:19.12 | cytrak | ManxPower: dman that didn't work ... I was just on a call with digium and the span got restarted and the call was dropped |
| 15:19.40 | ManxPower | cytrak, put your zapata.conf on pastebin.ca |
| 15:20.19 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 15:20.26 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 15:20.30 | ManxPower | [TK]D-Fender, AEL in 1.4 is supposed to be non-experimental |
| 15:20.54 | littleball | hello, i heard that msn messenger support sip and can be connected to asterisk. But i cannot find such configuration option in msn messenger. who can help? |
| 15:21.01 | MrChimpy | well not moving away from dialplan will always leave the "but asterisk's dialplan looks like it was designed by monkeys sky high on PCP" argument. |
| 15:21.27 | MrChimpy | better to move towards something sane and keep legacy support |
| 15:21.52 | *** join/#asterisk stillbourne (n=jdgeier@72.16.203.209) |
| 15:21.55 | mercestes | What's wrong with the Asterisk dial plan? I kinda like it. |
| 15:22.04 | mercestes | it's better than say.............coppercom. |
| 15:22.10 | [TK]D-Fender | ManxPower : What do you think the impact of it will be through the community? |
| 15:22.29 | ManxPower | [TK]D-Fender, It's a hell of a lot less error prone than the current system |
| 15:23.00 | CoffeeIV_ | To answer my own question about the s,n, thing -- apparently you can't start with s,n, it has to start with s,1, and then you can use n thereafter |
| 15:23.10 | [TK]D-Fender | ManxPower : Could be, but its free-form nature makes coding style more of a problem. |
| 15:23.32 | [TK]D-Fender | CoffeeIV_ : "n" = NEXT. Can't have a NeXT without a START :) |
| 15:23.40 | nettie | ManxPower I also have a problem with another carrier :) damn I only have problems eheh.. when I call a number I hear only the user pick the phone up and talking but not the ringing before he picks up. If the phone is busy I just dont hear anything. I remember I checked into call progressing but I didnt find any solution yet. It's very strange. As far as I know call progressing is passed to the phone in the form of RTP packets, exaclty like |
| 15:23.48 | ManxPower | [TK]D-Fender, theres also res_perl and res_js |
| 15:23.51 | mercestes | CoffeeIV_: That's a common error. I do it often in hasty copy paste operations. |
| 15:24.10 | cytrak | ManxPower: I will in a second |
| 15:24.12 | nettie | Do you think could be a good idea mirror the phone port on the switch to my laptop, fireup ethereal to figure out what could be wrong? |
| 15:24.12 | ManxPower | nettie, make sure you have /etc/asterisk/indications.conf |
| 15:24.21 | nettie | ManxPower I do |
| 15:24.44 | nettie | it's there, it default to US as country |
| 15:25.11 | *** join/#asterisk privalac1 (i=user90@Kitchener-HSE-ppp3571565.sympatico.ca) |
| 15:27.25 | [TK]D-Fender | ManxPower : I should get around to those as well except I only do PHP in Linux so far. |
| 15:27.35 | privalac1 | Got a big problem. A site is down. Inter sip phone calls works, but if I dial let's say *98 for voicemail, I do not hear the playback of the sound files |
| 15:27.35 | *** join/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 15:27.49 | [TK]D-Fender | ManxPower : And I think even mentioning res_php is sacriledge around here ;) |
| 15:28.19 | ManxPower | privalac1, how long has this system been in production? |
| 15:28.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:28.48 | KranZ | res_php! |
| 15:29.24 | KranZ | privalac1: what does the CLI indicate when you dial *98 |
| 15:29.28 | KranZ | file not found? |
| 15:29.41 | privalac1 | 3-4 months |
| 15:29.51 | ManxPower | privalac1, what did you change? |
| 15:30.14 | *** join/#asterisk dextro (n=dextro@suffrage.itfreedom.com) |
| 15:30.14 | privalac1 | Nothing. It started doing this on it's own arounf 9h30 tis morning |
| 15:30.43 | ManxPower | privalac1, did you try rebooting? |
| 15:30.55 | littleball | hello, who can help to configure windown messenger to connect to asterisk? |
| 15:31.21 | ManxPower | privalac1, We have to reboot any Asterisk that has an analog card in it at least once per week |
| 15:32.27 | MrChimpy | manx: i don't remember that one being on the brochure :) |
| 15:32.52 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.76.47) |
| 15:32.54 | privalac1 | The card is a Sangoma's quad pri with echo |
| 15:33.29 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:36.01 | privalac1 | Rebooted a few times, Sangoma's guy checked did not find anything. Telco checking PRI's |
| 15:36.59 | shiznatix | can anyone help me get ghost script to give me a standard consistant page size. if i do: pdf -> tiff then fax that then take that same fax and fax it back to asterisk and do tiff -> pdf then fax then do pdf -> tiff and fax it back to the fax machine it does not work and i am fairly certain it is because of the changed page size |
| 15:39.16 | CoffeeIV_ | when you use Gosub, how do you make it go back to the return address it saved ? |
| 15:41.00 | Qwell | CoffeeIV_: return? |
| 15:41.12 | *** part/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 15:41.41 | privalac1 | Kranz: The usual as if the system would be working. If I do an Echo test I can hear myself fine... |
| 15:42.03 | *** join/#asterisk pbx1 (n=pbx1@58.69.229.213) |
| 15:42.09 | CoffeeIV_ | it mentions the Return() in the show application gosubif documentation but not for gosub |
| 15:43.42 | gandhijee | anyone know if there is away to make the polycoms default to numeric input on a text field? |
| 15:44.50 | *** join/#asterisk camelon (n=chiardon@200.71.58.39) |
| 15:45.05 | camelon | Morning |
| 15:46.45 | cytrak | ManxPower: here http://pastebin.com/713655 |
| 15:49.42 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:50.22 | cytrak | dman now my include => iax-users won't work |
| 15:50.47 | SplasPood | How would one go about reloading manager.conf short of reloading everything? |
| 15:51.26 | Hmmhesays | manager reload? |
| 15:51.51 | Hmmhesays | guess not |
| 15:54.16 | *** part/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 15:54.22 | SplasPood | Hmmhesays: Yea, that was my first try :) |
| 15:54.30 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:54.33 | KranZ | reload [whatever the manager module filename is] |
| 15:55.08 | SplasPood | yea, dunno that part either |
| 15:55.13 | SplasPood | its not *manager* |
| 15:55.19 | SplasPood | only thing I hit there is cdr_manager.so |
| 15:55.44 | dextro | I have an interesting voice mail bug I wanted to know if anyone has experienced before I submit a bug report: When leaving a voice mail for a user with the 'operator=yes' flag set, if I dial '0' while recording a message I am promted with "dial '1' to accept the recording or wait to while i try that extension". If I dial '1' the message is saved and I am forwarded along no problem; however if I wait for the timeout, the message is deleted as e |
| 15:55.44 | dextro | xpected but the msgXXXX.txt file is not removed and the voice mail system thinks there is a new message. Anyone experienced this issue before? |
| 15:56.25 | Damin | dextro: What version are you using (of asterisk?) |
| 15:57.00 | *** join/#asterisk heka (n=heka@82.114.68.123) |
| 15:57.07 | blitzrage | dextro: yes -- its a bug |
| 15:57.29 | heka | anybody can help me building oh323? |
| 15:57.35 | Damin | dextro: I believe I saw a patch for that applied to 1.2 SVN post 1.2.7.1 release.. so you might want to grab the latest SVN from 1.2 |
| 15:57.41 | *** join/#asterisk [vmwarez]dotcom (n=jjones@216.147.224.254) |
| 15:57.48 | Damin | blitzrage: Wasn't that patched though? |
| 15:58.01 | blitzrage | Damin: I think its still in the bug tracker |
| 15:58.04 | dextro | I am using 1.2.7.1 but am compiling TRUNK now to test |
| 15:58.07 | blitzrage | http://bugs.digium.com/view.php?id=7125 |
| 15:58.10 | SplasPood | lame, I think the only way to reload manager.conf is with a full reload |
| 15:58.13 | dextro | thanks blitzrage |
| 15:58.20 | blitzrage | dextro: don't use trunk -- use 1.2 SVN |
| 15:58.25 | SplasPood | can't do that now cause i'll screw up my polycom hint/subscription status |
| 15:58.25 | heka | Im getting an error while trying to build asterisk-oh323-0.7.3, here is the log http://pastebin.com/713675 |
| 15:58.28 | sevard | SplasPood: the only way to reload a lot of stuff is with a full reload |
| 15:58.35 | blitzrage | svn co http://svn.digium.com/svn/asterisk/branches/1.2/ |
| 15:58.47 | SplasPood | sevard: I've found that for 99.9% of what I need to do thats not true tho |
| 15:58.57 | Damin | Make sure that when someone 0's out while recording a msg and then chooses to DELETE the recorded file, the .txt file isn't left around by itself to cause problems later. #7061 (dimitripietro reporting, blitzrage confirmed) |
| 15:58.58 | sevard | SplasPood: what about queues and agents |
| 15:59.06 | SplasPood | sevard: reload app_queue.so |
| 15:59.09 | blitzrage | SplasPood: unload / load the appropriate module |
| 15:59.11 | SplasPood | sevard: reload chan_agent.so |
| 15:59.12 | sevard | sweet! |
| 15:59.21 | Damin | blitzrage: It was applied to 1.2 revsion 23985 |
| 15:59.22 | blitzrage | SplasPood: do show modules to see if the manager has it's own module, then just reload it |
| 15:59.22 | sevard | i didn't think that reloaded the conf files |
| 15:59.24 | SplasPood | blitzrage: which is what? I'm looking and I don't see it |
| 15:59.28 | blitzrage | Damin: oh yah? cool |
| 15:59.37 | blitzrage | SplasPood: its at the CLI -- run show modules |
| 15:59.47 | dextro | so the 1.2 "branch" is the latest revision akin to CVS-HEAD? |
| 15:59.53 | blitzrage | probably be app_manager or res_manager or something.... |
| 15:59.59 | blitzrage | dextro: absolutely not |
| 16:00.22 | Damin | dextro: It's a feature frozen branch that is suitable for most production uses.. |
| 16:00.27 | SplasPood | blitzrage: nope, its not... I already did an ls on /usr/lib/asterisk/modules .. only manager-esq thing is cdr_manager.so |
| 16:00.49 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 16:00.52 | Damin | dextro: TRUNK is the new "cvs head" and it is most definitely undergoing some significant changes.. |
| 16:00.59 | blitzrage | dextro: 1.2 has it's own branch which changes go into -- then at certain points, a release is made called a 'tag'. The CVS-HEAD (old name) is now called trunk, and is located at http://svn.digium.com/svn/asterisk/trunk/ which is the development branch |
| 16:01.11 | Damin | dextro: If you are running TRUNK in production, you are a lunatic.. ;) |
| 16:01.15 | blitzrage | yah -- difference between 1.2 and trunk are pretty staggering |
| 16:01.28 | privalac1 | Carrier checking the loop. But I see a lot of Avoiding deadlock for 'Zap/14-1' any hints for anyone? |
| 16:01.29 | dextro | i am not running trunk, i just wanted to check and see if the bug was in there still |
| 16:01.45 | blitzrage | right -- but your best bet is to try the 1.2 branch, not jump all the way to trunk |
| 16:01.51 | dextro | okay, i think i get it now; i was a little mixed up with tags and branches and trunks |
| 16:01.57 | Damin | dextro: I run 1.2 SVN (not the actual snapshot releases) on my production boxes.. |
| 16:02.12 | Blackthorn | Hello, I have an issue dialing sip phone out the local pri with a dual ring. Dialing sip to sip, pri to disa back out, and all other combo's works fine. I've tried the dial with -r and it only does single ring but then I don't get busy tones etc etc. |
| 16:02.13 | SplasPood | blitzrage: yea just took a glance at the source.. seems the only time reload_manager is used is on a full reload |
| 16:02.15 | dextro | i typically run the latest tagged version |
| 16:02.17 | blitzrage | but you have to monitor the changes going in |
| 16:02.18 | gandhijee | whats better to use to connect to MySQL, the unixODBC driver or the one in the addons package for mySQL |
| 16:02.28 | blitzrage | SplasPood: then I guess that's what you gotta do :) |
| 16:02.42 | syle | what is most calls asterisk box can accept currently? no registrations, straight ip address connects via sip to diff asterisk boxes |
| 16:02.42 | blitzrage | no idea -- I use PostgreSQL because it's better :) |
| 16:02.43 | SplasPood | blitzrage: if it wasn't for these damn polycoms I would... Gotta wait till after hours now |
| 16:02.51 | dextro | so technically the 1.2 "branch" could/will be more up to date than my 1.2.7.1 "tag" |
| 16:02.56 | blitzrage | syle: not enough information to telly ou |
| 16:03.03 | syle | what do you need? |
| 16:03.09 | *** join/#asterisk stratacom (i=strataco@area51.ntg.more.net) |
| 16:03.11 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 16:03.16 | blitzrage | dextro: it is more up to date -- but there can still be errors / bugs, etc... |
| 16:03.23 | gandhijee | i was gonna use that orginially, then i saw a buncha stuff that said mySQL was better supported =/ |
| 16:03.24 | blitzrage | it's still technically a development branch |
| 16:03.42 | blitzrage | I guess... unless you're using ODBC and func_odbc |
| 16:04.02 | blitzrage | syle: are you doing transcoding? what kind of processor on the box? |
| 16:04.03 | Damin | blitzrage: I've gone a bit nuts w/ commeting out ast_log messages at the warning level for crap that I can't do anything about and really don't need to know about.. :) Things like Comfort Noise Notifications, VAD warning and that damn new "Forcing marker bit" crap that was just added.. |
| 16:04.18 | syle | i'm just wondering if their is some kind of hard limit in asterisk or not, otherwise it would just depend on testing based on memory, cpu, I/O and bandwidth right |
| 16:04.23 | blitzrage | Damin: I hear yah man -- some of that stuff is just annoying |
| 16:04.32 | blitzrage | syle: yep |
| 16:04.43 | blitzrage | no hard limit that I'm aware of |
| 16:04.47 | syle | ok thx |
| 16:05.35 | syle | i am not sure about transcoding yet, but i've been asked to emulate calls between different servers to find that out |
| 16:05.50 | blitzrage | transcoding is going to really eat CPU -- so you better figure it out :) |
| 16:05.57 | syle | alright |
| 16:06.31 | syle | from your experience what percentage less calls can be made because trancoding was being performed? |
| 16:07.11 | blitzrage | if you end up with dual-Xeon 3.8GHz, you can probably get up to about 200 ulaw->g729 transcoded calls |
| 16:07.51 | syle | expensive server :) |
| 16:07.59 | blitzrage | transcoding is an expensive process |
| 16:08.04 | cytrak | can zapata.conf interfere with extension.con dialplans ? |
| 16:08.07 | cytrak | includes ? |
| 16:08.17 | blitzrage | ummmm... no, unless you've really screwed something up :) |
| 16:08.23 | stratacom | Does anyone know about support of sending CallerID information to the CO from the * platform? |
| 16:08.34 | syle | how about those 200ulaw->g729 calls without transcoding? |
| 16:08.36 | blitzrage | well... I guess it could if you're directly your Zap calls into a context that it shouldn't be |
| 16:08.53 | blitzrage | syle: the fact you're changing codecs MEANS you're transcoding |
| 16:08.59 | cytrak | I have an include => iaxlocal under my [zappriin] context that is no longer working |
| 16:09.07 | *** join/#asterisk ToTo (n=ToTo@host21-83.pool8260.interbusiness.it) |
| 16:09.10 | cytrak | it was working earlier |
| 16:09.16 | syle | no i;m just getting the before figures so i can figure out the percentage question i asked you :) |
| 16:09.22 | privalac1 | What does "Avoiding initial deadlock for 'Zap/19-1'" means on a PRI? |
| 16:09.23 | syle | probably won;t be |
| 16:09.27 | blitzrage | syle: I don't know what you mean |
| 16:09.40 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
| 16:09.40 | blitzrage | privalac1: just a warning message -- can safely ignore it |
| 16:10.11 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 16:10.21 | *** join/#asterisk watchy (n=watchy@70.238.57.237) |
| 16:10.36 | watchy | anyone got documentation only provisioning sipuras? |
| 16:10.39 | syle | if it takes a dual-XEon 3.8ghz to do transcoding on those 200 calls, what least expensive server could accomplish the same thing without the transcoding, 1 2.6 ghz cpu for example |
| 16:10.40 | watchy | on |
| 16:10.55 | blitzrage | syle: you mean straight ulaw->ulaw then |
| 16:10.59 | syle | yes |
| 16:11.05 | privalac1 | Yeah but I have big problem on this server. Can not dial out or dial-in. I just tried an IAX trunk and I can dial out using this trunk. I can hear the other end, but they don't hear me... |
| 16:11.15 | blitzrage | syle: good question -- let me know once you've figured it out :) |
| 16:11.16 | dlynes_ | watchy: Are you a service provider? |
| 16:11.18 | privalac1 | This server has been working fine for 3 months... |
| 16:11.45 | watchy | dlynes: nope |
| 16:11.55 | stratacom | all: I need to send specific CID on outbound calls so called party can call back directly to user |
| 16:12.04 | stratacom | all: can this be done? |
| 16:12.24 | dlynes_ | watchy: Yeah, if you were, then you'd be able to get the administrator's manual for the sipura |
| 16:12.26 | syle | blitzrage: one last question from your experience, what are generally the bottlenecks in asterisk on a large volume of calls? cpu, memory, then I/O? assuming a single IDE 2.6 ghz cpu with 500 megs of ram |
| 16:12.27 | watchy | dlynes: i just wanna have them config from a ftp/tftp server. is that possible? |
| 16:12.37 | dlynes_ | watchy: However, you can still download the user's manual from their website |
| 16:13.00 | watchy | why are they bitches about the admin manual? |
| 16:13.24 | watchy | i'm gonna have to buy different atas i guess |
| 16:13.27 | dlynes_ | watchy: most companies are when it comes to voip gear |
| 16:13.34 | syle | i am just trying to get an idea of what i should be watching for before i generate all those calls |
| 16:13.45 | dlynes_ | watchy: Sipura's far from the only company that does that |
| 16:13.48 | syle | for the emulator, so i don;t crash any machines |
| 16:13.49 | watchy | the manual has to be posted somewhere |
| 16:14.05 | dlynes_ | watchy: it is, but you need to get a login account from sipura |
| 16:14.26 | dlynes_ | watchy: try emailing support@sipura.com to see if you get anywhere |
| 16:14.29 | watchy | i would think it would be on some other website that someone posted besides sipura |
| 16:14.33 | watchy | i mean it is the internet |
| 16:14.51 | syle | i am going to try to emulate 250 simultaneuos calls per box are the specs |
| 16:14.54 | dlynes_ | watchy: Just bs them about how you're planning to buy a hundred ata's or something and you want to make sure that their ata's are going to be easy to deploy |
| 16:15.04 | cytrak | does exten => _1XXX,1,Dial,Zap/g2/${EXTEN} take precedence over an " include => " that has exten => 1541,1,Macro(zaptoiax,${EXTEN}) ? |
| 16:15.05 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 16:15.25 | watchy | it is possible to make them grab configs from tftp/ftp isnt it? |
| 16:16.00 | cytrak | for some reason that include works on a local context but it won't work if the call comes from the PSTN |
| 16:16.06 | _Sam-- | hey if im using chanspy to listen to calls, how can i record those calls? |
| 16:16.13 | KranZ | watchy: yes |
| 16:16.13 | dlynes_ | watchy: tftp, http, and https |
| 16:16.19 | dlynes_ | watchy: but not ftp |
| 16:16.25 | watchy | tftp is fine |
| 16:16.32 | watchy | now i just need infos on how to do it |
| 16:16.51 | dlynes_ | watchy: like i said...email them and hyperbolize the truth a bit |
| 16:17.10 | dlynes_ | watchy: you should be able to get a login id and password if you're good at the art of hyperbolizing :) |
| 16:17.11 | KranZ | watchy: since they are a linksys/cisco company |
| 16:17.27 | KranZ | you'll probably have to be a registered service provider with linksys before they'll give you access |
| 16:17.42 | gaupe | watchy: you'll find all the info you need for provisioning sipura ATA on voip-info.org and google, it's not hard at all |
| 16:17.46 | dlynes_ | KranZ: No, not for the sipura units...that's all done through sipura |
| 16:18.11 | KranZ | coo |
| 16:18.12 | Maxxed | hey any of you guys know how to fix this phonenumber@ipaddress thing with the new 8.2 sip firmware on the 7960's ? |
| 16:18.14 | dlynes_ | gaupe: yeah, but he won't be able to get the configuration compiler, or anything like that |
| 16:18.25 | Maxxed | when sombody calls is num@ipaddress of the pbx |
| 16:18.29 | gaupe | that's right, you need to apply for that |
| 16:18.35 | Maxxed | i wana strip off the @ipaddress |
| 16:18.44 | Maxxed | i would think its somthing in the config |
| 16:18.47 | Maxxed | but what |
| 16:18.52 | KranZ | Maxxed: that's part of the sip standard |
| 16:19.06 | Maxxed | i didnt have a prob with it before |
| 16:19.21 | KranZ | you probably misconfigured the domain on the device or pbx |
| 16:19.24 | dlynes_ | watchy: anyways...it's kinda stupid the way they do it...you need to get a login id and password to access like 3 or 4 files that you can't access on the ordinary support page |
| 16:19.25 | Maxxed | i noticed when i upgraded the phone started doing it |
| 16:19.32 | Maxxed | i didnt change any configuration at all |
| 16:19.40 | Maxxed | just the firmware on a few phones |
| 16:19.46 | KranZ | but the phone's conf might have changed |
| 16:19.49 | Maxxed | the 7. what ever phones are still working fine |
| 16:19.51 | watchy | dlynes: damn them |
| 16:19.51 | Maxxed | on the same pbx |
| 16:19.54 | KranZ | check for a domain setting |
| 16:20.02 | dlynes_ | watchy: grandstream's even worse...you can't get squat except directly from them...you can't even get a login page to do it |
| 16:20.03 | watchy | i'll just configure it from the freakin website |
| 16:20.18 | Maxxed | domain setting? |
| 16:20.28 | KranZ | watchy: it doesnt hurt to send an email |
| 16:20.35 | watchy | i;m gonna do that also |
| 16:20.41 | watchy | but right now i need these bitches confed |
| 16:20.50 | dlynes_ | watchy: Aastra's one of the few voip gear companies i know of where you don't need to register with the manufacturer as a service provider |
| 16:21.01 | KranZ | Maxxed: the pbx could see the @ip as not a local domain and reject the call |
| 16:21.07 | dlynes_ | watchy: and you still get provisioning tools |
| 16:21.21 | KranZ | is polycom that way? |
| 16:21.46 | *** join/#asterisk S4w (n=saw@adsl-3-65-52.mia.bellsouth.net) |
| 16:21.50 | dlynes_ | no idea |
| 16:22.02 | dlynes_ | I haven't had the pleasure of dealing with polycom yet |
| 16:22.04 | S4w | hey guys any of you know the bandwidth requirements for the GSM codec? |
| 16:22.12 | KranZ | <64kbit |
| 16:22.14 | privalac1 | What would cause Asterisk to stop playing any wav file? |
| 16:22.22 | privalac1 | Or gsm files... |
| 16:22.34 | KranZ | the codec wasnt loaded |
| 16:22.37 | sevard | e heck did the sun,mon,tues, etc sound files go? |
| 16:22.38 | KranZ | the files aren't there |
| 16:22.42 | Maxxed | i dont have any probelems droping/missing calls |
| 16:22.46 | sevard | s/e /where/g |
| 16:22.50 | Maxxed | its just the callerid info on the phone |
| 16:22.51 | privalac1 | They are there... |
| 16:23.06 | S4w | KranZ: sure? thats more than the g726 codec :-| |
| 16:23.08 | privalac1 | I uploaded one and it can be played fine |
| 16:23.09 | KranZ | what does the CLI say when * plays the file |
| 16:23.22 | KranZ | S4w: I said it was less than 64kbit |
| 16:23.27 | KranZ | i dont actually know |
| 16:23.28 | KranZ | heh |
| 16:23.36 | S4w | KranZ: isnt that too general? :-P |
| 16:23.46 | privalac1 | For exemple: Executing Playback("IAX2/600@600/1", "demo-congrats") in new stack |
| 16:23.47 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 16:23.52 | diclophis | hello all |
| 16:24.03 | KranZ | S4w: http://www.openh323.org/docs/bandwidth.html |
| 16:24.15 | diclophis | so... anyone have some tips for faxing with asterisk + PRIs |
| 16:24.15 | diclophis | ? |
| 16:24.23 | KranZ | privalac1 your verbosity is turned up right? |
| 16:24.36 | KranZ | and sounds are in /var/lib/asterisk/sounds |
| 16:24.50 | privalac1 | I just set it to 9999 |
| 16:24.59 | S4w | KranZ: sweet man, thank you |
| 16:25.03 | mercestes | diclophis: Yea, turn off silence supression, VAD, and Echocanceling on EVERYTHING. |
| 16:25.11 | mercestes | diclophis: and use Ulaw... |
| 16:25.13 | diclophis | ... how do I do that? |
| 16:25.26 | diclophis | mm not using sip or iax.. just incoming and outgoing on PRIs with spandsp |
| 16:25.34 | KranZ | S4w: yeah, that table is hard to read |
| 16:25.35 | diclophis | with a 4port digium card |
| 16:25.43 | mercestes | diclophis: Yea, I was just reading the +PRIs. |
| 16:25.54 | diclophis | i have gotten rex to work ..ok, but tx is crapping out halfway through |
| 16:25.55 | S4w | KranZ: i'll get it ;) |
| 16:26.02 | diclophis | doesnt make sense to me |
| 16:26.06 | KranZ | S4w: but the bits/sec column is what you're looking for |
| 16:26.25 | diclophis | and it craps out with .tifs that were made with the app_rxfax |
| 16:26.30 | KranZ | so... 13.2 not including ip overhead |
| 16:26.37 | diclophis | so its not like it is a bad encoding or something |
| 16:27.19 | *** join/#asterisk ToTo (n=ToTo@host21-83.pool8260.interbusiness.it) |
| 16:27.25 | S4w | KranZ: got it ;-) thanx fot the info |
| 16:27.28 | privalac1 | Kranz: Just tried with an other file: -- Executing Playback("IAX2/600@600/3", "you-entered") in new stack |
| 16:27.28 | privalac1 | <PROTECTED> |
| 16:28.15 | privalac1 | -rw-r--r-- 1 asterisk asterisk 1518 Jan 17 2004 /var/lib/asterisk/sounds/you-entered.gsm |
| 16:29.58 | watchy | hmm why wont this ata turn on dhcp |
| 16:32.30 | SplasPood | How does one change the default callerid of 'asterisk' when no CID info is available to something else? callerid= in the global section of sip.conf doesn't seem to be doin it |
| 16:32.38 | SplasPood | for inbound calls |
| 16:34.10 | sevard | Do we not have days of the week in /var/lib/asterisk/sounds ? |
| 16:35.09 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:35.15 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 16:35.26 | *** join/#asterisk Lino` (n=Lino@i577BCA13.versanet.de) |
| 16:35.52 | carrar | Morning kids!! |
| 16:37.49 | diclophis | howdy |
| 16:38.07 | mercestes | Sevard: check /var/lib/asterisk/sounds/digits |
| 16:38.48 | mercestes | Sevard: I think it's day-0 day-1 day-2 etc. etc. etc. |
| 16:39.10 | mercestes | Sevard: Kind of a stupid place ot put it tho, huh? |
| 16:39.16 | sevard | yeah :| |
| 16:39.47 | sevard | they ought to be named sunday.gsm etc |
| 16:40.10 | camelon | more than the 30% of my *Box used to get the busy tone (2E1s + TE4xxp + 2ATAs) and happen only with zap no sip . . . and the same for the outgoing calls . . .some idea? TIA |
| 16:40.30 | *** part/#asterisk S4w (n=saw@adsl-3-65-52.mia.bellsouth.net) |
| 16:40.48 | camelon | sorry: more than the 30% of my incoming and aoutgoing calls |
| 16:41.39 | sevard | mercestes: what about am / pm ? :) |
| 16:42.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:44.25 | *** part/#asterisk extremis (i=extremis@unon.net) |
| 16:46.02 | Damin | Uhh Ohh.. just had Asterisk lock solid.. |
| 16:46.16 | privalac1 | My problem all seem to come from outbound voice... Seems like my asterisk box can not output sound anymore... |
| 16:47.07 | *** join/#asterisk keyhack (n=keyhack@68.236.93.245) |
| 16:49.40 | sleepy_one | Damin, what did you do to it? :-) |
| 16:50.13 | *** join/#asterisk saftsack (n=saftsack@p54A7CC39.dip.t-dialin.net) |
| 16:50.46 | KranZ | sevard: they're named that so you can use variables to call the files |
| 16:50.56 | KranZ | day-${daynum} |
| 16:51.07 | sevard | ah |
| 16:51.30 | sevard | i'll just put symlinks to the dates for my purposes :) |
| 16:51.36 | KranZ | heh |
| 16:51.37 | pythos | I have gotten as far as a dial-tone on my FSX lines, what is next for using the FXO ports for something? <PS, Im a total newbie, as you can probably tell> |
| 16:51.39 | KranZ | that works too |
| 16:51.57 | KranZ | fxo ports connect to your phone provider |
| 16:52.28 | pythos | right, I have two FXO's and two POTS lines, Im not sure of what to do from there |
| 16:53.00 | KranZ | fxs -> phone, fxo -> local phone line |
| 16:53.16 | KranZ | for fxo to be useful, you need to already have phone service |
| 16:53.21 | CunningPike | pythos: Other way around -> smoke |
| 16:53.22 | sleepy_one | pythos, configure asterisk to use them ~wiki |
| 16:54.15 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 16:54.31 | KranZ | pythos: if you plan on using a voip provider and dont already have local phone service, the fxo lines are pointless |
| 16:54.37 | pythos | cunning, I have a headset to test which ports make dial-tone, the other two are th eones for telco, I would believe |
| 16:55.16 | sleepy_one | if you have the TDM400p green = FXS red = FXO |
| 16:56.54 | pythos | kranz, could be usefull for answer machine/voicemail |
| 16:59.05 | pythos | I guess |
| 17:00.18 | pythos | loosing servers? |
| 17:00.48 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 17:03.00 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 17:04.51 | Blackthorn | Hello, I have an issue dialing sip phone out the local pri with a dual ring. Dialing sip to sip, pri to disa back out, and all other combo's works fine. I've tried the dial with -r and it only does single ring but then I don't get busy tones etc etc. |
| 17:08.18 | aetius | Blackthorn: hard phone, softphone, or both? |
| 17:08.48 | aetius | (I don't know what your problem could be, but that would seem to be a good troubleshooting step). |
| 17:10.12 | *** join/#asterisk carlos-the-man (n=carlos@201.155.235.25) |
| 17:10.51 | Blackthorn | only sip phone out through pri. |
| 17:11.20 | carlos-the-man | guys I just installed a new mandriva and asterisk RPM binaries, totally newbie and want to get started with a softphone setup, are there any newbie instructions for me on the internet? |
| 17:11.27 | Blackthorn | sip to voice pulse, pots to pri and back out via disa works fine as well. |
| 17:11.57 | Blackthorn | yes quite a few carlos. go to google type asterisk install |
| 17:12.10 | aetius | www.asteriskdocs.org has a book |
| 17:12.48 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 17:12.59 | FuriousGeorge | yo |
| 17:13.04 | aetius | Blackthorn: right, but a hard sip phone, soft sip phone, or both (or different sip softphones)? |
| 17:13.19 | Blackthorn | i'm using sipura-2000's |
| 17:14.03 | Blackthorn | which i do not know by your question is hard or soft. :\ |
| 17:14.34 | aetius | heh, good point |
| 17:14.37 | aetius | I'd say hard. |
| 17:14.49 | aetius | so the next step would be to try another sip client and see if it has the same problem. |
| 17:14.57 | aetius | like one of the free softphones. |
| 17:14.58 | *** part/#asterisk rnovotny22 (n=Bob@198.57.19.126) |
| 17:15.09 | FuriousGeorge | how good is linux's support of amd64 architecture |
| 17:15.10 | aetius | that'll isolate the problem to either the sipura or the asterisk box. |
| 17:16.24 | FuriousGeorge | im getting a new mb/cpu/mem for one of my servers, and im finding the mobile barton chips i used to buy are just as expensive as athlon 64 now |
| 17:18.16 | Blackthorn | ahh.. good idea. |
| 17:18.19 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.161.Dial1.SanJose1.Level3.net) |
| 17:18.53 | Blackthorn | anyone know of software i could load on a pc to duplicate a sip phone? (soft i guess as you would say?) |
| 17:19.02 | nahirean | x-lite |
| 17:19.05 | nahirean | sjphone |
| 17:19.05 | coppice | amd64 linux works beautifully |
| 17:20.01 | Zodiacal | anyone know how asterisk can handle an alarm system? if the alarm system plugs directly into my pots, is there a way i can still use that line with asterisk? |
| 17:20.23 | Zodiacal | can i plug that line back into asterisk Fxo port? what if the alarm system uses the line, will asterisk know that that line is in use and to use another one? |
| 17:20.28 | Blackthorn | i show a x-lite 1.01 available on tucows. think that is latest version? |
| 17:20.39 | nahirean | google x-lite and get it from xten |
| 17:21.19 | aetius | yeah, the current version is 2.0 |
| 17:21.27 | aetius | http://www.xten.com/index.php?menu=download |
| 17:23.14 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-58.apple.com) |
| 17:24.43 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-58.apple.com) |
| 17:25.20 | *** join/#asterisk ToyMan (n=stuq@74-32-76-147.dsl1.mdl.ny.frontiernet.net) |
| 17:31.39 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 17:32.43 | Blackthorn | ok I setup the xlight, and placed a call. and it did not do a double ring. |
| 17:32.55 | Blackthorn | so guess it's in the sipura-2000 units |
| 17:33.09 | Blackthorn | x-light to * out pri |
| 17:36.38 | Zodiacal | any ideas? |
| 17:36.41 | Zodiacal | about the alarm system |
| 17:36.43 | gandhijee | anyone here familiar with writing web pages for polycoms? |
| 17:39.07 | Zodiacal | would * know if a pots line was in use, but via another device other than asterisk |
| 17:41.12 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 17:41.24 | Renacor | can I use multiple extensions in the goto statement? |
| 17:41.49 | Renacor | i.e. exten => s,1023,Goto(internal_numbers,extension1 extension2,1) |
| 17:43.55 | sleepy_one | cya all l8r :-D |
| 17:45.49 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 17:46.07 | *** join/#asterisk ToTo (n=ToTo@host21-83.pool8260.interbusiness.it) |
| 17:47.47 | gandhijee | my page seems to make the phone reboot, but its just standard XHTML |
| 17:48.21 | _Sam-- | does anyone know where to download ringtones that are regular ring tones, and not music ring tones |
| 17:48.37 | _Sam-- | like ones that sound like regular phones ringing |
| 17:49.06 | brodiem | _Sam-- look at the old news on voip-info.org, there was a set of 10 or so posted sometime in the last week or so |
| 17:50.15 | Blackthorn | well.. don't know what else to do. I basicly played with all the settings int he spa-2000. It only double rings going through * to the local pri. It dosn't double ring spa to spa or spa to voicepulse |
| 17:51.25 | gandhijee | is there a guide on how to upgrade the polycom firmware anywhere? |
| 17:51.29 | CunningPike | Wow - this may be old news, but I've just 'discovered' that SIP 1.6.5 lets the IP501 have BLF icons :) |
| 17:51.48 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 17:51.58 | CunningPike | gandhijee: Use an FTP provisioning server |
| 17:52.32 | gandhijee | is there on that you can recommned? |
| 17:52.46 | CunningPike | gandhijee: We use vsftp |
| 17:52.52 | gandhijee | thanks |
| 17:53.12 | gandhijee | CunningPike: you ever write any pages for you poy? |
| 17:53.15 | gandhijee | *poly |
| 17:53.20 | *** join/#asterisk Kernel_Core (n=I@116.230.dial-up.xter.net) |
| 17:53.30 | CunningPike | gandhijee: No - what are you trying to do? |
| 17:53.40 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 17:54.20 | gandhijee | write a page that will query the CDR database, then display the results on the phone |
| 17:54.42 | gandhijee | i haven't gotten there yet, i'm still tryin to figure out why my phone decides to reboot sometimes |
| 17:55.02 | file | Pisa International Airport = uh this is an airport? |
| 17:55.25 | CunningPike | gandhijee: What's the ultimate goal? Company directory on your phone? |
| 17:55.37 | gandhijee | i just got past the 404 error it gave me after i put in stuff in a text field |
| 17:56.00 | file | eep |
| 17:56.10 | file | I'll be back online from Paris... |
| 17:56.17 | gandhijee | CunningPike: its for a small motel, ultimate goal is to have it total the # of calls, mutilply it by a cost, and display total charges |
| 17:56.31 | gandhijee | CunningPike: then let them print it to a printer on the network |
| 17:56.43 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:56.53 | CunningPike | gandhijee: From the phone? Why not just use the Asterisk CDR? |
| 17:57.43 | file | security didn't even check my boarding pass... |
| 17:57.45 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:57.50 | gandhijee | cunningpike: the phone will be querying the Asterisk CDR =þ |
| 17:58.00 | file | so if anyone wants to... I dunno... threaten to blow up a set of departure gates... choose Pisa |
| 17:58.17 | gandhijee | lol@file |
| 17:58.32 | gandhijee | that only works if you are white, if u were brown, they would have searched you already |
| 17:58.36 | CunningPike | gandhijee: OK, so your guest can check their call totals? |
| 17:58.42 | file | this is true |
| 17:58.52 | file | plus I'm Canadian, so watch out - I'll hurt you with a moose... or a beaver |
| 17:59.03 | Corydon-w | file: you're not on a plane? |
| 17:59.10 | file | not yet |
| 17:59.10 | gandhijee | CunningPike: no, this is from the main console, small hotel, no way i am puttin poly's in every room here |
| 17:59.20 | file | scheduled to depart in 45 minutes |
| 17:59.43 | ghenry | hi again. is it possible to use exten s in a sip context? ie a sip users calls a number from outside your network and gets a menu, or do you need a nromal number to dial? |
| 17:59.45 | ghenry | brb |
| 17:59.55 | gandhijee | CunningPike: its for the hotel owner to check the number of calls, and see what the cost is. |
| 17:59.56 | CunningPike | gandhijee: But why not write a web page that goes directly against your CDR - I'm not sure what part the phone needs to play...... |
| 18:00.16 | file | I'm rushing to try to 1. Download xcode in time so I can write code on the plane and 2. Download the code in question |
| 18:01.40 | file | we're starting to board soon though :\ |
| 18:01.41 | gandhijee | CunningPike: like i said its a small motel, and the people that run it(my folks) have a hard enough time tryin to browse the web |
| 18:01.55 | [Airwolf] | Can someone tell me if this is good syntax for IFTIME, because i don't really understand the documentation on the wiki. |
| 18:01.59 | [Airwolf] | exten => s,2,Set(time=${IFTIME(9-17?office:nooffice)}) |
| 18:02.35 | gandhijee | CunningPike: having the interface directly on the phone emulates the traditional phone consoles of the mitels, etc |
| 18:02.48 | docelmo | Say what module in asterisk controls the Manager API? |
| 18:03.00 | CunningPike | I see - well, good luck with all that ;) |
| 18:03.06 | gandhijee | but first i gotta get around the phone rebooting problem |
| 18:03.17 | gandhijee | i;m starting to this it runs out of memory |
| 18:03.45 | gandhijee | *that |
| 18:04.24 | gandhijee | and apparently it doesn;t like ! in the text fields |
| 18:08.40 | CunningPike | This is a 601, I take it |
| 18:09.04 | gandhijee | yea |
| 18:09.31 | gandhijee | i'm just gonna load the firmware via tftp, i don't feel like configuring vsftpd right now |
| 18:09.51 | gandhijee | do the files have to be in a special directory or anything? |
| 18:09.57 | gandhijee | or just unzip in the ftp root? |
| 18:12.16 | mercestes | gandhijee The umm..tftp root. |
| 18:12.31 | *** part/#asterisk terrapen (n=cjs@166.70.183.109) |
| 18:12.39 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) |
| 18:12.44 | gandhijee | nm, it all set |
| 18:13.22 | docelmo | come on.. someone gots to know what module control's the manager API |
| 18:13.23 | CunningPike | gandhijee: The effort to setup ftp will be worth it eventually - you can set the Polycom to poll for updates etc |
| 18:13.44 | file | there's a group with kids behind me... |
| 18:13.47 | file | I hope they are not on my flight |
| 18:13.59 | Blackthorn | I saw a list on the wiki last night that showed which modules did what. but dont' know where it is now |
| 18:14.07 | gandhijee | true, but right now i just want to get this thing working and not rebooting |
| 18:14.41 | gandhijee | i just don't want to spend the time on it at this moment, i still have to get CDR w/my sql working right |
| 18:15.06 | CunningPike | http://www.voip-info.org/wiki-Asterisk+modules |
| 18:16.33 | Blackthorn | Do you know why an sipura spa-2000 would have double rings when calling out through * then pri? but not spa to spa? |
| 18:17.29 | brad_mssw | are you passing 'r' to your Dial() command? |
| 18:17.37 | CunningPike | Blackthorn: What he said :) |
| 18:19.14 | Blackthorn | nope |
| 18:19.46 | *** join/#asterisk littlejohn (n=little@host24-75.pool8716.interbusiness.it) |
| 18:19.49 | Blackthorn | x-light to * out pri works fine, pots to pri to * disa back through pri works fine as well. |
| 18:22.01 | *** part/#asterisk Kernel_Core (n=I@116.230.dial-up.xter.net) |
| 18:22.48 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 18:22.49 | SpaceBass | hey folks |
| 18:22.58 | SpaceBass | I'm having problems trying to get AGI scripts to work |
| 18:23.20 | SpaceBass | I can execute the script from the commandline as the asterisk user just fine, but its not working in my dialplan |
| 18:23.41 | elg | how does one check status, e.g. VMSTATUS? |
| 18:29.35 | camelon | Hi |
| 18:30.12 | camelon | if i'm looking for a voip console recepconist . . .wich you recommend? |
| 18:31.12 | tuxd00d | do you mean, receptionist console? |
| 18:31.29 | SpaceBass | arrruuuggg stipid AGI just will not work |
| 18:31.34 | camelon | yepppp . . sorry |
| 18:31.52 | camelon | tuxd00d . . .yeeeppp |
| 18:31.56 | tuxd00d | hardware or software? |
| 18:32.09 | camelon | both . . please |
| 18:33.04 | camelon | the best on the shop i hd and sf |
| 18:34.09 | tuxd00d | I don't have any experience... but polycom has good phones, and they have a hardware console add-on |
| 18:34.30 | camelon | but it works with asterisk? |
| 18:34.47 | tuxd00d | why wouldn't it work? |
| 18:35.02 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:35.19 | SpaceBass | should I be able to execute a BASH script with an arguement using AGI? |
| 18:35.28 | camelon | some asterisk list someone report ucomlete funcionality with * bt i don't remeber where |
| 18:35.52 | tuxd00d | I'm pretty sure the polycom one's work |
| 18:36.05 | camelon | OK . .thanks |
| 18:36.06 | tuxd00d | but you will have to research first |
| 18:36.22 | camelon | and about software? |
| 18:36.30 | camelon | soft console |
| 18:36.41 | *** join/#asterisk fjean (n=fjean@201.29.140.206) |
| 18:36.47 | tuxd00d | no experience with any softphones or the like |
| 18:37.25 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-58.apple.com) |
| 18:37.52 | camelon | OK . .thanks |
| 18:39.11 | fjean | hey guys, I have a little problem here and I would like to share it with you guys som you might have a suggestions on where I should start looking for, it's about SIP calls that just stays there even when conversation is finished |
| 18:39.59 | fjean | often, during the day, I have CDRs that show calls that are lot longer than the actual conversation, it might be a few minutes or more, see hours |
| 18:40.40 | fjean | even have calls that finishes after the start of a second call for the same user |
| 18:41.36 | fjean | the asterisk box is communicating with one SER box |
| 18:42.25 | fjean | only, but this happens often ; I know that there are some delays because its on the internet, so sometimes it might go over 500ms |
| 18:42.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:43.09 | fjean | but this is a real problem here ; does anyone has an idea on why this is happening ? SIP BYEs that are getting lost, etc |
| 18:44.19 | fjean | any hint is welcomed... |
| 18:44.22 | fjean | thanks |
| 18:47.21 | *** part/#asterisk trumpetinc (n=irc_kevi@1Cust45.VR1.PHX1.broadband.uu.net) |
| 18:48.02 | FuriousGeorge | so im shopping around for a new mb cpu for my asterisk server and im looking at the amd64 939 socket chips |
| 18:48.19 | FuriousGeorge | they say they got a 2ghz ht bus now, yet all the motherboards say 1ghz |
| 18:48.22 | FuriousGeorge | wadup with that |
| 18:48.43 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 18:49.42 | SpaceBass | get the pent d 805 |
| 18:49.50 | SpaceBass | and overclock it to 4.1 ghz |
| 18:49.59 | SpaceBass | articles all over digg.com and slashdot.com about it yesterday |
| 18:50.25 | FuriousGeorge | scardinal: if i wanted to OC and had an unlimited budget i would definately go with the opteron dual core series :) |
| 18:50.31 | FuriousGeorge | i hear they hit 3 ghz |
| 18:50.31 | bkw__ | its .org boi |
| 18:50.37 | bkw__ | but .com works but its really .org :P |
| 18:50.55 | SpaceBass | the 805 is a $120 cpu...2.2ghz and OCs to 4.1 and runs stable...just a suggestion |
| 18:51.23 | FuriousGeorge | SpaceBass: arent pentium-d's dual core 64 bit too? |
| 18:51.30 | FuriousGeorge | or are they 32 bit? |
| 18:51.34 | SpaceBass | FuriousGeorge, believe so....but not an expert |
| 18:51.35 | KranZ | they do emt64 |
| 18:51.44 | SpaceBass | just read the article yesterday and it piqued my interest |
| 18:51.46 | FuriousGeorge | KranZ: which is not really 64 bit or something |
| 18:51.59 | KranZ | i've installed 64bit gentoo on a p4 |
| 18:52.06 | FuriousGeorge | hmmm |
| 18:52.21 | FuriousGeorge | dual cores, 64 bit, and cheaper than amd.. how is this possible |
| 18:52.33 | KranZ | tho there's not much point unless you need >4gb memory |
| 18:53.13 | FuriousGeorge | to what? 64 bit? tbh i dont care if its 32 bit or 64, but the fact is that amd doest really make new 32bit chips, and thats usually what i buy |
| 18:55.12 | x86 | i like 32 bit x86 chips better... |
| 18:55.36 | SpaceBass | i ordered that 804 dual core and a mobo...plan on OCing it 3.8 or so and seeing how stable it is |
| 18:55.36 | x86 | only good 64 bit chips out are the Alpha, PPC64 (G5 and POWER), and Sparc64 ;) |
| 18:56.10 | x86 | i'm on a G5 iMac right now ;) |
| 18:56.14 | FuriousGeorge | SpaceBass: im looking at that chip now. a great price but the fsb is only 533 mhz (133 x 4), although each core has a mef of cache |
| 18:56.30 | SpaceBass | FuriousGeorge, i mean, for the price its a pretty nice little chip |
| 18:56.38 | SpaceBass | ok, anyone have expirence with AGI ? |
| 18:56.39 | FuriousGeorge | SpaceBass: that it is, im tempted |
| 18:56.53 | x86 | FuriousGeorge: get Pentium D's |
| 18:57.13 | x86 | dual core, 2mb cache per core (iirc), and 800mhz FSB |
| 18:57.15 | SpaceBass | FuriousGeorge, newegg.com had it for $128 i think |
| 18:57.23 | FuriousGeorge | x86: one meg each |
| 18:57.28 | FuriousGeorge | SpaceBass: thats where im looking |
| 18:57.56 | gandhijee | Conroe is pretty badass though |
| 18:58.04 | *** join/#asterisk bahamat (n=bahamat@207.67.145.230) |
| 18:58.09 | x86 | alpha was the best arch ever, imho |
| 18:58.13 | x86 | too bad HP killed it :( |
| 18:58.18 | SpaceBass | I got really excited about the chip yesterday, then I start pricing out a relitivly barebones box and was at $500 before I knew it |
| 18:58.19 | gandhijee | the K8's aren't bad though |
| 18:58.32 | gandhijee | the lead alpha designer helped design the k7 and k8's |
| 18:58.34 | SpaceBass | and considering that I'm getting my nokia 770 internet tablet today, I think Mrs Spacebass woul;dnt be too happy |
| 18:59.01 | FuriousGeorge | yeah, otoh the cheapest dual core amds are 300 bucks |
| 18:59.17 | gandhijee | AMD dual cores are better IMHO |
| 18:59.44 | gandhijee | no bus contingencies |
| 18:59.51 | gandhijee | what an intel dual core? |
| 19:00.02 | FuriousGeorge | yeah the pentium-d |
| 19:00.07 | gandhijee | the current intel dual cores are garbage and a 100% hack job |
| 19:00.12 | FuriousGeorge | which i know nothing about being a long time amd fan |
| 19:00.16 | gandhijee | they just did it to say the had the first dual core |
| 19:00.25 | SpaceBass | arrruuuggg I'm typing a blog entry on how to do something .... suggested people use nano over vi...I feel so dirty and cheap |
| 19:00.32 | gandhijee | they just put 2 cores on one package. |
| 19:00.43 | gandhijee | the still share the FSB |
| 19:00.48 | *** part/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 19:00.52 | *** join/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 19:00.54 | FuriousGeorge | yeah i always heard the intels dont dissipate heat well for that reason |
| 19:01.14 | gandhijee | hell even some of the intel guys don't like them. |
| 19:01.27 | gandhijee | but they are all jived up about Core |
| 19:01.48 | gandhijee | i have on in my laptop |
| 19:01.59 | gandhijee | its was pretty badass w/ linux, but i had to return it =( |
| 19:02.03 | fjean | is there a documentation on which SIP events are treated and how it's performed within * :-) |
| 19:02.09 | gandhijee | all that hardwork ended up goin down the drain |
| 19:03.08 | *** part/#asterisk bahamat (n=bahamat@207.67.145.230) |
| 19:03.55 | gandhijee | the pentium d i have next to me doesn't have sse3 either |
| 19:04.10 | FuriousGeorge | what pentiumd is that? |
| 19:04.29 | gandhijee | 920 i think |
| 19:04.35 | gandhijee | 3 GHz |
| 19:04.43 | FuriousGeorge | gandhijee: at the same time, you gotta admit that an application like telephony would lend it self well to multiple cores |
| 19:04.57 | FuriousGeorge | since even one conversation has two ends |
| 19:05.07 | gandhijee | true, but IMHO it would be better on AMD as the cores don't share the same bus |
| 19:05.08 | FuriousGeorge | not that you need two cores to make one call but you know what i mean |
| 19:05.54 | gandhijee | even with seperate physical processors the AMD would be a better solution, each CPU has dedicated bandwidth to the memory |
| 19:06.17 | FuriousGeorge | is it really that bad to have 2 tdmp400s in one box, if they dont share irq's with other devices |
| 19:06.32 | gandhijee | 2 is supposta be the max |
| 19:06.44 | FuriousGeorge | so no more than 2, right, gotcha |
| 19:07.01 | gandhijee | thats what i've heard, but it i would be very wrong |
| 19:07.19 | gandhijee | that should be i could be |
| 19:07.29 | SpaceBass | anyone have expirence with AGI ? |
| 19:07.35 | gandhijee | not it i would be |
| 19:07.37 | FuriousGeorge | i might have to call my boss and tell him hes getting a dual core athlon whether he likes it or not; ill mumble something about "memory bandwidth" and "doing it right" then hangup before he knows what im talking about. |
| 19:07.57 | gandhijee | lol |
| 19:08.29 | gandhijee | if u get him a board that supports dual procs, you can tell him he can have a quad processor machine =o |
| 19:08.45 | FuriousGeorge | now that would be beautiful overkill |
| 19:08.47 | FuriousGeorge | :) |
| 19:08.53 | gandhijee | the company i work for makes an amd ebx board w/ stackable hypertranport modules |
| 19:08.56 | gandhijee | very swank |
| 19:09.20 | FuriousGeorge | im not entirely sure what that is, but it sounds delicious |
| 19:09.28 | gandhijee | http://www.win-ent.com/MB-06047.htm |
| 19:09.44 | gandhijee | it lets you add another card on top and go dual processor when you want |
| 19:10.04 | gandhijee | we are working on a pass through HT bus so you can stack more cards on top and add infiniband |
| 19:10.27 | FuriousGeorge | ahh, ebx is for embedded devices |
| 19:10.29 | FuriousGeorge | pretty cool |
| 19:10.38 | gandhijee | anything is there is not PCI bus, PCIe only =/ |
| 19:10.47 | FuriousGeorge | gotcha |
| 19:10.52 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:11.13 | gandhijee | plays battlefield like a champ |
| 19:11.14 | *** part/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 19:11.21 | *** join/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 19:13.32 | cytrak | strange .. if I configure my zapata.conf and zaptel.conf to only use bchan=49-52 and leave the other ones unconfigured , shouldn't zap show channels only display those 4 ? |
| 19:13.40 | cytrak | ztcfg does |
| 19:13.47 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 19:14.12 | CunningPike | cytrak: I would have thought so...... |
| 19:14.43 | cytrak | hmm not with reload I just found out I have to stop and start * |
| 19:15.05 | kaz0358 | i have a small annoying talker echo problem. i have isolated the problem--the linksys WIP300 has an extremely sensitive microphone and if the earpiece volume is turned way up, the mic can pick up the audio. i can reduce or eliminate the echo by turning down the ear piece volume |
| 19:15.37 | gandhijee | kax0358: how are you liking that phone? i have 2 of them that i still need to setup |
| 19:15.48 | kaz0358 | however, i'm more concerned if we roll this out to other people.. especially those that need the volume turned up quite a bit for one reason or another. if the call is going from asterisk as voip to another voip long distance carrier, is there a way to eliminate the echo? |
| 19:15.57 | cytrak | the other thing is I can't find out what's my d channel .. I got my 3 span connecting to a PRI on a siemens PBX and even though I should be getting 24 channels I can only get 1-19 the others show as inactive |
| 19:16.45 | kaz0358 | gandhijee, it is a pretty cool phone. there are several things that i would like to see improved... the battery case should be more difficult to pull off. you should be able dial SIP urls. you should be able to dial a number that has an asterisk in it, but not necessary start with an asterisk |
| 19:16.51 | Damin | gandhijee: Didn't I run into you at Clue Con? |
| 19:17.26 | gandhijee | probably |
| 19:17.27 | gandhijee | i was one of the 2 indian people there |
| 19:17.31 | gandhijee | and i wasn't surj |
| 19:17.45 | sevard | what's the easiest way of listening to the wav files Monitor() generates? I was thinking a MoH class that played the directory, but a ControlPlayback would be neat |
| 19:17.52 | sevard | i just can't think of a way to tie it together |
| 19:18.01 | gandhijee | oddly enough we were both patel's which is quite ironic |
| 19:18.14 | Damin | Yeah.. I think Kielhofner and I talked to you.. doesn't someone in your family own Win-Ent? |
| 19:18.18 | kaz0358 | gandhijee, i'd also like to see the option to have the phone roam onto any open access point, which might be a security risk. but it would be nice to have the option. i don't have any problem with WEP or WPA. overall, it functions very much like a cell phone and it seems to have good battery life |
| 19:18.21 | gandhijee | yeah |
| 19:18.35 | gandhijee | i was gonna send him some old PiCA boards we had |
| 19:18.56 | gandhijee | he ended up not wanting them cuz i we couldn't find the firmware that had console redirect on them |
| 19:19.08 | gandhijee | i was gonna give him a couple for free |
| 19:19.27 | Damin | gandhijee: You should send a couple of those PL-06051 boxes over to Switchvox so they can certify them w/ their software and start selling them... |
| 19:19.41 | gandhijee | lemme see which one that is |
| 19:20.16 | gandhijee | i know he wanted the eSoft box, but we can't outright sell that one too him cuz of some licencing issues, and it would have been pretty expensive |
| 19:20.32 | gandhijee | Damin: we have something better than PL-06051 now =O |
| 19:21.01 | Damin | gandhijee: Cool.. I often need a not quite totally OEM solution for some clients.. |
| 19:21.16 | gandhijee | we just finished a box for intel, has pentium m on it w/ Xscale 465. workin on gettin HMP running on it |
| 19:21.17 | Damin | gandhijee: Something that looks different enough from a desktop PC that it will scare them away from it.. |
| 19:21.33 | gandhijee | and some of the intel guys should be talking to the asterisk folks about it |
| 19:22.17 | gandhijee | IRC the 6051 doesn't have any PCI slots, so no PSTN connectivity on that one |
| 19:22.19 | Damin | Hey.. anyone in the Baltimore, DC area? |
| 19:22.28 | gandhijee | dude |
| 19:22.36 | gandhijee | i was until last week |
| 19:22.56 | SpaceBass | anyone have expirence with AGI ? |
| 19:22.57 | Damin | I have free passes to ISPcon for anyone that wants them.. |
| 19:23.05 | *** part/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 19:23.27 | gandhijee | SpaceBass: you read the AsteriskTFOT book? it has some primer stuff in there |
| 19:23.58 | SpaceBass | no, been meaning to get the o'riley book |
| 19:24.03 | SpaceBass | I'm just trying to call a bash script |
| 19:24.10 | SpaceBass | or rather make it execute |
| 19:24.25 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 19:24.38 | Dr-Linux | question, does cisco 7940 have any button to make it boot? i plugged the power cable but it's not getting up! |
| 19:24.49 | gandhijee | Damin: you going to cluecon again this year? |
| 19:24.54 | SpaceBass | Dr-Linux, no button...using poe or the power cube? |
| 19:24.56 | jpabuyer | yeah.. the power button :) |
| 19:25.54 | Damin | gandhijee: Yea.. I'll be there.. |
| 19:26.11 | Dr-Linux | SpaceBass: i'm using power cable, but phone is not booting, whats wrong |
| 19:26.12 | gandhijee | i'll prolly see you there then |
| 19:26.23 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 19:26.25 | Damin | gandhijee: Cool.. cool.. |
| 19:26.28 | SpaceBass | Dr-Linux, could be the cable or the phone...guess that doesnt help much |
| 19:26.39 | SpaceBass | are you sure the cable is working? try the ole tongue test? |
| 19:26.50 | gandhijee | gonna try to get my uncle to send some sponser money to cluecon |
| 19:27.03 | gandhijee | maybe bring some of the newer hardware we have |
| 19:27.07 | Damin | gandhijee: Good! |
| 19:27.28 | Dr-Linux | SpaceBass: how can i do ole tongue test? :P |
| 19:27.46 | CunningPike | cytrak: A full PRI is 24 channels - 23B + 1D. Conventionally, the D-Channel is the 24th |
| 19:28.00 | mercestes | Put your tongue to it and see if it tickles. |
| 19:28.05 | CunningPike | So, even if you only have 3B channels, they are 1-3, and your D-Channel is 24 |
| 19:28.17 | SpaceBass | Dr-Linux, ^^^ what mercestes said |
| 19:28.52 | Dr-Linux | hhm.. lolz i have no handly access on the phone, the lady is doing that |
| 19:28.55 | CunningPike | cytrak: If you have 2 3-channel spans, the channels would be 1-3, 24 and 25-27, 48 - make sens? |
| 19:28.59 | CunningPike | sense |
| 19:29.02 | CunningPike | whatever |
| 19:29.06 | Damin | blitzrage: You wanna see something really ugly? |
| 19:29.10 | *** join/#asterisk flynux (n=prout@2a01:38:0:0:0:0:0:1) |
| 19:29.12 | mercestes | It doesn't hurt.....it's only a few volts...tell the woman to stick her tongue to it and tell you what hapens. |
| 19:29.18 | sevard | what's the easiest way of listening to the wav files Monitor() generates? I was thinking a MoH class that played the directory, but a ControlPlayback would be neat |
| 19:29.20 | sevard | i just can't think of a way to tie it together |
| 19:29.21 | sevard | gah |
| 19:29.39 | Dr-Linux | SpaceBass: do you know, the adaptor name, that we can use for cisco phone |
| 19:29.47 | mercestes | you talking about the in and out channels, Sevard? |
| 19:30.10 | mercestes | I always used soxmix to put them together.....there is a umm......script somewhere on the asterisk wiki...2wav2mp3 it's called I believe. |
| 19:30.12 | sevard | mercestes: sure, but i'm not even to the mixing part yet |
| 19:30.44 | mercestes | It showed up under asterisk + record + calls or asterisk + monitor or some nonsense on asterisk wiki...nice user contribution. |
| 19:30.54 | SpaceBass | Dr-Linux, I'm using PoE mostly...and knockoff power adaptors from ebay |
| 19:31.06 | jake1932 | here's an orig http://www.voipsupply.com/product_info.php?products_id=139 |
| 19:31.22 | mercestes | hang on..I'll find it. |
| 19:31.36 | SpaceBass | Dr-Linux, since you cannot see the phone....i wonder if its really not powering on....sounds like when my mother calls and tells me the entire internet is broken :) |
| 19:31.50 | gandhijee | rofl |
| 19:31.50 | sevard | mercestes: I suppose mixing these files would be neat but I'm talking about just playing them |
| 19:32.40 | gandhijee | SpaceBass: my parents call me and tell me the email is gone, when they minimized it to the tast bar |
| 19:32.44 | gandhijee | *task |
| 19:32.46 | Dr-Linux | SpaceBass: lol, i have configured more then 20 Cisco phones, but never seen any in real :P |
| 19:33.31 | SpaceBass | Dr-Linux, thats nuts! |
| 19:33.40 | SpaceBass | gandhijee, mine too...so I got them a mac for Christmas |
| 19:34.01 | SpaceBass | got tired of "i double right clicked on that HTML email and now my blah blah blah blah" |
| 19:34.18 | gandhijee | i was gonna load linux w/ fluxbox. but they supposedly needed to run some lame ass application for this new hotel they are building |
| 19:34.34 | gandhijee | so i had winblows on there, and they never ran the app |
| 19:34.40 | SpaceBass | yeah- I'm totally convenced that linux is not ready for the desktop |
| 19:34.56 | gandhijee | i think i could have had them rocking w/ fluxbox |
| 19:35.03 | gandhijee | its too easy to messup |
| 19:35.11 | SpaceBass | what is fluxbox |
| 19:35.20 | gandhijee | a very very minimal windows manager |
| 19:35.24 | SpaceBass | ahhh |
| 19:35.29 | gandhijee | just a menu and a fake me out taskbar |
| 19:35.30 | SpaceBass | get em a mac :) |
| 19:35.45 | sevard | SpaceBass: have you seen GXL or whatever it is |
| 19:35.46 | sevard | XGL |
| 19:35.52 | SpaceBass | not yet |
| 19:35.56 | gandhijee | god forbid, they would prolly be confused again |
| 19:36.00 | sevard | it's totally ready :) |
| 19:36.50 | SpaceBass | i guess I say its not ready b/c there are just some killer apps that are missing...and some functionality... |
| 19:37.15 | gandhijee | what ever happened to those guys that were making the windows type desktop? |
| 19:37.24 | SpaceBass | lindows? |
| 19:37.30 | sevard | like Open Office SpaceBass? because that's frieckn awesome |
| 19:37.32 | SpaceBass | they are around...selling out of Wallmart |
| 19:37.35 | gandhijee | nah i don't think it was them |
| 19:37.40 | sevard | I use linux for my desktop daily |
| 19:37.47 | gandhijee | the actual interface mimicked windows |
| 19:37.55 | SpaceBass | sevard, I think OpenOffice is great for anyone geeky...but its not ready for joe sixpack |
| 19:37.58 | SpaceBass | i use it daily |
| 19:38.07 | gandhijee | sevard: you are also in the asterisk room my friend |
| 19:38.19 | SpaceBass | exactly |
| 19:38.21 | aetius | on IRC, heh |
| 19:38.23 | gandhijee | SpaceBass: acutally i have my parents on OpenOffice |
| 19:38.25 | SpaceBass | that too |
| 19:38.28 | aetius | on freenode ... |
| 19:38.29 | gandhijee | yea |
| 19:38.30 | sevard | SpaceBass: My mom uses excel and word and databases at this school every day, she's been doing it for like 15 years. I sat her infront of Open Office and she was _right at home_ |
| 19:38.51 | sevard | now, if somebody can use Word for LONG LONG time, as long as that and be right at home with Open Office, I'm convinced it's ready |
| 19:39.04 | SpaceBass | I'm sure my mother could use OpenOffice but I don't want to take the calls every time somone sends her a word 2003 doc |
| 19:39.26 | sevard | last time i checked it could open ms word docs :P |
| 19:39.28 | SpaceBass | my problem with openoffice on the mac is that it runs as an x11 app, not mac (coca) native |
| 19:39.31 | gandhijee | my folks are also from india and have never really used a computer |
| 19:39.45 | SpaceBass | sevard, 2003? I know it can open 2000 and 87 |
| 19:39.46 | gandhijee | SpaceBass: there is a port of it to coca |
| 19:39.56 | sevard | SpaceBass: you can run X11 applications in the native coca enviroment without having to run an X11 WM |
| 19:40.00 | SpaceBass | gandhijee, oooooo really?....off to google I go |
| 19:40.03 | gandhijee | i just don't remember the name |
| 19:40.14 | SpaceBass | I've been trying thinkfree.com |
| 19:40.16 | SpaceBass | i like it a lot |
| 19:40.21 | gandhijee | yeah. i tried running it on my g4, i wanted to shoot myself |
| 19:40.37 | gandhijee | OS X takes up to many system resources |
| 19:41.00 | sevard | You have to have a _lot_ of RAM for OS X |
| 19:41.03 | CunningPike | SpaceBass: NeoOffice |
| 19:41.07 | sevard | it's very RAM hungry. |
| 19:41.16 | gandhijee | yeah apparently a gig doesn't cut it |
| 19:41.22 | SpaceBass | CunningPike, thanks |
| 19:41.37 | sevard | gandhijee: wow, i had 640mb ram and it ran pretty great |
| 19:41.40 | SpaceBass | ok...one last shot at my agi problem...then off to get my Nokia 770 .... |
| 19:41.46 | gandhijee | i dunno what it is |
| 19:41.58 | SpaceBass | I'm trying to run a bash script via AGI...i can execute it from the bash prompt, but it does nothing when I try as an AGI |
| 19:42.03 | sevard | SpaceBass: N 770!! |
| 19:42.06 | gandhijee | i upgraded ram first |
| 19:42.07 | SpaceBass | * CLI says it exits zero |
| 19:42.07 | sevard | SpaceBass: :( give me one :( |
| 19:42.17 | gandhijee | then went from a 450 to 1GHz, |
| 19:42.21 | SpaceBass | sevard, that baby is going to be my a/v system remote control...cannot wait! |
| 19:42.30 | sevard | SpaceBass: oh man i want one so bad |
| 19:42.36 | gandhijee | then hacked an nVidia card to run Quartz on it. |
| 19:42.41 | SpaceBass | sevard, I've been coveting it for a while |
| 19:42.42 | sevard | SpaceBass: I've wanted one since i heard about them 2 years before the launch |
| 19:42.50 | gandhijee | still horrible, im gonna give it the linux treatment soon |
| 19:42.58 | SpaceBass | I'm excited about 2006 os for it...SIP client! |
| 19:43.03 | mercestes | is it in the agi-bin directory? |
| 19:43.03 | sevard | SpaceBass: no way! |
| 19:43.16 | SpaceBass | mercestes, yeah, and the asterisk user owns it |
| 19:43.19 | sevard | SpaceBass: :D you'll have to tell me how it performs |
| 19:43.21 | SpaceBass | sevard, yeah...but no dates on when its coming out |
| 19:44.13 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 19:44.44 | KranZ | any word on when 1.4 is due? |
| 19:45.10 | sevard | so, is sox and soxmix two different apps? |
| 19:45.13 | KranZ | yes |
| 19:45.18 | *** join/#asterisk SajiD_KhaN (n=RusteD@203.145.159.44) |
| 19:45.23 | sevard | i guess so |
| 19:45.32 | sevard | didn't know about soxmix |
| 19:45.46 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
| 19:45.51 | SpaceBass | when I get back I was going to finish my dial plan to control iTunes with Asterisk...but something tells me I'll be on the Nokia 770 in the back yard with a beer |
| 19:46.04 | gandhijee | lol |
| 19:46.05 | KranZ | its good for combining the in and out recordings after monitoring a channel |
| 19:46.28 | KranZ | SpaceBass: that's expensive |
| 19:46.34 | sevard | SpaceBass: i'm jealous |
| 19:46.46 | KranZ | or is it wi-fi voip capable |
| 19:46.56 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 19:47.08 | SpaceBass | KranZ, update coming soon to support SIP |
| 19:47.31 | SpaceBass | And speaking of WiFi voip...I ordered my stinkin WIP330 OVER A MONTH AGO and voipsupply.com hasnt shipped |
| 19:48.16 | gandhijee | SpaceBass: don't feel bad, i waited like 3 months for mine |
| 19:48.32 | SpaceBass | how is it? |
| 19:48.43 | gandhijee | acutally i have the WIP300, didn't want that janky MS crap |
| 19:48.43 | brad_mssw | SpaceBass: dunno, they're saying late may for shipping ... voipsupply is usually good, I doubt it's their fault |
| 19:48.46 | SpaceBass | I'm starting to hate my Hitachi IP5000 |
| 19:49.02 | gandhijee | but i still had to wait 3 months for it |
| 19:49.32 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-26.wiline.com) |
| 19:49.37 | kaz0358 | gandhijee, you'll have to let me know what you think of the WIP300 when you get it up and going. i think its pretty good. i haven't used any of the first gen wifi voip stuff |
| 19:49.44 | SpaceBass | ok...off to the store |
| 19:50.24 | gandhijee | i have a ZyXEL here too |
| 19:50.24 | SpaceBass | my IP5000 works pretty well...doesnt always move b/t APs well...but it doesnt support WPA, which means I'm running a 2nd WIFI subnet |
| 19:50.31 | SpaceBass | I have the zyxel...it just died |
| 19:50.41 | gandhijee | what happened? |
| 19:50.41 | mercestes | I have a UTStarcom. |
| 19:50.57 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 19:50.59 | kaz0358 | spacebass, do you have the option of roaming on open access points with the ip5000? the linksys wip300 doesn't seem to have that option |
| 19:51.27 | gandhijee | any of you guys checked out the roamAD stuff? |
| 19:51.38 | gandhijee | thats what you need |
| 19:52.15 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 19:52.17 | gandhijee | lets you pass the audio stream from one access point to another |
| 19:52.25 | gandhijee | also some very groovy stuff |
| 19:53.03 | SpaceBass | kaz0358, yeah, it can roam, but its usually pointless since most open APs have captitive portals |
| 19:53.26 | SpaceBass | But the IP5000 will roam with in my WiFi network to different APs....fairly well |
| 19:53.50 | gandhijee | SpaceBass: you do it while you are talking? |
| 19:53.50 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 19:54.49 | SpaceBass | gandhijee, yeah |
| 19:54.50 | SpaceBass | used it |
| 19:55.00 | gandhijee | work well? |
| 19:55.19 | SpaceBass | I have 4 APs in my house for my LAN, but I run them at WPA and since the phone won't do WPA, its on its own open AP on seperate subnet |
| 19:55.32 | SpaceBass | it worked ok |
| 19:55.35 | SpaceBass | ok...off to the store |
| 19:55.47 | gandhijee | later |
| 19:57.54 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 19:58.01 | *** part/#asterisk fjean (n=fjean@201.29.140.206) |
| 20:07.00 | *** join/#asterisk drega (n=dforeman@p54A0A63F.dip0.t-ipconnect.de) |
| 20:10.11 | blebleble | anyone have or know of any good documentation on fine tunning the sipura 2002's? i have one that has choppy voice no matter what, and its connected directly to the internet, not behind any other devies |
| 20:12.48 | jake1932 | blebleble: is it choppy ata-> asterisk or only ata->asterisk->ITSP? |
| 20:13.20 | blebleble | ata->asterisk->ITSP, but my other lines (different parts of the state) all work fine its just this one |
| 20:14.02 | jake1932 | but you get a clean sound ATA->asterisk? |
| 20:15.38 | jake1932 | bbl |
| 20:15.39 | blebleble | no issues there too |
| 20:16.47 | zwelch | wow, i just put together a list of SIP related RFCs. going just off of the announcement list on one site (http://www.cs.columbia.edu/sip/news.html), i found 79 RFCs; while some are quite tertiary, it's somewhat daunting to see how many standards there are just for this one protocol ;) |
| 20:17.44 | zwelch | the question that i have is... to what extent does asterisk conform with each of them? |
| 20:18.24 | zwelch | ... and should i be looking elsewhere (than RFCs) for SIP-related standards? |
| 20:19.37 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 20:19.51 | lzhang | what are some good good php industry websites? |
| 20:20.32 | drega | eh can't remember off the top of my head but it a magazine |
| 20:20.37 | drega | phparch.com I think |
| 20:20.42 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 20:21.19 | drega | and phpmag.com |
| 20:21.22 | drega | sorry .net |
| 20:21.31 | FuriousGeorge | now i need a motherboard |
| 20:21.31 | lzhang | drega: thanks :) |
| 20:21.47 | drega | np |
| 20:21.52 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 20:21.55 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 20:22.12 | *** join/#asterisk dlynes_ (n=dlynes@216.251.149.66) |
| 20:23.15 | dlynes_ | Good afternoon, peeps |
| 20:23.24 | *** join/#asterisk iPBX (n=owned@68-169-204-147.agstme.adelphia.net) |
| 20:23.38 | iPBX | hi #asterisk |
| 20:23.45 | kaz0358 | hi ipbx |
| 20:24.07 | iPBX | i've got a big headache... my router doesn't let me open port ranges, just individual ports |
| 20:24.21 | iPBX | do i really have to open 10000 individual ports for RTP? |
| 20:24.29 | kaz0358 | okay.. and you are using asterisk behind a firewall and wanting to use sip? |
| 20:24.30 | iPBX | or can i just open like 10 if i only expect 10 rtp streams |
| 20:24.37 | iPBX | sip and iax |
| 20:24.54 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 20:24.55 | iPBX | i'm setting up a new server for a customer, and it's actually their router |
| 20:25.01 | kaz0358 | do you really need sip? iax2 is most often times a viable alternative |
| 20:25.01 | iPBX | their router is a windows server 2003 box |
| 20:25.12 | dpryo | lol |
| 20:25.23 | dpryo | iPBX: You probably need to open individual ports :D |
| 20:25.31 | iPBX | yeah, we're using eyebeam, and GXP2000's |
| 20:25.39 | *** join/#asterisk algorithmn (n=algorith@ool-45722b4c.dyn.optonline.net) |
| 20:25.40 | dlynes_ | iPBX: Why do you need to open ports, period? |
| 20:25.45 | iPBX | IAX is for voicepulse trunk |
| 20:25.48 | kaz0358 | well, you could also put ser in front of asterisk |
| 20:26.05 | dlynes_ | I've only ever had to do portmappings once |
| 20:26.07 | kaz0358 | and then you most likely wouldn't need to mess with the firewall. |
| 20:26.15 | dlynes_ | And that was because of buggy firmware in a particular router |
| 20:26.17 | iPBX | so that the server is accessible across the net, all the users are remote |
| 20:26.33 | dlynes_ | ah |
| 20:26.41 | dlynes_ | ok...that would be different, then :) |
| 20:26.56 | iPBX | i have my server in the DMZ, so it works ok :-p |
| 20:27.04 | iPBX | used ip tables and i'm good with that |
| 20:27.18 | iPBX | this one they're using the routing functions in winserver 2003 |
| 20:27.29 | iPBX | which allows NAT and port exceptions/forwarding |
| 20:27.37 | iPBX | but stupid crap doesn't let you specifiy a port range, |
| 20:27.44 | iPBX | so I'm concerned about RTP |
| 20:27.47 | kaz0358 | ipbx, you might take a look at ser http://www.voip-info.org/wiki-SIP+Express+Router .. or you can mess with your firewall |
| 20:28.06 | iPBX | even with Ser, won't i still need to open all those RTP ports? |
| 20:28.22 | kaz0358 | ipbx, no it has ways around most firewalls |
| 20:28.24 | dlynes_ | I still don't understand why you need to open so many rtp ports |
| 20:28.38 | dlynes_ | Each phone should only need one rtp port, non? |
| 20:28.46 | iPBX | dlynes_ I read RTP uses ports 10000-20000 |
| 20:28.54 | algorithmn | grandstream BLF w/hint groups giving "INVITE with REPLACEs" w/also "Remote host can't match request BYE"... idea's? |
| 20:29.03 | dlynes_ | It'll use anything in that range...you don't need to use the whole range, though |
| 20:29.08 | *** join/#asterisk harlequin516 (n=sham@65.39.84.194) |
| 20:29.22 | dlynes_ | iPBX: Take a look at rtp.conf |
| 20:29.22 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 20:29.32 | iPBX | good idea :-> |
| 20:29.37 | dlynes_ | iPBX: You can modify the range |
| 20:29.49 | dlynes_ | iPBX: Most ip phones allow you to change the port as well |
| 20:30.01 | iPBX | so lets see how would i figure out how many ports i need to actually open... |
| 20:30.02 | harlequin516 | Can I get a t1 provisioned a few lines for data and a few lines for POTS lines? |
| 20:30.15 | kaz0358 | harlequin, yes.. |
| 20:30.18 | harlequin516 | I mean IP traffic for data |
| 20:30.26 | iPBX | 5 phones... 1 IAX trunk that supports up to 4 simunateous calls... |
| 20:30.27 | dlynes_ | harlequin516: yes, but you generally need to use a channel bank and specialized equipment like an adtran 850 or something |
| 20:30.39 | Blackthorn | Hi, I'm still working on my double ring issue. It only does it sipura spa-2000 units talking through * to the local pri. The x-light client on the computer does not. I get a ring, silence, lower ring then 1/2 the way through that ring a loud ring again. Any suggestions what to adjust? |
| 20:31.04 | Blackthorn | i think i've switched, changed, modfied every setting in the spa-2000 today :\ |
| 20:31.05 | kaz0358 | blackthorn, i get that too, but i haven't worked on solving it |
| 20:31.08 | harlequin516 | dlynes_: Is that through my local telco? |
| 20:31.19 | dlynes_ | harlequin516: no...you would buy that stuff yourself |
| 20:31.43 | dlynes_ | harlequin516: or you can pay the telco or an interconnect to do it for you |
| 20:31.52 | dlynes_ | harlequin516: but then it's usually not worth it :) |
| 20:32.04 | Blackthorn | kaz: reading through several web searches i've kind of theorized it's that the ata places the call to * and starts rining, then you hear the ring from the pri as well. |
| 20:32.41 | harlequin516 | Okay So What kind of company provides this kind of service? I used to work for a company that had a single T1 that they could configure as many phone channels or data as they wanted. |
| 20:32.48 | Blackthorn | ata to ata, pri to * (disa) back to pri works fine. |
| 20:32.55 | Blackthorn | as well as ata to voice pulse. |
| 20:33.11 | kaz0358 | blackthorn, yeah.. i've wondered if that might not be the case |
| 20:33.13 | dlynes_ | harlequin516: You can either call your local telco, or look in the phone book under 'phone systems' |
| 20:33.33 | dlynes_ | harlequin516: the odd interconnect company (phone system installer) will do it, as well |
| 20:33.53 | dlynes_ | harlequin516: also, some of the more specialized networking services companies will do it |
| 20:34.32 | dlynes_ | harlequin516: I suspect you're probably in the states, so I wouldn't be able to suggest somewhere to go |
| 20:34.35 | harlequin516 | Is it cheaper to get it all setup in my house, or at a hosting company? |
| 20:34.54 | harlequin516 | Yes, am in Phoenix, AZ |
| 20:34.59 | dlynes_ | harlequin516: a colo facility is usually considerably cheaper for a pri/t1 cost |
| 20:35.03 | kaz0358 | blackthorn, and you have played with notifyringing in sip.conf? |
| 20:35.18 | dlynes_ | harlequin516: however, you have to pay a locker cost on top of that |
| 20:35.29 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:35.30 | dlynes_ | harlequin516: so unless you have a need for a locker, it might not be worth it |
| 20:35.49 | dlynes_ | harlequin516: and if you're at a telco colo, they'll already give you a pri/10MB ethernet split |
| 20:35.51 | harlequin516 | Ick , is there a resopurce on the web that will guide me through this process? |
| 20:36.23 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:36.23 | dlynes_ | harlequin516: I wouldn't know about the US, but in Canada, there's www.gt.ca (group telecom) |
| 20:36.52 | dlynes_ | harlequin516: actually |
| 20:37.00 | dlynes_ | harlequin516: level 3 might over that service in the states |
| 20:37.05 | dlynes_ | s/over/offer/ |
| 20:37.11 | Blackthorn | not yet |
| 20:37.34 | dlynes_ | Blackthorn: was that a response to me? |
| 20:37.54 | dlynes_ | guess not |
| 20:37.54 | Blackthorn | to kaz |
| 20:38.10 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:38.10 | kaz0358 | blackthorn, i'm doing a real quick test on it |
| 20:39.02 | kaz0358 | blackthorn, that fixed it |
| 20:39.32 | dlynes_ | man....major geek speak going down in #perl |
| 20:39.47 | Blackthorn | really? what did you set? |
| 20:40.16 | lzhang | I've noticed that 'show queues' doesn't show realtime queues... is there any way around that |
| 20:40.40 | lzhang | I've also tried it through the manager interface |
| 20:41.34 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 20:42.18 | kaz0358 | blackhorn, notifyringing = no in sip.conf |
| 20:46.01 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:46.30 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:46.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:47.54 | Blackthorn | kaz: :( didn't work for me. But thanks for trying and sharing the info |
| 20:49.34 | *** join/#asterisk tuxd00d (n=tuxinato@69-169-11-49.lmdaca.adelphia.net) |
| 20:50.29 | kaz0358 | blackhorn, are you sure it isn't something you can configure on the sip phone? i'm just guessing here. |
| 20:53.35 | SpaceBass | back |
| 20:53.40 | SpaceBass | with the nokia 770 in hand |
| 20:55.32 | SplasPood | w00000 Polycom firmware 1.6.6.. Seems to remove the arbitrary 7 monitored extension limit on the 601! |
| 20:55.37 | SplasPood | gotta te |
| 20:55.39 | SplasPood | test |
| 20:57.17 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:58.12 | caio1982 | does asterisk supports that grandstream gxv3000 video phone? it seems to support only the h264 format but asterisk only accept h263 right? or are they some level compatible? |
| 20:58.51 | dlynes_ | SpaceBass: How is that phone? |
| 20:59.22 | *** part/#asterisk [vmwarez]dotcom (n=jjones@216.147.224.254) |
| 21:00.42 | SpaceBass | its not a phone |
| 21:00.46 | SpaceBass | just a tablet |
| 21:00.48 | SpaceBass | so far, so good |
| 21:01.41 | dlynes_ | Yeah...that's the one that's a tablet with a phone in it, and uses Linux for the OS, right? |
| 21:01.51 | SpaceBass | right, but no phone |
| 21:02.01 | dlynes_ | ah...ok...thought there was a phone with it |
| 21:02.08 | SpaceBass | no...sip client coming soon |
| 21:02.11 | dlynes_ | stupid nokia...don't they know they're a phone company? :) |
| 21:02.22 | dlynes_ | SpaceBass: But I've already got a sip client for my Nokia 6670 |
| 21:02.30 | SpaceBass | :) |
| 21:02.37 | SpaceBass | i didnt buy this for voip but it would be nice |
| 21:02.46 | dlynes_ | SpaceBass: and for my Sharp Zaurus :) |
| 21:03.15 | dlynes_ | But trying to do voip on a sharp zaurus without a wireless network card is an exercise in frustration |
| 21:03.20 | dlynes_ | the usb link is extremely slow |
| 21:03.37 | dlynes_ | there's an irc client for that pda, too |
| 21:03.46 | dlynes_ | but not like i want to try typing in irc on it :) |
| 21:04.32 | SplasPood | my buddy bought a 770.. it was cute, albeit useless seeming |
| 21:05.05 | *** join/#asterisk Coriantum (n=asdfkle@71-213-5-22.slkc.qwest.net) |
| 21:05.23 | Coriantum | Is pbx_builtin_setvar_helper gone in 1.2? |
| 21:06.03 | drega | SplasPood I thought ov getting one for light weight shitter reading |
| 21:06.07 | drega | ;) |
| 21:07.49 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:08.13 | SplasPood | drega: thats what mah blackberry is for :) |
| 21:08.37 | Mystiq | but your blackberry can't view video streaming :) |
| 21:08.48 | Mystiq | the n770 can as a matter of fact |
| 21:08.53 | Coriantum | Anyone know what causes this: |
| 21:08.56 | Coriantum | warning: implicit declaration of function `ast_separate_app_args' |
| 21:09.10 | SpaceBass | well the n770 cannot get on my wifi network yet...so Im about to throw a fit |
| 21:09.25 | Mystiq | SpaceBass: how come it can't ? |
| 21:09.55 | *** join/#asterisk tobmoox (n=xoombot@24-72-198-209.cm-dynip.usadig.com) |
| 21:09.56 | SpaceBass | have 2 wifi networks...one wpa2 with 64bit key...the other is open but with mac filtereing...trying to get on the open one, but cannot find the damn mac on this thing |
| 21:10.11 | *** part/#asterisk tobmoox (n=xoombot@24-72-198-209.cm-dynip.usadig.com) |
| 21:10.45 | Mystiq | SpaceBass: can't you just use ifconfig in a terminal ? |
| 21:10.53 | SpaceBass | dont have terminal installed yet |
| 21:10.57 | SpaceBass | doesnt come with it |
| 21:11.15 | harlequin516 | Okay I am trying to find a company that will host my asterisk PBX in Phoenix, AZ with a T1 PRI from the localTelco. |
| 21:11.21 | harlequin516 | Where dfo I start looking? |
| 21:12.13 | Coriantum | harlequin516: I'll msg you about that |
| 21:12.20 | Mystiq | SpaceBass: http://770.fs-security.com/xterm/ |
| 21:12.25 | harlequin516 | Okay |
| 21:12.31 | Mystiq | SpaceBass: www.maemo.org for alot more goodies |
| 21:12.33 | SpaceBass | yeah, know i can get one |
| 21:12.46 | Mystiq | ah, ok :) |
| 21:12.48 | SpaceBass | been following maemo for a while....just thought itd be easy to get this thing up and running |
| 21:13.11 | SpaceBass | didnt want to pair with my bt phone either |
| 21:14.01 | Mystiq | well, mine broke a month ago.. doesn't even want to boot anymore |
| 21:14.03 | Mystiq | *g* |
| 21:14.21 | SpaceBass | no warrenty? |
| 21:14.34 | Mystiq | yes, but too lazy :p |
| 21:14.38 | SpaceBass | lol |
| 21:15.15 | *** join/#asterisk niter3 (n=klutch@d57-102-239.home.cgocable.net) |
| 21:16.29 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 21:16.58 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) |
| 21:19.09 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011868204.cable.net.co) |
| 21:21.17 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 21:21.40 | Blackthorn | thanks kat. i'll look into it. time to close office |
| 21:22.36 | SpaceBass | ok...still need some * help... |
| 21:22.45 | SpaceBass | trying to execute an BASH script via AGI |
| 21:22.53 | rpm | is there anything i can use for asterisk which generates calls? |
| 21:23.04 | SpaceBass | I can execute it fine from the bash prompt, but the AGI call just returns 0 but does nothing |
| 21:23.07 | SpaceBass | rpm, callfiles |
| 21:31.51 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011868204.cable.net.co) |
| 21:34.44 | *** join/#asterisk my007ms (n=my007ms@196.202.70.1) |
| 21:34.58 | my007ms | hello all |
| 21:35.56 | my007ms | can i limit time of call in one trunk |
| 21:36.53 | my007ms | for example make ppl can not make more then 1 min when thy call out from exact trunk |
| 21:38.12 | tuxd00d | anyone else having trouble accessing sellvoip.net? |
| 21:40.58 | cytrak | is there a way to reload voicemail.conf ? |
| 21:42.01 | De_Mon | reload app_voicemail.so |
| 21:45.05 | cytrak | cool thanks |
| 21:47.31 | dlynes_ | tuxd00d: someone was on here the other day complaining about it, too |
| 21:48.00 | De_Mon | use calleveryone their prices are great if you buy in bulk! |
| 21:48.18 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-72-39.telkomadsl.co.za) |
| 21:48.54 | dlynes_ | you get what you pay for, too :) |
| 21:50.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:56.07 | tuxd00d | dlynes: I just called sellvoip... they were unaware of the problem... but they see it now... silly |
| 21:56.40 | tuxd00d | De_Mon: why the chuckles? |
| 21:59.41 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 22:01.19 | *** join/#asterisk Harlyman^ (n=skrot@torino.crystalnet.no) |
| 22:02.22 | Malthus | hi all |
| 22:02.50 | Malthus | my T1 lines start ignoring DTMF after asterisk has been running a while |
| 22:03.15 | Malthus | they start working again after I restart asterisk |
| 22:03.29 | Malthus | how can I debug this? |
| 22:04.05 | *** part/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 22:04.10 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 22:04.10 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 22:04.11 | Malthus | it happens on a e&m wink telco line as well as an fxo_ls channel bank connection |
| 22:04.38 | SplasPood | hrm... whats the trick to getting the contact directory working on the polycom 601.. I know I've done it before, but I forgot what I did... Works fine on my 501s.. |
| 22:05.09 | CunningPike | SplasPood: Should be the same - I'd be interested in seeing if the limit has gone away too..... |
| 22:05.55 | dlynes_ | Malthus: regular t1, or pri? |
| 22:06.23 | SplasPood | CunningPike: Thats what I'm working towards, but now I can't even get the directory to work.. Although it does on my 501 |
| 22:06.24 | dlynes_ | Malthus: ah..nvm...didn't see your last line |
| 22:06.51 | CunningPike | SplasPood: What problem are you having? |
| 22:06.55 | Malthus | heh |
| 22:07.11 | Malthus | DNID comes in missing end digits too :) |
| 22:07.32 | dlynes_ | Malthus: sounds like your gains might need to be adjusted, then |
| 22:08.01 | SplasPood | CunningPike: It just doesn't save my entries, and also if I manually create the xml I see it pull it via HTTP on boot, but it doesn't do anything with it |
| 22:08.01 | dlynes_ | Malthus: your rxgain/txgain can be adjusted from -100 to 100 |
| 22:08.18 | SplasPood | CunningPike: However from the 501 I was able to just go an add one and it worked no problem |
| 22:08.20 | dlynes_ | Malthus: I believe the defaults are 0 and 0 |
| 22:08.24 | Malthus | when I switch from em_w to featd, it says its not featd and its switching to em_w, and DNID works perfectly |
| 22:08.38 | CunningPike | SplasPood: That's odd - you're entering from the keypad, or from a file? |
| 22:08.45 | SplasPood | CunningPike: I've tried both |
| 22:08.48 | Malthus | when I switch back to em_w in zapconf it stops working again :) |
| 22:08.50 | dlynes_ | Malthus: yeah...you're talking greek to me, now |
| 22:09.00 | Malthus | oops, sorry |
| 22:09.02 | CunningPike | SplasPood: Let me make sure ours works..... |
| 22:09.03 | dlynes_ | Malthus: I understand em_w is em and wink |
| 22:09.15 | dlynes_ | Malthus: and i know it's some proprietary signalling method |
| 22:09.16 | Malthus | featd is a variant of e&m wink |
| 22:09.20 | dlynes_ | Malthus: but that's it |
| 22:09.43 | Malthus | you think the gains could be the prob? |
| 22:09.56 | Malthus | why would it work when asterisk is just started? |
| 22:10.25 | CunningPike | SplasPood: Ours is working fine - Directory..Contact Directory..More..Add.. fill in the form and hit Save |
| 22:11.24 | SplasPood | CunningPike: Yes I know the procedure. its not working :P |
| 22:11.35 | CunningPike | SplasPood: :P |
| 22:11.44 | dlynes_ | Malthus: could just be line degradation, too |
| 22:11.44 | SplasPood | I remember I had this problem with another 601 I setup |
| 22:11.48 | SplasPood | but I forget the fix |
| 22:11.57 | SplasPood | and looking at the config files for that phone.. everything is the same |
| 22:12.30 | Malthus | then why would a restart of asterisk solve the prob? |
| 22:12.40 | dlynes_ | Malthus: ah...no idea then |
| 22:13.41 | dlynes_ | Malthus: You coudl try something like debug channel Zap/1-1 though, too |
| 22:13.53 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 22:14.07 | Malthus | will do |
| 22:14.17 | Malthus | didn't realize zap had a debug command |
| 22:14.33 | dlynes_ | Malthus: it doesn't...that's just a general debug |
| 22:14.33 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-80.z142-153-67.customer.algx.net) |
| 22:14.36 | Malthus | k |
| 22:14.40 | dlynes_ | Malthus: it's channel independent |
| 22:16.13 | bzbw | hi, I have 3 groups and want each member extension in the group to be able to pickup calls from same group ONLY, how do I set up the hint group? |
| 22:16.42 | dlynes_ | Malthus: you need to adjust your debug level, too |
| 22:16.47 | dlynes_ | Malthus: set debug 10 say |
| 22:17.55 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 22:17.56 | Malthus | ok |
| 22:18.03 | Malthus | I have something |
| 22:18.10 | Malthus | where can I paste? |
| 22:18.14 | dlynes_ | ~pb |
| 22:18.15 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 22:18.32 | CunningPike | Ouch |
| 22:18.37 | dlynes_ | heh |
| 22:18.55 | dlynes_ | actually |
| 22:19.02 | dlynes_ | someone should modify the text for ~pb |
| 22:19.15 | dlynes_ | There's three or four other pastebins, too |
| 22:19.29 | De_Mon | hrm, CLI> dial <context>,<patern> doesn't work |
| 22:19.31 | Malthus | http://pastebin.com/714340 |
| 22:19.35 | Malthus | check that out |
| 22:19.47 | De_Mon | oh help dial works |
| 22:19.48 | dlynes_ | De_Mon: Do you have chan_alsa or chan_oss loaded? |
| 22:19.57 | Malthus | I get NULL frame messages when I send DTMF |
| 22:20.15 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-202.cybersurf.com) |
| 22:20.21 | dlynes_ | Malthus: yeah...i'm not the one to help you probably...I suspect i wouldn't even have a clue what i'm looking at |
| 22:20.30 | Malthus | heh |
| 22:20.35 | Malthus | I am so lost |
| 22:20.42 | dlynes_ | I'll take a look |
| 22:20.48 | lzhang | Malthus, how did you get that frame debugging output? |
| 22:20.49 | dlynes_ | But, I'm not going to promise anything |
| 22:21.18 | Malthus | lzhang: debug channel Zap/3-1 |
| 22:21.28 | lzhang | cool thanks |
| 22:21.36 | Malthus | dlynes_ thanks alot |
| 22:21.48 | dlynes_ | Malthus: yeah...that's completely meaningless to me |
| 22:21.52 | Malthus | if you start me off in the right direction I'm willing to look at code |
| 22:21.56 | De_Mon | dlynes_ its dial patern@context |
| 22:21.57 | Malthus | :) |
| 22:22.31 | dlynes_ | De_Mon: yeah, but afaik, the dial application from within the cli uses the alsa or the oss channel modules |
| 22:23.05 | De_Mon | yeah, your point? |
| 22:23.06 | dlynes_ | so you actually need to have one or the other of those two channel modules loaded |
| 22:23.11 | dlynes_ | or you won't have the dial command |
| 22:23.21 | dlynes_ | You said the dial command wasn't working |
| 22:23.27 | De_Mon | it was the wrong syntax |
| 22:23.28 | dlynes_ | ah |
| 22:23.34 | dlynes_ | thought it just wasn't working, period |
| 22:23.38 | De_Mon | not 'doesnt exist' just .. yah |
| 22:24.12 | Malthus | heh, somebody deleted that bug from asterisk mantis |
| 22:24.44 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 22:24.48 | SplasPood | CunningPike: I think the phone is real anal about the formatting on that .xml... I deleted the one I created, reset the phone, and now it seems to let me add entries from the phone |
| 22:25.01 | Manipura | I start asterisk, by just typing asterisk, and my IAX phone doesn't work, I type asterisk -vvvvgc and my IAX phone can log in again. Anyone know why this is? |
| 22:25.07 | CunningPike | SplasPood: OK- great |
| 22:28.03 | *** join/#asterisk kippi (n=nlabla@cpc1-hatf3-0-0-cust211.lutn.cable.ntl.com) |
| 22:28.04 | kippi | hey |
| 22:28.16 | kippi | has anyone used Tiger with asterisk cdr? |
| 22:29.31 | Malthus | would there be a way to find out if DTMF detection has been turned off? |
| 22:29.37 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-145-155.gdrpmi.dsl-w.verizon.net) |
| 22:30.32 | DarKnesS_WolF | i have a little strange issue i'm trying RealTime with MySQL but it's not working and nothing on the logs .. i can see that i'm connected to Mysql but when i try to register to my asterisk server i linphone saying not implemented .. and don't ask for apssword or anything |
| 22:31.16 | dlynes_ | Malthus: you could try filing a bug report (bugs.digium.com), but first make sure nobody else has reported a similar error |
| 22:32.11 | dlynes_ | Malthus: if somebody has reported a similar error, maybe post a confirmation message that you too have encountered the error, and then extrapolate by saying how it manifests itself on your system slightly different from the other poster, or exactly the same, ... |
| 22:32.30 | gandhijee | /j windows |
| 22:32.41 | *** part/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 22:34.15 | Malthus | dlynes: I was hoping it wouldn't come to this :( |
| 22:34.57 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 22:35.16 | *** join/#asterisk zeroten (n=cas@213-63-26-86.static.jdsl.net.artelecom.pt) |
| 22:35.27 | dlynes_ | Malthus: they've been covering off any new bugs pretty quick lately |
| 22:35.40 | dlynes_ | Malthus: They're aggressively trying to get the bug count down |
| 22:35.45 | Malthus | k |
| 22:35.53 | *** part/#asterisk iPBX (n=owned@68-169-204-147.agstme.adelphia.net) |
| 22:35.55 | Malthus | I'll have to upgrade to 1.2.7 and try then |
| 22:36.15 | Malthus | but its in emergency production |
| 22:36.16 | dlynes_ | Oh...yeah...don't bother reporting a bug, unless you're using the latest version (1.2.7.1) |
| 22:36.26 | dlynes_ | They might not deal with it, otherwise |
| 22:36.32 | Malthus | didn't want to take it down |
| 22:36.38 | dlynes_ | Well, you can report it |
| 22:36.51 | Malthus | I'll upgrade it |
| 22:36.53 | DarKnesS_WolF | got it i removed my sip.conf :) so i don't have binding port or address |
| 22:37.02 | dlynes_ | but your mileage may vary how much attention they'll give it |
| 22:37.23 | dlynes_ | yeah...they've got it down to 252 bugs now |
| 22:37.32 | dlynes_ | And that's for the code that will become Asterisk 1.4 |
| 22:37.33 | Malthus | thats not bad |
| 22:37.48 | SplasPood | CunningPike: IT WORKS!!!!!! |
| 22:37.53 | Malthus | lemme try to get it back up to 253 :) |
| 22:37.56 | SplasPood | CunningPike: Polycom fixed it! |
| 22:37.58 | CunningPike | More than 7??????? |
| 22:37.59 | dlynes_ | lol |
| 22:38.02 | CunningPike | WAHOO!!! |
| 22:38.04 | SplasPood | CunningPike: Yea, I've got 9 in there |
| 22:38.09 | CunningPike | Fantastic |
| 22:38.17 | SplasPood | yes, up to 48 total now |
| 22:38.21 | SplasPood | only on the 601 tho |
| 22:38.24 | SplasPood | 8 on the 501 and 301 |
| 22:38.25 | CunningPike | You should update the wiki |
| 22:38.29 | SplasPood | CunningPike: Gonna |
| 22:38.38 | CunningPike | Great - thanks for testing |
| 22:38.52 | dlynes_ | CunningPike: btw |
| 22:39.04 | dlynes_ | CunningPike: Have you encountered any flakiness with blf's? |
| 22:39.38 | CunningPike | dlynes_: Not much - sometimes they go away for a short time, but they've been way better since 1.6.5 |
| 22:39.46 | dlynes_ | 1.6.5? |
| 22:39.55 | SplasPood | 1.6.6 |
| 22:39.56 | dlynes_ | that must be a polycom firmware version or something |
| 22:40.18 | SplasPood | 1.6.6 is the first version with the increase in limits |
| 22:40.23 | SplasPood | must've been released in the last week or two |
| 22:40.46 | zeroten | is there a page showing the great new features of 1.4? |
| 22:42.17 | dlynes_ | zeroten: svn co http://svn.digium.com/svn/asterisk/trunk asterisk-trunk-pre-1_4 ; cd asterisk-trunk-pre-1_4 ; svn log |
| 22:42.54 | dlynes_ | zeroten: that's probably about the closest you're going to get to a new feature list for 1.4 |
| 22:43.06 | dlynes_ | zeroten: until shortly before or after its release |
| 22:43.16 | zeroten | well, i don't i don't all svn check ins.... |
| 22:43.24 | zeroten | don't want that |
| 22:43.34 | dlynes_ | zeroten: then you're going to have to wait |
| 22:44.01 | dlynes_ | but shared line appearances is one of the big new improvements |
| 22:44.04 | dlynes_ | and AEL2 |
| 22:44.29 | zeroten | what about that ajax stuff mark was doing? will it be for 1.4? |
| 22:44.34 | dlynes_ | and of course better SIP compatibility |
| 22:44.47 | dlynes_ | zeroten: no idea...isn't ajax web stuff? |
| 22:44.54 | Malthus | ajax? |
| 22:44.55 | zeroten | yeah |
| 22:45.03 | dlynes_ | zeroten: what does ajax have to do with asterisk? |
| 22:45.22 | wunderkin | manager stuff i think |
| 22:45.25 | zeroten | all i know, is that he was doing some gui stuff |
| 22:45.27 | dlynes_ | ah |
| 22:45.34 | zeroten | similar to ajax |
| 22:45.38 | dlynes_ | yeah..i don't know anything about it |
| 22:45.42 | dlynes_ | one second, zeroten |
| 22:45.43 | zeroten | ajam |
| 22:46.09 | zeroten | http://www.asterisk.org/node/73 |
| 22:46.16 | zeroten | http://svn.digium.com/view/asterisk/team/oej/test-this-branch/doc/ajam.txt?rev=17044&view=markup |
| 22:46.21 | dlynes_ | zeroten: Try here: http://svn.digium.com/view/asterisk/team/markster/asterisk-sla/ |
| 22:46.22 | Malthus | lol |
| 22:46.31 | dlynes_ | zeroten: that's Mark Spencer's svn branch |
| 22:46.40 | zeroten | yes |
| 22:47.06 | dlynes_ | or is it spenser? |
| 22:47.09 | dlynes_ | I can't remember |
| 22:47.20 | Malthus | how did that call limit question get in that thread? |
| 22:47.35 | dlynes_ | Malthus: huh? |
| 22:47.43 | Malthus | http://www.asterisk.org/node/73 |
| 22:48.12 | dlynes_ | oh |
| 22:48.16 | dlynes_ | no idea...that's hilarious |
| 22:49.12 | dlynes_ | anyways |
| 22:49.16 | dlynes_ | if it's in trunk |
| 22:49.28 | dlynes_ | It's almost a sure thing that it will be in 1.4 |
| 22:49.32 | dlynes_ | it's not guaranteed |
| 22:49.34 | dlynes_ | but it's pretty sure |
| 22:49.54 | zeroten | someone get some screenshots :) |
| 22:50.11 | dlynes_ | russellb, afaik is the one that makes the decision as to whether something makes it to release or not |
| 22:50.38 | dlynes_ | And oej afaik, is the fellow that decides whether something makes it into trunk or not |
| 22:50.40 | SplasPood | dlynes: wait.. 1.4 is actually gonna have proper SLA? |
| 22:51.01 | dlynes_ | SplasPood: Did I say proper? No, I don't think I did....I just said it was going to have sla :) |
| 22:51.21 | dlynes_ | SplasPood: 1.2.7.1 has bla...is it proper bla? far from it :) |
| 22:51.36 | SplasPood | dlynes: heh.. well ok |
| 22:52.51 | *** join/#asterisk ozverenm (n=ozverenm@73.27.103-84.rev.gaoland.net) |
| 22:52.57 | ozverenm | hello all |
| 22:53.26 | ozverenm | have some questions |
| 22:53.38 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:54.26 | dlynes_ | The answer is no |
| 22:54.48 | CunningPike | ozverenm: Ignore dlynes_ ;) |
| 22:55.36 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:55.45 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.132.Dial1.SanJose1.Level3.net) |
| 22:55.48 | CunningPike | ozverenm: Ask, already! |
| 22:55.59 | dlynes_ | Maybe he doesn't want to :) |
| 22:57.15 | dlynes_ | Malthus: btw, subscription support is better supported on 1.2.7.1 |
| 22:57.33 | *** join/#asterisk juice (n=juice@mo-67-77-176-48.dyn.sprint-hsd.net) |
| 22:58.32 | Malthus | well, it didn't fix the missing digits on the e&m link |
| 22:58.51 | ozverenm | I am testing a junghanns.net card with test equipment |
| 22:59.03 | Malthus | missing DNID that is |
| 22:59.18 | ozverenm | Its a double E1 card |
| 23:00.06 | ozverenm | but I can constat loss of Q931 packets |
| 23:00.32 | ozverenm | and Q921 packets |
| 23:01.01 | ozverenm | Does anyone knows about intensive testing of E1 cards ? |
| 23:01.56 | cytrak | this sucks my call are been dropped like crazy |
| 23:02.13 | cytrak | every 3 minutes it drops a call |
| 23:03.08 | *** join/#asterisk NeonLevel (i=HydraIRC@200.52.142.184) |
| 23:04.28 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 23:05.12 | *** join/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 23:06.35 | *** join/#asterisk Dr-Linux (n=huh@202.59.73.131) |
| 23:07.33 | ManxPower | cytrak, Diagram your setup |
| 23:08.26 | cytrak | you mean visio style ? |
| 23:08.38 | cytrak | or just explain in words ? |
| 23:08.57 | dlynes_ | cytrak: like sip phone->asterisk->internet->asterisk->pri or whatever |
| 23:09.03 | cytrak | ok |
| 23:09.29 | *** join/#asterisk Smi|k (n=smilk@netblock-72-25-103-165.dslextreme.com) |
| 23:09.31 | dlynes_ | cytrak: where it says internet, replace that with sip/iax/h323, or whatever it is you're using |
| 23:09.41 | Smi|k | anyone here worked on heavy integration of ecommerce and voip? |
| 23:09.53 | dlynes_ | cytrak: and try to extrapolate on what codecs you're using |
| 23:09.58 | ManxPower | You can't make money with VoIP! |
| 23:10.01 | dlynes_ | cytrak: and your bandwidth |
| 23:11.07 | dlynes_ | cytrak: how many simultaneous calls you have on your server would be helpful, too |
| 23:11.18 | dlynes_ | cytrak: and what cpu(s) you're using |
| 23:11.28 | cytrak | first: pstn->span1->asterisk->iax2->idefisk |
| 23:11.28 | cytrak | second: pstn->span1->asterisk->span2->SiemensPBX->HicomPhones |
| 23:11.42 | cytrak | bandwidth is a T1 |
| 23:11.42 | Dr-Linux | hi all |
| 23:11.53 | cytrak | the machine is just a simple P4 |
| 23:12.29 | cytrak | P4 2.80Ghz mem 1.5GB |
| 23:12.30 | dlynes_ | cytrak: span 1 is an E1? |
| 23:12.42 | dlynes_ | cytrak: or pri? |
| 23:12.48 | cytrak | pri |
| 23:12.53 | ManxPower | cytrak, and you do NOT have busydetect=yes or callprogress=yes in zapata.conf? |
| 23:13.06 | cytrak | they are all pris |
| 23:13.20 | cytrak | span1, span2 and span3 |
| 23:13.28 | cytrak | actually I think i do |
| 23:13.32 | cytrak | let me check |
| 23:13.38 | ManxPower | DON'T DO THAT! |
| 23:13.51 | ManxPower | both options should be renamed randomlydisconnectmycalls=yes|no |
| 23:14.08 | *** join/#asterisk Percz (n=Miranda@megazirt.gotadsl.co.uk) |
| 23:14.10 | cytrak | hehe |
| 23:14.17 | cytrak | I do have callprogress=yes |
| 23:14.27 | cytrak | I should set that to no |
| 23:14.34 | ManxPower | that would cause randomly disconnected calls |
| 23:14.48 | ManxPower | correct, do not set it at all or set it to no |
| 23:15.08 | Dr-Linux | woww MultiTech voip gateway rocks with asterisk :) |
| 23:15.42 | cytrak | I've been working on the box since mornning so my users were complaining about disconnects , some of them I caused by me but others I didn't cause it. |
| 23:16.21 | cytrak | I remember you told me to set resetinterval to something greater than 3600 and that helped. the span are no longer restarted like crazy |
| 23:16.59 | cytrak | that alos caused the calls to be dropped, but for example I've been on hold with digium now for 10-15min using my idefisk and no drops |
| 23:17.15 | dlynes_ | Dr-Linux: Are you actually using them, or just checking out their web page? |
| 23:18.28 | Dr-Linux | dlynes_: today i configured my asterisk with my 2 MultiTech gateways |
| 23:18.28 | dlynes_ | ah |
| 23:18.29 | dlynes_ | Dr-Linux: what're their prices like? |
| 23:18.39 | cytrak | cool digium picked up now |
| 23:18.44 | cytrak | brb |
| 23:19.07 | dlynes_ | Dr-Linux: nvm....bloody expensive |
| 23:19.12 | ozverenm | sangoma cards are goods or not ? |
| 23:19.16 | ozverenm | reputation ? |
| 23:19.18 | dlynes_ | Dr-Linux: i don't know how the hell you guys could afford them :) |
| 23:19.25 | dlynes_ | ozverenm: reputation's good, yeah |
| 23:19.30 | Malthus | how many ports on those gateways? |
| 23:19.46 | ozverenm | DIGIUM ? |
| 23:19.50 | dlynes_ | Malthus: http://www.multitech.com/PRODUCTS/Families/MultiVOIP/ |
| 23:19.57 | dlynes_ | ozverenm: also good reputation |
| 23:19.58 | Dr-Linux | dlynes_: we have 4 MultiTech gateway 810 model |
| 23:20.05 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 23:20.09 | ozverenm | eicon ? |
| 23:20.15 | *** join/#asterisk achandra (n=achandra@12.44.122.130) |
| 23:20.24 | Dr-Linux | dlynes_: bcoz we unaware before if there is something named Asterisk :D |
| 23:20.25 | Malthus | I just got a bunch of audiocodes gateways |
| 23:20.51 | achandra | hello...I have some questions about res_snmp.conf in asterisk...is that file compiled by default? I cant seem to find it. |
| 23:21.24 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
| 23:21.37 | dlynes_ | ozverenm: no idea |
| 23:21.54 | dlynes_ | achandra: i think it's an addon |
| 23:22.05 | dlynes_ | achandra: I'm not sure where you get it from |
| 23:22.23 | ozverenm | did anyone have tried to make a ISDN bridge ? |
| 23:22.46 | achandra | hmm...been searching for it using standard find by name command in linux...but no beuno...it has some cool features to port into cacti. |
| 23:24.19 | Malthus | any telecom lawyers in the house? |
| 23:24.49 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:24.55 | dlynes_ | ozverenm: No, but I'll sell you a Golden Gate Bridge |
| 23:26.46 | dlynes_ | achandra: looks like it's only in trunk |
| 23:26.57 | dlynes_ | achandra: and if not trunk, then oej's test-this-branch |
| 23:27.20 | ManxPower | achandra, did you check in /path/to/source/asterisk/configs |
| 23:28.55 | Malthus | is there a way to check asterisk's uptime? |
| 23:29.10 | Netgeeks | show uptime |
| 23:29.17 | X-Rob | wpm4l-gw*CLI> show uptime |
| 23:29.17 | X-Rob | System uptime: 1 week, 2 days, 1 hour, 14 minutes, 25 seconds |
| 23:29.17 | X-Rob | Last reload: 19 hours, 10 minutes, 12 seconds |
| 23:29.17 | X-Rob | wpm4l-gw*CLI> |
| 23:31.18 | Malthus | thanks |
| 23:31.26 | ManxPower | pbx-1*CLI> show uptime |
| 23:31.26 | ManxPower | System uptime: 6 weeks, 15 hours, 44 minutes, 9 seconds |
| 23:31.35 | Malthus | hah |
| 23:31.41 | Malthus | System uptime: 7 minutes, 56 seconds |
| 23:31.42 | Malthus | Last reload: 4 seconds |
| 23:33.23 | cytrak | ManxPower: as soon as I'm done here I'll comment it out |
| 23:33.35 | achandra | ManxPower: I will check that..but say I can find the file (ie the sample one) in source. Can I simply drop it into the /etc/asterisk file and give it a go...or? |
| 23:33.43 | cytrak | ManxPower: my call through IAX2 to PRI hasn't dropped though |
| 23:34.03 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 23:35.46 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 23:35.47 | a1fa | hey |
| 23:35.53 | achandra | ManxPower: The config doesnt show it...but the google has "source site" on it...can I simply use that one? |
| 23:35.54 | a1fa | anoybody using mythtv? |
| 23:36.00 | a1fa | anoybody using mythtv+mythphone ? |
| 23:36.30 | GreyFoxx | myth yes, mythphone no |
| 23:36.36 | a1fa | GreyFoxx : lol |
| 23:36.47 | a1fa | get out-a-here ;P |
| 23:37.03 | a1fa | i dont get it |
| 23:37.23 | achandra | alfa: not sure how youd seperate the mic from earphone/speaker part without getting horrible echo.. |
| 23:38.06 | achandra | ive seen that module..it works like any softphone right..so if its sitting in front of your tv...are you going to have this long ass cable with a earphone bud on it? |
| 23:39.24 | The_Isle_of_Mark | lo all |
| 23:40.36 | The_Isle_of_Mark | anyone knwo where the distinctive ring configuration howto is? |
| 23:41.16 | a1fa | achandra : no, i am not using it to make phone calls |
| 23:41.22 | a1fa | achandra : i am using it to see caller ID |
| 23:41.31 | a1fa | achandra : so i dont have to get up off my couch |
| 23:42.31 | a1fa | it works for the first couple of seconds, then it dies off |
| 23:43.12 | De_Mon | if I include => local_numbers in [local] can I Goto(local,<a local_numbers_patern>) |
| 23:47.22 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |