| 00:02.09 | [TK]D-Fender | Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT |
| 00:02.15 | [TK]D-Fender | CAUSE : No such context/extension |
| 00:04.00 | [TK]D-Fender | CALLED NUMBER : 4123235000 |
| 00:04.07 | [TK]D-Fender | http://www.junctionnetworks.com/knowledgebase/pstn-gateway/pbx-configuration/asterisk-configuration-iax |
| 00:04.31 | [TK]D-Fender | Their own samples don't seem to suggest that the number format you provided is the kind they want you to give it in |
| 00:13.23 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe93:fa45) |
| 00:14.29 | philfry | i figured it out |
| 00:14.37 | philfry | i re-created the trunk…its happy now |
| 00:14.53 | *** join/#asterisk SqRt7744 (~smuxi@S010698fc1148194f.du.shawcable.net) |
| 00:14.56 | [TK]D-Fender | And the difference is...? |
| 00:15.24 | philfry | ?? it dials out right away like i would expect now |
| 00:15.24 | philfry | no idea |
| 00:15.37 | [TK]D-Fender | [20:14]philfryi figured it out <- you didn't |
| 00:16.06 | [TK]D-Fender | You just randomly pushed buttons and for "some reason" it currely works in whatever capacity you've tested it. |
| 00:16.30 | SqRt7744 | hi, I'm trying to set an asterisk server that my dad can call to leave dictations (he'll call a sip account). I want him to be able to record voicemail, but he needs to be able to control the recording (pause, continue, erase, etc). Is that possible? |
| 00:17.36 | [TK]D-Fender | SqRt7744: Not via Voicemail(), but coding your dialplan for him, sure. |
| 00:18.45 | SqRt7744 | ah ok, is there any way to do it w/o hacking it myself? It seems unlikely that I'm the first/only person who would be interested in this functionality... |
| 00:18.52 | jpsharp | Use the Dictate command. |
| 00:19.17 | SqRt7744 | alright, i'll have a look (sorry I'm totally new to asterisk) |
| 00:22.05 | SqRt7744 | the dictate command looks awesome, thanks jpsharp |
| 00:38.49 | *** join/#asterisk gusto (~gusto@ppp-93-104-88-10.dynamic.mnet-online.de) |
| 00:59.45 | *** part/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
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| 02:01.22 | *** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-jlzfxvjfaevdvwwh) |
| 02:01.34 | rdegges | Yo guys >:) |
| 02:11.39 | *** join/#asterisk TSM2 (~the_softw@fw-lon1.wenn.com) |
| 02:34.17 | ChannelZ | whistles |
| 02:35.05 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
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| 03:48.14 | cj | seraphie: the one and only ;-) |
| 03:48.26 | cj | there's a jc over on efnet, but yeah. no. I'm not him. |
| 03:48.54 | cj | seraphie: oops. that was for SerajewelKS |
| 03:49.07 | cj | so. any of you familiar with zyxel DSLAMs? |
| 03:56.53 | *** join/#asterisk rroche (~rroche@201.140.135.226) |
| 04:11.10 | rroche | CDR isn't being called after call completion, once the call has been transferred, Asterisk 1.8.8.0, call can be either initiated from an extension or coming from trunk, but it has to be then transferred to another Extension |
| 04:11.19 | *** join/#asterisk sustav (~vpp@12.230.214.130) |
| 04:11.30 | rroche | Any ideas on how I can get the billable seconds of that call? |
| 04:36.35 | ChannelZ | I think it shows up as two separate calls |
| 04:41.12 | ChannelZ | I don't have to deal with billing but perhaps CEL is a better solution? https://wiki.asterisk.org/wiki/display/AST/Channel+Event+Logging+%28CEL%29 |
| 04:41.25 | rroche | thanks let me read on that |
| 04:53.26 | *** join/#asterisk barkbus (~barkbus@89-96-128-52.ip12.fastwebnet.it) |
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| 06:55.53 | *** join/#asterisk jits (737357e2@gateway/web/freenode/ip.115.115.87.226) |
| 06:56.17 | jits | hi guys.. I would like to accept anonymous calls only from 1 ip address, how do i accomplish this? |
| 06:56.59 | kaldemar | jits: make a device with type=peer, host=<ip> and insecure=port,invite |
| 07:03.02 | *** join/#asterisk oquidave (~oquidave@41.210.129.28) |
| 07:04.32 | *** join/#asterisk vassilux (~vassilux@LPuteaux-156-14-48-156.w82-127.abo.wanadoo.fr) |
| 07:07.18 | vassilux | hi alls, I got some troubles with res_http_post with Debian Squeeze. Debian package gmime is installed any idea ? |
| 07:07.50 | kaldemar | what troubles? |
| 07:09.51 | vassilux | make menuselect indicate that res_http_post can't be seelcted. I execute ./configure and gmime is not find. |
| 07:12.13 | kaldemar | install the -dev package of libgmime |
| 07:12.42 | kaldemar | that's what res_http_post depends on. |
| 07:12.43 | vassilux | so in my own Squeeze box I haven't troubles. I installed dev packages :-) I'm retrying |
| 07:13.25 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 07:16.19 | vassilux | sory my fault I installed libgmime-2.0.-2-dev with libgmime-2.4-dev it works ... |
| 07:20.15 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-xtoppvgajnwmfdcn) |
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| 08:02.44 | *** join/#asterisk sundancer (~rizvicm@2a00:ee0:3a:800:9067:de6e:13e7:a49) |
| 08:03.30 | sundancer | hi |
| 08:03.46 | sundancer | i have this problem with ALSA and im not sure what could be wrong |
| 08:03.50 | sundancer | Connected to Asterisk 1.8.13.1~dfsg-1 currently running on raspberrypi (pid = 2996) |
| 08:03.53 | sundancer | [Oct 26 08:01:11] ERROR[2996]: chan_alsa.c:180 alsa_card_init: snd_pcm_open failed: No such file or directory |
| 08:03.56 | sundancer | [Oct 26 08:01:11] ERROR[2996]: chan_alsa.c:276 soundcard_init: Problem opening alsa capture device |
| 08:03.59 | *** join/#asterisk fredericve (~fes@host-212-68-194-46.brutele.be) |
| 08:04.19 | sundancer | in modules.conf i have: |
| 08:04.19 | sundancer | noload => chan_oss.so |
| 08:04.19 | sundancer | load => chan_alsa.so |
| 08:04.19 | sundancer | any suggestions appretiated, thank you |
| 08:04.33 | v0lZy | hm |
| 08:04.49 | sundancer | asterisk process runs under userid of 'asterisk' which is in 'audio' group just like /dev/snd/* files are |
| 08:04.54 | v0lZy | problem opening alsa capture device.. |
| 08:04.55 | sundancer | crw-rw---T 1 root audio 116, 16 Oct 25 20:15 pcmC0D0p |
| 08:05.15 | v0lZy | sundancer: aplay -l and aplay -L |
| 08:05.19 | sundancer | this is on raspberry pi if it makes any difference |
| 08:05.39 | sundancer | **** List of PLAYBACK Hardware Devices **** |
| 08:05.39 | sundancer | card 0: ALSA [bcm2835 ALSA], device 0: bcm2835 ALSA [bcm2835 ALSA] |
| 08:05.51 | *** join/#asterisk barkbus (~barkbus@2.176.43.195.dsl.static.ip.kpnqwest.it) |
| 08:06.00 | sundancer | # aplay -L |
| 08:06.01 | sundancer | null Discard all samples (playback) or generate zero samples (capture) |
| 08:06.01 | sundancer | sysdefault:CARD=ALSA bcm2835 ALSA, bcm2835 ALSA Default Audio Device |
| 08:06.47 | fredericve | Hi, is there any way to disable directmedia in the dialplan? usecase: when doing call recording we obviously need the audio to pass through asterisk |
| 08:06.54 | sundancer | sound works OK with VLC .. i tried to stream from internet radios |
| 08:07.00 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 08:07.13 | sundancer | v0lZy: any other idea? |
| 08:07.26 | sundancer | or anyone else, thank you again ;) |
| 08:07.30 | v0lZy | hm |
| 08:07.36 | v0lZy | so your alsa sound works, but not with asterisk... |
| 08:08.01 | v0lZy | you're on rpi... archlinux? |
| 08:08.36 | sundancer | v0lZy: its that official debian release. im not sure if it has codename for rpi |
| 08:08.42 | v0lZy | ok. |
| 08:08.43 | v0lZy | hm |
| 08:08.51 | *** join/#asterisk _omer (~omer@58-27-166-195.wateen.net) |
| 08:08.57 | v0lZy | Im not familiar with rpi really... nor debian that much |
| 08:08.57 | v0lZy | but |
| 08:09.03 | _omer | hello |
| 08:09.18 | v0lZy | snd_pcm_open failed no such file |
| 08:09.32 | v0lZy | that would to me imply that somethings wrong with the PCM |
| 08:09.40 | v0lZy | sundancer: if u are using alsamixer |
| 08:09.45 | v0lZy | can u control PCM? |
| 08:13.03 | sundancer | v0lZy: i can set up volumes, mute on playback and recording sliders |
| 08:13.06 | sundancer | so yes |
| 08:13.30 | sundancer | and there is this bcmNNNN chip for soundcard |
| 08:13.32 | v0lZy | u have a slider on PCM? |
| 08:13.36 | sundancer | yes |
| 08:13.39 | v0lZy | hm. |
| 08:14.49 | v0lZy | https://issues.asterisk.org/jira/browse/ASTERISK-14927?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel |
| 08:19.25 | sundancer | v0lZy: hm i see issue is fixed but i cant find the solution.. is it me and my blindness or there is not solution described :) |
| 08:20.21 | v0lZy | ditto. |
| 08:21.22 | v0lZy | from what i google |
| 08:21.35 | sundancer | im not sure what is the difference between ALSA and OSS so this may be a stupid question: can i switch to OSS? |
| 08:21.46 | v0lZy | they mention either alsa.conf or asound.rc being the culprits |
| 08:22.24 | v0lZy | try |
| 08:22.28 | v0lZy | modprobe snd-pcm-oss |
| 08:23.36 | *** join/#asterisk danfromuk (~IceChat77@2.27.12.9) |
| 08:26.48 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 08:27.12 | sundancer | pv0lZy: modprobe worked, asterisk started with chan_oss.so but when i dialed the Dial("Console/dsp" ... hanged raspi :) its not even responding to arping so it must be hard |
| 08:27.18 | sundancer | i will see what hapened when i get home :) |
| 08:27.28 | sundancer | thanks for assistance |
| 08:27.39 | sundancer | sorry, thats v0lZy |
| 08:28.22 | v0lZy | uh, sorry sundancer :( |
| 08:28.51 | sundancer | anyway what im trying to achieve is to get headless softphone so i tried linphonec-nox and twinkle but had problems with that too so im trying asterisk + console now |
| 08:28.54 | sundancer | np |
| 08:31.11 | v0lZy | i think freeswitch can work as a softphone too |
| 08:33.32 | sundancer | will try |
| 08:37.46 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
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| 09:07.29 | *** join/#asterisk rahail (~irC@67.214.121.189) |
| 09:07.34 | rahail | Hi every one i got question |
| 09:07.42 | rahail | i try run asterisk -vr that dont work |
| 09:07.54 | rahail | however when i run the asterisk with asterisk -c or asterisk -vccc |
| 09:08.03 | rahail | it work and when i exit out from putty |
| 09:08.06 | rahail | asterisk stoped |
| 09:08.07 | rahail | :( |
| 09:08.11 | rahail | how can i solve this |
| 09:10.21 | kaldemar | define "work" |
| 09:10.41 | kaldemar | what does "asterisk -vr" tell you? |
| 09:11.39 | rahail | when i run asterisk like that it say u sure asterisk ctl running |
| 09:11.44 | rahail | something like that msg kaldemar |
| 09:13.46 | kaldemar | so, you don't have asterisk running. that makes is impossible to attach to a running process with -r. asterisk -c starts a new process and provides console, but it does not run as a daemon in the background. how did you install asterisk? |
| 09:14.14 | rahail | i instal manuly |
| 09:14.27 | kaldemar | there are many ways to install "manually". what do you mean? |
| 09:14.34 | rahail | wget url |
| 09:14.37 | rahail | ./configure |
| 09:14.38 | rahail | make |
| 09:14.40 | rahail | make install |
| 09:14.43 | rahail | make sample |
| 09:15.16 | kaldemar | "make config" will install an init script which you can use to start asterisk. |
| 09:15.32 | rahail | i can do that now |
| 09:15.37 | rahail | or i have to do everything from begning |
| 09:15.47 | kaldemar | no |
| 09:15.57 | rahail | no on which one |
| 09:16.00 | rahail | do all from begning |
| 09:16.09 | kaldemar | the last. you don't have to do it all, just "make config". |
| 09:17.46 | kaldemar | then you can start it with "/etc/init.d/asterisk start" or "service asterisk start" or whatever works on your platform. |
| 09:17.59 | rahail | now i try asterisk -vr |
| 09:18.00 | rahail | i get his |
| 09:18.01 | rahail | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 09:19.04 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:cdc3:def9:77fb:e5be) |
| 09:19.15 | rahail | kaldemar |
| 09:19.55 | kaldemar | did you start asterisk first? |
| 09:20.01 | rahail | how do i start |
| 09:20.13 | kaldemar | i just told you. |
| 09:20.18 | rahail | i did this |
| 09:20.21 | rahail | service asterisk start |
| 09:20.34 | rahail | thank u so much man |
| 09:20.36 | rahail | :) |
| 09:20.46 | rahail | last 5 hour i was keep reading to find solution |
| 09:20.50 | rahail | i got it now |
| 09:21.02 | rahail | thank u again you do not know how much gratefull i am i cant express it |
| 09:21.15 | sundancer | rahail: do you understand what 'service asterisk start' means? |
| 09:21.29 | sundancer | understand what is that command for |
| 09:21.34 | rahail | to start asterisk |
| 09:21.39 | sundancer | yes |
| 09:21.43 | sundancer | ok |
| 09:22.20 | sundancer | so last 5 hours you didnt know asterisk process was not running? |
| 09:22.30 | rahail | it was runing |
| 09:22.35 | rahail | but every time i loged out from ssh |
| 09:22.37 | rahail | it get stop |
| 09:22.53 | rahail | on forum blog so many placed i was getting wrong hints |
| 09:22.55 | rahail | :) |
| 09:23.01 | rahail | or maby i was searching wrong way |
| 09:23.51 | rahail | so for me with kaldemar help "make config" did the work |
| 09:23.56 | rahail | which i missed on installation |
| 09:25.04 | rahail | ok i guess now i can go back to sleep |
| 09:25.11 | rahail | thank u again kaldemar good night / good morning |
| 09:26.17 | kaldemar | no problem. |
| 09:31.25 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
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| 09:49.36 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 09:53.17 | BorjaGVO | Hi everyone! Is it possible to "come back" from dial() app once the call has been made? I mean, if a user comes in into a queue, and DTMFs 1, the call is transferred dialing some number. My question is, once dial() has been executed, if there is no answer on that phone, can I play a sound afterwards to the user that was on the queue? |
| 09:58.14 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 10:01.19 | *** join/#asterisk jaxon007_ (~jay@123.252.144.92) |
| 10:01.22 | *** join/#asterisk danfromuk (~IceChat77@2.27.12.9) |
| 10:02.05 | jaxon007_ | How to Identify (Detect) FAX on Outbound call? |
| 10:04.18 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 10:09.31 | jaxon007_ | How to Identify (Detect) FAX on Outbound call?. My asterisk version is Asterisk 1.8.17.0. I am using SIP trunk for Outbound calling |
| 10:10.19 | *** join/#asterisk barkbus (~barkbus@2.176.43.195.dsl.static.ip.kpnqwest.it) |
| 10:10.45 | danfromuk | A client's got an ISDN card in an asterisk server. The ISDN receives calls, and is also connected to a Windows based fax server. How do I go about diagnosing where the problem is. |
| 10:17.05 | *** join/#asterisk einstein99 (~msa@80.146.245.99) |
| 10:17.32 | *** join/#asterisk x1user (~User@212.36.13.6) |
| 10:18.04 | x1user | Hi, all. I got gsm gates overloaded, how can i make failove dialplan using SIP as a failover trunk =/ |
| 10:18.47 | *** join/#asterisk natschil (~nathanael@stgt-5d84878e.pool.mediaWays.net) |
| 10:20.16 | natschil | Hello. I've installed asterisk and it seems to work fine. Since I do not own my internet connection, and can only listen on certain ports on the internet, I'd like asterisk to listen to sip requests on port 10023. Is there a) some way to achieve this and b) does anony know any software phones that support changeing the server (not client side) sip port? |
| 10:28.18 | jaxon007_ | Is there any way in asterisk to get the sip codes using some asterisk application or function or any other way? |
| 10:29.39 | kaldemar | jaxon007_: what do you mean by "sip codes"? |
| 10:30.38 | jaxon007_ | <PROTECTED> |
| 10:31.46 | kaldemar | jaxon007_: funtions SIP_CAUSE or HANGUPCAUSE |
| 10:32.10 | jaxon007_ | <PROTECTED> |
| 10:33.59 | kaldemar | natschil: you can define a port in sip.conf, like udpbindaddr=1.2.3.4:10023. remember you also need ports for RTP, those are configure in rtp.conf. i have yet to see a phone that does not support non-5060 ports for SIP. |
| 10:37.01 | jaxon007_ | can someone answer this question: How to Identify (Detect) FAX on Outbound call?. My asterisk version is Asterisk 1.8.17.0. I am using SIP trunk for Outbound calling |
| 10:43.23 | jaxon007_ | kaldemar: there is no such functions available in asterisk 10. |
| 10:45.22 | BorjaGVO | Question: In this line of dialplan: "exten => s,n,Set(GROUP()=foo)", what is GROUP()? I mean, it has a parenthesis...what is it for? |
| 10:47.33 | kaldemar | jaxon007_: oh yes, SIP_CAUSE was a hash. ${HASH(SIP_CAUSE,${CHANNEL(name)})} will have it if you have storesipcause=yes in sip.conf. |
| 10:48.06 | BorjaGVO | It's first appeareance in the dialplan is in this line |
| 10:48.16 | kaldemar | jaxon007_: and the HANGUPCAUSE variable. |
| 10:48.23 | jaxon007_ | kaldemar: Thanks :) |
| 10:48.33 | kaldemar | BorjaGVO: "core show function GROUP" |
| 10:48.52 | BorjaGVO | kaldemar: Thank you! |
| 10:48.56 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
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| 11:11.16 | *** join/#asterisk k610 (~Instantbi@94.139.41.14) |
| 11:15.08 | jmls1 | yay! Asterisk SVN-branch-11-r374932 , System uptime: 1 week, 2 days, 2 hours, 39 minutes, 14 seconds , 101166 calls processed |
| 11:15.34 | jmls1 | 11 is the most stable version of asterisk I have used over the past 7 years. Good job, guys. |
| 11:24.04 | *** join/#asterisk sustav (~vpp@12.230.214.130) |
| 11:30.57 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
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| 11:46.55 | fredericve | anyone can hint me on how to disable parallel builds on asterisk 11? |
| 11:49.17 | fredericve | is it just in the make options? |
| 11:49.19 | *** join/#asterisk hachque (quassel@2600:3c01::f03c:91ff:fe96:5060) |
| 11:50.31 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:53.45 | vassilux | I use snom and aastra phones with my asterisk. Asterisk auto provosioning work well with snom and a pnp script . I use tftp for aastra but use asterisk auto provisioning for aastra like snom. Any idea ? |
| 11:54.02 | *** join/#asterisk kalib (~kalib@177.19.119.68) |
| 11:55.00 | kalib | hello people, I can't restart my asterisk service or even start my CLI (rvvvg). When I tried /etc/init.d/asterisk restart my server became crazy... itś just repeating. |
| 11:55.15 | kalib | "mpg123: no proccess to kill |
| 11:55.23 | kalib | asterisk ended with exti status 1 |
| 11:55.32 | kalib | asterisk died with code 1 |
| 11:55.47 | kalib | automatically restarting asterisk |
| 11:55.56 | kalib | and that's it.. a loop of these messages.. |
| 11:56.05 | kalib | when trying to start the cli (-rvvvvg) |
| 11:56.07 | kalib | I just got: |
| 11:56.43 | kalib | unable to connect to remote asterisk (does /var/run/asterisk/asterisk.cli exist?) |
| 11:56.50 | kalib | but yes, exist.. already checked. |
| 11:58.00 | kaldemar | .cli or .ctl? |
| 11:58.34 | kalib | ops..ctl |
| 11:58.37 | kalib | but I have the file |
| 11:58.40 | kalib | any tip? |
| 11:58.55 | kaldemar | if you have safe_asterisk running, it may be responsible for restarting asterisk. kill it and start asterisk manually with asterisk -vvvc to see why it dies. |
| 12:00.19 | kalib | how can I check this safe_asterisk? |
| 12:00.59 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:01.03 | jaxon007_ | how to run "FaxDetect Dialplan" application in asterisk 1.8 |
| 12:01.20 | kalib | when trying to run asterisk -vvvc I can it tried to start asterisk but got an error about dahdi |
| 12:01.47 | kalib | asterisk has detected a problem with your dahdi configuration and will shutdown for your protection. |
| 12:01.57 | kalib | but it was working just fine until yesterday. |
| 12:02.14 | kalib | but we had a huge blackout on the city.. no energy.. |
| 12:02.21 | kalib | just this morning.. the server is crazy. |
| 12:02.23 | kalib | any tip? |
| 12:02.37 | kaldemar | ps ax | grep safe_asterisk |
| 12:03.02 | kaldemar | if it is running, "killall safe_asterisk" |
| 12:03.27 | kaldemar | then start asterisk manually and grab the error you see. |
| 12:04.00 | kalib | let me check.. |
| 12:04.22 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:05.27 | *** join/#asterisk k610 (~Instantbi@94.139.41.14) |
| 12:05.41 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 12:05.51 | *** join/#asterisk Anderson-D (~anderson@relay.primocollect.com.ua) |
| 12:05.59 | Anderson-D | Hello. How can I disable 2nd line in twinkle? |
| 12:06.31 | Anderson-D | Our calling system just calls to two lines of each user almost all the time |
| 12:06.42 | Anderson-D | And half of all calls is simply missed |
| 12:06.45 | *** join/#asterisk wonderworld (~wonderwor@dsdf-4db55384.pool.mediaWays.net) |
| 12:10.27 | kalib | kaldemar, killed safe_asterisk.. |
| 12:10.37 | kalib | and then tried asterisk -vvvc |
| 12:10.39 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:10.44 | kalib | ang got the same dahdi error |
| 12:10.59 | kalib | asterisk has detected a problem with your dahdi configuration and will shutdown for your protection |
| 12:11.10 | kalib | it gives me options.. |
| 12:11.18 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 12:11.19 | kaldemar | pastebin the error. |
| 12:11.35 | kaldemar | ~pb |
| 12:11.35 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:11.48 | kalib | just a second.. |
| 12:13.40 | kalib | ok..installed.. |
| 12:13.43 | kalib | how to use it now? |
| 12:13.53 | kalib | can I redirect asterisk -vvvr to it? |
| 12:13.55 | BorjaGVO | Is it possible to "come back" from dial() app once the call has been made? I mean, if a user comes in into a queue, and DTMFs 1, the call is transferred dialing some number. My question is, once dial() has been executed, if there is no answer on that phone, can I play a sound afterwards to the user that was on the queue? |
| 12:16.15 | kaldemar | if a call is in a queue, app Dial is not involved. |
| 12:16.43 | kaldemar | both of those can be used with a timeout however. |
| 12:16.53 | jaxon007_ | can someone tell how FaxDetect application works in asterisk 1.8 |
| 12:17.04 | [TK]D-Fender | BorjaGVO, And pressing "1" does not implicitly allow anything to happen there that you didn't configure it to |
| 12:17.33 | kalib | kaldemar, http://pastebin.com/pZapPep6 |
| 12:18.56 | kaldemar | kalib: time to look at your chan_dahdi.conf |
| 12:19.06 | BorjaGVO | [TK]D-Fender: obviously it is configured |
| 12:19.25 | kalib | kaldemar, on it right now.. but it was working just fine.. |
| 12:19.48 | [TK]D-Fender | BorjaGVO, If pressing "1" does something while in queue, YOU mpointed it to a context for exit purposes that allows it to go there |
| 12:20.29 | BorjaGVO | sorry.. |
| 12:20.35 | BorjaGVO | I messed things up |
| 12:20.51 | BorjaGVO | when pressing one, the caller is in a IVR |
| 12:21.16 | BorjaGVO | and pressing 1 implies transferring that call to trunk and dialing out |
| 12:21.48 | BorjaGVO | the question would be if is it possible to do playback a sound if call is not answered... |
| 12:22.06 | fredericve | Can you use T.38 on IAX? |
| 12:23.33 | [TK]D-Fender | BorjaGVO, "core show application dial" <--- |
| 12:23.38 | [TK]D-Fender | fredericve, No. |
| 12:23.47 | fredericve | thanks |
| 12:25.13 | kalib | kaldemar, http://pastebin.com/qX5xCGR6 |
| 12:25.59 | kalib | running right now dahdi_test |
| 12:29.47 | kaldemar | kalib: do you have dahdi and the module(s) for your card(s) loaded? |
| 12:30.15 | robl^ | hrmm. channel topic needs to be updated. asterisk 11.0.0 LTS is published on website (no longer listed as beta) |
| 12:32.01 | kalib | kaldemar, let me check the modules.. as I said, everything was working until yesterday.. but we had a huge balckout on the city. :/ |
| 12:36.09 | *** join/#asterisk LemensTS (~matthew@12.187.73.218) |
| 12:37.02 | kalib | kaldemar, how can I know which modules should be running? with lsmod | grep dahdi I can only see one running |
| 12:37.20 | kalib | ops.. more than one |
| 12:37.39 | kalib | dahdi_transcode 15116 0 |
| 12:38.07 | kalib | dahdi 200528 3 wctdm,wct4xxp,dahdi_transcode |
| 12:38.18 | kalib | crc_ccitt 10561 1 dahdi |
| 12:38.20 | kalib | just that |
| 12:39.51 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 12:41.19 | *** join/#asterisk barkbus (~barkbus@2.176.43.195.dsl.static.ip.kpnqwest.it) |
| 12:43.40 | barkbus | is there any way to subscribe at MS Lync presence service?, some experimental resource module? |
| 12:44.59 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 12:55.36 | jmetro | hm.. I have a dialplan that includes an Option 2 at the auto attendant, and extensions in the 200 range. How can I shorten the wait time that asterisk, where it is waiting for additional digits before following the dialplan for option 2? |
| 12:56.40 | [TK]D-Fender | jmetro, "core show function TIMEOUT" |
| 12:57.54 | *** join/#asterisk Rumbles (~stock@77.107.183.144) |
| 12:59.28 | jmetro | Perfect.. now the optimal place to put this would be on m exten => XXX instead of the exten => 2, correct. |
| 13:00.16 | *** join/#asterisk serafie (~erin@nat/digium/x-jbxpjkssgxltweah) |
| 13:01.25 | jmetro | Or on reading this more closely, I should put it at the beginning of the AA, exten=>s,1,Set(TIMEOUT(digit)=2) for example. |
| 13:02.05 | [TK]D-Fender | jmetro, You do this before the poitn where it awaits any kind of input. |
| 13:05.35 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:06.51 | *** part/#asterisk kalib (~kalib@177.19.119.68) |
| 13:07.57 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 13:09.00 | einstein99 | Hi, someone here who has some knowledge about libpri and clip-no-screening on E1-PRI? |
| 13:12.46 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:14.58 | *** part/#asterisk barkbus (~barkbus@2.176.43.195.dsl.static.ip.kpnqwest.it) |
| 13:15.19 | *** part/#asterisk jaxon007_ (~jay@123.252.144.92) |
| 13:15.33 | WIMPy | ~ask |
| 13:15.34 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:15.43 | WIMPy | einstein99: ^^ |
| 13:18.06 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 13:23.16 | Rumbles | Hello, just wondering, can anyone give me any help with MixMonitor not mixing the calls correctly? On the inbound calls I'm testing with the recipient talks first, but then there's a 5 second gap before the callers voice recording plays, so they're out of sync :/ |
| 13:24.01 | Rumbles | i.e. the recipients recording is 5 seconds earlier than it should be, or visa versa |
| 13:24.35 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:28.15 | einstein99 | I've got a problem with incoming calls what are masked with the clip-no-screening option. |
| 13:28.47 | einstein99 | Asterisk sees (on libpri-level) the screened number, but does not forward this number to be used. |
| 13:28.55 | einstein99 | sry for delay. I was afk. |
| 13:31.47 | einstein99 | On PRI-Debug i can see "Presentation permitted, user number passed network screening" |
| 13:32.26 | einstein99 | imho this bug report is correlated, but i dont find the connected one. https://issues.asterisk.org/jira/browse/ASTERISK-3445 |
| 13:32.33 | WIMPy | Are you saying you want both numbers but only getting one? |
| 13:33.21 | einstein99 | not exactly, i want only the screened number, but get the one provided by the network. |
| 13:34.31 | WIMPy | Looks like I never tried with chan_dahdi. I assumed it would provide both. |
| 13:35.31 | WIMPy | The real issue with this however is that the expected behaviour is unknown. There's no real documentation on what the meaning of the various caller ID field is. |
| 13:37.24 | einstein99 | if i see it correctly, the pri-level just "decides" to forward to asterisk only one of them. |
| 13:37.31 | einstein99 | the false one in my case. |
| 13:40.00 | WIMPy | Interesting. I'll try in a few moments. |
| 13:40.21 | einstein99 | @WIMPy: thanks. |
| 13:40.47 | einstein99 | is afk for some minutes. |
| 13:42.24 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 13:47.42 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 13:51.14 | WIMPy | einstein99: I get both numbers just the way I'd expect to see them. |
| 13:52.16 | *** join/#asterisk Johnny- (~John@23-24-48-117-static.hfc.comcastbusiness.net) |
| 14:01.06 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b8af:1c5a:dfdb:17d6) |
| 14:05.26 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:17.46 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:22.44 | einstein99 | Which variable are you using in extensions.conf to show them? |
| 14:24.00 | WIMPy | CALLERID(num) and CALLERID(ani-num) |
| 14:25.03 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:25.06 | einstein99 | i tried those two, but in both variables i had only the network number |
| 14:26.27 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 14:27.43 | WIMPy | It's not even a current version. It was in february. |
| 14:28.08 | einstein99 | i am using 1.8.10 |
| 14:29.43 | einstein99 | i will try to find the pri debug in logs again. |
| 14:34.47 | einstein99 | _I've got it. |
| 14:36.06 | einstein99 | using pri-debug i have "aPresentation permitted, user number not screened (0)" and then "Presentation allowd of network provided number (3)" |
| 14:36.07 | WIMPy | The debug or the result? |
| 14:36.54 | einstein99 | thats the debug lines. As a result i can only access (ani-num or num) only the network provided number. |
| 14:37.16 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:37.53 | WIMPy | I'd try currente versions before digging deeper. |
| 14:38.53 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:39.54 | einstein99 | i hoped, someone has the solution already. the blogs and bugtracker only showed results from 2005, so i assumed, i just dont see the correct switch how to use this. |
| 14:40.11 | einstein99 | i will try to compile latest version and test it again. |
| 14:40.19 | WIMPy | I'd try current versions before digging deeper. |
| 14:41.03 | einstein99 | thanks anyway for checking with me. |
| 14:42.02 | einstein99 | if it still behavesstrange, i will contact you again and ask you for some pri debug compares, if you dont mind. |
| 14:42.41 | WIMPy | make it so :-) |
| 14:51.29 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:51.39 | Katty | knitknitknit |
| 14:52.36 | jmetro | I hear knitting needles make good earbuds , very clear audio. |
| 14:57.46 | LemensTS | SIP server 87.238.224.40 |
| 14:57.47 | LemensTS | proxy 87.238.224.47 someone gave me these to setup a voip trunk, ive never had 2 different ip's for a voip trunk...whats the difference? |
| 14:57.58 | WIMPy | You don't use needles when knitting telephone networks. |
| 14:58.33 | Katty | but knitting telephone networks is my specialty :< |
| 14:58.39 | Katty | should knit a little asterisk plushie |
| 14:58.51 | Katty | stitches My First PBX into the side. |
| 14:59.00 | Nugget | heh |
| 14:59.03 | jmetro | wouldnt that be trixbox? |
| 14:59.04 | [TK]D-Fender | LemensTS, They need you to label the TO: as their specific server but send it to their proxy (which will carry it onward) |
| 14:59.08 | Katty | gives to leif, even tho he's not here |
| 14:59.17 | Katty | hai nugget |
| 14:59.21 | Katty | you little morsel of awesomeness |
| 14:59.27 | Katty | hugs on Nugget |
| 14:59.40 | *** join/#asterisk bytemaster (~qweq@host81-150-217-167.in-addr.btopenworld.com) |
| 14:59.41 | [TK]D-Fender | mechanically separates Nugget |
| 14:59.46 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 15:00.21 | Katty | thwaps [TK]D-Fender |
| 15:00.25 | Katty | MY NUGGET> I WAS HERE FIRST |
| 15:00.39 | Nugget | huggles Katty |
| 15:00.47 | Katty | also, my coffee has gone cold :< |
| 15:01.08 | Katty | off to the nukerwave! |
| 15:03.26 | *** join/#asterisk Devon_ (~chatzilla@adsl-65-8-195-27.mia.bellsouth.net) |
| 15:05.25 | LemensTS | TKD-Fender, http://pastebin.com/P4Tr3ZKW that is the sip context I created for it...I am unsure what to do with the proxy IP they gave me...you say send it to their proxy, I am unsure how to do that |
| 15:05.47 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 15:06.07 | LemensTS | I just DIAL(SIP/CONTEXTNAME/PHONENUMBER) ... |
| 15:09.54 | Katty | i need a cute little design for my new koozie i'm knitting |
| 15:10.37 | LemensTS | Katty, * <-- what better than Asterisk symbol |
| 15:11.56 | Katty | i can't imagine that'd be very challenging |
| 15:12.10 | Katty | grey koozie, orange asterisk |
| 15:12.25 | Katty | ponders |
| 15:12.30 | Katty | i know i can get 4 letters into 29 stitches |
| 15:12.33 | Katty | and my koozie is 40 |
| 15:12.46 | Katty | i might even be able to get the whole word "asterisk" to wrap around it |
| 15:13.06 | LemensTS | I have the whole work ASTERISK tattoo'd down my ribs |
| 15:13.07 | Katty | however i might get sued for copyright infringement |
| 15:13.15 | Katty | that's a terrible idea |
| 15:13.18 | Katty | it hurts. |
| 15:13.19 | Katty | a lot |
| 15:13.25 | Katty | due to bones and such. |
| 15:13.32 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 15:13.37 | LemensTS | Yea it did. They can try to take it off me lol. |
| 15:14.11 | Katty | that's what kat von d makeup is for. |
| 15:14.16 | Katty | that stuff can cover /anything/ |
| 15:14.24 | Katty | concealer++ |
| 15:14.32 | [TK]D-Fender | LemensTS, outboundproxy= <---- |
| 15:15.15 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 15:16.47 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:16.48 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:20.38 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:22.37 | LemensTS | TKD-Fender, thank you. So one IP points to their SIP Server, and one to their SIP Proxy. The SIP Proxy server sets up the call and information, and the SIP Server handles the actual Voice Data? Am I correct? |
| 15:23.23 | [TK]D-Fender | LemensTS, No, You put the address on the envelope and hand it to their courrier. |
| 15:23.51 | [TK]D-Fender | LemensTS, How they respond and where audio will come from is anyone's guess. But they want you to hand it to their proxy first |
| 15:26.17 | LemensTS | That is a good analogy. So if it goes to the proxy first, and it decides where to go after that, than what is the Sip Server IP for they gave me? |
| 15:26.44 | [TK]D-Fender | tells their proxy where to pass it on to. |
| 15:27.28 | [TK]D-Fender | "To: Mike" -> on a letter you hand to his assistant. |
| 15:27.30 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:27.31 | *** join/#asterisk wonderworld (~wonderwor@dsdf-4db53c82.pool.mediaWays.net) |
| 15:27.48 | [TK]D-Fender | Basically they don't want that server DIRECTLY exposed |
| 15:28.13 | [TK]D-Fender | Or they want to track them by forcing a middleman, etc |
| 15:29.58 | LemensTS | Ok that makes more sense, I will read up on it some more. Thanks. |
| 15:42.52 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
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| 16:02.46 | *** join/#asterisk nny (~Scott@174.107.223.14) |
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| 16:07.08 | *** join/#asterisk LedZeplin (jbearer@shell.atvnc.com) |
| 16:07.34 | nny | http://pastebin.com/xtm8v2Di |
| 16:09.08 | LedZeplin | alright, I'm on trixbox 2.8 and I'm looking for a change. Willing to consider anything, questions is where do I start from here? I'd prefer to use CentOS as the OS, and I'd love for extension info to be pulled from our corporate ldap. |
| 16:10.33 | [TK]D-Fender | FreePBX does not pull from anywhere else |
| 16:11.12 | nny | sorry http://pastebin.com/xtm8v2Di <-- getting odd ARG1/ARG2 swapping on two macros. (I know Gosub > Macro, this is an older dialplan). ARG1 seems to go from value A to value B without any direct change to it after the fact, rather it seems to pick up ARG1 from the initial routine. Thoughts? |
| 16:16.35 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:19.06 | jmetro | Do Timeout(Digit) and WaitExten work on the entire length of time to enter the extension, or start a new timer in between each tone? |
| 16:19.47 | [TK]D-Fender | digit is inter-digit. the parm on Waitexten is teh RESPONSE. |
| 16:20.10 | *** join/#asterisk mobile_gordita (~Robert@66-87-93-143.pools.spcsdns.net) |
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| 16:32.29 | *** join/#asterisk radpandaz (~root@38.127.98.104) |
| 16:32.36 | radpandaz | hey guys have some questions about asterisk |
| 16:32.47 | jmetro | ~book ? |
| 16:32.47 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:33.28 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:33.29 | [TK]D-Fender | radpandaz, Ask away |
| 16:33.39 | *** join/#asterisk ctaloi (~ctaloi@50-56-202-179.static.cloud-ips.com) |
| 16:33.43 | radpandaz | Well I am setting it up for my first time |
| 16:33.53 | radpandaz | How do I set up a IVR? |
| 16:34.07 | radpandaz | Do I just get a VOIP phone and how do I integrate that into the Asterisk application? |
| 16:34.07 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 16:34.15 | radpandaz | How do I make the mariage between the inbound and outbound calls? |
| 16:35.39 | [TK]D-Fender | radpandaz, the dialplan (extensions.conf) |
| 16:36.08 | [TK]D-Fender | radpandaz, ther is no "integrate". Each piece does it's own thing and what you let any device dial, and what it does is entirely up to you |
| 16:36.22 | [TK]D-Fender | radpandaz, All call processing = dialplan = extensions.conf |
| 16:37.19 | radpandaz | Hmm |
| 16:37.41 | radpandaz | Thanks! Let me take a look, one second |
| 16:39.55 | jmetro | The IVR is basically just some sound files playing to the caller, while waiting for them to press a button. Then you have exten => (what they pushed) going somewhere else to ring a phone, go to voicemail, dial by name directory, enter a queue, etc. |
| 16:42.19 | radpandaz | extensions.conf is this massive file |
| 16:42.59 | [TK]D-Fender | radpandaz, Only the psycho sample file..... which you should copy to a temp file and start from scratch. |
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| 16:43.20 | [TK]D-Fender | radpandaz, Your's could end up being 5 lines long. Or 500. |
| 16:44.10 | [TK]D-Fender | radpandaz, Your typical IVR consists of setting timeouts, playing sound file(s), and awaiting input. that part = 4 lines maybe. then more lines for the actual options. |
| 16:44.25 | radpandaz | i just need to know how to call from the server |
| 16:44.50 | radpandaz | how do i associate a ph# to the application? |
| 16:45.13 | [TK]D-Fender | radpandaz, careful on your wording... that sounds liek ahving the server decide of it's own accord to dial out without being a reaction to a call IN to it from a device like an attached phone... |
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| 16:45.31 | [TK]D-Fender | radpandaz, So start that thought over... |
| 16:48.42 | radpandaz | Sorry I'm a little new to this stuff, I'm not a big phone guy. |
| 16:48.57 | radpandaz | Thanks for your help and patience |
| 16:49.23 | pabelanger | radpandaz: Have you read the book? |
| 16:49.24 | pabelanger | ~book |
| 16:49.24 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:50.43 | radpandaz | Hmm, I have not. I just want to do a few simple things, I think... |
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| 16:51.05 | [TK]D-Fender | radpandaz, The book helps describe the basic concepts you'll eed to know. |
| 16:51.17 | radpandaz | If I buy a VoIP phone or similar hardware how do I associate a number with my asterisk server... how do I place calls to an IVR set up on that machine? |
| 16:51.32 | radpandaz | Thank you [TK]D-Fender |
| 16:51.40 | radpandaz | I'll order it |
| 16:51.52 | [TK]D-Fender | radpandaz, free to read online |
| 16:52.56 | [TK]D-Fender | radpandaz, short version : Devices you configure with * each have their own config files (sip.conf , chan_dahdi.conf, iax.conf, etc). That just tells * how to indentify the device and then where to go in the dialplan to process calss they place. It also defines the device-name you use to CALL them within the dialplan. |
| 16:53.57 | [TK]D-Fender | radpandaz, So If you have a SIP phone, and you use an IAX2 ITSP you would set up a sip.conf entrly like [dave], and point that device to a context in your dialplan (eg "context=phones") |
| 16:55.02 | [TK]D-Fender | radpandaz, in the dialplan you'd make a [phones] section and make EXTEN lines to match the kinds of numbers they might dial. One (or more) of these might lead to a Dial() command that tells * to call out to the IAX2 provider via the entry you created in iax.conf for them |
| 16:55.37 | [TK]D-Fender | radpandaz, so it isn't so much "integration" and jsut shoving the call in a place whose steps lead to the outcome you want. |
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| 17:00.23 | radpandaz | There's probably some piece of the puzzle I'm not understanding |
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| 17:01.15 | radpandaz | I have a Asterisk server setup....and I want to use it to call phones. Heh. I know that sounds dumb, I recognize this. I just feel like I need a better conceptual top-down understanding of that process. Hmm. |
| 17:01.51 | [TK]D-Fender | radYou need to break it up instead. The dialplan is what takes the actions of one kind of thing and leads to another |
| 17:02.57 | [TK]D-Fender | radpandaz, Yes you want to call phones. but is this where just your SERVER calls them and the person on the other side only taks to some sort of menu? Or is it that you want a person with a device in hand to PLACE a call, have * take that request then then call OUT? |
| 17:03.32 | [TK]D-Fender | radpandaz, Its the difference between two people talking and one person interactive with just an IVR for example |
| 17:04.01 | [TK]D-Fender | radpandaz, Which is why you want to be VERY clear in your description of what you want. |
| 17:05.13 | [TK]D-Fender | Making * accept calls from the PSTN by some means and falling into an IVR to fill out a questionnaire = easy. Having your server call OUT in a robo-calling manner is another layer of work entirely. Direction matters here. |
| 17:05.36 | [TK]D-Fender | radpandaz, Now that was 2 directions worth of only ONE side being a "person" |
| 17:05.53 | [TK]D-Fender | radpandaz, So ... start over :) |
| 17:08.06 | slav3_kitten | i've not done a ton of googling. but assuming i can get my cisco 7921 working properly. is it possible to have an incoming call ring both the 7921 and my sip 7960 at the same time then transfer it to whichever picks up; easily possible i add? |
| 17:08.58 | [TK]D-Fender | slav3_kitten, Just dial both. |
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| 17:10.29 | slav3_kitten | well there is a simple solution that should have just smacked me in the face lol |
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| 17:10.50 | Dovid | hi all |
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| 17:12.34 | radpandaz | is there a good beginner document i can use for asterisk? |
| 17:13.11 | PipBoy | Hey, I am looking do strip + off of some outbound calls before my billing system process' the call. I have the general idea of how my dialplan should look but I need a bit of help. Anyone got a few mins? |
| 17:13.35 | jmetro | use the book radpandaz, i linked it above |
| 17:13.38 | [TK]D-Fender | PipBoy, Variables 101 .... |
| 17:13.58 | [TK]D-Fender | jmetro, We was linked to it TWICE and acknowledged it already ..... |
| 17:14.17 | [TK]D-Fender | radpandaz, And the book is it. |
| 17:14.42 | [TK]D-Fender | radpandaz, the best thing we currently have out there. Until maybe I write one myself.... |
| 17:16.00 | PipBoy | Thanks <[TK]D-Fender> for letting me know its easy lol.. Already figured it wasnt super complicated |
| 17:16.24 | jmetro | RadPandaz: It sounds like youre just trying to setup outbound dialing. you are going to need a SIP trunk probably. |
| 17:16.26 | [TK]D-Fender | PipBoy, Also shouldn't take minutes. Just show your sample case... |
| 17:16.36 | [TK]D-Fender | PipBoy, And we'll help refresh you... |
| 17:16.59 | [TK]D-Fender | jmetro, Don't go assuming what tech's he'll be using and planting ideas in his head. |
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| 17:19.29 | PipBoy | http://pastebin.com/0kqS67f7 is the dialplan and I know that something along the lines of exten => _+X!,1,Noop is the start of removing the + .. but I am having issues putting it together properly |
| 17:20.00 | PipBoy | I dont dig around in my dialplan as much as I should >.> |
| 17:24.18 | [TK]D-Fender | PipBoy, seeing a2b there tells me you likely learned as little as humanly possible while trying to become the next calling card business.... |
| 17:24.29 | PipBoy | Wow your awesome |
| 17:24.30 | [TK]D-Fender | PipBoy, And I see only 1 pattern in tehre. |
| 17:24.45 | [TK]D-Fender | exten => _X., <- 1 patern only |
| 17:25.01 | [TK]D-Fender | start with a digit, have 1 or mor chars after it. |
| 17:25.31 | [TK]D-Fender | Now, I asked you for a sample of what you wanted to DO, not to see what you already did. |
| 17:25.54 | [TK]D-Fender | I want to accept XYZ and in the end process 123 |
| 17:25.56 | [TK]D-Fender | ^ |
| 17:26.06 | [TK]D-Fender | before -> after |
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| 17:26.41 | [TK]D-Fender | exten => _X.,1,Set(CALLERID(num)=${CALLERID(NUM)}) <--- this already does precisely nothing |
| 17:27.19 | PipBoy | Firstly.. I am not an Asterisk Guru, However I have worked with it for a few years.. And I currently run a small voip business with a few PRI's and I dont appreciate your rude remarks |
| 17:27.22 | [TK]D-Fender | exten => _X.,n,Set(DYNAMIC_FEATURES=) <-- was this somehow set GLOBALLY elsewhere? If not... it's already blank, and what do we think this would do? |
| 17:27.59 | [TK]D-Fender | Things to think about in this bit you did show us. |
| 17:28.11 | [TK]D-Fender | So..... whats the A>B you're looking to do? |
| 17:29.08 | PipBoy | I am taking calls from a uc320 that forces +1 on local calls.. I want to strip the + and pass the rest through to my Billing program. I later add the + depending on who my LCR sends it too |
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| 17:30.11 | PipBoy | the code I linked via pastebin is the dialplan that takes calls from some of my customers and routes it into my billing.. However I was planning to make a sepperate context to handle the "+1" calls from my cisco customers |
| 17:35.11 | [TK]D-Fender | PipBoy, So make a pattern to matcht eh + in front and goto the exten stripping off the first char as you do. |
| 17:37.12 | PipBoy | Yea, I've done it before... Keep messing it up this time. Figured id ask for some help because I have a full schedule. Ill figure out what im doing wrong |
| 17:37.36 | philfry | how do phones detect if there is a new voicemail for an extension? the phones aren't detecting a new voicemail for some reason |
| 17:38.03 | jmetro | the message waiting indicator - it is sent by the server |
| 17:38.31 | [TK]D-Fender | PipBoy, Can you show me a pattern that would even jsut match it? |
| 17:39.15 | jpsharp | But you have to tell the server what mailbox is tied to what SIP device with the mailbox= line in sip.conf |
| 17:39.16 | [TK]D-Fender | philfry, Phones don't "detect" it. Asterisk TELLS them as per your peer definition being told to tell them about it. |
| 17:40.41 | philfry | oh i see |
| 17:42.20 | philfry | well the weird thing is i have a digium phone and it sees the voicemails correctly and if on a polycom phone i dial the voicemail feature code it does show a new voicemail |
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| 17:45.02 | [TK]D-Fender | philfry, Thsoe are 2 very different sounding descriptions. |
| 17:45.16 | [TK]D-Fender | one is SEE, the other is DIAL. |
| 17:45.22 | philfry | right |
| 17:45.26 | [TK]D-Fender | and we aren't seeing any of what was set up |
| 17:45.35 | [TK]D-Fender | You are comparing apples & oranges across the board. |
| 17:46.04 | philfry | well i just mean that even though dialing shows the voicemails for that line but the phone indicator doesn't light up |
| 17:46.45 | philfry | so would sip_additional.conf shed any light on the configuration? |
| 17:47.40 | [TK]D-Fender | philfry, Partly. that and "voicemail show users" |
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| 17:57.16 | philfry | ok i think i figured it out each extension was set to use extension@device and changed it to extension@default |
| 17:57.28 | cusco | hi... I'm trying to optimize DTMF between two asterisks via SIP |
| 17:58.21 | cusco | I dialed 9 digits, and only got 92847 ... debug: http://paste.debian.net/203890/ |
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| 18:08.13 | cusco | and .. I should say, I'm using X-Lite -> asterisk-1 -> asterisk-gw -> (PSTN) --back-again--> asterisk-gw -> asterisk-1 |
| 18:08.27 | cusco | the end machine, asterisk-1 is reading up only those digits |
| 18:08.41 | cusco | but in gw, I can place a IVR and read all the 9 digits |
| 18:08.52 | cusco | (I am typing them fast in the softphone) |
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| 18:23.06 | cusco | this is while using Read() |
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| 18:54.36 | natschil | kaldemar: thanks for your reply regarding non-standard sip ports earlier! |
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| 18:56.36 | nny | http://pastebin.com/xtm8v2Di I have a macro that is being called after another macro (dials multiple phones, someone answers, transfers using the second macro). Somehow even though the second macro is setting ARG1 the second macro is intermittently using ARG1 from the first macro, creating issues. I am aware the macro is depreicated in favor of gosub (old dialplan) but afaik it should still work. |
| 18:57.07 | nny | oddly enough the second macro always uses ARG2 as defined |
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| 19:01.39 | nny | is ARG1 being overwritten when the other two channels are closed (ZOMBIE)? |
| 19:02.55 | nny | eh nm, just gonna remove the first macro, superfluous anyways |
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| 19:31.21 | sherl0k | hi, anyone here have experience in DTMF tones and how they register in asterisk? |
| 19:34.02 | [TK]D-Fender | sherl0k, What is your issue/goal? |
| 19:34.52 | sherl0k | I have people in another office dial into our conference system, but when they press keys to use the options in the menu, asterisk isn't recognizing them |
| 19:35.09 | sherl0k | kicks back the "we have not received a valid response" message |
| 19:36.06 | [TK]D-Fender | Conferencing does not forward DTMF AFAIK |
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| 19:37.10 | sherl0k | Well we used to run Trixbox 2.8.0.4 and never had this problem |
| 19:37.41 | sherl0k | Believe the Asterisk version of that was 1.4? i can log in and look |
| 19:38.13 | sherl0k | 1.6, my mistake |
| 19:38.30 | sherl0k | We're now on 1.8 |
| 19:39.56 | [TK]D-Fender | Well if you're referring to conference ends passing between each, it shouldn't. If you're referring to oterh direct use then your mode is likely jsut not right |
| 19:40.17 | sherl0k | User dials one of our numbers, it passes to an IVR |
| 19:40.38 | sherl0k | They punch in a conference number and enter the PIN, and they're placed in it |
| 19:40.45 | sherl0k | This works for about 80% of the people that dial in |
| 19:41.12 | sherl0k | DTMF passes through this, I've turned on DTMF debugging in the Asterisk console and I see it take place |
| 19:41.22 | [TK]D-Fender | Then you have a basic DTMF issue, nothing specific to conferencing. |
| 19:42.19 | sherl0k | Correct, it's DTMF related |
| 19:42.52 | natschil | Can anyone point me to documentation as to howto setup an asterix server with the iax2 protocol? It's very hard to find documentation that is up to date, as a lot seems to have changed at some point recently with the asterix code base (or maybe I'm wrong). |
| 19:43.54 | [TK]D-Fender | natschil, Only thing that changed in the past 5 years is CALLTOKEN which is well documented. |
| 19:43.59 | sherl0k | I've tried multiple DTMF modes - inband, auto, rfc2833 |
| 19:47.26 | natschil | [TK]D-Fender: hmm, I must be wrong with that respect then. There is a lot of documentation on the internet regarding files such as sip_nat.conf or such which do not exist on my server, and changeing the respective options in sip.conf works too, so I deduced that the config file structure must have changed. If that is not the case, then I will try creating iax.conf, and hope that that will work. |
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| 19:49.21 | [TK]D-Fender | natschil, sip_nat.conf is OLD FreePBX BS, and nothing to do with IAX2 |
| 19:49.57 | [TK]D-Fender | natschil, Even comparing SIP to IAx says that your head isn't on straight about this. |
| 19:51.52 | natschil | Takapa: oh okay. No I wasn't saying they're the same thing, I was saying that that was an example where I (wrongly) deduced that config file layout had changed. |
| 19:53.55 | natschil | My head probably isn't straight about this either though. I've discovered that sip over a nat is a pain to implement, and hence I'm now hoping I can connect to asterix with something like sflphone over iax2, as that makes NAT easier. I don't care that it may be a dirty hack, all I need is to transmit voice over some protocol that is encrypted. |
| 19:54.24 | natschil | and if the encryption happens client side, such as with zrtp, It doesn't matter how badly the server side is implemented. |
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| 20:04.01 | drmessano | Damn Damn Damn |
| 20:05.22 | jpsharp | Except if you encrypt client side via zrtp and then pass the call off via IAX, Asterisk will decrypt the call before sending it over the IAX pipe. |
| 20:09.12 | jpsharp | So either your zrtp endpoints need to be able to talk directly to each other *or* you need to set up IAX encryption. |
| 20:11.36 | sherl0k | [TK]D-Fender: any thoughts on my DTMF issue? Any suggestions? |
| 20:12.16 | [TK]D-Fender | Don't know what is on the other side or what they are supposed to be suppotring |
| 20:12.43 | [TK]D-Fender | And you should be doing a more comprehensive test. Go make an IVR to test it over a longer scale |
| 20:15.19 | sherl0k | How much more comprehensive can I get than "have people dial the IVR" |
| 20:16.11 | sherl0k | I'm watching the logs, it's not showing the tones being parsed |
| 20:18.20 | [TK]D-Fender | What is on the other end? Get a solid configrmation of what they are configured to support. Check your own side and prove the peers are matched right. If it's anything OOB (rfc2833, INFO, etc), then there is a more serious issue. If it's inband... it should only be from a crappy provider and I'd start looking for alternatives. |
| 20:20.58 | [TK]D-Fender | confirmation* |
| 20:39.31 | sherl0k | Sometimes I never know. I know one office it's a VOIP provider, but in other cases it's a third porty. Could be a cell phone, another VOIP provider, POTS, who knows |
| 20:40.22 | sherl0k | The annoyance is that this worked fine before I upgraded, and I've done as much as I could to mimic the old configuration but it appears there are defaults that have changed and I don't know what the settings are supposed to be now./ |
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