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| 00:07.13 | LemensTS | anyone have problems on polycom phones writing logs to apache directory? Im using webdav on a directory, and it writes the app.log to it, but not the boot.log, i tail -f the apache log and dont see anything different about the boot.log then the app.log |
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| 00:41.20 | riddlebox | has anyone ever heard of Xblue Networks? My boss has me using one of their systems at home to see if I like it so we can start selling it? |
| 00:41.34 | riddlebox | I didnt actually mean that second part as a question, sorry |
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| 01:33.26 | jamko | Good evening. I added a 2nd ip address to my nic, to allow for communication with mysql on the lan. I specified asterisk to only bind to the original ip address, which is a static public address. Ever since doing this, I have been experiencing NAT type issues, one way audio mainly. |
| 01:35.28 | jamko | sip debug shows communication strictly with the public side, however it is randomly showing an alternate wan ip address in some of the "received" .. This WAN ip is one of mine, but should not be showing up in the context that it's in. |
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| 01:43.01 | jamko | I am convinced asterisk is sending traffic out the private side, because it is receiving some of the traffic on the public ip that is assigned to my router, and not asterisk. |
| 01:54.21 | jamko | yes that was it. I specified a gateway on the second address. stupid. |
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| 03:01.36 | dzup2 | hello, i need a review in how is Digium TE410P Quad T1 4E1 PCI ECO ? |
| 03:05.29 | *** join/#asterisk Besticles (~Besticles@ip68-104-111-21.lv.lv.cox.net) |
| 03:07.27 | Besticles | I got a new Digium Card today, TE420 /w Echo Cancelling. DAHDI Tool reports that the T1 I plugged in and configured is OK, and Asterisk sees the channels I setup. But when I originate, AMI comes back with Status: Error. It's not giving me enough information to diagnose. Any suggestions? |
| 03:08.46 | pabelanger | Besticles: Use the CLI command Originate and see what happens |
| 03:15.09 | Besticles | I sent the command via CLI, and after 20 / 30 seconds it comes back with -- Hungup 'DAHDI/1-1'. The only feedback. |
| 03:15.22 | Besticles | But didn't dial my cell. |
| 03:15.57 | Besticles | Wait wait wait, I'm a moron. |
| 03:16.01 | Besticles | Thanks for the help. |
| 03:16.31 | Besticles | Can't believe I was that dumb. |
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| 03:27.22 | Besticles | Nevermind, I'm still having the same symptom. Can't think of anything else that I might have forgotten to setup. |
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| 04:17.47 | recluze | hello everyone |
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| 04:21.22 | recluze | I'm having a little trouble. I'm trying to send a fax through a php script. I can place a callfile in the spool directory and call the fax-tx macro ... I have no idea though how to tell fax-tx about the target fax machine number. Any hints? |
| 04:21.56 | recluze | (I'm using digium's FFA and I've already gone through the manual and the forums... if there's a link, that would be appreciated too.) |
| 04:22.55 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:28.39 | ChannelZ | what fax-tx macro? |
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| 04:38.15 | recluze | ChannelZ: the one given in the digiumn FFA manual ... |
| 04:38.21 | recluze | (sorry, got dced) |
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| 04:54.33 | ChannelZ | hmm well what I see is not a macro but just a context with extensions.. in which case you'd dial the number in your call file via Channel: and have it run the Extension: s in the Context: fax-tx starting at Priority: 1 |
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| 05:41.22 | pabs3 | is there a way to list all calls to a specific context & extension |
| 05:41.24 | pabs3 | ? |
| 05:41.59 | pabs3 | sip show channel doesn't seem to list the context/extension |
| 05:45.52 | ChannelZ | core show channels |
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| 05:53.27 | pabs3 | hmm, that seems to truncate the channel name |
| 05:54.35 | pabs3 | I guess I need an XML-RPC or similar interface to that |
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| 05:56.28 | ChannelZ | core show channels concise will show the whole name |
| 05:56.57 | ChannelZ | and it's a little more parseable |
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| 05:58.15 | pabs3 | excellent |
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| 06:28.35 | mahiti-irc | hi |
| 06:29.12 | mahiti-irc | how do u configure dahdi for TE122P pri card for 32 channels? |
| 06:29.59 | mahiti-irc | which echo canceller shud i choose? |
| 06:38.21 | mahiti-irc | anyone here |
| 06:38.34 | mahiti-irc | echo... |
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| 06:46.19 | mahiti-irc | somone can help on dahdi here? |
| 06:46.32 | mahiti-irc | or is there a separate irc for it |
| 06:46.36 | mahiti-irc | ??? |
| 06:47.42 | [netman] | I think it's here |
| 06:48.16 | mahiti-irc | oh, can you help me man? |
| 06:48.57 | kaldemar | mahiti-irc: have you tried to configure it yet? |
| 06:49.16 | mahiti-irc | nope |
| 06:49.25 | mahiti-irc | i cant find a doc for it |
| 06:50.10 | kaldemar | have you installed dahdi? |
| 06:50.19 | mahiti-irc | yup |
| 06:50.50 | mahiti-irc | dahdi_scan and dahdi_hardware show me my pri card all wel |
| 06:51.42 | mahiti-irc | i think my next step will configuring a echo canceller in system.conf ! |
| 06:51.50 | mahiti-irc | but which shud i choose |
| 06:52.16 | mahiti-irc | mg2, kb1,sec...? |
| 06:53.02 | coppice | the only good free EC is OSLEC |
| 06:54.10 | *** join/#asterisk mr_ian (~mr_ian@96.54.2.132) |
| 06:54.13 | kaldemar | there you go. for the mean time, choose mg2 for example to get the forward with the setup. you can install oslec afterwards if you notice echo problems. |
| 06:56.19 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 06:57.07 | shamelessn00b | mg2 |
| 06:57.49 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 06:58.40 | coppice | someone should clean that stuff up. MG2 isn't good, but the other options it gives you are completely useless |
| 06:59.49 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 06:59.53 | schmidts | good morning |
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| 07:06.32 | mahiti-irc | ok will check that |
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| 07:13.13 | shamelessn00b | sangoma? |
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| 07:46.49 | hurdman | anyone is working with unimrcp here ? |
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| 10:13.45 | dandate2 | anyone know what the difference is between cisco spa8000 and iad2430 series ? |
| 10:15.26 | schmidts | i havent looked but it could be the linksys cisco change thing |
| 10:15.52 | dandate2 | an 8 port spa8000 runs $200 on ebay, the iad2430 runs $800 ?? |
| 10:16.05 | dandate2 | i looked at the specs and it seems that the iad2430 has one extra wan port... |
| 10:16.09 | schmidts | maybe the spa8000 was one of the linksys products and the iad2430 is the "new" cisco product which is just the same but more expensive :D |
| 10:17.22 | dandate2 | iad2430 also requires a rackmount |
| 10:17.34 | schmidts | have a look at spa942 and spa504 there you would find not so a big difference but these phones are nearly the same |
| 10:17.39 | dandate2 | do we seriously get more quality out of this in an industrial environment |
| 10:18.06 | schmidts | maybe the spa8000 is offical EOL so you will get no updates or support on this |
| 10:18.52 | coppice | the IAD2430 offers more scalable revenue for cisco |
| 10:18.59 | dandate2 | lol |
| 10:19.10 | dandate2 | its a damn 1-u server |
| 10:25.05 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 10:27.55 | c0rnoTa | Hello all |
| 10:28.12 | c0rnoTa | My problem with call drops on PRI, followed by 'FRAME_CONTROL (8)' was successfully solved. The issue was in callprogress=yes option in chan_dahdi.conf. It seems that I wrong when I wrote configuration file and confused it with usecallingpres option. |
| 10:28.15 | dandate2 | is there a cheap 16 port router out there? |
| 10:28.34 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 10:35.28 | shamelessn00b | My asterisk is crashing at ~125 calls, logs show this message |
| 10:35.30 | shamelessn00b | kernel: [1346981.172486] asterisk[4253]: segfault at 0 ip 7f1d67c1bfe2 sp 440cda88 error 4 in libc-2.7.so[7f1d67b9f000+14a000] |
| 10:35.39 | shamelessn00b | Using asterisk 1.6.2.6 |
| 10:35.53 | shamelessn00b | kernel version 2.6.26-2-amd64 |
| 10:35.59 | dandate2 | i guess what i should ask is, what is the difference between an ATA and voice gateway |
| 10:37.05 | schmidts | shamelessn00b please try a newer version like 1.6.2.13 and if you can reproduce this problem open an issue in the tracker (issues.asterisk.org) |
| 10:37.26 | shamelessn00b | ok |
| 10:37.38 | schmidts | shamelessn00b and also have a look at your doc/backtrace.txt file how to get the needed debug information |
| 10:38.03 | shamelessn00b | ok sure |
| 10:38.10 | shamelessn00b | thanks :) |
| 10:38.12 | schmidts | thanks ;) |
| 10:38.17 | schmidts | np |
| 10:51.11 | *** join/#asterisk Ouchi (~kvirc@200.150.66.162) |
| 10:51.22 | kaldemar | dandate2: that's just terminology. |
| 10:54.51 | *** join/#asterisk petern_ (~petern@lachesis.fuzzle.org) |
| 10:55.36 | petern_ | Okay, so I got sphinx2 working... except it doesn't :( |
| 10:57.04 | *** join/#asterisk tomodachi (~mateuszm@triton.dsv.su.se) |
| 10:57.18 | tomodachi | hi ive noticed that my metmee conferences sometimes stay open |
| 10:57.22 | tomodachi | eventhough all users have left |
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| 11:13.30 | shamelessn00b | tomodachi: use confbridge |
| 11:13.55 | tomodachi | shamelessn00b: whats if confbridge? |
| 11:14.00 | tomodachi | doesent seem to be a linuxbinary |
| 11:14.09 | tomodachi | or a command in the asterisk console (im using 1.2) |
| 11:14.23 | shamelessn00b | lol, idk if it would be available in 1.2 |
| 11:16.45 | shamelessn00b | ok its not |
| 11:16.59 | tomodachi | nope ... |
| 11:17.07 | tomodachi | i guess uppgrading would solve (since it seems to be a bug) |
| 11:17.08 | shamelessn00b | its for 1.6.2.* |
| 11:17.11 | tomodachi | yeah.. |
| 11:17.21 | tomodachi | but upgrading would be a VERy bothersome process right now .. |
| 11:17.22 | tomodachi | :) |
| 11:17.36 | shamelessn00b | yeah, you will also need to change your dialplan |
| 11:17.44 | shamelessn00b | there are syntax variances |
| 11:17.51 | tomodachi | mm |
| 11:17.52 | shamelessn00b | for some applications |
| 11:21.20 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:21.20 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:24.15 | shamelessn00b | \o/ |
| 11:24.48 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:24.52 | tomodachi | shamelessn00b: hmm any other idea of whay i mighty try? |
| 11:25.02 | tomodachi | kicking the last member of the conference does nothing |
| 11:25.08 | tomodachi | its like a zombie conference user.. |
| 11:25.13 | tomodachi | just doesent die when you kick it |
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| 11:27.01 | shamelessn00b | 'x' â close the conference when last marked user exits |
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| 12:05.05 | *** join/#asterisk mahiti-irc (~mahiti@122.166.127.74) |
| 12:05.10 | mahiti-irc | hi |
| 12:05.19 | mahiti-irc | i installed asterisk 1.4 and dahdi |
| 12:05.43 | mahiti-irc | in centos and dahdi_cfg -vvv is showing the output well |
| 12:05.59 | mahiti-irc | but i am not able to figure which trunk shud i be using |
| 12:06.09 | mahiti-irc | how do i find that? |
| 12:06.36 | mahiti-irc | am calling other telephones using .call file |
| 12:06.54 | mahiti-irc | in /var/spool/asterisk/ |
| 12:08.23 | mahiti-irc | guys |
| 12:09.32 | russellb | have you configured /etc/asterisk/chan_dahdi.conf ? |
| 12:10.09 | mahiti-irc | yup |
| 12:10.26 | russellb | well you call whatever you configured in there, heh |
| 12:10.49 | mahiti-irc | oh |
| 12:10.55 | mahiti-irc | did u mean group=0 ? |
| 12:11.04 | russellb | well i don't know what you want, but sure |
| 12:11.07 | russellb | DAHDI/g0 |
| 12:11.18 | mahiti-irc | k right |
| 12:11.25 | mahiti-irc | i will chk that and brb |
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| 12:41.30 | mahiti-irc | thanks russellb its working classic |
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| 12:47.27 | *** join/#asterisk fprior (~chatzilla@200.43.69.102) |
| 12:55.54 | fprior | Hi all. my trouble: when a PC with SIP client (ex. Zoiper) shutdown unexpected (caused by blackout), on Asterisk server "sip show peers" continue showing agent as "connected" for some minutes. Is possible set this refreshing timeout ? Thanks |
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| 13:00.44 | Naikrovek | fprior: do you have qualify turned on for that extension? |
| 13:01.07 | Naikrovek | i don't now how often it checks, but i'm sure it's shorter than "some minutes" between checks. |
| 13:01.42 | Naikrovek | qualify=yes will make the peer "unreachable" if it can't respond to SIP messages in 2 seconds or less |
| 13:01.44 | *** join/#asterisk oryxtec (~test@116.71.176.64) |
| 13:02.10 | oryxtec | in order to uninstall asterisk i only need to delete these files and folders? |
| 13:02.37 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 13:02.42 | kaldemar | it will check with 60 second frequency unless otherwise defined with qualifyfreq in sip.conf. |
| 13:02.53 | oryxtec | ---/etc/asterisk----------/etc/zaptel.conf-------- /var/log/asterisk--------/var/lib/asterisk------/var/spool/asterisk----/usr/lib/asterisk |
| 13:02.57 | oryxtec | thats all? |
| 13:03.11 | oryxtec | and asterisk will get completly uninstalled? |
| 13:03.24 | kaldemar | oryxtec: zaptel.conf is not a part of asterisk, but zaptel. how did you install? |
| 13:03.56 | oryxtec | i donwload asterisk package from asterisk.org web site and installed it |
| 13:04.08 | oryxtec | with dahdi... |
| 13:04.17 | oryxtec | i wrote zaptel by mistake |
| 13:04.22 | kaldemar | what kind of a package? source package? .rpm, .deb? |
| 13:04.29 | oryxtec | source |
| 13:04.55 | kaldemar | use the uninstall target in the makefile. i.e. give command "make uninstall" in asterisk directory. |
| 13:05.04 | kaldemar | in the source directory, that is. |
| 13:05.09 | oryxtec | ooh |
| 13:05.12 | oryxtec | let me try |
| 13:05.41 | oryxtec | by the way i have already deleted these dir |
| 13:05.52 | oryxtec | which i just wrote |
| 13:06.01 | oryxtec | it wont make any differnce |
| 13:06.02 | oryxtec | rite |
| 13:06.15 | kaldemar | right. |
| 13:06.31 | fprior | Naikrovek: thanks for your info, http://tinyurl.com/cjobt say all: "qualify=xxx|no|yes ", check every 2 seconds or xxx milliseconds |
| 13:06.56 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 13:07.06 | [TK]D-Fender | that is not a check FREQUENCY |
| 13:07.10 | Naikrovek | right |
| 13:07.30 | Naikrovek | i think it's 1 minute check frequency, what you specify with the qualify parameter is the timeout |
| 13:07.47 | Naikrovek | the maximum time to wait before considering the phone to be offline |
| 13:07.58 | Naikrovek | 2000 is entirely reasonable, so just set it to yes |
| 13:08.56 | oryxtec | guys which ver of asterisk 1.4 is more stayable? |
| 13:08.58 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:10.25 | fprior | [TK]D-Fender: "Asterisk will send a SIP OPTIONS command regularly to check that the device is still online", why do you say is not FREQUENCY ? |
| 13:11.33 | kaldemar | oryxtec: start with the latest. |
| 13:11.39 | [TK]D-Fender | fprior: Because it isn't |
| 13:11.51 | [TK]D-Fender | oryxtec: the latest obviously |
| 13:12.00 | leifmadsen | fprior: I don't understand your question |
| 13:12.29 | Naikrovek | fprior: the qualify parameter does not specify how OFTEN Asterisk checks for phone reachability |
| 13:12.45 | [TK]D-Fender | fprior: That is how long * will wait for a RESPONSE to the OPTIONS packet, not the frequency at which it will SEND them. |
| 13:12.47 | Naikrovek | qualify determines what is considered an unreachable phone |
| 13:12.47 | kaldemar | fprior: the page you linked to does not say that asterisk would check every 2 seconds. |
| 13:12.56 | Naikrovek | how is this hard to understand? |
| 13:13.05 | *** join/#asterisk Jareeta (~ahailes@WHITE-METEO.MIT.EDU) |
| 13:13.12 | Naikrovek | i'm not a sharp man, but this is pretty simple |
| 13:13.13 | Naikrovek | to me |
| 13:13.27 | leifmadsen | cuts his hand on Naikrovek |
| 13:13.32 | adyn | I am trying to undertand the way trunking works, I have two SIP trunks into an Asterisk PBX, is it possible to route outbound calls through either one of them regardless of who actually handles the incoming DID's? |
| 13:13.36 | Naikrovek | heh |
| 13:13.42 | Naikrovek | thanks, leifmadsen |
| 13:13.48 | leifmadsen | adyn: you mean SIP connections. There are no SIP trunks. |
| 13:13.55 | [TK]D-Fender | adyn: You can do whatevery you want with every call going to your server |
| 13:13.57 | adyn | ok... SIP connections |
| 13:14.05 | leifmadsen | adyn: and yes, just send the call to whoever you want.... |
| 13:14.07 | leifmadsen | it's just dialplan |
| 13:14.08 | [TK]D-Fender | adyn: And tehre is no "conenction". There are only "calls" |
| 13:14.19 | [TK]D-Fender | gah, can't type today |
| 13:14.40 | Naikrovek | it's okay, none of us can type most days |
| 13:14.44 | adyn | I'm more worried about CID, so I send a call out SIP A and it would show the same CID as SIP B |
| 13:15.00 | leifmadsen | adyn: that is a function of the SIP provider allowing you to pass CID information. Ask them. |
| 13:15.05 | adyn | ok thank you |
| 13:15.09 | leifmadsen | Asterisk will allow you to do it, but the provider probably won't. |
| 13:15.18 | leifmadsen | it's a liability concern |
| 13:15.37 | [TK]D-Fender | adyn: You'll send whatever you choose to. Your provider will accept whatever they choose to. |
| 13:15.59 | adyn | yeah we route emergency calls through the DID provider but I'm having call quality issues and we believe its the provider so want to try sending the calls through another provider. |
| 13:16.55 | [TK]D-Fender | adyn: Its your dialplan, do whatever you want |
| 13:17.24 | adyn | yup, got that. I just didn't understand the technology to know if it was possible to do once it leaves my system. Thanks for the info. |
| 13:17.57 | [TK]D-Fender | adyn: First, it never leaves your system |
| 13:18.17 | [TK]D-Fender | adyn: Call comes in. Your dialplan does what you set it up to do. * sits in the MIDDLE |
| 13:18.17 | fprior | yes, I understand now, "qualifyfreq" is frecuency to send OPTIONS command. |
| 13:18.33 | adyn | ok maybe I'm not phrasing it correctly but I'm talking about |
| 13:18.40 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:18.47 | adyn | erg maybe I'm not asking the question correctly |
| 13:20.03 | adyn | I think you answered the question though, so thank you. I'll have to experiment :) |
| 13:20.37 | fprior | Naikrovek: sorry but, if something is simple for you not mean is simple for everyone. |
| 13:30.07 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:30.36 | Naikrovek | fprior: yeah i guess. |
| 13:30.53 | Naikrovek | i just consider myself to be borderline retarded, is all |
| 13:31.19 | Naikrovek | or, itellectually challenged, if the R word is offensive to you |
| 13:31.27 | Naikrovek | intellectually* |
| 13:31.47 | [TK]D-Fender | Naikrovek: What do you mean ... "borderline"? ;) |
| 13:32.10 | Naikrovek | well if i say "i'm stupid" people will be like "aw don't be hard on yourself" and crap like that |
| 13:32.12 | *** join/#asterisk elzid (~IceChat7@host81-143-42-174.in-addr.btopenworld.com) |
| 13:32.51 | Naikrovek | brb |
| 13:32.58 | elzid | hello all,from phpagi anyway I can play audio whilst I call a function and wait for response? |
| 13:33.15 | [TK]D-Fender | Naikrovek: j/k ... actually from what I've seen from here you very far from "stupid". May not have the most technical knowledge but your mind appears to be one of the most open around... |
| 13:34.23 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:36.34 | elzid | I need to make a curl hit out to a 3rd party which takes up to 20secs to return a value, can I play audio in the meantime? |
| 13:40.26 | fprior | Naikrovek: I'm not offended, my first msg for you today was "thanks for info"; only I'm not agree with your comment "how is this hard to understand?" to a user that ask a question. It's all, I won't start discussion, really |
| 13:41.43 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 13:52.24 | oryxtec | please help... http://pastebin.com/eRyX0wmp |
| 13:52.34 | oryxtec | when i do outbound call |
| 13:53.08 | oryxtec | on asterisk it say trying but on other side call is getting connect... |
| 13:53.23 | oryxtec | on asterisk cli i can see premission denied error msgs. |
| 13:53.53 | oryxtec | plz help. |
| 13:54.58 | kaldemar | oryxtec: what user is running asterisk? |
| 13:55.19 | oryxtec | i guss root |
| 13:55.47 | *** join/#asterisk recluze (~recluze@119.153.84.242) |
| 13:56.53 | recluze | hello all. |
| 13:57.27 | recluze | I'm trying to send fax through digium's FFA and having a final hicup in the setup. I can receive faxes alright and have setup a script for sending faxes through call files and a context. |
| 13:57.52 | recluze | When I send the fax, I get an error saying "failed to start FAX session". |
| 13:57.57 | recluze | See the whole output here: http://pastebin.com/QuXq8wUJ |
| 13:58.02 | kaldemar | oryxtec: don't guess, check. ps u -C asterisk |
| 13:58.21 | recluze | Can anyone provide any hints please. I've googled the heck out of my brain for this. |
| 13:58.39 | [TK]D-Fender | oryxtec: g: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied <--- double //'s |
| 13:58.41 | recluze | (If there's a way to get more debug info, that would be much appreciated too.) |
| 13:59.00 | oryxtec | http://pastebin.com/BJVJdqkA |
| 13:59.06 | [TK]D-Fender | oryxtec: And "guess" isn't good enough. Go find out |
| 14:00.17 | schmidts | recluze you do a hangup in your macro maybe thats why |
| 14:00.38 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 14:00.41 | kaldemar | oryxtec: your asterisk user has no privileges to read those files. |
| 14:01.00 | oryxtec | rite.. how can i give permission? |
| 14:01.07 | recluze | schmidts: No. I have nothig after SendFAX |
| 14:01.22 | oryxtec | so asterisk can read those files |
| 14:01.47 | recluze | http://pastebin.com/5vWuBAu8 |
| 14:02.07 | recluze | that's the context |
| 14:02.28 | drmessano | oryxtec, http://www.zzee.com/solutions/linux-permissions.shtml |
| 14:02.44 | schmidts | sorry this was just the macro you run in the h extension ;) |
| 14:02.58 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:03.03 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 14:03.08 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:03.16 | recluze | :O |
| 14:03.35 | oryxtec | there is no file |
| 14:03.37 | oryxtec | :S |
| 14:03.42 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:04.38 | drmessano | Asterisk is unable to read nothingness |
| 14:04.45 | drmessano | app_zen is coming in 1.10 or so |
| 14:04.56 | recluze | so... any ideas? |
| 14:05.14 | coppice | Dan Brown readers can read nothingness, why not asterisk? |
| 14:05.23 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:05.58 | [TK]D-Fender | ~asterisk-non-root |
| 14:05.58 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 , and for Debian : http://www.thinkdebian.org/archives/828 |
| 14:06.01 | [TK]D-Fender | oryxtec: ^^^^ |
| 14:06.11 | drmessano | Is that like the monk that walked into the burger joint and told them "Make me one with everything" |
| 14:06.17 | [TK]D-Fender | oryxtec: Go read the book. You are trying to run * as "asterisk", but intalled as root |
| 14:06.21 | oryxtec | let me go through this |
| 14:06.22 | oryxtec | thanks |
| 14:06.53 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:07.00 | drmessano | oxyten, maybe you should read the book in its entirety |
| 14:08.49 | recluze | looks around |
| 14:09.11 | [TK]D-Fender | drmessano: He's runnign a GUI install with a2billing. .... |
| 14:10.43 | *** join/#asterisk csnook (~chris@138.210.3.1) |
| 14:10.55 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 14:11.05 | Naikrovek | that thinkdebian.org link doesn't work anymore |
| 14:11.14 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-qikqzjylrgjvszee) |
| 14:11.16 | Naikrovek | and he (dustybin) moved the site to wizbox.org but now it's down too |
| 14:11.49 | recluze | begins to think he's on mute in the channel |
| 14:11.55 | drmessano | [TK]D-Fender, nevermind the dandruff |
| 14:12.03 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:12.03 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:12.14 | drmessano | recluze, perhaps nobody has an answer for you? |
| 14:12.23 | recluze | :'( |
| 14:12.43 | recluze | but maybe someone knows how to help me get more debug info |
| 14:12.50 | recluze | come on... have some mercy |
| 14:13.09 | recluze | you must remember what it was like when you were an asterisk newbie :) |
| 14:14.37 | drmessano | Yes, I do.. and the more I yelled "hey guys, what about my question!!!!" when it didn't get answered, the more I was shunned and outcast by the community, my family, and my country |
| 14:15.16 | recluze | lol |
| 14:15.55 | recluze | but since you turned out to be a guru, I guess I should continue on this track ... :D |
| 14:16.14 | recluze | i'm kidding... please don't ridicule me more |
| 14:16.16 | recluze | i will shut up now |
| 14:17.03 | drmessano | recluze: Try this: http://tinyurl.com/3ydtrze |
| 14:17.27 | Naikrovek | wtf lol |
| 14:18.17 | Naikrovek | i love meme generator but i hate URL shorteners |
| 14:18.59 | Naikrovek | recluze: don't worry about it, hang out, lurk, maybe learn smoething, and ask again in a while |
| 14:20.49 | recluze | k ... thanks |
| 14:23.32 | drmessano | http://tinyurl.com/2d27rbb |
| 14:25.20 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 14:25.23 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 14:38.29 | elred_ | Hello, i am using Asterisk and i want IT to send a Re-INVITE when i activate moh (the snom phone yet send it a new invite with "a=sendonly" in the SDP). But Asterisk (1.6.2.5-0ubuntu1) is not forwarding any reinvite. If i activate canreinvite=yes, it just goes wrong since it try to bridge my (inside lan) phone with the outbound trunk :/ On http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite it's told canreinvite=no don't block re-i |
| 14:38.30 | elred_ | nvite of moh. But it seems it in fact does. Any suggestion ? Thx a lot |
| 14:39.21 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 14:43.11 | elred_ | nobody ? :-( |
| 14:48.17 | Naikrovek | not directed at anyone but this will be useful in here i think: http://memegenerator.net/Advice-Dog/ImageMacro/3104593/cant-read-or-use-google-demand-help-from-IRC |
| 14:48.27 | [TK]D-Fender | elred_: reinvites where NAT is involved = DOA |
| 14:48.34 | *** join/#asterisk BANSAL (~bansal@117.199.125.19) |
| 14:49.17 | elred_ | yep |
| 14:49.19 | elred_ | i know |
| 14:49.34 | elred_ | but for some reason i need to test a reinvite to the outbound trunk |
| 14:49.40 | elred_ | moh was just a way to like "trigger one" |
| 14:50.11 | elred_ | if i could ask asterisk to change codec, for instance, or any others stuff can could lead to a reinvite from ASTERISK and the operator (and not the on-lan phone <-> operator), it would be just good for me |
| 14:53.57 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:55.50 | [TK]D-Fender | elred_: not happening |
| 14:56.58 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 14:57.32 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 14:57.33 | [sr] | howdy |
| 14:57.49 | thehar | meep |
| 14:58.04 | [sr] | I have a SMC router now, and SIP connectthis is not really asterisk related |
| 14:59.02 | [sr] | i have a SMC router now, and i use a software from my mobile company to perform connection using the softphone that uses SIP, and now, the persons on the other side don't hear me |
| 14:59.30 | [sr] | forgot to say something, my network is: SMC=>Linux NAT Machine=>internal network |
| 14:59.35 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 14:59.41 | [sr] | on the linux machine i have the sip modules loaded |
| 14:59.52 | [sr] | with the previous router everything worked OK |
| 14:59.59 | *** join/#asterisk zoid_ (~awainer@190.2.14.213) |
| 15:00.09 | [sr] | is there some port that i can forward directly to the linux box? |
| 15:01.17 | zoid_ | Hi, I just installed my tdm400 clone card w/2 fxos. One works, the other is mute, switching the modules in the card gives the same result, is there any way this is not ha hardware issue? |
| 15:03.51 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:06.04 | [TK]D-Fender | zoid_: Is it the module, or the port on the card? |
| 15:07.04 | zoid_ | the module |
| 15:07.21 | zoid_ | swapping the two modules makes the other port work |
| 15:07.41 | [TK]D-Fender | zoid_: then the module is clearly dead |
| 15:07.56 | zoid_ | I'm about to RMA the module, but since this is my first time with asterisk... |
| 15:08.07 | zoid_ | that's what I tought :( |
| 15:08.24 | zoid_ | the weird thing is, it detects the call, answers, but I hear nothing |
| 15:09.56 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:11.19 | zoid_ | thank [TK]D-Fender |
| 15:11.25 | zoid_ | *thanks |
| 15:13.17 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 15:16.01 | *** join/#asterisk csnook (~chris@138.210.3.1) |
| 15:28.26 | ariel_ | anyone know how to test an fxo on a digium board to see what the voltage is on the line? |
| 15:29.58 | drmessano | voltmeter |
| 15:30.16 | ariel_ | drmessano: it would be nice if it was a location system |
| 15:30.27 | drmessano | huh? |
| 15:32.08 | drmessano | It would be nice if it were a duck too |
| 15:32.11 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 15:33.00 | ariel_ | I am trying to find out if there is an actual line plugged in to the port |
| 15:33.18 | ariel_ | and if it's set to reverse polarity |
| 15:37.12 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 15:40.29 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 15:42.24 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:45.30 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 15:45.49 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 15:47.03 | *** join/#asterisk jasonwert-work (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 15:58.42 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:00.17 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:05.44 | *** join/#asterisk jasonwert-work (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 16:11.46 | Naikrovek | remember those mountain goats climbing the dam wall? here are some more pictures |
| 16:11.47 | Naikrovek | http://io9.com/5664476/mountain-goats-scramble-up-a-near+vertical-wall-in-italy/gallery/ |
| 16:14.43 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:15.00 | bmoraca_work | Naikrovek: I half expected them to be building a pile of themselves to get up...you know, stacking themselves up |
| 16:15.13 | Naikrovek | goat pyramid |
| 16:15.30 | bmoraca_work | seeing the pictures, though, it's a very rough, non-vertical wall. i'm not surprised they're able to climb it |
| 16:15.44 | Naikrovek | i'm not either |
| 16:15.47 | Naikrovek | but it's really cool |
| 16:17.53 | bmoraca_work | hrm...Hulu finally coming to the Roku, but it's only Hulu Plus :( |
| 16:18.42 | Naikrovek | how many officially endorsed tv sites now |
| 16:18.43 | Naikrovek | netflix |
| 16:18.45 | Naikrovek | hulu |
| 16:18.47 | Naikrovek | abc.com |
| 16:18.51 | Naikrovek | what else |
| 16:18.54 | Naikrovek | big ones |
| 16:19.37 | Naikrovek | i would almost like to see them join up, but then the price for a subscription will skyrocket, but as it is now, you have to subscribe to two or three to see all the shows you wanna ssee |
| 16:19.39 | Naikrovek | see |
| 16:19.57 | Naikrovek | neither are ideal solutions. |
| 16:20.20 | bmoraca_work | netflix is great for older shows, but they don't have current seasons on it |
| 16:20.28 | bmoraca_work | since I don't have cable, that's what I'm missing |
| 16:20.29 | Naikrovek | i'm A-OK with paying for online TV access, but I don't want to fiddle with 8 accounts and 8 subscription fees. (the number 8 here is an example) |
| 16:20.40 | Naikrovek | hulu basic has new stuff usually |
| 16:21.00 | Naikrovek | but not everything |
| 16:21.05 | bmoraca_work | right, but I can't watch Hulu Basic on my Roku |
| 16:21.11 | Naikrovek | netflix seems to drop TV shows in by batches |
| 16:21.14 | bmoraca_work | only Hulu Plus |
| 16:21.18 | Naikrovek | you'll see a lot of new stuff appear then nothing for a while |
| 16:22.05 | evils_unon | hey guys, I've got an issue with the latest version of asterisk I got off the yum repos (1.6.2.12). When I dial a number there's a 3-4 second delay until asterisk actually processes the call. Anybody have ideas where I'd start with this one? |
| 16:22.27 | Naikrovek | look in /var/log/asterisk/full |
| 16:22.38 | Naikrovek | and then the dialplan to see where the delay is |
| 16:23.07 | evils_unon | I see no such log... is there something I have to enable for that log? |
| 16:23.12 | Naikrovek | ... |
| 16:23.21 | evils_unon | all I have is event_log messages and queue_log |
| 16:23.33 | Naikrovek | are you running asterisk or some out-of-the-box distro like trixbox or something |
| 16:23.33 | evils_unon | and then some cdr files |
| 16:23.50 | evils_unon | asterisk straight from the official yum repos |
| 16:23.52 | Naikrovek | look in /var/log/asterisk |
| 16:23.56 | Naikrovek | is that directory there? |
| 16:23.58 | Naikrovek | okay |
| 16:24.03 | Naikrovek | are you on asterisknow? |
| 16:24.10 | evils_unon | no |
| 16:24.12 | Naikrovek | okay |
| 16:24.14 | Naikrovek | just checking |
| 16:24.22 | evils_unon | just base centos install with straight asterisk |
| 16:24.22 | Naikrovek | is /var/log/asterisk (directory) there? |
| 16:24.24 | evils_unon | yea |
| 16:24.31 | Naikrovek | okay inside that folder is there a 'full' file |
| 16:24.32 | evils_unon | in there is event_log messages and queue_log |
| 16:24.35 | Naikrovek | hm |
| 16:24.35 | Naikrovek | okay |
| 16:24.37 | Naikrovek | then |
| 16:24.44 | Naikrovek | run asterisk -rvvvvvvv |
| 16:24.49 | Naikrovek | as root or the user asterisk runs as |
| 16:25.00 | evils_unon | yea |
| 16:25.04 | Naikrovek | the amount of 'v's you need will vary |
| 16:25.08 | Naikrovek | okay |
| 16:25.13 | Naikrovek | make a call that causes the delay and watch |
| 16:25.22 | Naikrovek | see if anything obvious shows up during the delay |
| 16:25.37 | Naikrovek | or you can pastebin the result and link it here, we'll take a look |
| 16:25.48 | Naikrovek | ~pb |
| 16:25.48 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 16:26.10 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 16:29.31 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 16:30.00 | *** join/#asterisk Besticles (~Besticles@ip68-104-111-21.lv.lv.cox.net) |
| 16:32.00 | *** join/#asterisk oryxtec (~test@119.152.77.23) |
| 16:32.30 | Besticles | I am having problems originating calls with my new card. It's a 4 port card, I plugged a T1 in, and I believe I have it configured properly. Asterisk recognizes the channels. When I originate through the cli, it does nothing for 10ish seconds. |
| 16:32.46 | evils_unon | sorry, I got pulled away for a minute... I'll get back to this in a bit... :\ |
| 16:32.49 | Besticles | But then comes back and says it hungup the request channel, without any reason code. |
| 16:33.15 | Besticles | I am out of ideas on how to fix this. Any suggestions? |
| 16:34.32 | bmoraca_work | Besticles: show your call log, as well as the command you're using to originate, as well as your configurations in a pastebin |
| 16:34.34 | bmoraca_work | ~pb |
| 16:34.34 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 16:35.45 | *** join/#asterisk jasonwert-work (~jasonwert@adsl-99-27-170-70.dsl.klmzmi.sbcglobal.net) |
| 16:36.33 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 16:44.02 | Besticles | Is cdr-csv the call log? I'm reading that asterisk by default writes to this file for complete call recording, yet mine is empty. |
| 16:44.21 | *** join/#asterisk xoveruk (~xover@193.220.59.2) |
| 16:44.34 | xoveruk | how can I verify that asterisk is working without using a phone to test it? |
| 16:44.43 | russellb | define "working" |
| 16:44.54 | xoveruk | I can see sip peers and iax trunks |
| 16:45.01 | xoveruk | the service is running |
| 16:47.50 | Naikrovek | without using a phone to test it? |
| 16:47.51 | Naikrovek | wow |
| 16:47.53 | Naikrovek | um.. |
| 16:49.08 | [TK]D-Fender | "How do I prove my car works? Put the key in the ignition, start the engine and DRIVE IT" |
| 16:49.27 | Naikrovek | use a softphone i guess, but you said no phones, so... why do you CARE if it works without any phones |
| 16:50.43 | evils_unon | ok... I'm back now... nothing that explains the delay in the call output: http://pastebin.ca/1963175 |
| 16:57.46 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:57.55 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:58.43 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 17:01.18 | *** join/#asterisk smth (~smth@199.84.137.3) |
| 17:01.32 | *** join/#asterisk anontreborsux (~teddf@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 17:02.36 | anontreborsux | I am putting together a 25 set asterisk system. I need voice mail and I have a pri at that location. I found a PRI card cheap on ebay and I made a server and installed asterisk. |
| 17:02.48 | anontreborsux | I cant decide on what ip sip phones to use |
| 17:03.00 | anontreborsux | so many companys are selling them in lots on ebay |
| 17:03.43 | anontreborsux | They are car dealerships so I need multiple lines accessed and many times 2 3 or 4 are on hold and must be able to take another |
| 17:03.44 | *** join/#asterisk jasonwert-work (~jasonwert@adsl-99-27-170-70.dsl.klmzmi.sbcglobal.net) |
| 17:04.15 | anontreborsux | if I get 7960s they can put on hold and handle up to 6 calls? |
| 17:04.30 | WIMPy | anontreborsux: If you fiond a phone that's goot at that, tell me. |
| 17:04.33 | anontreborsux | I asume they can transfer |
| 17:04.49 | *** join/#asterisk josephnexus (~josephnex@71-209-40-81.bois.qwest.net) |
| 17:04.58 | anontreborsux | well my 2 decades old merlin legends can |
| 17:05.15 | anontreborsux | wimpy??? |
| 17:05.30 | WIMPy | My ISDN phone of similar age can as well... |
| 17:05.34 | anontreborsux | on a 7960 when a call comes in you cant put it on hold? |
| 17:05.46 | adyn | yeah you can |
| 17:05.47 | anontreborsux | then anser another |
| 17:06.03 | adyn | I haven't used 6 lines but I have used up to 3 on a 7960 |
| 17:06.04 | WIMPy | Ok, teh Snom also does it, but the keys aren't located next to the display. That's not optimal. |
| 17:06.18 | anontreborsux | snom? |
| 17:06.22 | adyn | simultaneous calls, 2 on hold with 1 incoming |
| 17:06.33 | josephnexus | hello everyone! I've got asterisk making a request via http using wget to a website... the website returns simply a 1 or a 0 depending on if the information provided by wget is correct. How can I have asterisk parse that 1 or a 0 and react accordingly? |
| 17:06.36 | anontreborsux | ya that is fine |
| 17:06.49 | WIMPy | Snom 360. You can use all 12 keys as line keys if you like. |
| 17:07.11 | WIMPy | But you need to remember yourself, who you got on what key. |
| 17:08.20 | anontreborsux | or label them |
| 17:08.28 | WIMPy | The SPA960 looks like it would do it in a good way, but I fond it rather painfull. |
| 17:08.51 | anontreborsux | if i used a snom 360 as operator and 7960s on the other extensions |
| 17:08.51 | adyn | I have a SPA962 that we use with a 4 line system that works well |
| 17:08.55 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 17:09.02 | anontreborsux | all 23 channels are accessable right |
| 17:09.15 | anontreborsux | so if call comes in operator can transfer |
| 17:09.21 | WIMPy | Sure |
| 17:09.44 | [TK]D-Fender | anontreborsux: Your phones call handling has no mapping to :lines" typically |
| 17:10.00 | anontreborsux | if she says 154 you have a call on (what does she say) can they type something to get the call from here on a page? |
| 17:10.04 | [TK]D-Fender | anontreborsux: If can handle X calls. These can be from any given resource |
| 17:10.15 | [TK]D-Fender | anontreborsux: Yes, that is called "parking" |
| 17:10.25 | anontreborsux | so she can park it on a code |
| 17:10.33 | [TK]D-Fender | anontreborsux: You park a call, it read s a lot # back to yuo and you yell out "Jim, call on 714!" |
| 17:10.33 | anontreborsux | then page that code for whoever |
| 17:10.45 | josephnexus | exactly |
| 17:10.49 | anontreborsux | he types 714 |
| 17:10.55 | [TK]D-Fender | anontreborsux: Yes |
| 17:10.57 | bmoraca_work | anontreborsux: if they're looking for a key system, they will be disappointed with asterisk |
| 17:11.00 | [TK]D-Fender | anontreborsux: All very normal stuff. |
| 17:11.30 | anontreborsux | and with a pri card does the operator just pic a number to park it on or is each channel designated a # that is constant? |
| 17:12.10 | josephnexus | anontreborsux, you tell it to park the call, it picks the first available lot and tells you the number |
| 17:12.15 | anontreborsux | I dont know wht we need yet |
| 17:12.22 | anontreborsux | I am information gathering |
| 17:12.33 | WIMPy | There is no relation between channels and parking slots. |
| 17:12.37 | anontreborsux | I think it should handle it |
| 17:12.41 | bmoraca_work | anontreborsux: additionally, you're going to want a hardware echo cancelling card with a telco PRI. |
| 17:12.42 | anontreborsux | ok |
| 17:12.54 | WIMPy | Channels are pretty randon and you don't have to care about them. |
| 17:13.02 | anontreborsux | kewl |
| 17:13.13 | anontreborsux | http://cgi.ebay.com/SpoTel-TE110P-TE110P-E1-T1-PRI-Asterisk-Trixbox-card-/170423407465?pt=LH_DefaultDomain_0&hash=item27ae06d369 |
| 17:13.27 | anontreborsux | is that all i need or will i need another card? |
| 17:13.48 | anontreborsux | I have a pri in place with a t1 card on the legend |
| 17:14.11 | bmoraca_work | that doesn't have HWEC |
| 17:14.23 | bmoraca_work | the T1 card in the legend will have HWEC |
| 17:14.45 | bmoraca_work | trust me on this, if you don't want screaming users, you want HWEC |
| 17:14.48 | anontreborsux | but deltacom is flexable about changing formats |
| 17:14.52 | [TK]D-Fender | Located in montreal.. LOL. I could drive over and piick one up :p |
| 17:14.53 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 17:15.07 | anontreborsux | HWEC? |
| 17:15.20 | anontreborsux | why would they scream |
| 17:15.27 | josephnexus | echos |
| 17:15.51 | bmoraca_work | anontreborsux: HWEC is HardWare Echo Cancellation. they WILL scream when they get echos on nearly all calls (not uncommon with a PRI) |
| 17:15.51 | WIMPy | HardWare Echo Cancellation |
| 17:16.10 | leifmadsen | screams... reams... eams...eamseamseamseeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 17:16.38 | bmoraca_work | anontreborsux: that said, if the PRI is not a real PRI and is delivered as part of an integrated service (you plug in to a T1 port on an Adtran or Cisco Total Access product), then you don't actually need echo cancellation, because that device will do it for you |
| 17:16.39 | *** join/#asterisk dreyfizzard (~dreyfizza@c-76-22-250-19.hsd1.tn.comcast.net) |
| 17:16.40 | Kobaz | so what do you guys use these days for analog->sip (fxo) |
| 17:16.59 | josephnexus | sangoma with echo cacncellation |
| 17:17.03 | josephnexus | the a200 I believe |
| 17:17.10 | Kobaz | not a line card... sip |
| 17:17.20 | bmoraca_work | Kobaz: SPA3201 |
| 17:17.29 | Kobaz | mm linksys |
| 17:17.35 | leifmadsen | Linksys SPAxxxx |
| 17:17.43 | bmoraca_work | sucks, but they're really not that bad |
| 17:17.45 | anontreborsux | so that card does not have echo cancelation |
| 17:17.45 | Kobaz | i was playing with a linksys spa4000, callerid doesn't work |
| 17:17.52 | [TK]D-Fender | Kobaz: Depends how many ports. More than 1, or actual serious use I'd go AudioCodes/Mediatrix |
| 17:17.54 | Kobaz | but other than that they work great |
| 17:18.13 | Kobaz | [TK]D-Fender: yeah, i have been going with audiocodes, now i'm finally getting burned from their lack of tech support |
| 17:18.18 | bmoraca_work | certainly, if you need many ports, an AudioCodes or Mediatrix would be better |
| 17:18.30 | Kobaz | 10 fxos... i need to replace these broken audiocodes boxes |
| 17:19.08 | Kobaz | anything i send to them gives me 500 internal server error |
| 17:19.12 | bmoraca_work | that said, if you needed a device which did a whole hell of a lot, an Adtran TA924e could be useful...they have one that does 8fxo, 16fsx, 2 pri and 2 data T1, plus two ethernet port router all in one |
| 17:19.33 | Kobaz | try to register, internal server error. try to place a call.. internal server error... try to log in on ssh to watch the console... disconnected due to packet error |
| 17:19.50 | Kobaz | bmoraca_work: yeah, i really love adtran, but that box does a little bit too much |
| 17:20.02 | bmoraca_work | it's also pretty expensive :P |
| 17:20.25 | WIMPy | 10 fxo? Wouldn't a pri be cheaper and easier? |
| 17:20.46 | drmessano | Hey now |
| 17:20.54 | drmessano | That 24 port card is badass |
| 17:21.08 | Kobaz | WIMPy: they are paying like 200 a month for 10 lines, it would be like 700 for a pri |
| 17:21.20 | drmessano | For a partial PRI? |
| 17:21.35 | WIMPy | Ouch |
| 17:21.49 | Kobaz | yeah i tried all kinds of setups, anything over t1 was just way more |
| 17:22.26 | bmoraca_work | Kobaz: why do you need to move it to SIP? |
| 17:22.28 | Kobaz | how come spa3201 doesnt have any hits on froogle |
| 17:22.40 | coppice | spa3102 |
| 17:22.42 | Kobaz | bmoraca_work: they are running a silly switchvox server |
| 17:22.42 | drmessano | Because its an SPA 3102 |
| 17:22.48 | Kobaz | ah |
| 17:22.51 | Kobaz | that would do it |
| 17:23.06 | anontreborsux | So I need a ti card that has echo cancelation |
| 17:23.07 | bmoraca_work | Kobaz: have you considered a T1 channel bank? |
| 17:23.49 | Kobaz | trying to minimize cost.. we already sold them the two audiocodes boxes... 4 port and an 8 port... i want to replace it with something equivalently priced |
| 17:23.53 | *** join/#asterisk jasonwert-work (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 17:23.55 | *** join/#asterisk wierdo (jimmy@77.78.3.197) |
| 17:24.04 | anontreborsux | so the cheap t1 cards will sound like crap? |
| 17:24.35 | bmoraca_work | anontreborsux: yes. |
| 17:24.38 | Kobaz | i've been staying away from line cards... so that would leave me with needing a channel bank and a sip-t1 box |
| 17:24.49 | Kobaz | which would be double of the audiocodes cost |
| 17:25.06 | bmoraca_work | Kobaz: a T1 line card and a T1 channel bank would work. |
| 17:25.07 | Kobaz | i've been bitten so many times by dahdi/zaptel issues, i just refuse to go back |
| 17:26.00 | anontreborsux | http://cgi.ebay.com/T1-card-E1-ISDN-PRI-TE110-TE110P-Digium-Asterisk-card-/170444588199?pt=LH_DefaultDomain_0&hash=item27af4a04a7 why is this bad? |
| 17:26.24 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 17:26.30 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 17:26.38 | bmoraca_work | anontreborsux: it will work, but you WILL have echo issues and your users WILL complain. |
| 17:26.52 | anontreborsux | gotchya i dont wnat that |
| 17:27.12 | anontreborsux | is there something i can use with that card to solve that or do i need an echo canceling card |
| 17:27.35 | Kobaz | it's best to do echo can on the card |
| 17:27.59 | Kobaz | if adtran made a purely 8 port fxo, i would be in love |
| 17:28.09 | WIMPy | anontreborsux: You can try in software, but you most probably want a card with HWEC> |
| 17:28.20 | bmoraca_work | Kobaz: you could get an Adtran Atlas 550 with two octal FXO cards and a T1 card |
| 17:28.56 | bmoraca_work | they're pretty cheap on eBay anymore |
| 17:29.03 | Kobaz | why would i need a t1 card? |
| 17:29.15 | Kobaz | or does it not do sip |
| 17:29.21 | bmoraca_work | because the Atlas 550 doesn't do SIP |
| 17:29.25 | anontreborsux | how do i search it t1 card echo nada |
| 17:29.31 | *** join/#asterisk kargig (~kargig@unaffiliated/kargig) |
| 17:29.39 | Kobaz | anyone have any experience with quintum? |
| 17:29.40 | anontreborsux | is there a name for echo cancelation besides hwec |
| 17:29.56 | Qwell | echo cancellation |
| 17:30.09 | bmoraca_work | anontreborsux: http://www.telephonydepot.com/Catalog/Digium-Digital-T1-E1-J1-Cards/Digium-TE121B |
| 17:30.11 | WIMPy | A DSP module or write it out. |
| 17:32.19 | Kobaz | mediatrix looks good, but they dont have high density fxo |
| 17:32.31 | anontreborsux | i need pci |
| 17:32.47 | bmoraca_work | anontreborsux: so go to that site and choose the PCI version |
| 17:33.51 | *** join/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 17:34.26 | coppice | I wonder what percentage of PCI cards they sell now |
| 17:34.35 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 17:34.40 | bmoraca_work | Kobaz: they have the 1204, which is 4 FXO...you'd need 3 of them... |
| 17:36.23 | kargig | has anyone succeeded in making multiple calls from the console using chan_alsa ? |
| 17:36.25 | anontreborsux | if i have a fxo card with hardware echo cancelation on it in the same computer does it work on the ti card |
| 17:36.34 | zoid_ | By any change, does anybody managed to get asterisk to work with a panasonic analog pbx? I can't make it detect the hangups |
| 17:36.36 | bmoraca_work | anontreborsux: no. |
| 17:36.39 | Kobaz | bmoraca_work: yeah, i know |
| 17:36.52 | Kobaz | bmoraca_work: it would be nice if they had an 8 or 16 |
| 17:36.55 | anontreborsux | man echo cancelation jumps the dinero |
| 17:37.28 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:37.28 | bmoraca_work | anontreborsux: asterisk is still cheaper than the alternatives. but telephony isn't cheap anyway, so i'm not sure what you were expecting |
| 17:38.01 | anontreborsux | i know |
| 17:38.15 | anontreborsux | and i can catch em on ebay for like 300 |
| 17:38.25 | anontreborsux | but 168 looked so nice |
| 17:38.42 | anontreborsux | it sounds like that is a bad thing to skimp on |
| 17:39.16 | josephnexus | it will come at night and haunt you if you dont |
| 17:39.19 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 17:39.20 | josephnexus | have echo cancellation |
| 17:40.34 | anontreborsux | man for under 2 gs i can have a 20 port phone system with voicemail how can i complain |
| 17:41.02 | bmoraca_work | anontreborsux: it's not going to be a very good phone system |
| 17:41.05 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:41.18 | anontreborsux | in what way? |
| 17:41.23 | bmoraca_work | in every way |
| 17:41.36 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:41.42 | *** join/#asterisk jasonwert-work (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 17:41.45 | anontreborsux | in comparison with the merlin legends how so |
| 17:42.08 | anontreborsux | what will i be lacking? |
| 17:42.10 | josephnexus | think of it this way, you will be depending on it every day... you want it good |
| 17:42.22 | bmoraca_work | that's something i couldn't tell you, because i'm not familiar with the merlin legend nor with your "2 gs" phone system |
| 17:42.32 | Naikrovek | wtf?! my ITSP just told me that I now have the option of using outgoing callerID that is not my own |
| 17:42.42 | Naikrovek | maybe that feature isn't as rare as I assumed it would be... |
| 17:43.03 | josephnexus | it isn't |
| 17:43.06 | josephnexus | :-P |
| 17:43.11 | anontreborsux | echo cancelling t1 pri card terabyte hard drive 4 gigs ddr2 quad core athlon |
| 17:43.13 | josephnexus | many provide that |
| 17:43.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 17:43.31 | anontreborsux | 7960 ciscos |
| 17:43.37 | Qwell | ebayed hardware. |
| 17:43.48 | anontreborsux | right |
| 17:43.57 | anontreborsux | not the computer |
| 17:44.03 | anontreborsux | that is built already |
| 17:44.07 | bmoraca_work | Naikrovek: it's a matter of policy...it's not hard to provide, but some places don't LIKE to provide it. i don't provide it as a matter of policy, not because I can't. |
| 17:44.09 | anontreborsux | and running asterisk |
| 17:44.35 | anontreborsux | are 7960s bad?? |
| 17:44.36 | Qwell | you're trusting the stability of your phone system on random crap you found on ebay. |
| 17:44.50 | bmoraca_work | anontreborsux: they're not great. i prefer the SPA500 series. |
| 17:44.56 | anontreborsux | wait aminute??? they are just phones they work or not |
| 17:45.06 | anontreborsux | so many companyies going out of business |
| 17:45.13 | Qwell | I'm not even talking about the phones. (Cisco are crap though) |
| 17:45.18 | Naikrovek | bmoraca_work: yeah it's just funny because i didn't ask for it, they just said "hey, we did a network upgrade, here's a list of things you can do now:" one of them was "use callerID of a phone number that is not assigned to your trunk." |
| 17:45.22 | anontreborsux | what are you talking about then? |
| 17:45.33 | Qwell | the other stuff you said you bought off ebay |
| 17:45.36 | bmoraca_work | anontreborsux: ... they're not "just phones"...there are many varying degrees of "works" in the VOIP world. |
| 17:45.40 | anontreborsux | the only thing on ebay was the phones? |
| 17:46.42 | anontreborsux | @qwell a new digium card is a digium card no matter wher ei got it |
| 17:46.51 | Qwell | No, it's not. |
| 17:46.56 | anontreborsux | wow |
| 17:46.59 | anontreborsux | ok |
| 17:47.09 | anontreborsux | anyway.. |
| 17:47.24 | Qwell | For instance, the ebay link you posted. |
| 17:47.28 | Qwell | CLEARLY not Digium. |
| 17:47.31 | Naikrovek | anontreborsux: Qwell works for Digium; he knows his hardware |
| 17:47.36 | anontreborsux | 7960s Do you guys think it can handle a small car dealership? |
| 17:48.15 | anontreborsux | IF someone sells me a new Digium card it does not matter where i got it (assuming it is actually a card) |
| 17:48.25 | fullstop | Hi all. In the "h" extension, is there a way to stop processing the dialplan, or do I have to jump to an "end" extension? |
| 17:48.28 | anontreborsux | I wasnt planning on buying a used one that would be hit or miss |
| 17:48.39 | Qwell | anontreborsux: It doesn't matter if it's new. |
| 17:48.44 | Qwell | Don't buy crap on ebay. |
| 17:49.15 | anontreborsux | ok now your being weird ill make sure i go by retail because you told me ebay was dangerous |
| 17:49.22 | anontreborsux | come on man |
| 17:49.47 | Qwell | I would bet you real cash money that there are currently 0 legitimate Digium cards on ebay right now. New or otherwise. |
| 17:49.49 | josephnexus | anontreborsux, i think it's just speaking of experience, your mileage may vary |
| 17:50.10 | anontreborsux | if i buy 10 7960s on ebay from a company going out of business (many are right now) ill be ok |
| 17:50.10 | fullstop | It's probably a brand new "Diguim" card. Just like the genuine Sorny and Panaphonic televisions. |
| 17:50.11 | *** join/#asterisk haryv (~haryv@154.5.144.132) |
| 17:50.23 | Qwell | fullstop: exactly |
| 17:50.37 | anontreborsux | http://cgi.ebay.com/Digium-TE121-w-Echo-Cancellation-/270645157569?pt=LH_DefaultDomain_0&hash=item3f03b55ec1 i am sure that is a card but it may or may not have an issue |
| 17:50.43 | josephnexus | you can risk it if you want, but we're just trying to make sure you have a successful deployment |
| 17:50.53 | anontreborsux | I understand |
| 17:51.00 | *** join/#asterisk thansen (~thansen@12.8.216.84) |
| 17:51.21 | josephnexus | it causes us much pain to see people having spent money on a card that isn't what they thought it was |
| 17:51.27 | *** join/#asterisk Marini (~Marini@80.90.80.78) |
| 17:51.28 | anontreborsux | right |
| 17:51.36 | haryv | which card |
| 17:51.45 | Marini | Hello |
| 17:51.47 | frigidzephyr | if you buy a digium card off ebay, it won't have a warranty btw |
| 17:52.01 | anontreborsux | that is legit |
| 17:52.05 | Qwell | says you. |
| 17:52.18 | anontreborsux | no warrenty is true |
| 17:52.29 | josephnexus | anyone have any ideas for how I can have asterisk visit a website and parse the one character that the webpage returns? |
| 17:52.34 | Marini | i want to match exten variable to some patterns, what is the best way, can I use that pattern chars we use in extensions? |
| 17:52.49 | wdoekes2 | josephnexus: func_curl? |
| 17:52.52 | josephnexus | and if a 0 is returned it stays in the loop, and if a 1 is returned it says "thanks" and hangs up? |
| 17:52.56 | anontreborsux | @qwell whats the warrenty on them from Digium |
| 17:53.02 | frigidzephyr | 5yr |
| 17:53.09 | josephnexus | i saw that I could use wget and curl, but I don't see how to have it read the information wdoekes2 |
| 17:53.09 | anontreborsux | thats strong |
| 17:53.30 | *** join/#asterisk ManxPower (~manxpower@244.sub-75-216-106.myvzw.com) |
| 17:53.30 | anontreborsux | no questions asked send card get card or get card then send card |
| 17:53.32 | haryv | One last step in trying out asterisk gui. Everything is up and running but the asterisk login page have said invalid username password. Cli says it does not exist yet, it does in manager.conf. http://pastebin.ca/1963227 any ideas? |
| 17:53.58 | anontreborsux | Do they send card and then you return card? |
| 17:53.58 | haryv | how long is the warentee for? |
| 17:54.08 | frigidzephyr | anontreborsux: for warranty RMA, technical support has to troubleshoot it first |
| 17:54.14 | ManxPower | Has anyone here setup a T-1 on one end and E-1 on the other end (international leased line)? |
| 17:54.16 | anontreborsux | i understand |
| 17:54.32 | anontreborsux | it is the do they send before return or after |
| 17:54.38 | wdoekes2 | josephnexus: if that is too hard for you, you should probably be reading some more first |
| 17:54.42 | wdoekes2 | ~book |
| 17:54.42 | 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 |
| 17:55.08 | josephnexus | ok... i'm fine with reading and such, but I don't know of a good location to look |
| 17:55.16 | fullstop | wow, check out this iPed! http://www.digginchina.com/apad-china-ipad-iped-mid-entertainment-and-internet-device-p-3344.html :-P |
| 17:55.42 | josephnexus | the books are nice, but does anyone know of any good web sources? |
| 17:55.47 | haryv | Anyone here installed the asterisk gui in the past? |
| 17:56.14 | anontreborsux | http://cgi.ebay.com/Digium-AEX430B-3-FXS-0-FXO-0-Echo-Cancellation-/330320457646?pt=LH_DefaultDomain_0&hash=item4ce8a22fae Thats not a real Digium card? |
| 17:56.34 | anontreborsux | Tech-R-us is listed as a vendor |
| 17:57.02 | Marini | PLEASE HELP: I want to match {$exten} variable in a if condition to some patterns? what is the best way to match patterns in an expression, can i use the same as for extensions ? |
| 17:57.25 | anontreborsux | Ok assumeing I get a working t1 echo cancelation card and 7960s. Assuming everything is working will it do the job? |
| 17:57.43 | wdoekes2 | Marini: regular expressions: ${EXTEN}=~... |
| 17:58.25 | Marini | wdoekes2: where can i find some examples? |
| 17:58.43 | josephnexus | also, if someone wants to help me create this howto and such, i'm more than willing to pay |
| 18:01.20 | wdoekes2 | Marini: *sigh* on google: http://www.google.com/#q=asterisk+regular+expression |
| 18:01.55 | kargig | ERROR[12137]: chan_alsa.c:708 alsa_read: Read error: Resource temporarily unavailable any ideas ? asterisk 1.4.31 on debian lenny |
| 18:04.54 | josephnexus | so curl is pretty easy to use, but I'm not finding anywhere (when looking at the dial plan section of the before mentioned book) where i can read the data that curl retrieved |
| 18:04.58 | [TK]D-Fender | anontreborsux: Where are yo getting the 7960's, and for how much? |
| 18:05.39 | anontreborsux | 7960s are everywhere |
| 18:05.48 | josephnexus | 7960s are a huge pain |
| 18:05.51 | [TK]D-Fender | anontreborsux: Now the actual answer is...? |
| 18:05.54 | anontreborsux | i can buy 30 from another dealership |
| 18:05.54 | josephnexus | especially if they don't come with the right firmware |
| 18:06.05 | anontreborsux | that were just pulled becaus ethey went out of business |
| 18:06.06 | [TK]D-Fender | anontreborsux: I can buy 10,000 from Cisco. Your point? |
| 18:06.08 | josephnexus | because cisco doesnt like actually giving the firmware out |
| 18:06.11 | petern_ | If they come with the wrong firmware, put the correct firmware on them... |
| 18:06.20 | Qwell | petern_: You can't. |
| 18:06.20 | anontreborsux | i realize i will have to change the firmware |
| 18:06.26 | [TK]D-Fender | anontreborsux: Cisco is not recommended. |
| 18:06.27 | Qwell | not legally, anyways |
| 18:06.47 | josephnexus | go with aastra if you can |
| 18:06.53 | josephnexus | my experience with them has been quite nice |
| 18:06.56 | [TK]D-Fender | Polycom > All |
| 18:06.57 | anontreborsux | so what is readily aivalable used and recommended? |
| 18:07.07 | josephnexus | i prefer aastra |
| 18:07.10 | josephnexus | easy to setup and maintain |
| 18:07.14 | josephnexus | and the cost is nice on the units |
| 18:07.30 | wdoekes2 | josephnexus: Set(x=${CURL(http://my.url)}) ; ... do something with ${x} |
| 18:07.38 | petern_ | Problem with the Ciscos is they have a stupidly large screen for no apparent reason. |
| 18:07.40 | anontreborsux | AASTRA M8314 |
| 18:07.44 | *** join/#asterisk jasonwert-work (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 18:07.50 | anontreborsux | damn nice prices |
| 18:08.09 | anontreborsux | AASTRA M8314 you think would do me? |
| 18:08.11 | josephnexus | ah, wdoekes2, so the x in this case would be the 1 or 0 that my webapp returns |
| 18:08.21 | josephnexus | that's quite clever... i was over thinking it |
| 18:08.24 | josephnexus | thanks for the pointer! |
| 18:08.35 | fullstop | So I have AsteriskFAX working on a TDM410P. Tomorrow, my "trunk" will go from the analog to a TE122. What sort of troubles might I have with the fax? The TE122 is a whole new beast for me. |
| 18:08.57 | anontreborsux | i need ethernet switch in them though |
| 18:09.07 | [TK]D-Fender | anontreborsux: FFS that is a stupid ANALOG PHONE |
| 18:09.15 | frigidzephyr | fullstop: the is the circuit going in the TE122 ? PRI ? |
| 18:09.15 | josephnexus | most of the aastra phones do that |
| 18:09.23 | anontreborsux | lol |
| 18:09.26 | josephnexus | i like the 5731i for entry level stuff |
| 18:09.35 | josephnexus | and the 5767i for the higher end phones |
| 18:09.37 | *** join/#asterisk jhirley (~chatzilla@c-75-74-13-194.hsd1.fl.comcast.net) |
| 18:10.35 | fullstop | frigidzephyr: yes. No additional hardware. We are moving from an ancient lucent merlin system, and it is a PRI. |
| 18:11.07 | anontreborsux | fullstop us too |
| 18:11.09 | frigidzephyr | fullstop: as long as timing is set correctly for the PRI span, then you should be good, |
| 18:11.21 | anontreborsux | but that legend has been such a work horse |
| 18:11.43 | anontreborsux | so cisco 7960s suck |
| 18:11.48 | fullstop | I had the line hooked up once and placed a few calls in... but I've never done it with the fax. |
| 18:11.52 | anontreborsux | ill have problems? |
| 18:12.10 | [TK]D-Fender | anontreborsux: How are you going to power them? That is an OLD model. |
| 18:12.21 | anontreborsux | power blocks |
| 18:12.28 | anontreborsux | they have them with them |
| 18:12.31 | [TK]D-Fender | anontreborsux: Fugly, and costly |
| 18:12.43 | fullstop | anontreborsux: The legend has been a good piece of hardware.. but the sets are outdated and we've had a few scares where it did boot correctly until after a few tries. |
| 18:12.44 | anontreborsux | the power blocks like to screw up? |
| 18:13.09 | anontreborsux | fullstop i have 30 processors and 100s of cards |
| 18:13.11 | petern_ | anontreborsux, they cook themselves. The Cisco power bricks I have run very hot. |
| 18:13.33 | fullstop | anontreborsux: ours is a much smaller setup. :) |
| 18:13.37 | anontreborsux | but ciscos are so aivalable and cheap |
| 18:13.54 | anontreborsux | fullstop we have 9 legends in 9 dealerships |
| 18:14.02 | anontreborsux | we have the cards for backup |
| 18:14.25 | petern_ | Which is odd, as a 7960 draws 6.3W from a PoE switch. |
| 18:15.02 | anontreborsux | how is comunicating betwwen asterisk machines? |
| 18:15.20 | anontreborsux | I realize i need to provide it with addiquate bandwith |
| 18:15.34 | anontreborsux | like nissan will have one system and accounting another |
| 18:15.38 | fullstop | petern_: I guess that means that the power brick is inefficient and is producing a lot of heat as it converts. |
| 18:15.40 | petern_ | I run a dozen 7960/7970s, but in my defence I didn't choose them :) |
| 18:15.43 | petern_ | fullstop, yup |
| 18:15.48 | anontreborsux | can an extension call an extension on the other |
| 18:16.16 | fullstop | They expect you to do POE and give you a genuinely shitty transformer. |
| 18:17.18 | petern_ | Plus you have to be wary of PoE switches -- they're pre-standard. |
| 18:17.28 | anontreborsux | what has a ethernet switch and sip and aivalable besides ciscos |
| 18:18.03 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 18:18.04 | WIMPy | And POE switches have a lower MTBF than non-POE ones. |
| 18:18.07 | haryv | I cooked my ether port tester on a rj45jack once because it was poe live. |
| 18:18.24 | petern_ | heh |
| 18:18.35 | haryv | 100 bucks poof |
| 18:18.48 | fullstop | They don't protect against that? |
| 18:18.50 | petern_ | Ah, not just a cable tester then... |
| 18:18.59 | nny | any polycom users experienced this issue? I have someone with numerous phones that when you put line 1 on hold and pick up line 2, afterwards when you resume line 1 you cannot hear the caller or visa versa... |
| 18:19.14 | petern_ | A plain cable tester will probably blow its LEDs, but then they're not 100 bucks. |
| 18:19.19 | haryv | Its only a cable tester. Fluke makes a great tester and port tester plus it can flash the port on the switch. |
| 18:19.29 | nny | still gathering data, but wondering if this is a polycom config issue or asterisk itself |
| 18:19.46 | fullstop | nny: is nat involved? |
| 18:19.46 | anontreborsux | and the recent changes in DMCA does that include phones because it covered many electronic devices of all kinds having legally obtained software put on it including firmware and software |
| 18:19.54 | nny | fullstop no |
| 18:19.55 | anontreborsux | iphone video game consoles ect |
| 18:20.03 | nny | fullstop same ip/ on LAN |
| 18:20.05 | anontreborsux | this did not include ip phones? |
| 18:20.06 | fullstop | nny: haven't seen it (yet!) |
| 18:20.17 | nny | fullstop: i'll let you know once I dig enough |
| 18:20.49 | fullstop | I have a bunch of SoundPoint 321 sets here, which have been fairly well tested... |
| 18:20.58 | fullstop | and I have not seen it. |
| 18:21.12 | nny | fullstop yeah I didn't set this up, suspecting 99% it's a config issue |
| 18:24.06 | anontreborsux | What will be the diffrence for me between 7940 and 7960 oh wait dod they both have ethernet to computer switch |
| 18:25.00 | anontreborsux | http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 18:25.46 | [TK]D-Fender | anontreborsux: Stop wasting your time with Cisco phones |
| 18:26.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 18:26.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:26.49 | anontreborsux | i am still looking for a suggestion |
| 18:26.57 | anontreborsux | they are so cheap and aivalable |
| 18:28.11 | anontreborsux | If I am on a 7940 can i put a call on hold then dial out another line? |
| 18:28.21 | anontreborsux | will it do call conference |
| 18:28.30 | anontreborsux | how many simultanious lines? |
| 18:28.30 | [TK]D-Fender | yes, jsut like just about EVERY phone out there |
| 18:28.46 | [TK]D-Fender | anontreborsux: http://www.telephonydepot.com/Catalog/Polycom-Phones |
| 18:28.57 | anontreborsux | 7940 will do 4 and 7960 6 is that right? |
| 18:29.19 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 18:29.33 | anontreborsux | So hard to figure out what model will do all i need |
| 18:30.28 | fullstop | I would purchase one and test drive it before I bought a case of them. :) |
| 18:30.50 | [TK]D-Fender | fullstop: He's already sold on them.... this looks like a waste of time |
| 18:30.59 | anontreborsux | i think i will buy a polycom 601 and a 7940 and see |
| 18:31.10 | nny | fullstop: looks like it's differenth then the user described (shock!). When a second call comes in the user can no longer hear the first caller's voice, even if they ignore it |
| 18:31.25 | nny | fullstop: some kind of rtp stream issue.. still waiting on ssh access lol |
| 18:31.54 | nny | i have never heard of the issue before, and I don't think any google query will turn up some relational data |
| 18:32.01 | nny | or least it hasn't yet |
| 18:32.07 | anontreborsux | [TK]D-Fenderwhat polycom model has a switch? |
| 18:32.23 | anontreborsux | I am lost not sold. |
| 18:32.36 | [TK]D-Fender | anontreborsux: 601 = discontinued. You seem to insist on looking at old model and running out of supoprt |
| 18:32.47 | anontreborsux | i am looking for used |
| 18:32.56 | [TK]D-Fender | anontreborsux: 331/335/450/550/650 |
| 18:34.35 | *** join/#asterisk tessier (~treed@kernel-panic/copilotco) |
| 18:35.07 | tessier | Anyone know of a way to monitor asterisk call quality? Packet loss, jitter, delay, etc? Traditional nagios type tools are designed for something entirely different. |
| 18:35.27 | tessier | I keep having people tell me our call quality is not so good but when I test it things are great. |
| 18:36.40 | leifmadsen | tessier: AQuA |
| 18:36.53 | leifmadsen | tessier: hard to monitor in real time though -- you can use RTCP for that |
| 18:38.06 | tessier | Actually, I might be able to do it with ethereal... |
| 18:40.23 | anontreborsux | Linksys SPA922 ???? |
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| 18:41.28 | tessier | leifmadsen: How would I use RTCP to monitor calls? |
| 18:41.42 | tessier | Is there software to do that or do I have to capture the rtcp with ethereal or something? |
| 18:42.11 | Besticles | I can make outbound calls. This whole time I though LIBPRI was optional for some reason to make dahdi originates. |
| 18:44.16 | anontreborsux | Linksys SPA922 works with asterisk any opinions? |
| 18:45.00 | [TK]D-Fender | anontreborsux: Sadly limited phone |
| 18:45.00 | Gugge | anontreborsux: my opinion would be yes |
| 18:45.15 | leifmadsen | tessier: probably through the AMI I would suspect. Not sure how accurate they will be or how to do it, but that is basically how you would have to do it. |
| 18:45.25 | Gugge | but i dont like the SPA phones :) |
| 18:51.22 | tessier | http://www.voip-info.org/wiki/view/Asterisk+RTCP |
| 18:51.25 | tessier | With I had access to that. |
| 18:51.33 | tessier | But we are running an old asterisk from fonality which I can't upgrade. :( |
| 18:55.04 | anontreborsux | <[TK]D-Fender> I need to have a recetionist answer the phone. Put it on hold maybe answer others. Page for someone to pick up or transfer the call. Will the Linksys do that? |
| 18:55.32 | anontreborsux | What would I be lacking with the linksys from a 7940 or 7960 |
| 18:55.51 | anontreborsux | the hardest part of this is choosing the phones |
| 18:56.05 | [TK]D-Fender | anontreborsux: That is a dumb shit phone that you can't even push a like-key to flip between the only 2 calls it can handle. It is no smarter than an ANALOG PHONE |
| 18:56.17 | anontreborsux | gotchya |
| 18:56.23 | anontreborsux | i dont want compaints |
| 18:56.26 | [TK]D-Fender | anontreborsux: Stop being a comlpete cheap-ass or you'll find yourself flushing money & time. |
| 18:56.53 | anontreborsux | god know admining 164 workstations and 7 servers for dealerships in 3 cities is complaint ridden already |
| 18:57.06 | anontreborsux | I am trying to find balance |
| 18:57.20 | anontreborsux | I dont want to pay over 200 |
| 18:57.23 | anontreborsux | a phone |
| 18:57.31 | anontreborsux | I cant would be mor elike it |
| 18:57.37 | leifmadsen | tessier: sorry, you're out of luck. Only way to do it then is to use AQuA and create a test where you perform calls automatically across the netowrk and keep testing |
| 18:57.57 | anontreborsux | Owe ya is there a way to record phone calls with asterisk? |
| 18:58.05 | [TK]D-Fender | anontreborsux: "A" phone, or for EACH phone? |
| 18:58.07 | leifmadsen | MixMonitor() |
| 18:58.13 | [TK]D-Fender | anontreborsux: Yes |
| 18:58.20 | anontreborsux | MixMonitor? |
| 18:58.28 | anontreborsux | kewl ill google it |
| 18:58.29 | leifmadsen | MixMonitor(). |
| 18:58.29 | [TK]D-Fender | anontreborsux: Dialplan app. Don't get ahead of yourself |
| 18:58.45 | anontreborsux | i just need to know it exsists first |
| 18:58.50 | leifmadsen | enables the echo canceller |
| 18:58.55 | leifmadsen | it exists |
| 18:59.00 | anontreborsux | kewl |
| 18:59.03 | Qwell | cancels leifmadsen |
| 18:59.11 | leifmadsen | kills Qwell with love |
| 18:59.29 | anontreborsux | I assume i can use a fxo card for the exsisting pagepac |
| 18:59.50 | [TK]D-Fender | anontreborsux: If that's what you have |
| 19:02.03 | anontreborsux | I can do this I just need to find some handsets |
| 19:02.25 | anontreborsux | because money is tight as you all might have noticed from the news in the car business |
| 19:02.41 | anontreborsux | fender thank you very much for the advice |
| 19:02.53 | anontreborsux | thank all of you by the way |
| 19:04.06 | anontreborsux | Cisco 3com nortel linksys polycom who else has something decent |
| 19:04.43 | Qwell | Polycom. |
| 19:04.53 | Kobaz | polycrum is awesome |
| 19:05.03 | Kobaz | i just wish they had more buttons |
| 19:05.40 | josephnexus | aastra |
| 19:05.44 | josephnexus | anontreborsux: I like aastra |
| 19:05.46 | Kobaz | those are nice too but more pricey |
| 19:05.47 | josephnexus | :-p |
| 19:06.19 | Kobaz | there's just no other phone with high sound quality and uber configurability in the 70-80 dollar range other than polycom |
| 19:06.40 | Kobaz | well, no phone that ive found so far |
| 19:07.03 | Naikrovek | there isn't one |
| 19:07.06 | Naikrovek | polycom > all |
| 19:07.18 | anontreborsux | 70-80 dollars what model? |
| 19:07.25 | Naikrovek | IP321 |
| 19:07.26 | Kobaz | 321 |
| 19:07.47 | Naikrovek | http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-321 |
| 19:07.58 | Naikrovek | they don't show you the price anymore but it's like $79.95 or something |
| 19:08.03 | Naikrovek | was last time i ordered anyway |
| 19:08.16 | anontreborsux | i have to hav ethernet switch |
| 19:08.20 | anontreborsux | in tem |
| 19:08.23 | anontreborsux | in the phone |
| 19:08.25 | Naikrovek | then IP 331 |
| 19:08.36 | Naikrovek | http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-331 |
| 19:08.40 | Naikrovek | or 335 if you can afford it |
| 19:08.53 | Naikrovek | 335 would be best in that series actually |
| 19:09.02 | Kobaz | boo fscking rah |
| 19:09.13 | Kobaz | my vendor finally got me new audiocodes firmware |
| 19:11.22 | Kobaz | lets hope this works |
| 19:11.35 | Kobaz | MP118_SIP_F6.00A.024.003.cmp |
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| 19:17.37 | drfreeze | Question - Is there a way to provide visual line presence on polycom phones? |
| 19:17.50 | Naikrovek | visual line presence? whassat mean |
| 19:17.51 | drfreeze | I have clients moving from analog system with fixed lines to PRI and polycom |
| 19:18.01 | Naikrovek | well they're going to have to change their mindset |
| 19:18.17 | drfreeze | They like the fact that they could tell if someone was on a line on the analog system |
| 19:18.25 | Naikrovek | THAT is possible |
| 19:18.27 | Naikrovek | but a per-line thing is not |
| 19:18.32 | Naikrovek | nor is it desireable |
| 19:18.47 | drfreeze | Naikrovek: how? |
| 19:18.52 | Naikrovek | there won't be any more "Doris! Your sister is on line 2!" |
| 19:18.55 | anontreborsux | if you park it whats the diffrence in real world use |
| 19:19.00 | anontreborsux | just perception |
| 19:19.01 | Naikrovek | exactly |
| 19:19.02 | Naikrovek | park it |
| 19:19.09 | Naikrovek | now the BLF thing is different |
| 19:19.12 | Naikrovek | you can see when someone is on the phone |
| 19:19.14 | anontreborsux | doris your sister in on 22 |
| 19:19.17 | Naikrovek | that's called BLF |
| 19:19.17 | drfreeze | parking the call disapears |
| 19:19.19 | anontreborsux | where ys parked it |
| 19:19.32 | Kobaz | your sisters brother's uncle's cousin is on 9277832 |
| 19:19.41 | drfreeze | a visual indicator of parked calls would be nice |
| 19:19.56 | Naikrovek | parking the call does not make it disappear, the phone will tell you where the call is. then you say "Doris! Your sister is on 72!" then doris dials extension 72 and her sis is there |
| 19:20.08 | WIMPy | drfreeze: That's possible, but still horrible. |
| 19:20.11 | Naikrovek | yes |
| 19:20.17 | Naikrovek | lose the "i wanna see everything" idea |
| 19:20.20 | Naikrovek | well |
| 19:20.22 | anontreborsux | hmm when the operator needs to get it back when no one picke dup the page how does she know it is still parked |
| 19:20.23 | Naikrovek | tell your peeps to lose that |
| 19:20.49 | anontreborsux | is there multiple parking |
| 19:20.51 | Naikrovek | the operator's (or whoever did the parking) phone will ring after i think 1 minute and it'll be the parked call calling back |
| 19:20.54 | Naikrovek | yes there is multiple parking |
| 19:21.09 | anontreborsux | Naikrovek i am sold |
| 19:21.12 | drfreeze | yes, we know all the basics of parking |
| 19:21.17 | anontreborsux | asterisk is good |
| 19:21.17 | Naikrovek | you transfer a call to (by default i think) extension 70, then the phone system says back to you "71" or "72" or "73" or whatever |
| 19:21.20 | drfreeze | but it is still out-of-sight-out-of-mind |
| 19:21.25 | drfreeze | that is why it reings back |
| 19:21.32 | Naikrovek | drfreeze: then stick to your analog system or buy a digital one |
| 19:21.46 | *** join/#asterisk csnook (~chris@va-76-1-132-194.dhcp.embarqhsd.net) |
| 19:21.55 | anontreborsux | rind back solves the issue |
| 19:21.58 | WIMPy | It's nopt out of sight if you set up buttons for the parking slots. |
| 19:22.07 | drfreeze | Naikrovek: what were you saying was possible with the line presence |
| 19:22.19 | drfreeze | WIMPy: oh, that would help |
| 19:22.21 | Naikrovek | you can set up BLF on your phone to see the status of other phones |
| 19:22.38 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-171-237.dsl.stlsmo.sbcglobal.net) |
| 19:22.38 | drfreeze | ok, I'll read up on BLF |
| 19:22.41 | drfreeze | thanks |
| 19:22.42 | anontreborsux | another question regardless of what ip phones i get they all can call for voicemail from asterisk right? |
| 19:22.44 | Naikrovek | my receptionist has a Polycom IP650 with a sidecar, everyone (important) has a button on the sidecar |
| 19:22.49 | gregd | hi guys, could someone enlighten me on chan_tapi? where can I find the possible config attributes for tapi.conf? |
| 19:22.58 | Naikrovek | so when they're on the phone, the light next to their name is red and the icon next to their name changes |
| 19:23.16 | Naikrovek | when they set DND it changes to a DND icon |
| 19:23.16 | Naikrovek | etc |
| 19:24.44 | LemensTS | I have an asterisk server. I have a SIP account with an ITSP on it, who I have DID's with. If a call comes in on a DID, and i forward it back out to them to someone's cell phone, it does not work. Any other number I try does. Yet I can call from the asterisk to the cell phone over the SIP account just fine. Any ideas? |
| 19:24.54 | [TK]D-Fender | anontreborsux: A call is a call is a call |
| 19:25.01 | anontreborsux | kewl |
| 19:25.07 | drfreeze | Naikrovek: is there a light for each position on a sidecar? |
| 19:25.08 | LemensTS | When I say does not work I mean no audio either way |
| 19:25.12 | Naikrovek | drfreeze: yes |
| 19:25.21 | Naikrovek | Polycom SoundPoint IP 650 - google it |
| 19:25.23 | anontreborsux | just trying to check all i can i dont want to get everything and get ow you cant do this |
| 19:26.20 | anontreborsux | I think I would be best off if I get a big feature rich phone with a sidecar that sees who is on the phone ect for the operator at each location |
| 19:26.28 | [TK]D-Fender | LemensTS: So ITSP>*>Same ITSP>Cell = no audio. SIP Phone>*>ITSP>Cell = OK? |
| 19:26.31 | *** join/#asterisk twanny796 (~twanny@80.71.106.213) |
| 19:26.45 | LemensTS | TKD-Fender: Exactly |
| 19:26.47 | anontreborsux | frnder how is compunication between extensions on diffrent asterisk boxes |
| 19:27.12 | twanny796 | 'low budget' isdn pci card? |
| 19:27.17 | [TK]D-Fender | LemensTS: If you "forward to a NON-cell #? Is it everything you bounce, or jsut stuff targeting a Cell specifically? |
| 19:27.27 | twanny796 | mISDN compatible |
| 19:27.30 | [TK]D-Fender | LemensTS: Basically sounds like a reinvite issue. make sure it's disabled |
| 19:27.56 | [TK]D-Fender | twanny796: There are a ton of cheap HFS-C cards out there that work. Beronet, etc |
| 19:28.29 | LemensTS | TKD-Fender: I have forwarded to Landlines and Cellphones n my tests. Im in Missouri, the cell having a problem is in Georgia is the only thing different. |
| 19:28.36 | LemensTS | Ill check that reinvite. |
| 19:29.11 | *** join/#asterisk asphere (~ardavis@hologram.homeip.net) |
| 19:29.20 | twanny796 | [TK]D-Fender: ok they changed name of shop! |
| 19:30.16 | *** join/#asterisk Porks (~BARAD-DUR@unaffiliated/porks) |
| 19:31.10 | [TK]D-Fender | twanny796: I have no personal experience with them, there are many usable cards out there cheap. |
| 19:31.36 | *** part/#asterisk Porks (~BARAD-DUR@unaffiliated/porks) |
| 19:36.10 | gregd | so what are the requirements to use tapi on windows? What kind of modem does it have to be? |
| 19:36.20 | *** join/#asterisk oryxtec (~test@119.152.77.23) |
| 19:43.33 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
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| 19:44.52 | anontreborsux | what was the program for recording calls |
| 19:45.57 | fullstop | Can I have an audio path like this: SIP -> IAX TRUNK -> SIP (302 redirect) -> SIP ? |
| 19:46.11 | *** join/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 19:46.22 | fullstop | The last two SIP legs are both on the same asterisk server. |
| 19:46.40 | fullstop | and on the same network |
| 19:46.45 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 19:48.10 | atan | Hey fellas, I'm trying to setup my first asterisk install on debian. I have it running, but I can't seem to get the asterisk-gui working. I have followed the guides found on Google, and they all seem so very basic, yet for some reason I cannot access it on port 8088. Locally (lynx) I can pull it up but it says nothing found. I'm trying to get the remote side of it working first. Anyone here been through this before? =) |
| 19:48.22 | anontreborsux | will mixmonitor alow me to record all calls in or out? |
| 19:48.29 | anontreborsux | what is the best way to do that? |
| 19:48.32 | beardy | Skip any GUI crap. |
| 19:48.39 | tomodachi | atan: if you really wanna learn asterisk |
| 19:48.44 | tomodachi | dont start with the gui stuff |
| 19:49.19 | beardy | That goes for everything, if you want to learn anything. |
| 19:49.34 | tomodachi | except for windows maybe |
| 19:49.50 | atan | tomodachi: okay, I do suppose. I want to set this up to act as my fax server. My provider will fix me up with a DID, and point it to my asterisk box... and from there I'm on my own. |
| 19:50.02 | anontreborsux | What would be the best way to record all calls in or out. |
| 19:50.04 | beardy | There isn't really things to learn with that, since.. you can't, aren't allowed, and so forth. |
| 19:50.24 | atan | All the config stuff seems very basic. Much like apache configs if you will. |
| 19:50.38 | beardy | atan: So you have it up and running in a basic fashion already?= |
| 19:50.43 | tomodachi | well it doesent sound like you need much of a dialplan |
| 19:51.08 | atan | I've already svnd the asterisk-gui thinger. What's the best way to remove the thing from it now? |
| 19:51.16 | atan | Or have I already filled the boat with too much water? |
| 19:51.54 | beardy | Where..? |
| 19:52.03 | beardy | If not in /usr/local pretty much. |
| 19:54.30 | anontreborsux | What would be the best way to record all calls in or out. |
| 19:58.22 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
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| 20:09.44 | bmoraca_work | anontreborsux: record them to a ram disk and then copy them to your destination after the fact |
| 20:09.56 | bougyman | or get good storage |
| 20:10.18 | bougyman | we use raids with write-cache and a lot of cache on the controller, keeps up with 300 concurrent calls just fine. |
| 20:15.57 | *** join/#asterisk csnook (~chris@va-76-1-132-194.dhcp.embarqhsd.net) |
| 20:25.34 | Kobaz | hmm |
| 20:25.38 | Kobaz | can you specify more than one bindaddr |
| 20:26.41 | [TK]D-Fender | Kobaz: No, one, or all |
| 20:26.59 | [TK]D-Fender | checkout time, BBIAB |
| 20:27.04 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-171-237.dsl.stlsmo.sbcglobal.net) |
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| 20:33.36 | Kobaz | so what am i doing wrong here |
| 20:33.47 | Kobaz | there's still something fundemental that i don't get about sip |
| 20:33.51 | Kobaz | http://pastebin.com/d4ACA0zH |
| 20:34.13 | Kobaz | i'm trying to accept an incoming call, and asterisk is telling me it's ignoring the invite |
| 20:34.37 | Kobaz | it's saying SIP/2.0 401 Unauthorized |
| 20:34.45 | Kobaz | but the passwords match |
| 20:37.09 | *** part/#asterisk josephnexus (~josephnex@71-209-40-81.bois.qwest.net) |
| 20:39.40 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 20:39.42 | citywok | ~itsp-us |
| 20:39.48 | citywok | ~itsp |
| 20:39.48 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:39.53 | citywok | ~itsplist-us |
| 20:39.53 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 20:42.49 | Kobaz | Ignoring this INVITE request |
| 20:42.55 | Kobaz | why why why... this is so god damn annoying |
| 20:43.05 | Kobaz | how c an i get it to not ignore the invite |
| 20:43.19 | Kobaz | i've never had this problem before |
| 20:44.16 | pabelanger | Kobaz: enable verbose and debug and see what is going on. Asterisk usually lists a reason |
| 20:45.04 | Kobaz | yeah |
| 20:45.10 | Kobaz | i'll try debug |
| 20:45.32 | pabelanger | Also looks like a routing problem. |
| 20:45.51 | pabelanger | Retransmitting #1 (no NAT) to 24.103.56.168:5060: |
| 20:46.07 | pabelanger | but I also see 127.0.0.1 as the IP |
| 20:47.31 | Kobaz | it's supposed to be going 127.0.0.1:5060 to 127.0.0.1:5061 |
| 20:47.42 | Kobaz | one asterisk talking to another |
| 20:47.51 | Kobaz | yeah, there shouldn't be any external ips involved |
| 20:47.52 | atan | Can anyone think of a simple SIP client for Windows off the top of your head? Just to test connections at this point. I'd prefer something completly free of bloat, ads, shareware-type junk if at all possible |
| 20:48.20 | *** part/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 20:48.40 | citywok | xlite |
| 20:48.44 | citywok | zoiper |
| 20:49.03 | pabelanger | Ekiga |
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| 20:54.32 | *** join/#asterisk oDesk (~f@94.97.56.103) |
| 20:56.04 | oDesk | is it possible to stream swf as music on hold in asterisk ? |
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| 20:57.42 | Kobaz | okay, that was the problem |
| 20:57.44 | Kobaz | the nat stuff |
| 20:57.50 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 20:58.43 | Kobaz | phew |
| 21:03.20 | citywok | nat caused problems w/ sip? no way! |
| 21:03.23 | citywok | :P |
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| 21:26.19 | Kobaz | with asterisk 1.6 i'm getting better errors now |
| 21:26.42 | [TK]D-Fender | Kobaz: Now you have have the most modern and fashionable flaws! |
| 21:26.47 | diegocn | hello ppl, how can i make a click-to-call script? |
| 21:26.48 | Kobaz | the audiocodes is complaining it can't find an endpoint, which is much better than some random internal server error |
| 21:26.51 | Kobaz | [TK]D-Fender: yeah |
| 21:26.53 | [TK]D-Fender | 's new screen is HUGE.... |
| 21:26.57 | Kobaz | heh |
| 21:27.32 | [TK]D-Fender | diegocn: Call-files, AMI / CLI Originate. Take your pick |
| 21:27.41 | Kobaz | now to fix the endpoint problem... i've never set up an audiocodes on a system with a non-standard sip port |
| 21:28.06 | citywok | diegocn: [TK]D-Fender listed the ways to originate a call. how you do your script depends on... what you are working with |
| 21:28.21 | diegocn | I'm trying with ami and php |
| 21:28.40 | diegocn | my script connects then make the call to destination |
| 21:28.42 | [TK]D-Fender | diegocn: That'll do |
| 21:29.51 | *** part/#asterisk TJ_stamm (~TJ@69.197.107.254) |
| 21:31.00 | citywok | Yep, that's very doable, sounds like you;ve got it all figured out! |
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| 22:09.42 | Kobaz | maybe not |
| 22:09.44 | Kobaz | er |
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