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| 00:22.07 | LemensTS | I can't figure out why dahdi is not building /usr/lib/asterisk/modules/chan_dahdi.so ...i have a pb of the cmd i did to install dahdi: http://pastebin.com/AHCE1p5s |
| 00:42.00 | LemensTS | nm had to compile dahdi before asterisk |
| 00:42.03 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
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| 01:18.01 | fozzmoo | Is someone available that could help me debug a simple SIP problem? |
| 01:21.56 | fozzmoo | I'm getting this "SIP/2.0 401 Unauthorized" when an incoming call comes in. |
| 01:22.08 | fozzmoo | Running 1.6 |
| 01:23.42 | *** join/#asterisk wdbl (~wdbl@ool-44c07005.dyn.optonline.net) |
| 01:24.02 | Kobaz | (gdb) p hh_new |
| 01:24.02 | Kobaz | $2 = (struct hanguphandler_item *) 0xdeadbeef |
| 01:24.04 | Kobaz | wtf |
| 01:28.30 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:29.24 | riddlebox | hello, is anyone using google voice+gizmo5+asterisk to make and recieve calls? |
| 01:29.58 | riddlebox | when I point my google voice config to my broadvoice account it works everytime but when I put my gizmo5 number or my sipgate number it wont call out? |
| 01:51.44 | russellb | Kobaz: haha. |
| 01:51.51 | russellb | Kobaz: MALLOC_DEBUG on? |
| 01:53.07 | russellb | Kobaz: MALLOC_DEBUG does a few things on top of just tracking allocations for finding memory leaks. |
| 01:53.13 | *** join/#asterisk wonderworld (~ww@port-92-201-125-20.dynamic.qsc.de) |
| 01:53.32 | russellb | Kobaz: It also allocates a "fence" around each allocation to help detect writes just before or after an allocation. It sets the fence to that value. |
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| 02:14.40 | Kobaz | yeah i figured |
| 02:14.47 | Kobaz | yeah malloc debug is on |
| 02:15.23 | Kobaz | russellb: i was casting a void* to the wrong thing... i was getting some funky behavior... heh |
| 02:17.05 | provolone | how can I use asterisk -x to detect when the phone has been answered in a shell script ? |
| 02:18.44 | provolone | is there something I can use that will block until it is answered ? |
| 02:18.59 | Kobaz | provolone: AMI |
| 02:19.41 | Kobaz | provolone: welcome to dialplan and event handling... I've been writing code to handle stuff just like that for about three years |
| 02:20.05 | provolone | where should I start for docs ? |
| 02:21.27 | Kobaz | The asterisk manager interface |
| 02:23.12 | russellb | ~newbook |
| 02:23.12 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 02:23.18 | russellb | there is an AMI chapter in there ... |
| 02:23.28 | russellb | it's not really documentation of the commands, but gives you an overview of what the interface is |
| 02:28.49 | Kobaz | oh man... so close |
| 02:30.27 | Kobaz | whew |
| 02:30.41 | Kobaz | two weeks of development, i now have proper transfer tracking for my callqueue |
| 02:31.01 | Kobaz | but only for slow attended transfers, still need to work on fast attended transfers |
| 02:31.51 | Kobaz | i have all the logic for fast attended transfers, just need to put it back in |
| 02:32.02 | russellb | what do you mean by fast? |
| 02:32.10 | russellb | where someone hangs up before waiting for the answer? |
| 02:32.44 | provolone | so most of you guys support call centers for marketing ? |
| 02:32.48 | Kobaz | russellb: yeah |
| 02:33.02 | russellb | we call that "blonde transfers" |
| 02:33.03 | russellb | :-p |
| 02:33.08 | Kobaz | hah |
| 02:33.09 | Kobaz | ooooh |
| 02:33.14 | Kobaz | that's what you guys were talking about last week |
| 02:33.46 | Kobaz | provolone: many of us hack at various things with no focus in particular... and then some people have a particular focus |
| 02:34.00 | Kobaz | i have about 6 projects that I work on |
| 02:34.06 | russellb | doesn't actually use asterisk :-p |
| 02:34.09 | Kobaz | yeah |
| 02:34.15 | Kobaz | he's more of a surfer |
| 02:34.20 | provolone | as you can see I am just at the edge of the pool with my feet in |
| 02:34.53 | russellb | I just try to talk to lots of people that do use it to get good feedback on what needs to be done. |
| 02:34.57 | russellb | seems to work out ok. |
| 02:35.15 | Kobaz | russellb: so the 'fast attended transfer'.. .the blonde transfers... dialplan wins up running directly on the transferee channel, instead of the intermediary channel |
| 02:35.46 | Kobaz | wins/winds |
| 02:36.34 | Kobaz | so i have to do some hacks to push our events and stuff as if it was running on the intermediary channel, to make things sane for processing something normalized |
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| 02:41.24 | Kobaz | russellb: i added a bunch more info to the sip attended transfer event to make this sort of stuff much easier to process... don't need to listen for masquerades and renames and etc to know what's going on... just need the one event |
| 02:41.32 | Kobaz | i should post up a patch for that |
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| 03:19.49 | adeel | anyone ever setup a2billing with voicemail per did/account? |
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| 04:04.32 | tash | strange, it seems that randomly a .call gets stuck in the asterisk outgoing directory. If I open the file, I see a bunch of delayed retry lines. Any ideas? |
| 04:04.33 | eMBee | good morning |
| 04:05.01 | eMBee | has some trouble with asterix thinking that an extension is busy, when it isn't |
| 04:05.37 | eMBee | the phone was moved to a different location in the net, and it seems like now asterix thinks there are two phones with the same extension |
| 04:05.50 | eMBee | erm, asterisk |
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| 04:07.23 | eMBee | is there a way i can clear the state of an extension in the commandline tool? |
| 04:07.59 | WIMPy | Asterisk does not support multiple devices on one account. |
| 04:08.21 | WIMPy | If a device registers, all info is replaced. |
| 04:10.52 | eMBee | ok, maybe i am missreading this, anyways, asterisk still thinks the extension is busy but it isn't can i force a reset without restarting? |
| 04:11.21 | WIMPy | There is nothing to reset. |
| 04:11.35 | WIMPy | Maybe it's unable to reach the phone? |
| 04:11.39 | adeel | try reloading the channel? e.g. sip reload |
| 04:11.45 | WIMPy | What does teh *cli say when it fails? |
| 04:12.35 | eMBee | well, in the cli when i run core show channels it shows up as SIP/808-00000dbd 6500@DLPN_DialPlan_D Up AppQueue((Outgoing Line)) |
| 04:13.57 | WIMPy | set verbose at least 3 and maybe debug to 1 and paste the message you get then. |
| 04:15.59 | eMBee | but at the same time the user managed to place a call from 808 to 807 and i saw: SIP/808-00000dd9 807@DLPN_DialPlan_DE Ring Dial(SIP/807) |
| 04:16.37 | eMBee | what is the difference between SIP/808-00000dd9 and SIP/808-00000dbd? |
| 04:16.59 | WIMPy | That's only the incoming call leg. No information about any attempt to reach a phone. |
| 04:17.21 | WIMPy | Another call serial number. |
| 04:18.00 | eMBee | ok |
| 04:18.23 | WIMPy | Maybe it's your queue acting up, not the device? |
| 04:18.53 | eMBee | possible |
| 04:19.06 | eMBee | the device seems to work just fine |
| 04:19.17 | eMBee | so i try to reset the queue? |
| 04:19.22 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:19.50 | WIMPy | I can't comment on queues. Never used them. |
| 04:20.12 | ChannelZ | Yeah. He cuts in line. |
| 04:22.06 | eMBee | lol |
| 04:22.35 | eMBee | yes, i see a DAHDI/6-1 6500@queues:1 Up Queue(6500) sticking around there too |
| 04:22.43 | eMBee | can i forse a hangup? |
| 04:22.47 | eMBee | force? |
| 04:23.02 | WIMPy | channel request hangup ... |
| 04:24.08 | eMBee | ah, yes, just found it in the help, thanks |
| 04:34.48 | eMBee | that seems to have fixed the problem |
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| 04:42.00 | adeel | is there a listing of the variables for * 1.8? |
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| 05:32.16 | adeel | Is there a way to make the voicemail system play back the DID that was dialed, rather than the mailbox number? that is, when you hear Allison say ' The person at extension <account # here> is unavailable' ....can i change the the account # to be the DID? |
| 05:38.11 | ChannelZ | well you could make people's mailbox numbers their DIDs |
| 05:38.22 | ChannelZ | it reads the mailbox number, not the extension dialed that got them there |
| 05:38.29 | ChannelZ | ...or you can hack the source |
| 05:39.02 | adeel | yeah, i was leaning towards hacking the source; maybe just add another parameter for the did that was dialed |
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| 05:39.28 | adeel | only problem is that there could be a number of different DID's associated with 1 account/extension |
| 05:40.18 | *** part/#asterisk provolone (~root@c-68-33-201-70.hsd1.md.comcast.net) |
| 05:48.46 | Kobaz | or just write your own wrapper in dialplan in front of VoiceMail() |
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| 06:07.42 | adeel | hmmm.... |
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| 06:54.29 | fakhir | hello. if you order IVR prompt from digium can you provide a list of names? |
| 07:05.15 | shapr | I really wish wireshark had some way to read asterisk CLI captures. |
| 07:06.12 | WIMPy | What part of them would you like in there? |
| 07:08.27 | *** join/#asterisk fross (~fross@68-112-140-187.dhcp.stcd.mn.charter.com) |
| 07:11.54 | fross | Is the new book on a pdf at all, or just the html? |
| 07:12.19 | shapr | WIMPy: It's more that I'd like to be able to filter and fold the messages. |
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| 07:18.48 | kaldemar | fross: the new book is not yet published. we'll see when it is. |
| 07:22.34 | fross | ahh ok, Its the only other book the "Asterisck: The Futrue of Telephony from 2005? |
| 07:23.47 | fross | I am just looking for a good reference to put on my kindle |
| 07:27.12 | kaldemar | the second edition of "Asterisk: The Future of Telephony" was written for asterisk 1.4, so it's quite outdated now. but many things still apply. i'd wait for the new book, it's supposed to be ready in "early 2011" according to http://asteriskdocs.org. |
| 07:27.49 | fross | kaldemar: ok thanks Ill just wait to get the new one |
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| 07:30.00 | schmidts | good mroning |
| 07:30.06 | schmidts | s/mroning/morning |
| 07:37.19 | wdoekes2 | wow.. no annoying line from the infobot :) morning btw |
| 07:37.36 | WIMPy | Syntax error :-) |
| 07:37.45 | wdoekes2 | drailing slash |
| 07:37.57 | wdoekes2 | too bad ;) |
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| 10:15.05 | hrhrhr | g'morning |
| 10:15.17 | hrhrhr | are there any handsets out there with an easy way to log out/log in |
| 10:15.23 | hrhrhr | hot desking scenario? |
| 10:16.55 | schmidts | hrhrhr not really AFAIK but you can just use dialplan logic for this |
| 10:17.23 | hrhrhr | i get that answer every time |
| 10:17.25 | hrhrhr | :D |
| 10:17.36 | hrhrhr | i just want a noob friendly way to log ppl out and in |
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| 10:18.52 | schmidts | maybe you can use a phone with a builtin webinterface |
| 10:19.04 | WIMPy | Or a phones browser if you want a fancy version. |
| 10:22.45 | schmidts | hrhrhr you can use IP subnets for example, every user has to change the IP of the phone and with this ip you have another user |
| 10:23.27 | schmidts | hrhrhr but i dont know if asterisk can support the same user but with different ips, i guess not, but you can use again dialplan logic for this :D |
| 10:24.38 | joobie | guys |
| 10:24.54 | joobie | anyone played with self-signed certs on a CISCO 500 series? |
| 10:25.12 | joobie | im going through a proces that you have to send the CSR to cisco to have them send back a cert |
| 10:25.19 | joobie | sounds a bit overkill - not sure if there is a simpler way |
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| 10:38.39 | binrush | Hello. I have a problem using asterisk 1.6 with realtime sips. When I add sip channel (my sip provider) to asterisk using realtime |
| 10:39.06 | binrush | incoming calls don't work for me. The message is NOTICE[19805]: chan_sip.c:21250 handle_request_invite: Sending fake |
| 10:39.08 | binrush | auth rejection for device "test" |
| 10:39.10 | binrush | <sip:test@my.sip-provider.org>;tag=as0af02b0c |
| 10:39.16 | binrush | Sorry for broken message |
| 10:43.50 | *** part/#asterisk aceking5 (~aceking5@71-94-133-68.static.mtpk.ca.charter.com) |
| 10:56.01 | kaldemar | binrush: realtime is not working. |
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| 10:57.40 | binrush | kaldemar: sorry ? |
| 11:01.32 | bjhaid | i would like to know if its possible that say i am trunking via a sip provider and i am provided several sip channels, if i have the channels put into a group (g), can i do, dial/g/08039269311, would this be correct, say i am hardcoding which PSTN line the extension should dial once the extension is dialed? |
| 11:08.43 | *** join/#asterisk coppice (~chatzilla@12.69.71.130) |
| 11:09.26 | schmidts | does wiki.asterisk.org work for anybody? |
| 11:20.29 | kaldemar | binrush: looks like asterisk has no information on a user by the name test, so it sends an authentication rejection. |
| 11:41.51 | *** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.16) |
| 11:42.00 | aiksa[LV] | hi |
| 11:42.36 | aiksa[LV] | can i have two different sip peers with the same username=... record, but different secret=.... records ? |
| 11:42.37 | shapr | hiya |
| 11:43.01 | aiksa[LV] | obviously [peername] part is different for the both |
| 11:44.29 | *** join/#asterisk binrush (~rush@194.165.18.23) |
| 11:45.21 | aiksa[LV] | or are there any other method to sync the process of a secret change through provision on both the pbx and enddevice |
| 11:45.26 | binrush | kaldemar: what's wrong with realtime ? |
| 11:46.24 | kaldemar | binrush: no idea. but it doesn't seem to be working. |
| 11:47.39 | kaldemar | binrush: do you have the realtime module (res_realtime.so) loaded and how did you configure it? |
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| 11:52.20 | z4nD4R | supports asterisk 1.4.23 TLS?? I find on mail-list that NO... but some official ... cant find.... |
| 11:53.46 | kaldemar | z4nD4R: no. |
| 11:54.17 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 11:54.35 | z4nD4R | kaldemar: even if i find some path? |
| 11:54.43 | z4nD4R | i find only this http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg26840.html |
| 11:55.04 | aiksa[LV] | or is there a way perhaps how asterisk could support multiple secerts for a single sip friend entry? |
| 11:55.37 | binrush | kaldemar: res_realtime.so Realtime Data Lookup/Rewrite 0 |
| 11:57.34 | kaldemar | z4nD4R: of course you can patch it to support TLS if there is such a patch. |
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| 11:59.10 | z4nD4R | kaldemar: hmm... crazy... i find patch with zrtp but only for 1.4 version... and 1.4 does not support TLS.. :D ... i dont thig, that some path for asterisk 1.4 with tls is |
| 11:59.30 | binrush | kaldemar: and I have sip.conf => odbc,asterisk,ast_config in my extconfig.conf |
| 12:03.06 | kaldemar | binrush: and you have configured also res_odbc, have the module loaded and the ODBC connector exists? |
| 12:04.00 | bjhaid | <PROTECTED> |
| 12:04.51 | kaldemar | binrush: http://ofps.oreilly.com/titles/9780596517342/ch16.html |
| 12:06.49 | binrush | kaldemar: yes. Even more - everything is work except incoming calls. |
| 12:08.23 | binrush | kaldemar: The problem is with host parameter in sip_buddies table. When I put hostname of my sip provider there, incoming calls are rejected by my asterisk with error message. When I put IP address there, they are working fine. |
| 12:08.53 | bjhaid | <PROTECTED> |
| 12:09.10 | binrush | kaldemar: And in case of non-realtime configuration, everythong works fine with hostnames as well |
| 12:10.59 | kaldemar | binrush: you instruct asterisk to read table ast_config in extconfig.conf, not sip_buddies. |
| 12:16.24 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 12:26.19 | *** join/#asterisk datacompboy (~datacompb@l49-200-67.cn.ru) |
| 12:26.57 | datacompboy | Hi! Anyone know security holes in asterisk 1.4 ? I'm receive messages like 'Content-Type:^@ application/sdp' where ^@ is zero byte. After that -- INVITE handled, without authorization! |
| 12:27.56 | z4nD4R | datacompboy: security holes? Use some protection on SIP or RTP? |
| 12:28.40 | datacompboy | z4nD4R: i have local peers (authorized by IP) and several remotes, that should REGISTER. I see call tries from remote IPs, which do not authtorize |
| 12:29.15 | datacompboy | I have changed dialplan so no more outgoing calls possible from unknown peers (~200 eur lost by calls to Hebrew, Zimbabwe etc in Dec2010) :) |
| 12:29.29 | datacompboy | Checked SIP config -- it still ok, but still new tries to call in logs. |
| 12:30.04 | z4nD4R | datacompboy: my question... if you use same protection on SIP ( tls ) or RTP ( SRTP ) .. or zrtp or VPN? |
| 12:30.41 | datacompboy | z4nD4R: plain SIP/udp + plain RTP |
| 12:31.20 | z4nD4R | datacompboy: so... this is realy BIG security hole |
| 12:31.36 | datacompboy | z4nD4R: i'm asking about other aspect. |
| 12:32.14 | datacompboy | z4nD4R: question is: is known bug about processing zero bytes in SIP/UDP messages or not. |
| 12:33.01 | z4nD4R | datacompboy: so... that i dont realy know... byt 1.4 version is out off date... but... i dont answer on your question... |
| 12:34.40 | datacompboy | z4nD4R: thanks, i know that 1.4 is outdated, but it handles SIP very well, and right now i have no req to upgrade it. Except of this noisy bug, which i want to get -- is it known or not. Google said nothing. |
| 12:37.24 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 12:38.19 | wdoekes2 | datacompboy: I very reluctant to believe that this is not a configuration issue on your end |
| 12:39.45 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 12:41.58 | datacompboy | wdoekes2: may be. configuration are same on several systems. hmmm... it 1.4.22! not .26! Looks i know answer |
| 12:45.19 | datacompboy | Well, can someone answer on newbie questions about 1.8? is it: 1) allows several subscriptions to single peer from different location? call to it should call all. 2) support app_confenrece-like conferencing? (not meetme) 3) allow to detect call termination due to lost of rtp packets ? |
| 12:45.56 | WIMPy | no, yes, yes |
| 12:46.15 | WIMPy | And I think 3 was already possible in 1.4. |
| 12:46.40 | datacompboy | WIMPy: 3 in 1.4? how?! |
| 12:46.58 | WIMPy | rtptimeout |
| 12:48.20 | datacompboy | WIMPy: Hmmm... Thanks :) Is it disable silence suppression when set? |
| 12:48.49 | WIMPy | no |
| 12:50.21 | datacompboy | ok, will do upgrade to 1.6 and look is that fixes problem. Thanks to everyone! |
| 12:50.21 | *** part/#asterisk datacompboy (~datacompb@l49-200-67.cn.ru) |
| 12:53.02 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 13:02.01 | leifmadsen | WIMPy: yep, because 1) would be SIP proxy functionality |
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| 13:55.28 | shortcircuit | I've been reading through AST.pdf in asterisk-1.8.2.3/doc, and I get the feeling it may not have been updated since 1.6.x. The compile instructions all say asterisk-1.6.X.Y.tar.gz and there's some discussion of syntax differences between 1.4 and 1.6. |
| 13:58.48 | shortcircuit | It looks like the PDF was an export from a wiki. (The first non-index page says "This is the home of the official wiki for The Asterisk Project", and has a Recently Updated index in the next section. Maybe someone just needs to export the wiki again? |
| 13:59.50 | WIMPy | Just read http://wiki.asterisk.org or |
| 13:59.54 | kaldemar | the migration for separate text files to the wiki export was done after 1.8.0. it has been updated, but maybe not on all parts. |
| 13:59.55 | WIMPy | ~newbook |
| 13:59.55 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 14:00.08 | kaldemar | s/for/from |
| 14:00.11 | *** join/#asterisk bjhaid (~abejide@41.155.40.208) |
| 14:00.49 | shortcircuit | infobot: Love to. :) |
| 14:00.49 | infobot | If you love to. :) so much, why don't you marry it? (oooooh) |
| 14:00.58 | shortcircuit | Heh. |
| 14:02.55 | shortcircuit | Hm. Is there an epub (or pdf) version of the new book? |
| 14:03.33 | WIMPy | I guess you will have to wait for the official release. |
| 14:05.11 | shortcircuit | The web interface isn't a real problem, it's just more convenient on my Nook. That's what I've been reading the AST.pdf with. |
| 14:06.50 | shortcircuit | I've bought the epub forms of four O'Reilly books so far, and expect I'll buy more in the future. :) |
| 14:13.20 | *** part/#asterisk clintc (~clintc@n128-227-2-246.xlate.ufl.edu) |
| 14:21.03 | *** join/#asterisk dms (~Adium@nat/digium/x-wcquqpccugcumfiv) |
| 14:31.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:35.04 | *** join/#asterisk Mhaddog__ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
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| 14:42.54 | *** join/#asterisk jkroon (~jkroon@dsl-241-252-219.telkomadsl.co.za) |
| 14:43.28 | jkroon | hi guys, when issueing an AMI Hangup request I don't seem to receive a Hangup event for that channel, only the bridged channel - is this the correct behaviour? |
| 14:44.03 | jkroon | ast 1.6.2.16.1 |
| 14:46.44 | aiksa[LV] | what type of channel? |
| 14:53.01 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:53.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:54.58 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:55.32 | jaytee | When is the new book out in print? |
| 14:59.42 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-smjuupksptrqnocw) |
| 15:02.19 | Kobaz | jkroon: you should get hangup events for all channels that were hung up |
| 15:02.24 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 15:05.38 | jkroon | Kobaz, interesting. i don't think i am. |
| 15:05.45 | jkroon | let me re-read the output more carefully. |
| 15:06.51 | shortcircuit | is liking the Asterisk book. Chapter 1 reads like the author's muse took him in a choke-hold and commanded the words to flow. |
| 15:08.17 | jkroon | Kobaz, if it only generates a Hangup event for the bridged channel (not the one the hangup was requested on) that's a bug right? that needs to be reported? |
| 15:08.21 | *** part/#asterisk skyion (~brad@tcs-gw.bulwer.thusa.net) |
| 15:08.34 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 15:08.41 | Amnesia | could anyone help me with setting up dahdi? |
| 15:09.08 | Katty | goooooooooooooood morning!!! |
| 15:09.10 | WIMPy | jkroon: Do you get a succes report for your request? |
| 15:09.17 | Amnesia | I've already set "context=outgoing" but when I type dahdi show channels, it keeps saying context: default |
| 15:09.22 | WIMPy | Good afternoon |
| 15:10.10 | WIMPy | Amnesia: All options must be before the channels they should be used on. |
| 15:10.25 | jkroon | WIMPy, yes. |
| 15:10.26 | Amnesia | hm, I've set it beneath [channels] |
| 15:11.22 | WIMPy | jkroon: I don't know if that's the way it's supposed to work, but it seems logical to me that you don't get another event for your action then. |
| 15:11.23 | jkroon | I am either missing call hangup events or somewhere I'm issueing a Hangup action but the call doesn't generate a Hangup event. re-reading the trace I do eventually (quite a bit down) locate the second Hangup event. |
| 15:11.55 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:12.00 | WIMPy | ok |
| 15:12.08 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:12.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:12.39 | Amnesia | WIMPy: http://pastebin.com/kb7Dkn01 |
| 15:12.45 | Amnesia | http://pastebin.com/8tYPatYi |
| 15:14.34 | WIMPy | Amnesia: You said you put it beneath [channels] but that paste shows it above. |
| 15:14.37 | jkroon | ok, added code to just remove the call from my call list if I issue a hangup request and get a non-success response. |
| 15:14.44 | Amnesia | WIMPy: I tried both |
| 15:14.59 | Amnesia | no success |
| 15:15.08 | WIMPy | It needs to be between [channels] and channel => |
| 15:15.39 | WIMPy | But your cli output doesn't show any real channels at all. |
| 15:16.13 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:16.13 | Amnesia | hm, did I forget some steps? |
| 15:16.27 | Amnesia | haven't done anything with /etc/dahdi/* really |
| 15:16.42 | Amnesia | except than dahdi_genconf and then change /etc/dahdi/system.conf |
| 15:16.50 | Amnesia | to "nl" instead of "us" |
| 15:17.09 | WIMPy | You haven't configured any channel. |
| 15:17.29 | Amnesia | hm |
| 15:17.45 | WIMPy | That was a sample configuration, not a generated one. |
| 15:18.23 | Amnesia | /etc/asterisk/chan_dahdi.conf or /etc/dahdi/system.conf? |
| 15:18.31 | Amnesia | how can I generate it? |
| 15:18.45 | WIMPy | system.conf |
| 15:18.52 | WIMPy | dahdi_genconf |
| 15:19.14 | WIMPy | Err, no. that chan_dahdi.conf. |
| 15:19.30 | Amnesia | root@asteriskpbx:/home/admin# dahdi_genconf |
| 15:19.30 | Amnesia | Empty configuration -- no spans |
| 15:19.30 | Amnesia | Empty configuration -- no spans |
| 15:19.30 | WIMPy | Sorry, I don't fiddle with dahdi that much. |
| 15:19.50 | Amnesia | np, ++ for the fact you're at leasst trying to help me:D |
| 15:20.04 | WIMPy | Guess you need to load kernel drivers for your hardware then. |
| 15:20.52 | Amnesia | they're already loaded |
| 15:21.05 | Amnesia | driver should be 'wcb4xxp' but is actually 'hfcmulti' |
| 15:21.05 | Amnesia | pci:0000:30:00.0 wcb4xxp+ 1397:08b4 Junghanns QuadBRI ISDN card |
| 15:21.38 | WIMPy | Oh, you need to make sute you load the one you want to use and only that. |
| 15:22.14 | WIMPy | So if you want to use dahdi, blacklis hfcmulti in /etc/modules.d/blacklist or somewhere in that area. |
| 15:22.16 | Amnesia | wcb4xxp and hfcmulti are both loaded |
| 15:22.33 | WIMPy | Ok, but only one of them can work. |
| 15:22.50 | WIMPy | Make sure the other doesn't get loaded. |
| 15:22.58 | Amnesia | removed hfcmulti, and dahdi_hardware doesnt state the same anymore, but dahdi_genconf does |
| 15:23.09 | WIMPy | prefers hfcmulti, but that depends on your needs. |
| 15:23.35 | WIMPy | You probably have to remove and re-load wcb4xxp. |
| 15:24.00 | Amnesia | sec:) |
| 15:24.04 | WIMPy | It won't magically recognize the hardware was freed. |
| 15:25.11 | Amnesia | hm wcb4xxp on itselfs seems to word |
| 15:25.13 | Amnesia | work* |
| 15:25.49 | Amnesia | thx:) |
| 15:31.58 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 15:33.14 | schmidts | hey guys, does anyone of you know a problem that dialplan execution hangs after a playback. but its only sometimes not everytime |
| 15:34.41 | Amnesia | WIMPy: /etc/dahdi/system.conf got generated |
| 15:34.56 | Amnesia | but /etc/asterisk/chan_dahdi.conf didnt change |
| 15:36.04 | WIMPy | No, it generates /etc/asterisk/dahdi-channels.conf that can be included in chan_dahdi.conf. |
| 15:38.14 | Amnesia | ah, it isnt included by default.. |
| 15:39.18 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:39.45 | Amnesia | what's the include syntax of asterisk? |
| 15:39.59 | *** join/#asterisk Abd4llA (~Abd4llA@41.35.187.189) |
| 15:40.26 | _Corey_ | Amnesia: include => xxxx |
| 15:40.34 | Amnesia | /etc/aste....? |
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| 15:54.02 | chazzam | ~book |
| 15:54.02 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
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| 16:00.20 | *** part/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
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| 16:02.51 | *** join/#asterisk odin917 (~Gavin_Sil@64.61.105.226) |
| 16:03.19 | odin917 | hey everyone |
| 16:04.18 | odin917 | my sip trunk provider is telling me my asterisk system is "registering funny", he sent me the output and on one of the lines the registration output on his end is " Contact: <sip:s@MY.IP.ADDRESS> " |
| 16:04.45 | odin917 | does anyone know what/where i set the "contact" |
| 16:05.15 | odin917 | he said it should read, myUserName@my.ip.address ; not s@my.ip.address |
| 16:05.31 | kyler | I'm still trying to find a SIP DID provider offering (incoming) SMS. I see Anveo and Google Voice but I'd like something with easy Asterisk integration. |
| 16:07.05 | _Corey_ | kyler: I think Vitelity is offering it, though I haven't used it myself |
| 16:14.20 | kyler | _Corey_: Thanks for the pointer! That sounds like what I want. I'm still trying to get details... |
| 16:19.09 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 16:19.51 | [Outcast] | Has anyone had issues with the AMI taking a really long time to process requests? |
| 16:22.04 | *** join/#asterisk baldrailers (~Adium@180.194.7.54) |
| 16:25.17 | [Outcast] | no one else has had issue with AMI executing commands slowly? |
| 16:32.12 | *** join/#asterisk carlopires (~carlo@187.115.76.237) |
| 16:34.02 | *** join/#asterisk carlopires (bb734ced@gateway/web/freenode/ip.187.115.76.237) |
| 16:35.31 | odin917 | does anyone know where I set "contact" in the sip registration? |
| 16:35.54 | odin917 | My provider is seeing Contact: <sip:s@myIPaddress> |
| 16:36.12 | odin917 | instead of myUser@myIPaddress |
| 16:36.49 | _Corey_ | odin917: I'm guessing you're doing a "register => user:password@host" |
| 16:37.01 | _Corey_ | try "register => user:password@host/user" |
| 16:37.33 | odin917 | _Corey_: thank you, trying now |
| 16:37.57 | carlopires | I was trying to create some extensions in extensions.lua but not of them is show on "dialplan show" cli |
| 16:38.07 | carlopires | I'm using *1.8 |
| 16:41.03 | carlopires | I noticed globals is set with extensions.lua but not extensions |
| 16:46.50 | *** join/#asterisk wonderworld (~ww@port-92-201-125-20.dynamic.qsc.de) |
| 16:49.08 | carlopires | does anyone know why extensions created in lua is not listed with dialplan command ? |
| 16:49.15 | *** join/#asterisk AppLync (~AppLync@mail.applync.com) |
| 16:49.22 | AppLync | afternoon |
| 16:49.31 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 16:50.37 | AppLync | I am trying to figure out a way to change the user state of a user so that the BLF's will light up. They wont actually be getting a call, be cause we are pushing the presence from somewhere else. |
| 16:50.53 | AppLync | We just want to remtoely light up the BLF's for an extension |
| 16:50.57 | LemensTS | Anyone here run asterisk as a dhcp server without internet connection? My concern is Polycom phones need a NTP server to update the date/time, without internet they can't. Only way I see is to install NTP server on the asterisk server...not sure if I am doing it proper, not much information on it.... |
| 16:51.33 | AppLync | dhcp server in box without internet is fine. |
| 16:51.55 | LemensTS | AppLync: yes, how do you update the date/time on the phones |
| 16:52.32 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 16:52.34 | _Corey_ | AppLync: Look into the custom device state function, I think that's what you need |
| 16:52.44 | AppLync | without internet, you will need to install NTP service as well, and just pint the dhcp server settign for ntp to itself. The phone siwll pull the time form the dhcp server |
| 16:53.18 | AppLync | That is what I am doing. but is doesnt seem to work. I can see the change in the devstate list command, but the BLF's dont light up. |
| 16:53.38 | LemensTS | AppLync: yea that is what i was trying to do, I will look for more information |
| 16:53.44 | asteriskmonkey | hey in asterisk 1.8.2 should i be able to see the channels of a pri if i do dahdi show channels? im seeing when i do dadhi_cfg -vvv .. channels are getting listed and there in /etc/dahdi/system.conf ok and /etc/asterisk/chan_dahdi.conf but in the cli i see nothing |
| 16:53.50 | _Corey_ | AppLync: What does your hint look like for one of these? |
| 16:54.02 | asteriskmonkey | i do a dahdi show status and i see the span is up and ok so why no channels showing? |
| 16:54.18 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:56.59 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
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| 16:58.58 | dacm_work | Hi guys. |
| 16:59.06 | dacm_work | Trying to diagnose 1-way audio. |
| 16:59.23 | dacm_work | Should I be worried about 401 - Unauthorized, or is that normal? |
| 16:59.44 | asteriskmonkey | anyone know why a pri wouldnt show channels but show ok in the asterisk cli? |
| 17:02.02 | dacm_work | It's coming after an INVITE. Later afterwards another INVITE is sent, this time with an Authorization header. (So I'm guessing it's expected behaviour. At least on asterisk's part.) |
| 17:04.49 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 17:07.17 | CoffeeIV | I am using asterisk 1.8. My users.conf has a lot of "host = dynamic" lines, and my log has a lot of failed DNS lookups for host "dynamic". Where can I find documentation on what is actually allowed and expected in users.conf, so I can fix this ? |
| 17:07.23 | LemensTS | Can you add Dahdi lines to the line buttons on Polycom phones, so if that line is in use, it is lit up? |
| 17:12.41 | kyler | _Corey_: Vitelity+S.MS looks like just what I needed. Thank you so much for the help! I've been thrashing with this for months trying to find a solution. |
| 17:13.23 | _Corey_ | kyler: No problem... I read a little about it and it seems like they have their act together. We've used some of their services for a while and they've been solid. |
| 17:13.47 | *** join/#asterisk wonderworld (~ww@port-92-201-125-20.dynamic.qsc.de) |
| 17:14.40 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 17:14.53 | kyler | _Corey_: I made a voice call to their support number first just to get a feel for their service. It seemed good. I think this is going to be a huge relief and save me a lot of money. |
| 17:14.54 | *** join/#asterisk m_tadeu (~quassel@static-b5-252-50.telepac.pt) |
| 17:15.35 | _Corey_ | Good luck |
| 17:16.32 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 17:17.01 | LemensTS | Or I guess it is called "Shared Lines" but I was wondering if it worked with Analog lines |
| 17:20.16 | [Outcast] | has anyone seen long delays when executing commands on the AMI? |
| 17:23.43 | pabelanger | With Background() DTMF can interrupt the prompt that is playing, anybody know how to do the same with SayDigits()? |
| 17:24.39 | russellb | pabelanger: you can't |
| 17:25.07 | russellb | someone should add it ... has been requsted many times before, but nobody has done it |
| 17:25.16 | russellb | background ability for all of the Say variants |
| 17:25.21 | WIMPy | create one file out of the digits you want to say and use background? |
| 17:25.22 | Qwell | Background() should have never been its own app, heh |
| 17:25.40 | pabelanger | russellb: yar! |
| 17:25.46 | russellb | WIMPy: gets more complicated when you want to say dates, times, a large range of numbers |
| 17:26.05 | WIMPy | Sure. But it's possible. |
| 17:26.11 | russellb | i guess so |
| 17:26.16 | pabelanger | Qwell: agree, we should have created a barge-in option |
| 17:26.30 | russellb | here's what you do ... |
| 17:27.20 | Qwell | originate a new channel, start Record(), call SayDigits(), throw the filename in a global variable, and Background() that file on the original channel |
| 17:27.23 | russellb | you Originate() a call that will use ChanSpy() in whisper mode to play the file to the channel, meanwhile you just use Background() with silence ... after background exits, you use the app that hangs up a channel to kill the call you originated |
| 17:27.26 | Qwell | haha |
| 17:27.44 | Qwell | russellb: ^5! |
| 17:27.47 | russellb | \o/ |
| 17:27.48 | AppLync | _Corey_: if I do a SIP show hintsa? |
| 17:27.55 | russellb | not convoluted at all |
| 17:27.56 | AppLync | core show hints, rather? |
| 17:35.11 | AppLync | vitelity defintiely has their act together. I ave also had great luck with didforsale. |
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| 17:41.45 | fross | Does SIP only work for internal pre defined extentions or call it make outgoing calls to any number? |
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| 18:32.58 | dacm_work | Hi guys. |
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| 18:33.53 | dacm_work | When redirecting a call from my ITSP back out to my ITSP, * is not sending out any RTP packets, although it is receiving both sets from my ITSP. |
| 18:34.15 | dacm_work | i.e. The call connects fine but there is no audio. |
| 18:34.35 | dacm_work | How can I go about debugging this further? |
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| 18:41.29 | daxt | hi room , i have asterisk setup server which now have two extensions , i can ring from one extension to another using two softphones and vise versa , but i cannot hear the call , what could be the issue ? |
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| 18:42.11 | daxt | skype works on both PCs were linphone (softphone ) is installed , so its not a voice related issue |
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| 18:57.53 | LemensTS | On IP phones, what are the line buttons for? On our analog systems, they show if a pots line is in use or not. |
| 18:58.05 | LemensTS | I mean on our analog systems phones... |
| 18:58.46 | paulc | LemensTS: They're usually used for instances of calls. So you can juggle multiple calls at once, or have a multiple extension numbers assigned to a phone (and be able to tell which one was called) |
| 18:58.49 | WIMPy | They are there to make it easier for ppl mentally stuck in that time. |
| 18:58.59 | WIMPy | Or to change accounts. |
| 18:59.01 | paulc | LOL @ WIMPy - ooh the burn! :) |
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| 19:00.13 | LemensTS | paulc: ok, so if i have 2 lines on an ip phone, and i create a sip account "800", then line 1 would be 800? |
| 19:01.04 | LemensTS | and most people would create another sip account "801" and asssign it as line 2? |
| 19:01.26 | LemensTS | or assign line 2 as 800 as well? |
| 19:01.48 | tonsofpcs | WIMPy: you mean like how I have to explicitly explain the difference between DIDs and BRIs to IT folk? |
| 19:01.53 | paulc | LemenTS: Sure, if you assign SIP account "800" to "line 1" on the phone. A lot depends on the phone. Like a Polycom phone, for example - you can have multiple buttons on the same account, makes call juggling easy because one line button = one call. But you can also configure multiple calls per button. So button 1 = 800, button 2 = 555, and each button can handle multiple calls (use the cursor keys to select the one you want to hold/retrieve/end/transfer |
| 19:01.59 | nny | ok really banging my head here. I have 2 boxes, 1 is a test box I peridoically bring online via IP switch in the process of testing an upgrade. This isn't my preffered method, but it works. the IP phones (polycom) keep registering with the tets box, even after the switch, as well as the production box. I have checked for any network oddities, and my remote phone (linksys) shows up in registry too. Users report that they are seeing voicemails from the test b |
| 19:02.07 | paulc | tonsofpcs: seriously? ouch - that's painful! |
| 19:02.24 | nny | my guts still says network, but just pinging for ideas |
| 19:02.27 | WIMPy | tonsofpcs: No idea what kind of ppl you have to deal with. |
| 19:02.48 | paulc | nny: what happens if you reboot the phones after switchover? |
| 19:02.55 | nny | paulc: same thing |
| 19:03.29 | _Corey_ | lol, BRIs and DIDs |
| 19:05.13 | _Corey_ | I had a major VoIP carrier mess up an order recently for adding 3 DIDs to a test trunk. The rep's response "OH! I was confused because I thought you wanted 3 PHONE LINES." |
| 19:05.38 | LemensTS | paulc: ok that makes sense. im assuming most people dont use line 2 button than, im not sure why i would need 2 sip accounts on 1 phone...? |
| 19:05.51 | paulc | headslaps - reps really should be technical, not just "sales people" |
| 19:06.08 | tonsofpcs | it depends on the use case |
| 19:06.26 | WIMPy | LemensTS: To route different DIDs, e.g.. |
| 19:06.42 | paulc | LemensTS: What kind of phone do you have? At home, maybe use it for 2 different providers (like a US one and a UK one, for example). In an office environment, use it for your published extension number that every knows, and the second line for a secret numbe for the special people in your life |
| 19:06.45 | tonsofpcs | I need a 'phone' that has 16 SIP connections... |
| 19:07.26 | paulc | tonsofpcs: Something with an add on module then (Aastra? Polycom?) or... front it with Asterisk, the phone is a single endpoint and Asterisk handles your 16 accounts? (or am I missing it?) |
| 19:08.10 | tonsofpcs | paulc: oh, I'm just saying for LemensTS. I'm looking at one of these: |
| 19:08.21 | tonsofpcs | http://www.telos-systems.com/vx/index.html |
| 19:09.05 | paulc | tonsofpcs: oh yes - Telos.. they make sexy shit.. love their gear and those new VX things are cool |
| 19:09.09 | paulc | radio involvement then? |
| 19:10.11 | tonsofpcs | ya |
| 19:10.12 | tonsofpcs | currently using a Symetrix 108 |
| 19:10.24 | tonsofpcs | is an engineer for a TV/radio station |
| 19:10.57 | tonsofpcs | I've been playing with getting a bunch of hardware codecs (many with SIP capabilities) to talk with each other |
| 19:11.03 | tonsofpcs | these APT boxes are a PITA, but I think they'll be good once I get them working |
| 19:11.34 | paulc | tonsofpcs: Have you seen http://www.bionics.co.uk/PhoneSystems/PhoneBOX.aspx |
| 19:11.47 | tonsofpcs | no |
| 19:11.48 | tonsofpcs | *clicks* |
| 19:11.56 | paulc | is a software developer with a passion for radio |
| 19:12.33 | tonsofpcs | is that yours? |
| 19:14.07 | paulc | tonsofpcs: nope, but it's interesting.. good British engineering.. and kinda turns things a bit upside down.. That said, I like the Telos VX stuff too.. |
| 19:14.31 | dacm_work | Is this the best place for help with debugging problems with *? Or is there a forum somewhere I should use> |
| 19:14.33 | dacm_work | ? |
| 19:14.40 | paulc | Interesting how these days it's becoming more and more about the software and the data behind it.. caller history, notes, etc.. building screening into the system as a whole etc |
| 19:14.48 | paulc | dacm_work: ask away |
| 19:14.49 | paulc | ~ask |
| 19:14.49 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:15.17 | dacm_work | <dacm_work> When redirecting a call from my ITSP back out to my ITSP, * is not sending out any RTP packets, although it is receiving both sets from my ITSP. |
| 19:15.17 | dacm_work | <dacm_work> i.e. The call connects fine but there is no audio. |
| 19:15.17 | dacm_work | <dacm_work> How can I go about debugging this further? |
| 19:16.42 | tonsofpcs | paulc: we use a win 9x machine running word full screen for call notes, with a scan converter feeding a video line in the studio about 80' away |
| 19:16.52 | dacm_work | It's as if the RTP bridging just isn't happening. |
| 19:17.55 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 19:18.11 | dacm_work | My * is behind a NAT. (My ITSP is outside my network of course.) When one end of the call is within my network then it seems to work ok. |
| 19:18.38 | paulc | tonsofpcs: yeah, that works :) Amazing what goes on behind the scenes eh? |
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| 19:38.50 | tonsofpcs | paulc: eh, we used to use a whiteboard and someone would hold it up to a window.... simple works :) |
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| 20:07.52 | shortcircuit | dacm_work: I'm just getting started in all this, but I'm pretty sure the NAT is your problem. |
| 20:09.13 | pabelanger | ~collectdebug |
| 20:09.13 | infobot | rumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 20:09.18 | pabelanger | infobot: ping? |
| 20:09.19 | infobot | ~pong |
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| 20:09.54 | pabelanger | dacm_work: ^ PB a complete debug log showing your problem |
| 20:12.02 | becks` | hi, somebody knows a software to call some 1000 numbers and play a pre-recorded message? I know there is a project for such purposes but I can't remember the name |
| 20:14.50 | fenrus | lol |
| 20:14.51 | dacm_work | shortcircuit: I thought so at first, but I'd expect to at least see the RTP packets _leaving_ my server. |
| 20:15.06 | fenrus | this sounds like a advertising scam |
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| 20:15.16 | becks` | it's not |
| 20:15.26 | _Corey_ | this is actually pretty common |
| 20:15.28 | dacm_work | pabelanger: Will do. Instructions are on that pagge right? |
| 20:16.08 | _Corey_ | becks: you can do it with asterisk without too much difficulty if you write a small external script to push call spool files at asterisk |
| 20:16.27 | becks` | ok thanks _Corey_ |
| 20:16.57 | _Corey_ | the trick with that technique is to send only as many as you have channels you want to dial, and then sleep for as long as the message takes to play |
| 20:17.08 | _Corey_ | you'll need to look at the AMD() application though |
| 20:17.17 | pabelanger | dacm_work: yes |
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| 20:27.08 | dacm_work | pabelanger: Ok, I have the log. Is there any sensitive info I should remove from it before sharing it? (Registration with my ITSP for example?) |
| 20:27.32 | pabelanger | dacm_work: you'll have to review the capture, but usually not |
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| 20:32.44 | dacm_work | pabelanger: Do you think it's worth masking the phone numbers? Also where should I upload to? The issue tracker or just some pastebin? |
| 20:32.53 | pabelanger | dacm_work: Again your call, use PB. Remember, anything you upload will be on the web for the world to view |
| 20:33.20 | dacm_work | ok I'll probably do a quick search and replace on them. |
| 20:33.24 | dacm_work | ``** SIP TIMER: Cancelling retransmit of packet" |
| 20:33.24 | dacm_work | ^ Is that suspicious? ;-) |
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| 20:42.00 | dacm_work | http://fpaste.org/Qr8Z/raw/ |
| 20:43.11 | dacm_work | I masked anything that I thought might be sensitive, but you should still be able to follow it. (i.e. Nothing was changed to the same `mask'.) |
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| 20:51.29 | voipnet-tech | question, have a server where the end-user has been busy building queues and extensions. come to findout their ext-queues context (freepbx) has -= 5457 extensions (16523 priorities) in 1 context. =- . from an asterisk perspective, can i expect this would be causing a performance hit? a syncronization script that runs hourly performs a reload and the logger dumps 8MB of log lines into the full log. found yesterday /var/log/ |
| 20:52.18 | drmessano-lt | 5457 extensions ? |
| 20:52.23 | drmessano-lt | wow |
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| 20:53.06 | voipnet-tech | this is the whole dialplan: -= 10787 extensions (32106 priorities) in 191 contexts. =- |
| 20:54.06 | _Corey_ | voipnet-tech: how many actual phones? |
| 20:54.34 | voipnet-tech | _Corey_: just under 300 |
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| 20:54.50 | _Corey_ | Only FreePBX could produce something that awful... ;) |
| 20:55.13 | voipnet-tech | i know right! |
| 20:55.38 | _Corey_ | Someone from Digium should be able to provide a definitive answer, but my understanding is that Asterisk 1.6+ deals with large dialplans much better than its predecessors |
| 20:56.01 | voipnet-tech | luckily on 1.6.2.16.1 as of yesterday |
| 20:56.15 | _Corey_ | The killer is the reloads |
| 20:56.31 | voipnet-tech | yea takes 2-3 mins |
| 20:56.35 | _Corey_ | We had a customer with a similar FreePBX mess who was hitting "Apply / Reload" every 10 seconds |
| 20:57.12 | dacm_work | pabelanger: (Sorry forgot to poke you earlier so you might not have seen.) I've uploaded that log to http://fpaste.org/Qr8Z/raw/ |
| 20:57.27 | carrar | voipnet-tech, thats not that much |
| 20:59.48 | carrar | however all that in 1 context I bet makes reloading a bit slow |
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| 21:03.53 | carrar | voipnet-tech, you could probably reduce your extensions count by using a db |
| 21:04.32 | carrar | which would also in turn reduce your priorities possibly |
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| 21:13.57 | voipnet-tech | carrar: ty |
| 21:14.29 | jaytee | I'm trying to add a third NIC to my Asterisk server. I have one NIC connected to my cable modem which is set as dhcp and the second NIC is set as static on my LAN. When I add the third card I can't get my sip provider to register. I use flowroute and the dns lookup fails and reports this on the console: chan_sip.c: Registration from '<sip:19853426@sip. |
| 21:14.29 | jaytee | <PROTECTED> |
| 21:16.19 | jaytee | I'm not sure why it's trying to use the loopback address when the third NIC is installs. |
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| 21:29.09 | shortcircuit | dacm_work: I don't profess that I'll be able to figure it out, but do you have, e.g. Call-ID strings to search against for calls that worked, and calls that didn't? |
| 21:29.12 | Dovid | anyone here host @ Telx ? |
| 21:31.47 | dacm_work | shortcircuit: Not in that log no. That just contains one call that didn't work. |
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| 21:34.05 | shortcircuit | dacm_work: You sure? A visual inspection shows at least five different Call-IDs. Or am I misunderstanding something? |
| 21:35.40 | dacm_work | shortcircuit: Well I only made one call whilst the debug logging was on. So unless asterisk can log retroactively... |
| 21:35.58 | shortcircuit | Hm. I wonder if I'm looking at the wrong paste. |
| 21:37.24 | shortcircuit | Ah. I'm just misreading this, I suspect. I see one "set of 180 Ringing" lines, but that's about 40% into the file. |
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| 21:38.01 | dacm_work | I can only find SIP/ITSP-00000002 and SIP/ITSP-00000003 which are the 2 sides of the same call I believe. |
| 21:39.04 | dacm_work | shortcircuit: http://fpaste.org/Qr8Z/raw/ is my log. |
| 21:40.45 | shortcircuit | Yeah, that's what I'm looking at. I think I'm just out of my depth. |
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| 21:49.38 | Alborracho | does anyone know how to deal with mpg123 zombies? i have a lot of them and i think they are crashing asterisk |
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| 21:51.39 | jaytee | hears the lyrics to All you zombies from The Hooters play in his head. |
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| 21:54.38 | shortcircuit | recalls the plot to R.A.H's All You Zombies. |
| 21:55.51 | shortcircuit | Alborracho: Zombie processes are usually caused by system calls that don't return. I usually encounter them under hardware failure circumstances, when a file I/O operation hangs for a while. |
| 21:56.45 | shortcircuit | Alborracho: My best suggestion would be to use 'lsof' and see what resources they're using which they haven't yet been cleaned up. That might give you clues as to where to look. |
| 22:00.22 | Alborracho | ok, ill look that, thx |
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| 22:09.54 | DrkShadow | hey, sometimes the Dial app returns a 503 "Service Unavailable" message to my Ass-tra phones. The phones, instead of treating the error as a server error, instead try to redial immediately -- ad infinium. This causes the provider to block the calls. |
| 22:10.32 | DrkShadow | I'm trying to find out how to detect the 503 Service Unavailable message in the dialplan and treat it differently -- anything differently -- just so the phones don't try to redial. Help? How can I figure out if app Dial failed with 503 Service Unavailable? |
| 22:12.31 | *** part/#asterisk pepe_lucho (~pepe_luch@unaffiliated/pepe-lucho) |
| 22:14.08 | WIMPy | You'd need to parse the sip headers yourself. But maybe HANGUPCAUSE is good enough. You can then modify it to something else. Or just configure the phones. |
| 22:14.30 | DrkShadow | configure the phones how? Seems to be a firmware bug to me, as the sip protocol is specific. |
| 22:15.04 | DrkShadow | I wish I could get DIALSTATUS after the fact... nothing is recorded for it. |
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| 22:16.09 | WIMPy | Use Verbose after the Dial. |
| 22:16.30 | DrkShadow | ? |
| 22:16.49 | DrkShadow | ah. |
| 22:16.50 | WIMPy | Use Verbose() after the Dial(). |
| 22:17.19 | DrkShadow | I want it for cdr database data though :-( oh well. |
| 22:17.52 | WIMPy | You should be able simply add it. |
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| 22:20.18 | DrkShadow | (we dial numbers that cause this issue very infrequently) |
| 22:20.37 | DrkShadow | (and I'm not about to test because it will likely result in our international calling being disabled practically immediately) |
| 22:21.13 | WIMPy | If you do it now, you at least know when it happens. |
| 22:21.40 | DrkShadow | I can do it after hours.. in three hours. |
| 22:22.07 | DrkShadow | and set it up now.. |
| 22:23.08 | DrkShadow | OH MY GOD |
| 22:23.33 | DrkShadow | I was just going to give the congestion tone when the call failed, if it failed with that reason, assuming that what happens is the error would stop. |
| 22:23.59 | DrkShadow | In actuallity, if the dial app returns, I _always_ call congestion(). This might be what's making the phones redail... |
| 22:24.03 | DrkShadow | puts Hangup() instead |
| 22:24.04 | WIMPy | You need to Hangup() with parameter. |
| 22:24.16 | DrkShadow | hmm. |
| 22:24.25 | WIMPy | The parameter is the important bit. |
| 22:24.49 | WIMPy | Try Hangup(1) |
| 22:25.21 | DrkShadow | Hangup(${HANGUPCAUSE}) ? |
| 22:25.35 | WIMPy | That will do nothin. |
| 22:25.41 | DrkShadow | ... |
| 22:26.11 | WIMPy | You need to get rid of whatever HANGUPCAUSE is set to and triggers that behaviour. |
| 22:26.25 | DrkShadow | I think congestion() is causing that behavior. |
| 22:27.04 | WIMPy | That could actually make sense. |
| 22:27.19 | DrkShadow | ... for local calls I use either congestion(), hangup(), or Busy() and then Hangup() |
| 22:27.53 | DrkShadow | maybe I'll try Congestion() Hangup() |
| 22:28.07 | DrkShadow | (that's actually what I do elsewhere, too. Huh) |
| 22:28.44 | WIMPy | has never used anythin other than Hangup. |
| 22:29.47 | DrkShadow | probably don't need to, but I do use busy() and congestion() depending on how DialStatus comes out... which I may not need to do, either. |
| 22:30.04 | DrkShadow | would love to have other people's comments :-) |
| 22:30.12 | WIMPy | No, that is taken care of by Dial. |
| 22:30.20 | DrkShadow | hmm |
| 22:30.28 | _Corey_ | Never underestimate a good hangup code |
| 22:30.35 | _Corey_ | :) |
| 22:30.43 | DrkShadow | even if the dial app continues through the dialplan if it doesn't succeed? |
| 22:30.44 | *** part/#asterisk dms (~Adium@nat/digium/x-wcquqpccugcumfiv) |
| 22:30.55 | DrkShadow | probably I shouldn't do that in these cases.. |
| 22:31.04 | WIMPy | Dial will set HANGUPCAUSE and that will be used by Hangup, unless you specify something different. |
| 22:31.19 | DrkShadow | oh shit. I do need it in one place.. but yeah, ok, that works :-) |
| 22:31.44 | WIMPy | So a Hangup after a Dial will reflect the sate f the Dial. |
| 22:31.46 | WIMPy | of |
| 22:31.59 | DrkShadow | cool :-D |
| 22:32.21 | justdave | is there a way to feed dundi a list of valid extension numbers other than to have the extensions exist in a context? |
| 22:32.49 | WIMPy | justdave: Like what? |
| 22:33.30 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 22:33.55 | justdave | well, what I'm trying to do: I have several offices that currently have a dundi mesh to figure out which office owns which extension number. I'm trying to set up a "backup" routing so that servers can report on a lower priority "I can see someone else who has this extension" rather than "I own it" |
| 22:34.42 | justdave | but putting in an _XXX extension with a DUNDILOOKUP() in it appears like it says "I own any three digit extension" instead |
| 22:34.49 | WIMPy | You can use multiple contexts with different priorities. |
| 22:35.08 | justdave | right, trying to figure out how to feed the lower-priority one |
| 22:37.35 | WIMPy | That scenario might be beyond the capabilities of the on-board instruments. |
| 22:38.23 | justdave | I tried making one with a switch=> in it that points back at the direct-routing dundi context, and that worked for a while, and then suddenly today it starts hosing asterisk when that dialplan line gets hit. Commenting out that line stopped it from hosing. Haven't changed asterisk version or anything recently so no idea why it suddenly started doing that. |
| 22:38.38 | justdave | was trying to figure out a dialplan-way to emulate switch=> I suppose |
| 22:39.22 | justdave | which works fine for direct calls with about 3-lines of dialplan including a DUNDILOOKUP function, but doesn't work so well for feeding the results back to another dundi context |
| 22:39.40 | WIMPy | There are functions for dundi lookups. But I never tried them. |
| 22:41.13 | WIMPy | Well, the concpt is clearly another one: Just route the lookups but always set up a direct connection. To generate somethig to route the connection via dundi meand, I hve no idea. |
| 22:41.30 | WIMPy | means |
| 22:42.42 | justdave | basically if there's a network disruption between paris and san francisco, but both of those servers can still see toronto, I want to route calls to san francisco from paris via the toronto server |
| 22:43.02 | justdave | which I actually already had working, using switch=> before switch started crashing on me |
| 22:43.46 | WIMPy | Yes, got that, but I can't see how to do that other than manually adding matching extensions. |
| 22:57.57 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 23:01.31 | justdave | of course if I'm writing dialplan functions anyway I suppose I could just write up a quick ami-enabled CGI and hit it via an agi script that hits it over https or something |
| 23:01.36 | *** part/#asterisk nny (~Scott_2@174.107.201.103) |
| 23:01.53 | justdave | bypass dundi altogether :) |
| 23:06.44 | justdave | my use case is considerably less complex than dundi was designed to deal with anyhow. :) |
| 23:06.59 | justdave | (and can all be accomplished over a secure network) |
| 23:25.21 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |