| 00:05.10 | *** join/#asterisk Akiraaa (n=Akiraa@79.112.31.181) |
| 00:05.40 | Akiraaa | Is it possible for Asterisk to auto-detect VoIP devices connected to the same LAN? |
| 00:05.50 | Akiraaa | Or to detect devices based on their individual IPs |
| 00:06.07 | p3nguin | It only detects what is sent to it. |
| 00:07.30 | Akiraaa | Having trouble configuring a hardware VoIP phone (could be non-working, actually), whereas it's painless to connect softphones like SJphone or X-Lite... Do you recommend a hardware device that you use yourself? |
| 00:08.00 | p3nguin | I use Cisco 7900 series hardphones, personally. |
| 00:08.53 | voipmonk | anyone selling an old 2g iphone? |
| 00:09.26 | voipmonk | Akiraaa: polycom, linksys, cisco :) |
| 00:10.03 | Akiraaa | May have gotten burned on some cheap Chinese crap |
| 00:14.34 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 00:16.23 | tzafrir__laptop | Akiraaa, do you have control over the DHCP server in the network? |
| 00:16.41 | Akiraaa | tzafrir_laptop |
| 00:16.47 | Akiraaa | tzafrir_laptop |
| 00:16.55 | Akiraaa | yes |
| 00:18.26 | Akiraaa | tzafrir__laptop: the problem is that the devices can be configured (through a web interface), but can't register to the local asterisk PBX, whereas the same settings work for a variety of equally configured softphones |
| 00:19.34 | tzafrir__laptop | One option is to provide them configuration through tftp/ftp/http |
| 00:20.04 | p3nguin | You've probably missed a setting that is required. |
| 00:20.21 | p3nguin | Maybe a proxy setting in addition to a registrar setting, for example. |
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| 00:33.57 | *** mode/#asterisk [+o mog] by ChanServ |
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| 01:20.57 | Rabenklaue | hi, I'm using asterisk 1.6.1.12 with zaphfc dahdi drivers. |
| 01:21.17 | Rabenklaue | I always get those messages in syslog: |
| 01:21.17 | Rabenklaue | [569136.296731] zaphfc[0]: b channel buffer underrun: 0, 0 |
| 01:21.17 | Rabenklaue | [569143.259781] zaphfc[0]: b channel buffer underrun: 1, 1 |
| 01:21.17 | Rabenklaue | [569143.259801] zaphfc[0]: b channel buffer overflow: 29, 29 |
| 01:22.34 | Rabenklaue | And with those (I think IRQ problems) my whole system is stucking (I'm also using a wifi card in master mode built inside my server, and with zaphfc loaded it is nearly unusable because both (HFC and WIFI are PCI cards)). |
| 01:22.44 | Rabenklaue | Does anyone know how I could fix my problem= |
| 01:22.44 | Rabenklaue | ? |
| 01:23.07 | florz | Rabenklaue: you know my patch? |
| 01:23.13 | Rabenklaue | my HFC card: 02:09.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 01:23.47 | Rabenklaue | I've read sth about the florz patch, but didn't applied it,I think |
| 01:24.11 | Rabenklaue | I'm just using the dahdi-linux-2.2.0.2_zaphfc patch to use my HFC card with dahdi |
| 01:24.23 | florz | it could help, if not directly, then there is an undocumented parameter in there that probably helps |
| 01:25.11 | florz | however, I'm not really sure how well it does apply on current versions of everything ... probably the one included with bristuff is the safest bet in that regard |
| 01:25.38 | Rabenklaue | I thought bri(stuff) is included within asterisk 1.6, or am I wrong? |
| 01:26.21 | florz | bristuff certainly not, otherwise I don't know |
| 01:26.24 | Rabenklaue | On http://zaphfc.florz.dyndns.org/ I only see a way patching the common zaphfc driver, not the one shipped with dahdi (with the patch mentioned above applied) |
| 01:26.57 | florz | they named it the same? sorry, I'm not on top of those things anymore |
| 01:27.55 | florz | but the zaphfc driver from kapejod as well as my patch for it are GPL, so they probably are not being distributed by digium |
| 01:29.44 | florz | what I can tell you is that that driver with my patch should be able to handle those buffer overflows and underruns, but no clue whether it would work with * 1.6 |
| 01:32.03 | Rabenklaue | I just loaded zaphfc driver with timer_card=1 |
| 01:32.27 | Rabenklaue | And actually (at least the last 30 seconds) there are no messages in syslog any more. |
| 01:32.57 | Rabenklaue | Those messages firstly appeared (as far I can say) after I plugged in my wifi card. |
| 01:33.53 | Rabenklaue | So perhaps there is something wrong with the timer settings, although I have no idea how this could happen |
| 01:34.29 | Rabenklaue | Thanks a lot anyway. If my problem persists, I'll try to apply your patch. |
| 01:36.41 | Rabenklaue | Well, I even found a dahdi compliant florz patch at: http://www.solidboot.com/~fabled/zaphfc-dahdi-flortz.diff |
| 01:41.49 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
| 01:41.49 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:42.05 | Rabenklaue | I see this is the one I posted above (dahdi-linux-2.2.0.2_zaphfc). |
| 01:42.10 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
| 01:42.10 | *** mode/#asterisk [+o denon] by ChanServ |
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| 01:43.15 | wonderworld | Hey |
| 01:43.25 | wonderworld | Merry xmas |
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| 01:46.02 | *** mode/#asterisk [+o denon] by ChanServ |
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| 02:55.16 | Rabenklaue | Can anyone recommend a free administration frontend for asterisk (web based or common gui client) |
| 02:56.17 | [TK]D-Fender | ~freepbx |
| 02:56.19 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 02:56.20 | [TK]D-Fender | Rabenklaue: ^^^ |
| 02:58.18 | Rabenklaue | [TK]D-Fender: Thanks, I'll have a look on FreePBX, respectively the applications used there. |
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| 03:45.17 | Micc | I was just about to purchase the HP echo canceler, but I was reading part of it and noticed it says its only for use with dahdi hardware. Did I read something wrong? I thought it was just software echo cancelation that would work on our SIP/IAX trunks. |
| 03:45.53 | coppice | you can't echo cancel a SIP/IAX trunk |
| 03:51.13 | hardwire | coppice: you can if you run it through TDMoE or TDM local first. |
| 03:51.14 | hardwire | :P |
| 03:54.58 | coppice | that's not a SIP/IAX trunk |
| 03:55.36 | coppice | if you call someone across the PSTN and they are using VoIP the EC at your end can go really funky |
| 03:56.34 | hardwire | SIP <-> TDM Loopback <-> IAX |
| 03:57.19 | hardwire | if you for some reason need to deal with EC on behalf of your SIP or IAX PSTN provider. |
| 03:57.36 | coppice | oh, you can set up a whole bunch of weird configurations, but only a limited subset of those provide useful functionality |
| 03:57.38 | hardwire | that would help, but it could become rather funky. |
| 03:57.54 | hardwire | at least if you were on a high jitter system |
| 03:57.58 | Nivex | play that funky music! |
| 03:58.00 | hardwire | err.. scenario |
| 03:58.00 | coppice | you simply *cannot* EC for your provider. it just won't work |
| 03:59.00 | hardwire | has to EC on PRI often. |
| 03:59.22 | coppice | duh, yeah, that's where is belongs :-) |
| 03:59.37 | hardwire | I take it you don't see the similarity? |
| 04:00.02 | coppice | I take it you don't see the key differences? |
| 04:00.19 | hardwire | I'm not trying to be rude. but I can see situations where EC on VoIP may be desirable. |
| 04:01.03 | coppice | oh, there are lots of places it would be desirable. there are very very few where is will work |
| 04:01.34 | Micc | Yes, we need to echo cancel our IAX2 calls. |
| 04:01.40 | hardwire | I'm guessing mostly because of RTT. |
| 04:02.00 | hardwire | now if there were a longer buffer for echo cancellation it may become more viable. |
| 04:02.02 | Micc | But I do have access to the server that has the PRI, but its a really old version of asterisk 1.2 |
| 04:02.28 | Micc | Maybe I can turn it on in the config over there. |
| 04:03.03 | hardwire | heh, well that would be a higher quality solution. |
| 04:04.01 | hardwire | coppice: if it's only a 20ms or so and low jitter between IAX and the provider. That would work out well. |
| 04:04.06 | coppice | if you have a path which is purely A-law or u-law *and* the path has an absolutely rock solid length, with not jitter buffer juggling by anyone in the path, *and* the packet loss rate is extremely low, *and* an missing packet time is accurately filled, then you can echo cancel over IP. How often are all four conditions properly fulfilled |
| 04:04.27 | hardwire | packet loss doesn't matter |
| 04:04.46 | coppice | why would that be? |
| 04:04.54 | hardwire | because if there is no audio. there is no echo. |
| 04:05.08 | coppice | huh? |
| 04:05.09 | hardwire | and echo cancellation doesn't need to be 100% accurate. |
| 04:05.43 | hardwire | once you loopback through TDM any loss is filled with silence. |
| 04:06.15 | hardwire | if what you were explaining were 100% the truth and the final argument.. we couldn't echo cancel on a PRI that is connecting to cellular phones. |
| 04:06.20 | coppice | in the best case just after each lost packet you'll get a rude noise. However, most systems don't fill the missing time accurately and the EC falls apart |
| 04:06.54 | coppice | hardwire: you can't you rely on the cellular system being very thorough at removing echo |
| 04:07.05 | hardwire | eh? |
| 04:07.16 | hardwire | -> food |
| 04:07.25 | Nivex | embrace the echo the echo the echo the echo |
| 04:07.56 | coppice | the cellular networks have very carefully designed EC arrangements to remove all echo within their network, because it simply can't be handled beyond their network |
| 04:09.51 | ChannelZ | Speaking of echo, I have a TDM card; Client calls in, I answer my SIP phone, and hear myself echoing for usually the first 15-20 seconds of the conversation. I'm *pretty* sure the calls that echo are from the remote end calling in on their own interally VoIP system which causes extra delay |
| 04:11.53 | ChannelZ | is the general idea to turn up the taps to hopefully get it to train faster, or is that then going to make other 'normal' calls (people calling in on their analog phones) echo? |
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| 04:58.29 | dlynes | ChannelZ: you only hear it for the first 15-20 s of the conversation, and after that it disappears? |
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| 05:33.56 | ChannelZ | yeah it slowly disappears |
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| 07:15.32 | jblack | On http://www.overpill.com/2009/12/21/soviet-scientist-turns-foxes-into-puppies/ I see "THIS SPACE LEFT INTENTIONALLY UGLY CUZ U BLOCKED R ADS BITCH" |
| 07:30.28 | p3nguin | What mechanism are you using to block their ads? |
| 07:58.26 | ChannelZ | Adblock Plus here.. googlesyndication.com/pagead/* blocked :) |
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| 08:06.57 | tschi | hallo |
| 08:07.16 | tschi | Ich briache bitte Hilfe mit dahdi |
| 08:12.13 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 08:13.42 | tschi | Hallo |
| 08:14.57 | tschi | kennt sich wer mit Asterisk 1.6 und DAHDI aus? Karte wird schon erkannt, aber wie mache ich das im Wählplan? |
| 08:19.19 | tschi | Hello |
| 08:19.38 | tschi | i need help with asterisk and dahdi |
| 08:22.16 | florz | then you are wrong here |
| 08:22.30 | florz | this channel is about wood and rainfall |
| 08:22.39 | fenrus | :) |
| 08:22.49 | fenrus | well, what seems to be the problem with dahdi ? |
| 08:24.44 | tschi | i hava a TDM800 analog card running with asterisk 1.6 |
| 08:25.36 | tschi | when i pick up the phone i get following message in asterisk: |
| 08:25.51 | fenrus | if long, use pastebin! |
| 08:26.04 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 08:26.07 | tschi | Starting simple switch on 'DAHDI/2-1' |
| 08:26.43 | tschi | after pressing a number the phone signals busy |
| 08:28.05 | tschi | at DAHDI/1-1 and DAHDI/2-1 is a phone installed |
| 08:28.55 | tschi | now the question: where can i define the extensions for this phones? |
| 08:33.12 | fenrus | in chan_dahdi.conf i guess |
| 08:33.28 | fenrus | and with extensions.conf |
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| 08:37.20 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 08:38.15 | shamelessn00b | hi guys |
| 08:39.33 | ManxPower | ~answers |
| 08:39.51 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 08:39.51 | ManxPower | ~book |
| 08:39.52 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 09:32.24 | Akiraa | Does "SIP-5002" mean anything? |
| 09:32.52 | ChannelZ | not without context no |
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| 09:34.35 | Akiraa | Has anyone worked with Wuchan brand phones? (PH 802) http://www.5111soft.com/productview.php?id=1&pid=7 |
| 09:35.53 | ChannelZ | hmm never even heard of em |
| 09:50.16 | shamelessn00b | hey! ChannelZ |
| 09:50.43 | fenrus | channelz? ;) |
| 09:50.59 | fenrus | Akiraa, nah, never - looks a bit cheap dont you think |
| 09:52.59 | coppice | what IP phone doesn't look fairly nasty? |
| 09:53.18 | fenrus | i like the simpleness of the 7911 |
| 09:53.21 | fenrus | (cisco) |
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| 10:10.06 | benngard | avaya 9650? |
| 10:11.03 | fenrus | Our avaya phones at work are real ugly |
| 10:11.27 | fenrus | 1616 |
| 10:12.47 | benngard | are u running with sip or h.323 software? |
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| 10:14.31 | shamelessn00b | hey guys, is there a way I could analyze data transmitted by my PRI cards |
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| 10:19.33 | fenrus | you use some kind of PRI-payload-analyzer hardware |
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| 10:30.11 | Akiraa | Do you have systems in place that optimize outbound calls based on carrier and subscription plans? |
| 10:32.17 | Akiraa | This problem is compounded by new EU rules that allow subscribers to keep their phone numbers, while changing carriers. |
| 10:33.07 | Akiraa | Asterisk would have to do some signal analysis to detect a specific audio sequence that alerts someone of leaving the current network. |
| 10:38.08 | *** join/#asterisk NateHB (n=noone@72.34.90.74) |
| 10:38.16 | NateHB | any one awake? |
| 10:39.16 | fenrus | Akiraa, hm, interesting.. this will probably develop in a little while then |
| 10:40.56 | Tech_Travis | NateHB: yes. |
| 10:42.45 | NateHB | could you doube check a firewall rule for me, see if you can access port 80 of 72.34.90.75 |
| 10:43.17 | benngard | telnet 72.34.90.75 80 |
| 10:43.17 | Nugget | telnet is eeeeeeevil! |
| 10:43.18 | benngard | Trying 72.34.90.75.. |
| 10:43.28 | benngard | no connect :( |
| 10:43.35 | NateHB | good, thanks |
| 10:43.40 | benngard | np |
| 10:44.50 | NateHB | I had to get rid of our edgemark, ended up using a dd-wrt on a linksys router, seting one of the lan ports to the same vlan as the wan, and pluging my asterisk box diectly to the internet, gave it its own ip |
| 10:45.43 | NateHB | then I figured it probably be a good idea to drop any packets on ports 80,9000(openfire),9001(webmin) |
| 10:46.15 | NateHB | Any other ports you guys see that I should be blocking |
| 10:46.59 | NateHB | +question mark, that was supposed to be a question mark |
| 10:47.27 | Tech_Travis | NateHB: If it's only an * box shouldn't you drop every thing but 5060 UDP and whatever range your RTP is using? |
| 10:47.31 | NateHB | -mark, the last word was not good |
| 10:47.40 | fenrus | drop all except what you want to let in. |
| 10:47.47 | fenrus | allow established tcp sessions from the inside |
| 10:51.00 | benngard | NateHB: why do u have: 139/tcp open netbios-ssn |
| 10:51.00 | benngard | 445/tcp open microsoft-ds |
| 10:51.09 | benngard | :) |
| 10:51.35 | benngard | and: 6881/tcp filtered bittorrent-tracker |
| 10:52.24 | NateHB | no shit, ive got a bittorrent tracker running on there? |
| 10:52.33 | NateHB | or just the port is open? |
| 10:52.37 | benngard | :) |
| 10:52.56 | benngard | should i try to connect? ;) |
| 10:57.47 | NateHB | take a look again |
| 10:58.19 | NateHB | I just (if i did this correctly) set to drop all tcp traffic not coming from my own subnet |
| 11:00.11 | NateHB | thats a decent solution, correct? |
| 11:08.52 | shamelessn00b | this may sound newbish but meh, I've configured 2 independant sangoma cards, one as network other as cpe (4 ports each), now I'm placing calls from first card to the second card and I want to capture audio transmitted from the cards like you can do with SIP calls and analyze it in wireshark |
| 11:09.10 | shamelessn00b | see the delay/quality |
| 11:09.12 | shamelessn00b | etc |
| 11:09.56 | shamelessn00b | I captured some packets using wanpipemon from interface w1g1 at receiver and exported that file into wireshark |
| 11:10.17 | shamelessn00b | but it just shows me LAPD or Q.931 packets |
| 11:10.22 | shamelessn00b | no RTP packets |
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| 11:57.10 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 12:05.17 | coppice | RTP is a VoIP packet format. if you E1 or T1 is configured as a traditional PSTN voice circuit you won't see RTP. You will see LAPD and Q.931 on the signalling channel, and alaw on the other 30 channels (E1), or ulaw on the other 23 channels (T1) |
| 12:26.52 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 12:28.01 | hluesea | hello channel when i tried to install asterisk clean installed centos 5.4 i take a error messaje when tries installing dahdi |
| 12:28.22 | hluesea | like that ; error while loading shared libraries: libc.so.6: cannot open shared object file: No such file or directory |
| 12:28.48 | hluesea | is anyone to know the that issue ? I searched google but i do not find a correct solution |
| 12:28.53 | shamelessn00b | coppice: how can I capture traffic from channels other than the signalling channel |
| 12:30.04 | coppice | I think wanpipemon lets you snoop on the audio in any time slot, but I can't remember how |
| 12:30.25 | shamelessn00b | is there a wanpipemanual online |
| 12:30.31 | shamelessn00b | its not very well documented |
| 12:30.43 | shamelessn00b | hluesea: install a package that provides libc.so.6 |
| 12:31.14 | *** join/#asterisk d-k-t (n=D@112.202.232.46) |
| 12:31.20 | coppice | "not very well documented" describes everything from sangoma. Its great when its set up, but getting it set up...... |
| 12:31.29 | shamelessn00b | lol |
| 12:31.37 | shamelessn00b | it aint that hard |
| 12:31.40 | shamelessn00b | :D |
| 12:32.32 | coppice | maybe you've been lucky and stumbled on what you need quite quickly. if you have problems, trying to resolve them has been a huge pain for years |
| 12:32.56 | shamelessn00b | what problems? |
| 12:34.27 | coppice | for example, their config generator only generates a few common types. if youre requirement doesn't fit it produces something useless you need to edit by hand. now, try figuring out how you need to change it :-) |
| 12:43.42 | *** join/#asterisk rare1980 (n=rare@119.152.85.182) |
| 12:43.59 | rare1980 | hi |
| 12:44.02 | rare1980 | all |
| 12:44.06 | rare1980 | how are you today? |
| 12:46.34 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 13:01.49 | shamelessn00b | coppice: lol,yeah |
| 13:02.39 | shamelessn00b | fortunately, what I configured falls in the 'common' categories |
| 13:03.03 | shamelessn00b | hi rare1980 |
| 13:22.33 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 13:25.38 | mace | hi, i'm having troubles with iax behind nat. the server behind the nat is 1.6.2.0, with the other end being a commercial provider (voiptalk). some calls (incoming and outgoing) are dropped after around 30 seconds. watching the channels, these calls invariably remain in the 'ringing' state, even though two way audio is being passed. the timeout is no doubt due to the timeout on the dial() command which passes the call to a handset |
| 13:26.10 | mace | i've watched the firewall, and am not seeing any obvious packets being blocked when this occurs |
| 13:26.21 | mace | any ideas? |
| 13:32.32 | fenrus | 1.6.2.0 is not valid rfc1918 addresses. |
| 13:32.37 | fenrus | you should not use them. |
| 13:32.51 | mchou | mace: your description is confusing |
| 13:32.52 | mchou | lol |
| 13:33.34 | mchou | mace: is the called picked up? |
| 13:34.15 | mchou | mace: are you saying even though the call is picked up the "state" of the call is still reported as ringing? |
| 13:35.58 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 13:36.16 | mchou | fenrus: at least you've got a good sense of humor |
| 13:36.45 | fenrus | begin solving your asterisk problem, then redesign your network according to standards ,) |
| 13:36.49 | fenrus | ;) |
| 13:46.04 | *** join/#asterisk smooth_penguin (n=smoove@115.240.103.171) |
| 13:46.46 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) |
| 14:00.00 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 14:05.03 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 14:20.33 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
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| 15:06.03 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 15:07.31 | mace | fenrus: 1.6.2.0 is asterisk version, not ip address |
| 15:08.03 | mace | mchou: yeah, call picked up and two way audio, but asterisk reports as still 'ringing' |
| 15:11.13 | *** join/#asterisk rare1980 (n=rare@119.152.85.182) |
| 15:22.59 | fenrus | mace, :D :D :D |
| 15:25.46 | *** join/#asterisk voipmonk (n=shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 15:27.53 | mace | fenrus: yeah got the joke after posting that |
| 15:28.10 | mace | is somewhat fattened from lunch :) |
| 15:30.04 | *** join/#asterisk inkubus08 (n=inkubus0@c-71-207-46-164.hsd1.pa.comcast.net) |
| 15:32.42 | rare1980 | yeh |
| 15:32.45 | rare1980 | how are you all |
| 15:32.55 | rare1980 | mace or fenrus |
| 15:33.08 | rare1980 | how is going on so far? |
| 15:33.12 | fenrus | quite all right thankyou. |
| 15:34.02 | rare1980 | good |
| 15:34.09 | voipmonk | very well |
| 15:38.50 | mace | indeed |
| 15:45.51 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:51.49 | *** join/#asterisk silv3r_m00n (n=enlighte@59.93.166.52) |
| 15:51.53 | silv3r_m00n | hi there |
| 15:52.52 | silv3r_m00n | I wanted this ... a call comes in ... number is identified .....and a script runs using the number as input ........and came across asterisk .......i want some basic tutorial as to how to use asterisk |
| 15:54.04 | fenrus | use System() to execute for example shell-scripts outside asterisk |
| 15:54.20 | silv3r_m00n | where do I run that command ? |
| 15:54.26 | silv3r_m00n | in the terminal ? |
| 15:54.49 | fenrus | in the extensions-config. |
| 15:55.24 | silv3r_m00n | well... first I want to understand what asterisk is, what hardware if any it needs and what all can it do ... |
| 15:55.38 | fenrus | then go read :) |
| 15:55.56 | silv3r_m00n | looking for some basic tutorial which can explain with some examples |
| 15:55.57 | fenrus | www.asterisk.org |
| 15:56.21 | fenrus | have a look at for example |
| 15:56.22 | fenrus | 16:27:53 mace: fenrus: yeah got the joke after posting that |
| 15:56.24 | fenrus | err |
| 15:56.28 | fenrus | http://www.automated.it/guidetoasterisk.htm |
| 15:56.40 | fenrus | or perhaps http://www.the-asterisk-book.com/unstable/installation-1.4-debian-4.0.html |
| 15:56.58 | fenrus | all depends on what OS you run |
| 15:57.22 | silv3r_m00n | fenrus: tell me 1 thing.....my computer has dialup modem , so can asterisk do something with that ? |
| 15:57.29 | fenrus | nah |
| 15:57.41 | fenrus | asterisk requires no hardware if you only intend to use it with SIP / Softphones |
| 15:58.04 | fenrus | if you need to use a hardphone consider buying an ATA box or some SIP-phone |
| 15:58.55 | silv3r_m00n | what is ata / sip phone ? |
| 15:59.02 | fenrus | if your connection to the POTS/PSTN isnt via SIP you might want to consider getting a SIP-provider, or getting a some kind of card to use a pstn-line |
| 15:59.29 | fenrus | analouge telephone adapter, to connect an classic telephone to a sip-network |
| 15:59.42 | fenrus | an sip phone is phone that has the SIP-part built in ;) |
| 15:59.55 | silv3r_m00n | "card" ....what sort of |
| 16:00.19 | silv3r_m00n | sip means , session initiation protocol ? |
| 16:00.42 | fenrus | http://www.digium.com/en/products/analog/ |
| 16:01.02 | fenrus | Yes |
| 16:01.22 | fenrus | the signalling protocol |
| 16:01.41 | silv3r_m00n | what is the difference between connecting phone line via a dialup modem and via those digium cards ? |
| 16:02.12 | fenrus | with a digium card you will be able to call |
| 16:02.15 | fenrus | with a modem you wont. |
| 16:02.25 | *** join/#asterisk Alagar (n=Administ@122.164.37.150) |
| 16:03.13 | fenrus | silv3r_m00n, http://www.digium.com/en/docs/misc/fxs_fxo_desc.php |
| 16:03.20 | silv3r_m00n | hmm... can't call with a modem...but I remember seeing a software many years ago ...thru which a call cud be made via a dialup modem |
| 16:04.42 | silv3r_m00n | so these call centers...where they receive calls on a computer ... they use something like this ? |
| 16:05.05 | silv3r_m00n | or calls cannot be received via a dialup modem |
| 16:05.59 | fenrus | call centers often use hardphones with some computer software connected to it |
| 16:06.18 | silv3r_m00n | fenrus: does this >> http://www.authenticsolution.com/voice-logger.html have any similarity to the digium cards |
| 16:06.24 | silv3r_m00n | I mean is there any relation |
| 16:06.28 | fenrus | well, calls can be made and recieved with a dialup modem.. but they wont interface with asterisk |
| 16:07.16 | fenrus | i'm not that "in" telephony - dont know what that is.. :D |
| 16:07.47 | silv3r_m00n | what is a hardphone and softphone ? |
| 16:07.48 | fenrus | the digium cards gives you the cupper-connections for T1/E1/POTS.. |
| 16:07.58 | fenrus | a hardphone is a physical telephone |
| 16:08.10 | fenrus | a softphone is a software in a computer |
| 16:08.14 | silv3r_m00n | hmm |
| 16:09.59 | silv3r_m00n | so to handle incoming calls .... I need one of those cards ? |
| 16:11.35 | fenrus | to handle an analouge telephone OR an analouge telehpone-line you need one of those cards. |
| 16:11.56 | silv3r_m00n | hmm |
| 16:12.17 | fenrus | if you want to call by SIP you dont |
| 16:12.53 | silv3r_m00n | and can this be done ....that the telephone line is connected to the computer as well as a handset ....when a call comes .....the computer just checks the incoming number , but doesn't pickup the call and starts a program...then a user sees the program output and picks up the phone |
| 16:13.32 | fenrus | the card recieves the call, and can do "whatever" with it |
| 16:13.44 | fenrus | route it to a hardphone |
| 16:14.10 | silv3r_m00n | what is the technical term for those cards ? if I go to the market what do I ask for ? |
| 16:14.31 | fenrus | if you want one that works with asterisk, i'd go for a digium card |
| 16:15.02 | fenrus | http://voip.weblogsinc.com/2005/07/14/use-a-v92-modem-as-an-fxo-card-on-asterisk/ |
| 16:15.04 | silv3r_m00n | and what about non -digium cards? |
| 16:15.22 | fenrus | then you need to read up about what cards asterisk support |
| 16:15.23 | silv3r_m00n | and can calls be recorded with these cards ? |
| 16:15.35 | fenrus | the recording takes place inside asterisk |
| 16:15.43 | fenrus | and that's possible |
| 16:16.22 | fenrus | have a look at this: http://www.voip-info.org/tiki-index.php?page=X100P+clone |
| 16:17.13 | fenrus | to recieve calls from a analug telephone line you need a FXO-card |
| 16:22.03 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 16:26.45 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
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| 16:35.54 | *** part/#asterisk silv3r_m00n (n=enlighte@59.93.166.52) |
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| 17:12.46 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 17:13.25 | bcrisp | k, favorite xmas gifts received? |
| 17:38.15 | *** join/#asterisk MaliutaLap (n=biteme@96.53.148.53) |
| 17:52.24 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 18:01.43 | mchou | ~sipnat |
| 18:01.44 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:02.06 | benngard | any1 that has tested lastest sip release (2.5) for avaya handseats? |
| 18:02.52 | benngard | cant get "SET DISPLAY_NAME_NUMBER 1" to work :( |
| 18:03.12 | mace | mchou: just reproduced iax 'ringing' problem, will try again with wireshark running - if thats of any use? |
| 18:05.34 | mace | mchou: interesting, can't reproduce with wireshark running - beginning to suspect race condition |
| 18:06.09 | mchou | mace: you can always downgrade |
| 18:06.17 | mchou | mace: or just use sip |
| 18:07.20 | mace | sip + nat strikes me as an even larger world of pain |
| 18:09.27 | *** join/#asterisk dkirker-openmobl (n=dkirker@adsl-99-161-109-81.dsl.pltn13.sbcglobal.net) |
| 18:09.52 | mchou | mace: you have iax phones? |
| 18:10.12 | mace | mchou: just sip |
| 18:10.34 | mchou | then why would you care about iax vs. sip? |
| 18:11.23 | mace | because i've almost never had sip working through a firewall |
| 18:11.36 | mace | iax on the other hand almost always works |
| 18:12.01 | mace | suspect in this instance though i'm banging against the bleeding edge too much |
| 18:12.47 | mchou | sip works just fine across firewalls as long as you don't try streaming media directly between endpoints |
| 18:13.25 | mchou | i.e. across multiple nats |
| 18:14.00 | [TK]D-Fender | [13:11]<mace>because i've almost never had sip working through a firewall <- The reverse in my experience |
| 18:16.59 | *** join/#asterisk lost_soul (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 18:18.37 | mace | nods |
| 18:22.55 | *** join/#asterisk sebbl (n=Momofu@109.192.162.148) |
| 18:23.32 | benngard | sip server at publik ip and endpoints behind nat seems to work fine, atleast for me |
| 18:30.29 | sun28 | moin |
| 18:32.37 | [TK]D-Fender | plus |
| 18:48.59 | teknoprep | SIP is very easy to make work through a NAT |
| 18:49.03 | teknoprep | with asterisk |
| 18:49.15 | teknoprep | there are a few prereq's but its pretty simple |
| 18:49.32 | teknoprep | SIP ALG or any NAT helpers / Proxies should be turned off on the firewall |
| 18:49.49 | teknoprep | forward proper SIP 5060 ports and proper RTP ports |
| 18:50.24 | teknoprep | setup your "externip =" "localnet =" |
| 18:51.45 | teknoprep | asterisk makes life pretty easy on NAT'ing of SIP... its SIP Proxies that get difficult to NAT sometimes... Port Randomnization can be a bitch |
| 18:54.06 | wwalker | anyone know how to see the jitterbuffer settings? I've added them to sip.conf, but sip show settings doesn't show anything. |
| 18:56.37 | lost_soul | afternoon everyone |
| 19:05.39 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 19:10.20 | ChannelZ | yawns |
| 19:29.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:44.17 | fenrus | 8~ |
| 19:47.09 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 19:53.22 | *** join/#asterisk cesar_CR (i=cesar@201.201.41.242) |
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| 19:59.41 | *** join/#asterisk dshap (n=dshap@64.79.144.7) |
| 19:59.53 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 20:05.16 | dshap | I'm trying to write an iPhone app that can record and upload audio files for playback over a phone call on an asterisk server. Could someone please take a look at these available audio formats and let me know which one will be the best quality that Asterisk is capable of playing without converting the file? http://bit.ly/7mWvfQ |
| 20:05.52 | fenrus | ulaw is neat |
| 20:07.10 | ManxPower-work | Asterisk'd WAV49 format is GSM audio wrapped in a MS header. It is a decent compressed codec and will play on Windows without issues. |
| 20:07.50 | ManxPower-work | The standard .WAV format in Asterisk us ulaw wrapped in a MS header. |
| 20:07.56 | ManxPower-work | s/us/is |
| 20:08.10 | fenrus | "show file formats " in the asterisk CLI shows you the available formats |
| 20:12.59 | *** join/#asterisk dshap (n=dshap@64.79.144.7) |
| 20:13.04 | ChannelZ | Hmmm. * 1.6.1 complains: "app_voicemail.c:10656 load_config: maxsilence should be less than minmessage or you may get empty messages" |
| 20:13.07 | dshap | sry my IRC client froze |
| 20:13.23 | ChannelZ | Yet the dist config files don't have "minmessage", it's "minsecs". |
| 20:13.25 | dshap | i know how to see the avialble formats, i'm really just asking in terms of what will be the best quality |
| 20:14.06 | dshap | uLaw 2:1 is an available format to record on the iPhone |
| 20:14.09 | dshap | would you recommend that one? |
| 20:15.42 | tzafrir | dshap, is disk space an issue? |
| 20:16.00 | dshap | tzafrir: assuming it's not, what would you recommend? |
| 20:16.12 | tzafrir | Also: do you use wide-band anywhere on your system? |
| 20:17.12 | dshap | not sure about that one |
| 20:17.22 | dshap | (i'm pretty new to this stuff) |
| 20:17.43 | dshap | i really just want to play an audio clip that a user can record on their iPhone and upload to the server |
| 20:17.51 | dshap | and i'd like it to sound as good as possible over a phone call |
| 20:18.10 | ChannelZ | ulaw |
| 20:18.39 | ChannelZ | 16bit, 8kHz |
| 20:19.59 | dshap | hm |
| 20:20.03 | dshap | i know how to specify the sample rate |
| 20:20.13 | dshap | not sure about the # of bits |
| 20:20.17 | dshap | that's the # of bits per sample, right? |
| 20:20.42 | dshap | oh |
| 20:20.45 | dshap | bit depth maybe? |
| 20:20.58 | ChannelZ | yes |
| 20:21.27 | ManxPower-work | ChannelZ: reread UPGRADE*.txt and the .sample configs that came with your version of Asterisk. Option names change all the time, but should be documented in the UPGRADE*.txt files. |
| 20:21.43 | ChannelZ | I know that. I have the right option, but * tells me the wrong one. |
| 20:21.54 | dshap | ChannelZ: is ulaw considered a linear PCM audio format? |
| 20:22.05 | ChannelZ | sort of |
| 20:22.13 | ManxPower-work | ChannelZ: then you 1) have discovered a bug in Asterisk or 2) the sample files you are looking at are for a different version of Asterisk. |
| 20:22.58 | ChannelZ | It's an inconsistency. One of many |
| 20:23.00 | dshap | well according to Apple's documentation I can only set the bit depth of linear PCM audio formats |
| 20:23.02 | ManxPower-work | You're not looking at the .sample files on voip-info.og, are you? |
| 20:23.16 | ChannelZ | no, local from the distribution |
| 20:23.32 | ManxPower-work | Ah. Too bad. If you installed from source I might be able to help you. |
| 20:23.58 | ChannelZ | I don't need help |
| 20:24.02 | ChannelZ | And it IS from the source |
| 20:24.13 | ManxPower-work | You also can't file a bug against packaged Asterisks |
| 20:24.26 | ManxPower-work | ChannelZ: what does "show version" or "core show version" give you? |
| 20:24.47 | ChannelZ | It's just the code wasn't fully changed, one warning tells you 'maxsilence should be less than minmessage' but another tells you 'minmessage has been depreciated' |
| 20:25.04 | ChannelZ | 1.6.1.12 |
| 20:28.24 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 20:31.36 | ManxPower-work | ChannelZ: report it on issues.digium.com |
| 20:33.28 | ChannelZ | yes I am |
| 20:39.08 | tzafrir | well, you sure can file a bug against packaged asterisk. If there are maintainers to that package |
| 20:41.34 | ChannelZ | (re: mine is not packaged, it was built by me from there source. The default configs I keep in a 'dist' directory inside /etc/asterisk for reference, which is what I meant by 'local distribution') |
| 20:41.52 | *** join/#asterisk dshap (n=dshap@64.79.144.7) |
| 20:48.53 | ManxPower-work | looks like 1.6.12 fixed some "errors" in 1.6.1.8 that I was relying on for my scripts. |
| 20:51.14 | [TK]D-Fender | 1.6.12 z0mg TIME TRAVEL! |
| 20:51.16 | imcdona | I am setting up NFAS on two pri's. I am able to make and receive calls ok but there is no audio. Does anyone have any tips? |
| 20:52.06 | [TK]D-Fender | imcdona: tip : provide complete details on your calls |
| 20:52.34 | imcdona | TK what do you mean? |
| 20:52.42 | imcdona | you want to see pastebin? |
| 20:53.03 | imcdona | moment |
| 20:58.13 | imcdona | here is the call: http://pastebin.com/m38ed9f95 |
| 20:58.58 | imcdona | I have been fiddling with the configs now I can make calls with no audio, and asterisk not answer incoming calls |
| 21:01.16 | imcdona | this is the zapata.conf: http://pastebin.com/m5d90e85f |
| 21:01.37 | [TK]D-Fender | imcdona: What makes you think I trust your SIP device> call IN to your PRI |
| 21:02.13 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 21:02.43 | *** join/#asterisk voipmonk (n=shido6@67.204.52.26) |
| 21:03.40 | [TK]D-Fender | imcdona: And your configs don't look like NFAS.. you skip right over ch 24 |
| 21:03.51 | imcdona | TK. I hear you. I have ruled out the SIP devices....Here is an inbound call: http://pastebin.com/m340e7a3d |
| 21:05.33 | ChannelZ | Anyone running 1.6.2 want to try a quickie for me |
| 21:06.06 | ChannelZ | or actually any random people running any version.. this could be platform-related |
| 21:06.42 | imcdona | What do you mean skip right over? |
| 21:07.59 | imcdona | I have the trunkgroups and spanmap setup in zapata.conf for an NFAS config. I am confused |
| 21:08.33 | [TK]D-Fender | imacFirst... d-chan and spanmap stuff is for ZAPTEL, not ZAPATA |
| 21:09.11 | [TK]D-Fender | imcdona: second : channel => 1-23 channel => 25-47 <- you really don't seem to have a clue what you're doing on the other end |
| 21:09.13 | imcdona | sorry...I meant zaptel |
| 21:09.21 | [TK]D-Fender | imcdona: You skip channel 24 which is the POITN of NFAS. |
| 21:09.55 | [TK]D-Fender | imcdona: And then you also failed to show me your zaptel.conf |
| 21:10.12 | imcdona | ahh! I see...so in nfas I specify all the channels instead of just the b channels? |
| 21:10.32 | [TK]D-Fender | imcdona: I also have no clue what is in that IVRo of your and its a flood of GUI crap I have little faith in |
| 21:11.02 | imcdona | zaptel.conf: http://pastebin.com/m20cd7915 |
| 21:11.05 | [TK]D-Fender | imcdona: the point of NFAS is to have multiple PRI's share a d-chan so you can recover +1 B / extra PRI added |
| 21:11.13 | [TK]D-Fender | imcdona: its a waste most of the time |
| 21:11.37 | [TK]D-Fender | imcdona: you configured those as INDEPENDENT PRI'S |
| 21:12.02 | [TK]D-Fender | imcdona: Time to go read the docs again |
| 21:12.32 | imcdona | I skipped channel 24 because channel 24 is the primary d cahnnel and 48 is the backup d cahnnel this is only 2 pri's in an nfas config. ARe you saying I need to include channels 24 and 48 even thourh they are d channels? |
| 21:13.40 | imcdona | TK...no. I understand that the gains of NFAS are not apparent in this config. As in gaining extra b channels. The telco set it up this way to add more capacity rather than adding a rollover on the inbound did's to a second PRI |
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| 21:19.18 | [TK]D-Fender | imcdona: it isn't required |
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| 22:15.51 | ManxPower-work | imcdona: Remember that just because you ordered NFAS doesn't mean the telco ACTUALLY set it up that way. NFAS is not all that common of a service in the USA. |
| 22:18.16 | jblack | ManxPower-work: You mean my cheeseburger may come without cheese on it? |
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| 22:31.41 | wwalker | anyone know how to see the jitterbuffer settings? I've added them to sip.conf, but sip show settings doesn't show anything. |
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| 22:41.26 | ChannelZ | hmm this is fun. |
| 22:41.43 | ChannelZ | sip show peers is showing my Name/username as "Bob/bitch". But I don't see where that is coming from.. |
| 22:46.01 | TJNII | I think someone is screwing with you. |
| 22:46.49 | ChannelZ | no, I remember putting that in somewhere as an auth username when I was testing something for someone I can't find anywhere where that is set now so I don't know where it's getting it from |
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| 23:28.10 | wwalker | ChannelZ: grep -ril bitch /etc/asterisk |
| 23:28.46 | wwalker | unless your peers are in realtime db |
| 23:28.49 | ChannelZ | yeah it's not there. It had to have been cached somewhere, which is strange because I've rebuilt 2 different versions of * and restarted since I was doing that testing |
| 23:29.06 | wwalker | ChannelZ: grep -ril bitch / |
| 23:29.12 | wwalker | :) |
| 23:29.26 | ChannelZ | heh |
| 23:30.09 | wwalker | or, to be rid of it but not locate it....: rm -rf/; /bin/reboot |
| 23:30.30 | ChannelZ | hmm my system is screw after running that command |
| 23:30.34 | wwalker | the missing space is for safety |
| 23:31.05 | wwalker | once ran his weekly report instead of cat'ing it.... |
| 23:31.18 | wwalker | not good for a sysadmin |
| 23:31.34 | TJNII | Why was the report chmodded +x? |
| 23:32.41 | TJNII | Man, I moved all my computers out of my office and now this room is cold! |
| 23:33.24 | ChannelZ | no global warming for you |
| 23:33.24 | wwalker | was 18 years ago, I'd been an admin for a few months. typing chmod 755 * was common bad habit, until that day. It was HP-UX, the real question is why was . in the PATH HP? |
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