| 00:11.18 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:11.18 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.1 (2012/01/19), 1.8.8.2 (2012/01/19), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 00:12.38 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:19.43 | *** join/#asterisk keepar (ayane@quality.webhosting.at.sunsaturn.com) |
| 00:34.56 | *** join/#asterisk cyborg-one (1000@31.31.99.227) |
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| 01:01.09 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 01:11.12 | Ice_Strike | Hello |
| 01:12.20 | Ice_Strike | I have managed to get asterisk working and it seem doing fine so far. |
| 01:12.43 | Ice_Strike | Now I want to develop a call center website to allow agents to use the website |
| 01:12.51 | Ice_Strike | Which PHP libabry do you recommend? |
| 01:36.36 | [TK]D-Fender | ~toywy |
| 01:36.37 | infobot | somebody said toywy was The one you write yourself. |
| 01:56.34 | carrar | I have that one |
| 01:56.43 | carrar | It's crap |
| 01:57.10 | carrar | And the author won't fix it |
| 02:15.27 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 02:50.15 | ChannelZ | hehe |
| 02:51.14 | *** join/#asterisk min3r (1000@173.81.252.114) |
| 03:20.40 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
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| 03:47.04 | WIMPy | Ha, I'm no longer alone. Someone googled for "dahdi pri G722". |
| 03:51.14 | [TK]D-Fender | misery loves company ;) |
| 03:54.22 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 04:01.12 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 04:02.07 | WIMPy | Look at the number of users in here. |
| 04:05.38 | [TK]D-Fender | rawr |
| 04:18.00 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 04:20.41 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 04:26.00 | Dovid | is a 180 with SDP "correct SIP"? |
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| 04:38.21 | bbourdage | Can you change the keys for delete message, and other items in voice mail, or are they hard coded ? |
| 04:38.59 | Dovid | bbourdage: Can you explain what you mean by "keys" |
| 04:39.03 | [TK]D-Fender | bbourdage: vi app_voicemail.c |
| 04:39.25 | [TK]D-Fender | Hard-coded. Fortunately you have the code |
| 04:39.35 | bbourdage | 7 for delete, would like to mimic our current system which is 3 |
| 04:39.43 | bbourdage | Thanks Fender, that helps a ton |
| 04:39.51 | Dovid | wow i am tired. didnt see the entire msg |
| 05:43.18 | *** join/#asterisk Linux4Eric (~chatzilla@cpe-71-72-172-65.woh.res.rr.com) |
| 05:46.38 | Linux4Eric | is dahdi still required now that the linux kernel timer can be used? |
| 05:46.55 | Linux4Eric | on 1.8 |
| 06:01.24 | ChannelZ | you still need it for audio mixing if you use MeetMe |
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| 06:47.27 | Hadi_ | Hello.. anyone have experience with Asterisk and Polycom "Watch Buddy"? |
| 06:48.14 | Hadi_ | We have a bunch of phones behind NAT (the asterisk machine is on a public IP outside) and having problems getting Watch Buddy to work on the Polycom IP 650 expansion module. |
| 06:50.52 | Linux4Eric | Hadi_, has this ever worked yet for you? |
| 06:51.49 | Hadi_ | yes it works fine |
| 06:52.01 | Linux4Eric | What I mean is, does the IP 650 make and receive calls? |
| 06:52.03 | Hadi_ | but when there is lots of call volume on the ext being monitored |
| 06:52.19 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 06:52.31 | Hadi_ | the lights stop working... they either stay lit.. (as if the users on the phone) or they do not indicate that the ext is on the phone |
| 06:52.48 | Hadi_ | when we reset the phone.. (reception phone) everything works... then after a while it stops again |
| 06:52.56 | Linux4Eric | I am not familiar with watch buddy but I assume it is some sort of Busy Lamp that can show status of extensions? |
| 06:53.08 | Hadi_ | yes exactly |
| 06:54.53 | Linux4Eric | I have had much difficulty with NAT and polycom phones, but it sounds like NAT is working properly since your phones all work correctly |
| 06:55.31 | Linux4Eric | so watch buddy uses asterisk BLF "Hints", is that correct? |
| 06:55.58 | Hadi_ | yes |
| 06:56.59 | Linux4Eric | you may be running out of bandwidth with too many calls and that packet that changes the BLF on the phone might get dropped |
| 06:57.23 | Hadi_ | there is a dedicated DSL line for 10 ext's |
| 06:57.29 | Hadi_ | and we are using g729 |
| 06:58.26 | Hadi_ | It seems that when its not workingÂ… the phone send a REGISTER request but does not send the SUBSCRIBE as it should be. The INVITE is coming to the phone before it manage to send the SUBSCRIBE.. In this case the PUBLISH and the NOTFY are not sent. |
| 06:58.55 | Hadi_ | when we restart the phone... it starts to work again for a little while |
| 06:59.22 | Linux4Eric | hmm. Once in a great while my BLF flip out for about 10 seconds but always return to normal. |
| 07:00.09 | Linux4Eric | By the way, I do the BLF on cisco/linksys phones and I have to change a setting that states its an Asterisk server. Anything like that on the IP650? |
| 07:00.31 | Hadi_ | I don't think so... |
| 07:00.42 | Hadi_ | The issue only with Polycom we have no issue with other phones |
| 07:02.57 | Linux4Eric | http://www.asteriskguru.com/archives/image-vp258046.html |
| 07:03.13 | Linux4Eric | This states a firmware upgrade solved someone's problem |
| 07:03.22 | Linux4Eric | a few posts down |
| 07:04.45 | Hadi_ | let me check |
| 07:04.56 | Hadi_ | weare running latest firmware |
| 07:05.19 | *** part/#asterisk bbourdage (~bbourdage@174.33.93.154) |
| 07:08.41 | Linux4Eric | I don't know then. Definitely seems like it's the phone since it tries to send subscribe before registration |
| 07:10.29 | Linux4Eric | or doesn't send subscribe, have you confirmed that asterisk is sending device state change information about the particular extensions from the console? WHat version of Asterisk are you running out of curiosity |
| 07:11.50 | Hadi_ | 1.6.1.20 |
| 07:18.41 | Linux4Eric | I am out of suggestions, seems like you've covered a lot of bases in your troubleshooting. It's probably not an Asterisk issue, but either network (nat or otherwise) or Phone firmware bug |
| 07:19.48 | Hadi_ | yes... it is not asterisk |
| 07:20.01 | Hadi_ | was wondering if someone had same issue... perhaps some changes to the phone config will resolve it |
| 07:21.46 | Linux4Eric | yes i hear ya, but the fact that it works fine until lots of traffic makes me think that the settings are right. Do you use QoS or any bandwith shaping on your DSL router? |
| 07:22.39 | Hadi_ | no since the DSL is dedicated to the phones |
| 07:22.50 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 07:23.06 | Hadi_ | I'm wondering if setting the reception phone under DMZ settings of the router will resolve the issue |
| 07:25.22 | Linux4Eric | it would remove nat and do more like port forwarding so try it and see if different result. |
| 07:26.07 | Linux4Eric | I gotta run, good luck |
| 07:31.56 | *** part/#asterisk Linux4Eric (~chatzilla@cpe-71-72-172-65.woh.res.rr.com) |
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| 11:22.57 | *** join/#asterisk webczat (webczat@193.200.46.16) |
| 11:23.05 | webczat | hello |
| 11:23.13 | webczat | How to get documentation on asterisk10? |
| 11:32.44 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
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| 11:42.42 | webczat | should I read sample configs? |
| 11:57.44 | wdoekes2 | webczat: read the CHANGES and UPGRADE files, they'll tell you what's different from 1.8 |
| 11:58.04 | wdoekes2 | ~book |
| 11:58.05 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 11:58.11 | webczat | are config examples and command/app references enough to understand things? |
| 11:58.27 | wdoekes2 | 10 isn't a completely different version |
| 11:58.30 | webczat | or wiki for 1.8, maybe? |
| 11:58.37 | wdoekes2 | it's just 1.8 with certain enhancements |
| 11:59.59 | webczat | okay. so wiki for 1.8? or maybe can I just read docs in sample configs and base on it? |
| 12:10.19 | webczat | hmm what is call parking? |
| 12:15.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 12:15.59 | jzaw | webczat: ive migrated more or less the same confs from 1.2 - 1.4 - 1.6 - 10 |
| 12:16.09 | jzaw | editing new / changed stuff as ive been going on |
| 12:16.35 | jzaw | there have been changes but they always come up in the console / logs as errors so you can edit them |
| 12:18.35 | webczat | I am actually new to asterisk in general |
| 12:19.43 | webczat | and trying to configure it using, but without, sample configs. like handwriting configs, but reading samples for it |
| 12:21.46 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 12:30.41 | jzaw | nods |
| 12:31.15 | jzaw | webczat: thats how i started |
| 12:31.17 | jzaw | actully |
| 12:31.21 | jzaw | actually |
| 12:31.32 | jzaw | what was useful was to take some asterisk distro |
| 12:31.35 | jzaw | with webgui |
| 12:31.44 | jzaw | and add / remove / edit extensions |
| 12:31.51 | jzaw | and see how the actual textual confs changed |
| 12:31.59 | jzaw | you get a feel for the syntax that way |
| 12:32.11 | jzaw | then i went for my own roll out |
| 12:32.41 | jzaw | also extensions.ael is much much easier to read and especially easier to write |
| 12:32.58 | webczat | mhm |
| 12:33.06 | jzaw | imho of course |
| 12:33.08 | webczat | now, I don't actually understand what are parking lots/etc |
| 12:33.29 | webczat | and transfers. I mean, can't I do a transfer using my sip client? |
| 12:50.06 | *** join/#asterisk RyuunoAelia (~any@2a02:120b:c3f6:a641:219:66ff:fedd:acbc) |
| 12:50.18 | RyuunoAelia | hello |
| 12:50.41 | RyuunoAelia | I have a somewhat long question : |
| 12:51.02 | RyuunoAelia | I have this fiber router http://www.swisscom.ch/res/internet/dsl/router/index.htm?languageId=en (which is in fact a rebrand of http://www.adbglobal.com/broadband/triple-play/prg-f4202n.html ). |
| 12:51.24 | RyuunoAelia | This router is imposed on me, and I am quite dubious about security. It has an Asterisk running as uid 0 facing internet (because the routeur has 2 analog rj-11 connectors for the phone). And when I connect via ssh (OpenSSH-4.2_p1 ...) it runs linux kernel version 2.6.16... and asterisk -V gives me "Asterisk" no version number... |
| 12:51.29 | RyuunoAelia | Is there a way to know which version/security issues this Asterisk has ? |
| 12:52.07 | RyuunoAelia | (I am more interested in the security issues as this asterisk is internet-facing and the only way they give me to use my phone number...) |
| 13:10.25 | *** join/#asterisk mchou (~mchou@unaffiliated/mchou) |
| 13:28.56 | webczat | what's the bridge module and what's the dial module? |
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| 13:31.21 | *** join/#asterisk Ashutto (~Ashetic@2001:b00:7c00:600:6ef0:49ff:fe7d:3d35) |
| 13:31.25 | Ashutto | Hello |
| 13:32.00 | RyuunoAelia | webczat: it gets complicated because it uses some kernel module for an ADC-DAC from broadcom (the whole router is a broadcom SoC) |
| 13:35.15 | RyuunoAelia | webczat: loaded modules are : res_adsi res_monitor res_features pbx_config chan_jdsp chan_sip app_dial app_openrg_cmd app_transfer app_voicemail |
| 13:35.47 | RyuunoAelia | chan_jdsp is the abstraction for the broadcom ADC-DAC |
| 13:37.41 | RyuunoAelia | I asked my ISP to get the source code (as it is GPL), but they take their time... delaying with "we have to ask the higher-ups how to do it" and so on |
| 13:39.48 | file | RyuunoAelia, a quick Google leads here - http://a1-forum.at/kombipaket-aonspeed-privatkunden/neue-prgav4202n-version-von-4-8-3-dwvv-tau-5-1-6-t970.html |
| 13:40.04 | file | lots of good info, maybe a link to source |
| 13:41.40 | RyuunoAelia | hard for me to tell I don't read german :D |
| 13:42.24 | webczat | hemm a question: extension contexts are groups for extensions, does it mean I as an user have the assigned extension context or what? |
| 13:46.00 | RyuunoAelia | file: is there no way to know except for me to find or get the sources ? |
| 13:46.57 | file | if the core show version won't tell you then the binary doesn't know |
| 13:47.08 | file | well, if you don't have a 'core show version' you can narrow it down |
| 13:47.23 | file | because the command changed from show version to core show version after... 1.4? I think? it was long ago |
| 13:47.46 | RyuunoAelia | aoutch... |
| 13:48.02 | RyuunoAelia | "No such command 'core' (type 'help' for help)" |
| 13:52.37 | RyuunoAelia | so I guess the kernel is not the only thing "really old" on my router... |
| 13:53.30 | RyuunoAelia | well there is openssh too ... version 4.2_p1 is quite old too ... |
| 13:54.47 | RyuunoAelia | that's something for a router that began *selling* 4 months ago ... |
| 13:56.14 | file | generally the software provided for socs contains really old software |
| 13:56.21 | file | that is 'confirmed' to work happily |
| 13:56.34 | file | and they don't want to go to the trouble of upgrading because of the cost in time and money |
| 13:59.51 | RyuunoAelia | yes... but I don't have any choice I am forced to use this thing... I can't event buy a fibre PCI-e card and plug it... plus the thing is a "firewall" (nice security)... and well the software provider is based in Israel ... the kind of things that makes me trust my hardware ^^ |
| 14:03.09 | RyuunoAelia | plus I had to buy this router... and they won't let upgrade it myself because everything is closed... you apparently you need to sign the firmware to get it into the SoC (I didn't even try because I don't even have the sources) |
| 14:04.37 | RyuunoAelia | well complaining here won't change anything ^^ |
| 14:06.49 | RyuunoAelia | and from a security point of view considering that it's an Asterisk 1.4 or something are there remotely exploitable security issues ? |
| 14:10.25 | *** join/#asterisk singler (~singler@84.15.129.49) |
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| 14:18.35 | Neptu | hej if i do a sequection in a plsql scrip the whole script is a transaction?? |
| 14:19.10 | Neptu | how can i commit on some points?? |
| 14:19.19 | Neptu | or rollback |
| 14:34.11 | Neptu | how can i do several commits over a loop in a pl/sql?? |
| 14:42.19 | RyuunoAelia | Neptu: maye asking a pl/sql channel will get you an answer ^^ |
| 14:42.58 | Neptu | there is a channel only for postgres pl/sql?? |
| 14:46.22 | RyuunoAelia | Neptu: well there probably a postgres channel ^^ |
| 14:49.33 | webczat | what happens if you set parkexcl_exclusive to yes? |
| 15:09.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:16.46 | webczat | I can't actually test everything yet and not everything is documented |
| 15:18.19 | [TK]D-Fender | webczat: Yes, devices are pointed to a context. That says what that device can dial |
| 15:19.16 | [TK]D-Fender | webczatwhat's the bridge module and what's the dial module? <- Bridge is a dialplan application that lets you hook into another call in progress. Like a way of forcing a 3-way call, etc. |
| 15:19.31 | [TK]D-Fender | webczat: Many other ways to use that tool for other similar means. |
| 15:19.56 | [TK]D-Fender | [07:33]webczatnow, I don't actually understand what are parking lots/etc |
| 15:19.58 | [TK]D-Fender | [07:33]webczatand transfers. I mean, can't I do a transfer using my sip client? |
| 15:20.30 | [TK]D-Fender | Parking lots are so you can trasfer a call you're on to a "holding area" so that you can pik it up froma completely different device by calling up the lot # |
| 15:20.48 | [TK]D-Fender | And not sure what you meant more precisely about your transfer question. |
| 15:26.15 | webczat | [TK]D-Fender: I understand parking lots. my question was about why there are multiple of them? and what's parkext_exclusive then? an intelligent parking or what? |
| 15:27.23 | *** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net) |
| 15:27.52 | [TK]D-Fender | Not sure on the finer points, but picture using your system to host multiple companies. You don't want company A picking up company B's parked calls. |
| 15:27.58 | [TK]D-Fender | As for that parm... lets take a looks.. |
| 15:29.28 | [TK]D-Fender | Hrm, wording is a little vague... |
| 15:30.01 | webczat | I understand it like I can have two parking lots, one set on 701 to 799, one from 800 to 899, and both using number 1 for call parking |
| 15:30.51 | [TK]D-Fender | doesn't sound right... |
| 15:31.34 | [TK]D-Fender | "* Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access." |
| 15:31.43 | [TK]D-Fender | this is a little off as well.. |
| 15:32.20 | webczat | maybe I'm right in my understanding. |
| 15:32.31 | webczat | hmm, the callee parks, not the caller, right? |
| 15:33.02 | [TK]D-Fender | The caller parks the other call |
| 15:33.12 | webczat | ahh |
| 15:33.25 | [TK]D-Fender | We are talking. I transfer you to the parking ext. it reads the lot #. I hang up and then you're handed off |
| 15:33.53 | webczat | and I can unpark then |
| 15:34.02 | webczat | what is the pickup dtmf? |
| 15:34.19 | [TK]D-Fender | You dial an exten that calls ParkedCall() |
| 15:34.43 | [TK]D-Fender | one is generated in the parking context based on that other setting to specify a pickup exten |
| 15:34.50 | [TK]D-Fender | but you can also make your own. |
| 15:35.05 | webczat | I meant the other parameter set to *8 by default |
| 15:35.40 | [TK]D-Fender | I'm not sure the point of features.conf for that method.. |
| 15:35.46 | [TK]D-Fender | Feels kinda pointless |
| 15:35.59 | [TK]D-Fender | wait.. that is CALL pickup, not parking |
| 15:36.13 | webczat | mhm |
| 15:36.14 | [TK]D-Fender | call pickup is to hijack a call that is actually ringing somewhere else |
| 15:36.25 | [TK]D-Fender | I see your phone ringing and want to answer from mine |
| 15:36.26 | webczat | like erm what? |
| 15:36.43 | webczat | ah so then what do I do? |
| 15:37.37 | [TK]D-Fender | For what? |
| 15:37.59 | webczat | How do I pick the call up? |
| 15:38.28 | [TK]D-Fender | *8 is the default. This also depends on your setting pickupgroup for your devices |
| 15:42.49 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
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| 16:05.26 | Ashutto | how can i have a call stack trace for my dialplan in asterisk console? |
| 16:08.35 | [TK]D-Fender | Ashutto: Can you clarify that request... |
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| 16:20.04 | *** join/#asterisk webczat (webczat@193.200.46.16) |
| 16:20.12 | webczat | hmm |
| 16:23.34 | *** join/#asterisk Azrael808 (~peter@dandlgreen.demon.co.uk) |
| 16:24.36 | webczat | the subscribecontext parameter means that I can subscribe to receive notifications only for those extensions that are in the context? |
| 16:25.13 | Ashutto | [TK]D-Fender, in my console, if i call my asterisk, i'd like to see the dialplan execution plan (like "Dial(xxxx)", "Hangup()") in the calling order |
| 16:32.31 | [TK]D-Fender | Ashutto: Then go connect to the * CLI and "core set verbose 10" |
| 16:41.06 | Ashutto | thanks ^_^ |
| 16:49.43 | *** join/#asterisk picard276 (~chatzilla@74.60.166.135) |
| 16:49.49 | picard276 | hey guys im having an issue with a2billing |
| 16:49.58 | picard276 | im not sure how to get it to pass to a2billing? |
| 16:50.10 | picard276 | i have my incoming DID and i have that set to a custon Extension: a2billing,s,1 |
| 16:50.21 | picard276 | but it looks like this http://pastie.org/3231635 |
| 16:50.24 | picard276 | when a call comes in? |
| 16:52.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:52.56 | *** join/#asterisk Azrael808 (~peter@dandlgreen.demon.co.uk) |
| 16:53.03 | Ashutto | [TK]D-Fender, it only says "Using SIP RTP CoS mark 5" |
| 16:55.22 | picard276 | what? |
| 16:57.34 | [TK]D-Fender | Ashutto: If you set the verbose level and you don't see anything then that is because it isn't even executing dialplan because a call (if any) is refused |
| 16:57.49 | [TK]D-Fender | Ashutto: At which point you should enable SIP DEBUG and look at the actual attempt |
| 16:57.55 | [TK]D-Fender | Ashutto: "sip set debug on" |
| 17:00.08 | picard276 | anyone have any ideas |
| 17:04.14 | [TK]D-Fender | I'd start by proving the target exists |
| 17:20.17 | *** join/#asterisk Vicksters (~Vicks@modemcable142.35-177-173.mc.videotron.ca) |
| 17:28.03 | *** join/#asterisk phunteltek (~androirc@cpe-184-59-139-30.neo.res.rr.com) |
| 17:29.08 | phunteltek | Hey guys it's good to be here |
| 17:31.50 | phunteltek | Can anyone recommend an app for android? |
| 17:33.47 | phunteltek | Can you would recommend a voice over ip application for android o s tablet |
| 17:35.49 | phunteltek | <PROTECTED> |
| 17:36.09 | ikevin | sipdroid |
| 17:37.06 | phunteltek | <PROTECTED> |
| 17:37.37 | phunteltek | Are there any issues with firewalls on the android tablet? |
| 17:37.50 | phunteltek | <PROTECTED> |
| 17:38.02 | ikevin | using 3g or wifi? |
| 17:38.09 | phunteltek | <PROTECTED> |
| 17:38.27 | ikevin | via wifi no firewall/nat problem on android |
| 17:38.49 | ikevin | (else if you have a firewall on your wifi equipment) |
| 17:39.34 | ikevin | by default android accept all incomings/outgoings connexion |
| 17:39.47 | phunteltek | <PROTECTED> |
| 17:40.08 | phunteltek | <PROTECTED> |
| 17:40.25 | phunteltek | <PROTECTED> |
| 17:40.26 | ikevin | i you use public wifi you maybe need a thing like vpn to receive calls |
| 17:40.34 | phunteltek | <PROTECTED> |
| 17:40.56 | ikevin | or a stun server |
| 17:41.08 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:41.33 | phunteltek | Do you typically need to root your android device? |
| 17:41.43 | ikevin | no |
| 17:42.20 | ikevin | stun server can be setup on sipdroid, and vpn can be setup via android |
| 17:42.39 | ikevin | (see settings > network > vpns) |
| 17:42.51 | phunteltek | <PROTECTED> |
| 17:43.01 | ikevin | = |
| 17:43.16 | ikevin | stun server can be more easy to find and configure |
| 17:43.58 | phunteltek | I would like to use vpn to get through my work network. They have an open wifi network there but I don't want them watching the traffic |
| 17:44.27 | ikevin | ok, maybe openvpn can be a good & simple solution |
| 17:44.37 | ikevin | so, vpn take a lot of battery |
| 17:45.00 | [TK]D-Fender | So far GV & Skype on your phone have nothing to do with your SIP agent on your phone vs your server. |
| 17:45.01 | phunteltek | <PROTECTED> |
| 17:45.39 | [TK]D-Fender | phunteltek: Right now the first thing is : Why should we trust your server environment is even sane, let alone all proper? |
| 17:45.43 | p3nguin | Use an encryption method. |
| 17:46.15 | [TK]D-Fender | phunteltek: Start further back and reall start looking at what is coming in and what you setup to allow it in. |
| 17:46.27 | p3nguin | If VoIP over WiFi is bad, encrypted VoIP over WiFi must be horrid. |
| 17:46.59 | [TK]D-Fender | p3nguin: Encryped should not be any worse. Encapsulated perhaps.... |
| 17:47.02 | phunteltek | I haven't had any problems with voice over wifi |
| 17:49.04 | phunteltek | <PROTECTED> |
| 17:49.29 | phunteltek | <PROTECTED> |
| 17:49.55 | phunteltek | <PROTECTED> |
| 17:52.15 | phunteltek | I have an android tablet with no phone or 3g |
| 17:53.11 | phunteltek | <PROTECTED> |
| 17:54.37 | phunteltek | My asterisk box is behind a firewall though |
| 17:55.02 | p3nguin | As it should be. |
| 17:56.35 | phunteltek | I use to get phone calls from a fake extension from ip in bulgaria |
| 17:56.47 | phunteltek | <PROTECTED> |
| 18:09.07 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 18:15.53 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:18.57 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:40.59 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 18:47.07 | *** join/#asterisk grkblood (~grkblood@c-71-226-163-238.hsd1.sc.comcast.net) |
| 18:47.34 | grkblood | my inbound calls are pretty choppy, the caller can hear me fine but im having trouble hearing them clearly, heres an example: http://files.joeshowradio.com/samples/inbound_sample.mp3 |
| 18:47.52 | grkblood | does anyone know what might be causing this? |
| 19:03.03 | p3nguin | poor downstream bandwidth, excessive downloads from PCs, no QoS on the network |
| 19:17.56 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 19:25.42 | grkblood | p3nguin, my speedtest results are 21.53Mpbs down/4.14 Mbps up/49 ms and im not downloading anything |
| 19:31.17 | ketas | packet loss? |
| 19:34.33 | grkblood | nope, it looks pretty solid |
| 19:35.15 | ketas | who's provider? |
| 19:35.57 | p3nguin | Who is a provider? |
| 19:36.45 | p3nguin | Oh, maybe you are asking who is the provider? |
| 19:36.57 | ketas | wtf? |
| 19:37.20 | grkblood | vitelity |
| 19:37.36 | grkblood | its prolly them, i need an east coast provider |
| 19:37.53 | Vicksters | anyone here has some experience with DAHDI? |
| 19:37.54 | ketas | is this some kind of asymmetric connection? |
| 19:38.38 | grkblood | callcentric is the only reputable provider i know of on the east coast, anyone in the southeast worth checking out? |
| 19:38.46 | p3nguin | "my speedtest results are 21.53Mpbs down/4.14 Mbps up" |
| 19:39.02 | p3nguin | Seems pretty asymmetric to me. |
| 19:39.19 | p3nguin | Call Centric isn't really reputable. |
| 19:40.05 | grkblood | whos worth checking out on the east coast then? |
| 19:40.05 | p3nguin | Vitelity should be fine on the east as long as you use a PoP in the east instead of one on Denver. |
| 19:40.24 | grkblood | p3nguin, thats there only PoP right now |
| 19:40.30 | grkblood | ive asked them about that |
| 19:40.58 | grkblood | they plan on adding more, but currently there only PoP is out of denver |
| 19:41.03 | grkblood | their* |
| 19:41.48 | p3nguin | Check out VoIP.ms, then. They have one in Atlanta and one in Tampa. |
| 19:43.03 | p3nguin | I'm using them nearly exclusively. I rarely have any issues with the service. |
| 19:44.04 | grkblood | ill check them out, hopefully they port toll free numbers |
| 19:45.41 | p3nguin | Toll-Free number $25 Per number 2 to 4 Weeks * 800/855/866/877/888 |
| 19:45.45 | p3nguin | Yep, they do. |
| 19:49.04 | grkblood | where do you see that they port numbers? im not seeign anythign about that in there faq |
| 19:50.27 | p3nguin | I'm logged in. I clicked on DID Portability. |
| 19:50.42 | p3nguin | Look in their wiki. |
| 19:51.41 | p3nguin | http://wiki.voip.ms/article/Porting_a_Number#Numbers_portable |
| 19:52.03 | p3nguin | They show a screenshot of what I just copied/pasted. |
| 20:35.47 | ChannelZ | Vicksters: For analog, yes |
| 20:37.19 | Vicksters | I can't get it working. Everything looks to be fine and detected except I have no dial tone when I look at dahdi_monitor |
| 20:38.34 | Vicksters | I've put the relevant configs on pastebin if you'd care to take a peek I could PM the links to you? |
| 20:44.56 | ChannelZ | yeah |
| 20:47.50 | ChannelZ | So when you Dial() you're trying to monitor with dahdi_monitor and get nothing? |
| 20:48.02 | Vicksters | Yes. |
| 20:48.08 | Vicksters | Nothing at all. |
| 20:48.56 | ChannelZ | What country? |
| 20:49.00 | Vicksters | canada |
| 20:49.51 | ChannelZ | hmm.. I assume they use kewlstart but I have no idea |
| 20:50.10 | Vicksters | its the same as the US |
| 20:50.43 | ChannelZ | What happens if you call in rather than out? Does it detect that OK? |
| 20:51.39 | Vicksters | nah it doesnt either |
| 20:52.18 | Vicksters | its like the card is totally dead but ive tried two of them and its the same result |
| 20:52.22 | ChannelZ | hmm |
| 20:53.48 | ChannelZ | and "dahdi show status" in the console shows the channel? |
| 20:53.55 | Vicksters | yeah. |
| 20:54.04 | Vicksters | theres 0 and 1 |
| 20:54.31 | Vicksters | er. thats show channels. status just shows that channel. |
| 20:55.57 | ChannelZ | Hmm. Not sure what to tell you. Perhaps the card is not right in the head... you're sure the line it's hooked up to/the cable works? :) |
| 20:56.42 | Vicksters | yeah ive tried the line with an actual phone and it works as advertised. i came here as a last resort. :/ |
| 21:04.34 | ChannelZ | Sorry not sure what else to suggest. I don't have any experience with the x100p but it seems there are a number of clones, some probably more problematic than others |
| 21:06.03 | Vicksters | yeah thats for sure. this thing is a piece of crap. :) |
| 21:06.30 | Vicksters | thanks for your help anyway. im just puzzled as to why it doesnt work. this card used to be fine in another box with basically the same setup. |
| 21:09.56 | *** join/#asterisk Dovid (~dovid@213.8.121.90) |
| 21:11.18 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 21:11.52 | *** join/#asterisk dandate2 (~dan@222.127.52.58) |
| 21:12.11 | Dovid | Hi all |
| 21:12.20 | dandate2 | i set my pap2-t voipgateway + router to bridge mode and now I cannot connect to its GUI, is this normal? |
| 21:15.10 | *** join/#asterisk chasing`Sol (~cS@197.132.55.173) |
| 21:34.46 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 21:42.56 | *** join/#asterisk mahlon (mahlon@martini.nu) |
| 21:58.22 | *** join/#asterisk s[X] (~s_x_@eth589.qld.adsl.internode.on.net) |
| 21:58.55 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:38.31 | *** join/#asterisk TehRabbitt (~TehRabbit@unaffiliated/tehrabbitt) |
| 22:38.54 | TehRabbitt | hey, my router died on me, and I had to replace it, ever since, now my asterisk setup refuses to work X_X any ideas? |
| 22:39.02 | TehRabbitt | I can dial out, but there's no audio, so i'm thinking NAT issue |
| 22:39.12 | TehRabbitt | but I have the ports fowared properly so i'm not sure what to check next |
| 22:41.10 | TehRabbitt | I have ports 10000-20000 fowarded as well as 5060-5061 |
| 22:43.31 | TehRabbitt | any ideas? |
| 22:44.39 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 22:45.26 | [TK]D-Fender | Idea : show us |
| 22:45.35 | [TK]D-Fender | Your forwarding, the failed calls with SIP debug. Everything |
| 22:46.51 | TehRabbitt | ok one sec |
| 22:48.09 | ChannelZ | Ugh. I can still smell the stink of the Comcast guy's BO who was in my office yesterday. |
| 22:48.22 | TehRabbitt | ChannelZ: tht sucks X_X |
| 22:49.25 | TehRabbitt | hm this is really weird, now it is randomly working.... i'm convinced NAT is voodoo shit |
| 22:49.26 | TehRabbitt | lol |
| 23:15.50 | *** join/#asterisk min3r (1000@173-81-252-114-pkvl.atw.dyn.suddenlink.net) |
| 23:25.09 | *** join/#asterisk dandate2 (~dan@222.127.170.179) |
| 23:25.40 | dandate2 | when I call out from my pap2 there is about an 8 second delay to connect the call, however when i call out from a softphone on the same network its instant? |
| 23:26.49 | Vicksters | dandate2: whats your dialplan like in the pap2? |
| 23:27.48 | [TK]D-Fender | Delay-ful |
| 23:28.15 | dandate2 | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 23:28.25 | Vicksters | and what number are you trying to call? |
| 23:28.44 | dandate2 | either an internal extention or through an outbound route is same delay |
| 23:29.36 | [TK]D-Fender | And I doubt either of those parrterns match what you've just shown us |
| 23:30.08 | dandate2 | the dial plan? |
| 23:32.41 | Vicksters | you need to modify your dialplan to match the numbers you try to call. here's a tutorial from a quick google search: http://www.netphonedirectory.com/pap2_dialplan.htm |
| 23:33.59 | dandate2 | hmm the factory default dial plan doesnt cut it eh |
| 23:34.04 | Vicksters | nope! |
| 23:34.19 | Vicksters | i mean you need to customize it to your specific situation |
| 23:34.51 | Vicksters | otherwise it just cannot know what your environment is like.. |
| 23:40.32 | eZz | hm, is there a way to record a full call flow (including ringing tones) using MixMonitor ? |
| 23:41.01 | eZz | sounds like MixMonitor is starting only when a channel is being bridged |
| 23:43.13 | *** join/#asterisk cstachris (~chrismylo@202.182.147.116) |
| 23:43.35 | [TK]D-Fender | eZz: answer the call first |
| 23:43.53 | eZz | I don't need to answer |
| 23:45.12 | eZz | I need a schema like this: 1) MixMonitor(/some/file), 2) Dial(Tech/Num ...), h,1,StopMixMonitor() |
| 23:46.17 | [TK]D-Fender | If you want the ringing you will have to ANSWER first |
| 23:46.36 | [TK]D-Fender | Otherwise it could still be held OOB and you'll get nothing |
| 23:47.16 | [TK]D-Fender | And there is no need to stop MixMonitor in "h". It should die automatically |
| 23:47.37 | dandate2 | what is the dial plan that softphones use? |
| 23:47.41 | p3nguin | none |
| 23:47.43 | eZz | If I will answer - then how can I will have a fact than the user was answered ? |
| 23:47.59 | eZz | than=that |
| 23:50.28 | [TK]D-Fender | Thy wording that again... |
| 23:50.31 | [TK]D-Fender | try* |
| 23:50.48 | [TK]D-Fender | And be clean about which call you are referring to. |
| 23:50.48 | eZz | btw I just tried answer 1-st, then dial |
| 23:50.53 | [TK]D-Fender | clear* |
| 23:50.54 | [TK]D-Fender | gah |
| 23:50.59 | eZz | the result wan't changed |
| 23:51.07 | [TK]D-Fender | Dial with "r" |
| 23:51.28 | [TK]D-Fender | that should gurantee it |
| 23:52.21 | eZz | I don't need a fake ringing |
| 23:54.04 | eZz | dang, even with 'r' the same result |
| 23:56.23 | p3nguin | You would execute MixMonitor, then Answer, then Dial. MixMonitor doesn't record when the line is not up. |
| 23:56.49 | eZz | exten => _X.,n,Answer() |
| 23:56.49 | eZz | exten => _X.,n,MixMonitor(/tmp/call-${STRFTIME(${EPOCH},,%Y-%m-%d)}-${ACTIVENUM}-${RAND(100,999)}.wav) |
| 23:56.52 | eZz | exten => _X.,n,Dial(SIP/9529,180) |
| 23:56.55 | eZz | what's wrong? |
| 23:57.19 | eZz | hm, I see |
| 23:58.23 | p3nguin | Of course if you Answer, then execute MixMonitor, it would begin recording right away because the line is up. |
| 23:58.47 | eZz | in this case I have channel up and extension (for user's call answer) is launching right after dial |