| 00:04.31 | *** join/#asterisk jagster` (~chatzilla@unaffiliated/jagster/x-9084543) |
| 00:04.53 | jagster` | anyone have any good reading material for best practices on CDR reporting? |
| 00:05.19 | Katty | phix: *hee* that's cute. |
| 00:08.56 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 00:09.09 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 00:12.54 | *** join/#asterisk bandroidx (~bandroidx@205.185.117.117) |
| 00:21.34 | *** join/#asterisk Cubber (~ronny@cpe-74-71-254-190.twcny.res.rr.com) |
| 00:24.03 | *** part/#asterisk ghost75 (~trechber@dslb-088-066-173-112.pools.arcor-ip.net) |
| 00:25.31 | phix | Katty: :) |
| 00:27.46 | *** join/#asterisk KNERD (~KNERD@24.175.249.177) |
| 00:29.43 | *** join/#asterisk deviantlinux (~ryan@unaffiliated/deviantlinux) |
| 00:30.41 | deviantlinux | I am compiling a source-based asterisk, to copy a module .so from it, and move it to an RPM-based asterisk. I tried the same compile options as the source RPM, but I can't seem to get it right. Can I fake asterisk into thinking my module was compiled with the same compile-time options as the RPM? |
| 00:31.51 | WIMPy | If it doesn't want to load it, that's probably better for you. |
| 00:31.55 | deviantlinux | No. |
| 00:32.07 | deviantlinux | I need this module, and the compile-time options are irrelevant for this module |
| 00:32.18 | deviantlinux | this is a res_failover.so module for an OpenVox product. |
| 00:32.54 | WIMPy | It's rarely a good idea to install different parts of one applicatin in different ways. |
| 00:33.24 | *** join/#asterisk Cubber (~ronny@cpe-74-71-254-190.twcny.res.rr.com) |
| 00:33.29 | WIMPy | If they were't relevant, it would probably work. |
| 00:34.17 | deviantlinux | no it wouldn't probably work, asterisk blindly states that if the compile time options are different, it refuses to load. |
| 00:48.17 | deviantlinux | Trying it now without a --prefix as it looks like asterisk hashes the configure options |
| 01:11.20 | *** join/#asterisk navaismo (~navaismo@189.191.229.207) |
| 01:13.36 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 01:15.33 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:16.44 | *** join/#asterisk DIdaho (~jorgensen@203.116.17.103) |
| 01:20.36 | DIdaho | hello all: I'm seeing a problem in 1.8.13 (latest available in ubuntu 13.04-beta) for an outbound sip call. The behavior seen is that I can place a call but the BYE fails due to authorization failure. What's happening behind the scenes is that the sip server (not looked up through SRV) has a list of IPs and we seem to get a different one each time(round robin?) and the Proxy-Authorization header includes the IP address as the domain |
| 01:21.01 | DIdaho | anyone else seen this and maybe knows if it is fixed in more recent code? |
| 01:22.41 | DIdaho | in general it seems annoying that the registration hops around from one IP to the next. does the hostname really need to be re-resolved each time (re-)registration occurs? seems like it would make more sense to only re-resolve it if registration times out |
| 01:24.19 | Rhomber | So, i have a Dial cmd that runs a macro.. I assume there is no way for the macro to set any channel variable or return variable that will end up available to the dialplan that invoked the original Dial? |
| 01:24.30 | Rhomber | or at least, that's what my testing shows.. just wanted to confirm |
| 01:37.33 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 01:40.53 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 01:45.55 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:45.55 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:50.18 | Rhomber | never mind, i was hanging up in the macro |
| 01:50.32 | Rhomber | not hanging up meant the macro_result propagated correctly to the dialstatus :) |
| 02:14.32 | *** part/#asterisk ryan_gpx (~ryan@unaffiliated/deviantlinux) |
| 02:17.09 | deo | hello everybody.. need some help/adivice here.. this morning our pabx went crazy.. whenever someone call from outside. our pbx will play the ivr but then whenever we dial an extension number its says "Thats not a valid extension" but the extension we dialled is actually valid.. |
| 02:17.41 | deo | this already worked before and i dont know why its not working today.. please need some help thanks guys.. |
| 02:27.27 | Rhomber | are you configuring asterisk yourself with like extensions.conf? or using FreePBX? |
| 02:27.37 | Rhomber | if your using FreePBX, best to ask them |
| 02:27.46 | tm1000 | #freepbx |
| 02:28.14 | deo | hi Rhomber tm1000 thanks .. im actually using FreePBX |
| 02:28.43 | tm1000 | deo: This channel is usually just either A) Plain vanilla Asterisk B) Specific Asterisk Questions if using freepbx |
| 02:29.08 | deo | thanks tm1000 |
| 02:31.13 | Rhomber | so i can propagate MACRO_RESULT as DIALSTATUS .. with certain values.. great |
| 02:31.29 | Rhomber | but when the called party hangs up, i can't propagate anything custom :( |
| 02:31.44 | Rhomber | DIALSTATUS always becomes ANSWER |
| 02:31.58 | Rhomber | what i want to do is return BUSY |
| 02:36.45 | *** join/#asterisk radic (~radic@dslb-178-002-221-148.pools.arcor-ip.net) |
| 02:36.46 | *** part/#asterisk mjordan (~mjordan@adsl-98-81-83-165.hsv.bellsouth.net) |
| 02:37.12 | Katty | moo |
| 02:42.20 | Rhomber | moo |
| 02:54.07 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.27) |
| 02:55.13 | ruben231 | hi guys any one have idea, i have DID# and set for inboudn but when i call the DID on my asterisk server calls does not comes in but i see this on logs----> http://pastebin.com/qppkrAAk |
| 02:55.16 | ruben231 | any idea guys |
| 02:57.45 | Rhomber | yeah |
| 02:58.20 | Rhomber | chances are your host= in sip.conf doesn't match |
| 02:58.49 | Rhomber | or in general the incoming can't be mapped to a SIP peer configuration |
| 02:58.57 | Rhomber | so it's ending up in default |
| 02:59.04 | Rhomber | and you don't allow guests (good) |
| 02:59.50 | Rhomber | i'm fairly new to Asterisk, but I had this problem anyway |
| 03:02.25 | ruben231 | Rhomber: how to resolve somehow..? |
| 03:03.58 | Rhomber | i said how? |
| 03:04.20 | Rhomber | you need to ensure your incoming matches a SIP peer. |
| 03:04.26 | Rhomber | so like, with host= |
| 03:05.08 | ruben231 | Rhomber: yes i setup the host already |
| 03:06.21 | ruben231 | http://pastebin.com/jw5axGXa |
| 03:10.22 | Rhomber | For 66.162.83.73? |
| 03:10.27 | Rhomber | anyway, lunch time |
| 03:16.46 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:32.43 | *** join/#asterisk Foxi352 (~Foxi352@v-172-252.access.restena.lu) |
| 04:32.44 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 04:33.36 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 04:33.53 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.27) |
| 04:35.58 | *** join/#asterisk FireAndIce (~FireAndIc@203.187.232.215) |
| 05:43.21 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:05.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 06:11.52 | *** join/#asterisk BlackDex (~BlackDex@ori.vyus.nl) |
| 06:13.18 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 06:15.15 | *** join/#asterisk BlackDex (~BlackDex@ori.vyus.nl) |
| 06:18.43 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 06:20.05 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:26.15 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 06:29.17 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 06:41.38 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 06:42.58 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 06:43.37 | bulkorok | hi |
| 07:00.20 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 07:15.05 | *** join/#asterisk kontinuity (~kontinuit@122.172.246.161) |
| 07:19.09 | *** join/#asterisk jsjc (~Adium@227.Red-2-136-107.dynamicIP.rima-tde.net) |
| 07:31.00 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:2cd4:f5e1:467b:3f54) |
| 07:32.40 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 07:34.26 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 07:41.55 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f5fc:7d14:5b66:cba0) |
| 07:52.40 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 07:54.25 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 07:54.32 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 08:01.25 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 08:06.54 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 08:15.46 | *** join/#asterisk _omer (~omer@116.71.181.26) |
| 08:15.52 | _omer | hi |
| 08:16.23 | _omer | How to extract SIP UserName from ${MESSAGE(to)} and ${MESSAGE(from)} ... ? |
| 08:19.24 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:20.43 | Martin` | [Apr 3 08:18:27] NOTICE[1430] chan_sip.c: Failed to authenticate device 41555<sip:41555@83.98.169.205>;tag=458b39ba |
| 08:20.52 | Martin` | can't fail2ban this :( |
| 08:31.50 | *** join/#asterisk amessina_ (~amessina@50-196-241-73-static.hfc.comcastbusiness.net) |
| 08:33.46 | *** join/#asterisk _omer (~omer@116.71.181.26) |
| 08:33.56 | _omer | <_omer> How to extract SIP UserName from ${MESSAGE(to)} and ${MESSAGE(from)} ... ? |
| 08:36.14 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:37.58 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:39.50 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 08:43.52 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:58.39 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 08:58.54 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 09:00.13 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 09:02.22 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 09:03.18 | *** join/#asterisk timahvo1 (~rogue@41.215.82.210) |
| 09:07.57 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 09:18.23 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:18.28 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:33.44 | *** join/#asterisk deo_ (~deo@112.198.90.88) |
| 09:42.11 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 09:48.56 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-173-226.pools.arcor-ip.net) |
| 10:20.44 | zamba | what do these mean: Sending fake auth rejection for device ? |
| 10:29.34 | *** join/#asterisk ollii (~risker@port-87-193-161-154.static.qsc.de) |
| 10:30.54 | ollii | hey, i'm using * 1.8.20.1 and want to call a macro with arguments from a queue, for example: Queue(TEST_QUEUE,,,,,,foomacro(arg1,arg2)), but an error occurs: WARNING[43286]: app_macro.c:309 _macro_exec: No such context 'macro-foomacro(arg1' for macro 'foomacro(arg1' |
| 10:31.15 | bulkorok | don't use macro... |
| 10:31.22 | ollii | so my question is, how can i push variables to a macro called from app Queue, like Dial(STRING,m(foobar,arg1,arg2) |
| 10:33.07 | bulkorok | <PROTECTED> |
| 10:36.20 | ollii | so, how should i call this ? Queue(TEST_QUEUE,,,,,,,foogosub,s,1(arg1,arg2)) |
| 10:36.50 | ollii | if i call gosub directly in an extension: exten => _X.,1,Gosub(foogosub,s,1(arg1,arg2)) |
| 10:40.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:42.36 | ollii | could you give me an example how to call gosub from app queue? |
| 10:44.55 | skrusty | do i need anything special installed to use ReceiveFax? It's compiled in and executes in the dialplan, but i never hear any fax tones (as i would expect) and it just times out |
| 10:45.07 | skrusty | on my old 1.6 box i used rxfax, and that was fine |
| 10:45.19 | skrusty | but not sure if there is something else i need to do on 1.8 |
| 10:45.49 | skrusty | oh, and a i get a load of UDPTL output (seq number and len) |
| 10:55.45 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 11:05.58 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 11:06.17 | *** join/#asterisk elmargol (~elmargol@host44-23-dynamic.20-79-r.retail.telecomitalia.it) |
| 11:07.21 | elmargol | I',m searching an embedded system for receiving fax messages over a dedicated fax line. Any suggestions? I need to forward all incoming messages to an email adress |
| 11:11.43 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 11:13.07 | bulkorok | skrusty: res_fax_spandsp is a good one |
| 11:14.28 | bulkorok | ollii: no I don't know how to implement a gosub in Queue |
| 11:15.04 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 11:16.51 | *** join/#asterisk apurvtwr (~apurvtwr@182.71.34.251) |
| 11:17.26 | bulkorok | skrusty: check the FFA admin guide for dialplan-examples (works with spandsp too... differences are spotted in the doc): http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf |
| 11:18.09 | skrusty | cheers guys |
| 11:18.38 | bulkorok | elmargol: asterisk can do that for you... |
| 11:20.01 | elmargol | bulkorok, is there "cheap" hardware out there to do this? |
| 11:20.19 | elmargol | the cheapest minipci isdn card i can find is about 200$ |
| 11:20.42 | bulkorok | elmargol: so you get your faxes via ISDN?! |
| 11:21.15 | elmargol | bulkorok, no over a simple phone line |
| 11:21.24 | bulkorok | analog one!? |
| 11:21.39 | elmargol | bulkorok, I guess I'm not 100% sure |
| 11:21.49 | bulkorok | you should! |
| 11:21.59 | elmargol | The fax i use has a 33.6kpbs connection |
| 11:22.51 | bulkorok | should be an analog line... so the best is to use an ATA |
| 11:23.19 | ollii | cisco spa112 (former linksys) worked fine for me |
| 11:23.50 | ollii | if you need an isdn line, look for 1port hfc cards, price is ~30 $ |
| 11:24.32 | elmargol | The problem i have is that i have to receive fax messages from a remote location |
| 11:24.36 | elmargol | If I'm not at the office |
| 11:24.38 | ollii | that is not a problem |
| 11:24.50 | elmargol | Thats why Im plaing to forward all messages to email |
| 11:25.29 | ollii | 1x spa112,1x small box with linux+asterisk + working email server + working internet |
| 11:25.31 | ollii | there you go |
| 11:25.49 | bulkorok | reminds me to what I want to do at home ;-) |
| 11:25.58 | apurvtwr | hi everyone. I have a Asterisk 1.8 PBX on which I have registered a SIP phone with a followme number. It works fine, only I want find out if the incoming call was answered on the SIP or it's followme. Is that possible ? Google doesn't seem to be of much help. |
| 11:26.25 | elmargol | Modem 33.6kbps Approx 3 seconds transmission |
| 11:26.26 | elmargol | Data Compression Mono: MH / MR / MMR Colour: JPEG <- thats what my fax supports |
| 11:27.23 | elmargol | I could run asterisk on virtualbox on my server. there are plenty resources left |
| 11:27.31 | elmargol | Guess spa112 is the ideal sulution for me... |
| 11:27.54 | elmargol | How does asterisk connect to the spa112? over ssl/tls ? |
| 11:28.06 | ollii | mh...its not...you need an fxo port, spa112 only supports fxs (end device, like phone or fax device).. you need an SPA3102 which supports fxo |
| 11:28.11 | ollii | via SIP |
| 11:28.28 | bulkorok | that'S what I wanted to write too... |
| 11:28.30 | ollii | its possible to use srtp with tls in combination with sip |
| 11:28.44 | ollii | http://www.cisco.com/en/US/products/ps10027/index.html |
| 11:28.51 | ollii | " One PSTN FXO port to connect to a Telco or PBX circuit" |
| 11:29.19 | bulkorok | either that or an internal fxo card |
| 11:30.08 | ollii | are there any cheap ones? |
| 11:30.13 | bulkorok | interesting: http://www.sangoma.com/products/usbfxo/ |
| 11:30.13 | elmargol | well I don't have a pc for this... |
| 11:30.47 | elmargol | The server is to far away to run the phone wire |
| 11:31.10 | elmargol | Cisco SPA3102 <- is cheap |
| 11:31.57 | bulkorok | sure |
| 11:33.02 | elmargol | OK I get 1Cisco SPA3102 connec tit to ethernet and to the phone line. 1 virtual machine (asterisk) I configure Cisco SPA3102 to forward everything to the asterisk box. And configure asterisk to forward the fax to email? |
| 11:34.55 | elmargol | I'm not 100% sure that there isn't box like the SPA3102 offering a built in email forwarder |
| 11:37.48 | ollii | could be possible |
| 11:38.18 | ollii | fritzbox 7270 could be an option? |
| 11:39.10 | ollii | or something like this: http://www.auerswald.de/de/produkte/telefonanlagen/home-office/compact-3000-analog/produktbeschreibung.html |
| 11:39.27 | ollii | its a small pbx with fxo integrated |
| 11:40.27 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 11:40.59 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 11:41.18 | *** join/#asterisk rhineheart_m (~chatzilla@unaffiliated/rhineheartm/x-283746) |
| 11:41.20 | *** join/#asterisk Kangal (~root@2001:41d0:8:390a::1) |
| 11:41.39 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 11:41.59 | ollii | bulkorok: after using dumpchan in my macro and noticing that calleridnum is set to answered agent, i found my solution ;) |
| 11:42.20 | ollii | 2nd argument is passed as Set(__ARGX=XX) |
| 11:42.27 | ollii | before app queue |
| 11:42.40 | ollii | its ugly but MEMBERINTERFACE seems not available in this state |
| 11:43.28 | bulkorok | ollii: great... but be aware that macro is deprecated after 1.8 |
| 11:43.53 | ollii | that is the problem of futre ollii |
| 11:43.57 | ollii | *future |
| 11:43.58 | ollii | ;) |
| 11:45.11 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:45.44 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 11:45.49 | *** join/#asterisk Yeretik (~Yeretik@nixe.com) |
| 11:49.14 | elmargol | OK I think i get the Cisco SPA3102 |
| 11:55.17 | apurvtwr | I have an agent for whom I have a SIP Phone and it's followme number (his mobile number). This agent is registered on an asterisk queue. When an inbound calls comes to this queue, it rings the agent's SIP phone first and if that is unanswered, the call is made to his followme number. Is it possible to find out if he answered the call on the SIP phone or the follow me? |
| 11:55.51 | apurvtwr | any pointers? |
| 11:57.03 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 11:58.00 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:02.16 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 12:03.53 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f5fc:7d14:5b66:cba0) |
| 12:09.18 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 12:09.45 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 12:09.45 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 12:12.31 | skrusty | is it possbile to block an IP after x number of failed registration attempts in a row? |
| 12:14.43 | pabelanger | skrusty: not specifically in asterisk, you need something like fail2ban |
| 12:14.52 | pabelanger | or some other script |
| 12:17.40 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-102.telkomadsl.co.za) |
| 12:18.34 | elmargol | are there special asterisk distributions. or should i just use debian stable and install asterisk? |
| 12:18.47 | elmargol | I'm going to use virtualbox |
| 12:18.49 | eirirs | freepbx distro |
| 12:19.08 | beardy | elmargol: Yes, use Debian stable. |
| 12:19.35 | elmargol | is this an appliance? |
| 12:21.10 | jkroon | what's "this"? |
| 12:21.23 | elmargol | freepbx |
| 12:21.33 | jkroon | no, it's a configuration application for asterisk. |
| 12:21.42 | elmargol | ah. I guess i just use debian... |
| 12:21.50 | elmargol | I don't like centos :) |
| 12:22.22 | jkroon | so basically it's a set of convoluted bits and pieces of dialplan that gets configured and managed via some (i think) php code, using some form of mysql (usually, i guess) backend. |
| 12:22.36 | jkroon | elmargol, i'd recommend you do that yes. and learn to write your own dialplan. |
| 12:22.36 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:23.20 | elmargol | I'm wondering if HylaFAX isn't the more fitting solution for me |
| 12:26.21 | bulkorok | elmargol: hylafax is just an endpoint... just like receivefax() from asterisk... |
| 12:26.56 | elmargol | just ordered the SPA3102 :) |
| 12:27.08 | bulkorok | so using asterisk and hylafax is usually incomming fax => asterisk => iaxmodem(or sth like that) => hylafax |
| 12:27.40 | ollii | my experience with hylafax + iaxmodem + asterisk 1.8+ is not so well...i prefer ReceiveFax as well |
| 12:27.43 | WIMPy | Except for that you can leave Asterisk out of the equation if you want. |
| 12:27.58 | ollii | advantage of hylafax is at sending faxes, there are print drivers for windows available |
| 12:30.36 | bulkorok | for asterisk you can implement some mail2fax instead of printerdriver... |
| 12:31.19 | ollii | indeed, i use that too, but a proper MIME Parser is not easy to find |
| 12:31.37 | ollii | if you want to send doc,docx,pdf,tiff and so on |
| 12:32.05 | *** join/#asterisk DND (5ec8071a@gateway/web/freenode/ip.94.200.7.26) |
| 12:32.18 | elmargol | I think sending PDF over FAX is all you need :) |
| 12:34.08 | ollii | haha, there are special customers out there |
| 12:34.48 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:36.06 | DND | hi guys i need help. how can i add another span? |
| 12:36.18 | DND | we have an existing pri card TE121 |
| 12:36.26 | DND | we added another same card |
| 12:36.30 | DND | but different telco |
| 12:36.45 | DND | dahdi says "ok" |
| 12:37.24 | bulkorok | long tim eago... I used zaptel... needed zapata.conf and zaptel.conf |
| 12:38.54 | elmargol | Are there easy solutions to send sms messages using asterisk? If a fax gets received or something |
| 12:39.14 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:39.27 | bulkorok | elmargol: why not sending email with the fax attached?! |
| 12:39.38 | ollii | http://www.voipon.co.uk/voismart-sip-gsm-sms-4-channel-p-703.html |
| 12:39.44 | ollii | something like this |
| 12:39.54 | ollii | or you take a old mobile phone and connect it via serial |
| 12:39.57 | bulkorok | elmargol: you can additionally send a sms too |
| 12:40.03 | bulkorok | or bluetooth... |
| 12:40.17 | elmargol | bulkorok, I was thinking something like. You got a fax please check mail :) |
| 12:40.40 | elmargol | can't i send sms using a sip provider? |
| 12:40.55 | ollii | dunno |
| 12:42.17 | bulkorok | there was a SMS() app... but I don't know if it's still the one |
| 12:43.04 | *** join/#asterisk davlefouAMD (~david@41.225.59.22) |
| 12:43.11 | bulkorok | https://wiki.asterisk.org/wiki/display/AST/SMS%28%29 |
| 12:43.40 | bulkorok | so... I think you have to look around what helps... |
| 12:43.48 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 12:46.30 | elmargol | bulkorok, thank you i look arround :) |
| 12:53.41 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 12:54.38 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:55.27 | *** join/#asterisk serafie (~erin@nat/digium/x-erfxqvtqkloporeg) |
| 12:56.38 | *** part/#asterisk ollii (~risker@port-87-193-161-154.static.qsc.de) |
| 13:06.47 | *** join/#asterisk zpotoloom (~toomas@2001:ad0:1:1:3ed9:2bff:fe5c:e027) |
| 13:09.40 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:13.13 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 13:14.45 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 13:15.08 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-144-164.user.veloxzone.com.br) |
| 13:23.53 | seik0 | Hi, people. I got into CDR and here is something like a mess. Could someone advice where can I read about CDR in deep details, not only what each "column" stands for ? |
| 13:25.17 | seik0 | I've read a bit, that with complicated configurations it can be quite ununderstandable |
| 13:27.10 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 13:27.40 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:27.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:27.46 | *** join/#asterisk melter (~Melter@2001:4930:116:0:c1ad:cbc9:dc9c:8c2a) |
| 13:29.51 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 13:34.33 | *** join/#asterisk apurvtwr (~apurvtwr@182.71.34.251) |
| 13:36.06 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 13:37.47 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 13:41.08 | *** join/#asterisk deo (~deo@112.198.90.205) |
| 13:49.00 | *** join/#asterisk Chinorro (~chino@38.215.117.91.dynamic.mundo-r.com) |
| 13:52.56 | igcewieling | " CDR and here is something like a mess" <--- you are correct, CDRs are a mess |
| 13:53.06 | igcewieling | Did you have any specific question? |
| 13:54.23 | v0lZy | id guess they are a mess because u have different billing charges based on what numbers are being called? |
| 13:54.28 | v0lZy | or am i missing something |
| 13:54.44 | v0lZy | isnt it just a simple table you can use grep and awk on? |
| 13:56.55 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mnukcofocwyfcprt) |
| 13:56.55 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:58.51 | *** join/#asterisk Jinxed- (c663818f@gateway/web/freenode/ip.198.99.129.143) |
| 13:59.03 | apurvtwr | hi . Is there a way to figure out if a call was answered on the SIP or it's follow me? |
| 13:59.16 | Jinxed- | anyone have a recommendation for a quality inexpensive GSM gateway? |
| 14:01.46 | igcewieling | huh? There is no cost information in the CDRs |
| 14:02.30 | igcewieling | v0lZy: normally CDRs are stored in a database |
| 14:02.37 | WIMPy | Interesting, isn't it? That's usually why you want them. |
| 14:02.59 | igcewieling | WIMPy: I want Asterisk to decide how much to bill for the calls? |
| 14:03.08 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-144-164.user.veloxzone.com.br) |
| 14:03.29 | WIMPy | No you want the costs of your provider stored in the CDRs. |
| 14:03.38 | jmetro | igcewieling wouldnt you make your own cost column based on math + billable time |
| 14:04.22 | igcewieling | jmetro: yes, but that would be something you do, not something Asterisk does automatically |
| 14:04.38 | igcewieling | We, for example, store the route ID and the account ID in custom CDR columns |
| 14:09.36 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:17.01 | v0lZy | WIMPy: But how do you store those records though... arent costs like from one country to another different if its from France to Germany or from France to lets say the US. |
| 14:18.03 | WIMPy | Off course. I was only talking about logging the information you get from your provider. |
| 14:20.02 | v0lZy | I dont think im quite getting it. |
| 14:20.29 | v0lZy | what information comes back from the provider? |
| 14:20.58 | WIMPy | "AOC" |
| 14:21.22 | WIMPy | That's wat's usually displayed on your phone during a call. |
| 14:22.22 | v0lZy | AOC? |
| 14:22.35 | v0lZy | (sorry, not versed in this stuff yet) |
| 14:22.39 | WIMPy | Advice Of Charce |
| 14:24.38 | wdoekes | *charge |
| 14:24.43 | v0lZy | do all providers provide that? |
| 14:24.47 | v0lZy | I mean, is that a standard thing |
| 14:24.52 | igcewieling | v0lZy: virtually no information comes back from the provider. YOU know how much YOU bill for the calls. Asterisk does NOT. |
| 14:24.53 | WIMPy | yes |
| 14:24.58 | WIMPy | yes |
| 14:25.08 | v0lZy | (first time i heard of it... dont see it on my mobile phone... and havent configured it on my sip phones... before that, i dont remember phones having displays :D :D |
| 14:25.51 | igcewieling | v0lZy: lets start over. Have you configured Asterisk to store CDRs and if so, where are you storing them? |
| 14:26.01 | WIMPy | If you do the provider thing, you decide. If you're the user, you just usually want to collect the data to assign the correct charges to departments or whatever. |
| 14:26.29 | v0lZy | igcewieling: yeah, that part makes sense to me... u want to create a bill for someone you export all references regarding his originating numbers, then sort them the way your billing is set up and charge based on how many minutes/seconds whatever your interval fall where in your scheme... that part is clear |
| 14:26.49 | igcewieling | v0lZy: good. Now exactly what is not clear. |
| 14:27.24 | v0lZy | igcewieling: well, for one if a person decides to resell their voip services... |
| 14:27.27 | WIMPy | You can even do it in realtime. Asterisk allows you to send AOC. |
| 14:27.59 | v0lZy | do you just wing it when it comes to costs u have vs charges you'll be charging or what |
| 14:27.59 | igcewieling | v0lZy: I'm not seeing a difference. |
| 14:28.44 | v0lZy | igcewieling: suppose I use you as my SIP trunk and resell my asterisk pbx to a tennant in the same building |
| 14:28.49 | igcewieling | v0lZy: We have a set rate we bill to customers for different types of calls. We have a set rate our wholesaler charges us. Management makes sure the first is never less than the last. |
| 14:29.12 | igcewieling | v0lZy: then you send me a bill for my usage. |
| 14:29.15 | v0lZy | igcewieling: I can divide the calls up between mine and the tennans and then charge the tenant his part |
| 14:29.44 | igcewieling | v0lZy: correct. |
| 14:29.45 | v0lZy | if i want to up the price... i can up on the same scheme the provider charges ... i just increase for say 1 cent on the interval charges they have or something |
| 14:29.56 | v0lZy | but what if i want to give a different scheme |
| 14:30.02 | *** join/#asterisk nephfl (6c09e362@gateway/web/freenode/ip.108.9.227.98) |
| 14:30.15 | v0lZy | then i have to get information from my provider on how much he bills for calls to each country |
| 14:30.25 | v0lZy | or each other provider in this country or what? |
| 14:30.44 | v0lZy | I mean.. we have multiple gsm networks here... calls to one could be cheaper then calls to another... |
| 14:31.01 | WIMPy | Should be lots of fun to dig through thousands of price lists. |
| 14:31.12 | v0lZy | yeah, my point exactly |
| 14:31.14 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 14:31.21 | igcewieling | v0lZy: welcome to the world of telecom billing. This is how it works. |
| 14:31.32 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:31.35 | v0lZy | how's this done.. its all done on what igcewieling said? fixed charges for when I call you... and you dont charge me differently if i use u to call germany or france? |
| 14:31.56 | raden | Is cracknig asterisk boxes just like a hoby for people ???? I'm getting tired of all the failed registration attempts a little ridiculous |
| 14:31.58 | WIMPy | So the easiest thing is to just forward the charges you receive and possibly multiplay them. |
| 14:32.02 | v0lZy | or you do charge me differently and i have to keep track of how much u charge for each destination? |
| 14:32.11 | igcewieling | v0lZy: we have massive price lists from our provider listing each different rate, that is put into our billing system |
| 14:32.19 | v0lZy | WIMPy: yeah, thats the easiest thing, but i was thinking.. is that what everyone does? |
| 14:32.29 | WIMPy | raden: No, it's big business. |
| 14:32.56 | raden | WIMPy, ? |
| 14:33.07 | raden | its ridiculous |
| 14:33.13 | igcewieling | raden: hacking PBXs and reselling stolen mins is big business |
| 14:33.23 | raden | UGH |
| 14:33.25 | v0lZy | raden: makes sense its big business.. crack a box, someone else pays for your calls... you resell those calls... sounds like lots of profit. |
| 14:33.25 | raden | im sick of it |
| 14:33.44 | WIMPy | raden: Find someone who misconfigured their system and make free calls. Off course not for yourself but as a resale business. |
| 14:33.46 | raden | If they have customers why cant they just make a freaking honest living :( |
| 14:33.57 | igcewieling | Every couple of months some new exploit is discovered (not usually in Asterisk) and our toll fraud software kicks in and I get a dozen or so e-mails |
| 14:34.15 | WIMPy | raden: Whay should they if they can earn even more? |
| 14:34.24 | raden | I understand |
| 14:34.25 | igcewieling | the most recent is some kind of exploit on Adtran routers. |
| 14:34.38 | raden | just a lot of work for people when they could be doing honest work and earning just as much |
| 14:34.43 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 14:34.49 | WIMPy | That's the way society wors and the internet as well. |
| 14:34.55 | raden | igcewieling, interesting |
| 14:35.03 | WIMPy | +k |
| 14:35.07 | igcewieling | raden: we have it easy. All our customers are on static IP addresses and 99% of our customers are on-net |
| 14:35.14 | raden | Im changing routers very soon so theses issues will go buhbye |
| 14:35.22 | igcewieling | so we use iptables to disable all access from off-net. |
| 14:35.36 | igcewieling | granted, that is not practical for most people. |
| 14:35.37 | raden | 90# of our customers are internal |
| 14:35.49 | raden | so ill allow external subnets i need to and block the rest |
| 14:36.11 | *** join/#asterisk Xaviertoor (~Xaviertoo@189.52.87.66) |
| 14:36.26 | v0lZy | what about radius authentication |
| 14:36.35 | v0lZy | or VPN for those that have dynamic addresses |
| 14:36.41 | WIMPy | What about that? |
| 14:36.51 | v0lZy | that would secure thing |
| 14:36.56 | WIMPy | Just shifts the issue elsewhere. |
| 14:36.56 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:37.30 | v0lZy | yeah, but i suppose cracking VPN's etc is a bit different then just burteforcing asterisk |
| 14:37.41 | WIMPy | But hiding stuff does help in practice. |
| 14:38.29 | WIMPy | Maybe, maybe not. But just because someone cracks your VPN, they may not have noticed that they could crack your Asterisk as well. |
| 14:39.34 | WIMPy | Just don;t try to focus on getting rid of the annoying log entries. That's not going to help you. Quite possibly the opposite might happen. |
| 14:41.16 | v0lZy | we'll if you allow only a single session at a time under a username password or alternatively encrypted key pair |
| 14:41.59 | v0lZy | its probably easy to notice |
| 14:42.04 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 14:45.37 | anonymouz666 | coppice: do you mind if I ask you a question about "data calls" v90 modem stuff through asterisk? you answered a mail in feb/2006 http://lists.digium.com/pipermail/asterisk-users/2006-February/138868.html about this subject. Since then, anything changed? Do you know any VoIP device (ATA) that could be used with a modem doing a v90 handshake? |
| 14:47.44 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:47.44 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:48.49 | coppice | anonymouz666: physics tends not to change |
| 14:49.04 | anonymouz666 | heh |
| 14:49.06 | anonymouz666 | nice |
| 14:50.33 | anonymouz666 | coppice: do you already saw any VoIP equipament in the path and this kind of negotiation working? |
| 14:51.37 | anonymouz666 | modem <-> FXS/ATA <SIP/RTP/G711a> asterisk <-> T1/E1 <-> PSTN |
| 14:51.38 | anonymouz666 | like this |
| 14:53.28 | [TK]D-Fender | anonymouz666, I've heard of low baud (2400) working "sometimes" for CC machines, etc. Wouldn't bet on it reliably |
| 14:53.41 | WIMPy | anonymouz666: Better connect that modem via some card in your server. |
| 14:54.17 | anonymouz666 | you mean: modem <-> FXS <-> asterisk <-> E1/T1 <-> PSTN |
| 14:54.20 | WIMPy | Otherwise X.75 seems to work pretty reliably over VOIP, but at very low speeds/massive delays, off course. |
| 14:54.21 | anonymouz666 | without RTP |
| 14:54.29 | WIMPy | indeed. |
| 14:56.18 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 14:56.32 | anonymouz666 | what's the main difference between the FAX stuff and the v.90 negotiation? I have a setup like: FAX machine <-> FXS/ATA <SIP/RTP/G711a> asterisk <-> T1/E1 <-> PSTN |
| 14:56.38 | anonymouz666 | working pretty well |
| 14:56.48 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 14:56.51 | anonymouz666 | even with multiple pages |
| 14:57.01 | WIMPy | I don't know if FXO cards allow timing cable to E1/T1 cards. If notm you might still get some timing glitches. |
| 14:58.28 | anonymouz666 | In my case I would be using a FXS card to plug into the modem (FXO) |
| 14:58.35 | igcewieling | sangoma supports clock sync between their t1/e1 and their analog cards with a special cable |
| 14:59.12 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 15:00.01 | WIMPy | Err, sorry, FXS, yes. |
| 15:01.05 | anonymouz666 | this clock sync should be make the TX more reliable |
| 15:02.10 | WIMPy | Unless there are still some hidden horrors in Asterisk, that should make it work perfectely. |
| 15:02.21 | anonymouz666 | lowering the speed in "data calls" should make things easier to get in sync? |
| 15:02.46 | anonymouz666 | i.e.: If I force the baud rate in the modem |
| 15:03.34 | WIMPy | Yes, but I don't think anyone would make any guess on how much that helps. |
| 15:04.00 | anonymouz666 | at least it is something to give a try, when nothing works. :-) |
| 15:04.39 | WIMPy | Better not to try that construction. |
| 15:05.20 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 15:06.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:06.50 | *** join/#asterisk nix8n82 (~AndChat27@216.67.162.172) |
| 15:09.30 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:13.15 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 15:13.32 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:18.51 | *** join/#asterisk kchehab (~david@77.42.241.66) |
| 15:19.04 | fling | [Apr 3 22:04:14] WARNING[31947]: pbx.c:4458 pbx_extension_helper: No application 'GoSub' for extension (hh |
| 15:19.12 | kchehab | guys is there a way to integrate viber with asterisk |
| 15:19.14 | fling | so gosub stopped working after an update |
| 15:19.21 | fling | what am I doing wrong |
| 15:19.36 | kchehab | viber module for asterisk or any thing |
| 15:20.30 | fling | umm |
| 15:21.23 | [TK]D-Fender | kchehab, no |
| 15:23.20 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:23.57 | igcewieling | joy. Asterisk 11 hardlocked on us |
| 15:30.25 | *** join/#asterisk fkurkowski (~fkurkowsk@186.215.23.194) |
| 15:31.35 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:36.03 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 15:40.51 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 15:42.17 | *** join/#asterisk SpenglerMBP (~spenglerM@dns1.aacc.edu) |
| 15:42.35 | SpenglerMBP | is there a hangup cmd to make the phone itself hang up |
| 15:42.56 | SpenglerMBP | for example ; i dont want to hear the busy sound when i enter an acd queue |
| 15:43.13 | WIMPy | That's up to the phone. |
| 15:44.47 | SpenglerMBP | is there a better way to login to the acd as an agent? Right now i have it set to dial *57 and that logs him in |
| 15:45.23 | [TK]D-Fender | Spengler1, You tell us this ... as that "*57" meant something explicit for us as a basis of comparison |
| 15:45.59 | [TK]D-Fender | s/as that/as if/ |
| 15:46.14 | [TK]D-Fender | MBP rather |
| 15:46.23 | [TK]D-Fender | curses his auto-complete... |
| 15:46.41 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 15:47.08 | SpenglerMBP | Agent picks up the phone , dials *57 , this logs him into ACD |
| 15:47.20 | [TK]D-Fender | SpenglerMBP, "logs in how" <- |
| 15:47.29 | SpenglerMBP | through the dial plan |
| 15:47.30 | [TK]D-Fender | SpenglerMBP, What function is that using? |
| 15:48.51 | SpenglerMBP | exten => *54,1,Verbose(2,Logging in the Queue Member) |
| 15:48.51 | SpenglerMBP | same => n,Set(MemberChannel=${CHANNEL(channeltype)}/${CHANNEL(peername)}) |
| 15:49.10 | SpenglerMBP | same => n,AddQueueMember(support,${MemberChannel}) |
| 15:49.10 | SpenglerMBP | same => n,Set(DEVICE_STATE(Custom:lamp1)=INUSE) |
| 15:49.10 | SpenglerMBP | same => n,Hangup |
| 15:49.34 | SpenglerMBP | Just wondering if this is the best way to do it |
| 15:49.34 | [TK]D-Fender | SpenglerMBP, * already is hanging up explicitly. |
| 15:49.39 | [TK]D-Fender | SpenglerMBP, What PHONES are you using? |
| 15:49.44 | SpenglerMBP | Digium D50 |
| 15:50.18 | [TK]D-Fender | SpenglerMBP, well that ends the call. What you are hearing is not a BUSY tone ... it is a REORDER tone. and that's just how the phone works. |
| 15:50.43 | [TK]D-Fender | Spengler1, the phone is "hung up, expect for the tone it decides to give you until you hang up the handset |
| 15:52.19 | SpenglerMBP | [TK]D-Fender ; how do your agents login to ACD? |
| 15:52.45 | [TK]D-Fender | SpenglerMBP, Same way. This is not an Asterisk problem. |
| 15:53.17 | [TK]D-Fender | SpenglerMBP, Your PHONE chooses to make that sound for you. That "hangup" ends the call. |
| 15:53.34 | SpenglerMBP | Gotcha |
| 15:56.09 | *** join/#asterisk RyanTG (~Thunderbi@mailer.americassuppliers.com) |
| 16:07.19 | jmetro | You can make your phone make goofy sounds using Toning though. I once made all our dialtones super high pitched |
| 16:07.37 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:08.07 | jmetro | and the goold ol starwars ringtone from the book |
| 16:17.48 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 16:31.17 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 16:34.55 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:41.36 | jagster` | whats a good book for voip/pbx intro? |
| 16:41.41 | jagster` | for sys admins |
| 16:42.06 | WIMPy | For Asterisk the to recommendation is the |
| 16:42.10 | WIMPy | ~book |
| 16:42.11 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:45.45 | *** join/#asterisk fkurkowski (~fkurkowsk@186.213.159.244) |
| 16:47.00 | igcewieling | whoo! whoo! the 3rd server for our asterisk cluster arrives on friday |
| 16:50.20 | WIMPy | grid |
| 17:04.09 | *** join/#asterisk navaismo (~navaismo@189.191.229.207) |
| 17:06.27 | jagster` | i get to mange a trixbox setup /wrists |
| 17:08.13 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:08.29 | igcewieling | jagster`: is it all that much worse than FreePBX, other than the moral issues Trixbox has. |
| 17:09.47 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 17:16.36 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:17.13 | *** join/#asterisk jkroon (~jkroon@41.13.112.247) |
| 17:17.37 | jkroon | hi guys, are there any known issues with chan_sip just not processing it's udp socket for port 5060? |
| 17:17.54 | igcewieling | jkroon: not generally |
| 17:18.01 | jkroon | netstat shows a high recv-q and i've restarted asterisk so many times now, even just "sip reload", and nothing seems to fix it. |
| 17:18.44 | igcewieling | We had 11.2 die on us last night. Had to killall -9 asterisk in order to stop asterisk. |
| 17:18.46 | jkroon | * version 11.3.0, had an older version too |
| 17:22.52 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:22.52 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:24.37 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 17:26.48 | jagster` | should i trust a website called asterisk.ru |
| 17:30.08 | jmetro | jagster: im guessing no, unless they made it multilingual |
| 17:30.23 | [TK]D-Fender | Trust it ... for WHAT? |
| 17:32.43 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:33.40 | jmetro | [TK]D-Fender: TO make breakfast obviously. |
| 17:34.25 | navaismo | ahahaha |
| 17:41.14 | danfromuk | In a multi-tenant situation, when the queue position announcement is played, is there any way to change the source files for the numbers? Is it always alison or whatever numbers are in the sounds directory? |
| 17:41.53 | danfromuk | If each tenant wants to supply their own queue_thereare files, they'll also need to supply their own numbers. |
| 17:44.28 | navaismo | hmmm not sure if can work but you can try to set the channel language to another, then create that folder with the desired files in it |
| 17:44.37 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 17:45.35 | danfromuk | Does the language have to be a real language, or can it be random digits or numbers thats significant to me? |
| 17:47.10 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 17:47.48 | leifmadsen | danfromuk: it can be anythign you want |
| 17:49.44 | paulc | and... so long as it's english, the numbers should be ok - but we had "interesting" times with foreign languages that "count differently" |
| 17:51.38 | jmetro | pascal counts differently |
| 17:52.25 | danfromuk | leifmadsen: thanks |
| 17:53.06 | paulc | as do the French.. and those olde worlde englishe types.. "four and twenty blackbirds, baked in a pie" - I mean, why not just say "24 birds" and be done with it?! :-) |
| 18:25.08 | *** join/#asterisk gbit (~gbit@unaffiliated/gbit) |
| 18:27.02 | gbit | Hello, what is recommended to make a wireless "cloud" to a company uses smartphones to make SIP calls? WDS + Access Point? The user needs to walk and the call must be on when changing AP's. |
| 18:28.03 | jkroon | gbit, good luck finding a phone that can deal with wifi hand-off. |
| 18:28.32 | gbit | So alcatel will win this? |
| 18:28.43 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:28.44 | jkroon | multiple APs with WDS is one way, but you don't *need* WDS, you can also cable the APs up. |
| 18:29.07 | jkroon | gbit, my experience with wifi based phones has not been good. doesn't mean no solution exists. |
| 18:29.34 | gbit | I'm planning to use android nexus or better |
| 18:30.00 | jkroon | smart phones in other words, not dedicated wifi phones. have you scoped battery life? |
| 18:30.58 | gbit | yes, it not bad. I will make it airport mode on, only wireless enabled, and block users to install any other software on it. To make it least |
| 18:32.44 | gbit | cisco phones are too expensives |
| 18:33.30 | gbit | they will have more gain? |
| 18:36.00 | *** join/#asterisk cbdev (~fnord@hieristdas.internetzuen.de) |
| 18:37.36 | gbit | There it must be an open source way to make this, no one has done this before? |
| 18:40.58 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 19:20.39 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:20.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.3.0 (2013/03/28), 10.12.2 (2013/03/27), 1.8.21.0 (2013/03/28), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 19:20.50 | jimw | 'dialplan show baz' sez there is one. |
| 19:21.51 | jimw | It's an echo test, but it seems legit. |
| 19:22.34 | igcewieling | the are missing something. pastebin the output of dialplan show baz |
| 19:22.55 | jimw | So I don't get confused, let me quote the message without the metanames: to extension '600' rejected because extension not found in context 'home'. |
| 19:23.27 | jimw | So foo = '600' and baz = 'home. I'll pastebin the result. |
| 19:23.46 | file | dialplan show 600@home |
| 19:24.52 | jimw | Pastebin -- gotta turn on a lot of JS :) |
| 19:26.13 | jimw | http://pastie.org/7304220; there's only 600 in this context. |
| 19:26.29 | jkroon | is there someone here that would be able to assist with a rather annoying chan_sip.so issue where a failing DNS server can take down chan_sip (and asterisk entirely for that matter since the reload/startup sequence will be disturbed) |
| 19:26.33 | jkroon | https://issues.asterisk.org/jira/browse/ASTERISK-21378 |
| 19:26.34 | LieutPants | [ASTERISK-21378] [Status: Triage] chan_sip - DNS based denial of service - https://issues.asterisk.org/jira/browse/ASTERISK-21378 |
| 19:28.26 | jimw | If I put that same sequence into another context (inhabited by polycom phones), it works. |
| 19:29.34 | file | chan_sip blocks on DNS, to make it do otherwise is... invasive |
| 19:31.26 | jimw | file: Just in case: http://pastie.org/7304326 |
| 19:32.24 | file | jimw, certainly looks fine... |
| 19:32.36 | *** join/#asterisk GameGamer43_ (uid5533@gateway/web/irccloud.com/x-qynpmidpttjbzapu) |
| 19:32.54 | jimw | Could be ekiga problem, I guess. I wish I understood this stuff. |
| 19:36.48 | jkroon | file, how invasive? |
| 19:37.06 | file | ...highly |
| 19:37.33 | jkroon | ok, so I'm now sitting with two major chan_sip issues, neither of which has a simple fix. |
| 19:38.07 | file | new SIP work in Asterisk 12 won't suffer from the same problem |
| 19:38.17 | file | so, yay for that |
| 19:38.24 | jkroon | indeed. |
| 19:38.44 | jkroon | but asterisk 12 is not a trivial upgrade from the looks of things. nor is it production ready by a long shot. |
| 19:39.03 | file | doesn't even really exist yet |
| 19:39.09 | jkroon | exactly |
| 19:41.03 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 19:42.46 | file | jkroon, what's the other issue? |
| 19:43.20 | igcewieling | jkroon: DNS/SIP issues are easily mitigated |
| 19:43.29 | jkroon | peers going missing during reload (https://issues.asterisk.org/jira/browse/ASTERISK-21194) |
| 19:43.30 | LieutPants | [ASTERISK-21194] [Status: Open] [Assigned: jkroon] chan_sip can fail to find a peer during reload - https://issues.asterisk.org/jira/browse/ASTERISK-21194 |
| 19:43.34 | jkroon | igcewieling, please explain |
| 19:43.48 | jkroon | file, i was going to attempt a fix on that, but time hasn't permitted yet. |
| 19:44.00 | file | ah yes that, very good! then I don't need to add anything to my check list |
| 19:44.18 | igcewieling | jkroon: make SURE all IP addresses configured in the system are listed in /etc/hosts, run a local caching only name server, and point /etc/resolv.conf at 127.0.0.1 |
| 19:45.02 | jkroon | igcewieling, done. except that you really can't put all names etc in /etc/hosts because they *can* change from time to time and negates the whole reason for having DNS in the first place. |
| 19:45.23 | jkroon | and you can't put SRV records in /etc/hosts either ... |
| 19:45.38 | igcewieling | jkroon: the host name for local ip addresses (as shown by ifconfig) change? |
| 19:45.41 | WIMPy | Indeed. It's bad enough with listing all hosts of a provider as seperate peers. |
| 19:45.47 | jkroon | igcewieling, they're there. |
| 19:45.57 | igcewieling | I'm ONLY talking about LOCAL IPs in /etc/hosts |
| 19:45.57 | WIMPy | Some day you find out it doesn;t work any more, because they added a new one again. |
| 19:46.03 | igcewieling | jkroon: every single one? |
| 19:46.09 | jkroon | igcewieling, yes. |
| 19:46.12 | jkroon | all eight of them. |
| 19:46.32 | igcewieling | Why in the world do you have 8 ip addresses on your PBX? |
| 19:46.42 | jkroon | rofl, many reasons :) |
| 19:47.01 | jkroon | primary being the ability to load balance over multiple links from other sites. |
| 19:47.17 | igcewieling | The issue "local sip calling breaks when the internet goes down" is solved by my method |
| 19:47.42 | igcewieling | as well and many other DNS related stuff. jkroon: are you running a local caching nameserver on the PBX as well? |
| 19:47.48 | jkroon | ok, the particular box in question has a 1Gbit up/downlink. |
| 19:48.04 | jkroon | if that link goes down, there is a backup box with a similar configuration in another datacenter. |
| 19:48.20 | igcewieling | jkroon: BGP |
| 19:48.27 | jkroon | yes, local caching nameserver running on the pbx itself, it's two upstream dns servers are in the same cabinet. |
| 19:48.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:48.58 | jkroon | and their peers are unfortunately about 10km away with a slightly under 2ms round-trip latency. |
| 19:50.05 | jkroon | issue here is that the entire iburst decided to go awol in it's entirety, so every lookup takes around 10s or so to time out, there are 34 lookups timing out. That's 340s or almost 6 minutes. |
| 19:51.40 | *** join/#asterisk itgrl (~IceChat77@12.25.151.58) |
| 19:53.24 | igcewieling | iburst? |
| 19:56.23 | jkroon | http://mybroadband.co.za/news/wireless/74657-truth-behind-iburst-network-problems.html |
| 19:57.08 | jkroon | iburst is a network provider in south africa, and we have a number of sip registrations going towards them. seems a bunch of their equipment got confiscated, effectively taking down their network. |
| 19:57.39 | jmetro | "You didnt pay your fees? Time to shut off south africas internet" |
| 19:57.53 | jkroon | rofl! or at least a big portion thereof! |
| 19:58.09 | igcewieling | Ah. Hard to deal with a catastrophic outage |
| 19:59.35 | jkroon | well, in the bigger scheme of things that's less than 1% of the overall configuration. so my problem remains. |
| 19:59.56 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:59.57 | jkroon | and the only solution I have is to externalize the DNS for chan_sip and just pass it IP addresses in it's config. |
| 20:03.00 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 20:03.57 | jkroon | either way - problem located and can now be managed. Will need to think about more permanent solutions, for the moment I've just commented out the config that's problematic. |
| 20:04.02 | *** join/#asterisk Cubber (~ronny@cpe-74-71-254-190.twcny.res.rr.com) |
| 20:07.08 | *** join/#asterisk Cubber (~ronny@cpe-74-71-254-190.twcny.res.rr.com) |
| 20:08.48 | newtonr | jkroon: if you want to add your and igcewieling's workaround ideas to the JIRA issue that may be helpful for others. I'm guessing since it's really invasive, always existed and will be resolved with chan_gulp that nothing will happen immediately on it. It may even already exist on JIRA somewhere. I'll have to look around. |
| 20:09.32 | newtonr | *really invasive to fix |
| 20:11.36 | igcewieling | considering this has been an issue since the beginning of Asterisk time...... |
| 20:12.12 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 20:21.27 | jkroon | newtonr, I'll add a comment indicating how to reduce the *risk* and impact of the problem, however, none of what we discussed really fixes the problem. |
| 20:24.25 | newtonr | Thanks |
| 20:26.17 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 20:29.03 | mjordan | I can say that while a patch that adds asynchronous DNS lookups to chan_sip would always be evalauted for inclusion, we at Digium won't try to provide that functionality in chan_sip. |
| 20:29.18 | mjordan | the risk far, far, far out weighs any benefit. |
| 20:30.00 | mjordan | even if a patch were to be provided, it would probably only be considered for inclusion in trunk - as file pointed out, it is going to be insanely invasive, and providing sufficient assurances against regression would be incredibly difficult |
| 20:30.18 | mjordan | which goes back to file's point that it's probably better to build this correctly in chan_gulp. |
| 20:31.24 | mjordan | If it's something someone is interested in pursuing I'm okay leaving the issue open, but otherwise I'd rather document ways that mitigate DNS lookups (local DNS caching, etc.) and close the issue. There's no point in keeping issues that document behavior that is less than ideal - but that are not bugs - open. It adds no value for the open source developer community. |
| 20:32.22 | WIMPy | There was tehe what is a bug question again. |
| 20:32.54 | mjordan | to be frank, it doesn't really matter. Even if it were a bug, the solution for that bug would create so many other bugs that we'd do no other bug fixing for the next 6 months |
| 20:33.03 | mjordan | is that worthwhile? |
| 20:33.30 | mjordan | by the time anyone actually got assigned to look at that problem, 11 would be EOL |
| 20:33.47 | WIMPy | that's an entirely different question. |
| 20:33.49 | mjordan | and then you have a replacement channel driver that doesn't suffer from this problem |
| 20:34.20 | WIMPy | I'm pretty curious to see what other issues it will have. |
| 20:34.26 | mjordan | a lot? |
| 20:34.34 | mjordan | every single DNS lookup has to not expect a result back |
| 20:34.43 | mjordan | it has to build in a mechanism to defer reception of the DNS result |
| 20:35.08 | mjordan | chan_sip is *not* designed with the idea that a transaction is deferred |
| 20:35.13 | mjordan | because it doesn't have a transaction layer |
| 20:35.19 | mjordan | or any layers, for that matter |
| 20:35.51 | WIMPy | I'd try to simply not wait and reschedule the operation if unsuccessfull. |
| 20:36.01 | mjordan | If it makes people feel happier to have an issue open and they want to go poke at it, that's great. I'd love to see a patch that adds that functionality and does so in a manner that we can verify it doesn't break everyone's world. But historically speaking, these kinds of patches always create a ripple of regressions |
| 20:36.25 | mjordan | WIMPy: and now everything has to have a scheduler, that it doesn't have now. In 35000 lines of spaghetti code |
| 20:36.44 | WIMPy | 's not arguing that one. |
| 20:36.47 | mjordan | :-) |
| 20:36.48 | igcewieling | mjordan: could dnsmgr be reworked a little to shorten timeouts? |
| 20:37.03 | WIMPy | I'm currently looking it to it. But for another reason. |
| 20:37.07 | mjordan | maybe, but it won't - unfortunately - uniquely solve the DNS caching problem |
| 20:37.35 | mjordan | if a change could be localized to dnsmgr, that would be fantastic. That's much easier to deal with then things that propagate into the channel drivers |
| 20:38.18 | mjordan | alas, I have to pop out. We can keep the issue open for awhile, and if it looks like someone want to tackle the problem I'd love to keep it open |
| 20:38.40 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mnukcofocwyfcprt) |
| 20:40.15 | newtonr | I've found three old issues for this so far, whee |
| 20:40.37 | newtonr | I'll link them to the new issue before I do whatever with it so they are easier to find |
| 20:40.51 | jkroon | mjordan, a change to dnsmgr might be more worthwhile pursueing. |
| 20:41.46 | jkroon | and I think that doing it there might actually solve more problems, including duplicate lookups. For example, instead of lookup up sip.iburst.co.za 17 times we look it up once, it times out (after say 1s), and we cache that failure for some period of time. |
| 20:42.11 | jkroon | so instead of timing out 34 times we time out once. |
| 20:42.20 | WIMPy | Makes sense. |
| 20:42.32 | WIMPy | Or put a scheduler in to dnsmgr. |
| 20:42.46 | jkroon | there is already one, for the refresh thing, if I'm not mistaken. |
| 20:42.56 | igcewieling | yup |
| 20:43.04 | WIMPy | I guess you're right. |
| 20:43.16 | jkroon | another change to dnsmgr could include re-using old results should new lookups fail. |
| 20:43.46 | WIMPy | Hohouring the TTL might be a good idea. |
| 20:43.50 | jkroon | is dnsmgr used for SRV lookups? |
| 20:43.52 | igcewieling | we used dnsmgr to do primitive load balancing on our endpoints -- until we discovered the adtrans will try reinviting existing calls to the new server when the IP changes. decided it did not give us enough advantage and use ONLY SRV now |
| 20:44.03 | WIMPy | needs auto correct. |
| 20:44.16 | jkroon | well, scheduling the refresh to co-incide with the TTL would be awesome instead of doing it at fixed intervals. |
| 20:44.46 | igcewieling | jkroon: you know Asterisk's SRV support sucks REALLY bad before 1.8 and sucks in 1.8+? |
| 20:45.17 | igcewieling | if you are on 1.8+ I can send you my AEL to use SRV to retry failed calls, etc. |
| 20:45.29 | igcewieling | it isn't pretty but it seems to work well |
| 20:45.30 | jkroon | yes, i know it's bad. still, as bad as it is it saved my ass on at least a handful of situations. |
| 20:45.47 | jkroon | igcewieling, currently on 11.3.0 |
| 20:46.08 | igcewieling | jkroon: At least with 1.8+ you can write dialplan to fail calls over to different priorities |
| 20:46.24 | jkroon | well, on one machine anyway, the rest (reasonable number) are all on some version of 1.8 or newer |
| 20:46.45 | jkroon | igcewieling, would be appreciated if you could send that on please. |
| 20:47.33 | igcewieling | jkroon: http://pastebin.ca/2348978 |
| 20:47.37 | jkroon | thanks |
| 20:48.56 | igcewieling | np |
| 20:49.33 | jkroon | hmm, you can improve on that still |
| 20:49.40 | *** join/#asterisk dipanjan (b495342b@gateway/web/freenode/ip.180.149.52.43) |
| 20:49.49 | jkroon | a SIP 404 response for example should remain a 404, no matter where you pass it :) |
| 20:50.08 | igcewieling | Hmm? |
| 20:50.54 | igcewieling | we have a policy to retry all calls at least once, since carriers are idiots and randomly fail calls for no reason |
| 20:50.56 | jkroon | if ${HANGUPCAUSE} == 1 then the destination doesn't exist, and this results in either CONGESTION or CHANUNAVAIL, not BUSY, ANSWER or NOANSWER. Line 74 of your code. |
| 20:51.04 | jkroon | hahaha, there is that |
| 20:51.21 | dipanjan | Can I specify conditions in a call file? For example, I want to goto a certain context if the channel is free but to some other if the channel is busy/call is waiting. |
| 20:51.52 | jkroon | dipanjan, no, but you can do similar things by originating via Local/ channels. |
| 20:52.13 | igcewieling | though I really should have some Wait(x)s in there between dial attempts to let any transient issues clear. |
| 20:52.41 | dipanjan | jkroon: Ah! I see. More learning curve :( Was sailing with the call files pretty smoothly! |
| 20:52.47 | dipanjan | jkroon: thanks! |
| 20:53.09 | igcewieling | Asterisk is like sex -- there is ALWAYS more to learn. |
| 20:53.27 | jmetro | youre doing it wrong <.< |
| 20:53.33 | tzanger | that should go on the cover of the next TFOT |
| 20:53.45 | tzanger | where's leifmadsen |
| 20:53.45 | igcewieling | tzanger: 8-) |
| 20:53.45 | dipanjan | igcewieling: +1 |
| 20:55.04 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 21:00.08 | jkroon | newtonr, mjordan - comment added - please check if this makes sense so that I can update if needed. Or perhaps if I missed anything: https://issues.asterisk.org/jira/browse/ASTERISK-21378#comment-204927 |
| 21:00.09 | LieutPants | [ASTERISK-21378] [Status: Triage] chan_sip blocks on DNS lookups - causing severe delays with registrations in certain scenarios - https://issues.asterisk.org/jira/browse/ASTERISK-21378 |
| 21:01.06 | jkroon | hmm, how's this for a "quick fix", *always* fail the DNS lookup and then update it via the callback once we get a result? |
| 21:01.09 | *** join/#asterisk _Corey_ (~corey@64.215.11.114) |
| 21:01.42 | jkroon | will probably cause more chaos than good so i'll scrap that for now ... |
| 21:02.21 | WIMPy | thinks that might be an idea |
| 21:02.33 | WIMPy | At least the first half :-) |
| 21:03.15 | *** join/#asterisk iprouteth0 (ccf60469@gateway/web/freenode/ip.204.246.4.105) |
| 21:03.26 | jkroon | newtonr, severe delays ... no, that's what you're not getting - I'm able to put asterisk into a FULL denial of service state with this. |
| 21:03.36 | jkroon | as in, chan_sip WILL NOT FUNCTION at all. |
| 21:03.37 | iprouteth0 | gotta say google calling with asterisk 11/motif definitely is a bit weird |
| 21:04.11 | WIMPy | I have an documentation issue. There doesn't seem to be any official syntax for switches. What I found says there's a pattern to match at the end but that doesn't seem to work. Can anyone shed some |
| 21:04.18 | WIMPy | light on that? |
| 21:05.13 | WIMPy | That seems to be contributing big parts to asterisk losing digits when dialing too fast. |
| 21:06.22 | iprouteth0 | Does google send the dialed digits at all or how does chan_motif identify between connections. I'm currently answering call with s,1 whereas in 1.8 I used to use username@gmail.com,1 |
| 21:06.24 | jkroon | i've found that my "losing digits" issue was related to FXS channels where I tried to utilize _[*0-9]! style patterns and the ! ended the reading of digits |
| 21:07.05 | WIMPy | is not using anythign analog. |
| 21:07.11 | file | iprouteth0, ; Incoming calls will first look for the extension matching the name of the endpoint in the configured context. If |
| 21:07.11 | file | ; no such extension exists the call will automatically fall back to the "s" extension. |
| 21:07.25 | WIMPy | And the ! surely shouldn't end reading digits. |
| 21:07.26 | jkroon | WIMPy, ISDN or sip with overlapdial? |
| 21:07.44 | jkroon | WIMPy, read the documentation carefully |
| 21:07.54 | iprouteth0 | I have the connection named only with my username, not username@gmail.clm |
| 21:07.55 | jkroon | the ! will match immediately and enter the dialplan. |
| 21:08.06 | WIMPy | The timing issue comes with ISDN. With SIP I get the wrong pattern matched sometimes. |
| 21:08.10 | file | iprouteth0, then use the username as the extension in the dialplan context you have it going to |
| 21:08.17 | file | without gmail.com that is |
| 21:08.54 | WIMPy | jkroon: Yes, it will match, but that doesn't automatically end dialling. Except for chan_sip which has to switch to DTMF at that point. |
| 21:08.57 | iprouteth0 | I'll give it a shot. I was wondering if that might be the case, and just as you say the s,1 is the fallback mechanism. Just wanted to bounce my idea's around before I go breaking things again |
| 21:09.13 | WIMPy | That's extremely annoying, but probably almos impossible to fix with chan_sip. |
| 21:09.18 | file | runs off |
| 21:09.34 | jkroon | WIMPy, switch off overlapdialing ... |
| 21:09.48 | newtonr | jkroon: I feel like DoS generally has the connotation of a purposeful attack. I get what you are saying. If you can think of a better way to word the title feel free to edit it. |
| 21:09.49 | WIMPy | No go |
| 21:10.11 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:10.21 | WIMPy | 1. I definitely don't want to and 2. it's not even always possible. |
| 21:10.34 | jkroon | newtonr, i hear you, and this is a hard one to purposefully exploit, unless you can inject some hostname into the victim system. |
| 21:11.02 | igcewieling | you europeans with your variable length phone numbers |
| 21:11.17 | jkroon | WIMPy, well then, just make sure of the difference between . and !, ! matches 0 digits, . matches 1 *or more* |
| 21:12.03 | jkroon | igcewieling, yea, 10-digit telephone numbers in SA is awesome, can match on 0[1-578]XXXXXXXX and I can know the whole country is covered :D |
| 21:12.13 | WIMPy | jkroon: I know. But chan_sip will match the shortest extension, not the most specific one. |
| 21:12.36 | jkroon | with ! yes it will, with . it will match most specific. |
| 21:13.11 | WIMPy | That's only true for chan_sip. |
| 21:13.29 | igcewieling | dialplan pattern matching should have nothing to do with chan sip |
| 21:13.30 | WIMPy | Other channels and WaitExten do the most specific thing. |
| 21:13.45 | WIMPy | True, but it does. |
| 21:14.04 | jkroon | no, WaitExten should function exactly the same irrespective of channel technology being used. |
| 21:14.17 | jkroon | if it does not, that's a bug. |
| 21:14.23 | WIMPy | jkroon: Yes. |
| 21:15.19 | WIMPy | Now I broke sip completely :-( |
| 21:17.37 | newtonr | don't file any more bugs :P not allowed |
| 21:17.49 | jkroon | what? my bugs are too hard to fix? |
| 21:18.07 | WIMPy | newtonr: Can I spam you instead? :-) |
| 21:18.43 | newtonr | jkroon: no more! |
| 21:18.48 | newtonr | WIMPy: as long as it's entertaining! |
| 21:19.21 | WIMPy | newtonr: D'oh. Extra work. |
| 21:19.51 | newtonr | jkroon: i linked your issue to five others! i think yours has most detail out of them all. if only i could give out gold stars. |
| 21:20.19 | jmetro | you could give out high fives |
| 21:20.29 | newtonr | high fives jkroon |
| 21:20.33 | jmetro | http://www.maniacworld.com/internet-high-five.jpg |
| 21:20.51 | newtonr | nonono don't touch my monitor. i hate it when people do that |
| 21:20.55 | newtonr | greasy prints all over the monitor |
| 21:20.59 | jkroon | newtonr, i think i've filed enough bugs for the year. at least I try to provide patches along with bug reports. |
| 21:21.00 | jkroon | not planning to file any more this week :p |
| 21:22.16 | newtonr | jkroon: your bug reports are definitely way easier to triage than much of the amazingly horrid stuff we see |
| 21:23.46 | jkroon | rofl @ internet high-five. thanks newtonr - and always a pleasure to help improve an already amazing piece of software (in spite of some of my nastier bug reports) |
| 21:24.45 | newtonr | we can only fix bugs that get reported! so it's definitely a huge help. |
| 21:26.24 | newtonr | maybe at the next Astricon we should give a shout out to some of the top bug reporters for the year |
| 21:26.48 | WIMPy | Then we need more documentation. |
| 21:27.00 | jkroon | when is the next astricon? and where? |
| 21:27.20 | WIMPy | I thinks I have enough stuff to report where I don;t know if it's a bug because I cannot find out what the intended behaviour is. |
| 21:27.50 | newtonr | WIMPy: i see a ton of that too. we need more people providing documentation patches. I've been trying to push issues through even if its just for documentation reasons. |
| 21:28.30 | jkroon | WIMPy, i find that the source code (other than chan_sip ...) is mostly readable and usually clarifies the behavior. |
| 21:29.10 | newtonr | jkroon: not for users and admins who are not developers |
| 21:29.10 | WIMPy | newtonr: But how con we contibute if we don;t have the information? |
| 21:29.47 | jkroon | newtonr, yea, guess i'm a bit of a somewhat unique freak in that respect. |
| 21:30.11 | WIMPy | jkroon: I can find the current behaviour by experimenting, But I still won;t know if it's intended or not. |
| 21:30.11 | newtonr | WIMPy: just open a bug report and make sure you mention that the issue isn't clear due to lack of documentation. Then we can throw it in the queue for someone to clarify. We need more community help with that though. |
| 21:30.19 | jkroon | ntw newtonr - the new subject is definitely more appropriate. |
| 21:30.49 | newtonr | jkroon: sweet |
| 21:31.11 | WIMPy | ok |
| 21:31.17 | newtonr | WIMPy: often we don't know what the original intent behind a feature or behavior was... sometimes the most we can do is at least document "this is how it works" |
| 21:31.41 | igcewieling | UPGRADE-*.txt, the sample Asterisk config files, output of "core show applications" and "core show functions", and ATFOT is about as close official documentation as you can get |
| 21:32.13 | WIMPy | Yes, but that only documents some aspects of Asterisk. |
| 21:32.36 | igcewieling | WIMPy: I feel it documents most aspects of asterisk |
| 21:32.59 | WIMPy | I fond enough things that are unclear. |
| 21:33.09 | newtonr | i find the most things lacking is documentation for specific .conf file options |
| 21:33.18 | newtonr | yeah and general lack of clarity |
| 21:33.20 | WIMPy | Like the switches I just mentioned. |
| 21:33.42 | iprouteth0 | for xmpp.conf, the priority setting lacks proper comments, but is very important |
| 21:33.51 | igcewieling | WIMPy: switch => seems to be a feature not often used. |
| 21:34.06 | WIMPy | But it's absolutely essential. |
| 21:34.10 | newtonr | iprouteth0: submit a documentation patch to issues.asterisk.org! |
| 21:34.39 | igcewieling | WIMPy: depends on how you design things. most people do not use switch => |
| 21:34.50 | iprouteth0 | @netwonr: in due time perhaps. I've send some info over to malcolmd@digiuim regarding the wiki documentation |
| 21:35.44 | WIMPy | igcewieling: Does only work if you only want to place national calls in a country with a closed numbering plan. |
| 21:35.55 | newtonr | WIMPy: in general, mjordan and the team are currently focusing a lot on making sure current work and especially new features are being well documented. For all the old stuff out there, we can always use more help from the community. So yell at everyone you see complain to submit a documentation patch or at least a documentation "bug". |
| 21:35.56 | igcewieling | I should revise that to: most people on #asterisk and asterisk-users mailing list don't use switch => If they did there would be many more questions about it. |
| 21:36.22 | malcolmd | yeah, sorry, i haven't been able to get to it yet, apologies. it's in my queue - regarding the xmpp / motif priority |
| 21:36.25 | WIMPy | newtonr: ok |
| 21:36.34 | igcewieling | WIMPy: I don't see how calling countries with variable length dialplans would require using switch => |
| 21:36.58 | iprouteth0 | malcolmd: no worries, I know you're often busy :) |
| 21:37.23 | WIMPy | switches are the only way to change extensions *while* dialing. |
| 21:38.06 | igcewieling | no, they are not |
| 21:38.18 | WIMPy | Tell me anoter one. |
| 21:38.38 | igcewieling | don't change extensions during dialing. |
| 21:38.43 | newtonr | jkroon: http://www.asterisk.org/community/astricon-user-conference details will be there. October. |
| 21:38.56 | WIMPy | *Bzzzt* |
| 21:39.00 | igcewieling | if by extensions you mean different patterns for exten => |
| 21:39.29 | igcewieling | Asterisk's support for variable length dialing isn't all that great anyway, so you don't have a lot to work with. |
| 21:39.30 | WIMPy | No, the thing is that I want to canonicialise the user input before processing it. |
| 21:39.49 | igcewieling | switch => makes asterisk query another server to find out if it handles that specific TN. |
| 21:39.58 | jkroon | WIMPy, then use Read() perhaps? |
| 21:40.09 | WIMPy | It's bad if your extensions don't match any more just because someone decided to dial the area code even though it was a local call. |
| 21:40.11 | jkroon | process it and then use it as part of a goto statement? |
| 21:40.52 | WIMPy | True, there are quite some issues when you try to get dialing to work. |
| 21:40.59 | igcewieling | We use an AGI to canonicialise user dialed TNs (as well as lots of other stuff) |
| 21:41.05 | jkroon | WIMPy, fortunately someone decided a long time ago in SA that 7-digit dialing will no longer function, and everyone is using 10-digit numbers, even for local calls now :) that helps. |
| 21:41.13 | WIMPy | And there are several kinds of switches. Loopback would be the one in the above example. |
| 21:41.25 | WIMPy | But I use dundi as well. |
| 21:41.35 | igcewieling | It sounds like your problem is more with Asterisk's support for variable length dialplans and not with switch => |
| 21:41.45 | WIMPy | jkroon: Goto stops dialing. So that's a no go. |
| 21:41.45 | jkroon | talking of dundi ... i go off to figure out what went wrong with dundi this morning. |
| 21:42.06 | jkroon | WIMPy, ok, so won't work for your situation then :). good luck. |
| 21:42.10 | igcewieling | I am SO glad we designed things to not need dundi 8-| |
| 21:42.16 | WIMPy | Same goes for AGI or anything dialplan actually. If you have a match, it's game over. |
| 21:42.45 | jkroon | igcewieling, i've got multiple servers, and a client can register to any given one. what other mechanism would you use to find them? |
| 21:43.18 | WIMPy | No, my issue with switch is that it looks like it shold have an optional pattern match, but that seems to be ignored. |
| 21:43.23 | igcewieling | jkroon: one option, why not have your client register to both of the servers |
| 21:44.03 | igcewieling | another option, if your client will accept calls from servers it is not registered to, then use Realtime so both servers know the current IP/port of the client. |
| 21:44.30 | WIMPy | The other issue is that the pbx core seems to lose digits when received while a dundi lookup is still in progress. |
| 21:44.42 | jkroon | igcewieling, because i have a significant number of clients, and i unfortunately can't control all their devices, more than half of which only supports registering to one server, and then in some cases a fall-back should the first fail. |
| 21:45.27 | jkroon | thus also why I publish SRV records, so that more of them actually utilizes the fall-backs. |
| 21:45.36 | igcewieling | I believe realtime also supports a server name, so you could find out which server the client was registered to using that and some dialplan |
| 21:45.38 | fkurkowski | I've been having some problems while trying to configure a dual digium card system. Basically, one card's FXO channels are unresponsive and only detecting nonexistent calls. Would anyone mind taking a look at this? http://pastebin.com/brMaJ2KL |
| 21:45.47 | jkroon | unfortunately, like asterisk, a lot of systems won't accept "anonymous" calls |
| 21:47.59 | jkroon | fkurkowski, multiple TDM analog cards in one system? I was tought to NOT do that because it causes problems ... which may or may not still be relevant. |
| 21:48.54 | fkurkowski | hum, that's what I was suspecting - since both of them were working properly individually. |
| 21:49.09 | jkroon | that having being said - FXO, so the PBX is providing the signalling? |
| 21:49.29 | fkurkowski | yeah, thats right. |
| 21:49.39 | WIMPy | Why shouldn't you be able to even have lots of analog cards? |
| 21:49.47 | jkroon | ok, each card is generating around 1000 interrupts per second (you can check /proc/interrupts) |
| 21:50.04 | WIMPy | Can tehy really be worse than the PRI ones? |
| 21:50.07 | jkroon | that's 2000 interrupts, and they're both sitting on the same PCI bus, albeing in different slots. |
| 21:50.54 | jkroon | WIMPy, not sure, but from what i've read it's OK to stack the ISDN cards, but NOT to stack analog - no idea why. just realized I question most things in life - but this thing I have not questioned yet. |
| 21:51.24 | jkroon | sruffell, would you perhaps be able to shed some light? |
| 21:52.03 | jkroon | the two cards isn't even sharing an interrupt, and the interrupts are being routed to different CPUs... |
| 21:52.18 | fkurkowski | I've posted the /proc/interrupts in the pastebin, each card is generating about 1kk interrupts if I'm not mistaken. |
| 21:52.55 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 21:52.59 | WIMPy | The famous timing cable should prevent the IRQs from multiplying with the cards. |
| 21:53.00 | fkurkowski | <PROTECTED> |
| 21:53.02 | jkroon | ok, i'd contact digium support if I were you |
| 21:53.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:53.35 | WIMPy | fkurkowski: Both on the same IRQ? |
| 21:53.38 | fkurkowski | yeah, I might as well do that. |
| 21:53.45 | fkurkowski | no |
| 21:53.50 | jkroon | WIMPy, i don't think the wctdm has support for timing cables ... ? |
| 21:53.54 | WIMPy | ok |
| 21:54.23 | WIMPy | has NFI. Just learned earlier on that the Sangoma analog cards do. |
| 21:54.38 | jkroon | fkurkowski, why didn't you get yourself an 8-port card? |
| 21:55.14 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 21:55.14 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:55.15 | fkurkowski | we already had a spare 4 port one |
| 21:55.31 | sruffell | looks back |
| 21:56.01 | fkurkowski | and since we had never entered this grounds, thought it wouldnt matter |
| 21:56.43 | jkroon | out of my depth, hopefully one of the other folks here can sort you out |
| 21:57.30 | WIMPy | fkurkowski: As jkroon suggested, try to make sure the IRQs go to different cores. |
| 21:57.39 | fkurkowski | i'm almost giving up, spent the last two weeks trying on different systems, os, hardware. |
| 21:57.55 | jkroon | WIMPy, they already do. |
| 21:58.02 | fkurkowski | they are right now |
| 21:58.08 | igcewieling | You can put multiple modern Digium cards in a server without issue |
| 21:58.26 | sruffell | stacking analog cards is fine from a physical perspective. The newer cards can handle small variations in interrupt latency (as opposed to the original TDM400). What can cause problems are "noisy" systems that are jammed with a bunch of analog cards. Is the FXO port that's not working on the TDM400 by any chance? |
| 21:58.33 | jkroon | fkurkowski, grab the serial numbers off the cards, go to support.digium.com, register the cards and contact digium direct. they're pretty much on the ball. |
| 21:58.40 | sruffell | I looked at the pastebin and there is a wctdm driver loaded. |
| 21:58.49 | jmetro | get on the ball. |
| 21:58.59 | fkurkowski | sruffell: no, its on the TDM410 |
| 21:59.45 | sruffell | looks again |
| 22:00.38 | igcewieling | jkroon: we designed our system from the ground up to be simple, possible because of our business model -- you need a static ip if you want our service and you generally need our equipment and we REALLY want you to be on-net for QoS. Makes things SO much more simple. |
| 22:01.01 | fkurkowski | i'm going to test both cards individually again just to make sure they are ok |
| 22:01.41 | sruffell | fkurkowski: ok…also, I didn't see any dmesg output in the pastebin. If you could paste that too.... |
| 22:01.54 | sruffell | (and I didn't notice what version of DAHDI you were using) |
| 22:01.55 | fkurkowski | sure |
| 22:02.19 | [TK]D-Fender | yup, so was there sun & Tue |
| 22:05.33 | *** join/#asterisk serafie (~erin@nat/digium/x-bpvfktzlfllxtuxz) |
| 22:06.24 | zamba | nker vi totalt 4000kr. Det kan deles opp i to innbetalinger, men betal for satan inn 2000kr så snart som mulig slik at vi får ordnet hastesaker billigst mulig. Overfør til konto 97105682690 (Roksøy). |
| 22:06.38 | zamba | gah.. wrong paste :) |
| 22:06.40 | zamba | "chan_sip.c:22546 in handle_request_invite: Sending fake auth rejection for device 202" |
| 22:06.48 | zamba | that.. what do these mean? |
| 22:07.14 | WIMPy | Money! |
| 22:07.37 | WIMPy | You enabled alwaysauthreject. |
| 22:07.46 | zamba | allowguest=no |
| 22:07.49 | zamba | alwaysauthreject=yes |
| 22:07.56 | zamba | i have those two in sip.conf |
| 22:08.13 | zamba | shouldn't those block these? |
| 22:08.54 | *** join/#asterisk thansen (~thansen@63-248-43-122.static.orml012.digis.net) |
| 22:09.09 | WIMPy | Quite the opposite. |
| 22:09.38 | thansen | I assume asterisk/sip with work with Yealink IP Video Phone w/HD Voice (YEA-VP530)? |
| 22:09.47 | zamba | i have allowguest = no.. that should be a safety precaution? |
| 22:09.48 | thansen | to do video specifically |
| 22:10.19 | thansen | additionally, does anyone have other recommendations for a hardware video phone? |
| 22:10.50 | zamba | http://www.voip-info.org/wiki/view/Asterisk+security |
| 22:10.51 | zamba | reading that |
| 22:10.59 | fkurkowski | sruffell: i'm using the latest DAHDI version. the cards are unfortunetely not on the original system anymore. however there is a very interesting message on dmesg now (i don't remember it on the former system though, gonna have to check it out tomorrow) |
| 22:11.04 | fkurkowski | wctdm24xxp 0000:01:05.0: ERROR: Unable to service card within 25 ms and unable to further increase latency |
| 22:11.23 | igcewieling | a bad day: http://imagebin.org/252694 |
| 22:11.26 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-tgjcyvgnjoxehknc) |
| 22:11.54 | WIMPy | zamba: voip-info can be a good starting point, but don't take things there for more than that. |
| 22:12.24 | zamba | WIMPy: then what is correct? |
| 22:12.59 | WIMPy | Using your brain :-) |
| 22:13.09 | igcewieling | thansen: the polycom VVX1500 is popular. Cameras have been released for the VVX600 and VVX500 recently. |
| 22:13.13 | zamba | well.. allowguest and alwaysauthreject seems to be set to sound values |
| 22:13.30 | WIMPy | Generelly wiki.asterisk.org is the better choice. |
| 22:13.48 | thansen | igcewieling: and asterisk/sip support video right? |
| 22:13.53 | sruffell | fkurkowski: yeah…that could explain what you're seeing. Something is preventing the card from servicing the interrupt quickly enough. Check for framebuffers, or disk in PIO mode. |
| 22:14.23 | fkurkowski | will do, thanks for your help |
| 22:14.25 | igcewieling | thansen: yes. We use a few VVX1500s, but I do not personally use them |
| 22:14.26 | sruffell | but it's most likely *not* the card. I bet the audio on the TDM400 sounds bad... |
| 22:14.30 | WIMPy | fkurkowski: How many cores do you have? |
| 22:14.36 | fkurkowski | two |
| 22:14.58 | sruffell | fkurkowski: what is the output of "cat /proc/cmdline" |
| 22:15.16 | fkurkowski | root=/dev/hda1 ro quiet |
| 22:15.47 | thansen | igcewieling: that baby is spendy! nice |
| 22:15.47 | sruffell | hda…I bet that's your problem. What is the output of /sbin/hdparm -t /dev/hda |
| 22:16.29 | igcewieling | yes, they are. look at the VVX500 and VVX600, polycom recently released camera add-ons for those models. |
| 22:16.43 | igcewieling | and they are a lot cheaper |
| 22:20.33 | fkurkowski | meh, I lost access to the system and there's no one at the office right now. but i dont think the original machine is hda, i issue the cat /proc/cmdline on the older one. |
| 22:21.30 | sruffell | ok…if the output of hdparm -t is low (like 5MB/s) then that most likely is the cause. If it's a SATA drive you can sometimes adjust things in the BIOS settings to put the controllered in native mode. |
| 22:22.07 | iprouteth0 | file: Thanks BTW. username without domain was it |
| 22:22.48 | fkurkowski | yeah im pretty sure its a sda, will take a look at the BIOS tomorrow |
| 22:22.57 | sruffell | kk |
| 22:23.42 | jkroon | fkurkowski, to check the disk IO mode use hdparm -I ... it'll tell you whether it's using DMA or PIO. |
| 22:23.47 | jkroon | PIO bad, DMA good. |
| 22:24.08 | fkurkowski | taking notes |
| 22:24.56 | sruffell | jkroon: sometimes I think with controllers in "COMBINED" mode that won't tell you how it's *really* configured. /sbin/hdparm -t does though becuase only PIO has such poor performance. |
| 22:26.22 | jkroon | i have never seen a COMBINED mode... wasn't even aware it existed. i've learned much today. |
| 22:27.07 | file | You are welcome. |
| 22:27.24 | jkroon | just outright prefers AHCI for anything SATA, and pretty much all SCSI controllers outperform SATA again, so that's my bottom line at the moment - if you give me AHCI incapable SATA - find someone else to support you, unless it's SCSI. |
| 22:28.29 | sruffell | heh |
| 22:31.05 | jkroon | broke DAHDI ... somehow I've managed to desync the public and private keys ... |
| 22:31.30 | jkroon | ok, does asterisk provide a convenient way of confirming that a specific public key is the right key for a given private key? |
| 22:34.40 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 22:34.41 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 22:37.45 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 22:37.46 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:45.13 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 23:02.29 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 23:06.05 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 23:17.56 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:755f:a004:67b8:7d00) |
| 23:25.29 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 23:29.52 | *** join/#asterisk Micc (~Mic@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 23:34.48 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 23:39.07 | *** part/#asterisk ghost75 (~trechber@dslb-088-066-173-226.pools.arcor-ip.net) |
