| 00:18.44 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 00:20.13 | *** join/#asterisk doppo (~doppo@107.191.100.185) |
| 00:40.44 | *** join/#asterisk jzaw (~jzaw@community-wifi.dzki.co.uk) |
| 00:48.07 | *** join/#asterisk Hexxis (~Hexxis@107.150.31.94) |
| 01:00.18 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:11.36 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 01:21.59 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 01:22.23 | *** join/#asterisk AlHafoudh (~AlHafoudh@echo.freevision.sk) |
| 01:45.00 | *** join/#asterisk erick2014 (~textual@cpe-174-096-091-109.carolina.res.rr.com) |
| 01:47.43 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 02:50.16 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 02:57.22 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:27.59 | fling | Are you guys using medooze mcu? |
| 03:29.36 | *** join/#asterisk burnbrighter (~kurt@50-79-211-169-static.hfc.comcastbusiness.net) |
| 03:30.01 | ChannelZ | I drink a lot, if that's what you meant. |
| 03:30.33 | Penguin | I don't drink any more. |
| 03:31.19 | ChannelZ | STRAIGHT EDGE! |
| 03:31.32 | Penguin | Well, no... |
| 03:31.36 | Penguin | I don't drink any less, either. |
| 03:32.30 | ChannelZ | Actually I don't drink, I just didn't understand the question and it sounded boozy. Cuz there was a "z" in it. |
| 03:32.45 | ChannelZ | But I GOOgled it |
| 03:40.43 | fling | Asterisk works beautifully but I'm getting issues configuring medooze :P |
| 03:40.49 | fling | needs some booze |
| 03:41.05 | fling | ÇzoopÇɯ |
| 03:47.28 | ChannelZ | or some NoDoze |
| 04:13.04 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 04:16.49 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 04:22.59 | *** join/#asterisk cyborg-one (~cyborg-on@85.238.103.28) |
| 04:23.14 | *** join/#asterisk cyborg-one (~cyborg-on@85.238.103.29) |
| 04:32.31 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 04:47.31 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 04:55.46 | *** join/#asterisk notzopsi (~zopsi@zopsi.com) |
| 04:56.11 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 04:58.01 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 05:06.30 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 05:17.53 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-56-12.rn.hr.cox.net) |
| 05:36.10 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 06:00.29 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 06:21.23 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:21.47 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 06:42.34 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 06:45.56 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 06:49.17 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 06:53.03 | *** join/#asterisk ruben23 (~OpenDial0@112.198.82.49) |
| 06:55.20 | *** join/#asterisk CeBe (~CeBe@port-92-206-123-83.dynamic.qsc.de) |
| 06:58.33 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 07:00.27 | ruben23 | guys i have an issue with my asterisk- all my calls like 10-20 calls are on asignle file of recording, how do i separate this per calls..? |
| 07:00.31 | ruben23 | any idea guys..? |
| 07:04.30 | ChannelZ | Are you not StopMonitor()ing anywhere? Or is your dialplan such that people can make multiple calls without hanging up? |
| 07:08.15 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 07:09.39 | *** part/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 07:23.25 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 07:27.58 | ruben23 | the setup is this - i have 2 asterisk server - asterisk1 - is where the call are executed by they login first on a confernce before call will be stablished then it wil pass on to asterisk 2 where the calls wil be sent out and be recorded.. |
| 07:28.37 | ruben23 | im thinking might be asterisk2 is detecting the conference and record all calls as 1 file...that might be..? |
| 07:32.30 | *** join/#asterisk jhlavacek (~jirka@jix.nextradsl.cz) |
| 07:32.43 | ChannelZ | So a person dials in to asterisk1 and gets placed in a conference call, and then by some other mechanism other calls are placed on asterisk2, the other end of which is connected to the conference on asterisk1? |
| 07:37.44 | *** join/#asterisk karamell (~maxi@46.45.3.62) |
| 07:37.54 | *** join/#asterisk pryshebliad (~aab@2a00:b1c0:0:3:e2cb:4eff:feb5:132b) |
| 07:38.36 | pryshebliad | Hello! Why you don't make packages for CentOS 7? |
| 07:39.03 | ChannelZ | Because that's usually the distro's job |
| 07:39.23 | ChannelZ | s/distro's/distros maintainer's/ |
| 07:42.51 | pryshebliad | ChannelZ: Why it makes for CentOS 6? |
| 07:45.40 | ruben23 | ChannelZ: asterisk1 basically stablished conference with the caller only in the room then dials a number which is pass to asterisk2 and be recorded and to be dialout successfully |
| 07:46.12 | ChannelZ | pryshebliad: that's all they feel like doing, I guess. |
| 07:56.03 | phix | hey gang |
| 07:56.47 | *** join/#asterisk u0m3_ (~u0m3@92.80.112.149) |
| 07:58.06 | pryshebliad | phix: Hey |
| 07:58.32 | pryshebliad | phix: Make us asterisk for Debian! |
| 07:58.41 | ruben23 | ChannelZ: ..? |
| 07:58.54 | pryshebliad | How are you? |
| 07:59.54 | phix | pryshebliad: There is asterisk for debian |
| 08:00.17 | pryshebliad | phix: I can't install Asterisk 13 on Debian 7 |
| 08:00.36 | phix | why not? |
| 08:00.45 | phix | and what's wrong with version 11? |
| 08:01.19 | pryshebliad | phix: There is no pjsip |
| 08:01.36 | phix | what's that? |
| 08:11.35 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:12.26 | wdoekes | pryshebliad: ask tzafrir ;) |
| 08:12.48 | wdoekes | morning |
| 08:13.17 | tzafrir | hi |
| 08:14.21 | *** join/#asterisk Geek-Linux (~GeekLinux@115-186-130-251.nayatel.pk) |
| 08:16.26 | Geek-Linux | Hello All, I am have an issue in confbridge. it has a wierd behaviour in conference some times i callers can hear the voice and sometimes not. |
| 08:19.14 | Geek-Linux | what could be the possible reasons. i am connected with the telco over sip. |
| 08:26.05 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:ccca:a225:254f:2a34) |
| 08:33.06 | babak | Hi, I have an application that calls duration is 15 sec, I want to know what caps is reasonable I assume in my dimentioning for example with 8E1=240/15=16 caps and with 16E1=480/15=32caps can be achioved by one ordinary server ? |
| 08:49.29 | Geek-Linux | Hello All, I am having an issue in confbridge. it has a weird behavior in conference. some times callers can hear the voice and sometimes not, but mainly not |
| 08:49.40 | Geek-Linux | <PROTECTED> |
| 08:52.25 | Geek-Linux | i am using codec g729 and asterisk version is 11.12.1 |
| 08:52.46 | Rac-on | Geek-Linux: asking your question twice does not result in a faster answer |
| 08:53.34 | pryshebliad | tzafrir: Hey! How about Asterisk 13 packages for CentOS 7 and Debian 7? |
| 08:54.06 | tzafrir | No idea about centos. |
| 08:54.09 | Geek-Linux | Rac-on: i thought i am missing some information, thats why reasked it with the codec and asterisk version. |
| 08:54.34 | tzafrir | I first need to get Jessie properly out the door. Then I'll see about backporting |
| 08:55.28 | *** join/#asterisk wanna (~wanna@194.183.244.5) |
| 08:56.12 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 08:56.31 | EmleyMoor | Is there a channel variable that can show me the IP address a SIP call is coming from? |
| 08:58.21 | *** join/#asterisk theahindle (~ahindle@li758-180.members.linode.com) |
| 08:58.22 | EmleyMoor | Ah, not urgent - I can tell IPv6 is working. |
| 08:58.35 | theahindle | Hi - do you think I'm best using 'repro' or iptables to forward SIP to another IP? |
| 08:58.45 | EmleyMoor | Is it possible to have Asterisk listen only to specific IPv6 addresses yet? (11) |
| 08:58.51 | theahindle | Linux server has a public IP, asterisk server doesn't, so I need to route everything from the public IP to the internal IP, and back, without issue :) |
| 08:59.17 | theahindle | At the minute I'm using repro but getting ''no reply to our critical packet '' |
| 09:14.50 | *** join/#asterisk areski (~areski@188.189.4.95) |
| 09:27.52 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:48.32 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 09:58.56 | *** join/#asterisk areski (~areski@188.189.4.95) |
| 10:03.12 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 10:04.15 | theahindle | I followed this http://www.voip-info.org/wiki/view/port+forwarding but my asterisk is still saying 'no reply to our ctritical packet' |
| 10:04.50 | theahindle | Though asterisk looks like it's trying to send the packet to the server forwarding the traffic to it, not sur eif it's supposed to, or whether it should send it directly |
| 10:05.55 | *** join/#asterisk jhlavacek (~jirka@195.70.143.8.adsl.nextra.cz) |
| 10:06.25 | theahindle | Now I'm not sure what to do |
| 10:06.43 | theahindle | I can create calls, but they get hung up on after 28 seconds :( |
| 10:10.31 | theahindle | Is there a company/person I can pay to sort this? |
| 10:12.27 | wdoekes | theahindle: ~30 seconds means that no ACK is received |
| 10:12.49 | wdoekes | (usually) |
| 10:13.18 | wdoekes | Alice sends INVITE, Bob sends 200, Alice sends ACK to the "Contact:" header in Bob's 200 |
| 10:13.51 | wdoekes | if the Contact header contains some RFC1918 IP, this goes sour |
| 10:13.55 | theahindle | wdoekes: Yea there is a timeout somewhere of 32000ms - I think it's a routing/networking thing as I'm using another linux PC to forward traffic on ports to the asterisk box |
| 10:14.14 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 10:14.53 | theahindle | I get this a lot: Retransmitting #9 (NAT) to 10.3.146.10:5060: |
| 10:15.16 | theahindle | But the 10.xx IP is internal, shouldn't it be sending to the client that initiated the call directly? |
| 10:15.22 | theahindle | Or should 10.3.146.10 route it to the user? |
| 10:15.36 | wdoekes | yes. tell Alice to ignore the Contact-header and reply to the received IP instead |
| 10:15.47 | wdoekes | see the nat= options |
| 10:16.10 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 10:16.48 | theahindle | Alice is stubborn though |
| 10:16.51 | theahindle | And also Alice is Twilio |
| 10:18.11 | theahindle | I'm using Twilios REST API to initiate a SIP call/INVITE to my linux server, which is forwarding 9060 and 10000-20000 to the asterisk box |
| 10:18.28 | wdoekes | if you can't change Alice, you must alter your own packets. see externip= |
| 10:18.47 | wdoekes | (externhost=_) |
| 10:19.01 | wdoekes | (or externaddr=) |
| 10:19.14 | theahindle | wdoekes: Thanks - I have externaladdr set: externaddr=41.160.211.254:9060 |
| 10:19.36 | wdoekes | which messages are you retransmitting then? |
| 10:20.47 | theahindle | http://pastebin.com/raw.php?i=vdAKZ3G3 |
| 10:22.54 | wdoekes | ok, so it's the 200 that doesn't even go out right. I hope you know what runs on 10.3.146.10 |
| 10:23.30 | theahindle | That's my server, forwarding ports 9060/10000-2000 to 10.3.146.135 |
| 10:23.34 | theahindle | 10000-20000, even |
| 10:23.54 | wdoekes | 135? or 35? |
| 10:24.19 | theahindle | 35 sorry |
| 10:24.34 | theahindle | http://pastebin.com/raw.php?i=cT5LcPRx - SIP_SRV_LAN_IP=10.3.146.35 |
| 10:26.12 | wdoekes | I don't know what your server did, but for DNAT, it shouldn't rewrite the source IP, which it apparently does |
| 10:26.17 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:27.19 | wdoekes | the received=IP should be 107.21.231.147 |
| 10:27.35 | theahindle | Ok |
| 10:27.51 | theahindle | Maybe I have some old iptables rule that is rewriting it or something |
| 10:27.54 | theahindle | Though I do iptables --flush |
| 10:30.27 | wdoekes | you'll want to do some tcpdumping on your server to see what's going on |
| 10:31.05 | theahindle | Thanks for your help wdoekes |
| 10:31.08 | theahindle | This will be a fun few hours :) |
| 10:34.16 | *** join/#asterisk jhlavacek (~jirka@jix.nextadsl.cz) |
| 10:35.30 | theahindle | iptables-save shows all sorts of old rules, and MASQUERADE setting which I presume is breaking it |
| 10:36.08 | theahindle | I thought --flush would get rid of them though :/ |
| 10:36.43 | wdoekes | iptables -t nat -F |
| 10:37.40 | wdoekes | possibly even -t mangle -F |
| 10:37.41 | theahindle | Masquerading is gone |
| 10:37.42 | theahindle | to test again! |
| 10:39.48 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 10:44.36 | theahindle | I removed the forwarding of port 80 and the masquerading and my call has gone on for 1:33 now! |
| 10:44.39 | theahindle | wdoekes: I could kiss you |
| 10:46.18 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 10:46.48 | *** join/#asterisk areski (~areski@188.189.4.95) |
| 10:47.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:48.11 | wasanzy | I want to create a call file and when I initiate a conference call, it keep playing a specified audio for anyone who joins the conference by dialing a code, will just listen to when the music got to before they join. |
| 10:48.19 | wasanzy | I need help with the call file first |
| 10:48.31 | wasanzy | what do I need to create it> |
| 10:48.32 | wasanzy | ? |
| 10:51.00 | theahindle | Can I send you money wdoekes? PayPal? |
| 10:51.55 | wdoekes | sure. glad that it works :) |
| 10:52.09 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 10:52.27 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 10:54.37 | *** join/#asterisk cjk (~cjk@vodsl-10825.vo.lu) |
| 10:55.05 | cjk | hi, it seems that asterisk has a limi for the maxmimum length of a hint or variable or extension line |
| 10:55.11 | cjk | does anyone know where this could be changed? |
| 10:56.12 | *** part/#asterisk theahindle (~ahindle@li758-180.members.linode.com) |
| 10:58.09 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:59.46 | cjk | it seems that the maxmimum variable size is set to 90 |
| 11:03.31 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:04.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:09.34 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 11:44.26 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 11:55.44 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 11:56.03 | *** join/#asterisk CeBe (~CeBe@port-92-206-123-83.dynamic.qsc.de) |
| 11:59.49 | marceloamorim | guys, I don`t know, last week I thought my problem was the transfer |
| 12:00.49 | marceloamorim | but I figured out that is the music, if I put the call on hold on my asterisk, the other asterisk the call came from, that call is dropped |
| 12:01.08 | marceloamorim | was happend every transfer, but now the call dropped on hold too |
| 12:02.41 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 12:05.01 | *** join/#asterisk areski (~areski@188.189.40.26) |
| 12:14.50 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:14.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:22.32 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1303:2acf:e9ff:fe1c:8fb) |
| 12:23.15 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 12:29.31 | wasanzy | I want to create a call file and when I initiate a conference call, it keep playing a specified audio for anyone who joins the conference by dialing a code, will just listen to when the music got to before they join. |
| 12:33.16 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 12:36.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:38.59 | *** join/#asterisk MadHatter42 (~tuwid@79.171.53.122) |
| 12:44.21 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 12:51.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 12:55.55 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1303:2acf:e9ff:fe1c:8fb) |
| 13:00.14 | marceloamorim | other peculiarity is the version on asterisk in the side dropped is 11.6-cert4 and the side that I`m transfer the call is 1.8.17.0 |
| 13:00.47 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:03.01 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 13:06.04 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 13:07.19 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 13:17.21 | *** join/#asterisk areski (~areski@188.189.40.26) |
| 13:20.57 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 13:22.39 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:22.48 | *** join/#asterisk iulhk (~iulhk@182.189.38.180) |
| 13:24.41 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 13:27.30 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:28.03 | WIMPy | marceloamorim: Does your music contain parts that get interpreted as DTMF and trigger features on the other side? |
| 13:28.54 | WIMPy | wasanzy: I don't understand your question. |
| 13:29.27 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 13:29.37 | iulhk | using asterisk-11.11.0 i hv sip account 1000 , when i register sip account 1000 at the console i can see "Registered SIP '1000' at 182.189.38.180:35164" in (db=asterisk) (table=sippeers.port:35164) and when i un-register account i can see at the console " Unregistered SIP '1000'" , but in databse port value still 35164 although it should change with 0.. any idea please ? |
| 13:29.45 | marceloamorim | well WIMPy, I use the default MOH on the asterisk |
| 13:41.47 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:44.42 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 13:52.23 | *** join/#asterisk ruben23 (~OpenDial0@112.198.77.98) |
| 13:53.54 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 13:58.13 | *** join/#asterisk MadHatter42 (~tuwid@109.69.2.62) |
| 14:03.41 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 14:04.10 | *** join/#asterisk areski (~areski@188.188.46.161) |
| 14:10.13 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:17.12 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:19.43 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-ltsbpytcxynrhyok) |
| 14:19.44 | *** join/#asterisk ipengineer (~zconkle@12.144.114.2) |
| 14:24.52 | ipengineer | I am having an issue in Asterisk 13 where asterisk does not consistently send the 180 Ringing back. It is very sproratic.. Sometimes it will send it other it wont.. https://gist.github.com/zconkle/7e483b66de2c92839439 |
| 14:24.59 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:25.49 | [TK]D-Fender | ipengineer: Show actual * CLI SIP debug for the full call |
| 14:26.05 | [TK]D-Fender | ipengineer: Doesn't doesn't send ringing back "just because" |
| 14:27.34 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pwptoibturowzccn) |
| 14:27.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:29.10 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-jewxozhosbetjhfc) |
| 14:31.05 | *** join/#asterisk nighty^_ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:34.52 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:34.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:36.26 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 14:38.26 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-ogimtcjogiadeizb) |
| 14:39.40 | ipengineer | [TK]D-Fender: https://gist.github.com/zconkle/ba2d0b8f89efe5c5bb43 I think it may have something to do with the Dial cmd and it not returning ringing. Instead it says âconnected line has changed. Saving it until answerâ. I think I usually see ringing there |
| 14:41.05 | [TK]D-Fender | ipengineer: You're missing relevant debug |
| 14:41.19 | [TK]D-Fender | Asterisk PASSES ringing. |
| 14:41.24 | [TK]D-Fender | It has to get it from the source |
| 14:41.28 | [TK]D-Fender | It doesn't jsut add it |
| 14:41.40 | [TK]D-Fender | And you aren't showing proof that it's coming in. |
| 14:42.57 | ipengineer | [TK]D-Fender: So what all do you need? I am setting sip debug on. What else? |
| 14:43.15 | [TK]D-Fender | Where do we see "ringing" coming in in the first palce? |
| 14:43.21 | [TK]D-Fender | place* |
| 14:44.20 | ipengineer | I dont see it anywhere.. That is the problem. If I keep calling this it will go back to ringing |
| 14:44.53 | [TK]D-Fender | * only passes what it gets. |
| 14:45.27 | [TK]D-Fender | "A" calls in. * dials "B". if "B" doesn't report back "ringing" then * will not send "A" "ringing". |
| 14:45.38 | [TK]D-Fender | SOunds like "B" is sporadic |
| 14:47.18 | ipengineer | I gotcha.. Ok let me check that side |
| 14:50.47 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 14:50.50 | wasanzy | hello |
| 14:51.23 | *** join/#asterisk areski (~areski@188.189.7.14) |
| 14:52.32 | wasanzy | I have this configuration: http://pastebin.com/T4erPdsU and looking at the call flow: http://pastebin.com/cf1353m4 I can't get the audio played in the conference. this line doesn't seem to be executing: exten => 200,1,Originate(Local/inject@myconference/n,exten,myconference,quiet_join,1) |
| 14:52.45 | wasanzy | anyone to help me? |
| 14:55.27 | [TK]D-Fender | Why should it? |
| 14:55.51 | [TK]D-Fender | Nothing in that output shows you dialing 200 |
| 14:59.10 | wasanzy | when I dial 200 I get : http://pastebin.com/e9hR8wb8 |
| 14:59.33 | wasanzy | I don't know No channel type registered for 'Local it is talking about |
| 14:59.38 | zaf | anyone in here have any insight on the asterisk yum repos? |
| 14:59.40 | ipengineer | [TK]D-Fender: FYI restarting asterisk fixed that ringing problem. A huge part of me thinks this is a deadlocking issue or something because it will work for days on end and then start that. A restart fixes it |
| 14:59.43 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:00.21 | [TK]D-Fender | ipengineer: Your description show us nothing really... |
| 15:00.53 | [TK]D-Fender | wasanzy: chan_local is clearly not loaded |
| 15:01.14 | ipengineer | [TK]D-Fender: I never implied that it did. I was stating what little information I know right now⦠Which isnt much |
| 15:01.39 | [TK]D-Fender | ipengineer: We end with what we began with. At least there is symmetry... |
| 15:03.35 | cjk | hello, it seems that the maximum variable size is set to 90. any idea how to change this? |
| 15:04.38 | wasanzy | is there a module called Bridge? |
| 15:05.46 | [TK]D-Fender | wasanzy: Do you see one? I see one for ConfBridge.... |
| 15:07.03 | wasanzy | I get this error that is why am asking: channel.c:5956 ast_request: No channel type registered for 'Bridge' |
| 15:07.30 | *** part/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 15:07.34 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 15:07.48 | wasanzy | [TK]D-Fender: you have any idea? |
| 15:10.34 | [TK]D-Fender | wasanzy: Have you looked in your modules folder? |
| 15:11.42 | wasanzy | There is bridge_multiplexed, bridge_simple, bridge_multiplexed and bridge_builtin_features and also ConfBridge. they are all loaded |
| 15:12.32 | [TK]D-Fender | show us the folder |
| 15:15.52 | ChannelZ | or what you're actually trying to do that causes this |
| 15:16.09 | wasanzy | http://pastebin.com/FkCmVkxm |
| 15:16.31 | [TK]D-Fender | chan_bridge.so <------- |
| 15:16.47 | pryshebliad | Why the packages for CentOS 7 isn't ready? |
| 15:17.51 | ChannelZ | Ran out of sugar. |
| 15:18.17 | ChannelZ | Just build from the source. It's not that hard. |
| 15:24.15 | wasanzy | I dial I don't really understand what is going on here; http://pastebin.com/3aFfH3Gr when I dial 200 |
| 15:24.31 | wasanzy | I guess it has something to do with slin192 |
| 15:27.58 | mjordan | what is the sample rate on your confbridge? |
| 15:29.00 | *** join/#asterisk alami (~ialami@unaffiliated/alami) |
| 15:29.14 | wasanzy | mjordan: I don't get you please |
| 15:31.12 | alami | when i issue dialplan remove extension 111@context the extension is removed, but after diaplan reload, it's show that again, is there a better way to do that ..? |
| 15:31.32 | alami | or do i have to delete it from extension.conf=? |
| 15:33.59 | [TK]D-Fender | alamior do i have to delete it from extension.conf=? <- correct |
| 15:34.44 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1303:2acf:e9ff:fe1c:8fb) |
| 15:35.28 | *** join/#asterisk ipengineer (~zconkle@12.144.114.2) |
| 15:35.54 | mjordan | wasanzy: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge#Asterisk13Configuration_app_confbridge-bridge_profile_internal_sample_rate |
| 15:35.59 | alami | [TK]D-Fender: what's the use case of diaplan remove exten..? |
| 15:36.21 | [TK]D-Fender | alami: Don't want to reload everything but need to make a quick run-time fix |
| 15:39.26 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 15:40.50 | Penguin | Why couldn't you remove that extension with dialplan remove ... and then dialplan save to save it? |
| 15:41.53 | [TK]D-Fender | Does that actually still work at all? |
| 15:41.57 | WIMPy | Have you ever tried 'dialplan save'? |
| 15:42.02 | Penguin | I think you must not have writeprotect=yes in extensions.conf for that to work, though. |
| 15:42.04 | [TK]D-Fender | IIR there was a "writeprotect" var you could set for that |
| 15:42.18 | WIMPy | That exists. |
| 15:42.19 | [TK]D-Fender | Ancient crap |
| 15:42.37 | [TK]D-Fender | First time I can recall it even being mentioned in here |
| 15:42.38 | WIMPy | But I wouldn't expect the saved file to be the same that was loaded. |
| 15:42.59 | WIMPy | Like with iptables-save. It may work fro some, but not for others. |
| 15:43.02 | Penguin | Usage: dialplan save [/path/to/extension/file] |
| 15:43.03 | Penguin | Example: dialplan save (/etc/asterisk/extensions.conf) dialplan save /home/markster (/home/markster/extensions.conf) |
| 15:43.56 | Penguin | 'dialplan save' will save to the default; 'dialplan save /other/file' will save it to another place. |
| 15:44.30 | WIMPy | just did it. Now let's see what it created... |
| 15:44.32 | alami | Penguin:Command 'dialplan save ' failed. |
| 15:44.54 | [TK]D-Fender | Wonder how much it'll massacre your orginal file if you tried it |
| 15:44.55 | *** join/#asterisk Zetten (~pvzetten@p5099910b.dip0.t-ipconnect.de) |
| 15:44.59 | WIMPy | It expects a path, not a filename. |
| 15:45.17 | Penguin | [tk]d-fender: I'm about to perform a definitive test on that in two minutes. |
| 15:45.18 | WIMPy | It's sorted/ |
| 15:45.23 | [TK]D-Fender | stripping comments, "exec"s' tha merged confgs on-load, etc |
| 15:45.26 | WIMPy | . |
| 15:45.26 | [TK]D-Fender | Sounds liek suicide |
| 15:45.42 | WIMPy | Definitely |
| 15:46.05 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-urkkjiakoqjtsawj) |
| 15:46.12 | [TK]D-Fender | alami: Why are you actually even considering this? |
| 15:46.33 | WIMPy | Maybe you can use it like google translate. Use 'dialplan save' then load it again, save again and see what changed :-) |
| 15:46.54 | Zetten | I'm having a bit of trouble with Cisco 7961 phones and the pjsip channel - they don't seem to ever try to authenticate even after getting the Unauthorized reponse with the realm info |
| 15:47.23 | file | Zetten, set force_rport=no |
| 15:47.37 | alami | [TK]D-Fender: i want to make script to delete unsed ext automaticly, now i check that i can't diaplan save /etc/asterisk/extencion.conf |
| 15:48.08 | Zetten | file: in the endpoint or globally? |
| 15:48.14 | Penguin | dialplan save /var/lib/asterisk/ |
| 15:48.14 | Penguin | Dialplan successfully saved into '/var/lib/asterisk//extensions.conf' |
| 15:48.21 | file | Zetten, you can't set things globally - has to be endpoint ^_^ |
| 15:48.22 | Penguin | Now to check them. |
| 15:48.28 | Zetten | fair enough :) |
| 15:49.35 | alami | Penguin: yes i can use /var/lib, and then include it in /etc/asterisk/extension.conf |
| 15:49.57 | [TK]D-Fender | alami: Did you set that value as required? |
| 15:50.28 | wasanzy | [TK]D-Fender: is this line valid in the first place: exten => 200,1,Originate(Local/inject@myconference/n,exten,myconference,quiet_join,1) |
| 15:50.30 | Penguin | This is very interesting. |
| 15:50.38 | alami | [TK]D-Fender:wich value? |
| 15:50.48 | [TK]D-Fender | alami: Both Penguin and I told you... |
| 15:50.50 | [TK]D-Fender | look up |
| 15:51.12 | [TK]D-Fender | wasanzy: Why wouldn't it be? |
| 15:51.16 | Penguin | It does change the order of contexts and extensions within them. |
| 15:51.38 | wasanzy | [TK]D-Fender: because of the errors I pasted earlier |
| 15:51.43 | Penguin | It does not copy any comments. |
| 15:51.44 | [TK]D-Fender | Penguin: functionally shouldn't matter though as long as includes are in proper order. |
| 15:52.06 | WIMPy | The includes are ... included. |
| 15:52.07 | [TK]D-Fender | Penguin: But everything that is parsed on load like "exec" would get trashed |
| 15:52.24 | Penguin | And it includes all #include -ed files in the main conf. |
| 15:52.30 | WIMPy | But the order is very random. |
| 15:52.37 | [TK]D-Fender | alami: Correction, I now see ***ONE*** use for this. |
| 15:52.44 | alami | Penguin: sorry i have miss everything |
| 15:52.57 | [TK]D-Fender | alami: You accidentally DELETED your dialplan file and this is ALL you have left to restore it. |
| 15:53.03 | Penguin | So dialplan save is not for everyone. |
| 15:53.33 | Penguin | Yeah, that would be a good use for it. You deleted extensions.conf, but still have a dialplan loaded... you could save the running config to a file. |
| 15:53.59 | WIMPy | Yes |
| 15:54.02 | Penguin | It's not ideal, but it would be better than nothing. |
| 15:54.11 | [TK]D-Fender | If only barely |
| 15:54.16 | alami | [TK]D-Fender:lol |
| 15:54.20 | alami | okay :-) |
| 15:54.30 | WIMPy | Pretty hard to read, but looking at it, I do have hopes it might work as before. |
| 15:54.31 | Zetten | file: I have no idea how that worked, but it did! thanks very much! |
| 15:54.56 | [TK]D-Fender | WIMPy: Well it should save as parsed.... |
| 15:55.30 | WIMPy | Yes. But I already know "should". |
| 15:55.56 | alami | Penguin: so dialplan save /etc/asterisk/extension.conf don't work with you? |
| 15:56.04 | Penguin | I don't recommend it. |
| 15:56.30 | WIMPy | I already mentioned iptables. I don't know the current state, but that didn't really work for a long time. |
| 15:56.31 | Penguin | I won't test it. |
| 15:58.21 | WIMPy | Ok. I will do it. |
| 15:58.31 | alami | okay, i will look for other way to do it |
| 15:59.42 | WIMPy | Ok, that gave two warnings about missing parenthesis. |
| 16:00.27 | Penguin | Who/What are you talking to? |
| 16:00.59 | WIMPy | Maybe to you. |
| 16:01.09 | alami | is there a way to jump this error I can't save dialplan now, (just for test)? |
| 16:01.33 | WIMPy | I can't compare the versions. They are in a totally different order. |
| 16:01.38 | Penguin | You have to write it to a path that asterisk has permission to write. |
| 16:02.03 | Penguin | dialplan save /var/lib/asterisk |
| 16:02.04 | Penguin | for example. |
| 16:03.53 | WIMPy | 'dialplan save' fails if you have an escaped ; in an argument. |
| 16:04.05 | alami | Penguin: is that a good idea: create a /var/lib/asterisk/costum.conf and then #include it into /etc/asterisk/extensions.conf? |
| 16:04.14 | Penguin | I wouldn't. |
| 16:04.42 | Penguin | But it would be /var/lib/asterisk/extensions.conf |
| 16:04.47 | Penguin | not /var/lib/asterisk/costum.conf |
| 16:06.11 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 16:10.17 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 16:10.50 | alami | Penguin: yes name.. i would try it :-) because it's the only easy way for me now :-) |
| 16:11.12 | ChannelZ | Did you ever figure out your traffic control WIMPy? |
| 16:11.19 | Penguin | (1051.17) <Penguin> It does change the order of contexts and extensions within them. |
| 16:11.22 | Penguin | (1051.43) <Penguin> It does not copy any comments. |
| 16:11.36 | Penguin | And it might do other bad things, too. |
| 16:11.42 | Penguin | So you've been warned. |
| 16:12.41 | WIMPy | ChannelZ: No. I tried ##networking, ##linux and #Netfilter, but so far I'm stuck. |
| 16:12.56 | Penguin | Have you rebooted? |
| 16:13.05 | WIMPy | None of the readmes/howtos mention any modules I don't have. |
| 16:13.35 | alami | Penguin: that's true but i use the include funtion, and work witout Problem |
| 16:14.20 | alami | Penguin:and Order is not nessesery in my project :-) |
| 16:15.47 | ChannelZ | Hrmm |
| 16:16.10 | WIMPy | ChannelZ: But at least it's an issue where I still have hope that I might have broken it myself. |
| 16:16.18 | Penguin | Have you rebooted? |
| 16:16.38 | WIMPy | Am I using Windows? |
| 16:16.55 | WIMPy | But the box does only have an uptime of 7 days. |
| 16:17.22 | Penguin | works, stops working, reboot, works again |
| 16:17.32 | Penguin | I'm asking if you rebooted it AFTER it stopped working. |
| 16:18.13 | Penguin | I know it was booted at some point, so we don't need to talk about that time. I only care if it was rebooted after it broke. |
| 16:18.14 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:18.34 | WIMPy | Unfortunatly I have no idea since when it has been broken. I usually don't have much traffic any more so it's likely gone unnoticed for a while. |
| 16:19.57 | Penguin | And the problem is that tc doesn't do anything at all? |
| 16:20.24 | WIMPy | No, classification works, it just ignores the total bandwidth limit. |
| 16:21.03 | WIMPy | The stats show both byte and paket counters increasing, but the rates are always 0. |
| 16:21.14 | WIMPy | ... which is obviousely wrong and likely related. |
| 16:22.04 | Penguin | My rate fluctuates, so is it possible you are looking at it when it's legitimately 0? |
| 16:22.22 | WIMPy | No. |
| 16:22.29 | WIMPy | As I said: The counters are increasing. |
| 16:22.46 | Penguin | root shaper 5080132426 0 4455316 0 |
| 16:22.53 | Penguin | 10 fair-queue 348859662 0 0 0 0 |
| 16:23.03 | Penguin | Both 0 for rate. |
| 16:23.13 | Penguin | and 10 is for RTP. |
| 16:23.27 | Penguin | If I start a call, it should increase. |
| 16:23.39 | Penguin | And I would expect it to go back to 0 when I hang up. |
| 16:23.45 | Penguin | tests that theory. |
| 16:24.50 | WIMPy | Yes, that's what I'd expect. |
| 16:25.46 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 16:26.23 | Penguin | 10 fair-queue 348968802 29048 0 0 0 |
| 16:26.33 | Penguin | RTP running, rate is non-zero. |
| 16:30.35 | *** join/#asterisk teknoprep (~Tek@23.31.254.90) |
| 16:30.37 | teknoprep | hi all |
| 16:30.55 | teknoprep | does anyone here know of either a web based application or a desktop application that can be used to monitor SIP Presence |
| 16:31.15 | teknoprep | we use polycom phones and prefer doing this with SIP and not the asterisk manager |
| 16:31.38 | Penguin | BLF? |
| 16:32.06 | teknoprep | yes .. Subscribe |
| 16:32.27 | Penguin | Polycom phones and asterisk already support BLF. You just have to configure the phones' buddy subscriptions and asterisk hints. |
| 16:32.34 | teknoprep | thanks for that |
| 16:32.46 | teknoprep | i need more presence than what the phonen can display on the current unites |
| 16:32.54 | teknoprep | they would like a desktop or web based application to monitor this |
| 16:33.03 | teknoprep | that's why i asked about desktop or web based application for this |
| 16:33.03 | Penguin | Need more sidecars. |
| 16:33.07 | teknoprep | no |
| 16:33.11 | teknoprep | desktop or web based application |
| 16:33.15 | teknoprep | its that simple |
| 16:33.19 | WIMPy | A web based one should be easy to do. |
| 16:33.43 | paulc | How about OpenFire - set up an XMPP server and use some IM clients on the desktop, then get the Asterisk integration plug in to make buddies show as busy when they're on the phone - would it be enough, or you need more granular? |
| 16:33.43 | teknoprep | any that you know of off hand |
| 16:33.50 | teknoprep | already have openfire with XMPP |
| 16:33.59 | teknoprep | the XMPP integration is not working as intended yet |
| 16:34.03 | teknoprep | still want SIP Subscribe |
| 16:34.51 | cjk | teknoprep, i understand what you want to do, drop me a line when you have a solution, all other solutions like xmpp, astmanager are not reliable or stable |
| 16:35.11 | paulc | So you're ideally looking for a softphone with SIP subscribe, or an IM client that can subscribe in a similar fashion? |
| 16:35.25 | cjk | a softphone without the voice functionallity |
| 16:35.31 | teknoprep | yes.. i was actually thinking of using counterpath and turning off the ringer |
| 16:36.09 | paulc | What are the issues you're having with Openfire? |
| 16:36.37 | teknoprep | http://icanblink.com/ |
| 16:36.40 | teknoprep | this might work |
| 16:36.58 | *** join/#asterisk riess82 (~riessma@194-166-84-59.adsl.highway.telekom.at) |
| 16:37.02 | cjk | paulc, one software to maintain, to install to patch, to keep up2date, to learn and so on |
| 16:38.03 | cjk | teknoprep, do you have some budget for this? |
| 16:38.05 | teknoprep | paulc i don't use asterisk |
| 16:38.14 | teknoprep | except as an application for call queues in small offices |
| 16:38.21 | teknoprep | i use SipXecs / eZuce |
| 16:38.41 | WIMPy | cjk: Why not use the web idea? Just have something on the server listen on AMI and update the states. |
| 16:38.45 | paulc | nods - ah, that'll be the integration challenge then |
| 16:38.48 | teknoprep | the issue is a bug currently on the latest version that is hindering monitoring of presence |
| 16:38.56 | paulc | What about |
| 16:39.06 | cjk | WIMPy, because secretaries dont have their browser running all the time |
| 16:39.06 | paulc | FOP - Flash Operator Panel.. that thing still out there? |
| 16:39.08 | teknoprep | but still.. i would prefer not having to have the client logged in if you want to monitor presence |
| 16:39.13 | teknoprep | FOP works with Asterisk |
| 16:39.15 | teknoprep | again i don't use it |
| 16:39.19 | teknoprep | except for call q's |
| 16:39.48 | WIMPy | cjk: It's probably quite easy to do an applicatipo that way, either. |
| 16:40.14 | cjk | WIMPy, yeah but still more unreliable than a sip-subscribe client |
| 16:40.21 | cjk | and not cheaper to develop |
| 16:40.31 | WIMPy | How could it be more unreliable? |
| 16:40.42 | cjk | WIMPy, i am heavily using asterisk manager |
| 16:41.24 | cjk | and i really think it has performance issues |
| 16:41.31 | WIMPy | I haven't had an issue with the manager, yet. Unlike sip subscriptions. |
| 16:41.42 | teknoprep | try 5000 registrations |
| 16:41.46 | teknoprep | it won't work |
| 16:41.58 | teknoprep | i have a client with 5800 registered devices |
| 16:42.03 | teknoprep | 400 just in the call center |
| 16:42.29 | WIMPy | Surely not an issue of AMI. |
| 16:42.29 | cjk | WIMPy, and of course the security issues of astman |
| 16:43.55 | WIMPy | If you only allow read access to the events you need, where's the security issue? |
| 16:44.21 | cjk | WIMPy, the output is far too verbose |
| 16:44.24 | WIMPy | And if you do an application, you can do some middleware if you're really concerned. |
| 16:44.30 | cjk | you have to pay attention to every noop done in the dialplan |
| 16:44.39 | cjk | WIMPy, ok you see the problem with the solution |
| 16:44.48 | cjk | too many points do debug |
| 16:44.51 | cjk | to maintain |
| 16:44.55 | cjk | to many layers |
| 16:44.59 | cjk | just too complex |
| 16:45.06 | cjk | sip sibscribe is easy |
| 16:45.08 | cjk | nothing to add |
| 16:45.17 | cjk | well i can understand teknoprep |
| 16:45.28 | cjk | teknoprep, do you have any budget for a SIP subscribe client? |
| 16:45.38 | WIMPy | Why do you want to enable dialplan at all? |
| 16:45.41 | [TK]D-Fender | Yup, 5800 is substantial. |
| 16:45.45 | teknoprep | yes of course |
| 16:48.01 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:221:6aff:feb8:e0b2) |
| 16:49.48 | *** part/#asterisk riess82 (~riessma@194-166-84-59.adsl.highway.telekom.at) |
| 16:50.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:52.44 | *** join/#asterisk teknoprep (~Tek@23.31.254.90) |
| 16:55.39 | sbrath | I'm having an issue on asterisk 11.10.0 where a custom written dialplan, that sends callers to a Q, sends them there but nothing rings, and then it times out and go's to a fail-over queue, and then finally rings the same phones... The customer dialplan was written circa Asterisk 8 I believe, it dosen't look like any syntax changed... Any sugestions on where to look? |
| 16:56.22 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 16:56.24 | [TK]D-Fender | sbrath: At the call. |
| 16:56.27 | sbrath | My calls come in on a Vega T1 box, and sometimes it looks like the Vega channel is added to the Queue, and sometimes not removed after a call is ended. |
| 16:58.01 | pzn | is there any channel specific to the "sip" protocol? I want to know if there is any sip packet to "change phone display" while a call is in course. I have a weird service that "makes a pseudo conference" and change the people that is talking between 2 (or more calls) during the same call. I would like to implement a "display mode" on the receivers phone |
| 16:59.41 | WIMPy | The only generic way I can think of is CONNECTEDLINE(). |
| 16:59.45 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-gbycswmnwatsqktf) |
| 16:59.48 | sbrath | [TK]D-Fender: I've watched the call come thru, and the log looks wierd. It calls Queue("SIP/Vega-000051fb", "q-cservice,tTwrk,,,45") in new stack..... Then it just sits around for 20 seconds, and says nobody picked up. BUT, durring that 20 seconds, it claims that one of the phones changed state to "Idle" but no app_queue.c SIP/xxx is ringing is listed. |
| 17:00.19 | [TK]D-Fender | sbrath: You should be showing us complete backup for this.... |
| 17:00.46 | [TK]D-Fender | sbrath: Because you're asking for an autopsy and not giving us the body. |
| 17:01.06 | sbrath | [TK]D-Fender: Do you mean you'd like a paste.bin of the log and dialplan portion. No problem. |
| 17:01.57 | sbrath | nice analogy ;) |
| 17:07.04 | sbrath | http://pastebin.com/MKSCRBGA |
| 17:08.11 | *** join/#asterisk zerick (~eocrospom@190.118.16.131) |
| 17:08.42 | [TK]D-Fender | sbrath: I'm not seeing it listed a specific device as being called. |
| 17:08.54 | [TK]D-Fender | sbrath: You should also have dumped the status of the queue, etc |
| 17:09.43 | [TK]D-Fender | sbrath: hopefully you are at verbose 10... |
| 17:10.09 | sbrath | On line 70, it calls Queue() into the 1st level queue, and it seems to timeout one cycle, and then finally ring someone. I'll add the queue status to the paste. |
| 17:12.03 | sbrath | http://pastebin.com/tN8tXJ8E |
| 17:12.47 | sbrath | I'm setting the Verbose, and will watch again. |
| 17:12.58 | [TK]D-Fender | that is the log FILE isn't it? |
| 17:13.18 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:14.25 | sbrath | Do you want the whole File? That was only the portion from the call starting to answer.. |
| 17:14.41 | [TK]D-Fender | NO, you should be loking at live * CLI only |
| 17:14.55 | [TK]D-Fender | not some arbitrary amound of what a logger config decided to output there |
| 17:15.27 | sbrath | I'll look at the live CLI and re-test. |
| 17:24.09 | sbrath | Well, I got a good trace of the start in CLI, and then flinched and ended the CLI. But the logging seems good.. Here it is anyway: http://pastebin.com/ZeWUC6hA |
| 17:25.25 | sbrath | I am confused as to how sip/Vega is a member of the Queue? |
| 17:26.36 | [TK]D-Fender | Where's the queue status dump prior and during? |
| 17:27.46 | sbrath | I guess I have to open more consoles, as it's an active call center and I'm just seeing live calls. Let me do that.. |
| 17:36.17 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-hkpckdnlttzfgfah) |
| 17:41.08 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:41.15 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:42.04 | sbrath | I think I got a good one: http://pastebin.com/vdMTyTFe |
| 17:44.17 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:44.23 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:50.25 | [TK]D-Fender | Curious that we're not seeing it state that it's calling SIP/4120 |
| 17:50.59 | [TK]D-Fender | I'd also look at the SIP debug for this. |
| 17:51.28 | [TK]D-Fender | It's looking like a call to 1 device was attempted but terminated as well |
| 17:51.47 | *** join/#asterisk saint_ (~saint@108.170.8.42) |
| 17:53.10 | sbrath | I still wonder why the Sip/Vega is listed as a member of the Queue? That isn't a valid device. But the prefix of my T1 device. I wonder if the Queue is trying to give it the call, but not logging it. |
| 17:53.19 | saint_ | hi all - does release 13 have a web interface too like 1.8 ? |
| 17:53.29 | sbrath | I'm going to force that out of the queue and watch again. |
| 17:53.46 | WIMPy | saint_: Asterisk itself has never had a web interface. |
| 17:55.01 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 17:55.09 | saint_ | wimpy : i m talking about this one: https://wiki.asterisk.org/wiki/display/AST/Asterisk+GUI |
| 17:56.48 | WIMPy | saint_: Asterisk-GUI has been abandoned AFAIK. *IF* you want a gui, the best (as in least bad) option should be FreePBX. But don't ask about it here. Go to #freepbx for that. |
| 17:56.59 | *** join/#asterisk ctaloi (sid34941@gateway/web/irccloud.com/x-oafnwsjvehvcebkz) |
| 17:57.18 | saint_ | never mind then.. i lll stay within the console and text files. thanks for the hint. |
| 17:58.08 | [TK]D-Fender | Asterisk-GUI was abandoned long before * 1.8 |
| 17:58.39 | WIMPy | Wow. That long ago? |
| 17:58.44 | [TK]D-Fender | yup |
| 18:01.31 | saint_ | BRB |
| 18:02.34 | sbrath | [TK]D-Fender: It seems like if I remove the Bogus Sip/Vega device from the Q it seems to be working better... What would have added Sip/Vega to the Queue Members? |
| 18:02.51 | [TK]D-Fender | Its your system.... |
| 18:03.03 | [TK]D-Fender | We haven't seen where you defined them or allowed them to be added |
| 18:04.39 | [TK]D-Fender | Actually... |
| 18:04.41 | [TK]D-Fender | <PROTECTED> |
| 18:05.02 | [TK]D-Fender | "Dynamic" <- you added it via either CLI or your dialplan |
| 18:05.28 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 18:08.28 | sbrath | Ya I see that, which is odd. As the way you join the queue is to call a special internal number, which "Logs you In". |
| 18:09.22 | WIMPy | sbrath: Could someone have transferred a call to that extension? |
| 18:10.36 | [TK]D-Fender | sbrath: nothing "special". It's all just your dialplan. |
| 18:28.19 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 18:32.04 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1303:2acf:e9ff:fe1c:8fb) |
| 18:38.07 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 18:50.01 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-xtynfzecujbfzyes) |
| 18:50.01 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:02.26 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-tdclhqyxzfwmgwxj) |
| 19:04.36 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:04.46 | *** join/#asterisk jsjc (~Adium@5.9.67.27) |
| 19:09.00 | *** join/#asterisk Cryptix_Durden (~Cryptix_D@41333089.cst.lightpath.net) |
| 19:23.37 | sbrath | WIMPy: Since it's a Ascii "Vega" extension, not even sure how someone would do that. It almost seems like a remote caller was trying to hack and tried to "Join" the queue remotely? |
| 19:26.35 | [TK]D-Fender | Never call a peer an "extension". |
| 19:26.50 | [TK]D-Fender | As for how it got there... you or your users did something improper at some point in time. |
| 19:27.11 | [TK]D-Fender | You might see where the possiblity was creted if it still exists in your dialplan. |
| 19:28.05 | WIMPy | Probably both. |
| 19:32.34 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1303:2acf:e9ff:fe1c:8fb) |
| 19:33.35 | *** join/#asterisk mlhess_ (~digiv@drupal.org/user/102818/view) |
| 19:35.14 | *** join/#asterisk smellis_werk (~smellis@2001:470:c452:2::9) |
| 19:37.48 | sbrath | [TK]D-Fender: I think the error is then in my queue-mgmt login/logout.. It adds/removes members, but it's possible that the SIP/Vega is showing up in it's expression, and then adding the SIP/Vega when someine logs into the queue... :( |
| 19:38.15 | [TK]D-Fender | So far only you know why it might appear there... |
| 19:39.02 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:43.24 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:51.40 | sbrath | so my bug is in my QueueMgmt Code ( That's really old ) http://pastebin.com/Xkyxhvnb |
| 19:59.12 | [TK]D-Fender | wonderfully free of validation.... |
| 19:59.36 | [TK]D-Fender | I suppose we can imagine where that would be accessable from... |
| 20:00.27 | sbrath | only available from a internal button. But I guess I have to rhobust it.. |
| 20:30.20 | *** join/#asterisk saint_ (~saint@184.95.43.162) |
| 20:30.55 | *** join/#asterisk ruben23 (~OpenDial0@112.198.77.98) |
| 20:31.08 | saint_ | if I do a make install of asterisk 13 on a server that used to run 1.8 , it is going to keep all the voice messages, right ? |
| 20:33.18 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1303:2acf:e9ff:fe1c:8fb) |
| 20:34.34 | ruben23 | hi guys i have 2 asterisk server --> asterisk one usersdial with conference then on that conference session he is only the one and from there dials out but this call are pass on to the second asterisk where phones extensions are registered then calls are recorded here also the dialout pattern is executed here until it dialsout....the issue is on this second asterisk it only records all calls in one audio file with multiple call session..this means i lik |
| 20:36.41 | *** join/#asterisk areski (~areski@91.177.220.190) |
| 20:40.01 | ruben23 | any idea guys..? |
| 20:42.02 | saint_ | nop ... |
| 20:45.26 | *** join/#asterisk pigpen (~pigpen@216-177-181-17.block0.gvtc.com) |
| 20:46.14 | pigpen | Hi all, I am using 11.3.0, I have dug around without any luck. I am wanting to enable faxdetect inbound per DID. |
| 20:46.51 | pigpen | I have a PRI, but I have several fax machines connected thorugh the PRI and handed off to an FXS card. |
| 20:47.07 | pigpen | If I do a global faxdetect, it will grab these too. |
| 20:47.19 | pigpen | which at this point, I donât want. |
| 20:47.58 | saint_ | when there is an error in asterisk wiki, who do we contact ? |
| 20:48.01 | pigpen | we have about 200-300 DIDâs and would like to enable it on a per number basis. |
| 20:48.41 | saint_ | pigpen just curious.. what's wrong with the global detecgt ? |
| 20:49.38 | pigpen | the global will grab faxes destined to real machines. |
| 20:51.04 | ChannelZ-Wk | I don't remember, if you have faxdetect on but no fax extension, does it vomit and hang up or just continue on in the dialplan? Because you could separate things by context then if not |
| 20:52.24 | pigpen | hmmâ¦not a bad idea. |
| 20:53.16 | pigpen | so like in incoming, just have it dump real fax machines off to their fxs, but have clients fall through to a context that has a fax exten. |
| 20:53.20 | pigpen | hmm... |
| 20:54.36 | ChannelZ-Wk | Well assuming it works it's the only way I can think of, to look at the DID in your dialplan and then immediately jump them to a different context |
| 20:54.52 | *** join/#asterisk gusto (vanessa@2a02:810d:8640:da4:21e:65ff:fe84:da46) |
| 20:54.54 | gusto | hi workingcats |
| 20:54.56 | gusto | e |
| 20:54.59 | gusto | hi WIMPy |
| 20:55.03 | gusto | WIMPy: how are you? |
| 20:55.29 | pigpen | ChannelZ-Wk: yeah, just strip off real fax machines before moving on, then pick up the tones in aother context. Iâll give it a try. |
| 20:55.30 | gusto | Penguin: hi, how are your mirrors on your truck? |
| 20:58.59 | saint_ | so.. I installed the new pj libraries |
| 20:59.25 | saint_ | then when i do a configure / make menuselect , the res_pjsip has XXX ... what did I do wrong so menuselect don't see them ? |
| 21:01.54 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:01.57 | ChannelZ-Wk | hmm you shouldnt have to install pj separately that I remember |
| 21:02.08 | saint_ | ChannelZ-Wk which i just did .. |
| 21:02.23 | ChannelZ-Wk | or are you trying 13 (I haven't yet) |
| 21:02.36 | saint_ | got it from github , configure , make , make install --prefix=/usr --shared-enable |
| 21:02.39 | saint_ | yeah |
| 21:02.47 | saint_ | i m trying 13 |
| 21:02.50 | ChannelZ-Wk | although I don't see how doing so would break it, unless installing it yourself makes Asterisk have Special New Powers |
| 21:03.04 | ChannelZ-Wk | well I downloaded it at home, let me see what configure says |
| 21:03.30 | ChannelZ-Wk | or I thought I did |
| 21:04.31 | saint_ | that is what i followed: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject |
| 21:04.54 | ruben23 | hi guys i have 2 asterisk server --> asterisk one usersdial with conference then on that conference session he is only the one and from there dials out but this call are pass on to the second asterisk where phones extensions are registered then calls are recorded here also the dialout pattern is executed here until it dialsout....the issue is on this second asterisk it only records all calls in one audio file with multiple call session..this means i lik |
| 21:04.56 | saint_ | ChannelZ-Wk no worries right now. i have to take care of the kids. I'll try again later. |
| 21:06.18 | saint_ | ChannelZ-Wk ha.. so this page says "installing the new repo from svn" ... then it says "installing the old repo from github" ... but under the Installation link , it says "pjproject is new blah blah blah.. you should install it from github" ... |
| 21:07.10 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 21:08.05 | wdoekes | saint_: you're right. the github version used to be the only working one for asterisk |
| 21:08.19 | wdoekes | until the svn version merged in the asterisk-specific changes |
| 21:08.27 | wdoekes | the svn version is the newest |
| 21:08.37 | saint_ | wdoekes ok .. maybe they should fix the wiki then .. |
| 21:09.13 | gusto | thats something i do not understand, that they pull this foreign packets like pjsip in |
| 21:09.16 | wdoekes | I did add the ref to svn, but then I became annoyed with the wiki-tool |
| 21:09.20 | ChannelZ-Wk | well, I'm not sure now. I thought when I first installed 12, it was a custom version of pjsip that was included, it wasn't stock. But I do see now I have some pjsip stuff in /usr/include but I don't remember building pjsip myself. |
| 21:09.39 | saint_ | yes, the "general installation" wiki link forwards you to https://wiki.asterisk.org/wiki/display/AST/New+SIP+channel+driver |
| 21:09.41 | gusto | i for my part am very happy about the modularity of asterisk so that i can keep it small |
| 21:09.44 | saint_ | and that one comes from Asterisk 12 |
| 21:10.07 | saint_ | but then Building and Installing pjproject says to use svn . damn it. |
| 21:10.34 | saint_ | gusto that s a new sip stack. better made. |
| 21:10.44 | mjordan | saint_: you can use the one on github, or the one from Teluu. Both are valid. If Asterisk isn't detecting pjproject, go to the troubleshooting steps |
| 21:10.49 | gusto | "better" |
| 21:10.56 | mjordan | saint_: which has specific instructions on what to look for when that occurs |
| 21:11.05 | saint_ | @mjordan will do. thank you sir. |
| 21:11.08 | mjordan | np |
| 21:11.57 | mjordan | gusto: it is better. It is an actual SIP stack. chan_sip was not. It was more of a single pancake. |
| 21:12.34 | saint_ | @mjordan where would be the troubleshooting page for this particulare issue ? I don t see that in the github install of pjproject |
| 21:13.00 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-Troubleshooting |
| 21:13.53 | saint_ | mjordan can you add a link to the page https://wiki.asterisk.org/wiki/display/AST/New+SIP+channel+driver to go to the one you just gave me ? |
| 21:14.10 | gusto | so it is more modular than chan_sip was, maybe ... but I do have chan_sip on asterisk running on routers with very little ram and it works |
| 21:14.36 | mjordan | saint_: Sure |
| 21:14.54 | mjordan | wait a second |
| 21:15.00 | mjordan | that's a development page you're looking at |
| 21:15.09 | mjordan | Yeah............ you probably don't want to use that at all. |
| 21:15.16 | mjordan | That's now rather old. |
| 21:15.35 | mjordan | If you ever find yourself looking at a wiki page that is under the "Development" heading, approach with caution. |
| 21:15.52 | saint_ | mjordan I followed this page: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject |
| 21:15.54 | gusto | something I would be happy about is a proper digest handling/implementation, maybe pjsip has it, maybe not, I would need to test it. because on chan_sip on MIPS arch routers it did not use random nonce, what is a problem |
| 21:16.20 | saint_ | mjordan which is under the tree getting started / Installing asterisk / Installing asterisk from sources |
| 21:16.27 | mjordan | saint_: yes, that's fine |
| 21:16.33 | mjordan | that has the troubleshooting steps I just linked you to on it |
| 21:16.48 | mjordan | the "New SIP Channel Driver" page was part of the development cycle that produced all this stuff, and hasn't been updated in awhile. It was dev notes. |
| 21:17.08 | mjordan | The information/tutorials/troubleshooting stuff is primarily under here: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 21:17.13 | saint_ | mjordan ok , it is linked at the very top though. so I followed it.. not super clear, lol .. |
| 21:17.40 | mjordan | eek |
| 21:17.44 | mjordan | I'll fix that :-) |
| 21:17.46 | mjordan | sorry about that |
| 21:18.04 | saint_ | no worries - so this pages says also that getting the (old asterisk) pjproject using github |
| 21:18.06 | mjordan | it does say it's for "background information", but we can probably just remove that |
| 21:18.20 | mjordan | " As such, Asterisk requires pjproject version 2.2 or later. Alternatively, an Asterisk compatible version of pjproject is available on github, or - depending on your Linux distribution - available as a package. " |
| 21:21.01 | saint_ | mjordan so you confirm that there is no difference between the svn or github version , right ? becuase the page says "For (old Astersik) version , use git" but it says "for latest pjproject use svn" |
| 21:23.22 | supersoaker | Hello! |
| 21:23.31 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 21:23.31 | supersoaker | Warning, about to paste 3 lines from a log I request help with |
| 21:23.47 | supersoaker | Please advise if this is okay |
| 21:24.27 | saint_ | GIve it a shot. See what happens. haha |
| 21:24.31 | saint_ | l |
| 21:25.14 | supersoaker | [Oct 29 17:17:59] DEBUG[5378] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM ast_sip_devices WHERE name = ? |
| 21:25.18 | supersoaker | [Oct 29 17:17:59] DEBUG[5378] res_config_odbc.c: Parameter 1 ('name') = 'rfozoonmehroffice' |
| 21:25.20 | supersoaker | [Oct 29 17:17:59] DEBUG[5378] res_odbc.c: odbc_release_obj2(0x7f9f2c0a15e8) called (obj->txf = (nil))y |
| 21:25.43 | supersoaker | So, rfozoonmehroffice is no longer a sip device on our network and was removed from the database |
| 21:26.03 | supersoaker | Looks like our provider or something keeps trying to complete a call and i think it involves this |
| 21:26.56 | supersoaker | also : That's odd... Got a response on a call we don't know about. |
| 21:27.04 | mjordan | saint_: the version of pjproject in github right now is 2.3. I think we may have pulled in a bug fix we pushed up. |
| 21:27.15 | mjordan | any difference between the two is bound to be very minor. |
| 21:27.20 | supersoaker | han_sip.c: Invalid SIP message - rejected , no callid, len 580 |
| 21:27.37 | saint_ | mjordan okay. i just switched to svn . no biggies . thank you for the info. |
| 21:29.03 | saint_ | i m going hardcore. doing an upgrade from 1.8 to 13 on a prod server. i m going to be so screwed up if it does not work. |
| 21:35.39 | mjordan | saint_: you did read all of the upgrade notes, right? |
| 21:35.44 | mjordan | like, all of them? |
| 21:37.31 | *** join/#asterisk MadHatter42 (~tuwid@79.171.53.122) |
| 21:37.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-pwptoibturowzccn) |
| 21:41.34 | newtonr | saint_, yeah, uh , that is a big jump. you'll want to read the upgrade notes for each version |
| 21:42.31 | newtonr | i'd expect something to break going that far |
| 21:42.35 | newtonr | almost certainly |
| 21:42.59 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:46.09 | ruben23 | hi guys |
| 21:47.39 | saint_ | newtonr it s a pretty basic installation ... just with auto attendant, ivr , and about 250 sip extensions .. |
| 21:47.41 | saint_ | voicemail ... |
| 21:47.55 | saint_ | i have all night to fix stuff if it does not go well .. |
| 21:48.04 | newtonr | good luck! :) |
| 21:48.28 | saint_ | newtonr you know what's the pain in the butt actually. it's the res_digium and digium phones ... |
| 21:48.50 | *** join/#asterisk d3nyal (~d3nyal@81.241.247.201.static.claro.com.sv) |
| 21:49.04 | *** join/#asterisk theron (~theron@199.201.65.135) |
| 22:12.54 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 22:16.08 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 22:29.05 | *** join/#asterisk jlnt (~lankthata@107-143-2-51.lightspeed.rcsntx.sbcglobal.net) |
| 22:31.35 | ChannelZ-Wk | Is the JSON stuff required by asterisk for ARI? |
| 22:32.12 | file | yes |
| 22:32.20 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 22:32.39 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 22:32.47 | ChannelZ-Wk | OK thanks |
| 22:39.13 | *** join/#asterisk areski (~areski@246.204-67-87.adsl-dyn.isp.belgacom.be) |
| 22:39.40 | *** join/#asterisk benighted (~adam@dhcp-1c-7e-e5-45-5c-af.cpe.wightman.ca) |
| 22:40.39 | benighted | Hey all, I"m setting up a new install of freepbx, I'm able to register my trunk, but once I do I'm unable to register local extensions |
| 22:55.23 | ChannelZ-Wk | #freepbx |
| 23:06.03 | apb1963 | If the provider/carrier/ITSP fails to complete a call (because credit is not available on the account), will there be a CDR created by * 1.8 ? |
| 23:07.50 | [TK]D-Fender | cdr.conf is pretty clear on the basics for this |
| 23:08.04 | [TK]D-Fender | there is a value to set if you want to log unanswered calls or not |
| 23:08.09 | *** join/#asterisk jhlavacek (~jirka@195.70.143.8.adsl.nextra.cz) |
| 23:08.29 | apb1963 | ah. Good point. I forgot that. Thank you |
| 23:11.06 | *** join/#asterisk Compy (~compy@c-73-43-66-143.hsd1.ga.comcast.net) |
| 23:13.09 | apb1963 | "mysql is available via the asterisk-addons" where do I find these addons please? |
| 23:13.23 | apb1963 | eh, I'll google |
| 23:13.36 | ChannelZ-Wk | make menuconfig |
| 23:13.47 | ChannelZ-Wk | see Add-ons |
| 23:16.36 | apb1963 | ty |
| 23:16.42 | apb1963 | not make menuselect? |
| 23:18.35 | ChannelZ-Wk | same diff |
| 23:20.19 | apb1963 | hmm. ok. ty |
| 23:29.05 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:31.54 | ruben23 | hi guys |
| 23:33.50 | Compy | 'lo all. I have an AGI script that I wrote that I can execute when I dial a certain extension. This works well, but is there any way I can tie it to a series of numbers a user can dial during the call? This AGI script records the call, collects a few other pieces of information, then ships it off to a file store. |
| 23:34.30 | Compy | I think I'm just missing the terminology. I'm searching through the rather in-depth documentation, so I'm sure its something simple. |
| 23:34.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:37.53 | ChannelZ-Wk | Well AGI is mostly just a way to control the dialplan externally.. so you'd still do whatever you'd normally do from the dialplan, using Read or Background or WaitExten |
| 23:38.28 | Compy | Ahh, see I was thinking about that all wrong then. |
| 23:38.31 | Compy | I'll read up on it! |
| 23:38.36 | Compy | Thanks a bunch, ChannelZ-Wk. |
| 23:39.31 | ChannelZ-Wk | In terms of triggering the script in the middle of a call.. hmm |
| 23:40.25 | ChannelZ-Wk | Not sure if you can do that with applicationmap in features? I've never actually used them |
| 23:42.40 | ChannelZ-Wk | And you weren't necessarily thinking about it all wrong, re-reading your question I might have gotten your intent a little wrong. |
| 23:42.42 | [TK]D-Fender | you can |
| 23:44.10 | Compy | Hmm. Yeah, I'm trying to think about how the "right way" to do it would be, then research further from there. Either have all calls run through the AGI script and wait for digits to actually invoke behavior (this sounds hacky), or spin up the script once, say *77 was dialed during a call. |
| 23:44.17 | ChannelZ-Wk | The 'collects a few other pieces of information' I read as you wanted to read digits in your AGI but sounds like that's not it at all.. so the applicationmap is really the answer to the question |
| 23:45.19 | ChannelZ-Wk | It sounds like you're just wanting to do a customized one-touch record type thing |
| 23:45.35 | Compy | That's exactly what I want to do, but I don't want to override the default recording mechanism. |
| 23:46.16 | ChannelZ-Wk | Yeah. So hooking with features.conf is appropriate |
| 23:47.44 | Compy | Alright! I'll take a look at that then! I really appreciate it. |
| 23:53.29 | ChannelZ-Wk | Sure good luck |