| 00:03.23 | exothermc | pabelanger: also when I call them they are busy. trying to figure out why. |
| 00:05.12 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-222-230.mia.bellsouth.net) |
| 00:05.23 | pfn | do people still use tdm400p's? I should get rid of mine |
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| 00:15.33 | danj1980 | Hello |
| 00:16.13 | danj1980 | Quick questions... can you use the extensions.conf file together with realtime? |
| 00:16.34 | xheliox | yes |
| 00:17.51 | danj1980 | even in the same context? |
| 00:18.49 | xheliox | I'm not certain, but I don't believe so. |
| 00:19.14 | xheliox | I can tell you in 3 seconds though. |
| 00:20.51 | danj1980 | ok thanks. |
| 00:22.13 | xheliox | YEs. |
| 00:22.34 | xheliox | I just added an extension in extensions.ael to a context I also have in extensions.conf |
| 00:22.37 | xheliox | and it merged them. |
| 00:22.44 | xheliox | that's something I always wondered too. ;) |
| 00:22.53 | danj1980 | thanks for trying it out |
| 00:22.56 | xheliox | I don't do very much with ael though, so it never really mattered. |
| 00:25.33 | danj1980 | and if it works with ael, it will work in the same way with realtime? |
| 00:26.45 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 00:26.46 | xheliox | presumably, couldn't say and can't test. |
| 00:27.03 | xheliox | my experience with realtime extensions has always been horrid++++. |
| 00:27.20 | xheliox | maybe they've gotten the kinks worked out, but it was bad. |
| 00:29.49 | danj1980 | thanks for your advice |
| 00:30.31 | xheliox | no problem. |
| 00:31.19 | xheliox | and even if the antenna is ok, a meter will tell you if it's interacting with something |
| 00:40.42 | pabelanger | exothermc: What DAHDI hardware? |
| 00:41.19 | xheliox | hey danj1980 |
| 00:54.10 | [TK]D-Fender | [20:17]<danj1980>even in the same context? <- sort of. Make a container context and include one linked to realtime, and the other static |
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| 01:01.38 | johnf | I'm getting an exit non-zero when using follow me, any pointers on how to debug this? |
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| 01:08.00 | danj1980 | thanks [TK]D-Fender |
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| 01:11.02 | sky1975 | Can somebody tell me where can I get a supprot for ooh323 |
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| 02:26.57 | drmessano | ooh323? |
| 02:27.15 | drmessano | Well, you get a Delorean |
| 02:27.25 | drmessano | A big ass roll of cable |
| 02:27.58 | drmessano | Find a nice, tall building near the town square |
| 02:28.50 | drmessano | Order a flux capacitor from Mouser. Needs to be rated to handle at least 1.21 Jigawatts |
| 02:30.16 | drmessano | Install said flux capacitor in Delorean. I would recommend arc welds over nut/bolt/lock washers |
| 02:30.39 | drmessano | Find a pic of your family, for reference. Tuck it in the pocket of your vest |
| 02:31.15 | drmessano | Point said Delorean to intersect cable at the moment you've achieved the following: |
| 02:31.19 | drmessano | * 88 miles per hour |
| 02:31.32 | drmessano | * Lightning strike |
| 02:32.25 | drmessano | I guess I left out the bit about how to rig the cable, but that's probably commented in the source |
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| 02:45.24 | variable_office | is there a way to tell via the api whether a given channel is on hold. or what the current active call is for a given sip peer? |
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| 02:46.44 | docelmo | Say anyone got a sec to help me figure out a FXO problem with a Verizon phone line? |
| 02:46.59 | docelmo | I have did everything I can think of now I am just plum out of ideas |
| 02:47.30 | xheliox | What's up? |
| 02:47.57 | docelmo | me? |
| 02:48.08 | xheliox | yeah. what's the problem? |
| 02:48.47 | docelmo | TDM400P.. Has worked fine forever.. Just updated software and now my FXO ports are not responding |
| 02:49.00 | xheliox | what did you update? |
| 02:49.12 | docelmo | When I call it just rings asterisk doesnt see it. When I try to dial out of the port it says its not available.. |
| 02:49.16 | docelmo | Asterisk/Dahdi |
| 02:49.45 | docelmo | Just FYI not a novice.. Just suck at configuring boards |
| 02:49.48 | xheliox | were you using dahdi previously or did you upgrade from zaptel? |
| 02:49.57 | sky1975 | Anybody to give me a help on ooh323 |
| 02:50.11 | docelmo | dahdi -> dahdi |
| 02:50.32 | xheliox | dahdi show status -- does that display anything? |
| 02:51.18 | docelmo | Yes.. |
| 02:51.30 | docelmo | Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 02:51.31 | xheliox | does it show the card? |
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| 02:52.47 | docelmo | Any thoughts? |
| 02:53.07 | xheliox | and dahdi show channels ? |
| 02:53.38 | docelmo | shows business as usual |
| 02:53.46 | pabelanger | docelmo: What version of DAHDI did you updated too? |
| 02:54.05 | docelmo | 2.2.1.1 |
| 02:54.22 | docelmo | Well thats what asterisk shows anyhow |
| 02:55.03 | pabelanger | And you restarted DAHDI after you installed the software? IE: /etc/init.d/dahdi restart |
| 02:55.14 | pabelanger | and then asterisk |
| 02:55.21 | docelmo | yep.. even went as far as to reboot the box |
| 02:56.17 | pabelanger | docelmo: any reason you didn't upgrade to DAHDI 2.3.0? |
| 02:56.30 | docelmo | RPM's Im lazy |
| 02:56.55 | pabelanger | Either way, have to bail. Try rolling back to previous version of DAHDI |
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| 08:36.25 | ruben23 | hi guys.. |
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| 09:13.42 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 09:13.42 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.8 (2010/06/01), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.32 (2010/06/01), *-Addons 1.6.1.4, 1.6.0.6 (2010/06/08), 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.2 (2010/06/08) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bug |
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| 09:32.36 | d4rkstar | ?book |
| 09:32.56 | d4rkstar | ~books |
| 09:33.04 | Blackvel | oh my fault. didnt escape \ ()/ signs |
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| 09:36.35 | nalbaxx | ~books |
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| 09:52.05 | ChannelZ | ~book |
| 09:52.06 | 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 |
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| 10:31.59 | [sr] | hola |
| 10:32.02 | [sr] | :) |
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| 10:34.06 | guza | hi, can someone help me, i can not find IVR configuration menu in asterisknow GUI |
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| 11:31.31 | gavimobile | folks, after I press *77 to record so I can sucessfully hear the record , but after that how do I save it.. it doesn't appear in the default folder /var/spool/asterisk/tmp or /var/lib/asterisk/sounds/tmp |
| 11:31.52 | gavimobile | does it save automatically? what I noticed is that it says in the docs pres *99 to hear therecording... which that doesn't work either. what I do is I press # then I hear a message asking me if I want to hear again by pressing 1 or rerecord by pressing 2 |
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| 11:52.41 | pabelanger | gavimobile: #freepbx |
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| 11:57.24 | gavimobile | askedthere already |
| 11:57.27 | gavimobile | zzZZzz |
| 11:57.30 | gavimobile | sleeping |
| 11:57.57 | pabelanger | gavimobile: Then wait, this channel is not for FreePBX support |
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| 13:15.32 | [sr] | hi [TK]D-Fender |
| 13:15.34 | [sr] | :) |
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| 13:53.01 | Dovid | j #asterisk-il |
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| 14:03.16 | [sr] | i'm sad |
| 14:06.08 | Dovid | boohoooo |
| 14:06.37 | [sr] | heeh |
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| 14:08.02 | Lantizia | Hey does anyone know of any guides for converting the releases .tar.gz in to debian packages? (not from SVN to debian) |
| 14:08.55 | Lantizia | I basically have my own things I like to put in menu select... but I've several phone systems to manage... and APT would be a good way to keep them updated |
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| 14:20.23 | maldous | hi. quick question. my sip.conf has a single [general] register=> that works. i only want asterisk to answer a call to the registered client. what goes in extensions.conf to answer everything/anything? |
| 14:20.58 | maldous | asterisk is reporting Call from '' to extension 'p123456789' rejected because extension not found. |
| 14:21.17 | maldous | (where p123456789 is my client) |
| 14:21.48 | maldous | exten => _p123456789,1,Playback(welcome-message) |
| 14:21.56 | maldous | no workee. any ideas? |
| 14:30.13 | pabelanger | Lantizia: http://svn.debian.org/viewsvn/pkg-voip/asterisk/ |
| 14:30.53 | pabelanger | maldous: check your context |
| 14:31.02 | maldous | do'h. left out '[default] |
| 14:31.06 | Lantizia | pabelanger, didn't I just say not SVN |
| 14:31.12 | maldous | thx |
| 14:31.32 | pabelanger | Lantizia: ok then... #debian |
| 14:32.38 | pabelanger | Lantizia: And if you bothered to look at the link I posted, it was the source code for how debian creates the asterisk .deb files. Just modify then for your needs |
| 14:32.51 | pabelanger | s/then/them |
| 14:35.21 | Lantizia | pabelanger, it's completely useless I've already seen it |
| 14:36.06 | pabelanger | Lantizia: Then you need to read how Debian packaging works |
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| 14:36.06 | pabelanger | http://www.debian.org/doc/maint-guide/ |
| 14:41.06 | maldous | ok, new question. ;) - my asterisk is just a sip client now. i want dial plan to answer and dial another number. answer works, dialplan has exten => _p.,n,Dial(SIP/p123456789); - asterisk says No such host: p123456789\n Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 14:41.12 | maldous | where do i create a SIP channel? |
| 14:42.12 | pabelanger | maldous: sip.conf |
| 14:43.01 | pabelanger | maldous: Where are you expecting the call to actual go? |
| 14:43.12 | pabelanger | s/actual/actually |
| 14:43.32 | maldous | p123456789 |
| 14:43.56 | maldous | my incoming extension is of that format - i'm in the dark here. |
| 14:44.11 | pabelanger | maldous: Yes, but what is p123456789? |
| 14:44.30 | maldous | an example phone number |
| 14:44.47 | maldous | (123) 456-789 |
| 14:45.11 | maldous | i could be completely wrong here |
| 14:45.28 | pabelanger | maldous: Yes, but for what? What is that number associated to? What is the path the call will take from you asterisk box to the far end user? |
| 14:45.55 | pabelanger | You need an interface to actually send calls over |
| 14:46.16 | pabelanger | IE: a SIP provider, Analog line, Digital (T1 / ISDN) |
| 14:46.22 | maldous | sip provider. |
| 14:46.36 | pabelanger | maldous: then setup a context in our sip.conf for it |
| 14:46.44 | maldous | yep. about to do that. ;) |
| 14:46.53 | pabelanger | IE: [my_sip_providor] |
| 14:47.18 | pabelanger | then you can do Dial(SIP/my_sip_providor/p123456789) |
| 14:47.23 | maldous | ah! |
| 14:47.26 | pabelanger | ~book |
| 14:47.27 | 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 |
| 14:47.29 | maldous | thx. will try. |
| 14:47.31 | pabelanger | maldous: ^^^ |
| 14:47.34 | maldous | :) |
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| 14:48.35 | cusco | hi... |
| 14:52.01 | [sr] | entao cusco? :P |
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| 15:10.05 | cusco | oi [sr],q uem és? |
| 15:10.49 | [sr] | cusco: lol ninguem que conheças...so tava a meter-me ctg |
| 15:10.55 | cusco | :-) |
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| 15:11.05 | cusco | usas asterisk? |
| 15:11.55 | [sr] | tem k ser em en |
| 15:12.09 | [sr] | cusco: still a begginer.. |
| 15:13.21 | tzafrir_laptop | [sr], hi |
| 15:13.55 | [sr] | tzafrir_laptop: hi |
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| 15:18.37 | maldous | ok, i'm stuck. |
| 15:18.49 | maldous | i've made a working call and capture the sip INVITE. |
| 15:19.02 | maldous | http://pastebin.com/eAEmqDmy has what worked - what should go into sip.conf to "emulate" that? |
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| 15:23.48 | maldous | aah. dummy defaults. let's try. |
| 15:28.15 | maldous | ah. my From: line has "Anonymous" |
| 15:28.22 | maldous | anyone know the config that defines this? |
| 15:28.39 | maldous | fromuser and defaultuser don't do it. |
| 15:29.32 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:39.51 | darg_ | if I have my ATA behind a firewall, can I still tell * to use reinvite to tell it to connect directly to a remote end? |
| 15:40.04 | darg_ | if not, is there something I can adjust in the firewall (nat) so that I can? |
| 15:42.24 | maldous | dang. Failed to authenticate on INVITE to '"Anonymous" <sip:username@hostname>;' |
| 15:50.02 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 15:54.32 | tzafrir_laptop | darg_, nat = yes ; in sip.conf? |
| 16:00.48 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 16:04.36 | maldous | anyone know how to change Anonymous? |
| 16:07.05 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 16:07.27 | *** join/#asterisk Tim_Toady (~moi@77.49.107.115.dsl.dyn.forthnet.gr) |
| 16:10.26 | maldous | my digest username is wrong. how, oh how, do i change this.. |
| 16:14.29 | *** join/#asterisk BANSAL (~bansal@117.199.118.104) |
| 16:21.50 | maldous | blah. i can dial out with x-lite to broadworks, but not asterisk to broadworks. |
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| 16:47.04 | pabelanger | maldous: then you have a configuration issue with your asterisk box |
| 16:47.48 | maldous | hmm |
| 16:48.01 | maldous | would unsupported codecs result in "SIP/2.0 401 Unauthorized" ? |
| 16:48.53 | pabelanger | maldous: Would have to see a SIP debug trace for the call. |
| 16:48.57 | pabelanger | ~collectdebug |
| 16:48.59 | infobot | methinks collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 16:49.34 | pabelanger | use pb to paste the log |
| 16:49.50 | maldous | yeah, yeah. :) |
| 16:50.33 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 16:51.50 | maldous | tomorrow. 2:51am here. tired. too. |
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| 18:18.06 | xheliox | Cherries ftw. |
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| 18:45.37 | docelmo | Anyone have any thoughts on why a TDM400P w/ FXO ports would not sense an incoming call or also allow outbound calls via that channel? |
| 18:48.21 | *** join/#asterisk MiserySoft (~lnd@217.155.155.134) |
| 18:48.40 | Mhaddog_Mac | have you revise your inbound routes? and trunks? |
| 18:50.32 | *** part/#asterisk mintux (~mrg@unaffiliated/mintux) |
| 18:51.11 | docelmo | Mhaddog_Mac: me? |
| 18:51.27 | Mhaddog_Mac | yeap... |
| 18:52.40 | docelmo | I cant even get asterisk to kick up and show a call coming in on the channel |
| 18:52.44 | *** join/#asterisk Alagar (~Administr@122.164.36.63) |
| 18:53.21 | docelmo | Routing calls via the dialplan is the last of my worries right now.. Im just trying to get calls to come in and out |
| 18:53.30 | *** join/#asterisk Alagar (~Administr@122.164.36.63) |
| 18:53.37 | Mhaddog_Mac | is the card being recognize on dahdi? have you do dhadi show channels? |
| 18:53.49 | Mhaddog_Mac | is the card being loaded up? |
| 18:54.02 | docelmo | yep.. everything is 100% |
| 18:54.08 | Mhaddog_Mac | I have not mentioned dial plan... |
| 18:54.32 | Mhaddog_Mac | I mentioned trunks and routes, that they are setup the right way to allow you to place a call.... |
| 18:55.04 | docelmo | Right now yes.. I am trying to push a call out and nothing happens on the channel. When I call it it just rings.. Nothing more |
| 18:59.35 | Mhaddog_Mac | you sure the telco line is fine? |
| 18:59.50 | Mhaddog_Mac | and when you place the call u see the channel become active |
| 19:10.27 | docelmo | When I place the call no.. It bombs.. Yes the line is good from the CO. I plugged my phone into it. |
| 19:10.46 | docelmo | <PROTECTED> |
| 19:10.58 | docelmo | thats what I get when I use ANY of my FXO channels |
| 19:15.12 | cusco | docelmo: what happens when tying to dial inboud? |
| 19:15.25 | cusco | does asterisk say anything? |
| 19:19.39 | pfn | so, does meetme still require dahdi devices anymore? |
| 19:26.03 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-67-231.home.otenet.gr) |
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| 19:46.31 | cusco | pfn: you can use pseudo |
| 19:55.31 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 19:59.28 | ChannelZ | yawns |
| 20:08.19 | pfn | cusco, that's the usb fake driver isn't it? |
| 20:09.25 | WIMPy | pfn: You can also use ConfBridge instead of MeetMe. That works without dahdi. |
| 20:10.18 | pfn | oh, is confbridge the "replacement" for meetme? |
| 20:10.46 | ChannelZ | hmm.. is dnsmgr in a module or part of the core? |
| 20:11.18 | WIMPy | pfn: I think so |
| 20:11.25 | WIMPy | ChannelZ: Module |
| 20:12.36 | ChannelZ | which? I can't figure out what seems relevant |
| 20:12.44 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-67-231.home.otenet.gr) |
| 20:13.06 | [sr] | WIMPy: LCR's working great :) TE mode configured and understood, only missing some time to test NT mode |
| 20:14.15 | WIMPy | ChannelZ: Hmm, maybe I'm actually wrong. I don't see any module like dns, but I thought I reloaded it some time when I had trouble with it. |
| 20:14.56 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 20:14.58 | WIMPy | [sr]: Great. But except for the interface configuration there is no difference in usage. |
| 20:15.06 | puzzled | hi |
| 20:15.25 | ChannelZ | I just grepped the source tree and it seems to be in main/ :( |
| 20:15.35 | [sr] | WIMPy: i see, for NT mode i just have to use a ISDN crossover type cable, correct? |
| 20:15.58 | WIMPy | [sr]: yes |
| 20:16.44 | [sr] | WIMPy: one thing came to my mind now, the Hour/Date information that my ISDN phones has on the display, this will work when connected to the card? or its just a feature from the conventional PBX? |
| 20:16.49 | WIMPy | prefers crossover adaptors however. Different cables can be irritationg. |
| 20:16.51 | WIMPy | -o |
| 20:17.51 | WIMPy | LCR will transmit date/time. If you have a Siemens phone, you will have to configure Siemens bug compatibility however. |
| 20:18.09 | [sr] | damn! yes i do have siemens phone's |
| 20:18.21 | [sr] | the PBX now is a siemens HIcom 100 |
| 20:19.53 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:19.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:19.55 | [sr] | WIMPy: where do i find info about that siemsns bug bypass/workarround? |
| 20:20.03 | WIMPy | Unfortunaletly that's configured per extension and not per port. |
| 20:21.03 | WIMPy | Parameter "seconds". |
| 20:22.50 | *** join/#asterisk Mango (~iMango@d154-20-89-230.bchsia.telus.net) |
| 20:23.29 | Mango | USD$5 by PayPal for anyone who can help me make Outbound MWI Subscriptions work! |
| 20:23.45 | Mango | http://forums.digium.com/viewtopic.php?f=1&t=74300 |
| 20:25.52 | [sr] | WIMPy: was trying to find on the pdf docs from lcr and dont see it |
| 20:25.58 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 20:27.53 | drmessano | $5? |
| 20:28.11 | ChannelZ | minimum wage |
| 20:28.25 | Mango | not if it takes five minutes |
| 20:28.27 | [sr] | $5 its not enought even for a dinner here :P |
| 20:28.43 | Mango | ok, what's it worth to you? :) |
| 20:28.53 | ChannelZ | well if it took 5 minutes you'd have probably figured it out yourself by now |
| 20:29.01 | drmessano | $37.50 an hour, rounded to the first hour... it just cost you $5 for me to open the link |
| 20:29.14 | Mango | sorry |
| 20:29.18 | Mango | you have a no fix no charge policy |
| 20:29.19 | Mango | :D |
| 20:29.49 | Mango | ChannelZ: Perhaps you're smarter than me. I'm willing to pay to be humiliated ;) |
| 20:29.51 | ChannelZ | needs breakfast/lunch. hmm.... |
| 20:30.08 | *** join/#asterisk x303 (~x303@188.255.100.97.cfl.res.rr.com) |
| 20:30.31 | ChannelZ | Mango: No, but just looking at your description my guess is it's going to take more work in digging through the source, as opposed to just some configuration option set wrong |
| 20:30.34 | drmessano | You've already humilitated yourself by offering $5 for a bounty.. the additional heckling is no charge |
| 20:31.06 | Mango | ChannelZ: No chance I missed something blindingly obvious? |
| 20:31.16 | drmessano | Yeah, you left out a zero |
| 20:31.23 | drmessano | $50 <-- or maybe two |
| 20:31.24 | Mango | no I didn't |
| 20:31.28 | Mango | I was offering $0 before :D |
| 20:32.31 | drmessano | You do realize it's almost insulting to everyone else here to offer $5 to fix a problem, as if that's all their time or skill is worth, right? |
| 20:32.35 | drmessano | Just sayin' |
| 20:34.28 | Mango | I was hoping someone would say, "You left out a semicolon...that'll be $5 please." I'm sorry if I offended anyone; I didn't intend to. Thank you all for looking at the link. |
| 20:34.59 | pfn | ugh, why is asterisk not responding to SIP 407 |
| 20:35.07 | pfn | it says auth required, then asterisk doesn't even bother sending a response |
| 20:39.47 | ChannelZ | Mango: only thing I can think of is it's something to do with the realm maybe, that its not matching the right peer to auth or something. I have no idea |
| 20:40.03 | ChannelZ | or its simply broken |
| 20:40.06 | Mango | Okay, I'll keep digging. Thanks for the pointer. |
| 20:40.11 | Mango | That's possible too :P |
| 20:40.22 | WIMPy | [sr]: No idea, where I could have found it, if not in the pdf. |
| 20:44.08 | [sr] | WIMPy: ok i'll search later, one thing, the card has a power jack, just like IDE disks for example, this doesn't need to be connected, or does it? the card can give power to the phone in NT mode withuot that |
| 20:46.10 | ChannelZ | It does for FXS ports |
| 20:46.27 | ChannelZ | (IE if you have FXS modules for analog phones connected to the card) |
| 20:46.59 | [sr] | ahh |
| 20:47.05 | [sr] | i see, makes sense |
| 20:47.16 | [sr] | to power the ports |
| 20:47.31 | [sr] | but for the NT mode, the port doesn't need that when we talk ISDN? |
| 20:47.32 | WIMPy | [sr]: Don't know about the openvox (it was openvox, wasn't it?). It's probably to supply power on NT ports, but it may be able to forward power from the TE to the NT port as well, if you have power |
| 20:47.37 | WIMPy | on the line. |
| 20:47.38 | ChannelZ | yeah. I think if you've got FXO only it's not necessary but don't quote me on that |
| 20:48.07 | ChannelZ | for what you've got I have no idea |
| 20:48.27 | [sr] | ChannelZ: think not, i have one machine with FXO cards and that is not connected, in fact i don't remember if it has this power jack |
| 20:49.46 | [sr] | WIMPy: yap openvox, i'll try NT mode without connecting this to see what happens |
| 20:50.46 | florz | [sr]: for providing power, you'll probably have to connect power - but depending on the device you may not need a powered bus at all |
| 20:51.01 | [sr] | WIMPy: one cool thing is that i have a few ISDN modems that used to connect to the net, that can be used for TE mode :) and they are really cheap |
| 20:51.47 | [sr] | florz: i see, i'll do the test :) |
| 20:51.51 | WIMPy | [sr]: Probably most people do, if they only need one or two ports. |
| 20:53.33 | [sr] | WIMPy: i'm going to collect all this old ISDN modems to work with LCR... it's cheaper than the VOIP cards!! |
| 20:53.47 | drmessano | I was told by someone, [TK]D-Fender maybe, that the cards should be powered no matter what.. that bus power isn't sufficient |
| 20:54.21 | drmessano | I had questioned the need if I had no FXS ports, but seems that's irrelevant |
| 20:58.01 | [sr] | hum |
| 20:58.12 | [sr] | one interesting thing, my 4 FXO port card |
| 20:58.39 | *** join/#asterisk exothermi (~exothermc@64.185.113.197) |
| 20:58.52 | [sr] | hum forget |
| 20:58.59 | [sr] | i was going to say nonsense |
| 21:01.04 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:03.01 | exothermi | Hi I'm trying to bring up 2 PRIs, using the dahdi driver. I get "PRI span 1/0: Provisioned, Down, Active", and when I start asterisk cat /proc/dahdi/1 reports the status of all channels as " (In use) " Where as before I start asterisk that isn't present. |
| 21:03.39 | exothermi | The other thing is CRC4 error count: 90432 which doesn't seem correct. |
| 21:03.41 | WIMPy | In use by Asterisk. So far so good. |
| 21:04.02 | exothermi | WIMPy: ahh that is what that means? |
| 21:04.39 | exothermi | WIMPy: ok so how about: "PRI span 1/0: Provisioned, Down, Active" from pri show spans ? |
| 21:04.54 | exothermi | The "Down" part doesn't look good. |
| 21:05.34 | *** join/#asterisk x303 (~x303@97.100.255.188) |
| 21:05.34 | WIMPy | Indeed, but I'm not that familiar with dahdi messages and heir meanings/causes. |
| 21:06.53 | [sr] | WIMPy: now that i know LCR, i'd love to have a PRI line!!! |
| 21:08.06 | WIMPy | [sr]: PRI with LCR will only work with HFC-E1 cards, AFAIK. |
| 21:08.13 | [sr] | WIMPy: just curious, how much cost a BRI line (2 channels) where you are? |
| 21:08.18 | pfn | ah, needed the defaultuser option |
| 21:08.40 | *** join/#asterisk Sedorox (brandon@smartserv/cna/Sedorox) |
| 21:09.13 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:09.39 | WIMPy | [sr]: Starts at about 24 EUR/Mon I think. |
| 21:10.02 | [sr] | WIMPy: hum i see, i pay 35â¬+VAT month for each :S |
| 21:10.38 | [sr] | but i'm going to move to another operator that charges 16⬠per each BRI line, with national limited call's |
| 21:10.44 | [sr] | unlimited i mean |
| 21:12.06 | [sr] | exothermc: my FXO card, with dahdi alsi show's "in use" when they are not in a call |
| 21:15.37 | exothermi | [sr]: Ok so that is normal. |
| 21:16.01 | exothermi | but pri show spans still show the span being down. |
| 21:16.42 | [sr] | well that dont know |
| 21:17.26 | [sr] | one think i noticed with LCR on my BRI lines, is that when they don't show the UP state in the LCR monitor tool, they came imediattly UP when i call on of the numbers, may be the same |
| 21:17.59 | *** join/#asterisk x303 (~x303@97.100.255.188) |
| 21:18.41 | exothermi | So ya to me LCR means Least Cost Routing. What are you referring to? |
| 21:19.00 | [sr] | nah |
| 21:19.23 | [sr] | its an interface like dahdi |
| 21:19.28 | [sr] | but different |
| 21:19.32 | WIMPy | No, PRIs are always active, Energi saving is only done on BRIs. |
| 21:19.56 | [sr] | WIMPy: cool, that is energy saving, leaned one more |
| 21:19.56 | WIMPy | Linux Call Router |
| 21:20.44 | WIMPy | That's the idea, but I don't know it it really saves any measurable amount of energy. |
| 21:21.31 | [sr] | it may be irrelevant..i think, to put the line up when a call arrives it has to have energy also |
| 21:22.20 | *** part/#asterisk pfn (pfnguyen@socal.hanhuy.com) |
| 21:23.41 | exothermi | actually reading the openvox manual, and There maybe some hardware switches I can make. |
| 21:26.45 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-67-231.home.otenet.gr) |
| 21:28.46 | [sr] | WIMPy: ah one thing, this HFC-4S could have 4 ports, but it's really a openvox model with only 2 ports, it has the place on the card... but the chip's the 2 other ports are missing |
| 21:30.32 | WIMPy | So it's upgradable :-) |
| 21:30.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 21:31.37 | [sr] | that |
| 21:35.07 | cusco | hi... |
| 21:35.31 | cusco | pri show spans is not showing on asterisk cli |
| 21:36.02 | cusco | only pri intense debug |
| 21:36.04 | cusco | why? |
| 21:39.31 | *** join/#asterisk danj1980 (~dan@91.108.17.42) |
| 21:39.44 | danj1980 | Hi all, I have a problem with realtime SIP.conf |
| 21:40.19 | danj1980 | I've added a SIP account into realtime db (and I have some others listed in the sip.conf file). |
| 21:40.42 | danj1980 | When the sip account makes a call, qualify lists it as REACHABLE. |
| 21:40.55 | danj1980 | However, a few seconds after the call has finished, it switches to UNREACHABLE |
| 21:41.12 | danj1980 | The SIP account listed in the sip.conf file are working without any problems. |
| 21:41.27 | danj1980 | Can anyone point me in the right direction? |
| 21:41.53 | cusco | danj1980: in your db there should beasetting called "qualify" |
| 21:41.56 | cusco | set it to yes |
| 21:42.29 | danj1980 | I have qualify=yes in the general section of sip.conf |
| 21:42.37 | danj1980 | Should that not cover it? |
| 21:43.06 | danj1980 | Im getting this error now: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'sipuser_201' |
| 21:46.18 | danj1980 | ok, enabled rtcachefriends. and it works now. |
| 21:46.21 | danj1980 | thanks |
| 21:49.45 | [sr] | WIMPy: which trademark do you advice for SIP desk phones, with and without PoE |
| 21:51.15 | WIMPy | From the ones I have tried I'm only convinced by the Snom 360/370. 320 is probably ok if you don't need a big display. |
| 21:55.44 | *** join/#asterisk cdahmedeh (~cdahmedeh@92.62.171.129) |
| 21:56.22 | [sr] | hum, the problem will be to find them here in PT |
| 21:57.10 | [sr] | hum they here distributers here |
| 21:57.19 | [sr] | going to check the phones how they look :) |
| 21:58.10 | [sr] | they have both PoE and non-PoE? |
| 21:58.51 | WIMPy | POE or 5V PSU. |
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| 22:13.09 | *** join/#asterisk lwizardl (~James@c-68-60-86-92.hsd1.mi.comcast.net) |
| 22:13.22 | lwizardl | hi |
| 22:14.27 | ChannelZ | OHHAI! |
| 22:14.35 | lwizardl | i'm looking to get my sip details from my magicjack so that i can use it in an asterisk pbx can someone help with that ? |
| 22:20.09 | [sr] | WIMPy: nice |
| 22:20.40 | [sr] | WIMPy: one thing in asterisk, the phone list per SIP account, it's not centralized in asterisk, is it? |
| 22:21.05 | ChannelZ | http://lmgtfy.com/?q=asterisk+magicjack |
| 22:21.08 | [sr] | at least the softphone i use, 3CX, it's created locally |
| 22:21.09 | *** join/#asterisk tehrabbitt (~tehrabbit@c-71-59-82-2.hsd1.nj.comcast.net) |
| 22:21.42 | [sr] | ChannelZ: cool site |
| 22:22.20 | ChannelZ | :) |
| 22:23.34 | lwizardl | ChannelZ, yes but issue with what google finds doesn't work anymore since the newer updates. all those pages are from 2008-2009 |
| 22:26.52 | [sr] | ChannelZ: do you know if the address book, can be centralized in asterisk? |
| 22:26.57 | [sr] | per SIP account |
| 22:27.29 | ChannelZ | What 'address book?' |
| 22:27.50 | *** join/#asterisk Mark22 (~mark@unaffiliated/mark21) |
| 22:28.02 | tehrabbitt | hey guys, not sure if you remember me or not.. ChannelZ might... Anyway, interesting thing I found with SIP... I was having all those issues behind my NAT router on comcast... and mysteriously about 2 weeks later it just *works* without me touching anything... so my thinking is comcast is blocking SIP traffic or at least shaping it |
| 22:28.58 | lwizardl | tehrabbitt, I wouldn't put anything past comcast. Comcast does what they please when ever they want |
| 22:29.25 | ChannelZ | is on Comcast business |
| 22:29.28 | Mark22 | Hello, is it possible (and if yes where should I look) to do something like the following: press your customer number and end with a # (the customer number is at least 2 digits, but could also be 6 digits, the first number for the customernumber is never a 0 (cleaning it from entered 0's isn't needed)) |
| 22:30.02 | [sr] | ChannelZ: how could i say, each SIP user can't create it's own addressbook, and that be centralized in asterisk? or doesnt work that way? for example, if i connect from differents phones |
| 22:30.19 | Mango | Mark22: Sounds like you want to create an IVR. |
| 22:30.20 | WIMPy | [sr]: I use the minibrowser for a central directory. But some phones (like e.g. the Snom) also support LDAP. |
| 22:31.02 | [sr] | WIMPy: i get it, no way that, that info be saved by SIP user |
| 22:31.06 | Mark22 | Mango: I know I need something like an IVR, I did even look at http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu (but I couldn't find a solution for ending it with a # so it could be multiple numbers long) |
| 22:31.18 | Mango | Mark22: http://www.voip-info.org/wiki/view/Asterisk+tips+IVR+menu might be a good place |
| 22:31.24 | Mango | ...to get you started :P looks like you already found it. |
| 22:32.08 | WIMPy | [sr]: SIP won't provide directy syncronisation or so, no. |
| 22:32.23 | [TK]D-Fender | Mark22: Or you could match a variable length # with a more open patterna and chk the validity after the fact |
| 22:32.27 | [sr] | WIMPy: i see, it could something interesting for future.. |
| 22:32.44 | [TK]D-Fender | Mark22: If you want to terminate with a "#' then, that's what Read() is for, vs an IVR |
| 22:33.28 | Mark22 | we use a xml file for directories (a single file per client, a client can have multiple phones), we provide the files using a tftp connection (also used to configure their phones, as I don't like using the web interface if I need to configure 10 phones at 1 day) |
| 22:34.16 | *** join/#asterisk tehrabbitt (~tehrabbit@c-71-59-82-2.hsd1.nj.comcast.net) |
| 22:34.24 | *** join/#asterisk QaDeS (~mklaus@p4FC72352.dip0.t-ipconnect.de) |
| 22:35.31 | *** join/#asterisk alancio (~Alancio@190.206.158.176) |
| 22:36.05 | alancio | hi people, how can I know from what iax channel a call is coming from? |
| 22:36.26 | Mark22 | [TK]D-Fender: thank you, I should be able to make something with Read() |
| 22:36.27 | [TK]D-Fender | alancio: Look at the channel name itself |
| 22:36.30 | WIMPy | alancio: Put them in different contexts. |
| 22:36.52 | alancio | WIMPy: thats what I'm trying to avoid |
| 22:37.04 | alancio | [TK]D-Fender: I tried that, and I get the incorrect channel name! |
| 22:37.48 | ChannelZ | incorrect how |
| 22:37.50 | alancio | [TK]D-Fender: instead of getting the right one, I always get the last defined peer |
| 22:38.02 | Mark22 | alancio: in some cases you could also use Include => some context |
| 22:38.31 | alancio | Mark22: yes, I'm using that |
| 22:38.50 | ChannelZ | what var are you looking at? |
| 22:38.51 | alancio | I think the reason why I get the incorrect channel name is because I'm using templates when defining the iax peers |
| 22:39.02 | [sr] | well going to sleep |
| 22:39.03 | [TK]D-Fender | alancio: From what you're saying it's matching the WRONG peer. Don't expect * to tell you ANOTHER one that that. |
| 22:39.15 | [sr] | stay ok folks |
| 22:39.32 | alancio | I'm printing ${CHANNEL} |
| 22:39.34 | [TK]D-Fender | alancio: fix your peers |
| 22:39.57 | alancio | if I use CHANNEL(peername) I don't get anything |
| 22:40.41 | Mark22 | what asterisk version do you use? |
| 22:40.44 | alancio | [TK]D-Fender: fix them how? should I not use templates? |
| 22:40.57 | alancio | I'm using asterisk 1.4.31 |
| 22:41.12 | [TK]D-Fender | alancio: How do you think we can advise at this point? You haven't shown us anything. |
| 22:41.26 | alancio | [TK]D-Fender: ok, I'll post something, hold a sec |
| 22:42.20 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 22:45.21 | alancio | http://pastebin.com/4YWTNAyG |
| 22:46.12 | [TK]D-Fender | alancio: Why did you creat 5 identical looking peers? |
| 22:46.43 | alancio | for ease of management |
| 22:46.43 | [TK]D-Fender | alancio: And I don't see any DEBUG for the incoming call |
| 22:47.15 | alancio | DEBUG? what do you mean? |
| 22:47.20 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 22:47.28 | [TK]D-Fender | alancio: I don't see the IAX DEBUG dump of the incoming call. |
| 22:47.49 | alancio | oh sorry, I only copied one line, let me put the whole thing |
| 22:51.44 | alancio | http://pastebin.com/JmatyK7u |
| 22:53.09 | [TK]D-Fender | alancio: Fill in proper "username" into each. |
| 22:53.50 | alancio | ok, I'll try that |
| 22:56.31 | alancio | YES! That did it! you are a genious |
| 22:56.38 | alancio | genius |
| 22:56.43 | alancio | thanks a lot |
| 22:57.05 | alancio | I'm using the channel name to add the appropriate prefix to the caller id |
| 22:57.24 | alancio | I think its more elegant than having different contexts, and repeat everything |
| 22:57.29 | *** join/#asterisk fofware (~fabian@190.7.25.160) |
| 22:58.35 | alancio | thanks |
| 23:02.53 | ChannelZ | Wait() will break if a channel hangs up yes? |
| 23:03.15 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:07.02 | Mark22 | ChannelZ: what would it break? the call or something else? |
| 23:08.22 | ChannelZ | well I mean asterisk won't just hang if I use a crazy Wait value |
| 23:08.48 | ChannelZ | Basically I want to 'ignore' calls from certain people, not even Answer() so I don't get billed for the call :) |
| 23:09.21 | ChannelZ | But if I just NoOp or something and drop through the end of the dialplan, it returns a BUSY to my ITSP which then in turn sends me an alert email |
| 23:12.03 | ChannelZ | looks like it works |
| 23:16.12 | *** join/#asterisk fuziontech (~fuziontec@c-71-59-3-118.hsd1.ga.comcast.net) |
| 23:23.33 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:24.05 | Mark22 | some action based on the caller id would be nice, i need to look for that (there is an option, I should be able to do it with some AGI scripting or System() I think) |
| 23:24.14 | Mark22 | do you need to pay for incoming calls? |
| 23:26.40 | ChannelZ | I'm just doing AGI, looking up CID numbers in a database to figure out what to do with specific calls |
| 23:27.03 | ChannelZ | And yes I pay for incoming calls. It's not much but for all these telemarketers I'd rather not even waste the penny |
| 23:31.23 | Mark22 | I am also thinking about a database to look it up to change the caller id (for outbound calls) or the name for a number (inbound calls) |
| 23:32.02 | ChannelZ | yeah I do that too for incoming, a lot of people's CNAM's show up as "MOBILE PHONE" or a city name or something else worthless |
| 23:36.06 | Mark22 | I just want to do it for the names I know (source: customer address books :P) |
| 23:37.54 | ChannelZ | right thats all I do. It's hooked up to my customer/client contact database |
| 23:38.41 | ChannelZ | like a few people call from their company so their normal caller ID name shows up as 'Company X' or whatever, but they have their own direct DID so I convert it into their name |
| 23:39.20 | *** join/#asterisk tehrabbitt (~tehrabbit@c-71-59-82-2.hsd1.pa.comcast.net) |
| 23:40.47 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:44.53 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 23:45.51 | joako | How can I set what codecs are used for a SIP channel outside of sip.conf? |
| 23:48.01 | [TK]D-Fender | joako: huh? |
| 23:48.34 | joako | [TK]D-Fender, My SIP provider allows g729 but does not announce it. So if I put disallow=all, allow=g729 I can use g279 |
| 23:48.51 | joako | but if I put disallow=all, allow=g729, ulaw it will never use g729 |
| 23:50.00 | [TK]D-Fender | joako: show the debug |
| 23:50.17 | [TK]D-Fender | joako: And remember it is a quesiotn of whose PREFERENCE is used |
| 23:50.26 | lwizardl | so anyone here know much about getting SIP details from a magicjack ? all the stuff online is no longer working on the newest updates. I have the password but not my username and proxy |
| 23:50.35 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 23:51.12 | joako | Hmm, today it is saying Capabilities: us - 0x1 (g723), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1 (g723) |
| 23:51.20 | joako | but before it was peer ulaw only |
| 23:52.49 | *** join/#asterisk krahe (~krahe@203-109-246-60.static.bliink.ihug.co.nz) |
| 23:53.31 | [TK]D-Fender | <PROTECTED> |
| 23:53.51 | [TK]D-Fender | Looking like someone SNAFU'd |
| 23:56.01 | joako | Either way I need to use 1 codec for voice calls and 1 codec for fax calls. How would I do that with the same sip peer? |
| 23:57.08 | krahe | hi, I am having problems to setup asterisk with mysql, it complains about pgsql, I am not sure if I can run without postgresql, I wonder if someone here can give me some help to setup asterisk |
| 23:57.14 | krahe | thanks |