| 00:01.07 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 00:04.26 | simplydrew | p3nguin: in my tftpboot directory, can I create a separate directory for the 7965 firmware files without running into issues? |
| 00:04.43 | simplydrew | since the sip firmware for the 7960's is in that directory itself, wasn't sure how I wanted to differentiate |
| 00:04.45 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-eaamqxapfyrqyozj) |
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| 00:09.21 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 00:16.26 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 00:16.52 | atan | Anyone happen to have Polycom IP300 config files for ftp/network config? |
| 00:20.11 | carrar | http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip300.html |
| 00:20.23 | atan | That's where I'm at now actually ^_^ thanks :D |
| 00:20.47 | atan | Happen to know if you can somehow warm transfer on the IP300? |
| 00:21.02 | carrar | What is a warm trasnfer |
| 00:21.09 | atan | Say you have a customer on line and want to transfer to someone else. You call then, talk, link the calls, introduce and then exit your portion but leave them connected |
| 00:21.43 | atan | The Cisco IP 7960 does it without any issue but I can't get the same thing to happen on the IP300 |
| 00:27.42 | p3nguin | simplydrew: There are some ways to use directories, like setting the directory in the phone itself, or setting it in the default file for loading the phone-specific files. |
| 00:28.18 | simplydrew | p3nguin: for the time being I just copied them into the root. created the config file for the phone, however it doesn't seem to be pulling an IP |
| 00:28.35 | simplydrew | it did the first time I plugged it in, but after I tried booting it again it seems like it's hung up |
| 00:28.47 | p3nguin | That's not the responsibility of the tftpd. |
| 00:29.11 | simplydrew | I know that. I'm just trying to figure out what's wrong in general. |
| 00:29.19 | simplydrew | Don't seem to have that problem with the 7960 |
| 00:40.18 | *** join/#asterisk nix8n82-phone (~AndChat@65.161.180.230) |
| 00:41.01 | *** join/#asterisk JuStIcIa_ (~artur0@190.167.46.103) |
| 00:54.13 | simplydrew | p3nguin: any further insight? |
| 00:58.43 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:17.01 | p3nguin | simplydrew: Check your dhcpd to make sure it can give out new leases; check your phone to make sure dhcp is enabled. |
| 01:17.17 | p3nguin | (not necessarily in that order) |
| 01:22.23 | *** join/#asterisk delki8 (~delki8@189.5.136.31) |
| 01:24.38 | simplydrew | p3nguin: looks like both are fine. when I look in the settings, it's pulling the right information. however, it's not pointing to the asterisk IP for tftp |
| 01:25.11 | p3nguin | You'll need to either override the tftp setting in the phone or set the right options in the dhcpd. |
| 01:25.25 | p3nguin | If you have only one phone, I'd set the alternate tftp in the phone for now. |
| 01:26.02 | simplydrew | p3nguin: that's what I was trying to do. I hit the key sequence to unlock the phone to change the settings, but it wouldn't let me hit the "edit" key to actually modify it |
| 01:26.28 | p3nguin | Did you get the unlock icon on the display after entering the key sequence? |
| 01:26.50 | simplydrew | yes |
| 01:27.14 | p3nguin | Scroll way down the list to find Alternate tftp. Change it to Enable. Then scroll back up to tftp address and press Edit. |
| 01:27.32 | simplydrew | ah okay, didn't see that setting. I'll give that a try in a minute |
| 01:28.22 | p3nguin | It's tricky like that. |
| 01:40.14 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 02:08.41 | p3nguin | Is there a function that does the opposite of ISNULL? |
| 02:09.38 | p3nguin | I want to validate a variable. If it has a value, the condition would be true; if it is null, it's false. ISNULL makes a null value true, which is reverse of what I'm after. |
| 02:13.19 | p3nguin | ISNOTNULL() would be perfect. |
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| 02:17.27 | *** join/#asterisk penguin (penguin@cpe-74-77-221-5.buffalo.res.rr.com) |
| 02:17.42 | simplydrew | p3nguin: changed the tftp IP successfully. now I'm hung up at "registering". hmm |
| 02:18.01 | p3nguin | Did you install and load chan_sccp? |
| 02:18.09 | p3nguin | And configure, of course. |
| 02:18.30 | simplydrew | I followed your instructions with the svn, ./configure, make, make install |
| 02:18.40 | simplydrew | I can't seem to tell if it's running though |
| 02:18.46 | p3nguin | But did you configure your sccp.conf and load the module? |
| 02:18.56 | p3nguin | It's not magic -- it's just a channel driver. |
| 02:19.45 | simplydrew | yes, in sccp.conf I defined that it needed to load chan_sccp |
| 02:19.53 | p3nguin | Uh, no. |
| 02:20.23 | simplydrew | actually, that was modules.conf |
| 02:20.33 | p3nguin | In sccp.conf you define your devices and lines. |
| 02:21.00 | p3nguin | There are some samples to see how that's done if you aren't familiar with it already. |
| 02:21.36 | simplydrew | I have a sample, I'm just not sure where sccp.conf is supposed to be |
| 02:22.14 | p3nguin | It goes with all of the other confs in /etc/asterisk/ |
| 02:22.53 | p3nguin | You'll have a general section, a device section for each phone, and a line section for each line on each phone. |
| 02:22.53 | simplydrew | okay, that's where I had it |
| 02:22.59 | simplydrew | just wanted to make sure |
| 02:23.17 | simplydrew | I did replace the sample information with the information for my particular phone that I'm trying to get working |
| 02:23.22 | simplydrew | and it's there |
| 02:23.31 | p3nguin | If you've configured your file, and it is in /etc/asterisk, just run module load chan_sccp.so and see what happens. |
| 02:23.44 | p3nguin | If you're using a sample, it's not going to be useful to you. |
| 02:23.56 | p3nguin | Sample files are for reference, not for use. |
| 02:24.32 | simplydrew | load it how? |
| 02:24.34 | simplydrew | via the cli? |
| 02:24.37 | p3nguin | yes |
| 02:25.04 | p3nguin | I was under the impression that you'd used asterisk before today. |
| 02:25.23 | simplydrew | I have, but I haven't done specific cli intensive work |
| 02:25.35 | simplydrew | I'm mainly used to trixbox and freepbx |
| 02:25.44 | p3nguin | barfs a little |
| 02:25.59 | simplydrew | yeah, I expected that |
| 02:27.24 | p3nguin | When I configure sccp.conf, I like to use the templates. It makes adding devices and lines a lot easier. |
| 02:27.40 | p3nguin | defaultdevice, defaultline, etc. |
| 02:32.22 | p3nguin | Hmm. Anyone know if $[ ! ${ISNULL()} ] is a valid condition? |
| 02:32.32 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 02:34.52 | simplydrew | p3nguin: Unable to load module chan_sccp.so |
| 02:34.53 | simplydrew | Command 'module load chan_sccp.so' failed. |
| 02:34.54 | simplydrew | [Sep 8 22:34:23] WARNING[8429]: loader.c:446 load_dynamic_module: Error loading module 'chan_sccp.so': /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: manager_event |
| 02:34.55 | simplydrew | [Sep 8 22:34:23] WARNING[8429]: loader.c:783 load_resource: Module 'chan_sccp.so' could not be loaded. |
| 02:35.15 | p3nguin | Interesting. |
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| 02:43.54 | *** join/#asterisk dlisenby (4bb63d33@gateway/web/freenode/ip.75.182.61.51) |
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| 03:18.30 | dlisenby | hello all. quick question. I'm testing callcentric. In sip.conf do I need a section for calls coming inbound and outbound for the same SIP line? |
| 03:19.15 | p3nguin | One single entry is enough, as long as you don't use type=user. |
| 03:20.07 | dlisenby | nah type = peer |
| 03:20.58 | dlisenby | outbounds are working fine. Inbounds are giving me the "extension not found in context 'default' when my sip.conf clearly calls it "DID_callcentric" |
| 03:21.27 | dlisenby | I'd say I'm intermediate with Asterisk... not new but not a "conf" guy either. This is my first go-round with sip |
| 03:22.36 | p3nguin | Your context for callcentric is set to DID_callcentric? |
| 03:22.45 | dlisenby | yes |
| 03:22.47 | p3nguin | What is the host set to? |
| 03:22.51 | *** join/#asterisk Supersaw_Hoover (~Supersaw_@kuriboh-08.dynamic2.rpi.edu) |
| 03:23.01 | *** part/#asterisk Supersaw_Hoover (~Supersaw_@kuriboh-08.dynamic2.rpi.edu) |
| 03:23.10 | dlisenby | the context=default does exist under the general section |
| 03:23.45 | dlisenby | but further down in the sip.conf is the [callcentric] heading with context = DID_callcentric |
| 03:23.57 | dlisenby | callcentric.com |
| 03:24.02 | dlisenby | outbound is working |
| 03:27.24 | p3nguin | My concern is that they are sending your calls to you from an IP address which is not resolved via callcentric.com. |
| 03:27.24 | p3nguin | I've seen this before. |
| 03:27.24 | dlisenby | oh.. |
| 03:27.24 | p3nguin | You can find out with a sip debug while a call is coming in. |
| 03:27.24 | dlisenby | I see in the error line "204.11.192.22:5060 instead of callcentric.com:5060 |
| 03:27.24 | dlisenby | maybe I should change that? |
| 03:27.34 | p3nguin | I'm thinking it's a problem with SRV records. |
| 03:27.41 | p3nguin | That IP address is alpha1. |
| 03:28.00 | p3nguin | It's one of their hosts used in the sip SRV. |
| 03:28.06 | dlisenby | not sure I know what you mean? alpha1? |
| 03:28.20 | p3nguin | alpha1.callcentric.com |
| 03:29.54 | dlisenby | ah |
| 03:30.15 | dlisenby | just changed it to the ip mentioned above. tried it again, now ip is 204.11.192.36 |
| 03:30.16 | p3nguin | If it is a problem with how asterisk is doing SRV lookups, the only way around it that I can think of is to define a peer for each server. I'd use templates to do it cleanly. |
| 03:30.34 | p3nguin | Is that the only IP address in the call? |
| 03:30.43 | p3nguin | Is there ever a second IP address on any of these calls? |
| 03:30.46 | dlisenby | hmm.. wonder if callcentric will show me all their potential server ips |
| 07:59.39 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 07:59.39 | *** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 08:15.07 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e2.bcn.adamo.es) |
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| 08:40.00 | *** join/#asterisk DanFromUK (DanFromUK@2.27.26.159) |
| 08:40.45 | DanFromUK | hi, has anyone set up a Cisco 7945 to run through asterisk? |
| 08:44.45 | *** join/#asterisk irroot (~irroot@196-215-124-168.dynamic.isadsl.co.za) |
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| 09:00.42 | *** join/#asterisk danfromuk (DanFromUK@2.27.27.75) |
| 09:01.07 | danfromuk | is it difficult to get a cisco 7945g phone to connect to asterisk? is there anything i should be aware of? |
| 09:01.33 | wdoekes2 | danfromuk: perhaps you should try and come back when you have a specific problem |
| 09:08.14 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 09:08.18 | eject_ck | Hi all |
| 09:08.20 | eject_ck | just get segfault at 7f1c11be94c4 ip 7f1c1196f6a2 sp 40f96228 error 7 in libspandsp.so.1.0.0[7f1c1193f000+a2000] |
| 09:08.27 | eject_ck | res_fax ? |
| 09:13.27 | *** join/#asterisk BuenGenio (~Gene@228.Red-81-37-22.dynamicIP.rima-tde.net) |
| 09:28.51 | wdoekes2 | which spandsp version? |
| 09:29.01 | wdoekes2 | eject_ck |
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| 09:50.22 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 09:50.33 | Boardy | I have 3 "register =>" directives in my sip.conf. Is it possible to set a different expiry for one of them? |
| 09:52.17 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 09:53.48 | kaushal | is there a converter for converting mp3 to ulaw available on linux ? |
| 09:55.07 | singler | kaushal: try sox |
| 09:59.35 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-76.telkomadsl.co.za) |
| 09:59.37 | wdoekes2 | Boardy: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] |
| 09:59.55 | jkroon | hi guys, i'm trying to get streaming audio working from moh. |
| 10:00.30 | jkroon | I'm not sure what it means if the moh process is in state T in top, looks like "stopped", which seems wrong, but would explain why I'm not getting any audio? |
| 10:01.37 | wdoekes2 | jkroon: is it possibly writing to stderr too? (and hanging because no one reads its output?) |
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| 10:09.09 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 10:10.28 | jkroon | wdoekes2, i guess. |
| 10:10.45 | jkroon | that would make sense, although, with arecord passing -q typically makes it quiet. |
| 10:11.05 | wdoekes2 | wrap a script around it with 2>/dev/null and see if it helps |
| 10:13.08 | jkroon | seems to work ... strange. |
| 10:14.16 | jkroon | yes thanks! |
| 10:14.55 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
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| 10:45.58 | devil_evoxxx | Hi, i've got a Main asterisk Machine 1.4 that is connected with a provider. |
| 10:45.58 | devil_evoxxx | This machine has got it's pubblic ip and is connected with an other asterisk (1.8.6.0) machien via sip trunk. The second machine ( asterisk 1.8.6.0) is natted 1:1 on a pubblic ip and the local ip of this machine is 172.16.171.10. |
| 10:46.02 | devil_evoxxx | Ok, now Let's call the main asterisk machine as "A" and the second machine (asterisk 1.8.6) as "B". |
| 10:46.05 | devil_evoxxx | The configuration as A side for connecting B machine is here: http://pastebin.com/PWRyAdZCc |
| 10:46.08 | devil_evoxxx | The Configuration as B side for connecting do A machine is here : http://pastebin.com/unM7n08C |
| 10:46.11 | devil_evoxxx | On the B machine i've setted in the general section : bindaddr=172.16.171.10 and externip= 213.203.123.227. |
| 10:46.14 | devil_evoxxx | The problem is that i still have a unidirectional audio. The call diagram is |
| 10:46.16 | devil_evoxxx | Mobile Phone Call #number# the call comes from provider to machine "A" . "A" forward call to B machine but the callee can hear , but the caller can't hear the callee |
| 10:46.56 | devil_evoxxx | i'm dumping udp packet and the rtp stream try to talk directoly with provider and not with machine A |
| 10:47.29 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 11:00.42 | schmidts | devil_evoxxx stupid question but have you configured machine B with nat=yes? |
| 11:01.29 | schmidts | devil_evoxxx sorry my fault on B you can see in the udp streams that B tries to talk directly with your provider, then you should try to disable directmedia |
| 11:03.50 | devil_evoxxx | yes on machine b is set to nat=yes |
| 11:03.54 | devil_evoxxx | now i try directmedia |
| 11:04.32 | schmidts | devil_evoxxx on machine A you also have to set nat=yes for the B peer, but imho it sounds like directmedia |
| 11:05.25 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 11:05.53 | devil_evoxxx | on machine A i set nat=yes for B peer, on the B machine i set nat=yes for A peer ( i'm not sure ) |
| 11:07.06 | devil_evoxxx | is correct? |
| 11:08.34 | devil_evoxxx | schmidts i've to set directmedia in sip general section? |
| 11:09.20 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 11:11.00 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 11:11.57 | devil_evoxxx | schmidts i've to set directmedia on A machine or B machine? |
| 11:16.16 | schmidts | yes the general section in your sip.conf |
| 11:17.19 | Boardy | wdoekes2: When I list the registry, I still see the default, not the 60s I want (I'm using v1.6.2.9 (Debian squeeze)) |
| 11:24.41 | devil_evoxxx | schmidts i've setted directmedia=no |
| 11:26.34 | devil_evoxxx | but |
| 11:26.55 | devil_evoxxx | enabling rtp debug i still have |
| 11:26.59 | devil_evoxxx | the same problem |
| 11:27.36 | devil_evoxxx | the rtp stream try to talk directly with provider and not with other asterisk machine |
| 11:31.43 | *** join/#asterisk irroot (~irroot@41.221.5.170) |
| 11:31.54 | schmidts | devil_evoxxx could you pastebin a sip debug of machine B |
| 11:32.58 | devil_evoxxx | sure |
| 11:33.39 | schmidts | devil_evoxx btw i am looking at your pastebin of the machine B sip config at you have directmedia=nonat in there |
| 11:33.45 | schmidts | have you tried to replace it with no? |
| 11:35.05 | devil_evoxxx | 1 |
| 11:35.24 | devil_evoxxx | http://pastebin.com/PGQNMKj0 |
| 11:36.00 | schmidts | devil_evoxx i mean a sip debug not rtp debug ;) |
| 11:36.32 | devil_evoxxx | i'm "melted" ..i'll do in a moment |
| 11:40.40 | devil_evoxxx | here the sip debug http://woki.as48500.net/dump.txt |
| 11:44.03 | Boardy | In sip.conf I register with "register => user:secret@host/extension~60, but the used expiry is still the same as the defaultexpiry. |
| 11:44.17 | schmidts | devil_evoxx 87.13.67.31 is machine A or B? cause if its A then i guess the problem is this: c=IN IP4 172.16.13.4 |
| 11:45.29 | devil_evoxxx | the ip 87.13.67.31 is the calleee |
| 11:45.31 | devil_evoxxx | callee |
| 11:46.33 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:47.18 | schmidts | so this: http://pastebin.com/ShfavNGQ is the invite from A to B right? |
| 11:49.59 | devil_evoxxx | the dump was made on b machine and, yes is the invite from A to B |
| 11:51.33 | schmidts | as i said the problem is A acts also like its behind NAT the rtp data is wrong from A |
| 11:52.34 | *** join/#asterisk poison (~poison@router.begen1.office.netnoc.eu) |
| 11:53.17 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:ec3d:ac06:21a2:ec00) |
| 11:53.52 | devil_evoxxx | shit..A is not behind nat, it'has his own ip address |
| 11:54.09 | devil_evoxxx | directly configured on the eth interface |
| 11:57.04 | devil_evoxxx | so, it can be the dirty connection track table |
| 11:57.11 | devil_evoxxx | on the router that have behind B machine? |
| 11:57.26 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 11:58.25 | devil_evoxxx | schmidt i've solved the problem |
| 11:58.27 | devil_evoxxx | ill'set |
| 11:58.35 | devil_evoxxx | canreinvite=no on the asterisk 1.4 machine (A side) |
| 11:58.38 | devil_evoxxx | now its ok.. |
| 12:00.12 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 12:03.03 | ijpalmer | I'm using realtime queues and want to use hints for my device states. When I add 'HINT:1641:devstate-test' to the state_interface column of the queue member table queue show shows that device as invalid and no calls get delivered. If I remove that field all is ok. Is the syntax wrong? |
| 12:03.32 | ijpalmer | oops sorry meant to be @ d |
| 12:05.53 | poison | hi all, how can I rewrite numbers so if I call local numbers 0xxxxxxx that they map to the international variant: +33xxxxxxx ? |
| 12:06.04 | aberrios | hmm, I have a problem where a user logs into a queue, received one call, then after the call has ended the device is constantly in use and eh doesn't get another call from the queue |
| 12:06.21 | aberrios | remove them from the queue and add them again same happens. one call then constantly busy. |
| 12:10.50 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:11.44 | devil_evoxxx | poision, you can rewrite ${EXTEN} |
| 12:12.46 | poison | devil_evoxxx: any clue how? |
| 12:16.35 | tuxx- | hiya, whats the best way to let a call jump out of the queue after 15 seconds? I tried giving the timeout parameter to the Queue() application, but the caller stays in the queue even after the timeout has exceeded. |
| 12:19.15 | tuxx- | oh nvm |
| 12:19.24 | tuxx- | i forgot a , in the Queue() application ;-P |
| 12:19.32 | tuxx- | *facepalm* |
| 12:25.52 | devil_evoxxx | something likes Set(${EXTEN}=+33${EXTEN:1} |
| 12:26.23 | devil_evoxxx | look ${EXTEN:1} i not remember how work substring |
| 12:27.03 | poison | ok I'll try, tnx! |
| 12:27.06 | kaldemar | don't rewrite EXTEN, it causes a jump in the dialplan. |
| 12:27.34 | kaldemar | Set(${EXTEN}=+33${EXTEN:1}) won't work anyway, you'd have to use Set(EXTEN=+33${EXTEN:1}) |
| 12:28.59 | leifmadsen | save the data to another channel variable first |
| 12:29.04 | kaldemar | do it in the dial command, e.g. exten => _0X.,1,Dial(tech/peerorchan/+33${EXTEN:1}) |
| 12:29.15 | leifmadsen | Set(thisExten=${EXTEN}) |
| 12:29.27 | kaldemar | or use another variable, depends on the surrounding dialplan. |
| 12:29.36 | leifmadsen | Dial(${GLOBAL(myITSP)}/+33${thisExten:1}) |
| 12:30.17 | leifmadsen | or use a subroutine (GoSub()) and pass ${EXTEN} as an argument |
| 12:38.08 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 12:39.26 | *** join/#asterisk bmg505 (~leon@196-209-7-35.dynamic.isadsl.co.za) |
| 12:39.35 | ijpalmer | what is the syntax for the state_interface using realtime queue members, I want it to be a hint using local/XXX@XXXXXX |
| 12:41.41 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 12:42.48 | irroot | ~thebook |
| 12:42.48 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 12:48.03 | *** join/#asterisk psilikon (~joel@rrcs-24-73-84-158.se.biz.rr.com) |
| 12:50.16 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 13:01.38 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 13:01.53 | *** part/#asterisk billmania (~bill@38.98.130.98) |
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| 13:28.20 | Boardy | Any ideas? I register with a specific expiry (~60), but after a "sip reload" allways the default registry is used (and the provider uses 60s, so after that period I'm disconnected) |
| 13:28.52 | Boardy | should be: default EXPIRY is used |
| 13:29.06 | *** join/#asterisk GreatSUN (~greatsun@188.20.12.162) |
| 13:29.07 | GreatSUN | re |
| 13:29.27 | Katty | hellloooo my pretties!!! |
| 13:31.51 | *** join/#asterisk Fritz09 (~Adium@pop1-3620.catv.wtnet.de) |
| 13:34.23 | Katty | where is everyone |
| 13:34.29 | Boardy | I'm here. |
| 13:34.39 | leifmadsen | Katty: I'm at home |
| 13:37.35 | *** join/#asterisk cyborg-one (1000@85-238-116-237.broadband.tenet.odessa.ua) |
| 13:38.32 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:38.48 | pabelanger | Boardy: which setting are using is sip.conf? |
| 13:38.55 | Kobaz | i'm home too |
| 13:38.57 | Kobaz | mm, yard sales |
| 13:43.47 | *** join/#asterisk bmg505 (~leon@196-209-7-35.dynamic.isadsl.co.za) |
| 13:43.55 | chuckf | was working |
| 13:44.06 | *** join/#asterisk bchia (~Adium@nat/digium/x-dsetazqbrcjdotgi) |
| 13:45.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:46.26 | Boardy | pabelanger: defaultexpiry=7200 and register => user:pwd@host/ext~60 |
| 13:48.43 | Boardy | pabelanger: http://pastebin.com/zDkT7nrn (the general section) |
| 13:48.56 | luckman212 | what's the current consensus on fixing 'res_musiconhold.c:659 monmp3thread: Request to schedule in the past?!?!' |
| 13:50.05 | Katty | hugs leifmadsen |
| 13:50.08 | Katty | hugs chuckf |
| 14:00.20 | *** join/#asterisk BuenGenio (~Gene@186.Red-81-34-134.dynamicIP.rima-tde.net) |
| 14:00.47 | tuxx- | hiya, does the queuemember penalty also work with the strategy 'rrmemory'? |
| 14:04.33 | leifmadsen | hugs on Katty |
| 14:04.50 | Qwell | tells Mr. Roberts |
| 14:04.53 | Qwell | Mrs* |
| 14:04.57 | Qwell | innocent typo |
| 14:05.13 | leifmadsen | Qwell: Mr. Roberts would be my father-in-law :) |
| 14:05.43 | Qwell | "leifmadsen *hugged* someone. On the *Internet*!" |
| 14:05.55 | leifmadsen | Qwell: that reminds me of the time I took Bill out to lunch to ask if it would be ok for me to marry his daughter -- he thought I was going to ask him how to break up with her... |
| 14:06.03 | leifmadsen | s/Internet/internet |
| 14:06.13 | tuxx- | s/internet/interwebs |
| 14:06.20 | tuxx- | hehe |
| 14:06.22 | leifmadsen | s/interwebs/the tubes! |
| 14:06.24 | tuxx- | ;D |
| 14:06.51 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:10.12 | tuxx- | ah well, penalties for queuemembers dont work with rrmemory, it just keeps calling the person with the lowest priority ;-( |
| 14:11.05 | p3nguin | Penalties work fine, but you have to understand them. |
| 14:11.59 | luckman212 | res_musiconhold.c:659 monmp3thread: Request to schedule in the past?!?! ........ any thoughts? |
| 14:12.11 | Qwell | music is deprecated |
| 14:12.15 | luckman212 | lol |
| 14:13.54 | p3nguin | Is ringing the new moh? |
| 14:14.00 | p3nguin | or silence? |
| 14:14.37 | p3nguin | Oh, maybe talk radio is the new moh. |
| 14:16.26 | luckman212 | Milliwatt() is the new MOH |
| 14:16.35 | luckman212 | didn't u hear? |
| 14:17.26 | Qwell | playtones, man |
| 14:18.32 | luckman212 | nah, I'm 'bout to go all ZapATeller() on you mo-fo's |
| 14:18.37 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 14:18.37 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:18.47 | Qwell | luckman212: https://issues.asterisk.org/jira/browse/ASTERISK-4329 |
| 14:18.49 | p3nguin | Any specific tones you prefer playing, or should it be randomized? |
| 14:19.01 | Qwell | p3nguin: ^^ |
| 14:20.05 | p3nguin | You didn't write that, right? |
| 14:20.38 | Qwell | sure I did |
| 14:20.43 | p3nguin | I don't know what jira's terms of reporter and participant mean. |
| 14:21.08 | p3nguin | I thought you knew English better thAn that. |
| 14:21.32 | Qwell | oh burn |
| 14:22.05 | Qwell | that was like 6 years ago. I didn't care much then. |
| 14:22.59 | p3nguin | oh |
| 14:23.17 | chuckf | and you do care now? |
| 14:23.49 | Qwell | I care more now then I did than. |
| 14:24.00 | p3nguin | lol |
| 14:24.11 | p3nguin | You're going to confuse people. |
| 14:24.15 | Qwell | :D |
| 14:24.46 | p3nguin | They're already wrong enough as it is, so we don't need to encourage them. |
| 14:25.21 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 14:30.09 | Boardy | Your right their, p3nguin. |
| 14:30.31 | Qwell | No, his not right at all. |
| 14:30.55 | p3nguin | I can feel a brain explosion coming on. |
| 14:30.57 | Qwell | He could of been right, but he wasn't. |
| 14:31.03 | Qwell | there you go |
| 14:31.12 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:35.10 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:35.23 | *** join/#asterisk moy_ (~moy@173.239.155.74) |
| 14:50.01 | titter | Can someone see why when calling into a Queue from my cell phone the dahdi channel doesn't hangup if I stay on the line with my cell phone? It repeats the call 3 times before it finally hangs up? http://pastebin.com/vitEZRRF |
| 14:50.27 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:57.03 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 14:57.46 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:58.21 | Katty | looks in |
| 14:58.22 | Katty | Qwell: YOU |
| 14:58.36 | Qwell | ME? |
| 14:58.41 | Katty | why i outta... |
| 14:58.42 | Katty | just... |
| 14:58.46 | Katty | HUG YOU TO BITS! |
| 14:58.49 | Katty | hugs Qwell to bits |
| 14:58.52 | Qwell | NO! Anything but that! |
| 14:59.13 | dwayne | sweeps up the Qwell-bits |
| 14:59.48 | dwayne | packages them and sends them to Antarctica |
| 15:04.16 | Katty | :< |
| 15:13.32 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:13.38 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:17.44 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:23.40 | *** join/#asterisk d_preston215 (~chatzilla@173-12-4-137-panjde.hfc.comcastbusiness.net) |
| 15:23.48 | d_preston215 | Is there a way to import annoucements into Asterisk? |
| 15:24.10 | p3nguin | What do you mean? |
| 15:24.19 | Katty | define 'annoucement' |
| 15:24.29 | p3nguin | Define import. |
| 15:25.22 | Katty | define asterisk |
| 15:25.27 | Katty | *hee* |
| 15:25.35 | *** join/#asterisk skrusty (~skrusty@83.166.171.74) |
| 15:25.48 | p3nguin | Define is. |
| 15:25.57 | Katty | define into. |
| 15:26.18 | p3nguin | Define dead horse. |
| 15:26.20 | file | coughs |
| 15:26.20 | p3nguin | :P |
| 15:27.27 | p3nguin | But seriously, what do you mean by import and announcements? |
| 15:28.07 | Katty | hugs file |
| 15:28.17 | file | hugs Katty |
| 15:29.33 | d_preston215 | recordings that asterisk plays as annoucements. |
| 15:29.46 | d_preston215 | Like a default IVR message. |
| 15:30.49 | Katty | well if they are the right format |
| 15:30.58 | Katty | you can dump them into a folder, and have asterisk Background() them |
| 15:32.31 | p3nguin | Okay, so now I know what announcments are, but what do you mean by import? |
| 15:32.44 | luke-jr | I don't see how one can use GROUP_COUNT to replace call-limit-- won't there be race conditions? |
| 15:33.03 | d_preston215 | Take them from one asterisk setup and put them in another asterisk setup. |
| 15:33.16 | p3nguin | I'd use rsync. |
| 15:33.34 | p3nguin | if both are still online, it'll do exactly what you need. |
| 15:34.34 | p3nguin | luke-jr: If you mean that one person can make a call and another person cannot, yes. That's what it does. |
| 15:35.28 | luke-jr | p3nguin: ⦠I don't know how you get that from what I said :| |
| 15:35.39 | Katty | winscp is nice if you're transfering from linux to windows |
| 15:35.41 | p3nguin | What other race condition is there? |
| 15:35.41 | Katty | and then back again |
| 15:35.46 | Katty | just remember to transfer binary |
| 15:36.26 | luke-jr | p3nguin: ChannelA sets its group to FOO at the same time as ChannelB sets its group to FOO, and both of them get 2 channel count |
| 15:58.51 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
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| 16:25.39 | *** join/#asterisk dobby156 (~joe@79.135.102.10) |
| 16:25.40 | dobby156 | hi |
| 16:25.48 | dobby156 | I am running asterisk 1.6 |
| 16:26.06 | dobby156 | and using sip notify command on the server is cause a memory leak |
| 16:27.48 | *** join/#asterisk dobby156 (~joe@79.135.102.10) |
| 16:27.58 | dobby156 | sorry acidentally disconnected |
| 16:28.10 | dobby156 | so anyway there seems to be a leak |
| 16:28.19 | *** part/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 16:28.29 | dobby156 | cause by a sip_alloc in chan_sip.c |
| 16:29.23 | dobby156 | it seem to originate from sip_cli_notify |
| 16:29.59 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 16:30.48 | dobby156 | is there a way to fix this, because of the deployment of this system I am unable upgrade to 1.8 or 10b |
| 16:31.15 | dobby156 | I have tried recompiling multiple times and the problem is the same |
| 16:31.45 | luke-jr | yay, Digium just wasted 2 days and $86 ⦠:| |
| 16:34.22 | *** join/#asterisk thedavidfactor (~david@nc-76-3-106-212.dhcp.embarqhsd.net) |
| 16:38.40 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:39.46 | luckman212 | luke-jr: ? |
| 16:40.56 | *** join/#asterisk jkroon (~jkroon@197.174.40.136) |
| 16:42.50 | luke-jr | luckman212: they diagnosed our problem as a hardware failure, so we paid the overnight shipping costs for a replacement, and it's still crashing with that |
| 16:43.01 | luke-jr | (and they didn't have any in stock, so it took a day + overnight) |
| 16:43.43 | luckman212 | :| |
| 16:55.18 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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| 17:03.10 | *** join/#asterisk cyford (cyford@50.13.199.119) |
| 17:03.11 | p3nguin | Did they diagnose the problem for you, and then wasted time and money? |
| 17:18.09 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 17:20.01 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
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| 17:36.05 | *** join/#asterisk _omer (~omer@182.185.220.157) |
| 17:37.24 | _omer | when I run asterisk it get crashed. asterisk -cvvvvg doesn't show error ... |
| 17:37.52 | p3nguin | What causes it to crash? |
| 17:37.59 | p3nguin | What channel drivers are you using? |
| 17:39.15 | _omer | yes I want to know the cause .... I have copy pasted my old asterisk configuration files |
| 17:39.27 | _omer | I am using SIP |
| 17:39.57 | _omer | the main thing is ... it get crashed as soon as I run it |
| 17:39.58 | Micc_ | Looks like 1.8.7 will support multi-tenant parking again, eh? |
| 17:40.01 | p3nguin | What are you doing when it crashes? Is it crashing while idle? |
| 17:40.24 | _omer | linux# asterisk |
| 17:40.43 | _omer | I am unable to run it |
| 17:40.51 | p3nguin | Show me. |
| 17:41.03 | p3nguin | Use a pastebin. |
| 17:41.05 | _omer | ok |
| 17:41.05 | p3nguin | ~pb |
| 17:41.05 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 17:44.13 | _omer | http://pastebin.com/2b0tRazv |
| 17:46.27 | _omer | p3nguin: ?? |
| 17:46.30 | *** join/#asterisk navaismo (~navaismo@187.170.4.20) |
| 17:46.45 | p3nguin | Run it with: asterisk -G asterisk -U asterisk -vvvvddddddddg |
| 17:46.53 | p3nguin | Show me everything it outputs. |
| 17:47.05 | _omer | ok |
| 17:50.57 | _omer | this time It popup error msg...and I have fixed it ... |
| 17:51.05 | _omer | thanks p3nguin |
| 17:55.34 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 17:57.28 | anonymouz666 | anyone know where I can find the queue_members SQL for use in realtime? |
| 17:57.35 | anonymouz666 | there's none in contrib/ |
| 17:57.49 | anonymouz666 | and res_config_odbc is complaning about some columns |
| 17:57.57 | p3nguin | _omer: What was the problem? |
| 18:01.56 | anonymouz666 | I expected to find the SQL anywhere but not in voip-info ;) |
| 18:06.40 | carrar | did you look in the contrib directory? |
| 18:07.08 | p3nguin | (1257.35) <anonymouz666> there's none in contrib/ |
| 18:07.36 | carrar | I see several there |
| 18:07.59 | carrar | well at leas 1 |
| 18:08.18 | carrar | CREATE TABLE queue_member_table |
| 18:09.02 | anonymouz666 | carrar: where? |
| 18:09.14 | anonymouz666 | contrib? what's the file name? |
| 18:09.27 | anonymouz666 | grep queue_member_table * - shows nothing |
| 18:09.29 | carrar | download lastest source? |
| 18:09.38 | anonymouz666 | latest than 1.8.7.0-rc1? |
| 18:09.52 | anonymouz666 | I find it anyway, thanks. |
| 18:10.01 | anonymouz666 | found |
| 18:10.35 | *** join/#asterisk Crowb4r (u1012@gateway/web/irccloud.com/x-wexdcwnbdqenxyaq) |
| 18:32.59 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 18:43.45 | Luyt | Anybody ever heard of "PortaUM"<sip:PortaUM@91.195.160.22:5064> ? |
| 18:52.43 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 18:52.52 | anonymouz666 | PortaUM sounds a portuguese word |
| 18:53.38 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:1426:b5c4:b598:4131) |
| 18:54.22 | Luyt | hmmm yes lemme geoip that IP |
| 18:55.10 | Luyt | hmmm, that IP is from Breezz, Netherlands. A large voip provider. |
| 18:55.17 | *** join/#asterisk oej_ (~olle@ns.webway.se) |
| 18:56.36 | Luyt | Probably some keepalive messages or so |
| 18:57.22 | *** join/#asterisk irroot (~irroot@197.173.241.226) |
| 19:04.42 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 19:15.20 | StaRetji | ehm folks, how can I make extension hangup if now answered after 15 seconds? |
| 19:15.59 | StaRetji | not* |
| 19:18.22 | singler | use timeout parameter in Dial() |
| 19:20.08 | StaRetji | singler: thx, it actually extension in queue |
| 19:20.10 | StaRetji | agents |
| 19:20.31 | navaismo | use timeout in queue |
| 19:20.33 | singler | use timeout parameter in queue config |
| 19:20.37 | navaismo | next step hangup |
| 19:20.59 | StaRetji | navaismo: timeout is 15 |
| 19:21.03 | StaRetji | but I use ringall |
| 19:21.18 | StaRetji | and after 15 it just continues to ring |
| 19:21.21 | StaRetji | is this okay> |
| 19:21.22 | StaRetji | ? |
| 19:21.36 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 19:21.46 | StaRetji | is 15 seconds or 15 rings? |
| 19:21.49 | singler | did you set it calling queue() or in config file? you need both to work correctly |
| 19:21.53 | singler | seconds |
| 19:22.12 | StaRetji | exten => 900000,1,Queue(custsrv_queue|tTr|||300) |
| 19:22.19 | p3nguin | singler: Queue is run from an extension, it doesn't run an extension. |
| 19:22.35 | p3nguin | unless your member is a Local channel, then it does. |
| 19:22.48 | StaRetji | yes, from ivr, i run queue |
| 19:23.20 | cerienjean | Hi - I have a nat issue |
| 19:23.28 | p3nguin | ~sipnat |
| 19:23.28 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 19:23.39 | cerienjean | I have two identicals setup: server (nat) internet (nat) client |
| 19:23.42 | cerienjean | same settinfs |
| 19:23.55 | cerienjean | in once case, the audio works fine, not in the other |
| 19:24.00 | StaRetji | I gotta go now, but when I return I demand solution |
| 19:24.01 | StaRetji | :) |
| 19:24.04 | StaRetji | hehehehhe |
| 19:24.07 | StaRetji | joking ofc |
| 19:24.10 | StaRetji | see ya folks |
| 19:24.30 | cerienjean | in the faulty case, sip set debug shows that the phone is nated (on the sip exchange) bu the RTP flow is sent to the local private address of the clienbt |
| 19:24.58 | p3nguin | Joking officer? |
| 19:25.14 | StaRetji | still here |
| 19:25.27 | StaRetji | I think I better stay and fix this |
| 19:25.29 | StaRetji | exten => 300000,1,Queue(custsrv_queue|tTr|||300) |
| 19:25.32 | StaRetji | what is 300? |
| 19:25.40 | p3nguin | timeout |
| 19:25.47 | StaRetji | omg |
| 19:25.49 | p3nguin | But you've done it all wrong. |
| 19:25.50 | StaRetji | 300 seconds? |
| 19:26.07 | cerienjean | Yes, I've seen the article, I am option #4 |
| 19:26.09 | StaRetji | well, it was not me who done it |
| 19:26.34 | cerienjean | But how can I debug more finely ? |
| 19:26.48 | p3nguin | You shouldn't be using t and T in your queue options. t maybe, but I can't imagine that you want the callers to be able to transfer calls. |
| 19:27.03 | p3nguin | And you have pipes where you should have commas. |
| 19:27.54 | StaRetji | thx |
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| 19:36.57 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 19:42.00 | luckman212 | does anyone in here know where the code is in the src that Asterisk uses to form the body of the SDP in its SIP INVITEs? |
| 19:42.17 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 19:45.25 | anonymouz666 | luckman212: what you are trying to do is a mistery |
| 19:47.03 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:47.07 | luckman212 | its not that mysterious. I have a context that I created that just plays MusicOnHold. I dont want to waste bandwidth by having all phones that dial in to this context sending audio when they dont need to. so I want to set the INVITE for that call to a=sendonly |
| 19:48.48 | treborsux | <PROTECTED> |
| 19:49.24 | p3nguin | treborsux: crontab |
| 19:51.03 | p3nguin | perhaps something like: 0 23 * * * asterisk -rx 'core restart when convenient' &>/dev/null |
| 19:55.34 | anonymouz666 | luckman212: and when your phone hits the MusicOnHold what are the INVITE content? |
| 19:56.09 | luckman212 | a=sendrecv |
| 19:56.32 | anonymouz666 | whe MOH answer the reply is sendrecv? |
| 19:56.57 | luckman212 | the dialplan is : Answer() and then MusicOnHold() |
| 19:57.06 | luckman212 | so the Answer() yes, sets a=sendrecv |
| 19:57.28 | luckman212 | If I don't Answer() first, then calling MusicOnHold() doesn't work |
| 19:58.05 | anonymouz666 | Answer sets sendrecv and MOH should REINVITE changing this attribute |
| 19:58.06 | *** join/#asterisk scolson (~scolson@c-68-40-184-237.hsd1.mi.comcast.net) |
| 19:59.18 | *** join/#asterisk adeeln (~adeel@184.175.36.92) |
| 19:59.18 | luckman212 | well when I do 'rtp set debug on' i can see the RTP still flowing 2-way from the device<->asterisk |
| 20:01.05 | scolson | here is one I am sure everyone sees every day. I have a hardphone that can't receive audio when making calls outbound and a softphone using the exact same extension that works just fine. both are behind the same NAT router. The hardphone is an aastra 6757i and I've tried toggling upnp and report to no avail |
| 20:01.14 | scolson | rport not report |
| 20:01.14 | scolson | t |
| 20:04.37 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 20:11.41 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 20:18.26 | *** join/#asterisk anymemm (~anymemm@89.215.135.208) |
| 20:19.03 | anymemm | hello everybody, greetings from sofia/bulgaria |
| 20:20.23 | anymemm | i am trying to register an asterisk 1.2.15 to a german fritzbox 7270 in order to receive inbound call via isdn |
| 20:20.43 | anymemm | is there anybody out there with experience on this? |
| 20:20.56 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:20.56 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:24.08 | anymemm | registration of the fritzbox is succeeding althogh on inbound call i get an Sep 9 22:23:26 NOTICE[20761]: chan_sip.c:10629 handle_request_invite: Failed to authenticate user <sip:[NUMBER CALLING FROM]@fritz.fonwlan.box>;tag=298C87F8441893F4 |
| 20:24.31 | anymemm | any ideas? |
| 20:25.19 | navaismo | the sip user doesnt exist or wrong password |
| 20:25.27 | *** join/#asterisk BuenGenio (~Gene@38.Red-83-41-148.dynamicIP.rima-tde.net) |
| 20:26.03 | navaismo | so you are connecting both via sip? |
| 20:28.00 | anymemm | yes, sip is what i'm using |
| 20:28.34 | navaismo | can you show us your sip registrations just remove ips and passwords |
| 20:29.12 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 20:30.20 | anymemm | yes, just a moment |
| 20:32.33 | anymemm | as the ip and password are generic, i am not removing them: [fritzbox] |
| 20:32.33 | anymemm | host=192.168.21.2 |
| 20:32.33 | anymemm | fromdomain=fritz.fonwlan.box |
| 20:32.33 | anymemm | fromuser=623 |
| 20:32.33 | anymemm | context=fritzbox-in |
| 20:32.33 | anymemm | type=friend |
| 20:32.35 | anymemm | dtmfmode=rfc2833 |
| 20:32.39 | anymemm | disallow=all |
| 20:32.41 | anymemm | allow=alaw |
| 20:32.43 | anymemm | allow=ulaw |
| 20:32.45 | anymemm | insecure=invite,port |
| 20:32.47 | anymemm | username=623 |
| 20:32.49 | anymemm | secret=000623 |
| 20:32.51 | anymemm | ;requirecalltoken=no |
| 20:32.53 | anymemm | nat=no |
| 20:32.55 | anymemm | canreinvite=no |
| 20:33.08 | anymemm | register => 624:000624@192.168.21.2/624 (this line i above the [fritzbox] block) |
| 20:33.45 | navaismo | use pastebin instead |
| 20:35.06 | navaismo | try to copy both peers registrations and the entire output when call arrive in the Pastebin |
| 20:35.16 | navaismo | http://pastebin.com/ |
| 20:36.22 | anymemm | sorry, you're right, http://pastebin.com/hquxrxf7 |
| 20:37.12 | anymemm | the entire outbug of sip debug peer follows in a second post |
| 20:38.23 | navaismo | brb |
| 20:41.45 | anymemm | that's the debug trace when i try to make an inbound call: http://pastebin.com/K9e2gd4G |
| 20:50.47 | navaismo | the registration string from the other pbx? |
| 20:51.11 | *** join/#asterisk luckman212_ (~irc@2001:470:1f07:1225:5880:fdc2:20d7:fcf6) |
| 20:51.20 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 20:54.42 | anymemm | well, the other pbx is a kind of cheap consumer device which only shows a "09.09.1122:37:36Internettelefonie mit 623 über 192.168.21.112 war nicht erfolgreich. Ursache: Forbidden (403)" in its log |
| 20:55.50 | navaismo | how do you set the connection in thata pbx? |
| 20:56.52 | anymemm | there are three fields: registrar: fritz.box, username: 623, password: 000623 |
| 20:58.31 | navaismo | fritz.box=192.168.21.112=asterisk? |
| 20:58.53 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:58.53 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:00.31 | navaismo | that asterisk version support the sendrpid and trustrpid? |
| 21:01.22 | anymemm | fritz.box is at 192.168.21.2, asterisk is as 192.168.21.112, asterisk registers at fritzbox (which "converts" a 2 channel isdn to voip) successfully; as an inbound call arrives at fritzbox asterisk fails authenticating it for some reason |
| 21:02.13 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:02.21 | anymemm | i have no idea about sendrpid and trustrpid, it is an older version 1.2.15 |
| 21:03.58 | navaismo | ok when you regiter from asterisk to germanpbx you send user:pass@germanpbxIP, but when you register from germanpbx to asterisk you send user:pass@germanpbxip<--- this should be user:pass@asteriskip or im wrong? |
| 21:09.15 | anymemm | well, registering fritz to asterisk is working well, without any problem, and i'm sure that i'm not mixing up id addresses (i'm using this to send faxes from a fax server which only supports isdn cards) |
| 21:09.32 | navaismo | ok |
| 21:10.05 | navaismo | are you tried with diferent pass? |
| 21:11.31 | anymemm | no yet, but as fritzbox is somehow not easy to debug (although it's good hardware running on linux) i'll try to recreate the sip accounts and try again |
| 21:17.56 | anymemm | ok, recreated the sip account on the fritzbox, but still getting "SIP/2.0 407 Proxy Authentication Required" |
| 21:18.25 | anymemm | and then: "SIP/2.0 403 Forbidden" |
| 21:18.49 | anymemm | i'll now try to register a softphone with that account |
| 21:22.18 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:22.38 | navaismo | ok |
| 21:24.16 | anymemm | well, i as assumed, both directions work using jitsi (former sip-communucator) when registered to fritzbox |
| 21:25.01 | anymemm | there is some parameter which has to be set within sip.conf when registering asterisk to fritz... |
| 21:26.20 | anonymouz666 | malcolmd: please document what dahdi_maint -s <spannum> output means |
| 21:26.22 | anonymouz666 | :) |
| 21:27.48 | navaismo | mmm the problem i think is the digest i guess is wrong |
| 21:28.10 | malcolmd | anonymouz666: -> Pete Engler: pengler@digium.com |
| 21:31.00 | anonymouz666 | who is pete? |
| 21:31.50 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 21:34.20 | navaismo | guys the md5 challenge for sip can be obtained with echo -n "user:realm:pass:nonce" | md5sum ??? |
| 21:37.50 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 21:38.06 | pabelanger | navaismo: not sure :nonce is needed |
| 21:39.11 | navaismo | im triyng to get the digest for the anymemm pastebin log using the user, real pass and the nonce in the log |
| 21:46.02 | *** join/#asterisk caveat- (~false@gateway/shell/bshellz.net/x-yotjkrkuzjkvasnr) |
| 21:46.41 | malcolmd | anonymouz666: product manager telephony cards (dahdi) |
| 21:47.46 | *** join/#asterisk kinko (~kinko@77.208.56.35) |
| 21:48.29 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:48.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:48.48 | kinko | hello |
| 21:50.29 | kinko | question: get one way audio or null audio if not monitor(g729,/dev/null) before Dial , I guess some codec issue here , any idea ? |
| 21:51.17 | ChannelZ | sounds more like firewall/NAT issues |
| 21:51.23 | kinko | I get a call SIP from peer, and Dial SIP to another peer, if no any fake monitor in the middle, no audio |
| 21:51.39 | kinko | ChannelZ, I full removed iptables , and same, seems not fw related |
| 21:52.01 | kinko | ChannelZ, also there no NAT |
| 21:52.40 | kinko | ChannelZ incomming leg SIP peer and outgoing leg SIP peer are both public IP, no NAT here |
| 21:53.38 | anonymouz666 | malcolmd: thank you |
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| 22:00.50 | *** part/#asterisk frem_ (~jamesgeck@64.207.3.161) |
| 22:01.44 | navaismo | arrggg anymemm i cant obtain the digest |
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| 22:11.28 | anymemm | navaismo: i also don't know how it is generated, the whole issue is completely strange as it simply stopped working suddenly without changning anything... neverthelsee, i thank you very much for your efforts to help me! |
| 22:15.41 | *** part/#asterisk droemel (~droemel@p4FCAC881.dip.t-dialin.net) |
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| 23:42.24 | leifmadsen | wow quiet night |
| 23:42.49 | p3nguin | Now you ruined it. Way to go! |
| 23:43.43 | leifmadsen | yay! |