| 00:06.09 | *** join/#asterisk adolfomaltez (~taro@190.62.233.253) |
| 00:13.37 | Mon|A|rch | so, in my IVR, sometimes i have to hit a DTMF tone twice for waitexten to pick it up. from what i can tell this behavior isn't consistent |
| 00:13.56 | Mon|A|rch | it happens on all of my waitextens, but never the same one twice |
| 00:14.07 | Mon|A|rch | sometimes it'll be all of them |
| 00:14.12 | Mon|A|rch | sometimes it'll just be one |
| 00:15.00 | Mon|A|rch | any ideas? |
| 00:15.07 | Mon|A|rch | why would this happen normally |
| 00:15.13 | Mon|A|rch | i can paste the relevant code |
| 00:15.23 | Mon|A|rch | but it's not very substantial |
| 00:24.00 | igcewieling | Mon|A|rch: SIP or POTS? |
| 00:24.34 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 00:29.26 | Mon|A|rch | SIP |
| 00:32.56 | igcewieling | confirm you are using rfc2833 ("sip show peer X") |
| 00:36.15 | igcewieling | Mon|A|rch: What version of Asterisk? |
| 00:38.47 | *** join/#asterisk Billiard (~jordan@cpe-24-208-88-79.new.res.rr.com) |
| 00:39.52 | Billiard | Which module provides the "Log" application? :/ |
| 00:41.40 | Mon|A|rch | igcewieling, 1.8 |
| 00:42.06 | Mon|A|rch | and it is rfc2833 |
| 00:42.15 | Mon|A|rch | it picks them up, just occasionally it doesn't |
| 00:42.18 | Mon|A|rch | it's really strange |
| 00:42.27 | igcewieling | Mon|A|rch: is the endpoint device also forced to rfc2833? |
| 00:43.09 | igcewieling | Mon|A|rch: the most common cause of DTMF issues on SIP is that the call is for some reason using inband. Make a call then do a sip show channel X to confirm rfc2833 |
| 00:44.21 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 00:46.24 | Mon|A|rch | alright |
| 00:46.26 | Mon|A|rch | thanks |
| 00:46.51 | igcewieling | Billiard: looks like app_verbose provides both Verbose and Log |
| 00:47.52 | Billiard | thanks, I just randomly tried that one and got it :p |
| 00:48.09 | Mon|A|rch | it's rfc alright |
| 00:48.15 | Mon|A|rch | hold on, I'll make a different kind of call |
| 00:48.18 | Mon|A|rch | see if that's the issue |
| 00:51.45 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 00:59.54 | *** part/#asterisk Billiard (~jordan@cpe-24-208-88-79.new.res.rr.com) |
| 01:02.25 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:02.25 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:06.51 | *** join/#asterisk DarthExpeditor (~DarthExpe@rrcs-71-43-76-226.se.biz.rr.com) |
| 01:10.07 | *** part/#asterisk ghost75 (~trechber@dslb-188-105-026-157.pools.arcor-ip.net) |
| 01:11.15 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:36.28 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 01:36.36 | *** join/#asterisk KJ4IPS (~KJ4IPS@96-38-107-69.dhcp.jcsn.tn.charter.com) |
| 01:36.39 | saint_ | is there a web site with theme songs for the indications.conf file ? |
| 01:36.44 | saint_ | I'm looking for star trek :) |
| 01:37.02 | *** join/#asterisk ankurgupta0001 (~ankurgupt@216-75-234-234.static.wiline.com) |
| 01:37.36 | ankurgupta0001 | ip not updating on sip peer on asterisk |
| 01:39.56 | KJ4IPS | Anyone had success with chan_motif and STUN (cant seem to recieve calls from gVoice when going through nat, and debuggibg rtp shows that it is sending the internal IP, and stun debug says that it is ignoreing the mapped address (that cant be right, can it )) |
| 01:44.16 | KJ4IPS | If anyone has any ideas: http://pastebin.com/gTa2wJqr |
| 01:44.18 | igcewieling | ankurgupta0001: Glad I could point you in the right direction |
| 01:44.30 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:44.59 | igcewieling | KJ4IPS: Asterisk's STUN support is not great. does chan_motif support things like externip and localnet? |
| 01:45.12 | ankurgupta0001 | yes |
| 01:45.18 | ankurgupta0001 | u were right |
| 01:45.22 | KJ4IPS | if it does, it has a different name for them (or i cant spell) |
| 01:45.27 | igcewieling | ankurgupta0001: Does the SIP peer register? |
| 01:45.45 | igcewieling | ankurgupta0001: i.e. host=dynamic |
| 01:48.06 | ankurgupta0001 | seems like it is not able to register when the call is going and ip changes |
| 01:48.30 | ankurgupta0001 | yes the host=dynamic |
| 01:49.02 | KJ4IPS | ankurgupta0001, how long is the registration (maybe not re-registering quick enough) |
| 01:49.09 | igcewieling | ankurgupta0001: I would expect bad things to happen if you have a call active and your IP changes. |
| 01:49.36 | KJ4IPS | :e rtp.conf |
| 01:49.40 | ankurgupta0001 | should I try reducing the min registration period |
| 01:49.44 | igcewieling | ankurgupta0001: your best bet is a VPN tunnel from your phone to the network your Asterisk is on and make THAT handle the IP changes. It makes things a lot more complicated though. |
| 01:49.59 | KJ4IPS | on another thought, that probably wont do any good... |
| 01:50.05 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:50.06 | igcewieling | It is up to your sip client to re-register when the IP changes. Have you tried any other sip clientes? |
| 01:50.40 | ankurgupta0001 | tried to use pjsip |
| 01:51.02 | ankurgupta0001 | seems like one of the best and all the other also uses the same |
| 01:51.04 | igcewieling | I don't know anything about softphones, in my opinion they all suck and we stick with hardphones |
| 01:51.21 | ankurgupta0001 | yes,u r right |
| 01:51.38 | ankurgupta0001 | they are not as good as hardphones |
| 01:57.09 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 01:57.15 | solitude88 | Is the dumb ass Jerry McCorckle in here |
| 01:58.48 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:58.48 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:59.04 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 02:08.15 | *** join/#asterisk piffey (~pfe@gateway/tor-sasl/piffey) |
| 02:11.06 | KJ4IPS | there is a goto in stun.c... |
| 02:22.05 | *** part/#asterisk adolfomaltez (~taro@190.62.233.253) |
| 02:22.13 | *** part/#asterisk piffey (~pfe@gateway/tor-sasl/piffey) |
| 02:22.13 | igcewieling | KJ4IPS: http://xkcd.com/292/ |
| 02:22.30 | *** join/#asterisk adolfomaltez (~taro@190.62.233.253) |
| 02:23.00 | KJ4IPS | whatever, i am going to impliment an externip patch... |
| 02:28.41 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 02:29.11 | saint_ | with a line like this: same => n,GotoIf($[${TEST} = 1]?weasels:iguanas) |
| 02:29.31 | saint_ | how do I know if weasels and iguanas are contexts or a label ? |
| 02:29.36 | saint_ | like should I do |
| 02:29.45 | saint_ | same => n(weasels) .... |
| 02:29.45 | saint_ | or |
| 02:29.48 | saint_ | [weasels] |
| 02:29.51 | saint_ | (...) |
| 02:30.53 | WIMPy | By reading the manual. |
| 02:31.12 | WIMPy | If the destination has only one atom, it's a priority. |
| 02:31.56 | igcewieling | weasels = priority or label. 1234,weasels = extension,priority and bob,1234,weasels = context,extension,priority. This applies to pretty much any place you can go to or gosum |
| 02:32.48 | saint_ | igcewieling: thanks |
| 02:33.44 | igcewieling | saint_: every time you ask a question answered in the asterisk book god kills a kitten (and our mental tally of your karma does down). If you don't watch out there will be no kittens and nobody will help you anymore. |
| 02:34.50 | WIMPy | core show ... might be quicker. |
| 02:35.22 | fling | -- Executing [001@hh:1] Dial("SIP/000-00000039", "iax2/000@u-kit,,Tt") in new stack -- Called iax2/000@u-kit |
| 02:35.25 | fling | [Jan 31 09:34:08] WARNING[21727][C-000000b1]: chan_iax2.c:11140 socket_process_helper: I don't know how to authenticate 000 to 10.0.0.36 |
| 02:37.18 | saint_ | igcewieling: found it in chapter 10. thanks. |
| 02:38.22 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 02:40.57 | *** join/#asterisk razorgank (~quassel@203.160.180.2) |
| 02:42.43 | razorgank | hello, based on my old sip logs, the "Contact" field still shows the public ip address, how do i force it to use the private ip on asterisk? |
| 02:42.51 | razorgank | http://pastebin.com/NZ79p3aA |
| 02:43.22 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 02:49.54 | igcewieling | razorgank: externip= and localnet= |
| 02:51.28 | razorgank | igcewieling: yup already defined that part : localnet=10.128.147.118/255.255.255.192 should i define externip also? the box is not intended for public |
| 02:51.43 | igcewieling | that is not a valid localnet. |
| 02:52.51 | igcewieling | razorgank: if you don't need to talk to the outside, then remove the localnet and externip, it is only needed to talk to devices outisde your network |
| 02:53.40 | KJ4IPS | can't spell NULL |
| 02:53.42 | igcewieling | devices with IPs inside the localnet will not get the externip address. What is the IP of the client. |
| 02:54.23 | KJ4IPS | sudo -u builduser make |
| 02:54.25 | igcewieling | I don't know for sure, but since 10.128.147.118/255.255.255.192 is not a valid local network specification Asterisk may not be accepting the setting. |
| 02:55.51 | igcewieling | I assume 10.128.147.118 is an IP on the asterisk server, which is not correct. On your asterisk box, do a netstat -rn and that will show you your connected networks |
| 02:57.43 | Kobaz | 10.128.147.118/255.255.255.192 is a legitimate block |
| 02:57.57 | Kobaz | should be written as 10.128.147.64/26 but it's valid nonetheless |
| 02:59.50 | igcewieling | Kobaz: Asterisk might accept it as valid, and assume 10.128.147.64, but that doesn't make it correct. |
| 03:02.18 | igcewieling | Try creating a route like that using the Linux "route" command: netmask doesn't match route address |
| 03:03.44 | Kobaz | true, some parsers are more strict than others |
| 03:18.46 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 03:48.00 | *** join/#asterisk din3sh (~din3sh@41.136.80.125) |
| 03:50.12 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 03:50.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 03:55.37 | KJ4IPS | darn, semes after rigging rtp to take its IP from the config file, it still does not like me |
| 04:08.32 | Kobaz | hmm |
| 04:08.44 | Kobaz | no direct way to clear the gosub stack |
| 04:10.09 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 04:10.25 | Kobaz | oh, hmm |
| 04:10.30 | Kobaz | good thing i wrote StackPeek |
| 04:21.47 | *** join/#asterisk radic (~radic@dslb-178-002-233-233.pools.arcor-ip.net) |
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| 04:28.27 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-oetmcfqdrirpzyzz) |
| 04:42.50 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 04:46.47 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 04:55.17 | *** join/#asterisk derkko (~afernande@184.sub-70-193-193.myvzw.com) |
| 04:55.47 | derkko | how can i playback a .wav file from a url? |
| 04:57.40 | ChannelZ | Hmm. |
| 04:58.43 | ChannelZ | I'd guess you'd have to use the CURL and FILE functions to actually write it to disk somewhere and then Playback()/Background() it from there. |
| 04:59.17 | ChannelZ | although looking at it FILE says "text", might puke on binary. |
| 04:59.43 | derkko | there is no way to stream it without having to download it? |
| 05:03.14 | razorgank | igcewieling: thanks for clearing that up, i'm new to subnets, do i have to define the network of the signalling and media gateway? i can successfully send OPTIONS method to media gateway but on the signalling, during call setup, the response 200 ok is not reaching the signalling, whenever i'm(asterisk) receiving an INVITE |
| 05:05.18 | igcewieling | razorgank: Asterisk uses localnet to determine of the client needs "nat fixups" (externip) or not. If the client is not inside the localnet then the fixups are done, if it isn't, then they aren't. |
| 05:05.51 | igcewieling | you can have multiple localnets if you have multiple networks behind the same NAT as asterisk (or if you have a VPN, etc) |
| 05:06.48 | ChannelZ | However * still won't send media to the right place unless it gets some from the client first. |
| 05:07.27 | igcewieling | ChannelZ: I've never found that to be a problem. |
| 05:08.14 | ChannelZ | I haven't tested it under 11 now that we have ICE and STUN and all that crap but it certainly used to be that way. |
| 05:11.19 | igcewieling | Most of our asterisk boxes are on public IPs, but we have a few assholes who insist on "bring your own broadband" and force us to be behind NAT. |
| 05:11.51 | igcewieling | Most of the time they can't even manage for port forward SSH to the PBX. |
| 05:11.53 | igcewieling | </rant> |
| 05:14.04 | igcewieling | ChannelZ: if it is a problem, port forwarding the RTP ports should solve it. |
| 05:14.52 | ChannelZ | yes |
| 05:16.53 | ChannelZ | I don't know much about STUN/ICE and whether or not they let * do the Skype-style NAT traversal so you don't really need to forward ports. |
| 05:20.04 | igcewieling | ChannelZ: In theory yes, in practice sometimes. At least from what I can tell from what I've seen on the mailing lists and JIRA |
| 05:26.56 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 05:29.41 | fling | ChannelZ: I just connected my two asterisk servers with iax2 :P |
| 05:30.06 | fling | [asterisk_A] <-iax2-> [asterisk_B] |
| 05:30.12 | fling | everything works! |
| 05:32.31 | ChannelZ | glad to hear it |
| 05:44.50 | fling | ChannelZ: what trunk=yes does? > http://etel.wiki.oreilly.com/wiki/index.php/Peering_two_Asterisk_servers_using_IAX |
| 05:44.59 | fling | ChannelZ: I do not use it and dial works… |
| 05:47.42 | ChannelZ | trunking in IAX multiplexes the traffic for multiple calls over the same connection rather than having a separate connection going for each call |
| 05:48.48 | ChannelZ | It reduces the overhead and thus uses less bandwidth as the number of simultaneous calls increase. |
| 05:49.26 | ChannelZ | (only if the calls are going between the same systems obviously) |
| 05:53.31 | fling | ChannelZ: great! I only have a single call max over iax2 atm… |
| 05:54.59 | ChannelZ | then there's no real benefit but not a detrement either I don't think |
| 06:01.22 | igcewieling | trunk=yes uses more bandwidth when you only have 1-2 (maybe 3) calls active between the same two IPs |
| 06:03.20 | igcewieling | These days using IAX2 is practically an affectation, like using a pocket watch or wearing a bow tie all the time |
| 06:05.26 | fling | igcewieling: right, I'm thinking about selling my watches on ebay… |
| 06:05.56 | igcewieling | keeping your bow ties and princess phones? |
| 06:06.00 | igcewieling | 8-0 |
| 06:06.07 | fling | haha :D |
| 06:08.17 | fling | igcewieling: princess phones… |
| 06:08.33 | fling | igcewieling: how to connect my intercom to asterisk? |
| 06:12.41 | igcewieling | first find a chicken..... |
| 06:13.10 | fling | igcewieling: a chicken? |
| 06:13.19 | fling | igcewieling: and then make a fine dish? |
| 06:13.35 | fling | igcewieling: so I have this device on the wall http://www.micronnet.ru/?page=1000065&rubrik=173&id=302702 |
| 06:13.46 | fling | igcewieling: simple, single button |
| 06:13.49 | igcewieling | Just answering a silly question with a silly statement |
| 06:14.37 | fling | igcewieling: it is ringing when somebody dials my apt number downstairs |
| 06:15.16 | fling | ChannelZ: may I connect this to asterisk somehow? |
| 06:15.36 | igcewieling | That is likely (but not guaranteed) an analog phone. Assuming it is an analog phone, then read up on connecting analog phones to Asterisk |
| 06:15.56 | igcewieling | goes back to upgrading servers |
| 06:16.02 | fling | igcewieling: right, it is probably analog |
| 06:16.12 | fling | I press a single button to open a door |
| 06:17.35 | *** join/#asterisk resist0r (~resist0r@gateway/tor-sasl/mrresist0r) |
| 06:18.22 | fling | looks like it is RING to open |
| 06:18.29 | fling | a single pulse click |
| 06:18.33 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 06:18.51 | fling | a stupid analog 15V phone :) |
| 06:22.39 | igcewieling | almost done upgrading 9 servers |
| 06:28.29 | *** join/#asterisk tamiel (~tamiel@c-67-169-76-114.hsd1.ca.comcast.net) |
| 06:38.14 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 06:39.11 | *** join/#asterisk Nickinator (~user@123-243-142-239.static.tpgi.com.au) |
| 06:44.31 | *** join/#asterisk din3sh (~din3sh@41.136.251.206) |
| 06:47.41 | din3sh | gd mrning all |
| 06:50.23 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 06:52.03 | fling | ChannelZ: are you using fail2ban |
| 06:52.09 | fling | din3sh: hey |
| 06:53.47 | ChannelZ | yes |
| 06:55.19 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 06:55.22 | fling | ChannelZ: is this looks good? > http://pastebin.com/raw.php?i=J1gGXyqj |
| 06:56.32 | fling | din3sh: why are you using siptosis? is it the best solution? better than skypiax+freeswitch? |
| 06:57.57 | ChannelZ | I have fewer regexes that cover all those variations but I guess it looks fine |
| 06:58.17 | ChannelZ | There's a fail2ban test script where you can feed it your filter and a log file and it will tell you what it found |
| 06:58.33 | fling | ChannelZ: somali people want to call cuba :P |
| 06:59.18 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:02.41 | ChannelZ | fail2ban-regex /var/log/asterisk/whatever /etc/fail2ban/filter.d/asterisk.conf |
| 07:02.42 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:02.51 | fling | ChannelZ: thanks |
| 07:07.38 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:09.03 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1003) |
| 07:09.15 | fling | din3sh: what skype version do I need? 2.0.0.x? don't I need oss? how will sound work? |
| 07:10.36 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:10.55 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:12.35 | din3sh | i havent tried skypiax+freeswitch |
| 07:12.36 | din3sh | :p |
| 07:12.48 | fling | din3sh: smart! |
| 07:12.57 | ChannelZ | I think it's pretty ghetto |
| 07:12.59 | din3sh | dont know if its the best solution but it works alright |
| 07:13.02 | din3sh | ;) |
| 07:13.10 | fling | ChannelZ: but it works ;] |
| 07:13.26 | ChannelZ | I mean specifically Skype<->Asterisk integration |
| 07:13.32 | ChannelZ | at least the hacky through-the-client way |
| 07:13.41 | fling | din3sh: I can't get how this pcm sound works in siptosis |
| 07:13.51 | ChannelZ | and SFA is dead. But you can pay Skype for SIP. |
| 07:14.31 | din3sh | what pcm sound? |
| 07:14.40 | fling | ChannelZ: or I may add another qemu vm to my server and setup siptosis on it :P |
| 07:14.48 | din3sh | yeah the other solution is SKype for SIP |
| 07:15.12 | din3sh | fyi i have not installed siptosis on linux/* server |
| 07:15.13 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:15.14 | fling | din3sh: skypiax is using fake oss device kernel module for skype |
| 07:15.23 | din3sh | rather on the windows clients |
| 07:16.41 | ChannelZ | fucking propriatary crapnets |
| 07:18.42 | fling | din3sh: what codec are you using? |
| 07:20.27 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:23.18 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:23.19 | *** join/#asterisk thansen (~thansen@63-248-42-154.static.orml012.digis.net) |
| 07:23.23 | din3sh | codec? |
| 07:23.25 | din3sh | alaw |
| 07:23.26 | din3sh | :p |
| 07:24.14 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:24.27 | thansen | anyone around? I just did a massive update from 1.8.x to 11.2.1 and I no longer hear anything from Playback |
| 07:24.34 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:24.48 | fling | din3sh: but it is bad |
| 07:25.05 | thansen | I don't hear ringing either |
| 07:25.21 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:26.45 | din3sh | what is your setup? |
| 07:26.58 | din3sh | can u recieve skype calls on ur * box? |
| 07:27.26 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:27.26 | ChannelZ | thansen: need more info than that |
| 07:28.10 | thansen | ChannelZ: what can I provide, I simply know enough to be dangerous with *...I'm no expert |
| 07:28.27 | ChannelZ | Any console errors? Did you read the UPGRADE-10.txt and UPGRADE.txt docs to track all the config options that might have changed? |
| 07:29.04 | fling | din3sh: skypes <-?-> [skypefarm] <-skypiax-> [freeswitch] <-/ not connected /-> [asterisk] <-iax2-> [asterisk] <-sip-> me |
| 07:29.23 | thansen | ChannelZ: everything starts up fine, but this was a distro upgrade, are those docs online? If so I'll dig them before I bother you any further |
| 07:29.24 | fling | din3sh: calls are received on freeswitch but it is not connected to my asterisk so I do not use it :p |
| 07:29.27 | ChannelZ | Are you trying to see how much latency you can get into a call or what? |
| 07:29.41 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:29.47 | fling | din3sh: and looks like siptosis is simplier, and support more features |
| 07:30.07 | ChannelZ | thansen: well those files are in the source, I assume distros put them somewhere like /usr/share/doc or whatevers. |
| 07:30.17 | fling | din3sh: also it is possible to use L16 codec with siptosis, right? |
| 07:30.51 | thansen | ChannelZ: just got my first audio out of the sucker...musiconhold works :) |
| 07:30.54 | thansen | sheesh |
| 07:31.20 | ChannelZ | thansen: I don't specifically know off the top of my head what changed from 1.8 through to 11 that might affect you, there's a myriad of possibilities of what actually stopped working |
| 07:31.42 | thansen | ChannelZ: well, I know MeetMe is gone so I'm reworking that |
| 07:31.49 | ChannelZ | Like is it firewall/NAT issues, or something as simple as sounds location, etc |
| 07:31.55 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:31.58 | thansen | but I have a pretty simple setup |
| 07:32.15 | thansen | anywho, give me a minute to dig through this doc and then I'll be back |
| 07:32.40 | ChannelZ | if the console with verbose turned up a little isn't showing a bunch of barf about "can't find file in any format" and things otherwise look normal, it sounds like a network/traffic problem |
| 07:33.38 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:33.50 | fling | ChannelZ: talking about formats, is it a good idea to convert all the sounds I use in dialplan to all allowed formats? |
| 07:34.15 | ChannelZ | Not necessarily. |
| 07:34.22 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:35.02 | ChannelZ | Asterisk will transcode when it has to, unless you're on an underpowered server with tons of calls going where the CPU usage of transcoding would hurt, it's not worth bothering with |
| 07:35.16 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:35.35 | fling | ChannelZ: no, cpu usage is really low, ok |
| 07:36.05 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:36.07 | igcewieling | I make sure my audio files are in ulaw, g729, g722, and sln16 formats |
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| 07:37.14 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:37.23 | igcewieling | But we handle 80 - 150 calls per server at the moment |
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| 07:41.16 | thansen | ChannelZ: this has got to be about the worst ever |
| 07:41.19 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:41.34 | thansen | I've been calling with my cell phone and the damn bluetooth has been activated |
| 07:41.41 | thansen | sound works fine |
| 07:41.43 | thansen | lol |
| 07:41.50 | thansen | facepalm |
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| 07:42.18 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:42.32 | ChannelZ | Troubleshooting Step 1: Is computer device plugged into electrify wall socket? |
| 07:42.48 | ChannelZ | :) |
| 07:42.49 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:43.02 | thansen | seriously |
| 07:43.16 | thansen | been that kind of night updating this thing |
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| 07:43.24 | ChannelZ | Could be worse, you could be elico apparently with his 3rd-world soviet internet connection |
| 07:43.35 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:43.47 | thansen | lol |
| 07:43.54 | fling | ChannelZ: wtf, I have 100Mb port at home in Russia |
| 07:44.09 | ChannelZ | I think we had this discussion once before |
| 07:44.27 | ChannelZ | How about 3rd-world north korean internet? |
| 07:44.28 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-162-252.red.bezeqint.net) |
| 07:44.37 | fling | ChannelZ: and it is cheap, ~$30/month |
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| 07:46.34 | ChannelZ | Actually I guess he's from isreal or someplace |
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| 07:49.48 | igcewieling | thansen: sounds like your actual problem is fatigue. |
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| 07:50.23 | igcewieling | fling: you likely have better internet than most people in the USA |
| 07:50.26 | thansen | yeah, been working on this server for hours and I starting thinking it would be a 'simple' upgrade |
| 07:51.14 | fling | igcewieling: right, here in siberia we have better internets than in moscow/europe |
| 07:51.16 | igcewieling | I upgraded 9 servers tonight, the last one, which I'm working on now I realized in VERY complex with lots and lots and lots of TNs which have to be re-routed. |
| 07:51.33 | fling | igcewieling: what is TN? |
| 07:51.38 | igcewieling | telephone number |
| 07:51.42 | fling | oh! ok :P |
| 07:52.59 | igcewieling | this company can't put all their TNs under ONE account, no that would be too easy, they need each TN to be on a separate account. If they were all on one account I could update them all with a simple SQL query. *grumble* |
| 07:55.21 | igcewieling | 358 numbers to change |
| 07:55.33 | igcewieling | PHP to the rescue! |
| 07:55.35 | fling | igcewieling: why are you changing them? |
| 07:55.51 | fling | igcewieling: what about a complex SQL query? |
| 07:55.57 | igcewieling | fling: very long story, but "change the routing" is the short version |
| 07:56.34 | igcewieling | fling: I'm not good enough with SQL for a complex SQL query. I'll use a couple of simple queries and some php |
| 07:57.47 | fling | igcewieling: good idea |
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| 08:03.24 | ChannelZ | DELETE FROM * WHERE 1=1 |
| 08:03.57 | igcewieling | the next query would then have to be delete from employees where name='manxpower'; |
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| 08:15.24 | igcewieling | Hmmm... good thing I backed up the database. 8-| |
| 08:17.18 | fling | Today I sent an email to sip provider support but got error 471 or something so I called them, they fixed their MTA/MDA and I received few emails they sent to me last November :P |
| 08:17.39 | Wiretap | fling: lol |
| 08:18.13 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
| 08:18.29 | fling | almost three months delay, never use microsoft email server in production! |
| 08:18.34 | fling | Wiretap: :p |
| 08:18.55 | Wiretap | microsoft exchange should only ever be used as a termination point |
| 08:19.04 | Wiretap | not as relaying MTA |
| 08:19.51 | fling | Wiretap: I use postfix, works fine… |
| 08:20.06 | Wiretap | I use a commercial filter appliance for my relaying |
| 08:20.40 | fling | Wiretap: what is it? |
| 08:23.51 | *** part/#asterisk thansen (~thansen@63-248-42-154.static.orml012.digis.net) |
| 08:36.46 | fling | which one US area code is better? 206,253,360,425? |
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| 08:42.16 | ChannelZ | Better how? Like a crap place to live, or.. |
| 08:42.24 | din3sh | lol |
| 08:42.57 | igcewieling | 206 is the closest to Russua. |
| 08:43.06 | ChannelZ | snickers |
| 08:46.11 | fling | igcewieling: how closer? they all are in Washington, right? |
| 08:46.43 | igcewieling | I was mostly joking, but seattle is close to the coast. |
| 08:46.57 | fling | igcewieling: and 206 is seattle? |
| 08:48.12 | igcewieling | yes. |
| 08:48.38 | igcewieling | you could try for an area code which starts with 7 + the first 2 digits of your area code just to confuse people |
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| 08:50.58 | fling | igcewieling: woot? you are confusing me now :} |
| 08:52.01 | zamba | i want to set up templates in sip.conf, instead of having to write all common options all the time.. how is this done? |
| 08:53.58 | ChannelZ | [barf](!) |
| 08:54.05 | zamba | hm? |
| 08:54.49 | ChannelZ | You make [barf](!) followed by all your common options |
| 08:54.54 | zamba | and then? |
| 08:55.12 | ChannelZ | then for a device, you can say [JoesPhone](barf) |
| 08:55.29 | zamba | and then add the differentiating options? |
| 08:55.31 | wdoekes | you could replace barf with a more sensible name, like PhoneTemplate if you want |
| 08:55.40 | ChannelZ | see the example sip.conf |
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| 08:59.27 | PbxMan | morning |
| 09:00.16 | fling | ChannelZ: may I do something similar for extensions.conf? |
| 09:00.44 | fling | PbxMan: hey |
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| 09:02.17 | wdoekes | fling: there you'll probably want 'include' |
| 09:02.34 | fling | wdoekes: thanks |
| 09:02.40 | ChannelZ | fling: not really, but I suppose macros are analogous |
| 09:02.51 | fling | is reading |
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| 09:05.11 | wdoekes | (avoid macros, use gosub) |
| 09:05.50 | fling | wdoekes: right, I may simplify things by adding some additional contexts… |
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| 09:16.23 | bogrd__ | hi, i'm trying to use ARA (Asterisk Realtime Architecture) for connecting MySQL via ODBC to my asterisk-11 running on ubuntu 12.04. I observed that none of the book which has come out on asterisk do NOT cover this topic and I am finding it real hard to get started. Is there any reason why the Asterisk Definitive guide does not say anything about ARA? Is it usable or deprecated? |
| 09:17.50 | fling | ChannelZ: I have simplified my sip.conf a lot with [barf](!) |
| 09:25.12 | *** join/#asterisk alsuren_ (~dlaban@80.169.133.251) |
| 09:27.41 | fling | igcewieling: right, 206 rocks! |
| 09:29.38 | kaldemar | bogrd__: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/I_section12_tt1465.html |
| 09:30.26 | kaldemar | bogrd__: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/I_section12_tt1465.html#Database_id243940 http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/I_section12_tt1465.html#Database_id276166 |
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| 09:30.51 | kaldemar | bogrd__: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html |
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| 09:32.10 | kaldemar | bogrd__: see contrib/realtime/ in the source package for some more table structure samples. |
| 09:36.54 | fling | [Jan 31 16:36:35] WARNING[21709]: chan_sip.c:29596 set_insecure_flags: Unknown insecure mode 'very' on line 79 |
| 09:39.08 | kaldemar | fling: "very" was deprecated in 1.4 and removed in 1.6.0. the current equivalent is insecure=port,invite |
| 09:39.24 | fling | kaldemar: thanks |
| 09:40.12 | kaldemar | fling: UPGRADE*.txt usually mentions removed and deprecated parameters and options, that's the place to look for if you use a current version and get a message like that. |
| 09:40.20 | fling | kaldemar: to define a default context for sip, I add a context=foo in [general] section of sip.conf? |
| 09:40.33 | kaldemar | fling: yep. |
| 09:40.38 | fling | ok, thanks |
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| 09:59.32 | vedic | Hello friends, is there any way to get IVR recording to the Fast AGI server? Currently the recording remains on the * server and the AGI is running on the remote server. What are options ? |
| 09:59.42 | ruben231 | hi guys |
| 10:00.42 | ruben231 | guys any help installing asterisk 1.8 and during make process this i encounter error_ any idea -------------> http://pastebin.com/x2izjxtq |
| 10:03.25 | kaldemar | ruben231: "Either run 'make menuselect' or remove the existing menuselect.makeopts file to resolve this issue." |
| 10:04.10 | kaldemar | ruben231: what is unclear about that? the 'make menuselect' part? |
| 10:05.41 | vedic | is there any way to get IVR recording to the Fast AGI server? Currently the recording remains on the * server and the AGI is running on the remote server. What are the options ? |
| 10:06.24 | fling | kaldemar: how to properly send busy without answer? |
| 10:06.28 | ruben231 | <PROTECTED> |
| 10:06.46 | kaldemar | fling: app Busy is one way. |
| 10:06.56 | fling | kaldemar: what is other way? |
| 10:07.08 | kaldemar | fling: hangup with a cause code. |
| 10:07.39 | kaldemar | fling: in general, there is no reason to answer a call if the destination is busy. |
| 10:07.46 | fling | kaldemar: same => n,Dial(SIP/000&iax2/foo/000,,t) |
| 10:07.49 | fling | kaldemar: right |
| 10:08.18 | fling | kaldemar: so when one of channels is busy, if it will answer before sending busy the call will be terminated |
| 10:08.28 | ruben231 | kaldemar: any idea somehow..? |
| 10:08.49 | kaldemar | fling: why would the channel answer if it is busy? |
| 10:09.05 | fling | kaldemar: for playtones, I'm removing it :p |
| 10:09.20 | fling | kaldemar: so now it will work fine |
| 10:09.47 | kaldemar | fling: playtones is the "unproper" way, it only indicates busy inband, but does not signal it. |
| 10:09.56 | fling | kaldemar: but without playtones, I do not hear any beeps from a softphone |
| 10:10.47 | kaldemar | ruben231: have you tried to other option it tells you? |
| 10:11.17 | kaldemar | ruben231: if you need res_config_ldap, then install the dependency for it and re-run the configure script. |
| 10:11.22 | vedic | How is asterisk-agi-audiotx which provides GET, PUT and ISEXISTING commands for AGI |
| 10:11.37 | kaldemar | ruben231: if you already have the dependency installed, the run the configure script and try again. |
| 10:11.39 | vedic | Is the stable and production ready? |
| 10:12.28 | vedic | ruben231: If you have done make before, you might want to do make clean first before starting again |
| 10:13.18 | kaldemar | vedic: that's not a part of asterisk, you'd need to ask "them". with a quick glance it seems like the project has been dead for 4 years and it is made for asterisk 1.4. |
| 10:14.09 | vedic | kaldemar: Yea, that is what worries me. No recent updates. Do you know any other way to transfer audio from * to Fast AGI server? |
| 10:14.22 | fling | kaldemar: is this regular? > [Jan 31 17:12:13] WARNING[21709][C-0000011b]: chan_sip.c:10131 process_sdp: Ignoring video stream offer because port number is zero |
| 10:14.34 | ruben231 | <PROTECTED> |
| 10:14.45 | vedic | which is stable and production ready |
| 10:15.00 | ruben231 | kaldemar:the problem what could be the dependency for this module, have no idea |
| 10:15.08 | vedic | ruben231: as kaldemar said, try ./configure again |
| 10:15.12 | fling | kaldemar: I see this warning when I dial a peer which is using linphone |
| 10:15.20 | fling | kaldemar: but the video is disabled in it |
| 10:15.47 | vedic | ruben231: and then in make menuselect don't select that module. If that is XXX it means there are missing dependencies. |
| 10:16.26 | kaldemar | ruben231: the LDAP development libraries, such as libldap2-dev or similar. |
| 10:16.51 | kaldemar | fling: probably. |
| 10:17.24 | ruben231 | vedic: yes i tried now still teh same error XXX res_config_ldap..missing dependencies still.. |
| 10:18.49 | din3sh | #shutdown -h now |
| 10:19.31 | vedic | ruben231: if you are able to keep the cursor on XXX marking of ldap, it will probably show at the bottom what is missing. You would need to install that if that module required. Also check if some other module needs ldap (but I don't think that as cause of error else error msg will show that up) |
| 10:20.47 | ruben231 | something is happening..hmmm |
| 10:23.09 | ruben231 | <PROTECTED> |
| 10:23.19 | ruben231 | vedic: thanks also |
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| 10:56.46 | vedic | Has anybody tried listening to network card or audio device using libpcap and record incoming and outgoing audios? |
| 10:57.09 | vedic | Something like oreka does. Or has anybody tried oreka with asterisk? |
| 11:00.21 | wdoekes | vedic: pcapsipdump |
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| 11:02.04 | vedic | wdoekes: thanks. Is there any way with this to store incoming and outgoing audio separately ? As it says, single session is stored as one file. If I have to get only incoming then what ? |
| 11:02.47 | *** join/#asterisk alsuren__ (~dlaban@80.169.133.251) |
| 11:04.01 | wdoekes | you can extract the audio with tcpdump -r ... -w ... src port ... |
| 11:05.47 | vedic | wdoekes: Have you tried sending the incoming audio to remote server instead of asterisk server? |
| 11:06.24 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:06.24 | vedic | wdoekes: as mentioned here: http://lists.digium.com/pipermail/asterisk-users/2010-February/244404.html |
| 11:07.34 | wdoekes | what does that have to do with anything? |
| 11:09.13 | vedic | wdoekes: using pcap to direct the audio coming to * server to some other server |
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| 11:09.28 | *** join/#asterisk resist0r (~resist0r@gateway/tor-sasl/mrresist0r) |
| 11:09.28 | wdoekes | if you want access to on-disk files from a remote machine: #nfs #httpd #ssh |
| 11:09.52 | vedic | wdoekes: This way there is no need to copy audio from * server to remote server as the audio copied directly to remote server |
| 11:10.54 | vedic | wdoekes: nfs is one option that would store audio to remote server but httpd and ssh will store the audio to * and then send it to remote server |
| 11:12.21 | wdoekes | you're not describing any problem to solve. |
| 11:13.41 | wdoekes | you begin with the solution: I want files on location Y. and then you create a problem: I don't want the files to touch location X, even though they should be created there |
| 11:14.15 | wdoekes | you could mirror all traffic to location Y. and sniff it there. |
| 11:14.23 | vedic | wdoekes: I don't want the files to touch X where * server is running. I want to get stored on Y where AGI scripts are running |
| 11:14.31 | vedic | wdoekes: I see |
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| 11:21.10 | *** join/#asterisk Anderson-D (~anderson@mail.primocollect.com.ua) |
| 11:21.12 | Anderson-D | Hello |
| 11:21.24 | Anderson-D | Is it possible to make twinkle register only 1 line on asterisk instead of 2? |
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| 11:26.06 | wdoekes | Anderson-D: to asterisk, there are no "lines". and I would expect it to register only once. you should probably ask in #twinkle |
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| 12:00.37 | fling | kaldemar: I have two peers on a single ip address |
| 12:00.54 | fling | kaldemar: and when first peer calls me, asterisk thinks it is second one |
| 12:02.31 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 12:07.29 | kaldemar | fling: how have you configured them? |
| 12:09.09 | fling | kaldemar: http://dpaste.com/901744/ |
| 12:10.07 | kaldemar | do they register to you? |
| 12:11.35 | fling | kaldemar: yes, I also have register => xxx:pass@sip.zadarma.com/xxx and register => yyy:anotherpass@sip.zadarma.com/yyy |
| 12:13.01 | kaldemar | why are you using a static host if they register to you? |
| 12:13.19 | fling | kaldemar: no, I register to them |
| 12:13.54 | kaldemar | so they do not register to you. |
| 12:14.17 | fling | I have register in my sip.conf, idk about them |
| 12:14.28 | kaldemar | find out. |
| 12:14.36 | fling | I can't |
| 12:14.46 | kaldemar | nosense. yes you can. |
| 12:15.02 | fling | but how |
| 12:15.46 | kaldemar | ask them. or look at sip traffic to your box. |
| 12:21.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:21.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:24.58 | fling | kaldemar: I only see my registrations in sip debug > REGISTER sip:sip.zadarma.com SIP/2.0 |
| 12:29.41 | fling | kaldemar: so what does this mean? |
| 12:36.33 | kaldemar | fling: that they don't register to you or the interval they try is big. |
| 12:37.44 | fling | kaldemar: I have set the same context for both peers, so I will route them right anyway |
| 12:39.48 | fling | kaldemar: call-limit=x is the sum of both incoming and outgoing? |
| 12:40.42 | kaldemar | you should really read the sample config... |
| 12:40.51 | fling | sorry, ok |
| 12:41.01 | fling | is now reading |
| 12:42.00 | kaldemar | call-limit is neither the sum of in+out nor preferred for any use nowadays. |
| 12:42.37 | fling | kaldemar: but isp guy said… |
| 12:43.10 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 12:43.30 | giany | hi, any idea what could be the reason for this message : "Timer T2 expired while waiting for next fax page" or how can I increase this timer? |
| 12:43.56 | kaldemar | fling: i don't care much what the isp guy says. :) |
| 12:44.36 | fling | kaldemar: they are also sending me wrong config examples. I will not listen to them :p |
| 12:51.11 | fling | Now I know how call-limit works, and it is not working how they think :] |
| 12:51.59 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 12:52.52 | bombev | hi huys:) |
| 12:52.58 | fling | bombev: hey |
| 12:53.03 | bombev | hows it going |
| 12:53.21 | *** part/#asterisk x4rlos (~karl@217.20.42.14) |
| 12:53.21 | fling | very well |
| 12:53.42 | bombev | can somebody take a look |
| 12:53.45 | bombev | at http://pastebin.ca/2309258 |
| 12:54.11 | bombev | and tell me why when I got 503 service unavailable the Hangup cause is 3 |
| 12:55.10 | bombev | when I checked here http://wiki.freeswitch.org/wiki/Hangup_causes : Hangup cause 3 is SIP 404 No route destination |
| 12:55.58 | Greenlight | This is Asterisk, not freeswitch though ... |
| 12:58.26 | bombev | ? |
| 12:58.47 | kaldemar | bombev: look around that message. in verbosity and in sip debug. |
| 13:00.52 | bombev | first message is Invite, 2nd 183 Session progres |
| 13:01.08 | bombev | and 3rd message is 503 Service unavailable |
| 13:01.41 | kaldemar | don't do that. put up proper pastebins if you want to show something. |
| 13:03.11 | bombev | it is strange to me because for message 503 service unavailable corresponds hangup causes 34, 38,41,42, and 4 |
| 13:03.11 | bombev | 44 |
| 13:05.53 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 13:08.26 | kaldemar | bombev: it's your box, you will see what causes it if you just bother to watch. |
| 13:09.10 | acidfoo | !book |
| 13:10.30 | *** part/#asterisk racho (~racho@46.40.123.204) |
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| 13:24.33 | *** join/#asterisk wasanzy (~wasanzy@41.57.206.3) |
| 13:25.38 | wasanzy | hello Folks |
| 13:26.25 | wasanzy | am getting this error any time I restart asterisk: http://paste.scsys.co.uk/226871 |
| 13:27.46 | *** join/#asterisk alsuren__ (~dlaban@80.169.133.251) |
| 13:28.16 | *** join/#asterisk alsuren__ (~dlaban@80.169.133.251) |
| 13:28.46 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 13:29.48 | as001 | Hello how can I send UserEvent to Manager from Dialplan with 2 or more variables (bodies)? |
| 13:30.38 | as001 | I use Asterisk 1.8 and following line does not work no metter if I put , or | between first and second variable |
| 13:30.49 | wasanzy | no one is here to help? |
| 13:31.08 | igcewieling | astra05: put something else between them |
| 13:31.22 | as001 | UserEvent(Name,SipID1: ${var1},SipID2: ${var2}) don't work |
| 13:31.56 | igcewieling | wasanzy: the error indicated those files do not exist. Did you confirm they actually exist? |
| 13:32.22 | igcewieling | astra05: what is actually seen in Manager? |
| 13:33.01 | igcewieling | Almost sounds like the issue I had with CELUserEvent, but need to know what was actually received by Manager |
| 13:33.37 | igcewieling | astra05: Does UserEvent(Name,SipID1: ${var1}XXXSipID2: ${var2}) work? |
| 13:33.47 | igcewieling | notice the lack of a second comma |
| 13:34.11 | wasanzy | igcewieling: they don't exist so what do I install to make them exist? |
| 13:34.26 | wasanzy | should install those modules from the menuselect? |
| 13:34.29 | igcewieling | wasanzy: did you build Asterisk from source or install from a package? |
| 13:34.47 | wasanzy | from source |
| 13:34.58 | igcewieling | wasanzy: enable them in menuselect and rebuild asterisk |
| 13:35.06 | wasanzy | ok |
| 13:35.09 | wasanzy | thank you |
| 13:35.16 | igcewieling | if you can't enable them in menuselect then some required -devel package is not installed in your system |
| 13:38.31 | as001 | just a sec let me try |
| 13:40.54 | as001 | igcewieling in menager everything is in first variable (with XXX) |
| 13:41.06 | wasanzy | igcewieling: I didn't see modules in menuselect |
| 13:41.47 | igcewieling | as001: Try UserEvent(Name,SipID1: ${var1}\,SipID2:\ ${var2}) |
| 13:42.14 | igcewieling | are you doing this from an AGI? |
| 13:43.42 | as001 | I am doing this from Dialplan and everything goes to SipID1 |
| 13:43.42 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:44.29 | igcewieling | that makes no sense. pastebin the actual manager event reveived from Asterisk |
| 13:45.04 | wasanzy | igcewieling: you are right, I found them now. But what are the dependencies res_curl family? |
| 13:45.16 | igcewieling | wasanzy: no idea. |
| 13:45.35 | igcewieling | but likely some google searches would be helpful. |
| 13:46.21 | *** join/#asterisk alsuren__ (~dlaban@80.169.133.251) |
| 13:46.28 | wasanzy | because I found xxx against them : XXX res_curl res_http_post res_odbc and so on |
| 13:46.32 | wasanzy | ok |
| 13:48.25 | as001 | I will do it later. Thanks anyway |
| 13:48.32 | *** part/#asterisk as001 (~uros@82.117.198.142) |
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| 14:17.25 | wasanzy | this is really bad, I can't seem to find the module dependencies |
| 14:26.12 | *** join/#asterisk vedic (~V@183.82.83.164) |
| 14:29.09 | igcewieling | something like this page (3rd google result) http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/9335 |
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| 14:35.14 | wasanzy | is that page for me? |
| 14:37.14 | igcewieling | yes |
| 14:38.37 | wasanzy | ok thank you, am looking at it |
| 14:44.19 | *** join/#asterisk carrar (tim@osburn.com) |
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| 15:06.46 | rue_house | can anyone help me with a mgcp problem? my second gateway wont work... |
| 15:08.02 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:08.02 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:10.29 | *** join/#asterisk F|ReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:10.43 | igcewieling | rue_house: try the asterisk mailing list, almost nobody uses MGCP. |
| 15:12.04 | rue_house | hmm |
| 15:12.24 | rue_house | are there any asterisk developers on the mailing list? |
| 15:15.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:15.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:19.12 | *** join/#asterisk viLeR (~viLeR___@190.145.50.27) |
| 15:20.36 | [TK]D-Fender | yes |
| 15:24.12 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:26.16 | *** join/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 15:29.20 | Qwell | rue_house: On the Asterisk Developers mailing list? |
| 15:29.25 | Qwell | no, none. |
| 15:31.10 | leifmadsen | that is a place best avoided. |
| 15:34.09 | [TK]D-Fender | Here there be dragons... |
| 15:37.47 | Qwell | signs leifmadsen up for the -biz and -users lists |
| 15:41.40 | *** part/#asterisk vedic (~V@183.82.83.164) |
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| 15:55.48 | Elleni | From time to time I get tcptls.c:101 ssl_close: SSL_shutdown() failed: 5 on asterisk console.. any one has an idea where that could come from. I am willing to learn to debug myself, but did not find anything usefull while google ing |
| 15:55.56 | Elleni | hi all btw :) |
| 15:59.48 | *** join/#asterisk klnikita (~klnikita@unaffiliated/klnikita) |
| 16:02.39 | klnikita | Hello, is it possible to have a list of modules that were "really" used during a period ? so I can disable the autoload and load just that list of modules |
| 16:02.55 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:02.55 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:03.22 | [TK]D-Fender | klnikita, No. |
| 16:07.58 | klnikita | it could be useful to have it, then I guess I will go back to my manual list, thanks for your answer |
| 16:11.08 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:19.40 | Qwell | Elleni: What version of Asterisk? |
| 16:19.40 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 16:21.27 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 16:33.12 | Elleni | Qwell, its 1.8.19.1 installed on a gentoo box... |
| 16:33.35 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 16:33.56 | Elleni | I am thinking of disable selinux to be sure it does not come from there, though I have it in permissive mode, where nothing should be blocked - just logged |
| 16:35.55 | Elleni | Now making a testcall from one softphone on a pc to another on iphone I get some other warning: WARNING[2624]: res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110. I configured asterisk to only allow srtp over tls |
| 16:37.51 | Elleni | call works well, no disconnect for 45 Mins now, just this messages on console... |
| 16:41.03 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 16:44.40 | *** join/#asterisk digsby (~digsby@65.172.54.10) |
| 16:45.57 | digsby | If anybody uses sipsak for testing, I have a problem where my from sip uri always shows to be the same as my to sip uri |
| 16:51.36 | Elleni | Qwell, have to leave now, but will come back soon.. :) |
| 16:51.41 | Elleni | bye all |
| 16:52.23 | *** part/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
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| 17:13.48 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 17:13.51 | *** join/#asterisk ircNewbz (~ircNewbz@unaffiliated/ircnewbz) |
| 17:13.52 | jmls | evening all |
| 17:14.02 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 17:14.17 | jmls | I am having some real problems with cisco79xx and the latest version of asterisk 11 |
| 17:14.56 | jmls | if firmware > 7.4, then there are all sorts of call quality issues. The call recordings themselves are perfect. |
| 17:15.20 | jmls | I initially suspected network issues between pbx and phone, but have ruled everything out. |
| 17:15.37 | jmls | If I revert back to 7.4 of sip, then the problem disapears. |
| 17:16.18 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:16.18 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:16.31 | jmls | this was _not_ a problem with asterisk 1.4, so there is probably some issues with the sip version / asterisk version matrix ;) |
| 17:17.23 | jmls | any clues on where to start looking ? 7.4 is not good for us, as call hold / pickup does not work with asterisk 11 and 7.4 ... |
| 17:21.32 | ChannelZ | What are the quality issues? Clicking and popping or just sounds bad or.. |
| 17:21.38 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:32.23 | jmls | ChannelZ : the users are saying "robotic" - and often they think the caller has "gone" because the line goes silent for a second or two |
| 17:34.13 | ChannelZ | I'd check that firmware hasn't defaulted to some other wacky codec (depends on your config if this can even happen) though I don't know what. |
| 17:36.08 | igcewieling | or some wacky packet size like 10ms or 30ms instead of the 20ms Asterisk expects by default. |
| 17:38.20 | *** join/#asterisk ideaman55 (~ideaman55@173-10-29-217-BusName-utah.ut.hfc.comcastbusiness.net) |
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| 17:42.05 | ideaman55 | Anyone... I need to use Asterisk for some real-time queueing for agents handling inbound calls in a support queue. Can 1.8 handle that just fine, or should I be trying a different branch? |
| 17:42.09 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 17:43.28 | Greenlight | 1.8 will handle that fine |
| 17:43.37 | saint_ | I just love my Linux box.. 11:40:21 up 588 days, 19:23, 1 user, load average: 0.28, 0.20, 0.18 |
| 17:43.43 | saint_ | Can't beat that with a Windows |
| 17:44.02 | pzn | Hi! I'd like asterisk to "retry" before unregistering the peer... if qualify is set to 60, it sends a package every 60 seconds. how can I ask it to retry once again if this package fails without unregistering the peer? |
| 17:46.11 | [TK]D-Fender | ideaman55, Yes |
| 17:46.32 | Qwell | pzn: Why not just set qualify to 120? |
| 17:46.33 | [TK]D-Fender | pzn, No... that is in MILLISECONDS |
| 17:47.10 | ChannelZ | ADD for phones |
| 17:49.46 | *** join/#asterisk mathi (Matthew@ip-62-235-214-212.dsl.scarlet.be) |
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| 18:14.11 | mathi | hi |
| 18:14.50 | mathi | if two servers are registered to an SIP provider for incoming calls, which server will take the call ? |
| 18:17.36 | navaismo | i think the last registered |
| 18:17.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:19.24 | igcewieling | mathi: totally depends on the provider, but usually, like navaismo said, the most recent registration will overwrite any previous registrations |
| 18:20.03 | mathi | it would be useful in the case I want one back up server, in the case one crash for example |
| 18:20.04 | mathi | right ? |
| 18:20.39 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 18:20.56 | igcewieling | pzn: Asterisk's qualify is interesting. I find using qualify=10000 and qualifyfreq=120 almost eliminates random UNREACHABLES, at the cost of taking a little longer to see that the peer went away. |
| 18:20.57 | [TK]D-Fender | <mathi> if two servers are registered to an SIP provider for incoming calls, which server will take the call ? depends on the provider |
| 18:56.30 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-148-077.pools.arcor-ip.net) |
| 18:59.47 | ChannelZ | OT: does German use quotation marks? |
| 19:02.50 | gusto | no idea |
| 19:04.34 | ChannelZ | looks like they do but they're lower and higher at the start/end and backwards. |
| 19:04.38 | ChannelZ | ugh |
| 19:04.40 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 19:16.40 | wdoekes | ChannelZ: what kind of quotation marks? |
| 19:18.46 | wdoekes | ah.. „Hast du den Artikel ‚EU-Erweiterung‘ gelesen?“ |
| 19:19.00 | wdoekes | that looks backwards indeed |
| 19:22.51 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:26.29 | kikohnl | jlmls: 7.4 firmware is ancient latest is 9.3.1SR1
I'm running 7941G-GE with 9.3.1SR1 with no voice quality issues |
| 19:28.00 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 19:29.34 | danfromuk | Hi, we've got a user inferface that allows clients to manage their dialplans themselves. A client recently misconfigured their dialplan and lost some calls because they pointed a DID to an extension which did not exist. Is there any way to catch all autofallthroughs and send a warning email to the client? |
| 19:30.16 | danfromuk | I was going to watch for calls where the status is FAILED but that doesnt help for calls that have been answered by an IVR for example |
| 19:34.18 | ChannelZ | wdoekes: yeah like that. For quoting someone saying something. |
| 19:36.47 | [TK]D-Fender | danfromuk, No |
| 19:38.05 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-148-077.pools.arcor-ip.net) |
| 19:38.41 | ChannelZ | Wouldn't an 'i' extension catch that? |
| 19:39.05 | danfromuk | Ok, so before every Dial(Local/) or Goto() command, is there a way to check whether extension exists? |
| 19:39.26 | danfromuk | GotoIf(extension exists,extension,elsewhere) |
| 19:39.26 | igcewieling | danfromuk: with recent versions of Asterisk see function DIALPLAN_EXISTS |
| 19:39.41 | igcewieling | ChannelZ: no. "i" only works for IVRs (i.e. background) |
| 19:40.08 | bparker | it depends on what danfromuk means by "pointed the DID" |
| 19:41.27 | danfromuk | All DIDs go to DID@sipserver.company.com. Then there is an initial line that says exten=>DID,1,Goto(Extension that the client wants) |
| 19:41.46 | danfromuk | So igcewieling's reply was perfect. |
| 19:41.56 | danfromuk | Thanks |
| 19:42.05 | igcewieling | no problem. |
| 19:42.28 | igcewieling | danfromuk: you should read through "core show applications" and "core show functions" You'll learn lots of cool stuff |
| 19:42.43 | bparker | 'i' should catch an invalid extension pointed to by Goto() |
| 19:42.50 | bparker | IIRC |
| 19:42.50 | *** join/#asterisk newtonr (~newtonr@64.34.219.47) |
| 19:42.51 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:43.22 | bparker | but maybe not |
| 19:43.28 | bparker | I'm not an expert |
| 19:46.26 | igcewieling | a quick search of the mailing list archives will likely provide the answer. |
| 20:06.47 | *** join/#asterisk techwerkz (~Justin@76.101.15.40) |
| 20:09.56 | saint_ | can someone explain to me something from the documentation : |
| 20:09.57 | saint_ | GoSub(subVoicemail,start,1(${JOHN},${EXTEN})) |
| 20:11.38 | *** join/#asterisk ircNewbz (~ircNewbz@unaffiliated/ircnewbz) |
| 20:11.44 | techwerkz | Recently upgraded to 11 and anywhere in my queues.conf that I use SIP/provider/2223332222 gets marked as invalid now in the queue status. If I use SIP/2223332222@provider it seems to work. Is this a change or a bug? DAHDI still seems to work as expected DAHDI/g3/2223332222 |
| 20:11.45 | saint_ | in here, ${EXTEN} is the number that we are calling ? or the called number ? How does the GoSub function read this ${ARG2} ? |
| 20:13.01 | navaismo | saint the ${EXTEN} is the number/string after 'exten =>' |
| 20:13.43 | navaismo | and in the sub its is passed as ARG2 because ARG1 is the variable JOHN |
| 20:14.08 | saint_ | navaismo: hhhaaaa, okay. thanks |
| 20:18.12 | igcewieling | navaismo: only if it isn't a pattern |
| 20:20.42 | navaismo | uh? |
| 20:21.29 | navaismo | exten => kaka ${EXTEN}=kaka exten=> _4XX ${EXTEN}= 4plus 2 digits |
| 20:21.41 | navaismo | I'm missing something? |
| 20:22.05 | navaismo | i don't understand you igcewieling |
| 20:22.26 | igcewieling | " saint the ${EXTEN} is the number/string after 'exten =>'" So if you have exten => _4XXX then ${EXTEN} will NOT be "_4XXX" or even "4XXX" |
| 20:22.46 | igcewieling | so what you said applies if you are not using a pattern match in your extension |
| 20:23.33 | navaismo | i still dont follow you. As I said exten=_4XX --->${EXTEN}= 4plus 2 digits |
| 20:23.40 | navaismo | between 0-9 |
| 20:23.44 | igcewieling | based on saint's previous questions it is an important thing to point out |
| 20:24.13 | navaismo | not sure if my english or you misunderstood the concept |
| 20:26.10 | navaismo | reading again all |
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| 20:26.14 | ChannelZ | given exten => 12345 ${EXTEN} is 12345. |
| 20:26.59 | ChannelZ | given exten => _4XXX then ${EXTEN} is 4XXX where XXX is whatever numbers they dialed that happened to match the extension. So if they dial 4123 it'd be that. |
| 20:27.01 | navaismo | ah got it igcewieling you are taking my statement literally, since we are in asterisk I suppose the people knows about dialpattern |
| 20:28.12 | danfromuk | Is it possible to get an output of all the commands that AMI is receiving? I think there is a problem with an ssh client i'm using because AMI is saying authentication failed. Other ssh clients work fine. |
| 20:28.24 | navaismo | so when i refer to the string after exten Im assuming people knows about pattern or not |
| 20:28.37 | navaismo | im not telling it will take 4XXX as value |
| 20:28.48 | navaismo | of course not |
| 20:29.43 | igcewieling | navaismo: if you had seen saint's previous 2 days of questions you would not suppose that. 8-| |
| 20:30.31 | navaismo | I believe that He already read the book |
| 20:31.24 | [TK]D-Fender | Telephony is not faith-based |
| 20:31.40 | ChannelZ | Sort of is. |
| 20:32.45 | ChannelZ | I hope things work when I pick up the phone but that isn't always the case, and I say "jesus christ" a lot |
| 20:33.00 | ChannelZ | and some other choice phrases |
| 20:36.19 | techwerkz | I am saying some nice phrases now |
| 20:42.33 | techwerkz | So anyone have any idea on my previous question? |
| 20:46.23 | igcewieling | We are experiencing issues with Polycom phones subscribed to a large number of extensions (for BLF). Have others seen this issue and if so did you ever manage to fix it? |
| 20:46.25 | ChannelZ | No idea. |
| 20:48.47 | navaismo | techwerkz, i dont know many phrases, is that your question? :P |
| 20:50.31 | techwerkz | All of a sudden after upgrading from 1.8 to 11 my queues that contained SIP/provider/number no longer work. queue show says the member is invalid. Changing to SIP/number@provider seems to fix it. |
| 20:50.54 | techwerkz | Just kind of weird for that to change |
| 20:50.58 | igcewieling | techwerkz: no mention of a change in the UPGRADE-XX.txt files? |
| 20:51.13 | techwerkz | Not that I saw |
| 20:51.39 | techwerkz | I did find a forum post of someone unanswered experiencing this on Asterisk 1.4 |
| 20:52.25 | techwerkz | dahdi still works as it always has in queues, not that I believe the other format would be valid |
| 20:52.54 | techwerkz | the dial command still works fine with SIP/provider/number as well |
| 20:54.03 | techwerkz | Not a huge deal as I did a quick regex fine/replace to update the conf ... more or less curious to figure this one out |
| 20:55.35 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 20:55.55 | gavimobile | is it good practice to not load modules which are not in use? |
| 20:55.55 | navaismo | well i never added a external number as a queue member |
| 20:56.20 | gavimobile | maybe as a way to save resources? |
| 20:57.07 | navaismo | in tinny devices yes, only compile and load the desired modules |
| 20:57.24 | ChannelZ | If there are a bunch of dialplan applications or functions you aren't using I suppose you'll scrape up a little bit of memory back. |
| 20:59.25 | navaismo | offtopic awesome http://www.sinologic.net/blog/wp-content/uploads/2013/01/dialup-final.jpg |
| 21:00.49 | techwerkz | navaismo: I do it as a quick one to always forward to a cell phone, and deal with multiple devices a user might have (home, office, vacation home, etc). My dialplan isn't tailored to use & as I have some other custom things that happen later downstream when a person is called. |
| 21:02.22 | navaismo | i see |
| 21:04.36 | igcewieling | gavimobile: I generally noload => chan_ modules and other modules I know for sure I'll never use and usually find out I'm wrong and end up using one of the modules I noload' |
| 21:04.42 | igcewieling | d usually after hours of troubleshooting |
| 21:06.09 | gavimobile | igcewieling: I just disabled the ones which the logs are always complaing about on asterisk restart |
| 21:06.20 | gavimobile | most of them seemed to be realted to cdr |
| 21:06.36 | gavimobile | I hope ill be alright but thanks for the advice igcewieling |
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| 22:00.04 | danfromuk | Is there an alison sound that says "please enter your mailbox number" ? |
| 22:06.22 | igcewieling | danfromuk: Usually in the asterisk sounds directory there are .txt files which list what all the sound files say |
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| 22:10.22 | DoSJustin | set |
| 22:11.25 | danfromuk | igcewieling: i didnt realise. thanks |
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