| 00:03.10 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 00:04.10 | sbingner | should this be decent? -- echo ratio = 0.0763 (850.3 / 11145.0) |
| 00:05.20 | sbingner | it says I should get it below 5% but I have no idea how to get it lower lol |
| 00:05.24 | sbingner | used the patched fxotune |
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| 00:18.47 | rbd | say I have a sip phone on extension 1234 in my dial plan. At extension 111, I have an agi script that runs and ends up streaming an mp3 to the caller ...could I use the asterisk manager API Originate call to connect the two so that the SIP phone will ring, and when the user picks up, he or she will hear that streamed audio? |
| 00:20.50 | CtRiX | rbd as i have already said: |
| 00:20.54 | CtRiX | !google "asterisk outgoing spool voip" |
| 00:21.14 | CtRiX | http://www.voip-info.org/wiki-Asterisk+auto-dial+out <---- |
| 00:21.54 | rbd | CtRiX: looking at that. looks good, but the manager API may be best in this case, as I would be doing this from a remote box (i.e. not the asterisk server) |
| 00:22.32 | rbd | and that way I wouldn't have to map the drive over... |
| 00:31.31 | rbd | CtRiX: okay tyanks for the help. It looks like this is doable if I use the 'Local' channel type for the origination call |
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| 00:36.55 | boon1 | are there any known issues with "make" on Freebsd 6.1 and 1.4.0-beta2? |
| 00:37.55 | boon1 | configure is fine, i then make and get the following; http://pastebin.ca/187360 |
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| 00:48.12 | netsurfer | lo ppl... anyone find that astman.Engine causes problems in the new AJAM demo in 1.4.0(b2) ? |
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| 01:09.43 | jtexter3 | Is there any easy way to determine the source of an alarm? I have a tie-line scenario with some Dialogic equipment, and periodically, the lines slip into alarm, and then back out. What I want to determine is if the alarm is starting on the Digium side, or the Dialogic side? |
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| 02:08.47 | richipoo | Need help using res_config_ldap.c ... currently http://svn.digium.com/svn/asterisk/team/group/res_config_ldap asterisk_with_ldap won't compile |
| 02:09.11 | richipoo | There a working version or snapshot? |
| 02:13.08 | richipoo | Error I get while compiling svn asterisk_with_ldap is "res_config_ldap.c:952: error: ‘STANDARD_HANGUP_LOCALUSERS’ undeclared (first use in this function)" |
| 02:16.00 | kratzers | missing some dependancy? |
| 02:16.10 | kratzers | dependency* |
| 02:29.02 | jtexter3 | richipoo: Is that the exact output? If so, looks like maybe the file is corrupted, maybe needing to call dos2unix on the file? |
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| 02:38.57 | RR_tX | need some help - I was working on an old box - 1.0.9 and had purchased some g729 licenses but didnt realize unti afterwards that digium only lists 1.2 and 1.4 -- Does anyone know if g729 is supported on 1.0.9? - thanks for your help |
| 02:42.25 | jtexter3 | 1.0.9 is pretty old, doubt you'll have much luck there |
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| 02:42.46 | jtexter3 | any reason to not upgrade to 1.2? |
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| 02:44.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:44.25 | RR_tX | they have a fairly complex setup and had some issues with their FXS ports -- was just going to bring the call in SIP and be done with it. If i have to upgrade I would need to run a parrallel system to ensure interopaility with old scripts |
| 02:45.02 | RR_tX | I mean fxo |
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| 02:55.22 | HaMYaI | hi, are there changes I have to make to the configuration files when upgrading from 1.2.12.1 to 1.4.0-beta? |
| 03:00.12 | hads | ./UPGRADE.txt |
| 03:05.15 | HaMYaI | hads: thanks, never knew that |
| 03:08.39 | HaMYaI | hads: I did "find ./ -name UPGRADE.txt -print" from installation dir but it reveals nothing |
| 03:09.26 | HaMYaI | not in zaptel, libpri either |
| 03:18.18 | De_Mon | HaMYaI where did you get that source from? |
| 03:26.29 | HaMYaI | De_Mon: I just found it, thanks |
| 03:27.12 | HaMYaI | De_Mon: forgot to untar the file =) |
| 03:47.24 | De_Mon | yeah, that usually helps! happy upgrading |
| 03:52.17 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 03:54.20 | gambolputty | Hi. Can * automatically answer a SIP call with text in the URL? (user@1.1.1.1) I want to have a context do something with the text. |
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| 04:14.23 | AJaymn | hey guys ive been using A@H for awhile so i somewhat understand the workings of the conf files. so im renting a rack server. Is there an easy precompiled ver of Asterisk for Fedora? |
| 04:16.31 | tuck3r | asterisk takes like 2 seconds to compile |
| 04:17.22 | AJaymn | well its being a pain in the apples! |
| 04:17.35 | tuck3r | uh, why? |
| 04:17.44 | tuck3r | make; make install |
| 04:17.49 | AJaymn | lol |
| 04:18.42 | AJaymn | is there a command to uninstall prev ver of Asterisk? |
| 04:18.43 | tuck3r | AJaymn: A tip for getting help on IRC, if your having problems compiling don't ask for a pre-compiled version just ask for help compiling ;) |
| 04:18.56 | tuck3r | no need it just overwrites |
| 04:19.00 | *** join/#asterisk KeNroM (i=Kayrom@72.22.146.245) |
| 04:19.09 | tuck3r | if you getting errors pastebin it |
| 04:19.16 | KeNroM | ok |
| 04:19.21 | tuck3r | ~pastebin |
| 04:19.22 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 04:19.41 | KeNroM | i got a wild question....... can a ip phone use with Asterisk |
| 04:20.09 | tuck3r | uh, duh |
| 04:20.21 | KeNroM | how? |
| 04:20.29 | tuck3r | ~book |
| 04:20.30 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:20.31 | KeNroM | can u lead me in the right direction |
| 04:20.44 | tuck3r | ~voip-info |
| 04:20.45 | jbot | methinks voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 04:20.49 | KeNroM | i`m evening willing to pay for consulating |
| 04:21.36 | tuck3r | read that book or pay a consultant |
| 04:23.29 | KeNroM | ok |
| 04:23.30 | KeNroM | kool |
| 04:24.20 | tuck3r | if you don't want to invest lots of time you should just pay somebody, asterisk is not the easiest thing to learn |
| 04:27.57 | KeNroM | ok |
| 04:28.27 | KeNroM | i have a server with vicidial on it... |
| 04:28.42 | KeNroM | but i want to create custom scripts |
| 04:29.15 | tuck3r | read the book |
| 04:29.36 | KeNroM | i bought the book |
| 04:29.42 | tuck3r | ooh |
| 04:29.55 | KeNroM | it doesnt really cover that area in depth |
| 04:31.39 | tuck3r | that area? |
| 04:31.50 | KeNroM | custom scripts |
| 04:31.57 | tuck3r | agi? |
| 04:32.03 | KeNroM | php |
| 04:32.10 | tuck3r | do you know php? |
| 04:32.33 | KeNroM | to a certain level |
| 04:32.51 | KeNroM | not that good thou... |
| 04:32.53 | tuck3r | you use the phpagi library i think |
| 04:33.29 | KeNroM | if i found book wid a example i can quickly learn to build custoim scripts |
| 04:33.38 | KeNroM | no |
| 04:33.50 | tuck3r | you don't? |
| 04:34.36 | KeNroM | not sure:( |
| 04:35.06 | tuck3r | http://phpagi.sourceforge.net/ |
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| 04:36.13 | KeNroM | ok |
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| 04:36.18 | KeNroM | and what that does? |
| 04:36.24 | tuck3r | php class |
| 04:37.03 | rene1 | hello, i wonder if 1.4 queues support setinterfacevar=yes when configured in realtime mode |
| 04:37.35 | tuck3r | did you try it? |
| 04:43.04 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 04:43.43 | AJaymn | Anyone use Asterisk on Fedora? |
| 04:43.59 | tuck3r | lots of people do |
| 04:44.11 | tuck3r | you never posted the log of your compile... |
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| 05:06.18 | imdabest | hello |
| 05:07.11 | imdabest | i need help can any one help |
| 05:07.21 | tuck3r | just ask |
| 05:07.28 | imdabest | oh thanks |
| 05:07.55 | imdabest | i am using ambeint md3200 card with asterisk caller id is not coming |
| 05:08.30 | tuck3r | i'm no help |
| 05:08.43 | imdabest | any one else? |
| 05:11.20 | tuck3r | i never even heard of that card |
| 05:16.27 | rene1 | imdabest: why bother, but if you must try at least with an intel clone. it is you best bet |
| 05:20.36 | imdabest | i can get callerid when i use other country in zaptel it works fine but when i use some other country it doesnt |
| 05:20.46 | kram | hrm, anyone here know brian mcmanus? |
| 05:21.37 | imdabest | like when loadzone= us works fine but when i change it to other country caller id doesnt apear can any one help |
| 05:21.57 | benjk | imdabest, which country are you int? |
| 05:22.01 | benjk | in |
| 05:22.28 | imdabest | Pakistan |
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| 05:22.38 | webman | hmmm, current svn 1.2.x branch won't compile for me "emacs.c: In function `em_kill_line':emacs.c:172: internal compiler error: Segmentation fault" using debian stable |
| 05:22.47 | benjk | I don't think there is any profile for PK |
| 05:22.55 | webman | and it fails in a different spot each time |
| 05:23.19 | imdabest | well there is no profile but i loaded in zonedata.c and recompile my zaptel |
| 05:23.20 | benjk | why don't you just use loadzone=us |
| 05:23.39 | imdabest | well i need to use local line + internation |
| 05:23.47 | imdabest | international i mean |
| 05:24.01 | benjk | the callerid schemes are different between countries |
| 05:24.14 | imdabest | http://bugs.digium.com/view.php?id=7012 |
| 05:24.44 | imdabest | this is the post from where i got the zoneinfo for Pakistan |
| 05:24.52 | benjk | you will need to make sure that your changes for PK will also take care of the PK callerid, whatever caller ID method is used in PK |
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| 05:26.01 | webman | anyone else had problems compiling the svn 1.2.x version of asterisk on debian stable??? |
| 05:26.42 | tuck3r | nobody uses debian stable :P |
| 05:27.55 | rene1 | imdabest: what do you mean? you can use loadzone=us and still mke a dialplan to do international & locla calls |
| 05:28.17 | benjk | he wants PK dial and indication tones |
| 05:29.11 | imdabest | yeah you are right benjk |
| 05:29.42 | benjk | does that US caller ID scheme work in PK? |
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| 06:03.23 | tzafrir_laptop | webman, what's the problem |
| 06:03.25 | tzafrir_laptop | ? |
| 06:04.18 | webman | tzafrir_laptop well, I think maybe it is just my machine, it compiles on my workstation, but not under on my test machine which is running under xen |
| 06:04.39 | webman | but now I am thinking it is a memory problem of some sort.... :( |
| 06:04.57 | tzafrir_laptop | that is the editline directory |
| 06:05.17 | tzafrir_laptop | strangely enough, there is a debian package for it |
| 06:05.29 | webman | maybe the editline part requires more memory than asterisk itself to compile or something..... ? |
| 06:05.38 | tzafrir_laptop | for starters: apt-get build-dep asterisk |
| 06:05.57 | tzafrir_laptop | this should give you an idea about packages you need |
| 06:06.04 | tzafrir_laptop | except libpri |
| 06:06.37 | tzafrir_laptop | that "emacs.c" is in the subdirectory editline |
| 06:06.54 | webman | well, I only need sip... I am actually trying to debut a problem with dropping calls on transfer when using sip :( |
| 06:08.00 | tzafrir_laptop | you may be able to skip vi.c , but not emacs.c , if you want the cli... |
| 06:08.25 | tzafrir_laptop | hmmm there should be a simple patch to use the debian editline package |
| 06:09.55 | webman | ouch, that build-dep wants 157 new packages !! incl X11 |
| 06:10.34 | x86 | i've got a SIP peer that can not register to my Asterisk 1.2.11 box... |
| 06:10.36 | tzafrir_laptop | Try applying it manuallt |
| 06:10.39 | tzafrir_laptop | http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/patches/sys_editline.dpatch?op=file&rev=0&sc=0 |
| 06:10.49 | x86 | all that sip debug gives me is SIP/2.0 401 Unauthorized |
| 06:10.53 | tzafrir_laptop | but editline is just about the first thing being built |
| 06:11.02 | x86 | I've verified the username, password, etc, but it still does not work.... |
| 06:11.11 | x86 | any ideas as to how i can fix this? |
| 06:11.15 | tzafrir_laptop | if you get a compiler errr there, something may be wrong with the compiler |
| 06:11.38 | tzafrir_laptop | X11 is from gtk, probably |
| 06:11.52 | tzafrir_laptop | which you don't need |
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| 06:31.32 | richipoo | Need help using res_config_ldap.c ... currently http://svn.digium.com/svn/asterisk/team/group/res_config_ldap asterisk_with_ldap won't compile |
| 06:31.39 | richipoo | There a working version or snapshot? |
| 06:31.42 | richipoo | Error I get while compiling svn asterisk_with_ldap is "res_config_ldap.c:952: error: ‘STANDARD_HANGUP_LOCALUSERS’ undeclared (first use in this function)" |
| 06:32.32 | x86 | ah figured it out |
| 06:33.01 | x86 | i have multiple external IPs bound to the asterisk box, and the IP the client was registering to is not the default IP of the machine |
| 06:33.27 | x86 | so client would send server a register request on to 1.2.3.4, and server would send back a reply from 1.2.3.5 |
| 06:33.56 | x86 | so i had to get creative with some SNAT rules in my server's iptables firewall to re-write the source IP address, and it works like a champ now :) |
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| 07:44.48 | richipoo | join #openldap |
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| 07:52.43 | carrar | Welcome to #openldap |
| 08:06.29 | richipoo | :-p |
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| 09:32.03 | kristalino | wich text-to-speech software can i use for french language ? |
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| 09:47.13 | tzafrir_laptop | festivsl? |
| 09:47.47 | kristalino | tzafrir_laptop, no french language available for it afaik |
| 09:48.15 | benjk | Cepstral has French text to speech |
| 09:49.08 | coppice | i think festival can do french with some separately maintained add-ons |
| 09:49.16 | robin_sz | just feed the text through a regex to remove the last letter from each word .. and there you go. |
| 09:49.22 | *** part/#asterisk richipoo (n=richip@S01060050ba83dd3f.cg.shawcable.net) |
| 09:49.29 | kristalino | coppice, really ? sounds interesting |
| 09:50.20 | benjk | festival can do anything because it is not a ready to use text to speech system, but it is a speech synthesis laboratory environment for research |
| 09:50.25 | kristalino | " English (British and American), Spanish and Welsh text to speech" |
| 09:50.40 | robin_sz | welsh? |
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| 09:50.54 | benjk | but it requires you to know about speech synthesis, you have to know what you are doing, then you can make it do just about anything |
| 09:51.57 | kristalino | did anyone here already achieved that ? |
| 09:52.22 | benjk | like I said its a lab environment for researchers to test new ideas |
| 09:52.48 | benjk | there will be plenty of French speech synthesis researchers who use festival for French TTS research |
| 09:53.29 | benjk | some of them will have published their work (incl their configs) but like I said, you have to know what you are doing |
| 09:53.33 | robin_sz | if they can make it do welsh, french should be EASY |
| 09:54.00 | benjk | you need speech synthesis skills, not unix skills |
| 09:54.16 | benjk | its an environment for scientists of that field |
| 09:54.17 | coppice | why would welsh be hard? |
| 09:54.17 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 09:54.20 | kristalino | no speech synthesis skills at all :( |
| 09:54.50 | benjk | in that case it is going to be difficult to teach it new tricks |
| 09:54.54 | kristalino | robin_sz, is welsh a difficult language ? |
| 09:54.57 | robin_sz | coppice, well, first youd have to find someone who a) spoke welsh and b) could use a computer. |
| 09:55.28 | benjk | you can always lure them with a bunch of sheep |
| 09:55.45 | robin_sz | the areas where welsh is spoken tend to be remote sheep farming bits. |
| 09:56.00 | kristalino | benjk, did you already succeded in learning festival another language it doesn't know by default ? |
| 09:56.09 | coppice | the languages which are supported reflect people's interests. sadly a massive amount of stuff has been done with festival, where the results have been disipated are are not available to anyone |
| 09:56.13 | benjk | no I gave up and use Cepstral |
| 09:56.26 | robin_sz | benjk, in england we have white and brown sheep. In wales they consider them to be blonde and brunette ;) |
| 09:56.33 | benjk | because I have no intention of becoming a speech synthesis scientist |
| 09:57.56 | benjk | there is another text to speech system that is based on a different approach, and its focus is on multi=lingual abilities |
| 09:58.43 | benjk | I don't recall now what it was called, but its based at a university in Belgium I think, so all the initial work was done in French |
| 10:01.08 | coppice | MBROLA |
| 10:02.18 | benjk | http://tcts.fpms.ac.be/synthesis/mbrola.html |
| 10:13.21 | benjk | I had a contractor working for me on several projects in the 90s, he was welsh |
| 10:13.44 | benjk | he was an incredible fast coder |
| 10:14.46 | benjk | one of his colleagues said, if you let Ian code it, you won;t need any code obfuscator |
| 10:14.57 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
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| 10:51.13 | edwar64896 | tzafrir - you on? |
| 10:59.30 | tzafrir_laptop | here |
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| 11:16.39 | RoyK | <PROTECTED> |
| 11:17.00 | RoyK | tzafrir: ping |
| 11:17.16 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:21.33 | tzafrir_laptop | pong |
| 11:24.48 | RoyK | did you get my email? |
| 11:25.10 | tzafrir_laptop | answered it |
| 11:25.32 | tzafrir_laptop | E1 is handled there |
| 11:26.15 | tzafrir_laptop | I only modified "whatever is not E1" |
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| 11:31.30 | RoyK | ok |
| 11:31.51 | RoyK | anyway - I don't have any BRI/T1 equipment around |
| 11:34.12 | webman | does anyone know a lot about sip and could look at bugid 8064 please? call drops on transfer |
| 11:39.07 | RoyK | webman: try rolling back to, say, 1.2.9 and see if the problem's there. if it's not, try 1.2.10 and so on to isolate the time when this was broken |
| 11:53.04 | edwar64896 | hey tzafrir. Am getting myself familiar with astribank - don't appear to be able to reboot my server and have astribank come up again normally if power remains connected to the astribank. |
| 11:53.37 | edwar64896 | is there any way I can 'reset' the unit on reboot so I can achieve a normal boot sequence? |
| 11:55.07 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
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| 11:58.38 | nassy | edwar64896: what is a channel bank if you dont mind me asking. i dont understand the explanation in my asterisk the future of telephony book |
| 12:00.05 | edwar64896 | nassy: ooh what a good question... |
| 12:00.08 | edwar64896 | I guess the answer is... |
| 12:00.23 | edwar64896 | a device which bundles a group of FXO or FXS devices into a single device |
| 12:00.31 | edwar64896 | and interfaces to a telephony processor. |
| 12:00.41 | edwar64896 | there are many classes of channelbank, |
| 12:01.06 | edwar64896 | various interfaces are T1/E1, PCI, ethernet or USB |
| 12:01.12 | edwar64896 | does this help? |
| 12:01.23 | nassy | yes |
| 12:01.32 | nassy | i didnt understand the interfaces part |
| 12:02.09 | edwar64896 | you have to interface the channelbank to a telephony proessor. If we are talking astrisk, you can get PCI channelbanks... |
| 12:02.16 | edwar64896 | (TDM2400P) |
| 12:02.24 | edwar64896 | Ethernet (SIP Gateway channelbank) |
| 12:02.28 | edwar64896 | or USB (Astribank) |
| 12:02.35 | nassy | the book has a T1 (PRI) connecting asterisk to the channel bank and then an FXx to the PSTN |
| 12:02.39 | edwar64896 | T1/E1 (Rhino) |
| 12:03.18 | edwar64896 | makes sense. Sounds like a Rhino or similar. |
| 12:03.29 | edwar64896 | That's the expensive way of doing it. ;-) |
| 12:03.49 | edwar64896 | T1/E1 interfaces are not cheap and motherboards aren't being shipped with that many PCI interfaces last time I looked. |
| 12:03.53 | nassy | doesnt the PRI have to go from the PSTN to asterisk |
| 12:04.05 | nassy | and not from asterisk to the channel bank |
| 12:04.05 | edwar64896 | If that's what you want to do. |
| 12:04.19 | edwar64896 | You can also use T1 to the channelbank if that is what the CB requires. |
| 12:04.29 | nassy | i thought the phone comany (CO) provided the PRI |
| 12:04.33 | edwar64896 | There's a difference between PRI and T1 |
| 12:04.47 | nassy | PRI is ISDN technology correct? |
| 12:04.47 | edwar64896 | PRI is more about the signalling interface. |
| 12:04.51 | edwar64896 | T! is the data interface. |
| 12:05.02 | edwar64896 | Yes PRI = Primary Rate ISDN |
| 12:05.19 | edwar64896 | but it is supplied over a T1 or E1 data interface. (G703) |
| 12:05.58 | RoyK | ops. new crash in chan_sip |
| 12:06.13 | nassy | ah, so with the appropiate PCI card asterisk can provide PRI signaling to the channel bank |
| 12:06.23 | edwar64896 | Nope. |
| 12:06.31 | edwar64896 | You don't need PRI signalling for a channelbank. |
| 12:06.48 | RoyK | nassy: yes, but you're probably better off with a multiport ATA |
| 12:06.49 | edwar64896 | you would use FXO signalling for an FXS channelbank, but the connection would be over T1 |
| 12:06.57 | nassy | but according to the diagram it is going from asterisk to the channel ban |
| 12:06.59 | RoyK | cheaper and far easier to setup |
| 12:07.10 | RoyK | nassy: yes, you can do that, but you do not want to |
| 12:07.31 | edwar64896 | nassy: how many FXS ports are you trying to use? |
| 12:07.49 | nassy | no i am just rying to understand the diagram in the book |
| 12:07.54 | edwar64896 | ok |
| 12:08.03 | edwar64896 | what page? |
| 12:08.05 | nassy | i thought only the phone company could provide PRI signaling |
| 12:08.09 | nassy | 24 |
| 12:08.22 | edwar64896 | Not at all. If you like, Asterisk can provide PRI signalling. |
| 12:08.29 | edwar64896 | this is how you might interface to a legacy PBX. |
| 12:08.40 | webman | nassy: no, zaptel can provide PRI signalling to another asterisk box, or another PBX etc... |
| 12:09.14 | nassy | ah ok, so in figure 2-2 n page 24 is that what they are illustrating |
| 12:10.00 | edwar64896 | Sure. |
| 12:10.06 | edwar64896 | this will be using FXO and FXS signalling. |
| 12:10.09 | nassy | and for me if i need to migrate from a toshiba strata ctx system in the office to an asteris system that is what i would use |
| 12:10.15 | edwar64896 | there is no PRI in this diagram |
| 12:10.47 | edwar64896 | don't know anything about the strata ctx, but if you are using PRI into the tosh then you will probably need a couple of ISDN spans and use PRI_NET signalling on one |
| 12:10.52 | nassy | the T1 in the diagram is not a PRI? |
| 12:10.56 | edwar64896 | nope |
| 12:11.11 | edwar64896 | just one of the spans on an ISDN card. |
| 12:11.40 | nassy | i dont understand what you mean by spans on ISDN card |
| 12:11.41 | edwar64896 | remember - difference between T1/E1 and PRI. they are not the same. |
| 12:11.57 | nassy | ok, the T1 is data |
| 12:11.58 | edwar64896 | check out the te405p on the digium site. This has 4 spans |
| 12:12.03 | edwar64896 | yes. T1 is data |
| 12:12.31 | edwar64896 | once you configure a span on the ISDN card for pri_cpe signalling, you will be connecting it to a PRI from the telco (or another asterisk box!) |
| 12:12.37 | nassy | so in that diagram they are not showing a T1 comming in from the provider to asterisk |
| 12:13.05 | edwar64896 | nope. you're right. they are not. |
| 12:13.20 | edwar64896 | this doesn't seem to show a telco connection. |
| 12:13.28 | edwar64896 | only a single FXO to a central office. |
| 12:13.39 | edwar64896 | not much for a head office connection, but there you are. |
| 12:13.46 | nassy | this is cool. i didnt know asterisk could provide all that stuff |
| 12:13.50 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 12:13.52 | edwar64896 | that and more. |
| 12:14.32 | nassy | i am ging to try to put in a proposal for asterisk to replace our toshiba system int he office (150 employees int he main office) |
| 12:14.47 | edwar64896 | where are you based? |
| 12:14.57 | nassy | there will also be a presentation to replace with an avaya system |
| 12:15.02 | edwar64896 | erk. |
| 12:15.02 | nassy | nyc, usa |
| 12:15.17 | edwar64896 | avaya are pretty agressive here in Aus |
| 12:15.34 | nassy | but i was hoping to gain about a year of experience before replacing |
| 12:15.55 | nassy | but we are moving to a new floor int he same building in jan |
| 12:15.59 | edwar64896 | one bit of advice - don't skimp on the interface cards and the phones that you use. |
| 12:16.03 | nassy | and CEO wants to replace then |
| 12:16.52 | nassy | avaya consutant is pretty good |
| 12:17.09 | nassy | and we may end up using that |
| 12:17.31 | edwar64896 | all ends up being the marketing and the relationship that the Avaya consultant gets to have with your CEO! |
| 12:17.40 | edwar64896 | its not about the tech! ;-) |
| 12:17.48 | nassy | its a good relationship |
| 12:17.59 | edwar64896 | always is. |
| 12:18.25 | nassy | i dont see avaya as having any technical advantages over asterisk |
| 12:18.29 | nassy | is that accurate |
| 12:19.00 | edwar64896 | that won't ever be the issue. |
| 12:19.11 | nassy | where i seem to lose out is the support |
| 12:19.17 | nassy | ie, when things go wrong |
| 12:19.35 | benjk | avaya doesn't deadlock |
| 12:19.49 | edwar64896 | Neither does my * box. |
| 12:20.04 | nassy | my angle is going to be though i can reinstall and be back up and running in an hour if we use trixbox or something until we becme more familar with it |
| 12:20.04 | benjk | then you were lucky |
| 12:20.24 | nassy | i see the 1.4 version of asterisk has some fixes for deadlocks |
| 12:20.35 | edwar64896 | 1.4 isn't usable yet. |
| 12:20.42 | benjk | some fixes isn't good enough |
| 12:20.50 | edwar64896 | just try and test it but don't rely on it for prod. |
| 12:21.03 | benjk | its a structural problem, requires major changes in the core |
| 12:21.10 | nassy | what would the symptoms of a deadlock be |
| 12:21.14 | webman | nassy: and how do you diagnose if the problem is your asterisk your telco or some weird bug in asterisk ??? a re-install won't fix a bug |
| 12:21.20 | benjk | everyting stops |
| 12:21.38 | benjk | because asterisk is waiting for itself |
| 12:21.42 | edwar64896 | what version are you using? |
| 12:22.25 | nassy | webman: same way if i use an avaya system. combo of consultants and troubleshooting on my end |
| 12:22.28 | *** join/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 12:22.49 | edwar64896 | you'll not be doing much that way. |
| 12:23.04 | edwar64896 | Standard behaviour of telco is to send guy out with |
| 12:23.05 | nassy | webman: my suspicion would be bugs would get fixed faster in asterisk |
| 12:23.09 | edwar64896 | a little PRI tester. |
| 12:23.18 | edwar64896 | S/He will tell you PRI is fine. |
| 12:23.22 | edwar64896 | It will then be up to you. |
| 12:23.35 | edwar64896 | You might want to consider the Business Enterprise version of * |
| 12:23.47 | nassy | yeah i looked at that |
| 12:23.48 | benjk | that gives you the worst of both worlds |
| 12:23.55 | edwar64896 | weally? |
| 12:24.02 | edwar64896 | why? |
| 12:24.03 | nassy | i sent in a request for a quote |
| 12:24.05 | benjk | you can't do anything yourslef cause you don't get sources |
| 12:24.21 | webman | nassy: I'm just suggesting you might be better off asking someone to install asterisk for you, and then you will have someone to call if there is a problem... as well as having the option to fix it yourself |
| 12:24.41 | benjk | the attraction of open source is so you can hire anybody you like to modify or fix things for you |
| 12:24.51 | benjk | you are not locked to a single vendor |
| 12:24.58 | benjk | with ABE you have vendor lock in |
| 12:25.10 | nassy | yeah, i am also checking out some consultants |
| 12:25.16 | webman | but no support if you modified it |
| 12:25.21 | benjk | people I have helped troubleshoot stuff in ABE claimed they didn't get sources |
| 12:26.15 | benjk | in any event, what's the point then if you lose support you may as well not pay for it |
| 12:26.17 | webman | benjk: could be right, I've never really looked at it a lot |
| 12:26.53 | webman | benjk: yeah, but for your normal pbx, would you really need to alter the source code anyway ?? |
| 12:27.15 | benjk | with asterisk you are more likely to requires changes, yes |
| 12:29.14 | *** join/#asterisk linux_newbie95 (n=ryan@cpe-72-226-110-116.nycap.res.rr.com) |
| 12:29.48 | linux_newbie95 | im having trouble installing asterisk |
| 12:31.23 | linux_newbie95 | can some one give me a complete rpm of asterisk |
| 12:31.59 | linux_newbie95 | hello |
| 12:32.45 | linux_newbie95 | help |
| 12:32.48 | linux_newbie95 | help |
| 12:33.32 | webman | royk: that sip bug seems to start in 1.2.9 as 1.2.8 works ok.... and 1.2.10, 1.2.11, 1.2.12 are all broken, and current svn is broken too |
| 12:34.04 | *** part/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:1) |
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| 12:51.24 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 12:55.05 | RoyK | webman: then report that to the bugtracker |
| 12:55.19 | *** join/#asterisk postel (n=jp@Wikimedia/Postel) |
| 12:55.40 | webman | royk: I did ... |
| 12:55.54 | RoyK | webman: if you can narrow it down further, closer to a svn rev, it'll speed up debugging even more |
| 12:57.57 | webman | royk: yeah.... is there some info on checking out a specific version of asterisk?? |
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| 13:04.01 | *** join/#asterisk lorinc (n=ang@caracas-5068.adsl.interware.hu) |
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| 13:13.24 | RoyK | webman: svn up -r revnum |
| 13:22.08 | webman | is there some way to know what svn version a particular 1.2.x release is? ie, I will need to test svn versions > 1.2.8 and < 1.2.9 but how to know which versions they are? |
| 13:23.14 | *** join/#asterisk conico (n=chatzill@88.224.254.41) |
| 13:23.39 | conico | hello |
| 13:25.30 | conico | i want to set up an asterisk server.and the clients wont be in my network.if i use iax2 will these clients's voice traffic will use my bandwidth and if i use sip not ? |
| 13:26.01 | webman | conico: the other way around... assuming the clients will only talk to each other |
| 13:26.15 | conico | ?? |
| 13:26.21 | conico | what you mean ? |
| 13:26.36 | benjk | the first 8-10 seconds the iax connection will go through the server |
| 13:29.11 | conico | so i need much bandwidth if i use iax2 than if i use sip ? |
| 13:29.25 | benjk | no |
| 13:29.34 | webman | conico: only if all your calls are less than 10 secs... |
| 13:29.37 | benjk | its only the first 8-10 seconds |
| 13:30.26 | RoyK | webman: back |
| 13:32.32 | conico | so if i use sip and these clients are all behind a nat device like as adsl modem/router and my asterisk server is on a public ip, should i forward all ports to the sip client on th adsl router ? |
| 13:32.34 | *** join/#asterisk edwar64896 (n=medwards@72.83.233.220.exetel.com.au) |
| 13:32.54 | conico | or ony certain ports? |
| 13:33.02 | webman | conico: usually you won't need to .... |
| 13:33.10 | conico | i read that rtp has nat problems |
| 13:33.16 | benjk | iax only uses one port |
| 13:33.19 | webman | just set nat=yes on asterisk sip.conf |
| 13:33.30 | conico | if i use sip ? |
| 13:33.45 | benjk | if you use sip you have to open a whole range of ports |
| 13:33.57 | webman | conico: if you use sip, it will normally work fine as long as you set nat=yes on asterisk |
| 13:34.08 | benjk | that depends |
| 13:34.34 | benjk | if the asterisk server is behind nat, you have to do some extra work |
| 13:34.38 | webman | benjk: I think you only do the port forwarding if the asterisk server is behind nat |
| 13:34.54 | benjk | if the clients are behind hat you may also have to do some work |
| 13:35.02 | webman | but if the clients are behind nat, then usually no work at all on the router |
| 13:35.32 | RoyK | webman: ping |
| 13:35.33 | RoyK | 1.2.8 2006-05-30 rev ~30860 |
| 13:35.33 | RoyK | 1.2.9 2006-06-06 rev ~32600 |
| 13:35.39 | conico | the clients are behind nat asterisk has public ip |
| 13:35.51 | RoyK | conico: then nat=yes should do the job |
| 13:36.01 | benjk | then reinvites are unlikely to work |
| 13:36.10 | benjk | you have to route all the traffic through the server |
| 13:36.10 | conico | for which one ? sip or iax ? |
| 13:36.49 | benjk | unless all clients are on the same LAN, even then you may find that they are getting the public address instead of the local one |
| 13:37.01 | benjk | for sip |
| 13:37.18 | conico | clienst are not on same lan |
| 13:38.10 | benjk | then you are going to find it challenging |
| 13:38.38 | conico | so if i use iax i will do no port forwarding on clients' adsl modems and they all use my bandwidth during their conversation ? |
| 13:38.41 | benjk | especially if you don't control all the NAT routers yourself |
| 13:38.46 | conico | onr just 8 sec ? |
| 13:39.29 | conico | well we have 2000 clients they are pharmacists |
| 13:39.43 | benjk | iax is designed to remove any peers in the middle |
| 13:39.44 | conico | pharmacy whole line seller we are |
| 13:39.58 | benjk | this usually takes 8-10 seconds |
| 13:40.01 | conico | i want to use asterisk for them |
| 13:40.23 | conico | but ii could not decide which one i should use |
| 13:40.27 | conico | sip or iax ? |
| 13:40.43 | benjk | if you have to ask, the chance is iax is better for you |
| 13:40.47 | webman | conico: use iax definitely and you will have much less nat problems, less bandwidth needed, and overall just better |
| 13:40.56 | conico | 2000 clients causes too much bandwidth iif they use my bandwidth |
| 13:41.02 | benjk | with sip you have to know what you are doing to get NAT traversal situations worked out |
| 13:41.19 | webman | conico: if you are worried about bandwidth then use iax |
| 13:41.26 | CtRiX | 2000 clients cause deadlock far before your BW is full ! |
| 13:41.28 | RoyK | webman: overall just better for some very few things, and bandwidth needs equals SIP |
| 13:41.52 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:41.59 | RoyK | CtRiX: we have ~2k clients on one box here. works for me (tm) |
| 13:42.05 | benjk | 2000 clients doesnt mean 2000 concurrent calls |
| 13:42.20 | webman | royk: no iax is less bandwidth for sip when the clients are talking to each other |
| 13:42.25 | CtRiX | RoyK, are you hunting deadlocks now? |
| 13:42.41 | webman | s/for/than/ |
| 13:42.46 | RoyK | webman: very, very, very little difference. rtp has a header consisting of, what, 12 bytes? |
| 13:43.06 | RoyK | and iax uses a similar header of what, 8 bytes? |
| 13:43.25 | benjk | the point is that if those clients have their own routers and he doesn't control them, then you are almost certain to have to forbid reinvites |
| 13:43.31 | RoyK | CtRiX: no. and it's been running quite stably for long |
| 13:43.47 | RoyK | CtRiX: I agree that asterisk is far from optimal, but it works |
| 13:44.02 | webman | royk: but iax audio will go direct between end points, whereas sip rtp will go via the asterisk server |
| 13:44.03 | CtRiX | ye dometimes it works |
| 13:44.31 | RoyK | webman: that way, yes..... |
| 13:45.26 | benjk | what kind of service is this anyway? |
| 13:45.31 | RoyK | webman: but then, bandwidth isn't really an issue. by the time you have 2k customers, it really does not count. also, what is the percentage of people calling others on the system, as compared to people calling out to PSTN? |
| 13:45.36 | RoyK | prolly an itsp :P |
| 13:45.43 | webman | conico: the question you should ask (of yourself) is how many calls will be from one client to another, and how many will be to your asterisk server (ie, outbound/external calls) |
| 13:45.43 | benjk | do you only have those clients call each other? |
| 13:45.47 | conico | clients will talk to each other and they have all their adsl routers and i will only supply an asterisk server on a public ip i will give them just their call ids /numbers |
| 13:45.56 | benjk | or do you have a gateway for PSTN calls too? |
| 13:46.11 | conico | no gateway for pstn for now |
| 13:46.16 | benjk | so its basically all peer to peer |
| 13:46.21 | conico | yess |
| 13:46.25 | benjk | in that case you don't need asterisk at all |
| 13:46.50 | benjk | set up enum in your DNS |
| 13:47.11 | benjk | then they can all call each other directly |
| 13:47.27 | benjk | need to have an IAX client that can make enum lookups |
| 13:47.33 | webman | conico: do your clients really call each other that often that this will be of benefit?? |
| 13:48.14 | conico | the clients will call our companies sellers,' too |
| 13:48.37 | conico | i wnna but viagra for ex |
| 13:48.46 | conico | buy |
| 13:49.07 | benjk | wrong channel |
| 13:49.13 | benjk | try #viagra |
| 13:49.23 | benjk | or #spammers |
| 13:49.27 | conico | i just want an enviroment like they can call each other and our companies selling people |
| 13:49.39 | conico | you rude benjk |
| 13:50.02 | benjk | hey, you brought up the rude word |
| 13:50.22 | conico | i dont know drug names |
| 13:50.29 | conico | i just recalled that one |
| 13:50.37 | conico | lol |
| 13:50.44 | webman | benjk: he is talking about drug clients.... |
| 13:51.01 | benjk | you seem to suffer from paranoia, maybe something against anxiety would help you |
| 13:51.09 | DrukenHME | i took viagra once... got stuck in my throught, had a stiff neck for hours... |
| 13:52.00 | *** join/#asterisk lorinc (n=ang@caracas-5068.adsl.interware.hu) |
| 13:52.05 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 13:52.35 | conico | our clients are pharmacists |
| 13:52.55 | benjk | I think a 0800 number will serve you much better |
| 13:55.09 | conico | in realty i just wants this.phrmacists will call our company for drug ordering.but some smart one come and say hey why we dont use this for them to talk each other.and at that that stage i dont want to say to him that we use iax and peer to peer talking uses our bandwidth too much |
| 13:55.36 | conico | so i ask if i use sip or iax |
| 13:55.38 | benjk | get an 0800 number |
| 13:55.55 | webman | conico: use iax |
| 13:55.59 | conico | cause i read that iax uses asterisk servers's bandwidth |
| 13:56.14 | conico | thank you webman i use iax |
| 13:56.37 | benjk | sip clients on different lans talking to each other need a server in between |
| 13:57.04 | conico | i have server on my network with public ip |
| 13:57.10 | conico | asterisk serve |
| 13:57.11 | conico | r |
| 13:58.01 | conico | i just want to cut pstn costs for calls for clientss |
| 13:58.05 | benjk | sure, if you have to be in the middle, it won't help you with saving bandwidth |
| 13:58.37 | conico | what wont help me ? |
| 13:58.45 | conico | iax ? |
| 13:58.58 | benjk | sip |
| 13:59.12 | conico | thanks |
| 13:59.55 | benjk | you should get an 0800 number |
| 14:00.03 | conico | what is 0800 |
| 14:00.07 | conico | number ? |
| 14:00.11 | conico | what for ? |
| 14:00.55 | benjk | for your pharmacists to be able to call you free of charge |
| 14:01.13 | conico | i want to involve in this new technology |
| 14:01.26 | *** join/#asterisk postel_ (n=jp@Wikimedia/Postel) |
| 14:01.27 | conico | newer than pstn |
| 14:01.41 | webman | royk: looks like that bug is introduced in r31520 ... |
| 14:02.08 | benjk | you rent an 0800 number that's delivered over sip or iax to you |
| 14:02.18 | benjk | then you set up an asterisk server to receive the calls |
| 14:02.30 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 14:02.39 | benjk | and feed it into your existing PBX or if you have, your IP deskphones |
| 14:02.52 | benjk | your clients use their existing phones to call you on the 0800 numbers |
| 14:02.55 | conico | i wanna use softphones |
| 14:02.56 | benjk | that makes sense |
| 14:03.03 | benjk | what you are trying to do makes no sense |
| 14:03.27 | benjk | you are going to lose sales calls |
| 14:03.47 | *** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it) |
| 14:03.50 | benjk | because people don't want to bother with stuff that's cumbersome |
| 14:04.04 | benjk | they want the ease of use of the telephone they are already used to |
| 14:04.13 | *** join/#asterisk Trionnis (i=lordkuri@12.206.2.116) |
| 14:04.19 | benjk | so they won't call you for sales, they call your competitor then |
| 14:04.57 | conico | more often we call them |
| 14:05.05 | Trionnis | howdy guys (and gals) |
| 14:05.09 | benjk | that wont work either |
| 14:05.17 | benjk | they may not run the softphone all the time |
| 14:05.24 | benjk | very likely they wont |
| 14:05.54 | conico | you are right |
| 14:06.08 | conico | a headphone and a speaker |
| 14:06.17 | conico | that is crap |
| 14:06.42 | benjk | even if you ship them an IP phone, the chance is that many of them won't even plug it in |
| 14:06.49 | conico | giving 2000 ip phones are too expensive |
| 14:07.00 | benjk | thats not the point |
| 14:07.24 | benjk | the point is that people are used to their existing phones, they are not going to have another phone or softphone just to call you |
| 14:07.42 | benjk | if every company they deal with does that, it will cause them a lot of headache |
| 14:08.02 | benjk | which kind of equipment or software do I have to use if I want to call company X ? |
| 14:08.06 | conico | ata devices ? |
| 14:08.09 | webman | benjk: I think that is the more important factor.... conico needs to have a reason why these people *want* to use his system |
| 14:08.21 | benjk | what kind of setting will I need to have active if I want to call company Y ? |
| 14:08.48 | benjk | give them an 0800 number, that is a reason |
| 14:08.57 | benjk | why would they want an ATA? |
| 14:09.17 | benjk | if every comapny they buy from gives them an ATA, how many ATA's and ohones will they have in their office? |
| 14:09.43 | conico | they can call other pharmacits free and our company calls them free? |
| 14:09.46 | benjk | people expect to be able to make ALL their calls from a SINGLE phone |
| 14:09.50 | Trionnis | there are some places you just can't cut costs, and you have to bite the bullet, heh |
| 14:09.56 | benjk | that's not a good reason |
| 14:10.07 | benjk | how many times are they going to call another pharmacist? |
| 14:10.20 | webman | conico: provide them with an asterisk pbx which will route your phone number via IP, and every other call via their pstn |
| 14:10.28 | Trionnis | lol |
| 14:10.32 | benjk | even that they won't take |
| 14:10.52 | benjk | if every comapny they deal with gives them a PBX, how many will they have? |
| 14:11.03 | Trionnis | hey benjk, you remember the wierd thing I had with the audio files in my IVR yesterday? |
| 14:11.16 | benjk | its ludicrous to think that your customers are enthusiastic about your favourite toy technology just because you are |
| 14:11.23 | benjk | for them there is no problem |
| 14:11.43 | benjk | you are trying to sell a solution for which there is no problem |
| 14:11.59 | webman | conico: you need to find a requirement or a need.... yes, benjk... exactly :) |
| 14:12.23 | benjk | if you get an 0800 number, then you have something they will like |
| 14:12.38 | benjk | and you still get to play with asterisk |
| 14:13.02 | benjk | because you get an 0800 number that's delivered via sip or iax to you, so asterisk will be your entry point for the calls |
| 14:13.14 | conico | does a 0800 not wotk without voip/asteris? |
| 14:13.28 | benjk | they can use their existing phones to call you free of charge, they will like that |
| 14:13.31 | webman | I've tracked my bug report down to a single change (svn version) does anyone know if there is anything more I could do to try and fix it or get it to progress ?? |
| 14:13.37 | conico | if it works with pstn then no need for asterisk ? |
| 14:13.42 | Trionnis | you can get a regular forwarded number, but they tend to be more expensive |
| 14:13.46 | benjk | you get the calls over voip from the provider of the 0800 number |
| 14:14.20 | benjk | its less costly to do it via voip because you don't have to have a bunch of PSTN lines |
| 14:14.33 | benjk | you can get multiple calls all over the ip connection |
| 14:14.43 | Trionnis | and the provider isn't paying for the incoming call, and the outgoing forward |
| 14:14.50 | benjk | indeed |
| 14:14.51 | *** join/#asterisk AdmoIRC (n=Miranda@user-0c93s2v.cable.mindspring.com) |
| 14:15.37 | conico | sound quality ? better than pstn or worse or depends? |
| 14:15.42 | benjk | and then, if you have some pharmacists who are technologically advanced, like some that do have their own IP-PBX, then you can publish a SIP URI or an IAX URI on which they can call you |
| 14:15.50 | conico | depends on my bandwidth for this 0800 |
| 14:15.52 | conico | ? |
| 14:15.55 | Trionnis | depends on the provider, bandwidth availiable, latency |
| 14:16.08 | benjk | sound quality generally the same as PSTN, but a lousy provider can make it worse |
| 14:16.16 | Trionnis | g711u can actually sound *better* than pstn in the proper scenario |
| 14:16.51 | benjk | only if you mean to say PSTN == analog |
| 14:17.01 | benjk | PSTN = (analog, ISDN) |
| 14:17.03 | Trionnis | well yes |
| 14:17.10 | conico | and with this way 0800 we can not call them via voip ? |
| 14:17.15 | Trionnis | thanks mr. pedantic lol |
| 14:17.15 | Trionnis | ;) |
| 14:17.15 | benjk | if you have ISDN, that uses g711, so it's the same |
| 14:17.56 | Trionnis | conico: you can route calls however you want, if you want to send it out an FXO port, you can, or you can send it out voip, or whatever |
| 14:18.57 | conico | how can i send out with voip to them ? |
| 14:19.06 | conico | they are connected to pstn |
| 14:19.06 | benjk | you can't |
| 14:19.16 | benjk | you have to call via the pstn |
| 14:19.26 | benjk | but you can use a VoIP-PSTN gateway service |
| 14:19.33 | benjk | that's usually cheaper |
| 14:19.34 | conico | no benefit and too much dependencies and this fails |
| 14:19.43 | conico | thank you lol |
| 14:19.43 | benjk | not really |
| 14:19.45 | benjk | its cheaper |
| 14:19.54 | benjk | and if you can't make a call, you can use a different one |
| 14:20.05 | benjk | and eventually fallback to your PSTN connection |
| 14:20.12 | benjk | so you increase failsafety |
| 14:20.20 | benjk | because you have more options to make a call |
| 14:20.52 | conico | is this for pharmacists? |
| 14:20.54 | benjk | in any event, what you want to do isn't going to work |
| 14:21.04 | webman | benjk: but it depends on the volume of calls :) |
| 14:21.21 | benjk | because most of your clients will not be running their softphones when you want them to |
| 14:21.22 | conico | is this failsafety for pharmacists |
| 14:21.30 | conico | yes i agree |
| 14:21.48 | benjk | the pharmacists will not see any change other than that they can call an 0800 to reach you |
| 14:21.57 | benjk | they will still use their existing PSTN phones |
| 14:22.35 | conico | all is failed |
| 14:22.37 | conico | lol |
| 14:22.42 | benjk | and if you want to call them cheaper, you can set your PBX to first try a VoIP call via a gateway provider |
| 14:22.58 | benjk | if that call fails, it can automatically try the PSTN route that you have now |
| 14:23.23 | benjk | so its not any worse than what it is now, but you have the chance to safe some money as most of the calls will go through |
| 14:23.29 | benjk | via VoIP |
| 14:23.42 | benjk | the pharmacists will receive those calls on their PSTN phones |
| 14:23.50 | benjk | they won't even know you are using VoIP to call out |
| 14:24.17 | conico | is this via 0800 too ? |
| 14:24.21 | benjk | no |
| 14:24.26 | benjk | the 0800 is for incoming |
| 14:24.45 | benjk | people call you on that 0800 number and you receive the calls over voip |
| 14:24.54 | conico | for outgoing? |
| 14:24.58 | conico | what should i do ? |
| 14:25.09 | Trionnis | does anyone know if * caches IVR menu audio files somewhere other than /var/lib/asterisk/sounds/ ? |
| 14:25.10 | benjk | when you make an outgoing call it may use the same provider that you got the 0800 number from or any other provider |
| 14:25.27 | benjk | in fact, you can sign up with multiple providers, which increases your failsafety |
| 14:25.29 | Trionnis | conico: I'd recommend you have a look at voipjet for outgoing |
| 14:25.40 | benjk | if one provider cannot deliver the call, asterisk will try the next provider etc etc |
| 14:25.56 | Trionnis | they're pretty inexpensive, and you can set outgoing CID so it looks like you're calling from the regular pstn line |
| 14:25.59 | conico | will i need fxo for this ? |
| 14:26.09 | conico | i am in turkiye |
| 14:26.09 | benjk | there are many providers who do plans that are entirely prepaid, no monthly basic fee |
| 14:26.18 | benjk | so if you don't use them you don't pay |
| 14:26.28 | benjk | don't use analog |
| 14:26.28 | Trionnis | if you want to fail over to pstn ultimately, then yes, you'll need an interface |
| 14:26.35 | benjk | analog is unreliable |
| 14:26.48 | benjk | you want pure VoIP or ISDN |
| 14:26.56 | benjk | you don't want anything with FXO ports |
| 14:27.17 | Trionnis | benjk: that might be ideal, but could be cost prohibitive |
| 14:27.18 | conico | i see... |
| 14:27.33 | benjk | messing around with analog gear is far more expensive |
| 14:27.34 | Trionnis | some telcos charge obscene prices for a single BRI |
| 14:28.04 | benjk | in my experience analog is about 10-50 times more expensive than ISDN |
| 14:28.13 | Trionnis | how so? |
| 14:28.43 | benjk | like messing around for 3 weeks fulltime before the damn FXO port has anything close to acceptable performance for the customer to sign off |
| 14:28.45 | webman | benjk: In aus ISDN is more expensive than analog by about 10-30% |
| 14:28.56 | benjk | or like messing around for 3 weeks and the customer cancels the contract |
| 14:29.05 | Trionnis | I can get a basic analog circuit here for ~$30/mo usd including taxes, as opposed to a single BRI being about $300 install, and ~$150/mo |
| 14:29.11 | webman | benjk: oh yeah, well certainly in that way :) |
| 14:29.15 | benjk | and you have to pay two engineers salaries for 3 weeks without any income |
| 14:29.25 | benjk | well, analog is always like that |
| 14:29.33 | benjk | when used in business environments |
| 14:29.47 | benjk | the only place where analog is ok is for home users |
| 14:29.57 | Trionnis | also I have to pay for incoming as well on a BRI |
| 14:30.11 | Trionnis | and yes, this is a business line :) |
| 14:30.17 | webman | any business should use ISDN anyway, it makes cabling easier, better quality, faster calls, CID is better, actually, everything is better :) |
| 14:30.27 | benjk | there you can say, OK, this Sipura costs you 75 USD, and its 50 USD for us to set it up, this is what you get, it aint going to be any better than this, take it or leave it |
| 14:30.55 | benjk | you can't do that with a business customer, not even a tiny one-man business |
| 14:31.34 | Trionnis | oh, I agree webman, I just can't justify those kind of costs when I can get analog failover for a lot less :) |
| 14:31.50 | Trionnis | a lot of it depends on the local telco, and how bad they want to jack you for the line |
| 14:32.08 | benjk | I charge 1500 USD installation and setup for a single analog POTS line with a Sipura 3000 |
| 14:32.19 | Trionnis | luckily, I have the joy of dealing with a small family owned ILEC, so they jack you around as much as possible |
| 14:32.25 | benjk | and I don't give any guarantee that it will work for more than 5 mins |
| 14:32.41 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 14:32.43 | benjk | if it breaks when we walk out the door, it will cost 100 USD per hour to come back |
| 14:32.51 | benjk | analog is gambling |
| 14:33.17 | benjk | you have been warned, if you are so fucking dumb that you want it anyway, you are going to get ripped off for that stupidity |
| 14:33.52 | benjk | bottom line is I don't want customers who insist on analog after I tell them they shouldn't want analog |
| 14:34.42 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 14:35.20 | webman | benjk: I guess you do have to admit that there are some locations where the cost difference between ISDN and analog is too high, and analog is a 'better' business option.... at least to try it first to see if it works :) some work without any problem |
| 14:35.36 | benjk | I don't want to serve such a location |
| 14:35.51 | benjk | unless the customer takes 100% of the risk |
| 14:35.59 | benjk | if it doesn't work, they pay me anyway |
| 14:36.13 | benjk | in that case I have no issue with wasting their time and money |
| 14:36.27 | benjk | if it is going to waste my time and my money I do have an issue |
| 14:37.29 | benjk | there is a reason why NEC charges 6000 USD for the smallest analog interface card for their smallest PBX |
| 14:37.47 | Trionnis | that would be called "taking advantage of people" ^^ |
| 14:37.51 | webman | speaking of my time and money, I have found a major bug in asterisk which is affecting two of my clients, and it has been in asterisk since 1.2.9.... is there any bug marshall or something that can look at it for me ? |
| 14:38.28 | benjk | no, its called thousands of man years of fiddling with stuff until it works under very possible and impossible circumstances |
| 14:38.36 | tzafrir_laptop | webman, have you files it yet in bugs.digium.com? |
| 14:39.10 | tzafrir_laptop | once you do that, you'll find bug marshals lurking at #asterisk-bugs |
| 14:39.23 | benjk | unless they don't like your bug |
| 14:39.39 | benjk | if they dont like your bug, they may just delete it |
| 14:40.17 | webman | tzafrir_laptop: yes, hours ago |
| 14:40.23 | benjk | "please dont abuse the bugtracker, purchase commercial support" |
| 14:40.37 | webman | for those interested see bugid 8064 |
| 14:40.38 | benjk | "this is a configuration issue" |
| 14:41.22 | webman | benjk: nope, it was introduced in asterisk svn revision 31520 ... I know the exact lines of code... just dunno how to fix them :( |
| 14:41.33 | benjk | like when asterisk tried to use 0.0.0.0 as a verbatim ip address |
| 14:41.42 | benjk | that was deemed a configuration issue |
| 14:41.46 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-197-7.red.bezeqint.net) |
| 14:42.07 | webman | benjk: oh.... great :) |
| 14:42.28 | benjk | I mean how come you do bindport=0.0.0.0 ? |
| 14:42.49 | benjk | its only documented as an option in the sip.conf sample file, how dare you actually use that |
| 14:43.45 | webman | benjk: well, for another one like that, see bugid 8065 which I posted tonight |
| 14:44.15 | benjk | nah, I already had dinner, don't want to lose it |
| 14:44.37 | webman | queues.conf.sample has eventmemberstatusoff = no but it should be eventmemberstatus = no |
| 14:45.19 | webman | tzafrir_laptop: yeah, I have done that already... |
| 14:45.39 | tzafrir_laptop | yeah, I saw that |
| 14:48.22 | RoyK | webman: I guess someone'll fix it quite soon, then |
| 14:49.30 | webman | royk: yeah, but it is sunday night here, and I'd like to go home knowing it is fixed :) |
| 14:53.13 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.210) |
| 14:55.52 | jtexter3 | Anybody here done much with ChanSpy on the 1.4 beta? |
| 15:02.12 | DarKnesS_WolF | jtexter3: chanspy on 1.4 ? or u mean with the normal patch method ? |
| 15:08.51 | *** join/#asterisk _ViperNetworks (n=Nitesh@c-68-61-148-94.hsd1.mi.comcast.net) |
| 15:09.04 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 15:21.47 | *** join/#asterisk rigid (n=The@port-212-202-77-8.dynamic.qsc.de) |
| 15:21.51 | rigid | re |
| 15:22.03 | rigid | is it possible to use skype with asterisk, yet? |
| 15:22.20 | rigid | has someone reverse engineered the protocol? |
| 15:23.49 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 15:28.41 | benjk | there are several skype hookups for asterisk, they all involve using the official skype client which has to run on the same machine |
| 15:28.59 | benjk | in other words, kludges |
| 15:32.04 | ai-a[afk] | however the protocol has been reversed in china :),,, so watch this space. |
| 15:32.20 | benjk | in China |
| 15:32.40 | benjk | where Skype wouldn't be able to get anywhere with a lawsuit |
| 15:32.48 | benjk | won't happen anywhwere else |
| 15:33.09 | CtRiX | and skype is lieky to change specs in a minute or so |
| 15:33.21 | CtRiX | as a closed protocol, every change wouldn0t be a problem. |
| 15:33.25 | coppice | i don't think the protocol has been reverse engineered in china, but there are kludge interfaces there too. |
| 15:33.34 | CtRiX | the best solution is simply avoiding the use of skype. |
| 15:33.36 | benjk | yeah, one update and your reengineering effort is f****d |
| 15:34.07 | ai-a[afk] | search google for reports of it then. |
| 15:34.34 | benjk | parts of it have been reengineered according to the folks who claim they did it |
| 15:34.37 | coppice | you can find lots of reports of this, but they all turn out to be kludging things |
| 15:34.52 | benjk | how credible their claims are is another matter |
| 15:35.03 | RoyK | some customers of mine have been asking for video conference á la skype. is this doable with asterisk? |
| 15:35.35 | ai-a[afk] | nothing wrong with a 'kludge' anyway,, you need need to feed it an audio device and manipulate it using skype api... simple and should work forever. |
| 15:35.48 | benjk | in any event, they are doing this so they can make their own Chinese skype service |
| 15:36.03 | benjk | so they don't really need to be interoperable if Skype decides to break their stuff |
| 15:36.15 | coppice | even biggish outfits like Pika claim to have a clean Asterisk to Skype interface. I was told that's just another kludge |
| 15:36.18 | benjk | there is no such thing as a skype api |
| 15:36.21 | CtRiX | RoyK, sip && h323 support video |
| 15:36.26 | benjk | its a skype softphone api |
| 15:36.50 | benjk | ever tried to hook a pbx into a cell phone as a replacement for an ISDN trunk? |
| 15:37.05 | benjk | that's what those folks are doing |
| 15:37.09 | benjk | its a kludge |
| 15:37.44 | benjk | coppice what they mean by "clean" is legally clean |
| 15:37.55 | benjk | because they use the softphone api legally |
| 15:38.08 | coppice | they also vaguely imply scalable :-) |
| 15:38.12 | benjk | and they have an arrangement with Digium for their closed source thing |
| 15:38.23 | RoyK | CtRiX: sure, I know, but they need something like the GUI stuff, showing each member etc |
| 15:38.33 | CtRiX | ah ... |
| 15:38.47 | benjk | well, you can always run virtual machines to host multiple skype softphones I guess |
| 15:38.51 | *** join/#asterisk _deg_ (n=deg@201-15-254-2.ctame704.dsl.brasiltelecom.net.br) |
| 15:39.21 | CtRiX | benjk, you can run several skypes in different X sessions... with different users, i think. |
| 15:39.42 | CtRiX | even X fake sessions... |
| 15:39.54 | benjk | not sure if they can all make phone calls at the same time as they would be competing for the audio device |
| 15:40.24 | CtRiX | i think you can configure it to a fake audio device |
| 15:40.24 | benjk | maybe you can round robin and then feed that into asterisk |
| 15:40.28 | benjk | good luck with that |
| 15:40.32 | *** join/#asterisk blop (n=blop@40.187-242-81.adsl-dyn.isp.belgacom.be) |
| 15:40.41 | benjk | it'd be a horrible kludge |
| 15:40.47 | CtRiX | yes |
| 15:40.50 | *** part/#asterisk blop (n=blop@40.187-242-81.adsl-dyn.isp.belgacom.be) |
| 15:40.52 | *** join/#asterisk blop (n=blop@40.187-242-81.adsl-dyn.isp.belgacom.be) |
| 15:41.42 | benjk | people are kidding themselves if they think there is going to be any kind of skype solution to gain access from a switching system |
| 15:41.45 | blop | hi |
| 15:41.47 | blop | =) |
| 15:41.51 | blop | has anyone here ever tried using TDM400P on a VIA EPIA board ? :) |
| 15:41.59 | benjk | they are on skype hype koolaid |
| 15:44.51 | *** join/#asterisk Ard0gx (n=adiaz0@200.87.54.142) |
| 15:45.29 | Ard0gx | hi, is somebody using chan_ss7 with * ? |
| 15:47.33 | [TK]D-Fender | blop : I know someone who did.... |
| 15:48.15 | coppice | Ard0gx: people seem to have a lot of problems with it |
| 15:48.18 | blop | i found some info on the mailing list :) http://lists.digium.com/pipermail/asterisk-dev/2004-September/006172.html |
| 15:48.49 | blop | it should work, but i need to get 3x TDM400P to work so i better choose a correct power unit |
| 15:51.03 | webman | blop: I think you are more likely to have a problem with your motherboard/PCI/interrupts ... :) |
| 15:52.10 | Ard0gx | and, someone using cluster config with it ? |
| 15:54.03 | blop | webman: i'm looking for a small system to run my 3x TDM400P, sth the most fanless i can find :) even wth hdd as i can boot/pxe and run asterisk from ram, you got any idea about such configurations? |
| 15:54.31 | benjk | forget it |
| 15:54.51 | benjk | get a sip-fxo gateway |
| 15:54.54 | blop | the size doesnt matter =) |
| 15:55.09 | webman | blop: I think you will have a lot of problems finding a system which will run 3 x TDM cards properly... you should use some sort of external ATA |
| 15:55.22 | blop | :/ |
| 15:55.59 | blop | what about 2 TDM ? |
| 15:56.08 | benjk | still not recommendable |
| 15:56.12 | blop | i could split in two system |
| 15:56.43 | benjk | get a gateway |
| 15:56.46 | blop | ATA's sux :/ i'd prefer zaptel hardware |
| 15:56.57 | benjk | mediatrix, audiocodes or whatever they're called |
| 15:56.59 | blop | as its controlled directly by asterisk, its faster |
| 15:57.10 | benjk | zaptel sucks more than those, especially for analog |
| 15:57.32 | benjk | get a channel bank |
| 15:57.40 | benjk | PRI card + channel bank |
| 15:57.56 | blop | channel bank ? |
| 15:58.00 | benjk | yes |
| 15:58.35 | benjk | they have configurable fxs/fxo ports and you talk to them over a T1 so you can hook it into a PRI card |
| 15:58.44 | benjk | they usually have 12 or 24 ports |
| 15:59.05 | benjk | ADTRAN |
| 15:59.08 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 15:59.20 | benjk | also, there's the astribank |
| 15:59.31 | benjk | its using USB for up to 16 ports |
| 15:59.54 | blop | well, i dont have any T1, or even PRI/BRI :) |
| 16:00.08 | benjk | its not as solid as a T1 channel bank, but still better than messing about with analog zaptel cards |
| 16:00.10 | blop | i only do analog atm, its a system which run @home =) |
| 16:00.21 | benjk | you are not listening |
| 16:00.26 | blop | well i'm trying |
| 16:00.32 | benjk | the T1 is not for using ISDN |
| 16:00.44 | benjk | you don't need any ISDN |
| 16:00.53 | benjk | you use the PRI card for connecting the channel bank |
| 16:01.21 | blop | ok, so its 2 different devices? |
| 16:01.46 | benjk | [asterisk]==[T1-interface-card]-------[T1-channel-bank]-----up to 24 fxs/fxo ports |
| 16:01.50 | blop | :) |
| 16:01.56 | blop | sounds nice |
| 16:02.00 | benjk | its solid |
| 16:02.22 | benjk | and you can get those T1 channel banks for a few hundred dollars on eBay |
| 16:02.52 | blop | well, i could sell the 2 tdm400p cards i already have and switch to that |
| 16:03.03 | benjk | that sounds like a plan |
| 16:03.05 | blop | :) |
| 16:03.13 | benjk | search for Adtran + channel bank |
| 16:03.34 | blop | and i should get a T1 interface card working in a VIA EPIA system? |
| 16:03.34 | benjk | there are probably other vendors too, but most people seem to talk about adtran |
| 16:03.39 | benjk | yes |
| 16:03.42 | blop | :) |
| 16:03.47 | benjk | a single span card should be fine |
| 16:03.50 | blop | thanks already for the informations :) |
| 16:04.03 | benjk | like a Digium T110 or a Sangoma A101 or OpenVox D110 |
| 16:04.21 | benjk | unless you want to hook up more than one channel bank |
| 16:04.23 | blop | sth like http://www.digium.com/en/products/hardware/te110p.php ? :) |
| 16:04.26 | blop | ok :) |
| 16:04.41 | blop | nice :) |
| 16:05.44 | benjk | that one, yes |
| 16:05.46 | benjk | or http://www.sangoma.com/datasheets/p_aft-et1-specs |
| 16:06.28 | blop | ouch :) its not that cheap :D |
| 16:06.54 | benjk | if you want cheap, check this out http://www.openvox.com.cn/products.php?genre_id=18 |
| 16:07.13 | benjk | D110 |
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| 16:09.01 | blop | hum :) there must be some performance differences btw those of course :) or should the OpenVox D110 work as well as the digium one? |
| 16:09.16 | benjk | it should be the same |
| 16:09.21 | blop | :) |
| 16:09.33 | benjk | this particular model is based on a GPL hardware design |
| 16:09.41 | benjk | you can make them yourself if you want to |
| 16:09.51 | benjk | http://www.zapatatelephony.org |
| 16:10.02 | blop | "D110P is 100% hardware compatible with TE110P." :) |
| 16:10.06 | benjk | the one up there is a 4 port card |
| 16:10.16 | benjk | called the Tormenta II |
| 16:10.24 | blop | hey :) fun |
| 16:10.36 | benjk | the D110 is a single port version of that (with some changes) |
| 16:11.42 | benjk | if you are looking for some companies in the US who specialise in printed circuit boards and protoypes, you will find that you could make your own batch of 10 for about half the retail price of the D110 |
| 16:12.09 | benjk | put the remaining 9 cards up for sale on eBay |
| 16:12.31 | benjk | or get 9 buddies to join you |
| 16:13.27 | blop | :) |
| 16:13.41 | benjk | that's how Digium started, btw |
| 16:13.50 | blop | well, i dont live in the US actually :D but i'll surely read all those stuff |
| 16:14.30 | benjk | that makes it a little more difficult because many of the companies who do very small volumes only deliver to US customers |
| 16:15.02 | benjk | where abouts are you? |
| 16:15.05 | benjk | which country |
| 16:15.27 | blop | belgium :) |
| 16:15.55 | blop | what's the "Rohs compliance" about? |
| 16:16.15 | coppice | if you are in belgium you should know |
| 16:16.54 | blop | :) |
| 16:17.00 | benjk | Its a new EU directive that requires all electronics manufactured in or delivered to the EU to be free of any components which have lead and other substances considered hazardous for the environment |
| 16:17.23 | blop | The RoHS Directive stands for "the restriction of the use of certain hazardous substances in electrical and electronic equipment". This Directive bans the placing on the EU market of new electrical and electronic equipment containing more than agreed levels of lead, cadmium, mercury, hexavalent chromium, polybrominated biphenyl (PBB) and polybrominated diphenyl ether (PBDE) flame retardants. |
| 16:17.25 | blop | okie :) |
| 16:17.26 | benjk | I think it came into effect on June 30 this year |
| 16:17.43 | blop | i had never heard of it :D |
| 16:17.45 | coppice | and a real piece of BS it is, too |
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| 16:18.15 | rob0 | Could say that about most laws / regulations / directives :) |
| 16:18.26 | benjk | well, forcing semiconductor makers to abandon lead and cadmium is in principle a good thing |
| 16:19.09 | blop | OpenVox seems reallly nice :) |
| 16:19.23 | coppice | not really, when they haven't worked out a proper alternative to lead solder |
| 16:19.34 | benjk | that's the point |
| 16:19.37 | benjk | now they have to |
| 16:19.49 | blop | :P |
| 16:20.35 | coppice | mercury still goes in lamps. they haven't worked out an alternative for that, so it has an exemption. somehow the electronics business got screwed. now makers have to cope with huge failure rates for anything sold in the EU |
| 16:21.07 | benjk | but that's only during the transition |
| 16:21.18 | coppice | transition to what? |
| 16:21.27 | benjk | until they work out the toothing problems |
| 16:21.43 | coppice | there is no current solution |
| 16:21.44 | benjk | once they have to, they will find alternatives |
| 16:21.58 | benjk | that's the point, the solution has to be found now |
| 16:22.23 | coppice | oh great. screw up all the electronics, and then look for a cure :-) |
| 16:23.10 | blop | benjk i could also use a A1200P in place of 3 TDM as it provides 12 modules port :D |
| 16:23.22 | benjk | I think we need far more painful laws to look that our children have a planet to live on |
| 16:23.36 | coppice | the lead in the EU mostly comes from water pipes. nobody is replacing those at any significant rate |
| 16:23.41 | benjk | blop, like I said, don't use analog cards |
| 16:23.52 | coppice | that's the one they should address |
| 16:24.01 | benjk | well, that's something that should be done then |
| 16:24.18 | coppice | and even though they recycle car batteries, the loss on each cycle is significant |
| 16:24.27 | benjk | the worst thing however is when everybody says "why me" |
| 16:25.11 | benjk | the more industries have been forced into something like RoHS, the fewer other industries can say "not us" |
| 16:25.14 | blop | benjk: well i do need to link 2 analog lines and lots of analog phones tough :/ i would rather change for digital but i cant replace everything, or maybe there are some issues with analog cards that i wont have with a channel bank ? |
| 16:25.32 | benjk | get an astribank then |
| 16:25.42 | benjk | its for your home, right? |
| 16:25.43 | coppice | well, when 90% of the people in London have to worry about the lead content of their water every morning, it might slightly piss them off to find their TV only lasts 2 years because some clown took the lead out without an alternative |
| 16:25.52 | blop | yup its at my place :) |
| 16:26.19 | benjk | TV is overrated anyway |
| 16:26.49 | blop | :P |
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| 16:27.12 | benjk | blop, I think an fxs/fxo gateway or an astribank is your best choice |
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| 16:27.30 | benjk | have you considered using IP phones instead of analog ones? |
| 16:27.53 | blop | well i have :) |
| 16:27.58 | benjk | because then all you'd need is two sipura-3000 adapters for the analog lines |
| 16:28.17 | benjk | or better still, get a BRI circuit |
| 16:28.29 | benjk | and use a single port BRI card (less than 50 USD) |
| 16:28.29 | blop | i've already worked with sipura 3000, but i dont like them, its slower than a TDM :D |
| 16:28.40 | benjk | its less troublesome |
| 16:28.54 | blop | i could use only 1 tdm with 2 fxo and switch the phones to ip |
| 16:29.14 | blop | i should check what's cheaper |
| 16:29.15 | benjk | ideally, you'd use all IP phones and use BRI instead of POTS |
| 16:29.51 | blop | yeah, but BRI's cost more every month :/ and i need 2 lines as i do have 2 adsl over them |
| 16:30.22 | benjk | maybe you can resell a DID or two to subsidise the cost of the circuit |
| 16:30.33 | blop | lol :) |
| 16:30.38 | blop | why not :D |
| 16:31.12 | hmmhesays | well i finally sold my les paul |
| 16:31.15 | benjk | analog is really shitty, once you have experienced the difference, you really don't want to go back |
| 16:31.19 | blop | i presume we can find channel bank for BRI too ? =) |
| 16:31.46 | benjk | well, you can use multiport BRI cards and configure the ports so that they can drive ISDN phones |
| 16:32.09 | benjk | but the multi-port BRI cards are significantly more expensive than those sub-50 USD single port ones |
| 16:32.44 | blop | openvox dont do BRI/PRI products? |
| 16:32.53 | benjk | the D110 is PRI |
| 16:33.03 | benjk | but they don't have a BRI card yet |
| 16:33.25 | coppice | openvox must be the only card maker that doesn't do cgheap BRI cards :-) |
| 16:33.31 | benjk | Billion in Taiwan make a BRI USB adapter |
| 16:33.41 | blop | "D110P is a single span E1/T1/J1 digital line telephony product." => so E1/T1 is considered as PRI ? |
| 16:33.52 | benjk | yes |
| 16:33.58 | blop | ok :) |
| 16:33.59 | benjk | PRI == primary rate ISDN |
| 16:34.21 | benjk | T1 in the US, CA, JP, TW and E1 elsewhere |
| 16:34.25 | blop | so PRI is always 32 channels max |
| 16:34.37 | benjk | 24 channels for T1, 32 for E1 |
| 16:34.43 | benjk | 1 channel is used for signaling |
| 16:35.01 | benjk | E1 also uses an additional channel for timing |
| 16:35.22 | benjk | thus 23 channels for voice on a T1 PRI and 30 channels for voice on a PRI E1 |
| 16:35.27 | blop | :) |
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| 16:35.58 | blop | and a channel can be linked either to a line or a phone it doesnt matter with digital devices ? |
| 16:36.00 | coppice | T1 in the US, CA, JP, TW, HK and E1 elsewhere |
| 16:36.11 | benjk | ok |
| 16:36.33 | benjk | a channel bank has configurable FXS and FXO ports |
| 16:37.02 | benjk | ah, misread your question, sorry |
| 16:37.33 | benjk | with PRI you cannot connect any phones, only switches |
| 16:38.06 | benjk | with BRI you can configure a port for connecting to another switch or to an ISDN phone |
| 16:39.07 | blop | ok, and "channel bank" always means gateway to analog system ? or you have channel bank to connect isdn phones also? |
| 16:39.48 | benjk | so if you have a multi-port BRI card you could configure one port to hook up to your BRI circuit from the phone company and the remaining ports to hook up ISDN phones |
| 16:40.02 | blop | ok |
| 16:40.16 | benjk | I have never seen the term channel bank used for ISDN phones |
| 16:40.39 | benjk | the Astribank is a USB device |
| 16:40.47 | blop | but lets say i buy a D110PG, can i link it to analog and digital phones and lines ? using one or more devices linked to it |
| 16:41.01 | benjk | it connects to your server via USB 2 and it gives you multiple FXO/FXS ports |
| 16:41.19 | benjk | there are several versions, 8, 12 and 16 ports I think |
| 16:41.55 | blop | yeah i've seen that :) i'm on their website |
| 16:42.02 | benjk | the D110 is a PRI card, you cannot connect any ISDN phones to it |
| 16:42.27 | blop | even with another device in btw them ? |
| 16:42.28 | benjk | PRI == only between switches |
| 16:42.42 | benjk | yes, there are such devices, but they are very expensive |
| 16:42.48 | blop | oki :) |
| 16:43.00 | benjk | Voxtream in Denmark make such a thing |
| 16:43.10 | benjk | E1 in and multiple BRI out |
| 16:43.23 | benjk | 1200 EUR or so |
| 16:43.28 | blop | ouch :) |
| 16:43.42 | benjk | and that's the one with the lowest number of ports |
| 16:43.50 | benjk | more ports and the price goes up |
| 16:44.00 | blop | but cant we link several isdn phones to 1 bri port ? |
| 16:44.10 | benjk | I dont remember exactly what the configuration was, but it was rather pricey |
| 16:44.33 | benjk | you can, but only 2 channels can be used on one port at a time |
| 16:44.55 | benjk | I think you can connect up to 7 ISDN phones on a single BRI port |
| 16:45.08 | benjk | but only two can actually use the circuit at any time |
| 16:45.26 | blop | yeah but that's fine for a personnal use :) |
| 16:45.43 | benjk | ok, I haven't thought about that |
| 16:45.56 | blop | but you wont hear the ongoing communication on the other phones like with analog phones in // right ? |
| 16:46.13 | benjk | in that case you could build a very cheap system by using two single port BRI cards |
| 16:46.20 | blop | :) |
| 16:46.35 | benjk | one for the BRI circuit from the telco, the other for up to 7 ISDN phones |
| 16:46.49 | blop | could be nice |
| 16:47.05 | benjk | you wouldn't be able to listen in on an ongoing conversation on anothe phone |
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| 16:47.14 | benjk | it would just be busy |
| 16:47.33 | blop | hummm :) |
| 16:47.41 | benjk | but yes, I suppose for a home setup, this wouldnt be a serious limitation |
| 16:48.03 | benjk | unless you have more than one teenager in your household |
| 16:48.13 | blop | but you cant call from one phone to another when their linked to the same BRI port of course .. :) |
| 16:48.19 | benjk | better use that sipura for the teenager |
| 16:48.40 | benjk | yes you can call between phones |
| 16:48.52 | benjk | but again, the limit of two channels applies |
| 16:48.59 | benjk | BRI is like a bus |
| 16:49.10 | benjk | bus system I mean |
| 16:49.22 | benjk | its actually called S0 bus interface |
| 16:49.37 | benjk | so all the devices on the bus can communicate with each other |
| 16:49.45 | benjk | but the limit of two concurrent channels appllies |
| 16:50.30 | blop | yeah but, one local phone to another local phone, and the 2 channels are busy (going throught asterisk?) or its only 1 channel ? |
| 16:50.55 | benjk | I am not sure, never done this |
| 16:51.00 | blop | :) |
| 16:51.06 | blop | anyway |
| 16:51.12 | benjk | I only use ISDN for connection to the telco |
| 16:51.25 | blop | i dont even have isdn phones =) |
| 16:51.37 | benjk | well, they are cheaply available |
| 16:51.46 | benjk | about 50 EUR or so |
| 16:51.49 | blop | so in the case i replace the old pstn ones, i could buy IP phones directly :D |
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| 16:52.02 | benjk | that's what folks here said, I think that's a second hand market price |
| 16:52.13 | benjk | yes |
| 16:52.15 | blop | :) |
| 16:52.49 | benjk | but if you can get second hand ISDN phones for 50 EUR a piece you'd get better hardware than you'd get for the same money spent on IP phones |
| 16:53.33 | blop | obviously :) |
| 16:53.36 | benjk | so its worth considering and researching |
| 16:54.15 | benjk | anyway, either way, try to avoid analog where you can |
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| 16:54.30 | blop | isdn phones need 4 wires instead of 2 right ? :) |
| 16:54.46 | benjk | they use RJ48 jacks |
| 16:55.00 | blop | 48 ? :) not 45 ? |
| 16:55.23 | benjk | well you can use RJ45 as well since its narrower it will fit in |
| 16:55.26 | dwclarkNU | hello all, having an issue where my 7960G phones wont automatically re-register to *...i can call out from them, but not into them, the only way i can resolve it is to reboot the phones so they re-register on start up |
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| 16:58.32 | blop | benjk: thanks a lot for your time :) i'll think about all that a bit now :) |
| 16:59.00 | benjk | welcome |
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| 17:18.22 | Trionnis | anyone have a suggestion of a DID provider that uses Level3 ? |
| 17:18.34 | Trionnis | recommendation actually |
| 17:29.39 | DrukenHME | Trionnis: you get that damn sipura workin? |
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| 17:53.35 | batphone | anyone else having problems w/teliax? |
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| 18:00.02 | Trionnis | anyone know what voicepulse charges for DID's? I can't seem to find it on their site anywhere |
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| 18:02.13 | batphone | ok so i go to bed last night after hanging up on a call through teliax |
| 18:02.19 | batphone | i wake up and hit redial |
| 18:02.28 | batphone | and now i get 407 Proxy Authentication Required |
| 18:02.59 | batphone | then SIP/2.0 503 Service Unavailable |
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| 18:06.31 | benjk | last time I checked, Voicepulse charged 11 USD per DID |
| 18:07.14 | Trionnis | ok, thanks |
| 18:08.33 | Trionnis | any experiences with exgn/Vitelity ? |
| 18:08.50 | Trionnis | e.g. reliability, support, etc |
| 18:11.20 | batphone | X-Asterisk-HangupCause: Unallocated (unassigned) number |
| 18:11.24 | batphone | this is in the debug |
| 18:11.33 | batphone | does it have anything to do with those error messages? |
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| 18:22.51 | *** join/#asterisk DEader (i=Messiah@ool-44c6aa92.dyn.optonline.net) |
| 18:23.12 | DEader | has any one experience one way audio when using a did with asterisk |
| 18:23.25 | DEader | can someone help me to fix this problem |
| 18:24.08 | batphone | DEader: that can be caused by a number of things |
| 18:24.12 | batphone | NAT would be one of them |
| 18:24.20 | batphone | what kind of phone are you using internally? |
| 18:24.27 | batphone | make sure there is a default gateway address in the phone.. |
| 18:24.52 | DEader | i'm using grandstream gvx 3000 |
| 18:25.02 | DEader | then i thought maybe the phone was broken |
| 18:25.17 | DEader | so now i'm using eyebeam softphone |
| 18:28.07 | DEader | which default gateway address i should use |
| 18:30.00 | batphone | ... eh |
| 18:30.15 | batphone | the default gateway for your network |
| 18:30.20 | batphone | like, 192.168.1.1 |
| 18:30.26 | batphone | whats the IP of your phones? |
| 18:30.43 | RoyK | DEader: just lookup the config from your PC |
| 18:30.47 | RoyK | if it's on the same net |
| 18:30.51 | RoyK | netstat -rn will show you |
| 18:30.55 | DEader | the ip is 192.168.2.2 |
| 18:33.04 | DEader | batphoen |
| 18:33.54 | batphone | iax works |
| 18:34.00 | DEader | batphone so the option in asterisk that said nat should i click no |
| 18:34.16 | batphone | you are using asterisk@home? |
| 18:34.20 | batphone | im not familiar with it |
| 18:34.43 | DEader | no i'm not using asterisk@home |
| 18:34.47 | DEader | i'm using trunk |
| 18:34.58 | batphone | um |
| 18:35.20 | batphone | one way audio... |
| 18:35.28 | DEader | sorry i mean in the sip config |
| 18:35.30 | batphone | put canreinvite=no in your sip.conf |
| 18:35.37 | batphone | under the phone's context |
| 18:35.44 | DEader | ok |
| 18:35.51 | DEader | canreinvite=no |
| 18:35.56 | DEader | what about the nat= |
| 18:36.16 | RoyK | always use nat=yes |
| 18:36.26 | RoyK | I rerally don't understand why there is a choice at all |
| 18:36.43 | RoyK | s/rerally/really/ |
| 18:36.57 | RoyK | nat=yes won't break non-nat clients |
| 18:37.09 | batphone | RoyK: ive seen it break it before |
| 18:37.46 | DEader | type=peer |
| 18:38.20 | DEader | so batphone do you think this problem is because i'm behind a router |
| 18:38.39 | RoyK | batphone: we have about 2k clients on this one box, all with nat=yes, and it works. 30% of them or so are not behind nat |
| 18:38.53 | batphone | how fortunate |
| 18:38.54 | RoyK | batphone: so I somehow doubt what you're saying |
| 18:39.16 | batphone | RoyK: ok man, I've built a few PBX's myself and im just telling you that I've seen it break before. |
| 18:39.24 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 18:39.33 | batphone | RoyK: for all I know it could be related to the version of asterisk on them... |
| 18:40.03 | DEader | hrm |
| 18:40.09 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.210) |
| 18:40.39 | DEader | do you think is the router batphone or royk |
| 18:40.42 | batphone | heh, what version was it where qualify=yes broke nat=yes |
| 18:40.55 | batphone | DEader: it might be, I assume you are using SIP |
| 18:41.05 | DEader | yeah i'm using sip |
| 18:41.17 | batphone | if you have access to the router's firewall logs might want to see if some RTP is getting dropped |
| 18:41.21 | batphone | which way is the audio? |
| 18:41.34 | batphone | inbound or outbound? |
| 18:41.48 | DEader | the audio is going outbound |
| 18:41.51 | DEader | but that inbound |
| 18:43.20 | *** join/#asterisk dwclarkNU (n=dwclark@c-24-128-111-200.hsd1.ma.comcast.net) |
| 18:43.23 | DEader | ? |
| 18:43.44 | DEader | what if i setup has iax will i still have that problem |
| 18:46.12 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:51.14 | *** join/#asterisk adorah (n=admin@87.68.168.128.cable.012.net.il) |
| 18:52.00 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 18:57.21 | *** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131) |
| 18:57.41 | Dr-Linux|work | what language is this? >> salah satu yang akan dibahas dalam artikel |
| 18:57.45 | *** join/#asterisk ToTo (n=ToTo@host138-138-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:11.45 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 19:12.45 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 19:21.40 | *** join/#asterisk rigid (n=The@port-212-202-77-8.dynamic.qsc.de) |
| 19:33.22 | Trionnis | when * attempts to register a SIP connection and it gets "SIP/2.0 407 Proxy Authentication Required", isn't it supposed to retry the registration with the authentication data? |
| 19:34.12 | jtexter3 | Okay, MOH question. Is there any way to not have the play stop and start when using native formats? Currently, I have some MP3's setup using mode=files, and every time a call is parked, the MOH starts from the beginning |
| 19:42.48 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.173) |
| 19:44.22 | *** join/#asterisk ManxPower (n=eric@71-8-11-111.dhcp.leds.al.charter.com) |
| 19:48.29 | *** join/#asterisk Vetaly (n=vetaly@adsl-49.ttn.ru) |
| 19:49.55 | Vetaly | hello all |
| 19:50.42 | *** join/#asterisk Borgon (n=l3orgon@12.173.143.25) |
| 19:52.05 | Borgon | Hello, is there like a sip to iax converter? am behind a double nat and i cant get incoming audio to work. IAx works fine but my application uses sip and am not hearing a thing, ive read about enabling rtp ports but its not working |
| 19:52.54 | ManxPower | Borgon: Asterisk will convert between voip protocols. |
| 19:53.12 | ManxPower | there is no reason for audio not to work if it is configured correctly. |
| 19:53.22 | ManxPower | Borgon: I assume you have a /etc/asterisk/rtp.conf |
| 19:53.48 | Borgon | ManxPower: The asterisk server is on a open full access vps... I am behind a hotel wifi and internal nat |
| 19:54.17 | ManxPower | Borgon: ah, so Asterisk server is not behind NAT? |
| 19:54.25 | ManxPower | Is the Asterisk server behind a firewall? |
| 19:54.58 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:55.21 | Borgon | ManxPower: the asterisk server is in a open vps.. remotely hosted in a datacenter server.... My program is a sip program on my laptop. |
| 19:55.57 | Borgon | my laptop is behind a hotel wifi, and internal nat.. i get a "symmetrical" router message when my application tries to configure sip |
| 19:56.05 | ManxPower | Borgon: do you have access to the asterisk server right now? |
| 19:56.09 | Borgon | yes |
| 19:56.12 | Borgon | am on the shell now |
| 19:56.15 | *** join/#asterisk rdlang (n=kvirc@82-171-126-45.dsl.ip.tiscali.nl) |
| 19:56.19 | ManxPower | edit sip.conf |
| 19:56.28 | Borgon | netstat -pan shows 5070 open but am seeing rtp shows it starts at 10000 and i got webmin on 10000 |
| 19:56.39 | Borgon | so let me close that see if that helps |
| 19:57.04 | ManxPower | put context=INVALID in the [general] section. put the correct context= line in the [device] section of sip.conf, also make sure you have nat=yes in the [device] section and you DO NOT CONFIGURE NAT SUPPORT ON THE SIP CLIENT |
| 19:57.33 | Borgon | ok |
| 19:57.33 | rdlang | any ppl here with experience with gtalk channel ? |
| 19:57.51 | ManxPower | the reason we put context=INVALID in [general] is to make SURE the call is matching the [device] section and not just being accepted without authorization |
| 19:58.08 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:58.45 | ManxPower | RoyK: me to |
| 19:58.46 | ManxPower | o |
| 19:58.55 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 19:59.40 | RoyK | my [default] context looks like exten => _X.,1,Verbose(don't use [default]) exten => _X.,n,Hangup |
| 20:00.31 | ManxPower | RoyK: I don't have any [default context] |
| 20:00.34 | *** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it) |
| 20:01.03 | RoyK | ManxPower: it's good to have it there in case someone messes up something, so I can see a call has come in |
| 20:01.15 | RoyK | i'm not the only one configuring these boxes |
| 20:02.47 | Vetaly | help me pleas i am have a problem vis transfer at asterisk |
| 20:02.57 | Vetaly | one way transfer !!! |
| 20:03.37 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 20:05.00 | Vetaly | SOMEBODY know this error ? |
| 20:05.01 | Vetaly | WARNING[20855] res_features.c: Did not read data. |
| 20:05.16 | Vetaly | this is error from transfer |
| 20:06.33 | ManxPower | Vetaly: I have never in my 5 years of working with Asterisk ever seen that message. |
| 20:06.42 | Vetaly | hello |
| 20:06.51 | ManxPower | Why don't you just use the builtin transfer features of your SIP phone? |
| 20:07.02 | hmmhesays | i bet it is a dtmf setting problem |
| 20:07.11 | Borgon | ManxPower: on sip debug when trying to register the sip user with firefly i receive |
| 20:07.13 | Borgon | Ignoring this INVITE request |
| 20:07.13 | Borgon | Retransmitting #2 (no NAT) to 192.168.50.15:5060: |
| 20:07.24 | Borgon | Via: SIP/2.0/UDP 192.168.50.15:5060;branch=z9hG4bK-d87543-0053370f9a443524-1--d87543-;rport;received=12.173.143.25 |
| 20:07.45 | Vetaly | ))) i use a AddPac |
| 20:07.48 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:07.51 | Vetaly | not a SIP phone |
| 20:07.56 | ManxPower | Borgon: then the registration does not match the sip.conf info. put your sip.conf ([general] and [device] up on pastebin |
| 20:07.59 | Borgon | ManxPower: the sip server has a different ip a 69.blah so i should bind it ? or make it static |
| 20:08.18 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-168-240.dsl.chcgil.sbcglobal.net) |
| 20:08.23 | Flauto | hi all |
| 20:08.25 | ManxPower | Borgon: I have no idea. |
| 20:08.25 | Borgon | ManxPower: there is no [device] in m y sip.conf |
| 20:08.32 | ManxPower | Borgon: then there is your problem. |
| 20:08.44 | ManxPower | you need a section for that sip account. |
| 20:08.51 | Vetaly | [hmmhesays] no i have a transfer at one side |
| 20:08.56 | Borgon | When i change wifis .. i can get sip working fine without configuring of no devices or anything. |
| 20:08.59 | ManxPower | where device=thesipuseridofyourclient |
| 20:09.07 | Borgon | but when am on this hotel wifi noene of my sip work |
| 20:09.10 | Flauto | is there anything extra that i need to do to be able to use jabber or googletalk when i compile asterisk svn? |
| 20:09.14 | Borgon | oh |
| 20:09.17 | Vetaly | ïðèâåò êòî þçàåò ðóññêèé? |
| 20:09.19 | Borgon | well yeah i have a user on it already |
| 20:09.21 | ManxPower | Flauto: I have no fucking idea. |
| 20:09.29 | ManxPower | Borgon: then pastbin that info |
| 20:09.30 | Flauto | hehe |
| 20:09.49 | Borgon | [general] |
| 20:09.49 | Borgon | context=INVALID |
| 20:09.56 | ManxPower | Borgon: ON PASTEBIN!!!!!!!!!!!!!!!!!!! |
| 20:09.59 | ManxPower | ~pb |
| 20:10.00 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 20:10.00 | Flauto | i saw the there are jabber.conf and gtalk.conf and jingle.conf in the sample config files |
| 20:10.24 | Borgon | iw ill paste it |
| 20:10.28 | Borgon | but i dont think it will make a difference |
| 20:10.30 | Flauto | so, it might be read to go after compiling asterisk? |
| 20:10.38 | ManxPower | Borgon: it can't hurt. |
| 20:10.58 | ManxPower | Flauto: the version of Asterisk that I use does not support those protocols. |
| 20:11.04 | Borgon | http://pastebin.ca/188134 |
| 20:11.34 | Flauto | okay |
| 20:13.35 | ManxPower | Borgon: http://pastebin.ca/188137 |
| 20:13.49 | ManxPower | Borgon: I assume you have NO nat settings on the client? |
| 20:14.17 | ManxPower | NAT configured on Server + NAT configured on client = Nat does not work. |
| 20:16.12 | Vetaly | ManxPower sorry |
| 20:16.30 | ManxPower | Borgon: perhaps the hotel nat router disallows UDP? |
| 20:19.38 | ManxPower | Borgon: sorry, I forgot to include nat=yes in the pastebin. make sure it's there |
| 20:19.54 | ManxPower | http://pastebin.ca/188144 |
| 20:20.07 | Vetaly | hey people have a RUSSIAN man at channel? |
| 20:20.13 | Vetaly | men ) |
| 20:23.03 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:23.42 | DrukenHME | ManxPower: would an iax client have problems with hotels and other shitty nat's ? |
| 20:24.16 | ManxPower | DrukenHME: that would depend on many things. In MY experience, I've never had to do IAX voa hotel nat |
| 20:24.27 | ManxPower | since I have my own SIP ATA/router I travel with |
| 20:24.59 | ManxPower | And I've made SIP calls via double nat on the client side and single nat on the server side. |
| 20:25.15 | DrukenHME | hmm... |
| 20:25.36 | DrukenHME | but wouldn't iax basically just work? no matter what ? |
| 20:25.46 | ManxPower | DrukenHME: in theory |
| 20:26.07 | ManxPower | but I don't have an IAX2 router+ata |
| 20:26.12 | DrukenHME | hmm... interesting |
| 20:29.07 | Vetaly | Ïðèâåò, A-Tuin|work! |
| 20:29.18 | DrukenHME | icky... |
| 20:29.24 | Vetaly | Hello, A-Tuin|work! |
| 20:32.52 | tzafrir_laptop | Vetaly, English, please... |
| 20:33.41 | *** join/#asterisk trumee (n=trumee@cpc6-cmbg1-0-0-cust742.cmbg.cable.ntl.com) |
| 20:33.55 | *** join/#asterisk adorah (n=admin@87.68.168.128.cable.012.net.il) |
| 20:34.59 | [Airwolf] | Can someone tell me why asterisk keeps connection to a diffrent port then I specify in the registry string ? |
| 20:35.58 | Borgon | ManxPower: ok i got it to register lets pray the sound works |
| 20:36.35 | trumee | guys, what rtp ports does asterisk use? |
| 20:37.05 | trumee | i guess sip is at 5060, but what rtp ports does it use? i need to open up ports on my router |
| 20:37.24 | [Airwolf] | trumee, 10000-20000 udp |
| 20:37.39 | *** join/#asterisk kpanik (n=spike@ip-100-55.sn2.eutelia.it) |
| 20:38.17 | trumee | [Airwolf]:is there any conf file where this is specified? |
| 20:38.33 | [Airwolf] | rtp.conf |
| 20:38.36 | [Airwolf] | :) |
| 20:38.40 | kpanik | re |
| 20:38.47 | trumee | is iax2 better than sip generally? |
| 20:39.04 | ManxPower | trumee: Define "better" |
| 20:39.10 | kpanik | any suggestion for zttest ? |
| 20:39.19 | trumee | ManxPower:umm, voice quality. NAT traversal etc |
| 20:39.44 | trumee | ManxPower: my ISP gives me an option between sip/iax2. i was wondering which one should i use? |
| 20:40.00 | ManxPower | trumee: voicequality: no difference. NAT traversal: IAX2 should be a little better. Hardware support: SIP all the way. |
| 20:40.13 | kpanik | I've a te410p on server with mainboard where don't setting IRQ in the BIOS |
| 20:40.36 | ManxPower | kpanik: is the card sharing IRQs ( cat /proc/interrupts ) |
| 20:40.44 | trumee | ManxPower:what about bandwidth usage, do they require same sort of bandwidth? |
| 20:41.10 | aptura | kpanik buy a motherboard that supports irq assignments |
| 20:41.12 | ManxPower | trumee: for more than 2 calls between the same two servers, IAX2 w/trunking will use less bandwidth. |
| 20:41.16 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:41.23 | ManxPower | outside of that setup, there is not a major difference |
| 20:41.30 | RoyK | nite |
| 20:42.13 | trumee | ManxPower:i am buying a PAP2T. and my plan is to use DECT phone<>PAP2T<>SIP<>ASTERISK BOX<>IAX2<>ISP. hopefully it should work. i am just worried about QOS |
| 20:43.01 | kpanik | <PROTECTED> |
| 20:43.01 | kpanik | <PROTECTED> |
| 20:43.01 | kpanik | <PROTECTED> |
| 20:43.01 | kpanik | <PROTECTED> |
| 20:43.01 | kpanik | <PROTECTED> |
| 20:43.01 | kpanik | <PROTECTED> |
| 20:43.01 | trumee | ManxPower:i have a WRT54GL using dd-wrt firmware. i am looking for best QOS settings. |
| 20:43.03 | kpanik | <PROTECTED> |
| 20:43.05 | kpanik | <PROTECTED> |
| 20:43.07 | kpanik | <PROTECTED> |
| 20:43.09 | kpanik | NMI: 0 0 0 0 |
| 20:43.11 | kpanik | LOC: 747467 747466 747465 747464 |
| 20:43.13 | kpanik | ERR: 0 |
| 20:43.15 | kpanik | MIS: 0 |
| 20:43.15 | ManxPower | ~pb |
| 20:43.17 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 20:43.36 | ManxPower | kpanik: your card is not hsaring an IRQ. good. |
| 20:44.25 | kpanik | yes but i give command lspci -vb i see irq 3 for my card and a controller e7520 DMA |
| 20:44.44 | kpanik | my zttest is very poor |
| 20:44.58 | kpanik | 93% 95% 97% max |
| 20:47.15 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 20:47.48 | ManxPower | kpanik: lspci is not correct for systems using ACPI |
| 20:48.03 | ManxPower | kpanik: disable the onboard ethernet devices and see if your zttest improves |
| 20:49.19 | kpanik | i've kernel 2.6.18 and zaptel driver 2.1.9.1 |
| 20:50.02 | kpanik | with the kernel version 2.6.17 and zaptel 1.2.7 the result of zttest is more good... |
| 20:54.04 | [Airwolf] | Can someone tell me if I use a port number for a sip registration and the port= setting why Asterisk still uses 5060 ? |
| 20:55.48 | ManxPower | [Airwolf]: because Asterisk still expects to receive signaling information on local port 5060 udp. |
| 20:56.03 | ManxPower | the port= option is pretty useless most of the time, and would refer to the source port from the FAR side. |
| 20:56.08 | *** join/#asterisk freebsd_fan (n=ebola@hdkbib3.hdk.gu.se) |
| 20:56.29 | macTijn | hmm |
| 20:56.31 | [Airwolf] | ManxPower, ok, so how can I let Asterisk listen on another port for just one connection ? |
| 20:56.43 | ManxPower | [Airwolf]: I don't know if you can. |
| 20:56.57 | macTijn | are there any known problems with rxfax and Asterisk 1.2.7.1 ? |
| 20:57.09 | ManxPower | [Airwolf]: your extensive search of the maling lists archives was not helpful? |
| 20:57.47 | ManxPower | macTijn: Results 1 - 10 of about 89 from lists.digium.com for 1.2.7.1 problem |
| 20:57.49 | [Airwolf] | I didn;t find anything relative. But I was looking for thing related to the exact provider. |
| 20:59.09 | macTijn | when using rxfax |
| 20:59.17 | macTijn | over an E1 |
| 20:59.37 | macTijn | my faxes keep getting cut in half |
| 21:00.28 | macTijn | and my AGI script that's supposed to mail me the fax isn't getting executed |
| 21:01.31 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 21:02.57 | ManxPower | macTijn: it sucks to be you. |
| 21:03.19 | [Airwolf] | ManxPower, I really love your answers sometimes :P |
| 21:04.39 | Vetaly | GOOD night! |
| 21:09.05 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 21:09.13 | aptura | seems the linksys wip300 and 330 have been given some decent reviews over other wifi voip phones. |
| 21:09.30 | aptura | anyone experainced with these phones care to comment? |
| 21:12.22 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:15.40 | kpanik | I'm sorry it's important in the kernel the RTC ? |
| 21:17.16 | hmmhesays | the wip300 is ok |
| 21:17.18 | hmmhesays | not great |
| 21:17.19 | hmmhesays | but ok |
| 21:23.07 | aptura | hmmhesays what phones have you tested |
| 21:24.16 | *** join/#asterisk freebsd_fan (n=ebola@200.32.71.142) |
| 21:27.36 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 21:37.26 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 21:41.01 | *** join/#asterisk ToTo (n=ToTo@host138-138-dynamic.2-87-r.retail.telecomitalia.it) |
| 21:44.59 | hmmhesays | wip-300 |
| 21:45.02 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 21:46.49 | *** part/#asterisk ManxPower (n=eric@71-8-11-111.dhcp.leds.al.charter.com) |
| 21:48.26 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 21:51.48 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 21:54.11 | *** part/#asterisk roo9 (i=roo9@is-god.net) |
| 21:55.42 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:56.23 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.97) |
| 21:57.17 | kpanik | what is the best kernel configuration and version for te410p and zttest good result ?? tnx in advance |
| 21:57.57 | dlynes_laptop | kpanik: 2.4 or higher |
| 21:58.05 | kpanik | yes |
| 21:58.22 | kpanik | i've 2.6.18 but my zttest is very poor |
| 21:58.35 | dlynes_laptop | kpanik: I personally use 2.6, but that's so I can use low latency optimizations |
| 21:59.25 | kpanik | with 2.6.17 litte more better |
| 21:59.48 | kpanik | yes with setpci ?? |
| 22:00.02 | dlynes_laptop | I use 2.6.15.5, myself |
| 22:00.37 | Juggie | i use 2.6.9-34.0.1.ELsmp w/ good success |
| 22:00.38 | dlynes_laptop | However, if you're getting poor results with zttest, it could be your motherboard, interrupt conflicts, or a myriad of other problems |
| 22:02.08 | kpanik | I've not sharing irq |
| 22:03.43 | dlynes_laptop | kpanik: are you using dma on your hard drives? |
| 22:03.57 | *** join/#asterisk Mike800 (n=mike800@pool-71-160-122-28.lsanca.dsl-w.verizon.net) |
| 22:04.02 | dlynes_laptop | kpanik: are you using ACPI, or APIC? |
| 22:04.16 | kpanik | I use sata disk |
| 22:04.30 | kpanik | ACPI and APIC |
| 22:04.37 | kpanik | ACPI |
| 22:04.40 | kpanik | sorry |
| 22:04.41 | dlynes_laptop | disable ACPI and APIC |
| 22:04.52 | dlynes_laptop | They both cause problems with Digium hardware |
| 22:05.24 | kpanik | disable in the kernel ? |
| 22:05.24 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:05.45 | dlynes_laptop | kpanik: in your lilo startup, you can do apic=no, acpi=no |
| 22:06.37 | sbingner | dlynes_laptop: do they always cause issues? mine seems ok with it on, in fact enabling it made it stop crashing from some sort of IRQ conflicts on another system |
| 22:07.07 | Mike800 | Anyone wanna see if they can help me with a problem I'm having with the Page() application? |
| 22:07.33 | dlynes_laptop | sbingner: no, they don't always cause issues |
| 22:07.36 | *** join/#asterisk frawd (n=francois@129.Red-81-33-44.dynamicIP.rima-tde.net) |
| 22:07.56 | dlynes_laptop | sbingner: but disabling everything, and working back from there is usually the best route to resolving an issue |
| 22:08.17 | kpanik | I try and put the result |
| 22:08.22 | kpanik | thanks |
| 22:08.28 | Mike800 | I have a topic on the forums, but no one knows how to fix the problem: http://forums.digium.com/viewtopic.php?p=32346#32346 |
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| 22:11.46 | Mike800 | anyone? |
| 22:13.39 | dlynes_laptop | Mike800: could it be a problem with those specific phones? maybe the sipura 941 doesn't do it properly? |
| 22:14.19 | sbingner | dlynes_laptop: there are about 3 people on that thread that say they have similar issues btw |
| 22:14.35 | frawd | Hi everyone, i have an issue on zap FXO lines (with an Openvox A400P), everytime Asterisk establishes an outgoing call, i hear some kind of pulse that cuts the first second of the call (can't hear when the person says "hi" or something directly). It doesn't appear to be related to the echo training that mutes the channel during 400ms (i tried without), but i have no clue what it could be... |
| 22:15.05 | Mike800 | dlynes_laptop: i dont think so. The problem is from asterisk, because its Asterisk that hangs up on the call. I tried using a softphone that autoanswers, and the same problem |
| 22:15.12 | frawd | anyone experienced this? |
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| 22:16.49 | jtexter3 | Mike800: FWIW, I've got a customer that uses the Page application CONSTANTLY on Polycom 501's without issue |
| 22:17.20 | Mike800 | hmm |
| 22:17.28 | jtexter3 | That being said, I'm only using the half-duplex option |
| 22:17.38 | Mike800 | i've tried both half-duplex, and full |
| 22:18.07 | Mike800 | im confident the problem comes from Asterisk, because asterisk is the one that hangs up the call. The phones answer and the page is broadcasted for 4 seconds |
| 22:18.42 | sbingner | Mike800: but you said it doesn't work at all when half-duplex? |
| 22:18.59 | Mike800 | no, it works for 4 seconds, then the audio cuts out |
| 22:19.37 | sbingner | Quote: "My problem is this: When users dial *0, all the availible SIP phones are added to a MeetMe, as intended by the Page application. When the page is initiated, users hear the "beep" and their Linksys SPA942's pickup. After this beep, however, they hear no sound. " <-- that got fixed somewhere along the line? |
| 22:19.50 | jtexter3 | Mike800: What kernel are you using? And do you have any hardware in the box, i.e. a T1/E1 card? |
| 22:20.22 | Mike800 | i think i meant to write that it works for 4 seconds |
| 22:20.34 | Mike800 | I have a tdm2400 with 8 fxo and 4 fxs |
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| 22:21.36 | sbingner | have you tried paging to only Zap interfaces? |
| 22:21.45 | Mike800 | im using kernel 2.6.17-1.2174_FC5smp |
| 22:22.16 | Mike800 | I dont have any Zap interfaces connected to it currently |
| 22:22.34 | jtexter3 | Mike800: It sounds like a MeetMe issue you would have if you didn't have a timing source. Have you tried loading the ztdummy module? |
| 22:23.15 | sbingner | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.997965 <-- his results from zttest for 18 passes |
| 22:23.16 | Mike800 | thats what I thought, but if music on hold works, and I can successfully have a meetme conference with multiple people in it, then would it still be a timing issue? |
| 22:28.00 | Mike800 | any ideas? |
| 22:29.37 | *** join/#asterisk edwar64896 (n=medwards@72.83.233.220.exetel.com.au) |
| 22:30.34 | edwar64896 | ping |
| 22:30.45 | edwar64896 | ~ping |
| 22:30.47 | jbot | pong |
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| 22:32.43 | DrukenHME | uhm... Mike800, you said you have a tdm2400, but you also said you don't have any zaptel devices... so which is it? |
| 22:33.05 | Mike800 | no...I meant I have nothing connected to the 2400 |
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| 22:33.29 | DrukenHME | shouldn't matter |
| 22:33.33 | Mike800 | i know |
| 22:34.33 | dsrtZrzmra | anybody using E1's with MFC / R2 ? |
| 22:35.45 | Mike800 | do you think its a bug in asterisk? |
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| 22:46.07 | DrukenHME | Oct 1 18:39:31 WARNING[31433]: channel.c:2752 ast_channel_make_compatible: No path to translate from SIP/lesnet-081d7750(256) to SIP/101-081d2210(4) |
| 22:46.10 | DrukenHME | anyone? |
| 22:46.28 | edwar64896 | yeah. |
| 22:46.33 | edwar64896 | incompatible codecs. |
| 22:46.37 | hads | DrukenHME: Asterisk can't transcode between your codecs. |
| 22:46.46 | edwar64896 | looks like you are trying to use g729 and don't have it installed |
| 22:46.59 | edwar64896 | or registered |
| 22:47.25 | DrukenHME | nope... don't have g729 on that box.... maybe the provider won't do anything but g729.... |
| 22:47.35 | hads | That's a posibility. |
| 22:48.23 | edwar64896 | definitely g729 problem. do a show codecs in your CLI |
| 22:48.42 | edwar64896 | you might be able to get them to change to ulaw or ilbc or something.... |
| 22:49.05 | hads | edwar64896: Not nessecarily. It could be due to the codec setup in sip.conf |
| 22:49.31 | edwar64896 | its past that. |
| 22:49.41 | edwar64896 | otherwise call setup would not occur |
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| 22:49.52 | sbingner | Mike800: try plugging in some zap devices and see if it works to them |
| 22:50.10 | Mike800 | sbingner: ok, any reason why that might work? |
| 22:50.28 | sbingner | no, but it would tell you it's some sort of sip issue with page if it did I expect |
| 22:50.52 | sbingner | and if it doesn't it'll remove SIP from the list of possible causes |
| 22:51.26 | hads | edwar64896: You can reproduce that symptom by using allow/disallow lines in sip.conf |
| 22:52.10 | Mike800 | sbingner: So you're saying, if I connect zap devices, and it works, then its a Zap issue. But if I connect Zap devices, and it still doesnt work, its a SIP issue? |
| 22:52.37 | Mike800 | i feel like its probably some random option I have enabled that is causing it to conflict |
| 22:52.40 | sbingner | Mike800: I'm saying if you connect zap devices and it works, it's a sip/page issue... if it doesn't it's NOT SIP related issue |
| 22:52.55 | Mike800 | ohh..ok |
| 22:53.22 | Mike800 | If this doesn't work, who can I contact regarding this problem? |
| 22:53.36 | sbingner | I'd suggest digium :b |
| 22:53.47 | Mike800 | ya..i was gonna call them this week |
| 22:53.51 | edwar64896 | DrunkenHME: do a show translation. do you have g729 installed? |
| 22:54.37 | sbingner | Mike800: of course people in here know * pretty well... I've been out of it for a while tho so don't trust me too much ;) |
| 22:54.52 | sbingner | usually somebody from digium in here during the week in my experience |
| 22:55.30 | Mike800 | ok |
| 22:55.31 | edwar64896 | Mike800: sorry - missed the earlier part of your conv. I have TDM2400 too - what's the issue? |
| 22:55.35 | Mike800 | well....i'll keep trying |
| 22:56.19 | Mike800 | edwar64896: http://forums.digium.com/viewtopic.php?p=32880#32880 |
| 22:56.24 | edwar64896 | ta. |
| 22:56.36 | Mike800 | :-) |
| 22:57.16 | sbingner | edwar64896: in a nutshell; page doesnt work after 4 seconds, meetme or regular calls do |
| 22:57.27 | edwar64896 | ok. |
| 22:57.30 | edwar64896 | reading... |
| 22:58.07 | Mike800 | its such a crappy problem. There is no good reason why this is happening |
| 23:06.09 | kpanik | ehmmm ... I try a insert into lilo.conf noapic and acpi=off, but the result it is gotten worse ... 99.93 |
| 23:06.09 | *** join/#asterisk _deg_ (n=deg@201-15-254-2.ctame704.dsl.brasiltelecom.net.br) |
| 23:07.52 | kpanik | any suggestion ? |
| 23:10.03 | *** join/#asterisk eidolon (n=eido@pdpc/supporter/monthlybronze/eidolon) |
| 23:10.51 | eidolon | hi folks, looking for a recommendation for the best free (or very cheap) softphone for the PC. (winXP to be precise). Been using Twinkle under Linux, hich is okay, but Linux alsa audio drivers, particularly for mic input, are terrible. So I'm changing to use my windows box as my softphone host. |
| 23:13.19 | Nivex | eidolon: you might be able to snag a copy of X-Ten's X-Lite. |
| 23:13.30 | eidolon | hmm |
| 23:13.32 | Mike800 | Ya...x-lite is pretty good |
| 23:14.28 | eidolon | hmmmmmmm. |
| 23:16.19 | eidolon | MAN i hate windows. |
| 23:16.27 | eidolon | "Oo, you just installed a simple VOIP client. REBOOT!" |
| 23:16.29 | eidolon | *clunk* |
| 23:16.45 | Mike800 | speaking of windows... Anyone know how to increase the buffer size in Putty? |
| 23:17.00 | eidolon | like, scrollback buffer? |
| 23:17.01 | Mike800 | so that i can scroll up A LOT and see more history |
| 23:17.03 | eidolon | ah. |
| 23:17.03 | Mike800 | ya |
| 23:17.14 | eidolon | dunno. been a while since i used it seriously. we have a securecrt license :) |
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| 23:17.26 | Mike800 | hehe |
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| 23:25.43 | Mike800 | Anyone figure anything out regarding the Page() issue? |
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| 23:34.17 | Flauto | anyone can tell me what is responsetimeout function now is what |
| 23:35.10 | Flauto | is it something like set(timeout(response)=) |
| 23:39.55 | edwar64896 | eidolon: change settings / window |
| 23:39.58 | edwar64896 | you'll be able to do it there. |
| 23:41.05 | edwar64896 | Mike800: sorry - I have not played with page and this maybe beyond me with out some serious experimentation... |
| 23:41.34 | Mike800 | edwar64896: well thanks for trying...i |
| 23:41.41 | Mike800 | i'm gonna call digium this week |
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| 23:42.15 | kink0 | hello |
| 23:43.50 | edwar64896 | hello |
| 23:44.36 | edwar64896 | Flauto: I think responsetimeout is deprecated now |
| 23:44.52 | edwar64896 | timeout(response) is the way to go I think |
| 23:47.24 | Flauto | what is the correct one then |
| 23:47.46 | Flauto | are you guys using gtalk? |
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