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| 00:40.56 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
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| 01:56.11 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.79.32) |
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| 02:17.08 | ruben23 | hi guys anyone can help ---->how to registered two asterisk server from each othere |
| 02:26.58 | *** join/#asterisk jpoz_ (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
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| 02:53.11 | TazzNZ | ruben23: I would setup a SIP friend between the two servers |
| 02:53.18 | TazzNZ | but what are you trying to do ? |
| 02:55.32 | ruben23 | TazzNZ: i plan to used iax on both server - so calls can pass on both server.. |
| 02:56.54 | TazzNZ | you can use SIP for that too, but IAX is good |
| 02:57.21 | TazzNZ | in that case, you need to edit iax.conf and add a friend in for the remote server |
| 02:57.31 | zopsi | I really do like chan_pjsip and the new DPMA ... nice job Digium + contributors. |
| 02:57.32 | TazzNZ | edit them on both servers |
| 02:57.55 | ruben23 | the problem i set configuration - i see btoh serevr from each oteh but got this ---> 192.168.7.125:4569 N register = <Unregistered> 60 Rejected <---server 1 |
| 02:58.23 | ruben23 | serevr 2 ---> 192.168.7.125:4569 N register = <Unregistered> 60 Rejected |
| 02:58.42 | TazzNZ | that means that you have something wrong with the username/password/ip |
| 02:59.01 | ruben23 | i should build and extension first..? |
| 02:59.08 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-210-61.socal.res.rr.com) |
| 02:59.19 | ruben23 | like iax 1000 |
| 02:59.27 | TazzNZ | shouldn't need it |
| 02:59.56 | TazzNZ | you can pass a call directly to the peer |
| 03:01.08 | ruben23 | how do i check the peer..? of each server..? |
| 03:01.20 | TazzNZ | e.g. .....Dial(IAX2/my_remote_server/12345) |
| 03:01.33 | TazzNZ | that will dial 12345 on the remote server |
| 03:01.40 | TazzNZ | iax2 show peers |
| 03:02.04 | ruben23 | i dont see anything with that its empty |
| 03:02.09 | ruben23 | iax2 show peers |
| 03:02.20 | TazzNZ | have your created your peers ? |
| 03:02.26 | TazzNZ | pastebin iax.conf |
| 03:03.29 | ruben23 | i think the problem is - the other one is elastix and the other one is asterisk <---does this matters..? |
| 03:03.46 | TazzNZ | no |
| 03:03.56 | TazzNZ | can you show us what is in that file ? |
| 03:04.45 | ruben23 | ok wair |
| 03:04.51 | ruben23 | on both..? |
| 03:04.56 | TazzNZ | yes, both |
| 03:06.33 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
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| 03:40.20 | stevePearPear | hi, I could connect to Asterisk through ws, however it seems that its not detecting any connection through wss. is the url = wss://xxx.xxx.xxx.xxx:8088/ws ? ws worked at ws://xxx.xxx.xxx.xxx:8088/ws |
| 04:01.15 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 04:03.29 | *** part/#asterisk vinhdizzo (~vinh@cpe-98-154-210-61.socal.res.rr.com) |
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| 04:35.28 | ruben23 | TazzNZ: still there..? |
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| 04:41.35 | *** part/#asterisk KVron (~KVron@187.194.45.100) |
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| 05:06.18 | ruben23 | TazzNZ: still there..? |
| 05:08.38 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 05:08.55 | billu | hi i have a asterisk version 11.6-cert1 with freepbx 2.11, i have a issue with voice, playback applic. ation have drop in sound and also have low voice on bridge channels. one thing that i notice the result of "core show translation" have increased translation rate e.g. ulaw->alaw 9150 while on asterisk 1.8 it is 1, and my same set up is working fine with asterisk 1.8. please help. |
| 05:30.34 | [TK]D-Fender | [23:03]ruben23i think the problem is - the other one is elastix and the other one is asterisk <---does this matters..? <- BOTH are Asterisk |
| 05:42.53 | ruben23 | [TK]D-Fender: the asterisk1 -----> autodialler/aut 192.168.56.3 (S) 255.255.255.255 4569 OK (2 ms) / 192.168.56.3:4569 N TESTast 192.168.56.2:4569 60 Registered |
| 05:43.28 | ruben23 | Elastix ---> TESTast/IAXTEST 192.168.56.2 (D) 255.255.255.255 4569 OK (537 ms) / 192.168.56.2:4569 N register = <Unregistered> 60 Rejected |
| 05:43.45 | stevePearPear | could Asterisk work with wss? |
| 05:44.19 | [TK]D-Fender | ruben23: aways PASTEBIN this stuff... and you were asked for your configs a while ago... |
| 05:44.21 | ruben23 | the asterisk1 resolved the registration already but the elastix is still rejected. |
| 05:44.31 | ruben23 | ok sending |
| 05:49.18 | ruben23 | [TK]D-Fender: this is the config -------> http://pastebin.com/Lczd9f2R |
| 05:49.45 | [TK]D-Fender | register=register => autodialler:dialler123@192.168.56.2:4569 <- of course this is going to fail.... |
| 05:50.27 | [TK]D-Fender | First your have "register=register =>"... clearly bad right from the start... |
| 05:50.33 | [TK]D-Fender | You dobled the word "register" |
| 05:50.50 | [TK]D-Fender | Then even if you fix that part you're not ALLOWED to register |
| 05:51.08 | ruben23 | why..? |
| 05:51.16 | [TK]D-Fender | [autodialler] host=192.168.56.3 <--- thiws peer has a FIXED host. You are not allowed to register to something with a fix host. |
| 05:51.37 | [TK]D-Fender | There is n need to register to the other server at all |
| 05:51.40 | [TK]D-Fender | no* |
| 05:52.13 | [TK]D-Fender | You've forgotten your basics here... |
| 05:53.49 | ruben23 | [TK]D-Fender: sorry, so no need to registered both..one onyl woudl work..? |
| 05:54.07 | [TK]D-Fender | [01:51][TK]D-Fender[autodialler] host=192.168.56.3 <--- thiws peer has a FIXED host. You are not allowed to register to something with a fix host. |
| 05:54.20 | ruben23 | ok ill chnage it to dynamic |
| 05:54.23 | [TK]D-Fender | What is the purpose of registration? |
| 05:54.25 | [TK]D-Fender | NO |
| 05:54.55 | [TK]D-Fender | One side HAS to Stop registering. Thre is no need |
| 05:56.30 | [TK]D-Fender | Both of your servers no doubt have fixed IP's on your LAN I see there... so there isn't even a point inregistring. |
| 05:56.34 | [TK]D-Fender | on EITHER side |
| 05:56.48 | [TK]D-Fender | Never 2, and you don't even need 1. |
| 05:57.16 | ruben23 | how to bridge calls in between with them..? |
| 05:57.27 | [TK]D-Fender | set your hosts. Dial. Done. |
| 06:04.59 | *** join/#asterisk Iamnach0 (~Iamnacho@ip98-186-184-132.ks.ks.cox.net) |
| 06:06.43 | billu | Hi asterisk ver 11.6-cert1 have increased translation rate? |
| 06:10.20 | [TK]D-Fender | No, that is calced based on CPU. |
| 06:10.27 | [TK]D-Fender | Sounds like a build or environment issue |
| 06:12.32 | stevePearPear | any idea why I’m able to connect to Asterisk on ws://xxx.xxx.xxx.xxx:8080/ws but not on wss://xxx.xxx.xxx.xxxx:8080/ws ? |
| 06:13.52 | billu | but i have build it 2 different environment one is host on VM and other is on stand-alone machine with core i-3 but both gives same translation. |
| 06:21.30 | billu | I am facing voice quailty issues on my asterisk-11 version. |
| 06:26.39 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 06:28.50 | [TK]D-Fender | heads to bed. |
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| 07:31.42 | billu | any one can help? |
| 07:33.55 | stevePearPear | anyone tried wss on Asterisk before? |
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| 07:42.30 | ruben23 | hi guys |
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| 08:08.08 | ruben23 | hi guys |
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| 08:14.18 | stevePearPear | hello, anyone tried Asterisk with wss and SIPML5 ? |
| 08:14.59 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-qxyayoaweusigfet) |
| 08:18.26 | wdoekes | ~poll |
| 08:18.27 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 08:18.29 | wdoekes | ~polls |
| 08:18.29 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
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| 08:49.37 | *** join/#asterisk _omer (omer@184.175.79.212) |
| 08:52.18 | _omer | anyone here? |
| 08:52.45 | _omer | how do I know if any digium card is connected/installed with system? |
| 08:54.44 | Tim_Toady | lspci, dahdi_hardware |
| 08:55.39 | *** join/#asterisk lordz_md (~lordz.md@gateway.13elements.com) |
| 08:58.43 | _omer | sorry I know nothing about checking digium cards. |
| 08:58.53 | _omer | is this some command ? |
| 08:59.41 | Tim_Toady | 2 different commands first is more generic, the second digium specific |
| 08:59.58 | _omer | let me try both one by one |
| 09:00.48 | _omer | [root@localhost asterisk]# dahdi_hardware |
| 09:00.49 | _omer | pci:0000:05:00.0 wcaxx+ d161:8010 Digium A4B |
| 09:00.58 | _omer | thanks. it works |
| 09:02.03 | _omer | Tim: once card is installed and PSTN analog line is plugged in, Then which channel should be used for incoming calls? DAHDI ? |
| 09:02.26 | _omer | i want to call at PSTN line and to see if call is coming in ... just like we do "sip set debug on" |
| 09:06.00 | ruben23 | hi guys - i have asterisk 1 and asterisk registered to each other, problem both of there extensions cant call to each other...any idea..? |
| 09:06.11 | Tim_Toady | _omer: you have to check your dadhi config, both in /etc/dadhi/system.conf |
| 09:06.24 | Tim_Toady | and in /etc/asterisk/chan_dahdi.conf |
| 09:08.27 | _omer | thanks Tim. Let me check |
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| 09:20.23 | stevePearPear | hello, anyone tried Asterisk with wss and SIPML5 ? |
| 09:24.17 | *** join/#asterisk whizzi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
| 09:24.18 | stevePearPear | i could connect to asterisk using SIPML5 with ws, however when i changed to wss://, i always got failed to connect error. It seems like Asterisk didn’t even recieve any wss message |
| 09:26.03 | whizzi | Good day everbody. Is there anything changed in Asterisk 11.10.0 with the logging? I can’t seem to get any logging output anymore on any logging channels? |
| 09:27.45 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
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| 09:36.48 | whizzi | never mind my question, it seems to be fixed and I’m just having trouble putting in nrs on a phone >_< |
| 09:45.48 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.76.92.218) |
| 09:49.06 | *** join/#asterisk woopstar (~woopstar@unaffiliated/woopstar) |
| 09:50.06 | woopstar | Hi all. I have video working with WebRTC + Asterisk, but the quality is very poor even though the clients are connected through LAN (1gbit). Are there any settings in Asterisk to look at? |
| 09:53.10 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
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| 10:16.31 | stevePearPear | hi woopstar, did ur webrtc go through wss? |
| 10:16.53 | woopstar | stevePearPear: yup. Running wss |
| 10:16.59 | woopstar | to webrtc2sip |
| 10:20.54 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
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| 10:38.33 | stevePearPear | icic i was trying out through wss, except directly through Asterisk :( And it seems like I couldn’t connect wss directly to Asterisk :( |
| 10:43.20 | Greek-Boy | why would by Polycom phones disregard SIPAddHeader(Alert-Info: Ring Answer) even when I have confiured it in the xml config file with the correct class? |
| 10:43.21 | Greek-Boy | :-( |
| 10:47.02 | woopstar | stevePearPear: had that problem too, so used the webrtc2sip gateway. do note, that to get wss to work, you need a valid not self signed certificate, or you have to import the self signed into your browser |
| 10:47.59 | stevePearPear | woopstar, using webrtc2sip gateway would mean loading up 1 more server in this case is it? I’ve a running webrtc2sip server too and I’m trying to do without it |
| 10:48.51 | woopstar | stevePearPear: yes, tried the same but never got it to work. Think problem is multiple, but pretty much related to this: https://issues.asterisk.org/jira/browse/ASTERISK-22961 and that Asterisk is not supporting VP8 which Chrome is using. |
| 10:49.44 | file | Asterisk has VP8 passthru support. |
| 10:50.18 | woopstar | file: oh... didn't knew. thanks. |
| 10:50.23 | stevePearPear | woopstar: i am using for web calling now only, thanks for the link, i’ll have a look |
| 10:51.04 | woopstar | stevePearPear: did you get webrtc to work with asterisk through wss and chrome 35? the calling part? |
| 10:51.12 | woopstar | or did i just read you wrong? |
| 10:51.16 | stevePearPear | i managed to get it working for ws |
| 10:51.26 | stevePearPear | i’m having issue getting it to work through wss :( |
| 10:51.43 | stevePearPear | it can’t even connect to my Asterisk v12.3 server |
| 10:52.54 | woopstar | heh, great :D |
| 10:53.33 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-vhqzqwopevvdqmcq) |
| 10:54.05 | stevePearPear | :( |
| 11:00.22 | *** join/#asterisk StaRetji (~vencuga@178.79.6.17) |
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| 11:11.30 | jaflong | hi., does anyone know the what the video rtp properties are required for compatibility with asterisk. I am sending the rtp from gstreamer, the audio is processed fine but video is not. |
| 11:12.36 | jaflong | This is a rtp debug line that show rtp is received. Got RTP packet from 10.1.1.40:44902 (type 99, seq 064200, ts 158009290, len 001348) |
| 11:13.08 | *** join/#asterisk x5eb (~seb@seb-hpws2.elen.ucl.ac.be) |
| 11:13.35 | stevePearPear | hey woopstar: u still around? it seems that its still an on-going patch discussion. If I were to use webrtc2sip, what does webrtc2sip does in that case? Do we require a good webrtc2sip server in that case? |
| 11:13.52 | *** join/#asterisk HeN_ (uid3747@gateway/web/irccloud.com/x-shtvgyhrysniyvpx) |
| 11:14.17 | woopstar | stevePearPear: yea... its a "proxy / media gateway" . I run it on a seperate server. http://webrtc2sip.org/ |
| 11:15.07 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.50) |
| 11:15.10 | stevePearPear | ic, I have a webrtc2sip server going on too. I’m just curious if it would take huge resources for webrtc2sip |
| 11:15.31 | stevePearPear | does it do video transcoding? or it would simply route the traffic to Asterisk which does most of the work |
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| 11:25.29 | StaRetji | hi, can some one help me finding example of diversion header config? I have problem with calls that are forwarded do not show original callerid |
| 11:25.50 | StaRetji | thanks ;) |
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| 11:29.44 | stevePearPear | woopstar: u there? |
| 11:30.16 | woopstar | yes |
| 11:30.59 | stevePearPear | have a question :) |
| 11:31.00 | stevePearPear | I have a webrtc2sip server going on too. I’m just curious if it would take huge resources for webrtc2sip. Does it do video transcoding? or it would simply route the traffic to Asterisk which does most of the work |
| 11:31.17 | woopstar | it does proxy, so resources are moved i believe. |
| 11:32.39 | cargill | I'm trying to set up dundi and whatever I do, dundi show peers is empty, but the peers are defined in dundi.conf, that should be enough according to all the documentation I could find |
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| 11:49.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:55.33 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 11:55.37 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 11:55.41 | cargill | what could be set wrong? |
| 11:57.37 | wdoekes | cargill: set verbosity, unload the dundi module, and then load the module |
| 11:58.03 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 11:58.06 | wdoekes | I'd start there |
| 12:02.38 | *** join/#asterisk TechAdam (c27dc475@gateway/web/freenode/ip.194.125.196.117) |
| 12:03.47 | cargill | wdoekes: thanks, the load/unload pointed me to DNS, the names were not resolvable but the system I was basing it on had the names specified in /etc/hosts |
| 12:04.05 | TechAdam | hey guys, if a person tries to call me but gets 30seconds of silence, followed by a busy signal, that suggests its a problem on their end, right? |
| 12:07.25 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 12:09.26 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.76.92.218) |
| 12:10.10 | Greek-Boy | I'm begging for someone to give an example of their Polycom sip.cfg whereby distinctive ringtones are confirmed to be working. Please... I have tried every working example I can find on the net and dont know where I am going wrong. :-( |
| 12:11.38 | Chainsaw | Greek-Boy: Need to know whether you are on SIP or UCS firmware. |
| 12:12.04 | Greek-Boy | I'm on sip |
| 12:12.40 | Greek-Boy | the firmware came packed in "SoundPoint_IP_SIP_3_2_7_release_sig_combined.zip" |
| 12:15.06 | Greek-Boy | but there was no bootblock file in that zip, there is just sip.ld and sip.ver. The phone doesn't seem to upgrade any firmware. The bootblock on the phone is 3.0.2.0024. Perhaps that's the problem? Old firmware? |
| 12:17.30 | *** join/#asterisk asterisco (~eth@87.18.111.33) |
| 12:17.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:17.37 | asterisco | hi at all |
| 12:18.02 | asterisco | i'm a new user of asterisk... |
| 12:18.20 | Greek-Boy | Chainsaw: SIP firmware on the phone is 3.3.3.0069 |
| 12:18.42 | asterisco | ...i've installed asterisk in a debian, and i could cal from internal numbers |
| 12:19.05 | asterisco | i've buy Grandstream gxw4104 |
| 12:19.13 | asterisco | i've configured it |
| 12:19.31 | asterisco | but when i try dial a external number i obtain this |
| 12:19.43 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 12:19.53 | asterisco | [Jun 11 14:11:00] WARNING[5145]: chan_sip.c:18036 handle_response_invite: Received response: "Forbidden" from '"Giuseppe Cell" <sip:101@10.0.1.111>;tag=as3233f385 |
| 12:19.59 | asterisco | someone could help me? |
| 12:20.18 | Greek-Boy | Chainsaw: Am I supposed to be getting the firmware from here? http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 12:20.29 | Greek-Boy | because that's under "UC" firmware |
| 12:22.06 | Chainsaw | Greek-Boy: I only have sample configs for UCS firmware. |
| 12:22.32 | Greek-Boy | ok. Will you pastebin it for me? |
| 12:22.35 | Chainsaw | Greek-Boy: But perhaps others still use SIP. Try Kobaz. |
| 12:23.48 | [TK]D-Fender | asterisco: Didn't I ask you to show both sides of your config a day or two ago? |
| 12:24.08 | asterisco | yes excuse me |
| 12:24.16 | asterisco | i put all in pastbin |
| 12:24.22 | asterisco | and pastpic |
| 12:27.06 | Chainsaw | Greek-Boy: There is no point in me putting UCS examples on pastebin for a SIP firmware deployment. |
| 12:27.11 | Chainsaw | Greek-Boy: It will not work. |
| 12:28.30 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 12:29.45 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 12:30.56 | Greek-Boy | Chainsaw: I understand. Could you point me to the latest SIP firmware for SoundPoint® IP 321/331? On http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip321_331.html#download I can not find the pack which is for my current firmware of 3.3.3.0069. |
| 12:31.35 | Greek-Boy | the only version that matches has the "UC" i nit |
| 12:31.46 | Chainsaw | Greek-Boy: That hardware is out of support, SIP firmware may be your only choice. |
| 12:32.10 | jaflong | jaf |
| 12:32.36 | [TK]D-Fender | Greek-Boy: the US *IS* SIP |
| 12:32.46 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 12:32.51 | [TK]D-Fender | UC |
| 12:33.06 | [TK]D-Fender | 4.1.1 Combined <------------- |
| 12:33.29 | Greek-Boy | [TK]D-Fender: Oh I se... Thank you for clearing that. It seems I'm getting confused with UC/UCS |
| 12:33.32 | [TK]D-Fender | And all the version below are also there |
| 12:33.40 | Chainsaw | Greek-Boy: At any rate, the config you need is http://pastebin.com/wsQLv6Sf |
| 12:33.54 | Chainsaw | Greek-Boy: That works on PolycomSoundPointIP-SPIP_670-UA/4.0.5.4853 |
| 12:34.04 | Greek-Boy | according to Polycom i *must* use it with Lync only |
| 12:34.06 | Chainsaw | Greek-Boy: If you run anything older it likely will not work. |
| 12:35.12 | Greek-Boy | Chainsaw: I am assuming alterinfo also needs to be configured or do I just send the name of the tone on altertinfo sip header? |
| 12:35.16 | [TK]D-Fender | My bad on there, go a version below |
| 12:36.03 | Chainsaw | Greek-Boy: SIPAddHeader(Alert-Info: Chime) |
| 12:36.05 | Chainsaw | Greek-Boy: As an example. |
| 12:36.59 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:37.19 | [TK]D-Fender | UC software 4.1.X Rev B is for use only with Microsoft® Lync® Server and using a single line registration. UC software 4.1.0 Rev B supports the following Polycom devices: SoundPoint IP 321, 331, 335, 450, 550, 560, 650, SoundStation IP 5000, Duo and VVX 500 phones. |
| 12:40.41 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 12:41.57 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:43.38 | jaflong | has anyone used gstreamer to send video rtp to asterisk? |
| 12:45.49 | *** join/#asterisk cw1972_ (~cw1972@145.255.240.32) |
| 12:47.15 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 12:49.41 | *** join/#asterisk marceloa1 (~Thunderbi@189-90-192-72.isimples.com.br) |
| 12:50.44 | [TK]D-Fender | jaflong: And what is it doing for call setup? |
| 12:52.26 | jaflong | call setup is done with sip and sdp |
| 12:52.52 | *** join/#asterisk alami (~ialami@unaffiliated/alami) |
| 12:52.57 | [TK]D-Fender | From gstreamer? |
| 12:53.28 | *** join/#asterisk jepperl (~jepperl@213.150.61.162) |
| 12:54.04 | asterisco | [TK]D-Fender: take a llook, i've putted all information here http://www.in-tv.it/gxw4104/ |
| 12:54.22 | jaflong | scripting is used to generate the command parameters and start gstreamer to send to rtp udp |
| 12:54.34 | asterisco | all config screen captur and sip.conf config.conf |
| 12:54.54 | *** join/#asterisk marceloamorim (~Thunderbi@189-90-192-72.isimples.com.br) |
| 12:55.50 | *** join/#asterisk gusto (~gusto@178.143.0.10) |
| 12:56.05 | *** join/#asterisk SGjunior (~sgjunior@out-pq-224.wireless.telus.com) |
| 12:57.07 | [TK]D-Fender | asterisco: https://docs.google.com/file/d/0B4tW1jaeiI6_UWJjMUtMYlQtX2s/edit |
| 12:57.13 | Katty | morning |
| 12:57.17 | [TK]D-Fender | asterisco: I don't see any of these users created in your sip.conf |
| 12:57.42 | asterisco | ehmm excuse me |
| 12:57.50 | asterisco | i must delete it |
| 12:58.06 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 12:58.19 | asterisco | in this guide: https://www.grandstream.com/products/gxw_series/gxw410x/documents/gxw410x_interop_asterisk.pdf |
| 12:58.28 | asterisco | i've used |
| 12:58.38 | [TK]D-Fender | asterisco: The settings you just showed me don't match |
| 12:58.42 | asterisco | Configure the GXW410x to function as a PEER gateway (no SIP accounts are |
| 12:58.42 | asterisco | needed; only used for one-stage dialing). |
| 12:58.52 | [TK]D-Fender | the GS has users defiend and you are not making * match |
| 12:59.08 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 12:59.12 | asterisco | yes... |
| 12:59.13 | asterisco | ...but i've do a lot of try |
| 12:59.32 | asterisco | now i delete 9001 and 9002 |
| 12:59.32 | asterisco | but i think that the problem is the same |
| 13:01.03 | asterisco | i've deleted it |
| 13:01.11 | asterisco | but there is the same problem |
| 13:01.25 | *** join/#asterisk Ulrar (~Ulrar@luwin.ulrar.net) |
| 13:01.39 | Ulrar | Hi, in the extensions.conf, how do I match numbers from 1000 to 1045 ? |
| 13:01.46 | Ulrar | would 1000-1045 work ? |
| 13:02.08 | asterisco | https://docs.google.com/file/d/0B4tW1jaeiI6_RkVQQTl4aWZoTlE/ |
| 13:02.44 | asterisco | [TK]D-Fender: take a look |
| 13:02.53 | asterisco | the asterisk error is the same |
| 13:03.06 | asterisco | [Jun 11 15:00:42] WARNING[5351]: chan_sip.c:18036 handle_response_invite: Received response: "Forbidden" from '"Giuseppe Cell" <sip:101@10.0.1.111>;tag=as043bc590' |
| 13:03.53 | *** join/#asterisk imox (~imox@p57A97874.dip0.t-ipconnect.de) |
| 13:04.02 | [TK]D-Fender | asterisco: https://docs.google.com/file/d/0B4tW1jaeiI6_YTRWUlJZUnNpMjQ/edit <- these IP's don't look right for your server... |
| 13:04.08 | file | raises eyebrow |
| 13:04.31 | [TK]D-Fender | Ulrar: No. pattern matching is on a per-character basis |
| 13:04.39 | asterisco | gulp |
| 13:04.43 | [TK]D-Fender | Ulrar: It's all in the book.... |
| 13:04.45 | asterisco | i'm an idiot |
| 13:04.47 | Ulrar | So there is no way to make what I want ? |
| 13:04.47 | asterisco | !!!! |
| 13:04.56 | asterisco | i try to change ip :) |
| 13:05.00 | asterisco | 10.0.1.111 |
| 13:05.03 | [TK]D-Fender | Ulrar: It would take more than 1 pattern |
| 13:06.10 | file | I wonder if exten => _10[0-4][0-5] would work |
| 13:06.30 | file | tries not to remember the PBX core |
| 13:06.57 | [TK]D-Fender | file: nope... |
| 13:07.00 | asterisco | [TK]D-Fender>: WORKKKKKK!!!!!!!!!!!!!!! |
| 13:07.02 | asterisco | Fineeee |
| 13:07.05 | asterisco | thank you |
| 13:07.29 | asterisco | is from two days that i'm crazing for an error so stupid!!! |
| 13:07.29 | Ulrar | 10[0-3]X and 104[0-5] ? |
| 13:07.35 | [TK]D-Fender | Ulrar: yes |
| 13:07.44 | Ulrar | Let's go with that then, thanks |
| 13:07.49 | asterisco | [TK]D-Fender: is possible change caller id? |
| 13:07.53 | [TK]D-Fender | Ulrar: You're welcome |
| 13:07.54 | file | oh, literal |
| 13:07.57 | file | yeah. |
| 13:07.58 | [TK]D-Fender | asterisco: No. |
| 13:08.07 | asterisco | :( |
| 13:08.10 | asterisco | oook |
| 13:08.13 | asterisco | is the same |
| 13:08.14 | asterisco | :D |
| 13:16.03 | asterisco | [TK]D-Fender: if i've a telephon number that i use with gxw4104 |
| 13:16.16 | asterisco | for exemple 095505050 |
| 13:16.24 | [TK]D-Fender | You don't use a "nuimber" with it... it uses your lines. |
| 13:16.32 | [TK]D-Fender | it knows nothing about "phone number" |
| 13:16.35 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-wvzsammsnjgdzcxk) |
| 13:16.54 | asterisco | c? |
| 13:17.43 | asterisco | i can't obtain redirect in an internal directly call 0955050501 |
| 13:17.44 | asterisco | ? |
| 13:17.58 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 13:18.04 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 13:19.08 | [TK]D-Fender | your wording is not clear. Reqphrase your question |
| 13:19.13 | asterisco | ok |
| 13:19.50 | asterisco | i would contact an internal phone directly call my telephon number+internal number |
| 13:20.03 | asterisco | my telefpn number is 123456 |
| 13:20.07 | [TK]D-Fender | Still not making any sense |
| 13:20.08 | asterisco | my internal is 1 |
| 13:20.36 | asterisco | i would that if someone call 1234561 ring only my internal number |
| 13:21.03 | asterisco | if someone call 1234562 ring another internal number |
| 13:21.09 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 13:21.37 | [TK]D-Fender | there is no "number" when they dial. This is dumb analog. your line will ring and if you set the rest up right it will send the call to Asterisk. |
| 13:21.42 | [TK]D-Fender | Go LOOK at what that call looks like |
| 13:24.37 | billu | Hi all, I have Asterisk 11.6-cert1 with freepbx 2.11 on ubuntu 10.04 have increased translation time e.g. ulaw->gsm 15000 . i am facing voice quailty issue any suggestion to improve it??? |
| 13:25.25 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 13:25.25 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:25.25 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:27.10 | Ulrar | [TK]D-Fender: 10[0-3]X should match 1001 for example ? Doesn't seem to work |
| 13:27.32 | [TK]D-Fender | You'd have to show the whole thing... |
| 13:27.56 | Ulrar | rejected because extension not found in context developers |
| 13:28.15 | [TK]D-Fender | your whole config line, enough of that context to prove its condition, etc |
| 13:28.33 | [TK]D-Fender | Enough of that error to prove what it was looking for. |
| 13:28.42 | Ulrar | http://pastebin.com/ikLnnX24 |
| 13:28.54 | [TK]D-Fender | Ulrar: that isn't a pattern at all |
| 13:29.03 | [TK]D-Fender | Ulrar: Because you forgot the most important part |
| 13:29.04 | Ulrar | [Apr 25 17:59:27] NOTICE[14374]: chan_sip.c:23999 handle_request_invite: Call from '1000' (192.168.48.1:25488) to extension '1001' rejected because extension not found in context 'developers'. |
| 13:29.15 | [TK]D-Fender | Ulrar: _ <------------------- |
| 13:29.19 | Ulrar | Right |
| 13:29.55 | Ulrar | Thanks :) |
| 13:30.21 | [TK]D-Fender | Ulrar: Dial(SIP/${EXTEN}, 160) <- also, never shove extra spaces inside of app parameters unless required |
| 13:31.29 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 13:32.00 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:33.20 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 13:36.45 | *** join/#asterisk Nivex (~kjotte@atlantis.home.nivex.net) |
| 13:42.44 | Katty | weeeeee |
| 13:43.51 | [TK]D-Fender | ~whee |
| 13:43.51 | infobot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 13:46.39 | *** join/#asterisk n3hxs (~Ed@pool-96-245-157-123.phlapa.fios.verizon.net) |
| 13:50.04 | *** join/#asterisk bhavikpatel6842 (~root@122.169.25.78) |
| 13:50.10 | bhavikpatel6842 | Hi All |
| 13:50.31 | bhavikpatel6842 | Is there any way to generate multiple call using originate command ? |
| 13:50.54 | bhavikpatel6842 | I have more then 2 no and trying to call all the number at a same time. |
| 13:51.12 | bhavikpatel6842 | I know about SIP/1234&SIP/34334 |
| 13:51.42 | bhavikpatel6842 | but this works like one extension pick other extension is hangup.i want all answer. |
| 13:51.49 | bhavikpatel6842 | Can any one suggest me ? |
| 13:53.20 | [TK]D-Fender | Those should obviously be MULTIPLE speatare Orignated channels |
| 13:53.36 | [TK]D-Fender | You want them acting independently? Then pace them independently |
| 13:53.47 | bhavikpatel6842 | like ? |
| 13:54.24 | [TK]D-Fender | 2 originates. |
| 13:54.33 | bhavikpatel6842 | $ASManager->Originate("SIP/1234",'', $context, '1',NULL, NULL,$TIMEOUT,'', NULL, 1,NULL); |
| 13:54.53 | bhavikpatel6842 | in this command how I can manage ? |
| 13:55.00 | [TK]D-Fender | that's ONE Originate... |
| 13:55.17 | bhavikpatel6842 | so multipal originate command ? |
| 13:55.19 | [TK]D-Fender | [09:54][TK]D-Fender2 originates. |
| 13:55.47 | bhavikpatel6842 | but I have issue like if one originate calls calls is flow not getting back to second one. |
| 13:56.28 | [TK]D-Fender | huh? |
| 13:56.29 | bhavikpatel6842 | I used PHP For loop and try to call originate command multiple times but only one time calls. |
| 13:56.54 | [TK]D-Fender | I can originate a hundred calls back to back instantly. |
| 13:57.01 | [TK]D-Fender | This isn't an Asterisk issue |
| 13:57.14 | bhavikpatel6842 | I used asterisk 11.10 |
| 13:57.22 | [TK]D-Fender | Version doesn't matter |
| 13:57.29 | bhavikpatel6842 | manager 1.3 |
| 13:57.54 | bhavikpatel6842 | give me some idea |
| 13:58.10 | bhavikpatel6842 | how I can manage such kinda? |
| 13:58.17 | [TK]D-Fender | idea : your code is wrong somehow and we can't see it all so we have no idea where you went wrong |
| 14:00.02 | *** join/#asterisk serafie (~erin@nat/digium/x-webelorobkdgtdbj) |
| 14:00.49 | bhavikpatel6842 | ok thanks |
| 14:00.54 | *** part/#asterisk bhavikpatel6842 (~root@122.169.25.78) |
| 14:02.43 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 14:05.37 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zaxfiadfdxmzavim) |
| 14:05.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:08.55 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 14:09.12 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 14:14.30 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
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| 14:19.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:26.49 | Ulrar | With asterisk 10.12, an exten with just answer and setmusiconhold should play the music, not hangup, right ? |
| 14:27.42 | Ulrar | Ha, I need MusicOnHold() |
| 14:27.48 | Ulrar | Sorry for the noise |
| 14:43.24 | *** join/#asterisk salz212 (~chatzilla@203.215.176.22) |
| 14:43.46 | *** join/#asterisk Rico (~rico@unaffiliated/rico29) |
| 14:43.48 | Rico | hello |
| 14:44.08 | Rico | is that normal that, il 1.8.25.0 release, subscribecontext is not defined in realtime mysql scheme ? |
| 14:47.34 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 14:48.58 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 14:50.57 | [TK]D-Fender | Could be. Try adding it and testing |
| 14:52.17 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:52.40 | *** join/#asterisk billu (b6444534@gateway/web/freenode/ip.182.68.69.52) |
| 14:57.16 | *** join/#asterisk raspberrypifan (~textual@c-98-254-226-109.hsd1.fl.comcast.net) |
| 15:03.53 | salz212 | hi, I am looking to have asterisk voicemail work with directory structure but with DB as well, like I do not want to store voicemails in DB but I need the information of current voicemails for a user in DB, also if someone delees his.her voicemail it also gets effected in DB. I am wondering if there is a build-in functionality of such type (currently I don't see any built in on forums and... |
| 15:03.55 | salz212 | ...mailing list) |
| 15:04.21 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:04.39 | [TK]D-Fender | No, you only get to use one module or the other, not both |
| 15:05.03 | [TK]D-Fender | If you want some info in a DB you'll have to make your own to poll the files. |
| 15:05.55 | salz212 | yes but that would be expensive to pool the entire storage if the mailboxes are more lets say 1000s |
| 15:07.48 | [TK]D-Fender | Do it per-box in a post voicemail/voicemailmain script |
| 15:11.35 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:15.10 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-ydhtbpixxkijdwih) |
| 15:15.19 | salz212 | but on deletion either way i'll be needing to have to check on storage. No? and moving from INBOX to OLD or any such thing |
| 15:16.55 | [TK]D-Fender | [11:07][TK]D-FenderDo it per-box in a post voicemail/voicemailmain script <---- |
| 15:18.30 | *** join/#asterisk raspberrypifan (~textual@c-98-254-226-109.hsd1.fl.comcast.net) |
| 15:20.29 | *** join/#asterisk fireman_biff (~biff@204.212.242.57) |
| 15:22.32 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 15:26.01 | fireman_biff | Hi all, I can see IAX2 traffic coming into a PBX using tcpdump but asterisk shows no output with iax2's debug on. I've reloaded chan_iax2.so already, and asterisk is definitely listening on port 4569. I know that restarting the PBX will solve the problem, but is there anything else I can try that won't result in downtime? |
| 15:26.25 | *** join/#asterisk x86 (~x86@i.am.leet.org) |
| 15:37.52 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 15:38.23 | x86 | [Jun 11 11:35:16] ERROR[4332][C-00000000]: chan_motif.c:821 jingle_add_google_candidates_to_transport: Unable to add Google ICE candidates as ICE support not available or no candidates available |
| 15:38.47 | x86 | no matter what I do, I can't get this to go away... icesupport=yes in rtp.conf |
| 15:39.24 | x86 | this is when Asterisk tries to dial out via Google Voice (chan_motif) |
| 15:39.46 | Qwell | First: Google Voice over XMPP is dead. |
| 15:39.59 | Qwell | Second: Did you build with pjproject installed? |
| 15:41.01 | x86 | so, asterisk + google voice is not possible anymore? |
| 15:41.51 | x86 | no pjproject, unless it's included with asterisk's source distribution, and compiled by default... although I don't remember seeing anything mentioning pjproject in menuselct |
| 15:41.59 | x86 | menuselect* |
| 15:42.44 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-pppfobhdsfssyjre) |
| 15:42.44 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:44.11 | x86 | if gvoice is no longer supported, I suppose it's not worth troubleshooting eh? :P |
| 15:46.49 | *** join/#asterisk Mango45 (~Mango45@d209-89-208-218.abhsia.telus.net) |
| 15:50.32 | Qwell | x86: Officially, Google Voice was supposed to no longer work, as of a few weeks ago. |
| 15:50.52 | Qwell | Whether the servers and such are still running is a whole other issue. In short, no, it's not worth troubleshooting. |
| 15:51.02 | x86 | in that case, who are people recommending for cheap (free?) domestic US termination? and origination? |
| 15:51.31 | Mango45 | Anveo Direct is pretty cheap, but they do IP auth only. |
| 15:51.47 | x86 | home use only, maybe 10-60mins usage/mo |
| 15:52.28 | x86 | Mango45: FiOS gives me a single, dynamic IP |
| 15:53.50 | Mango45 | Localphone is pretty cheap. http://www.localphone.com/call/united_states |
| 15:54.04 | Mango45 | Actually it looks like their rates aren't far off Anveo Direct |
| 15:54.32 | x86 | I wonder also if Google will allow me to port the DID they issued me to another carrier |
| 15:54.36 | Mango45 | CallWithUs allows direct audio between you and the carrier, if you like doing that. |
| 15:54.52 | Mango45 | They will - https://support.google.com/voice/answer/1316844?hl=en |
| 15:56.42 | x86 | nice |
| 15:57.16 | Mango45 | Question, how can I suppress output when doing a dialplan reload? |
| 15:57.55 | x86 | $0.75 per month for 250 minutes.... how would that even be worth paying the credit card transaction fees? |
| 15:57.56 | Qwell | Mango45: turning off verbosity can reduce it |
| 15:57.58 | x86 | :) |
| 15:59.08 | x86 | Mango45: not possible to fully suppress it that I'm aware of... |
| 15:59.36 | Mango45 | Qwell: I have a cron job that does rasterisk -x "core set verbose 0"; rasterisk -x "dialplan reload". Unfortunately I still see the output. |
| 15:59.59 | Mango45 | However if I type those commands into the CLI, it works. Trying to figure out how to automate it. |
| 16:00.05 | Qwell | Don't do that. |
| 16:00.14 | x86 | yeah that's horrible |
| 16:00.21 | Mango45 | Ok, what should I do instead? |
| 16:00.25 | Qwell | Ever. Please stop. file kills a kitten every time Asterisk gets reloaded by a cron job. |
| 16:00.41 | file | moo |
| 16:00.47 | Mango45 | I didn't reload Asterisk, just the dial plan. |
| 16:00.49 | file | I don't mind that |
| 16:00.54 | Qwell | file: diaf |
| 16:00.56 | x86 | Mango45: what keeps turning verbose on? easier solution would be to isolate and eliminate the cause, not the symptom :) |
| 16:01.12 | Mango45 | I need it for other purposes. :) |
| 16:01.31 | Qwell | There is no reason you need to reload with any frequency. |
| 16:01.38 | x86 | right |
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| 16:02.32 | Mango45 | Qwell: I have a file of variables that changes frequently. What do you suggest I do differently? |
| 16:02.46 | Qwell | Don't change them frequently. |
| 16:02.57 | x86 | Mango45: http://www.localphone.com/prices/monthly_subscription/US_ML_250 |
| 16:03.18 | Mango45 | Qwell: Then the PBX would not perform its intended function. |
| 16:03.20 | x86 | key point: This service requires you to dial a local landline number, that means you may also pay your phone provider local charges. ;) |
| 16:03.37 | Qwell | Then the intended function is implemented improperly. |
| 16:03.37 | file | reload all the things, every second |
| 16:03.47 | Mango45 | sighs |
| 16:04.06 | Mango45 | x86: http://www.localphone.com/services/voip |
| 16:04.25 | Nivex | Mango45: sounds like you need realtime. |
| 16:04.31 | x86 | Mango45: perhaps you want realtime |
| 16:05.09 | Mango45 | x86: Seems overkill for a 70-line text file. |
| 16:05.23 | x86 | Mango45: alternatively, you could store the variables in AstDB (or ODBC, or ...), and use a dialplan application to query the DB |
| 16:05.27 | Nivex | you could also potentially leverage the database and use remote commands to update it: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Internal+Database |
| 16:05.34 | x86 | Nivex: lol |
| 16:05.37 | Nivex | x86: we seem to be on the same wavelength |
| 16:05.55 | Qwell | Well, it's a solved problem, so... |
| 16:08.48 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 16:10.09 | Nivex | I'm migrating my network to IPv6, and one of the laggards are my VoIP devices. I have two ATAs and a hardphone. Any recommendations for IPv6 capable kit to replace them? |
| 16:11.23 | pabelanger | I think snom has ipv6 support |
| 16:11.35 | pabelanger | but, I've never used it. |
| 16:11.55 | Nivex | Snom does seem to be coming up most in my searches. Don't know anything about them either. |
| 16:12.18 | x86 | Cisco devices support IPv6 fairly well with SCCP anyway... I've not tried them with SIP (or with anything besides CUCM) |
| 16:12.35 | x86 | Nivex: I've heard only good things about Snom |
| 16:20.56 | Nivex | http://wiki.snom.com/FAQ/Which_phones_are_IPv6_capable%3F That's pretty much most of their lines |
| 16:21.21 | Nivex | the m9r (DECT cordless) appears to be the only thing capable |
| 16:21.56 | Qwell | Nivex: poorly written page. note the period. |
| 16:22.38 | Qwell | Looks like it's saying 7xx and 8xx are IPv6-capable. 3xx, PA1, and Meetingpoint are not. |
| 16:22.49 | Nivex | ah, yes, it was hiding :) |
| 16:23.02 | Qwell | yeah, switching to German makes it more clear. |
| 16:23.09 | Qwell | Alle Telefone aus der 7xx und 8xx-Reihe unterstützen IPv6. (ab Firmware-Version 8.7.5.8) |
| 16:23.28 | Qwell | even if you don't speak it, it's pretty obvious what it's saying. |
| 16:23.39 | Nivex | Ach! Sehr gut! Danke. |
| 16:24.48 | Nivex | Ich hab zwei jahr im oberschule Deutsch gelernt, aber das war zwanzig jahre ... uhh... ago. |
| 16:25.33 | Qwell | Mein Deutsch ist sehr Flugzeug... |
| 16:25.52 | x86 | #myplanet sprechen zie Deutsch :) |
| 16:26.04 | Nivex | That's fine if you're trying to talk to a flight attendant (which one of my classmates went on to become). |
| 16:26.17 | x86 | mein Deutsch ist ein schlampe fotze ;) |
| 16:26.32 | Nivex | aaaand x86 just lost me |
| 16:26.43 | x86 | eine* (?) |
| 16:26.44 | Qwell | x86: Very nice. |
| 16:26.52 | x86 | Qwell: haha |
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| 17:47.32 | billu | hi all, asterisk 11.6 "core show translation" have increased value (ulaw->gsm 15000) any ideahow to fix????? |
| 17:52.05 | newtonr | billu, What are you really trying to fix? |
| 17:52.40 | newtonr | billu, that is a very fast translation time. The same as on my machine and probably many machines. Are you having audio issues or another problem? |
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| 18:00.52 | *** join/#asterisk CGMChris (~CGMChris@64.233.255.242) |
| 18:01.36 | billu | ya I have audio issue low voice quailty on bridge channels |
| 18:02.24 | CGMChris | I recently upgraded to Asterisk 11 and now my Directory() application is not able to locate anyone's name. "voicemail show users" lists out all of the extensions and names correctly, and the contexts listed match the one I am passing to Directory(). Are there any known issues w/ this or did something change in v11? |
| 18:05.09 | newtonr | billu, have you tried different codecs and different phones to see if you can narrow down when it occurs? Also what do you mean by bridge channels? |
| 18:06.05 | *** join/#asterisk JonathanDavis (~solyce@cpe-98-28-193-254.woh.res.rr.com) |
| 18:06.26 | JonathanDavis | asterisk noob |
| 18:06.40 | JonathanDavis | I cannot get asterisk 12.2 to call out using google voice |
| 18:06.42 | billu | i mean by bridge i dial the number and on answer of this i again dial the other number and both communicate |
| 18:06.45 | JonathanDavis | errr sorry |
| 18:06.58 | JonathanDavis | I cannot get asterisk 12.2 to receive calls using google voice |
| 18:07.03 | JonathanDavis | inbound |
| 18:07.08 | JonathanDavis | I think that’s the term |
| 18:07.32 | JonathanDavis | I’ve got like 15 v’s in the command-line to load up the daemon |
| 18:08.05 | newtonr | billu, I didn't understand. A dials B, B answers, and then what happens? B dials ? |
| 18:09.11 | JonathanDavis | When I call to GV number it does ring my SIP phone… I can answer the SIP phone, but the calling phone continues to “ring”… |
| 18:09.25 | JonathanDavis | no connection is ever established |
| 18:09.26 | asterisco | how can i choice an out going line randomly |
| 18:09.28 | asterisco | = |
| 18:09.30 | asterisco | ? |
| 18:09.51 | JonathanDavis | Not sure if it is a NAT issue |
| 18:10.29 | [TK]D-Fender | asterisco: "core show function RAND" |
| 18:10.33 | JonathanDavis | IN the console I see: Called SIP/101 |
| 18:10.39 | JonathanDavis | SIP/101-00000007 is ringing |
| 18:10.53 | JonathanDavis | SIP/101-00000007 answered Gtalk/+############-a### |
| 18:10.56 | *** join/#asterisk aness (~aness@cm-84.215.80.229.getinternet.no) |
| 18:10.59 | billu | <PROTECTED> |
| 18:11.04 | JonathanDavis | Sending DTMF '1' to the calling party |
| 18:11.19 | newtonr | JonathanDavis, GV dropped xmpp support a couple weeks ago |
| 18:11.36 | JonathanDavis | ahh |
| 18:11.48 | JonathanDavis | and that’s it huh |
| 18:11.53 | JonathanDavis | but I can still make calls out with it? |
| 18:12.20 | newtonr | JonathanDavis, you can google around to see the news on it... no idea how it will actually behave now |
| 18:12.36 | newtonr | i'd assume that it won't work as previously expected |
| 18:12.46 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 18:14.11 | billu | we have PSTN gateway which supports ulaw so i test my stuff with ulaw. my same set up is working well with asterisk version 1.8 |
| 18:14.14 | newtonr | billu, I don't know why you would get "low voice" in that specific scenario. I'd try out different phones and codecs to see where it occurs consistently. |
| 18:16.12 | CGMChris | [TK]D-Fender, heard anything about any problems with Directory() not working in Asterisk 11? |
| 18:17.28 | [TK]D-Fender | CGMChris: No... but you haven't really shown us the issue so I wouldn't go assuming it's a bug. |
| 18:17.58 | billu | @newtorn i seen only difference in translation in asterisk 1.8 and asterisk 11. other wise all debug logs are same |
| 18:19.59 | newtonr | billu, using same phones, codecs and sip.conf configuration in both versions? |
| 18:21.13 | billu | ya exactly same thing |
| 18:22.01 | newtonr | billu, are you running 1.8 and 11 on different hardware? |
| 18:23.35 | billu | Both server are hosted on VMware and on ubuntu-10.04 server |
| 18:23.50 | CGMChris | [TK]D-Fender, While preparing to do a pastebin, I just realized that the voicemail.conf I was using (a template that shipped with asterisk 1.4) had 2 default sections, one as an example and one I have been using all along. Apparently how that discrepency is handled has changed at some point. In any case, fixed. |
| 18:25.04 | [TK]D-Fender | Any reference to 1.4 = automatic user-error :) |
| 18:26.14 | newtonr | billu, if you can reproduce it on a third server, by installing 1.8, setting up your config and a test scenario for the call and then simply just installing 11, retesting and reproducing without changing config, phones or network; then I would file a bug. |
| 18:27.23 | newtonr | billu, otherwise. I have no idea :) |
| 18:29.49 | billu | @newtorn okey i have standalone server with asterisk 1.8 over ubuntu 10.04. i will test it first on same and then upgrade it to asterisk 11 and share the outcome with you |
| 18:32.26 | *** join/#asterisk KNERD (~KNERD@24.175.253.226) |
| 18:32.45 | malcolmd | translation times from 10 and forward aren't analogous to the translation times reported in 1.8 and prior, iirc |
| 18:34.00 | billu | #newtorn don't you have any reason to suspect on the result of "core show translation" for asterisk 1.8 and 11 seems have huge difference |
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| 18:42.44 | malcolmd | iirc, all of that was changed during the work to revamp formats in asterisk 10, so what's seen by >=10 isn't analogous to what was seen in <=1.8 |
| 18:44.50 | newtonr | billu, not necessarily with identical configuration. i do testing on the same hardware with identical configurations between 1.8 and 11 almost every day and I don't have any audio differences with ulaw/gasm |
| 18:45.37 | newtonr | malcolmd, ah cool |
| 18:45.52 | salz212 | [TK]D-Fender: As per your suggestion of doing db operation on post voicemail /voicemailmain to get voicemail data in db. My concern is how can I get the voicemail's moves, deleted and manipulated by users in voicemailmain app,.. like I do not see any channel variable. |
| 18:52.28 | [TK]D-Fender | salz212: You save the files & folders and compare them |
| 18:53.11 | [TK]D-Fender | salz212: There isn't going to be a variable. It would be just about insane to expect * to log these as vars for the 100 transactions a used could be doing in there |
| 18:56.47 | salz212 | [TK]D-Fender: Well files and folders are saved by asterisk itself, and manipulated as well by itself. I do not see a point to check storage directories in asterisk voicemail call, unless I am missing something. |
| 18:57.10 | billu | @newtonr I dont know but i with asterisk 11 have issue you know while using playback a gsm it sounds slow and drop or some noise, and same file on 1.8 plays clear |
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| 19:20.31 | mic_ | hello |
| 19:20.40 | mic_ | does anyone have any experience with SIP GSM gateways? |
| 19:21.00 | mic_ | I need one (4ch) and it would be fantastic to hear some recommendations. |
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| 19:22.53 | jmmills | can someone quickly explain to me in asterisk CEL records what the difference between cid_num, cid_ani, cid_rdns, and cid_dnid are? |
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| 19:31.24 | igcewieling1 | Asterisk 11.10.0 Has anyone had problems with Asterisk no longer logging verbose items to the log file? |
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| 19:40.10 | pabelanger | mic_, rumor is sangoma has some good gear for that |
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| 19:41.38 | mic_ | pabelanger: we are currently running on a G400E from OpenVox |
| 19:41.49 | mic_ | pabelanger: but I am not sure what is worse - the card or the driver |
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| 19:45.16 | Dovid | lol |
| 19:45.27 | Dovid | I never had luch with OpenVox |
| 19:45.34 | Dovid | any Asterisk dev's here/ |
| 19:45.42 | Dovid | . ?* |
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| 21:03.53 | burnbrighter | Can anyone tell me if the cisco presence patch wipes out your existing configurations? |
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| 21:13.22 | zopsi | mjordan: nice job with DPMA + pjsip. Really enjoying my new setup. |
| 21:15.55 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 21:15.57 | cj | moo |
| 21:16.42 | cj | can someone tell me how to match source address on inbound SIP request and direct to a particular context on match? |
| 21:16.56 | cj | gil, you about? |
| 21:17.03 | cj | carrar? |
| 21:18.19 | wdoekes | cj: core show function CHANNEL (look for recvip or something) |
| 21:18.30 | wdoekes | and 'core show application GotoIf' |
| 21:18.55 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 21:19.06 | cj | danke shön wdoekes |
| 21:19.41 | cj | schön? my german sucks so bad that I can't even tell you how to say that my german sucks in german. |
| 21:19.44 | mjordan | zopsi: thanks, it was a team effort. Sorry I missed your message yesterday, I guess that means you got it all set up and working? |
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| 21:21.25 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 21:25.56 | cj | how do I print something a value to the console ? |
| 21:26.00 | cj | is there a function for that? |
| 21:26.10 | cj | I looked through the list of functions and didn't see anything apropos |
| 21:30.52 | newtonr | cj, Verbose and NoOp |
| 21:31.08 | cj | ah, I didn't see NoOp, but I thought I remembered it existing. |
| 21:31.59 | cj | and exten => s,NoOp("${CALLERID(num)}") is a non-terminating rule, right? |
| 21:32.20 | newtonr | cj, it doesn't affect the executing channel at all |
| 21:32.26 | cj | ok, thanks |
| 21:32.32 | newtonr | No Operation is what it stands for |
| 21:32.35 | newtonr | i believe |
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| 21:32.50 | cj | oh, I'm sure. it's like the assembly instruction nop |
| 21:33.06 | cj | http://en.wikipedia.org/wiki/NOP |
| 21:33.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:33.13 | newtonr | cj, Verbose is a little nicer, since you can fit it into whatever verbosity levels you use |
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| 21:42.55 | cj | what's the operator to match a regular expression? |
| 21:43.06 | cj | <PROTECTED> |
| 21:43.34 | cj | I don't want to do = "303" but instead =~ /${customer0}/ |
| 21:44.13 | cj | opens the asterisk book |
| 21:44.17 | cj | wanders out to the shop first |
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| 21:50.04 | cj | good thing I know where my pdf copy is, I'm tellin' you what. |
| 21:51.12 | asterisco | what is the best international voip provider? |
| 21:51.30 | [TK]D-Fender | YES |
| 21:51.40 | [TK]D-Fender | ~best |
| 21:51.40 | infobot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
| 21:51.53 | asterisco | for international call |
| 21:52.06 | [TK]D-Fender | Iternation From whom? To whom? |
| 21:52.09 | asterisco | if you can give me some provider |
| 21:52.11 | [TK]D-Fender | every plan is different |
| 21:52.21 | asterisco | i know |
| 21:52.21 | [TK]D-Fender | ~itsplist-us |
| 21:52.21 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 21:52.44 | asterisco | and for italy? |
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| 21:54.58 | asterisco | ~itsplist-it |
| 21:56.35 | [TK]D-Fender | Not many italy-based providers mentioned here |
| 21:56.39 | [TK]D-Fender | Go google around |
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| 22:18.35 | burnbrighter | Can anyone tell me if the cisco presence patch wipes out your existing configurations? |
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| 22:27.25 | [TK]D-Fender | asterisco: Keeping in mind that just because you may find a provider BASED in Italy, they won't necessarily be a better provider, or off cheaper rates. |
| 22:28.20 | asterisco | you think that search a voip in Usa or in china is not a good idea? |
| 22:34.24 | [TK]D-Fender | I think you should read what I have told you again a few times... |
| 22:37.32 | TazzNZ | asterisco: also remember - you will be sending voip packets to them over the public internet - you have 0 control over the quailty or priority |
| 22:38.28 | TazzNZ | having said that - I currently do that with a provider and I havn't had issue....yet |
| 22:42.41 | file | the internet is a wild place |
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| 23:02.00 | f0ner00t | <PROTECTED> |
| 23:02.11 | f0ner00t | <PROTECTED> |
| 23:02.14 | f0ner00t | Sorry |
| 23:02.16 | f0ner00t | for the last link |
| 23:04.22 | [TK]D-Fender | answered in #freepbx |
| 23:04.34 | f0ner00t | Thank you sir |
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