| 00:00.00 | [TK]D-Fender | phpboy: Tissue box is on the left |
| 00:00.32 | phpboy | [TK]D-Fender: SILENCE! |
| 00:00.32 | phpboy | :P |
| 00:00.59 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 00:01.13 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 00:01.23 | *** mode/#asterisk [-v phpboy] by [TK]D-Fender |
| 00:01.38 | [TK]D-Fender | phpboy: Granted! |
| 00:01.40 | phpboy | no nice :( |
| 00:02.10 | *** mode/#asterisk [+v phpboy] by [TK]D-Fender |
| 00:02.12 | phpboy | [TK]D-Fender: type the following command "/mode #asterisk -v [TK]D-Fender" |
| 00:02.15 | phpboy | :D |
| 00:03.00 | phpboy | peace be with you my brothers |
| 00:03.04 | phpboy | I'm going home now :( |
| 00:03.23 | phpboy | Thanks to the stupid Telco I'm dealing with :( |
| 00:04.29 | phpboy | [TK]D-Fender: thanks :D |
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| 01:04.41 | satish2437 | Hi all, I need help in choosing a proper channel type to configure on zaptel.conf for ss7 network. |
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| 01:38.06 | *** join/#asterisk edibrac (n=elusive4@64.241.37.140) |
| 01:39.34 | edibrac | do i have this right -- if I go with a hosted asterisk "solution" then I'm exposing myself to things like latency problems between my office and their servers? Running your own T1 to your office bypasses this problem? |
| 01:39.58 | edibrac | but the latency problem can be worked around technically w/ QOS? |
| 01:40.27 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 01:45.14 | _ShrikE | edibrac: latency is less of an issue than things like jitter and packet loss. |
| 01:45.24 | mmattice | if you have QoS all the way between you and your "hosted" box |
| 01:49.11 | _ShrikE | edibrac: and so long as ANY internet is involved, you can throw QOS out the window. |
| 01:49.43 | *** part/#asterisk Daejeo (n=chatzill@118.221.248.29) |
| 01:54.51 | edibrac | is there a common term for the "hosted setup" versus "local loop setup" ? |
| 01:55.05 | edibrac | i mean, "and" not versus |
| 01:57.03 | *** join/#asterisk Segnale007 (n=Pietro@host128-254-dynamic.35-79-r.retail.telecomitalia.it) |
| 02:04.13 | edibrac | if you use something like Aretta (www.aretta.com) -- their data center is in Atlanta, GA. If I order a PRI for area code 510 in California, how do they configure their asterisk box in Georgia for that a block of 510 DIDs? |
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| 02:05.46 | edibrac | does my LEC have to talk to the LEC in Atlanta to make this happen? |
| 02:08.12 | *** join/#asterisk Hanif08 (n=bucoo77@netop.jaring.my) |
| 02:12.33 | _ShrikE | edibrac: They get the numbers through wholesale interconnects with nationwide carriers like verizon, level 3, etc.. You connect to them over the public Internet. |
| 02:13.44 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 02:14.19 | jaytee | does MagicJack suffer less from jitter since it uses "magic" packets? :-) |
| 02:14.37 | zr0 | haha |
| 02:15.11 | edibrac | say i already have a block of DIDs -- can I keep those and "duplicate" them so that Aretta (or a similar hosting company) also uses them? |
| 02:15.58 | jaytee | you can probably transfer them to Aretta but you can't use two providers for the same DID |
| 02:20.48 | edibrac | if it's deemed that they suck and we want to move it somewhere that sucks less -- is that something providers can do, say over the weekend? |
| 02:22.17 | _ShrikE | edibrac: The port process can take several days even with an expedite |
| 02:22.29 | jaytee | if you find a provider that "doesn't suck" then you can have them do the DID number transfer to their system but it needs to be scheduled well in advance |
| 02:23.16 | _ShrikE | edibrac: I would suggest you work with a provider who is willing to offer a trial before you go off porting numbers. |
| 02:23.17 | edibrac | is it something where both providers have to agree upon? like, one provider cannot just take over at will, without consent? |
| 02:23.22 | `Sean | Does anyone know where i can find some consultants to help me with my dilemna?? with the cisco 7970?? |
| 02:25.47 | _ShrikE | edibrac: You as the customer file an LOA to move the numbers between carriers. |
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| 02:33.42 | *** join/#asterisk ex-parrot (n=michael@116.90.128.177) |
| 02:33.59 | ex-parrot | hey all.... can someone suggest why astrundir is being ignored on my 1.4.21.2 install? |
| 02:34.08 | ex-parrot | it still tries to create things directly in /var/run |
| 02:36.19 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.29) |
| 02:42.32 | Daejeo | how to steal switchvox gui? |
| 02:42.39 | Daejeo | i like it :) |
| 02:46.18 | ex-parrot | n/m I worked it out |
| 02:46.25 | ex-parrot | need to put astrundir in the [directories] section |
| 02:46.26 | *** part/#asterisk ex-parrot (n=michael@116.90.128.177) |
| 02:52.16 | Daejeo | do you have a pro edition ? |
| 02:52.27 | Daejeo | standard edition ? |
| 02:54.42 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.29) |
| 02:54.59 | Daejeo | did i miss anything ? |
| 02:57.38 | eppigy | DI DI MAO |
| 03:02.23 | cvnet | If interconnecting using H.323 are you able to support H.245 Alpha Numeric on DTMF? <-- does asterisk supports this? |
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| 03:08.56 | [TK]D-Fender | cvnet: JFGI, I already found an answer. |
| 03:09.36 | cvnet | [TK]D-Fender im lost |
| 03:10.18 | [TK]D-Fender | cvnet: http://www.google.ca/search?hl=en&q=asterisk+H.245+dtmf&start=10&sa=N |
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| 03:26.24 | Hanif08 | possible to integrate asterisk with realtime billing? or someone has done it b4.. |
| 03:29.07 | cvnet | Hanif08: asterisk2billing.org |
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| 03:33.11 | troubled | Any ideas about 1.6.0.2 build choking on "manager.c -> manager.o"? âSENTINELâ undeclared |
| 03:34.01 | troubled | seems im missing iLbc header as well, but I skipped that for now. but still cant seem to get past the manager file, even after disabling the manager cdr option in make menuselect |
| 03:34.34 | Juggie | troubled, manager cdr is cdr_manager not manager.c |
| 03:34.44 | troubled | Juggie: ya, kinda figured |
| 03:34.59 | Juggie | could you pastebin the full error |
| 03:35.06 | troubled | sure, one sec |
| 03:35.55 | troubled | hmm, got a better pastebin? pastebin.ca is taking it time for me |
| 03:36.13 | troubled | !paste |
| 03:36.46 | troubled | Juggie: http://debian.pastebin.com/d5d90a631 |
| 03:37.28 | troubled | nice option with the menuselect btw |
| 03:38.44 | troubled | anyways. so far I have dahdi and tools installed and running okay. seems like it picked up my tdm400p np, and the genconf option is great, so im hoping that the problem isn't related to that end of the install. |
| 03:41.05 | Juggie | troubled, is that 1.6.0.2 from a tar.gz? |
| 03:41.50 | Juggie | or is it a debian version, etc |
| 03:44.56 | troubled | Juggie: svn http://svn.digium.com/svn/asterisk/tags/1.6.0.2 |
| 03:45.39 | troubled | i could switch that to trunk if you like though. just not sure what ver and state trunk might be at |
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| 04:17.06 | VoipForces | Hi, anyone knows why a SIP carrier (namely saba voip) would take up to 17 seconds to send a BYE when a hangup occurs ? |
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| 04:29.05 | Hanif08 | thank cvnet |
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| 04:49.59 | troubled | Juggie: "Asterisk has successfully been built"! :) |
| 04:50.20 | troubled | Juggie: seems that 1.6.0.3-rc1 works fine. thanks for replying at least though. cheers |
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| 05:32.05 | Daejeo | se |
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| 06:11.22 | Daejeo | hello guys, how can I access conf files on switchvox |
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| 07:47.49 | BBHoss | does anyone know if asterisk queues asynchrouse originate manager commands if the pbx has a maximum call number set? |
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| 07:50.39 | BBHoss | or rather, does asterisk queue originates, or do i need to handle that myself in my own app |
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| 09:01.54 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 09:02.15 | hi365 | anyone know the makeup of the command that you can call after you call the a2billing agi? |
| 09:02.41 | hi365 | i.e. AGI(a2billing.php|bla|bla|bla) |
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| 10:43.51 | BBHoss | anyone ever seen this: Channel allocation failed: Can't create alert pipe! |
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| 10:59.11 | sosoriri | anyone seen the sound delay problem? |
| 11:00.02 | sosoriri | for example, when user enter the conference, he will listen the sound "Please enter your conference number followed by the pound key.", but "your conference number followed by the pound key.". |
| 11:00.51 | sosoriri | i heard it from my friend, but we can't catch it again. |
| 11:01.04 | *** join/#asterisk unasi7 (n=unasi7@169-94.104-92.cust.bluewin.ch) |
| 11:01.04 | sosoriri | anyone solve it? |
| 11:02.33 | tzafrir_laptop | from which type of phone did he call into the conference? |
| 11:02.37 | *** join/#asterisk Maliuta_CA (n=biteme@206.47.36.150) |
| 11:07.26 | mikkel | Anyone know of the problem with SIP phones that come in a state, where the caller can hear but the SIP phone can't hear ? |
| 11:08.01 | mikkel | I'm using sangome BRI ISDN card and asterisk 1.4.21 |
| 11:09.15 | sosoriri | what's the problem? or how can i catch it again? |
| 11:09.52 | sosoriri | from pstn and IP phone. |
| 11:09.58 | sosoriri | manybody told me something like this, but i can't catch it again. |
| 11:11.28 | sosoriri | it's not first problem. when i heard this, i have no favor about it. |
| 11:11.41 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:11.51 | sosoriri | but now i can't pass this problem. |
| 11:12.59 | *** join/#asterisk qdk (n=qdk@79.138.245.169.bredband.3.dk) |
| 11:15.30 | angryuser | sosoriri: try to make a sip trace |
| 11:16.16 | sosoriri | ?? |
| 11:16.18 | angryuser | sosoriri: look if codecc negotiation goes well, do you have this issue with only one type of phones ? can you test it with softphone ? |
| 11:17.01 | angryuser | sosoriri: oh all this goes to mikkel :) |
| 11:17.09 | sosoriri | i tested it many times, but i can't catch it again. |
| 11:17.28 | angryuser | sosoriri: so you dont have it now ? |
| 11:18.16 | mikkel | angryuser: How do I make a sip trace, you mean debug ? |
| 11:18.21 | angryuser | mikkel: is there any nat involved ? |
| 11:18.36 | mikkel | angryuser: no |
| 11:18.43 | *** join/#asterisk Cy (n=Cy@45.Red-81-35-252.dynamicIP.rima-tde.net) |
| 11:19.20 | angryuser | mikkel: do you have any firewalls involved ? are your phones has the ports set withing asterisk port range ? |
| 11:19.25 | sosoriri | now, i can't catch this case again? |
| 11:19.47 | sosoriri | if i know why it occurs, i will dig it. |
| 11:19.51 | sosoriri | but now,.... |
| 11:19.53 | mikkel | angryuser: It happens like 1 time a month or so, and a combination of restarting wanrouter/asterisk and the linksys SIP box makes it work again. |
| 11:20.26 | sosoriri | mikkel: i think so. "It happens like 1 time a month or so" |
| 11:21.50 | sosoriri | i think it's problem of codec, server load. |
| 11:22.01 | sosoriri | sorry. i think it's not problem of codec, server load. |
| 11:22.18 | mikkel | angryuser: No firewall, all is on the same LAN as the asterisk server. How is the port range defined ? |
| 11:22.57 | angryuser | mikkel: by default it is 10 000 - 20 000 udp |
| 11:23.33 | mikkel | angryuser: I have no restriction on ports on the LAN. |
| 11:24.09 | sosoriri | angryuser: it's not problem about router/firewall.. |
| 11:24.21 | sosoriri | i believe so. |
| 11:25.13 | mikkel | angryuser: Here is some debug http://pastebin.com/m62aa7986 |
| 11:26.20 | angryuser | mikkel: sip:1000@192.168.30.242:5060 it the called person ? |
| 11:26.54 | mikkel | angryuser: That is the Linksys SIP box, yes the called person |
| 11:27.52 | angryuser | mikkel: well * is trying to send CANCEL and the sip box is not responding, so it is destroing the dialog, i dont see a problem with one way audio here |
| 11:28.33 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 11:29.17 | mikkel | angryuser: http://pastebin.com/d65ffac9f |
| 11:30.05 | angryuser | mikkel: need moree after that |
| 11:31.02 | mikkel | angryuser: What do you mean ? |
| 11:31.49 | angryuser | mikkel: and dont see any issues, normal invite, you need to capture some more sip dialog with the audio issue |
| 11:32.41 | *** join/#asterisk The_Lightside (n=Lightsid@dsl-241-86-116.telkomadsl.co.za) |
| 11:35.00 | echinos | When I use softphones, I get call quality issues |
| 11:35.17 | angryuser | mikkel: try to disable all codec's but one, and disable canreinvite if you have it for that phone |
| 11:35.21 | echinos | When I call in from POTS etc. and relay back out, sound seems a-ok |
| 11:35.25 | echinos | Normal? |
| 11:40.44 | *** join/#asterisk mort_gib (n=mjensen@164.236.120.212.dsl.dynamic.gibconnect.com) |
| 11:42.34 | mikkel | angryuser: How do I do that, don't have any codes specified, just using default. don't even have a codecs.conf file. |
| 11:46.30 | kaldemar | mikkel: it's not done in codecs.conf, but in sip.conf for SIP. look into options disallow and allow. |
| 11:47.52 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 11:49.55 | gambler1 | I have a strange problem with * server, we have been able to see a call that has not been answered (dest phone rings) but * cdr shows status as answered, we have been using 1.6.0.1, does anyone have similar problem? |
| 11:50.41 | *** join/#asterisk DelphiWorld (n=Miranda@41.221.26.96) |
| 11:50.55 | DelphiWorld | hi my friends |
| 11:51.05 | DelphiWorld | please is it pocible to store my extentions in a rdbms ? |
| 11:51.06 | angryuser | mikkel: also in sip linksys box |
| 11:51.28 | *** join/#asterisk carranca (n=carranca@pampero.itba.edu.ar) |
| 11:51.29 | kaldemar | gambler1: do you have an answer or an application that issues an answer in your dialplan? |
| 11:52.20 | carranca | how can i compile asterisk with a verbose output? i have a segfault with an application im adding and i want to know how its being compiled |
| 11:52.24 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 11:54.10 | gambler1 | kaldemar: no I do not have an answer, just remote side rings, but in * cdr it says it has been answered |
| 11:54.51 | gambler1 | kaldemar: I use sip upstream provider and it seems that we have a problem only with one country |
| 11:56.12 | mikkel | angryuser: Here is a debug from a call where I can hear her (her is the SIP phone), but she can't hear me: http://pastebin.com/d5a783d86. And only g726 codec is allowed |
| 11:58.32 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-4a34369d70a3e617) |
| 12:05.09 | yang | Aynone using the Snom phones ? |
| 12:05.44 | yang | I have a busy light field issue |
| 12:06.19 | *** join/#asterisk hakr (i=hakr@pdpc/supporter/active/hakr) |
| 12:08.02 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 12:10.31 | mikkel | angryuser: Have to go to the location, will log on later. Thanks for all the help so far. |
| 12:10.36 | *** part/#asterisk Cy (n=Cy@45.Red-81-35-252.dynamicIP.rima-tde.net) |
| 12:13.48 | *** join/#asterisk jonavogt (n=jonavogt@u51-229.dsl.vianetworks.de) |
| 12:15.11 | jonavogt | Hi everybody |
| 12:18.45 | jonavogt | I got a problem I'm not able to resolve. All the sudden asterisk thinks it doesn't have any free misdn channels. Restart of asterisk didn't help, neither did reboot. Hardware seems to work fine since I can see the call infos... I'm sorta lost here. What else can I check? |
| 12:20.07 | *** join/#asterisk wonderworld (n=w@p57BDB74E.dip.t-dialin.net) |
| 12:22.14 | wonderworld | hi, i am trying to change the outgoing callerid on our asterisk for some sip extensions. clip no screening has been enabled by our telco, but it doesn't seem to work. when i change the callerid in sip.conf to the number, that should be presented to the called person, the called person sees the main number of our pri, not the number i set. |
| 12:22.47 | wonderworld | the way out is SIP -> asterisk -> zap -> E1 -> PSTN |
| 12:23.47 | jonavogt | wonderworld: Are you allowed to set this specific number? I once had the same when using a number which wasn't assoiciated with that line. |
| 12:25.29 | *** join/#asterisk DelphiWorld (n=Miranda@41.221.26.210) |
| 12:25.31 | wonderworld | yes, thats why i had them enable "clip no screening" which actually allows to send any number |
| 12:25.45 | DelphiWorld | hi my friends |
| 12:25.56 | *** join/#asterisk path_ (n=path@15-106-21-190.adsl.terra.cl) |
| 12:25.59 | DelphiWorld | please ho to store my asterisk data into a RDBMS ? |
| 12:26.16 | kaldemar | wonderworld: some networks also require things like including your area code (possibly without zeroes in front) in the CLID. |
| 12:26.18 | jonavogt | wonderworld: wasn't sure if that allowed any number.. |
| 12:26.24 | wonderworld | i think i set the outgoing number in the wrong config maybe? could it be that i would have to alter my zapata.conf? or set the number with "Set" in the dialplan? |
| 12:27.20 | wonderworld | kaldemar: thanks i'll try that |
| 12:27.25 | *** join/#asterisk jaime_B (n=jaime_B@p548F4744.dip.t-dialin.net) |
| 12:29.41 | jonavogt | what should "misdn show channels" output on a working misdn system? |
| 12:30.56 | jaime_B | good day everybody. I want to connect two asterisk boxes with iax protocol via a vpn that is only up on demand. does asterisk handle this non-permanent link by itself? |
| 12:31.10 | wonderworld | ok, didn't work either. btw. what would be the correct place to set that number? sip.conf, dialplan or maybe even zapata.conf? |
| 12:31.37 | jonavogt | wonderworld: I set my numbers in the Dialplan and that works |
| 12:31.56 | jonavogt | wonderworld: which doesn't mean the other two are not legid. |
| 12:32.40 | *** join/#asterisk mort_gib (n=mjensen@164.236.120.212.dsl.dynamic.gibconnect.com) |
| 12:32.48 | kaldemar | wonderworld: sip.conf or dialplan. i used to use shorter numbers in sip.conf and prefix them in dialplan before sending to PSTN. zapata.conf settings would only apply to incoming calls. |
| 12:34.37 | mort_gib | Hi Yang, happy new year |
| 12:35.50 | kaldemar | wonderworld: except the pridialplan in zapata.conf might be something to change. |
| 12:36.44 | wonderworld | its "unknown" atm |
| 12:39.27 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:51.18 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:56.31 | jonavogt | <PROTECTED> |
| 12:58.35 | *** join/#asterisk path_ (n=path@15-106-21-190.adsl.terra.cl) |
| 13:00.45 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:01.04 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
| 13:05.35 | *** join/#asterisk telnettech (n=telnette@199-2-117-121.setardsl.aw) |
| 13:08.19 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:08.32 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 13:08.42 | aiksa[LV] | hi everyone. |
| 13:09.14 | phpboy | howdy |
| 13:09.17 | aiksa[LV] | any idea why i have correct BLF status indication for incomming calls to SIP extension, but not for Outgoing? |
| 13:09.33 | aiksa[LV] | type - friend |
| 13:09.55 | aiksa[LV] | limitonpeers and call-limit set accordingly to voip info |
| 13:10.20 | aiksa[LV] | sip show subscriptions keep the idele status upon active outgoing call from this extension |
| 13:11.39 | aiksa[LV] | Asterisk 1.4.22 with bristuff patch from xorcom |
| 13:12.09 | *** part/#asterisk jaime_B (n=jaime_B@p548F4744.dip.t-dialin.net) |
| 13:13.20 | BBHoss | can anyone tell me if I can execute system commands from agi? |
| 13:13.20 | *** join/#asterisk mort_gib (n=mjensen@164.236.120.212.dsl.dynamic.gibconnect.com) |
| 13:14.38 | aiksa[LV] | BBHoss: OS system commands? |
| 13:14.41 | BBHoss | yes |
| 13:14.49 | BBHoss | i recieve this: EXEC system /usr/bin/touch /var/alertnet/recordings/test.touched |
| 13:14.57 | BBHoss | in the agi debug, but it doesnt do it |
| 13:15.10 | BBHoss | and of course returns 0 |
| 13:15.33 | aiksa[LV] | hmm, you send EXEC back to * through AGI? |
| 13:15.40 | BBHoss | yeah |
| 13:15.48 | BBHoss | but it wont execute system() |
| 13:16.30 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:17.02 | aiksa[LV] | you could execute any system command from the AGI script itself |
| 13:17.14 | aiksa[LV] | without turning back to asterisk |
| 13:17.22 | BBHoss | yes but it limits the input to 0 charachters |
| 13:17.25 | BBHoss | 60 |
| 13:17.34 | BBHoss | anything after that it ignores |
| 13:18.16 | aiksa[LV] | no no |
| 13:18.47 | aiksa[LV] | you make a script (php, bash, python c++, whatever) which you will call from AGI |
| 13:19.12 | aiksa[LV] | and in this script, using the standart functionality of the language it is written in, you can execute the local OS commands |
| 13:20.06 | *** join/#asterisk path_ (n=path@15-106-21-190.adsl.terra.cl) |
| 13:20.51 | aiksa[LV] | howdy tzafrir_laptop |
| 13:21.12 | tzafrir_laptop | aiksa[LV], hi |
| 13:21.41 | tzafrir_laptop | what's up? |
| 13:21.53 | BBHoss | well in this case in the agi script i'd have to hand it the id of the db object so it can go over HTTP and fetch the xml for the object and then parse it out and generate the audio via swift |
| 13:22.13 | BBHoss | because its on a different server |
| 13:22.41 | BBHoss | i'd rather just hand it the text directly instead of making ANOTHER unneeded call back the the webserver |
| 13:23.28 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:23.45 | aiksa[LV] | tzafrir_laptop: could it be that bristuff patch to asterisk 1.4 which comes with astribank doesnt have limitonpeers functionality from asterisk 1.4? |
| 13:24.16 | jonavogt | nobody got a idea about blocked misdn channels and where to continue? |
| 13:24.22 | BBHoss | but still, why won't asterisk EXEC System, did someone disable it for security reasons or something? |
| 13:24.37 | BBHoss | i just tried EXEC playback and it works fine, but it won't EXEC system |
| 13:24.39 | aiksa[LV] | BBHoss: dont know, have never done it this way |
| 13:24.50 | BBHoss | aiksa[LV]: thanks anyways |
| 13:25.14 | aiksa[LV] | BBHoss: i usually stick to the standart method of the language |
| 13:25.18 | tzafrir_laptop | aiksa[LV], when was that added? |
| 13:25.22 | aiksa[LV] | gives far more controll over the process |
| 13:25.37 | tzafrir_laptop | Though I don't remember anything explicitly about it |
| 13:25.40 | BBHoss | aiksa[LV]: yes but surely you've had situations where you can't do things the "standard" way :) |
| 13:25.48 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 13:26.13 | tzafrir_laptop | Right now the patch we include is rather minimal and doesn't touch chan_sip.c |
| 13:27.24 | aiksa[LV] | tzafrir_laptop: not sure about the exact minor version of 1.5 |
| 13:27.30 | aiksa[LV] | of 1.4 |
| 13:28.25 | aiksa[LV] | tzafrir_laptop: why I was asking this was because of hint status |
| 13:30.05 | aiksa[LV] | although the limitonpeers (joins incomming and outgoing call limits for a SIP friend) is set to yes for a specific peer its status is still shown as idle when it perfoms an outbound call |
| 13:31.11 | tzafrir_laptop | any idea if this is also the case with plain 1.4.22? |
| 13:31.13 | aiksa[LV] | and that option doesnt shop up on "sip show peer XX" ; though i am not sure if it should on non patched asterisk |
| 13:32.14 | aiksa[LV] | tzafrir_laptop: I have no option of testing this at the present moment. I could perform a couple of tests later today when I get home |
| 13:33.05 | aiksa[LV] | BBHoss: of course, but not on this track |
| 13:35.55 | aiksa[LV] | tzafrir_laptop: on the other hand we could ask anyone here (with 1.4.22) to perform a "sip show peer" |
| 13:36.26 | aiksa[LV] | so anyone here on asterisk 1.4.22 w/o bristuff patch applied? |
| 13:36.37 | wonderworld | how would i make asterisk play the "original" busy signal from the telco when calling out, instead of just signaling "BUSY" or "CHANUNAVAIL" etc? |
| 13:37.07 | aiksa[LV] | wonderworld: there is a dialplan command named Busy |
| 13:37.13 | aiksa[LV] | you could take look at that |
| 13:38.33 | tzafrir_laptop | aiksa[LV], how exactly does it show up? |
| 13:39.07 | wonderworld | aiksa[LV]: nope, i want to hear the original sounds when calling out to somewhere. my asterisk just hangs up when someone is not reachable and gives me the errorcode. |
| 13:39.37 | wonderworld | like i call a not connected number, i want to hear the anouncement from the telco |
| 13:39.53 | wonderworld | instead of just getting the status code and a hangup |
| 13:41.27 | telnettech | qustion......the libpri and zaptel.....do they need to be the same version as the asterisk software |
| 13:41.57 | telnettech | <PROTECTED> |
| 13:42.25 | telnettech | or should load the latest libpri and zaptel even i i am using asterisk 1.2.28 |
| 13:43.29 | *** part/#asterisk VoipForces (n=courchea@mail.net-forces.com) |
| 13:45.16 | *** join/#asterisk DarkRift (n=dark@65.92.249.26) |
| 13:45.54 | aiksa[LV] | wonderworld: what do you mean by the announcement. a person telling you number is busy? |
| 13:46.23 | aiksa[LV] | tzafrir_laptop: there aint no limitonpeers setting switch when i execute "sip show peer XX" |
| 13:46.35 | aiksa[LV] | i suspect it might me present on unpatched version |
| 13:47.02 | jonavogt | I get a "NO FREE CHAN IN STACK" without having any misdn calls at all... I have no clue what is going on... anybody got an idea?!? |
| 13:47.12 | aiksa[LV] | someone with non patches 1.4.22 could just launch this command on any of his sip friends who has limitonpeers set to yes |
| 13:48.26 | aiksa[LV] | jonavogt: what does misdn show stacks says? |
| 13:48.40 | tzafrir_laptop | aiksa[LV], strings /usr/lib/asterisk/modules/chan_sip.so | grep limitonpeers |
| 13:48.48 | *** join/#asterisk r00ttolo (n=Alex1@85-18-14-41.fastres.net) |
| 13:48.52 | jonavogt | asterisk*CLI> misdn show stacksBEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:4 * Port 2 Type NT Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:4 * Port 3 Type NT Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:4 * Port 4 Type NT Prot. PMP L2Link DOWN L1Link:UP Blocked:0 Debug:4 |
| 13:49.00 | jonavogt | ups |
| 13:49.06 | tzafrir_laptop | aiksa[LV], is it there in the first place? |
| 13:49.12 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-77.static.insightbb.com) |
| 13:49.15 | aiksa[LV] | a sec. |
| 13:49.15 | r00ttolo | hello, someone can help me? i need a step by step guide for asterixnow |
| 13:49.43 | tzafrir_laptop | r00ttolo, step 1: it's called "asterisk". Step 2: try #asterisknow |
| 13:49.59 | r00ttolo | ops sorry |
| 13:50.24 | aiksa[LV] | tzafrir_laptop: yes as "global_limitonpeers" |
| 13:50.26 | r00ttolo | i've already try it but noone answer |
| 13:50.45 | aiksa[LV] | jonavogt: misdn show channels |
| 13:50.55 | r00ttolo | i setup all and update asterisk-gui but now i don't know how to do |
| 13:50.57 | tzafrir_laptop | only that? If so, it means that chan_sip does not parse that keyword? Who does, then? |
| 13:51.04 | jonavogt | aiksa[LV]: Chan List: (nil) |
| 13:54.03 | NovceGuru | asstricks |
| 13:56.31 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 13:56.44 | SuPrSluG | tzafrir_laptop, is ztxen still out there somewhere? |
| 13:57.14 | tzafrir_laptop | isn't there an open bug in Zaptel? |
| 13:57.39 | tzafrir_laptop | Subdolus, I never really followed it |
| 13:57.45 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 13:59.05 | *** join/#asterisk kinnaz (n=nnscript@no.life.ee) |
| 13:59.11 | *** join/#asterisk suvir (n=suvir@ppp-124-120-135-54.revip2.asianet.co.th) |
| 14:00.54 | jonavogt | aiksa[LV]: what should misdn show channels show? |
| 14:01.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:02.05 | aiksa[LV] | jonavogt: sorry - off for a while |
| 14:02.22 | jonavogt | aiksa[LV]: no problem |
| 14:02.34 | aiksa[LV] | tzafrir_laptop: - have no clue |
| 14:02.48 | aiksa[LV] | nevertheless that was against the patched chan_sip |
| 14:03.46 | telnettech | question......the libpri and zaptel.....do they need to be the same version as the asterisk software i am installing? example: libpri 1.2.4, zaptel 1.2.21, asterisk 1.2.28 or should load the latest libpri and zaptel even if i am using asterisk 1.2.28 |
| 14:05.15 | [TK]D-Fender | telnettech: You should be using the lastest one released since your * version, but before the next |
| 14:05.16 | Nugget | telnet is eeeeeeevil! |
| 14:05.58 | [TK]D-Fender | telnettech: And we're on 1.6 now, you definitely should not be on 1.2 |
| 14:06.10 | jonavogt | misdn just stopped working suddently if that helps anyone |
| 14:06.51 | wonderworld | aiksa[LV]: yes, like any anouncment. i want the signaling from the telco, not asterisks. like "number has changed", " number is busy", original busy signal from telco side, etc. |
| 14:06.56 | telnettech | TK thanks.....i have to use 1.2.28 cause that is the only version we have certified to use with our other application for customers....i understand that we will have 1.4 certified by the end of 1st quarter |
| 14:07.22 | telnettech | TK.....where can i find out of the libpri and zaptel is a version compatible with the 1.2.28 version |
| 14:07.36 | [TK]D-Fender | telnettech: the CHANGELOGS |
| 14:07.38 | echinos | When I use softphones, I get call quality issues |
| 14:07.42 | echinos | When I call in from POTS etc. and relay back out, sound seems a-ok |
| 14:07.46 | echinos | Normal? |
| 14:07.53 | echinos | or... common? :/ |
| 14:07.54 | [TK]D-Fender | echoSoftphones & headsets SUCK |
| 14:08.29 | [TK]D-Fender | echinos: Shit sound cards, shit headsets. All sorts of extra interferences, etc |
| 14:08.43 | echinos | k |
| 14:09.04 | telnettech | TK....is that part of the software? |
| 14:09.04 | echinos | time for a desk set then |
| 14:09.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:09.32 | [TK]D-Fender | telnettech: No its a TEXT FILE the they put out with every release |
| 14:09.56 | telnettech | ok i will search for it |
| 14:10.01 | telnettech | thanks TK |
| 14:13.14 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
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| 14:23.10 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 14:23.15 | timeshell | Can someone tell me the current build number of GUI2? |
| 14:24.13 | phpboy | exten => 3000,4,GotoIfTime(08:00-18:00,mon-fri,*,*?officehours-cartrack,${EXTEN},1) <--- that's how my office hours work at the mo |
| 14:24.23 | phpboy | what's the best way to introduce cutting holidays off? |
| 14:27.18 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 14:33.49 | yang | FREE USA DIDs if anyone is interested - http://www.ipcomms.net/html/FREEDID_Landing2.htm |
| 14:37.07 | *** join/#asterisk mort_gib (n=mjensen@164.236.120.212.dsl.dynamic.gibconnect.com) |
| 14:39.43 | yang | hi mort_gib |
| 14:40.08 | mort_gib | Hi Yang, happy new year |
| 14:40.21 | mort_gib | -Did your resolve your Snom issue?? |
| 14:40.44 | yang | hm |
| 14:40.47 | yang | no |
| 14:40.55 | *** join/#asterisk stmaher (n=stephen@mateus.province5.tv) |
| 14:41.14 | stmaher | Hello everyone.. I have 6 analog POTS lines.. which card would you recommend with Asterisk ? |
| 14:41.15 | mort_gib | What's up?? |
| 14:41.20 | yang | On my GS phones I can dial *8 (send) to take the unanswered call, dunno how to do it with Snom |
| 14:41.34 | mort_gib | yang: same |
| 14:41.42 | yang | there is no Send key on snom |
| 14:41.59 | mort_gib | the trick is in the pickupgroup and callgroup in sip.conf |
| 14:42.13 | yang | yep |
| 14:42.40 | yang | I would like to know which key substitutes SEND key |
| 14:42.56 | jonavogt | yang: what was the problem? |
| 14:43.33 | yang | I see on Snom Unavailable *8 |
| 14:43.45 | yang | when I do *8 (Confirmation key) |
| 14:44.00 | yang | BLF works to some extent |
| 14:45.20 | jonavogt | I at least got BLF working: <sip:number@asterisk>|*8 which dials *8number when pressed and ring |
| 14:45.21 | jonavogt | ing |
| 14:46.46 | yang | mort_gib: indeed i sorted out pickup and callgroup now it works |
| 14:47.13 | jonavogt | yang: then I got _*8number,1,Pickup(number) in my dialplan... |
| 14:47.24 | jonavogt | yang: you could also use a regex here |
| 14:48.39 | jonavogt | yang: but no clue about *8, i get unavaible *8 to.... but i don't have a call rining right now... and some other problems which prevent me from generating one. |
| 14:49.23 | mort_gib | jonavogt: you might want to use another key for pickup |
| 14:50.48 | jonavogt | mort_gib: yeah, i'll try that, when I got misdn working again... |
| 14:51.00 | jonavogt | mort_gib: thanks |
| 14:52.40 | jonavogt | anyone, how can I debug misdn channel not avaible problems the best way? |
| 14:53.42 | jonavogt | or better no channel avaible problems |
| 14:54.01 | *** join/#asterisk riddlebox (i=80fc732b@gateway/web/ajax/mibbit.com/x-fae60c8f3adcec89) |
| 15:02.54 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-6cee8008bf5380bc) |
| 15:02.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:03.27 | *** part/#asterisk Hanif08 (n=mail@pub01.comm.jaring.my) |
| 15:04.56 | *** join/#asterisk ptinsley (n=pdt@c-68-52-120-169.hsd1.tn.comcast.net) |
| 15:06.15 | yang | another thing, The lights on snom do they only blink when the call is activated on the other side? Or can you set it to light permanently to see if the other extension is online? |
| 15:07.05 | yang | jonavogt: mort_gib ? |
| 15:07.40 | *** join/#asterisk mort_gib (n=mjensen@164.236.120.212.dsl.dynamic.gibconnect.com) |
| 15:08.03 | jonavogt | i haven't tried that... mine is on when the other extension is on the phone and blinking if the extension is ringing... |
| 15:08.25 | yang | yes |
| 15:08.30 | yang | i would like it like that |
| 15:09.19 | yang | In Snom setting you have selected BLF ? or some other option ? |
| 15:09.22 | kinnaz | hello, i have audiocodes mediant 1000, tryng to install asterisknow to it via pxeboot, when i get to boot menu my screen goes wild and i get alot of these OIH15 text like all over the screen, i was wondering how could fix it |
| 15:10.22 | yang | I have it like....when the extension is in a call it lights, and when the remote is ringing it blinks |
| 15:10.31 | mort_gib | kinnaz: Check line speed |
| 15:10.41 | kinnaz | mort_gib not sure how to set line speed in that pxeboot menu thingie |
| 15:11.01 | mort_gib | You would have the problem on your minicom or similar |
| 15:11.30 | kinnaz | mort_gib my connection to device itself has rigth speed, otherwise i wouldnot be able to run around in setup |
| 15:11.41 | kinnaz | mort_gib the problem rises when i get to the part i should boot from network |
| 15:11.51 | jonavogt | yang: i got the same as you have... |
| 15:12.09 | mort_gib | I use Soekris devices a lot |
| 15:12.36 | mort_gib | When I instal OpenBSD the device uses 19200 but the install uses 9600 |
| 15:12.56 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:13.22 | Katty | hugs anthm |
| 15:13.30 | anthm | hello |
| 15:13.52 | yang | jonavogt: becouse on GS its different , the lights light all the time green, when remote is disconected they light red |
| 15:14.35 | jonavogt | yang: unfortunally the snoms just have one color |
| 15:14.43 | yang | kinnaz: any particular reason why you don't go with asterisk installation from the distribution ? |
| 15:14.59 | yang | jonavogt: not a problem, just wanted to know how they behave |
| 15:15.04 | kinnaz | yang you mean exactly what ? |
| 15:15.14 | yang | kinnaz: do you need asteriskNOW ? |
| 15:15.18 | kinnaz | yes |
| 15:15.36 | yang | kinnaz: becouse AsteriskNOW is different than Asterisk |
| 15:15.51 | yang | it has a GUI made by digium |
| 15:15.59 | yang | which sits on top of asterisk |
| 15:16.00 | kinnaz | basicaly my problem is that i dont know how to force tty speed to that bootloader |
| 15:16.11 | kinnaz | yang yeah i know, thats the reason we are using in that audicodes device |
| 15:16.24 | yang | ok |
| 15:16.44 | kinnaz | but will see tomorrow i shall get cdrom to try to boot from cdrom, maybe that will end better |
| 15:17.08 | yang | ~asterisknow |
| 15:17.09 | jbot | methinks asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 15:17.29 | The_Lightside | hi all, having an issue where MOH stops playing when system reaches approx 90 active calls |
| 15:19.10 | The_Lightside | any ideas anyone? |
| 15:20.22 | yang | The_Lightside: have you checked the musiconhold.conf ? |
| 15:21.50 | The_Lightside | in terms of? |
| 15:22.08 | yang | for any unusual settings |
| 15:23.20 | The_Lightside | other than: [GEMS-INFO] |
| 15:23.21 | The_Lightside | mode=files |
| 15:23.21 | The_Lightside | directory=/var/lib/asterisk/moh/GEMS-INFO |
| 15:23.27 | *** join/#asterisk mog (n=mog@nat/digium/x-372346cb8a7505d3) |
| 15:23.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:26.24 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:26.50 | jameswf | argh vacation over |
| 15:27.59 | jjshoe | :P |
| 15:28.57 | *** join/#asterisk Sargun_screen (n=sargun@64-142-51-251.dsl.static.sonic.net) |
| 15:29.16 | Sargun_screen | Now, I actually need an SS7 link, any ideas on how to get one, or how to get an ss7 "proxy" |
| 15:30.14 | Sargun_screen | I need it for SS7 MAP. |
| 15:31.04 | riddlebox | yang: AsteriskNow uses Freepbx gui |
| 15:31.40 | *** join/#asterisk kerx (n=kerx@adsl-69-104-78-246.dsl.irvnca.pacbell.net) |
| 15:32.17 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:34.47 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-2b490db207c7b86e) |
| 15:36.59 | Katty | oh boy. |
| 15:37.03 | Katty | my siste found religion. |
| 15:37.06 | Katty | sister. |
| 15:37.26 | jameswf | Katty: Was it lost? |
| 15:37.58 | [TK]D-Fender | I have witnessed many things.... but never Jehovah |
| 15:37.59 | Katty | jameswf: apparently it was worthy of a 3 page facebook post ^_- |
| 15:38.10 | Katty | [TK]D-Fender: not that religion |
| 15:38.20 | Katty | [TK]D-Fender: tho i wouldn't be surprised....if...eventually... |
| 15:38.23 | Katty | well, let's not go there |
| 15:38.33 | Katty | i'm still trying to fathom HOW |
| 15:38.38 | jameswf | is on most of the social networks but has a seperate blog, no way i would maintain all of em |
| 15:39.28 | jameswf | Jehovahs witnesses came to my house Dec 27th I asked em if they had a nice christmas |
| 15:39.36 | Katty | i can't say i'd want most people reading my blog. |
| 15:39.37 | riddlebox | lol |
| 15:39.47 | Katty | jameswf: be careful |
| 15:39.56 | Katty | jameswf: that sort of question will lead into a 30 min discussion with them |
| 15:40.18 | Katty | jameswf: after 18 years of forced religion, i know how their brains work. |
| 15:40.23 | jameswf | You would think as decked out as my house was with christmas lights my door would be the wrong one to knock on |
| 15:40.32 | Katty | by far |
| 15:40.35 | Katty | you need help |
| 15:40.40 | Katty | you are TOP priority |
| 15:41.07 | jameswf | I would only talk religion If we used my bibles i have enough for everyone... they left |
| 15:41.42 | jameswf | mormons are more fun atleast they pretend to listen |
| 15:42.01 | *** join/#asterisk travis_h (n=Travis@70.15.96.160.res-cmts.eph.ptd.net) |
| 15:42.08 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 15:42.09 | Katty | i don't know what would happen if the mormons showed up |
| 15:42.12 | Katty | if the JWs showed up... |
| 15:42.21 | Katty | they'd probably just say hi, and then leave due to recognizing me |
| 15:42.29 | Katty | well, no, probably they wouldn't |
| 15:42.43 | ricko73 | jameswf: what's better is when they show up at a Lutheran Pastor's house... |
| 15:42.45 | Katty | they'd probably ask me a thousand question about why i don't go to the "Kingdom Hall", even those it's been over 10 years. |
| 15:42.55 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:42.56 | jameswf | my wifes grandma is JW we are her dirty little secret :) |
| 15:43.04 | Katty | oh yeah. |
| 15:43.17 | Katty | my /entire/ family was. |
| 15:43.22 | travis_h | hello all.... I am using freePBX and for some reason my trunk to my service provider is not registering. Is this where I could find some help? |
| 15:43.23 | Katty | mom and dad, and the grandparents |
| 15:43.34 | Katty | travis_h: you should try #freepbx |
| 15:43.42 | travis_h | okay... thanks |
| 15:43.43 | Katty | jameswf: my head was royally screwed up |
| 15:44.27 | jameswf | so i joined myyearbook.com and have like $13k to bad its all fake... |
| 15:44.41 | Katty | :< |
| 15:46.34 | travis_h | seems #freepbx is just about dead :( |
| 15:47.16 | *** join/#asterisk chendy (n=chatzill@121.35.51.73) |
| 15:47.28 | rickross | anyone have experience with echo on headests using Asterisk 1.4.x and Polycom phones? No echo at all on handset, but serious echo from headset (a Hello Direct "HelloSet Pro") |
| 15:47.34 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-18d0f6b47d94ead3) |
| 15:47.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:50.11 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 15:50.24 | jameswf | rickross: adjust volume on headset |
| 15:51.00 | [TK]D-Fender | rickross: there are AEC settings to tweak if you cannot do this reasonably on the headset itself |
| 15:51.07 | rickross | jamesswf - we'll give it a try |
| 15:51.19 | [TK]D-Fender | rickross: And * has nothing to do with this |
| 15:52.03 | rickross | [TK]D - good to know |
| 15:52.05 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 15:52.31 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:52.41 | rickross | I had seen this, and wasn't sure if it might be something adjustable in * - http://www.mail-archive.com/asterisk-users@lists.digium.com/msg130504.html |
| 15:54.37 | jonavogt | what does NO USERUESRINFO mean? |
| 15:54.46 | *** join/#asterisk syrusfrost (n=none@adsl-224-238-33.asm.bellsouth.net) |
| 15:54.48 | seanbright | jonavogt: where are you seeing that error message? |
| 15:55.02 | [TK]D-Fender | rickross: Yes, those are the setings the enable |
| 15:55.12 | *** part/#asterisk travis_h (n=Travis@70.15.96.160.res-cmts.eph.ptd.net) |
| 15:55.12 | jonavogt | in asterisk trying to use a misdn line |
| 15:55.36 | jonavogt | seanbright: full line is "P[ 4] NO USERUESRINFO" |
| 15:55.47 | seanbright | is that a direct copy/paste? |
| 15:55.55 | jonavogt | yup |
| 15:56.01 | seanbright | strange that it's spelled wrong |
| 15:57.15 | seanbright | what version of asterisk? |
| 15:57.25 | jonavogt | seanbright: 1.4.21.1 |
| 15:58.08 | seanbright | there is a variable MISDN_USERUSER that you can set |
| 15:58.11 | seanbright | from dialplan |
| 15:59.07 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 15:59.29 | jonavogt | seanbright: so this has nothing to do with the second next line which says: "P[ 4] !! NO FREE CHAN IN STACK" ?? |
| 15:59.42 | Katty | nothing quite like a lil hiccupy rotweiller puppy to brighten the morning |
| 15:59.48 | seanbright | jonavogt: i have no idea |
| 15:59.58 | seanbright | jonavogt: i'm just pulling all of this out of my ass. |
| 16:00.23 | jonavogt | seanbright: no problem... thought I'd ask. |
| 16:01.17 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-196-172-rrdg-esr-2.dynamic.isadsl.co.za) |
| 16:01.55 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:04.45 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 16:07.18 | *** part/#asterisk wacky_ (n=abourget@IP-208-88-110-46.mtl.fibrenoire.ca) |
| 16:08.07 | *** join/#asterisk km2 (n=x@mobile-166-217-156-075.mycingular.net) |
| 16:11.30 | *** join/#asterisk af_ (n=getsmart@88-149-240-130.dynamic.ngi.it) |
| 16:13.02 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.29) |
| 16:13.12 | Daejeo | hello * :) |
| 16:13.29 | jonavogt | Daejeo: hi |
| 16:14.11 | Daejeo | hello jonavogt |
| 16:15.01 | *** join/#asterisk reneger (n=reneger@dslb-092-073-025-142.pools.arcor-ip.net) |
| 16:15.28 | Daejeo | is there any Indian guy? |
| 16:16.05 | *** join/#asterisk Gnutoo (n=gnutoo@host74-134-dynamic.21-79-r.retail.telecomitalia.it) |
| 16:17.16 | *** join/#asterisk tobias (n=tobias@cpe-076-182-095-118.nc.res.rr.com) |
| 16:18.10 | Daejeo | Restrictions on Internet telephony removed by TRAIâs |
| 16:18.40 | Daejeo | http://www.trai.gov.in/trai/upload/PressReleases/594/pr18aug08no70.pdf |
| 16:26.31 | kannan | Daejeo , i'm Indian |
| 16:27.31 | *** join/#asterisk Serge8888 (n=Serge@76.66.214.130) |
| 16:28.19 | Daejeo | did hear the new about voip termination to pstn and vice-versa? |
| 16:28.34 | kannan | yes, but only ISPs can offer it |
| 16:28.41 | Daejeo | no no no |
| 16:28.56 | Daejeo | there is one more guy |
| 16:29.29 | *** join/#asterisk maddog01 (n=minotaur@mail.upperjamestoyota.ca) |
| 16:29.50 | kannan | Daejeo, what one more guy? |
| 16:29.50 | Daejeo | National Long Distance (NLD) -STD-PCo guy |
| 16:29.57 | kannan | yes of course |
| 16:30.29 | kannan | but no commercial offers so far , thay all use it to competitively mower their rates, as there is no more access deficit charges levied |
| 16:30.52 | kannan | know any one giving foreign origination on Ip to PSTN india so far? |
| 16:31.23 | Daejeo | well, you will see soon |
| 16:31.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:32.13 | kannan | but we find a large number of DID providers on the lists hah ha |
| 16:32.31 | Daejeo | maybe not like that |
| 16:33.40 | Daejeo | people will start small business- max 100 DIDs |
| 16:34.25 | Daejeo | because the current internet infrastructure is not really good |
| 16:35.29 | *** join/#asterisk kippi (n=chriso@83-244-164-130.cust-83.exponential-e.net) |
| 16:35.31 | kippi | hey |
| 16:35.59 | Daejeo | do you know how much is the license fee for STD-PCO? |
| 16:36.10 | kippi | i need to trace one user to see if there phone rings when I dial a main number, is there a simple way to do this, its just the console is a bit busy |
| 16:38.48 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:39.21 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:39.21 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:40.04 | *** join/#asterisk voxter (n=voxter@190.241.15.218) |
| 16:42.49 | *** part/#asterisk jonavogt (n=jonavogt@u51-229.dsl.vianetworks.de) |
| 16:43.38 | simond | I'm trying to get a g729 pass-thru call from a SIP handset (g711|g729) to a SIP trunk (g729) via a Dial() in asterisk, however the SIP messages seem to indicate that asterisk has settled on g711 with the handset before the Dial(SIP/number@trunk,<no options>) is attempted. Is there a way around that without restricting the handset to only use g729 in sip.conf? |
| 16:43.45 | riddlebox | it really sucks that I cannot find a ITSP to port my home number |
| 16:44.29 | *** join/#asterisk op3r (n=op3r@124.104.89.249) |
| 16:45.00 | op3r | hello is there a certain way to block an IP trying to register on asterisk? |
| 16:46.12 | simond | op3r: not that I know of, you have to do it externally using iptables, or whever your OS supplies for packet filtering. |
| 16:47.03 | simond | er, I mean the way I know if is iptables; not possible to block within asterisk as far as I know. |
| 16:47.45 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:47.55 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr) |
| 16:48.00 | maddog01 | riddlebox: are you trying to get sip service at home? |
| 16:48.35 | op3r | simond: but if you do it on iptables will it block other ips trying to register also? |
| 16:48.46 | simond | op3r: not if you're specific about it. |
| 16:49.31 | op3r | -A RH-Firewall-1-INPUT -m udp -p udp --dport 5060 -j ACCEPT |
| 16:49.31 | op3r | <PROTECTED> |
| 16:49.50 | op3r | -A RH-Firewall-1-INPUT -m udp -p udp --dport 5060 -j Block -s ip here |
| 16:49.53 | op3r | ? |
| 16:50.08 | simond | something like that; except you need to specify the ip as well |
| 16:50.17 | simond | I believe the option is --source |
| 16:50.30 | simond | er, so yes. as you said :P |
| 16:51.06 | simond | takes another sip of his coffee. |
| 16:51.07 | op3r | -A RH-Firewall-1-INPUT -m udp -p -s 192.168.0.0 udp --dport 5060 -j REJECT |
| 16:51.38 | op3r | ? |
| 16:51.54 | simond | yes, except with a valid IP, or specify the netmask if you are blocking a range of IP addresses. |
| 16:52.06 | op3r | yep but the IP is just an example |
| 17:02.21 | op3r | ocnarf |
| 17:02.31 | op3r | !seen ocnarf |
| 17:04.03 | riddlebox | maddog01, yes |
| 17:04.31 | _ShrikE | riddlebox: is it because it is home service, or are you in a smaller market? |
| 17:05.00 | riddlebox | _ShrikE: it seems that it is because it is a smaller market |
| 17:05.06 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 17:05.09 | Katty | AHhhhhhhhhhhieeeee!!! |
| 17:05.13 | Katty | asplodes |
| 17:05.26 | _ShrikE | ducks |
| 17:05.27 | riddlebox | but like broadvoice has numbers available in areas that are smaller than the town I live in |
| 17:05.41 | _ShrikE | riddlebox: if you don't mind me asking, what town is it? |
| 17:05.48 | riddlebox | Trenton, Illinois |
| 17:06.13 | Katty | that's not far from here |
| 17:06.22 | riddlebox | Katty, where are you? |
| 17:06.44 | Katty | 2hours directly south of stl along the river |
| 17:06.46 | Katty | cape girardeau |
| 17:06.54 | riddlebox | ohh we just did an install there |
| 17:06.58 | Katty | in cape? |
| 17:07.01 | riddlebox | yup |
| 17:07.04 | Katty | sweet |
| 17:07.06 | Katty | who was it? |
| 17:07.11 | riddlebox | not asterisk though.... |
| 17:07.16 | riddlebox | avaya IP Office |
| 17:07.24 | Katty | that's okay. we do more than asterisk too (unfortunately) |
| 17:07.25 | riddlebox | MidAmerica Hotels |
| 17:07.44 | Katty | mount auburn |
| 17:07.47 | Katty | yep, know exactly where that's at |
| 17:07.54 | Katty | drive past it every day ;) |
| 17:08.02 | Katty | right by the mall |
| 17:08.04 | riddlebox | really |
| 17:08.10 | riddlebox | what do you guys sell? |
| 17:08.12 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:08.14 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 17:08.15 | Katty | we're mostly a copier dealer |
| 17:08.19 | watchy | hey sexy |
| 17:08.26 | Katty | but we do some sprint phones, some furniture |
| 17:08.27 | watchy | what kinda connector is on a sangoma analog card? |
| 17:08.29 | riddlebox | you sell asteisk solutions? |
| 17:08.32 | Katty | some asterisk stuff... some samsung stuff |
| 17:08.34 | watchy | on the phone cable |
| 17:08.37 | Corydon76-dig | copies Katty |
| 17:08.47 | Katty | riddlebox: yeah, but only when directly asked ^_- if that makes sense |
| 17:08.51 | Katty | riddlebox: they don't really market it |
| 17:08.56 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr) |
| 17:08.57 | Katty | riddlebox: we're a toshiba and samsung partner |
| 17:09.02 | riddlebox | Katty: cool, do you do any in the stl area? |
| 17:09.08 | Katty | riddlebox: and with /just/ me it's hard to do a lot of bigger installs. |
| 17:09.10 | riddlebox | asterisk |
| 17:09.12 | Katty | riddlebox: not yet |
| 17:09.22 | Katty | riddlebox: we've stuck with only cape so far |
| 17:09.28 | riddlebox | if you do, I would be glad to help |
| 17:09.37 | watchy | is it a Rj11 or what on a sangoma analog card cable? |
| 17:09.52 | Katty | riddlebox: really? you solo install them? |
| 17:10.00 | Katty | riddlebox: or is it associated with a larger company? |
| 17:10.04 | riddlebox | I have a couple that I have done on the side |
| 17:10.09 | Katty | sweet (= |
| 17:10.15 | Katty | <PROTECTED> |
| 17:10.52 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
| 17:11.03 | watchy | is there an rj4 connector? |
| 17:11.18 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 17:12.43 | Corydon76-dig | watchy: http://en.wikipedia.org/wiki/4P4C |
| 17:13.01 | watchy | yjsm |
| 17:13.02 | watchy | thanks |
| 17:13.13 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-181-200.phil.east.verizon.net) |
| 17:13.52 | watchy | i need the ends to make cables for Sangoma cards |
| 17:13.52 | riddlebox | _ShrikE: do you have any ideas on a provider? |
| 17:14.08 | Corydon76-dig | watchy: Graybar carries them |
| 17:14.15 | watchy | thanks bro |
| 17:14.19 | watchy | i owe u a hug |
| 17:14.23 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 17:14.36 | Assid | anyone here using asterlink by chance? |
| 17:14.43 | *** join/#asterisk Ward1983 (n=ward@91.178.23.198) |
| 17:15.49 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:16.10 | _ShrikE | riddlebox: no sorry |
| 17:16.26 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:16.27 | riddlebox | yeah it sucks |
| 17:16.55 | Ward1983 | any distros you can recommend where i can choose what features i need? |
| 17:17.11 | watchy | build your own |
| 17:17.12 | Ward1983 | (basically i only beed voip and a webinterface) |
| 17:17.25 | Ward1983 | watchy, nah im a home user |
| 17:17.37 | watchy | pretty much all ready made asterisk distros are for special needs people |
| 17:17.55 | watchy | everyone in this channel is pretty much anti gui |
| 17:18.05 | Ward1983 | watchy, ok so what OS would you recommend using? and are there any trouble using xen? |
| 17:18.09 | watchy | gentoo |
| 17:18.24 | watchy | get the most non overhead linux distro made |
| 17:18.40 | Ward1983 | thats gentoo? :o |
| 17:18.41 | Ward1983 | lol |
| 17:18.55 | watchy | yea, its so low overhead it doesnt even have a instaler |
| 17:18.59 | *** join/#asterisk Sargun (n=Sargun@66.151.148.225) |
| 17:19.01 | watchy | or come with ftp |
| 17:19.12 | Assid | anyone suggest a good incoming provider? |
| 17:19.21 | Ward1983 | watchy, and a debootstrapped debian would that givem uch trouble? |
| 17:19.27 | Sargun | Assid, Where do you DIDs for? |
| 17:19.42 | Assid | Sargun: incoming tollfree |
| 17:19.55 | watchy | i never used debian, i don't much care for it though |
| 17:19.55 | Ward1983 | watchy, thats very minimal too |
| 17:20.02 | watchy | i hate the nicknames of their distros |
| 17:20.03 | Assid | asterlink going down is starting to be painful |
| 17:20.06 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 17:20.29 | Ward1983 | watchy, and xen? any problems with it i sohuld know about? |
| 17:20.36 | pdfhacker | Assid: I've been very happy with Iaxtel |
| 17:20.44 | watchy | never used xen, i don't even know what it is |
| 17:20.45 | pdfhacker | going on two years now |
| 17:20.57 | watchy | is it some new linux distro? |
| 17:21.02 | Ward1983 | watchy, you dont know what xen is :o |
| 17:21.13 | watchy | its probably some gui driven retard distro |
| 17:21.14 | Ward1983 | watchy, its to virtualise systems |
| 17:21.17 | watchy | like ubuntu |
| 17:21.18 | mcargile | xen does not have a realtime clock so ztdummy will not work |
| 17:21.35 | Ward1983 | mcargile, any workarounds for that? |
| 17:21.41 | Sargun | Assid, I personally like flowroute |
| 17:21.54 | Sargun | pdfhacker, I've had a lot of issues with them. |
| 17:21.56 | Ward1983 | watchy, ok i'm not ever gonna ask advice from you again, you're ignorant |
| 17:22.10 | Ward1983 | stating xen is probably a GUI based distro like ubuntu....... |
| 17:22.11 | Assid | Sargun: yeah, but they dont have 6/6 for incoming |
| 17:22.18 | mcargile | there is a patch i wrote over a year ago in the bug tracker i think which makes ztdummy use the kernel tick but this is far less accurate especially virtualized |
| 17:22.20 | Assid | and they have that whole minimum stuff or something |
| 17:22.21 | Sargun | Assid, 6/6? |
| 17:22.30 | Sargun | Assid, Their customer service is really nice, their prices are ok. |
| 17:22.34 | mcargile | and you have to have the kernel tick set to 1000hz |
| 17:22.37 | Assid | Sargun: 6 seconds pulse with 6 seconds increments |
| 17:22.52 | Sargun | Assid, 6 second pulse? |
| 17:22.52 | Ward1983 | mcargile, :( then ill probably have to use my old athlonXP |
| 17:23.10 | mcargile | you might try looking at openvz |
| 17:23.14 | Sargun | Assid, I haven't heard of this, you mean 6 second billing? |
| 17:23.46 | mcargile | it does not actually do virtualization but nifty stuff with chroots and kernel patches to keep the roots from interfering with each other |
| 17:24.08 | mcargile | also this is only an issue if you are using anything that needs a zaptel timer |
| 17:24.15 | mcargile | such as meetme or IAX trunking |
| 17:24.57 | Ward1983 | mcargile, i only plan to use a SIP trunk (i use it right now with elastix but that distro opens a TON of ports by default) |
| 17:25.17 | mcargile | not positive but it might work |
| 17:25.28 | Sargun | Assid, We do only about $100/mo of toll-free traffic |
| 17:25.43 | Ward1983 | mcargile, well its running in xen allready and it seems to work fine |
| 17:25.50 | mcargile | thats good |
| 17:25.59 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 17:26.10 | Ward1983 | mcargile, so i shouldnt need ztdummy? |
| 17:26.16 | Ward1983 | jsut to make sure i understand you correct |
| 17:26.16 | mcargile | like i said you will have limited functionality. anything that needs a proper timing source will not function |
| 17:26.44 | Ward1983 | mcargile, i'm not sure what needs a timing source and what not |
| 17:26.53 | Katty | poofs |
| 17:27.29 | mcargile | conferences, iax trunking, and music on hold will not work |
| 17:27.39 | Sargun | Assid, It's very cost effective for us, we were on sixtel/iax.cc, then we went to teliax. They kind of suck for in-depth/complicated applications |
| 17:28.45 | Assid | Sargun: yes.. 6 second billing.. thats pretty much the norm across the boardd |
| 17:28.48 | Assid | incoming and outgoing |
| 17:28.58 | Assid | i know flowroute just did it for outgoing |
| 17:29.21 | Assid | but everyone does origination as well |
| 17:29.26 | Sargun | Assid, It doesn't effect our bill significantly, maybe it's just restricted to our application. |
| 17:29.44 | Sargun | Assid, The total different was like $15 MAX (over 6 months) |
| 17:30.10 | Sargun | Assid, And that's when we peeked out at $300/mo |
| 17:30.43 | mcargile | not sure what your application is but one of the biggest cost savings for us was to find a sip provider that did not bill for ring time. we offer services to call centers |
| 17:31.04 | *** join/#asterisk piper69 (n=chatzill@unaffiliated/piper69) |
| 17:31.21 | jjshoe | mcargile really? on outbound you mean? |
| 17:31.26 | Sargun | mcargile, We do customer service for complaints through our 800-number |
| 17:31.29 | mcargile | so there is a number of origination that do not lead to answers. |
| 17:31.39 | Sargun | mcargile, Usually calls last 20-40 minutes. |
| 17:31.42 | mcargile | yes on outbound |
| 17:31.46 | jjshoe | hrm nice |
| 17:32.26 | mcargile | if you do a lot of outbound with very few connects not having to pay for the non-connects is nice |
| 17:32.53 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 17:33.30 | Sargun | mcargile, I think we pay 1 minute fee whether or not connected. |
| 17:33.58 | Assid | Sargun: all providers are 6/6 including your upstream |
| 17:34.33 | Sargun | Assid, I honestly don't care about price |
| 17:34.37 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
| 17:35.01 | Assid | Sargun: whats your incoming charges |
| 17:36.03 | Sargun | Assid, Let me login and see |
| 17:37.06 | Sargun | Assid, I don't remember.. You can call FL and see what yours would be |
| 17:37.23 | Sargun | It's like $0.01/minute or something like that |
| 17:37.29 | Sargun | +/- 1 cent |
| 17:37.33 | Sargun | that's not for our toll free |
| 17:38.11 | Ward1983 | thanx mcargile i needed to go for a moment |
| 17:38.50 | Sargun | 1.95c/m |
| 17:39.06 | Sargun | Which was slightly higher than our existing one. |
| 17:39.12 | Ward1983 | anyone know a good trunk for egypt? |
| 17:39.20 | Ward1983 | good / cheap |
| 17:39.29 | Sargun | but the amount of headache headache I got from other providers hurt me more. |
| 17:40.27 | Ward1983 | i mean to call to egypt, not from |
| 17:41.59 | Assid | hrmm got vmail |
| 17:44.32 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 17:46.24 | simond | I'm confused at how asterisk is negotiating codecs, is there an explanation anywhere online? |
| 17:46.35 | Sargun | simond, voip-info.org |
| 17:46.47 | Sargun | If that was turned into a book, it'd be worth thousands of dollars. |
| 17:47.39 | simond | my hardphone has ulaw/g729 enabled, yet asterisk is inviting the trunk in my Dial() command with gsm/ulaw/alaw. |
| 17:48.43 | Sargun | simond, did you write disallow=all |
| 17:48.49 | Sargun | and then allow=ulaw underneath it? |
| 17:49.39 | simond | I have tried a few combinations. One was: disallow=all;allow=ulaw,g729 for the handset, and disallow=all;allow=g729 for the trunk. For some reason, asterisk couldn't find a codec path in that case. |
| 17:51.25 | simond | I'm aparently misunderstanding how the decision of what codec to use is being arrived at, as that seems like a valid setup. |
| 17:51.59 | Sargun | simond, do you have reinvite off? |
| 17:52.10 | simond | no, I have reinvite on everywhere. |
| 17:54.33 | simond | I'm using a plain Dial(), no extra options other than the destination. No extraneous Answer()'s |
| 17:56.54 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:57.05 | simond | oh, I added disallow=all;allow=g729,ulaw,alaw to the [general] section, that seems to have made it possible to get through. |
| 17:57.24 | simond | Haha, however, now local g711 calls no longer work. |
| 17:58.58 | simond | I suppose that is necessary in the general section, as asterisk is a back-to-back user agent, right? |
| 18:01.07 | simond | Hmm, forcing the codec to g711 using the Set(SIP_CODEC=ulaw) for calls terminated on the local asterisk instance does not seem to work either. chan_sip states in the logs that it is switching to ulaw |
| 18:01.27 | simond | however when it goes to play back audio prompts, it complains of not being able to find a path from g729 to gsm. |
| 18:02.13 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:03.30 | [TK]D-Fender | simond: which is normal |
| 18:04.28 | simond | Does the SIP_CODEC=ulaw statement not do what I think it does? |
| 18:05.07 | [TK]D-Fender | simond: Perhaps is the codec is not already set |
| 18:05.22 | [TK]D-Fender | if* |
| 18:05.26 | simond | Upon Answer(), the packet shows that the call is using ulaw. |
| 18:05.49 | simond | a=rtpmap:0 PCMU/8000 |
| 18:05.50 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
| 18:06.32 | simond | but then channel.c seems to still be trying to talk g729. |
| 18:08.10 | *** join/#asterisk sasargen (n=chatzill@174-153-129-22.pools.spcsdns.net) |
| 18:08.58 | *** join/#asterisk Ward1983 (n=ward@91.176.70.30) |
| 18:10.36 | *** join/#asterisk markgreene (n=markgree@209.12.142.5) |
| 18:12.31 | markgreene | Can someone answer me this question: Does PRI technology allow for me to manipulate the NAME that I send out? |
| 18:12.52 | *** join/#asterisk bsdfreak (n=alex@ninja.paranode.net) |
| 18:13.04 | bkw_ | markgreene: usually NO |
| 18:13.54 | mog | some providers allow it |
| 18:13.57 | mog | most dont |
| 18:14.10 | markgreene | ok but it IS possible if the provider accepts it? |
| 18:14.29 | mog | yes, you can send callerid name on pri |
| 18:14.34 | mog | most providers just toss it |
| 18:15.12 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 18:15.48 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 18:15.54 | markgreene | mog, yeah that's what I'm dealing with here. My provider use to take whatever name I pulsed. Now they don't and they are telling me they never did :-) |
| 18:16.03 | Assid | doesnt iaxtel have a website? |
| 18:16.15 | markgreene | Assid, you mean teliax? |
| 18:16.17 | mog | markgreene, they might have changed their providers |
| 18:16.25 | Assid | pdfhacker: said iaxtel |
| 18:17.03 | markgreene | Assid, I've only heard of teliax |
| 18:17.11 | Assid | hrmm k |
| 18:17.19 | Assid | lemme check teliax ten |
| 18:17.24 | Assid | i actually prefer sip based |
| 18:17.39 | markgreene | Assid, they support both :D |
| 18:17.54 | Assid | hows their quality? |
| 18:18.43 | markgreene | Assid, I've actually had an account with them for 2 years now. Quality is great. Reliability for me has been all I need. They had a small outage once for an hour. |
| 18:19.02 | markgreene | Assid, and that was between 2 and 3 AM |
| 18:19.08 | Assid | well recently in the past 3 months asterlink gone down 2-3 times |
| 18:19.23 | Assid | its down as we speak |
| 18:19.43 | Assid | 1 hr 15 mins already KNOWN |
| 18:19.45 | markgreene | I've never used them, but I'll keep your review in mind |
| 18:20.18 | markgreene | Assid, Teliax has a good prepaid service. You could load the account with a few bucks and use it as a failover. |
| 18:20.31 | Assid | markgreene: need it for incoming |
| 18:20.51 | Assid | crap teliax is expensive for tollfree origination |
| 18:20.57 | markgreene | Assid, port the numbers if Teliax and move on :-) |
| 18:21.07 | markgreene | *if=to |
| 18:21.07 | Assid | markgreene: 5c /min ? |
| 18:21.10 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 18:21.14 | Assid | no freaking way |
| 18:21.34 | markgreene | Assid, ?!?! One second let me see this. |
| 18:21.45 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 18:21.47 | *** join/#asterisk kerx_ (n=pre_prog@38.118.193.10) |
| 18:22.04 | Assid | so far.. im considering flowroute, voicepulse and another |
| 18:22.13 | kerx_ | Flowroute is a great provider |
| 18:22.29 | Assid | thats cause you work for them |
| 18:22.48 | kerx_ | No, I've used them and continue to do so. They have intelligent staff and very friendly |
| 18:23.09 | Assid | yeah.. the guy i spoke to was nice.. i do have an account with them.. |
| 18:23.11 | kerx_ | I would love to work for them as an intern :) |
| 18:23.13 | Assid | just dont use them as much |
| 18:23.16 | markgreene | Assid, I have never noticed the 5 cent/min rate for Toll calls. All my DIDs are in the 2 cent / minute range. Sorry about that |
| 18:25.45 | Assid | man.. i cant do any other work.. cause my client keeps asking me is it up yet? is it up yet |
| 18:26.32 | kerx_ | That sucks |
| 18:26.39 | kerx_ | Let him know things will be up when they get up |
| 18:27.31 | kerx_ | I'm confused when registering Agents (SIP clients) why there is redundant work involved, having to place them in sip.conf, agents.conf, and users.conf ? |
| 18:28.17 | *** join/#asterisk SlicerDicer (n=SlicerDi@24-119-155-26.cpe.cableone.net) |
| 18:28.49 | SlicerDicer | I have a spa3102 I am trying to get my asterisk.adsi to download to a phone I have. Is it not possible with the spa3102? do I need different hardware? |
| 18:28.57 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:29.10 | SlicerDicer | I was looking at some cards that have fxs modules but I dont want to pull the trigger if I have something wrong :) |
| 18:29.59 | *** join/#asterisk riddlebox (i=80fc732b@gateway/web/ajax/mibbit.com/x-347b7a4909d6a28e) |
| 18:30.12 | riddlebox | Katty: you back? |
| 18:30.17 | SlicerDicer | exten => 1234,1,ADSIProg() |
| 18:30.19 | Sargun | kerx, you work for bayan? |
| 18:30.23 | SlicerDicer | should be simple enough is what I have read? |
| 18:30.33 | [TK]D-Fender | kerx_: agents.conf has nothing to do with your phones. Users.conf if GUI crap. |
| 18:30.35 | SlicerDicer | but when I dial it says that there is no adsi device attached? |
| 18:30.44 | kerx_ | Sargun, I would love to if he offered me an internship :P |
| 18:30.50 | [TK]D-Fender | kerx_: Not needed unless you're hadcuffing yourself to that POS |
| 18:30.55 | Sargun | kerx_, Ah, yeah |
| 18:30.55 | kerx_ | I see |
| 18:31.10 | Sargun | kerx, Are bayan and Jordan here? |
| 18:31.11 | kerx_ | [TK]D-Fender, so really the only thing required is sip.conf to have clients register |
| 18:31.23 | kerx_ | Sargun, I have no idea |
| 18:31.55 | [TK]D-Fender | SlicerDicer: I'm not sure ADSI works on anything other than Zap FXS |
| 18:32.03 | SlicerDicer | thats what I was thinking :/ |
| 18:32.10 | [TK]D-Fender | SlicerDicer: And sucks anyways |
| 18:32.14 | SlicerDicer | but I just wanted to be sure before I pulled the trigger on buying it :) |
| 18:32.20 | SlicerDicer | why does it suck? |
| 18:32.33 | [TK]D-Fender | SlicerDicer: * doesn't do anything for ADSI really. |
| 18:32.53 | [TK]D-Fender | SlicerDicer: * cannot be really interactive with it except for that one useless bit of Voicemail. |
| 18:33.05 | [TK]D-Fender | SlicerDicer: Total waste on a digital phone wanna-be |
| 18:33.19 | SlicerDicer | I really just wanted the voicemail features :) |
| 18:33.22 | SlicerDicer | lol... |
| 18:33.55 | [TK]D-Fender | SlicerDicer: Your phone came programmed from the telco, right? |
| 18:34.05 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-196-172-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:34.34 | [TK]D-Fender | SlicerDicer: because the vast majority of those phones are locked where * cannot load anything on to them |
| 18:34.36 | SlicerDicer | [TK]D-Fender: of course not.. I purchased it blank and not locked to any telco |
| 18:34.52 | [TK]D-Fender | SlicerDicer: Hope so.... but its blows anyways... |
| 18:35.27 | SlicerDicer | yep one of them I brought to Australia and it worked there :) |
| 18:35.29 | SlicerDicer | lol... |
| 18:35.33 | Sargun | Hm, anyone know anyone who provides SS#7 MAP access over VoIP? |
| 18:35.34 | SlicerDicer | 2002 |
| 18:38.31 | Katty | consumes yummy subway sammich |
| 18:38.35 | Katty | riddlebox: ping! |
| 18:38.47 | eppigy | i hate subway |
| 18:38.49 | Katty | :< |
| 18:38.51 | eppigy | QUIZNOS |
| 18:38.53 | Katty | ewww |
| 18:38.54 | eppigy | OR FIREHOUSE |
| 18:38.55 | Katty | quiznos makes me sick |
| 18:38.57 | eppigy | 8[] |
| 18:39.02 | Katty | especially that rosemary bread |
| 18:39.04 | eppigy | then you must hate delicous |
| 18:39.06 | eppigy | =( |
| 18:39.07 | SlicerDicer | [TK]D-Fender: thanks for your help :) |
| 18:39.21 | Katty | their sammies aren't too bad tho |
| 18:39.23 | SlicerDicer | I will likely get the FXS stuff to make it work :) even if its not full featured |
| 18:39.34 | eppigy | thats all i eat is their sammies |
| 18:39.37 | SlicerDicer | the board is chump change by comparsion to what I spent on the phones |
| 18:39.38 | Katty | and i like their chip select much more than subway |
| 18:39.39 | eppigy | wait |
| 18:39.47 | eppigy | sandwichs |
| 18:39.48 | Katty | flatbread sammies |
| 18:39.50 | [TK]D-Fender | SlicerDicer: A greater insult... |
| 18:39.50 | Katty | not sammiches |
| 18:39.51 | eppigy | ugh |
| 18:39.57 | Katty | the bread is just... bleh |
| 18:39.58 | eppigy | :[ |
| 18:40.01 | SlicerDicer | [TK]D-Fender: eh? |
| 18:40.02 | Katty | too much rosemary |
| 18:40.06 | Katty | makes me sick, like cilantro |
| 18:40.07 | eppigy | i only eat their white bread |
| 18:40.11 | [TK]D-Fender | SlicerDicer: Expensive analog phones |
| 18:40.16 | Katty | qdoba-- |
| 18:40.28 | SlicerDicer | [TK]D-Fender: I have had them for a while.. just recently started playing with asterisk |
| 18:40.33 | eppigy | TRABAJO |
| 18:40.36 | SlicerDicer | likely I will replace them before too long though |
| 18:40.40 | Katty | whassat |
| 18:40.57 | Sargun | Katty, is your real name Katty? |
| 18:41.05 | Katty | Sargun: no |
| 18:41.35 | [TK]D-Fender | SlicerDicer: Zap FXS sucks, I highly advise against investing in it |
| 18:41.42 | Sargun | How'd you come up with Katty? |
| 18:41.50 | SlicerDicer | [TK]D-Fender: is it unreliable? |
| 18:42.08 | [TK]D-Fender | SlicerDicer: Pricey, less flexible, etc |
| 18:42.15 | SlicerDicer | ahh |
| 18:42.23 | Katty | Originally, it was Karina, a vulcan from Star Trek. But i also adore animals, so it was switched to Katrina on slashnet.org. Katrina was taken on freenode, so I just made it Katty. |
| 18:42.29 | [TK]D-Fender | SlicerDicer: Dependant on your server, hardwired to it, etc |
| 18:42.37 | Sargun | Katty, Ah, cool. |
| 18:44.07 | Katty | Sargun: my real name is Josh, and I am from Indiana*** |
| 18:44.10 | Katty | ***the above is not true. |
| 18:44.50 | Sargun | That was random, and off. |
| 18:44.51 | Sargun | er |
| 18:44.51 | Sargun | dd |
| 18:44.53 | Sargun | odd |
| 18:45.02 | file | tackles Katty |
| 18:45.09 | Katty | that however, is Josh. |
| 18:45.12 | Katty | hugs file |
| 18:45.16 | file | indeed |
| 18:45.19 | file | hugs Katty |
| 18:45.55 | Sargun | file == some VoIP provider |
| 18:46.02 | Katty | file is my twin. |
| 18:46.02 | file | I am not. |
| 18:46.11 | Katty | he is my brother, from another mother. |
| 18:46.12 | file | (some VoIP provider) |
| 18:46.19 | Corydon76-dig | hugs file |
| 18:46.21 | Sargun | Are you sure? |
| 18:46.25 | Sargun | I talked to you before. |
| 18:46.25 | Katty | yes. |
| 18:46.27 | Sargun | I don't know why |
| 18:46.37 | Katty | was something broken? |
| 18:46.40 | file | long ago I was with a VoIP provider |
| 18:46.42 | Sargun | No, I'm talking to File. |
| 18:46.46 | file | but now I am with Digium |
| 18:46.51 | Sargun | file, that toll-free provider, that was really crappy? |
| 18:47.04 | Sargun | and used questionable marketing schemes? |
| 18:47.26 | Katty | well that only narrows it down to about 80% |
| 18:47.26 | Corydon76-dig | Sargun: You're not being specific |
| 18:47.35 | Sargun | Katty, haha |
| 18:47.37 | *** join/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 18:47.47 | Sargun | file, Give me the name, and I'll see if I remember or not |
| 18:47.56 | file | refuses to |
| 18:48.00 | Sargun | Grr. |
| 18:48.03 | Katty | why can't you love file for who he is )= |
| 18:48.04 | Sargun | I'll find out, grr. |
| 18:48.24 | Sargun | Man, I remember it was one of the original toll-free providers for Asterisk out there |
| 18:48.39 | Sargun | AstriCrap? |
| 18:48.44 | Sargun | :-P |
| 18:48.49 | *** join/#asterisk jeffgus (n=jeffgus@static-173-51-181-4.lsanca.ftas.verizon.net) |
| 18:48.59 | Katty | pokes jaytee |
| 18:49.02 | file | naps on Katty's shoulder |
| 18:49.11 | Katty | applies blanket to file |
| 18:49.12 | jaytee | ow!!! |
| 18:49.18 | Katty | jaytee: you're aliveeeee!!! :> |
| 18:49.22 | Katty | hugs jaytee |
| 18:49.27 | Sargun | file, Does digium have an SS7 link? |
| 18:49.32 | jaytee | hugs Katty |
| 18:49.36 | Sargun | *me opens his eyes widely* |
| 18:49.37 | Katty | jaytee: how're the ladies. |
| 18:49.42 | file | Sargun: ...no? |
| 18:49.45 | Sargun | What is this, ##socialites? :-P (kidding) |
| 18:49.51 | Katty | jaytee: besides big and beautiful. |
| 18:50.06 | jaytee | they're stuck indoors because it's chilly outside |
| 18:50.10 | Katty | Sargun: of course. |
| 18:50.14 | Sargun | file, Considering you work at VoIP middle earth, do you know of any people that do SS7 over SIP? |
| 18:50.14 | Katty | jaytee: ooh, bummer )= |
| 18:50.22 | Katty | jaytee: probably bored )= |
| 18:50.30 | Katty | jaytee: you should go shake a leaf at them. |
| 18:50.33 | file | Sargun: ...no |
| 18:50.39 | jaytee | Katty, yep they get restless by the end of winter |
| 18:50.57 | Katty | jaytee: do they whine a lot? |
| 18:50.58 | jaytee | but they get to paint and do inside games |
| 18:51.05 | Katty | hot dog. |
| 18:51.07 | Katty | i wanna paint |
| 18:51.16 | jaytee | Katty, i'ts more like a deep rumbling noise |
| 18:51.22 | Katty | nods |
| 18:51.26 | Sargun | file, or MAP over SIP/IAX? |
| 18:51.44 | Katty | jaytee: i bet there's some really interesting noises there. |
| 18:51.49 | jaytee | but they also trumpet loudly, at least Sophi does more than the rest but the others will if they get pissed |
| 18:52.04 | file | Sargun: and no. |
| 18:52.10 | Sargun | file, great. |
| 18:52.18 | Katty | jaytee: they don't get mean with the keepers do they? |
| 18:52.41 | Katty | jaytee: or do the keepers make it a general rule to stay clear when they're pissy |
| 18:52.47 | jaytee | yeah, on occassion I can hear Mwangi roaring really loud. His enclosure is like 1200 feet from my office but his roar carries |
| 18:53.01 | file | Katty: are you feeling better? |
| 18:53.10 | Katty | jaytee: do they announce lunch time? |
| 18:53.12 | Sargun | Why doesn't any SIP provider provide Ghetto setups for guys like me? |
| 18:53.14 | Katty | file: yes'r. |
| 18:53.24 | Katty | file: the lymphnodes have stopped hurting, but still a bit of a sinus infection |
| 18:53.36 | Katty | file: lot of coughing at night :< |
| 18:53.41 | Sargun | Katty, Ouch. I feel bad for you |
| 18:53.43 | Assid | btw.. anyone here played with a cisco 7960 |
| 18:53.45 | jaytee | Katty, no they don't usually get mean although a couple in the herd tend to be once in awhile. Those don't get to go for walks on the access roads or interact with people. |
| 18:53.56 | file | Katty: :( |
| 18:54.01 | Katty | jaytee: :< |
| 18:54.15 | jaytee | Katty, was work what you were dreading last nite? |
| 18:54.20 | Katty | nods sadly. |
| 18:54.40 | Katty | but it's been a pretty good day so far. |
| 18:54.45 | Katty | waits for phone to ring |
| 18:55.03 | eppigy | HOLLA |
| 18:55.16 | jaytee | "dave" |
| 18:55.38 | eppigy | hello jaytee |
| 18:55.52 | jaytee | TRABAJO |
| 18:55.54 | beek | hello jaytee |
| 18:55.58 | jaytee | hi beek |
| 18:55.58 | eppigy | TRABAJO |
| 18:56.00 | Katty | i need to spend my days bandaging puppies. |
| 18:56.04 | eppigy | DONDE ESTA |
| 18:56.14 | eppigy | I will show you fear in a handful of dust |
| 18:56.21 | Katty | infinate dust? |
| 18:56.21 | mort_gib | Joder, puedes continuar en Ingles!! |
| 18:56.25 | jaytee | I need to spend my days firing rocket launchers at pirate whalers |
| 18:56.36 | mort_gib | :-) |
| 18:56.38 | Katty | you know, i dream about water a lot. |
| 18:56.41 | Katty | Oceans, lakes. |
| 18:56.45 | Katty | Whales. |
| 18:56.47 | Katty | it's weird. |
| 18:56.57 | eppigy | I always dream of the surface of the sun |
| 18:56.59 | beek | Katty: me too -- and then I have to get up and go to the bathroom. |
| 18:57.06 | Katty | beek: hehe. |
| 18:57.13 | Katty | eppigy: do you wake up hot? |
| 18:57.37 | eppigy | i am always hot |
| 18:57.41 | eppigy | YOU KNOW WHAT I MEAN |
| 18:57.48 | Katty | k |
| 18:57.50 | jaytee | muy, muy caliente! |
| 18:57.55 | eppigy | SE |
| 18:57.59 | eppigy | MUY BIEN |
| 18:57.59 | Katty | (si) |
| 18:58.04 | eppigy | SI ME GUSTA |
| 18:58.09 | Katty | (kthx) |
| 18:58.21 | eppigy | QUE JAME |
| 18:58.45 | Katty | so. |
| 18:58.49 | Katty | jaytee |
| 18:58.55 | Katty | do I want a civil union, or a military wedding |
| 18:59.06 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr) |
| 18:59.30 | jaytee | sometimes I just wanna get a gun and run wild in my neighborhood shooting hispanics and screaming "You're all the reason my Lumenvox IVR is twice as difficult as it has to be!!!!" |
| 18:59.40 | Katty | ah hahahahaa |
| 19:00.00 | jaytee | Katty, is your beau in the military? |
| 19:00.05 | Katty | leave a couple alive so you can still have mexican down at the local place ;) |
| 19:00.13 | Katty | jaytee: honorable discharge |
| 19:00.21 | Katty | jaytee: but he was in for 10 years |
| 19:00.32 | jaytee | I really wouldn't shoot them, they're mostly very nice people |
| 19:00.35 | Katty | jaytee: we'd probably still qualify ^_- |
| 19:01.06 | jaytee | my neice had a military wedding. her hubby is a naval officer |
| 19:01.30 | jaytee | they had the guys lined up with an archway of swords thing |
| 19:02.49 | Katty | yeah |
| 19:02.51 | Katty | and the traditional firing of guns |
| 19:03.13 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:03.17 | Katty | probably shoot a pidgeon out of the sky )= |
| 19:03.35 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr) |
| 19:06.52 | jaytee | we didn't have any gun fire, we did have a piper though |
| 19:08.06 | jaytee | and my neice got married in one of the oldest churches in the US in Plymouth, MA. Right behind the church is the first cemetery of the Plymouth Colony and most of the original colonists from the Mayflower are buried there. |
| 19:08.30 | Assid | would anyone be kind enough to help me configure a cisco 7960 |
| 19:08.51 | jaytee | I would if I knew how |
| 19:09.07 | jaytee | but sadly....or joyfully depending on your viewpoint I don't |
| 19:09.29 | Daejeo | any hack for switchvox? |
| 19:09.48 | Katty | jaytee: piper? |
| 19:09.57 | jaytee | Daejeo, did you try Google? I hear it's really good! |
| 19:10.19 | jaytee | Katty, yeah a piper, y'know, a guy in kilts with a bagpipe? |
| 19:10.38 | Katty | oh |
| 19:10.40 | Katty | that kind of piper |
| 19:10.45 | Katty | well that's neat. |
| 19:10.54 | Daejeo | jaytee: i found some link , but after reading i lost the hope |
| 19:11.01 | Daejeo | :( |
| 19:11.17 | jaytee | Daejeo, hope springs eternal so just wait till spring? |
| 19:11.44 | Daejeo | The switchvox systems are VERY tightly locked down. Kerry Garrison said |
| 19:12.24 | Daejeo | http://www.trixbox.org/forums/trixbox-forums/open-discussion/free-switchvox-now-available |
| 19:12.31 | jaytee | oh, sure. now we're believing what Kerry Garrison says? (vomits) |
| 19:13.29 | jaytee | all Kerry needs is a clown suit to be a full blown assclown because he's already got the ass part down pat. |
| 19:13.43 | Daejeo | maybe some people already hacked, however they don't want to share it |
| 19:13.52 | Assid | anyone here played with cisco phones? the link i got ends up pointing to a pdf file instead of a zip file where im supposed to get the files |
| 19:14.03 | *** join/#asterisk mindCrime (n=chatzill@74.213.159.129) |
| 19:14.17 | Daejeo | cisco phone which one? |
| 19:15.52 | *** join/#asterisk WHYS (i=lpfm@137.28.94.209) |
| 19:16.28 | *** join/#asterisk _darkseid (n=Sanjeeva@cpc1-seve11-0-0-cust90.popl.cable.ntl.com) |
| 19:17.28 | Assid | 7960 |
| 19:17.41 | Daejeo | what are you trying to achieve ? |
| 19:19.51 | Assid | i got a 7960 with me, i need to configure and start using it with asterisk |
| 19:22.07 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:22.08 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:22.12 | Assid | cisco says to go to : http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 and pick up the firmwares, but i cant geyt anything there |
| 19:22.36 | Daejeo | ah |
| 19:22.51 | Daejeo | do you have an account ? |
| 19:23.01 | Daejeo | http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx |
| 19:23.11 | Daejeo | you can try instead |
| 19:23.11 | lmadsen | you can't get cisco firmware without an appropriate account |
| 19:24.09 | Assid | should i stick to sip? or should i let it use skinny? |
| 19:24.14 | Daejeo | i have 8.9 |
| 19:24.58 | Daejeo | i am using sip, and it is working fine |
| 19:25.45 | Assid | okay its got 6.0 (4.0) |
| 19:26.07 | Assid | the screen is actually broken.. i can barely make squat.. |
| 19:26.15 | Daejeo | oh |
| 19:26.45 | Assid | Daejeo: would you mind a pm? |
| 19:27.03 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:27.10 | Daejeo | np |
| 19:29.54 | jjshoe | I've worked with Kerry, it's not what I would call fun and intelligence packed |
| 19:37.32 | *** join/#asterisk E|nyPRI (n=sales@dsl-208-81-2-10.les.net) |
| 19:44.53 | SlicerDicer | anybody aware of a barebones basic sip phone that is not expensive that I can get? |
| 19:45.06 | SlicerDicer | want to get one for a neighbor to let them play with it heh |
| 19:48.29 | phpboy | [Jan 5 21:45:44] NOTICE[3995]: chan_zap.c:8534 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 <---- I hate that my TElco keeps blaming me for that error, and I strongly doubt it's my fault |
| 19:48.50 | phpboy | Does anybody have any info on such an notice? |
| 19:48.59 | kerx_ | Anyone know what AgentCallbackLogin has been replaced with in Asterisk 1.4.x ? It is mentioning that this function is deprecated |
| 19:49.15 | *** join/#asterisk ebannister (n=ebannist@12.187.84.162) |
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| 19:50.18 | *** part/#asterisk ebannister (n=ebannist@12.187.84.162) |
| 19:50.20 | *** join/#asterisk dieguito84 (n=diego@host118-190-dynamic.12-79-r.retail.telecomitalia.it) |
| 19:50.57 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 19:51.13 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:51.14 | phpboy | kerx: It does work in 1.4. but I'd recommend local channels instead |
| 19:51.19 | phpboy | IT's worth the conversion |
| 19:53.45 | *** join/#asterisk Raceman (n=raceman@45-49-223.ftth.xms.internl.net) |
| 19:55.49 | Raceman | Hi all, first of all happy new year. I've Asterisk behind a router/firewall (NAT), and my problem is that the other party can't hear me. I can hear the other party. Wat could be the problem? |
| 19:56.06 | [TK]D-Fender | ~sipnat |
| 19:56.06 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:56.11 | [TK]D-Fender | Raceman: ^^^^^^^^ |
| 19:56.16 | Raceman | ok thanks |
| 19:56.58 | beek | [TK]D-Fender: you typed that reply so quickly I assume it was muscle memory. |
| 19:57.16 | [TK]D-Fender | beek: more than that |
| 19:59.03 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 19:59.13 | beek | [TK]D-Fender: Ah... a bot to reply for you?! |
| 19:59.26 | beek | The [TK]-bot |
| 20:02.44 | kerx_ | ast_streamfile: Unable to open agent-pass (format 0x4 (ulaw)): No such file or directory |
| 20:02.54 | kerx_ | Why does this happen on AgentLogin() |
| 20:03.01 | kerx_ | Not sure why it's trying to play any audio |
| 20:03.13 | Raceman | [TK]D-Fender: thanks a lot, it's working now! :-) |
| 20:03.15 | kerx_ | file.c:602 ast_openstream_full: File queue-youarenext does not exist in any format |
| 20:03.16 | kerx_ | Or this |
| 20:03.52 | [TK]D-Fender | kerx_: when you get around to actually looking at what you're doing, you'll know |
| 20:04.25 | kerx_ | Heh, having a hard time to understand the relationship w/ a few things in queues.conf, agents.conf, extensions.conf to setup basic ACD w/ rrmemory and Login/Logout functionality. |
| 20:04.31 | kerx_ | Either its me or the documentation. |
| 20:04.34 | kerx_ | I'ts probably me |
| 20:09.27 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 20:09.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:13.41 | *** part/#asterisk delcoyote (n=delcoyot@hispla.com) |
| 20:14.17 | eppigy | lol |
| 20:14.27 | eppigy | 14:02 < kerx_> ast_streamfile: Unable to open agent-pass (format 0x4 (ulaw)): No such file or directory |
| 20:14.31 | eppigy | 14:02 < kerx_> Why does this happen on AgentLogin() |
| 20:14.33 | eppigy | 14:03 < kerx_> Not sure why it's trying to play any audio |
| 20:14.48 | eppigy | [TK]D-Fender: your thoughts? |
| 20:16.28 | [TK]D-Fender | eppigy: KILL |
| 20:21.22 | eppigy | TRABAJO |
| 20:22.51 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:29.22 | Assid | needs help setting up this cisco 7960 phone |
| 20:29.36 | Assid | tftp says ttransmission failed for whateveer reason |
| 20:33.45 | Assid | hrmm. |
| 20:33.48 | Assid | checks whats up |
| 20:38.28 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
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| 20:53.19 | *** join/#asterisk mindCrime (n=chatzill@74.213.159.129) |
| 20:53.37 | Raceman | bye |
| 20:53.39 | *** part/#asterisk Raceman (n=raceman@45-49-223.ftth.xms.internl.net) |
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| 21:02.48 | eppigy | hello |
| 21:02.51 | eppigy | i am dave |
| 21:03.31 | beek | eppigy: hello dave |
| 21:03.54 | eppigy | hello |
| 21:05.24 | nix8n82 | hi dave |
| 21:06.10 | *** join/#asterisk vader-- (n=me@c-71-225-195-86.hsd1.nj.comcast.net) |
| 21:09.45 | eppigy | hello |
| 21:13.56 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552227.dsl.bell.ca) |
| 21:14.23 | WHYS | Assid: Get tshark and watch the Tftp transfer on your tftp server (tshark port 69) |
| 21:15.44 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:16.27 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-4c141cda8329a832) |
| 21:27.41 | jjshoe | http://twitter.com/kerrygarrison |
| 21:28.01 | jjshoe | ripppppppppppppppppppppppppp |
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| 21:32.53 | aiksa[LV] | hi anyone has seen error like this before: chan_dahdi.c: 9 Write to 68 failed: Unknown error 500 ? |
| 21:40.33 | Sargun | How many people here are actually service providers? |
| 21:40.51 | E|nyPRI | waves |
| 21:41.39 | aiksa[LV] | Sargun: service providers like telco or like integrators? |
| 21:41.51 | Sargun | Telcos/Sip providers |
| 21:42.10 | Sargun | People who peer with ILECS/CLECS |
| 21:42.41 | aiksa[LV] | Sargun: ok. |
| 21:44.20 | aiksa[LV] | ok, off for a sleep. see you all later. |
| 21:45.36 | *** join/#asterisk SparFux (n=raoul@e182018145.adsl.alicedsl.de) |
| 21:46.14 | Sargun | Gr, basically the only people who sell clean/straight/simple stuff is verisign: http://www.verisign.com/products-services/communications-services/connectivity-and-interoperability-services/ss7/gsm-map-ansi-41/index.html |
| 21:46.21 | Sargun | which means it'll be very expensive |
| 21:49.46 | SparFux | Is there a software phone which can directly connect to an mISDN or capi device? |
| 21:53.11 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 22:01.47 | E|nyPRI | has anyone here fooled with asterisk-ss7 first-hand, with an a-link or d-link ? |
| 22:03.42 | *** join/#asterisk seanmh (n=Mike@207.114.199.107) |
| 22:04.08 | seanmh | does asterisk support SCCP and how well does it work? |
| 22:08.04 | Corydon76-dig | seanmh: see chan_skinny |
| 22:09.14 | *** join/#asterisk timothy_jones (n=chatzill@207.90.4.3) |
| 22:09.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:11.10 | timothy_jones | hey guys, quick question: When issue Originate command via AMI with ActionID set, I get respose with same actionId. But subsequent events (newstate, hangup, etc) do have actionID. Is there work around for that? |
| 22:11.47 | seanmh | Corydon76-dig: do you have any comments on the reliability of chan_skinny |
| 22:12.51 | Corydon76-dig | Nope, never used it |
| 22:12.54 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 22:12.59 | Corydon76-dig | I hate Cisco phones with a passion |
| 22:13.02 | russellb | timothy_jones: you can't do it. |
| 22:13.08 | russellb | it just doesn't work that way ... |
| 22:14.07 | timothy_jones | russellb: :( what do other people do? |
| 22:15.14 | timothy_jones | I mean if you want to track when the call ends. Is there another way? |
| 22:15.22 | [TK]D-Fender | timothy_jones: Live happier lives not neurosing about their calls ;) |
| 22:17.03 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 22:17.18 | timothy_jones | He-he, I wish. Basically, I have a small automated dialer app. You click a button and Originate is issued. I wanna pop up a screen once the phone call is over. Any idears? |
| 22:19.12 | [TK]D-Fender | timothy_jones: Have your dialplan spit out an AMI even listing the channelid then monitor AMI for the hangup event |
| 22:19.56 | phpboy | Have there been any enhancements from the current zaptel drivers to dahdi? |
| 22:20.06 | [TK]D-Fender | phpboy: Yes |
| 22:20.27 | phpboy | [TK]D-Fender: Where can I get some docs on how to get dahdi going? |
| 22:20.34 | phpboy | I need to move from zaptel to dahdi |
| 22:20.38 | [TK]D-Fender | phpboy: in the tarball |
| 22:20.44 | path_ | http://debian.pastebin.com/d5e88a2b2 anyone willing to help ? |
| 22:21.54 | phpboy | [TK]D-Fender: you are too kind :D |
| 22:22.32 | [TK]D-Fender | phpboy: I know... a character flaw I'll soon correct... |
| 22:24.23 | timothy_jones | [TK]D-Fender: I am using astmanproxy, which filters by ActionID. I had crazy idea of patching asterisk and storing ActionID in a channel variable and then in Hangup event adding it to the response. Too crazy or what? |
| 22:25.55 | path_ | thanks in advance btw |
| 22:26.51 | [TK]D-Fender | timothy_jones: better :pass the actionid as a variable in your originate and then when you see the New-chan, map the two. When you see the hangup, you'll know which... |
| 22:28.19 | timothy_jones | [TK]D-Fender: Sorry, not tracking on "map the two" part |
| 22:31.27 | *** join/#asterisk Get_The_Fish (n=IceChat7@75.151.94.189) |
| 22:36.04 | [TK]D-Fender | timeshell: Have your AMI watch-process associate the actionID & channelID #'s and watch for the hangup |
| 22:36.16 | [TK]D-Fender | timothy_jones: ^^ |
| 22:36.28 | Get_The_Fish | yo... I cant seem to find this anywhere... does anyone have a link or can someone explain how I can change the state of an agent (as defined in agents.conf) from available to busy either with SIP or programmatically |
| 22:36.44 | [TK]D-Fender | Get_The_Fish: you DON'T |
| 22:37.27 | timeshell | TK: What?? |
| 22:37.38 | path_ | is there a way to see my trunks through CLI ? |
| 22:38.13 | [TK]D-Fender | timeshell: bad autocomplete |
| 22:38.17 | timeshell | oh |
| 22:38.56 | path_ | thanks, I'm kind of invisible |
| 22:53.08 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
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| 23:02.11 | phpboy | hmmm, that's odd, dahdi won't auto configure my spans |
| 23:02.24 | phpboy | however if I manually configure them they seem to work? |
| 23:02.26 | phpboy | very weird |
| 23:02.38 | Sargun | Be happy they work at all. |
| 23:04.29 | PanGoat | Are there any specific known vulnerabilities with the management interface? I'm about to write a bot that does some basic management tasks, but want to be as secure as possible from the getgo. |
| 23:05.07 | PanGoat | It's pretty straightforward, but maybe there's something I've not thought of in terms of security |
| 23:06.08 | [TK]D-Fender | PanGoat : Its plain-text so anyone seeing the packets will know everything |
| 23:06.16 | [TK]D-Fender | PanGoat: security = 0 |
| 23:11.31 | PanGoat | [TK]D-Fender: yeah, I was looking at this http://www.voip-info.org/wiki/view/Asterisk+Manager+Proxy |
| 23:12.03 | PanGoat | there are a couple different routes to go |
| 23:12.16 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 23:12.43 | PanGoat | Does anyone know if there are any official * projects working towards a secure interface? |
| 23:12.45 | [TK]D-Fender | PanGoat: Centralized insecurity, woohoo! |
| 23:13.06 | PanGoat | [TK]D-Fender: lol, a lose-lose situation. |
| 23:14.55 | phpboy | PanGoat: What security are you looking for? |
| 23:16.21 | PanGoat | in general, I really like the idea of integrating * control into other apps (e.g iphone/android control app) |
| 23:16.48 | PanGoat | something that can return data and doesn't invole punching through numbers to control the system |
| 23:17.09 | PanGoat | The management interface fits the bill in that sense |
| 23:17.26 | PanGoat | however, the plaintext thing is a really big downside |
| 23:18.09 | PanGoat | Proxies are one way, but as [TK]D-Fender pointed out, that's a single point of failure/insecurity |
| 23:20.47 | [TK]D-Fender | PanGoat: Make an account whose shell does the telnet locally, and SSH into that. |
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