| 00:02.41 | SuperNull | gonna end up searching source uhg. good times. |
| 00:07.17 | carrar | SuperNull, probavly between 1.2 and 1.4.X |
| 00:07.20 | carrar | probably |
| 00:15.16 | SuperNull | yeah its fun the source does a select against * ... derp. so its making me work at finding it |
| 00:18.04 | SuperNull | 1.2.17 has realtime .. yet.. 1.2.5 doesnt seem so. |
| 00:18.26 | p3nguin | What's the problem with that? |
| 00:18.41 | SuperNull | 1.2.17 source isn't available. heh. |
| 00:18.46 | p3nguin | So 1.2.5 is too old and they added it later. Seems okay to me. |
| 00:18.46 | p3nguin | Oh! |
| 00:18.51 | SuperNull | ;) |
| 00:18.54 | SuperNull | bastards huh ? heh. |
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| 00:19.36 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:19.38 | SuperNull | i think... its just time to upgrade the server and force legacy sip customers to get firmware upgrades (it was kept due to some real old sip hardware that failed on 1.4 .. may work now on newer stuff) |
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| 00:19.42 | p3nguin | http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-1.2.17.tar.gz |
| 00:19.47 | p3nguin | What's wrong with this? |
| 00:19.58 | SuperNull | isn't realtime in the ast-addons ? |
| 00:20.04 | p3nguin | I have no idea. |
| 00:20.06 | SuperNull | or was that later moved heh. |
| 00:20.18 | p3nguin | I expect you to know that. |
| 00:21.03 | SuperNull | thats kind of what im trying to figure out .. im going into the old days of ast. |
| 00:21.41 | *** join/#asterisk wltjr (~wltjr@unaffiliated/wltjr) |
| 00:23.17 | SuperNull | alright.. well for future reference.. it is in fact with the asterisk primary source not addons at least in 1.2.17 |
| 00:24.50 | p3nguin | I never really cared too much about those things. I just used whatever they gave me when it was time to install. If something was changed, I changed to accommodate the differences. |
| 00:25.54 | SuperNull | the guy who previously worked here thought using just released features as a business model was a good idea. |
| 00:26.33 | SuperNull | realtime is somewhat useless for a provider if it doesnt update which server it is on. how can i point to it the right one on inbound without manual assignment |
| 00:26.46 | p3nguin | That's always a good idea. Use the newest possible things you can get your hands on before anyone else has time to test it. |
| 00:28.23 | SuperNull | he was a tester i will give him that.. but still. |
| 00:29.02 | p3nguin | It has to be put into production immediately so that any problems will be more evident due to serious outage. |
| 00:29.14 | SuperNull | exactly. |
| 00:29.20 | p3nguin | If you use it in a test environment, problems could easily be overlooked. |
| 00:29.35 | SuperNull | i implemented some intense mysql app stuff and it started failing due to intensity.. heh. |
| 00:29.37 | p3nguin | And that would be a waste of time. |
| 00:31.01 | SuperNull | im out .. long day.. later gents. |
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| 03:00.49 | SeRi | wow |
| 03:00.52 | SeRi | this is nice |
| 03:01.52 | p3nguin | Que? |
| 03:02.03 | SeRi | jelapanos en el taco? |
| 03:02.24 | SeRi | lol |
| 03:02.27 | p3nguin | No. Salsa verde en el taco! |
| 03:02.35 | SeRi | hell yea! |
| 03:02.36 | SeRi | lol |
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| 03:05.19 | SeRi | p3nguin: I am happy today |
| 03:05.21 | SeRi | no tears |
| 03:05.36 | p3nguin | I'm certain that your pillow will be happy tonight. |
| 03:05.41 | SeRi | my line works. I still cpmlained about the tech mess. I got one month credit |
| 03:05.56 | SeRi | s/cpmlained/complained/ |
| 03:06.15 | p3nguin | What did they end up doing to fix things? |
| 03:07.15 | SeRi | They had to work on the central hub for the naighborhood. |
| 03:07.24 | SeRi | I am sure thats not the right term |
| 03:07.33 | SeRi | but thats what they told me |
| 03:07.55 | p3nguin | I'm surprised no one else complained about the problems before. |
| 03:08.59 | SeRi | because I am the only one left with comcast |
| 03:09.06 | SeRi | my naighborhood is very small. |
| 03:09.10 | SeRi | about 4 houses |
| 03:09.25 | SeRi | when the storm hit and zapped all comcast customers they had issues like me |
| 03:09.32 | SeRi | they all split away from comcast |
| 03:09.44 | SeRi | hard headed me I stay and battle the beast |
| 03:10.11 | SeRi | and not 4 houses |
| 03:10.15 | SeRi | I meant about 40 houses |
| 03:11.32 | p3nguin | What other cable company is available there? |
| 03:14.24 | SeRi | in my area? |
| 03:14.25 | SeRi | none |
| 03:14.33 | SeRi | other isp is att |
| 03:14.40 | SeRi | I refuse to ever touch them |
| 03:17.03 | p3nguin | It seems that DSL has higher latency. |
| 03:17.05 | SeRi | p3nguin: you ever used flowroute? |
| 03:17.11 | p3nguin | Yes, every day. |
| 03:17.21 | SeRi | Is that who you use? |
| 03:17.28 | p3nguin | One of them. |
| 03:17.44 | p3nguin | I terminate through multiple providers. |
| 03:18.02 | p3nguin | I have DIDs with I think three. |
| 03:18.15 | SeRi | I see. I have a "test" account with them with a did as well. |
| 03:18.20 | p3nguin | VoIP.ms is my primary provider. |
| 03:18.50 | SeRi | right now my main provider is voip.ms and backup is flowroute |
| 03:18.59 | SeRi | I might switch it as flowroute is cheaper |
| 03:19.13 | p3nguin | They are? |
| 03:19.23 | *** join/#asterisk saxa (~saxa@201-50-97-161.user.veloxzone.com.br) |
| 03:19.31 | p3nguin | I thought they were slightly more. |
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| 03:19.48 | SeRi | .0098 vs .0125 |
| 03:19.55 | p3nguin | hmm |
| 03:20.08 | p3nguin | I'll have to login someday and check it out. |
| 03:20.47 | SeRi | yeap. |
| 03:21.04 | SeRi | the expensive part would be all the porting |
| 03:21.23 | p3nguin | I have no reason to move my DIDs. |
| 03:21.24 | SeRi | 7 did's with voip.ms |
| 03:21.32 | p3nguin | I doubt you do, either. |
| 03:21.42 | SeRi | agree |
| 03:21.57 | p3nguin | Origination service does cost more with flowroute. |
| 03:22.14 | SeRi | That is true |
| 03:22.21 | leifmadsen | flowroute is great for T.38 faxing |
| 03:22.35 | leifmadsen | at least the inbound in my testing |
| 03:22.39 | SeRi | leifmadsen: I didnt know that |
| 03:22.51 | p3nguin | Leave your DIDs where they are and use flowroute for termination if they really are cheaper. |
| 03:24.03 | SeRi | yeap... |
| 03:24.11 | SeRi | man work got hit so bad with a damn malware |
| 03:24.32 | SeRi | in the process I found a bug with officescann lol |
| 03:24.44 | SeRi | and I dont work with the windows stuff |
| 03:24.46 | SeRi | hahahaha |
| 03:25.46 | p3nguin | Sounds like someone needs fired for not having antivirus on the systems. |
| 03:25.59 | p3nguin | I'd fire my Windows guy over that. |
| 03:26.21 | p3nguin | First rule of Windows is to use antivirus. |
| 03:26.21 | SeRi | the virus came from a pc on the uk network |
| 03:26.27 | SeRi | usb thumb drive |
| 03:26.30 | p3nguin | Second rule of Windows is to use antivirus. |
| 03:26.34 | jpsharp | The first rule of Windows is don't use Windows? |
| 03:26.41 | SeRi | +1^^ |
| 03:26.45 | SeRi | lol |
| 03:26.56 | jpsharp | Says me, as I sit here at my Windows 7 machine ;) |
| 03:27.07 | SeRi | booooo! |
| 03:27.10 | p3nguin | blasphemer |
| 03:27.16 | SeRi | LMAO |
| 03:27.32 | jpsharp | I tried installing Mint, but the damn installer wouldn't boot on my machine. |
| 03:28.13 | p3nguin | Of course not, it's Ubuntu. |
| 03:28.22 | p3nguin | First rule of Linux is to avoid Ubuntu. |
| 03:28.40 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 03:28.44 | jpsharp | Gentoo? |
| 03:28.46 | p3nguin | You want something that works, use Mandriva or Arch. |
| 03:29.02 | p3nguin | If you like Windows, you should like Mandriva. |
| 03:29.22 | jpsharp | I just want shit that works and works well. |
| 03:29.30 | p3nguin | It's been a few years since I touched it, but I tested out Mandriva One and it was pretty nice feeling (if you like KDE). |
| 03:29.41 | jpsharp | sells the Windows box, gets an SGI machine. |
| 03:31.43 | SeRi | And if you want control use slackware... Just as Arch does. |
| 03:32.01 | SeRi | runs! NOOOO NO SGI!!!! |
| 03:32.20 | p3nguin | If you're a masochist, use CRUX. |
| 03:32.27 | SeRi | lmao |
| 03:32.35 | p3nguin | or OpenBSD. They feel about the same. |
| 03:33.49 | SeRi | omg openbsd... well people swares by them |
| 03:34.04 | SeRi | I just have to say its a pita |
| 03:34.15 | p3nguin | It's inconvenient, but it's a great OS. |
| 03:34.27 | SeRi | once running |
| 03:34.38 | p3nguin | It doesn't take long to get it running. |
| 03:34.40 | SeRi | I used licensed FreeBSD. |
| 03:34.46 | SeRi | back in 3.1 |
| 03:34.48 | leifmadsen | I just finished installing F17 on my new Lenovo laptop |
| 03:34.51 | leifmadsen | soooo good :) |
| 03:34.54 | p3nguin | But it does take a while to make it useful. |
| 03:35.01 | leifmadsen | i7 w/ 16GB and SSD |
| 03:35.03 | SeRi | leifmadsen: I heard is nice |
| 03:35.05 | leifmadsen | very |
| 03:35.08 | leifmadsen | T430s |
| 03:35.37 | leifmadsen | and now I'm done for the night... ruby all ready to go, but it's friday night, so I'm out! |
| 03:36.19 | SeRi | peace |
| 03:36.30 | p3nguin | Piece! |
| 03:41.34 | SeRi | out? |
| 03:42.14 | p3nguin | Piece out, my brahtha! |
| 03:46.02 | SeRi | lol |
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| 06:07.49 | SeRi | man this conf call is killing me. |
| 06:13.06 | ChannelZ | Maybe you should get off it before you die. |
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| 06:56.23 | SeRi | ChannelZ: I am trying. |
| 06:56.37 | SeRi | I finally got out but now I was redirected to a group chatt |
| 06:56.42 | SeRi | man do I hate lotus notes |
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| 08:08.46 | pppingme | looking for an autodialer that will take a list of phone numbers and play a .wav (or whatever format), prefer something non-gui, for use for something along the lines of closings, etc (think snow-school closings) |
| 08:12.07 | [TK]D-Fender | pppingme: This is a minimal amount of scripting to write yourself |
| 08:13.59 | pppingme | on the asterisk server directly, or as a sip client on a 2nd computer? my preference would be on the 2nd computer, with the autodialer acting as a sip extension |
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| 08:57.57 | ChannelZ | pppingme: look at 'call files' |
| 08:58.46 | ChannelZ | Much easier to do it server-side than involve a client. Though I suppose maybe there is a scriptable SIP client. But really, why bother |
| 09:01.00 | pppingme | playing with droping something in the outgoing folder |
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| 11:04.04 | Chotaire | Hi all. I've been trying to install a fresh asterisk current on an opensuse 12.2 vm that runs on ESXI. i installed libpri, dahdi and the asterisk source, installed the sample configuration files, added one test sip account and made a random call, once the client is trying to dial out, I get an immediate segmentation fault with the following syslog message: |
| 11:04.05 | Chotaire | Dec 1 11:48:20 to kernel: asterisk[12793]: segfault at b5b3837c ip b674ae9d sp b5b382c0 error 6 in chan_sip.so[b6725000+d5000] |
| 11:04.59 | Chotaire | despite the syslog message, there is no further details in asterisk console, it immediately segfaults. any idea? |
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| 11:24.49 | funky1 | hi all, i'm having some trouble with my audio during calls, probably NAT related but not sure, suggestion here was to run rtp and sip debug during call which i did, but have trouble identifying the issue in the logs, would someone be so kind to have a look and point me on some issues? my problem described and log output can be found here: http://lists.digium.com/pipermail/asterisk-users/2012-November/276279.html |
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| 11:43.50 | Chotaire | the segfault was caused by LOWMEMORY setting btw. |
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| 12:08.46 | m_tadeu | hi...I have a register entry in sip.conf. when asterisk tries to register in that remote sip server, I'm getting timeout. This was working a few days ago and no changes were made on my asterisk. My provider says there's nothing wrong with my account on the remote server |
| 12:09.23 | m_tadeu | using wireshark I see my asterisk machine sending the register request, but I get no response |
| 12:10.13 | m_tadeu | what can I do to check what the problem is |
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| 12:40.54 | ghost75 | what phone displays, is callerid(name) ? |
| 12:50.41 | ghost75 | nice, there is an LookupCIDName cmd |
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| 13:20.04 | ghost75 | any1 uses phones over wlan? |
| 13:20.46 | WIMPy | Yes. You don't want to. |
| 13:21.12 | ghost75 | because i see under sip peers it has 200ms delay |
| 13:21.44 | ghost75 | so i guess its unusable |
| 13:23.08 | WIMPy | It does not mean that audio would have the same delay. |
| 13:24.05 | WIMPy | OTOH the packetization would come on top of all other delays. |
| 13:24.21 | ghost75 | so why i wont use it |
| 13:38.34 | *** join/#asterisk qakhan (~qakhan@70-90-90-130-BusName-dc.hfc.comcastbusiness.net) |
| 13:39.37 | qakhan | hi all when i call out and run any IVR i get this message |
| 13:39.39 | qakhan | [Dec 1 08:28:13] WARNING[16187]: file.c:1297 waitstream_core: Unexpected control subclass '-1' |
| 13:39.51 | qakhan | and i cannot hear IVR |
| 13:55.33 | ghost75 | is it enough to put this in incoming context: exten => s,1,LookupCIDName() |
| 13:56.25 | p3nguin | That doesn't seem to be a valid app. |
| 13:56.33 | p3nguin | And what are you trying to accomplish? |
| 13:56.50 | *** join/#asterisk kontinuity (~kontinuit@122.178.197.186) |
| 13:57.24 | ghost75 | Looks up the Caller*ID number on the active channel in the Asterisk database (family 'cidname') and sets the Caller*ID name. |
| 13:57.41 | p3nguin | What version of asterisk do you use? |
| 13:57.47 | ghost75 | 1.8 |
| 13:57.57 | p3nguin | 1.8 isn't a version. |
| 13:58.16 | WIMPy | Not an Application I came across. And my Asterisk doesn't know it, either. |
| 13:58.28 | p3nguin | 1.8 is a branch. The 1.8 branch has many versions. I'm asking you what version. |
| 13:59.03 | p3nguin | I use the 1.8 branch and it's not an app in my version. |
| 13:59.08 | ghost75 | 1.8.13.1~dfsg-1~bpo60+1 |
| 13:59.30 | ghost75 | i just found the cmd on a old webpage |
| 13:59.34 | p3nguin | Back to the other question. What are you trying to accomplish? |
| 13:59.51 | ghost75 | (14:58:03) ghost75: Looks up the Caller*ID number on the active channel in the Asterisk database (family 'cidname') and sets the Caller*ID name. |
| 13:59.53 | WIMPy | And does YOUR Asterisk know it? |
| 14:00.35 | p3nguin | If you can't go to the CLI and get the usage of it by doing core show application SomeAppName, it isn't valid for your system. |
| 14:00.48 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 14:02.06 | ghost75 | oh i saw it can be done with this now: Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) |
| 14:02.51 | WIMPy | Interesting use of "now". |
| 14:03.31 | p3nguin | If you already have caller id numbers and names stored in the astdb, that could be useful. |
| 14:03.57 | ghost75 | yes, i was not aware the lookup can be done so easily :) |
| 14:05.10 | p3nguin | If you don't already have that data in your astdb, the Set() will be useless. |
| 14:05.19 | ghost75 | i have |
| 14:08.45 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 14:08.45 | *** mode/#asterisk [+o file] by ChanServ |
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| 14:49.51 | ghost75 | p3nguin: to come back about originate call from yesterday, if done from dialplan the first cmd is dial SIP/ext and the second Dial(Local/{EXTEN}@itsppeer right? |
| 14:50.58 | ghost75 | first cmd of course originate, not dial |
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| 15:03.28 | qakhan | hi all when i call out and run any IVR i get this message |
| 15:03.31 | qakhan | [Dec 1 08:28:13] WARNING[16187]: file.c:1297 waitstream_core: Unexpected control subclass '-1' |
| 15:03.33 | qakhan | and i cannot hear IVR |
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| 15:13.12 | ghost75 | aha now originate over web works :) |
| 15:13.28 | ghost75 | first local, not second |
| 15:15.21 | kontinuity | does anyone have an unfavourable experience with Kamailio? I am planning to deploy it in front of Asterisk for abuse mitigation |
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| 15:18.52 | carrar | Why not opensips? |
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| 15:28.37 | wasabi | Howdy folks. So, I'm new to actually working with Asterisk, and have some questions regarding queues. How are agent logins handled? Does Asterisk just store away the callback number, and ring that? or does the user have to be on an asterisk registered extension? |
| 15:29.41 | wasabi | For instance, I'm wondering how to get it to work properly with SIP addresses external to Asterisk. On another phone sytem. |
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| 15:57.24 | m_tadeu | is it possible to register asterisk in another sip proxy through cli? |
| 16:00.22 | file | no. |
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| 16:44.15 | Chotaire | hm, what happened to the command soft hangup? |
| 16:44.29 | WIMPy | channel request hangup |
| 16:45.08 | Chotaire | let's try that... |
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| 16:48.09 | Chotaire | thanks, php updated. |
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| 17:59.30 | g_r_eek | hello channel, i installed asterisk on centos according to this guide: http://blogs.digium.com/2012/11/05/how-to-install-asterisk-11-on-centos-6/ but it does not seem to run |
| 17:59.32 | g_r_eek | [root@localhost asterisk]# service asterisk start |
| 17:59.32 | g_r_eek | Starting asterisk: |
| 17:59.32 | *** join/#asterisk kontinuity (~kontinuit@122.178.197.186) |
| 17:59.33 | g_r_eek | [root@localhost asterisk]# |
| 18:00.09 | g_r_eek | i think its a user/security issue but am not sure is there is need to do with the permissions in order to start it? |
| 18:01.53 | ghost75 | did u check running process |
| 18:02.12 | *** join/#asterisk greenwolf (4257021f@gateway/web/freenode/ip.66.87.2.31) |
| 18:04.52 | WIMPy | I see nothing that says it doesn't run. |
| 18:06.19 | [sr] | offhi WIMPy |
| 18:06.31 | WIMPy | Moin [sr] |
| 18:06.31 | [sr] | ops |
| 18:06.35 | [sr] | hi WIMPy |
| 18:06.48 | [sr] | dont mind the extra "off" |
| 18:06.54 | ghost75 | switches on |
| 18:06.59 | [sr] | :) |
| 18:08.35 | g_r_eek | ghost75: let me check |
| 18:09.13 | g_r_eek | no it does not appear in any process |
| 18:09.22 | ghost75 | then check error log |
| 18:09.39 | g_r_eek | ghost75: in /var/logs? |
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| 18:09.47 | ghost75 | yes |
| 18:09.54 | g_r_eek | ok |
| 18:10.55 | zamba | i'm looking for a ISDN card to use for my asterisk box.. any recommendations? WIMPy? you had some earlier? |
| 18:11.26 | WIMPy | Any cheap one. |
| 18:11.34 | zamba | i have four lines, so i'm looking for a BRI interface card, i guess |
| 18:11.48 | zamba | WIMPy: you had a concrete suggestion as to what so search for on ebay? what was that again? |
| 18:12.22 | WIMPy | 4 lines or 4 channels? |
| 18:13.27 | zamba | hm.. not sure about the difference.. |
| 18:13.39 | zamba | but there's no problem having two PCI cards |
| 18:14.33 | WIMPy | 4 channels = 2 lines, 4 lines = 8 channels |
| 18:14.41 | zamba | 2 lines, then |
| 18:14.48 | zamba | to T1 boxes |
| 18:14.51 | zamba | two* |
| 18:14.52 | zamba | or whatever they're called |
| 18:15.21 | WIMPy | Ok, then if you have two PCI slots to spare ou can use the real cheap HFC-S things. |
| 18:15.31 | WIMPy | NT or NTBA. |
| 18:15.38 | zamba | NT i meant.. yeah |
| 18:15.46 | zamba | HFC-S yielded no results on ebay.com |
| 18:16.14 | WIMPy | I guess so. |
| 18:16.28 | ghost75 | whats an NT ? |
| 18:16.38 | WIMPy | Network Terminator |
| 18:16.50 | ghost75 | here in germany i only seen NTBA |
| 18:17.23 | WIMPy | That specifically a Network Terminator for Basic Access. |
| 18:17.55 | WIMPy | And if you're "here", you might want to join #asterisk-de . |
| 18:18.08 | zamba | http://www.ebay.de/itm/1-BRI-ISDN-Karte-CologneChip-HFC-S-S0-fur-Asterisk-Trixbox-Elastix-/130811735991?pt=DE_Computer_Peripherie_Netzwerk&hash=item1e74fcafb7#ht_500wt_1156 |
| 18:18.11 | zamba | is that a guy buy? |
| 18:18.18 | zamba | eerr.. where's my english today? |
| 18:18.19 | zamba | good* |
| 18:19.04 | WIMPy | It's ok, but if you can wait for an auction you should be able to get it considerably cheaper. |
| 18:19.17 | zamba | when are the auctions? |
| 18:19.42 | carrar | snip it! |
| 18:19.50 | carrar | snipe |
| 18:19.52 | WIMPy | Tell ebay to show auctions only. Then sort by end time. |
| 18:20.07 | g_r_eek | http://pastebin.com/BdAFxhSN |
| 18:20.14 | WIMPy | Did I just explain how to use ebay? |
| 18:20.23 | zamba | hehe |
| 18:20.26 | g_r_eek | any clue? |
| 18:21.02 | WIMPy | g_r_eek: Looks like you tried to start Asterisk twice. |
| 18:22.49 | zamba | WIMPy: i have actually been looking at ebay.de now and then, and have never found any auctions with those cards |
| 18:22.57 | zamba | WIMPy: you speak about them like they happen more or less every day :) |
| 18:23.19 | WIMPy | Not everyone mentions the chip. |
| 18:23.50 | zamba | then i'd be really happy if you wouldn't mind sharing some of your ebay ninja skills regarding acquiring these cards :) |
| 18:23.57 | zamba | what you search for, for instance |
| 18:23.57 | WIMPy | Just keep away from the AVM Fritz cards. |
| 18:24.40 | ghost75 | fritz card </3 |
| 18:27.21 | WIMPy | Hmm. Tons of the AVM stuff :-( |
| 18:27.44 | g_r_eek | WIMPy: hmmm rebooted when i try to connect to console with asterisk -rvvv i get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 18:29.44 | zamba | WIMPy: i'm giving the seller an offer of 10 EUR per unit |
| 18:29.50 | zamba | WIMPy: we'll see what he responds |
| 18:31.02 | WIMPy | zamba: So far only 390499936239 |
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| 18:31.19 | WIMPy | Maybe they have all been bought by Asterisk users? |
| 18:31.46 | zamba | i checked half a year ago.. same then |
| 18:32.16 | zamba | WIMPy: so that card is a good buy? |
| 18:32.31 | WIMPy | ISA, PCMCIA or USB :-( |
| 18:32.52 | WIMPy | Well, USB does work, but I personally have ssues with using USB stuff on a server. |
| 18:33.00 | zamba | same here |
| 18:33.02 | ghost75 | g_r_eek: check your port setup |
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| 18:34.02 | SeRi | dijib: hey! |
| 18:34.31 | g_r_eek | ghost75: i did the default install, i check the conf files for port? |
| 18:34.48 | WIMPy | Ther definitely used to be more of them on ebay. |
| 18:35.25 | ghost75 | ERROR[22463] pbx_dundi.c: Unable to bind to 0.0.0.0 port 4520: Address already in use |
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| 20:21.23 | slav3_kitten | i'm 1/2 tempted to call this number |
| 20:21.38 | slav3_kitten | 441212791341 |
| 20:21.58 | slav3_kitten | it's shown up 5 times from 2 different IPs trying to get outbound calling |
| 20:26.08 | ChannelZ | Is that a international number? (44?) |
| 20:26.47 | ChannelZ | UK |
| 20:27.41 | WIMPy | that's the most called number range. |
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| 20:31.40 | wltjr | anyone know why a cisco 501g shows busy on all lines when only 1 is in use? |
| 20:32.53 | [TK]D-Fender | wltjr: Show us |
| 20:33.16 | [TK]D-Fender | wltjr: Also clarifying your definition of "busy" |
| 20:35.02 | wltjr | [TK]D-Fender: http://pastebin.com/SAhH1waJ returning busy status it seems |
| 20:35.25 | wltjr | [TK]D-Fender: has something to do with the damn call waiting feature, turning that back on allows it to not be busy, starting to hate these phones |
| 20:35.37 | slav3_kitten | yea it's uk |
| 20:35.46 | wltjr | [TK]D-Fender: trying to get lines to ring in order, when they are on one, if ext 1 busy, ring 2, etc |
| 20:36.18 | [TK]D-Fender | wltjr: Right now you are dialing multiple devices simultaneously, not sequentially. |
| 20:39.02 | wltjr | [TK]D-Fender: its both, one sec |
| 20:40.09 | wltjr | [TK]D-Fender: I know this is not the best way to do this stuff I am going to work on a better way |
| 20:41.11 | [TK]D-Fender | I see your swapping 2 memebers in the 2nd dial |
| 20:41.22 | wltjr | [TK]D-Fender: http://pastebin.com/VJh7jjK6 |
| 20:41.26 | [TK]D-Fender | and 2 again after that |
| 20:41.44 | [TK]D-Fender | wltjr: Indded there is a better way to do this |
| 20:42.15 | [TK]D-Fender | because you are issuing up to 3 calls to 2 of the same members, when what you really want to do is stop dialing X, and add Y as you dgo, but leave Z ringing the whole time |
| 20:42.43 | wltjr | [TK]D-Fender: on sec on phone, |
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| 20:54.53 | wltjr | [TK]D-Fender: damn clients, want to discuss things when I am trying to work on them :) |
| 20:55.21 | wltjr | [TK]D-Fender: p3nguin provided me a snippet yesterday I think is more along the lines of what I should be doing, but I could not really understand the snippet |
| 20:55.45 | wltjr | http://pastebin.com/pvy0QEkn |
| 20:56.07 | wltjr | I know I will need to check the status of each line on each phone before I ring that line on that phone, will be a pita, but its the proper way I guess |
| 20:56.22 | [TK]D-Fender | wltjr: Umm.. that lokos nothing like your multi-dial... |
| 20:56.27 | WIMPy | To do what? |
| 20:56.50 | [TK]D-Fender | And is an overkill AsTDB Driven process |
| 20:56.54 | wltjr | [TK]D-Fender: no clue that was the example that was provided, and I figured I would have to adapt it, but it did seem quite different |
| 20:57.08 | wltjr | [TK]D-Fender: part of what I was wondering about the db vars and stuff |
| 20:57.27 | wltjr | starting to dislike these phones though, I have not seen others act like these do |
| 20:57.35 | [TK]D-Fender | wltjr: You seemed to simply want to ring a bunch of phones. Some not ringing as long as others, some being delayed fom joining in, etc.... |
| 20:58.23 | wltjr | [TK]D-Fender: yes, 2 goals, ring all phones when call comes in, then a couple phones are designated for people to actually answer them, the front desk, those I want to ring multiple times, 8 extensions per, if they are on line 1, I want line 2 to ring, etc |
| 20:59.07 | [TK]D-Fender | wltjr: What are these "line 1" and "line 2: things exactly? |
| 20:59.10 | WIMPy | Why do you want to change "line"? |
| 20:59.13 | wltjr | [TK]D-Fender: I had lots of problems ringing additional lines/extensions when one was in use, I had to add stuff like call-limit to sip.conf, but seems that was cause of call waiting being enabled in phones, but without call waiting enabled, if any 1 extension is in use, all 8 show busy, grrr |
| 20:59.42 | wltjr | [TK]D-Fender: sorry, next extension on the same phone, so if the first extension is in use, they are on that, ring the next one when a call comes in |
| 20:59.43 | WIMPy | Makes sense to me. |
| 21:00.03 | WIMPy | Again: Why? |
| 21:00.15 | [TK]D-Fender | wltjr: Most people tend to only create a single SIP Peer per phone.... not multiple separate SIP users to call. |
| 21:00.28 | wltjr | think call center, front desk people are responsible for answering phones, and many times more than one call is coming in, and only one person, thus put one on hold answer next line |
| 21:00.28 | [TK]D-Fender | wltjr: The concept of 1 identitiy, but supporting multiple calls |
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| 21:01.17 | wltjr | [TK]D-Fender: that is a new concept to me, I have always done multiple per phone, maybe legacy idk, I get what your saying, inbound is a sip peer/trunk and we get unlimited lines/connections over that |
| 21:01.24 | WIMPy | Let the phone do it instead of giving yourself a hard time. |
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| 21:01.39 | wltjr | WIMPy: how? :) |
| 21:01.55 | WIMPy | Just don't care. |
| 21:02.01 | [TK]D-Fender | Indeed start with that. undo having multiple registration on each phone and then we can help you fix your dialing pattern |
| 21:02.05 | WIMPy | The phone will just do it. |
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| 21:02.37 | wltjr | [TK]D-Fender: so one registration per phone? their config seemed to want 2, I had to go to advanced menu to provide ones for others |
| 21:03.22 | WIMPy | Whose config? And what "others"? |
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| 21:04.05 | WIMPy | Yo can have multiple calls even on only one account. |
| 21:04.07 | wltjr | WIMPy: the spa 501g configuration, in its web interface, it had a place to enter the proxy, and user id for the first 2 lines, but not the remaining 6, I had to go to advance menu to provide proxy and user for the remaining 6 |
| 21:04.34 | WIMPy | Using multiple accounts can make sense if you want different ringtones for different destination or such things. |
| 21:04.39 | wltjr | WIMPy: sure, but then how do I control where its ringing on the phone? like what extension/line on the phone |
| 21:04.41 | [TK]D-Fender | wltjr: 501g's are pretty basic dumb phones. Just supply 1 and it should cascase to the other buttons. |
| 21:05.05 | WIMPy | You only configure one of them. |
| 21:05.19 | WIMPy | And you don;t control anything. That's up to the phone. |
| 21:05.25 | [TK]D-Fender | yup |
| 21:05.28 | wltjr | client won't like that |
| 21:05.49 | wltjr | they are expecting things to be like the old key phone system, which I know can't happen |
| 21:05.53 | WIMPy | But if it's as clever als the the old SPA962s it might stop at two calls. |
| 21:06.24 | WIMPy | Don't make our brains hurt. |
| 21:06.56 | WIMPy | But really that should be left to the phone. If you don't like the UI, go for another type of phone. |
| 21:07.01 | wltjr | I wanted to control what lines/extension on the phones rang, and have things mapped to certain lines/extension |
| 21:08.28 | wltjr | like on front desk phones, if call comes in via local number ring right 4 lines in order if in use, and if 800 number ring left, office phones are different |
| 21:08.55 | WIMPy | Ok, but what you describes it to use different accounts for the same extension. |
| 21:09.05 | wltjr | office phones local rings ext/line 5 on each phone, 800 rings ext/line6, 1-4 are used for calls being transfered or other calls |
| 21:09.26 | wltjr | WIMPy: yes why I have a sip account per extension |
| 21:09.38 | WIMPy | Yes, you can do that. But that was not what I read above. |
| 21:09.59 | wltjr | WIMPy: sorry I might not be explaining it correctly |
| 21:11.34 | WIMPy | With what you're saying now you'd have exactely one account per extension. |
| 21:11.55 | wltjr | WIMPy: yes, that is what I have 8 sip accounts per phone, 8 lines/extensions per phone, big sip.conf :) |
| 21:12.52 | wltjr | WIMPy: but the phones allow more than one call for the same extension/line and it will ring that same one not others, thus I thought I had to do it this way to get the other extensions to ring, not even sure how you would answer the call coming in on the same line already in use |
| 21:13.17 | wltjr | to get any one line to be busy, I had to introduce call-limit for each sip account, but that broke call transfering |
| 21:13.19 | WIMPy | I don;t know what the current FW will allow you to do, but fot the 4 keys for transferred call, you should still use one account and just map it to 4 keys. |
| 21:13.51 | wltjr | WIMPy: how do I map one sip account to multiple keys/buttons? |
| 21:14.19 | WIMPy | On the old SPA962 you couldn't. Don;t know the newer models. |
| 21:14.41 | wltjr | WIMPy: well I have them all mapped to each directly, its the rest of the function I am working on |
| 21:14.58 | wltjr | WIMPy: without call-limit, a line in use never shows as busy |
| 21:15.05 | wltjr | even when checking status |
| 21:15.26 | WIMPy | Yes, but you don;t need to know if you let the phone handle it. |
| 21:15.32 | wltjr | I can get rid of call-limit by disabling call waiting, but then the phone can only receive a single call, 1 line in use, all 8 show as busy |
| 21:16.10 | wltjr | WIMPy: the phone doesn't handle it well, it doesn't activate the other keys/extensions, just rings on the one your already on |
| 21:16.12 | WIMPy | Sure |
| 21:16.36 | WIMPy | Maybe it's not the right phone for your situation. |
| 21:16.45 | wltjr | WIMPy: starting to think that might be the case :( |
| 21:16.47 | wltjr | damit |
| 21:17.14 | wltjr | had to be a reason why the ciscos were the cheapest, guess I should have gone grandstream, but I had one of those fail before |
| 21:17.31 | WIMPy | ~gs |
| 21:17.31 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:17.55 | wltjr | not impresed with their ata's either, the spa 112, it works but tends to take multiple tries for analog stuff, like fax |
| 21:18.10 | wltjr | infobot: thanks :) confirms my feelings on them |
| 21:18.10 | infobot | pas de quoi, wltjr |
| 21:19.22 | WIMPy | People here seem to like Polycom. And I can tell you that the old Snoms allow you to assign line keys to accounts. |
| 21:19.28 | wltjr | I think I might be able to do what I want with the phones I am using now, just will have to configure and hack around their quirks, likely some ugliness vs elegance, will consider replacing the phones, already regretting the no lcds :) |
| 21:20.45 | wltjr | WIMPy: I liked the snoms except for price, haven't worked with polycoms, just older cisco and linksys phones, 7960s I believe |
| 21:21.08 | wltjr | though the 7960s and Linksys ones I did a sip account per line/extension on the phone |
| 21:21.12 | WIMPy | I haven;t seen a Polycom so far, either. |
| 21:21.15 | Chainsaw | I went from Cisco 7960 to Polycom IP 670. |
| 21:21.23 | Chainsaw | There isn't enough money in the world to make me switch back. |
| 21:22.01 | wltjr | I have heard good stuff about the polycoms, I think their prices might have been really close to Ciscos so just went that route, and got Linksys ones before Cisco rebranded them, but those were ok |
| 21:22.07 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 21:22.26 | wltjr | spa-942, I thought these were similar or the same |
| 21:22.57 | Chainsaw | wltjr: The 7960 was SCCP hardware. The SIP retrofit had an awkward UI; it was slow and the protocol adherence left to be desired. Rebranded Linksys is probably a better bet. |
| 21:23.21 | wltjr | Chainsaw: yes I recall having to change their firmware for SIP support |
| 21:23.34 | wltjr | used ciscos that ran on the skinny protocol before that, horrible mistake :) |
| 21:23.41 | wltjr | but they worked :) |
| 21:23.55 | WIMPy | I've got a 962 catching dust here. Don't find it usable. |
| 21:24.01 | Chainsaw | wltjr: Just a very annoying situation to have phone hardware that only really did SCCP well and made sort of a half-hearted attempt at SIP... |
| 21:24.21 | Chainsaw | wltjr: Especially when interfacing with an Asterisk server that only really does SIP well and made sort of a half-harted attempt at SCCP. |
| 21:24.26 | Chainsaw | wltjr: Worst of both worlds. All the time. |
| 21:24.38 | wltjr | Chainsaw: I really haven't had any issues with the 7960, clients have been using them for years without issue |
| 21:25.19 | wltjr | probably should have got some of those used or something for this client, vs going with new low end/price phones |
| 21:25.27 | Chainsaw | wltjr: As long as you keep to u/aLaw codecs with SIP over UDP and there never is any intermittent packet loss, you might be fine. |
| 21:25.51 | Chainsaw | wltjr: *Any* loss during attended transfers and you are in a world of hurt. Nothing works. |
| 21:25.52 | *** join/#asterisk k611 (~K610@cable-78.29.241.186.coditel.net) |
| 21:26.10 | wltjr | Chainsaw: yes no transcoding all ulaw, and just udp, no tcp, doesn't seem to be packet loss, though I do experience that at times with comcast, waiting for a tech to come out tomorrow, though its resolved itself... |
| 21:26.34 | Chainsaw | wltjr: Don't try to enable SIP over TCP with a 7960. It can't do it. |
| 21:26.34 | wltjr | Chainsaw: they don't transfer much if at all, thus never had problems there, could be why I still like them :) |
| 21:27.00 | wltjr | Chainsaw: ok I wont :), but stuff is working for that client, has for years, so I don't touch anything, just upgrade * and the system :) |
| 21:27.51 | wltjr | just gotta dial in stuff for this other client before they kill me or maim me |
| 21:40.05 | slav3_kitten | what about 7690's? |
| 21:40.41 | Chainsaw | slav3_kitten: That's post-SCCP. It should be... less awful. |
| 21:40.53 | slav3_kitten | 7960* |
| 21:41.02 | slav3_kitten | had som dyslexia |
| 21:41.35 | Chainsaw | slav3_kitten: But still... if the Cisco is a car the Polycom is a spaceship. I know what I'd pick. |
| 21:42.19 | slav3_kitten | prints out a bunch of labels reading "polycom" then sticks it over the 6 phones he just installed |
| 21:42.29 | slav3_kitten | i got the starship titanic over here |
| 21:42.53 | slav3_kitten | but they are poe via my old cisco non 802.3af inline power switch |
| 21:43.02 | SeRi | Polycom are awesome :P |
| 21:43.05 | slav3_kitten | sip firmware does leave much to be desired |
| 21:43.22 | slav3_kitten | documentation even more so |
| 21:45.08 | slav3_kitten | i really want to pick up a polycom to replace the 7911 i have here in the living room |
| 21:45.18 | Chainsaw | SeRi: The HTTP provisioning especially. I have PHP scripts to drive things. |
| 21:45.33 | SeRi | Chainsaw: nice! |
| 21:45.35 | slav3_kitten | so A) i can learn polycom, an B) a full featured speaker phone |
| 21:45.41 | SeRi | I am old school. tftp :P |
| 21:45.50 | slav3_kitten | i love tftp |
| 21:45.52 | Chainsaw | SeRi: TFTP went out with the Ciscos. |
| 21:46.00 | SeRi | Chainsaw: lol! |
| 21:46.06 | slav3_kitten | what's wrong with tftp Chainsaw ? |
| 21:46.09 | SeRi | slav3_kitten: a nice 560 would do |
| 21:46.14 | Chainsaw | SeRi: Seriously. I'd have to pre-program a phone or know its MAC address ahead of time. |
| 21:46.47 | slav3_kitten | ah, i could see that being a problem in a business setting |
| 21:46.57 | Chainsaw | SeRi: A Polycom just requests a specific URL that I specify in DHCP to get provisioned. By virtue of a rewrite rule and the URL it uses, I know the MAC and generate the config file on the fly. |
| 21:47.06 | slav3_kitten | SeRi, a polycom 560? |
| 21:47.11 | Chainsaw | SeRi: Which I then write out, because that phone now has a station ID and is provisioned. |
| 21:47.48 | Chainsaw | SeRi: This is a business setting and my users are, for lack of a better word, spoiled. They expect to plug a phone in and have it working. That it spends a few minutes updating itself is already an inconvenience. |
| 21:48.05 | SeRi | slav3_kitten: Yes. |
| 21:48.14 | Chainsaw | SeRi: (And when I mean plugging a phone in, I mean an unconfigured, boxed unit from the store room) |
| 21:48.29 | SeRi | Chainsaw: good job man! |
| 21:48.29 | slav3_kitten | SeRi, too big, i need one physically the same size as a 7911 or smaller |
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| 21:48.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:48.42 | SeRi | slav3_kitten: 320 |
| 21:49.30 | slav3_kitten | does it support cisco poe? |
| 21:51.40 | *** join/#asterisk timeshell (~timeshell@184.151.190.202) |
| 21:51.44 | timeshell | Hi |
| 21:52.22 | timeshell | When making 1.8.15.0 app_page is XXX'd for selection. What is the requirement for this app to be selectable in menuselect? |
| 21:56.29 | WIMPy | MeetMe -> dahdi |
| 21:56.52 | Chainsaw | timeshell: And try with 1.8.18.0; 1.8.15.0 is now three versions out of date and known insecure. |
| 21:57.23 | timeshell | you need dahdi? |
| 21:57.36 | timeshell | For SIP paging? |
| 21:57.39 | WIMPy | MeetMe needs dahdi, yes. |
| 21:57.48 | WIMPy | That's why we have ConfBridge now. |
| 21:58.16 | timeshell | Yah, but asterisk-gui doesn't seem to know confbridge |
| 21:58.43 | timeshell | Still refers to page |
| 21:58.55 | WIMPy | ConfBridge in 1.8 was rather rudimental anyway. |
| 21:59.12 | timeshell | so just chan_dahdi? |
| 21:59.58 | WIMPy | No, dahdi itself. Not sure if you need chan_dahdi. |
| 22:02.20 | timeshell | dahdi itself means dahdi drivers/utils? |
| 22:02.56 | WIMPy | The kernel module. |
| 22:03.34 | WIMPy | As you don't have anything to configure anything, you don't need the utils. |
| 22:07.58 | SeRi | slav3_kitten: Yes it does. I have two 320's and one 560 and soon a 670. |
| 22:08.00 | SeRi | I brb |
| 22:08.06 | kostucha | hi |
| 22:08.33 | wltjr | great so I am starting to see where some of my problems come from the 501g, has 8 lines, and can have 2 calls per line, grrr |
| 22:09.55 | WIMPy | just like the old ones. |
| 22:10.20 | wltjr | explains why when a line is in use, it doesnt' show as busy, I guess each one needs 2 calls to become busy... |
| 22:10.29 | WIMPy | And don't try to change between multiple calls on different accounts. It hurts. |
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| 22:11.05 | wltjr | WIMPy: how do you change calls on the same line? :) |
| 22:11.41 | WIMPy | IIRC just press that line button. |
| 22:11.43 | wltjr | lesson learned, don't buy or use spa 501g phones |
| 22:11.57 | wltjr | WIMPy: if your already on that line? what happens to current call placed on hold? :) |
| 22:12.08 | WIMPy | Sipura have never been known for good software. |
| 22:12.15 | WIMPy | yes |
| 22:12.54 | wltjr | sure, but this has Cisco's name on it ;) lazy, didn't improve, just changed names bad Cisco |
| 22:13.24 | slav3_kitten | SeRi, wait an you plug it into pre standard poe it just works? |
| 22:13.25 | wltjr | WIMPy: the linksys ones I worked with before were not this bad, or maybe they were and I just did not do advanced stuff with them |
| 22:14.34 | WIMPy | I can't use it. |
| 22:15.04 | WIMPy | A friend once asked if he could use it and I said yes, but you don't want to. |
| 22:15.24 | WIMPy | Well, he did take it, but a few weeks later he brought it back. |
| 22:15.41 | wltjr | do polycoms have good native sip support? guess I need to look into alternatives to replace the 501g phones maybe |
| 22:16.09 | wltjr | though their shared line thing is kinda cool, I could replicate function of the old key phone system, staff would like that :) |
| 22:16.31 | Chainsaw | wltjr: By the book, everything works SIP support, yes. |
| 22:16.43 | Chainsaw | wltjr: They also have documentation that'll actually match what the device does. |
| 22:17.13 | wltjr | are the digium phones any good? |
| 22:17.30 | wltjr | Chainsaw: thats quite nice, acurate documentation gotta love it |
| 22:17.46 | Chainsaw | wltjr: Haven't used them, but they look like Polycom OEMs to me. |
| 22:18.05 | Chainsaw | wltjr: Shape of the handset & case, HD voice branding, etc... |
| 22:18.34 | WIMPy | And a binary module that smells like security through obscurity. |
| 22:18.50 | Chainsaw | WIMPy: Well I was trying to be nice by not mentioning that aspect. |
| 22:19.15 | Chainsaw | But yes, in all honesty... put a picture of an IP 670 (Polycom) next to the Digium phone offering and ask yourself how "custom" it really is. |
| 22:23.03 | ChannelZ | They look similar in a way that phones tend to look similar. |
| 22:24.04 | Chainsaw | Not really. Put a Cisco or a Grandstream next to it. |
| 22:24.18 | Chainsaw | Those don't share any design elements. |
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| 23:13.32 | ghost75 | any1one tried this: http://sourceforge.net/projects/ast-php-vm/ |
| 23:13.55 | Chainsaw | ghost75: Seems abandoned. Last update 3 years ago? |
| 23:14.32 | ghost75 | the perl script in asterisk is from 2002 |
| 23:15.43 | Chainsaw | ghost75: But by virtue of shipping with the sources, you know it has been updated to work. |
| 23:16.33 | ghost75 | tried any? |
| 23:17.48 | ghost75 | the php script was written because the perl script needs setuid root |
| 23:17.49 | Chainsaw | ghost75: I'm cruel enough to make my users call into their voicemail boxes. |
| 23:18.08 | wltjr | Chainsaw: evil, I make mine check their email :) |
| 23:18.31 | ghost75 | i have both, just want to try it for fun |
| 23:18.35 | Chainsaw | wltjr: Yes, by default they get e-mail as well. Some have just the e-mail and no on-phone notification. |
| 23:18.58 | Chainsaw | wltjr: My favourite, I'll never forget this, is the accountant putting blu-tac over the red voicemail notification light. |
| 23:19.37 | wltjr | Chainsaw: nice, I was trying to eliminate only one person hearing or checking voice mails at random times, since email tends to be more frequently monitored |
| 23:22.16 | ghost75 | the perl script says "comedian mail login" oO |
| 23:22.51 | Chainsaw | That's the name of the default voicemail backend, yes. |
| 23:26.52 | ghost75 | that perl script still kind of works |
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