| 00:00.51 | SomethingISODD | sorry alexshell i have some but not with festival |
| 00:06.09 | alexshell | SomethingISODD, what can you recommend me? I'm no closed to festival |
| 00:12.34 | alexshell | I would like to be able to run those phpagi examples |
| 00:13.22 | *** join/#asterisk mumtazah (n=mumtazah@203.82.79.101) |
| 00:14.22 | alexshell | I'm running Asterisk 1.4.22-3 |
| 00:22.01 | hardwire | woot |
| 00:26.05 | alexshell | is woot = what? |
| 00:26.45 | hardwire | woot = woot |
| 00:27.13 | hardwire | I was not wooting at thee |
| 00:27.55 | *** join/#asterisk Knightfal (n=Knightfa@75.142.144.171) |
| 00:28.51 | Knightfal | Everyone recovered from Astricon?!? |
| 00:30.29 | jaytee | yep, took an aspirin and a napped for an hour |
| 00:31.32 | Knightfal | nice |
| 00:34.16 | Knightfal | any one have any advice on pausing an agent after every call. I really only need this to take action on specific queues. |
| 00:37.06 | jaytee | there's a setting for giving the agent time to "cleanup" in the queues.conf file |
| 00:37.12 | Knightfal | ya |
| 00:37.15 | Knightfal | wrap |
| 00:37.45 | Knightfal | I like it but I have a specific need to actually throw them into a trackable state |
| 00:38.11 | jaytee | hmm, not sure how you could do that |
| 00:38.14 | Knightfal | without them having to use our ui |
| 00:38.38 | jblack | Has anyone seen "WARNING[19808]: channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to...." with iax2 on 1.6.2 ? |
| 00:47.45 | *** join/#asterisk chendy (n=chatzill@58.251.103.35) |
| 00:49.27 | jblack | Oh, looks like an open bug. |
| 00:54.10 | Katty | hugs jblack |
| 00:54.13 | Katty | how're you tonight dear |
| 00:57.34 | jblack | hey there |
| 00:57.49 | jblack | You first. |
| 00:58.20 | jblack | Katty: Hi. :) |
| 00:59.28 | Katty | oh |
| 00:59.29 | Katty | why |
| 00:59.32 | Katty | i have a google voice invite |
| 00:59.47 | jblack | Oh, cool. |
| 00:59.52 | jblack | I could use one. |
| 01:00.01 | Katty | someone sent me an invitation |
| 01:00.06 | Katty | i will see if i have any to send out tho |
| 01:00.08 | jblack | Because my life is a mess, and I want to hear your good news more than I want to tell you my bad news. |
| 01:03.44 | Katty | 5732006004 |
| 01:03.55 | Katty | is my google voice number. |
| 01:04.08 | Katty | this is neat. |
| 01:05.02 | jblack | need a test? |
| 01:05.46 | Katty | sure if you want |
| 01:06.10 | Katty | hmm. i don't have a mic. hopefully it rings the mobiel |
| 01:06.12 | Katty | mobile. |
| 01:07.30 | Katty | also, does anyone know David McNett is? |
| 01:07.37 | jblack | there you go. ring ring |
| 01:07.50 | jblack | heh |
| 01:08.34 | jblack | yeah. I think verizon has been shaping as of late. |
| 01:08.39 | Katty | kinda staticy )= |
| 01:08.47 | jblack | that's almost certainly this side. |
| 01:09.11 | jblack | eitehr that, or the provider I used. I suspect voipstreet sucks horridly bad. |
| 01:09.28 | Katty | who knows. it's amazing that it works tho! |
| 01:09.34 | Katty | happy dances |
| 01:10.16 | KavanS | doh.. |
| 01:10.20 | KavanS | verizon is shaping? |
| 01:10.33 | Katty | neat. you can record all incoming calls if you want. |
| 01:10.47 | jblack | nope. it's voipstreet. they suck _that_ bad. |
| 01:11.15 | jblack | I need to convince t-mobile to give congestion on no-answer/reject |
| 01:11.26 | jblack | They're dropping into their voicemail system. |
| 01:11.54 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 01:11.59 | jblack | maybe I could convince them to forward to a non-existant number. that might congest. |
| 01:12.30 | Katty | http://imagebin.org/68104 |
| 01:12.47 | jblack | It's too small to read |
| 01:12.53 | jblack | nm. I'm a dork |
| 01:15.12 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 01:17.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:21.07 | jblack | Katty: So, how are you doing? |
| 01:30.31 | *** join/#asterisk sahafeez (n=sahafeez@65.119.47.100) |
| 01:31.43 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.103) |
| 01:34.44 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 01:41.07 | Sandheaver | is watching DJ Hero videos on youtube. WANT |
| 01:50.40 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 01:50.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:55.42 | CcRnp | CEL WORKS PERFECT AND KEEPS TRACK OF EVERY CALL !!!!!! |
| 02:00.19 | *** join/#asterisk dmz (n=dmz@64.203.207.101) |
| 02:03.36 | *** join/#asterisk perlmonkee (n=perlmonk@anthony.kepler.net) |
| 02:03.54 | *** join/#asterisk dssman (n=no@174.112.35.38) |
| 02:05.17 | perlmonkee | I've had the same asterisk system up and running for several years now - I have a great many devices registered with it from a variety of locations - all behind NAT. I'm trying to get connected with the "sjphone" soft phone from a hotel in thailand and I can only RECEIVE calls... I can't make them - any ideas? |
| 02:08.52 | *** join/#asterisk dkirker (n=dkirker@99.2.134.30) |
| 02:14.53 | KavanS | perlmonkee, not sure... |
| 02:15.05 | KavanS | perlmonkee, iphone? |
| 02:17.10 | *** join/#asterisk Carlos_Tico (n=grillo_v@98.201.162.34) |
| 02:31.19 | dssman | is there an arguemnt that I can use in my extensions.conf file that will use the name of the client initiating the call?? For example, I want when a user dials 1000 that it takes them into their voicemail box (or password prompt). I can use the callerid(num) one, but my caller IDs are set by extension, so if I call from 2001 my actual CID is "JRS Consulting" 1905695xxxx |
| 02:45.57 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:48.10 | TJNII | perlmonkee: No audio issues? |
| 02:49.14 | TJNII | perlmonkee: And the phone registers OK? |
| 02:51.18 | *** join/#asterisk psilikon (n=psilikon@221-116.96-97.tampabay.res.rr.com) |
| 02:54.27 | psilikon | I'm trying to understand * + voicemail + IMAP. Is it possible to just have a voice mail message sent to a gmail account with only configuring an IMAP client on the * box? |
| 02:56.32 | TJNII | You'd need a script to do that, I think. |
| 02:57.20 | TJNII | AFAIK the * voicemail app can call a script or sendmail/sendmail clone. |
| 02:57.33 | jblack | I just got whomped with $15 of calls from callcentric. Tonight. over the space of 2 hours. |
| 02:57.42 | TJNII | Nice. |
| 02:57.45 | TJNII | Hacked? |
| 02:57.52 | jblack | Nope. |
| 02:57.55 | psilikon | Ok. The whole concept is fuzzy to me right now. Initially I was under the impression you could just have voicemail message sent as an attach to any email address; that is not the case right? |
| 02:58.15 | TJNII | psilikon: It uses smtp to email the message to the address. |
| 02:58.21 | psilikon | jblack, I use callcentric too. They're not bad. |
| 02:58.23 | jblack | I'm playing let's short-shrift the cell phone company with fav-5 games, and I think the 1.6 * server here ended up with a few open sockets. |
| 02:58.39 | jblack | psilikon: I'm actually really impressed, yeah. |
| 02:58.45 | TJNII | psilikon: It doesn't handle smtp natively, it calls sendmail. |
| 02:58.46 | jblack | I'm seeing latency of 2 seconds. That's incredible. |
| 02:59.09 | jblack | pardon, 2 ms. |
| 02:59.29 | TJNII | Okay, the other side of incredible. |
| 02:59.33 | jblack | I wouldnt mind getting iax from them, but <shrug> |
| 02:59.39 | psilikon | TJNII, so what is all this IMAP stuff I am reading out of TFOT? I skimmed over it but I am pretty sure it is telling me to setup and imap client _and_ server? |
| 03:00.18 | TJNII | psilikon: Not sure. I don't remember reading about IMAP in TFOT, but I haven't read it in a couple years. |
| 03:00.20 | TJNII | Link? |
| 03:00.22 | jblack | My bet is that 1.6 blocked up here, leaving a handful of sockets from test calls open, and they just let the clock run on about a dozen calls for a couple hours. :( |
| 03:00.42 | TJNII | Nice. |
| 03:00.43 | jblack | nothing like debugging nat. |
| 03:07.26 | jblack | btw, voipstreet is awful. skip them. |
| 03:08.46 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:09.57 | dssman | jblack... they dont auto drop lines if there is no voice data? |
| 03:11.46 | psilikon | any guides for implementing sendmail? |
| 03:12.00 | *** join/#asterisk effigee (n=jenshens@S0106001d7e52bc10.vc.shawcable.net) |
| 03:12.05 | dssman | yea lots |
| 03:12.06 | dssman | google it |
| 03:12.22 | jblack | dssman: Apparently not |
| 03:12.52 | dssman | problem I am having is it doesnt support ssmtp so I cannot use my mail server as it requires user name and pwd... I wanna install sendmail clone ssmpt which does it |
| 03:13.10 | dssman | jblack: eek... shitty, I guess you cant win either way |
| 03:13.40 | dssman | I know my telco disconnects me after 30 sec of silence, so I took the bite and disabled silence supression |
| 03:14.22 | dssman | higher bandwidth, but if I ever drop a call and the chan is open (for whatever reason) it will auto disconnect |
| 03:18.10 | jblack | Yeah, I'll give 'em the 15 bucks. |
| 03:18.35 | jblack | I emailed 'em all the asme to check thigns out on that side. a one time event is fine, but a repeat event would be bad. |
| 03:27.01 | dssman | oh well |
| 03:27.08 | dssman | u running a box for home or work? |
| 03:29.04 | KavanS | jblack, try voicepulse or vitelity |
| 03:29.15 | KavanS | they seem to be decent |
| 03:31.56 | p3nguin | jblack: Interex just had to ditch CallCentric because they can't keep quality service for his phone calls. |
| 03:32.15 | *** join/#asterisk came0 (n=came0@93.147.121.70.cfl.res.rr.com) |
| 03:32.24 | jblack | huh. THey seem excellent here. |
| 03:32.29 | p3nguin | jblack: He uses freeswitch instead of asterisk, though. |
| 03:33.07 | p3nguin | Apparently the two don't play well together. |
| 03:33.12 | *** part/#asterisk came0 (n=came0@93.147.121.70.cfl.res.rr.com) |
| 03:34.02 | p3nguin | After paying the $25 to port his number to them, he was pretty unhappy having to pay another $25 to port to flowroute. |
| 03:34.23 | p3nguin | Why does it cost so much for number porting? |
| 03:34.42 | jblack | because people want it. |
| 03:37.55 | jblack | Do you two do gtalk? |
| 03:38.03 | p3nguin | When you were saying your latency is 2 ms, were you using ping time or some tool from within asterisk to calculate it? |
| 03:38.13 | jblack | qualify in asterisk |
| 03:38.31 | jblack | callcentric/17772550003 204.11.192.23 5080 OK (2 ms) |
| 03:38.51 | jblack | I'm sure linode and callcentric are using the same hosting facility. |
| 03:39.06 | p3nguin | Is gtalk the google xmpp thing? |
| 03:39.16 | TJNII | doesn't even have 2ms to the phone in the next room.... |
| 03:39.26 | jblack | gtalk is AIM for google. |
| 03:39.49 | jblack | TJNII: That's why I gave 'em the 15 bucks. that's just. wow |
| 03:40.21 | p3nguin | I think I may have signed in once when trying to figure out how to login to my own jabber using pidgin, but other than that, no, I don't use the service. |
| 03:40.21 | jblack | from here, on the other end of dsl, I see about 40-50ms. |
| 03:40.36 | jblack | Oh, ok. |
| 03:40.52 | jblack | I have an android phone these days, so I've got google-everything. |
| 03:41.02 | TJNII | I'm getting between 110-140ms to my off-site sip accounts. |
| 03:41.12 | jblack | I think google even knows my underwear size by now. |
| 03:41.51 | TJNII | Google's targeted ads frignten me. It advertized digi-key to me on a work machine, on my second day. How did it know! How! |
| 03:42.16 | jblack | if you guys ever want to hit me up on one of their things, I'm james.d.blackwell@gmail.com |
| 03:42.21 | jblack | tjnii: Heh. |
| 03:42.25 | p3nguin | I guess I need to enable qualify on my peers so I can check their times, too. |
| 03:42.41 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) |
| 03:46.05 | p3nguin | ipcomms 38 ms, callcentric 66 ms, and voipms 60 ms. |
| 03:53.26 | *** join/#asterisk Nukemizer (n=Nukemize@15.249.sfcn.org) |
| 03:54.24 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 03:54.33 | Nukemizer | Is there a way to format the information from Asterisk out a T1 card to a legacy PBX ? |
| 03:54.35 | Nukemizer | Clear channel e&M Wink is the circuit type |
| 03:54.37 | Nukemizer | I need to send the following format *ANI*DNIS* out the T1 to the legacy PBX |
| 03:54.39 | Nukemizer | The PBX currently rejects the call unless I accept the call on the PBX with NO caller ID. |
| 03:55.45 | p3nguin | What the crap? I have over 100 ms between * and a phone on the same LAN, but only 31 ms between * and a peer across the internet. How is that even possible? |
| 03:56.12 | KavanS | christ |
| 03:56.15 | KavanS | wtf is call centric? |
| 03:56.21 | p3nguin | an ITSP |
| 03:56.27 | p3nguin | callcentric.com |
| 03:56.56 | KavanS | going to ping test on that... |
| 03:56.59 | KavanS | anyone else use them? |
| 03:57.34 | KavanS | hrm, 108ms to that ip jblack...so slow for me ;( |
| 03:57.51 | jblack | to which ip? |
| 03:57.51 | KavanS | voicepulse gives me 35ms |
| 03:58.03 | KavanS | 204.11.192.23 |
| 03:58.05 | KavanS | you got another host? |
| 03:58.15 | p3nguin | He did mention that his asterisk was likely in the same facility as callcentric's peer. |
| 03:58.47 | jblack | ..23, that' the one. |
| 03:58.58 | TJNII | I think you connection plays a huge part. I use Qwest DSL and I've gotten a little over 100ms to my outside sip devices across 3 moves in 2 states. |
| 03:59.08 | jblack | from home, I get about 55ms or so. |
| 03:59.19 | KavanS | ahh yeah |
| 03:59.40 | jblack | I'm using the new york linode facility, and that's about 55ms too. |
| 04:02.19 | p3nguin | Looks like all of their machines are 204.11.192.x, so I guess they don't have other locations for better times. |
| 04:02.43 | KavanS | doh... |
| 04:02.56 | KavanS | yeah using SIP + 1 analog for now...might get a T1 down the road and ditch these voip providers |
| 04:03.01 | jblack | who? |
| 04:03.11 | KavanS | using voicepulse and vitelity for SIP |
| 04:04.50 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 04:08.53 | *** join/#asterisk friartuck (n=pmccary@66.162.90.57) |
| 04:14.46 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 05:21.03 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-26-97.dsl.lsan03.sbcglobal.net) |
| 05:24.24 | *** join/#asterisk esaym153 (n=esaym153@cpe-24-174-176-203.satx.res.rr.com) |
| 05:24.33 | esaym153 | what is a good iax softphone? |
| 05:24.44 | esaym153 | that works in linux |
| 05:27.51 | mchou | Zoiper |
| 05:27.52 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 05:28.11 | mchou | anyone use voip.ms? |
| 05:30.17 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) [NETSPLIT VICTIM] |
| 05:37.52 | *** join/#asterisk ltd (n=z@202.55.149.246) |
| 05:44.45 | *** join/#asterisk xpot (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 05:49.13 | esaym153 | hmm |
| 05:49.49 | ChannelZ | things that make you go |
| 05:56.07 | p3nguin | mchou: I use it. |
| 05:56.21 | p3nguin | mchou: Did you have a question, or just taking a poll? |
| 05:56.52 | mchou | p3nguin: on inbound calls, does voip.ms do a proper cname lookup? |
| 05:57.20 | mchou | in other words, do you actually get the name of the caller? |
| 05:57.35 | mchou | or just a location? |
| 05:57.39 | p3nguin | It does a lookup and does provide the name... but I cannot attest to the accuracy of it. |
| 05:57.48 | mchou | or even worse, just the phone # |
| 05:58.03 | p3nguin | Did you turn on queries? |
| 05:58.09 | mchou | hmm?? |
| 05:58.18 | p3nguin | You have to enable cnam lookups on your DIDs. |
| 05:58.24 | mchou | no, I dont subscribe yet |
| 05:58.32 | p3nguin | They charge you per lookup, so you have to enable it. |
| 05:58.33 | p3nguin | oh. |
| 05:58.45 | mchou | i justed wanted to know if that was even an option |
| 05:59.03 | p3nguin | Oh, yeah. They do offer CNAM lookups. |
| 05:59.31 | p3nguin | I only pay them for lookups on my toll-free DID. On my regular DID with another provider, I do the lookups myself. |
| 05:59.58 | mchou | p3nguin: lookup yourself with what DB? |
| 06:00.26 | p3nguin | The script I use checks with three different online query services. |
| 06:00.37 | mchou | bah, I've seen that |
| 06:01.06 | mchou | are those sources even decent? |
| 06:01.32 | p3nguin | The first two sources almost always fail, with success on the third. |
| 06:03.10 | mchou | p3nguin: can you actually tell the voice quality difference betw. "premium" and "value" in a blind test? :) |
| 06:03.39 | p3nguin | nope |
| 06:04.17 | p3nguin | I haven't been able to tell any difference using ulaw on SIP or IAX2, so I use the value route all the time. |
| 06:04.42 | mchou | so no anomalies like dropped calls, etc? |
| 06:06.11 | mchou | so I need another favor.... what does their CDR reports look like? If you could pastebin a few lines (you can blank the phone numbers) I'd appreciate it |
| 06:07.46 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 06:10.45 | *** join/#asterisk lozarythmic (n=lpraties@e1-1.ns500-1.ts.milt.as9105.net) |
| 06:11.37 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 06:22.49 | p3nguin | mchou: http://imagebin.org/68118 |
| 06:24.09 | mchou | p3nguin: thanks |
| 06:25.16 | p3nguin | The printable CDR looks the same, but without all the HTML. |
| 06:25.25 | mchou | p3nguin: that's cool |
| 06:26.39 | mchou | p3nguin: you happen to know if you can register multiple SIP devices simultaneously to voip.ms using the same credentials? |
| 06:27.03 | p3nguin | I wouldn't know why you would want to. |
| 06:27.18 | p3nguin | Create some sub accounts instead. |
| 06:27.27 | mchou | p3nguin: cause it's easier on the end user |
| 06:27.44 | p3nguin | I don't see how. |
| 06:28.26 | mchou | ring multiple locations simultaneously w/o hassle of remember different credentials and whatnot |
| 06:29.25 | p3nguin | Why not configure a single Asterisk server where the DID comes in, then route calls to the various SIP devices from there? |
| 06:29.38 | mchou | lol |
| 06:29.59 | mchou | cause the end user doesnt want to maintain/run asterisk? |
| 06:30.08 | p3nguin | That's why they don't need to. |
| 06:30.23 | p3nguin | You maintain the server. They connect to your server. |
| 06:30.32 | mchou | what? |
| 06:31.06 | p3nguin | The end users want to have phones (softphones, even). Right? |
| 06:32.07 | p3nguin | Either give each person a sub account and allow him to connect directly to VoIP.ms, or put an Asterisk server somewhere and let each person connect to it instead. |
| 06:32.10 | mchou | if j6p wants an ITSP acct (say one one number for now), why the hell would he want to run asterisk or have differnt credentials for multiple devices? |
| 06:33.14 | mchou | say he has ATA at home, and is on the go with lappy softphone/or iphone-sip.... |
| 06:33.42 | mchou | wth would he NEED 3 different credentials? |
| 06:34.10 | p3nguin | because he has three devices, obviously. |
| 06:34.18 | mchou | just because * cant do it doesnt mean it makes sense |
| 06:34.46 | p3nguin | Just because you think it doesn't make sense doesn't mean it actually doesn't make sense. |
| 06:34.47 | drmessano | Theres a provider that allows multiple endpoints registered to one account? |
| 06:34.55 | mchou | yup |
| 06:35.07 | mchou | in fact they're free |
| 06:35.11 | mchou | ekiga |
| 06:35.18 | mchou | try it |
| 06:35.31 | drmessano | I think that's a violation of the RFC |
| 06:35.44 | mchou | what? |
| 06:35.46 | p3nguin | among other things. |
| 06:35.50 | drmessano | Endpoints should all be using their own credentials |
| 06:36.05 | p3nguin | That's what I tried to say. |
| 06:36.07 | mchou | OK, cite me chapter and verse :) |
| 06:36.32 | drmessano | How are we negotiating the call again? |
| 06:36.42 | mchou | drmessano: hmm? |
| 06:36.55 | drmessano | Its a simple question |
| 06:37.01 | mchou | drmessano: I dont understand your question |
| 06:37.01 | drmessano | How are we negotiating the call? |
| 06:37.38 | drmessano | Explain to me the process in how a SIP call is negotiated with multiple endpoints registered to the same peer entity |
| 06:39.41 | p3nguin | It certainly wouldn't make all the devices ring when the DID is called. |
| 06:39.52 | mchou | lol |
| 06:39.58 | mchou | you guys are rich |
| 06:40.06 | mchou | register with ekiga |
| 06:40.08 | p3nguin | At best, it might allow the last one registered to receive the call. |
| 06:40.12 | mchou | it works |
| 06:40.18 | mchou | all 3 devices ring |
| 06:40.41 | p3nguin | If you're so confident, why the fuck are you bothering us about it? |
| 06:40.48 | mchou | lol |
| 06:41.02 | mchou | p3nguin: why you getting so testy? |
| 06:41.04 | p3nguin | I mean, actually... why? |
| 06:41.34 | mchou | p3nguin: I just explained why from and end user perspective why it makes sense |
| 06:41.45 | p3nguin | That's not what I asked you. |
| 06:42.00 | drmessano | End users shouldnt be providing endpoint credentials anyway |
| 06:42.26 | drmessano | SIP registration should be transparent to them. Configured by an outside mechanism |
| 06:43.01 | mchou | drmessano: you know any softphones that do that? :) |
| 06:43.26 | drmessano | I think Bria does |
| 06:44.54 | p3nguin | I still don't understand why you're bothering us about how to manage SIP registrations if you're already confident in your own procedures. |
| 06:45.25 | mchou | p3nguin: I asked you whether voip.ms was capable of it |
| 06:45.31 | p3nguin | It's not. |
| 06:45.38 | p3nguin | And no other ITSP is, either. |
| 06:45.44 | mchou | p3nguin: and you go non-linear on a simple question |
| 06:45.46 | drmessano | p3nguin: He said "f**k you" |
| 06:45.47 | p3nguin | That's not how SIP works. |
| 06:46.01 | drmessano | (paraphrasing) |
| 06:46.13 | drmessano | I wouldnt take that from him |
| 06:46.15 | p3nguin | :/ |
| 06:46.34 | drmessano | Cut him off at the knees, you know? |
| 06:46.43 | mchou | p3nguin: last I checked ekiga is an ITSP |
| 06:52.13 | p3nguin | Can you use your ekiga account to call my phone number, which does not contain an @ symbol nor a domain name? Can I pick up my phone and call your ekiga address? |
| 06:53.13 | mchou | drmessano: http://queue.acm.org/detail.cfm?id=1229909 |
| 06:53.50 | mchou | drmessano: in the "early media" paragraph |
| 06:54.59 | mchou | drmessano: tey use a technique called parallel forking |
| 06:55.00 | drmessano | What the fuck does that page have to do with the RFC? |
| 06:55.23 | mchou | drmessano: a SIP proxy isn't conformant to RFC? |
| 06:55.27 | drmessano | I can implement and document any extensions to any protocol, doesnt mean it follows the RFC |
| 06:56.08 | drmessano | Why does a SIP proxy have to conform to the RFC? |
| 06:56.13 | drmessano | Asterisk doesnt |
| 06:56.16 | drmessano | M$ doesnt |
| 06:56.26 | *** join/#asterisk dkirker (n=dkirker@99.2.134.30) |
| 06:56.27 | mchou | "Forking is a primary source of power in SIP's rendezvous function. It also is the source of the greatest complexity in the protocol." |
| 06:56.28 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 06:56.42 | mchou | perhaps that's why MS and asterisk doesnt do it |
| 06:56.54 | drmessano | Again, reading me from a page that could be chocolate chip cookie recipies |
| 06:57.16 | drmessano | Theres nothing official or authoritative about the link you posted |
| 06:57.19 | mchou | drmessano: I did ask to to cite me chapter and verse from RFC |
| 06:57.42 | drmessano | mchou: No, do your own research.. you brought her, you fuck her |
| 06:57.47 | drmessano | Simple as that |
| 06:57.59 | mchou | lol |
| 06:59.02 | drmessano | Instead you link some iffy documentation from what could be the SIP implementation in Bonzi Buddy, for all I know |
| 06:59.23 | drmessano | Whats the matter, couldnt find a relevent and highly questionable wikipedia link? |
| 06:59.54 | p3nguin | wikipedia |
| 06:59.55 | p3nguin | lol |
| 07:00.20 | mchou | lol |
| 07:00.32 | p3nguin | mchou: Can you use your ekiga account to call my phone number, which does not contain an @ symbol nor a domain name? Can I pick up my phone and call your ekiga SIP address? |
| 07:00.38 | tzafrir_laptop | mchou, Asterisk isn't exactly a SIP proxy |
| 07:00.42 | mchou | I ask you for proof and you use a not so artful dodge |
| 07:00.49 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 07:01.07 | mchou | p3nguin: yes |
| 07:01.08 | p3nguin | As if the burden of proof is on him or something? |
| 07:01.26 | drmessano | It wasnt a dodge.. the burden of proof is on you, and if you're going to be an evasive douche about it, then figure it out yourself |
| 07:01.30 | p3nguin | How will I dial your ekiga SIP address from my phone? |
| 07:01.41 | mchou | p3nguin: I dont have any outbound $ with ekiga though atm |
| 07:02.16 | mchou | myusername@ekiga.net (or my DID) |
| 07:02.39 | drmessano | s/I dont have any outbound \$ with ekiga though atm/I dont have a let to stand on\, atm/ |
| 07:02.54 | p3nguin | I tried entering your SIP address, but I don't have letters nor an @ symbol on my phone. |
| 07:02.54 | mchou | p3nguin: just like virtually any ITSP |
| 07:03.25 | mchou | p3nguin: are you being daft? I also said my DID |
| 07:03.50 | drmessano | No, he's being obtuse.. there is a clear difference |
| 07:04.05 | drmessano | Roget's FAIL |
| 07:04.40 | p3nguin | I can't see where ekiga provides DIDs. |
| 07:05.32 | drmessano | or a business model? |
| 07:09.42 | ChannelZ | ...and, scene. |
| 07:11.41 | mchou | p3nguin: you can get an DID with ANY ITSP |
| 07:12.02 | p3nguin | Show me where ekiga provides DIDs. |
| 07:12.24 | mchou | p3nguin: wow. Your abe being intentionally obtuse |
| 07:12.30 | mchou | are* |
| 07:12.48 | mchou | either that or you are a total dunce |
| 07:13.11 | p3nguin | I looked. I failed to locate. You claim they have it; you provide me the link to it. |
| 07:13.38 | p3nguin | I could be a dunce. Does that change anything? |
| 07:13.45 | mchou | income call => (ANY ITSP DID) =>via sip=>ekiga.net=>devices |
| 07:13.53 | mchou | incoming* |
| 07:14.06 | mchou | which part dont you grok? |
| 07:14.31 | mchou | p3nguin: http://ubuntuforums.org/archive/index.php/t-295007.html |
| 07:14.44 | p3nguin | I'm not asking about ubuntu. |
| 07:14.48 | mchou | lol |
| 07:15.03 | p3nguin | I want to know where does ekiga provide DIDs. |
| 07:15.07 | perlmonkee | mchou: if anyone doubted your toolishness before... your ham handed attempt at using "grok" in converrsation just sealed the deal. |
| 07:15.08 | mchou | just because it's on an ubuntu site doesnt mean it's about ubuntu |
| 07:15.30 | mchou | p3nguin: keep playing dodgeball |
| 07:15.51 | perlmonkee | mchou: nobody likes you or thinks you're pretty or will ask you to prom. |
| 07:16.03 | p3nguin | I still can't figure out how to dial that damned @ symbol to call you on ekiga. |
| 07:16.28 | mchou | p3nguin: I said about 3x alread. _DID_ |
| 07:16.33 | mchou | already* |
| 07:16.45 | mchou | perlmonkee: I way too sexy for you! |
| 07:16.52 | mchou | I'm* |
| 07:16.54 | p3nguin | And I'm still waiting for you to show me where ekiga has DIDs. |
| 07:17.14 | mchou | p3nguin: you're beoyond help. |
| 07:17.25 | mchou | p3nguin: [00:13:44] <mchou> income call => (ANY ITSP DID) =>via sip=>ekiga.net=>devices |
| 07:17.26 | p3nguin | You're deflecting. |
| 07:17.40 | mchou | I'm not deflecting jack |
| 07:18.05 | mchou | p3nguin: you're saying my dial flow wont work? |
| 07:19.03 | mchou | especially WITHOUT @ sign? |
| 07:19.09 | mchou | lol |
| 07:19.47 | perlmonkee | mchou: the point being made is that you seemed to be claiming you had a DID "through ekiga" |
| 07:19.49 | perlmonkee | you don't |
| 07:19.50 | perlmonkee | you can't |
| 07:19.54 | p3nguin | What I said was I cannot pick up my phone and dial your ekiga SIP address because I cannot dial words and symbols. You've made it into something much larger. |
| 07:20.03 | perlmonkee | you MIGHT have a DID being forwarded through a 3rd party |
| 07:20.42 | p3nguin | I would even be interested in how you forward a DID to an ekiga SIP address. |
| 07:20.46 | perlmonkee | which is what you eventually claimed, while failing to include anything along the lines of "what I meant to say was..." |
| 07:20.50 | p3nguin | If you want to disclose it, that is. |
| 07:21.17 | *** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
| 07:21.33 | perlmonkee | p3nguin: sipphone/gizmo will allow you to forward all calls to your DID to another SIP address - for one example. |
| 07:21.39 | mchou | p3nguin: http://wiki.ekiga.org/index.php/Calling |
| 07:21.58 | mchou | p3nguin: seach for "diamondcard" on that page |
| 07:22.30 | mchou | p3nguin: now STFU |
| 07:22.35 | *** join/#asterisk Tim_Toady (n=moi@adsl45-72.kln.forthnet.gr) |
| 07:23.06 | perlmonkee | mchou: did you just instruct someone to shut up in the place you came seeking an answer to a question? |
| 07:24.56 | mchou | perlmonkee: and your point is? if p3nguin wants to play dodgeball, despite my repeated attempts to provide links, it doesnt mean I need to engage him |
| 07:25.20 | mchou | perlmonkee: if he still is in deep denial |
| 07:25.38 | p3nguin | Okay, so I see how you can use ekiga for "PC-To-Phone calls at very low rates all over the world." So what? |
| 07:26.24 | mchou | p3nguin: skip down 2 paragraphs |
| 07:26.43 | mchou | "You can actually use any H.323 or SIP ITSP provider, including your own PBX at work. However we recommend using the integrated provider." |
| 07:27.22 | p3nguin | Okay, that's good to know. Now I can call you from my phone while you are on ekiga. |
| 07:27.37 | p3nguin | You have advanced past step 1. |
| 07:27.50 | ChannelZ | wonders why anyone would want to call him |
| 07:28.07 | mchou | p3nguin: you're in "special" needs at this rate |
| 07:28.08 | p3nguin | *shrug* |
| 07:28.44 | mchou | it's liek I'm on step 205 while you're on step 0.5 |
| 07:28.47 | mchou | like* |
| 07:28.47 | p3nguin | Does it make you feel better insulting someone who you consider to be retarded? |
| 07:29.04 | mchou | no, it brings me no pleasure |
| 07:29.19 | mchou | I acknowledge your help earlier. |
| 07:29.50 | mchou | I then ask about simul regs using multiple endpoints and you go off in the deep end |
| 07:30.20 | p3nguin | No, you've confused me with yourself. |
| 07:30.48 | mchou | not only that, instead of think and reflecting on what I posted earlier (like the ubuntu link), you persisted in being obtuse |
| 07:30.54 | p3nguin | I politely said that it didn't make sense to do that. |
| 07:31.26 | mchou | p3nguin: and I politely disagreed, citing a use case |
| 07:32.17 | mchou | p3nguin: not only that I cited someplace where such a thing worked (and is descibe by someone at acm.org |
| 07:32.24 | mchou | described* |
| 07:33.02 | mchou | that should have tipped you off I'm not a wacko |
| 07:33.04 | p3nguin | At which point I questioned your motive for asking for help when you were confident that your method was the best way. |
| 07:34.13 | mchou | umm, I started off by asking about features about an ITSP where I wasnt subscribed |
| 07:34.33 | mchou | I didnt mention anything about "method" |
| 07:35.10 | mchou | I just asked whether that ITSP was capable of reg'ing multiple endpoints w/ same credentials |
| 07:35.14 | p3nguin | To my knowledge, ITSPs are not going to want you registering lots of devices to a single (probably main) account. |
| 07:36.02 | mchou | at which point you went on a non sequitur crusade |
| 07:36.37 | mchou | a simple "yes" or "no" would have sufficed |
| 07:36.48 | p3nguin | sigh |
| 07:36.55 | mchou | or enven "I dont know" |
| 07:36.59 | mchou | even* |
| 07:37.38 | p3nguin | Is it possible? Maybe. Will it work (correctly)? Doubtful. Do I know? No, because I do not recommend nor utilize such behavior with regard to SIP devices. |
| 07:37.54 | mchou | p3nguin: that's fine by me |
| 07:38.33 | mchou | p3nguin: we'll just have to agree to disagree on multiple registrations |
| 07:38.37 | p3nguin | In my experience, the last registered SIP device is the one that gets the call. |
| 07:39.37 | mchou | p3nguin: I agree that's certainly the case with asterisk |
| 07:39.48 | p3nguin | Maybe it's a limitation of the PBX software, maybe it's configuration. |
| 07:39.54 | mchou | nope |
| 07:40.03 | mchou | it's a limitation with * |
| 07:40.04 | p3nguin | VoIP.ms uses Asterisk and OpenSER. |
| 07:40.28 | mchou | afaik openser allows multiple regs |
| 07:40.41 | mchou | at least if iirc |
| 07:43.28 | mchou | http://www.voip-info.org/wiki/view/Asterisk+at+large |
| 07:43.39 | mchou | "Multiple registrations: |
| 07:43.39 | mchou | Asterisk doesn't support multiple registrations. A registrar proxy |
| 07:43.39 | mchou | will be able to do this, allowing such things as forking (sequential / |
| 07:43.39 | mchou | parrellel / combinations of both). |
| 07:53.55 | *** join/#asterisk Ashura (n=ashura@89.119.206.194) |
| 08:13.32 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:26.16 | mchou | drmessano: youre probaly no longer awake.... |
| 08:26.23 | mchou | http://www.faqs.org/rfcs/rfc3428.html |
| 08:26.45 | mchou | 6. Proxy Processing |
| 08:26.45 | mchou | <PROTECTED> |
| 08:26.45 | mchou | <PROTECTED> |
| 08:26.45 | mchou | <PROTECTED> |
| 08:26.45 | mchou | <PROTECTED> |
| 08:26.47 | mchou | <PROTECTED> |
| 08:26.49 | mchou | <PROTECTED> |
| 08:26.51 | mchou | <PROTECTED> |
| 08:27.32 | mchou | drmessano: that RFC is co-authored by H. Schulzrinne himself |
| 08:28.59 | mchou | granted it is _MAY_, but it is in a standards track RFC |
| 08:30.01 | mchou | drmessano: heh, a co-author on that RFC worked at M$ |
| 08:47.13 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
| 09:07.08 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 09:10.22 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 09:10.44 | FlaPer87 | have any of you configured sms through asterisk? |
| 09:25.53 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:51.19 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:d1d5:dc5:fbed:55fc) |
| 10:10.29 | *** join/#asterisk _trine (n=psybnc_a@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 10:13.12 | _trine | is it possible to receive calls to asterisk if it's connected to the internet via a wifi link from another distant router; in this situation this would mean that the asterisk would not have a direct 'outside' WAN IP address |
| 10:14.04 | _trine | rather it would have a lan IP address given to it with DHCP from the distant router |
| 10:29.24 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 10:41.43 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 10:50.08 | *** join/#asterisk Ad-Hoc (n=nimbus@193.92.197.35) |
| 10:55.30 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 11:03.21 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 11:28.20 | *** join/#asterisk cuco (n=cuco@bzq-82-81-32-128.red.bezeqint.net) |
| 11:48.27 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 11:48.34 | phix | hey, werd up |
| 11:48.41 | phix | quick question |
| 11:48.51 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 11:49.04 | phix | I want to dial a number, say something via festival or a pre-recorded file, then hang up |
| 11:49.13 | phix | now go! :) solution! :) |
| 12:15.26 | wonderworld | phix: create a call file. move it to the spool directory. pass a number as variable with it. send it into an extension. call the number with that extension. play the pre-recorded-file. hang up. |
| 12:16.19 | *** join/#asterisk tamiel (n=tamiel@abo-212-152-68.bdx.modulonet.fr) |
| 12:50.29 | *** join/#asterisk magic_1 (n=magic@unaffiliated/magic1/x-836121) |
| 12:51.06 | magic_1 | hi guys |
| 12:51.46 | magic_1 | can anyone help or point me in the right direction, want to get a push to talk server going with asterisk |
| 12:52.26 | magic_1 | all help is greatly appreciated |
| 13:02.34 | *** join/#asterisk retentiveboy (n=pdugas@74.95.28.37) |
| 13:16.11 | *** part/#asterisk esaym153 (n=esaym153@cpe-24-174-176-203.satx.res.rr.com) |
| 13:16.18 | *** join/#asterisk bbt (n=Sam@180.189.138.90) |
| 13:22.20 | *** join/#asterisk Jumpie (i=n3rdz@c-76-100-241-4.hsd1.md.comcast.net) |
| 13:22.26 | Jumpie | ugh this authentication crap is annoying |
| 13:22.28 | Jumpie | lol |
| 13:26.58 | *** join/#asterisk freckle (n=chatzill@195.74.96.123) |
| 13:27.35 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:27.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:32.05 | *** join/#asterisk garymc (n=garymc@host86-159-106-40.range86-159.btcentralplus.com) |
| 13:35.57 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:02.03 | *** join/#asterisk retentiveboy (n=pdugas@74.95.28.37) |
| 14:07.41 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
| 14:17.41 | cusco | hello |
| 14:17.48 | cusco | Kobaz: are you arround? |
| 14:20.28 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 14:21.59 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 14:24.49 | FlaPer87 | have any of you configured sms through asterisk? |
| 14:25.01 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 14:29.32 | cusco | hi |
| 14:29.46 | cusco | I am having troubles enabling 2nd span |
| 14:30.00 | cusco | dahdi_scan shows the second span: alarms=RED |
| 14:30.02 | cusco | why? |
| 14:39.50 | *** join/#asterisk RypPn (i=TuMbL@188.220.36.226) |
| 14:46.22 | FlaPer87 | does anyone here uses asterisk with messagenet for recieving and making calls? |
| 14:47.21 | p3nguin | phix: exten => 2200,1,Festival('stuff that festival says') |
| 14:48.57 | p3nguin | phix: I'm sure you can manage the hangup line on your own without my typing it out for you. |
| 14:51.15 | *** join/#asterisk dongs (i=1000@l212168.ppp.asahi-net.or.jp) |
| 14:51.24 | dongs | um hi |
| 14:51.33 | dongs | asterisk drops my calls after exactly 4 minutes 20 seconds. |
| 14:51.39 | dongs | i wish i was joking. |
| 14:51.40 | dongs | but wtf? |
| 14:51.53 | p3nguin | lol |
| 14:52.05 | dongs | is there some log or something i can see for this. |
| 14:52.37 | p3nguin | Show us your configs and debug of a call. |
| 14:53.46 | FlaPer87 | I'm having some problems configuring my messageneet account. If i configure it for inbound and outbound calls it doesn't work |
| 14:53.50 | dongs | how do i generate that. |
| 14:53.56 | FlaPer87 | I mean, I can make calls but not recieve them |
| 14:54.08 | FlaPer87 | I don't understand why |
| 14:54.22 | dongs | p3nguin: ? |
| 14:54.23 | FlaPer87 | has this happened before to any of you? |
| 14:54.46 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:56.36 | dongs | where can i see reason for dropped call etc. |
| 14:58.03 | *** join/#asterisk cuco (n=cuco@bzq-82-81-32-128.red.bezeqint.net) |
| 14:58.25 | dongs | hello. |
| 14:58.40 | [TK]D-Fender | dongs: at CLI with channel debug enabled |
| 14:59.02 | dongs | this is sip |
| 14:59.08 | dongs | also i dont know of anything you just said. |
| 14:59.15 | dongs | stuff drops, i dont have scrollback enough to save it im sure. |
| 14:59.19 | dongs | how can i dump all this to afile |
| 14:59.20 | dongs | or wahtever. |
| 15:02.40 | [TK]D-Fender | dongs: use wireshark or something then to isolate SIP traffic and capture everything... or get a bigger scrollback |
| 15:02.46 | dongs | rofl what. |
| 15:02.52 | dongs | are you saying asterisk has no way to put alog into a file? |
| 15:03.00 | dongs | wireshark on what? |
| 15:03.00 | [TK]D-Fender | dongs: is the call dropped a while into an actual conversation? |
| 15:03.03 | dongs | yes. |
| 15:03.06 | dongs | after 4mins20sec. |
| 15:03.10 | dongs | it ALWAYS does this. |
| 15:03.21 | dongs | i want to knwo the rason. |
| 15:03.23 | dongs | reason. |
| 15:04.02 | [TK]D-Fender | dongs: well if you know it always happens that far in just look at * CLI with SIP debug enabled at that time. |
| 15:04.20 | [TK]D-Fender | dongs: no need for anything external if the point of failure is predictable |
| 15:04.23 | *** join/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 15:08.15 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 15:09.33 | PMantis | Hello everyone! I'm building a new platform for use with my customers, and I'm wondering... which config method should I use? What's going to be the "preferred" way to config * going forward? |
| 15:09.52 | *** join/#asterisk raden (n=raden@66-168-15-19.dhcp.stpt.wi.charter.com) |
| 15:11.26 | PMantis | I see: realtime, .conf, .ael, .lua and a #exec /path/to/gen-config.sh |
| 15:11.38 | PMantis | What's the "best" to use? |
| 15:13.10 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 15:26.36 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp119-35.adsl.forthnet.gr) |
| 15:30.18 | *** join/#asterisk scalex000 (n=chatzill@170puntacana02.codetel.net.do) |
| 15:30.32 | [TK]D-Fender | PMantis: .conf |
| 15:30.34 | scalex000 | hello there |
| 15:31.50 | scalex000 | <PROTECTED> |
| 15:32.25 | [TK]D-Fender | scalex000: You can't |
| 15:33.17 | PMantis | [TK]D-Fender: You're the expert, but I can't help but ask why. Why do ael & lua exist if the old .conf is better? |
| 15:33.23 | scalex000 | sorry, they pickup the incoming call, how to prevent this matter |
| 15:33.40 | PMantis | ael looks pretty neat |
| 15:34.51 | scalex000 | TK, do you understand what I want to say? |
| 15:35.16 | [TK]D-Fender | PMantis: Because ael only gets compiled back and therefor doesn't add actual functionality and introduces another point of failure. and limits you to its syntax on top of extensions.conf |
| 15:35.34 | [TK]D-Fender | scalex000: Yes I do understand your issue and there is NO way to prevent this |
| 15:35.45 | [TK]D-Fender | scalex000: Get a DIGITAL circuit where this won't happen |
| 15:36.18 | scalex000 | TK, Thanks |
| 15:38.10 | [TK]D-Fender | PMantis: realtime doesn't support hints or includes IIRC which still rely on .conf anyway, and also adds another point of failure |
| 15:38.21 | PMantis | [TK]D-Fender: lua gets compiled back to .conf, too? |
| 15:38.49 | scalex000 | TK, I setup to Asterisk using IAX and I use encrypt option in have place I get the notice packet failed to decrypt |
| 15:39.46 | magic_1 | hi guys, anyone have anyluck getting a push to talk server up and running with asterisk |
| 15:40.18 | PMantis | magic_1: Kinda sounds like paging functionality :) |
| 15:40.24 | [TK]D-Fender | PMantis: Don't think so. I believe I head from Russellb or Qwell that it was a direct pipe into app execution, but I haven't touched it personally. Go investigate and see what you can and can;t do |
| 15:40.34 | [TK]D-Fender | PMantis: like hints I believe were left out.. |
| 15:41.20 | magic_1 | hhhmmm |
| 15:41.47 | magic_1 | PMantis: have you used the push to talk over cellular |
| 15:41.56 | magic_1 | trying to get that going through asterisk server |
| 15:42.54 | PMantis | magic_1: Yes, it's pretty neat. Don't think * supports that, but you might be able to mimic the functionality. |
| 15:43.38 | magic_1 | PMantis: yep thats my thinking, i wouldnt think by default, but sure there must be a way to mimic it |
| 15:44.43 | *** join/#asterisk zamba (i=marius@flage.org) |
| 15:44.57 | PMantis | [TK]D-Fender: so bottom line, .conf offers more functionality than any other config option, loads the fastest and has the least dependencies. |
| 15:46.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:46.33 | *** join/#asterisk alexshell (n=abc@unaffiliated/alexshell) |
| 15:46.39 | *** join/#asterisk johnakabean (n=john@static-173-50-101-13.nrflva.east.verizon.net) |
| 15:46.39 | scalex000 | TK, this NOTICE[30443]: chan_iax2.c:8382 socket_process: Packet Decrypt Failed! |
| 15:46.45 | johnakabean | hey everyone; I reinstalled asterisk and, before, I had a longer vm_general.inc file that had more options such as allowing callback of people that left messages. How do I get this file back? |
| 15:47.03 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 15:47.09 | johnakabean | apparently asterisk supports more vm options than is in the vm_general.inc |
| 15:47.31 | johnakabean | this is for 1.4.26.2 |
| 15:47.56 | [TK]D-Fender | johnakabean: that file is not an * config file |
| 15:48.23 | johnakabean | well, where is a list of all the voicemail.conf options possible on default asterisk |
| 15:48.44 | [TK]D-Fender | johnakabean: in the sample config file |
| 15:48.52 | johnakabean | I think one was "allowcallback=1" |
| 15:49.21 | [TK]D-Fender | johnakabean: and "all the voicemail.conf options possible" and "default asterisk" no not mix |
| 15:49.25 | [TK]D-Fender | do* |
| 15:49.38 | [TK]D-Fender | johnakabean: "default" does not exist either |
| 15:50.23 | PMantis | heh, * is really just a programming language and a set of binaries to "compile" a PBX from config files. :) |
| 15:50.43 | johnakabean | can you point me somewhere to a dialplan for voicemail that allows people to call them fromt he voicemail |
| 15:51.46 | [TK]D-Fender | johnakabean: CALLBACK <- read the sample config |
| 15:53.43 | johnakabean | I just need a dialplan that I can modify quickly; I have to leave in 20 minutes |
| 15:54.00 | johnakabean | I don't have time to make one; it would take me 40 or 50 minutes |
| 15:54.57 | [TK]D-Fender | johnakabean: "A dialplan"? Whats there to show. Just shove and exten in the context you specify to dial out whatever you dial out |
| 15:55.34 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
| 15:55.40 | [TK]D-Fender | johnakabean: 40 minutes for, 1 VM box option, 1 context header, and 1 dial command? |
| 15:55.43 | [TK]D-Fender | johnakabean: hillarious |
| 15:55.51 | [TK]D-Fender | johnakabean: 2.5 lines of text <- |
| 15:56.09 | [TK]D-Fender | exaggerated the '.5" |
| 15:57.01 | johnakabean | and 10 minutes to surf through freepbx's dialplans to find the vm context |
| 15:57.12 | [TK]D-Fender | johnakabean: johnakabean Wrong |
| 15:57.42 | [TK]D-Fender | johnakabean: if you can take the CID as-is to call them back then you already have a context you could dumpt them in. |
| 15:57.56 | [TK]D-Fender | jahAnd this has NOTHING to do with "making dialplan" since it already exists <- |
| 16:00.49 | johnakabean | vm_CIDNUM ??? |
| 16:00.53 | johnakabean | variable |
| 16:01.54 | johnakabean | bye thanks |
| 16:07.24 | yang | on "sip show registry" it gives me result "Auth sent" instead of Registered - I don't auth correctly, can I get some assistance - http://pastebin.ca/1625262 |
| 16:08.41 | [TK]D-Fender | yang: pastebin the SIP debug of your register attempts |
| 16:08.42 | [TK]D-Fender | ~pb |
| 16:08.43 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:08.45 | [TK]D-Fender | ^^^^^^^^^^^ |
| 16:11.03 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) |
| 16:11.12 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 16:14.31 | yang | [TK]D-Fender: http://pastebin.com/f40f1a247 |
| 16:16.31 | [TK]D-Fender | yang: Full of obvious errors of 404 & 401 |
| 16:16.39 | [TK]D-Fender | yang: bad domains, and other auth |
| 16:17.05 | yang | so, sip.messagenet.it is a wrong entry ? |
| 16:18.00 | yang | funny, they advertise it on their website |
| 16:18.52 | FlaPer87 | yang: are you able to recieve and make calls with messagenet? |
| 16:19.12 | FlaPer87 | If I configure a peer for messagenet I'm not able to recieve incoming calls |
| 16:19.14 | FlaPer87 | =( |
| 16:19.56 | yang | FlaPer87: I am not able to do anything with them, I just recharged my account (it wasn't in use for more than a year) |
| 16:29.01 | [TK]D-Fender | yang: Comment out your realm |
| 16:32.36 | yang | [TK]D-Fender: I looked for "realm" and I cannot find it |
| 16:33.21 | p3nguin | search feature is not working? |
| 16:34.45 | yang | eric:/etc/asterisk# grep -r "real" astersip.conf |
| 16:34.45 | yang | eric:/etc/asterisk# grep -r "real" sip.conf |
| 16:35.06 | yang | no hits |
| 16:36.04 | [TK]D-Fender | yang: http://pastebin.ca/1625262 <- line 18 |
| 16:36.30 | p3nguin | lol |
| 16:36.59 | yang | [TK]D-Fender: the first section is a VoIP provider's example - my config is under ### MY SETTINGS |
| 16:37.50 | p3nguin | If it's in the same file, you're using anything that is uncommented. |
| 16:39.12 | [TK]D-Fender | yang: Your's shows a peer FOLLOWED by a REGISTER which is illegal, and we don't see the rest of your [general] settings. Pastebin the REAL CONFIGS |
| 16:39.53 | yang | register string is inside sip.conf, the rest is inside astersip.conf |
| 16:40.10 | yang | but usually all my other VoIP uplinks work, like that |
| 16:41.51 | [TK]D-Fender | yang: You aren't going to get real help until you cooperate with those trying to see what's really going on. |
| 16:43.22 | *** join/#asterisk Inhuman (n=somethin@S0106001d7e47087f.su.shawcable.net) |
| 16:43.38 | scalex000 | hello, there If i get a notice from iax like this packet failed to decrypt what its mean? |
| 16:44.52 | Inhuman | I have a computer with a fax modem and some analog phone sets. What else do I need in order to set up Asterisk and have an automated attendant answer my phone when someone calls my house? |
| 16:45.17 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 16:45.18 | Inhuman | Will I need any other hardware? |
| 16:45.21 | TJNII | Yes |
| 16:45.30 | TJNII | A fax modem is no good for * |
| 16:45.35 | TJNII | It is for DATA not VOICE |
| 16:45.49 | [TK]D-Fender | Inhuman: A supported FXO interface... which your modem is not |
| 16:45.59 | Inhuman | ah, that's unfortunate |
| 16:46.13 | Inhuman | so I need a modem that supports data and voice or just voice? |
| 16:46.14 | yang | [TK]D-Fender: I think that sip.conf is always read before astersip.conf, since that is an #include /etc/asterisk/astersip.conf, so the register string comes before the actual # |
| 16:46.18 | yang | [messagenet-outgoing] |
| 16:46.53 | [TK]D-Fender | yagI don't know whats in that file where it is positioned relative, and am not going to waste my time guessing |
| 16:47.16 | [TK]D-Fender | Inhuman: No. Anything labeled as a "modem" is not going to be supported by *. |
| 16:47.16 | p3nguin | asterisk*CLI> iax2 reload |
| 16:47.16 | p3nguin | [Oct 17 11:46:08] NOTICE[15697]: chan_iax2.c:9102 iax2_poke_peer: Still have a callno... |
| 16:47.28 | p3nguin | What does this "callno" thing? |
| 16:47.36 | p3nguin | what is, rather |
| 16:47.43 | Inhuman | Fender, so what is it that I do need? |
| 16:49.14 | TJNII | I think the SPA-3000 is popular for what you want |
| 16:49.38 | TJNII | I use its cousin, the PAP2, but that won't let you connect to the telco line. |
| 16:50.04 | [TK]D-Fender | Inhuman: Linksys SPA-3102, or a card like Digium's, Sangoma's, Rhino's, etc |
| 16:50.39 | [TK]D-Fender | p3nguin: Stop mentioning discontinued products from companies that no longer exist :p |
| 16:50.47 | p3nguin | ? |
| 16:50.58 | [TK]D-Fender | p3nguin: SPA-3000 = Sipura |
| 16:51.03 | p3nguin | ? |
| 16:51.05 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 16:51.18 | [TK]D-Fender | p3nguin: Before they got eaten by Linksys. Which then got eaten by Cisco :p |
| 16:51.19 | *** part/#asterisk Inhuman (n=somethin@S0106001d7e47087f.su.shawcable.net) |
| 16:51.23 | p3nguin | ? |
| 16:51.30 | TJNII | [TK]D-Fender: That was me |
| 16:51.36 | TJNII | I did a quick google for it. |
| 16:51.48 | [TK]D-Fender | TJNII: YEAH YOU!!!! AND DON;'T YOU FORGET IT! |
| 16:51.51 | [TK]D-Fender | :P |
| 16:52.09 | p3nguin | I just asked what this "iax2_poke_peer: Still have a callno..." is/means. |
| 16:52.13 | [TK]D-Fender | p3nguin: You get off easy this time... BUT I'M WATCHING YOU! |
| 16:52.17 | TJNII | But if you want to keep yelling at p3nguin I won't complain. |
| 16:52.18 | p3nguin | hahaha |
| 16:54.03 | TJNII | But I love discuntinued hardware! I just bought a box of ISA NICs on eBay for $8! What a bargain! |
| 16:55.42 | p3nguin | I have another situation after I had to change that one peer to IAX2 from SIP. The dial plan for _2XX dials SIP/${EXTEN}. So should I put any IAX2 peers in the 300s and create _3XX Dial(IAX2/${EXTEN}), or is there a better way to handle it? |
| 16:57.09 | [TK]D-Fender | p3nguin: Yes... hard-code them and use a common dial macro passing the full tech to dial |
| 16:59.11 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:59.34 | p3nguin | With it that way, if 204 does not answer the phone, it then jumps into the _2XX match. So in the Dial() do I need to add the j option? |
| 17:00.39 | [TK]D-Fender | p3nguin: NO. |
| 17:00.47 | [TK]D-Fender | p3nguin: ew... Priority jumping was phased out in 1.2 |
| 17:01.05 | [TK]D-Fender | p3nguin: and I'm saying hard-code ALL OF THEM |
| 17:01.13 | [TK]D-Fender | p3nguin: not just his singular peer |
| 17:01.17 | [TK]D-Fender | this* |
| 17:01.23 | p3nguin | You want me to hard code every exten number? |
| 17:01.35 | *** join/#asterisk superbeef (n=IMP-IT@74.84.194.4) |
| 17:01.38 | *** join/#asterisk tamiel (n=tamiel@abo-212-152-68.bdx.modulonet.fr) |
| 17:01.39 | ChannelZ | how many extensions do you hae? |
| 17:01.47 | p3nguin | Won't that get tedious as I scale up? |
| 17:02.11 | p3nguin | Right now, it's not a big deal with only a dozen extensions, but as it grows, that would get ugly. |
| 17:02.16 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp69-203.adsl.forthnet.gr) |
| 17:03.26 | [TK]D-Fender | p3nguin: To what... 30? 50? big deal... |
| 17:03.44 | p3nguin | You hard code 50 extens? |
| 17:03.55 | p3nguin | This surprises me. |
| 17:03.55 | [TK]D-Fender | p3nguin: I've done far more |
| 17:04.06 | ChannelZ | Are you constantly changing them? Or adding 30 extensions in one whack? It's not that big a deal |
| 17:04.28 | [TK]D-Fender | p3nguin: whats a 1-shot copy/paste/tweak? |
| 17:06.27 | p3nguin | Let's say I'm really not capable of doing that. How do I make the single hard-coded exten exit on failure? Dial(M(GOTO^internal^204^101)) or what? |
| 17:07.21 | p3nguin | Check dialstatus and then use Goto()? |
| 17:13.53 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 17:14.54 | [TK]D-Fender | p3nguin: p3nguin check SIPPEER() to see if its a SIP peer otherwise you can ssume its IAX2 |
| 17:15.11 | [TK]D-Fender | p3nguin: but yours still allws you to dial an invalid device |
| 17:15.21 | [TK]D-Fender | p3nguin: which is a great reason why I don't recommend it |
| 17:15.40 | [TK]D-Fender | p3nguin: Person calls your IVR, tries to dial what they thought was a valid # and get dumped |
| 17:16.51 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 17:18.18 | p3nguin | I'm with you on that. I accept your suggestions. |
| 17:18.41 | p3nguin | On the other hand, I think this is a valid solution: |
| 17:18.43 | p3nguin | exten => 204,n,GotoIf($["${DIALSTATUS}" = "NOANWSER"]?:followme|204|1) |
| 17:19.16 | p3nguin | So now it goes to a completely different context rather than trying to match the _2XX in the same context. |
| 17:21.28 | [TK]D-Fender | p3nguin: EW |
| 17:21.36 | [TK]D-Fender | p3nguin: look at the list of possible values... not good |
| 17:21.45 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 17:21.49 | Dovid | j #asterisk-dev |
| 17:22.17 | p3nguin | Is ANSWERED a possible value? Can I do $["${DIALSTATUS}" != "ANWSERED"] ? |
| 17:22.29 | p3nguin | I don't know where the list of values is. |
| 17:23.09 | p3nguin | core show something? |
| 17:23.34 | Dovid | how about testing it |
| 17:23.42 | Dovid | use Noop to see the results of it |
| 17:24.20 | p3nguin | I would rather know where the list is. |
| 17:24.44 | Dovid | http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
| 17:24.46 | [TK]D-Fender | p3nguin: "core show application dial" |
| 17:24.50 | TJNII | debates being a smart-ass and telling him "/usr/src/asterisk" |
| 17:25.04 | [TK]D-Fender | p3nguin: and that wonderful CHANNELVARIABLES doc |
| 17:25.26 | [TK]D-Fender | TJNII: No.. that'd simply be "smart" :) |
| 17:25.38 | me|ong | anybody know why i dial the number to a phone, hear ringing, but the physical machine isn't ringing? |
| 17:25.52 | me|ong | from softphone to hard |
| 17:25.59 | p3nguin | Oops. I didn't scroll up high enough to see the list of dialstatus choices. |
| 17:26.00 | ChannelZ | maybe you called a parallel universe |
| 17:26.06 | me|ong | i hope not :( |
| 17:26.20 | me|ong | i got my cisco 2600 to register.. and the phone supposedly registers |
| 17:26.22 | ChannelZ | The ringer is off on the physical machine |
| 17:26.29 | me|ong | i dial the thing on the fxs port, but it doesn't ring |
| 17:26.32 | me|ong | naw |
| 17:26.37 | me|ong | no lights or anything |
| 17:26.49 | me|ong | if i try to dial from it.. i get dialtone, but nothing after i hit digits |
| 17:26.52 | TJNII | me|ong: Make sure the softphone doesn't play a ringing sound by default after you disl |
| 17:27.18 | me|ong | hmm well i'm debugging through cli, and i see the call being put through |
| 17:27.33 | TJNII | I usually turn that off because it can create weirdness with early media ringing as well. |
| 17:27.33 | [TK]D-Fender | me|ong: we can't... |
| 17:28.01 | me|ong | fender, i'm wondering if you have an answer of where to look for the prob |
| 17:28.20 | me|ong | unless you know how to configure asterisk owork with cisco IOS to an FXS port? |
| 17:28.30 | me|ong | to work* |
| 17:28.35 | ChannelZ | so you're saying you dial an extension, hear a ring, but the extension it's self never rings.. ever |
| 17:28.42 | me|ong | exactly |
| 17:29.08 | ChannelZ | and the console says.... ? |
| 17:29.13 | me|ong | sip show peers shows me the router registered and the phone registered (but unmonitored) |
| 17:29.40 | me|ong | everything you would see when you dial a soft phone |
| 17:29.42 | me|ong | just no pickup |
| 17:30.03 | me|ong | i can even see through the show sip calls that it's being processed |
| 17:30.08 | me|ong | but i can't figure out what's wrong |
| 17:30.21 | ChannelZ | and you pick up the physical device while it's supposed to be ringing but aren't connected? |
| 17:30.44 | drmessano | Who here uses voip.ms? |
| 17:30.58 | TJNII | Claerly the problem is that the pairs in the wire to the phone are twisted too tightly and the bits can't get through. You need to pull out the wire and un twist them. (Or you could pastebin a debug so we can see what it really wrong.) |
| 17:31.03 | p3nguin | raises hand |
| 17:31.17 | eppigy | hello |
| 17:32.08 | me|ong | just tried to answer while calling. Nothing |
| 17:32.14 | me|ong | yet i still hear it ringing on the laptop |
| 17:32.22 | me|ong | TJNII |
| 17:32.29 | me|ong | do you know how to read? |
| 17:32.39 | TJNII | me|ong: Do you know how to pasetbin? |
| 17:32.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:32.55 | TJNII | me|ong: Don't troll me when you refuse to give details. |
| 17:33.00 | ChannelZ | so you're not dialing the right channel, though I can't think of what you could be dialing that would ring. Or your phone is fucking busted |
| 17:33.07 | p3nguin | drmessano: I'm assuming there was more to it than just wondering who uses it. |
| 17:33.24 | TJNII | me|ong: If your powers of debugging were so godd you wouldn't be here asking for f*cking help. |
| 17:33.28 | me|ong | phone's not busted |
| 17:33.33 | TJNII | s/godd/good/ |
| 17:33.38 | me|ong | ffs |
| 17:33.38 | drmessano | p3nguin: Clearly |
| 17:33.47 | me|ong | gimme a sec to pastebin so i can shut TJNII the fuck up |
| 17:34.04 | ChannelZ | /ignore also works |
| 17:35.43 | drmessano | Wondering what the overall opinion of voip.ms is.. I still don't see a lot of google hits for them, good or bad, and I hear good things in here.. they look good on paper.. |
| 17:36.32 | p3nguin | I haven't had any trouble with them so far. |
| 17:36.44 | p3nguin | I have a toll-free DID with them as well as termination. |
| 17:37.25 | p3nguin | My only complaint is that using the value route, my CID doesn't always pass, but they disclaimed that accordingly, so it wasn't a surprise. |
| 17:37.34 | *** join/#asterisk effigee (n=jenshens@S0106001d7e52bc10.vc.shawcable.net) |
| 17:37.34 | [TK]D-Fender | drmessano: I've heard mostly good about them as well.. |
| 17:37.41 | drmessano | Awesome |
| 17:38.28 | drmessano | I was really hoping to get a few friends ported to Flowroute, but it doesnt look as if they'll ever get numbers in this area |
| 17:38.46 | p3nguin | They will also offer you a refund on your unused funds if you decide that you don't like their service. |
| 17:38.57 | drmessano | nice |
| 17:39.13 | p3nguin | I had a friend of mine go to flowroute from callcentric just the other day. |
| 17:39.55 | p3nguin | Well, maybe "offer" was the wrong term. If you don't like the service, you can ask for your unused finances to be returned, and they will comply. |
| 17:39.56 | drmessano | Flowroute service is great.. |
| 17:40.23 | me|ong | http://pastebin.ca/1625473 |
| 17:40.35 | me|ong | the cisco output is during a call |
| 17:40.39 | me|ong | where the phone doesn't ring |
| 17:40.43 | drmessano | They popped up, it worked well, seemed ambitious.. They expanded into new areas.. supposedly were adding more.... then...... crickets |
| 17:40.48 | KavanS | do any of you guys do trunk monitoring for your SIP provider? |
| 17:41.23 | drmessano | I dont have any trunks, only peers |
| 17:41.39 | KavanS | ok, peer monitoring? |
| 17:42.01 | KavanS | i.e. interfacing with nagios or some other monitoring system to let you know when SIP provider is down |
| 17:43.03 | [TK]D-Fender | me|ong: show * SIP debug. that pastebin offers nothing. Show a COMPLETE call |
| 17:43.09 | drmessano | I did have it interfaced to a MIDI sequencer to play "Popcorn" by Hot Butter when they went down |
| 17:43.14 | drmessano | But that was hardly remote |
| 17:43.27 | me|ong | tk |
| 17:43.37 | me|ong | wasn't done |
| 17:43.42 | me|ong | forgot the config on the router |
| 17:43.43 | me|ong | http://pastebin.ca/1625481 |
| 17:44.02 | [TK]D-Fender | me|ong: And we are debugging *, not your Cisco |
| 17:44.13 | drmessano | Beep-Boop-Beep-Boop-Boop-Boop-Boo-Beep-Boop-Beep-Boop-Boop-Boop-Boo |
| 17:44.22 | [TK]D-Fender | me|ong: me|ong so go look at what * is doing. |
| 17:44.27 | me|ong | Asterisk*CLI> show * SIP debug |
| 17:44.28 | me|ong | No such command 'show * SIP debug' (type 'help show * SIP' for other possible commands) |
| 17:44.31 | me|ong | everytime you suggest |
| 17:44.33 | me|ong | i try |
| 17:44.34 | TJNII | Who is 172.32.100.21 |
| 17:44.39 | me|ong | it's the router |
| 17:44.45 | me|ong | learn2network |
| 17:44.51 | [TK]D-Fender | me|ong: I never said that was the CLI command, and clearly you haven't looked yourself |
| 17:44.55 | me|ong | err asterisk |
| 17:45.00 | me|ong | the call manger |
| 17:45.01 | [TK]D-Fender | me|ong: "sip set debug: |
| 17:45.08 | me|ong | ahh |
| 17:45.10 | me|ong | thanks TK :) |
| 17:45.15 | mbrevda | ive got this really werid issue. Im using a fxo-sip adpter (audiocodes mp-114) and I have three pots lines atached. |
| 17:45.19 | mbrevda | Corospondingly, I have three registrations from the ata to asterisk (5001,5002,5003). if the calls some in on line 1, they get sent to <did_a> (not the realy name). Line 2-3 o <did_b>. |
| 17:45.25 | mbrevda | When a call comes in on line one, I dont get callerID (apperently due to an issue in authentication - I dont really understand that part) - but if it comes in on line2, everything works fine! |
| 17:45.25 | mbrevda | All lines are configures exactly the same, here is the sip debug for did_a: http://pastebin.ca/1625477 and for did_b http://pastebin.ca/1625478 |
| 17:45.58 | me|ong | tk, that's not working either |
| 17:46.21 | me|ong | i'm at the cli and it shows the call process, is that not what you're looking for? |
| 17:46.27 | [TK]D-Fender | me|ong: show us |
| 17:47.13 | me|ong | http://pastebin.ca/1625492 |
| 17:47.28 | TJNII | mbrevda: So the call comes in on the did and the phone connected to the line 1 port doesn't show caller ID? |
| 17:47.34 | [TK]D-Fender | facepalms |
| 17:47.45 | [TK]D-Fender | me|ong: FS show us the attempt to ENABLE SIP DEBUG |
| 17:47.56 | me|ong | Asterisk*CLI> show * SIP debug |
| 17:47.56 | me|ong | No such command 'show * SIP debug' (type 'help show * SIP' for other possible commands) |
| 17:47.56 | me|ong | Asterisk*CLI> set sip debug |
| 17:47.56 | me|ong | No such command 'set sip debug' (type 'help set sip' for other possible commands) |
| 17:47.56 | me|ong | Asterisk*CLI> sip set debug |
| 17:47.56 | me|ong | No such command 'sip set debug' (type 'help sip set debug' for other possible commands) |
| 17:47.58 | me|ong | Asterisk*CLI> sip set debug: |
| 17:48.00 | me|ong | No such command 'sip set debug:' (type 'help sip set debug:' for other possible commands) |
| 17:48.02 | me|ong | Asterisk*CLI> |
| 17:48.15 | [TK]D-Fender | me|ong: Do not spam like that again.... |
| 17:48.24 | p3nguin | s/spam/flood/ |
| 17:48.26 | [TK]D-Fender | me|ong: "sip debug" |
| 17:48.36 | p3nguin | spam is that unwanted email we get. |
| 17:48.41 | me|ong | no such command |
| 17:48.54 | [TK]D-Fender | me|ong: "help sip" <- go look at the list yourself |
| 17:48.59 | ChannelZ | sip set debug |
| 17:49.09 | mbrevda | TJNII: no. the ata forwards the pots call to the did set up in asterisk (this is an analog fxo) |
| 17:49.12 | [TK]D-Fender | ChannelZ: syntax changed between versions |
| 17:49.23 | me|ong | points to his third attempt for channelz to see |
| 17:49.35 | TJNII | mbrevda: Then I'm confused. Who isn't seeing the caller ID? Someone on the outside? |
| 17:50.12 | me|ong | sip set debug ON |
| 17:50.14 | me|ong | sighs |
| 17:50.28 | mbrevda | TJNII: sorry, though i spelled everything out. post -> ata(ID)->*(no ID) |
| 17:51.23 | TJNII | Aah. |
| 17:52.03 | me|ong | http://pastebin.ca/1625502 |
| 17:52.07 | TJNII | Okay, I missed "FXO adapter." Makes more sense now. |
| 17:52.18 | mbrevda | TJNII: thing is, ata IS sending callerid - if you look thru the sip debug, * isnt accepting the call that way |
| 17:52.32 | TJNII | is looking... |
| 17:53.29 | TJNII | mbrevda: Could you pastebin your sip.conf, too? May help. |
| 17:55.19 | [TK]D-Fender | me|ong: Where is the complete call attempt |
| 17:55.32 | me|ong | that's the debug from before i called to when it hung up on me |
| 17:55.36 | TJNII | mbrevda: Yea, pastebin your sip. Lines 108-114 are very suspicious! |
| 17:55.49 | mbrevda | sure, 1 sec |
| 17:56.15 | mbrevda | TJNII: line 108-114 in which debug? |
| 17:56.16 | mbrevda | http://pastebin.ca/1625513 |
| 17:56.37 | p3nguin | Hard-coding the extens certainly does make things easier to deal with, but it gives me more stuff to manage. |
| 17:56.44 | TJNII | mbrevda: The one you tahhed did_a |
| 17:57.43 | mbrevda | doesnt see the problem; obviously hasnt learned to read a sip debug |
| 17:57.54 | TJNII | mbrevda: Is that your complete sip.conf? |
| 17:58.22 | mbrevda | TJNII: aside for the trunks and their registrations |
| 17:58.35 | p3nguin | Is there a way to use functions from the console? |
| 17:59.13 | TJNII | mbrevda: No [general]? Do you want the settings at the bottom to only apply to [5003]? |
| 17:59.44 | mbrevda | TJNII: freepbx. I just copied the relevant parts |
| 18:00.09 | *** join/#asterisk traxx (n=traxx@91-64-130-76-dynip.superkabel.de) |
| 18:00.12 | TJNII | Aah. |
| 18:02.35 | mbrevda | TJNII: a lightly different question is why 5003 is involved at all - (Found peer '5003' for '<cellnum>' ) as that /shoudl/ only be used on calls from/to the theird line |
| 18:02.40 | mbrevda | *slightly |
| 18:02.54 | TJNII | mbrevda: Well, I see a couple issues. |
| 18:04.08 | TJNII | Is 1234 the real secret or did you use that in place of the real password? |
| 18:04.51 | mbrevda | TJNII: real! y? |
| 18:05.06 | mbrevda | want unlimited log didtance? ;) |
| 18:05.20 | mbrevda | *distance (/me cant spell tonight) |
| 18:06.43 | TJNII | mbrevda: Don't get too excited. If you look at lines 282-336 you'll see freepbx is firing off a whole lot of agis. Any one could lop off the callerID. |
| 18:06.50 | TJNII | ~freepbx |
| 18:06.50 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:07.11 | jblack | SO, callcentric refunded that 15 bucks. |
| 18:07.12 | TJNII | We have no way to know what those agis are doing, as we are uninvolved in them. |
| 18:07.37 | jblack | Apparently somebody with skills works late friday/early saturday there. |
| 18:08.25 | mbrevda | TJNII: if you look at the very first lines you will see that the callerid is being set TO 5001. Hence something is being recived WRONG. also, notice that for did_b, the callerID comethru alright (both traces are coming from the same callerid) |
| 18:08.35 | TJNII | mbrevda: But, if 1234 is your real secret, consider giving each line its own unique secret. Probably not your problem, but a good idea. 5003 is involved because it shares the IP. * is trying it first, failing, and the moving on to 5001. |
| 18:09.02 | TJNII | mbrevda: True. But a whole bunch of agis are called between the call coming in from the ATA and coing to your phone. |
| 18:09.25 | mbrevda | TJNII: if you would like, I can insert a line or two to the dialplan to prove the poitn |
| 18:11.58 | TJNII | mbrevda: I'm not going to argue with more information, but keep in mind it is a freepbx dialplan. Are you sure it is doing what you think it is doing? I'm not. |
| 18:12.45 | TJNII | And I don't know enough about what freepbx does to give you good advice. Sorry. |
| 18:15.47 | [TK]D-Fender | 13:55]<me|ong>that's the debug from before i called to when it hung up on me - no, we do not see the original INVITE, nor the dialplanexecution |
| 18:19.31 | me|ong | that's what i'm trying to tell you. I started the debug, I dialed, i waited for the softphone to hear a pickup or voicemail or soemthing. Instead it just finished with "hung up" |
| 18:21.39 | [TK]D-Fender | me|ong: that isn't the entire call. Go show the COMPLETE attempt |
| 18:21.55 | [TK]D-Fender | me|verbose 10 |
| 18:22.15 | me|ong | can you add verbose? |
| 18:22.31 | me|ong | if not, you're clearly not askign the right question or too stubborn to listen to what i'm telling you |
| 18:22.46 | me|ong | i started the debug 10 seconds BEFORE i started/initiated/made the call |
| 18:23.05 | me|ong | i watched.. nothing. i THEN dialed on my softphone 2033 to TRY and reach the hard phone attached to the cisco 2600 |
| 18:23.12 | me|ong | i get ringing on the soft phone |
| 18:23.15 | me|ong | nothing on hard phone |
| 18:23.33 | me|ong | all the debugs/pastebins/outputs from router and asterisk are all done from the time i started.. during.. and after |
| 18:23.52 | [TK]D-Fender | me|ong: your pastebin starts with a 200 OK, not the initial INVITE. we don't see what * is processing in the dialplan either |
| 18:24.11 | me|ong | did you read the pastebins i posted earlier? |
| 18:24.25 | me|ong | if not.. well sir you need to scroll up and see that i answered your questions already. |
| 18:24.28 | [TK]D-Fender | me|show me one meeting the points I asked for |
| 18:24.32 | me|ong | k |
| 18:24.40 | me|ong | define in steps exactly how you want this done then |
| 18:24.55 | me|ong | 'cause either you speak an entirely different language and we have one hell of a text barrier |
| 18:25.11 | me|ong | or you're just don't really care to help |
| 18:25.14 | me|ong | you* |
| 18:25.30 | me|ong | i speak french/english |
| 18:25.31 | [TK]D-Fender | I've said it about a dozen times now. Enable SIP debug, Set verbose to 10, pastebin the complete call attempt from beginning to end. |
| 18:26.09 | me|ong | k maybe i wasn't clear when i asked about adding verbose.. 'cause there's no other extra switches when turngin debug on |
| 18:26.10 | me|ong | apart form peer |
| 18:26.15 | me|ong | which i have done already |
| 18:26.19 | [TK]D-Fender | me|ong: http://pastebin.ca/1625502 <- doesn't start with the original INVITE, doesn't have verbose of * apps being executed |
| 18:26.19 | me|ong | <me|ong> can you add verbose? |
| 18:26.33 | [TK]D-Fender | me|ong: set verbose 10 |
| 18:27.07 | me|ong | No such command 'set verbose 10' (type 'help set verbose' for other possible commands) |
| 18:27.11 | me|ong | 0/5 now |
| 18:27.41 | me|ong | man this cli is frustrating.. is there a guide or command set that isn't built in? |
| 18:27.49 | ChannelZ | help |
| 18:27.53 | me|ong | tried |
| 18:28.07 | me|ong | it's like talking to a downsyndrome 90 year old with alzeimers |
| 18:28.09 | [TK]D-Fender | core set verbose 10 |
| 18:28.14 | [TK]D-Fender | and YES, "help" does just that |
| 18:28.56 | me|ong | score! thanks TK :) |
| 18:28.57 | me|ong | Verbosity was 3 and is now 10 |
| 18:29.49 | me|ong | give me a sec |
| 18:30.02 | me|ong | i'm doing logging through securecrt so you get all the commands and outputs |
| 18:31.05 | [TK]D-Fender | me|ong: I'm quite sure that SecureCRT doesn't show * CLI any better than Putty, standard SSH, etc... |
| 18:31.34 | me|ong | http://www.adcmelong.ca/debug.txt |
| 18:31.44 | me|ong | when it does that |
| 18:31.44 | me|ong | yeah |
| 18:31.49 | me|ong | ;) |
| 18:31.52 | me|ong | i work at cli for a living |
| 18:32.25 | me|ong | was never a fan of putty's buffers either |
| 18:32.45 | ChannelZ | I (hump) Putty |
| 18:32.49 | me|ong | from when i turned on the verbose |
| 18:32.53 | [TK]D-Fender | me|ong: Ok, speaking of Alzheimer's.... lets try this again.... THE ENTIRE CALL |
| 18:32.58 | me|ong | to when my little softphone said "hung up" |
| 18:33.00 | me|ong | I DID |
| 18:33.02 | me|ong | sighs |
| 18:33.04 | p3nguin | What's wrong with PuTTY's buffering? |
| 18:33.06 | me|ong | fucking read |
| 18:33.07 | me|ong | please |
| 18:33.07 | me|ong | I DID |
| 18:33.14 | me|ong | turned verbose to 10 |
| 18:33.19 | me|ong | you saw me type "score thanks" |
| 18:33.25 | me|ong | i then turned left to my laptop |
| 18:33.26 | me|ong | and DIALED |
| 18:33.31 | me|ong | all while that output was being taken |
| 18:33.50 | me|ong | ... so.. clearly.. if what you're looking for isn't showing up, that must be the problem... |
| 18:33.57 | *** join/#asterisk fofware (n=chatzill@host156.190-225-62.telecom.net.ar) |
| 18:34.08 | me|ong | i don't know what that problem is, so what can it be? This is why i'm here asking the "experts" |
| 18:34.18 | p3nguin | This seems familiar to me. |
| 18:34.27 | [TK]D-Fender | me|ong: holy fucking shit the call start with SIP/200 calling in! I want to see from the fucking instant SIP/200 placing tits fucking call. SIP/200 can be AT FAULT HERE. |
| 18:34.33 | [TK]D-Fender | me|ong: Are you reading me now? |
| 18:34.41 | [TK]D-Fender | me|ong: The call STARTS with SIP/200 |
| 18:34.46 | [TK]D-Fender | me|ong: so wake the hell up |
| 18:34.53 | me|ong | from the soft phone |
| 18:35.04 | [TK]D-Fender | YES FROM THE FUCKING SOFT PHONE |
| 18:35.10 | me|ong | i just started |
| 18:35.11 | mchou | haha |
| 18:35.12 | me|ong | do i hear an echo? |
| 18:35.20 | me|ong | stated* |
| 18:35.20 | mchou | wh's on first? |
| 18:35.26 | me|ong | lol |
| 18:35.30 | mchou | who's* |
| 18:35.33 | TJNII | Third base! |
| 18:35.44 | p3nguin | I Don't Know |
| 18:35.54 | me|ong | no, who's on third |
| 18:35.58 | p3nguin | no |
| 18:36.03 | me|ong | lol |
| 18:36.03 | [TK]D-Fender | me|ong: I wan to see the ENTIRE call starting from SIP/200's friggen initial INVITE |
| 18:36.14 | mchou | lol |
| 18:36.18 | me|ong | want me to video tape what i'm doing? |
| 18:36.28 | me|ong | 'cause short of you seeing physically |
| 18:36.33 | mchou | [TK]D-Fender: feeling a tad frustrated? |
| 18:36.38 | me|ong | he is?! |
| 18:37.00 | [TK]D-Fender | me|ong: mchou Yes, I have seen smarter dirt. It's being put to good use as potting soil for my plants |
| 18:37.06 | mchou | [TK]D-Fender: try not to pop a vein :) |
| 18:37.06 | TJNII | mchou: I don't know how he does this all day. |
| 18:37.06 | me|ong | give me a sec |
| 18:37.15 | me|ong | i'll make a call from softphoen to softphone |
| 18:37.18 | me|ong | then make a call right after |
| 18:37.40 | me|ong | giving "all" the debug he's apparently asking for but doesn't "See" when i do it just for the one phone |
| 18:38.32 | [TK]D-Fender | me|ong: http://pastebin.com/mf80513 |
| 18:39.04 | mchou | [TK]D-Fender: you have a point there |
| 18:39.17 | mchou | [TK]D-Fender: the moron part :) |
| 18:39.22 | me|ong | so, let me see if i understand your half assed english |
| 18:39.22 | [TK]D-Fender | mchou: The potting soil, or my pastebin? |
| 18:39.24 | me|ong | you want me to dial |
| 18:39.34 | me|ong | and leave it? no answer? CAUSE i'm NOT ANSWERING |
| 18:39.35 | me|ong | :D |
| 18:39.37 | mchou | [TK]D-Fender: the pb |
| 18:39.39 | me|ong | i can't. |
| 18:39.49 | me|ong | http://www.adcmelong.ca/debug.txt |
| 18:39.51 | me|ong | there |
| 18:39.52 | me|ong | two calls |
| 18:40.02 | me|ong | one from soft-soft |
| 18:40.03 | me|ong | answers |
| 18:40.05 | me|ong | i can hear |
| 18:40.14 | [TK]D-Fender | me|ong: Your stupi PB starts after SIP/200 call COMES IN. I want even the SIP debug from the START of that fucking call. How God damned dense are you? |
| 18:40.15 | me|ong | then i hung up and right after, I called the landline on the router |
| 18:40.44 | me|ong | i'm dense? i'm trying to tell you what you're asking for isn't happening.. i've tried to get that to output.. and it's not |
| 18:40.53 | me|ong | i call.. nothing shows up in debug... i'm aksing YOU why |
| 18:41.13 | [TK]D-Fender | me|ong: Isn't happening? |
| 18:41.13 | *** join/#asterisk gardo (n=gardo@121.97.141.237) |
| 18:41.26 | mchou | breaks out the popcorn for real now |
| 18:41.27 | [TK]D-Fender | me|ong: -- Executing [2033@users:1] Macro("SIP/201-095795b8", "phone") in new stack <---- this isn't the first thing that * outputs here... |
| 18:41.54 | [TK]D-Fender | me|ong: this is after * negotiates the call with SIP/200. You can't seem to find the SIP debug for this |
| 18:42.14 | me|ong | apparently not, i turn on the debug before any negotiations happen |
| 18:42.22 | [TK]D-Fender | me|ong: -- Executing [2033@users:1] Macro("SIP/201-095795b8", "phone") in new stack <- where's the negotiation that lead to this? |
| 18:43.33 | me|ong | that's the output from when debugging isn't on, correct? |
| 18:43.44 | mchou | haha!! |
| 18:43.51 | [TK]D-Fender | me|ong: its from after the point where I asked you to start |
| 18:44.00 | [TK]D-Fender | LONG after |
| 18:44.43 | [TK]D-Fender | me|ong: Where's the INVITE from SIP/200? |
| 18:46.50 | me|ong | http://pastebin.ca/1625588 |
| 18:46.52 | me|ong | line 100 |
| 18:47.02 | me|ong | that's the same output you were seeing earlier |
| 18:47.35 | me|ong | line 71 |
| 18:47.35 | [TK]D-Fender | me|ong: no, this is from MUCH earlier |
| 18:47.36 | me|ong | # |
| 18:47.36 | me|ong | Asterisk*CLI> |
| 18:47.36 | me|ong | # |
| 18:47.36 | me|ong | <--- SIP read from UDP://172.32.100.31:62460 ---> |
| 18:47.36 | me|ong | # |
| 18:47.38 | me|ong | INVITE sip:2033@asterisk.adcmelong.local SIP/2.0 |
| 18:47.47 | me|ong | this is the debug i was providing |
| 18:47.49 | me|ong | i sifted through it |
| 18:47.52 | [TK]D-Fender | me|ong: Do NOT spma like that again |
| 18:47.52 | me|ong | and cut it OUT for you |
| 18:48.01 | [TK]D-Fender | spam* |
| 18:48.04 | p3nguin | It's a flood, not spam. |
| 18:48.07 | me|ong | lrn2read |
| 18:48.11 | p3nguin | spam comes in email. |
| 18:48.16 | [TK]D-Fender | me|ong: and I asked from SIP./200's invite and you only provided trimmed off crap |
| 18:48.22 | p3nguin | flooding it lots of lines pasted to the channel. |
| 18:48.25 | me|ong | sighs |
| 18:48.39 | me|ong | that's the problem then... why is 200 not doing the invite? |
| 18:48.42 | [TK]D-Fender | me|ong: but yes, this LAST Pb finally has what I asked for. |
| 18:48.51 | me|ong | fender |
| 18:48.53 | me|ong | listen.. |
| 18:49.00 | me|ong | i took that from the original long ass debug i made |
| 18:49.03 | me|ong | i pulled it out |
| 18:49.08 | me|ong | i cut it from a text |
| 18:49.09 | me|ong | pasted in a new |
| 18:49.31 | [TK]D-Fender | me|ong: http://pastebin.ca/1625502 <- your original. Didn't have the originating invite |
| 18:49.36 | me|ong | can we move on from fucking arguing and please try to figure this out? why would 200 not send an invite back (is it's supposed to) |
| 18:50.16 | me|ong | wtf |
| 18:50.24 | me|ong | i swear i pulled it out from the debug.txt dude |
| 18:50.34 | me|ong | i went in and ctrl+f the fucker to find invite |
| 18:50.34 | [TK]D-Fender | me|ong: and I see your hosted link has changed contents |
| 18:50.35 | me|ong | :| |
| 18:50.54 | me|ong | ah fuck |
| 18:50.56 | me|ong | sighs |
| 18:51.07 | me|ong | i have to apologize now.. i had two fucking debugs running |
| 18:51.10 | me|ong | it overwrote it |
| 18:51.14 | me|ong | sighs |
| 18:51.31 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 18:51.46 | me|ong | sorry bud... forgot i set on the debug when i first got it |
| 18:51.54 | me|ong | err nto debug |
| 18:51.55 | me|ong | logging |
| 18:51.55 | [TK]D-Fender | me|ong: You are the smartest dumb person I've talked with all week. You can claim a plushie from the top shelf now... |
| 18:52.06 | mchou | haha |
| 18:52.11 | me|ong | bastard :P |
| 18:52.21 | me|ong | so, the last one ... what's missing from it |
| 18:52.29 | me|ong | or did that give you an idea of wtf is up? |
| 18:52.31 | [TK]D-Fender | me|ong: that's "Mr. Fucking Bastard" to you :p |
| 18:52.33 | *** join/#asterisk coppice (n=chatzill@host86-132-26-86.range86-132.btcentralplus.com) |
| 18:52.40 | me|ong | i'll stick to bastard. |
| 18:52.55 | mchou | "fat bastard" |
| 18:53.02 | [TK]D-Fender | me|ong: Now SIP/200 IS accepted after responding to the INVITE challenge, and * processes the call |
| 18:53.46 | ChannelZ | jizzes |
| 18:53.54 | [TK]D-Fender | me|ong: http://pastebin.ca/1625588 <- 70 shows the retry w/ auth |
| 18:54.15 | [TK]D-Fender | me|ong: http://pastebin.ca/1625588 <- 133 the call is accepted and is in the dialplan |
| 18:54.28 | [TK]D-Fender | 134 it pretty much immediately dials out. |
| 18:55.15 | [TK]D-Fender | me|ong: So fine... what's wrong prior, and past this point? |
| 18:55.40 | me|ong | so everything looks like it's working then? |
| 18:55.46 | me|ong | is that what you're saying? |
| 18:56.37 | TJNII | What, exactly, is 2033 again? Cisco hardphone, a FXS router, ? |
| 18:56.41 | [TK]D-Fender | me|ong: well I'm describing what I'm seeing. What do you qualify as "bad" about the call? |
| 18:57.28 | me|ong | the "bad" is that, i dial 2033 on the softphone (ext 200) and all i get is ringing. The actual hard phone attached to the router doesn't ring |
| 18:57.38 | me|ong | it's a working phone, i use it as my regular home phone |
| 18:57.48 | [TK]D-Fender | me|ong: well the Cisco is reporting progress |
| 18:58.01 | [TK]D-Fender | me|your problem is with your Cisco. * is doing its job fine |
| 18:58.21 | [TK]D-Fender | me|ong: ----> #cisco |
| 18:58.29 | me|ong | i know my * is working :P that's what i was trying to say earlier |
| 18:58.30 | [TK]D-Fender | me|ong: no move along..... |
| 18:58.38 | [TK]D-Fender | me|ong: ... and don't forget your plushie! |
| 18:58.44 | me|ong | i was trying to get some help to figure out what the problem may be |
| 18:59.11 | me|ong | i usually do reasearch before pestering the experts :P hence why i came to you guys hoping for assistance |
| 18:59.47 | [TK]D-Fender | me|ong: Oh we are experts in advising people to stay away from Cisco.... |
| 19:00.04 | me|ong | TJNII, it's a Cisco 2600 with 2fxo and 2fxs ports, |
| 19:00.22 | me|ong | i plugged my phone into the fxs1/0/0 port to get call routing setup |
| 19:00.22 | mchou | [TK]D-Fender: that include linksys too? |
| 19:00.46 | me|ong | fender, free router beats money spent on phone cards |
| 19:00.48 | [TK]D-Fender | mchou: No, Linksys aren't generally uncooperative bitchy and difficult to work with |
| 19:01.01 | me|ong | and ... it's fun to get technologies working together.. you learn alot ;) lol |
| 19:01.29 | [TK]D-Fender | me|ong: Yes.... Edison said something similar about inventing the lightbulb as well..... |
| 19:01.35 | me|ong | linksys was only bought out by cisco to put theri name on it |
| 19:02.13 | me|ong | either or, do you have any suggestion of why the call setup is being made, but nothing is happening on the connection end? |
| 19:02.15 | mchou | it's only a matter of time b4 cisco screws up linksys too |
| 19:02.21 | me|ong | physical connection* |
| 19:02.29 | me|ong | they aren't touching it |
| 19:02.30 | *** join/#asterisk peterthesing (n=peter@s55907487.adsl.wanadoo.nl) |
| 19:02.34 | me|ong | it's strictly for "name" |
| 19:02.40 | mchou | sure they are |
| 19:02.49 | mchou | try to find linsys docs |
| 19:02.53 | peterthesing | hi |
| 19:02.54 | mchou | linksys* |
| 19:02.55 | TJNII | Make 100% sure you're talking to the port you think you are talking to would be my suggestion. |
| 19:02.57 | me|ong | who needs linksys docs? |
| 19:02.59 | me|ong | click |
| 19:03.01 | me|ong | done |
| 19:03.05 | me|ong | wow welcome to WINDOWS |
| 19:03.54 | me|ong | should be talking.. it registers itself |
| 19:04.36 | peterthesing | i have a problem such as asterisk will not connect to mysql |
| 19:05.13 | me|ong | http://pastebin.ca/1625608 |
| 19:06.11 | *** join/#asterisk clovervidia (i=614c4a45@gateway/web/freenode/x-mzyuenwcgyaidoci) |
| 19:06.31 | clovervidia | hi can someone help me set up asterisk? |
| 19:06.52 | peterthesing | what do you need? |
| 19:07.12 | clovervidia | i just installed it on my mepis laptop and i dont know how to launch it or configure it |
| 19:07.51 | me|ong | :| |
| 19:07.51 | peterthesing | it should start using a runlevel |
| 19:07.51 | me|ong | if you installed it |
| 19:07.51 | me|ong | it tells you all the details when you're finished. |
| 19:08.01 | clovervidia | i installed it with the package manager |
| 19:08.03 | peterthesing | configuring is is a major pain |
| 19:08.10 | me|ong | ew. |
| 19:08.14 | TJNII | me|ong: Does a call the other way work? Cisco -> softphone |
| 19:08.19 | me|ong | nerp |
| 19:08.22 | TJNII | ~book |
| 19:08.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:08.23 | me|ong | doesn't. |
| 19:08.28 | TJNII | clovervidia: Start here ^ |
| 19:08.36 | me|ong | it's a useful book ;) |
| 19:08.41 | TJNII | clovervidia: Or at least skim it. |
| 19:08.46 | clovervidia | sounds good |
| 19:08.50 | clovervidia | toodles |
| 19:10.03 | TJNII | me|ong: To take a page from [TK]D-Fender's book, can you pastebin a complete sip debug for that direction? |
| 19:10.07 | clovervidia | hmm.... but i installed it with a package manager.. i dont really know where it went off to... @_@ |
| 19:10.32 | TJNII | clovervidia: Which distro |
| 19:10.37 | clovervidia | mepis |
| 19:10.40 | me|ong | nothing shows up in asterisk |
| 19:10.44 | me|ong | i've tried that already |
| 19:10.48 | p3nguin | You should be installing with the package manager so that you can manage it correctly later. |
| 19:10.53 | clovervidia | i did |
| 19:10.55 | TJNII | me|ong: Nothing at all? |
| 19:10.56 | p3nguin | I know. |
| 19:10.58 | p3nguin | You should have. |
| 19:11.02 | me|ong | let me try again |
| 19:11.06 | me|ong | i've changed a few things... |
| 19:11.14 | TJNII | me|ong: Turn qualify on and make sure the cisco is actually registered. |
| 19:11.26 | clovervidia | should there be an option to launch it from the menu? |
| 19:11.27 | p3nguin | clovervidia: So what was the problem? If you installed it, you're ready to configure it and use it. |
| 19:11.29 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
| 19:11.42 | p3nguin | clovervidia: Look at /etc/asterisk/ or /usr/local/etc/asterisk/ for the configs. |
| 19:11.43 | peterthesing | i have tried reinstalling mysql and asterisk rewrote res_mysql.conf but still i got errors when asterisk attempts to connect |
| 19:11.45 | me|ong | it's registered |
| 19:11.45 | TJNII | clovervidia: * generally runs as a daemon. |
| 19:11.48 | me|ong | that i'll guarantee |
| 19:12.44 | p3nguin | clovervidia: It's not a regular "program" that you just run when you feel like it. |
| 19:12.44 | me|ong | seems like ti doesn't know it's own number |
| 19:12.44 | me|ong | qtf |
| 19:12.44 | p3nguin | clovervidia: It's a system program that runs all the time, so you will need to configure it and run it from the command line. |
| 19:12.45 | clovervidia | ah cli it is then.. |
| 19:12.49 | TJNII | clovervidia: What do you want * for? |
| 19:13.13 | clovervidia | im doing things with the phones in the house |
| 19:13.19 | *** join/#asterisk cuco (n=cuco@bzq-82-81-32-128.red.bezeqint.net) |
| 19:14.00 | clovervidia | okay i've found all the conf settings... @_@ big lot.. |
| 19:14.02 | [TK]D-Fender | clovervidia: * is a SERVICE which you tend to leave running all the time, and not on a machine that you shut down. |
| 19:14.45 | clovervidia | if i turn it off will all the phones go down or will they just revert to normal use? |
| 19:15.03 | TJNII | clovervidia: How do you plan to connect the phones to it? |
| 19:15.25 | clovervidia | i will run it through google voice |
| 19:15.30 | p3nguin | If you are using only IP phones, they will not work if your Asterisk box goes down. |
| 19:15.47 | clovervidia | how would i connect a standard rj-11 phone then? |
| 19:15.50 | p3nguin | clovervidia: Good luck with that. My Google Voice number only does a forward to another number. |
| 19:16.33 | p3nguin | If you want to connect regular phones, you need a special card with jacks for regular phones... or you can use a special adapter to connect between the phone and the computer. |
| 19:16.44 | [TK]D-Fender | clovervidia: How does a piece of SOFTWARE magically control phones you don't have hardware interfaces for? |
| 19:16.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.19 | [TK]D-Fender | clovervidia: And to let * use an analog phone : ATA's |
| 19:17.21 | [TK]D-Fender | ~ata |
| 19:17.22 | infobot | methinks ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 19:17.43 | clovervidia | so i have the dect phones that only need one jack all the other ones will work with this? |
| 19:18.16 | [TK]D-Fender | clovervidia: You have a single receiver base for them? |
| 19:18.38 | clovervidia | yup |
| 19:18.46 | [TK]D-Fender | clovervidia: So only 1 device to be plugged in? |
| 19:18.58 | clovervidia | or so they claim |
| 19:19.01 | [TK]D-Fender | clovervidia: Ok, you have a regular analog POTS line as well? |
| 19:19.12 | TJNII | clovervidia: And before you ask, no a fax-modem won't work. |
| 19:19.12 | [TK]D-Fender | clovervidia: so "they" claim? Who is "they"? |
| 19:19.13 | clovervidia | pots? |
| 19:19.19 | [TK]D-Fender | ~pots |
| 19:19.20 | infobot | [~pots] POTS (Plain Old Telephone Service) is the term for a common analog phone line service as is used world-wide. The "phone company" is called FXO (~fxo), and the user end-point (or phone) is called FXS (~FXS). POTS supports 1 channel, and possibly call-waiting, 3-way calling, CID, as signalled to the telco. |
| 19:19.27 | clovervidia | i wasn't going to ask but that would be helpful |
| 19:19.31 | clovervidia | oh that pots |
| 19:19.50 | clovervidia | aha... so thats how it is |
| 19:20.32 | mbrevda | does asterisk have issues with multipal peers on the same ip? |
| 19:20.44 | p3nguin | clovervidia: Also, PSTN is a good term to know. |
| 19:20.47 | p3nguin | ~pstn |
| 19:20.48 | infobot | [pstn] Public Switched Telephone Network, or "please stop the nonsense" |
| 19:20.58 | clovervidia | aha.. |
| 19:21.07 | clovervidia | well thanks for the info and the links |
| 19:21.18 | clovervidia | if anything/everything blows up ill be back here |
| 19:21.27 | clovervidia | -lifts hat- |
| 19:23.26 | TJNII | So I called myself on a softphone, just screwing around. I answer and WTF? Why am I hearing U2? Then I remember "Oh yea, stereo hooked up to the sound card." Still made for a huge WTF moment.... |
| 19:25.27 | [TK]D-Fender | TJNII: Well yuo still haven't found... what you're looking for :p |
| 19:25.53 | TJNII | I think that was the song..... |
| 19:25.59 | TJNII | I on'y cought the tail end of it |
| 19:26.08 | TJNII | (Stereo is set to the radio) |
| 19:26.42 | coppice | why would anyone choose to hear U2? :-\ |
| 19:27.00 | scalex000 | TK. about your sugestion to use digital phones, I would like but here if you need lines you need order for T1, and this place not need 24 lines |
| 19:27.28 | peterthesing | while reading the book is it true that mysql is not supported and postgrsql only using odbc? |
| 19:27.57 | peterthesing | that from mysql i have from personal experience |
| 19:29.43 | [TK]D-Fender | scalex000: Who said T1 was the only option? Or that you had to get 24 channels on it? or that you neededd to personally use TDM at all? |
| 19:30.01 | [TK]D-Fender | peterthesing: No, it is not true |
| 19:30.14 | peterthesing | o??? |
| 19:30.49 | me|ong | ~unf |
| 19:30.50 | infobot | i heard unf is the Universal Noise of Frustration, it is also washort's favourite word, after heh . something else related to pr0n movies |
| 19:30.50 | peterthesing | realtim mysql status tells me differently?! |
| 19:30.54 | me|ong | :D |
| 19:30.57 | scalex000 | Tk, I mean I can setup a Digital station buy and card. but the lines will continue analog. |
| 19:31.12 | me|ong | sorry, couldn't help it |
| 19:31.27 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:c10c:1abb:687b:8c67) |
| 19:33.29 | peterthesing | is there a way to configure asterisk so it can connect to mysql on localhost or 127.0.0.1 |
| 19:34.40 | TJNII | peterthesing: I've never toyed with odbc, so don't ask me anything specific, but did you build * with mysql support? |
| 19:34.56 | peterthesing | yes i did |
| 19:36.40 | [TK]D-Fender | scalex000: "digital station" <- huh? |
| 19:36.55 | [TK]D-Fender | peterthesing: what about ODBC support? |
| 19:37.24 | [TK]D-Fender | peterthesing: Because there are plenty of docs showing ODBC being used. Also * does not know what ODBC is doing in the background. Nor does it care |
| 19:37.52 | [TK]D-Fender | peterthesing: of course there is the question of why you want to put ODBC between * and MySQL anyway since it can work direct |
| 19:37.56 | *** join/#asterisk DavidR2008 (n=chatzill@71.0.16.133) |
| 19:38.06 | peterthesing | well if it can't connect directly to mysql on the same machine perrhaps obdc might work |
| 19:38.08 | scalex000 | TK, with digital station, the problem I explain to you not perform. |
| 19:39.25 | me|ong | http://pastebin.ca/1625651 |
| 19:39.46 | me|ong | i'm tinkering... why the heck is the auth not sync'ed? |
| 19:40.28 | me|ong | unless the username is the extension also? |
| 19:40.56 | peterthesing | funny thing this * c, c# perl java php connect directly * doesn't |
| 19:47.27 | [TK]D-Fender | peterthesing: "if"? Are you guessing about problems you might run into or do you have something to show us? |
| 19:48.07 | jblack | [TK]D-Fender: Yo rock! |
| 19:48.13 | jblack | You too. :) |
| 19:48.22 | peterthesing | coming up |
| 19:49.56 | peterthesing | i don't get it all day i have been tinkering getting to connect to mysql and now it is connected |
| 19:50.18 | peterthesing | if i have trolled anyone then i am deeply sorry |
| 19:50.38 | [TK]D-Fender | peterthesing: Look at your bank balance... it might double ;) |
| 19:50.57 | peterthesing | ok thanks!!!! |
| 19:51.54 | peterthesing | signing off |
| 20:05.24 | carrar | Hi!! |
| 20:05.25 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:09.54 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:13.33 | ChannelZ | HEY! |
| 20:14.30 | carrar | very rainy here in Seattle today |
| 20:15.51 | mbrevda | TJNII: needed insecure=port,invite in all three trunks, now it works. thanks for trying |
| 20:21.22 | me|ong | tk |
| 20:21.33 | me|ong | who wrote the startup script for asterisk? |
| 20:21.52 | TJNII | mbrevda: Glad you figured it out |
| 20:25.40 | ChannelZ | it says in the header I thought |
| 20:36.09 | ZenBSDi | Hey guys.. is it possible for asterisk to be playing a sound file and dialing at the same time? |
| 20:36.28 | ZenBSDi | "Please hold an agent will" .. *hello* .. get that effect ya know? |
| 20:38.58 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 20:44.05 | ZenBSDi | nevermind.. it's the m class ... |
| 20:44.43 | *** join/#asterisk RobH_ (n=RobH@216.38.133.254) |
| 20:45.15 | RobH_ | Does anyone happen to have any experience powering polycom 501 series phones off netgear poe switches? |
| 20:46.05 | *** join/#asterisk traxx (n=traxx@91-64-130-76-dynip.superkabel.de) |
| 20:48.48 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 20:49.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:50.02 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 20:50.55 | *** part/#asterisk perlmonkee (n=perlmonk@anthony.kepler.net) |
| 20:51.27 | ZenBSDi | RobH, I do .. whats up? |
| 20:51.54 | ZenBSDi | not netgear POE though.. we have linksys switches with POE |
| 20:52.01 | ZenBSDi | works like a charm.. |
| 20:54.22 | RobH | I have the 501 plugged directly into the POE netgear with this cable |
| 20:54.25 | RobH | http://tinyurl.com/yk52gby |
| 20:54.38 | RobH | but it doesnt appear to power up the phone |
| 20:54.51 | RobH | i have two waps, and a linksys phone powered from the same switch |
| 20:54.59 | RobH | I am just wondering what I am doing wrong. |
| 20:55.51 | *** join/#asterisk fofware (n=chatzill@host156.190-225-62.telecom.net.ar) |
| 20:56.37 | RobH | ZenBSDi: So other than the netgear switch, the 501 with the cable I linked should be all I need correct? |
| 20:56.57 | ZenBSDi | Thats odd if it's not working.. |
| 20:57.18 | ZenBSDi | but that cable you're using .. |
| 20:58.42 | RobH | I have two kinds of cable. The large one with the inline transformer for PoE, as well as the standard one with nothing but an AC port on it. (If I understand correctly, the second cable with only the AC port isn't good enough to filter PoE for the 501 |
| 21:00.33 | *** join/#asterisk implicitlyrics (n=implicit@mar92-9-82-237-74-58.fbx.proxad.net) |
| 21:03.04 | ZenBSDi | I remember something about having to take into account how many watts those ports put through |
| 21:03.19 | ZenBSDi | You'll need to find out how many watts the netgear is putting out .. it might not be enough |
| 21:06.49 | RobH | Yea, it says the netgear is ieee 802.3af compliant |
| 21:06.56 | RobH | and that is what the polycom 501 requires with that cable |
| 21:07.22 | RobH | its capable of putting out 195 Watts |
| 21:07.35 | RobH | up to 15.4W per port. |
| 21:07.46 | ZenBSDi | Ok, well then .. I'm stumped too :p |
| 21:07.55 | *** join/#asterisk implicitlyrics (n=implicit@82.237.74.58) |
| 21:07.58 | ZenBSDi | it should work |
| 21:08.09 | RobH | Indeed, how frustrating. |
| 21:08.14 | RobH | >_< |
| 21:08.24 | RobH | atleast it seems that it should work to more than just me =] |
| 21:08.31 | RobH | <= feels less crazy. |
| 21:08.57 | TJNII | You don't have to enable PoE on a per-port basis or anything like that do you? |
| 21:09.01 | ZenBSDi | you have the correct converter cable.. you have a 15.4W per port POE netgear 802.3af compliant switch .. |
| 21:09.26 | TJNII | (Though I'll admint I am guessing. I can't afford PoE stuff) |
| 21:09.48 | ZenBSDi | TJNII, POE switches are POE switches without having to enter a manager and enable it |
| 21:09.51 | RobH | TJNII: out of the box it enables PoE to any port that it senses requires it |
| 21:10.08 | ZenBSDi | RobH, and just for shoots and grins.. you went through all your ports? |
| 21:10.12 | RobH | if i wanted a different behaviour, I would have to enable them to NOT give PoE to folks |
| 21:10.13 | *** join/#asterisk blake (n=blake@rupert.ekalb.net) |
| 21:10.40 | RobH | I plugged into multiple ports. the bad part is i have a linksys phone which has PoE internally regulated (unlike polycom 51) and i plug it into the port the polycom was on and it works great |
| 21:11.11 | RobH | so yep, tried multiple ports, two different PoE switches, and multiple polycom cables. |
| 21:11.22 | ZenBSDi | The 501 just won't turn on huh |
| 21:11.23 | Chainsaw | There are two ways of doing PoE though. The right way and the Cisco way. Which does the Polycom expect, and what is the Netgear configured for? |
| 21:11.51 | RobH | Chainsaw: interesting. |
| 21:12.04 | RobH | well, it is the 802.3af standard, which is what the polycom says it requires |
| 21:12.18 | ZenBSDi | RobH, does the switch have a management interface? |
| 21:12.23 | RobH | yep |
| 21:12.33 | ZenBSDi | Did you look around through it? |
| 21:12.47 | Chainsaw | You should find two PoE modes. Try to flip the setting. |
| 21:15.46 | [TK]D-Fender | IP 501 is *not* native PoE <- |
| 21:17.15 | RobH | [TK]D-Fender: Yep, I am using the power cable http://tinyurl.com/yk52gby |
| 21:18.19 | RobH | I am not seeing a setting for toggling what mode the PoE is in |
| 21:18.22 | RobH | other than on/off |
| 21:18.38 | [TK]D-Fender | RobH: Only supports 802.3af IIRC |
| 21:18.57 | RobH | The polycom only supports 802.3af you mean? |
| 21:19.07 | RobH | That is what the poe switch should be using and the phone should expect |
| 21:19.49 | RobH | is stumped, still looking through it though |
| 21:20.09 | [TK]D-Fender | RobH: What model of switch? |
| 21:20.23 | RobH | netgear fs728tp |
| 21:20.23 | *** join/#asterisk netpro25__ (n=alvarado@c-68-62-234-14.hsd1.fl.comcast.net) |
| 21:20.24 | [TK]D-Fender | RobH: Most Netgears only power HALF of the ports |
| 21:20.37 | RobH | I only have 5 devices plugged into it now though. |
| 21:20.48 | RobH | and I have 3 of them to spread out power if I reach capacity on any single one. |
| 21:21.08 | [TK]D-Fender | RobH: Do you have other phones working on it now? |
| 21:21.16 | RobH | I have a linksys phone working on it yes |
| 21:21.25 | RobH | linksys spa942 |
| 21:21.39 | [TK]D-Fender | RobH: Ok, that jsut about settles that... |
| 21:21.41 | RobH | as well as a couple of WAPs, so the PoE delivers power to them just fine |
| 21:21.56 | [TK]D-Fender | RobH: Perhaps the adapter cable is dead. it works on the wall-wart cable, right? |
| 21:22.31 | RobH | [TK]D-Fender: I have tried two brand new adapter cables, fresh out of their boxes. I tried plugging them directly into the switch to ensure it was not faulty cabling in the walls. |
| 21:22.52 | RobH | Neither one works, when I plug it in, the green LED on the adapter cable illuminates for a second, then goes off |
| 21:23.02 | RobH | not even fully illuminating. |
| 21:23.45 | netpro25__ | So I am still fighting with asterisk. Asterisk is has a public IP while my client is behind a NAT. I am now on a different network and the problem persists so I doubt it's an issue with the router on the client side. Here is my pastebin: http://pastebin.com/m142238f1 |
| 21:24.39 | netpro25__ | ^ I cannot connect to asterisk from behind a NAT that is |
| 21:25.25 | [TK]D-Fender | netpro25_Unless you have a firewall on the client side its the client that has an issue, or part of the link is filtering you |
| 21:26.50 | netpro25__ | [TK]D-Fender: just realized server has IPtables how should I ensure it is not an issue |
| 21:27.14 | *** join/#asterisk implicitlyrics (n=implicit@mar92-9-82-237-74-58.fbx.proxad.net) |
| 21:27.59 | *** join/#asterisk came0 (n=came0@93.147.121.70.cfl.res.rr.com) |
| 21:28.28 | netpro25__ | [TK]D-Fender: I have Accept UDP 5060 and 100001:20000 |
| 21:32.07 | [TK]D-Fender | netpro25_Trash it completely and test |
| 21:32.16 | netpro25__ | [TK]D-Fender: yea did not help |
| 21:32.38 | [TK]D-Fender | netpro25_Then either an ISP is filtering, or the remote end is screwing up. |
| 21:33.19 | carrar | 100001:20000? |
| 21:33.25 | carrar | isn't that backwares |
| 21:33.43 | [TK]D-Fender | And should be 10000, not 10001 |
| 21:33.44 | netpro25__ | extra zero |
| 21:33.45 | carrar | yeah |
| 21:33.47 | netpro25__ | type |
| 21:33.49 | netpro25__ | o |
| 21:33.52 | carrar | invaid number |
| 21:34.18 | netpro25__ | Is it possible it is a bug with 1.4.21 |
| 21:34.25 | netpro25__ | I am using an old ass version |
| 21:35.04 | [TK]D-Fender | netpro25_No |
| 21:43.38 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 21:46.04 | *** part/#asterisk came0 (n=came0@93.147.121.70.cfl.res.rr.com) |
| 21:49.58 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 21:51.23 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 22:16.07 | netpro25__ | So I have a better pastebin if anyone has a sec |
| 22:16.11 | netpro25__ | http://pastebin.com/m41395bf2 |
| 22:16.25 | netpro25__ | It shows the client and the server debugs |
| 22:18.24 | netpro25__ | For some reason the packets are hitting the router and it is just dropping them because it does not know where to send them |
| 22:18.32 | netpro25__ | at least that is what I think |
| 22:18.55 | [TK]D-Fender | netpro25_ * isn't getting those register packets... |
| 22:19.26 | [TK]D-Fender | netpro25_ : the problem is it isn't RECEIVING |
| 22:19.31 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:19.38 | netpro25__ | should the packets coming from asterisk have the internal IP address of the device |
| 22:19.48 | [TK]D-Fender | netpro25_: forget about worrying where to respond to... it doesn't even have anything to respond to. |
| 22:19.57 | netpro25__ | [TK]D-Fender: right |
| 22:20.07 | [TK]D-Fender | netpro25_: those options packets are irrelevant |
| 22:20.26 | [TK]D-Fender | netpro25_:ignore them. You have a clear issue of packets not making it from the client to * |
| 22:20.44 | [TK]D-Fender | netpro25_: so either an ISP is filtering, or your client equipment is screwed up |
| 22:20.46 | netpro25__ | [TK]D-Fender: right and I can forward and it works but it should work without forwarding |
| 22:20.51 | [TK]D-Fender | netpro25_: there is no "but" about this. |
| 22:21.08 | [TK]D-Fender | netpro25_Forwarding has NOTHING to do with a packet making it to * |
| 22:21.29 | netpro25__ | I mean on the client side |
| 22:21.40 | [TK]D-Fender | netpro25_: If there is any impact I'd start blaming the router on their side immediately |
| 22:22.30 | netpro25__ | I have had the same issue everywhere I have tried to connect which is about 4-5 different place |
| 22:23.11 | netpro25__ | not to mention that I can establish SIP connections with other providers on these connections no problem |
| 22:23.19 | *** join/#asterisk came0 (n=came0@70.121.147.93) |
| 22:23.45 | netpro25__ | I gtg, I will look at it later |
| 22:24.30 | *** join/#asterisk naS_ (i=417a0fa9@gateway/web/freenode/x-onleqbrsqhnfkegr) |
| 22:31.10 | *** part/#asterisk came0 (n=came0@70.121.147.93) |
| 22:39.31 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 22:58.45 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 23:14.07 | p3nguin | Looks like the ringtones on the Cisco 7912G phone cannot be changed. Can anyone confirm this? |
| 23:16.28 | *** join/#asterisk raden_work (n=raden@66-168-15-19.dhcp.stpt.wi.charter.com) |
| 23:30.35 | *** join/#asterisk came0 (n=came0@93.147.121.70.cfl.res.rr.com) |
| 23:32.57 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 23:34.18 | *** join/#asterisk naS__ (i=417a0fa9@gateway/web/freenode/x-uxxeqjllnmyevunq) |
| 23:39.07 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 23:44.33 | p3nguin | Correction. It looks like the ringtones on the 7912G cannot be changed when using the SIP image, but could be if I used skinny and callmanager. |
| 23:46.18 | Chainsaw | p3nguin: But then you'd have to work with the rather limited chan_skinny or the wholly undocumented chan_sccp2. |
| 23:46.39 | p3nguin | I would rather use SIP and keep the silly ringtone. |
| 23:46.43 | Chainsaw | p3nguin: Neither filled me with joy when I looked at them. (I'm on 7960G handsets, but the XML support is rather... neutered on the SIP load) |
| 23:48.06 | p3nguin | I love the ease of the 7940G's configuration and capability on SIP. |
| 23:48.33 | p3nguin | It's just the 7905/7912 that is REALLY limited. |
| 23:48.37 | Chainsaw | Until you want to keep state between XML pages. |
| 23:48.47 | Chainsaw | Or configure soft keys. Or do anything remotely exciting. |
| 23:48.56 | p3nguin | I don't know what I would need any of that for. |
| 23:49.00 | Chainsaw | You'll read about it, and it won't work. Oh sorry, not on SIP. |
| 23:49.29 | p3nguin | All the soft keys do what they are supposed to do. And there's no browser, so... |
| 23:49.45 | p3nguin | For XML stuff, I change it on the server and let the phone pull it from there. |
| 23:52.08 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |