00:02.06 | *** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it) |
00:02.21 | *** join/#asterisk ZeMMaD (n=ZeMMaD@209.59.105.69) |
00:03.07 | *** join/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net) |
00:03.47 | effape | it's kinda big - AMP |
00:05.12 | *** part/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net) |
00:05.51 | *** join/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net) |
00:06.44 | argentas | okay, AMP is nasty, gonna make diagnosing the issue nigh on impossible.. |
00:07.00 | Ariel_ | argentas, it's not nasty |
00:07.00 | justinu | heh |
00:07.29 | Ariel_ | he needs to get his country codes and setup for his location in the zapata.conf correct. |
00:07.49 | argentas | Ariel_: it is from the point of view of trying to figure out what the hell is going on in the dialplan |
00:08.04 | Ariel_ | argentas, I have no problem with the dial plans |
00:09.03 | argentas | yes, I agree, problem is *probably* the zap setup, but polarity switch is usually correct for the UK, but with that his SIP phone is not ringing |
00:09.30 | Ariel_ | his sip rings just when he pickup the line hangs up |
00:09.58 | Trazz | Ariel, |
00:10.01 | Ariel_ | he can change the dial sec's from the default 15 to something like 30 or more |
00:10.08 | Trazz | Do you recommend any good softphones ? |
00:10.18 | effape | well here it is http://pastebin.com/507463 |
00:10.24 | franck | Trazz: xten |
00:10.38 | franck | Trazz: ekiga (aka gnomemeeting) |
00:10.41 | Ariel_ | Trazz, xten makes a good one for most use. I use xlite for testing and most setups |
00:10.57 | argentas | 24 ; when polarity is used the phones doesn't rign and the asterisk doesn't pick up. |
00:11.18 | effape | isn't it polarity in the uk though? |
00:12.16 | argentas | effape: I thought so, yes, although looking again at what you have said, it does seem you are coming closer to it working with hist |
00:12.22 | effape | took a while for me to get to the point when asterisk didn't need to pickup the call to divert it to the sip. Now it rings on the incomming call end (and doesn't get picked up) |
00:12.48 | effape | yeah it's just that it drops when i pick it up |
00:13.02 | argentas | is that with polarity or hist? |
00:13.03 | effape | usehist works better than polarity but it just then get's hungup |
00:13.35 | effape | polarity doesn't do anything usehist get's to the sip but then hangup on pickup of the sip |
00:13.43 | justinu | analog telephony.... what a pain in the ass |
00:13.49 | effape | yeah tell me about it. |
00:13.55 | shmaltz | anybody tried this: |
00:13.57 | shmaltz | http://www.voipsupply.com/product_info.php?products_id=1009 |
00:14.01 | effape | I tried to get them to use isdn at least ;) |
00:14.10 | shmaltz | its an aastra 9112 |
00:14.33 | justinu | i've used aastra 480i |
00:14.35 | justinu | that's it |
00:14.37 | *** join/#asterisk rstandy (n=rastandy@d83-176-4-141.cust.tele2.it) |
00:14.58 | effape | i'm wondering if it's something to do with caller id not coming in. I dunno if it's setup or not on the line but i don't know how to check. |
00:15.07 | argentas | one sec, i'm gonna see if i've got a backup of the config from the analogue setup i had connected to BT (I'm not using only PRI) |
00:15.15 | franck | is it possible to use asterisk to look in enum first if the phone is available and then dial via PSTN if not? |
00:15.15 | argentas | s/not/now |
00:15.16 | Ariel_ | shmaltz, it's an ok phone. But I would recommend the Sipura 941 better |
00:15.46 | shmaltz | Ariel_, comparing those 2, Audio who wins? |
00:15.46 | Ariel_ | franck, yes |
00:15.56 | Ariel_ | 941 |
00:15.58 | franck | Ariel_: pointer on config? |
00:16.09 | effape | ok cool cheers |
00:16.13 | shmaltz | ARiel_, nat support who wins? |
00:16.17 | Ariel_ | franck, the enum web site has samples rules |
00:16.22 | Ariel_ | sipura |
00:16.27 | Ariel_ | hands down |
00:16.32 | shmaltz | Ariel_, ease of use, who wins? |
00:16.40 | shmaltz | Ariel_, feel and look? who wins? |
00:16.41 | franck | Ariel_: e168.org? |
00:17.05 | *** join/#asterisk websae (n=websae@CPE-24-167-204-30.wi.res.rr.com) |
00:17.07 | Ariel_ | sipura all around. But the 9112 looks more like a phone. But the sipura 941 has it beat. |
00:17.33 | websae | Ariel_: you like sipura? |
00:17.33 | Ariel_ | the sipura looks like a Cisco phone |
00:17.36 | websae | I have the 841 |
00:17.40 | shmaltz | Ariel_, i have been having massive problems lately with sipura, so I'm not sure I want to try them again, what do you think? |
00:17.47 | Ariel_ | 841 is ok but it not like the 941 |
00:17.57 | websae | what's good about the 941? |
00:18.00 | effape | hmm does this indicate i don't have it dialparties.agi: callerid = unknown / |
00:18.03 | Ariel_ | lots |
00:18.24 | *** part/#asterisk korihor (n=humberto@200.35.210.134) |
00:18.40 | Ariel_ | shmaltz, what problems? the 841 was in my view there first try and was not up to part. Better then the GS but not by much |
00:18.51 | websae | what's a good phone for a main call center location (secretary) with multiple lines...inexpensively speaking..? |
00:19.02 | Ariel_ | the 941 is a different phone all together the speaker even works great on it. |
00:19.15 | Ariel_ | Polycom 501 best all around phone for the price |
00:19.23 | justinu | yeah |
00:19.25 | websae | multi-line? |
00:19.26 | argentas | effape: does it work if you say usecallerid=no ? |
00:19.29 | justinu | that's still basically true |
00:19.32 | mog_home | polycom rocks |
00:19.35 | Ariel_ | Polycom, polycom |
00:19.35 | mog_home | i love that phone |
00:19.36 | effape | let me give it a go |
00:19.45 | justinu | i love my ip601 |
00:19.45 | mog_home | i am gonna get a 501 for home |
00:19.50 | justinu | get a 601 |
00:19.56 | websae | i need one with 6 lines |
00:20.03 | justinu | 601 = 6 lines |
00:20.11 | Ariel_ | 601 are great as well but for the overall price the 501 is better value |
00:20.28 | Equinox | How are the 301's? |
00:20.56 | Ariel_ | 301 and 300 from polycom are cheap and work great but are really cheap |
00:20.56 | justinu | ariel: can you use 501 with CDP? |
00:21.11 | Ariel_ | OK what is cdp |
00:21.19 | mog_home | what does 601 have that 501 doesnt |
00:21.24 | justinu | cisco's verion of PoE |
00:21.29 | Ariel_ | poe, web screen |
00:21.40 | Ariel_ | 601 can add the side cars |
00:21.42 | mog_home | 501 has poe i thought |
00:21.52 | Ariel_ | 501 yes but only with there cable |
00:21.55 | justinu | mog_home: you need some stupid cable that costs 40 bucks |
00:22.02 | wunderkin | has anyone here worked with sphinx? |
00:22.05 | mog_home | meh |
00:22.09 | justinu | yeah, meh |
00:22.11 | mog_home | i thought 501 could do xml |
00:22.12 | justinu | that's why I got a 601 |
00:22.21 | Ariel_ | only for a directory |
00:22.22 | justinu | i have a 24 port dlink PoE switch at home |
00:22.29 | mog_home | how much is 601? |
00:22.30 | Ariel_ | nice justinu |
00:22.34 | justinu | 250, i think |
00:22.36 | Ariel_ | less the 300 |
00:22.39 | mog_home | damn |
00:22.42 | mog_home | thats 100 more |
00:22.50 | wunderkin | were thoses mehs to me? heh |
00:23.00 | Ariel_ | yes it's more that is why I say 501 is best for the price |
00:23.01 | justinu | if you can get a 501 for 150, that's a smokin deal |
00:23.08 | mog_home | ebay |
00:23.12 | mog_home | i saw some a while ago |
00:23.15 | mog_home | at that price |
00:23.16 | Ariel_ | new for 171 |
00:23.27 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
00:23.48 | Equinox | New from voipsupply mine was about 200 |
00:24.03 | justinu | voipsupply is a rip |
00:24.09 | justinu | try atacomm.com |
00:24.09 | mog_home | yeah |
00:24.19 | mog_home | i need to find a friend at polycom |
00:24.23 | mog_home | see if one can fall of a truck |
00:24.26 | mog_home | around my house |
00:24.36 | justinu | i found the phone to be worth the money |
00:24.38 | msw | those trucks |
00:24.39 | Ariel_ | voipsupply is not a rip. give them a call |
00:24.52 | justinu | voipsupply's prices are high...... |
00:24.56 | msw | always things falling out of them |
00:24.58 | justinu | thats fact |
00:25.02 | Ariel_ | justinu, on the web give them a call |
00:25.06 | mog_home | yup msw |
00:25.14 | justinu | i'd rather not talk to anyone :P |
00:25.17 | justinu | i'm in telecom |
00:25.36 | Ariel_ | ok so your in telecom and don't want to talk... |
00:25.52 | Ariel_ | justinu, they can set you up with var prices on the phone then you can order via the web. |
00:25.57 | justinu | ok |
00:26.00 | justinu | i'm just teasing |
00:26.23 | Ariel_ | ask cory he is always on list |
00:26.30 | wunderkin | but if one falls off of a truck thats another story |
00:26.40 | mog_home | well i hate talking them |
00:28.35 | *** join/#asterisk iq (n=iq@71-38-78-98.omah.qwest.net) |
00:28.50 | *** join/#asterisk _ke4qqq (n=ke4qqq@srv.fgp.com) |
00:29.16 | GarryH | Any one using Snom 360? Is it a good phone for business use? |
00:29.28 | justinu | it's ok |
00:29.43 | justinu | it has it's strong points and weak points |
00:30.01 | mog_home | i loved mine |
00:30.05 | mog_home | till the screen broke |
00:30.07 | Ariel_ | the 360 is a good phone but I feel it's over priced. |
00:30.09 | GarryH | Justinu: What are the weak points? |
00:30.13 | mog_home | but it was a "truck phone" |
00:30.22 | mog_home | but its a good phone |
00:30.34 | justinu | weak points: sound quality, funky interface, odd design most people don't seem to like |
00:30.50 | *** join/#asterisk effape (n=nick@81.5.150.54) |
00:30.57 | Ariel_ | in my view for buz go with Polycom.... can't go wrong with them 501 and above |
00:30.59 | effape | sorry connection issues again |
00:31.05 | effape | it still disconnects |
00:31.16 | mog_home | yeah 501 is cool as shit |
00:31.16 | justinu | i'd have to agree with Ariel_ |
00:31.20 | mog_home | i liked snom 190 |
00:31.23 | mog_home | and 320 |
00:31.24 | mog_home | as well |
00:31.31 | effape | Exception on 23, channel 3 / Got event Polarity Reversal(17) on channel 3 (index 0) / Hangup due to Reverse Polarity on channel 3 |
00:31.38 | Ariel_ | too light feel like plastic hell but they work |
00:31.41 | mog_home | i still wish someone made a phone equal in ergonamics to 7960 |
00:31.43 | effape | Which appears to be caused by picking up the sip phone |
00:31.55 | argentas | effape: mind if i pm with you? |
00:32.12 | effape | sure if you tell me how (new to irc) |
00:32.16 | Ariel_ | mog_home, do you know any settings for tdm400 and UK??? |
00:32.29 | *** join/#asterisk Aughey (n=jha@ns1.washucsc.org) |
00:32.29 | justinu | what kind of trunk? |
00:32.30 | mog_home | there is a setting for hangup detect with bt |
00:32.35 | mog_home | but it shouldnt matter |
00:32.37 | justinu | some BT stuff is DASS2! |
00:32.39 | mog_home | and there is some cid stuff |
00:32.45 | GarryH | Justinu: We are going to upgrade all office phones from Nortel Meridian to new SIP phones. We are considering cisco 7960, polycom601 and snom 360. How your guys think? |
00:32.57 | justinu | garryh: polycom 601, no doubt. |
00:32.57 | mog_home | id go polycom 601 |
00:33.06 | Ariel_ | 601 polycom, polycom |
00:33.08 | mog_home | but 7960 has best ergonamic feel |
00:33.14 | mog_home | polycom close second |
00:33.17 | effape | how do i reply? |
00:33.18 | Ariel_ | 7960G only not plain |
00:33.20 | mog_home | and polycom has most features |
00:33.36 | Ariel_ | there are two 7960 and 7960g |
00:33.49 | argentas | effape: don't worry, just try the suggested config.. |
00:33.59 | shmaltz | GarryH, have you looked at citel.com? |
00:34.01 | effape | hehe yeah that's what i had |
00:34.03 | justinu | polycom is just gonna cost you 100 bucks a phone more |
00:34.10 | argentas | ah, ok |
00:34.12 | Ariel_ | G is best.. But I still like the Polycom better due to you get to upgrade the sip for free and don't have to spend extra for sip |
00:34.16 | justinu | s/polycom/cisco/ |
00:34.27 | effape | i'll give it a go with all the other stuff out |
00:34.34 | mog_home | how is that true? |
00:34.45 | shmaltz | Plycom is the best LAN phone out there |
00:34.52 | mog_home | how much is a 7960g with sip? |
00:34.57 | justinu | 300 |
00:35.05 | justinu | at the lowest |
00:35.14 | mog_home | so how is that 100 bucks more for a polycom? |
00:35.15 | shmaltz | mog_home, if you have to ask you cant afford it |
00:35.21 | mog_home | lol |
00:35.28 | justinu | then you need a 802.3af -> CDP converter |
00:35.30 | mog_home | i only want 2 phones at home |
00:35.31 | justinu | if you want poE |
00:35.33 | mog_home | i want a polycom |
00:35.34 | justinu | those are 20 bucks each |
00:35.36 | mog_home | and a wisip |
00:35.48 | mog_home | but really waiting on the wiax |
00:35.54 | mog_home | but no one is making one yet |
00:35.57 | Ariel_ | you have to pay 150 for the sip lic |
00:36.22 | GarryH | Mog_home: The Meridian system one good feature: Each phone has 5 line buttons bridged to all five pstn lines. Is this feature supported by *? |
00:36.28 | Ariel_ | Cisco don't come with sip licence, it's extra |
00:36.35 | mog_home | not really GarryH |
00:36.40 | mog_home | asterisk is not a keysystem |
00:36.43 | Ariel_ | gambolputty, no |
00:36.47 | mog_home | but you can make it like one if you want |
00:36.55 | mog_home | you can have line apperances |
00:37.05 | Ariel_ | hint |
00:37.15 | argentas | effape: did you say it was a TDM400 you were using? |
00:37.19 | mog_home | and there is stuff in bug tracker to do really cool stuff with hints |
00:37.22 | effape | yeah |
00:37.23 | mog_home | that should be in tree soon |
00:37.30 | GarryH | mog_home: How? what's line apperances? |
00:37.49 | mog_home | its a way for you to see the status of a device |
00:37.58 | mog_home | so if bob is on the phone |
00:38.02 | mog_home | a polycom 601 can have a light light up |
00:38.06 | franck | mog_home: look at www.planet.com.tw in their Internet Telephony products |
00:38.08 | mog_home | to tell you |
00:38.10 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-156-168-59.mia.bellsouth.net) |
00:38.15 | GarryH | shmaltz: The Citel is too expensive. Not worth |
00:38.23 | effape | brb |
00:38.29 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-156-168-59.mia.bellsouth.net) |
00:38.37 | shmaltz | GarryH, I'm not so sure you are right |
00:39.04 | *** join/#asterisk xtr-II (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
00:39.07 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
00:39.11 | shmaltz | GarryH, consider this: |
00:39.13 | shmaltz | ~$200 * 24 phones for a polycom 501, vs $125 * 24 for citel |
00:39.25 | mog_home | what you want me to see franck |
00:39.53 | franck | mog_home: 2FXS to sip adapters with PoE |
00:39.57 | GarryH | shmaltz: We only have 17 meridian phones. |
00:40.17 | *** part/#asterisk websae (n=websae@CPE-24-167-204-30.wi.res.rr.com) |
00:40.24 | mog_home | i just want to put stuff in my house |
00:40.35 | Ariel_ | it's best if you can replace all phones instead of using part from one type then from others. |
00:40.36 | shmaltz | GarryH, well one citel box = $3000, 17 * 200 = $3400 |
00:40.43 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
00:41.16 | Ariel_ | wow missed extra point.. Game is getting good... |
00:41.19 | hugo-v6 | gd morning |
00:41.38 | Ariel_ | morning wow, I am sure glad it's not morning yet |
00:41.52 | GarryH | mog_home: Are you talking about the hint feature in *? |
00:42.10 | mog_home | yes hints are how you trigger a line appearence |
00:42.24 | hugo-v6 | Ariel_: 1h42 am here ;)= |
00:42.35 | Ariel_ | hugo-v6, nice |
00:43.31 | *** join/#asterisk a1fa (n=a1fa@24.144.50.173) |
00:43.34 | a1fa | yo yo yo |
00:43.38 | hugo-v6 | Ariel_: well then u still have a few hours |
00:43.44 | GarryH | mog_home: As I know hint can only indicate the status of a channel by button led, but if I press that button, can I pick up the call? |
00:43.47 | a1fa | what is running on port: 2727? |
00:44.13 | a1fa | udp: 2727, 4520, and 4569? |
00:44.21 | a1fa | per asterisk conf? |
00:45.23 | ravenpi | How do I have an call ring multiple extensions simultaneously? |
00:45.37 | hugo-v6 | q: if i call someone which is busy, i get forbidden on my sip-phone. is there a way to tell the phone that the other side is busy? |
00:46.00 | *** join/#asterisk Soul (n=Soul@87-196-39-131.net.novis.pt) |
00:46.18 | hugo-v6 | ravenpi: dial(foo|bar|bam works here) |
00:46.26 | ravenpi | hugo-v6: is this a local (SIP to SIP) call, or are you going through the PSTN? |
00:46.44 | a1fa | has anybody tried voice-changer? |
00:46.45 | hugo-v6 | ravenpi: through pstn |
00:47.05 | *** join/#asterisk effape (n=nick@81.5.150.54) |
00:47.58 | effape | still doesn't work. When i take out hanguponpolarityswitch=yes it won't disconnect on pickup (but still registers a polarity switch) however it won't disconnect at all now. |
00:48.23 | mog_home | that part you config on phone GarryH |
00:48.30 | effape | So if i call and don't answer it will still run though the rules and won't hangup |
00:49.47 | a1fa | i dont want to fuck with my production asterisk |
00:49.54 | a1fa | has anybody tried the voice changer yet? |
00:50.59 | effape | there must be some way to make this work ;) |
00:51.25 | effape | It also takes about 5 rings before it gets to the sip |
00:51.35 | effape | which is kinda anoying too |
00:52.16 | effape | don't suppose you have anymore insight? |
00:53.02 | argentas | one sec, just having another look for my old configs.. |
00:53.09 | effape | excellent ta |
00:53.11 | Trazz | Ariel_, i installed the servercd.. is there a config command to run through the servername, dns, iip adress, routing still? |
00:54.28 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
00:55.52 | ke4qqq | anyone have experience integrating toshiba strata systems with asterisk?? |
00:55.56 | Ariel_ | Trazz, ifconfig |
00:56.14 | shmaltz | ke4qqq, yes I do |
00:56.22 | Ariel_ | ke4qqq, take toshiba sell on ebay works for me |
00:56.50 | justinu | ke4qqq: what's happening? |
00:56.54 | shmaltz | ke4qqq, you take the toshiba and use it as a door stopeer until you leave the room where you install asterisk, on the way out you take the toshiba with you and dump it |
00:57.03 | Trazz | Ariel_ ok i can do that.. do you have a cheat sheet you use now to get asterisk on the box and up and running now? |
00:57.19 | ke4qqq | lol.....I wish I could....however they don't want to replace all at once.....want to slowly migrate.... |
00:57.45 | ke4qqq | justinu: just having problems deciphering what little documentation there is with this system... |
00:57.50 | shmaltz | ke4qqq, so you can do the simple |
00:58.01 | shmaltz | does it support a T1 card? |
00:58.02 | msw | ke4qqq: that your callsign? |
00:58.09 | ke4qqq | yes |
00:58.28 | shmaltz | ke4qqq, then you buy a dual span t1 card for your asterisk box, |
00:58.35 | msw | ke4qqq: suppose you asked for it? (they didn't do that when I got my ticket) |
00:58.36 | shmaltz | ke4qqq, and a channel bank |
00:59.00 | shmaltz | put asterisk like this: |
00:59.02 | shmaltz | pstn <channel bank> t1 < asterisk > t1<toshib> |
00:59.12 | shmaltz | unlesss they use a PRI in which case it would look like this: |
00:59.23 | ke4qqq | schmaltz. already have a t1 communicating across...my big problem is telling the toshiba which ports are tie lines and then that certain extension are on asterisk.... |
00:59.24 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
00:59.25 | shmaltz | PSTN/PRI <asteirsk> t1 <toshiba> |
00:59.58 | shmaltz | ke4qqq, it's not a problem at all, configure the moved extensions as remote extensions and have asterisk capture it |
01:00.06 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
01:00.12 | shmaltz | remote meaning off net |
01:00.22 | shmaltz | every pbx I have worked with supports this |
01:00.23 | argentas | effape: don't seem to have a copy of my old configs anymore - sorry. This thread might help: http://www.voipuser.org/forum_topic_2743.html |
01:00.35 | shmaltz | so I'm sure toshiba does as well, unless it's not a bpx |
01:01.12 | shmaltz | ke4qqq, you got the idea? |
01:01.15 | ke4qqq | yeah it does....at least my interpretation of their weird documentation says so.... |
01:02.12 | shmaltz | gtg |
01:02.14 | shmaltz | c yas |
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01:18.59 | effape | hi ho, me again :0 |
01:19.12 | effape | I'm banging my head against the wall now :) |
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01:20.23 | effape | I just don't understand why it's getting the polarity reversal - that's what seems to be causing the problem.... Why would SIP/104-4c94 answered Zap/3-1 ... Took Zap/3-1 off hook ... Exception on 21, channel 3 ... Got event Polarity Reversal(17) on channel 3 (index 0) ... Hangup due to Reverse Polarity on channel 3 |
01:20.25 | effape | happen? |
01:21.03 | *** part/#asterisk Ironmask (n=joe@n120s119.bbr1.shentel.net) |
01:21.16 | effape | isn't the polarity to do with the zap channel? Why would the sip answer cause it to happen |
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01:29.58 | argentas | effape: I can only assume that the SIP answer is causing the zap channel to be answered, and perhaps the telco are then reversing the polarity at this point (causing the channel to be disconnected) |
01:30.22 | *** part/#asterisk rabar (n=putteepu@cpe-68-175-40-207.nyc.res.rr.com) |
01:31.09 | argentas | did you see the URL I posted for you? |
01:31.33 | gambolputty3 | anyone use the IAXPEER function? |
01:32.29 | *** join/#asterisk harry8 (n=harryyeh@Z-f1-0-0-90-S1.gw3.van1.rogerstelecom.net) |
01:32.42 | harry8 | does callerid not work on FXO ports? |
01:32.47 | harry8 | analog that is |
01:33.03 | harry8 | i have a TDM400P |
01:33.26 | brookshire | it should :) |
01:33.46 | harry8 | hmm |
01:33.49 | effape | hmm |
01:33.58 | harry8 | http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
01:34.10 | harry8 | it says it doesn't for analog lines but that doesn't make sense |
01:34.23 | harry8 | i have callerid analog phones and that receives caller id |
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01:41.07 | bn-7bc | pleace excuse all the bs i bosted her earlyer, it seem rhat asterisk only captures the i\first key in the keymap (found that out the hard way) so ehen i har * for atxfer and *2 for atomon non of them worked, is tis a known issue in v1.2.1? |
01:41.20 | bn-7bc | and is ther a warkaround |
01:42.00 | bn-7bc | that was posted not bosted |
01:43.40 | mog_home | ???? |
01:43.53 | mog_home | is your keyboard jammed? |
01:45.31 | bn-7bc | no fingers in a tangle |
01:45.48 | bn-7bc | 2:45 i shold be a sleap now |
01:46.53 | JunK-Y | sleep(x); |
01:47.30 | Ikarus | bn-7bc.sleep(60*60*8); |
01:47.31 | bn-7bc | it seem that asterisk only captures the first key in the keymap (found that out the hard way) so when I hare * for atxfer and *2 for atomon non of them worked, is this a known issue in v1.2.1? |
01:47.49 | bn-7bc | that shold be mouch clearer |
01:48.49 | justinu | le vrai moyen n'a pas besoin de sommeil |
01:48.54 | bn-7bc | Ikarus: yep in a few minutes |
01:50.40 | bn-7bc | well is it a known problem or is my setup just scrowed? |
01:51.05 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
01:51.49 | bn-7bc | ok it's late I'll ask again in aprox 5hrs |
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01:56.46 | *** mode/#asterisk [+o drumkilla] by ChanServ |
01:57.18 | dily | please, can anyone explane me what is this warning |
01:57.19 | dily | Jan 16 02:53:17 WARNING[4013]: chan_sip.c:2523 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8) |
01:57.39 | *** join/#asterisk GD_ (n=GD@ppp74-adsl-147.ath.forthnet.gr) |
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02:00.34 | harry8 | has anybody here had problems with Callerid for TDM400P FXO analog? |
02:00.48 | Qwell | harry8: only when the line doesn't support cid |
02:01.15 | harry8 | hmm |
02:01.23 | harry8 | is there anything special that you have to setup? |
02:01.32 | Qwell | callerid=asreceived |
02:01.42 | harry8 | in zapata.conf? |
02:02.00 | Qwell | I always get the two confused. whichever is in /etc/asterisk/, I believe |
02:02.56 | harry8 | yes |
02:03.01 | harry8 | zaptel.conf is in /etc |
02:03.04 | harry8 | :0 |
02:05.16 | justinu | zaptel.conf is just for phsyical line stuff |
02:05.28 | justinu | zapata.conf is where you put most of the signalling related things |
02:06.16 | justinu | i've taken a crash cource in PRI/CAS circuit turnup with zaptel hardware in the past week |
02:09.01 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
02:11.17 | GD_ | hello... could someone tell me what groups the asterisk user should participate in (apart from dialout), in order to run asterisk as asterisk:asterisk? |
02:11.42 | *** join/#asterisk aditya-bdg^_^ (i=EvilInLo@ws1.bratatex.melsa.net.id) |
02:12.28 | brookshire | that should work |
02:12.54 | GD_ | there's something wrong with starting asterisk from init.d and I believe it must have sth to do with permissions and stuff... |
02:12.56 | brookshire | just have to make sure all the right directories have the correct permissions. |
02:13.18 | GD_ | well i've done a chown -R asterisk:asterisk /etc/asterisk... what else should I do? |
02:13.24 | brookshire | like /var/log/asterisk |
02:13.30 | GD_ | the log files' permissions are ok as well |
02:13.41 | *** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it) |
02:13.46 | brookshire | what's the error? can you pastebin it? |
02:13.49 | GD_ | well.. |
02:13.59 | dily | can onyone explane me this WARNING: Jan 16 03:13:18 WARNING[4491]: chan_sip.c:2523 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8) |
02:14.00 | GD_ | there are no errors if i start asterisk from the command line |
02:14.13 | GD_ | but if I start it from init.d |
02:14.14 | dily | PLEASE |
02:14.31 | GD_ | my sip phone won't ring on incoming calls |
02:14.39 | GD_ | although it can dialout... |
02:14.44 | brookshire | even if you run asterisk as that user? |
02:14.51 | brookshire | like su - asterisk |
02:15.00 | brookshire | asterisk -vvvvvvvvvcg |
02:15.07 | GD_ | yes.. i've got the same problems if I do asterisk -U asterisk -G asterisk -vvvvvvvvvvvvvvvvvc |
02:15.15 | GD_ | only asterisk -vvvvvvvvvvvvc works fine |
02:15.46 | ast_new-B | hi all |
02:16.13 | GD_ | su-ing is the same are using the -U and -G options right? |
02:16.20 | GD_ | are=as |
02:16.24 | ast_new-B | can anyone tellme howto setup call limit by duration talk, extention, local & long distance call in dialplan ? |
02:18.16 | GD_ | local&long distance is relatively easy... use a pattern matching scheme to limit a user... i believe that's fairly well documented at www.voip-info.org |
02:19.42 | GD_ | brookshire can you think of anyway to debug this? the only thing I find different as far as asterisk's startup messages is concerned is a WARNING[7111]: db.c:47 dbinit: Unable to open Asterisk database |
02:19.54 | GD_ | as well as a WARNING[7111]: pbx_wilcalu.c:70 autodial: Autodial: Unable to open file message |
02:20.02 | h3x | except you need an area code shitlist |
02:20.21 | GD_ | i googled a bit about the asterisk database thing but to no avail... |
02:20.58 | brookshire | i dunno.. i got it working once |
02:21.05 | brookshire | but i haven't played with it since |
02:21.12 | brookshire | it must be something simple |
02:21.38 | brookshire | but if you can't get it working.. you can always try chroot :) |
02:21.38 | GD_ | thanks anyway.. i'll let u know if i find anything :-) |
02:21.54 | ast_new-B | thanks GD_, but how can grouped extentions eg : 110,111,112 they can call to long distance and another limited ? |
02:22.02 | ast_new-B | and how about call durations ? |
02:22.30 | GD_ | i can't be of much help for the call duration thing |
02:23.14 | GD_ | I thought you wanted to totally ban long distance calls... |
02:23.15 | h3x | show application dial tells you how to limit duration |
02:23.35 | h3x | rtfclh |
02:24.00 | Luke-Jr | wtf :( |
02:24.03 | Luke-Jr | my PAP2-NA died ? |
02:24.09 | ast_new-B | ok h3x and All thanks, i'll try it |
02:27.27 | Luke-Jr | Any idea on recovering a deadish PAP2-NA? ethernet/blue & red lights are dead |
02:28.57 | Luke-Jr | argh... ditto for router now too, this is not my night -.- |
02:29.08 | Aughey | Anyone have recommendations on dealing with needing 8 FXO ports. Options are two TDM04B cards or a 8-port FXO Analog VoIP gateway. (or any other options) |
02:29.19 | h3x | you shouldnt have connected that lightning rod on your roof to your ethernet network :D |
02:31.30 | Luke-Jr | I didn't |
02:31.38 | *** join/#asterisk alf (n=alfredwo@dsl-202-173-191-109.qld.westnet.com.au) |
02:32.10 | Luke-Jr | so any ideas on fixing a PAP2-NA? |
02:32.34 | h3x | heh. you are screwed. |
02:33.11 | Luke-Jr | ... -.- |
02:33.15 | Luke-Jr | wrong answer |
02:33.27 | h3x | no blinkey lights, no workie |
02:33.38 | GD_ | brookshire: I followed instructions on http://www.voip-info.org/wiki-Asterisk+non-root and it works like a charm now! it was easy indeed but I didn't know there were so many directories asterisk needs write access to... |
02:33.42 | alf | anyone knows a way to make the snom phone leds to show you how many busy lines you have at any moment? thanks! |
02:35.00 | brookshire | gd: awesome :) |
02:35.13 | alf | anyone knows a way to make the snom phone leds to show you how many busy lines you have at any moment? thanks! |
02:35.24 | Luke-Jr | h3x: as I said, two LEDs are lit |
02:35.55 | GD_ | now if only I could get voipbuster to work with IAX :-P |
02:36.32 | ast_new-B | i try to put some function like this exten => 9, GotoIf(${$(Callerid=110,111,112)100?101}) |
02:36.48 | ast_new-B | but why it only match for ext 110 |
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02:41.05 | ravenpi | Anyone know how to turn off the occasional MWI "warble" on Polycom? (And, no, not the stutter dialtone -- I know how to nix that.) Thanks! |
02:44.43 | khemir | how start a message when the user pick up the phone? |
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03:05.21 | strtok | hi |
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03:09.05 | SkramX | Hiya |
03:09.33 | warthog | having trouble with udev after yum on centos, zaptel devices not there, readded the rules and permissions but still do not have /dev/zap dir, all channels in asterisk do not work after yum update of udev. anyone have any ideas on this? |
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03:10.55 | Demo_G | Hello, every body |
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03:11.31 | warthog | not to active at the moment. |
03:11.34 | wunderkin | any sphinx gurus yet? hehe. :D |
03:11.40 | SkramX | oh god |
03:15.18 | warthog | if I follow the instruction in README.udev for zaptel on a 2.6 kernel distro like centos 4.2, will the /dev/zap directory exist on reboot? |
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03:18.15 | zavala | greeings all... |
03:18.46 | zavala | I've come for a bit of wisdom from those who know all things SCCP... anyone? |
03:19.32 | warthog | is that skinny? |
03:19.39 | zavala | yeah... |
03:20.02 | zavala | I've got the damnded problem... and it's getting the best of me.. |
03:20.04 | warthog | I heard that cisco was now supporting SIP with more vigour |
03:20.29 | zavala | true, and I can make Asterisk sing with SIP... the problem is that I need to use the 7914 sidecar |
03:20.46 | zavala | and that can only be used right now with Skinny (SCCP) |
03:21.20 | zavala | that's not the issue though... it's the blasted phone... I've got a 7960 phone with the SCCP image on it and it registers with * just fine |
03:21.34 | warthog | sorry, I know nothing about skinny, but noone is talking right now, so I though I would.... |
03:21.49 | warthog | you know much about udev issues? |
03:22.04 | zavala | what issue are you having? |
03:23.42 | warthog | after yum update in centos 4.2, none of my channels work in head. looks like udev, I did the README.udev in /usr/src/zaptel but I still get an error about /dev/zap during startup and all sterisk channels do not work still. |
03:24.24 | zavala | have you done a modprobe zaptel to see if it's there? |
03:24.42 | warthog | yeah, driver and 24xxx card load no problem |
03:25.17 | warthog | when I dial and extention via sip, iax or zap, I get nothing... |
03:26.13 | zavala | by extention you mean an external interface on like the PRI? |
03:26.23 | zavala | or do you mean even SIP -> SIP calls fail? |
03:26.35 | warthog | even SIP -> SIP call fail. |
03:27.16 | Trazz | Ariel, still there? |
03:27.39 | zavala | hmm that goes beyond the /dev/zap driver... that looks like a configuration issue.. |
03:28.11 | zavala | zap is only needed if you're going to be using PRI based cards (or zaptel based card).... or for timing for things like meetme and moh |
03:28.20 | zavala | in which case you can compile ztdummy |
03:28.21 | *** join/#asterisk welles (n=welles@222.90.141.49) |
03:28.22 | Trazz | zavala, i just installed fedora core 4 from scratch and want to install asterisk properly. which files do i need to go grab and install? |
03:28.30 | *** join/#asterisk wellng (n=welles@222.90.141.49) |
03:28.31 | warthog | I was not sure if timer would do all this. |
03:28.40 | *** part/#asterisk wellng (n=welles@222.90.141.49) |
03:28.43 | zavala | no, timer will only effect meetme |
03:28.43 | argentas | anyone using backports.org zaptel on Debian sarge? |
03:28.49 | welles | hi all |
03:29.15 | warthog | looks like I will be doing a reinstall.... |
03:29.21 | zavala | trazz, you'll need the base package for sure.... |
03:29.37 | zavala | hello wellls |
03:29.52 | *** join/#asterisk goh (n=goh@60.49.6.190) |
03:30.07 | Trazz | i am going to ftp this one up then asterisk-1.2.1.tar.gz |
03:30.20 | zavala | the zaptel package is for FXO/FXS based cards and the libpri is for... well, pri.. |
03:30.36 | Trazz | ok i dont have any cards in it |
03:30.56 | zavala | then you will just need the base, the sounds, and addons if you want to play.. |
03:31.15 | Trazz | ok do i need to have all of those at the time i compile? |
03:31.31 | zavala | if you plan on wanting to do conference with the 'meetme' package you'll need to download the zaptel package and compile ztdummy |
03:31.43 | CoffeeIV | I am looking for recommendations for a VoIP origination service to so that I can recieve faxes over VoIP. Any experiences ? |
03:32.05 | Trazz | ok great.. let me get started :) thanks |
03:33.02 | zavala | trazz, you'll just need to have the base package to start withi |
03:33.08 | warthog | zavala, if I want to reinstall all asterisk packages, can I get away with just removing all asterisk stuff from /usr/src and rm the /var/lib/asterisk and the spool/asterisk dirs? then download and compile again! |
03:33.26 | Trazz | i have the base now and just gunzip and untard |
03:33.34 | zavala | warthog, that will work |
03:33.42 | goh | I would like to implement voip tht can bridge existing PBX in multiple branch. what hardware are recommented? |
03:33.54 | warthog | thanks, sorry I don't know about skinny to be of assistance |
03:34.02 | zavala | warthog, np |
03:34.16 | zavala | goh, how are the remote sites connected back today? private PRI/TI? |
03:34.38 | zavala | coffeeIV, you want to fax over voip? |
03:34.39 | goh | broadband |
03:34.39 | Trazz | zavala, is there a good cheat sheet to use to get this compiled, configured and up and running? |
03:34.56 | zavala | goh, so they are VoIP pbx's already? |
03:35.33 | goh | there are got PSTN line I just want to have a few trunk that go to IP bridge another PBX. |
03:36.06 | CoffeeIV | zavala: yes, just receiving for now |
03:36.09 | goh | so that branch call branch is free |
03:36.15 | zavala | Trazz, if you want to install asterisk all in one directory (other then the rood of your filesystem) then you'll need to edit Makefile and set INSTALL_PREFIX to something |
03:36.25 | goh | no they dun have IP-PBX yet. |
03:36.26 | welles | hi zavala |
03:36.46 | Trazz | yes i would prefer that.. |
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03:37.18 | goh | zavala, do you think ATA device work able in this situation? |
03:37.20 | zavala | just make a directory like /opt/voip or something like that and then set your INSTALL_PREFIX to be that directory in the Makefile.. |
03:37.31 | Trazz | okay thats easy enough |
03:37.37 | zavala | goh, ATA devices will only trunk one call at a time... |
03:37.44 | Luke-Jr | Any idea on recovering a deadish PAP2-NA? ethernet/blue & red lights are dead |
03:37.58 | zavala | if you need multiple calls trunked together you're looking at real hardware on both ends |
03:38.12 | goh | ic |
03:38.37 | zavala | you could have miltiple ATA devices, but then you reach a point where you might as well just impliment a server |
03:38.42 | welles | hi zavala , |
03:38.46 | zavala | goh, do your remote sites have a PBX today? |
03:38.50 | zavala | he wells |
03:39.00 | zavala | sorry I missed your first hello |
03:39.01 | welles | i have problems |
03:39.03 | goh | got pbx already. |
03:39.21 | zavala | do the PBX's have T1 cards? |
03:39.25 | goh | that why I think use back exsiting pbx. |
03:39.25 | zavala | what's up wells? |
03:39.42 | goh | I the HQ got T1 but not the branch |
03:39.43 | welles | zavala, it seems that there is a bug in asterisk1.2.1 's meetme |
03:40.01 | zavala | what are you experiencing? |
03:40.05 | Corydon76-home | Have you reported it on the bugtracker, yet? |
03:40.15 | welles | zavala, i can not use ilbc as the codec |
03:40.38 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
03:40.41 | welles | zavala, the voice became very bad |
03:40.45 | drumkilla | welles: try the 1.2 branch from svn |
03:40.47 | goh | if implement IP-PBX then they will got tow PBX in the office. |
03:40.48 | drumkilla | it should be fixed there |
03:41.04 | goh | do all branch also need IP-PBX in this situation? |
03:41.53 | zavala | welles, some good advice from Corydon and drumkilla... look through the bug list already... and try the SVN 1.2.1 branch |
03:41.58 | ast_new-B | can anyone tell me what best hardware [CPU] recomendation for about 100 phone line extentions |
03:42.10 | welles | zavala, ok |
03:42.17 | zavala | goh, it depends on what you want to do... if you're looking for toll bypass, then yes |
03:42.28 | Corydon76-home | Not 1.2.1.... 1.2... |
03:42.36 | zavala | ahh... point taken |
03:42.48 | Corydon76-home | It's the tree that will become 1.2.2... and 1.2.3 after that, etc... |
03:43.13 | zavala | ast_new-B... good question... I know that some of the Dell hardware has some conflicts with chipsets... other then that, it depends on what you're going to be doing.. |
03:43.26 | goh | zavala, they just want to make internal call though VOIP. |
03:43.33 | *** part/#asterisk khemir (i=khemir@200.56.189.30) |
03:44.21 | zavala | goh, unless each branch location has a large volume of people... this is going to be expensive all for the sake of someone being able to say "my branch calls are over VoIP" |
03:44.23 | goh | What the diff between IP-PBX with the VOIP gatewap? |
03:45.10 | ast_new-B | my scenario is replacing our PBX |
03:45.25 | zavala | and IP based PBX generally means that the PRI gets terminated into a hard(or soft) PBX and then from the PBX to the handset is all done over TCP/IP |
03:45.30 | ast_new-B | and what;s good hardware are recomended |
03:46.05 | zavala | rock solid with room to grow, do Dell 2850 or Compaq DL380... |
03:46.25 | a1fa | has anybody tried the voice changer yet? |
03:46.30 | a1fa | grr |
03:46.37 | a1fa | ctrl+v |
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03:46.40 | CoffeeIV | anyone have any experience using asterisk with voipxpress.com, particularly with receiving faxes ? |
03:46.49 | zavala | a VoIP Gateway generally means a device that's going to take calls destined for a certain route and convert them to VoIP... |
03:47.11 | zavala | a VoIP gateway can also be used to do backhaul voice between two sites... |
03:47.12 | *** join/#asterisk mina (n=mina@modemcable175.94-70-69.mc.videotron.ca) |
03:48.04 | a1fa | VoIP Gateway =~ ATA |
03:48.18 | ast_new-B | zavala : how about bandwidth if it;s will connect to another asterisk server ? |
03:48.46 | goh | actually I got 37 branch tat need to call back the HQ everyday. That is costly. so how many IP-PBX need to location? do I IP PBX at HQ will work able? |
03:49.02 | a1fa | 37 voips |
03:49.08 | a1fa | 37 sip or iax accounts 1 pbx |
03:49.12 | zavala | keep in mind though that routing calls over the internet is a 'best effort' approach... |
03:50.07 | a1fa | goh: pay me 3k in cash |
03:50.08 | ast_new-B | yes, but how about bandwidth consuming for all codecs ? |
03:50.11 | a1fa | and i got you hooked up :P |
03:50.18 | zavala | their is no quality of service for commodity internet... so if a path is long, or someone backhoes through a line... forget about phone calls |
03:50.33 | *** join/#asterisk bmg505 (n=leon@c1-175-7.rndf.isadsl.co.za) |
03:50.42 | Equinox | zavala - If you go with same provider at both ends you can get a QoS guarantee. |
03:50.43 | goh | a1fa, why? |
03:50.54 | zavala | bandwith for codes depends on what you want and what you think people can live with others sounding like.. |
03:50.55 | Equinox | XO Communications gives me 60ms nationwide full rate node to node. |
03:51.01 | zavala | default is 64K per channel |
03:51.12 | jaike | g729 works excellent for us |
03:51.29 | jaike | make sure your internet connection is ok though |
03:51.41 | zavala | you can get latecy garantee's... which is what XO is promissing... |
03:51.58 | goh | internet should be ADSL |
03:52.00 | zavala | you can't get QoS guarantees unless you're doing MPLS or ATM |
03:52.04 | Trazz | zavala, should i use the make samples ? |
03:52.20 | zavala | yes, make samples is your sample config files to learn from |
03:52.22 | Equinox | Never much used MPLS. |
03:52.27 | goh | it's QOS is workable if all terminal implement it? |
03:52.45 | Equinox | goh- Problem is you generally don't control the middle of the equation ;) |
03:53.04 | zavala | equinox is right... |
03:53.14 | Equinox | But I've had good luck with most providers. |
03:53.21 | zavala | with XO you have some favor if your XO all over as all of your traffic will be on XO's private lines |
03:53.29 | Equinox | It's when you hop provider to provider and end up going through a peering point 2 states away you get owned. |
03:53.32 | zavala | same with Sprint |
03:53.46 | Equinox | Cogent is good too if it's available. |
03:53.48 | ast_new-B | so 64K in my country costly 200$ in a month |
03:54.11 | zavala | wow... you in latin america? |
03:54.30 | ast_new-B | so if 20 simulatanous phone will consume 20X64K |
03:54.35 | jaike | ast: thats expensive |
03:54.40 | jaike | where u at? |
03:54.44 | ast_new-B | indonesia |
03:54.45 | Math` | I've a client who uses voip over VSAT, works pretty fine |
03:54.58 | zavala | drumkilla... Corydon... you still on? |
03:54.59 | jaike | ast: im in the philippines..asterisk works great for us |
03:55.08 | Math` | there's delay, but no echo and no cutting-sound |
03:55.09 | ast_new-B | that;s very expensive .... |
03:55.16 | zavala | anyone with SCCP Skinny knowledge? |
03:55.45 | jaike | we have an E1 line...$950..can push 50-60 g729 channels through it |
03:56.09 | zavala | like I said... 64K is the top limit... you can go down to as low as 7k |
03:56.26 | ast_new-B | maybe i have to use g729 with 8L |
03:56.30 | ast_new-B | maybe i have to use g729 with 8K BW |
03:56.44 | jaike | ast: make sure latency is 250ms and below...and number hops to a minimum |
03:57.10 | Trazz | zavala, got that all done.. where should i dive in next ? |
03:57.21 | Trazz | i didnt start any processes yet .. |
03:57.24 | ast_new-B | jaike: how about quality if there's 20 simultan in/out phone ? |
03:57.25 | zavala | do you have phones? |
03:57.35 | Trazz | softphone |
03:58.06 | ast_new-B | jaike: is there any delay or echo tail ? |
03:58.10 | zavala | then you should start with the sip.conf and the extentions.conf |
03:58.24 | jaike | make sure you have a good iax provider in the US....on an athlon 64 2800 with 2Gb ram, even 50 channels is no sweat |
03:58.30 | zavala | echo isn't a result of latency on the line |
03:58.41 | zavala | you get echo the closer you are in physical locality, etc |
03:58.50 | jaike | the biggest problem we had was choppy lines, which was due to latency and packet losst |
03:58.53 | jaike | loss |
03:59.04 | jaike | once we solved that part, everything went smooth |
03:59.37 | zavala | for an IAX provider in the US I'd look at Jeremy's NuFone... or VoicePulse.. |
03:59.50 | jaike | we have multiple asterisk servers running |
04:00.18 | jaike | also dont limit yourself to iax providers..asterisk does sip pretty well too..as long as youve public ips |
04:00.19 | Math` | packet loss is the big problem, latency doesnt really matter as long as there is no echo |
04:00.22 | ast_new-B | jaike: for how many users your asterisk serve ? |
04:00.29 | zavala | anybody have exp. with Skinny (SCCP) and Cisco phones? |
04:00.36 | jaike | were a call center...150 seats |
04:00.38 | Math` | zavala just ask |
04:00.41 | Luke-Jr | Any idea on recovering a deadish PAP2-NA? ethernet/blue & red lights are dead |
04:00.53 | jaike | expanding to 500 by the end of the year...all on asterisk |
04:00.54 | jaike | :) |
04:01.42 | Trazz | zavala, i reviewed those two files and they have tons of stuff in them already enabled |
04:01.44 | zavala | I've building a SCCP setup and the 7960 will register with *, but when I try to make a phone call it only takes the first number dialed and then returns a fast busy |
04:02.15 | zavala | from * I can issue 'dial 3000' and it rings the SCCP 7960... |
04:02.26 | zavala | but I can't call anything from the 7960 |
04:02.27 | ast_new-B | jaike: that's heavy asterisk server |
04:02.36 | jaike | ast: actually we have multiple asterisk servers |
04:02.46 | jaike | all interconnected |
04:02.55 | zavala | I figured it was in a dialplan somehwere... but I've been snooping and SCCP doesn't pickup the dialplan.xml and beyond that I have nowhere left to look... |
04:03.09 | zavala | Math' any ideas? |
04:03.24 | Math` | whats the output ob the cli |
04:03.28 | Math` | on |
04:03.49 | zavala | with debug on it shows that the phone only transmitted one number.. |
04:03.55 | zavala | so I know it's not * dropping it |
04:04.16 | zavala | no debug it looks like: |
04:04.22 | zavala | <PROTECTED> |
04:04.27 | zavala | <PROTECTED> |
04:04.30 | ast_new-B | jaike: i'm really in newbie, so i have a lot of questions but i'll first reading about voip |
04:04.46 | ast_new-B | and asterisk the miracle |
04:04.46 | zavala | when in reality I dialed 8500 and hit 'dial' |
04:05.09 | jaike | ast: took me a while to get the hang of it...been doing asterisk for almost a year now |
04:05.18 | jaike | but its exciting |
04:05.50 | *** join/#asterisk centrix (n=centrix@dhcp148.wireless.fiber.dcdi.net) |
04:06.03 | ast_new-B | jaike: is there any problem with asterisk running now ? |
04:06.35 | Katty | hi. |
04:06.39 | zavala | hello |
04:07.19 | jaike | ast: 1.0.* was buggy before..but 1.2.* seems pretty stable |
04:07.35 | Math` | zavala: did you set the proper context in skinny.conf? |
04:08.23 | jaike | zavala: experienced the same thing with our polycoms...send the number before the complete number was dialed |
04:08.38 | Math` | with polycoms? |
04:08.38 | zavala | I'm using chan_sccp |
04:08.46 | jaike | had to disable digitmapping...dunno if 7960 has something like it |
04:08.47 | zavala | jaike: what as the resolution? |
04:09.11 | msw | when you set the digitmap properly in the polycom, it works great |
04:09.12 | Math` | jaike: skinny has its name for a reason, its very very lite |
04:09.13 | jaike | now...you have to press send to dial..so you can key in as many numbers as you like |
04:09.19 | msw | the default digitmap is no good though |
04:09.32 | Math` | no digit map, nothing, just sends digits 1 by 1 to the sccp server |
04:09.57 | jaike | ive not experience with skinny..sorry |
04:10.01 | zavala | trazz.... cant pm, not registered |
04:10.01 | jaike | no |
04:10.33 | zavala | yeah... math' .... if I lead off with an * or a # it will take multi-digits (even though they go nowhere)... if I lead of with a number, any number, I only get to dial one |
04:10.37 | Trazz | zavala, simple do /msg nickserv resgister somepassword |
04:10.53 | welles | zavala, thanks very much |
04:11.04 | zavala | no problem wells |
04:11.12 | welles | zavala, the problems fixed |
04:11.22 | *** join/#asterisk _SwM_ (n=admin@digitaldatabits.net) |
04:12.38 | a1fa | omfg |
04:12.39 | a1fa | ;) |
04:12.43 | a1fa | everybody stop for a second |
04:13.35 | zavala | so what's the rush a1fa? |
04:13.44 | jaike | ? |
04:14.52 | a1fa | :P |
04:14.55 | a1fa | two things |
04:15.07 | a1fa | anybody tried the voice changer patch on asterisk 1.2.1? |
04:15.10 | a1fa | 2nd thing, |
04:15.19 | *** join/#asterisk Jammy (i=jammy@CPE0008740429bc-CM001404df6f46.cpe.net.cable.rogers.com) |
04:15.19 | a1fa | udp: 2727, 4520, and 4569? |
04:15.29 | a1fa | whats on 2727 4520, 4569 |
04:16.21 | chet | ~docs |
04:16.25 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
04:16.50 | jaike | isnt 4569 iax2 |
04:16.51 | jaike | ? |
04:19.01 | Luke-Jr | Any idea on recovering a deadish PAP2-NA? ethernet/blue & red lights are dead |
04:19.22 | zavala | lukee-jr: I have no idea what you're talking about |
04:19.37 | a1fa | yeah |
04:19.41 | a1fa | 4569 is iax |
04:19.46 | a1fa | but, what is 2727? |
04:19.47 | Jammy | hey guys, got a quick question, I always get this error parsing a certain section of my zapata.conf file 'Ouch ... error while writing audio data: : Broken pipe' cant seem to figure out what keeps causing it to break at that point.. from the point on, it tries to configure the internal zap fxs |
04:19.52 | a1fa | ~2727 |
04:19.56 | a1fa | ~4520 |
04:20.01 | a1fa | i am trying to patch this server down |
04:20.18 | jaike | firewalls huh |
04:20.26 | Jammy | hmmm..heh.. nvr mind |
04:20.28 | Jammy | fixed itself |
04:20.51 | jaike | 2727 i think is mgcp |
04:21.12 | jaike | not sure though |
04:22.15 | a1fa | can i disable mgcp? |
04:22.28 | a1fa | i had somebody trying to bruteforce my asterisk |
04:22.39 | Math` | unload chan_mgcp.so |
04:23.01 | Math` | or add a noload directive for chan_mgcp.so in modules.conf |
04:23.23 | zavala | bingo noload=chan_mgcp |
04:24.35 | a1fa | what does mgcp do? |
04:24.42 | remmo | rtmf |
04:24.53 | a1fa | ufcs |
04:24.59 | *** part/#asterisk warthog (n=nvadekar@69.17.198.58) |
04:25.00 | a1fa | ufcsdttrtmf |
04:25.03 | a1fa | get it? |
04:25.09 | remmo | yes |
04:25.11 | remmo | did you? |
04:25.16 | a1fa | yes |
04:25.22 | zavala | no, seriously... some level of rtfm has to be done here.. |
04:25.26 | remmo | so then you worked it out |
04:25.44 | a1fa | ah |
04:25.46 | remmo | can someone tie my shoe laces? my arms are too short |
04:25.51 | a1fa | no dude |
04:25.57 | a1fa | your iq is below your shoe size |
04:26.07 | a1fa | its ok, daddy will hook you up |
04:26.10 | a1fa | lay back |
04:26.19 | remmo | maybe i should bend over for you? |
04:26.30 | a1fa | no, i aint that kind of daddy |
04:26.54 | zavala | oh give me a break... mgcp -- media gateway control protocol |
04:27.11 | a1fa | yeah |
04:27.14 | a1fa | i just read it |
04:27.18 | a1fa | i dont need that |
04:27.35 | zavala | if you don't know what it's for, your clearly can't miss it when you turn it of then can you? |
04:27.40 | remmo | anyone using vierling gsm gateways here? |
04:28.09 | a1fa | what is on port 4520? |
04:28.24 | a1fa | there is nothing in documentation about it |
04:28.59 | remmo | maybe dundi |
04:29.13 | a1fa | 2727 is mgcp |
04:29.24 | remmo | DUNDI |
04:30.01 | goh | if using leased line connection do I able connect 37 branch together with any lake? |
04:30.08 | a1fa | yup |
04:30.10 | a1fa | its dundi |
04:30.22 | a1fa | so chan_dundi? |
04:31.12 | zavala | it's not a channel driver.. |
04:31.24 | zavala | just comment it out in the dundi.conf |
04:31.41 | a1fa | :* |
04:32.18 | zavala | however, after having just looked at the conf.. it's on port 4520 by default |
04:32.25 | a1fa | everything is commented out |
04:32.34 | rajiv | exten => 500,2,VoiceMailMain(s${CALLERIDNUM}) results in callers hearing 'goodbye' then hangup. why might that be? |
04:32.36 | a1fa | but it still binds to the port |
04:33.00 | a1fa | rajiv: what version? ${CALLERIDNUM} is not used anymore |
04:33.10 | a1fa | ${CALLERIDNUM} is obsolete |
04:33.12 | rajiv | 1.0.10 |
04:33.17 | a1fa | well maybe not |
04:33.26 | a1fa | when did they take off ${CALLERIDNUM}? |
04:33.29 | a1fa | 1.2.0 tree? |
04:34.05 | rajiv | the variable is working correctly as i see in the console debug: Executing VoiceMailMain("SIP/110-38ff", "s110") |
04:34.17 | a1fa | cool |
04:34.27 | a1fa | its obsolete in 1.2.x |
04:34.36 | a1fa | do you have voicemail.conf configured |
04:35.04 | Trazz | zavala, can i force the zaptel to get installed in /opt/voip too ? |
04:35.31 | zavala | yes you can... however it's best to just let it install in the usual places as it needs to modify the kernel, etc |
04:35.43 | rajiv | a1fa: http://pastebin.com/507712 is the logs ... voicemailmain() with no args works just fine so i doubt it is a voicemail.conf issue |
04:36.48 | a1fa | i see |
04:37.02 | a1fa | beats me.. it could be a bug in 1.0.10 |
04:38.03 | Trazz | i would rather have it install in that same directory .. what do i need to modify for that zavala? |
04:38.35 | zavala | one last request.. and then I'll stop beating the horse and leave.... anyone have any working knowledge of the Cisco SCCP image on cisco phones? |
04:39.20 | a1fa | zavala: fyi : pbx_dundi is the module |
04:39.47 | zavala | a1fa, thanks |
04:39.54 | a1fa | np :p |
04:40.05 | a1fa | zavala: run netstat -nap | grep asterisk |
04:40.16 | a1fa | and paste to pastebin.ca |
04:41.15 | zavala | tcp 0 0 10.40.6.40:2000 10.40.15.51:51241 ESTABLISHED18716/asterisk |
04:41.15 | zavala | udp 0 0 0.0.0.0:2727 0.0.0.0:* 18716/asterisk |
04:41.15 | zavala | udp 0 0 0.0.0.0:4520 0.0.0.0:* 18716/asterisk |
04:41.16 | zavala | udp 0 0 0.0.0.0:5060 0.0.0.0:* 18716/asterisk |
04:41.18 | zavala | udp 0 0 0.0.0.0:4569 0.0.0.0:* 18716/asterisk |
04:41.20 | Math` | pastebin!!! |
04:41.20 | zavala | unix 2 [ ACC ] STREAM LISTENING 46606 18716/asterisk /opt/voip/var/run/asterisk.ctl |
04:41.21 | Math` | pastebin!!! |
04:41.23 | zavala | doh |
04:41.28 | a1fa | yeah man |
04:41.31 | Math` | :P |
04:41.38 | a1fa | do you really need that much exposure |
04:42.02 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
04:42.03 | a1fa | Jan 10 09:57:21 NOTICE[29465] chan_sip.c: Registration from '<sip:90093@213.249.97.123>' failed for '202.164.44.75' - Username/auth name misma |
04:42.11 | a1fa | some guy kept trying to brake in |
04:42.45 | a1fa | there is like half a million entries |
04:43.07 | a1fa | too bad my passwords are uber hashed + md5 stored |
04:43.09 | Math` | iptables -A INPUT -s 213.249.97.123 -j DROP |
04:43.13 | a1fa | no |
04:43.22 | a1fa | iptables -A INPUT -j DROP |
04:43.25 | Math` | lol |
04:43.44 | Luke-Jr | Any idea on recovering a PAP2-NA? ethernet/blue & power/red lights are lit solid |
04:43.49 | rajiv | wacky. if i put a wait(1) in between answer() and voicemailmain(s${CALLERIDNUM}) then things works properly |
04:43.49 | a1fa | $IPTABLES -A INPUT -p tcp -j REJECT --reject-with tcp-reset |
04:43.49 | a1fa | $IPTABLES -A INPUT -j REJECT |
04:44.02 | a1fa | thats the way to go man |
04:44.16 | Math` | yeah except -j REJECT uses your bandwidth if you get dos'd |
04:44.24 | a1fa | well true |
04:44.36 | a1fa | gotta keep it clean tho |
04:44.52 | a1fa | so it dont show up in port scanning as "ports filtred" |
04:45.11 | a1fa | :P |
04:45.30 | a1fa | btw, his ip changed 10 times |
04:45.36 | a1fa | different sources/subnets |
04:45.42 | a1fa | so i only allowed my subnet |
04:47.17 | a1fa | smart way to do it |
04:47.21 | a1fa | also allowedguest=no |
04:47.30 | *** join/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net) |
04:47.34 | a1fa | i suggest everybody set that to no |
04:48.34 | zavala | a1fa if you want to get technical about it... stand the SER server up infront of asterisk and then make SIP users authenticate to an LDAP or Kerberos backend |
04:49.09 | a1fa | zavala: not a bad idea |
04:49.16 | a1fa | sLDAP rather |
04:49.19 | zavala | yeah |
04:49.21 | zavala | :_D |
04:49.39 | a1fa | i am running my pbx from internet colo |
04:50.02 | a1fa | time=35.8 ms |
04:50.05 | zavala | ahhh... there is a recipe out there somewhere to stand up SER infront of asaterisk on the same box |
04:50.05 | a1fa | to my sip peer |
04:50.10 | zavala | nice times though |
04:50.19 | zavala | ;-) |
04:50.41 | a1fa | thats only my asterisk->sip provider |
04:50.46 | a1fa | but my home->asterisk |
04:50.47 | a1fa | 19ms |
04:50.48 | a1fa | :P |
04:50.50 | a1fa | its still slow |
04:50.50 | zavala | bugger.. this sccp thing is going to drive me up the damn wall... I've been looking for an answer for the past three hours |
04:51.01 | zavala | what's total time? |
04:51.07 | zavala | end to end? |
04:51.53 | a1fa | dunno |
04:51.55 | a1fa | bout 50s |
04:51.57 | a1fa | 50ms |
04:52.09 | a1fa | freaking broadvoice |
04:52.13 | a1fa | their shit is always slow tho |
04:52.18 | zavala | hahaa... |
04:52.24 | zavala | ever tried VoicePulse? |
04:52.57 | a1fa | i've heard about them |
04:53.02 | a1fa | but they dont offer the plan i need |
04:53.07 | a1fa | free western europe :P |
04:53.12 | a1fa | for $19 |
04:53.21 | zavala | ahh... nope... broadvoice it is.. |
04:53.57 | a1fa | <PROTECTED> |
04:54.04 | a1fa | is the fastest gateway they have |
04:54.09 | a1fa | and it is 35ms |
04:54.13 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-59-48.dsl.tul2ok.sbcglobal.net) |
04:54.19 | Math` | its not that bad |
04:55.49 | *** join/#asterisk [hC] (n=hardcore@S0106000e9b96114f.vf.shawcable.net) |
04:56.13 | a1fa | yeah |
04:56.22 | a1fa | i need to send a SIP phone to bosnjia |
04:56.25 | a1fa | Bosnia |
04:56.32 | a1fa | i wonder if they will have problems connecting to my SIP |
04:56.41 | a1fa | since they will be getting 100ms round trip |
04:56.51 | zavala | ouch.... |
04:57.19 | Math` | I've a VSAT SIP phone working #1 |
04:57.20 | *** join/#asterisk liew (n=liew@60.49.6.190) |
04:57.26 | Math` | as long as you got no packet loss, you're ok |
04:57.31 | Math` | the round trip is about 500ms |
04:57.46 | a1fa | i hope not |
04:57.52 | Luke-Jr | Is there a decent priced open source replacement for a PAP2-NA? |
04:57.53 | a1fa | udp is not very reliable :P |
04:58.02 | liew | may I know who is from malaysia that able help me implement asterisk for my client? |
04:58.08 | a1fa | Luke-Jr: i bought a PAP2-NA for $50!! |
04:58.17 | Luke-Jr | a1fa: where? |
04:58.22 | Math` | I bought a mediatrix 2102 for 35$cad (ebay:P) |
04:58.23 | a1fa | www.voipsupply.com |
04:58.29 | Math` | (worth 150$) |
04:58.32 | Luke-Jr | a1fa: $60 there... |
04:58.40 | a1fa | well $10 more will not kill you |
04:58.51 | a1fa | go to OfficeDepot and pickup PAP2 |
04:58.52 | a1fa | and hack it |
04:58.54 | Luke-Jr | a1fa: well, I was hoping to find something with open firmware |
04:59.01 | a1fa | hm! |
04:59.05 | a1fa | there is no such thing :P |
04:59.08 | Luke-Jr | :\ |
04:59.10 | Luke-Jr | why not? |
04:59.17 | *** join/#asterisk goh (n=goh@60.49.6.190) |
04:59.20 | a1fa | Luke-Jr: asterisk is open :P |
05:00.00 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
05:00.08 | Luke-Jr | a1fa: but it's not a low-power device, nor cheap for ATA abilities |
05:00.48 | a1fa | well IaxY |
05:00.49 | a1fa | then |
05:00.57 | a1fa | get iaxy |
05:01.01 | a1fa | its open source |
05:01.03 | wunderkin | pa168 phones or something like that |
05:01.07 | h3x | it aint open source |
05:01.11 | a1fa | it aint? |
05:01.12 | h3x | no |
05:01.14 | a1fa | bastards |
05:01.16 | h3x | the fuckin firmware is binary |
05:01.47 | zavala | on the other side of the coin, everybody's got to put food on the table some how |
05:01.49 | a1fa | suX |
05:02.05 | a1fa | liew: how much are you paying for remote assistance? |
05:02.20 | Math` | they get money for the hardware, whats the use of not opening the platform |
05:02.25 | a1fa | paypal that $$$ upfront.. and you get remote administration |
05:02.55 | liew | a1fa, where is ur stay right now? I want local people for assistance |
05:03.08 | liew | a1fa, ar u own a company |
05:03.13 | zavala | because there is nothing really special in the iaxy other then how they impliment the code |
05:03.57 | a1fa | liew: yes.. |
05:04.10 | a1fa | liew: You get remote assitance.. and you can watch what I am doing, and learn |
05:04.11 | liew | ar you from Malaysia? |
05:04.14 | a1fa | no |
05:04.22 | a1fa | :P |
05:04.56 | liew | that mean I still need to configure all the thing until can remote access |
05:06.25 | a1fa | money upfront :P |
05:10.36 | *** join/#asterisk mkl1525 (n=daniel@212.80.239.117) |
05:13.57 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
05:15.47 | zavala | for just some simple sip to sip phone calls inside of one network.. it shouldn't take you more then 30 minuts to an hour after you read the docs |
05:17.45 | *** join/#asterisk jcrock7 (n=jared@ool-44c1bb21.dyn.optonline.net) |
05:20.16 | a1fa | hm |
05:20.21 | a1fa | anybody used that dialpad editor |
05:20.26 | a1fa | java editor |
05:22.34 | a1fa | Luke-Jr: http://www.tomsnetworking.com/Sections-article153.php |
05:22.37 | a1fa | check this shit out man |
05:22.41 | a1fa | just what you were looking for |
05:22.44 | a1fa | embedded pbx |
05:23.26 | *** join/#asterisk jcrock7 (n=jared@ool-44c1bb21.dyn.optonline.net) |
05:31.34 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
05:34.07 | *** join/#asterisk wellng (n=welles@222.90.141.49) |
05:34.46 | Math` | thats nice |
05:35.25 | *** join/#asterisk cyber (n=kani@220.247.245.2) |
05:38.24 | *** join/#asterisk slan (n=lba@user-12lml5g.cable.mindspring.com) |
05:42.39 | a1fa | yah |
05:42.43 | a1fa | too weak tho |
05:42.56 | a1fa | cant do anything with that bad boy but make 1 call at the time |
05:42.56 | a1fa | :P |
05:43.08 | jbroome | you need a cluster of them! :) |
05:43.10 | a1fa | good if you have 400 of these |
05:43.17 | a1fa | thats what i was about to say |
05:43.25 | a1fa | AsterCluster |
05:43.26 | Math` | you can go up to 6 calls in ulaw |
05:43.31 | Math` | if you read the article thru :P |
05:43.34 | a1fa | i did |
05:43.50 | Luke-Jr | Are PAP2s currently unlockable? |
05:44.01 | a1fa | Luke-Jr: http://www.tomsnetworking.com/Sections-article153.php |
05:44.02 | a1fa | yes |
05:44.09 | a1fa | they are unlockable |
05:44.34 | Luke-Jr | what's the current method? :) |
05:44.41 | Math` | its on the wiki |
05:44.48 | Luke-Jr | which wiki? |
05:44.51 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
05:44.59 | a1fa | voip-info.org |
05:45.02 | Math` | http://www.voip-info.org/wiki/view/Linksys+PAP2+Unlocking+Methods |
05:45.16 | *** join/#asterisk |omni| (n=rob@net98.limelyte.net) |
05:45.56 | *** join/#asterisk GiRL[23] (i=SeRDaR@server.ivinskis.kursenai.lm.lt) |
05:46.36 | *** join/#asterisk Fire (n=Lady_Han@server.ivinskis.kursenai.lm.lt) |
05:46.38 | a1fa | http://www.voip-info.org/wiki/view/Linksys+PAP2+Unlocking+Methods |
05:48.02 | a1fa | night |
05:48.31 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
05:49.19 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
05:49.42 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
05:49.53 | Netgeeks | digium mail lists down? |
05:50.58 | Netgeeks | wow, quiet here too, I knew the aliens were gonna come and abduct all the asterisk people |
05:51.22 | *** join/#asterisk yeawsing (i=yeawsing@218.50.182.187) |
05:52.26 | yeawsing | Hi I have a question about License Fee for Asterisk. Anyone able to help me out |
05:52.42 | Trazz | if i am going to use pure sip and no hardware how do i get teh extensions configured ? |
05:54.49 | Math` | in sip.conf |
05:54.58 | Math` | yeawsing: whats the question |
05:55.10 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
05:56.08 | yeawsing | If I use the Asterisk Server for commercial use, do i need to buy the license $995 from digium. |
05:56.10 | Trazz | Math, what do i need in there? I have the x-lite up and running and logged in but cant dial anyting because i didnt configure any extensions i dont think |
05:57.22 | Math` | Trazz: well configure them |
05:57.24 | Math` | yeawsing: no |
05:57.29 | Trazz | i get this on the cli |
05:57.29 | Trazz | Jan 16 00:09:19 NOTICE[5415]: pbx.c:1731 pbx_extension_helper: Cannot find extension context 'internal' |
05:57.57 | Math` | Trazz: you need to define extensions contexts, then point your SIP devices to that context |
05:58.00 | Qwell | Trazz: Then the context internal doesn't exist |
05:58.30 | *** part/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
05:58.35 | Math` | yeawsing: you can use asterisk for any purpose without having to buy Asterisk Business Edition from digium |
05:59.01 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
05:59.55 | *** join/#asterisk Netgeeks (n=root@68-185-24-8.static.mdfd.or.charter.com) |
06:00.27 | yeawsing | ok thanks Math |
06:01.32 | *** join/#asterisk pengyong (n=lala@222.188.141.234) |
06:01.47 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
06:02.59 | yeawsing | If I need more 120 simultaneous call, do i need to purchase the License. |
06:03.18 | Netgeeks | no |
06:03.27 | Math` | yeawsing: *ANY* purpose :P |
06:03.45 | Netgeeks | you need to purchase the license if you want to have digium provide you software support |
06:03.52 | Math` | you do need to purchase licenses for g729 channels (at 10$/channel) |
06:04.11 | justinu | yikes |
06:04.12 | yeawsing | I just want to make sure that I do the proper way. |
06:04.15 | Math` | digium is gonna give support anyways... just going to charge you per-hour |
06:04.25 | justinu | what does digium charge/hr? |
06:04.51 | Math` | I have no idea |
06:05.45 | Math` | In addition to the free installation support for Digium hardware, Digium provides full support for the entire Asterisk software suite, including hourly rates with no commitments. For more information on commercial Asterisk support or any other Digium professional service, please contact sales@digium.com or call us toll free at 877-LINUX-ME (877-546-8963). |
06:06.57 | yeawsing | When I search the Digium License, the FAQ and brochures answer is not specified or very open. |
06:08.13 | yeawsing | So let say I want to purchase the License, is that mean $995 is just per/server or I can install many server using SINGLE license |
06:08.22 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
06:09.25 | yeawsing | I thought Digium Sales team is here. |
06:10.00 | Math` | you should contact sales directly |
06:10.22 | yeawsing | ya. I guess so. Thanks Math, appreciate that. |
06:10.29 | Netgeeks | it's currently sunday evening around midnight at digium. Even if some of their sales team hangs out here, I doubt they are here now |
06:10.40 | yeawsing | ha.. |
06:10.42 | justinu | wussies |
06:11.07 | Netgeeks | lol |
06:11.49 | Math` | lol |
06:14.21 | yeawsing | If I want to be a service provider using Asterisk to make PC to PSTN call, what do I need and who should I connect to? |
06:14.43 | Qwell | you need knowledge of telephony |
06:14.50 | Qwell | in other words...you'd know the answer to that question |
06:15.42 | justinu | lol |
06:16.09 | yeawsing | When I read on the net, they mention about IXC. |
06:16.46 | justinu | call someone like at&t and order a PRi |
06:17.49 | yeawsing | So r u guys connecting to at&t,sprint or MCI |
06:18.04 | justinu | none of the above |
06:18.47 | yeawsing | So how can we make cheap long distance call? |
06:19.02 | justinu | you need money |
06:19.08 | yeawsing | :) |
06:19.19 | justinu | credito |
06:19.33 | justinu | probably $50,000 |
06:20.02 | Luke-Jr | yeawsing: Voipjet or Voxee seem good |
06:20.05 | yeawsing | is that for buying Hardware or buying time. |
06:20.11 | justinu | buying time |
06:20.13 | Luke-Jr | yeawsing: Voipjet is 1.3 cents/min or Voxee is 1.1 cents/min |
06:20.14 | justinu | hardware is cheap |
06:20.39 | justinu | you need a rock solid IP link |
06:20.40 | Luke-Jr | yeawsing: but they're both VoIP, so you'll need bandwidth too |
06:20.47 | yeawsing | I've check with VoIPjet, but I still not sure about their call quality |
06:20.58 | Math` | voipjet is great |
06:21.13 | justinu | i'm talking about rates 1/4 of that |
06:21.18 | justinu | maybe 1/6th |
06:21.19 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
06:21.32 | Luke-Jr | justinu: where? |
06:21.37 | yeawsing | It seem to be VoIPjet is very popular. |
06:21.48 | justinu | wholesale, level3 or global crossing |
06:22.05 | Math` | justinu: IP or PRI? |
06:22.21 | justinu | ip |
06:22.41 | justinu | pri is a big PITA |
06:23.23 | Math` | and you say level3 can do 0.0025$/min to us/ca? |
06:24.25 | yeawsing | When u guys talking about Level3 is that mean Layer |
06:24.30 | Luke-Jr | no |
06:25.00 | yeawsing | Or just the H/W Level3,4 and 5 |
06:25.05 | Math` | Level3 is a company :P |
06:25.14 | Math` | www.level3.nety |
06:25.16 | Math` | www.level3.net |
06:25.27 | justinu | math: yeah, around that |
06:25.31 | yeawsing | oh. thanks for clear that up. |
06:26.20 | h3x | level3 costs a hell of a lot more than that |
06:26.23 | Math` | justinu: wow, contacting level3's sales dep tomorrow :P |
06:26.27 | h3x | unless you commit to 100 grand a month |
06:26.36 | justinu | so what's the problem? |
06:26.46 | yeawsing | wow |
06:27.24 | h3x | and it sucks coz it aint two way |
06:27.24 | justinu | eh? |
06:27.26 | Math` | eh? |
06:27.31 | h3x | inbound only |
06:27.32 | justinu | origination and termination are seperate |
06:27.35 | Math` | go on the website, it says both origination and termination |
06:27.39 | *** join/#asterisk dasuberdavid (n=david@pcp01534754pcs.huntsv01.al.comcast.net) |
06:27.41 | h3x | termination requires you to lock it to a device basically |
06:27.55 | h3x | and its # minutes per device/account |
06:27.56 | justinu | you use sip proxies |
06:28.06 | yeawsing | So if we want to connect to PSTN we can use VoIPjet. |
06:28.07 | justinu | i dunno where you get that |
06:28.24 | h3x | l3 has two different wholesale products |
06:28.33 | h3x | the one with inbound, outbound, and e911 is turnkey |
06:28.40 | h3x | like the one that packet 8 uses |
06:28.58 | h3x | but at that point all you are doing is pushing paper |
06:29.12 | justinu | oh, we're not doing that |
06:29.13 | h3x | their LI product is by the minute, but they dont have decent outbound termination to accompany itr |
06:29.27 | h3x | you may as well use long distance termination |
06:29.46 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
06:30.03 | yeawsing | Is VoIPjet can consider as long distance termination |
06:30.16 | h3x | voipjet illegally exports your calls to canada :P |
06:30.37 | h3x | grey market anyway |
06:30.46 | yeawsing | oh.. |
06:30.56 | h3x | thats why its so cheap |
06:31.13 | Math` | illegally export calls.... |
06:31.20 | Luke-Jr | illegally? |
06:31.20 | Math` | how come |
06:31.25 | yeawsing | but we can use it to make call using our Asterisk server |
06:31.29 | h3x | well its going to another country and coming back |
06:31.32 | Math` | yeawsing: yeah you can |
06:31.47 | Luke-Jr | h3x: nothing illegal there... |
06:31.57 | Math` | h3x: nothing illegal there but... where is it going? :P |
06:32.11 | h3x | well |
06:32.20 | h3x | it might be since voipjet dosent have a FCC 214 license |
06:32.21 | h3x | but its voip so |
06:32.33 | mishehu | and the mpaa says that you're illegally copying movies when you record them off of the tv. |
06:32.34 | Math` | voipjet is canadian |
06:32.37 | *** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com) |
06:32.42 | Luke-Jr | pfft, FCC |
06:32.43 | Math` | mishehu: lol |
06:32.46 | mishehu | illegal is in the eye of the beholder. |
06:32.56 | mishehu | or the gavel of the judge |
06:33.12 | h3x | they have equipment in the us |
06:33.12 | h3x | heh |
06:33.14 | Math` | and I can say nobody has the right to log the text I say without my express written consentement |
06:33.28 | Math` | and everyone here that is logging this channel is performing an illegal act passible of jail |
06:33.37 | yeawsing | ha |
06:33.38 | mishehu | Math`: you can say that, but enforce it? |
06:33.39 | mishehu | pssh |
06:33.43 | mishehu | I'm logging you anyway. |
06:33.45 | Math` | well... maybe :P |
06:33.46 | Luke-Jr | Math`: bs |
06:34.29 | mishehu | now, you do the Math` |
06:34.32 | mishehu | *rimshot* |
06:35.06 | Math` | lol |
06:35.47 | jaike | this channel is logged..i saw a page somewhere with all the messages here |
06:35.54 | yeawsing | Math what provider are u using if u don't mind I asking? |
06:36.08 | jaike | even read my old messages |
06:36.18 | Math` | yeawsing: for what? |
06:36.19 | justinu | there's some interesting drug talk in those logs |
06:36.20 | Math` | termination? dids? |
06:36.21 | Trazz | how do i give my x-lite two extensions so i can call myself? |
06:36.31 | Math` | Trazz: you can call your own extension |
06:36.39 | Math` | its just gonna ring on another line |
06:36.48 | Math` | tho its pointless |
06:36.51 | Trazz | ok |
06:37.33 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
06:41.21 | *** join/#asterisk Myk3 (i=Myk3@cpe-67-9-95-36.hot.res.rr.com) |
06:41.24 | Myk3 | hello all |
06:41.38 | Trazz | Math, i was trying to call myself to see how voicemail works |
06:41.44 | Trazz | it never picks up |
06:41.46 | Myk3 | i have 2 lines how can i configure each number to ring a different phone? |
06:42.19 | Math` | Trazz: well you need to configure your dialplan properly |
06:42.48 | Myk3 | when both numbers are dialed only one of the phones ring |
06:43.08 | Myk3 | can i set it up to ring different numbers for differenet phones? |
06:43.14 | Math` | DIal(TECH/ext1&TECH/ext2) |
06:43.15 | Myk3 | "useing softs to test" |
06:43.53 | Myk3 | any thoughts? |
06:45.03 | Myk3 | anyone? |
06:46.50 | Myk3 | ok there where people talking now there aint did i scar them away? |
06:46.54 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
06:48.08 | Trazz | Math, is voicemail and ivr built in and turned on by default ? |
06:48.26 | Math` | you need to tell asterisk everything... |
06:48.37 | Math` | it wont go to voicemail if you don't pickup automaticly |
06:48.40 | Math` | you need to make it go there |
06:48.55 | Math` | Dial(SIP/phone1|30) |
06:49.02 | Math` | VoiceMail(u1000) (for example) |
06:49.12 | Myk3 | how can i make asterisk got this phone for this number? |
06:49.15 | Trazz | ok |
06:51.05 | Trazz | so that would be exten => 244,2,Dial(SIP/sam,30) and exten => 244,2,VoiceMail(u1000) ? |
06:51.26 | Myk3 | anyone? |
06:51.32 | Myk3 | help please |
06:53.31 | yeawsing | Thanks guys. Talk to u guys later. |
06:53.34 | *** join/#asterisk liew (n=liew@60.49.6.190) |
06:53.45 | *** part/#asterisk yeawsing (i=yeawsing@218.50.182.187) |
06:54.35 | *** part/#asterisk liew (n=liew@60.49.6.190) |
06:54.50 | *** join/#asterisk goh (n=goh@60.49.6.190) |
06:55.05 | Trazz | Math, does that look correct? |
06:55.06 | Trazz | so that would be exten => 244,2,Dial(SIP/sam,30) and exten => 244,2,VoiceMail(u1000) ? |
07:00.15 | Qwell | Trazz: no, you can't have two of the same priority |
07:00.44 | Trazz | ok gotta make it 3 then |
07:02.46 | *** join/#asterisk welles (n=welles@61.150.60.123) |
07:05.32 | *** join/#asterisk meriad_ (i=Turtle@24.83.211.78) |
07:05.47 | *** part/#asterisk meriad_ (i=Turtle@24.83.211.78) |
07:06.48 | ast_new-B | somebody please help me with dialplan that limit call by duration, group ext that can dial in long distance or international ? |
07:09.01 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
07:10.06 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
07:10.21 | jahani | hi |
07:10.56 | jahani | from where i can get the codec g723.1 for asterisk ? |
07:12.02 | *** join/#asterisk bzbw (n=wlwzhang@ip-33-107-134-202.rev.dyxnet.com) |
07:12.32 | goh | for extension call to another extension within same asterisk pbx will audio path go though asterisk server? are the audio path go peer to peer. and only the sip state will go through asterisk server? |
07:12.46 | bzbw | has anyone connect * to a traditional PBX? |
07:13.06 | bzbw | this PBX require to dial 9 before getting to the outside line |
07:13.36 | bzbw | but even if I send a 9, the tranditional pbx seems just looping back to my *. |
07:13.42 | bzbw | it really puzzles me. |
07:15.02 | ast_new-B | try to set ignorepat => 9; |
07:15.07 | ast_new-B | in exentions.conf |
07:15.48 | *** join/#asterisk leto3 (n=l@car75-1-81-57-13-34.fbx.proxad.net) |
07:18.04 | *** join/#asterisk EriSan (n=erisan@151.8.109.90) |
07:20.03 | jahani | where i can find the codec g723.1 for asterisk ? |
07:21.13 | Qwell | jahani: legally? |
07:21.42 | Qwell | tip: nowhere |
07:22.42 | jahani | yes |
07:22.46 | jahani | legally |
07:23.40 | Trazz | Qwell, any idea why i get these errors? |
07:23.40 | Trazz | Jan 16 01:35:05 NOTICE[6347]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
07:23.40 | Trazz | Jan 16 01:35:05 WARNING[6347]: pbx.c:1690 pbx_extension_helper: No application 'VoiceMail2' for extension (macro-vmessage, s, 1) |
07:24.17 | ThaZZa_Work | hey all. |
07:24.19 | GarryH | exit |
07:24.22 | *** part/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net) |
07:24.51 | ThaZZa_Work | How do i fix this? Accepting UNAUTHENTICATED call from X.X.X.X: |
07:25.35 | Qwell | ThaZZa_Work: give the user a password |
07:25.58 | Qwell | Trazz: VoiceMail2 obviously doesn't exist, and something is wrong with your Dial line, or...simply, the phone isn't reachable |
07:26.32 | Qwell | jahani: please don't msg me |
07:26.35 | Qwell | and the answer is, you don't |
07:27.22 | ThaZZa_Work | Qwell: It is an * box connecting via iax to another * box. Should i have 2 register commands.. One at both ends? |
07:27.47 | jahani | so how can i use g723.1 with asterisk ? |
07:27.55 | Qwell | * boxes shouldn't really register with each other...make them static |
07:28.02 | Qwell | jahani: passthrough, and that's it |
07:28.23 | jahani | what u mean by passthrough ? |
07:28.51 | Qwell | asterisk doesn't decode/encode the audio, it just passes it through |
07:30.27 | jahani | i know but where i get the codec i buy buy from digium g729 but i need also g723.1 |
07:30.33 | ThaZZa_Work | Qwell: Can you refresh memory on where to look to make static? |
07:30.43 | Qwell | jahani: You don't |
07:30.56 | jahani | i don't ?? |
07:31.01 | Qwell | ThaZZa_Work: iax.conf, just give them passwords, and put in the IPs instead of host=dynamic |
07:31.09 | Qwell | jahani: for the third time, right |
07:31.28 | jahani | i don't understand what u mean by i don't !!! |
07:31.45 | Qwell | You don't use g723 with asterisk, unless it's done in passthrough |
07:32.15 | jahani | how to passthrough ? |
07:32.18 | ThaZZa_Work | Qwell: Cool thanks mate.. :D |
07:32.33 | Qwell | call another phone that does g723 |
07:34.10 | jahani | its not work |
07:34.26 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-118.claranet.co.uk) |
07:34.38 | jahani | it say commune client and gateway have no commun codec |
07:35.02 | Qwell | then g723 isn't an allowed codec |
07:35.11 | ThaZZa_Work | jahani: sounds to me like both phones dont' have a compatable set of codec's to use. |
07:35.25 | Qwell | ThaZZa_Work: You'd think that would be obvious |
07:35.30 | niZon | poor Qwell, bombarded with questions |
07:35.56 | ThaZZa_Work | Qwell: I saw it, and i am classifed as dumb. lol |
07:36.05 | *** join/#asterisk svenna_ (n=svenna@p548D09D9.dip0.t-ipconnect.de) |
07:36.17 | *** join/#asterisk zAmifage (n=g4l3ku5@c-67-187-20-28.hsd1.tx.comcast.net) |
07:36.25 | ThaZZa_Work | niZon: Poor Qwell earning his keep of being allow to get really drunk tonight.. ;-) |
07:37.09 | Poincare | If I want to do a Dial(SIP/user&ZAP/g1/1234), is it possible to set outgoing callerid only for the ZAP part? |
07:37.17 | jahani | ok |
07:37.25 | Qwell | Poincare: I don't think so |
07:37.34 | Poincare | bummer :-( |
07:37.39 | Qwell | but I don't know for certain |
07:37.43 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
07:38.18 | Qwell | Poincare: You might be able to dial a Local/ exten, which sets cid |
07:38.22 | tehdely | lilo: ---> zAmifage spamming in every chan |
07:38.52 | Poincare | Qwell: can i do a Dial(SIP/user&somelocalextension) then? |
07:39.09 | Qwell | so like, replace Zap/g1/1234 with Local/1234, and have like exten => 1234,1,Set(CALLERID(number)=12345) then 1234,2,Dial(Zap/g1/1234) |
07:39.12 | Qwell | That may work, but...ymmv |
07:39.33 | Poincare | i will try... |
07:39.39 | *** join/#asterisk Fire (i=user226@server.ivinskis.kursenai.lm.lt) |
07:39.49 | Poincare | the LOCAL has to be in the same context? |
07:40.00 | Qwell | You can do Local/1234@context |
07:40.10 | Poincare | ok, I'll let you know :-) |
07:41.17 | *** join/#asterisk Oden (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
07:45.13 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
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07:46.10 | zAmifage | Whoa look at this.. http://www.progenic.com/vote/?id=Galekus |
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07:46.19 | zAmifage | Check it out man! http://www.progenic.com/vote/?id=Galekus |
07:46.19 | Qwell | umm |
07:46.21 | zAmifage | http://www.progenic.com/vote/?id=Galekus |
07:46.23 | zAmifage | Hey vote for my site http://www.progenic.com/vote/?id=Galekus |
07:46.25 | zAmifage | Dude help me out a little and vote for my site.. will you? http://www.progenic.com/vote/?id=Galekus |
07:46.27 | zAmifage | Hey vote for my site http://www.progenic.com/vote/?id=Galekus |
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07:46.31 | zAmifage | http://www.progenic.com/vote/?id=Galekus |
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07:46.37 | zAmifage | Check it out man! http://www.progenic.com/vote/?id=Galekus |
07:46.38 | Qwell | time for a DDoS :P |
07:46.39 | zAmifage | Check it out man! http://www.progenic.com/vote/?id=Galekus |
07:46.41 | zAmifage | http://www.progenic.com/vote/?id=Galekus |
07:46.45 | zAmifage | Whoa look at this.. http://www.progenic.com/vote/?id=Galekus |
07:46.47 | zAmifage | Dude help me out a little and vote for my site.. will you? http://www.progenic.com/vote/?id=Galekus |
07:46.48 | MrChimpy | op me! |
07:46.49 | zAmifage | Check it out man! http://www.progenic.com/vote/?id=Galekus |
07:46.53 | zAmifage | Whoa look at this.. http://www.progenic.com/vote/?id=Galekus |
07:46.55 | zAmifage | Hey can you vote for my site? thanks http://www.progenic.com/vote/?id=Galekus |
07:46.57 | _vic | shut it down |
07:47.05 | _vic | yeah ! |
07:47.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
07:47.07 | MrChimpy | yeah. good |
07:47.41 | _vic | (only thinked. i have black magic 0=) |
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07:53.33 | [av]bani | =0 |
07:56.10 | Poincare | Qwell: it works! Thanks :-) Let me know when you're in the neighborhood, I'll buy you some drinks... |
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08:02.55 | ThaZZa_Work | anyone use festival? |
08:04.59 | Netgeeks | So, is the digium mail lists down? It's been awfully quiet in my inbox |
08:05.19 | Qwell | Netgeeks: would seem that way |
08:06.02 | Netgeeks | Glad to see my nightmare of a world conspiracy against me is once again delayed! |
08:09.13 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
08:09.23 | Qwell | Netgeeks: the conspirists took the weekend off for the holiday |
08:09.54 | MrChimpy | currently worrying about connecting TDM400 to PBX here. blowing stuff up would not be popular. |
08:10.14 | *** part/#asterisk jaike (n=a@203.131.137.76) |
08:10.28 | MrChimpy | floor ports are RJ45, need to go to RJ11 on the card |
08:11.04 | MrChimpy | i shall let our telephone dude worry about it |
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08:13.35 | justinu | Rj11's plug right into RJ45s, you know |
08:13.47 | Netgeeks | MrChimpy don't worry, the TDM400 card is safe, it's too cheap a component to be the one to blow up. Just ding around find the most expensive or most unavailable component, and it will be the one to go |
08:14.00 | Netgeeks | dig round |
08:14.07 | Netgeeks | around |
08:14.15 | Netgeeks | damn beer and fingers |
08:14.53 | Qwell | justinu: new cards are rj11 I hear |
08:15.08 | Netgeeks | Just because the rj11 will plug into a rj45 doesn't mean they are wired right |
08:16.51 | dippo_ | man i wish there was some way i could communicate with a registered sip phone and get it to dial a number and engage the speakerphone |
08:17.04 | dippo_ | i have a reigstered sip device in close proximity to an alarm on a door that was just tripped |
08:17.09 | dippo_ | i'd love to hear if there's anything going on :P |
08:18.30 | Netgeeks | if you had a cisco7960 for example and set up one of it's lines as a speakerphone auto answer, you could just dial that line |
08:18.38 | Netgeeks | lot of if's there |
08:18.46 | dippo_ | yeah |
08:18.53 | dippo_ | this is a grandstream budgetone |
08:19.04 | dippo_ | not exactly equipped with a long list of extras |
08:19.26 | Netgeeks | however, I always make the assumption that I'm not the only person in the world who configures one line on all my cisco 7960s as auto-answer speakerphone |
08:19.36 | Netgeeks | yep, not much there in that budgetel |
08:20.27 | dippo_ | hm |
08:20.31 | dippo_ | interestingly there is auto answer |
08:20.35 | Netgeeks | But don't worry, no one is there, if they opened the door, the draft from the movement of the door alone would have blown the budgettone off it's desk/table and if it fell more than 2 cm, it would have broken adn un-registered |
08:20.37 | dippo_ | <PROTECTED> |
08:20.39 | dippo_ | the call and turn the speaker on |
08:20.44 | dippo_ | heh |
08:21.09 | Netgeeks | or I should say, failed to re-register given you have the reg timeout at some sane value |
08:21.18 | dippo_ | i am sure it was a false alarm. the cops said they secured the area (10 mins after the alarm was tripped, on the second floor of a warehouse downtown behind 3 other locked doors. yeah, right.) |
08:21.34 | Netgeeks | unfortunately you probably don't want the speaker auto-answer to be it's single line default mode of operation |
08:22.03 | dippo_ | indeed |
08:22.08 | dippo_ | but i can change it temporarily from here |
08:22.17 | dippo_ | here goes nothin |
08:22.19 | Netgeeks | now I have an open web browser staring me in the face and I done forgot what I was going to have it do |
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08:22.31 | Netgeeks | calling the phone? maybe the perp will answer? |
08:22.39 | Snake-Eyes | happens to the best of us |
08:22.41 | dippo_ | bingo |
08:23.32 | dippo_ | all quiet on the western front |
08:23.48 | Netgeeks | thats good news for the eastern front then |
08:23.53 | MrChimpy | netgeeks: yep. not so worried about tdm400, more worried about our PBX :) |
08:24.33 | justinu | rj11 and rj45 wiring are compatible |
08:24.40 | Qwell | dippo_: That may very well have been the first report of asterisk being used to detect burglars :P |
08:24.46 | MrChimpy | at least the wiring is standard for the TE411 |
08:24.50 | Netgeeks | I would suggest the handy use of a butt set to verify the right configuration on both sides before hooking them up |
08:25.04 | MrChimpy | ju: but physically incompatible |
08:25.17 | Netgeeks | you should be able to become pretty comfortable you've got everything right with a little checking |
08:25.24 | dippo_ | heh |
08:25.30 | dippo_ | pretty handy, i have to say |
08:26.06 | MrChimpy | yeah. difficult though when I don't know if the TDM will give the right signals, given I don't have it configured through not connecting it to anything :) |
08:26.15 | MrChimpy | I got the lights to come on. that was good. |
08:26.17 | Netgeeks | they say you can generally solve the physically incompatability with copious amounts of alcohol |
08:26.31 | justinu | anyone know the difference between a Tie Trunk and a DID trunk? |
08:26.35 | justinu | in traditional PBXes |
08:26.52 | Netgeeks | I could only guess |
08:26.57 | Netgeeks | thus I won't |
08:27.23 | Netgeeks | but my empty browser has a google input... lemme see what it says |
08:27.50 | Netgeeks | tie trunk: A telephone line that directly connects two private branch exchanges (PBXs). |
08:28.07 | MrChimpy | whilst people seem awake, when I do the TDM411 stuff I'll need to be able to tell what number the caller dialled, so I can tell which service to direct them to (not using extensions). What does asterisk term this functionality? |
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08:28.33 | Qwell | called number identification, I think |
08:28.42 | MrChimpy | sweet. ta. |
08:28.49 | Qwell | ~cnid |
08:29.04 | justinu | DNIS, usually |
08:29.08 | Qwell | ~dnis |
08:29.10 | jbot | dnis is probably A telephone service that identifies the number that the caller dialed for the receiver of the call. DNIS is a common feature of 800 and 900 services, and can identify the number originally dialed when multiple 800 or 900 numbers terminate on the same destination trunks. DNIS works by passing the dialed number to the destination device, which can ... |
08:29.25 | Qwell | which can?! |
08:29.32 | justinu | cliff hanger |
08:29.52 | MrChimpy | ok, i'll search for both. how exciting. |
08:29.55 | *** join/#asterisk wellng (n=welles@61.150.60.123) |
08:29.56 | Netgeeks | bad jbot |
08:29.59 | Qwell | which can act upon this data to control its routing, queuing, IVR, or other call behavior. DNIS is typically used to separate call treatment for different inbound campaigns or help desk numbers, whether in one enterprise or at a service bureau. |
08:30.03 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
08:30.14 | MrChimpy | DNIS is exactly what I mean. |
08:30.19 | justinu | DNIS is when you have a block of 100 DIDs on a PRI |
08:30.32 | justinu | it's how you tell which DID someone dialed |
08:30.35 | Qwell | not always possible with analog lines |
08:30.49 | Qwell | but, with analog lines it's kinda easy |
08:30.53 | Netgeeks | *shrug* Analog lines are the devil spawn |
08:30.54 | MrChimpy | yeah, this'll be on E1s though |
08:30.55 | Qwell | you know which port it's coming from |
08:31.06 | *** join/#asterisk ReX (n=ReX@AMarseille-252-1-13-217.w83-197.abo.wanadoo.fr) |
08:31.10 | ReX | hi all |
08:31.14 | justinu | they have/had analog DID lines at one time |
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08:31.29 | Netgeeks | Hello ReX from france |
08:31.29 | justinu | but i think those are all gone |
08:31.54 | ReX | is anybody can help me about chan_btp compilation plz?? |
08:32.10 | Netgeeks | not I, sorry |
08:32.16 | MrChimpy | anyone actually configged the TE411? is it especially tricky? |
08:32.29 | Netgeeks | shouldn't be |
08:32.36 | MrChimpy | i hope not :) |
08:32.51 | Netgeeks | you've configured other digium T1/E1 cards? |
08:32.57 | MrChimpy | nope. |
08:33.02 | Netgeeks | ah, okay |
08:33.18 | MrChimpy | i'm a poor linux engineer who has been thrown at this stuff |
08:33.27 | Netgeeks | follow the example confs, pretty simple, you set up your T1/E1 parameters in /etc/zaptel.conf |
08:34.01 | Netgeeks | for t1 that would be the esf,b8zs stuff, for E1 it's some C's and S's but I don't remember off the top of my head |
08:34.07 | MrChimpy | cool. attempting with dev card and normal lines first |
08:34.26 | Netgeeks | you also tell it how to configure the channels in /etc/zaptel.conf as well |
08:34.44 | ReX | i have connected my cellphone on asterisk by BT, i just want asterisk dial and forward call when i call the cellphone number, is it possible? |
08:34.50 | Netgeeks | once you are done, (make sure you load the right module, someone here can pipe up the exact module name) |
08:35.02 | justinu | the digium cards aren't bad to setup |
08:35.18 | Netgeeks | and then run ztcfg -v, you shouldn't get any errors, if you do, something is amiss |
08:35.28 | MrChimpy | i'll doc what I do. i'm a good chimp. |
08:35.41 | Netgeeks | then I always run zttool just to make sure I've got no red or blue or yuellow alarms (with the T1/E1 connected) |
08:36.11 | Netgeeks | once you get that far, you can move on to zapata.conf which is not as simple, but if you can read you should be fine |
08:36.36 | MrChimpy | i can read. it's thinking that makes my head hurt. |
08:36.48 | Netgeeks | drink beer, it helps |
08:37.04 | Netgeeks | it helps stop the hurt, that is... for a while |
08:37.10 | Netgeeks | doesn't help the thinking tho |
08:37.12 | MrChimpy | it's 8.30AM here. i try not to start so early. |
08:37.42 | Netgeeks | hrm, NZ, AU? |
08:37.52 | Netgeeks | wait no, wrong side |
08:38.16 | Netgeeks | GMTish |
08:38.27 | MrChimpy | UK |
08:38.55 | MrChimpy | London to be precise |
08:40.03 | Netgeeks | Ah, one of my not so favorite places. The two times I visited it was over 90 degrees F (sorry dunno what that is in celcius) and I had a tiny little hotel room with no AC and a window that opened to the exaust from some seafood restraunt |
08:40.53 | MrChimpy | sounds familiar |
08:40.58 | Netgeeks | haha |
08:41.38 | MrChimpy | air conditioning is like strange voodoo to us. the tube during summer is so pleasant. |
08:42.20 | Netgeeks | I did however end up sitting at dinner right next to George Lucas and his daughter |
08:42.43 | MrChimpy | still, it's feckin freezing or raining so it's bearable for the one week we actually get summer |
08:42.59 | iDunno | hmm |
08:43.47 | Netgeeks | Then I got to ride this thing you called a train from london to Middlesborough... I called it a can of smoke... *sigh* |
08:44.00 | Netgeeks | but middlesbrough (spelling) was quite nice |
08:44.53 | iDunno | You have: tempF(90) |
08:44.53 | iDunno | You want: tempC |
08:44.53 | iDunno | <PROTECTED> |
08:45.04 | Netgeeks | thanks iDunno |
08:45.06 | iDunno | :) |
08:45.25 | MrChimpy | aye. out of the big smoke places are quite nice. i'm just here for the cash ;) |
08:45.27 | iDunno | now it's time to go to work :) |
08:45.34 | Netgeeks | it was the fifth decimal place 2 that broke the camels back... if it was just 32.22220 I would have been fine |
08:46.11 | MrChimpy | you have a very specific comfort zone. |
08:46.32 | Netgeeks | how true that is.... |
08:47.06 | MrChimpy | hm. 15 mins until everyone else gets in and starts bothering me |
08:47.31 | Netgeeks | I moved to this little speck on the map because it has very nice weather year around, well, good luck |
08:47.40 | Netgeeks | I'm going to hit the sack myself |
08:47.44 | MrChimpy | where are you? |
08:48.11 | Netgeeks | Brookings, Oregon, about 30 meters from a cliff that drops into the pacific |
08:48.31 | MrChimpy | ah nice. |
08:48.39 | MrChimpy | well, later.... |
08:48.48 | MrChimpy | thanks for the help |
08:48.54 | Netgeeks | hope it works out okay |
08:49.15 | MrChimpy | i'll let you know in about a week when i've torn my remaining hair out |
08:49.41 | Netgeeks | it grows back |
08:49.45 | Netgeeks | most of it |
08:51.03 | benjk | voipbuster announced end of life |
08:51.19 | benjk | oh well |
08:53.49 | ReX | nobody knows "Ext. Phone >> Cellphone BT (incoming call) >> Asterisk >> VoIP (outgoing call)" here? (without the use of cellsocket of course) |
08:57.33 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
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08:59.12 | benjk | cellsocket won't work in the UK, its US only |
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08:59.22 | benjk | you need a GSM gateway |
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08:59.41 | benjk | Siemens makes those |
08:59.45 | benjk | and some company in CZ |
09:00.33 | ReX | thx benjk, i try to configure asterisk with a bluetooth cell phone, but i don't arrive to compile chan_btp, i don't know what is the problem |
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09:02.53 | ReX | chan_bluetooth is installed and works fine (http://www.crazygreek.co.uk/content/chan_bluetooth) |
09:03.25 | ReX | but for the incoming call to cellphone, i suppose i need chan_btp |
09:03.51 | ReX | very sorry for my bad english :(( |
09:08.50 | ReX | benjk, do you mean this one http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html is a good way? |
09:11.30 | Skumling | I've got af simple question... how can I forward a call on asterisk? |
09:11.48 | [av]bani | bring a shrubbery |
09:11.48 | Skumling | I'm using ISDN phones connected through a HFC card in NT-mode |
09:12.13 | thazza | ReX: Hmmm.. Yummmy.. bluetooth to asterisk.. :D |
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09:15.17 | *** join/#asterisk llirk (n=majestic@210-84-11-13.dyn.iinet.net.au) |
09:15.21 | llirk | hi |
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09:20.17 | bsb | hi all! question to all, cisco 7914 is support SIP? |
09:20.31 | justinu | i've stumbled into another timezone |
09:20.40 | ReX | Skumling, exten => AAAAAAAAAA,1,Dial(SIP/BBBBBBBBBB@yourcontext), where AAAAAAAAA is your incoming call do you want to fw and BBBBBBBBBB, your forwarded number, i suppose |
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09:22.38 | Skumling | ReX: uhm okay, well now I just realize, that what I ment was "how do I transfer a call" :) |
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09:24.39 | benjk | ReX: yes, voiceblue, that's the one |
09:25.37 | ReX | if i don't arrive to install chan_btp, i must to buy a gateway gsm, but it's more expensive :( |
09:25.46 | benjk | as I understand it, it goes GSM network ----GSM---> Voiceblue-box ---SIP---> Asterisk |
09:25.59 | *** join/#asterisk nicksin (n=nicl@c1-144-7.ndn.isadsl.co.za) |
09:26.07 | benjk | its more expensive, but it's solid |
09:26.24 | nicksin | Good morning all |
09:26.32 | ReX | i suppose the simcard must to be introduce into this gateway gsm |
09:26.35 | benjk | chan_btp may not even do what you want |
09:26.48 | ReX | sure? |
09:26.50 | benjk | because it is only for detecting *presence* |
09:27.07 | benjk | hence btp, p for presence |
09:27.19 | benjk | so Asterisk knows that you are in the vicinity |
09:27.48 | benjk | there is also a bluetooth thingie to make a bluetooth headset or phone a terminal of Asterisk |
09:27.51 | ReX | I thought that chan_btp was the same as SIP for the bluetooth connexion |
09:28.28 | benjk | well I am not sure, but last time I looked at it this was the status back then |
09:28.50 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
09:29.05 | benjk | anyway, I think a proper gateway is the only solid thing to do |
09:29.05 | ReX | wich is the configuration file for incoming call by bluetooth, do you know? |
09:29.15 | benjk | no |
09:29.19 | ReX | okay |
09:29.26 | ReX | thx a lot benjk |
09:29.37 | benjk | and yes on the SIM card, the gateway needs to have it inserted |
09:29.46 | benjk | there are also gateways with multiple SIMs |
09:29.54 | ReX | cool |
09:29.56 | benjk | so you could have one SIM for each network |
09:30.07 | ReX | very cool :o)) |
09:30.52 | benjk | this is meant for least cost routing but these days with LNP on mobile networks I am not sure how you (or that gateway) would know which number belongs to which network |
09:32.12 | *** part/#asterisk secure75 (n=mic@p549A18C0.dip0.t-ipconnect.de) |
09:32.33 | benjk | anyway, I am getting a new Bluetooth capable mobile soon so if you check back in a while I will be able to tell you more about what the Bluetooth add-ons for Asterisk can do |
09:32.56 | *** part/#asterisk shawarma (n=sh@3E6B503C.rev.stofanet.dk) |
09:34.37 | nicksin | Im looking for some assistance in configuring 2 duxbury ISDN bri modems in a box. 1 for incoming and the other to the old PABX. |
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09:36.31 | ReX | benjk (private msg) ;) |
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09:45.10 | littleball | hello, any good tutorial about real time? |
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09:47.57 | MrChimpy | holy crap! I got a dial tone and busy tone out of my dev kit. wonders will never cease. |
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10:00.31 | jahani | what mean this error : Spawn extension (users-sip, h, 1) exited non-zero on 'SIP/1001-710a' ? |
10:01.52 | trixter | the hangup extention for users-sip exited with a value other than 0, it normally means that there was an error in what it called |
10:02.01 | trixter | but often can be ignored even if there was an error |
10:02.16 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
10:02.24 | trixter | dial iirc returns -1 if there is a hangup, app_conference I know returns 0 if the user hangs up |
10:02.33 | trixter | er doesnt return 0 |
10:02.37 | trixter | 0 is only if they press # |
10:03.17 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
10:04.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
10:04.18 | *** join/#asterisk ManxPower (i=ewieling@10.sub-70-197-2.myvzw.com) |
10:04.28 | jahani | when i call its ring 1 time and the call is terminate |
10:05.06 | puzzled | morning all |
10:06.32 | Modcuts | morning, if i have two sip lines, one of which has a local geographic number on it, does it depend on the provider if i want the incoming line to work with both sip lines? |
10:08.29 | bn-7bc | it seem that asterisk only captures the first key in the keymap so when I hare * for atxfer and *2 for atomon non of them worked, is this a known issue in v1.2.1? |
10:12.56 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
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10:15.55 | festr_ | hello |
10:16.03 | festr_ | isnt there problems with digium mailing lists? |
10:16.16 | festr_ | i've last message from date 15.1. |
10:16.29 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
10:21.49 | MrChimpy | hmm! |
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10:22.21 | MrChimpy | i have handset connected and getting dial tone etc from ast dev card |
10:22.36 | MrChimpy | ast can see me pick up and put down the phone |
10:22.56 | *** join/#asterisk ]]]papatya[[[[[ (i=E-sHeee@server.ivinskis.kursenai.lm.lt) |
10:23.04 | MrChimpy | handset is on channel 1 |
10:23.19 | MrChimpy | in zapata.conf I have |
10:23.27 | MrChimpy | [channels] |
10:23.27 | MrChimpy | context=test |
10:23.27 | MrChimpy | usecallerid=yes |
10:23.27 | MrChimpy | cidsignalling=v23 |
10:23.27 | MrChimpy | hidecallerid=no |
10:23.28 | MrChimpy | immediate=no |
10:23.30 | MrChimpy | <PROTECTED> |
10:23.32 | MrChimpy | context=internal |
10:23.34 | MrChimpy | signalling=fxo_ks |
10:23.36 | MrChimpy | echocancel=yes |
10:23.38 | MrChimpy | group=1 |
10:23.40 | MrChimpy | channel=1 |
10:24.10 | MrChimpy | I have an internal config in dialplan |
10:24.35 | MrChimpy | whcih answers, does a background(enter-ext-of-person), waits and hangs up |
10:25.08 | MrChimpy | it doesn't though. I start ast and all I ever get on picking up the handset is a dialtone, then busy if i dial anything |
10:25.31 | MrChimpy | any pointers? |
10:26.45 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
10:26.56 | ckruetze | Hi |
10:27.38 | ckruetze | I've a question regarding sendURL(). is sending and url something only * can do or is that sip standard? |
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10:29.14 | trixter | is it a sip standard even? |
10:29.40 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
10:33.12 | thazza | MrChimpy: What happens in the CLI? Have you asterisk -vvvvvvr |
10:35.34 | MrChimpy | cor. that'd be rather verbose then |
10:35.52 | MrChimpy | i pick up, it says -- Starting simple switch on 'Zap/1-1' |
10:36.37 | thazza | MrChimpy: When you have issues.. The more verbose the better. |
10:36.46 | MrChimpy | aye :) |
10:37.00 | MrChimpy | all I see is : |
10:37.07 | MrChimpy | <PROTECTED> |
10:37.07 | MrChimpy | <PROTECTED> |
10:37.24 | MrChimpy | that corresponds with me picking up and hanging up |
10:38.49 | MrChimpy | show channels |
10:38.49 | MrChimpy | Channel Location State Application(Data) |
10:38.49 | MrChimpy | Zap/1-1 s@internal:1 Rsrvd (None) |
10:38.49 | MrChimpy | 1 active channel |
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10:39.54 | MrChimpy | so that looks fine. maybe it's just audio playing that's broken. the file is there... does it normally work out of the box? it's just one of the standard ones. |
10:40.30 | MrChimpy | [internal] |
10:40.30 | MrChimpy | exten => s,1,Answer() |
10:40.30 | MrChimpy | exten => s,2,Background(enter-ext-of-person) |
10:40.30 | MrChimpy | exten => s,3,Wait(2) |
10:40.30 | MrChimpy | exten => s,4,Hangup() |
10:40.37 | *** join/#asterisk jluk (n=jon@80-235-135-92.cable.ubr07.nail.blueyonder.co.uk) |
10:40.59 | MrChimpy | it's in /var/lib/asterisk/sounds/enter-ext-of-person.gsm |
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10:42.12 | MrChimpy | i did set my defaultzone to uk, so maybe it's a country thing? |
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10:49.14 | MrChimpy | aha! |
10:49.33 | MrChimpy | i added echo test to extension 600 - that works |
10:49.46 | MrChimpy | it's s that doesn't work |
10:49.55 | cricalixwood | hi. Does anyone know if there is a problem with the Asterisk-users mailing list? I have not received anything since noon (GMT) yesterday. |
10:50.24 | MrChimpy | cric: apparently there is |
10:50.50 | cricalixwood | ok, thanks. Just confirming that it was not my end that was at fault. |
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10:53.57 | agx | hello, i cannot use rx_fax i get this message " ast_set_read_format: Unable to find a path from slin to unknown" ... have i to enable some extra codec or format in modules.conf? i'm using an isdn card via chan_modem to handle it |
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11:00.51 | agx | <agx> hello, i cannot use rx_fax i get this message " ast_set_read_format: Unable to find a path from slin to unknown" ... have i to enable some extra codec or format in modules.conf? i'm using an isdn card via chan_modem to handle it |
11:01.25 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
11:02.26 | MrChimpy | does the standard demo setup play a sample when you pick up the handset? |
11:02.28 | effape_ | so we have any zap experts in here? |
11:02.50 | effape_ | so/do |
11:03.02 | pengyong | any receive maillist from asterisk? |
11:03.09 | trixter | bbs news is covering the labia presidency |
11:03.11 | trixter | er liberia |
11:03.14 | pengyong | i don't got mails since last day |
11:04.09 | trixter | no one is getting mail from the lists |
11:05.33 | *** join/#asterisk rowter (n=woot@201.145.5.26) |
11:06.35 | tzafrir_laptop | effape_, they're out for lunch. Ask your question anyway |
11:06.59 | MrChimpy | currently I have the demo set up working, but I have to press 2 to get away from dial tone. (this is to handset connected to dev card) |
11:07.10 | MrChimpy | is that normal |
11:07.12 | MrChimpy | ? |
11:08.08 | RoyK | hm |
11:08.16 | RoyK | is there such a thing as a billsec timeout? |
11:08.26 | RoyK | absolutetimeout isn't really suited for callingcards |
11:08.33 | effape_ | aha ok. Basically the problem i'm having is with a tdm400p. When i get an incomming call with cidstart=polarity the call doesn't get routed to sip but if it's cidstart=usehist it does. However with usehist when i pickup the sip phone it gets a polarity switch and disconnects the zap channel |
11:08.43 | effape_ | it's all very odd |
11:11.18 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
11:11.19 | effape_ | pastebin.com/507986 seems to be where it all happens |
11:11.45 | effape_ | any ideas? |
11:12.33 | pif | hi, anyone tried using alaw files for musiconhold? |
11:12.50 | pif | with good results? |
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11:14.14 | effape_ | tzafrir_laptop who should i be asking about it? |
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11:19.30 | trixter | RoyK: execute a macro on connect that sets the timeout from the connection perspective -- absolutetimeout(current used - current available) which should equal bill seconds left |
11:19.42 | trixter | er flip those two numbers ... that would result in a negative number likely |
11:20.01 | trixter | well actually shouldnt that be a plus instead? I think so |
11:20.04 | rowter | !seen |
11:21.17 | ReX | is anybody can help me about cellphone BT connexion plz? |
11:21.41 | *** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com) |
11:22.16 | ReX | i've installed asterisk with chan_bluetooth, the cellphone is correctly connected and i can make outgoing call |
11:22.24 | ReX | my problem is for incoming call |
11:23.51 | ReX | wich configuration file i must to use to configure incoming bluetooth calls? |
11:23.57 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
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11:24.39 | ReX | I thought that chan_btp was the same as SIP for the bluetooth connexion, but i'me not sure and i don't arrive to compile chan_btp |
11:24.42 | RoyK | trixter: absolutetimeout includes call setup and ringing, which is somehow sub-optimal |
11:25.54 | trixter | RoyK: ok, then use the macro, although I am not sure how to get the currently used time, other than maybe some variable set when the call is answered |
11:26.01 | trixter | which should be accurate +/-1 sec |
11:26.26 | RoyK | perhaps i should write a TIMEOUT(billsec) |
11:26.47 | trixter | perhaps I was looking at this from the 5 minute solution perspective :P |
11:27.08 | RoyK | i'm using absolutetimeout how |
11:27.27 | RoyK | but it's not good enough for a commercial solution |
11:27.29 | RoyK | imho |
11:27.49 | trixter | well if that is your argument there are a lot of other things that need to be tossed out as well :P |
11:28.17 | trixter | a macro that sets the timeout based on currently used time + available time and that macro is called when dial connects |
11:29.01 | trixter | seems simple enough until something else cna be done, if this isnt something that has to be fixed in under 5 minutes then a proper solution would be better although take longer, especially for the testing phase |
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11:36.36 | MrChimpy | <PROTECTED> |
11:37.23 | trixter | congrats |
11:42.04 | *** join/#asterisk wellng (n=welles@222.90.15.246) |
11:42.15 | effape_ | i'm guessing noone yet then ;) |
11:42.56 | RoyK | hm |
11:43.38 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
11:44.17 | effape_ | i mean really if i just knew who i was looking out for :) |
11:45.06 | backblue | anyone with dundi? |
11:45.28 | mut | effape_? |
11:46.06 | mut | o nvm |
11:47.22 | effape_ | basically why is this happening? http://pastebin.com/507986 |
11:47.53 | effape_ | when i answer the sip it drops the zap chanel |
11:49.50 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
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12:07.28 | effape_ | sigh |
12:08.58 | fugitivo | effape_: i'm sure the problem is codec related |
12:09.13 | fugitivo | what codec is your sip device using? |
12:09.36 | thazza | effape_: Looks like you have reverse poltirty hangup turned on. Try turning it off. |
12:09.56 | effape_ | yeah but if i turn it off it never gets dropped. |
12:10.03 | fugitivo | effape_: codec |
12:10.21 | effape_ | What i don't understand is why it gets a reverse polarity on sip pickup., |
12:10.26 | effape_ | just checking codec now |
12:10.43 | effape_ | i have allow=ulaw and alaw |
12:10.46 | llirk | if I want to use the newest version of ooh323 with 1.2.1, is it best i pull down the latest CVS? or am i able to just upgrade the channel driver itself? |
12:10.56 | fugitivo | effape_: disallow=all allow=ulaw for your sip device? |
12:11.04 | thazza | effape_: Perhaps a hardward issue. this line makes me wonder. - Exception on 23, channel 3 |
12:11.06 | effape_ | yeah |
12:11.17 | fugitivo | Jan 16 03:30:06 DEBUG[2325]: Ooh, format changed from unknown to ulaw |
12:11.24 | effape_ | yeah but that follows the ulaw |
12:11.25 | effape_ | yeah |
12:11.38 | fugitivo | it's a problem with codecs |
12:12.04 | effape_ | what should i try? |
12:12.06 | fugitivo | what sip device are you using? |
12:12.26 | effape_ | for testing its xlite but in production it will be snom360 |
12:12.40 | fugitivo | setup xlite to use only ulaw |
12:14.30 | effape_ | i'm sorry but how do i do that? :/ Can't find option |
12:14.46 | effape_ | is that g711u? |
12:16.10 | fulgas | yes |
12:16.35 | effape_ | aha cool. Cheers. I can't try that just yet but i'll give it a go |
12:16.49 | effape_ | would that be why it's getting polarity reversal though? |
12:17.19 | sivana | morning |
12:18.00 | effape_ | mornin' |
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12:28.37 | [av]bani | anyone done a kind of 'virtual extensions'? where employees log in and out of a phone |
12:29.05 | [av]bani | eg employee sits down at a phone, hits *something to log in, then his pin and access code |
12:29.12 | [av]bani | when he leaves, hits *something to log out |
12:29.29 | sivana | [av]bani: you can do that with queues |
12:29.35 | [av]bani | ugh |
12:29.43 | thazza | [av]bani: Sounds like queues and agent logins |
12:30.04 | [av]bani | and if i dont want to use queues :) that would mean i'd have to setup a queue for every individual employee? |
12:30.06 | thazza | [av]bani: Yet i suppose you could do it. |
12:30.59 | mut | ooooooweeee |
12:31.02 | [av]bani | it would be a sort of roaming feature, like you can do with wifi |
12:31.05 | mut | i made some strong coffee this morn |
12:31.47 | *** join/#asterisk znoG_ (n=gs@33-138-114-200.fibertel.com.ar) |
12:32.30 | fugitivo | [av]bani: use the phone login |
12:32.39 | [av]bani | eh |
12:33.29 | mut | just make some whacked out dialplan for it |
12:33.38 | [av]bani | fugitivo: ??? |
12:34.48 | mut | dbput 1234 5432 |
12:34.49 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
12:34.53 | mut | dbput 5432 1234 |
12:35.01 | mut | dial 5432. dbgeet 5432 |
12:35.03 | mut | dial 1234 |
12:35.05 | mut | :P |
12:35.19 | [av]bani | o_O |
12:35.37 | mut | heh well it's an option |
12:36.56 | [av]bani | fugitivo: what do you mean use the phone login? |
12:37.17 | mut | he means login to the phone, wherever it's logged in, assuming it's sip or some such and not analog |
12:37.40 | effape_ | yeah i know the snom 360's i have have a login button to enter username and password |
12:37.48 | fugitivo | if it's analog he can use the ata/gateway login |
12:37.52 | effape_ | rather than config in the phone to a fixed login |
12:38.10 | *** join/#asterisk amir (n=amir@gentoo/developer/amir) |
12:38.17 | mut | fugitivo: the what? |
12:38.58 | [av]bani | fugitivo: eh? |
12:39.21 | mut | fugitivo: analog being like a channel bank |
12:39.32 | fugitivo | mut: oh |
12:39.38 | mut | not a phone hooked to an ata |
12:39.43 | [av]bani | login to the phone, you mean webinterface and change the config? |
12:39.54 | mut | [av]bani: man you're good |
12:40.14 | [av]bani | so give the employees the admin pw to the phone... |
12:40.38 | mut | eh they're all options |
12:40.42 | mut | we've given ya a few |
12:40.46 | fugitivo | so? it's just a phone |
12:40.48 | fugitivo | not a server |
12:40.50 | effape_ | no |
12:40.52 | mut | quite an array of choices if i do say so |
12:41.04 | [av]bani | well the queue idea sucks the least |
12:41.16 | effape_ | on the snom phones you have a screen. You can set them to require login |
12:41.18 | [av]bani | the others are varying degrees of horrible |
12:41.30 | mut | well thems ya options |
12:41.34 | fugitivo | you can code a web interface to post the data to the webinterface of the phone |
12:41.44 | fugitivo | so you don't need to give admin passwd to your users |
12:41.52 | mut | ^ there ya go |
12:41.54 | [av]bani | assuming they have access to a puter where the phone is, great |
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12:42.05 | mut | program a dial plan to do it |
12:42.09 | mut | and execute an agi script |
12:42.17 | fugitivo | there you go |
12:42.18 | effape_ | am i just taking rubbish? |
12:42.27 | fugitivo | the agi script is going to post the data to the webinterface |
12:42.36 | fugitivo | and login the user to the phone |
12:42.53 | mut | hot damn we make a good team fugitivo |
12:42.55 | mut | :P |
12:42.56 | fugitivo | see? more brains, more ideas |
12:42.59 | backblue | anyone with dundi? |
12:43.00 | fugitivo | hell yeah |
12:43.07 | [av]bani | yeah, just wondering which is less work, setting up piles of queues or writing agis |
12:43.21 | fugitivo | [av]bani: agi is going to be easy |
12:43.23 | mut | both are fairly simple.. |
12:43.33 | fugitivo | i think i'm going to code it myself, i liked the idea |
12:43.34 | [av]bani | i thought of agi but eh |
12:43.52 | mut | k well i must get to organizing my week of work |
12:43.54 | mut | bbfew |
12:44.02 | fugitivo | me too, i have a meeting |
12:44.03 | fugitivo | cya |
12:44.13 | [av]bani | changing config on phones all the time isnt really appealing |
12:44.24 | [av]bani | queues seems more 'natural' for asterisk |
12:44.25 | effape_ | why do you need to? |
12:44.29 | fugitivo | not config, just login information |
12:44.33 | effape_ | yes |
12:44.35 | fugitivo | that's why it's there |
12:44.37 | fugitivo | to login users |
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12:54.15 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
12:54.56 | warthawg | i have a really stupid question: how much money can a small business < 50 employees save by using Asterisk instead of a commercial PBX? |
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12:55.44 | fugitivo | warthawg: that answer can't be answering because there're a lot of variables on each company |
12:56.18 | warthawg | fugitivo, thanks. is it fair to say thousands of dollars would be typical? |
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12:56.33 | fugitivo | warthawg: again, it depends on your company |
12:56.48 | warthawg | ok, gracias amigo, i hope they don't catch you |
12:57.18 | fugitivo | warthawg: the idea is not saving costs, but adding funcionalities at less cost |
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12:57.29 | warthawg | aha, ok |
12:57.31 | xheliox | Is there any particular reason why an IAXy wouldn't natively bridge (IAXy -> * -> IAX Carrier) --- they're using the same codec on both sides... Asterisk says "attempting to xfer" and then fails, that's the only error that's received. |
12:58.23 | mut | anyone ever expiramented with using ucspi-tcp with asterisk? |
12:58.41 | mut | block annoying users who login 100 times a minute and fail |
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13:00.34 | many | heyya. is there anything to obeye if one uses asterisk+misdn+(bristuff+)rxfax? it seems rxfax connects correctly when being called by sip, but there is silence when being called by misdn |
13:00.48 | *** join/#asterisk littleball (n=littleba@cm78.epsilon175.maxonline.com.sg) |
13:00.56 | Asterisk_newbie | s |
13:01.04 | littleball | helo. does asterisk 1.2.1 support realtime on postgresql? |
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13:02.22 | Asterisk_newbie | hello anyone have a manual on howto install asterisk on debian and postgresql as database |
13:05.15 | fugitivo | good, people using postgresql and not the other crappy toy sql server :) |
13:08.23 | littleball | fugitivo, but UnixODBC postgresql is bad, right |
13:08.24 | littleball | ? |
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13:08.46 | fugitivo | i don't use realtime, for me, realtime is bad |
13:09.44 | mut | i wish i had time to dev my own module |
13:10.21 | mut | altho there is a appmysql isn't there |
13:10.24 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
13:11.56 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
13:13.54 | RoyK | ~seen zoa |
13:14.04 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 3d 16h 14m 13s ago, saying: 'apt-get install libssl-dev'. |
13:14.06 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
13:14.19 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
13:15.20 | *** join/#asterisk Druken (n=blowme@static.abss.ca) |
13:15.52 | Druken | morning peoples |
13:16.57 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
13:19.16 | hackeron | hey, is there anything built in to asterisk that displays the average latency, the % packet loss, etc? |
13:19.43 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:22.53 | *** join/#asterisk skamp|afk (n=keiner@minasmorgul.stuwo-steinweg.de) |
13:23.26 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
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13:25.05 | pigpen | in the cli: sip show peers |
13:25.13 | pigpen | shows you the latency...no packet loss. |
13:25.49 | trixter | hackeron: no there isnt that will show packet loss and jitter and such, latency based on each qualify is shown if you set qualify=yes (or a non 0 value) in your sip.conf |
13:26.13 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
13:26.26 | trixter | there are add-ons to ethereal and such that will show jitter, as for packet loss that is harder to see from a network perspective, the sniffer can tell if there is a missing sequence number but not what was actually received |
13:26.28 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
13:26.40 | hackeron | trixter: our grandstream phones crap out if I set qualify=yes :( |
13:27.40 | trixter | well the latency shown by qualify=yes is not network latency only, it includes the time it takes to generate sip packets and such since it measures that by sending an invite |
13:28.03 | RoyK | er |
13:28.05 | RoyK | wtf? |
13:28.05 | RoyK | <PROTECTED> |
13:28.36 | pigpen | trixter, thanks for the clarification....I learned something... |
13:29.30 | trixter | um ok |
13:29.39 | trixter | sounded like you already knew that based on your earlier comment |
13:29.48 | *** join/#asterisk Soul (n=Soul@87-196-39-131.net.novis.pt) |
13:31.09 | *** join/#asterisk CuTe^G (n=cuteg@202.63.195.79) |
13:31.10 | pigpen | I thought is was latency only..not the additional time to create the sip packet, etc.. |
13:31.44 | CuTe^G | hello guys i need help ... i m making Asterik base IP-PABX for domestic |
13:31.45 | pigpen | now the built in packet loss feature in the cli would be nice. |
13:31.52 | CuTe^G | i need FXO cards |
13:32.04 | pigpen | how many ports? |
13:32.06 | CuTe^G | which 4 ports FXO PCI card which i best and cheap |
13:32.13 | CuTe^G | ? |
13:32.19 | CuTe^G | 4 / 8 / 16 FXO |
13:32.20 | CuTe^G | any |
13:32.26 | tzanger | I only know of one four port FXO PCI card |
13:32.36 | pigpen | I haven't used any others then digium, they have a 4 port and the new 24 port modular |
13:32.50 | pigpen | 4 port is modular too |
13:33.07 | CuTe^G | how in one PCI Card 24 ports ? |
13:33.22 | CuTe^G | i m new user of Asterisk |
13:33.25 | pigpen | the 24 port is in increments of 4 similar functions (fxo/fxs) |
13:33.33 | pigpen | I am ordering one today. |
13:33.58 | pigpen | nice thing about the 24 port guy, is that you can get it with echo cancellation built on. |
13:34.20 | fugitivo | pigpen: did you get one? |
13:34.21 | CuTe^G | i need simple |
13:34.36 | fugitivo | CuTe^G: check the tdm2400 |
13:34.42 | pigpen | ordering one today. but it sounds great....I hear it is better on faxing too. |
13:34.43 | CuTe^G | 4 port FXO |
13:34.43 | CuTe^G | i put in 4 PSTN Lines |
13:34.55 | CuTe^G | and thru SIP Softphone my 4 agents will received the calls |
13:34.58 | fugitivo | pigpen: really? it's not a big tdm400? ;) |
13:35.02 | trixter | I had considered adding a res_snmp thing which would require counters for things like packet loss, throughput, etc.. just never got around to it |
13:35.13 | fugitivo | CuTe^G: only 4? get a tdm400p with 4 fxo modules |
13:35.16 | pigpen | fugitivo, not from what I hear...it is a complete redesign. |
13:35.27 | trixter | I think that if those counters existed (no matter where) traps could be sent easily enough and certainly anything could pull the info |
13:35.34 | CuTe^G | TDM2400 how much ?? |
13:35.46 | CuTe^G | then i need |
13:35.49 | trixter | $1800-2100 full depending on fxs or fxo |
13:35.55 | fugitivo | CuTe^G: maybe $300, where are you located? |
13:35.56 | CuTe^G | TDM400P FXO |
13:35.59 | pigpen | right. |
13:36.03 | trixter | the tdm400p is much cheaper |
13:36.09 | CuTe^G | I am from PAKISTAN |
13:36.15 | CuTe^G | how much TDM400P |
13:36.22 | fugitivo | CuTe^G: no idea then |
13:36.32 | trixter | I think those run full about $300-400 incl shipping |
13:36.44 | fugitivo | a TDM04B (tdm400 with 4 fxo) is $450 in Argentina |
13:36.45 | *** join/#asterisk Assid (n=assid@203.115.64.5) |
13:36.46 | Assid | heya |
13:37.13 | trixter | well there may be duties on that which I dont consider shipping charges |
13:37.22 | CuTe^G | and how much 1 FXO port ? |
13:37.29 | fugitivo | $180 |
13:37.31 | trixter | if you ship one to the UK for example its 3% duty + 17% vat on top, ireland is 21% vat 3% duty |
13:37.44 | trixter | so that can add a considerable amount, I have no idea what, if any, pakistan would charge |
13:37.49 | pigpen | dam! I am glad I don't have to screw with that.... |
13:37.58 | fugitivo | trixter: it's always better to find a local reseller |
13:38.05 | fugitivo | trixter: they'll have lower prices |
13:38.13 | trixter | that may not always be an option |
13:38.25 | Assid | hrmm.. anyone here using a cisco 7960 ? |
13:38.38 | trixter | and I was selling electronics to europe mainly becuase I could undercut local resellers by $100 on the same exact item and still profit like $100 |
13:38.39 | fugitivo | if i have to buy a card in the US, I have to add shipping + 50% custom taxes + 21$ iva (local taxes) |
13:38.43 | trixter | sometimes local resellers arent cheaper |
13:38.49 | trixter | especially if there is little competition |
13:39.08 | trixter | where are you that there is 50% duties? |
13:39.09 | CuTe^G | for domestic call center TDM400 is best ? |
13:39.14 | fugitivo | trixter: Argentina |
13:39.17 | trixter | ahh |
13:39.28 | ReX | anyone hre using a bluetooth cellphone for inbound and forward calls? ;) |
13:39.33 | trixter | yeah its 20-25% for europe so that isnt as bad but still... |
13:39.45 | trixter | I was selling mp3 players for a $100 profit and couldnt get em fast enough a couple years ago |
13:39.49 | fugitivo | trixter: custom taxes are a pain in the ass |
13:40.18 | CuTe^G | anyone help me plz msg me in pvt |
13:40.24 | trixter | the EU lets you do a little arbitrage though, once you get it into the EU the item can generally move around without additional duties |
13:40.35 | trixter | so you locate the cheapest country send em there then redistribute |
13:40.40 | trixter | but that doesnt work well for 1 item |
13:40.57 | CuTe^G | Asterisk X100P FXO $25.95 |
13:40.59 | CuTe^G | Asterisk X100P FXO $25.95 ??? |
13:41.21 | [TK]D-Fender | CuTe^G : Avoid the X100.... How many lines are you looking at bringing into *? |
13:41.22 | fugitivo | custom taxes should exist for items made in your own country, if the item is not made in your country, why must i pay taxes for it? |
13:41.31 | fugitivo | CuTe^G: the tdm400 will be the best option |
13:41.42 | CuTe^G | i need 8 lines |
13:42.00 | fugitivo | now you need 8? |
13:42.01 | CuTe^G | there is any 8 lines port PCI card for FXO ? |
13:42.22 | fugitivo | no |
13:43.19 | fugitivo | CuTe^G: you have 2 options, 2 TDM04B for around $450 each, or a TDM2402 for around $1000 ($1200 with echocan) |
13:43.23 | cfh | is possible monitor the state of isdn line on a sip phone? |
13:43.25 | Druken | CuTe^G: just the 24 :) |
13:43.58 | CuTe^G | why 1 port FXO for 26$ and 4 port FXO TDM400P for 450$ ?? |
13:44.12 | fugitivo | CuTe^G: the x100p is a bad card |
13:44.19 | fugitivo | CuTe^G: it's a cheap modem |
13:44.33 | CuTe^G | Digium TDM400P |
13:44.35 | CuTe^G | $141.40 |
13:44.35 | CuTe^G | Digium TDM400P |
13:44.35 | CuTe^G | $141.40 |
13:44.46 | Druken | fugitivo: i tell ya... i use the damn x100p's more than the tdm, because my tdm is a peice of shat |
13:45.06 | fugitivo | CuTe^G: that's the card without modules, add 4 fxo modules |
13:46.20 | fugitivo | Druken: my x100p cards suck, maybe i got the bad ones |
13:46.20 | fugitivo | Druken: echo, no hangup, etc |
13:46.20 | Druken | you got the shitty x100p, i got the shitty tdm :) |
13:46.20 | fugitivo | hehe |
13:46.29 | Skumling | When using a Siemens Gigaset on at HFC-S in NT-mode, how can I then transfer a call? I think it's difficult to find docs on this, it seems that everyone but me knows how to it? When I hit # on the handset, I just hear the DTMF tones |
13:46.32 | ManxPower | CuTe^G, That's $141.40 for the carrier card. You still need the FXO modules |
13:46.34 | [TK]D-Fender | CuTe^G : Other options are FXO->SIP gateways (Mediatrix,AudioCodes,etc...) or a Sangoma A200 PCI solution (2 cards a little cheaper than TDM400 solution) |
13:46.35 | Druken | fugitivo: my x100p's give me no echo, but my tdm echo's back my voice, and the tdm barely works... |
13:46.47 | fugitivo | Skumling: press flash on the phone |
13:47.12 | fugitivo | Druken: that's weird |
13:47.18 | fugitivo | Druken: you should return that card |
13:47.31 | Skumling | fugitivo: the phone has no flash button... or... eh... hrm. Some of the handsets actually has... |
13:47.35 | Druken | i've had it for a year, don't think they would take it back... hehe |
13:47.39 | fugitivo | Skumling: yes it has, i have one |
13:48.15 | Skumling | fugitivo: there's many different handsets... |
13:48.26 | ManxPower | Skumling, FLASH is sometimes called RECALL |
13:48.37 | fugitivo | Skumling: what model is your siemens? |
13:48.58 | CuTe^G | anyone guide me complete solution i m very new |
13:49.03 | Druken | flash or link in canada here :) |
13:49.20 | *** join/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu) |
13:49.27 | Skumling | fugitivo: I've got 3000C, 3000S and 4000 Micro |
13:49.31 | Druken | CuTe^G: www.voip-info.org bone up :) |
13:49.35 | *** part/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu) |
13:49.52 | fugitivo | CuTe^G: we already told you, 2 tdm400p with 4 fxo modules each, or an fxo->sip gateway (more expensive) |
13:50.05 | fugitivo | Skumling: i have the 4000 |
13:50.29 | fugitivo | Skumling: and the flash button, is a small circle at the inferior left side |
13:50.34 | fugitivo | Skumling: with 2 arrows |
13:51.14 | ManxPower | Telecom is expensive. If you can't handle that then don't mess with it. |
13:51.29 | fugitivo | Skumling: did you find it? :) |
13:52.23 | Druken | ManxPower: are you in a mood this morning?? :) |
13:52.27 | fugitivo | hehe |
13:53.15 | ManxPower | Druken, I just get tired of newbies expecting to be able to put a bunch of ports in a box for less money than most of us spend on COFFEE in a week. |
13:53.28 | fugitivo | lol |
13:53.43 | Druken | ManxPower: you must have an iv drip for your coffee.... |
13:54.02 | [TK]D-Fender | ManxPower : You clearly need a better coffee distributer! |
13:54.08 | Druken | but yeah i know what u mean :) |
13:54.14 | fugitivo | or he's going to die soon |
13:54.15 | ManxPower | Druken, I was referring to people that go to a coffeehouse. |
13:54.38 | ManxPower | $4/latte x 5 days per week |
13:54.40 | Druken | tim hortons gets expensive :) |
13:54.46 | MrChimpy | bored now. got my dev card working now it's dull. |
13:54.49 | fugitivo | coffee is going to kill you |
13:55.01 | fugitivo | drink tea or mate |
13:55.02 | MrChimpy | waiting for a server to stick the E1 card in |
13:55.05 | ManxPower | The nearest coffeehouse if like 30 miles from me. I don't go to them very often. |
13:55.07 | Katty | morning. |
13:55.30 | Ikarus | ManxPower: it is bullshit that that isn't the case yet, the actual technology isn't expensive, the cost of getting it tested is, bunch of rip-off test labs |
13:55.34 | Druken | ManxPower: get yourself an assistance, and send him to get you coffee :) |
13:56.09 | tzanger | ManxPower: I just use the free company coffee. :-) |
13:56.17 | ManxPower | Druken, Where can I get an assistant to work for free? |
13:56.22 | Skumling | fugitivo: is it a 4000 C or M? |
13:56.36 | tzanger | and I make a point to say how much easier it is to get a coffee from Tim Horton's than Starbucks when I'm picking up my sweetheart's coffee |
13:56.39 | fugitivo | Skumling: errr |
13:56.43 | tzanger | (I call it her pretentious coffee, heh) |
13:56.50 | fugitivo | Skumling: it just says siemens 4000 at the back side |
13:56.51 | [TK]D-Fender | Katty: Mew. |
13:56.56 | ManxPower | I brew my own. |
13:57.28 | Skumling | fugitivo: okay... hrm... I'm gonna google for the specs on my handset... thanks |
13:57.40 | fugitivo | Skumling: you don't have that button? |
13:57.45 | Katty | [TK]D-Fender: :> |
13:58.02 | Katty | [TK]D-Fender: apparently everyone else is too busy to say hi. |
13:58.12 | tzanger | Good morning my dear |
13:58.20 | Skumling | fugitivo: on my 3000C's I have a "R"-button - should be flash |
13:58.23 | Katty | tzanger: too late! |
13:58.24 | Druken | ManxPower: dunno, what countries have you looked in? i hear africa will work for mere food... :) |
13:58.33 | Skumling | fugitivo: but on my 4000 M I don't seem to have that button... |
13:58.38 | tzanger | Katty: oh well, I'll have to make it up to you another time |
13:58.45 | Katty | tzanger: k |
13:59.11 | fugitivo | Skumling: on my 4000, i have 2 tiny circle buttons at sides of the mic, do you have those circle buttons? |
13:59.35 | Druken | Katty: was on the phone, and in another program, so good mOOrning :) |
14:00.05 | Druken | Skumling: wouldn't "r" be release? |
14:00.10 | Katty | Druken: (= |
14:00.47 | trixter | hi kattykat |
14:00.51 | Ahrimanes | anyone here done mass config of snom phones? |
14:00.59 | *** join/#asterisk _rehash (n=rehash@ppp-48-118.atnr.ro) |
14:02.36 | Katty | trixter: hihi. |
14:04.17 | ManxPower | Gads! People are dying left and right. First my significant other's father, now the mother of my tech contact at my largest customer. |
14:06.11 | *** join/#asterisk pengyong (n=lala@222.185.197.28) |
14:06.17 | RoyK | anyone here running a rev1 te410p on 1.2? |
14:06.56 | ManxPower | RoyK, I am. |
14:07.01 | *** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net) |
14:07.03 | trixter | Katty: hows it goin? |
14:07.11 | trixter | did you may those mushroom thingys? |
14:07.17 | tzanger | RoyK: I am running oldschool TE405 with svn trunk |
14:07.26 | Katty | trixter: i'm sorta in a weird awakeful state between naps. |
14:07.29 | *** join/#asterisk paljas (n=paljas@sarastro.cs.uu.nl) |
14:07.38 | trixter | Katty: how cat like :P |
14:07.39 | Skumling | fugitivo: seems like it's what they call "the net carrier key" on 4000 Micro :) |
14:07.56 | Katty | trixter: i have a new favorite quote! |
14:08.01 | trixter | whats that? |
14:08.02 | Skumling | Druken: here in Denmark, the R-keys generally equals to flash |
14:08.03 | Katty | trixter: Now Men......Men are like a fine wine. They begin as grapes, and it's up to women to stomp the gutts out of them until they turn into something acceptable to have dinner with. |
14:08.14 | trixter | haha |
14:08.33 | trixter | and if they dont find a woman to stomp them they shrivel up and turn into rasins? |
14:08.36 | MrChimpy | don't stomp on my grapes! |
14:09.15 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
14:09.25 | trixter | science channel has a thing on now about the comet dust |
14:10.04 | trixter | <-- dreading when the mailing lists come back from the 1359013951 emails asking if the list is down |
14:10.08 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
14:10.24 | Katty | haha. |
14:10.58 | trixter | wow by placing quail brains in a chicken while its in the egg they have been able to get chickens to cluck like qualis and even mimik their movements and stuff |
14:11.15 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
14:11.42 | [TK]D-Fender | trixter : Big deal.... China made pigs that glow in the dark! |
14:11.46 | MrChimpy | trix: sounds worthwhile |
14:11.56 | MrChimpy | chickens have terrible personalities |
14:12.09 | trixter | heh |
14:12.36 | trixter | wellk they also mapped like 30% of the wooly mamoth genome so far which means that jurisic park can have those too |
14:13.16 | Katty | you're about to - you're about to - you're about to enter an echo test! |
14:13.23 | trixter | they just better get their operators correct and not miscount genders! |
14:13.28 | Katty | my co-workers are having too much fun this morning. |
14:13.33 | *** join/#asterisk basta (n=basta@213-156-52-98.fastres.net) |
14:13.57 | ManxPower | I found a new way to torture users! |
14:14.12 | tzanger | ManxPower: ? |
14:14.50 | ManxPower | tzanger, a script to disable their mailbox if 1) it has the default password or 2) they did not record all three (busy, unavail, name) greetings. |
14:14.55 | basta | is the mailing list server down ? I'm not receiving mails at asterisk-users since two days ... |
14:15.13 | trixter | basta: nah it works fine |
14:15.47 | tzanger | ManxPower: why would you do #2? |
14:16.16 | ManxPower | tzanger, We get 2 or 3 trouble tickets per week where the user complains about the "the user at extension xxx is busy" messages. |
14:16.31 | ManxPower | or complains that they are not listed in Directory |
14:16.35 | *** join/#asterisk welles (n=welles@222.90.92.113) |
14:16.44 | tzanger | how is that a trouble ticket? they didn't do what they needed to do |
14:17.00 | ManxPower | tzanger, Correct. |
14:17.31 | ManxPower | Then we have to explain to them that only morons don't read the instructions we provided to them when the mailbox was setup. |
14:17.32 | tzanger | so you lock them out as a preemptive measure. I like it. :-) |
14:17.49 | bn-7bc | whi is it tha res:heatures omly evere works with single key kombinations (ex: etxfer works hen i map it to * but not when mapped to *1)? |
14:17.55 | tzanger | wow the lists have been down for a couple days now |
14:19.19 | bn-7bc | does anyone else have the problem (nv 1.2.1)? |
14:19.56 | sivana | tzanger: you just wake up? :) |
14:20.17 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
14:20.19 | tzanger | sivana: of course |
14:20.22 | sivana | I'm coming down tomorrow... taking a sick day? |
14:20.24 | trixter | this is silly.. this guy does a robbery, and kills his sons girlfriend who turned him in, then when in jail for life for that murder he orders the murder of witnesses for the murder he was serving time for. Now that he has spent 23 years on death row fighting his case the argument is that 'he is old now so he shouldnt be put to death'.. while I am not commenting on whether or not its right or wrong for the death penalty I think that is a silly reas |
14:20.24 | trixter | on to ask for clemency. which means late tonight/early tomorrow he will be put to death beucase the courts dont seem to accept that spending years fighting your case for which you are found guilty of when it involves multiple murders is grounds |
14:20.31 | tzanger | I can't, I have to be in Niagara Falls |
14:20.54 | tzanger | hahahaha |
14:20.58 | tzanger | that is one fucked up situation |
14:21.25 | trixter | personally I think the death penalty should only be used when its cheaper, which 99% of the time its not so ... and when sentenced to death you get an automatic appeal, most defendants are getting court appointed lawyers and it takes at least 10 years often longer to get all that done, so generally it costs 2-3 times what life would cost |
14:22.31 | Skumling | hrm... how do I make an "s"-entry in my dialplan, that gives me a dialtone, and then allows me to enter the digits and call out? |
14:22.34 | trixter | but the guy is legally blind, deaf, and now cant walk (all of those things he wasnt when he ordered the murder of witnesses, killed one testifying against him and the underlying robbery that started it all.. |
14:22.50 | trixter | Skumling: what medium are you connecting with? |
14:23.00 | trixter | voip protocols typically send the whole number dialed not digit by digit |
14:23.08 | Skumling | trixter: ISDN phone via Zaptel |
14:23.09 | trixter | fxs ports will send digit by digit though |
14:23.12 | Skumling | HFC |
14:23.16 | ManxPower | Skumling, almost nobody needs to do that. |
14:23.30 | trixter | what signalling is the isdn doing? doesnt that normally send the entire number as data? |
14:23.40 | trixter | becuase of the out of band signalling that is used |
14:23.48 | trixter | or does that one send it digit by digit? |
14:24.17 | enemy^x | What should "show hints" actually display incase of a client beeing on the phone? In my case it displays "241 : SIP/241 State:Idle Watchers 0" ..... |
14:24.26 | Skumling | ManxPower: maybe... but humm, the issue is, that when I transfer a call, it runs the "s", when hitting flash immidately? |
14:24.53 | ManxPower | Skumling, it should not do that unless you have something like immediate=yes |
14:24.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
14:25.21 | Skumling | ManxPower: ah... |
14:25.22 | shmaltz | looks like the lists are down |
14:25.36 | [TK]D-Fender | enemy^x : 8247 : SIP/8247 State:InUse Watchers 0 |
14:26.05 | Skumling | ManxPower: my zapata.conf has a immediate=no string... |
14:26.32 | shmaltz | Skumling, what type of zap card do you have? |
14:26.36 | Skumling | HFC-S |
14:26.40 | Skumling | using Bristuff |
14:26.57 | trixter | shmaltz: nah they work fine |
14:27.07 | shmaltz | immediate is used when the signaling (like DID) gets done in touchtones |
14:27.17 | shmaltz | trixter, what the lists? |
14:27.22 | Skumling | I tried AMP, and when I had it running, I got a dialtone... but the dialplan in AMP is very complex (to me), and it really doesn't fit my needs, so I've decided to do custom config |
14:27.23 | trixter | sure why not |
14:27.46 | [TK]D-Fender | Skumling : CONGRATULATIONS |
14:27.53 | shmaltz | trixter, take a look: |
14:27.55 | shmaltz | http://lists.digium.com/pipermail/asterisk-users/2006-January/date.html |
14:28.02 | shmaltz | <PROTECTED> |
14:28.04 | shmaltz | Archived on: Sun Jan 15 05:17:07 CDT 2006 |
14:28.12 | trixter | people just took off for the holiday here |
14:28.19 | Skumling | [TK]D-Fender: huh? |
14:29.08 | [TK]D-Fender | Skumling : For getting away from AMP |
14:29.29 | [TK]D-Fender | Skumling : What do you really need out of *? |
14:29.33 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
14:29.36 | Skumling | [TK]D-Fender: hehe... thanks ;) |
14:30.45 | fugitivo | Skumling: congratulations too! |
14:30.56 | Skumling | [TK]D-Fender: I need a not-so-complicated system, but I need to run 4 separate (very small) companys on it, with different welcome etc, but I also need to be able to transfer calls between local phones, hold music, voicemailboxes etc. |
14:31.15 | Skumling | voicemail, hold music etc. should not be an issue |
14:31.21 | Skumling | I also need a telemarketing torture script :-D |
14:31.37 | [TK]D-Fender | Skumling : All very easy.... |
14:32.22 | [TK]D-Fender | Skumling : A days work tops. |
14:32.25 | Skumling | [TK]D-Fender: maybe ;)... I'm a asterisk newbie, and I've spend quite some hours reading docs, and I want to have a basically understanding of how things work |
14:32.27 | *** join/#asterisk barinoff (i=izida@82.162.60.62) |
14:32.42 | barinoff | can i ask dummy question? |
14:32.55 | [TK]D-Fender | barinoff : Hasn't stopped enough people yet... shoot! |
14:32.58 | [TK]D-Fender | ;) |
14:33.11 | Skumling | [TK]D-Fender: my only problem right now, is that when I hit the flash-key to transfer a call, the "s" is executed immediately |
14:33.26 | shmaltz | trixter, nah, I don't think that in england they have a MLK holiday |
14:33.32 | shmaltz | unless it's a bank holiday :P |
14:33.36 | Ahrimanes | anyone tried 4 or 8 port ata's ? |
14:33.46 | [TK]D-Fender | Skumling : What interface? |
14:33.49 | Skumling | [TK]D-Fender: I would like some kind of prompt letting me enter a number etc. |
14:34.03 | Skumling | [TK]D-Fender: ISDN Gigaset handset on a HFC-C in NT-mode |
14:34.05 | barinoff | :) I first time install this soft and now have a big question - have it http or gui iterface for configure? |
14:34.10 | shmaltz | Ahrimanes, yes |
14:34.16 | shmaltz | mediatrix |
14:34.17 | Skumling | [TK]D-Fender: using Bristuff |
14:34.25 | [TK]D-Fender | Skumling : Don't know ISDN.... wish I could help there. |
14:34.27 | Ahrimanes | scardinal: ok, know the pricing? did they work well? |
14:34.44 | barinoff | as i understand - it's workable with webmin? |
14:35.08 | [TK]D-Fender | barinoff : There are web-based GUI's for setting up * but they all create crappy dialplans and take away the control from you. |
14:35.46 | [TK]D-Fender | barinoff : No WebMin. there is AMP, ScopServ, FirstLane, and a few otehrs, none of which I suggest unless you are looking at a large install. |
14:36.03 | Skumling | [TK]D-Fender: damnit... but can't I make a "s"-entry in my dialplan? |
14:36.14 | barinoff | [TK]D-Fender where i can read about it? |
14:36.33 | _upsite | hey guys |
14:36.55 | shmaltz | D-Fender, barintoff, it's thirdlane, not first lane |
14:37.06 | shmaltz | also, thirdlane is a webmin module |
14:37.40 | barinoff | shmaltz - some link? |
14:37.41 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
14:37.54 | [TK]D-Fender | Skumling :Yes in the context that your phones use. I'd suggest you not mix other kinds of phone into that context however |
14:37.56 | shmaltz | thirdlane.com |
14:38.11 | [TK]D-Fender | shmaltz : Shit by any other name :) |
14:38.13 | _upsite | if i register my * like " register => user:pass@some.dialup.host.dyndns.org" an this host changes it's ip address ..* is not re-resolving the new ip address. any fixes or workarounds for that? |
14:38.14 | Skumling | [TK]D-Fender: that's no problem |
14:39.12 | [TK]D-Fender | Skumling : You can get dial-tone, but the problem is you want to use this for TRANSFER, not to place another direct call... |
14:39.25 | backblue | anyone with DUNDi working? |
14:39.45 | [TK]D-Fender | Skumling : if all you wanted to do was use "flash" to place another call you'd use DISA in your "s" pointing to the same context. |
14:40.10 | backblue | _upsite: that its a problem, and there is no correction avaliable yet. |
14:40.28 | [TK]D-Fender | Skumling : so in [myphones] you'd have exten => s,1,DISA(myphones,nopassword) |
14:40.34 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
14:40.36 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
14:41.03 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
14:41.40 | [TK]D-Fender | (or whatever the nopassword parameter is) |
14:41.56 | Skumling | phone, brb |
14:42.11 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
14:42.58 | *** join/#asterisk secure75 (n=mic@dslb-084-057-040-132.pools.arcor-ip.net) |
14:43.41 | enemy^x | [TK]D-Fender: http://pastebin.ca/36970 (could you please look at this, I`m not getting why my show hints doesnt work out :( ) |
14:44.02 | Skumling | [TK]D-Fender: okay... |
14:44.04 | shmaltz | anybody from Digium is up already? |
14:44.08 | shmaltz | kevin? |
14:45.01 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
14:45.06 | [TK]D-Fender | enemy^x : Which one isn't working? |
14:45.13 | [TK]D-Fender | (specific #) |
14:45.33 | pif | what is the g711-alaw sample rate? |
14:45.51 | enemy^x | [TK]D-Fender, none of them... I`m working with my own, 236 for the moment |
14:45.52 | [TK]D-Fender | pif : 8000 |
14:46.19 | [TK]D-Fender | enemy^x : pastebin your sip.conf |
14:46.23 | trixter | shmaltz: the list took a holiday, let it have its vacation |
14:46.28 | pif | funny, when I listen to the files from MusicOnHold they sound too fast |
14:46.50 | shmaltz | trixter, only if it shares the beer with me, and maybe dances with me |
14:46.54 | *** part/#asterisk secure75 (n=mic@dslb-084-057-040-132.pools.arcor-ip.net) |
14:47.28 | Skumling | [TK]D-Fender: hrm, when placing the DISA-stuff into my dialplan, I get a very funny sounding and jittering dialtone |
14:47.32 | trixter | hey its free you cant expect it to actually work too!! (the open source mantra) |
14:47.39 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
14:47.41 | shmaltz | lol |
14:47.48 | Skumling | [TK]D-Fender: and when dialing a local number nothing happens :-/... hrm. |
14:48.07 | [TK]D-Fender | Skumling : Pastebin your extensions.conf |
14:48.13 | enemy^x | [TK]D-Fender: http://pastebin.ca/36971 |
14:48.14 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
14:49.16 | shmaltz | this guy is an idiot: |
14:49.17 | shmaltz | http://video.google.com/videoplay?docid=-1532509206579317476&q=George+Galloway |
14:49.30 | [TK]D-Fender | enemy^x : So if 236 is on the phone it won't show you the hint for it? |
14:49.46 | enemy^x | [TK]D-Fender: correct |
14:49.49 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:49.50 | *** mode/#asterisk [+o anthm] by ChanServ |
14:50.26 | Skumling | [TK]D-Fender: http://pastebin.ca/36972 |
14:50.40 | [TK]D-Fender | enemy^x : paste the line that "show hints gives for it |
14:51.30 | Skumling | [TK]D-Fender: the dialplan is only for testing so far :) |
14:51.52 | [TK]D-Fender | Skumling : sunno.... |
14:52.00 | [TK]D-Fender | dunno.. |
14:52.10 | Skumling | [TK]D-Fender: what I expect to be able to, is to hit the flash key, then forward a call to eg. local 12 or 15 |
14:52.38 | enemy^x | [TK]D-Fender: http://pastebin.ca/36973 |
14:53.18 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
14:54.01 | [TK]D-Fender | Skumling : Could be becuase you have 2 entries for SIP/236 <- You shouldn't double them up like that... |
14:54.20 | *** join/#asterisk Davey|Work (n=davey@unaffiliated/davey) |
14:54.27 | Johnnie | Anyone handy with ztdummy installations? |
14:54.31 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc07j.dialup.mindspring.com) |
14:54.32 | Davey|Work | Is it possible to have a remote SIP client with a dynamic IP? |
14:54.41 | Johnnie | I'm having a problem which I can't seem to nail down. |
14:54.42 | [TK]D-Fender | Skumling : Well DISA won't help you transfer like I said. Try using the "#" method in your dial commands for it instead. |
14:54.50 | [TK]D-Fender | Davey|Work : Sure |
14:55.15 | shmaltz | for some reason I can't stop laughing at this guy: |
14:55.17 | shmaltz | http://www.fool.com/News/mft/2006/mft06011321.htm |
14:55.18 | shmaltz | I know it sounds rude, but I can't help it |
14:55.22 | Skumling | [TK]D-Fender: hum okay. any hints for where to search for docs? |
14:55.23 | Davey|Work | [TK]D-Fender, the guy who configured this server, set defaultip to the guys dyndns address, does that sound sane? |
14:55.50 | shmaltz | Funny Quote of the Day - Lucille Ball - "The secret of staying young is to live honestly, eat slowly, and lie about your age." |
14:56.28 | shmaltz | Davey|Work, it should actualy work, there should be a patch for asterisk that should support dyndns |
14:56.35 | [TK]D-Fender | Davey|Work : The remote sip extension should have "host=dynamic". |
14:56.51 | [TK]D-Fender | Davey|Work : That will let him register from "wherever |
14:57.20 | [TK]D-Fender | Davey|Work : don't use defaultIP. |
14:57.50 | Davey|Work | [TK]D-Fender, I figured that was the case, as it never worked ;) |
14:58.20 | Davey|Work | [TK]D-Fender, so the phone knows where the asterisk server is and when its IP changes, it automatically tells Asterisk? |
14:58.29 | Johnnie | Anyone? :) |
14:58.40 | Davey|Work | or, you know, he can power-cycle the phone or something? |
14:58.55 | shmaltz | Dictionary.com Word of the Day - capricious: whimsical; changeable. |
14:58.56 | shmaltz | http://dictionary.reference.com/wordoftheday/archive/2006/01/16.html |
14:59.15 | *** join/#asterisk tengulre (n=tengulre@219.144.204.16) |
14:59.38 | shmaltz | we gota put this one for asterisk as well: |
14:59.39 | shmaltz | Quote of the Day - Michael Jordan - "I can accept failure, but I can't accept not trying." |
14:59.44 | [TK]D-Fender | Davey|Work : It'll register wherever it is and thats that. |
15:00.02 | Davey|Work | [TK]D-Fender, OK, so I remove the defaultip, and set host=dynamic instead? |
15:00.27 | [TK]D-Fender | Davey|Work :: Typically if it looses contact it'll try to re-register at least a few times which will eventually succeed and update * as to its location |
15:00.33 | [TK]D-Fender | Davey|Work : Correct |
15:00.54 | Davey|Work | [TK]D-Fender, it already has host=dynamic, is the defaultip=hostname breaking it? |
15:02.42 | barinoff | i really cant uderstand how can i add account that sip software phone connect to asterisk? it write me that wrong username/password and in console sip show users is empty |
15:02.51 | *** join/#asterisk BeHappy_ (n=willy@host54-203.pool877.interbusiness.it) |
15:03.20 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
15:04.32 | [TK]D-Fender | Davey|Work : no, just reload in CLI |
15:04.42 | Davey|Work | see, I have no idea how to do that :) |
15:05.11 | [TK]D-Fender | Davey|Work : "asterisk -r" and then "reload" |
15:05.16 | Skumling | asterisk -r should pickup the process |
15:05.17 | Davey|Work | thanks :) |
15:05.26 | Skumling | [TK]D-Fender: sorry ;) |
15:05.52 | Davey|Work | is there a way to do like apachectl configtest, to do a sanity check before I do that? :) |
15:07.14 | [TK]D-Fender | Davey|Work : Sanity is sold seperately, not part of the default install :) |
15:07.32 | Davey|Work | wait, we can buy sanity now? Man, nobody told me :/ |
15:07.33 | [TK]D-Fender | Davey|Work : See "Terms & Conditions" |
15:08.34 | backblue | omg no one uses dundi, how the hell do you make a asterisk cluster without ser? |
15:08.45 | *** join/#asterisk gvag11 (n=g@ppp71-adsl-133.ath.forthnet.gr) |
15:08.50 | gvag11 | hi guys |
15:09.31 | gvag11 | Looking for recommended TYAN mobos... Or any other good mobo ... any idea ???? |
15:09.50 | ManxPower | backblue, You mean like as documented in the README files. |
15:09.53 | ManxPower | and on the wiki |
15:09.56 | ManxPower | and on the mailinglists |
15:13.08 | Skumling | [TK]D-Fender: seems like the DISA-stuff can do the trick... |
15:14.18 | xheliox | Can't dial out from Teliax, anyone else experiencing this too? |
15:14.18 | MrChimpy | i got music on hold working! my life's mission is complete! now people can dial in and hear "your mother's got a penis" at will. |
15:14.43 | [TK]D-Fender | Skumling : But that doesn't have anything to do with "transfer" which is what you were actually hoping for. |
15:15.14 | dippo_ | yes xheliox |
15:15.15 | [TK]D-Fender | Skumling : You should use the "#" transfer feature in your dial statements instead for those phones |
15:15.16 | dippo_ | we are down entirely |
15:15.24 | dippo_ | getting retries on outgoing stuff via the IAX trunk |
15:15.30 | dippo_ | i filed a ticket, no response yet |
15:15.34 | dippo_ | i have not been too pleased with teliax so far |
15:15.47 | xheliox | This is fairly unusual, in my experience. |
15:16.03 | dippo_ | our service has been okay, but I get very little in the way of responsiveness from them in person |
15:16.06 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
15:16.11 | dippo_ | as far as answering billing/support questions |
15:16.20 | xheliox | yeah, they have staffing issues |
15:16.33 | Skumling | [TK]D-Fender: okay, but how do I get to enter the tones without "s" being executed as soon as I hit flash? |
15:17.30 | xheliox | Their support number doesn't go through from my mobile. |
15:17.43 | [TK]D-Fender | Skumling : You don't USE FLASH. |
15:17.52 | *** join/#asterisk Splas (i=jwb@206.252.198.100) |
15:18.03 | [TK]D-Fender | use DTMF "3" and add the "t" to your dial lines |
15:18.27 | Skumling | [TK]D-Fender: humm okay :-/... |
15:18.30 | ManxPower | Skumling, There are several ways to do Transfers in Asterisk. |
15:18.55 | ManxPower | There is the evil stupid ugly hack of usint "t" or "T" on the Dial line to handle devices that are too brain dead to support their own transgers. |
15:19.15 | Skumling | ManxPower: okay... tell me all about it, I haven't been able to find much docs on transfers |
15:19.15 | ManxPower | There is the correct method of transfer for the device (FLASH for analog), Transfer button for SIP. |
15:19.24 | [TK]D-Fender | ManxPower: he's running ISDN phones whose "flash" button starts a seperate call. So I think that qualifies as "dumb" |
15:19.29 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
15:19.36 | ManxPower | Skumling, Since very few people use BRI phones with Asterisk, you won't get a lit of useful help here. |
15:19.55 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
15:20.00 | ManxPower | [TK]D-Fender, How do you know it's not just a config issue with the phone? |
15:20.11 | *** join/#asterisk skamp|tee (i=skambar@p54841A45.dip0.t-ipconnect.de) |
15:21.13 | Skumling | ManxPower: okay... humm, but for wireless use, I don't think there's a nice solution without using DECT |
15:21.28 | ManxPower | Skumling, your phone acts weird with you are trying to do a transfer. It might be best if you talk with people that actually have an ISDN BRI phone. Many of them hang out on #asterisk-drinkers. |
15:22.20 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
15:22.38 | Skumling | ManxPower: I think the problem is, that the ISDN base station (where I hook up with BRI to the HFC-card) itself is a small PABX... |
15:22.39 | dorphalsig | Hey! Anybody knows if I can use ASterisk real Time with * 1.0.10? |
15:22.41 | [TK]D-Fender | ManxPower : I don't. |
15:23.08 | ManxPower | dorphalsig, NO! Realtime is only for 1.2 |
15:23.17 | Skumling | What should the flash-button do, when it acts properly? |
15:23.49 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
15:24.23 | ManxPower | Skumling, That depends on the device. For most zap devices it sends a FLASH to Asterisk, for SIP devices it handles everything internally. |
15:24.47 | ManxPower | Skumling, I assume you enabled threeway and transfer in zapata.conf? |
15:24.52 | *** join/#asterisk secure75 (n=mic@dslb-084-057-039-021.pools.arcor-ip.net) |
15:24.57 | Skumling | I have transfer=yes |
15:25.16 | file[laptop] | you need to enable threesomes er I mean threeway calling |
15:25.31 | tdonahue | good morning all |
15:26.01 | *** join/#asterisk _cleric_ (n=dacleric@p5482BFC6.dip0.t-ipconnect.de) |
15:26.05 | Skumling | and if I define "s" to dial another local phone, then when I hit flash the other local phone rings, I pickup, have connection between the two local phones, and when I hangup, the call is routed between the external party and the "new" local phone |
15:27.44 | *** join/#asterisk Bambr (n=Bambr@213-35-236-199-dsl.end.estpak.ee) |
15:28.13 | Skumling | humm wtf. now on a newer base, the flash key doesn't work at all... |
15:28.26 | Skumling | (I've got two different ISDN base stations for testing) |
15:29.53 | ManxPower | Skumling, Well if you insist on not solving the underlying problem then use DISA. |
15:29.55 | jbalcomb | I have a 'ast_expr2.fl: ast_yyerror():' in my log. Is there are way to know which file the syntax error is in? |
15:30.30 | sivana | is T.38 suppose to work with traditional analog fax machines, or completely different hardware? |
15:30.33 | backblue | can i can test extensions? in asterisk console? like dial, just to know its working. |
15:30.57 | Skumling | ManxPower: I would love to solve the underlying problem, but I don't really know where to start :-/ |
15:31.10 | *** join/#asterisk ibob63 (n=hp@bb-87-82-25-51.ukonline.co.uk) |
15:31.18 | ManxPower | Skumling, so use DISA |
15:31.55 | *** join/#asterisk azzie (n=az@azzie.net) |
15:32.22 | ManxPower | Skumling, a file said you also have to enable threeway |
15:33.05 | ibob63 | Does asterisk have a billing module? I am planning to install it in my studio which is shared by four friends. Could we then work out the bill for each phone? |
15:33.24 | iCEBrkr | dooo be doobe dooooooo |
15:33.28 | *** join/#asterisk microcode (n=microcod@70.88.212.117) |
15:33.47 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
15:34.00 | *** part/#asterisk microcode (n=microcod@70.88.212.117) |
15:34.03 | iCEBrkr | jbalcomb: I'm thinking that error you're getting is in a GotoIf() statement |
15:34.12 | tmccrary | Is there something syntactically wrong with this line: exten => 2501,1,Dial(SIP/2501&SIP/2503,12,Ttr) |
15:34.27 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
15:34.36 | iCEBrkr | tmccrary: Looks good.. |
15:34.47 | tmccrary | because it only calls 2501, never 2053 |
15:34.51 | *** join/#asterisk stef2_ (n=stef@65.39.228.5) |
15:34.52 | *** join/#asterisk microcode (n=microcod@70.88.212.117) |
15:35.04 | tmccrary | exten => _25XX,1,Dial(SIP/${EXTEN},12,Ttr) |
15:35.07 | tmccrary | I also have that |
15:35.17 | tmccrary | Would that cause a problem? I need both. |
15:35.22 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
15:35.32 | Katty | uhm. |
15:35.35 | Katty | how do i save in vim? |
15:35.39 | Katty | :sq? |
15:35.40 | ravenpi | iCEBrkr: I *think* it's supposed to be a "|" between the two numbers, not a "&". (I haven't tested it, and could be wrong.) |
15:35.41 | tmccrary | :wq |
15:35.42 | Katty | thanks. |
15:35.43 | tzanger | Katty: :wq |
15:35.44 | fugitivo | :x |
15:35.50 | iCEBrkr | ravenpi: Nope. |
15:35.51 | tzanger | :w is write, :q is quite |
15:35.53 | tzanger | er quit |
15:35.59 | tzanger | if you just want to write use :w |
15:36.00 | fugitivo | :x = shortcut to :wq |
15:36.01 | Katty | fugitivo: :< |
15:36.03 | stef2_ | hello, i'm looking for version 1.3.4 of PWLIb and 1.9.4 of oh323 in order to compile the asterisk oh323 driver, do you have a mirror ? |
15:36.06 | tmccrary | I hate vim with a passion :) |
15:36.11 | iCEBrkr | tmccrary: That may cause a problem. |
15:36.16 | microcode | Anyone familiar with Asterisk on Mac OS X? |
15:36.23 | tmccrary | iCEBrkr: what would? |
15:36.27 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
15:36.28 | tmccrary | oh the pattern? |
15:36.37 | drumkilla | microcode: what's your problem |
15:36.41 | iCEBrkr | tmccrary: _25XX + 2501 |
15:36.55 | tmccrary | I have the 2501 listed first in the config. |
15:37.12 | iCEBrkr | tmccrary: I'm thinking it may 'match' 2501 first, but I'm not 100% sure. |
15:37.15 | pif | anyone using a cisco 7920 wi-phone? |
15:37.27 | microcode | dumkilla: thanks. just curious on how media processing (tone detection and announcement streaming) is done on the Mac. My understanind this that there is no zaptel lib, correct? |
15:37.32 | tmccrary | I wonder if there's anyway to force an exception like that |
15:37.45 | tmccrary | I have 2501 listed first in the config |
15:37.50 | microcode | drumkilla: i am using a softphone to connect to Asterisk via SIP right now |
15:37.53 | drumkilla | microcode: yeah, that has nothing to do with zaptel, actualy. |
15:37.59 | microcode | oh |
15:38.11 | iCEBrkr | tmccrary: I'm thinking it'll match first since it's absolute. |
15:38.13 | stef2_ | Anyone familiar with asterisk-oh323 ? |
15:38.15 | iCEBrkr | not a regexp match |
15:38.18 | tmccrary | I wonder if I go 2501,1 then _25XX,2 ? |
15:38.20 | microcode | so, media processing is in the core of Asterisk then? |
15:38.23 | tmccrary | no that wont work |
15:38.37 | tdonahue | if i have an analog fax machine attached to a channel bank, should I have faxdetect enabled or disabled? |
15:38.44 | drumkilla | microcode: well, most of the time. app_meetme, for example, uses zaptel to do conferencing |
15:38.57 | iCEBrkr | tmccrary: What the hell are you trying to do? |
15:39.17 | tmccrary | I have a set of extensions, 25XX, that I need to be able to call like normal. |
15:39.30 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
15:39.30 | tmccrary | However, when someone calls 2501, I need it to ring two extensions. |
15:39.38 | microcode | drumkilla: thanks. so, in the conf. call w/ zaptel, the zaptel lib provides access to a hardware implementation of the conf. media processing? |
15:39.49 | *** join/#asterisk discofish (n=Rob@rrcs-24-97-85-209.nys.biz.rr.com) |
15:39.51 | backblue | drumkilla: hi, can you give me a hand with dundi? i cant get help from anywhere! |
15:39.52 | Curus | SetCallerPres(prohib) works great on Zap |
15:39.56 | iCEBrkr | tmccrary: Why can't you make it 2601 or 1501 or something different? |
15:40.13 | Curus | When I try it on a different PBX connected via SIP, I have no luck with it |
15:40.14 | tmccrary | I'm thinking I'll have too, man asterisk really needs work still. So many hacks and work arounds. |
15:40.31 | [TK]D-Fender | tmccrary : You don't want to use 25XX. That allows you to dial non-valid extensions. You also DEFINATELY don't want to mix 25XX and 2501 together. |
15:40.32 | tmccrary | It really shouldn't be 1.0 yet I don't think. |
15:40.33 | drumkilla | microcode: well, it uses zaptel's timing to assist in the conferencing implementation. there is no hardware conferencing available with zaptel at this point, but there could be in the future ... |
15:40.36 | iCEBrkr | tmccrary: Hacks and work arounds? Just don't do shit all jacked-up and you'll be fine. |
15:40.38 | file[laptop] | it's rather easy to do conferencing with zaptel... you create a pseudo channel for the conference, write in signed linear and read in signed linear (you get it mixed) |
15:40.51 | *** part/#asterisk barinoff (i=izida@82.162.60.62) |
15:41.05 | tmccrary | Well, there should be a mechanism for calling two extensions at once, that's very common. |
15:41.08 | stef2_ | Where can I download version 1.3.4 of PWLib |
15:41.13 | Curus | Isn't the Zap-connected asterisk supposed to see that the CLID presentation has been prohibited in the SIP call? |
15:41.23 | microcode | drumkilla: thanks for the help! |
15:41.33 | drumkilla | microcode: no problem :) |
15:41.40 | iCEBrkr | tmccrary: You people forget, just because it's a highly configurable PBX software that you think it can do anything you want. Think about all the other PBX's out there that can't do what Asterisk does.. |
15:41.49 | tmccrary | Thank you for the help though iCE, I don't want to come off as rude. |
15:42.03 | iCEBrkr | No, not rude.. Just nieve. |
15:42.05 | drumkilla | backblue: try this link, it's very helpful ... http://leifmadsen.com/papers/dundi-intro.pdf |
15:42.06 | tmccrary | I know, some of my code is in asterisk right now (albeit not a lot) |
15:42.07 | iCEBrkr | :P |
15:42.45 | iCEBrkr | There are a lot of people who come in here with all these great expectations cuz Asterisk is so configurable and you can make it do jumping jacks.. but yet they really don't understand WTF they're trying to do. |
15:42.52 | *** part/#asterisk stef2_ (n=stef@65.39.228.5) |
15:42.55 | tmccrary | Honestly, I think I'm going to write something to allow calling multiple extensions, I really would like it. |
15:43.11 | iCEBrkr | ...and on top of that, it seems like people want to be all secretive about their what they're trying to do so we can't help them do it the right way |
15:43.38 | iCEBrkr | tmccrary: It dials multiple extensions the way you have it.. It's just you're whacked on your extension naming. |
15:43.41 | backblue | drumkilla: dundi looks for one existing extension, or it can looks for one REGISTERED peer? i have a problem because i have 2 asterisks and mobile extensions, that have to register on both servers. |
15:43.44 | jbalcomb | iCEBrkr: i see. its an unexpected TOK_EQ then says Input: = 0800. I did a grep on the configs. 71 lines with 0800. |
15:44.00 | backblue | drumkilla: i'm trying to take a look on dundi if it solves my problem. |
15:44.15 | jbalcomb | iCEBrkr: they all says "exten => s,6,GotoIf($[${dnd} = 0800]?10:7)" |
15:44.35 | iCEBrkr | tmccrary: and seriously-- depending on how many extensions you have, don't be lazy and use _25XX for matching.. Give each extension a 1 liner to a macro |
15:44.43 | tmccrary | Yeah, but there should be a mechanism for dialing multiple extensions at once. Right now, I have to create all kinds of crazy extra extensions and generally hack around the config to get things to work. Just because its the way asterisk does it right now, doesn't mean it's the best way. |
15:44.48 | drumkilla | backblue: it looks for the existance of an extensions |
15:44.51 | iCEBrkr | jbalcomb: ${dnd} appears to be blank.. |
15:44.57 | [TK]D-Fender | tmccrary : You CAN dial both at the same time, and your first pasted sample was FINE. it was just CONFLICTING with your pattern match! |
15:45.12 | iCEBrkr | tmccrary: What [TK]D-Fender said. |
15:45.13 | [TK]D-Fender | tmccrary : Do NOT mix 2501 and 25XX! |
15:45.27 | backblue | drumkilla: theres is nothing to lookup for registered peers on the time we try to make a call? |
15:45.44 | iCEBrkr | jbalcomb: exten => s,6,GotoIf(X$[${dnd} = X0800]?10:7) |
15:45.49 | iCEBrkr | jbalcomb: Try that instead |
15:46.00 | drumkilla | backblue: well, there is ... now bare with me here :) |
15:46.14 | jbalcomb | iCEBrkr: this is the line before that "exten => s,4,DBget(dnd=dnd/SIP/63)" |
15:46.28 | drumkilla | backblue: are you using realtime? |
15:46.30 | iCEBrkr | jbalcomb: It's still possible that DND won't be set. |
15:46.42 | dpryo | The ?: notation in asterisk, is the same as the one in c? ( <true/false> ? <if true> : <if false> ) |
15:46.55 | iCEBrkr | jbalcomb: Just stuff an X before the variables you're comparing |
15:47.01 | iCEBrkr | jbalcomb: like I pasted |
15:47.15 | jbalcomb | iCEBrkr: will do. whats the big x do? |
15:47.37 | iCEBrkr | jbalcomb: Nothing.. It's just a place holder to fill in any NULL or nothing values |
15:47.42 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
15:47.58 | [TK]D-Fender | jbalcomb : You are using < 1.2 commands and trying to use 3 dimensions in DB1 ! "dnd/SIP/63" is NOT valid. |
15:48.06 | iCEBrkr | jbalcomb: It's just a warning you're getting anyhow, so it's not really hurting anything |
15:48.16 | [TK]D-Fender | jbalcomb : DB1 is FAMILY/KEY. |
15:48.36 | tmccrary | Wouldn't it make more sense for asterisk to process non matching rules first and THEN match any regex'd type rules? |
15:48.42 | tmccrary | I'm going to write a patch. |
15:48.54 | tmccrary | or at least have an option to do so. |
15:49.03 | iCEBrkr | tmccrary: See, that's your problem right there, you're hell-bent on doing it the fucked up way. |
15:49.20 | tmccrary | wtf are you blathering about |
15:49.33 | wunderkin | anyone else getting mailing list posts? |
15:49.48 | anderiv | wunderkin: nope |
15:49.52 | tmccrary | I would prefer to have similar extensions for everyone in the same area, that's just common sense. |
15:49.56 | iCEBrkr | I'm blathering about you breaking Asterisk :P |
15:50.16 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
15:50.17 | tmccrary | How exactly would following common sense break asterisk? |
15:50.20 | backblue | drumkilla: i'm not, but i can use it, and i pretend to, my problem its nat issues. i dont know how is realtime with nat. |
15:50.21 | iCEBrkr | Why would you make a series of extensions for people and then have a GROUP in the same series? |
15:50.34 | tmccrary | Exceptions are handy. |
15:50.58 | drumkilla | backblue: nat would have no effect on using realtime. it's just storing the config in a database as opposed to text files. |
15:51.07 | jbalcomb | [TK]D-Fender: ok, i don't get the /programming/ part of Asterisk yet but this is a setup I am taking over from people who no longer exist. :/ Thanks. |
15:51.11 | iCEBrkr | Joe, Jim, Bob, and Mike have 2501 to 2504. Why would you make 2505 ring a group of phones when 250X are designated as individual's extensions. |
15:51.35 | backblue | drumkilla: hoo, ok i can use database storage, no problem with that, but how that solves my problem? |
15:51.36 | iCEBrkr | [TK]D-Fender: Yea, Jim was shoved into this Asterisk project without any notes :) |
15:51.39 | tmccrary | Lets say, sally who is 2505, needs to be reachable in two locations. |
15:51.56 | jbalcomb | iCEBrkr: thanks man. im trying to clean things up as I go about learning this setup. |
15:51.57 | tmccrary | But Sally SHOULDNT need two different extensions. |
15:52.01 | iCEBrkr | tmccrary: That I understand. |
15:52.02 | drumkilla | backblue: well, it should work with "rtupdate" turned on in sip.conf |
15:52.06 | drumkilla | backblue: another option is this ... |
15:52.14 | [TK]D-Fender | tmccrary : Pastebin your whole dial-plan and I'll see what I can do. |
15:52.18 | [TK]D-Fender | ~pb |
15:52.21 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
15:52.28 | tmccrary | okay one sec, thank you. |
15:52.31 | drumkilla | backblue: you can use DUNDi in combination with using the regexten feature in sip.conf. That will, the extension will only exist when the peer is registered |
15:52.39 | iCEBrkr | tmccrary: So is there a problem with creating 2501, 2502, 2503 extensions individually?? |
15:52.59 | *** join/#asterisk blitz[class] (n=lmadsen@199.227.185.35) |
15:53.08 | tmccrary | nope |
15:53.21 | tmccrary | really, I just want to be able to create an exception within a matching rule. |
15:53.27 | iCEBrkr | I mean if you have 100 extensions (2500-2599) doing that could be a pain in the ass.. |
15:53.47 | iCEBrkr | But if you only have 10 (2500-2509) then, what's the issue? |
15:54.17 | tmccrary | I don't see why that would cause a problem?. |
15:54.24 | iCEBrkr | jbalcomb: what version of asterisk they got over there? |
15:54.34 | Curus | Hmm actually SetCallerPres(prohib) deletes all caller id information and replaces it with unknown |
15:54.40 | iCEBrkr | tmccrary: The problem is having to type in all that shit :) |
15:54.47 | Curus | That's entirely useless, we have to send it for emergency services |
15:55.00 | tdonahue | if i have an analog fax machine attached to a channel bank, should I have faxdetect enabled or disabled? |
15:55.12 | iCEBrkr | tmccrary: and your 'exception' / 'patch' would just confuse people more than they already are.. |
15:55.17 | Curus | (For SIP, that is, for Zap it works perfectly) |
15:55.39 | tmccrary | but its just for special cases, so lets say most phone/calls just use the pattern match rule. However, when you want to do special things within that range of extensions, you can create a rule that is checked first. |
15:55.59 | tmccrary | so like 2-3 phones have special handling for calls (for multiple recipients, etc) |
15:56.10 | iCEBrkr | tmccrary: and your 'exception' / 'patch' would just confuse people more than they already are.. |
15:56.13 | iCEBrkr | ^^^^ |
15:56.20 | *** join/#asterisk MommomeryCliff (n=willy@host230-24.pool873.interbusiness.it) |
15:56.23 | [TK]D-Fender | tmccrary : Pastebin it.... |
15:56.25 | tmccrary | I'm sure you could say that about anything that makes something better. |
15:56.42 | backblue | drumkilla: can we talk better in private? |
15:56.48 | tmccrary | Don't fear change my friend. :) |
15:56.55 | ManxPower | If they are in the same context, then the closest matching exten => line will be the one that's used. |
15:57.17 | tmccrary | But that's not the case Manx. |
15:57.18 | iCEBrkr | tmccrary: I fear confusion.. |
15:57.22 | ManxPower | extensions included with include => are always considered "least specific" |
15:57.36 | dippo_ | xheliox: you heard anything from teliax yet? |
15:57.38 | tmccrary | I have one rule that is 2501 and it gets ignored for a _25XX rule. |
15:57.48 | [TK]D-Fender | tmccrary : How many extensions do you have? |
15:57.50 | ManxPower | tmccrary, in the same context? |
15:57.53 | tmccrary | 2501 is much more explicit than _25XX |
15:57.54 | tmccrary | yes |
15:58.00 | iCEBrkr | tmccrary: and if you hang out in here long enough, you'll find out that people come up with the craziest shit and they're severly confused on how things work in extensions.conf |
15:58.02 | jbalcomb | iCEBrkr: Asterisk 1.2.1 |
15:58.07 | ManxPower | tmccrary, is 2501 before the pattern in extensions.conf? |
15:58.11 | iCEBrkr | jbalcomb: Ok yea, you're gonna have to update the DBGet() stuff. |
15:58.15 | iCEBrkr | jbalcomb: It's completely changed. |
15:58.16 | tmccrary | Manx: Yes. |
15:58.25 | iCEBrkr | jbalcomb: Checkout the Wiki for the new syntax |
15:58.30 | ManxPower | tmccrary, I don't know what the problem is, since *I* use this feature all the time. |
15:58.33 | [TK]D-Fender | iCEBrkr : I've said as much including the error in formatting |
15:58.51 | iCEBrkr | hrrm? |
15:58.54 | ManxPower | For example I have an exten => _XXXX,1,Macro(disconnected) at the bottom of my extensions.conf |
15:58.56 | *** join/#asterisk Run (n=BIGGIRL^@85.108.145.224) |
15:59.18 | jbalcomb | iCEBrkr: sounds like fun. immature software mixed with lazy ass admins is good for creating new nueral patterns. |
15:59.28 | ManxPower | tmccrary, what happens when you comment out the _25XX lines? |
15:59.38 | tmccrary | let me paste the config |
15:59.47 | iCEBrkr | jbalcomb: naaa, I dunno about that.. 1.0.x code is significantly differen than 1.2.x |
15:59.49 | ManxPower | tmccrary, what happens when you comment out the _25XX lines? |
16:00.00 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
16:00.13 | iCEBrkr | jbalcomb: ...and it's all for the better.. You'll figure it out, Just dig through the Wiki for the DB() functions. |
16:00.14 | tmccrary | http://pastebin.com/508268 |
16:00.22 | iCEBrkr | I gotta head to lunch to beat the parade. |
16:00.22 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
16:00.32 | jbalcomb | iCEBrkr: voip-info.org wiki? |
16:00.38 | [TK]D-Fender | tmccrary ..... is that an AMP created context? |
16:00.39 | iCEBrkr | jbalcomb: yeah |
16:00.49 | jbalcomb | iCEBrkr: thanks. good on ya mate. |
16:00.50 | iCEBrkr | jbalcomb: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
16:00.56 | iCEBrkr | Specifically |
16:01.02 | tmccrary | No |
16:01.11 | backblue | drumkilla: that really solves my problems, nice, but i still have one problem that its mailbox, where its mailbox located in the end? |
16:01.19 | iCEBrkr | ok, I'm really leaving now :) |
16:01.21 | [TK]D-Fender | tmccrary : Hom many extesions do you have, and no-one has voicemail? |
16:01.27 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
16:01.29 | jbalcomb | iCEBrkr ciao. |
16:01.51 | tmccrary | No, I disabled voicemail for the time being. |
16:02.09 | *** part/#asterisk ibob63 (n=hp@bb-87-82-25-51.ukonline.co.uk) |
16:02.14 | [TK]D-Fender | and the # of extensions? |
16:02.21 | flujan | hi guys... We are setting up a test environment to work with asterisk and hard voip phones... So asterisk will work as a gateway between the phones. Actually we have 150 internal analog phone points. We expect to change this all to voip. |
16:02.30 | tmccrary | 3 |
16:02.34 | flujan | I know its also possible to work with adapters... |
16:02.39 | ManxPower | tmccrary, Do you REALLY want callers to be able to transfer themselves? |
16:02.41 | flujan | which is the best solution? |
16:02.42 | [TK]D-Fender | tmccrary : Not worth and pattern match. |
16:02.59 | tmccrary | I don't want to hack it up and being adding every extension manually. |
16:03.01 | flujan | and which is the cheaper solution? |
16:03.16 | ManxPower | tmccrary, nevermind, aswer my other question first. |
16:03.31 | tmccrary | I'd rather do this the RIGHT way, which apparently is impossible without the ability to add a rule that overrides the match (i.e. 2501,1) |
16:03.47 | ManxPower | tmccrary, people do it all the time. |
16:03.54 | [TK]D-Fender | tmccrary : You have 3... if you jumped to 10 its 7 more lines... whats the big deal? its a cut & paste job. Pastebin you entire extensions.conf and sip.con and I'll work a small miracle for you. |
16:04.17 | tmccrary | Again, I'd rather have it be done the right way. I'm sure I could come up with all kinds of crazy work arounds. |
16:04.31 | *** join/#asterisk m160858 (n=ubuntu@200.89.12.46) |
16:04.51 | tmccrary | I don't want to add every extension manually, whats the point of having a matching/regex feature in asterisk if I have to add them manually. |
16:04.54 | dippo_ | xheliox: we appear to be back up on teliax fwiw |
16:05.02 | m160858 | hi everyone |
16:05.03 | tmccrary | Sounds like something is broken to me.... |
16:05.04 | ManxPower | tmccrary, I see you are not listening. |
16:05.13 | m160858 | i have problems with my asterisk |
16:05.19 | tmccrary | Sorry manx, I am talking to Fender. |
16:05.20 | [TK]D-Fender | tmccrary : There is no way to prove the extensions validity! So forget pattern match! |
16:05.21 | m160858 | y try to install asterisk at home |
16:05.22 | tmccrary | What was your other question? |
16:05.29 | m160858 | but i can't turn on |
16:05.41 | ManxPower | For the THIRD time, what happens when you comment out the _25XX line and issue a "reload" on the asterisk CLI? I'm not asking again. |
16:05.58 | [TK]D-Fender | m160858 : For A@H plase go to #amportal |
16:05.58 | tmccrary | Chances are, that will work. |
16:06.01 | SDGL | First off, I appologize for the lengthy question... here it goes. I've been investigating a random deadlock within Asterisk for about a month, and I seem to have it narrowed it down to the call parking feature. We park about 50 calls per day, and the feature seems to be working pretty good, except that once in a while, while parking a call, the Asterisk box completly deadlocks, and a killall -9 asterisk is the only thing that will bring it back to life. It |
16:06.06 | ManxPower | tmccrary, Try it. |
16:06.08 | tmccrary | Let me try it for you and reiterate. |
16:06.09 | *** join/#asterisk daCount (n=cosg@19-103.241.81.adsl.skynet.be) |
16:06.14 | m160858 | apears like it doesn't exists |
16:06.34 | m160858 | oh |
16:06.36 | m160858 | thanks |
16:07.16 | [TK]D-Fender | tmccrary : Add an "i" exten in your phones context and check if the EXTEN matches 25XX and playback "thanks for trying to dial something that LOOKS like an extension but isn't!" |
16:07.20 | flujan | hi guys... We are setting up a test environment to work with asterisk and hard voip phones... So asterisk will work as a gateway between the phones. Actually we have 150 internal analog phone points. We expect to change this all to voip. |
16:07.42 | flujan | I want some recomendation to work with hard phone... a specific product |
16:07.42 | ManxPower | flujan, Great! |
16:07.50 | ManxPower | I recommend Polycom |
16:08.09 | [TK]D-Fender | flujan : Best solution change them all to PoE voip hardphones on a dedicated LAN. |
16:08.11 | *** join/#asterisk SDGL (n=sdgl@64.5.206.131) |
16:08.24 | SDGL | Sorry... lost my connection for a minute |
16:08.27 | daCount | any idea about a soft phone wich works with openbsd without too much hassle? |
16:08.29 | [TK]D-Fender | flujan : I will second ManxPower's suggestion for Polycom. |
16:08.42 | ReX | what are the libraries necessary to compile chan_btp plz? |
16:09.14 | ManxPower | ReX, almost nobody uses that channel since it's still alpha quality. The README did not give any clues? |
16:09.28 | flujan | ok |
16:09.41 | ReX | no ManxPower |
16:10.01 | ManxPower | ReX, how about the mailing list archives? |
16:10.10 | mut | paycheck taxes are teh suck |
16:10.11 | flujan | ManxPower: which is the best sollution? Work with adapter in the analog phones or change all to the voip hard phones. |
16:10.24 | ManxPower | flujan, that is ip to you. |
16:10.44 | ReX | I did not find anything |
16:10.53 | ManxPower | flujan, you should build a prototype system first |
16:11.25 | ManxPower | Results 1 - 10 of about 302 from lists.digium.com for bluetooth. (0.32 seconds) |
16:11.40 | trixter | I like to use a rapid prototyping system when making prototypes, that way I dont have to wait for someone to mail me a case instead I can have a custom case made right on the sopt :P |
16:12.21 | [TK]D-Fender | flujan : Like I said, change them all for VoIP hard-phones (suggest Polycom) using PoE on a dedicated LAN. |
16:12.42 | ManxPower | [TK]D-Fender, seems like overkill to me. |
16:12.51 | *** part/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
16:14.30 | *** join/#asterisk GillesR_IMG-IT (n=GillesR_@64.5.206.131) |
16:14.56 | [TK]D-Fender | ManxPower : He asked what the BEST solution was. Not the CHEAPEST. |
16:15.03 | flujan | ManxPower: yes, I will bought one... :D and [TK]D-Fender what to you mean by PoE? |
16:15.48 | jbalcomb | PoE is awesomeness. Power over Ethernet, no AC adapter |
16:15.49 | flujan | [TK]D-Fender: Actually we work with fix ip address in the computers |
16:15.50 | [TK]D-Fender | flujan : Power Over Ethernet. This is a standard that allows you to power your phones off of a special switch instead of having to plug them into a power bar at each desk. |
16:15.57 | *** join/#asterisk razu (n=razu@80-235-90-19-dsl.prn.estpak.ee) |
16:16.36 | [TK]D-Fender | tmccrary : Offer still open for me to re-vamp your setup.... |
16:16.44 | jbalcomb | [TK]D-Fender I heard the polycom phones dont do DTMF via SIP INFO, true? |
16:16.54 | flujan | [TK]D-Fender: with hard voip phones we will need to create proportional IP address. is it right? |
16:17.26 | [TK]D-Fender | jbalcomb : Not sure, but I know they do RFC2833 which is preferred. |
16:17.50 | [TK]D-Fender | flujan : They can use DHCP or static IP. I suggest using DHCP for everything personally. |
16:18.45 | Luke-Jr | Any idea on recovering a PAP2-NA? ethernet/blue & power/red lights are lit solid |
16:19.08 | jbalcomb | [TK]D-Fender gotcha. we seem to be using 'via SIP INFO' but I don't know if there is a real reason. |
16:19.14 | flujan | And about adapters to analog phones... I also need suggestions since I must create to proposal to my boss. :P |
16:20.00 | [TK]D-Fender | jbalcomb : programmer error <- :d |
16:20.22 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
16:20.37 | [TK]D-Fender | flujan : Another which might be easier for you would be to buy Norstar Digital sets and use them on Citel SIP gateways. |
16:20.51 | [TK]D-Fender | flujan : Check out http://www.atacomm.com |
16:20.53 | flujan | well, In order to build a prototype env two hard ip phones and a machine is enought, isn't it? |
16:21.08 | [TK]D-Fender | flujan : For just a test of what * can do, sure. |
16:21.11 | jbalcomb | [TK]D-Fender haha.. i dont doubt it. i like the most that know one here has any idea why we are doing anything the way we are doing it |
16:21.20 | jbalcomb | s/know/no/ |
16:21.31 | backblue | when i make reload somemodule.so it does not shows "parsing ... blah" why? |
16:21.53 | jbalcomb | jbot please dont do that anymore. its annoying. |
16:22.49 | [TK]D-Fender | jbalcomb : Just don't add the trailing "/" then |
16:23.05 | [TK]D-Fender | s/Just/Ignored ! |
16:23.07 | Johnnie | Anyone familiar with ztdummy and strange compiling issues? |
16:23.10 | [TK]D-Fender | see? |
16:23.19 | [TK]D-Fender | jbalcomb : Just don't add the trailing "/" then |
16:23.23 | [TK]D-Fender | s/Just/Ignored/ |
16:23.25 | flujan | [TK]D-Fender: thanks... I will buy this and set up a prototype env. |
16:23.28 | [TK]D-Fender | there you have it |
16:23.37 | flujan | ManxPower: thanks for the help. |
16:23.42 | [TK]D-Fender | flujan : What are you looking to buy? |
16:24.08 | jbalcomb | [TK]D-Fender gotcha |
16:24.15 | flujan | [TK]D-Fender: two voip phones and two adapters to test with our analog system. |
16:24.19 | dippo_ | is iax2 trunking encrypted at all? if one could tcpdump an iax2 stream could you decode audio packets from it? |
16:24.26 | jbalcomb | s/gotcha/jbot is a tool of Satan/ |
16:24.35 | flujan | [TK]D-Fender: first, we will set up a env to work with the internal phones |
16:24.59 | *** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it) |
16:25.03 | [TK]D-Fender | flujan : I typically don't suggest analog, but it is cheaper. For your test I suggest 1 x SPA-2002, and 2 Higher-end Polycom phones (501 or 601) |
16:25.08 | dily | hi@all |
16:25.15 | flujan | [TK]D-Fender: but we have 3 E1 and later we will adapt it to do calls outside the internal phone system... |
16:25.45 | [TK]D-Fender | flujan : You are using channel-banks on your E1's internally? |
16:25.51 | *** join/#asterisk fulgas (n=fulgas@209.8.233.229) |
16:26.48 | jbalcomb | [TK]D-Fender if im reading this righ, DBGet(dnd=dnd/SIP/63) is depricated for Set(dnd=${dnd(SIP/63)})? |
16:28.50 | Davey|Work | which ports do you typically need to open for SIP? |
16:29.42 | flujan | [TK]D-Fender: hum... I don't know... There is a company wich take care of our telepnony... We will make it internal now. |
16:29.56 | *** join/#asterisk dily_ (n=dily@ip-85-108.sn2.eutelia.it) |
16:29.57 | brookshire | davey: lots of them :) |
16:29.57 | [TK]D-Fender | jbalcomb : Very wrong. |
16:29.58 | flujan | [TK]D-Fender: is there a way to me discover that? |
16:30.10 | tdonahue | if i have an analog fax machine attached to a channel bank, should I have faxdetect enabled or disabled? |
16:30.33 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
16:30.34 | *** mode/#asterisk [+o denon] by ChanServ |
16:30.44 | flujan | [TK]D-Fender: we can receive external calls with private real numbers in the company. |
16:30.46 | dily_ | i recevice an error when i hangup an incomingcall, -- Incoming call: Got SIP response 503 "Server error" back from xxx.xxx.xxx.xxx. Any ideas? |
16:30.50 | [TK]D-Fender | jbalcomb dnd/SIP/62 does not look right unless the last "/" gets included in the KEY. Which looks confusing at best |
16:31.03 | ravenpi | Couple Polycom questions: 1) Any idea how to disable that occasional MWI "warble"? [Not the stutter -- I got rid of that.] 2) How about re-programming the keys? Found the place in the manual, but it's not very descriptive -- a sample of the <keys> section would help (want to have it dial the paging extension). |
16:32.25 | flujan | [TK]D-Fender: for instance, if someone wants to speak with me internally they just type 502 |
16:32.45 | flujan | [TK]D-Fender: to receive external calls... they use a real telephone number |
16:33.27 | backblue | Davey|Work: you can say that in your conf files. |
16:33.49 | backblue | you just specify a range and add the range to your firewall rules |
16:34.30 | [TK]D-Fender | ravenpi : Which "keys" are you looking to reprogram? |
16:35.01 | ravenpi | [TK]D-Fender: Preferably, the "Services" key (#31 on the 501, according to the docs). |
16:35.02 | *** join/#asterisk SDGL_ (n=SDGL@64.5.206.131) |
16:35.18 | *** part/#asterisk SDGL_ (n=SDGL@64.5.206.131) |
16:35.19 | Davey|Work | backblue, Ok, thanks |
16:35.24 | *** join/#asterisk SDGL_ (n=SDGL@64.5.206.131) |
16:35.26 | Davey|Work | backblue, which config? :) |
16:35.30 | [TK]D-Fender | flujan : Ok I'm not sure what kind of equipement you have now. Your wording is confusing where you are mixing up equipment used for your internal phone and technology used for your outside phone link |
16:35.37 | ravenpi | [TK]D-Fender: Though I'll use the bottom-most soft key if I need to... |
16:36.17 | *** part/#asterisk GillesR_IMG-IT (n=GillesR_@64.5.206.131) |
16:36.21 | *** part/#asterisk SDGL_ (n=SDGL@64.5.206.131) |
16:36.51 | [TK]D-Fender | ravenpi : Not sure how you would go about changing that key.... And what do you mean the "bottommost soft-key"? |
16:36.51 | Mimmus | does anyone use Digium's Asterisk Enterprise Edition in the world? |
16:37.19 | *** join/#asterisk GillesR_IMG-IT (n=GillesR_@64.5.206.131) |
16:37.24 | *** join/#asterisk SDGL (n=SDGL@64.5.206.131) |
16:37.42 | backblue | Davey|Work: rtp.conf, we need on ip_conntrack_rtp :P |
16:37.48 | backblue | s/on/one |
16:38.18 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
16:39.01 | ravenpi | [TK]D-Fender: *hmmm* Oh, well. The three keys on the left (two of which I have assigned to extensions) -- I'll make the third one dial away. |
16:40.03 | [TK]D-Fender | ravenpi : If you don't use the "messages" button to pick up voicemail you could use that.... |
16:40.27 | ravenpi | [TK]D-Fender: we do, alas. But thanks... |
16:40.30 | *** join/#asterisk ping1 (n=DLBaker@67-133-167-72.dia.cust.qwest.net) |
16:41.16 | [TK]D-Fender | ravenpi : oh well.. Acutally what DOES "services" do on an IP50x? Its not supposed to have the microbrowser.... |
16:41.43 | *** part/#asterisk SDGL (n=SDGL@64.5.206.131) |
16:42.05 | ravenpi | [TK]D-Fender: darned if I know. Pushing it does zilch -- which is why, combined with its ambiguous name, I thought it was an ideal candidate for re-programming. |
16:42.18 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
16:42.21 | *** join/#asterisk ping1 (n=DLBaker@67-133-167-72.dia.cust.qwest.net) |
16:42.28 | [TK]D-Fender | ravenpi : Could be. They may also be planning MB support... who knows. |
16:42.47 | dily_ | anyone know how to resolve this error -- Incoming call: Got SIP response 503 "Server error" back from xxx.xxx.xxx.xxx. ? plz |
16:43.05 | flujan | [TK]D-Fender: we use http://www.zox.com.br/tz20.htm |
16:43.13 | *** join/#asterisk IMG-SD (n=IMG-SD@64.5.206.131) |
16:43.14 | ravenpi | [TK]D-Fender: Thanks for the help. So: any idea on the MWI "warble"? I know *some* user's gonna want it disabled... |
16:43.21 | flujan | [TK]D-Fender: I don't know how to call this in english... :D |
16:43.47 | [TK]D-Fender | ravenpi : I looked it up, but don't see anything for the warble... |
16:44.08 | festr_ | hello, i've problem with ringing tone between transfering two fxs. Scenario-> IAX -> asterisk FXS1 (ringing) press # and transfer to FXS2 (ringing, but this ring is not heared in IAX channel), what can be wrong? is it bug? |
16:44.34 | ravenpi | [TK]D-Fender: Yeah, I've been Googling for a while... Oh, well: when I only have two things to whine about after a rollout, that's a pretty good day. |
16:44.37 | [TK]D-Fender | flujan : You already own a lot of those? |
16:44.46 | [TK]D-Fender | ravenpi : hold on... |
16:45.03 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-102.dynamic.qsc.de) |
16:46.37 | backblue | Davey|Work: well you have a patch for iptables if is your case. |
16:46.38 | [TK]D-Fender | ravenpi : I think you can kill it in sip.cfg under the <ALERTING> section by setting it to "silence" |
16:47.02 | [TK]D-Fender | ravenpi : Also under "miscellaneous" |
16:47.32 | ravenpi | [TK]D-Fender: AWESOME. I owe you a <beverage of choice>. |
16:47.42 | [TK]D-Fender | ravenpi : no biggie |
16:48.14 | *** join/#asterisk jimmy_deanPB_ (n=jhodapp@indianalifesciences.com) |
16:49.50 | *** join/#asterisk juice (n=juice@209.33.105.105) |
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16:55.43 | *** part/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu) |
16:56.05 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
16:56.33 | *** join/#asterisk jero (n=sflphone@savoirfairelinux.net) |
16:57.00 | *** part/#asterisk secure75 (n=mic@dslb-084-057-039-021.pools.arcor-ip.net) |
16:58.56 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
16:59.11 | *** join/#asterisk Jedirl (n=hhgds4@213.162.200.226) |
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17:00.49 | *** join/#asterisk BombeR (n=GuRBeT@stjhnf0112w-142162196108.pppoe-dynamic.nl.aliant.net) |
17:01.26 | flujan | [TK]D-Fender: sorry, I was researching about our telephony system. :) |
17:01.51 | flujan | [TK]D-Fender: we have a dialogic d/600 jct - 2 e1 card |
17:02.01 | flujan | [TK]D-Fender: and a d/300 jct |
17:02.20 | flujan | [TK]D-Fender: both from dialogic |
17:02.45 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
17:04.02 | [TK]D-Fender | flujan : Well You can probably forget about that card in your new setup. |
17:04.54 | *** part/#asterisk rstandy (n=rastandy@d83-176-116-85.cust.tele2.it) |
17:04.59 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
17:05.14 | flujan | [TK]D-Fender: which card do you recommend in the new setup? First to a prototype env. |
17:05.37 | *** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
17:05.42 | nextime | anyone using ooh323c? |
17:05.44 | flujan | [TK]D-Fender: and later on to replace them. :D |
17:06.08 | [TK]D-Fender | flujan : Ok what do you use for phones right now, and what do you use for lines? Can you run new Cat5 for VoIP hard phones? |
17:06.28 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
17:06.51 | flujan | what to you mean by Cat5 |
17:06.52 | flujan | ? |
17:06.59 | docelm0 | SIGH ETHERNET! |
17:07.06 | docelm0 | 10-100-1000 BASE T |
17:08.00 | [TK]D-Fender | flujan : And how many total phones / lines? |
17:08.13 | jarrod | anyone familiar with using cisco as5400/7x00 as media gateways? |
17:08.28 | jarrod | how do I take an incoming call on a PRI and turn around and forward out another channel? |
17:08.40 | flujan | [TK]D-Fender: 90 phone lines... :) |
17:09.04 | flujan | [TK]D-Fender: we use headsets like that I showed you. |
17:09.19 | [TK]D-Fender | flujan : Ok, that'd be the 3 E! you were mentioning earlier? |
17:09.45 | [TK]D-Fender | flujan : Are the headsets attached to another phone or are they actually full phones themselves? |
17:10.15 | jpablo | hey people, I'm getting this error when trying to ztcfg -v a dual e1 card: |
17:10.17 | jpablo | SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
17:10.18 | jpablo | 31 channels configured. |
17:10.18 | jpablo | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
17:10.37 | jarrod | check /etc/zaptel.conf |
17:11.03 | jpablo | i got: |
17:11.06 | jpablo | span=1,1,0,cas,hdb3 |
17:11.07 | jpablo | cas=1-15:1101 |
17:11.07 | jpablo | dchan=16 |
17:11.07 | jpablo | cas=17-31:1101 |
17:11.09 | [TK]D-Fender | jpablo : Maybe youshould paste the line it says is bad. <- |
17:12.16 | jpablo | [TK]D-Fender, excuseme ? |
17:12.17 | *** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
17:12.31 | flujan | [TK]D-Fender: the can work as both. Depends on the software configuration... http://www.zox.com.br/download/tz-20-manual.doc |
17:12.44 | *** join/#asterisk UlbabraB_ (n=salama@host241-43.pool8172.interbusiness.it) |
17:12.48 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
17:13.01 | flujan | [TK]D-Fender: in this link You have full specification about our produtcts... |
17:13.09 | PoWeRKiLL | hi |
17:14.12 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
17:14.53 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
17:15.04 | flujan | [TK]D-Fender: I found a english version http://www.zox.com.br/ingles/download/tz-20-manual.doc |
17:15.25 | PoWeRKiLL | someone have a the Asterisk Sound List in french ? |
17:15.52 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
17:15.56 | [TK]D-Fender | jpablo : I don't see "cas" as a valid parameter anywhere.... |
17:16.07 | jbalcomb | [TK]D-Fender this DB1 is tad confusing. the entry i have is "/SIP/Registry/6310.0.101.181:5062:3600:63:sip:63@10.0.101.181:5062 |
17:16.47 | [TK]D-Fender | jbalcomb : Thats created automatically by * to store SIP registration info. |
17:16.49 | *** join/#asterisk MommomeryCliff (n=willy@host230-24.pool873.interbusiness.it) |
17:16.55 | jbalcomb | [TK]D-Fender so i dont get how that correlates to a check on dnd/SIP/63 matching 0800 |
17:17.12 | [TK]D-Fender | jbalcomb : it DOESN'T have anything to do with your DND. |
17:17.19 | jpablo | [TK]D-Fender, you really don't know about configuring e1s, do you ... ? how else do i configure a cas group ? |
17:17.48 | [TK]D-Fender | jpablo : Check the WIKI and the sample file. I don't know them by heart for that tech... only N/A PRI |
17:17.52 | *** join/#asterisk roulduke_ (i=bhcj49ft@p508D28AE.dip0.t-ipconnect.de) |
17:18.10 | [TK]D-Fender | jbalcomb : Pastebin your extensions.conf and I'll take a lok |
17:18.39 | jbalcomb | [TK]D-Fender hrmm.. ok, so dnd/SIP/63 doesnt have anything to do with /SIP/Registry/63 |
17:18.56 | [TK]D-Fender | flujan : Of how many of those do you have already? |
17:19.05 | [TK]D-Fender | jbalcomb : correct. |
17:19.32 | IMG-GR | Hi guys, I keep getting "/usr/bin/ld: cannot find -ltonezone; collect2: ld returned 1 exit status; make[1]: *** [chan_zap.so] Error 1; make[1]: Leaving directory `/usr/local/src/libpri-1.2/channels'; make: *** [subdirs] Error 1" error while compiling Zaptel. Any idea, since I've done all the required step to compile it? |
17:20.04 | jbalcomb | [TK]D-Fender is there anyway to think of family, keytree, and key, as database, table, and values? |
17:20.34 | flujan | [TK]D-Fender: 150 :D |
17:20.51 | [TK]D-Fender | jbalcomb : not like that. DB1 is only family/key. not 3 dimensionsal. I mentioned this earlier. Pastebin your entire extensions.conf and I'll see if I can fix it up for you. |
17:21.21 | jbalcomb | [TK]D-Fender i dont know pastebin. :/ one sec. |
17:22.00 | [TK]D-Fender | flujan : ok, then my suggestion is to get AudioCodes 24 FXS SIP gateways and keep your current headsets. Much cheaper since you already have the phones (which is what those usints are and I'm sure they weren't too cheap either...) |
17:22.00 | ManxPower | ~pb |
17:22.03 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
17:22.04 | [TK]D-Fender | ~b |
17:22.05 | jbot | picobot: c |
17:22.21 | *** join/#asterisk Kato41 (n=Miranda@p54BEE46D.dip.t-dialin.net) |
17:22.30 | Kato41 | hi |
17:22.36 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
17:23.33 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
17:23.34 | flujan | ok thanks |
17:23.41 | jbalcomb | [TK]D-Fender our extensions.conf has five includes for 'ease of some crap' ill paste bin the include where the trouble is. |
17:24.16 | *** join/#asterisk [D]rea[M] (n=tR@stjhnf0112w-142162196108.pppoe-dynamic.nl.aliant.net) |
17:24.21 | Kato41 | got a problem in running two hfc-based cards in a asterisk@home setup |
17:24.30 | [TK]D-Fender | jbalcomb : ok, pastebin first. If it makes me cringe I might ask you to e-mail the whole thing |
17:24.43 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
17:26.21 | [TK]D-Fender | flujan : a full solution including 6 AudioCodes gateways and a Sangoma A104d card for your E1's would be about $17,800 USD |
17:26.32 | [TK]D-Fender | flujan : And not require massive rewiring. |
17:26.49 | jbalcomb | [TK]D-Fender well, i gaurantee you're gonna cringes atleast from lack of macros |
17:27.37 | [TK]D-Fender | jbalcomb : We'll see how generous I feel on seeing it :) |
17:28.02 | ManxPower | Yet another thing in New Orleans that isn't working. A friend's mother died and the cemetary where their family tomb is located is not open. |
17:28.41 | Kato41 | which OS/asterisk combination would be the best to run asterisk with two hfc isdn cards in euroisdn? |
17:29.02 | ManxPower | Kato41, Yes. |
17:29.55 | [TK]D-Fender | ManxPower : 42? |
17:31.16 | *** join/#asterisk ToTo (n=ToTo@host16-146.pool872.interbusiness.it) |
17:31.35 | jbalcomb | [TK]D-Fender http://pastebin.com/508395 |
17:31.43 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfk1f.dialup.mindspring.com) |
17:31.57 | *** join/#asterisk juice (n=juice@209.33.105.105) |
17:32.38 | Mimmus | can any good will man/woman give a look at this 'pri debug' output http://pastebin.com/508398 ? |
17:32.39 | [TK]D-Fender | jbalcomb : 1 phone/queue? |
17:33.01 | many | does anybody know how to read Hz value in linux systems? |
17:33.56 | *** join/#asterisk wpayne (n=wpayne@vgateway.libertyrms.info) |
17:34.36 | wpayne | Hello everyone |
17:34.54 | Mimmus | a call starting from a legacy Alcatel PBX connected to * is dropped with NOANSWER |
17:35.03 | jbalcomb | [TK]D-Fender i dont think so. as i understand it we have a host of 800's that go to a group of reps who answer based on callerid |
17:35.35 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
17:35.35 | [TK]D-Fender | jbalcomb : Just trying to figure out who the DND applies to.... where does it get set? What does it really mean? |
17:37.59 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
17:38.31 | jbalcomb | [TK]D-Fender another include, applications.conf, has this: http://pastebin.com/508407 |
17:39.49 | ping1 | many: cat /proc/cpuinfo ? |
17:39.57 | [TK]D-Fender | jbalcomb: what does 0800 indicate? |
17:40.10 | Kato41 | no fee |
17:40.18 | many | ping1: no. Not CPU Freq, Hz Value. zgrep HZ /proc/config.gz solved it for me. thanks anyway =) |
17:40.22 | [TK]D-Fender | jbalcomb: and why the "SIP" in that DB entry? |
17:40.48 | many | (but config.gz was to trivial to be intuitive *g*) |
17:42.24 | [TK]D-Fender | jbalcomb : I'm suspecting that it only needs a minor rename and a 5 minute fix |
17:42.51 | *** join/#asterisk RoadRunnR (n=MrRoadRu@213.187.82.17) |
17:44.10 | Kato41 | ^^ why dont i have got a /dev/zap ? |
17:44.50 | RoadRunnR | hi all |
17:44.56 | wpayne | is the zaptel module loaded Kato41? |
17:44.57 | RoadRunnR | is the mISDN guru arround? |
17:45.31 | Kato41 | yes, wpayne |
17:46.28 | many | teh misdn guru? |
17:46.34 | RoadRunnR | well, cricher or crich as he is called in beronet's mantis |
17:46.50 | RoadRunnR | sorry, crichter |
17:47.18 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfk1f.dialup.mindspring.com) |
17:48.05 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
17:48.51 | crich1999 | here i am ! |
17:49.13 | Kato41 | i am looking for a howto for asterisk 1.2 and two hfc-isdn cards |
17:49.33 | crich1999 | well 3 possibilities: 1. mISDN 2. bristuff 3. visdn |
17:49.48 | RoadRunnR | crich1999: great, we have be bouncing arround messages on the bug #174 |
17:49.56 | Kato41 | euroisdn komp plz |
17:50.02 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
17:50.13 | crich1999 | thats you RoadRunnR |
17:50.23 | RoadRunnR | might be faster this way ;-) |
17:50.42 | crich1999 | did you restart yet |
17:50.50 | RoadRunnR | crich1999: i can give you a login on the box, if you want |
17:50.55 | RoadRunnR | box came up just now |
17:51.34 | RoadRunnR | outgoing works again |
17:51.46 | crich1999 | Kato41: just read http://www.voip-info.org/wiki/view/chan_misdn |
17:51.54 | crich1999 | hehe thought so |
17:51.59 | Kato41 | thx crich1999 |
17:52.13 | RoadRunnR | incoming also ... |
17:52.28 | crich1999 | Kato41: get the mqueue branch RoadRunnR has got it working now i think |
17:52.36 | Skumling | Kato41: I was in your situation until a week ago... tried some different things before ending up with downloading bristuff, wich has a installation routine that downloads zaptel, lipbri and asterisk and compiles the whole bunch itself |
17:52.37 | wpayne | anyone using cisco 79xx phones with asterisk? |
17:52.55 | blitz[class] | lots of people |
17:53.13 | RoadRunnR | crich1999: this is still the same version that crash logs are from, i guess i can trash it again if i keep the connection open logn enough |
17:53.43 | RoadRunnR | wpayne: get snom 320 or 360, they are great |
17:53.44 | *** join/#asterisk Abbas (i=Abbas@203.81.194.208) |
17:54.05 | wpayne | I had my 7940 setup up with chan_sccp then my system died and I reinstalled everything. now I'm trying with chan_skinny with varying results. |
17:54.19 | Kato41 | Skumling: bristuff doesnt do it ... the way i used it... |
17:54.31 | jbalcomb | [TK]D-Fender the 0800 indicates 8 AM which is 'that guys' way of noting that the phone is set to after hours |
17:54.41 | crich1999 | RoadRunnR: what do you mean with still the same .. you mean when your calls are longing too long they crash ? |
17:54.42 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
17:55.02 | crich1999 | Kato41: whats the problem with bristuff ? |
17:55.06 | [TK]D-Fender | jbalcomb : Ok, you're running 1.2.x? |
17:55.35 | jbalcomb | [TK]D-Fender i do not know why 'that guy' chose the confusing DB entry using dnd and SIP as family/key or however that is structured |
17:55.46 | *** join/#asterisk saftsack (n=oliver@p54A7E695.dip.t-dialin.net) |
17:55.46 | jbalcomb | [TK]D-Fender yes, Asterisk 1.2.1 |
17:56.33 | wpayne | I think I've messed up the tftboot stuff, unfortunately I have forgotten the way I had set it up before with chan_sccp. it should be the same with chan_skinny right? |
17:57.22 | wpayne | I wish it was just a sip phone. |
17:57.26 | saftsack | someone here who has experiences with iaxmodem? |
17:57.37 | RoadRunnR | crich1999: what i mean, that this is the new version (svn current) that before show the problems, and those problems only occured after a call was terminated abnormaly, the first calls after a power-cycle use to work before as well, so my guess is that the same problem still is in there somewhere |
17:58.19 | saftsack | https://sourceforge.net/projects/iaxmodem does the page work for you? |
17:58.22 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
17:58.57 | crich1999 | RoadRunnR: i see. I'm testing the mqueue stuff only on my dev box now, there i very often restart either asterisk or the whole machine .. so i didn't get this behaviour yet |
17:59.22 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
18:00.06 | RoadRunnR | crich1999: i see, i have the trace on now, i'll update the bug as soon as i see the problem again, anything else i should log? |
18:00.56 | frenzy | hey all... |
18:00.56 | frenzy | do I need to purchase g729 license for able to connect g729 (UAs) ? |
18:00.56 | frenzy | or is it only used when asterisk acts as UA? |
18:00.57 | frenzy | channel.c:2685 ast_channel_make_compatible: No path to translate |
18:01.35 | frenzy | ? |
18:02.14 | jbalcomb | [TK]D-Fender In the Asterisk CLI executing ‘database put family key value’ produces the database entry ‘/family/key: value’ |
18:02.14 | ManxPower | frenzy, You need a G729 license for most things. |
18:02.20 | [TK]D-Fender | jbalcomb : heres a better sample for you to use - exten => _*62,n,Set(DB(dnd/62)=0800) |
18:02.24 | crich1999 | RoadRunnR: just the trace and syslog, you can mail me the whole trace when the problem happens, then i can check what happened before. How long does it work till the problem happens ? |
18:02.50 | frenzy | my UA is g729 however i'm terminating using ulaw |
18:03.04 | ManxPower | If Asterisk does not have to transcode (both devices using G729), doesn't have to listen to DTMF (t/T/w/W) or play any sound (voicemail, playback, background) then you prolly don't need a license. |
18:03.09 | frenzy | I'm getting no path to translation |
18:03.15 | RoadRunnR | crich1999: varies, seems to depend to the call duration |
18:03.16 | ManxPower | frenzy, that would be called transcoding. |
18:03.21 | ManxPower | you need a license to transcode. |
18:03.52 | frenzy | ohh |
18:04.13 | *** join/#asterisk juice (n=juice@209.33.105.105) |
18:04.18 | frenzy | and how are ports counted? |
18:04.20 | crich1999 | RoadRunnR: when i've finished finding the hfcmulti bug (which causes the unloading crash) i'm going to install the mqueue stuff on our pbx. Thats a good Testenv :) |
18:04.25 | frenzy | If I'm using g729 to connect and at the same time using it to terminate |
18:04.30 | frenzy | is that two ports |
18:04.30 | frenzy | or one port? |
18:04.34 | ManxPower | frenzy, you purchase license for X concurrent channels |
18:04.39 | frenzy | as its one call |
18:04.44 | ManxPower | usually 1 call = 1 channel |
18:04.53 | frenzy | oki |
18:04.55 | frenzy | thanks |
18:05.04 | [TK]D-Fender | jbalcomb : And check for it like - exten => 1,1,GotoIF($[${DB(dnd/62)}=0800]?4) |
18:05.09 | ManxPower | But why not just spend the money on a couple of licenses |
18:06.01 | ManxPower | If you want a three-way call then you would need 2 licenses for it. |
18:06.27 | ManxPower | And any MeetMe conference uses 1 license per G729 user |
18:08.02 | jbalcomb | [TK]D-Fender so my line exten => s,5,GotoIf($[${dnd} = 0800]?9:6) becomes exten => s,5,GotoIF($[${DB(dnd/62)}=0800]?9:6) |
18:09.15 | [TK]D-Fender | jbalcomb : no need for the "6". thats implied. exten => s,5,GotoIF($[${DB(dnd/62)}=0800]?9) |
18:09.25 | [TK]D-Fender | jbalcomb : An you don't need the dbget that preceds it |
18:10.41 | ManxPower | Yes, but at some verbose or debug levels it complains if you don't specify both priorities. |
18:10.48 | [TK]D-Fender | jbalcomb : use Set(DB(dnd/62)=nope) to disable |
18:11.01 | [TK]D-Fender | ManxPower : Really? how anal.... |
18:11.25 | [TK]D-Fender | ManxPower : Since when does any other language care that you don't have an "else" ? |
18:11.28 | ManxPower | [TK]D-Fender, I always do it just to be explicit. I like explicit. |
18:11.54 | [TK]D-Fender | ManxPower : I like explicit too, but thats just wrong.... and a PITA if you have to renumber everything... |
18:12.21 | ManxPower | [TK]D-Fender, that MIGHT have changed when they added the "n" priority. I don't know. |
18:12.43 | *** join/#asterisk Chiardon (n=yo@200.71.58.39) |
18:12.57 | Chiardon | justinu, --> Hey |
18:13.07 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
18:16.33 | jbalcomb | [TK]D-Fender ah, gotcha on the fall thru. excellent on taking out the DBGet. |
18:17.17 | jbalcomb | [TK]D-Fender apparently we are using IPswitchboard to let people turn the dnds on and off so we are stuck with the family/key:value structure and the creating and deleting of the DB entry |
18:18.00 | *** part/#asterisk techie (i=gus@antibala.com) |
18:22.06 | *** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com) |
18:22.26 | *** join/#asterisk bangawanga (n=ahecker@ppp-82-135-70-13.mnet-online.de) |
18:22.33 | *** join/#asterisk Chiardon (n=pirch@200.71.58.39) |
18:22.38 | bangawanga | hello guys |
18:22.42 | [TK]D-Fender | jbalcomb : Well thats not a bad thing, but you can jsut turn them on/odd from a phone... not sure why you'd use IPSwitchboard to change them... monitor perhaps... |
18:23.14 | *** part/#asterisk m160858 (n=ubuntu@200.89.12.46) |
18:23.37 | Kato41 | crich1999? |
18:23.44 | [TK]D-Fender | jbalcomb : as for the overall design of your setup I think it can be massively reduced with a few more variables. |
18:23.49 | *** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
18:23.54 | crich1999 | Kato41! |
18:24.25 | Kato41 | see the priv chat? |
18:24.34 | crich1999 | no |
18:26.04 | Kato41 | could you try to invite me instead? |
18:27.19 | crich1999 | Kato41: just opened a private dialog ot you |
18:27.31 | Kato41 | u dont see what i write? |
18:27.35 | *** join/#asterisk darkskiez (n=darkskie@bb-195-172-51-236.ukonline.co.uk) |
18:27.36 | Luke-Jr | Linksys refuses to honor their warranty |
18:27.37 | Luke-Jr | wtf |
18:27.53 | *** join/#asterisk jimmy_deanPB_ (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
18:28.07 | crich1999 | Kato41: no |
18:28.18 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
18:28.52 | jbalcomb | [TK]D-Fender yeah, i wasnt aware that we were using IPswitchboard until 30 minutes ago. i assume with macros and variables our config could be more that cut in half but it'll be some time before i 'get it' enough to do that. |
18:29.41 | Trazz | I can get extension 500 to work but i can get voicemail or anything else to work ... |
18:29.48 | crich1999 | Kato41: just offering you a dcc chat, you don't get it i assume ? |
18:29.49 | jbalcomb | [TK]D-Fender i really appreciate the info/assist. i assume that understanding the DB and use of it is a big step toward making good use of Asterisk/VoIP |
18:30.23 | docelm0 | Anyone in here using Mera MVTS? |
18:30.25 | *** join/#asterisk razu (n=razu@213-35-170-76-dsl.trt.estpak.ee) |
18:30.42 | darwin_35 | <PROTECTED> |
18:30.43 | darwin_35 | Jan 16 11:30:42 WARNING[71144]: loader.c:554 load_modules: Loading module app_dbodbc.so failed! |
18:30.43 | [TK]D-Fender | jbalcomb : a lot of the time you don't need db for anything, but it has a few nifty places for things exactly like what you're using them for |
18:31.29 | [TK]D-Fender | jbalcomb : I think I could severely shick the size of your setup and make it easier to process... if you're willing to sub-contract for a small fee :) |
18:31.32 | darwin_35 | input pls |
18:31.36 | Kato41 | @crich1999 got a priv chat and a dcc open with you |
18:31.58 | crich1999 | crich1999: can't be i have none |
18:32.26 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
18:32.34 | darwin_35 | I need this fixed for mysql |
18:32.48 | file | darwin_35: 1. Rude 2. What version of Asterisk |
18:32.56 | darwin_35 | 1.2.1 |
18:33.07 | darwin_35 | sorry file forgot to say good morning |
18:33.10 | jbalcomb | [TK]D-Fender perhaps. we have a consultant right now that is a jacknut. my boss asked me to decide if we should bring him back. if not i'm to decide if we should get someone else. |
18:33.12 | darwin_35 | my bad |
18:33.17 | jbalcomb | [TK]D-Fender are you local to 216? |
18:33.21 | *** join/#asterisk backblue_ (n=moo@87-196-15-214.net.novis.pt) |
18:33.32 | [TK]D-Fender | jbalcomb : Nope, but obtainable over SIP / 1-800 |
18:33.35 | jbalcomb | [TK]D-Fender that being Cleveland, OH ;) |
18:33.56 | file | darwin_35: you're using a version of app_dbodbc that uses something that is a patch on the bug tracker, and is not in Asterisk |
18:34.32 | [TK]D-Fender | jbalcomb : Montreal, QC here... |
18:35.05 | [TK]D-Fender | SSH / SIP don't care too much about distance, only latency :) |
18:35.13 | darwin_35 | is there a working ver of app_dbodbc |
18:35.14 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
18:35.14 | [TK]D-Fender | ... and bandwidth :D |
18:35.22 | FuriousGeorge | hey all |
18:35.32 | darwin_35 | for asterisk |
18:35.33 | file | darwin_35: there might be... go Google, browse the bug tracker |
18:35.58 | FuriousGeorge | i notice my iax minutes provider doesnt like certain 800 numbers, is this common to many voip accounts or is it just me |
18:36.26 | file | darwin_35: next time Google for ast_direct_realtime, because it led me right to the bug number and bug report for it |
18:36.27 | jbalcomb | [TK]D-Fender ah, I don't know if we are allowed to work with Canadians. Sorry. =) |
18:37.42 | [TK]D-Fender | :O |
18:38.53 | Kato41 | after making mqueue misdn i got to cp the channels-dir to /usr/src/asterisk/ ? |
18:39.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:39.31 | *** join/#asterisk MommomeryCliff (n=willy@host230-24.pool873.interbusiness.it) |
18:40.51 | crich1999 | Kato41: no you just need to go into asterisk/channels/misdn, then type make and go back to /usr/src/asterisk and type make install |
18:40.54 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
18:41.03 | g4m | howdy |
18:41.22 | FuriousGeorge | anyone else here using nufone? |
18:41.35 | FuriousGeorge | 4 outbound |
18:41.49 | sivana | FuriousGeorge: yes |
18:41.50 | g4m | Anyone have any suggestions on using vonage and asterisk together? |
18:41.54 | *** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com) |
18:42.12 | sivana | g4m: I don't think you can |
18:42.14 | fugitivo | g4m: yes, choose another provider |
18:42.17 | *** join/#asterisk Assid (n=assid@59.183.31.83) |
18:43.19 | g4m | heh |
18:43.48 | g4m | fugitivo: like? |
18:44.16 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
18:44.22 | fugitivo | teliax, voicepulse, broadvoice, errr, a lot |
18:44.48 | Lots | fug what you think of nuvio |
18:45.06 | fugitivo | didn't try it |
18:45.26 | Lots | just wondering it has the highest ratings on this site. |
18:45.27 | *** join/#asterisk jimmy_dean__ (n=jhodapp@indianalifesciences.com) |
18:45.34 | Kato41 | i think miranda doesnt support whispering ^^ |
18:45.39 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
18:45.45 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
18:45.49 | crich1999 | Kato41: probably ;) |
18:45.56 | crich1999 | Kato41: try xchat |
18:46.00 | tzafrir_laptop | the list is back... |
18:46.06 | Trazz | is there some basic configs i can look at that has a few extensions and voice mail working? |
18:46.38 | Lots | http://www.dslreports.com/gbu |
18:46.58 | tzafrir_laptop | mozilla/firefox with the irc extension |
18:47.09 | *** join/#asterisk [Sanem] (n=PisiKoLo@stjhnf0112w-142162198201.pppoe-dynamic.nl.aliant.net) |
18:47.20 | fugitivo | Trazz: asterisk examples |
18:47.21 | tzafrir_laptop | (for windows folks , that is |
18:47.24 | tzafrir_laptop | ) |
18:47.34 | fugitivo | bitchX baby! |
18:47.36 | *** join/#asterisk FastJack (i=fastjack@p5091E335.dip.t-dialin.net) |
18:47.39 | Trazz | yes i need some examples that work.. i am not getting anything but extesion 500 to work |
18:47.54 | fugitivo | Trazz: make samples after you compile asterisk |
18:48.15 | fugitivo | ? |
18:48.47 | crich1999 | I am afraid of bitchX users |
18:48.54 | fugitivo | why? |
18:49.01 | *** join/#asterisk Katonka (n=Katonka@p54BEE46D.dip.t-dialin.net) |
18:49.03 | fugitivo | i'm afraid of windows users |
18:49.10 | jbroome | their horrible part messages scare me |
18:49.12 | crich1999 | you should be! |
18:49.32 | Katonka | test |
18:49.58 | Katonka | how to whisper in xchat? |
18:50.18 | crich1999 | Katonka: are you Kato41 |
18:50.20 | crich1999 | ? |
18:50.26 | Katonka | yes |
18:50.40 | _jpk | is asterisk-user and -devel down at the moment? |
18:50.45 | crich1999 | you have multiple personalities ? |
18:50.47 | Katonka | i think i whisper allready? :) |
18:50.57 | crich1999 | no i don't think so |
18:51.14 | _jpk | proxy:~ # dig mx lists.digium.com |
18:51.22 | _jpk | lists.digium.com. 52m10s IN MX 10 69.16.138.164.digium.com. |
18:51.29 | _jpk | and that dows not resolve... |
18:51.53 | wunderkin | it was fixed |
18:52.02 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
18:52.05 | _jpk | when? |
18:52.28 | wunderkin | about an hour ago, i just started getting a couple messages |
18:52.45 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
18:53.11 | _jpk | ic. Just flushed the dns cache. |
18:53.33 | trixter | I am thinking that there will be a bunch of queued messages that get delivered soon, and I fully expect 21359139513 emails over the next hour |
18:55.15 | Trazz | fugitivo, i have the examples in and reloaded it. is there a cheat sheet on configuring x-lite softphone with asterisk with voicemail, etc ? |
18:56.21 | fugitivo | Trazz: sip.conf |
18:58.51 | *** join/#asterisk mikeyb_work (n=michael@66-193-82-211.gen.twtelecom.net) |
18:59.46 | [TK]D-Fender | Trazz : There is a specific example in the sample sip.conf |
19:01.07 | mikeyb_work | I have a sip channel that did not properly disconnect from asterisk when the call ended. I do not want to have to restart asterisk... is there any way to reset a sip channel? the output from "show channel SIP/172-22-12-57-8850" tells me "Blocking in: ast_waitfor_nandfds" |
19:01.12 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
19:01.34 | Katonka | crich1999, test |
19:01.55 | crich1999 | Katonka, test back |
19:02.08 | Katonka | crich1999: is this direct now? |
19:03.03 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
19:03.19 | jbalcomb | iCEBrkr I can't seem to find anything that discusses the use of that X, do you know where I should look? |
19:04.20 | crich1999 | Katonka: no i think not |
19:06.53 | Luke-Jr | Warning to anyone planning to buy Linksys stuff |
19:06.57 | Luke-Jr | they don't honour their warranties |
19:07.03 | jbalcomb | [TK]D-Fender do you know about the X iCEBrkr suggested using to avoid the warning when the value is null? I can't find references to its use. |
19:07.46 | *** part/#asterisk jimmy_dean__ (n=jhodapp@indianalifesciences.com) |
19:08.20 | Katonka | crich1999, svn hast to be executed where? to insert it in /usr/src/asterisk... |
19:09.25 | [TK]D-Fender | jbalcomb : didn't follow you there... |
19:11.10 | crich1999 | Katonka: you will need to get the whole asterisk via svn. svn is the replacement of cvs. just do that: svn co http//svn.digium.com/svn/asterisk/team/crichter/0.3.0 asterisk-0.3.0 and follow the instructions on the wiki page |
19:11.29 | tzanger | hahaha |
19:11.34 | infinity1 | where can we get the latest polycom firmware? i don't see it on polycom's website |
19:11.46 | Trazz | can you hook skype up to asterisk? |
19:11.47 | [TK]D-Fender | jbalcomb : If you meant his sample where he shoved an "X" in front of the DB(), thats completely unnecessary for your sample |
19:11.48 | Katonka | crich1999, i see. thx |
19:11.55 | [TK]D-Fender | infinity1 : Check with your reseller |
19:12.04 | crich1999 | Katonka: no problemo |
19:12.32 | infinity1 | [TK]D-Fender: oh geez. there use to be links online. |
19:12.36 | crich1999 | Trazz: not yet i think, there's a page on voip-info.org regarding this question |
19:12.39 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
19:12.55 | [TK]D-Fender | infinity1 : Polycom doesn't hand out its firmwares to just anybody. You can ge older versions here : http://www.freedomphones.net/polycom/files/ |
19:13.12 | [TK]D-Fender | Trazz : No Skype.... |
19:13.29 | infinity1 | [TK]D-Fender: i just got this 601 and it has more current firmware than what i can download. but my other polycom is still old |
19:13.42 | infinity1 | can someone dcc me the latest polycom firmware ? :) |
19:13.52 | [TK]D-Fender | infinity1 : What version do you have on it? |
19:14.12 | infinity1 | [TK]D-Fender: br 3.1 and siup 1.6.2 on the new one |
19:14.20 | [av]bani | i'm so disappointed polycom doesnt support xml on the 501 >:( wtf is the point of the bitmapped lcd? |
19:14.23 | infinity1 | [TK]D-Fender: br 2.6.1 and sip 1.5.2 on the old one |
19:14.53 | *** join/#asterisk justinu (n=j2@72.18.13.34) |
19:15.16 | [TK]D-Fender | infinity1 : You can get 1.6.2 off the site I gave you, and just ask your reseller to give you 1.6.3 |
19:15.28 | [TK]D-Fender | [av]bani : a pretty logo :)\ |
19:15.34 | [av]bani | >:( |
19:15.43 | [TK]D-Fender | [av]bani : Thats why my setup is 100% 60x :D |
19:15.50 | [av]bani | i get a pretty logo with gxp-2000 :P |
19:15.58 | infinity1 | [TK]D-Fender: atacomm should just email it when you buy a fone. heh |
19:16.11 | Lots | just curious if there is anyone here who is in the voip solution provider business that i could ask a few questions? |
19:16.22 | [TK]D-Fender | infinity1 : yeah and keep in mind how many mailboxes would reject an attachment that size.... |
19:16.32 | infinity1 | [TK]D-Fender: yes yes. :P |
19:16.43 | [TK]D-Fender | Lots : just ask the question and spin the wheel! |
19:16.52 | justinu | spin the bottle |
19:17.35 | infinity1 | ahah ...atacomm runs * ... |
19:17.38 | infinity1 | go figure |
19:17.41 | justinu | why not? |
19:17.45 | Trazz | TK, any webbased gui's that work ? |
19:17.57 | fugitivo | anyone using a commercial ASR? |
19:17.58 | jbalcomb | [TK]D-Fender ok. thanks. just for educational purposes though, would there be some place to learn about its use? |
19:17.58 | [TK]D-Fender | Trazz : GUI for what? |
19:18.14 | [TK]D-Fender | jbalcomb : ... huh?! |
19:18.15 | Trazz | gui to configure system and administer it |
19:18.21 | Lots | tkd uhh ok, well, i'm just trying to get business owners interested in voip, and was wondering if anyone knew of some good roi calculators i could use when selling voip to clients. |
19:18.32 | [TK]D-Fender | Trazz : AMP. Though I don't suggest it or any other.... |
19:18.37 | [TK]D-Fender | (for you) |
19:18.53 | Lots | infinity, is atacomm a good reseller? I haven't read any good reviews about them. |
19:19.07 | [TK]D-Fender | Lots : Just get pricing for other solutions and you'll have enough to show how much they can save. |
19:19.13 | infinity1 | Lots: they get recommended a lot. i have no complaints |
19:19.16 | [TK]D-Fender | Lots : You need to learn about the competition. |
19:19.17 | Trazz | Tk, i was hoping the gui would help to visualize the config for extensions, voice mail and ivr setup |
19:19.40 | infinity1 | Lots: descent prices. and they don't try to fuck you at the end of your online order with additional fees |
19:19.49 | [TK]D-Fender | Trazz : Forget GUI... a basic setup is dead easy. you just need to take it on 1 file at a time. |
19:20.25 | iCEBrkr | Yay! More 1wk deadlines |
19:20.41 | [TK]D-Fender | iCEBrkr : But are they retro-active?! ;) |
19:20.47 | infinity1 | Lots: i'm on the fone with atacomm right now. they are giving me the ftp login for the firmware |
19:20.57 | justinu | nice |
19:21.28 | [TK]D-Fender | infinity1 : While you're at it can you ask for admin guides for the Citel gateways? ;) |
19:22.00 | infinity1 | [TK]D-Fender: shit. i just hung up before i looked at the screen agin. i was trying the ftp |
19:22.04 | [TK]D-Fender | :O |
19:22.10 | infinity1 | [TK]D-Fender: did you check their ftp site? |
19:22.17 | [TK]D-Fender | nope, credentials? |
19:22.47 | infinity1 | [TK]D-Fender: i msged ya |
19:25.13 | Lots | tkd well i was just trying to find a average comparison chart somewhere on the net that had the comparison of a traditional TDM system vs. a VOIP system. |
19:25.22 | [TK]D-Fender | I may just phone them up for it.... since I'm going to need that info to promote them to other clients who can't change their wiring... |
19:26.40 | [TK]D-Fender | Lots : I think there has beena few like that floating around, but get some specific comparisons and remember cost isn't everything. Its what they get vs what it costs vs what they're prepared to spend. |
19:26.47 | [av]bani | http://bani.anime.net/phonez/ |
19:26.49 | [av]bani | updated \o/ |
19:26.50 | jbalcomb | [TK]D-Fender just looking for some reference material that explains the use of that 'X' |
19:27.06 | [TK]D-Fender | jbalcomb : was it the one in the GotoIF? |
19:27.14 | infinity1 | Lots: feature wise its probably the same. |
19:27.30 | infinity1 | Lots: but you get SIP with VoIP, which is where things get interesting |
19:27.38 | jbalcomb | [TK]D-Fender yes'm |
19:27.55 | Lots | mostly i just want a price chart as part of my selling points, not the entire presentation. |
19:28.01 | Assid | hrmm.. remind me never to get a cisco phone |
19:28.08 | Assid | i just tried my friends 7960 |
19:28.12 | *** join/#asterisk svenna_ (n=svenna@p548D1EDB.dip0.t-ipconnect.de) |
19:28.15 | Jammy | hey guys got a quick question... for incoming calls i want zapteller to answer, forgot how to enable that in extensions.conf and cant find documentation on that particular function.. i have exten => s,1,Zapteller so far...anyw ideas whats wrong? |
19:28.22 | iCEBrkr | I dunno man, this sucks.. I've kinda slacking on these projects cuz I was informed we don't need the Asterisk box.. Now all of a sudden.. We need it |
19:28.23 | Assid | the polycoms are like soooooo easy to provision |
19:28.25 | jbalcomb | cisco is the most awesome ever. |
19:28.37 | iCEBrkr | jbalcomb: Liar |
19:28.39 | *** join/#asterisk [Sanem] (n=LoGo@stjhnf0112w-142162201040.pppoe-dynamic.nl.aliant.net) |
19:28.43 | Assid | jbalcomb: the craziest to configure |
19:29.07 | jbalcomb | iCEBrkr HA! just because you have to have a brain to understand Cisco doens't mean it ain't the most awesome. |
19:29.26 | jbalcomb | Assid Agreed but therein lies the power. |
19:29.43 | iCEBrkr | jbalcomb: I dunno man, some of their router stuff isn't the bestest out there :P |
19:29.57 | jbalcomb | People who think Cisco is too complicated probably don't watch movies you have to read either, |
19:30.40 | jbalcomb | iCEBrkr hrmm.. such as? Juniper touches them in the highend 7xxx/12xxx market but otherwise I haven't seen a contender. |
19:30.52 | iCEBrkr | All I'm saying is to look into Foundry |
19:31.03 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
19:31.18 | jbalcomb | iCEBrkr Foundry is indeed decent, next in line I'd say. |
19:31.35 | iCEBrkr | It's one thing to say you have a 24 port gigabyte switch but the problem is the backplane can't really handle the through-put of all 24 ports at the same time.. |
19:31.52 | jbalcomb | iCEBrkr part of the joy of Cisco is in having their entire product line available. |
19:32.02 | iCEBrkr | Deep pockets.. |
19:32.39 | jbalcomb | iCEBrkr often cheaper to only spend life learning one CLI/IOS than mad labor hours trying to keep track of three or four product lines. |
19:32.51 | [TK]D-Fender | jbalcomb : there is nothing to say about that X. Unneccesary |
19:33.15 | iCEBrkr | jbalcomb: I hear that, but like I was saying.. A switch just isn't a switch and a router just isn't a router. |
19:33.18 | jbalcomb | [TK]D-Fender okidoki. I'll drop it from my thoughts. Thanks again. |
19:33.29 | [av]bani | part of the grey hair going bald of cisco is they DONT FREAKING REGRESSION TEST RELEASES >:( >:( |
19:33.35 | Lots | tkd you a solution provider? |
19:33.54 | [av]bani | in fact i suspect cisco has no test methods at all... |
19:33.55 | [TK]D-Fender | jbalcomb : glad to help |
19:33.58 | *** join/#asterisk [hC-] (n=hardcore@209.153.195.139) |
19:34.00 | jbalcomb | [TK]D-Fender is the best Asterisk consultant I've ever seen. |
19:34.07 | *** join/#asterisk squinky86 (n=ASGjon@unaffiliated/squinky86) |
19:34.20 | mog_work | three cheers for fender! |
19:34.20 | [TK]D-Fender | [av]bani : You should add that there IS PoE for Polycom IP 30x, 50x with an add'l adapter cable |
19:34.27 | jbalcomb | highly recommended if you need lengthy expert assistance. |
19:34.30 | iCEBrkr | jbalcomb: Oh yeah, and your 'X' question.. It's just a place holder.. It's not a command. and you don't use it in your DB() function. |
19:34.45 | iCEBrkr | jbalcomb: [TK]D-Fender is a frick'n nerd man.. |
19:34.47 | [av]bani | fender, or $15 if you use any off the shelf poe adapter setup |
19:35.01 | [av]bani | i dont include optional poe if they're just silly kits |
19:35.09 | *** part/#asterisk Davey|Work (n=davey@unaffiliated/davey) |
19:35.10 | fugitivo | jbalcomb: there's always paypal to help people that helps |
19:35.13 | [TK]D-Fender | iCEBrkr.... don't make be break you :D |
19:35.25 | iCEBrkr | :P |
19:35.33 | [av]bani | i include them if they're weird nonstandard poe like cisco |
19:35.36 | [TK]D-Fender | [av]bani : Its a selling point thoug, just reverse of the Cisco method... |
19:35.42 | jbalcomb | iCEBrkr frick'n nerds are my favorite people to contract. they waste unbillable hours fixing things for me and take such pride in the Quality of thier work that I don't have to hire someone else to doub check it. |
19:36.00 | [av]bani | fender, why include a $50 when i can do it myself with off the shelf for $15 |
19:36.01 | iCEBrkr | jbalcomb: We know this! |
19:36.05 | [av]bani | self defeating |
19:36.20 | [TK]D-Fender | jbalcomb : And for the good stuff my rates are very acceptable! ;) |
19:36.46 | [TK]D-Fender | [av]bani : You have to mention that the input requirements ALLOW you to use such methods. Not all do.... |
19:36.54 | Lots | jbal yep thats a habit i have to get away from in my business. |
19:36.55 | [av]bani | the great thing about fender's advice here is that if you're dissatisfied you can get a money back guarantee |
19:37.00 | jbalcomb | [TK]D-Fender :) perhaps I'll delve into a terrible conflict of interests and contract you through my company to this place. ;) |
19:37.14 | Lots | i find myself doing the same thing then i think to myself wait... am i getting paid for this crap? |
19:37.15 | Lots | lol. |
19:37.16 | [TK]D-Fender | jbalcomb : Paypal works.... |
19:37.33 | mog_work | lol |
19:37.47 | *** join/#asterisk slak- (i=slak@rewted.biz) |
19:37.49 | jbalcomb | Lots yeah, I don't do /extra/ work. We agreed to this, its done, I recommend fixing this too, new contract. |
19:37.52 | slak- | hi :) |
19:37.54 | [TK]D-Fender | I did a big "heads up" for a guy just for free and he was rather thankful and paind me anyways... |
19:38.00 | slak- | where should i begin looking for a voice t1 line.. |
19:38.02 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
19:38.06 | slak- | already have a data t1 here |
19:38.07 | [av]bani | snom would be an easy choice if they did xml... |
19:38.20 | tzanger | slak-: contact your local RBOC, then after the sticker shock wears off contact a bunch of the ILECs in your area |
19:38.29 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
19:38.48 | [TK]D-Fender | [av]bani : Too true! If you don't like my free advice you get DOUBLE your money back! |
19:38.50 | jbalcomb | [TK]D-Fender you'll get no 'thank you' money from me. ;) my lawyer and accountant would not approve. |
19:39.00 | iCEBrkr | LOL |
19:39.11 | rob0 | Double your money back, less a restocking fee of course |
19:39.16 | znoG | is there a list of options available for the VoicemailMain "0" option? (change pass, etc) |
19:39.18 | Lots | tkd, i'm new to this voip stuff, i quoted a guy for a 24-phone system with phones cabling and configuration about 10,500.00 for a hosted ip-pbx solution.. |
19:39.19 | [TK]D-Fender | jbalcomb : I meant in compensation were I to completely renovate your config :) |
19:39.28 | [TK]D-Fender | iCEBrkr : I think I've done PLENTY for you already ;) |
19:39.37 | Lots | thats seems on the cheap side to me though. |
19:39.40 | jbalcomb | do keep in mind that if someone does work for you and you do not pay them they are not responsible by law, atleast in Ohio. |
19:39.42 | iCEBrkr | lol |
19:40.04 | jbalcomb | [TK]D-Fender I think we've all /done/ plenty for iCEBrkr |
19:40.10 | [TK]D-Fender | znoG : its all on the WIKI... |
19:40.18 | slak- | how much can i expect to pay for a ~10 channel t1 line |
19:40.26 | slak- | i already have a data t1 can i use that? |
19:40.27 | iCEBrkr | jbalcomb: oh come'on |
19:40.31 | [TK]D-Fender | slak- : Depends where and what company.... |
19:40.37 | tzanger | slak-: depends entirely on your company and location |
19:40.37 | iCEBrkr | jbalcomb: you havent' done anything for me but give me a headache |
19:40.38 | *** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
19:40.40 | slak- | okay roughly? |
19:40.52 | [TK]D-Fender | tzanger : tag! |
19:40.53 | tzanger | I, for example, am in a Rate 4 group with Bell Canada. That's the "don't use the lube" rate group |
19:40.54 | jbalcomb | slak- if you can setup Multi-Link PPP i beleive can run data/voice on the same PRI |
19:40.56 | tzanger | [TK]D-Fender: haha |
19:41.01 | iCEBrkr | <-- phone |
19:41.13 | tzanger | jbalcomb: you don't need MLPPP for that |
19:41.20 | slak- | jbalcomb: is that something that my current data t1 provider would have to agree and do for me |
19:41.33 | jbalcomb | iCEBrkr *sniff* don't you remember all those 'GET * FROM *' queries I killed for you? |
19:41.37 | [TK]D-Fender | tzanger : Mine is at $700 or so for full PRI. I'm in a great area though |
19:41.42 | slak- | tzanger: who or what assigns phone numbers to t1 channels |
19:41.44 | tzanger | slak-: yes, and remember you'd only have 11x64kbps availbale for data then |
19:41.53 | tzanger | slak-: your provider. |
19:41.55 | slak- | okay that sucks |
19:42.05 | slak- | so its best to just get another line |
19:42.16 | tzanger | wwell maybe 12... depends on whether you're getting CAS or CCS T1 |
19:42.34 | tzanger | CAS T1 = 24 56kbps channels. CCS T1 = PRI = 23 64kbps + Dchan |
19:42.45 | [TK]D-Fender | tzanger : CAS? You can do split signalling with PRI can't you? CAS blows... |
19:42.47 | slak- | how much better is this voice t1 config to what i'm doing with sip accounts |
19:42.53 | *** join/#asterisk [}btorch] (n=kvirc@208.63.19.172) |
19:43.01 | slak- | s/to/than |
19:43.02 | *** part/#asterisk [}btorch] (n=kvirc@208.63.19.172) |
19:43.36 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
19:43.38 | [TK]D-Fender | slak- : Depends, what are you doing with "sip accounts"? incoming lines? |
19:43.52 | slak- | yes incoming for conferencing |
19:44.06 | }btorch{ | hey anyone here knows a good and cheap USB phone that I can use with * (SIP) |
19:44.08 | [TK]D-Fender | [av]bani : USe atacomm pricing for your snom entries... a fair bit cheaper... |
19:45.00 | [av]bani | last time i checked atacomm didnt have snom |
19:45.02 | [av]bani | i'll check again |
19:45.15 | [TK]D-Fender | [av]bani : THEY HAVE FOR A long TIME... |
19:45.40 | [TK]D-Fender | [av]bani : And for BLF you can add Polycom to the list for those... Lines - 1 |
19:45.43 | [av]bani | [TK]D-Fender: YOUR caps lock IS BROKEN |
19:45.58 | [TK]D-Fender | WhAt ArE YoU tAlKiNg AbOuT?! |
19:46.11 | [av]bani | if you refresh page you'll notice blf is listed on all the polycoms |
19:46.17 | [TK]D-Fender | [av]bani : I work a lot in caps here so if I slip up, just chill.... |
19:46.41 | [TK]D-Fender | [av]bani : not yet.... |
19:47.06 | wunderkin | ya fender has that problem a lot :P |
19:47.25 | [av]bani | there, got the new prices in |
19:47.26 | [TK]D-Fender | wundaboy : I prefer to think of it as a "feature" :F |
19:47.47 | Trazz | TK, i cant seem to find this script usr/src/asterisk/addmailbox |
19:47.52 | Trazz | yet it says its supposed to be there |
19:47.55 | [TK]D-Fender | [av]bani : and I mixed up the columns... |
19:48.04 | [av]bani | \o/ |
19:48.08 | [TK]D-Fender | Trazz : says who? |
19:48.09 | *** join/#asterisk samueltc (n=sam@69.156.67.214) |
19:48.12 | samueltc | hi |
19:48.15 | [av]bani | columnar mixture is <3 |
19:48.24 | Trazz | i looked all throgh my directories |
19:48.35 | fugitivo | v |
19:48.35 | fugitivo | j |
19:48.51 | [TK]D-Fender | [av]bani : You need to update the "B/L" for Polycom... |
19:48.56 | wunderkin | fender types with his toes and his eyes closed hehe.. |
19:49.03 | samueltc | anybody got app_prepaid_* working with postgresql? |
19:49.10 | [av]bani | fender, provide #'s |
19:49.23 | [TK]D-Fender | 2 hands tied behind my back.... in winter... 20' of snow... uphill ..... BOTH WAYS! |
19:49.38 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
19:49.39 | samueltc | I've got the one from commoveo |
19:49.40 | [TK]D-Fender | [av]bani : BLF & B/L = [Lines -1] |
19:49.59 | [TK]D-Fender | so if you are on a Polycom 601, you can have 5 BLF / programmable keys |
19:50.03 | [TK]D-Fender | max |
19:50.08 | [av]bani | why -1 ? |
19:50.13 | [TK]D-Fender | unless you add a side caddy |
19:50.22 | wunderkin | good thing they aren't called milf ... |
19:50.29 | iCEBrkr | o/~ BLF! Uh-huh.. o/~ |
19:50.39 | Trazz | Tk, where should that addmailbox be ? |
19:50.49 | [TK]D-Fender | [av]bani : the way line keys work on them. You can give a single registration 1 line key supporting multiple call/key, and use the remaining 5 for BLF / speed-dials |
19:50.59 | [av]bani | :/ |
19:51.10 | jbalcomb | [TK]D-Fender iCEBrkr What is the bestest phone known for Asterisk/VoIP? |
19:51.12 | [TK]D-Fender | Trazz : there is no "magic" script for that, you just write an entry into voicemail.conf! |
19:51.35 | Trazz | ok i did that.. i was reading where it said it make some sample greetgins. etc |
19:51.36 | [TK]D-Fender | jbalcomb : "bestest"..... Grammar very is english good yours!!! |
19:51.45 | [av]bani | speak like yoda fender does |
19:51.53 | iCEBrkr | jbalcomb: I dunno, I prefer ATA's :P |
19:52.07 | [TK]D-Fender | Do or do not... there is no tryhhhmmmMMMM!!! |
19:52.16 | wunderkin | [TK]D-Fender, sucky suky looong time! |
19:52.17 | jbalcomb | [TK]D-Fender =) I have a small fascination with slightly silly or rediculous character. |
19:52.38 | iCEBrkr | [TK]D-Fender: in other words, he retarded |
19:52.46 | jbalcomb | iCEBrkr would i think that clever or funny if i knew what ATA's were? |
19:52.55 | [TK]D-Fender | jbalcomb : I'd say the Cisco 7960G is probably the best phone, but the Polycom is a better value. |
19:52.56 | iCEBrkr | jbalcomb: oh yeah |
19:52.59 | Lots | anyone know of a good "VOIP Business" irc channel or a good forum to look at for the VOIP Business? |
19:53.12 | jbalcomb | iCEBrkr yes, 'he retarded' ... you picking up on the ebonics? |
19:53.15 | [av]bani | what's the lcd rez of the 7960g? |
19:53.24 | iCEBrkr | jbalcomb: www.sipura.com SPA2000 |
19:53.39 | [TK]D-Fender | jbalcomb : SPA-2002 <- Be current! |
19:53.52 | iCEBrkr | Yea yeah |
19:53.54 | iCEBrkr | :P |
19:54.18 | iCEBrkr | I got three of these SPA-2000's |
19:54.21 | iCEBrkr | Not sure how I got three. |
19:54.39 | [av]bani | o_O |
19:54.52 | [TK]D-Fender | iCEBrkr : I run SPA-2000, 3000, 941 at home and have 2000,2001's here at work |
19:55.06 | jbalcomb | any feelings about the SNOM phones? |
19:55.16 | [av]bani | fender, any idea how to increase the dialing speed of the 3000? |
19:55.39 | [av]bani | like you can do with eg modems, ATS11=40 or whatnot |
19:56.13 | [TK]D-Fender | [av]bani : Another update the IP 30x's LCD is measured in characters, not pixels. you may want to comment that in... |
19:56.45 | [TK]D-Fender | [av]bani : not the diaplan timeout, but rather the DTMF transmission rate to PSTN? |
19:56.54 | [TK]D-Fender | [av]bani : if so, not offhand |
19:56.56 | [av]bani | fixed |
19:57.02 | [av]bani | yes, the dtmf rate |
19:57.09 | [av]bani | its pretty lethargic |
19:57.40 | [av]bani | page updated for 301 |
19:57.46 | docelm0 | hay iCEBrkr is the geek convention on the 26th? |
19:58.10 | [av]bani | bummer no backlight on any of the polycoms. a tilted display doesnt help if the phone is in shadow |
19:58.16 | XIN01OZ | sup fellow asterisk boxers |
19:58.20 | ruud_org | [av]bani: are you talking about the SPA-3000? |
19:58.25 | [av]bani | ruud_org: yes |
19:58.40 | ruud_org | if so, the option you want is called "PSTN Dial Digit Len:" in the PSTN Line page |
19:58.51 | ruud_org | (i believe) |
19:58.54 | jbalcomb | hrmm.. no backlight is a bummer. some of the people here pitched a fit to get us to turn on the backlight on thier grandstreams |
19:58.56 | [av]bani | its .1/.1 which should be pretty fast, but its not... |
19:59.07 | ravenpi | [av]bani: Yeah, that's the -one- thing I wish they did differently with the Polycom. |
19:59.07 | [TK]D-Fender | [av]bani : I think SNOMS's have backlight as well as GS's, but thats it... |
19:59.10 | [av]bani | fender, aastra |
19:59.11 | *** join/#asterisk acidfoo (n=acidfoo@Kitchener-HSE-ppp3578579.sympatico.ca) |
19:59.19 | [TK]D-Fender | [av]bani : Ah yes... |
19:59.39 | ruud_org | on my spa-3000, i don't hear it dialing on the pstn, so i don't know whether the dalay in call set up is a result of slow dtmf or something else |
19:59.50 | [av]bani | you can hear it muting the digits as it dials out |
19:59.52 | XIN01OZ | what is suggested for voip to make termination to the PSTN cost wise .. a T3? |
19:59.54 | ruud_org | i guess i could listen in with a second phone in parallel on the pstn line |
20:00.04 | [TK]D-Fender | [av]bani : Hey, why don't you break upthe LCD column into multipl colums? res, type, backlight, etc? |
20:00.08 | [av]bani | a bit faint but you can still hear some bleed through |
20:00.14 | XIN01OZ | and be able to easily profit |
20:00.29 | ruud_org | [av]bani: interesting... never noticed that, but to be honest never tried to pay attention to it either |
20:01.05 | [av]bani | ruud_org: it's kind of annoying the spa-3000 connects the voip to pstn during hte dialing.. i wish it wouldnt do that till it was finished |
20:01.09 | *** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com) |
20:01.12 | [TK]D-Fender | [av]bani : Cisco 7905G supports Cisco PoE <- |
20:01.28 | [TK]D-Fender | [av]bani : yeah the muted DTMF bit annoys me too... |
20:01.48 | [av]bani | thats how i was able to tell it was dialing sluggish |
20:02.02 | [av]bani | you can blast out dtmf on modems and most switches will pick it up |
20:02.15 | [av]bani | ut the spa-3000 dials out all pokey and slow |
20:02.26 | *** join/#asterisk RussCC (n=face@216.157.205.211) |
20:02.33 | [TK]D-Fender | [av]bani : I'll check it out when I get home |
20:02.35 | ruud_org | [av]bani: strange, never noticed that it did... i hear the spa-generated dial-tone, dial the number, wait a while, and then finally hear the ringing... i don't hear anything regarding the PSTN dialtone or DTMF tones, but maybe this is all so faint that i didn't notice it |
20:02.54 | [av]bani | its possible if you've dorked with the gain settings |
20:02.58 | [av]bani | that you wont hear it |
20:03.13 | jbalcomb | iCEBrkr For "Unable to handle indication 3" am I looking at a HANGUPCAUSE issue? |
20:03.54 | [av]bani | 7905g updated |
20:04.11 | ruud_org | [av]bani: those are default (and call quality is OK), but if i turn up handset volume in my phone, i can hear it faintly in the background, now that you mention it |
20:04.16 | [av]bani | why wont cisco say the lcd rez of the 7960g... |
20:04.19 | ruud_org | learn something new every day :) |
20:04.22 | [av]bani | :) |
20:04.37 | RussCC | Hello, I have a question about Cisco configurations to allow traffic coming from my provider any ideas? |
20:04.54 | RussCC | Cisco Pix |
20:05.37 | RussCC | I am able to make outbound calls just fine |
20:05.52 | [av]bani | yay .05/.05 works |
20:06.11 | [av]bani | dials 7 digits in under a second :) |
20:06.23 | *** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
20:06.40 | RussCC | and when I have trafic completly open it works but once I start to lock it down I lose the ability to make a call |
20:07.31 | RussCC | the number I am calling rings but when the handset is picked up the line is quiet and if hung up the line will ring again |
20:08.06 | fugitivo | RussCC: udp 5060, udp 10k to 20k |
20:08.15 | fugitivo | (for sip) |
20:08.24 | *** join/#asterisk Katonka_ (n=Katonka@p54BEE46D.dip.t-dialin.net) |
20:08.32 | jbalcomb | [TK]D-Fender For "Unable to handle indication 3" am I looking at a HANGUPCAUSE issue? |
20:08.51 | [hC-] | This is odd, when i dial local calls on my pri, it generates ringing tones, and so does asterisk (double ring) without specifying the r option to dial. However, calling 800 numbers on it, i get no ringing at all |
20:08.56 | RussCC | fugitivo: thank you I will try that out |
20:09.03 | [av]bani | [TK]D-Fender: wonder if i should list skinny phones on there. cisco has some nice non-sucky 79xx models (eg do real 802.3af, not crippled, etc) but they only do skinny |
20:10.03 | [av]bani | and you can get used nortels for cheap, which speak the proprietary nortel (for which there is a * driver :) |
20:10.05 | RussCC | so I need to allow ports 5060 and 10k - 20k? Is that correct? |
20:10.15 | fugitivo | RussCC: for sip, yes |
20:10.19 | fugitivo | RussCC: udp |
20:10.26 | RussCC | fugitivo: thank you |
20:10.48 | [TK]D-Fender | jbalcomb : That seems to be a ringing indication mismatch |
20:11.09 | [TK]D-Fender | [av]bani : Screw Skinny.... experimental and proprietary... |
20:11.11 | Trazz | Jan 16 14:23:03 NOTICE[13437]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
20:11.12 | Trazz | Jan 16 14:23:03 WARNING[13437]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '2001' |
20:11.23 | [av]bani | experimental? i'm sure cisco would take issue with that :) |
20:11.24 | Trazz | i have edited my voicemail.conf, extensions.conf and sipconf |
20:11.31 | [TK]D-Fender | [av]bani : Experimental in * ! |
20:11.51 | fugitivo | Trazz: you didn't do it correctly thn |
20:11.52 | [TK]D-Fender | [av]bani : Don't make me "trout" you..... |
20:11.52 | fugitivo | then |
20:12.40 | Trazz | Fugitivo, i added it [local] |
20:12.41 | Trazz | 2000 => 1234,Joe Blow,joe@gmail.com |
20:12.47 | Trazz | thats in the voicemail.conf |
20:12.52 | Trazz | did reload |
20:12.59 | Trazz | it was already there |
20:13.23 | fugitivo | Trazz: it should be [default], not [local] |
20:13.32 | Trazz | ok |
20:13.34 | fugitivo | unless you specify @local when calling voicemail |
20:14.23 | [hC-] | is there a defacto way to handle/troubleshoot double ring/no ring issues over pri? |
20:14.33 | [hC-] | should i be checking tone zones, or... ? |
20:14.36 | [hC-] | (im in canada) |
20:19.33 | tzanger | [hC-]: first off |
20:19.41 | tzanger | use "progressinband=no" in zapata.conf |
20:19.47 | tzanger | stop/restart * and see if that fixes it |
20:19.52 | tzanger | who is your PRI provider? |
20:20.01 | [hC-] | okay. Allstream (formerly at&t canada) |
20:21.17 | [hC-] | its gonna be tricky to restart this thing, i gotta do it over lunch when nobody's using it |
20:21.48 | tzanger | [hC-]: yep. or just issue a "stop when convenient" and wait, setting your terminal to beep on silence. :-) |
20:21.56 | [TK]D-Fender | [hC-] : I'm with Allstream, and there was nothing funny to set here.... |
20:21.59 | tzanger | I don't trust restart when convenient because I'm not sure that a restart is enough |
20:22.48 | [hC-] | modifying zapata.conf should be sufficient to use restart, no? |
20:23.03 | tzanger | [hC-]: well as I said I'm not sure. that's why I am suggesting a good solid shutdown |
20:23.26 | [hC-] | [TK]D-Fender: hmm. I get double-ring it seems on local 604 calls, *seemingly* only from certain brands of phones (i hear it on sipura, but not cisco) and on 800 calls, i get no ringing tones at all. |
20:24.10 | [TK]D-Fender | [hC-] : this is after it connects the 2 endsand begins ringing? |
20:24.17 | Katonka_ | crich1999, installed asterisk03, but the misdn.conf is missing |
20:25.10 | [hC-] | [TK]D-Fender: for the double ring? yeah, after it passes it off to the zap channel, i hear two ringing tones that over lap one another, slightly off-timed |
20:25.28 | *** join/#asterisk l1nux (i=moi@214.138.103-84.rev.gaoland.net) |
20:25.37 | l1nux | hi :) |
20:26.08 | [TK]D-Fender | [hC-] : Sounds a bit like echo.... |
20:26.28 | [TK]D-Fender | [hC-] : You you using "r" in your dial command? |
20:26.34 | [hC-] | [TK]D-Fender: nope. |
20:26.39 | l1nux | asterisk-xmpp ready for testing ? |
20:26.46 | [hC-] | [TK]D-Fender: and i dont think its echo, neither parties notice echo in their voice during the call. |
20:27.02 | l1nux | svn-8106 ! |
20:27.37 | [TK]D-Fender | [hC-] :/ |
20:27.44 | *** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com) |
20:28.53 | katakefalos | i am looking for support for a newly purchased TE110P |
20:30.39 | katakefalos | i dont see anyone chatting its hard to believe with so many users online there must be something wrong can someone msg me directly? |
20:30.52 | [hC-] | arg. brb. |
20:31.00 | [TK]D-Fender | katakefalos :It gets quiet at time. How about you just ask your question.... |
20:31.05 | katakefalos | oh ic :) |
20:32.07 | *** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
20:32.30 | katakefalos | i need to configure my TE110P to provide a PRI to a legacy PBX system and am on hold for about 20 minutes on digiums phone support sp i logged in here |
20:32.33 | [TK]D-Fender | [hC] : Care to pastebin your zapata.conf? |
20:32.51 | [TK]D-Fender | katakefalos : While continuing to hold I hop? |
20:33.14 | katakefalos | sure one sec |
20:33.46 | katakefalos | is it pastebin.org or so? |
20:34.23 | [TK]D-Fender | katakefalos : Wasn't a message for you. What is that actual problem? |
20:34.48 | [TK]D-Fender | And for general reference : |
20:34.49 | [TK]D-Fender | ~pb |
20:34.51 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
20:35.09 | RussCC | fugitivo: If I have a static map and I am allowing any UDP traffic from my VOIP provider I still am unable to make a call |
20:35.36 | [TK]D-Fender | RussCC : You're behind NAT I take it? |
20:35.37 | katakefalos | my zapata.conf : http://pastebin.com/508676 |
20:36.04 | Katonka_ | is it possible to use misdn with asterisk@home? |
20:36.22 | [TK]D-Fender | katakefalos : You'll find AMP support hard come-by.... |
20:36.29 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
20:36.30 | RussCC | [tk]d-fender: yes I am using nat |
20:36.49 | katakefalos | and my etc/zaptel.conf : http://pastebin.com/508679 |
20:36.55 | [TK]D-Fender | RussCC : You need to fill in either EXTERNIP or EXTERNHOST, and LOCALNET. Have you don't all of these? |
20:37.31 | RussCC | [tk]d-fender: I don't quite follow you |
20:38.35 | [TK]D-Fender | RussCC : You need to set those values in the [general] section of SIP.CONF for it to work. |
20:38.39 | *** join/#asterisk Davey|Work (n=davey@unaffiliated/davey) |
20:38.54 | RussCC | [tk]d-fender: ok thank you |
20:39.16 | [TK]D-Fender | katakefalos : Your channel declaration looks wrong, it should be "channel => 1-13". You were missing the ">" |
20:39.23 | Davey|Work | Hi there, I'm having issues with one of our SIP clients, when I call him, he gets the ringing, but I get one ring and straight to his voicemail, he cannot dial out or to voicemail, any suggestions where to look? |
20:40.08 | [TK]D-Fender | Davey|Work : Did his setup ever work? |
20:40.31 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
20:40.32 | Trazz | TK, whats the easies way to test music on hold? |
20:40.48 | katakefalos | Fende: tehy just picked up |
20:41.15 | [TK]D-Fender | Trazz : exten => 1,1,MusicOnHold |
20:41.25 | Davey|Work | [TK]D-Fender, for outbound only |
20:41.50 | [TK]D-Fender | Davey|Work : You said he can |
20:42.02 | [TK]D-Fender | 't "dial out". What is this "outbound" that is working? |
20:42.04 | Davey|Work | the only change since then is a new router/DSL modem, so I'm guessing the problem is there. But he has opened up all the ports for both TCP/IP and UDP traffic |
20:42.17 | Davey|Work | [TK]D-Fender, he *could* dial out, now he can't |
20:42.22 | RoyK | drumkilla: ping |
20:42.33 | [TK]D-Fender | Davey|Work : So he's behind NAT, and what about you? |
20:42.51 | Davey|Work | [TK]D-Fender, Asterisk is on our gateway machine |
20:43.06 | *** join/#asterisk [hC-] (n=hardcore@209.153.195.139) |
20:43.13 | [TK]D-Fender | Davey|Work : Pastebin his sip account info |
20:43.24 | Davey|Work | [TK]D-Fender, sure, anywhere in particular? |
20:43.24 | [TK]D-Fender | [hC] : Care to pastebin your zapata.conf? |
20:43.29 | [TK]D-Fender | ~pb |
20:43.30 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
20:43.42 | [hC-] | [TK]D-Fender: yeah sure, give me just a sec. |
20:43.43 | Trazz | Tk, no music on hold just silence with that |
20:43.47 | RoyK | ~seen drumkilla |
20:43.54 | jbot | drumkilla is currently on #asterisk. Has said a total of 15 messages. Is idling for 4h 51m 23s, last said: 'backblue: you can use DUNDi in combination with using the regexten feature in sip.conf. That will, the extension will only exist when the peer is registered'. |
20:44.04 | [TK]D-Fender | Trazz : you may have to specify a context with that. |
20:44.25 | [TK]D-Fender | Trazz : You may also have to shove an Answer in front, etc... |
20:44.31 | Trazz | ok |
20:44.41 | Davey|Work | [TK]D-Fender, http://pastebin.com/508689 |
20:44.53 | Davey|Work | [TK]D-Fender, essentially the same as all our other clients |
20:45.09 | [TK]D-Fender | Davey|Work : What is he using as a phone? |
20:45.16 | Davey|Work | Sipura something or other |
20:45.30 | [TK]D-Fender | Davey|Work : Sipura should use INFO for the DTMF mode |
20:45.35 | Davey|Work | OK, let me try that |
20:45.42 | [TK]D-Fender | and get rid of defaultip |
20:46.06 | Katty | mister fender. |
20:46.10 | Davey|Work | does he need to re-connect his phone for Asterisk to find it? |
20:46.14 | [TK]D-Fender | Ms. Katty. |
20:46.35 | [TK]D-Fender | Davey|Work : perhaps, and he should go into the ATA's web config to set the DTMF mode. |
20:46.36 | Trazz | TK, i did that and nothing on console and call answers but its dead air |
20:46.48 | Davey|Work | [TK]D-Fender, OK |
20:46.53 | Davey|Work | thats the web config, right? |
20:46.57 | [TK]D-Fender | Trazz : Do you have MP3's in the appropriate folder and MPG123 installed? |
20:47.21 | Trazz | i installed mpg123 yes |
20:47.26 | Trazz | let me check on files |
20:48.54 | argentas | Trazz: what does 'mpg123 --version' show? |
20:49.37 | Trazz | mpg123: Unknown option "version". |
20:50.07 | argentas | ok, that's cool, just wanted to check it wasn't your distro installing mpg321 |
20:50.17 | Katonka_ | what has asterisk -r to say which version ist is if i installed the mqueue misdn? |
20:50.41 | Trazz | yep. i downloaded it and installed it seperately |
20:50.47 | argentas | ok, good. |
20:51.09 | Trazz | file is installed in /usr/local/bin |
20:51.24 | argentas | when you dial the extension, does 'ps ax' show that mpg123 is running? |
20:51.39 | [TK]D-Fender | Davey|Work : yes, the SPA's web config |
20:51.54 | Trazz | this is on console now |
20:51.56 | Trazz | Jan 16 15:00:45 WARNING[13369]: res_musiconhold.c:336 spawn_mp3: /var/lib/asterisk/mohmp3 is not a valid directory |
20:51.56 | Trazz | Jan 16 15:00:45 WARNING[13369]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player |
20:52.12 | [TK]D-Fender | Trazz : I think that error pretty much explains itself... |
20:52.22 | Trazz | my install is not in /var/lib its in /opt/voip |
20:52.42 | Davey|Work | [TK]D-Fender, no luck |
20:52.43 | [TK]D-Fender | Trazz : non-standard... bad start. |
20:53.09 | [TK]D-Fender | Davey|Work. And you reloaded your config, had him make the chages and reboot his device? |
20:53.18 | Trazz | i was told i could make everything run from 1 directory so i changed install_prefix |
20:53.46 | fugitivo | where can i get cheap soundpoint 501 and 601? |
20:53.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
20:53.54 | Davey|Work | [TK]D-Fender, correct |
20:54.03 | [TK]D-Fender | Davey|Work : repastebin his config |
20:54.16 | [TK]D-Fender | Trazz : then you picked the wrong folder in musiconhold.conf |
20:54.38 | [TK]D-Fender | fugitivo : www.atacomm.com has the lowest price I've seen anywhere |
20:54.43 | argentas | Trazz: you'll need to edit the path in etc/asterisk/musiconhold.conf |
20:54.46 | Trazz | hehehe i just found that in my grep for mohmp3 :) |
20:54.48 | Davey|Work | [TK]D-Fender, http://pastebin.com/508712 |
20:54.50 | Trazz | thanks |
20:54.55 | crich1999 | Katonka_: you'll need to either make samples again or just copy asterisk03/configs/misdn.conf.samples /etc/asterisk/ |
20:54.58 | fugitivo | [TK]D-Fender: thanks |
20:55.33 | Katonka_ | crich1999, rgr |
20:55.45 | [TK]D-Fender | Davey|Work : pastebin the [sip] context in extensions.conf |
20:56.42 | Davey|Work | uh, I don't appear to have one. |
20:56.53 | [TK]D-Fender | Davey|Work : that would be a rather severe problem then :) |
20:57.08 | Davey|Work | but we have like 6-8 phones that work just fine :) |
20:57.14 | Davey|Work | so uhm. eck. |
20:57.48 | [TK]D-Fender | Davey|Work : I think you should take a closer look for aminute.... |
20:58.21 | Davey|Work | [TK]D-Fender, looking :) |
20:58.46 | *** part/#asterisk rue_work (n=not@h24-207-96-50.cst.dccnet.com) |
20:58.54 | Davey|Work | grep -R "\[sip\]" ./ in the asterisk config dir brings up nothing. eeek |
20:59.14 | Trazz | argentas, ok thats fixed but still when i call i get dead air when connected.. |
21:00.07 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
21:00.30 | *** join/#asterisk areski (n=areski@156.Red-83-44-65.dynamicIP.rima-tde.net) |
21:00.32 | Curus | In the asterisk realtime voicemail table, what is customer_id used for? |
21:00.46 | argentas | Trazz: ok, now check whether 'ps ax | grep mpg123' shows anything whilst you are listening to the dead are |
21:01.03 | jpablo | anyone has experience using unicall ? |
21:01.27 | Davey|Work | [TK]D-Fender, OK, so essentially this is entirely b0rked and the fact any of our phones work is entirely a fluke? |
21:01.31 | Trazz | 14203 ? S 0:03 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 |
21:01.31 | Trazz | 14208 ? S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 |
21:02.09 | [TK]D-Fender | Davey|Work : Dunno, pastebin everything and maybe I'll notice something you're not... |
21:02.44 | Davey|Work | [TK]D-Fender, thanks, but well, I don't know what half of this stuff is and I don't want to reveal anything I shouldn't :) |
21:03.59 | Trazz | Argentas, i actually hear a tiny bit of music for a split second when i first connect and then dead air the rest |
21:03.59 | Katonka_ | misdn.conf ports are my cards or the channels of the isdn cards... like 1-3 card 1 and 4-6 card 2 (2 B-Chan, 1 D-Chan) |
21:04.05 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
21:04.29 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
21:05.10 | argentas | Trazz: hmm, you're not maybe starting to play the moh then hanging up are you? |
21:05.21 | Trazz | nope |
21:05.22 | *** join/#asterisk [hC-] (n=hardcore@209.153.195.139) |
21:05.25 | Trazz | its connected the whole time |
21:05.30 | argentas | k |
21:05.39 | Trazz | exten => 5000,1,Answer |
21:05.40 | Trazz | exten => 5000,2,MusicOnHold |
21:05.42 | Trazz | thats teh test exten |
21:07.12 | argentas | ok, add 'exten => 5000,3,Wait(10)' to that |
21:07.29 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
21:07.54 | Trazz | same thing |
21:07.55 | [TK]D-Fender | Davey|Work : Sure, whatever :) |
21:08.17 | Curus | Hmm maybe it's telling that a recursive grep for customer_id in the * source comes up empty |
21:08.19 | argentas | ok, couldn't remember where MusicOnHold blocks or backgrounds |
21:08.34 | Curus | I wonder if deleting the column will work |
21:09.34 | crich1999 | Katonka_, what means rgr ? |
21:09.39 | argentas | Trazz: came into this part way through, have you already put the console output in pastebin? |
21:09.54 | jpablo | crap, i have a problem with unicall, when i search in google for the error i find a paste bin: http://fr.pastebin.ca/comments.php?bin_id=14873 |
21:10.07 | jpablo | that's my problem, anyone knows how to fix that ? |
21:10.08 | Trazz | not yet.. do i need to turn on something first? |
21:10.11 | Katonka_ | crich1999, Roger=I will do so. |
21:10.17 | Davey|Work | [TK]D-Fender, you have been most helpful, but really, why does this have to be so freaking complex? Heh |
21:10.20 | Katonka_ | crich |
21:10.35 | Davey|Work | where the GUI dammit? :D |
21:10.36 | crich1999 | Katonka_, i see |
21:10.49 | argentas | Trazz: 'set verbose 20' first.. and we'll see what that shows us |
21:10.51 | Zodiacal | anyone know the # to call for verizon lines that tell you what number your calling from? |
21:11.07 | Trazz | ok |
21:11.42 | Katonka_ | crich1999, but i am confused with the ports. i got my 2 hfc cards found by misdn-init config . now in misdn.conf i have to choose port 1 for extern and port 2 for intern? |
21:12.12 | Trazz | gonna pastebin now |
21:12.36 | SwK[Work] | anyone polycom certified around? |
21:13.06 | Trazz | argentas - http://pastebin.com/508749 |
21:13.15 | Trazz | i am still connected btw |
21:13.46 | crich1999 | Katonka_, you can choose the section name like you wish. It's just a sample, name it like you think its best |
21:14.01 | Trazz | argentas, this is after i manually hang up http://pastebin.com/508751 |
21:14.36 | Katonka_ | crich1999, but i got 2 ports or 6 with 2 isdn-cards? |
21:15.53 | crich1999 | Katonka_, you got 2 ports with 2 hfcpci based cards |
21:16.09 | [TK]D-Fender | Davey|Work : this is not complex.... trust me.. its peanuts... |
21:16.15 | argentas | Trazz: taking a look now... |
21:16.18 | *** join/#asterisk clive- (n=pirch@dsl-165-158-250.telkomadsl.co.za) |
21:16.19 | [TK]D-Fender | SwK : Whats your question? |
21:16.21 | Trazz | ok |
21:17.34 | Davey|Work | [TK]D-Fender, heh, you apparently have a lot more time on your hands than I :) |
21:17.56 | Katonka_ | crich1999, that misdn.conf is used by asterisk after a reboot? or how is asterisk now using my hfc-cards? is zaptel and bristuff still used when misdn is installed? are you speaking german? ^ |
21:17.56 | [TK]D-Fender | Davey|Work : I jsut read the WIKI.... and I've been playing around with * for almost 2 years.... |
21:18.10 | [TK]D-Fender | Davey|Work : Books aren't worth much.... |
21:18.15 | Davey|Work | yeah, see that 1 years 11 months and 2 weeks more than me ;) |
21:18.30 | [TK]D-Fender | Davey|Work : You need better samples and a decent teacher... |
21:18.30 | crich1999 | Katonka_, yes i do |
21:18.40 | Davey|Work | [TK]D-Fender, want a job? LOL |
21:18.46 | SkramX | lol |
21:18.47 | [TK]D-Fender | I can provide both at a very reasonable price! ;) |
21:19.05 | crich1999 | Katonka_, join asterisk-misdn |
21:19.12 | SkramX | God, I had a AOL employee approach me for Asterisk consulting last night. |
21:19.14 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
21:19.26 | argentas | Trazz: is it definately a moh issue, what happens if you replace MusicOnHold with for example 'SayDigits(123456789)' ? |
21:19.31 | Davey|Work | SkramX, *For* AOL? |
21:19.37 | SkramX | Davey|Work: Maybe.. |
21:19.54 | SkramX | LOL, no.. it was for a company he wanted to start himself. |
21:20.11 | Druken | AOL is a pain in the ass |
21:20.27 | Trazz | ya its says the numbers |
21:20.48 | SwK[Work] | [TK]D-Fender is about their certification program... |
21:20.51 | SkramX | This employee wanted consulting for about 10 dollars an hour |
21:20.52 | SkramX | Oh well. |
21:20.58 | SkramX | <-- Not dirt. |
21:21.21 | Davey|Work | $10/hr? wow, thats like... crap |
21:21.29 | Davey|Work | my wife for more than that answering phones for UPS. |
21:21.32 | Davey|Work | quite a bit more |
21:21.38 | SkramX | Yeah |
21:21.39 | SkramX | heh |
21:21.47 | SkramX | Speaking of UPS.. im waiting for my package! |
21:22.14 | shmaltz | anybody ever heard of stratasoft? |
21:22.27 | jdv79 | what is brown doing for you? |
21:22.36 | jdv79 | i think they're off today - somme of them |
21:22.54 | Davey|Work | I got an unexpected package in the mail today, I heard when Ic alled my wife at lunch, she opens it up and I get all excited... turns out my mom had just ordered some stuff she couldn't get shipped to the UK and wanted it to go via me first. Hinges. |
21:23.10 | Davey|Work | SkramX, its a bank holiday, they may be off |
21:23.16 | Trazz | argentas, it says the numbers ok |
21:23.35 | SkramX | hmm |
21:23.51 | SkramX | im gonna jet for a bit |
21:23.54 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
21:24.00 | SkramX | tootles, feel free to pm me for whatever |
21:25.08 | znoG | whats the best way, from an AGI script, to find out of a particular EXTEN has a voicemail configured_ |
21:25.11 | znoG | ? |
21:25.35 | znoG | running "show voicemail users" and doing a grep on that would work, but from an AGI script how can I run such command? |
21:25.46 | znoG | unless I do an asterisk -rx and run it that way |
21:26.08 | [TK]D-Fender | znoG : you can run CPI commands from AGI direct.... |
21:26.29 | znoG | how so? |
21:27.41 | [TK]D-Fender | CLI* |
21:27.52 | [TK]D-Fender | znoG : Don't recall the exact command, but its there... |
21:28.02 | jbalcomb | hrmm.. significant lack of information specific to 'Unable to handle indication 3' |
21:28.13 | [TK]D-Fender | znoG : you can also parse voicemail.conf direct as well.. |
21:28.27 | Trazz | Tk, any ideas on why my music on hold would be broken? |
21:28.30 | znoG | [TK]D-Fender: yea, could do that too. Was hoping via AGI |
21:29.59 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
21:30.50 | [TK]D-Fender | znoG : znoG : You can use either a System call, or AMI, or parse the file direct. take your pick. I use AMI on my Polycom Idle scripts for queue stats & voicemail. |
21:31.20 | znoG | AMI? |
21:31.28 | znoG | oh |
21:31.31 | znoG | asterisk manager interface |
21:31.51 | [TK]D-Fender | yup, ok, gtg, back later |
21:32.02 | *** join/#asterisk oogle_ (n=oogle@63.215.127.17) |
21:33.26 | *** join/#asterisk nicknick (n=nicknick@81-86-107-241.dsl.pipex.com) |
21:33.37 | oogle_ | i'm having a big problem parking sip calls with moh-native. I initiate the transfer to 700 and the person being parked starts hearing music on hold, then when the channel is parked, the music on hold stops and starts again really loud and distorted |
21:33.45 | *** join/#asterisk wulf814 (n=lorentz@216.48.0.4) |
21:34.59 | Trazz | argentas you there ? |
21:35.49 | Trazz | can someone review this http://pastebin.com/508809 |
21:35.51 | wulfy814 | ok, silly question I just changed ISP's at the office |
21:36.00 | wulfy814 | and my GS2000 won't register |
21:36.04 | wulfy814 | with my * at home |
21:36.13 | wulfy814 | it is still showing the old IP from the old ISP |
21:36.27 | wulfy814 | how do I kill it? so that it can register from the new ISP |
21:42.22 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
21:42.23 | a1fa | hi |
21:42.57 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
21:43.45 | a1fa | on 1.2.1 |
21:44.39 | justinu | funny, i named a computer wiggum also |
21:45.30 | a1fa | sue them |
21:46.01 | a1fa | trademark infirngement |
21:46.03 | a1fa | !!!! |
21:46.05 | a1fa | yo yo yo |
21:46.09 | a1fa | z;) |
21:46.30 | devel | cool, justinu |
21:47.46 | a1fa | aiiiighhhhhhhhhhht |
21:48.11 | *** part/#asterisk oogle_ (n=oogle@63.215.127.17) |
21:48.28 | *** join/#asterisk gml (n=rm@66.193.229.9) |
21:49.41 | gml | hey when i do "zap show channel 1" it says "Dialing: no" and i get Don't know what to do w |
21:49.41 | gml | ntrol frame 15 |
21:49.41 | infinity1 | can someone familiar with polycoms tell me the point of the server tab in the web UI when there is a server associated with each configurable line? |
21:49.41 | gml | don't know what to do with control frame 15 |
21:49.46 | gml | is that the CO side or is that me? |
21:50.47 | justinu | infinity: in case the line doesn't have a server associated with it, it'll fall back to the "generic" server section |
21:51.20 | lesouvage | tazz: check your /etc/asterisk musiconhold.conf. There shoudl be a class default under [classes] |
21:52.02 | a1fa | lol sucks that asterisk wants termcap |
21:52.25 | *** part/#asterisk RussCC (n=face@216.157.205.211) |
21:52.38 | infinity1 | justinu: ic thanks :) |
21:53.53 | lesouvage | gml: I have the same while calling over the internet. I guess it has to do with the provider. |
21:54.23 | infinity1 | justinu: should register be set to Yes for the default server? i'm don't follow the significance. some poeple say set it to no.... confusing |
21:56.01 | sivana | anyone here in 902 area code? |
21:56.03 | Vyeperman | Does anyone have a recommendation for a cheap ip phone that I can throw in all of kids rooms, I have two nice ones for my business, but I'd like to be able to stick on in each room, and I'm not willing to pay $160 a piece for it. |
21:56.37 | Ikaruszszszsz | Vyeperman: budgetone |
21:56.58 | justinu | infinity: register=0 means that the phone can call out even if it can't register with a proxy |
21:57.15 | justinu | register=1 means that if there are no succesful registrations, it won't allow outbound calls |
21:57.23 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
21:57.29 | Trazz | any good information on how to build IVR ? |
21:58.15 | infinity1 | justinu: is all this documented someone? or are you the only one that knows? heh |
21:58.21 | clive- | trazz, use asterisk |
21:58.23 | infinity1 | er s/someone/somewhere/ |
21:58.30 | clive- | :0 |
21:58.32 | jarrod | hmm |
21:58.41 | jarrod | on all my calls im getting this 'ssssshhhh' |
21:58.44 | jarrod | 'sssshhh' |
21:58.53 | Trazz | clive- yes i am messing with it now. i would like to see some samples configs / examples on how to build IVR |
21:59.00 | justinu | infinity: it's in the polycom admin guide. |
21:59.05 | clive- | trazz check the wiki |
21:59.23 | Vyeperman | Ikaruszszszsz, thnx |
21:59.24 | justinu | afk |
21:59.55 | Ariel_ | Trazz, there is a great sample you can fallow of an ivr in your setup now. it's located at /usr/src/asterisk/configs/extensions.conf.sample |
21:59.56 | infinity1 | justinu: i swear i searched the admin guide for the word 'register' and didn't find anything good. i'll look again. |
22:00.48 | Trazz | Ariel, great your back..i can't music on hold working |
22:01.30 | *** join/#asterisk libila (n=vye@ip68-8-174-154.sd.sd.cox.net) |
22:01.38 | infinity1 | justinu: yer right. i just had to dig harder. |
22:02.00 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
22:02.01 | Ariel_ | Trazz, how did you configure the moh |
22:02.08 | *** join/#asterisk skamp|afk (i=skambar@p5484080E.dip0.t-ipconnect.de) |
22:02.13 | Trazz | i dl it , make linux and then make install |
22:02.17 | blitz[class] | tzanger: oh happy day... Canada is moving right wing on Jan 23rd it sounds like :( |
22:02.34 | Beirdo | bah |
22:02.34 | libila | I have 3 phone lines, and probably about 8-10 ip phones. Which card would fit my needs best? |
22:02.41 | Beirdo | we shall see, blitz[class] |
22:02.57 | Ariel_ | right wing left wing.. just a bunch of bs if you really think about it.. |
22:03.02 | Beirdo | and if we get a change of lying scum, that's a good thing in my mind, time for new lying scum |
22:03.05 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
22:03.14 | Umaro | hey guys.. anyone here using asterisk with telasip? |
22:03.19 | jarrod | why do calls thru asterisk suck |
22:03.24 | Trazz | Ariel, i dl it , make linux and then make install, when i dial that extension i get no music |
22:03.36 | *** join/#asterisk saitech (n=admin@85.235.237.14) |
22:03.47 | Trazz | yet it says its playing with verbose on 20 |
22:03.58 | tzanger | blitz[class]: ? |
22:04.01 | Ariel_ | Trazz, did you install mpg123 or what are you using for moh |
22:04.12 | Trazz | yes i installed mpg123 |
22:04.23 | tzanger | I'm voting green, I've had enough of the liberals, even though they're the only financially sound choice. the PC will never get my vote and hte NDP's a joke |
22:04.43 | Beirdo | oh, and blitz[class], you can't blame me this time, I voted in advanced polls... for the pinko bastard NDP as they actually have a real good chance of taking my riding. |
22:05.08 | saitech | i'm having a problem on my asterisk production box. It keep spams we with "chan_sip.c: failed to grab lock... trying again" in my debug log. The actually problems is, that it makes my box hang, while this debug is posted in the log.Can anyone help, or does anyone identify this problem ? |
22:05.14 | Beirdo | Liberals financially sound? have you slept through the scandals, tzanger? :) |
22:05.24 | Ariel_ | wow ndp, pinko's, green sounds like fun games for kids |
22:05.52 | Beirdo | voting Green Party's a good thing :) |
22:06.02 | tzanger | Beirdo: no. what I am saying is that they are *excellent* at turning a country around. However they take all that surplus and pocket it. That's the downside. The PC or NDP could never do what the libs did in terms of fiscal responsibility on a national level. |
22:06.14 | Beirdo | heh |
22:06.19 | Beirdo | dunno about that |
22:06.24 | tzanger | so I figure it's time to let some fresh blood in and if they fuck up the country, vote the liberals back in to clean up but get 'em out before they keep all the money |
22:06.32 | Beirdo | heh |
22:06.36 | Beirdo | sounds like a plan |
22:06.38 | jarrod | anyone had a problem with receive 'hiss' noises? |
22:06.39 | *** join/#asterisk rue_work (n=not@h24-207-96-50.cst.dccnet.com) |
22:06.52 | Trazz | Ariel, when i dial my extension i get a split second of medicine and then silence |
22:06.53 | rue_work | quick!? how do I find out how many active calls are on a machine???? |
22:07.02 | jbalcomb | jarrod I am having troubles with hiss, static, and echo across the board |
22:07.11 | jarrod | jbalcomb: asterisk 1.2.1? |
22:07.13 | rue_work | I think our pstn machine jsut went fooie, that or all our lines are in use... |
22:07.25 | tzanger | but no I'm voting green because I want to get fresh blood and new ideas in. They haven't got a hope of getting majority so it's "safe" and if I can get ONE seat green then I'm hoping ot see the greens at the next debates and really see what they can do. They are hellishly disorganized but let's give their crackpot ideas some voice and see what happens |
22:07.30 | tzanger | jarrod: no not me |
22:07.42 | Beirdo | hehe, cool |
22:07.54 | Beirdo | wonder if the Marijuana Party will land a seat? |
22:08.02 | rue_work | ANYONE!? |
22:08.16 | blitz[class] | rue_work: how about 'show channels' |
22:08.30 | tzanger | rue_work: nobody can tell but you |
22:08.47 | Trazz | brb |
22:09.11 | rue_work | show channels might be right.... |
22:09.29 | jarrod | asterisk quality is sucking |
22:09.31 | rue_work | I need to know if I need to reboot the pstn mahine or if the lines really are all used up |
22:09.54 | tzanger | rue_work: show channels |
22:10.00 | rue_work | hoooo, looks like we just used up all our lines |
22:10.03 | blitz[class] | well... the PC scares the hell outta me... and I'd rather the Liberals get back in since everyone should be watching exactly what they are doing -- would be pretty hard to steal money I would hope |
22:10.06 | rue_work | that freaked me |
22:10.19 | tzanger | blitz[class]: nah it's as easy as ever |
22:10.27 | saitech | i'm having a problem on my asterisk production box. It keep spams we with "chan_sip.c: failed to grab lock... trying again" in my debug log. The actually problems is, that it makes my box hang, while this debug is posted in the log.Can anyone help, or does anyone identify this problem ? |
22:10.56 | blitz[class] | tzanger: although I'm not too impressed with the Liberals trying to make handguns illegal -- criminals don't register their guns |
22:11.20 | a1fa | blah |
22:11.23 | blitz[class] | you know what... if you think the last minority gov't was a joke, this time around is going to be even worse |
22:11.42 | Umaro | no one here is using telasip? :/ |
22:11.48 | Beirdo | this time will be FUN, blitz[class] |
22:12.04 | Beirdo | I hope we get NDP minority so they have to bend over for everyone to get anything done |
22:12.06 | tzanger | blitz[class]: yeah that's a fool's errand used to placate the old and dumb |
22:12.14 | tzanger | I LIKE minority governments |
22:12.20 | blitz[class] | tzanger: ditto |
22:12.24 | tzanger | but I fucking hate the bullshit tactics that the conservatives did to end it |
22:12.25 | Johnnie | Hmmm... |
22:12.29 | Beirdo | me too, they tend to be more accountable |
22:12.30 | bkw_ | saitech, those are bad messages.. I can do a literal translation of the message if you lilke? |
22:12.31 | Beirdo | bah |
22:12.31 | tzanger | fucking waste my fucking money just ot get the exact same thing |
22:12.40 | Johnnie | Anyone here intimately familiar with zaptel sources and ztdummy? |
22:12.43 | blitz[class] | tzanger: if the PC's get a majority gov't I might as well just move to the US :) |
22:12.51 | Beirdo | and bah again |
22:12.57 | sivana | tzanger: they all suck and just fall into the same rut, year after year |
22:13.01 | blitz[class] | tzanger: actually... I'm considering moving to SF some August |
22:13.01 | Beirdo | and I will be anywyas |
22:13.04 | tzanger | blitz[class]: you won't have to, they basically said they would become a state |
22:13.12 | blitz[class] | tzanger: thats true |
22:13.15 | tzanger | sivana: which is why, like diapers, government needs frequent changing |
22:13.28 | tzanger | I want to get rid of the two-party system |
22:13.32 | bkw_ | too bad it never happns |
22:13.41 | bkw_ | do they make Governmental Drano? |
22:13.42 | Beirdo | we don't have a 2-party system, silly |
22:13.43 | saitech | bkw_ i have looked in the code, and it seems like its trying to retrieve some owner-lock from a sip call. but failed all the time. im not sure, but it seems that the lock is the header . |
22:13.44 | tzanger | I fully expect if the PC win for Harper to unzip his skin and Preston Manning to jump out |
22:13.47 | Beirdo | we have a 4-party system |
22:13.52 | tzanger | Beirdo: yeah right. |
22:13.56 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
22:14.03 | sivana | tzanger: I'm voting green party, what else do we have to lose |
22:14.06 | Beirdo | and Preston Manning has a lot more integrity than Harper :) |
22:14.08 | saitech | bkw_ if you have a better perspective i would be glad to hear so |
22:14.12 | tzanger | sivana: exactly :-) |
22:14.27 | libila | Does anyone have anything against this interface card: http://www.voicetronix.com.au/openpci.htm I have 8-10 ip phones & 3 phone lines I want to hook up, and of course I'll be using asterisk(FreeBSD). |
22:14.34 | Beirdo | or Stockwell Day. HE was a joke |
22:14.39 | tzanger | haha |
22:15.00 | tzanger | the PC are jokes... they don't have a platform nor a personality... they are the "we are not the liberals" party |
22:15.06 | tzanger | that's their entire platform |
22:15.11 | Beirdo | not true |
22:15.18 | tzanger | and the NDP is "we'll give money to everyone and everything to get in" party |
22:15.24 | blitz[class] | Kim Campbell was my hero |
22:15.28 | Beirdo | they are just chosing to not state any policy to not piss people off |
22:15.42 | tzanger | Beirdo: which indicates to me that they have no platform |
22:15.47 | Beirdo | no |
22:15.49 | tzanger | if you're not gonna tell me then why the fuck would I vote you in |
22:15.50 | Beirdo | they have one |
22:15.56 | sivana | it's just not public |
22:16.04 | Beirdo | but one that many people who are sitting on the fence may not like |
22:16.09 | blitz[class] | "vote us in and we'll tell you what we're going to do after we have already done it" |
22:16.15 | Beirdo | and one that half the people IN the party won't like |
22:16.21 | sivana | hehe |
22:16.25 | blitz[class] | Beirdo: thats terrible, irresponsible politics |
22:16.41 | Beirdo | more honest than the Liberals who promise piles of stuff and don't do any of it :) |
22:17.00 | blitz[class] | I hate consevative values |
22:17.03 | *** join/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
22:17.11 | Beirdo | we still have the GST that Chretien promised to get rid of |
22:17.15 | blitz[class] | I'm pretty much pro everything they are against |
22:17.39 | blitz[class] | Beirdo: whatever... you replace the GST with higher income tax -- no matter what happens you still don't get that money |
22:17.45 | blitz[class] | at least with GST I can choose not to buy stuff |
22:17.46 | SarahEmm | hiya |
22:17.58 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
22:18.10 | blitz[class] | SarahEmm: hoi-oh |
22:18.17 | Beirdo | SarahEmm: ahoy! |
22:18.23 | sivana | SarahEmm: yo |
22:18.32 | SarahEmm | hi :) |
22:18.34 | Beirdo | evil pirate wench :) |
22:18.36 | Beirdo | heh |
22:18.40 | SarahEmm | :) |
22:18.46 | *** part/#asterisk clive- (n=pirch@dsl-165-158-250.telkomadsl.co.za) |
22:18.51 | Beirdo | we STILL spend too much time in htat damn game |
22:19.02 | Beirdo | come buy some black cloth from us on DN |
22:19.04 | Beirdo | heh |
22:19.07 | tzanger | I'm addicted to Civ4 |
22:19.14 | blitz[class] | thank goodness I don't play games or I'd never get anything done |
22:19.16 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
22:19.17 | SarahEmm | DN is home for me :) |
22:19.20 | tzanger | blitz[class]: that's what I said too |
22:19.27 | Beirdo | me too, SarahEmm. |
22:19.32 | SarahEmm | anyone remember who the one here working on TTY/TDD stuff is, other than me? |
22:19.38 | bsdfreak | HEH |
22:19.40 | Beirdo | sorry, not off hand |
22:20.22 | Beirdo | well, I should go home anyways. got some cleaning, puzzle pirates, etc to do :) |
22:20.24 | saitech | bkw_: what would you translate the message into ? |
22:21.09 | Beirdo | seeya all. Don't forget to vote, regardless of which lying scum ya vote for :) |
22:21.13 | Beirdo | hehe |
22:21.25 | *** join/#asterisk x[Girl] (n=FunGirl_@stjhnf0112w-142163115151.pppoe-dynamic.nl.aliant.net) |
22:22.07 | SarahEmm | heh :) |
22:24.00 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
22:25.56 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
22:26.00 | gml | hey i'm trying to dial out on a Zap interface and i keep getting "Unallocated (unassigned) number" back |
22:27.27 | *** join/#asterisk cpm (n=Chip@68-66-23-191.chvlva.adelphia.net) |
22:28.11 | infinity1 | how can you make the polycom register on using a random poort? when i do 'sip show peers' its always 5060 |
22:28.31 | SarahEmm | why would you want to, infinity1? |
22:28.58 | libila | http://www.voicetronix.com.au/openpci.htm that interface card says Bus Type: PCI2.2, is that the common 32-bit PCI bus? |
22:29.08 | infinity1 | SarahEmm: because i have multiple polycoms behind nat with asterisk on the other side |
22:29.33 | SarahEmm | libila: one of the common types. your slots could be 1.x or 2.x... |
22:29.40 | SarahEmm | infinity1: ahhh... |
22:29.50 | SarahEmm | infinity1: the source port is 5060 you mean? |
22:30.39 | libila | SarahEmm: any easy way to tell short of looking up the mobo? |
22:30.48 | SarahEmm | does the mobo have ISA slots too? |
22:30.56 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
22:31.06 | *** part/#asterisk ping1 (n=DLBaker@67-133-167-72.dia.cust.qwest.net) |
22:31.24 | infinity1 | SarahEmm: yes. i'm assuming thats what is shown in sip show peers. all the other phones are random |
22:31.29 | SarahEmm | ahh.. |
22:31.35 | libila | SarahEmm: I don't believe so, it's not that old. |
22:31.45 | infinity1 | SarahEmm: except the polycom. |
22:31.49 | SarahEmm | okay. it's likely 2.2 then, not sure how you can be sure. |
22:31.59 | sivana | SarahEmm: you aren't registered to my server |
22:32.47 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
22:32.48 | SarahEmm | sivana: my box is down with a hardware failure right now |
22:32.54 | sivana | oh ok :) |
22:32.55 | libila | SarahEmm: if you had 3 phone lines to connect to 8-10 ip phones running asterisk on FBSD would you buy this card? http://www.voicetronix.com.au/openpci.htm |
22:33.12 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
22:33.38 | blitz[class] | SarahEmm: unfortunately there isn't really any easy way to tell if its 2.2 or 2.1 unless you check the MB manual |
22:33.53 | SarahEmm | i'm not sure libila, i don't have much experience with analog fxs/fxo cards :) |
22:34.08 | libila | SarahEmm: Neither do I, and I don't have a clue. |
22:34.13 | blitz[class] | fxs/fxo....ewww :) |
22:34.19 | blitz[class] | analog is gross :D |
22:34.25 | SarahEmm | i'd go with something a bit better supported than voicetronix.. they need exsternal binary-only drivers, no? |
22:34.32 | libila | blitz[class]: how else do you hook it up to your phone lines? |
22:34.44 | infinity1 | justinu: how do you make the polycom use a random source port, instead of the default 5060? |
22:36.28 | Aughey | libila: I'd suggest http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
22:36.46 | SarahEmm | anyone here familiar with the internals of fsckmodem.c? |
22:36.48 | SarahEmm | fskmodem.c rather |
22:37.11 | *** join/#asterisk saftsack (n=saftsack@p54A7F6F8.dip.t-dialin.net) |
22:37.16 | saftsack | hi |
22:39.12 | MstlyHrmls | infinity1: volpProt.local.port |
22:39.22 | MstlyHrmls | infinity1: in sip.cfg |
22:39.35 | infinity1 | MstlyHrmls: thanks! |
22:40.15 | infinity1 | MstlyHrmls: it says if you put 0, it still uses 5060. |
22:40.20 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
22:40.36 | MstlyHrmls | infinity1: yes, the Polycoms don't do the random source address thing |
22:40.43 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
22:40.51 | infinity1 | MstlyHrmls: hm. what do you suggest? |
22:40.55 | MstlyHrmls | infinity1: you'll have to assign a unique port to each phone behind the NAT |
22:41.30 | infinity1 | :/ ..k |
22:42.10 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
22:42.13 | *** join/#asterisk Flauto (n=zhao@c-71-194-194-48.hsd1.il.comcast.net) |
22:42.16 | libila | Aughey: Alright, well since I don't know what FXS is I don't think I'll need it, I'll probably get the TDM04B package. |
22:42.18 | ManxPower | WHERE does that myth come from???? |
22:42.25 | Flauto | hello everyone |
22:42.28 | ManxPower | There is no reason to assign each phone a different port. |
22:42.29 | Flauto | how you guys doing |
22:42.35 | saftsack | some iaxmodem experts here? |
22:42.40 | ManxPower | Unless your NAT router sucks, of course. |
22:43.18 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
22:43.20 | Flauto | is there anyone who can help me with a2billing? |
22:43.24 | Flauto | i tried a couple of times |
22:43.28 | Flauto | but did not work out |
22:43.39 | ManxPower | Flauto, Sorry, I don't bill for calls. |
22:43.46 | Flauto | hehe |
22:43.48 | ManxPower | ~fxofxs |
22:43.51 | jbot | well, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
22:43.51 | Flauto | manxpower |
22:43.52 | SarahEmm | ManxPower: meow! |
22:43.53 | Flauto | i dont' either indeed |
22:43.58 | Flauto | but i just wnat to play with it |
22:44.05 | ManxPower | hellop, SarahEmm |
22:44.11 | *** join/#asterisk |omni| (n=rob@net98.limelyte.net) |
22:45.42 | infinity1 | argh. when using the mac to assin the phones names in sip.conf, you can't use the default buttons to connect to voicemail. hmm |
22:45.55 | infinity1 | if its not one thing, its another. hah |
22:48.14 | dpryo | Somebody know how to use the feature buttons on Avaya 4620, under asterisk/sip? |
22:48.27 | jarrod | i get hiss noises on all channels, zap, sip, mgcp |
22:50.15 | *** join/#asterisk Katonka (n=Miranda@p54BEE46D.dip.t-dialin.net) |
22:50.29 | libila | if you just have ip phones but your using a normal phone line you jus need fxo ports right? |
22:50.39 | ManxPower | infinity1, Huh? |
22:50.41 | fugitivo | ~fxsfxo |
22:50.42 | jbot | i heard fxsfxo is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
22:50.44 | ManxPower | infinity1, Um, what phone? |
22:51.37 | libila | k so you do need both... |
22:51.50 | fugitivo | libila: no |
22:52.03 | fugitivo | libila: you already have ip phones |
22:52.21 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
22:52.22 | fugitivo | libila: you only need fxo ports to connect the line |
22:52.45 | *** join/#asterisk HuSeyiN (n=_Yalniz-@stjhnf0112w-142163117148.pppoe-dynamic.nl.aliant.net) |
22:53.16 | *** join/#asterisk Strom_C (n=strom@adsl-69-221-57-46.dsl.chcgil.ameritech.net) |
22:53.23 | Lots | whatd the diff between asterisk@home and regular asterisk? |
22:53.40 | fugitivo | Lots: well |
22:53.49 | ManxPower | Lots, Asterisk@HOME is some lame ass GUI that people like. |
22:53.53 | fugitivo | Lots: asterisk@home is a complete distribution, that includes asterisk |
22:54.06 | fugitivo | ManxPower: no, it's not a gui |
22:54.12 | fugitivo | amp is a gui |
22:54.20 | ManxPower | In nay case, we don't really talk about it here. |
22:54.24 | ManxPower | ~amp |
22:54.25 | jbot | somebody said amp was NOT supported here! people using it should join #amportal |
22:54.39 | fugitivo | Lots: asterisk@home is a distro that includes linux + asterisk + amp (gui) and it sucks |
22:55.05 | fugitivo | Lots: regular asterisk is the way to go |
22:55.31 | fugitivo | Lots: you pick the distro you like more, compile asterisk from source and configure it using plain text files |
22:55.43 | *** part/#asterisk l1nux (i=moi@214.138.103-84.rev.gaoland.net) |
22:55.56 | saftsack | fugitivo, hi, do you know iaxmodem? |
22:56.03 | Lots | fug thats where i'm at right now, running the most recent asterisk on debian |
22:56.14 | Lots | but reading this article about asterisk@home. |
22:56.50 | fugitivo | saftsack: no |
22:57.08 | infinity1 | ManxPower: polycom |
22:57.20 | *** join/#asterisk saftsack (n=saftsack@p54A7F6F8.dip.t-dialin.net) |
22:57.32 | fugitivo | Lots: asterisk@home is a try to bring a pbx to regular users, and "I" think, regular users shouldn't mess with a PBX |
22:57.40 | Lots | lol |
22:57.48 | ManxPower | infinity1, You're smarter than you look. Using Polycoms and setting up the phones with their MAC as their sip.conf username. |
22:57.49 | *** join/#asterisk Snooker (n=klayton@201.4.213.114) |
22:58.07 | ManxPower | infinity1, as a special bonus, do you want my config files? |
22:58.17 | infinity1 | ManxPower: for sure! |
22:58.25 | ManxPower | hold on. |
22:59.33 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
23:01.28 | ManxPower | infinity1, I'll include my config files, one set of config files for one of my phones, as well as the sip and bootrom that I use. |
23:01.44 | infinity1 | ManxPower: k. email? dcc? |
23:01.55 | ManxPower | infinity1, I'll put it on a web site. |
23:02.05 | infinity1 | ok. thanks! |
23:02.20 | ManxPower | www.fnords.org/~eric/tmp/poly.tar.gz |
23:02.25 | infinity1 | for my current problem, i think setting a "callback attribute" is necessary. |
23:02.28 | ManxPower | it's about 11MB with all the polycom stuff |
23:03.38 | ManxPower | infinity1, most of my users are too stupid to try anything fancy like pressing the voicemail key. |
23:03.57 | infinity1 | hah |
23:04.09 | crich1999 | do the Polycom phones allow Sendtext ? |
23:04.19 | ManxPower | crich1999, not when I tried it. |
23:04.36 | ManxPower | infinity1, one of my users actually told me they don't use text messaging because "it's too complicated" |
23:04.45 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
23:04.55 | crich1999 | ManxPower, hm .. the Snoms do, but you need to press the Message button to retrieve the text info, which really sucks |
23:05.19 | crich1999 | i search a nice sip phone which directly displays the text sended by sendtext |
23:05.52 | umay | i can't find one of those, either |
23:05.57 | Nugget | make one that has backlight and I'll order 100. |
23:06.05 | crich1999 | hehe |
23:06.08 | SarahEmm | hehe |
23:06.29 | umay | ADSI allowed for it ages ago |
23:06.31 | Darwin35 | ast_load became ast_conf_load right |
23:06.41 | crich1999 | isdn phones can do that too |
23:08.06 | *** join/#asterisk ToTo (n=ToTo@host16-146.pool872.interbusiness.it) |
23:08.28 | *** join/#asterisk backblue (n=moo@87-196-15-214.net.novis.pt) |
23:10.10 | Darwin35 | did ast_load become ast_conf_load in odbc |
23:10.53 | Darwin35 | trying to get dbodbc to work |
23:12.56 | *** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
23:13.03 | darwin_35 | afetrnoon all |
23:13.35 | darwin_35 | <PROTECTED> |
23:13.36 | darwin_35 | Jan 16 16:13:41 WARNING[94919]: loader.c:554 load_modules: Loading module app_dbodbc.so failed! |
23:13.43 | darwin_35 | neeed help with this error |
23:14.26 | file[laptop] | I've been over this before |
23:14.31 | file[laptop] | use ast_config_load |
23:14.50 | infinity1 | ManxPower: so 3509 is your voicemail app? |
23:15.27 | darwin_35 | there where 2 lines to change I could not find them |
23:15.34 | darwin_35 | File thats why I asked |
23:16.38 | file[laptop] | well, you'll have to find them... they're in app_dbodbc's source code |
23:16.39 | darwin_35 | what whas the destroy line file ? |
23:16.57 | file[laptop] | ast_config_destroy?"" |
23:17.19 | saitech | i'm having a problem on my asterisk production box. It keep spams we with "chan_sip.c: failed to grab lock... trying again" in my debug log. The actually problems is, that it makes my box hang, while this debug is posted in the log.Can anyone help, or does anyone identify this problem ? |
23:17.26 | ManxPower | infinity1, correct |
23:17.45 | *** join/#asterisk Jaxx[18f] (n=SiGn@stjhnf0112w-142162207029.pppoe-dynamic.nl.aliant.net) |
23:18.18 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-48-187.cybersurf.com) |
23:19.13 | Zodiacal | anyone know the advantage asterisk can give you by using a fax machine on an fxs port? insted of just wiring a fax machine to the phone line directly? |
23:20.02 | darwin_35 | thanks file |
23:20.12 | darwin_35 | I will treat you to dinner |
23:21.02 | Skumling | humm, I just had a very odd problem with *... when trying to call out using zap/g1 it kept failing with "Channel 0/1, span 1 got hangup, cause 42" - I was able to ring in to asterisk via the same zap-interface using my cellphone |
23:21.37 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
23:22.28 | Skumling | restart/reload of asterisk including ztcfg -vv didn't help... then I tried changing extensions.conf to zap/1 and it worked... also it worked with zap/2, and when changing back to zap/g1 that also works - looks like something in the dynamic channel allocation screwed up? |
23:22.46 | darwin_35 | it worked |
23:24.08 | *** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
23:24.19 | *** join/#asterisk corposant1 (i=corpo@dyn-shp-225-67.dyn.columbia.edu) |
23:24.57 | Skumling | If I should hook up a SIP-phone to Asterisk, would the Grandstream GXP-2000 be a decent choice? |
23:27.27 | corposant1 | anybody here have a moment to give me some input on an asterisk issue? |
23:27.55 | corposant1 | i'm having trouble with a phone being "UNREACHABLE" from the asterisk server, but it's able to make calls |
23:28.01 | corposant1 | just not receive them |
23:28.22 | Skumling | iDunno: humm okay, looks cool, but pricy too... |
23:28.24 | Corydon-w | corposant1: SIP phone? |
23:28.30 | corposant1 | yessir |
23:28.36 | corposant1 | asterisk has a public ip |
23:28.39 | corposant1 | as does the sip phone |
23:28.40 | Corydon-w | corposant1: set qualify=no |
23:28.42 | corposant1 | no nat |
23:28.45 | corposant1 | ah...k |
23:29.06 | Corydon-w | corposant1: the problem is that your SIP phone is not properly responding to an OPTIONS request from Asterisk |
23:29.33 | Corydon-w | If it doesn't respond to OPTIONS, Asterisk sees it as unreachable |
23:29.50 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
23:29.52 | Corydon-w | Even an error response to OPTIONS works... but no response is bad |
23:30.03 | corposant1 | that was indeed it |
23:30.07 | corposant1 | it's working |
23:30.18 | corposant1 | thanks! |
23:30.43 | corposant1 | next question would be how would I make it work if the phone were behind a nat? |
23:31.09 | corposant1 | doesn't the qualify setting act as a keep-alive for the nat? |
23:33.16 | *** join/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com) |
23:36.03 | [av]bani | http://bani.anime.net/phonez/ |
23:36.37 | *** join/#asterisk areski (n=areski@192.Red-83-60-99.dynamicIP.rima-tde.net) |
23:40.56 | Katty | hi. |
23:42.15 | Katty | the vonage commercial is annoying. |
23:42.51 | xachen | Vonage sucks |
23:42.59 | xachen | They are much like Microsuck |
23:44.14 | twisted[asteria] | woohoo woo hoo hoo |
23:44.17 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
23:44.44 | jdv79 | uhm, was it that good for you? |
23:46.33 | Darwin35 | yes it all works |
23:46.48 | Darwin35 | new fax machine rocks |
23:46.55 | Darwin35 | this kicks ass |
23:51.17 | Katty | oh boy, gateway media center! |
23:51.21 | Katty | just what i /always/ wanted! |
23:51.28 | twisted[asteria] | lol |
23:51.49 | infinity1 | anyone have a polycom configured with something like voxee? asterisk is easy, but i added a button for voxee, and it doesn't work |
23:52.19 | twisted[asteria] | ow |
23:52.24 | twisted[asteria] | my eye! |
23:52.57 | xachen | fax over Voip? :S |
23:53.06 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.103.247) |
23:53.26 | *** part/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com) |
23:53.29 | Katty | twisted[asteria]: i'd never poke you in the eye silly. |
23:54.18 | twisted[asteria] | Katty, mmkay. |
23:54.57 | iDunno | twisted[asteria]: that should make you ask where Katty *is* poking you. |
23:55.07 | Katty | gosh. |
23:57.05 | iDunno | owww. |
23:57.11 | iDunno | that 'urt. |
23:57.25 | iDunno | was that really neccessary? |
23:57.56 | [av]bani | how good is voxee? |
23:58.48 | justinu | you aint' all that beautiful either |
23:58.55 | justinu | so what does that leave... |
23:59.13 | Katty | be nice. |
23:59.44 | justinu | ok, ok... it's just that I'm not all that attracted to him. |
23:59.48 | twisted[asteria] | justinu, you're quite the mirror breaker yourself from what I hear. |
23:59.58 | Katty | woah |