| 00:12.16 | pabelanger | citywok: Most of the contents of doc/ was moved to http://wiki.asterisk.org |
| 00:12.58 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 00:13.00 | citywok | that's annoying for a file you want to just cp doc/asterisk-mib.txt :( |
| 00:13.29 | pabelanger | I don't see why it could not live in contrib |
| 00:13.34 | asteriskmonkey | do jitterbuffers help avoid echo? |
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| 00:13.56 | pabelanger | asteriskmonkey: no, just packet loss |
| 00:14.13 | citywok | pabelanger: yea idk, but it would be nice if it were somewhere so one could just cp it |
| 00:14.25 | asteriskmonkey | pabelanger: thanks |
| 00:14.27 | citywok | on the other hand since i can't get snmp to work anyways it's a mootp oint. lol |
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| 00:14.47 | pabelanger | citywok: proposed a patch for contrib ;) |
| 00:15.24 | asteriskmonkey | this is kinda left feild, but i have a carrier that always presents an echo situation to 1 particular exchange is there anyway to compensate using 1.6 or 1.8 ie run a second echo can on there sip channel for that area code/ |
| 00:15.56 | citywok | don't send calls to that area code on that carrier. seems easier lol. |
| 00:17.13 | pabelanger | asteriskmonkey: what interface are you using? |
| 00:17.24 | pabelanger | SIP, I assume |
| 00:17.28 | asteriskmonkey | yes |
| 00:17.35 | asteriskmonkey | SIP |
| 00:18.10 | pabelanger | asteriskmonkey: So, chances are, the problem lines in there equipment. If that is the case, not much you can do from Asterisk point of view |
| 00:18.22 | pabelanger | Maybe open a support ticket and submit a recording? |
| 00:18.35 | pabelanger | s/there/their |
| 00:19.23 | asteriskmonkey | pabelanger: gah thought as much :P might as well just roll em up my tcpdump file while im at it.. thanks though :) |
| 00:20.02 | pabelanger | asteriskmonkey: If you were using DAHDI hardware, it would be a different story. |
| 00:21.06 | asteriskmonkey | pabelanger: yeah i know, id have echo can selection galore plus hardware echo can opts... no sense me banging my head anymore on this one though.. clearly a carrier issue.. (probably also runnging an asterisk box with the pri too hot) |
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| 00:40.02 | pecenipicek | would this be a right and proper way of doing what i want? http://pastebin.com/vEcHQhA6 |
| 00:41.35 | pecenipicek | or am i missing something crucial here? |
| 00:47.08 | asteriskmonkey | <PROTECTED> |
| 00:47.22 | pecenipicek | dundi is in use as well. |
| 00:47.46 | pecenipicek | incoming calls land on the incomingdundi context before hitting the context where that ends up. |
| 00:47.48 | asteriskmonkey | yah i dont see where your actually calling dundi though |
| 00:48.08 | asteriskmonkey | your setting a variable but not using it |
| 00:48.35 | asteriskmonkey | Dial(SIP/${EXTEN}) will always result in the _XXX match values |
| 00:49.03 | pecenipicek | http://pastebin.com/eUcmbii7 |
| 00:49.25 | pecenipicek | i'm just trying to get the data out of a lookup via the DUNDILOOKUP function. |
| 00:50.04 | pecenipicek | hence the NoOp with the var which doesnt get displayed when called.. |
| 00:51.00 | LemensTS | <PROTECTED> |
| 00:51.02 | asteriskmonkey | what does the cli show you |
| 00:51.11 | asteriskmonkey | LemensTS: cepstral |
| 00:51.13 | pecenipicek | if you werent here, i'm actually trying to pull out the tech value out of the dundi lookup. |
| 00:51.55 | pecenipicek | the extension itself works fine, but not all phones are gonna be handled with SIP, and i'm looking for a way to automatically pull the protocol via which the call must proceed to get to the desired phone. |
| 00:52.18 | asteriskmonkey | why not just set in astdb keys? |
| 00:52.27 | asteriskmonkey | or switch to realtime and do a mysql check |
| 00:52.31 | *** part/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 00:52.34 | asteriskmonkey | dundi is used for box-box comm |
| 00:52.45 | asteriskmonkey | with unknown numbers |
| 00:54.08 | pecenipicek | i did try with realtime but trunking between boxes blew up then. |
| 00:54.28 | pecenipicek | and it still didnt handle the fact that not all extensions will be sip calls |
| 00:57.06 | pecenipicek | the basic problem is discovery of which tech is used at the called extension. |
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| 01:45.00 | jmackey | Anyone able to give me a hand with some dialplan problems? |
| 01:45.11 | leifmadsen | ~ask |
| 01:45.12 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:47.52 | jmackey | Ok then. When I have just the Dial application in a dialplan, it works just fine, but if I add anything before that. I get a "The number you have dialed is not in service." I do adjust the priority numbers accordingly. |
| 01:48.29 | jmackey | verbose in the CLI claims that Dial is working and the line is ringing, but I still get the service message. |
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| 01:57.58 | nix8n82 | you should pastebin your log of your call...if it's trixbox or freepbx you will probably be told to go to another channel |
| 01:58.18 | jmackey | its pure asterisk. |
| 01:58.55 | jmackey | call log: http://pastebin.com/jdP00whW |
| 01:59.38 | jmackey | dialplan: http://pastebin.com/5tBYAmfP |
| 02:18.06 | jmackey | ok, going through the SIP debug logs, it appears that my provider is sending a cancel for unknown reasons |
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| 02:29.08 | pecenipicek | good god. |
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| 02:30.08 | pecenipicek | is there really no proper and easy way to be able to figure out the technology of the number that is getting called? |
| 02:31.18 | WIMPy | You have to know it when you write your dialplan. |
| 02:33.12 | pecenipicek | There is no possible lookup command to look at a registered extension and to figure out which technology it should use? |
| 02:34.11 | WIMPy | Extensions are configured. Devices register. |
| 02:37.49 | pecenipicek | And figuring out the technology used for the device channel is impossible why? |
| 02:38.05 | pecenipicek | Since extensions can be arbitrarily found and matched via patterns. |
| 02:38.10 | pecenipicek | _XXX and the like |
| 02:38.20 | pecenipicek | why cant the technologies be matched the same way? |
| 02:38.39 | WIMPy | For devices the channeltype is always known. |
| 02:39.15 | *** join/#asterisk coppice (~chatzilla@m121-203-198-14.smartone-vodafone.com) |
| 02:39.25 | WIMPy | A device does not make sense without the channel type. |
| 02:40.20 | pecenipicek | thats probably one of the most counterintuitive things about asterisk then. |
| 02:41.35 | WIMPy | Well, I'd certainly prefer some abstraction layer there, but where do you have an issue with that? |
| 02:42.25 | WIMPy | or: Why (where) do you want to know the technology? |
| 02:43.10 | pecenipicek | the following three lines are excerpts from my extensions.conf |
| 02:43.23 | pecenipicek | exten => _XXX,2,NoOp(internal context called on Server B to ${EXTEN}, on channel ${CHANNEL}) |
| 02:43.23 | pecenipicek | exten => _XXX,n,Dial(SIP/${EXTEN},10) |
| 02:43.23 | pecenipicek | exten => _XXX,n,Hangup |
| 02:44.02 | pecenipicek | not all phones i have here support SIP. |
| 02:44.25 | pecenipicek | basically my hurdle is how to get the damn thing to figure out on its own should it dial via SIP or some other protocol. |
| 02:44.44 | WIMPy | Well, in most cases you won't get away that easily anyway. |
| 02:45.18 | WIMPy | There can't be a waybecause that's just what you need to tell it. |
| 02:45.33 | *** join/#asterisk maxagaz (dce7256a@gateway/web/freenode/ip.220.231.37.106) |
| 02:45.39 | maxagaz | Hi |
| 02:45.40 | pecenipicek | which is very counterintuitive when you compare it to the behavior of the rest of the stuff. |
| 02:45.59 | WIMPy | The technology is not a property of the device. The name is a property of the technology, giving a device. |
| 02:46.17 | maxagaz | Can I check in the console that sRTP is compiled in asterisk ? |
| 02:46.20 | pecenipicek | if it can find the number via the pattern, it should be able to also find out which tech is in use. |
| 02:46.35 | WIMPy | But you can just use all technologies that might be used. |
| 02:47.09 | pecenipicek | there i hit another hurdle. when the SCCP phone gets called via SIP it doesnt die immediately, it lingers for 20 seconds before dieing. |
| 02:47.32 | WIMPy | Usually you won't name the devices the same as the extensions, so that won't work anyway. |
| 02:47.57 | pecenipicek | i can of course always use _XXX,n,Dial(SCCP/${EXTEN}&SIP/${EXTEN},10), but thats a bit too dirty for my liking. |
| 02:48.23 | WIMPy | That would do what you want. |
| 02:48.34 | pecenipicek | emphasis on the last part. |
| 02:48.42 | WIMPy | That's what you get for not wanting to configure extensions. |
| 02:48.48 | pecenipicek | basically, i'm looking for a bit more elegant solution |
| 02:49.31 | pecenipicek | that wont actually die when phones dont end up registering on just one box. |
| 02:49.32 | WIMPy | Use a database, either directly or to generate a configuration. |
| 02:50.25 | WIMPy | So what's that multiple boxes thing now? |
| 02:50.29 | pecenipicek | it still doesnt really cover the problem of the extension calling. |
| 02:50.46 | pecenipicek | all this is in preparation of a multiple server deployment. |
| 02:51.08 | WIMPy | It does. extensions.conf maps from extensions to devices or other actions. Use it. |
| 02:51.59 | pecenipicek | i got the boxes talking via dundi about extension registered up there, got IAX2 trunking between them, but getting it to actually call a phone other than a single tech without the _XXX,n,Dial(SCCP/${EXTEN}&SIP/${EXTEN},10) workaround is the hurdle. |
| 02:52.10 | WIMPy | If you want to catch everything with just a few patterns, organize your extensions so that one digit gives you the technology. |
| 02:53.32 | WIMPy | You might be able to query device states. |
| 02:54.04 | pecenipicek | so for example, to segment sip phones into a 2xx pattern and sccp phones into 3xx pattern? so the dialing patterns would look like the following: |
| 02:54.04 | pecenipicek | _3XX,n,Dial(SCCP/${EXTEN},10) |
| 02:54.04 | pecenipicek | _2xx,n,Dial(SIP/${EXTEN},10) |
| 02:54.27 | pecenipicek | right? |
| 02:54.27 | WIMPy | That would work. |
| 02:55.12 | pecenipicek | makes a mental note to explain to people for who we are doing this that there will be some changes in internal numbers... |
| 02:55.20 | WIMPy | At least as long as you don;t want to swap hardware without changing the extension. |
| 02:55.54 | pecenipicek | mmkay. |
| 02:56.47 | WIMPy | I'm pretty sure you don't only need extensions to map to exactely one device. |
| 02:57.06 | pecenipicek | the worst part is that what i have in mind would likely work if the dundilookup returned the actual device info behind the iax trunk, and not the iax stuff itself... |
| 02:57.07 | WIMPy | So you'd better generate your configs or use realtime. |
| 02:59.01 | pecenipicek | number segmentation solves that particular idiocy... |
| 02:59.10 | pecenipicek | oh well. |
| 03:00.23 | pecenipicek | the pbx's dont even need to care much about the particularities of a device until they hit the dial app. |
| 03:03.34 | maxagaz | I'm trying sRTP, it's set in extensions.conf and in sip.conf, but I get the following error message in asterisk 1.8 : ERROR[13411]: chan_sip.c:27972 setup_srtp: No SRTP module loaded, can't setup SRTP session. |
| 03:04.03 | maxagaz | Can someone help about this ? |
| 03:04.53 | *** join/#asterisk jmackey (184463fe@gateway/web/freenode/ip.24.68.99.254) |
| 03:07.11 | jmackey | New question for you guys. What would cause asterisk not to run in daemon mode, but to run perfectly fine in foreground mode? When in daemon mode, it shows up as a running process, but no logs are made and I am unable to connect to it at all. |
| 03:08.20 | WIMPy | maxagaz: You obviousely have no rtp support loaded, possibly not compiled in. |
| 03:09.15 | maxagaz | WIMPy: it was checked during the compilation, like this [*] |
| 03:09.46 | WIMPy | module show like srtp |
| 03:11.36 | maxagaz | WIMPy: I can still see res_srtp checked in: make menuselect > Resource Modules |
| 03:12.10 | WIMPy | Then maybe it's just not loaded. |
| 03:12.30 | maxagaz | WIMPy: I thought modules were loaded automatically ? |
| 03:13.03 | WIMPy | Only if you configured it that way. See modules.conf. |
| 03:13.05 | maxagaz | WIMPy: Oh, but there are loaded at computer start, not when asterisk starts, right ? |
| 03:13.20 | maxagaz | s/there/they |
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| 03:13.56 | WIMPy | That has nothing to do with kernel modules. They are loaded into Asterisk. |
| 03:14.54 | maxagaz | WIMPy: okay, then, how can I debut this ? |
| 03:15.15 | WIMPy | See modules.conf. |
| 03:17.31 | WIMPy | You can try to load it manually with 'module load res_srtp' to see if it works. |
| 03:19.59 | maxagaz | WIMPy: [Jan 18 11:19:28] WARNING[13510]: loader.c:387 load_dynamic_module: Error loading module 'srtp': /usr/lib/asterisk/modules/srtp.so: cannot open shared object file: No such file or directory |
| 03:20.36 | WIMPy | Looks like it wasn't installed then. |
| 03:24.15 | helloritesh | Folks, has someone figured out how to detect DTMF inside Dial Macro "M()"? I am going nuts with it.. all commands execute properly but the "Read" just refuses to capture DTMF from the caller party is there a issue /bug posted on this? http://pastebin.com/F3XmGgnE looks really harmless but just refuses to work.. i have spent over 20 hours on these 5 lines and 100s of test calls.. no luck |
| 03:25.06 | maxagaz | WIMPy: I have res_srtp.so in /usr/lib/asterisk/modules |
| 03:25.59 | maxagaz | But if I try to load it, I get this : |
| 03:26.04 | maxagaz | [Jan 18 11:25:39] WARNING[13510]: loader.c:387 load_dynamic_module: Error loading module 'res_srtp': /usr/lib/asterisk/modules/res_srtp.so: cannot restore segment prot after reloc: Permission denied |
| 03:29.19 | WIMPy | maxagaz: I have not seen that message before. Did you restart Asterisk after installing that module? |
| 03:30.24 | maxagaz | WIMPy: yes I did |
| 03:30.44 | maxagaz | WIMPy: it's a fresh install of asterisk 1.8 |
| 03:31.15 | maxagaz | WIMPy: it's actually a default install with install of libsrtp |
| 03:31.54 | maxagaz | WIMPy: libsrtp installed before to have res_srtp checked in menuselect |
| 03:38.30 | maxagaz | I can't see any module related to srtp when I do this: CLI> module show like rtp |
| 03:44.43 | WIMPy | Sure, if it doesn't load. |
| 03:44.53 | WIMPy | But I have no idea why it doen't load. |
| 03:51.56 | JerJer | maxagaz: are you running as a non-root user ? |
| 03:52.38 | maxagaz | JerJer: no... |
| 03:53.21 | JerJer | what about selinux ? |
| 03:53.23 | maxagaz | JerJer, WIMPy: I get this in /var/log/asterisk/messages => [Jan 18 11:48:59] WARNING[5585] loader.c: Error loading module 'res_srtp.so': /usr/lib/asterisk/modules/res_srtp.so: cannot restore segment prot after reloc: Permission denied |
| 03:54.00 | JerJer | change "enforcing" to "disabled" in ''/etc/selinux/config'' and reboot. |
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| 03:59.04 | JerJer | as for a specific fix, no idea |
| 04:04.16 | maxagaz | JerJer: I can now see the module in 'module show like rtp' indeed |
| 04:08.23 | JerJer | res_srtp is doing something selinux doesn't like, so an exception needs to be made, but I don't know which/how |
| 04:08.48 | JerJer | seems like a good topic for a blog post |
| 04:08.57 | JerJer | :) |
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| 06:48.17 | maxagaz | Now that the rtsp module is loaded, I get the following error message when I hang up the phone: WARNING[5811] chan_sip.c: Can't provide secure audio requested in SDP offer |
| 06:48.28 | *** join/#asterisk domzinick (7d63ff96@gateway/web/freenode/ip.125.99.255.150) |
| 06:48.54 | domzinick | Hi! I'm looking for a speech recognition engine for asterisk. Is there a free alternative to lumenvox? |
| 06:49.39 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 06:52.20 | shapr | cmu sphinx? |
| 06:52.22 | domzinick | I just found AsteriskUniMRCP, the connector bridge for asterisk and UniMRCP. Is it good enough to recognize simple words, like months, major cities in US, etc? |
| 06:53.52 | domzinick | shapr: Isn't cmu sphinx something for the developers? or is there a connector for cmu sphinx with asterisk? |
| 06:55.59 | domzinick | shapr: Well, the website says it's easy to integrate with asterisk. I'd like to know which is better and why. asteriskUniMRCP or cmu sphinx? |
| 07:03.26 | coppice | they do totally different things |
| 07:24.01 | domzinick | coppice: I'm looking for the user to interact with asterisk by saying simple words. Which setup is best suited for it? |
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| 07:35.51 | maxagaz | Does this warning means that the phone is not encrypted with sRTP or that it is encrypted, but without some password ? => WARNING[6022] res_srtp.c: SRTP unprotect: authentication failure |
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| 07:45.05 | joachim_- | Hi. Any good documents out there about Asterisk server clustering? Have multiple asterisk servers on different networks, and need a good plan for redundancy? (If one server goes down, the other takes over) I also got multiple sql database servers also on different networks.. |
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| 08:26.31 | schmidts | good morning |
| 08:36.17 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 08:36.56 | shamelessn00b | hi, is it possible running asterisk on the cloud, openstack/eucalyptus/EC2 or any other platform |
| 08:37.30 | beardy | ... |
| 08:38.23 | shamelessn00b | I know its a very naive question to ask, you need to right specialized software to run on the cloud |
| 08:38.26 | *** join/#asterisk ed1 (~ed1@carmel8.bb.netvision.net.il) |
| 08:38.35 | ed1 | hello |
| 08:38.38 | shamelessn00b | but I fail to convince my retarded CTO |
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| 08:39.28 | shamelessn00b | he insists that we can run off-the-shelf pre-compiled open source software without any mods on a cloud |
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| 08:40.38 | ed1 | i need help with client loosing connection to * after some time when behind nat |
| 08:42.18 | beardy | He seems to have convinced you that "the cloud" is a wording to be used. Anyway, yes, of course it will run as long as the machine runs the architecture and OS your pre-compiled package is built for. |
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| 08:44.15 | shamelessn00b | beardy: does that mean that I can just set-up openstack platform and install asterisk on it and it would work |
| 08:49.54 | beardy | shamelessn00b: No idea. Depends what it actually runs. Is it a virtual machine frontend? |
| 08:52.42 | beardy | shamelessn00b: Looks like it. |
| 08:53.45 | shamelessn00b | The virtual machines would never span two different physical machines |
| 08:53.57 | shamelessn00b | thats what I learned just now |
| 09:02.12 | asterisk-learner | hello |
| 09:02.27 | asterisk-learner | is it normal to see "Blocked in : ast_waitfor_nandfds( )" when doing core show channels ? |
| 09:02.47 | asterisk-learner | or does this mean a problem in that channel ? |
| 09:02.52 | asterisk-learner | (stucked or so... ? ) |
| 09:03.45 | coppice | domzinick: CMU Sphinx is a speech recognition system. UniMRCP is just an interface to a speech recognition system. It is typically used with one of the commercial speech recognition engines |
| 09:03.47 | joachim_- | Hi. Any good documents out there about Asterisk server clustering? Have multiple asterisk servers on different networks, and need a good plan for redundancy? (If one server goes down, the other takes over) I also got multiple sql database servers also on different networks.. |
| 09:07.29 | schmidts | joachim_ hello the best way would be to use a proxy in front |
| 09:24.10 | *** join/#asterisk moy (~moy@70.25.53.39) |
| 09:33.11 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 09:41.10 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 09:45.56 | joachim_- | schmidts: What kind of proxy u mean? |
| 09:47.51 | shamelessn00b | I just finished writing my own connector for sphinx 4 and asterisk |
| 09:47.59 | shamelessn00b | anyone wanna try |
| 09:48.01 | shamelessn00b | ? |
| 09:49.06 | schmidts | joachim_ a sip proxy like kamailio |
| 09:51.04 | shamelessn00b | Try a load balancer, like LVS |
| 10:08.03 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 10:16.48 | merlin8282 | TimeRider: Hi. I modified your script, but just saw that it doesn't install mISDN. |
| 10:20.54 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 10:24.04 | maxagaz | what should be the "exten =>" in extensions.conf to redirect a call with number 8003 to sip user foo (already registered) ? |
| 10:24.23 | maxagaz | on asterisk 1.8.2 |
| 10:25.09 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:29.03 | TimeRider | merlin8282 : you added that to it? |
| 10:30.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:33.10 | Tozz_ | maxagaz: exten => 8003,1,Dial(SIP/foo) |
| 10:33.15 | kaldemar | maxagaz: exten => 8003,1,Dial(SIP/foo) <-- redirection is actually a completely different thing technically, but is this what you meant? |
| 10:33.52 | merlin8282 | TimeRider: not now, I'm first trying to install it manually. |
| 10:34.00 | TimeRider | merlin8282 : chanmsdn and lcr_driver apparently are what you need to be looking for |
| 10:34.17 | TimeRider | chan_misd doesn't work in 2.6.27 |
| 10:34.25 | TimeRider | are the modules in the modules directory for it? |
| 10:35.22 | maxagaz | Tozz_, kaldemar : thanks! |
| 10:35.49 | TimeRider | merlin8282 : try: module load chan_lcr |
| 10:36.01 | *** join/#asterisk marco_support (~marco_sup@95.232.237.32) |
| 10:36.06 | marco_support | hello to everyone |
| 10:36.21 | marco_support | i need help with asterisk 1.8 and res_calendar_ews |
| 10:37.40 | merlin8282 | TimeRider: failed, and nothing found with `locate chan_lcr` |
| 10:37.41 | *** join/#asterisk m_tadeu (~quassel@89.181.47.213) |
| 10:38.02 | marco_support | res_calendar_ews.c:530 send_ews_request_and_parse: Unable to communicate with Exchange Web Service at 'https://remote.pmsweb.it/ews/exchange.asmx': Could not read status line: connection timed out |
| 10:38.13 | TimeRider | chan_misd is in the source - maybe needs to be setup when compiling asterisk - menuconfig? |
| 10:38.13 | marco_support | this is the asterisk error |
| 10:39.14 | merlin8282 | TimeRider: can't select it : XXX |
| 10:39.28 | TimeRider | dependency problem? |
| 10:39.51 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:40.58 | merlin8282 | wait, I didn't have kernel sources in /usr/src -_- |
| 10:42.51 | TimeRider | I think maybe dependency issue.. you can try compiling it and see what's missing, be good if you can fix so we can script it in :) |
| 10:43.01 | TimeRider | I bbs |
| 10:44.57 | merlin8282 | What does (E) or (M) mean ? "Embedded" or "Module" ? |
| 10:48.38 | marco_support | res_calendar_ews.c:530 send_ews_request_and_parse: Unable to communicate with Exchange Web Service at 'https://remote.pmsweb.it/ews/exchange.asmx': Could not read status line: connection timed out |
| 10:49.09 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 10:50.09 | marco_support | anybody can Help me? |
| 10:55.20 | *** join/#asterisk sourcode (~code@ppp-58-8-67-24.revip2.asianet.co.th) |
| 10:55.57 | *** join/#asterisk mpe (~mpe@office.ipvision.dk) |
| 10:56.23 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 10:57.25 | mpe | in asterisk 1.6 / 1.8 is it posible to :n |
| 10:58.25 | mpe | set shared_lastcall for a queue |
| 10:59.40 | mpe | in the sektione for a specifik queue and not only in the general sektion |
| 11:00.49 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:01.00 | verywiseman | i want to make one person when he enter his extension and password , context of his extension is running rather than the context of the phone he use |
| 11:15.44 | *** part/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 11:20.19 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:35.44 | marco_support | http://downloads.sourceforge.net/project/freeassociation/libical/libical-0.44/libical-0.44.tar.gz?r=&ts=1295203713&use_mirror=freefr |
| 11:35.58 | marco_support | res_calendar_ews.c:530 send_ews_request_and_parse: Unable to communicate with Exchange Web Service at 'https://remote.pmsweb.it/ews/exchange.asmx': Could not read status line: connection timed out |
| 11:36.10 | marco_support | excuse me for repetition |
| 11:36.17 | *** join/#asterisk shtoom (shtoom@14.96.164.158) |
| 11:42.20 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 11:46.00 | gregd | guys, could someone suggest me something similar to spa 3102 but with better echo management? the price can be higher as well |
| 11:46.55 | *** join/#asterisk coppice (~chatzilla@m121-203-200-166.smartone-vodafone.com) |
| 11:47.01 | marco_support | OS 40 siemens |
| 11:47.06 | marco_support | or os 60 siemens |
| 11:47.59 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 11:50.25 | gregd | it does not seem like siemens os40 is an ATA |
| 11:56.10 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:01.32 | AdvoWork | if im trying to dial a number, i have a dialtone, but it just hangs and doesnt call, what logs would that be in? struggling to find anything.. |
| 12:02.49 | kaldemar | AdvoWork: attach to asterisk's CLI with "asterisk -vvvr" and see what happens when you make a call. |
| 12:03.28 | *** join/#asterisk coppice (~chatzilla@m121-203-200-166.smartone-vodafone.com) |
| 12:21.58 | *** join/#asterisk schmidts (~schmidts@213.235.212.193) |
| 12:28.36 | marco_support | res_calendar_ews.c:530 send_ews_request_and_parse: Unable to communicate with Exchange Web Service at 'https://remote.pmsweb.it/ews/exchange.asmx': Could not read status line: connection timed out |
| 12:29.49 | wdoekes2 | marco_support: what do you want to know? |
| 12:30.14 | marco_support | i have this error to connect to exchange calendar 2007 |
| 12:30.43 | wdoekes2 | it seems like it can't connect.. neither can I.. it seems like the host does not exist or is internal to your network only |
| 12:30.59 | wdoekes2 | $ host remote.pmsweb.it |
| 12:30.59 | wdoekes2 | Host remote.pmsweb.it not found: 3(NXDOMAIN) |
| 12:31.20 | marco_support | i know is linked with hosts file |
| 12:31.37 | wdoekes2 | in that case you should track why the connection times out |
| 12:31.43 | wdoekes2 | I doubt it is a problem in asterisk |
| 12:31.57 | marco_support | but if i use web |
| 12:32.05 | wdoekes2 | web.. from the same machine? |
| 12:32.06 | marco_support | i reached the page black |
| 12:32.09 | wdoekes2 | or from your desktop? |
| 12:32.14 | marco_support | from my desktop |
| 12:32.19 | wdoekes2 | ... |
| 12:32.23 | marco_support | and if i ping from asterisk I reach the ip |
| 12:32.41 | wdoekes2 | and if you 'wget' it? |
| 12:33.38 | marco_support | wait |
| 12:33.41 | marco_support | exit |
| 12:34.17 | marco_support | with wget i received |
| 12:34.29 | marco_support | that CA autority is not correct |
| 12:35.12 | marco_support | wget https://remote.pmsweb.it/ews/Exchange.asmx --no-check-certificate |
| 12:35.12 | marco_support | --2011-01-18 13:35:00-- https://remote.pmsweb.it/ews/Exchange.asmx |
| 12:35.12 | marco_support | Risoluzione di remote.pmsweb.it... 67.215.77.132 |
| 12:35.12 | marco_support | Connessione a remote.pmsweb.it|67.215.77.132|:443... connesso. |
| 12:35.12 | marco_support | AVVERTIMENTO: il nome comune di certificato "*.opendns.com" non corrisponde al nome dell'host richiesto "remote.pmsweb.it". |
| 12:35.14 | marco_support | HTTP richiesta inviata, in attesa di risposta... 302 Found |
| 12:35.16 | marco_support | Posizione: http://guide.opendns.com/?url=remote%2Epmsweb%2Eit%2Fews%2FExchange%2Easmx [segue] |
| 12:35.18 | marco_support | --2011-01-18 13:35:01-- http://guide.opendns.com/?url=remote%2Epmsweb%2Eit%2Fews%2FExchange%2Easmx |
| 12:35.20 | marco_support | Risoluzione di guide.opendns.com... 208.69.34.136 |
| 12:35.22 | marco_support | Connessione a guide.opendns.com|208.69.34.136|:80... connesso. |
| 12:35.26 | marco_support | HTTP richiesta inviata, in attesa di risposta... 200 OK |
| 12:35.28 | marco_support | Lunghezza: non specificato [text/html] |
| 12:35.30 | marco_support | Salvataggio in: "Exchange.asmx" |
| 12:35.32 | marco_support | <PROTECTED> |
| 12:35.35 | marco_support | 2011-01-18 13:35:02 (175 MB/s) - "Exchange.asmx" salvato [1355] |
| 12:35.37 | marco_support | ls |
| 12:35.38 | kaldemar | ~pb |
| 12:35.38 | 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. |
| 12:36.19 | marco_support | wdoekes2 any idea? |
| 12:36.20 | wdoekes2 | hmm.. a 302 as well.. that might trigger the "time out".. supply http://guide.opendns.com/?url=remote%2Epmsweb%2Eit%2Fews%2FExchange%2Easmx to asterisk instead |
| 12:36.23 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 12:36.34 | eject_ck | what this message means ? [2011-01-18 12:34:12] NOTICE[4638]: rtp.c:1808 ast_rtp_read: Unknown RTP codec 120 received from |
| 12:36.46 | Chainsaw | eject_ck: Generally, it means you're using X-Lite. |
| 12:37.02 | Chainsaw | eject_ck: It sends those purposefully invalid messages as a keepalive of sorts. |
| 12:39.05 | marco_support | ok |
| 12:39.34 | marco_support | Risoluzione di remote.pmsweb.it... 195.43.186.138 |
| 12:39.34 | marco_support | Connessione a remote.pmsweb.it|195.43.186.138|:443... connesso. |
| 12:39.34 | marco_support | AVVERTIMENTO: impossibile verificare il certificato di remote.pmsweb.it, rilasciato da "/CN=pms-SERVER-CA": |
| 12:39.34 | marco_support | <PROTECTED> |
| 12:39.34 | marco_support | HTTP richiesta inviata, in attesa di risposta... 401 Unauthorized |
| 12:39.36 | marco_support | Autorizzazione fallita. |
| 12:39.42 | marco_support | now is correct |
| 12:39.53 | marco_support | but the error in asterisk is the sam |
| 12:39.57 | marco_support | e |
| 12:40.07 | eject_ck | Chainsaw: I'm using nokia :) |
| 12:40.12 | wdoekes2 | 401? that is not a valid page, now is it? |
| 12:40.20 | Chainsaw | wdoekes2: 401 is authorisation required. |
| 12:40.26 | wdoekes2 | or does that calender-thingy support basic-auth |
| 12:40.48 | marco_support | what kind of authorization over exchange? |
| 12:41.28 | wdoekes2 | I have no idea.. I'm just helping you debug your timeout |
| 12:42.57 | wdoekes2 | but.. that's a normal http-connection.. you can do a tcpdump while connecting with asterisk |
| 12:43.00 | wdoekes2 | (or ngrep) |
| 12:43.03 | wdoekes2 | that should tell you more |
| 12:43.06 | marco_support | https connection |
| 12:43.42 | wdoekes2 | that guide.opensdns.com redirect wasn't |
| 12:44.11 | wdoekes2 | but you can indeed still check what kind of packet flow it does for the https connection |
| 12:45.01 | wdoekes2 | oh wait.. did you actually read what was happening with your wget |
| 12:45.10 | wdoekes2 | it's not supposed to redirect to http://guide.opendns.com/?url=remote%2Epmsweb%2Eit%2Fews%2FExchange%2Easmx |
| 12:45.16 | wdoekes2 | (I think) |
| 12:46.10 | marco_support | the only think is to know if res_calendar_ews support self CA ssl certificate |
| 12:46.40 | marco_support | or need CA certificate signed |
| 12:46.45 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 12:46.48 | wdoekes2 | I doubt that it needs a proper cert |
| 12:47.33 | marco_support | with ical google i have no problem |
| 12:47.34 | wdoekes2 | or if it does, it prints that as a warning |
| 12:48.10 | wdoekes2 | do you run chrooted? |
| 12:48.33 | marco_support | chrooted? why? |
| 12:48.54 | wdoekes2 | no.. never mind if you don't |
| 12:49.57 | leifmadsen | Katty: I got your card last night! :) |
| 12:50.05 | leifmadsen | wdoekes2: morning |
| 12:51.23 | wdoekes2 | afternoon leif |
| 12:52.25 | wdoekes2 | marco_support: do you have SOAP enabled in your exchange server? |
| 12:53.18 | marco_support | i think no |
| 12:53.54 | marco_support | you intend WEBDAV? |
| 12:53.58 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 12:54.00 | wdoekes2 | no |
| 12:54.31 | wdoekes2 | the error occurs after doing a SOAP POST request |
| 12:55.08 | marco_support | WARNING[30912]: res_calendar_exchange.c:408 exchangecal_request: Unknown response to CalDAV calendar ECHANGEcal2003, request SEARCH to https://192.168.0.134/exchange/: Could not read status line: connection timed out |
| 12:55.17 | marco_support | this error occurs also on exchange 2003 server |
| 12:56.21 | wdoekes2 | you should tcpdump and see what goes on with those packets, I think |
| 12:56.39 | marco_support | give me the command to do this |
| 12:56.49 | wdoekes2 | tcpdump -nn port 443 |
| 12:57.09 | wdoekes2 | possibly add -iany if you got multiple interfaces |
| 12:57.33 | marco_support | only one |
| 13:00.23 | marco_support | 13:59:37.000854 IP 192.168.0.134.443 > 192.168.0.222.57702: S 354776810:354776810(0) ack 1070002256 win 64240 <mss 1460,nop,wscale 0,nop,nop,timestamp 0 0,nop,nop,sackOK> |
| 13:00.29 | marco_support | ops |
| 13:00.36 | wdoekes2 | *gone* |
| 13:00.45 | marco_support | 13:59:37.000854 IP 192.168.0.134.443 > 192.168.0.222.57702: S 354776810:354776810(0) ack 1070002256 win 64240 <mss 1460,nop,wscale 0,nop,nop,timestamp 0 0,nop,nop,sackOK> |
| 13:01.02 | marco_support | 13:59:37.232091 IP 195.43.186.138.443 > 192.168.0.222.52594: . ack 851 win 260 <nop,nop,timestamp 24623164 2093905> |
| 13:02.36 | marco_support | 14:01:37.005603 IP 192.168.0.134.443 > 192.168.0.222.57702: R 1:1(0) ack 1126 win 0 |
| 13:02.51 | marco_support | 14:01:37.105164 IP 192.168.0.222.52594 > 195.43.186.138.443: F 851:851(0) ack 1 win 183 <nop,nop,timestamp 2123906 24623164> |
| 13:03.09 | *** part/#asterisk BlackBishop (dexter@d3xt3r01.tk) |
| 13:07.19 | *** join/#asterisk apalos (~shiny@host2.gennetsa.ondsl.gr) |
| 13:07.42 | apalos | hello, is there a way to make asterisk's dialplan respect min and max values? |
| 13:08.08 | apalos | for example, a telephone destination might have 7 digits as minimun and dial it immediately if the number is > 14 digits |
| 13:08.24 | apalos | but a dest can have anything between 7-14 digits |
| 13:12.04 | *** join/#asterisk PoTe (~PoTe@rev-200-40-119-222.netgate.com.uy) |
| 13:12.11 | PoTe | Hello everyone :) |
| 13:12.43 | PoTe | Question: |
| 13:13.11 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:13.34 | PoTe | <PROTECTED> |
| 13:13.55 | PoTe | Even with the call busy parameter set as no in the queues, and busy limit set to 1 in sip.conf. |
| 13:14.13 | PoTe | They are using xlite soft phones, tho I think that shoulndt make a difference |
| 13:14.24 | PoTe | anyone has any idea of what might be going on? |
| 13:14.27 | PoTe | Thanks in advance! |
| 13:15.25 | *** part/#asterisk marco_support (~marco_sup@95.232.237.32) |
| 13:15.57 | *** join/#asterisk tftech (~tftech@ip65-47-56-98.z56-47-65.customer.algx.net) |
| 13:22.36 | creativx | PoTe: x-lite has two or more lines.. asterisk obviously does not respect busy limit.. |
| 13:23.40 | PoTe | Ah, so it might be an issue with x-lite configuration then? I'll try to replicate it on my end. |
| 13:23.51 | creativx | no not really |
| 13:24.18 | creativx | as long as x-lite is in a call on line 1, it will tell asterisk its free to take another call on line 2 |
| 13:24.25 | creativx | we had this problem too with 1.2, dont recall what we did |
| 13:24.35 | creativx | but people generally dont like being bugged with calls from a queue on line 2 |
| 13:33.03 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 13:38.47 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:42.25 | *** join/#asterisk Lukkie (lucas@darnassus.vanschouwen.nl) |
| 13:44.19 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:44.37 | Lukkie | eya, is there a reason for the missing rpm packages for 1.8.2 ? |
| 13:44.46 | Tozz_ | HAI HOI |
| 13:44.54 | Lukkie | its TeleTozz! |
| 13:45.04 | Lukkie | you are everywhere :-) |
| 13:45.10 | Tozz_ | yes! |
| 13:45.23 | *** join/#asterisk RamaskeZ (~RamaskeZ@ip-87-86-154-70.easynet.co.uk) |
| 13:46.22 | RamaskeZ | Hi, can anyone tell me if there is a max size limit for astdb? |
| 13:47.11 | Lukkie | Tozz_: but you do not know the answer I presume? :) |
| 13:47.50 | Tozz_ | nope |
| 13:47.55 | Tozz_ | compile from source! ;) |
| 13:48.08 | Lukkie | could do that |
| 13:48.21 | Lukkie | it does fixes my problem ... |
| 13:48.40 | wdoekes2 | RamaskeZ: is there an underlying problem that causes you to ask said question? |
| 13:49.42 | Tozz_ | Lukkie: Je wil wel perse ISDN van VoIP is eng, maar je draait wel beta versie van je PBX? |
| 13:49.47 | Tozz_ | s/van/want/ |
| 13:50.06 | Tozz_ | uhmmm? |
| 13:50.07 | RamaskeZ | wdoekes2: no, it was more thinking of a future project which I may use astdb to cache some information to save having to do a SQL lookup, however I think this will get pretty big pretty quick. |
| 13:50.30 | Lukkie | Tozz_: first; it was not my choice to keep ISDN; secondly there is nothing really wrong with 1.8 (it is not beta is it?) |
| 13:50.30 | wdoekes2 | RamaskeZ: use SQLite |
| 13:51.29 | Tozz_ | Lukkie: google says its not recommended to use 1.8 on production systems |
| 13:51.46 | Lukkie | Tozz_: don't believe everything Google is telling you :) |
| 13:51.51 | Tozz_ | 42! |
| 13:52.00 | RamaskeZ | wdoekes2: great idea. Never even thought of that. Cheers :) |
| 13:52.52 | tuxx- | omg, lukkie hier ook al! |
| 13:53.00 | Lukkie | tuxx-: nee |
| 13:53.08 | tuxx- | oh. dan heb ik het fout. |
| 13:53.15 | RamaskeZ | wdoekes2: out of curiosity do you know if there is a limit anyhow as im now curious. :) |
| 13:54.17 | Lukkie | Tozz_: but sure, I could run 1.4 and build mISDN |
| 13:54.43 | Lukkie | but there is nothing wrong with the ISDN portion of my system :) |
| 13:54.53 | Tozz_ | besides that its ISDN |
| 13:55.02 | tuxx- | audiocodes all the way \o/ |
| 13:55.14 | Lukkie | ah shut it, you are just sad that we did not order a SIP trunk with you and did all our dialing via you :) |
| 13:55.20 | Tozz_ | jup! |
| 13:55.31 | tuxx- | tozz een voipprovider? |
| 13:55.39 | Lukkie | Tozz_: and so am I |
| 13:55.41 | Tozz_ | yes |
| 13:55.43 | Lukkie | Tozz_: but it was not up to me |
| 13:55.49 | Tozz_ | Lukkie: maybe in a couple months ;) |
| 13:56.21 | Lukkie | Tozz_: if they give me the green light you know you'll be the first I contact |
| 13:56.49 | Tozz_ | oke ;) |
| 13:56.57 | tuxx- | whats the name Tozz_ ? |
| 13:57.04 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 13:57.05 | wdoekes2 | RamaskeZ: astdb is a (modified) http://en.wikipedia.org/wiki/Dbm |
| 13:57.07 | tuxx- | our company is always looking for more sipproviders ;p |
| 13:57.54 | wdoekes2 | what does your company do? |
| 13:58.52 | Lukkie | Tozz_: if you can fake KPN invoices ... I might be able to shuff you some business ;) |
| 14:01.37 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 14:02.46 | RamaskeZ | wdoekes2: Thanks for your help :) |
| 14:03.04 | Tozz_ | Lukkie: difficult ;) |
| 14:03.07 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 14:03.17 | *** join/#asterisk tftech (~tftech@ip65-47-56-98.z56-47-65.customer.algx.net) |
| 14:03.21 | tuxx- | wdoekes2: development of a pbx, and implmentation of it. |
| 14:03.35 | *** join/#asterisk w0ls0n (~w0ls0n@94-120-181-66.dsl.sacoriver.net) |
| 14:03.57 | w0ls0n | can someone recommend a good quality mutiline phone that will work with * |
| 14:04.10 | tuxx- | w0ls0n: aastra, snom, polycom |
| 14:04.18 | Tozz_ | cisco |
| 14:04.19 | tftech | w0ls0n - personally I like the polycom |
| 14:04.22 | tuxx- | true :) |
| 14:04.33 | tuxx- | <- aastra fan ;x |
| 14:04.40 | Tozz_ | <- cisco fan :) |
| 14:04.42 | tuxx- | doesnt need to reboot after new configuration has been pushed |
| 14:04.48 | tuxx- | still a big fail of the polycom |
| 14:04.49 | tuxx- | hehe |
| 14:04.49 | tuxx- | :P |
| 14:04.55 | tftech | I dont like the gui on the Polycom as much, but the voice quality is really good |
| 14:05.02 | tuxx- | true |
| 14:05.02 | tftech | specially if people like to be on speaker |
| 14:05.04 | Tozz_ | mm thats nice. Cisco does reboot after each config change |
| 14:05.15 | tftech | Polycom reboots too |
| 14:05.35 | Tozz_ | bye! |
| 14:05.41 | tuxx- | aastra doesn't, and aastra supports config via XML push and pull =] |
| 14:05.41 | tuxx- | pretty darn sweet |
| 14:05.46 | tuxx- | netsplit horrorssss! |
| 14:05.55 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 14:06.09 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:06.15 | tftech | I like the aastra cordless |
| 14:06.33 | tuxx- | which one, the 2.4ghz? |
| 14:06.36 | tuxx- | those are illegal in the eu >_< |
| 14:07.35 | tuxx- | well, back2coding. gotta make the boss happy =P |
| 14:08.19 | tftech | tuxx - yes, where the base is IP and the cordless handset is 2.4 |
| 14:08.52 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 14:08.54 | *** join/#asterisk fnordus (~dnall@S01060023693bfad4.va.shawcable.net) |
| 14:09.05 | tuxx- | right, we cant have those in the eu :-( |
| 14:09.14 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:21b:63ff:feb5:e13b) |
| 14:09.52 | tftech | tuxx - I thought they had a 5gig one too |
| 14:10.05 | *** join/#asterisk ronr (~ron@62.177.179.146) |
| 14:10.31 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 14:10.47 | tuxx- | hm, gonna look into that. the idea behind that phone is really nice. |
| 14:10.49 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 14:10.58 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-vzwtkhhmhmccjltn) |
| 14:11.17 | *** join/#asterisk timholum1 (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
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| 14:12.28 | w0ls0n | soooooooooo |
| 14:12.33 | leifmadsen | loooooooooong |
| 14:12.40 | w0ls0n | I am looking on newegg for phones |
| 14:12.44 | w0ls0n | :-) |
| 14:12.54 | *** join/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 14:12.56 | leifmadsen | for polycom phones on newegg? |
| 14:13.02 | w0ls0n | I just don't know what to choose |
| 14:13.03 | w0ls0n | yea |
| 14:13.08 | w0ls0n | I just want one for trial |
| 14:13.16 | w0ls0n | even used is ok |
| 14:13.18 | tftech | voipsupply |
| 14:14.36 | w0ls0n | Cisco CP-7910G (VSRF) |
| 14:14.50 | w0ls0n | I guess one line is fine for the warehouse |
| 14:15.01 | w0ls0n | but I can use that just for testing. Thoughts? |
| 14:15.09 | leifmadsen | Use Polycom |
| 14:15.18 | leifmadsen | the Cisco phones can be a pain to configure and you can't do it centrally |
| 14:15.28 | leifmadsen | well... actually you can, but the firmware is a pain to get regardless |
| 14:15.38 | *** join/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 14:15.39 | leifmadsen | it's been years since I've used my 7960 |
| 14:15.42 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 14:15.49 | rushowr | sorry all, I got reset apparently |
| 14:16.10 | rushowr | did the msg with the short version of my issue come through? |
| 14:16.37 | w0ls0n | Polycom IP301 AC (VSRF) |
| 14:16.40 | w0ls0n | $70 |
| 14:16.42 | w0ls0n | 2 lines |
| 14:17.12 | rushowr | Asterisk is reporting lines in my dialplan as having unknown directives on certain lines, even though the code has previously worked on the same version with diff machines |
| 14:17.16 | Tozz_ | leifmadsen: There are 2 Cisco IP Phone series |
| 14:17.21 | rushowr | log snippet available if needed |
| 14:17.35 | Tozz_ | leifmadsen: the 'Small business' and the callmanager series (79XX) |
| 14:17.48 | Tozz_ | the Cisco SBS can be used in * without problems and can be centrally configured |
| 14:18.15 | leifmadsen | rushowr: show us the lines |
| 14:18.19 | rushowr | [Jan 18 09:07:21] WARNING[18116] pbx_config.c: ==!!== Unknown directive: exten _.,n,Set(ARRAY(cnamid,age,cname) at line 79 -- IGNORING!!! |
| 14:18.27 | rushowr | appropo line coming |
| 14:18.33 | leifmadsen | rushowr: no way that line ever worked |
| 14:18.41 | rushowr | exten _.,n,Set(ARRAY(cnamid,age,cname)=${ODBC_CNAMCHECK(${BTN})}) |
| 14:18.54 | leifmadsen | looks like your ODBC is not returning anything |
| 14:18.59 | leifmadsen | that would be my guess anyways |
| 14:19.02 | leifmadsen | oh |
| 14:19.04 | leifmadsen | exten => |
| 14:19.08 | leifmadsen | not exten _. |
| 14:19.10 | rushowr | ah...thanks! |
| 14:19.11 | *** join/#asterisk Defraz (~Defraz@63-226-95-152.dia.static.qwest.net) |
| 14:19.11 | Tozz_ | :> |
| 14:19.12 | rushowr | JESUS! |
| 14:19.16 | leifmadsen | *facepalm* |
| 14:19.21 | rushowr | sorry, I use AEL almost exclusively |
| 14:19.24 | leifmadsen | yuck |
| 14:19.28 | *** join/#asterisk dwayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:19.33 | leifmadsen | I tried it once and immediately went back to extensions.conf |
| 14:19.43 | rushowr | you should try it now that Murf fixed it up |
| 14:19.51 | rushowr | I contributed a lot of requests ;-) |
| 14:19.54 | leifmadsen | the last time Murf worked on it was like 2 years ago |
| 14:19.59 | rushowr | yep |
| 14:20.02 | leifmadsen | AEL isn't updated very much |
| 14:20.02 | *** join/#asterisk coppice (~chatzilla@210.17.255.115) |
| 14:20.06 | w0ls0n | how about a Polycom Soundpoint IP 301 2200-11301-001 Phone |
| 14:20.08 | rushowr | true true |
| 14:20.10 | leifmadsen | I'll stick with my exten => and same => |
| 14:20.10 | rushowr | thanks leif |
| 14:20.14 | *** join/#asterisk arielb27 (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 14:20.15 | rushowr | :D |
| 14:20.19 | rushowr | I can't believe I missed that |
| 14:20.20 | rushowr | cheers! |
| 14:20.22 | w0ls0n | I found that on ebay for $30 plus shipping |
| 14:20.23 | leifmadsen | peas |
| 14:22.53 | *** join/#asterisk dacm_work (~dan@host86-145-64-67.range86-145.btcentralplus.com) |
| 14:23.02 | dacm_work | Hi guys. |
| 14:24.02 | dacm_work | I recently updated asterisk and my phones are now getting call waiting beeps rather than simply ringing on the next available phone. |
| 14:24.10 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 14:24.11 | dacm_work | Is there some setting that would cause this? |
| 14:25.03 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:27.14 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 14:30.48 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 14:33.56 | dacm_work | callwaiting=no fixes it it seems |
| 14:36.19 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 14:37.30 | [Outcast] | what is the best way to make this go away when AGI script written PHP? utils.c:1131 ast_carefulwrite: write() returned error: Broken pipe |
| 14:40.04 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 14:40.29 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:43.20 | wdoekes2 | [Outcast]: fread stdin until eof |
| 14:43.51 | [Outcast] | ok |
| 14:44.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:44.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 15:00.52 | [Outcast] | wdoekes2: all better thanks for the tip |
| 15:01.15 | wdoekes2 | good :) |
| 15:01.27 | *** join/#asterisk BesarKon1ol (~BesarKont@host-88-80-14-228.cust.prq.se) |
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| 15:01.54 | *** join/#asterisk phr0zen (phr0zen@blk-224-135-210.eastlink.ca) |
| 15:02.06 | phr0zen | hello, i am wondering if anyone can help me with echo cancellation? |
| 15:02.57 | phr0zen | i've set the rx/tx gain and have echo cancellation set to yes |
| 15:03.15 | phr0zen | however I cannot seem to get the random echo to stop, it happens on some calls, not others |
| 15:03.15 | tzafrir | phr0zen, what device? What echo canceller do you use? |
| 15:03.24 | tzafrir | (software? hardware?) |
| 15:03.29 | phr0zen | i have a digium te410p card |
| 15:03.44 | *** join/#asterisk Zairus (~Zairus@226.109.165.83.dynamic.mundo-r.com) |
| 15:03.47 | phr0zen | software cancellation |
| 15:04.18 | phr0zen | some calls appear perfect, but others, watching with ztmonitor spike to like 17000 and such |
| 15:04.22 | phr0zen | on an echoy call |
| 15:04.37 | phr0zen | the rx/tx gains don't seem to affect it |
| 15:04.59 | phr0zen | PRI lines |
| 15:05.23 | merlin8282 | mmm, why is mISDN disabled when I do make menuconfig, when compiling asterisk ? When doing ./configure, one line says "checking linux/mISDNdsp.h presence... no". How can I correct that ? |
| 15:06.21 | merlin8282 | I'm on debian lenny, experimental is in sources.list, and I installed following packages: libmisdn-dev, linux-headers-misdn, misdn-source and misdn-utils |
| 15:06.36 | BesarKon1ol | Hi. When using dundi with a switch statement and there is multiple peer replies to a lookup, asterisk tries every single peer in the lookup. Is it possible to make it only dial one? |
| 15:06.59 | leifmadsen | yes, use the DUNDILOOKUP() function |
| 15:07.13 | leifmadsen | errr sorry, the DUNDIQUERY() and DUNDIRESULT() functions |
| 15:07.26 | leifmadsen | don't use the switch statement because it's not going to work like you're expecting it |
| 15:07.53 | leifmadsen | BesarKon1ol: http://ofps.oreilly.com/titles/9780596517342/ch23.html |
| 15:11.24 | BesarKon1ol | leifmadsen: thanks alot! looks like just what i need |
| 15:12.50 | leifmadsen | ya using switches is generally not an ideal method |
| 15:12.59 | leifmadsen | (for anything, not just dundi) |
| 15:13.13 | leifmadsen | switch => has fallen out of favour over the years as better methods have been developerd |
| 15:14.45 | merlin8282 | Can I use mISDN version 2 with asterisk 1.8 and a junghanns card (HFC-4S) ? |
| 15:14.57 | merlin8282 | Or do I have to use mISDN v1 ? |
| 15:18.44 | *** join/#asterisk freckle (~viperdude@2001:5c0:1000:b::731b) |
| 15:20.16 | freckle | does anyone know if there is a limit on the number of SIP accounts Asterisk can register against? |
| 15:22.21 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:22.26 | c0rnoTa | hello everyone |
| 15:22.42 | c0rnoTa | i want to use unistim channel on 1.8 with Russian charset |
| 15:23.04 | c0rnoTa | I'v downloaded patch from bugtracker and "ru.po" file |
| 15:23.26 | c0rnoTa | how to use "ru.po" file? Where should i place it? |
| 15:26.14 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:27.11 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:27.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:28.08 | WIMPy | merlin8282: You can use both or even dahdi. Depends on waht features you want. |
| 15:28.18 | phr0zen | is there a way to see the echo cancellation at work? |
| 15:28.25 | c0rnoTa | I'v found it |
| 15:28.27 | phr0zen | to confirm that it is doing what it is supposed to |
| 15:28.51 | c0rnoTa | ru.po file should be placed in /var/lib/asterisk/unistimLang directory |
| 15:28.57 | merlin8282 | WIMPy: ah, that means I don't even absolutely *need* mISDN, if I already have dahdi working ? |
| 15:29.17 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:29.46 | WIMPy | merlin8282: Depends on your needs. You may find more info on http://voice.yeti.dk/Asterisk_vs_ISDN |
| 15:30.59 | russellb | WIMPy: thanks for all of your help in #asterisk lately, btw. :-) |
| 15:31.48 | WIMPy | russellb: np, I put a rant in betweeen :-) |
| 15:33.59 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 15:34.55 | russellb | heh, as long as they are constructive rants. |
| 15:35.03 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 15:35.37 | *** join/#asterisk plundra (1000@v0.article.se) |
| 15:35.47 | WIMPy | #1 would be that SIP is probably the worst choice for VoIP. |
| 15:36.17 | merlin8282 | thanks WIMPy |
| 15:36.20 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-ezsrbswsaprmprvo) |
| 15:36.24 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:36.42 | WIMPy | And #2 that Asterisk and the PSTN or classic phone technology don't mix very well. |
| 15:37.38 | WIMPy | I guess I should add a serious issues part, especially for misdn1. |
| 15:38.08 | *** join/#asterisk espiceland (~erin@nat/digium/x-lmfosvtoekceyfqy) |
| 15:38.10 | WIMPy | merlin8282: That's for you as well: With misdn1 never reject a call, it will kill your Asterisk. |
| 15:45.41 | phr0zen | guys, just had a crash on my asterisk server |
| 15:45.45 | phr0zen | wondering if i can get some help |
| 15:46.10 | phr0zen | CPU 1: Machine Check Exception: 4 Bank 5: b200001040100e0f |
| 15:46.50 | leifmadsen | sounds like a kernel issue |
| 15:46.52 | WIMPy | That looks like a hardware fault. |
| 15:46.55 | leifmadsen | or that |
| 15:47.02 | leifmadsen | fan still running? :) |
| 15:47.05 | phr0zen | yea |
| 15:47.11 | phr0zen | fans still going everything fine |
| 15:47.15 | phr0zen | i can reboot and it works fine |
| 15:47.20 | phr0zen | it is seemingly random |
| 15:47.39 | WIMPy | Probably a memory issue. |
| 15:47.44 | phr0zen | i thought so too |
| 15:47.48 | phr0zen | removed one stick |
| 15:47.55 | phr0zen | it took 4 days then crashed |
| 15:48.08 | WIMPy | Could be the cache memory as well. |
| 15:48.27 | WIMPy | Try to compile gcc. That's the best memory test, I know :-) |
| 15:48.28 | phr0zen | if it is the cache, how can that be addressed? |
| 15:48.58 | WIMPy | Try another CPU. |
| 15:49.55 | phr0zen | intel xeon 5130 |
| 15:49.58 | phr0zen | is what is in there now |
| 15:50.44 | phr0zen | just curious if there is a way to decode that message to something a bit more... specific |
| 15:55.36 | phr0zen | is there a way to test the cache memory? |
| 15:57.09 | *** join/#asterisk tobi- (~tobi@85.220.135.170) |
| 15:57.45 | WIMPy | doesn't lnow any real tests. But you could try to repeatedly verify a big archive file. |
| 15:58.21 | WIMPy | If you get errors at the same place, it's RAM if you get errors in random places, it's cache. |
| 15:58.30 | WIMPy | its |
| 15:59.39 | phr0zen | it is seemingly random |
| 15:59.50 | phr0zen | but, by 'same place'.. could you elaborate |
| 16:00.10 | WIMPy | Same place in the archive when you verify it. |
| 16:00.26 | WIMPy | It needs to fint into your RAM, obviousely. |
| 16:00.40 | tobi- | Is there a way to get rid of the "ztscan > /path/to/ztscan.conf" at the first login after fresh asterisk-gui install? |
| 16:01.18 | phr0zen | so, what would be my best steps to find the source of this error (CPU 1: Machine Check Exception: 4 Bank 5: b200001040100e0f) |
| 16:01.34 | phr0zen | if you have any suggestions |
| 16:02.06 | phr0zen | or, is there a way to have it so it doesnt lock up? (as in, sees the error but doesn't care) |
| 16:02.14 | WIMPy | You should try to search for someone with more knopwledge of PC hardware and MCEs. |
| 16:02.50 | phr0zen | google proved, not to good in that respect.. at least finding post about said error |
| 16:03.02 | phr0zen | most said RAM error (bad chip) |
| 16:03.28 | WIMPy | Seems likely indeed. |
| 16:03.48 | WIMPy | But it could happen on the way. |
| 16:04.03 | phr0zen | saw a few posts say swapping a ram chip worked for them |
| 16:04.14 | phr0zen | just curious if it could just be the slot it is in? |
| 16:04.35 | WIMPy | Living on the edge. |
| 16:05.02 | phr0zen | lol |
| 16:05.34 | phr0zen | it's an intel sr2500 in case u cared |
| 16:05.38 | phr0zen | :P |
| 16:06.03 | WIMPy | is not the PC hardware guy. |
| 16:06.30 | phr0zen | knows this, but just needs to chat it out, might spark an idea like in House |
| 16:06.41 | phr0zen | :) |
| 16:07.58 | *** join/#asterisk m_tadeu (~quassel@89.181.47.213) |
| 16:08.03 | WIMPy | You will have to fiddle with the hardware and stress test it. |
| 16:09.37 | phr0zen | would memtest be a good start? |
| 16:09.40 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 16:10.04 | WIMPy | If it detects your problem. |
| 16:10.20 | phr0zen | ah.. if's... my worst enemy |
| 16:10.46 | tobi- | anybody who could help me with an asterisk-gui first-login problem? |
| 16:11.04 | WIMPy | Otherwise as suggested before, try to compile gcc. If that works without issues, your hardware is most likely working perfectly. |
| 16:11.36 | *** join/#asterisk Tim_Toady (~moi@178.128.134.192.dsl.dyn.forthnet.gr) |
| 16:11.59 | WIMPy | I have been able to spot faulty memory that war that memtest didn't find. |
| 16:12.10 | phr0zen | ok, sounds like a good plan |
| 16:12.11 | WIMPy | s/war/way/ |
| 16:12.35 | phr0zen | any estimate on how long that usually takes? |
| 16:12.49 | phr0zen | 4 gig ram / xeon 5130 / sata drives |
| 16:13.12 | phr0zen | (just trying to plan how much time i need to allocate) |
| 16:13.13 | WIMPy | I's been a while I tried that. |
| 16:13.23 | *** join/#asterisk sled-dog (~luser@adsl-074-165-241-009.sip.msy.bellsouth.net) |
| 16:13.41 | WIMPy | But don't forget to start at least as many threads as you have CPU cores+HT. |
| 16:20.29 | *** join/#asterisk IdleOne (~IdleOne@ubuntu/member/idleone) |
| 16:20.35 | *** part/#asterisk IdleOne (~IdleOne@ubuntu/member/idleone) |
| 16:21.09 | sbrath | Anyone running a PRI with Time Warner.... I'm having a bunch of Echo problems the the TImeWarner PRI and am looking for options. |
| 16:21.09 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:22.39 | jaytee | sbrath, do you have a hardware echo cancel module on your T1 card? |
| 16:22.48 | sbrath | no I'm using OSLEC. |
| 16:22.52 | sbrath | at 256 taps. |
| 16:23.43 | sbrath | I'm just using a simple TE110 type card. |
| 16:24.14 | jaytee | was it working ok without echo and just started happening or has this been since you installed it? |
| 16:25.26 | sbrath | I've seen discussions on hardware vs software on the echo canceler, but with 1-2 channels running shouldn't the software canceler be enough ? Or is Time Warner causing?? more echo.. The service is terminated to me on a Cisco device from their Fibre. |
| 16:27.55 | jaytee | I've used TW's PRI circuits with * and a TE212 card with hardware echo cancellation and no problems but I've never used OSLEC so I couldn't say. |
| 16:28.08 | jaytee | might try 128 taps instead of 256 |
| 16:28.51 | *** join/#asterisk deeperror (~deeperror@76.226.171.215) |
| 16:29.45 | jaytee | anyone remember how to strip commented lines from a .conf file? been awhile since I've done it and I lost my notes for it. think I used sed at the time |
| 16:30.15 | deeperror | Been having what we call a dead-air dial where we hookflash a call and when making the 3way call it rings 1 or 2 times then dead air. In the CLI it shows Answered but we hear nothing. Other times it works fine. 1.4.38 |
| 16:31.31 | deeperror | jaytee: could use sed/awk? |
| 16:31.33 | sbrath | jaytee: When I tested it when it was installed, it sounded great. But the client claims that "Some" people who call them get a huge echo, but the far-end dosen't hear the echo, only the Asterisk side. Initially it was runing mg2 echo canceler with defaults, and I just compliled and configured OSLEC with 256 ... |
| 16:32.30 | sbrath | Maybe if I just leave OSLEC at defaults.... Can making it 256 taps have a chance to make it worse? |
| 16:33.02 | jaytee | possible |
| 16:33.31 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 16:34.19 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.1 (2010/01/18), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 16:34.59 | *** join/#asterisk jkroon (~jkroon@dsl-241-243-199.telkomadsl.co.za) |
| 16:34.59 | leifmadsen | Asterisk Security Releases for 1.4, 1.6.1, 1.6.2, and 1.8 branches are now available. Please see the announcement at http://www.asterisk.org/node/51557 and http://www.asterisk.org/security |
| 16:35.41 | w0ls0n | you think a Polycom Soundpoint IP 301 2200-11301-001 Phone would work ok with * |
| 16:38.41 | hc | oh wow. i had though the asterisk.org/security page was not maintained anymore. so few updates recently |
| 16:39.25 | leifmadsen | hc: that's a good thing :) |
| 16:39.37 | leifmadsen | it's maintained, just no security issues for nearly a year |
| 16:40.19 | *** join/#asterisk aree (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 16:40.22 | aree | hi there |
| 16:40.29 | hc | i suspected that much for some time - now i have a confirmation :) |
| 16:41.18 | aree | i don't know, i setup kind of voucher redeem system when i try to dial extension 102 : this is what i got on asterisk cli : <S0IP/102-00000001> Playing 'prepaid-enter-pin-number.gsm' (language 'en') but on the phone i don't hear any voice ask me to enter the pin code, i test with two phone, using xlite softphone, and also using linksys pap2 device, there is no voice out |
| 16:41.28 | aree | what could be the issue ? |
| 16:43.16 | hc | leifmadsen: the 1.6 asterisk source on asterisk.org/downloads - is the patch already included there? |
| 16:43.21 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 16:43.22 | QubeZ | hello all |
| 16:43.35 | leifmadsen | yes |
| 16:43.40 | leifmadsen | did you look? :) |
| 16:43.45 | QubeZ | How can I blacklist outbound dialing to specific numbers? I have a list of 2 numbers that I do not want to be called from our system. |
| 16:43.45 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:43.49 | leifmadsen | although you say 1.6, which doesn't exist |
| 16:44.04 | leifmadsen | QubeZ: add them to the dialplan and send them somewhere else |
| 16:44.14 | leifmadsen | QubeZ: most specific will match first |
| 16:44.26 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 16:44.56 | hc | well, i meant the latest version whose version string starts with 1.6 :) - 1.6.2.16.1 |
| 16:45.10 | leifmadsen | 1.6.2.15.1, 1.6.2.16.1, and 1.6.1.21 |
| 16:45.24 | QubeZ | leifmadsen: was hoping there would be a better way like in the Asterisk DB or something. But I'll explore that. |
| 16:45.28 | sbrath | QubeZ: I guess you could add the numbers to the outbound dialing plan and drop the call. |
| 16:45.39 | leifmadsen | we just released 1.6.2.16 on friday, so we made 1.6.2.15.1 as well so people wouldn't necessarily be forced to upgrade to get the security fix |
| 16:46.02 | leifmadsen | QubeZ: you can certainly do that -- use BLACKLIST() , or use ODBC to check from an external database. There are many ways to accomplish it. |
| 16:46.39 | leifmadsen | DB(), BLACKLIST, func_odbc. res_ldap, func_curl...... MANY ways |
| 16:46.47 | leifmadsen | it' |
| 16:46.52 | leifmadsen | it's all about the dialplan you implement |
| 16:47.05 | QubeZ | leifmadsen: thanks |
| 16:47.42 | hc | i just applied the patch to 1.6.2.7-rc2 - think that's a bad idea? |
| 16:48.47 | leifmadsen | I have no comment. |
| 16:49.04 | leifmadsen | runs off to lunch |
| 16:49.15 | hc | okay, i'll just try it then - i don't really want to risk upgrading atm |
| 16:51.24 | hc | seems to have worked alright. |
| 16:54.45 | deeperror | Dialing outbound 3-way call, line rings 1-2 times, then silence. CLI shows answered but hear nothing. Making 1000's of outbound calls only occasionally does this occur. |
| 16:57.03 | *** join/#asterisk Adrellias (c4d355a5@gateway/web/freenode/ip.196.211.85.165) |
| 16:57.34 | Adrellias | sup guys sorry to bother is there a way i can make a test call from the command line via a leased line ? instead of inhouse sip or aix ? |
| 16:58.29 | Adrellias | instead of using a spool file |
| 16:58.33 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:59.01 | WIMPy | 'channel originate' |
| 16:59.24 | Adrellias | hrmm cool let me check into that |
| 17:01.55 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:10.18 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 17:21.18 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 17:23.56 | *** join/#asterisk Russ (~russ@149-169-253-20.nat.asu.edu) |
| 17:27.38 | Adrellias | hey guys i have dahdi installed and started but it does not show up in asterisk for some reason. |
| 17:27.41 | Adrellias | any ideas |
| 17:32.25 | Adrellias | it does find the dahdi configfiles etc so no effin idea why aint giving me the dh |
| 17:32.29 | Adrellias | dahdi command |
| 17:34.45 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v008-227.mobile.uci.edu) |
| 17:40.20 | ruyo | Is there any problem in having a RTP port range of about 500? |
| 17:40.38 | ruyo | Besides, I assume, being limited to 250 calls. |
| 17:41.05 | ruyo | Or 125.. |
| 17:51.34 | *** join/#asterisk hdiogenes (~humberto@201.76.154.143.intranet.digi.com.br) |
| 17:55.45 | tzafrir | Adrellias, how can you tell it is installed? |
| 17:56.08 | tzafrir | what is the output of: dahdi_test -c3 ? |
| 17:56.11 | deeperror | Adrellias: may need to 'make clean', configure, and make asterisk again with the new dahdi installed |
| 17:56.32 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 17:57.03 | tzafrir | Adrellias, in asterisk: what is the output of 'dahdi show channels' ? |
| 17:58.25 | Adrellias | hey tzafrir deeperror it was reading the channel config files one startup. but i figured out one of the channels are dead and the conf file wasnt done correctly |
| 17:58.45 | Adrellias | its showing up now :) |
| 17:58.47 | Adrellias | now ust to figure out how to use cli originate to test a call with it |
| 17:59.17 | Adrellias | or a spool file |
| 17:59.55 | *** join/#asterisk thews (~thews@24-119-172-234.cpe.cableone.net) |
| 18:02.59 | Adrellias | tzafrir do you know how to make a outgoing call using a spool file and a dahdi/zap trunk |
| 18:04.09 | tzafrir | I suppose, but I'm on my way out, so I suggest you ask others |
| 18:04.33 | Adrellias | thanks |
| 18:04.37 | Adrellias | :) |
| 18:09.29 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 18:10.00 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v008-227.mobile.uci.edu) |
| 18:12.28 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 18:23.13 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
| 18:23.42 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:25.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:26.01 | *** join/#asterisk devmod (~devmod___@c-76-100-208-204.hsd1.md.comcast.net) |
| 18:33.55 | jaytee | is there an easy way to control the default musiconhold volume? |
| 18:37.01 | *** join/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 18:37.55 | citywok | jaytee: modify the source? :P |
| 18:39.19 | jaytee | citywok, that's what I was thinking. I installed Audacity on my Win 7 station and I edited prompt file earlier. Now I'm trying to play that file or a moh file and Audacity isn't giving me any output. If I play the same file in Media Player I can hear it. I've got the beta version of Audacity. |
| 18:41.20 | *** join/#asterisk clintc (~clintc@n128-227-5-241.xlate.ufl.edu) |
| 18:42.11 | jaytee | gotta reboot, brb |
| 18:42.15 | devmod | does the gtalk chan support video ? |
| 18:45.21 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 18:45.38 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 18:46.52 | sawgood | rtp.c 1373 ast_rtp_read: Unknown RTP codec 72 received from xxx.xxx.xxx.xxx |
| 18:47.04 | sawgood | can someone tell me what this Asterisk NOTICE error is about? |
| 18:47.12 | sawgood | it just started happening today |
| 18:48.36 | sawgood | I think I might have found it ... |
| 18:51.02 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 18:53.40 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 18:58.16 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:c62c:3ff:fe2d:b7db) |
| 19:00.01 | aree | hello |
| 19:00.03 | aree | there |
| 19:00.09 | carrar | HARRO |
| 19:00.17 | aree | is it possible to turnover asterisk as music server ? |
| 19:00.26 | Tozz_ | 'turnover' ? |
| 19:00.28 | aree | with play rewind forward option ? |
| 19:00.31 | aree | turn over |
| 19:00.47 | Tozz_ | why not use a music server instead? |
| 19:01.13 | aree | music server ? |
| 19:01.17 | tzafrir | Tozz_, what would you require of a music server? |
| 19:01.32 | Tozz_ | i'm confused |
| 19:01.41 | tzafrir | Tozz_, one problem: Asterisk tends to use telephony-quality audio |
| 19:01.50 | Tozz_ | tzafrir: I dont want a music server |
| 19:02.00 | Tozz_ | that was aree |
| 19:02.12 | tzafrir | aree, what would you require of a music server? |
| 19:02.33 | aree | i simply want to centralized music datat |
| 19:02.40 | aree | that's all |
| 19:02.42 | tzafrir | for a LAN? |
| 19:03.00 | aree | yes for example but not only that |
| 19:03.09 | tzafrir | sanity check: is mpd good enough for you? |
| 19:03.38 | aree | what ? |
| 19:03.50 | aree | first i want to know if is it possible ? |
| 19:03.54 | tzafrir | http://mpd.wikia.com/wiki/Music_Player_Daemon_Wiki |
| 19:04.01 | aree | then i got an other question |
| 19:04.19 | tzafrir | There are a bunch of other alternatives |
| 19:04.23 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 19:04.29 | tzafrir | Specifically, Asterisk won't handle playlists and such |
| 19:04.56 | Tozz_ | why not use SqueezeCenter for example? |
| 19:05.04 | carrar | Asterisk is a PBX, not a playlist manager :) |
| 19:05.13 | aree | no you don't get my point |
| 19:05.20 | tzafrir | And then again, you can just drop off all the files on a file server |
| 19:05.24 | aree | i want to hear music on phone |
| 19:05.30 | aree | on rest time |
| 19:05.39 | Qwell | For the archives: bkruse is the awesomest person ever. |
| 19:05.44 | aree | by dialing a number |
| 19:05.58 | aree | an also want to receive a call if that possible |
| 19:06.02 | tzafrir | Qwell, regarding? |
| 19:06.15 | Qwell | tzafrir: You know how they say TANSTAAFL? |
| 19:06.21 | Zairus | Class icon --> create a trayicon who launches a JDialog(null, true) --> Can I access the icon class properties from JDialog code? |
| 19:06.35 | Zairus | How can retrieve the ancestor object? |
| 19:06.38 | carrar | aree, asterisk can play music but not control it, FF RW next etc... |
| 19:06.54 | Zairus | sorry, I made a mistake with the channel |
| 19:07.23 | tzafrir | Qwell, well? |
| 19:07.29 | aree | are you sure ? carrar |
| 19:07.35 | Qwell | tzafrir: bkruse says otherwise |
| 19:07.56 | carrar | no, perhaps join #Asterisk_we_are_sure to really find out |
| 19:08.16 | aree | are you Mark spencer ? carrar |
| 19:08.37 | carrar | I could be |
| 19:08.40 | thehar | mark is kram |
| 19:08.42 | carrar | how would youknow |
| 19:08.52 | carrar | I have a pic of him with me |
| 19:08.57 | thehar | well that's good |
| 19:08.58 | carrar | does that count for anything? |
| 19:09.02 | Tozz_ | yes! |
| 19:09.03 | carrar | heh |
| 19:09.28 | carrar | http://pics.osburn.com/photo/35514/original |
| 19:09.35 | carrar | thats HOT++ |
| 19:09.48 | carrar | heh |
| 19:10.11 | aree | so impossible ? |
| 19:10.17 | aree | am i right ? |
| 19:10.25 | carrar | without changing the code yes |
| 19:11.04 | aree | what i have to change on the code ? |
| 19:11.14 | carrar | if you just want 1 channel of music you could use mpg123 to play your music and maybe have to do things to yor stream |
| 19:12.09 | aree | okay and is it possible to hear music and get incomming call on the same line ? car |
| 19:12.15 | aree | carrar: |
| 19:12.31 | carrar | incoming call would ring second line |
| 19:12.35 | Tozz_ | you could stream the MoH from a music server, then use featuremap to execute Macro's |
| 19:12.45 | Tozz_ | where the Macro's would send some command to the music server |
| 19:12.56 | aree | no i need on the same line |
| 19:13.08 | Tozz_ | same line? |
| 19:13.13 | carrar | aree, without using the music on hold functionality of asterisk you might be able to hack something up |
| 19:13.38 | leifmadsen | you could probably use AstDB along with the Bridge() application |
| 19:13.48 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 19:14.02 | Tozz_ | but WHY would you want to do this? :) |
| 19:14.11 | carrar | Cause you can! |
| 19:14.14 | carrar | maybe |
| 19:14.17 | leifmadsen | or, just use a MeetMe() room and play music until someone else joins, then when someone calls, put them into the conference room too, which would stop the MOH |
| 19:14.31 | carrar | but he wants to control the playback |
| 19:14.34 | carrar | FF |
| 19:14.34 | carrar | RW |
| 19:14.37 | leifmadsen | anything is possible with Asterisk if you're clever enough |
| 19:14.38 | carrar | next previos |
| 19:14.41 | carrar | previous |
| 19:14.46 | leifmadsen | carrar: MusicOnHold() allows that I believe |
| 19:14.55 | leifmadsen | so use Bridge() and MusicOnHold() |
| 19:15.00 | leifmadsen | (I've done this before to simulate a queue) |
| 19:15.08 | carrar | oh |
| 19:15.08 | leifmadsen | in fact it's how Bridge() was originally created |
| 19:15.10 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v008-227.mobile.uci.edu) |
| 19:16.15 | aree | it sound like mark spencer ihere , thanks a lot ok i understand leifmadsen |
| 19:16.44 | leifmadsen | I don't have any idea what you mean by the first part of your sentence, but you're welcome anyways |
| 19:16.50 | Tozz_ | :> |
| 19:17.13 | aree | which one ? leifmadsen |
| 19:17.19 | Tozz_ | the left |
| 19:17.21 | Tozz_ | or right |
| 19:17.24 | leifmadsen | again, not following at all |
| 19:17.42 | Tozz_ | i'm guessing he doesnt either |
| 19:18.28 | Faustov | Yo mamma so fat, she sat on a binary tree and flattened it into a linked list in O(1) time. |
| 19:18.43 | aree | ok leave it then |
| 19:18.45 | aree | thanks a ot |
| 19:18.47 | aree | lot |
| 19:18.56 | WIMPy | LOL |
| 19:19.05 | aree | bye |
| 19:19.11 | *** part/#asterisk aree (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 19:20.07 | *** join/#asterisk jetlag (~jetlag@pool-173-61-239-155.cmdnnj.east.verizon.net) |
| 19:20.23 | carrar | MusicOnHold doesn't mention anything about Fast forward, rewind, next previous |
| 19:21.11 | leifmadsen | I know I've seen it somewhere, I just can't remember where |
| 19:21.32 | carrar | seems like a nice idea |
| 19:23.30 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:23.30 | paulc | ControlPlayback does it, no? |
| 19:23.44 | carrar | wonder if MP3Player can |
| 19:24.07 | carrar | using a M3U playlist |
| 19:24.28 | carrar | nothing about selection controls |
| 19:24.28 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 19:24.55 | carrar | ah yeah ControlPlayback looks like it does |
| 19:25.12 | carrar | not list of files |
| 19:25.12 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 19:26.27 | JonnyD_work | is it posable to kick someone out of a queue if they don't answer when their phone is being rung? |
| 19:28.39 | citywok | JonnyD_work: there is a pause member option |
| 19:29.13 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 19:29.35 | JonnyD_work | citywok: how does that work? do they have to log out and back in? |
| 19:31.43 | tobi- | how is it possible to play a message if all agents in a queue are either inuse oder unavailable? |
| 19:37.24 | citywok | JonnyD_work: not sure, one of the other guys probably knows. my guess would be it unpauses them if they make an outbound call, but i could be wrong. |
| 19:37.40 | plundra | tobi-: You can make it fall through the Queue()-command when that happen. And then check condition afterwards and then playback. |
| 19:37.44 | citywok | autopause=yes is the option. disclaimer:i've never used it |
| 19:38.05 | citywok | tobi-: there are also periodic announcement messages |
| 19:41.04 | JonnyD_work | tobi-: yes you can do that if eveyone in the queue is on the phone you can kick the call out of the queue and play any message you want with playback |
| 19:42.28 | tobi- | awesome, is there an example somewhere to be found? |
| 19:49.19 | JonnyD_work | well you just let it fall through use the timeout option on the queue exten => 1589,4,Queue(testq|t|||45) |
| 19:50.22 | JonnyD_work | so that will let the caller sit in queue for 45 seconds then it will fall thorough to the next step where you can use play a sound ex: exten => 1589,5,Playback(sorry-it-is-taking-so-long) |
| 19:50.37 | JonnyD_work | then you could add them to the queue again |
| 19:55.35 | tobi- | no other way? dont realy want ppl to wait x seconds first |
| 19:58.33 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 19:58.47 | jaytee | and once again sox saves the day |
| 19:59.37 | jaytee | I just had a weird incident where inbound calls from flowroute worked and internal extension to extension calls worked but outbound calls would fail with no indication. |
| 20:00.18 | jaytee | this is on asterisk 1.6.2.14 |
| 20:10.45 | leifmadsen | tobi-: http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id36004667 |
| 20:11.04 | leifmadsen | tobi-: that section is about announcement control in Queue() |
| 20:11.45 | pabelanger | Anybody else running 1.8 notice: [Jan 18 04:27:12] ERROR[18715]: res_timing_timerfd.c:171 timerfd_timer_ack: Read error: Resource temporarily unavailable |
| 20:12.03 | leifmadsen | hmmm I haven't seen that, but I don't think I'm using timerfd |
| 20:13.50 | WIMPy | What happened to chan_dahdi? That seems completely borked ATM. |
| 20:15.43 | WIMPy | Apart from other things I cannot get any audio on bridged calls any more. |
| 20:21.53 | leifmadsen | WIMPy: which version? |
| 20:22.28 | WIMPy | libpri, dahdi-linux and Asterisk all from svn, updated a few minutes ago. |
| 20:22.30 | JonnyD_work | tobi-: yes there are other ways what specificly did you want to do? |
| 20:24.22 | leifmadsen | WIMPy: 1.4 branch for libpri and 1.8 branch for asterisk I presume? |
| 20:24.26 | leifmadsen | hmmmm wonder what changed |
| 20:24.39 | leifmadsen | I just made 1.8.3-rc1 today too, so should check that versus 1.8.2 |
| 20:24.43 | WIMPy | Yes |
| 20:24.54 | leifmadsen | any chance you could check 1.8.2 to see if the problem reverts itself? |
| 20:25.38 | WIMPy | Actually I wonder if that issue was present already when I last tried a few days ago and I might have drawn fals conclusions then. |
| 20:25.50 | WIMPy | I will try 1.8.2. |
| 20:26.22 | *** join/#asterisk thx2000 (~thx2000@netblock-208-127-148-73.dslextreme.com) |
| 20:26.26 | leifmadsen | ok cool |
| 20:31.31 | WIMPy | Meanwhile I notice that ECT behaves strangely. I only seems to work between bridged calls. Trying to transfer to something like Echo() or MusicOnHold() results in the call being dropped instead. |
| 20:34.10 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 20:35.04 | leifmadsen | carrar: per musiconhold.conf.sample |
| 20:35.05 | leifmadsen | ;digit=# ; If this option is set for a class, then when callers are |
| 20:35.05 | leifmadsen | ; ; listening to music on hold, they can press this digit, and |
| 20:35.05 | leifmadsen | ; ; they will switch to listening to this music class. |
| 20:36.38 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 20:37.49 | carrar | heh |
| 20:39.07 | leifmadsen | I effin' knew there was an option to do that :) |
| 20:41.23 | carrar | hehe |
| 20:43.09 | tzanger | "please use the 7 key to switch the type of ad you listen to while waiting for customer service." |
| 20:43.49 | leifmadsen | :D |
| 20:44.21 | leifmadsen | "On a level of 1 to 10, how invasive of an ad would you like? Press 1 for extremely invasive, up to 10 for infinitely invasive" |
| 20:47.46 | _Corey_ | Some big company must have offered this at one point... I had a customer ask about this about 5 years ago |
| 20:48.41 | WIMPy | If you are interested in travel, press 1, if not, press 2. |
| 20:48.46 | WIMPy | If you are interested in mobile phones, press 1, if not, press 2. |
| 20:49.03 | WIMPy | If you are interested in finance, press 1, if not, press 2. |
| 20:49.08 | WIMPy | ... |
| 20:50.12 | _Corey_ | I think the company gave them a choice of music, though I think prompting them to choose their favorite advertising would be entertaining if you monitored the channels |
| 20:51.18 | WIMPy | leifmadsen: No go with 1.8.2, either. |
| 20:51.41 | WIMPy | Could that be dahdi itself, not Asterisk? |
| 20:52.15 | WIMPy | tries 1.8.1.2 |
| 20:53.18 | Defraz | Has anyone tried the Aastra and Yealink phones? |
| 20:54.37 | citywok | Defraz: i like the aastra phones myself, for the price they're pretty good, and really easy to configure/provision compared to polycom/cisco gear |
| 20:54.45 | leifmadsen | WIMPy: thanks, keep me posted. Would be good to track down where the issue was introduced |
| 20:55.05 | Defraz | yea, I used them a while back and they just seemed kinda cheap and was wondering if they got any better. |
| 20:55.12 | Defraz | the on off hook busted on them over time. |
| 20:55.17 | WIMPy | I shouldn't have given ex-gf the faster machine :-( |
| 20:55.22 | Defraz | YEa they seemed easy to configure. |
| 20:56.01 | leifmadsen | WIMPy: been there done there |
| 20:56.03 | leifmadsen | that* |
| 20:56.27 | citywok | Defraz: with the 57i after a year i haven't had any break |
| 20:56.35 | citywok | i have 60 or 70 of them |
| 20:58.12 | Defraz | okay, cool cool, just read some on teh yealink phones and was curious if any has tried them. |
| 20:58.47 | citywok | nope, but i saw them at astricon and they looked like cisco knockoffs |
| 20:59.03 | Defraz | yea see we have cisco phones now |
| 20:59.18 | Defraz | Kinda thought that myself. |
| 20:59.20 | WIMPy | They certainly look nice, but they don't seem to be supported. |
| 20:59.54 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
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| 21:04.18 | WIMPy | leifmadsen: No luck with 1.8.1.2 either. Will try dahdi 2.4.0 next. |
| 21:06.22 | jaytee | WIMPy, don't make the dumb mistake I'd often make which was not to recompile * after upgrading DAHDI or ZAPTEL |
| 21:06.56 | WIMPy | I won't :-) |
| 21:07.16 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 21:07.39 | jaytee | I already knew about the compile order and that I had to do it but when I was deep into troubleshooting and trying fixes I'd forget :-) |
| 21:07.48 | jaytee | getting effin old |
| 21:11.30 | WIMPy | leifmadsen: Both 1.8.1.2 and trunk work ok with dahdi 2.4.0. |
| 21:14.01 | WIMPy | Does anyone know if that is driver dependent or general? |
| 21:18.11 | leifmadsen | WIMPy: hmmm thanks for the info. Gotta reboot but will look shortly. I would check the changes to chan_dahdi and see if anything looks like it may have affected it. Then test just before and at that revision if you can. Then we can narrow down exactly which revision, and then we can mark it as a blocker/regression for the next versions |
| 21:18.17 | leifmadsen | russellb: ^^^ |
| 21:18.54 | WIMPy | I will try to see if I find something. |
| 21:20.36 | leifmadsen | brb |
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| 21:30.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:31.25 | WIMPy | leifmadsen: The issue with dahdi was actually fixed a few minutes after I spotted it. |
| 21:31.36 | leifmadsen | oh really! |
| 21:31.50 | leifmadsen | sounds like I might need to do an RC2 to include those fixes.... |
| 21:32.20 | WIMPy | Just another 'svn up' would have saved some 45 minutes, I guess. Bad luck. |
| 21:32.39 | WIMPy | That was dahdi, not Asterisk. |
| 21:33.29 | leifmadsen | oh was it? ok good to know |
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| 21:33.40 | leifmadsen | thanks for looking into it -- sorry for the wild goose chase |
| 21:33.41 | *** part/#asterisk mountainm2k (~msturtz@www.booyahnetworks.com) |
| 21:35.30 | WIMPy | Let's feel lucky it was fixed already. |
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| 22:18.17 | pabelanger | WIMPy: Don't worry, I was trying to debug an issue earlier this week; it was fixed more then 4 months ago. I don't know how I managed to let my local branch get so outdated. |
| 22:21.35 | Katty | peeks in |
| 22:26.13 | n3hxs-wk | spots Katty and waves. |
| 22:31.01 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
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| 22:32.19 | ClintGoudie-Nice | Greetings all. I'm looking at the Asterisk SRTP install guide here: http://www.voip-info.org/wiki/view/Asterisk+SRTP and it says to do an svn get of asterisk-srtp here ( svn co http://svn.digium.com/svn/asterisk/team/group/srtp asterisk-srtp ) but alas, that server is responding that the directory doesn't exist. |
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| 22:32.31 | ClintGoudie-Nice | Can anyone point me to where that svn has moved? |
| 22:33.11 | WIMPy | Asterisk 1.8 has srtp included. |
| 22:33.30 | WIMPy | But it can cause crashes. |
| 22:33.47 | *** join/#asterisk DennisG (~DennisG@541E88D0.cm-5-7c.dynamic.ziggo.nl) |
| 22:34.38 | ClintGoudie-Nice | the AsteriskNOW install I used appears to have installed 1.6 |
| 22:36.54 | thews | responsibility of maintaining our voip system(s) has been handed over to me, we have a mixture of polycom 430s and 501s, Off hand do any of you know if the two's firmware and config files can co-exist on the same tftp server? |
| 22:37.13 | *** join/#asterisk voudras (~voudras@cpe-67-253-160-235.rochester.res.rr.com) |
| 22:37.16 | voudras | hi all |
| 22:37.29 | thews | I am reading through some of the documentation, but have no experience setting these up from scratch |
| 22:37.44 | voudras | download the free orielly book |
| 22:37.53 | ClintGoudie-Nice | has the srtp project been moved somewhere else in svn? |
| 22:38.05 | thews | thews: responsibility of maintaining our voip system(s) has been handed over to me, we have a mixture of polycom 430s and 501s, Off hand do any of you know if the two's firmware and config files can co-exist on the same tftp server? |
| 22:38.16 | thews | is there a free book for that? :D maybe a blog |
| 22:38.52 | Qwell | ClintGoudie-Nice: It is in svn as of Asterisk 1.8 |
| 22:39.04 | voudras | yea one sec |
| 22:39.43 | voudras | i believe its here but its loadng slow for me www.asteriskdocs.org/ |
| 22:40.08 | voudras | yea thats it - click the link "To download the entire book (in PDF format), click here, or on the book cover image!" |
| 22:40.23 | WIMPy | ~book |
| 22:40.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 22:40.28 | WIMPy | or |
| 22:40.33 | WIMPy | ~newbook |
| 22:40.33 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 22:40.48 | voudras | ok i got a question. first off im new to asterisk - but i've been reading a lot, second im stuck using 1.2.4. my question regards a dialplan fall through |
| 22:41.32 | voudras | i want to verify that if i have a context for a local phone pickup that does, say - 4 steps for 4 different things (example, long distance, local extension, etc) |
| 22:41.57 | raden | Naikrovek, yo |
| 22:41.57 | voudras | i am trying to place a notification AGI script, im wondering if there is a way i can place a notice at the begining and end |
| 22:42.00 | raden | Katty, BOO :) |
| 22:42.08 | Katty | eeks |
| 22:42.20 | voudras | rather than between each dial statement |
| 22:42.59 | voudras | does my question make sense or should i clarify |
| 22:50.37 | thews | thanks for the link to that book, it's pretty helpful |
| 22:52.33 | *** join/#asterisk Faithful (~Faithful@carame.lnk.telstra.net) |
| 23:00.57 | voudras | thews: indeed |
| 23:01.48 | raden | how Katty |
| 23:06.52 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:08.57 | ClintGoudie-Nice | Ok, I'm trying to upgrade to 1.8 for SRTP, but chan_sccp isn't compiling. First error is: chan_sccp.h:44:26: error: asterisk/rtp.h: No such file or directory |
| 23:09.07 | ClintGoudie-Nice | I've got asterisk-devel installed |
| 23:10.18 | Qwell | Ask the chan_sccp folks to update it for 1.8. |
| 23:11.19 | ClintGoudie-Nice | I'm stuck between a rock and a hard place now. I can either have srtp from 1.8 or I can have sccp :S Is there any way to get access to the previous SRTP module that would build with 1.6? |
| 23:11.37 | *** join/#asterisk NuclearLucifer (gavroche@gavroche.pl) |
| 23:12.40 | Qwell | It never worked properly. |
| 23:12.45 | Qwell | So, that would be quite futile. |
| 23:26.40 | ClintGoudie-Nice | Is there an alternative to chan-sccp in 1.8? I've got some older cisco phones that dont have sip firmware to apply :S |
| 23:28.12 | Qwell | There's always been an alternative. chan_skinny has worked quite well since 1.4. |
| 23:29.37 | ClintGoudie-Nice | Thanks Qwell. I'll have to do some reading on that |
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| 23:36.02 | nightwalk | I have a tdm410 I'd like to separate out into it's own system (I'm told X & asterisk don't get along very well). |
| 23:36.55 | nightwalk | Any recommendations on embedded boards & enclosures (ex: microtik's) that'd be able to handle running a SOHO asterisk install? |
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| 23:54.01 | DarkRift | Anyone have made the sphinx speech recognition plugin ported on asterisk 1.8.1 or any other speech recognition available other than lumenvox ? |
| 23:55.11 | *** part/#asterisk wizard171 (~wizard171@h97.51.20.98.dynamic.ip.windstream.net) |
| 23:56.24 | frigidzephyr | DarkRift: there is also Vestec, besides lumenvox |
| 23:57.37 | ClintGoudie-Nice | What coul dcause chan_skinny to reject a device with the message "Device not found" |
| 23:57.45 | ClintGoudie-Nice | I've followed the sample config |
| 23:57.49 | DarkRift | yeah well I wanted something free |
| 23:58.51 | ClintGoudie-Nice | the only thing I haven't specified is host. Do I have to specify the host IP of the phone? |
| 23:58.57 | ClintGoudie-Nice | or can I specify a range? |