| 00:01.06 | Mon|A|rch | igcewieling, if I find a tuned version that works, is there anyone that would want to see it? I feel like if I'm going to struggle with it, other people shouldn't have to |
| 00:03.27 | darkdrgn2k | <PROTECTED> |
| 00:03.39 | darkdrgn2k | so codec? |
| 00:03.46 | [TK]D-Fender | darkdrgn2k: So go deal with that |
| 00:03.48 | [TK]D-Fender | NOT CODEC |
| 00:04.15 | [TK]D-Fender | [18:52]darkdrgn2kis it complainig about codec? [18:53][TK]D-Fenderdarkdrgn2k: No. |
| 00:04.28 | [TK]D-Fender | darkdrgn2k: What part of this is unclear? |
| 00:04.32 | *** join/#asterisk stevetodd (~blurryrun@199.87.121.1) |
| 00:05.43 | igcewieling | darkdrgn2k: you have an auth issue |
| 00:06.41 | igcewieling | Mon|A|rch: There are a few things which have such high monetary value nobody gives them away. Answering Machine Detection is one of those. |
| 00:06.56 | Mon|A|rch | i see |
| 00:06.57 | igcewieling | I wish you the best of luck though |
| 00:07.06 | darkdrgn2k | igcewieling: 403: The server understood the request, but is refusing to fulfill it. |
| 00:07.06 | darkdrgn2k | <PROTECTED> |
| 00:07.24 | igcewieling | Mon|A|rch: there is incredible amounts of money to me made from telemarketing |
| 00:07.25 | Mon|A|rch | well, i actually just pasted some values from a forum, and it successfully detected the voicemail, and me picking up the phone and talking |
| 00:07.34 | Mon|A|rch | let's see if it can handle me not talking, and will just do it's job |
| 00:07.44 | Mon|A|rch | igcewieling, i see |
| 00:08.46 | igcewieling | darkdrgn2k: the version I looked at didn't say that, but good catch. |
| 00:09.12 | darkdrgn2k | :) yeh well i guess its back to the drawing board.. at least i have an idea where i should look |
| 00:09.20 | darkdrgn2k | [TK]D-Fender: thanx for the tought love |
| 00:09.27 | [TK]D-Fender | darkdrgn2k: Go read the RFC |
| 00:09.33 | Mon|A|rch | hm, but not when I don't say anything |
| 00:09.39 | Mon|A|rch | ugh |
| 00:11.03 | Mon|A|rch | so, what exactly does AMD do to detect this sort of thing? listen for silence and hangup if the silence is long enough? |
| 00:11.10 | Mon|A|rch | or some combination of silence, sound and beeps? |
| 00:11.36 | WIMPy | A beep should be a clear thning. |
| 00:12.07 | WIMPy | Otherwise it's about length of voice before silens and a good portion of luck. |
| 00:13.23 | Mon|A|rch | evidently people get better results by using background() beforehand? |
| 00:16.50 | *** part/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 00:37.49 | darkdrgn2k | i tried comparing a successfully call and the failed call.. |
| 00:38.21 | darkdrgn2k | only thingi i can see is one line missing |
| 00:38.22 | darkdrgn2k | a=rtpmap:8 PCMA/8000 |
| 00:39.36 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 00:39.37 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 00:41.50 | volga629 | Hello Everyone, I see in log "Using SIP RTP CoS mark 5". Is mark 5 equal priority 5 under lldp med configuration on cisco ? |
| 00:42.10 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 00:47.50 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 00:47.52 | saint_ | greetings all |
| 01:29.33 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:29.33 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:29.42 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 01:38.36 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:42.44 | *** join/#asterisk nc11235 (~textual@pool-71-97-75-217.dllstx.fios.verizon.net) |
| 01:45.09 | saint_ | what's the best way to have someone record sounds files remotely on asterisk ? |
| 01:52.22 | igcewieling | call an extension and leave voicemail. |
| 01:52.28 | igcewieling | where best=easiest |
| 01:57.15 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 02:04.19 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 02:04.49 | *** part/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 02:10.56 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 02:23.38 | ketas | saint_: setup recorder extension :P |
| 02:23.47 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 02:25.40 | ketas | ahahaha... someone called to my ip |
| 02:25.48 | ketas | i wonder if scammer |
| 02:26.26 | ketas | 00972599224230 |
| 02:26.31 | ketas | :) |
| 02:26.38 | deo | let me call you :D |
| 02:27.05 | ketas | :P |
| 02:27.38 | deo | but im not a scammer :p |
| 02:29.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 02:29.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:31.49 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 02:42.00 | *** join/#asterisk hegars (~hegars@061092248178.ctinets.com) |
| 02:45.13 | volga629 | what is mean Looking for 1 in from-internal |
| 02:45.16 | volga629 | <PROTECTED> |
| 02:48.52 | ketas | notices that question mark has some annoying leading space |
| 02:50.41 | [TK]D-Fender | volga629: It means it's looking for a match for 1 in [from-internal] ... just like it says |
| 02:54.16 | volga629 | yes I found thank you |
| 02:55.39 | volga629 | res_rtp_asterisk.c: RTP Read too short Is possible that can be cause by incorrect value on lldp med network policy, where priority ? |
| 02:57.36 | volga629 | the value was incorrect, I looked on asterisk log and set the same on switch |
| 02:57.41 | volga629 | Network policy 2 |
| 02:57.43 | volga629 | ------------------- |
| 02:57.45 | volga629 | Application type: voiceSignaling |
| 02:57.47 | volga629 | VLAN ID: 300 tagged |
| 02:57.49 | volga629 | Layer 2 priority: 5 |
| 02:57.51 | volga629 | DSCP: 46 |
| 02:58.46 | volga629 | In asterisk log I see Using SIP RTP CoS mark 5 |
| 03:00.12 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 03:06.18 | *** join/#asterisk hegars (~hegars@061092248178.ctinets.com) |
| 03:14.08 | *** join/#asterisk gajini (~gajini@61.12.12.132) |
| 03:15.33 | saint_ | is it possible to have a digium phone on a private network 192.x.x.x , to connect to an asterisk which is in another network 192.x.x.x through the internet , knowing that the ports 5060tcp/udp and 10000:20000 are forwarded on the 2nd network ? |
| 03:24.39 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 03:25.21 | *** join/#asterisk din3sh (~din3sh@41.136.86.189) |
| 03:25.27 | din3sh | hey all |
| 03:25.38 | din3sh | wct4xxp 0000:0b:08.0: TE210P: RECEIVE slip NEGATIVE on span 2------>help!!!!! |
| 03:29.24 | [TK]D-Fender | saint_: networking is networking.... |
| 03:29.58 | [TK]D-Fender | you can use IP p[hones over the internet. So unless your scenario has something else whacked about it you're not telling us.... it's all just packets as usual |
| 03:30.36 | ketas | saint_: a 192.*.*.* is not private :P |
| 03:31.14 | coppice | true, but it does have private parts :-) |
| 03:31.20 | hegars | lol |
| 03:31.20 | ketas | ew |
| 03:31.25 | ketas | ! |
| 03:31.59 | saint_ | ketas: 192.168.1.x i m sorry |
| 03:32.15 | ketas | saint_: usually connect such things via some vpn... then you don't need forwarding too |
| 03:32.19 | saint_ | so i have a phone in a 192.168.1.x network , and another phone in another 192.168.1.x |
| 03:32.30 | ketas | s/connect/you connect/ |
| 03:32.41 | saint_ | ketas: but we have no vpn here . |
| 03:32.50 | ketas | hm |
| 03:33.02 | ketas | go forward ports then |
| 03:33.02 | saint_ | so in one netwoork, i have digium ext 100 and asterisk |
| 03:33.13 | saint_ | on the other network (B), I have digium 200 |
| 03:33.30 | saint_ | on the network A , i forwarded 5060tcp/udp and 10k-20k to asterisk |
| 03:33.38 | saint_ | what else needs to be done ? |
| 03:33.38 | ketas | lot of fun with audio ports + double nat |
| 03:33.53 | saint_ | i can see the 200 registering to asterisk , but no audio is working |
| 03:34.01 | ketas | there you go |
| 03:34.17 | ketas | forward some ports into asterisk |
| 03:34.35 | din3sh | wct4xxp 0000:0b:08.0: TE210P: RECEIVE slip NEGATIVE on span 2------>1st span connected to E1, 2nd span connected to nortel |
| 03:34.42 | ketas | oh |
| 03:34.43 | saint_ | ketas: i forwarded 5060tcp/udp and 10k-20k to asterisk |
| 03:34.45 | ketas | hm |
| 03:35.01 | ketas | well, does asterisk even use them? |
| 03:35.04 | din3sh | am having timing slip between * and nortel on 2nd span |
| 03:35.34 | din3sh | ketas: ofc asterisk uses them |
| 03:35.39 | hegars | saint_, what the ups on bot the phones? |
| 03:35.56 | ketas | saint_ |
| 03:35.59 | hegars | saint_, what is the ip's on both of the phones? |
| 03:35.59 | ketas | :P |
| 03:36.12 | saint_ | hegars: they have private IPs (192.168.1.x) |
| 03:36.24 | ketas | saint_: also, where exactly does the audio go= |
| 03:36.26 | ketas | ? |
| 03:36.27 | hegars | phone one is say 10 |
| 03:36.33 | ketas | where did audio go... and bunny |
| 03:36.34 | hegars | the other is say 50 |
| 03:36.57 | hegars | look at the inside of one of the networks and you have another host with 10 |
| 03:37.01 | ketas | maybe you need directmedia=no too= |
| 03:37.03 | ketas | ? |
| 03:37.08 | hegars | if so you'll find all your voice packets going therer |
| 03:37.34 | saint_ | i have directmedia=no already |
| 03:37.46 | hegars | they each register to a different system tho don't they? |
| 03:37.48 | ketas | change range of one network |
| 03:37.52 | saint_ | hegars: no |
| 03:38.01 | hegars | oh |
| 03:38.04 | ketas | OR, put phone into it's own network or ip range |
| 03:38.08 | ketas | ! |
| 03:38.29 | saint_ | network A = 192.168.1.x , with 1 digium ext 100 , and 1 asterisk - Ports 5060tcp/udp and 10k-20k are forwarded from the public ip to asterisk |
| 03:38.55 | saint_ | network B = 192.168.1.x with 1 digium ext 200, registering to the public IP of network A |
| 03:39.06 | saint_ | that part works. when one call the other , it rings . |
| 03:39.13 | saint_ | it's when they pickup that it does not ring anymore. |
| 03:39.44 | hegars | yeah signalling and media are two different things |
| 03:40.07 | ketas | saint_: start tcpdumping to see where media goes |
| 03:40.08 | hegars | are you able to use IAX? |
| 03:40.23 | saint_ | hegars: i only have 1 asterisk |
| 03:40.25 | saint_ | so no iax |
| 03:40.31 | ketas | saint_: hack phone or something |
| 03:40.36 | saint_ | ketas: will do.. |
| 03:40.50 | saint_ | let me run tcpdump |
| 03:41.06 | hegars | what phones/clients are they? |
| 03:41.20 | saint_ | hegars: digium phones |
| 03:41.33 | hegars | they would have to support IAX |
| 03:42.31 | ketas | hahaha |
| 03:42.48 | hegars | huh really |
| 03:42.54 | hegars | thanks digium |
| 03:43.02 | ketas | phones talk iax? |
| 03:43.14 | hegars | looks like they don't |
| 03:43.16 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 03:43.35 | ketas | that would make it idx |
| 03:43.48 | hegars | still |
| 03:44.09 | saint_ | first of all, does it seem possible to do that ? or would I need to have a proxy that would have to handle everything ? |
| 03:44.19 | hegars | wouldn't take much to have a little arm chip running full blown asterisk |
| 03:44.27 | [TK]D-Fender | saint_: Your description is very bad. Start over |
| 03:44.45 | saint_ | [TK]D-Fender: I have 2 networks, A and B. Both are 192.168.1.x |
| 03:44.46 | [TK]D-Fender | saint_: You haven't told us how those subnets actually route to each other. |
| 03:44.56 | ketas | they don't :P |
| 03:45.10 | ketas | well, they use nat |
| 03:45.13 | saint_ | [TK]D-Fender: they are home network (actually fire companies), provided by comcast |
| 03:45.37 | saint_ | [TK]D-Fender: so in each building, we have a comcast router and gives us IP addresses 192.168.1.x |
| 03:45.47 | saint_ | In network A, I have 1 asterisk and 1 phone (digium) |
| 03:45.53 | saint_ | in network B I have 1 phone |
| 03:45.57 | [TK]D-Fender | saint_: Leave the phones and asterisk out of this |
| 03:46.09 | [TK]D-Fender | saint_: You have not establish how the subnets are linked |
| 03:46.38 | ketas | saint_: replace routers with machine with vpn & asterisk :P |
| 03:46.48 | ketas | problem solved |
| 03:47.02 | hegars | with discrete subnets |
| 03:47.08 | saint_ | [TK]D-Fender: they are not linked. |
| 03:47.47 | saint_ | [TK]D-Fender: on network B, I just give the public IP address of network A (router A) to the phone, to it can reach asterisk |
| 03:47.48 | ketas | if you wish, put different router there... maybe embedded, maybe pc |
| 03:48.21 | hegars | what is the ip registered on the peer tho in asterisk |
| 03:48.23 | [TK]D-Fender | saint_: and....? |
| 03:48.44 | saint_ | [TK]D-Fender: so the phone on network B can register on asterisk |
| 03:48.54 | [TK]D-Fender | saint_: Fine. What do you see on *? |
| 03:49.00 | saint_ | if we call each other, the phones ring |
| 03:49.00 | ketas | can you edit routing table of phone |
| 03:49.06 | saint_ | it's the voice that does not go through |
| 03:49.07 | ketas | damn, this leads nowhere... |
| 03:49.11 | [TK]D-Fender | saint_: Show us the configs and the call debug. |
| 03:49.21 | saint_ | [TK]D-Fender: stand by |
| 03:49.57 | saint_ | [TK]D-Fender: when you write phone config, you mean from sip.conf , or the config in the phone itself ? |
| 03:49.58 | ketas | saint_: can you repurpose asterisk machine as router? |
| 03:50.11 | ketas | looks like easiest solution |
| 03:50.24 | saint_ | ketas: not now.. but i can think about taking SmoothWall and try to work with it |
| 03:51.26 | [TK]D-Fender | saint_: sip.conf |
| 03:51.33 | saint_ | [TK]D-Fender: ok, hold on |
| 03:52.06 | ketas | saint_: well, i take it as you don't know how to setup asterisk + routing / nat into same machine? |
| 03:52.38 | ketas | this might give you better routing experience too maybe :P |
| 03:52.45 | saint_ | ketas: easy here yound jedi. i'm not a network guy nor an asterisk guy. i've been playing with asterisk for a month only and swallowed linux + php + apache at the same time. |
| 03:53.05 | saint_ | ketas: i think i did pretty good for now. |
| 03:53.11 | ketas | :) |
| 03:53.18 | [TK]D-Fender | saint_: configs. Debug. Now. |
| 03:53.29 | hegars | hahaha |
| 03:53.29 | saint_ | [TK]D-Fender: yeah yeah.. stand by please. |
| 03:53.38 | dpilon | hahaha..no small talk in between..time is money |
| 03:54.15 | dpilon | reminds me of xmovies when he asks for help then goes to lunch |
| 03:54.53 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 03:54.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 03:55.05 | saint_ | [TK]D-Fender: sip.conf : http://pastebin.com/ZuYqcLpX |
| 03:55.42 | [TK]D-Fender | saint_: You have done NOTHING that is required for Asterisk to work from behind NAT in there,. |
| 03:55.50 | ketas | thinks about how cats land always on paws |
| 03:56.16 | [TK]D-Fender | saint_: canreinvite=no <- this parameter doesn't even exist in 1.6+ |
| 03:56.18 | saint_ | [TK]D-Fender: i thought nat=yes was enough ? |
| 03:56.19 | hegars | missing nat=yes |
| 03:56.32 | saint_ | hegars: it is here, in the template |
| 03:56.35 | [TK]D-Fender | saint_: You are very mistaken |
| 03:56.44 | hegars | nope there it is |
| 03:57.11 | [TK]D-Fender | You have not defined your LOCALNET |
| 03:57.16 | [TK]D-Fender | You have not defined your EXTERNADDR |
| 03:57.30 | [TK]D-Fender | You did not set NAT=YES under [general] |
| 03:57.40 | [TK]D-Fender | You have not prevented reinvites with the PROPER parameter. |
| 03:58.12 | saint_ | [TK]D-Fender: gee you are killing me. is this explained anywhere in the book or on asterisk.org ? |
| 03:58.24 | [TK]D-Fender | sip.conf.sample <------- |
| 03:58.46 | [TK]D-Fender | needs to rehost his site |
| 03:59.11 | Kyosh | Hi. I am using Asterisk 1.4.21 in RealTime as a voice gateway. Dialplans, Extensions, Contexts are all stored in MySQL. Clients can easily call outbound to trunks and trunk providers are sending traffic to my clients without problems. I recently started working on Inbound and Outbound IVR functions with this system. Inbound IVRs' work fine. Outbound IVRs' (using AMI and an autodialer I wrote) do not seem to be working. The Aut |
| 04:00.04 | [TK]D-Fender | Kyosh: Your description is getting cut off. Dump it in a pastebin. Trying to paste a story his is not working for you (or us) |
| 04:00.15 | Kyosh | k |
| 04:00.27 | ketas | . The Aut |
| 04:00.41 | dpilon | that is where is ends |
| 04:01.41 | apb1963__ | Obviously he had planned to say "The Autobahn is where I like to drive". |
| 04:01.55 | Kyosh | [TK]D-Fender: http://pastebin.com/MY0bfRzT |
| 04:01.55 | ketas | hahaha |
| 04:02.48 | [TK]D-Fender | Kyosh: Your description on "not ebing played" is lacking.... |
| 04:02.51 | ketas | i remember i chose very short host for my irc bot so i could send longest possible lines |
| 04:03.02 | [TK]D-Fender | kyis the call TRYING to play them and you're just not getting audio? |
| 04:03.05 | Kyosh | [TK]D-Fender: how so? |
| 04:03.13 | Kyosh | oh |
| 04:03.21 | ketas | it was x.x.xx |
| 04:03.26 | apb1963__ | I had a short hostess once... she also had a flat head. |
| 04:03.36 | ketas | hah |
| 04:03.37 | Kyosh | he outbound ivr call, it dials, when the other side picks up, none of he prompts are played, the ivr just hangs up |
| 04:03.43 | Kyosh | he/the |
| 04:04.04 | [TK]D-Fender | Kyosh: First, never use the term "IVR call". That term doesn't mean anything. |
| 04:04.05 | ketas | apb1963__: looks like something you can screw |
| 04:04.16 | [TK]D-Fender | Kyosh: And if your call is DYING... then the problem isn't the prompts. |
| 04:04.17 | ketas | apb1963__: with flat head screwdriver? |
| 04:04.31 | [TK]D-Fender | Kyosh: its the CALL... and you should already be looking at its DEBUG |
| 04:04.36 | [TK]D-Fender | Kyosh: And ahve that to show us |
| 04:04.47 | Kyosh | the call is not "dying", it completes. but when the callee picks up, the prompts are not played, the ivr just hangs up |
| 04:05.01 | [TK]D-Fender | Kyosh: Show us the actual problem. |
| 04:05.10 | Kyosh | fine. bbiab |
| 04:07.41 | din3sh | wct4xxp 0000:0b:08.0: TE210P: RECEIVE slip NEGATIVE on span 2------>1st span connected to E1, 2nd span connected to nortel , anyone?????????????????? |
| 04:07.43 | din3sh | :x |
| 04:09.10 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 04:11.52 | ketas | hmm, is there variable for parent context? |
| 04:12.13 | [TK]D-Fender | ketas: That does not make any sense. |
| 04:12.22 | ketas | why |
| 04:12.40 | [TK]D-Fender | ketas: Contexts don't have "parents" |
| 04:12.46 | ketas | i want to know where did goto'd from |
| 04:12.58 | [TK]D-Fender | ketas: You can't |
| 04:13.00 | ketas | i can have variables from there, somehow |
| 04:13.08 | ketas | bad idea? |
| 04:13.14 | [TK]D-Fender | Variables have no scope within a channel |
| 04:13.25 | [TK]D-Fender | More like "that idea does not exist" |
| 04:13.29 | *** join/#asterisk FireAndIce (~FireAndIc@175.100.158.245) |
| 04:13.36 | ketas | mh? |
| 04:13.37 | [TK]D-Fender | Invalid concept. |
| 04:13.42 | ketas | what is channel? |
| 04:13.47 | [TK]D-Fender | the call itself |
| 04:13.52 | ketas | oh |
| 04:13.59 | [TK]D-Fender | they are CHANNEL variables... not "context" variables. |
| 04:14.03 | [TK]D-Fender | They are not bound by context |
| 04:14.33 | ketas | well i want to know what was context the call originally came into |
| 04:14.37 | ketas | not current one |
| 04:15.09 | ketas | so i did Set(ORIG_CONTEXT=${CONTEXT}) |
| 04:15.24 | [TK]D-Fender | ketas: that is what you'd have to do. |
| 04:15.37 | [TK]D-Fender | There is no "parent" and no other tracking of where it started |
| 04:15.58 | ketas | call originating context variable could be useful |
| 04:16.05 | ketas | however well... |
| 04:16.51 | ketas | it's bad idea to have variable for context where call actually entered into dialplans |
| 04:16.54 | ketas | ? |
| 04:17.27 | [TK]D-Fender | That doesn't sound like an actual question.... |
| 04:17.35 | *** part/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 04:17.49 | [TK]D-Fender | And I suppose it's a question about why you care and what you intend to do about it. |
| 04:17.52 | ketas | questions doesn't need to have questiosentences |
| 04:17.54 | ketas | hmm |
| 04:18.29 | ketas | why... well i need this data... and i have workaround too |
| 04:18.32 | ketas | actually |
| 04:21.25 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 04:25.22 | saint_ | [TK]D-Fender: how about this one : http://pastebin.com/3q12C04f |
| 04:26.24 | [TK]D-Fender | saint_: MUCH better looking |
| 04:26.36 | [TK]D-Fender | saint_: No show an actual call with SIP DEBUG enabled. |
| 04:26.39 | [TK]D-Fender | now* |
| 04:26.44 | igcewieling | generally you want rfc2833 dtmfmode |
| 04:27.27 | [TK]D-Fender | I would recommend setting the mode for your devices. |
| 04:27.34 | igcewieling | ketas: there is nothing wrong with Set(ORIG_CONTEXT=${CONTEXT}) |
| 04:27.43 | [TK]D-Fender | Most popular SIP devices use rfc2833 as suggested. |
| 04:27.53 | ketas | igcewieling: indeed |
| 04:28.09 | [TK]D-Fender | Which is what he's already doing.... |
| 04:28.50 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:28.51 | ketas | MixMonitor(call-context-${ORIG_CONTEXT}-from-${CALLERID(number)}-to-${EXTEN}-forward-${MM_FWD_SUF}-${UNIQUEID}.wav,,b) |
| 04:28.54 | ketas | :P |
| 04:29.07 | saint_ | [TK]D-Fender: just a quick question before i do the traces. do i need to forward any ports on the B network to the phone ? |
| 04:29.14 | [TK]D-Fender | saint_: No |
| 04:29.41 | igcewieling | ketas: 1) CALLERID(number) is deprecated. Also what is going to happen if the Callerid number is 555 555 1212 (i.e. has spaces) |
| 04:29.56 | igcewieling | you can use the FILTER function to remove unwanted characters |
| 04:30.14 | ketas | oh indeed it was filter... i forgot it |
| 04:30.21 | [TK]D-Fender | igcewieling: News to me... |
| 04:30.29 | [TK]D-Fender | igcewieling: When did this happen? |
| 04:30.30 | ketas | deprecated, eh |
| 04:31.34 | [TK]D-Fender | igcewieling: And the number SHOULDN"T have spaces.... even though we maye have seen broken exceptions.... |
| 04:31.58 | igcewieling | [TK]D-Fender: around 1.4 I think. CALLERID(num) it may very well be grandfathered in |
| 04:32.11 | [TK]D-Fender | Never hurts to sanitize you input .... remember the tale of Bobby Tables.... |
| 04:32.15 | igcewieling | [TK]D-Fender: never underestimate what can be in the callerid number. 8-) |
| 04:32.20 | [TK]D-Fender | igcewieling: Both are fully supported |
| 04:32.53 | [TK]D-Fender | if1.4 killed ${CALLERIDNUM}. The function from 1.2+ always took both |
| 04:33.07 | [TK]D-Fender | igcewieling: 1.4 killed ${CALLERIDNUM}. The function from 1.2+ always took both |
| 04:33.07 | igcewieling | [TK]D-Fender: number is not documented in the CALLERID function docs |
| 04:33.24 | [TK]D-Fender | igcewieling: It's been documented all over the place and they aren't pulling so far as we can see |
| 04:33.40 | [TK]D-Fender | igcewieling: Perhaps on on the immediate docs... |
| 04:34.44 | igcewieling | We often put dashes in the callerid number to make it look better. |
| 04:36.05 | ketas | eh |
| 04:36.13 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 04:36.36 | ketas | igcewieling: so you don't want to be seen to anybody? |
| 04:36.46 | ketas | :P |
| 04:37.12 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.248) |
| 04:37.20 | ruben231 | hi guys |
| 04:38.37 | igcewieling | [TK]D-Fender: any idea why setting Allow Login With DB Credentials to True and AUTHTYPE=database would not make FreePBX use the DB for auth? |
| 04:39.19 | saint_ | [TK]D-Fender: sip debug: http://pastebin.com/LS9yDVkL |
| 04:39.24 | [TK]D-Fender | igcewieling: How2.8-? |
| 04:39.41 | saint_ | [TK]D-Fender: A.B.C.D being the Public IP of network A , where Asterisk + phone A are located |
| 04:39.41 | igcewieling | 2.9.mumble, but it was upgraded from something older at some point in the past |
| 04:39.44 | [TK]D-Fender | saint_: New PB, mask NOTHING |
| 04:39.51 | [TK]D-Fender | igcewieling: 2.9 should be DB-only |
| 04:40.06 | igcewieling | [TK]D-Fender: *nod* but it doesn't seem to be doing that |
| 04:40.29 | igcewieling | doesn't even appear to be using ANY auth now. |
| 04:40.40 | [TK]D-Fender | igcewieling: Not sure on that... I'd ask a dev #you_know_where |
| 04:40.49 | ruben231 | hi guys i have an asterisk server any chance i can restrict publci registration of my phone extension, only local are allowed any idea how to do it..? |
| 04:41.18 | *** join/#asterisk radic (~radic@dslb-094-216-243-142.pools.arcor-ip.net) |
| 04:41.33 | [TK]D-Fender | ruben231: PERMIT/DENY |
| 04:41.41 | [TK]D-Fender | ruben231: read the sip.conf.sample |
| 04:41.43 | igcewieling | ruben231: iptables to block all access to port 5060/udp from the internet is the most obvious thing. other than that as [TK]D-Fender says permit/deny |
| 04:43.34 | ruben231 | igcewieling: if i block port 5060, my voip carrier will be block also |
| 04:44.42 | igcewieling | ruben231: you can always allow your provider's IP in iptables. Personally I'd do both if it was my server, both iptables and permit/deny. |
| 04:44.56 | ketas | hah, and someone suggested me a paper asterisk book... would be horrible to search from |
| 04:44.57 | igcewieling | But I can be overly paranoid at times. |
| 04:45.34 | [TK]D-Fender | saint_: the call looks OK. no audio? |
| 04:45.43 | saint_ | correct, no audio |
| 04:46.01 | [TK]D-Fender | saint_: Verify your forwarding on the * side |
| 04:46.51 | [TK]D-Fender | saint_: feel free to actully show us as well. |
| 04:47.17 | saint_ | [TK]D-Fender: it s on an Airport Extreme. Beside a screen shot, I don t know how I could show you .. |
| 04:47.33 | [TK]D-Fender | saint_: I've heard a lot of problems with those routers... |
| 04:47.55 | [TK]D-Fender | saint_: If you have any viable substitute I highly recommend testing a swap |
| 04:48.09 | saint_ | [TK]D-Fender: i dont .. |
| 04:48.22 | [TK]D-Fender | Just Say No To Fruit-Based Technology |
| 04:48.34 | ketas | eat them |
| 04:51.07 | saint_ | [TK]D-Fender: at the line Reliably Transmitting (NAT) to , I see the remote public IP address with port 5060 .. you said that NO port should be forwarded in the remote router, to the phone, right ? |
| 04:52.09 | ruben231 | permit/deny woudl wrok on asterisk 1.4 right..? |
| 04:52.17 | [TK]D-Fender | saint_: Correct |
| 04:52.21 | [TK]D-Fender | ruben231: Yes |
| 04:55.28 | ketas | damn |
| 04:55.40 | ketas | i configured asterisk and i forgot to eat... |
| 04:55.48 | ketas | for like a day?! |
| 04:56.19 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 04:57.05 | ruben231 | guys whats teh difference between permit/deny and bindaddr on sip.conf..? |
| 04:57.55 | [TK]D-Fender | the interface you bind your ASTERISK to has nothign to do with the IP of a rREMOTE DEVICE you wish to allow ... or not. |
| 04:58.22 | igcewieling | Also bindaddr will likely break your asterisk setup. |
| 04:59.04 | ruben231 | igcewieling: why..? |
| 04:59.24 | igcewieling | ruben231: because it overrides what the operating system and asterisk thinks is the right thing to do. |
| 04:59.25 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 04:59.53 | ruben231 | coz now my setting is bindaddr=0.0.0.0 |
| 05:00.02 | igcewieling | there are a few situations like if your box has multiple IPs on the same network on the same physical interface. |
| 05:00.15 | [TK]D-Fender | ruben231: you don't seem to understand what binding an interface even means |
| 05:00.18 | igcewieling | ruben231: bindaddr=0.0.0.0 is the same as not having a bindaddr set. |
| 05:00.28 | [TK]D-Fender | ruben231: that is what addresses... ON YOUR SERVER will listen for conenction |
| 05:00.44 | [TK]D-Fender | ruben231: it is not a restriction as to who can cantact your server |
| 05:00.50 | igcewieling | [TK]D-Fender: doesn't it also determine the source ip of outgoing packets. |
| 05:00.59 | ruben231 | ok so it useless if i bind=0.0.0.0 then do some permit/deny policies |
| 05:01.04 | [TK]D-Fender | igcewieling: Let him break one thing at a time, ok? ;) |
| 05:01.13 | [TK]D-Fender | ruthey have nothing to do with each otheer |
| 05:01.20 | [TK]D-Fender | ruben231: they have nothing to do with each otheer |
| 05:01.22 | ketas | source ip is with routing |
| 05:02.05 | ketas | well bind to 127.0.0.1 helps with security a bit |
| 05:02.09 | ruben231 | ok |
| 05:02.09 | WIMPy | It does both. |
| 05:02.22 | WIMPy | Haha |
| 05:02.23 | igcewieling | ketas: so does unplugging the network cable |
| 05:02.49 | ketas | yes, don't let your asterisk to access network |
| 05:04.15 | saint_ | what does it mean : Call from 200 to extension '100' rejected because extension not found in context 'default'. |
| 05:04.29 | saint_ | i have the phones in the context LocalSets |
| 05:04.35 | saint_ | what the heck is this default ? |
| 05:04.48 | ketas | incoming call didn't match those |
| 05:04.48 | igcewieling | saint_: that means the incoming call did not match any entries in sip.conf |
| 05:05.22 | ruben231 | 192.168.40.0/255.255.255.0 <--------with space to specify many networks right..? |
| 05:05.36 | igcewieling | ruben231: no. multuple entries |
| 05:05.51 | ketas | igcewieling: maybe it did match generic one but it was default anyway? |
| 05:05.52 | ruben231 | how to add multiple network..? |
| 05:06.09 | ketas | comma? |
| 05:06.10 | igcewieling | ruben231: multiple permit= lines |
| 05:06.14 | WIMPy | And CIDR notation is allowd |
| 05:06.27 | ruben231 | comma..? |
| 05:06.39 | ketas | forget commas |
| 05:06.43 | igcewieling | use a comma if you want to TOTALLY SCREW IT UP. |
| 05:07.19 | ruben231 | multiple permit= lines ( used space here in between network) |
| 05:07.30 | ketas | theoatmeal had nice howto on how to use punctuation for self defence irl |
| 05:07.42 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 05:08.47 | ruben231 | <PROTECTED> |
| 05:09.19 | ketas | anyone finds cidr harder to read? |
| 05:09.26 | ketas | although somewhat simpler |
| 05:10.37 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 05:11.49 | ketas | i wonder why is that softphone is only thing that always sucks? |
| 05:12.10 | ketas | well, not only |
| 05:12.59 | [TK]D-Fender | souncards suck. most pc headsets suck |
| 05:13.08 | igcewieling | thanks, now I'm going to waste 4 hours on theoatmeal. |
| 05:13.17 | ketas | good luck |
| 05:13.30 | ketas | igcewieling: it does feel good after it, right? |
| 05:13.32 | *** part/#asterisk rue_house (~rue@24-207-103-226.eastlink.ca) |
| 05:13.38 | [TK]D-Fender | [00:04]saint_what does it mean : Call from 200 to extension '100' rejected because extension not found in context 'default'. <-dialplan error |
| 05:13.44 | igcewieling | ruben231: like this: |
| 05:13.45 | igcewieling | permit=192.168.1.0/255.255.255.0 |
| 05:13.45 | igcewieling | permit=10.0.0.0/255.0.0.0 |
| 05:13.54 | igcewieling | see? multiple lines, no spaces. |
| 05:13.58 | [TK]D-Fender | saint_: You have no match for "100" in [default} where its looking |
| 05:14.27 | saint_ | [TK]D-Fender: i had no default actually. i was trying with an asterisk "in the cloud", and i have the same issue. signaling is working, but no voice. |
| 05:14.40 | saint_ | i guess my project for the fire house goes in the water.. |
| 05:14.52 | ketas | pun intended |
| 05:15.19 | [TK]D-Fender | saint_: You are mixing topics |
| 05:15.32 | [TK]D-Fender | saint_: You dialplan error you just showed is completely separate from your audio issues |
| 05:15.46 | saint_ | [TK]D-Fender: i fixed the dialplan error |
| 05:15.50 | [TK]D-Fender | saint_: show us screenshots for your forwarding |
| 05:15.59 | saint_ | [TK]D-Fender: we were able to call each other. it's just when the other party picks up, there is no audio |
| 05:16.34 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 05:16.48 | ruben231 | igcewieling: how about deny all incoming sip..? from public..? deny=0.0.0.0..? |
| 05:17.37 | igcewieling | usually deny=0.0.0.0/0.0.0.0 having multiple permit lines makes sense, multiple deny lines is just crazy. |
| 05:18.44 | [TK]D-Fender | igcewieling: Not necessarily... but very rarely coudl I see cases where I'd imagine doing it |
| 05:19.28 | igcewieling | [TK]D-Fender: almost every stupid thing has an edge case where it is not a stupid thing. |
| 05:20.17 | igcewieling | killing is one example |
| 05:21.10 | ruben231 | <PROTECTED> |
| 05:21.16 | [TK]D-Fender | And some things not so fine an edge ;) |
| 05:21.17 | saint_ | [TK]D-Fender: the other guy i m working with went to bed. we'll resume tomorrow. thanks for the hand. |
| 05:21.34 | [TK]D-Fender | saint_: Get another router in the meantime |
| 05:21.45 | saint_ | [TK]D-Fender: any recommendation ? |
| 05:21.57 | igcewieling | ruben231: I would use a netmask of /255.255.255.255 in that case, but I don't think it is required. |
| 05:22.22 | ruben231 | igcewieling: just put publci ip address would do..? |
| 05:22.34 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-bohwyrxfvwluobcw) |
| 05:22.38 | igcewieling | you would have to ask someone who knows . |
| 05:22.51 | igcewieling | or you could use /255.255.255.255 for the netmask and KNOW it will work. |
| 05:23.02 | igcewieling | is off to sleep. |
| 05:24.51 | [TK]D-Fender | saint_: Linksys tend to play nic. |
| 05:25.59 | ketas | saint_: get sleep too :P |
| 05:26.08 | saint_ | yeah.. thanks ... |
| 05:29.03 | din3sh | \clear |
| 05:33.32 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
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| 05:39.11 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 05:39.38 | din3sh | WARNING[17945] abstract_jb.c: SIP/121-00000086 received frame with invalid timing info: has_timing_info=0, len=0, ts=0, src=dahdi_read |
| 05:39.44 | din3sh | What does this mean? |
| 05:51.44 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
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| 06:09.49 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
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| 06:51.05 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 06:52.35 | teloniusz | hi. Is there a way to disable jumping to 'h' extension on caller hangup? I've written a lua application for extensions.lua and I don't want it to be interrupted in the middle |
| 06:54.52 | *** join/#asterisk resist0r (~resist0r@c-174-57-0-74.hsd1.nj.comcast.net) |
| 07:04.44 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
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| 07:13.35 | ChannelZ | LOL - http://blog.krisk.org/2013/02/packets-of-death.html |
| 07:14.08 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:16.14 | resist0r | yeah I just read this too [ like 10 min ago ] :: http://tech.slashdot.org/story/13/02/06/2024251/intel-gigabit-nic-packet-of-death |
| 07:18.17 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:18.45 | ketas | hahaha |
| 07:18.52 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:20.22 | ketas | OWWW i have that in laptop |
| 07:20.28 | kaldemar | ChannelZ: "it's ptime!" |
| 07:21.42 | ketas | and i had this weird error once |
| 07:21.55 | ketas | disabled all offloadings |
| 07:27.05 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.73) |
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| 07:59.47 | Kobaz | hm, |
| 07:59.53 | Kobaz | i'm going to have a problem |
| 08:00.21 | Kobaz | you can't do reply/forward of a voicemail to a voicemail context outside your own |
| 08:02.31 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 08:04.51 | ketas | Kobaz: move mailbox? :P |
| 08:05.19 | Kobaz | i should probably add some sort of voicemail context include |
| 08:10.50 | *** join/#asterisk dijib (~dijib@67-204-199-13.eastlink.ca) |
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| 08:44.47 | nunne_ | calendar integration to exchange via ews. is there anyway to specify the user/calendar? I wan't to use a account which can view/read all users calendars.. Since I don't really want all the peoples real username and password in the calendar.conf. Plus they have a policy to change that. |
| 08:45.00 | x1user | Hello. I got group: [Feb 7 10:35:29] NOTICE[4011][C-00001009]: chan_sip.c:24742 handle_request_invite: Failed to authenticate device "0899995770" <sip:0899995770@85.118.193.134>;tag=as1962b365, my trunk is insecure=port,invite =/ ? |
| 08:45.05 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:46.27 | kaldemar | x1user: you need to look around that notice. what peer does the invite match? |
| 08:47.36 | x1user | kaldemar: the peer 85.118.193.134 is port,invite . Is that what you mean? |
| 08:48.26 | kaldemar | x1user: no. i mean you need to look at your CLI and find the line where it says which peer the incoming call matches. |
| 08:49.07 | kaldemar | x1user: if you can't, enable verbosity and sipdebug and pastebin what you get in CLI. |
| 08:56.40 | x1user | kaldemar: http://codepad.org/At6WeINQ this is max verbose and debug log |
| 08:57.58 | kaldemar | x1user: "sip debug" |
| 08:58.26 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
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| 09:00.38 | kaldemar | that does not include it. anyway, it seems to match a peer by the name GLOBUL-NEW and fail due to your deny/permit parameters. |
| 09:01.38 | *** join/#asterisk deo_ (~deo@203.177.214.75) |
| 09:02.16 | *** join/#asterisk din3sh (~din3sh@41.136.253.215) |
| 09:02.23 | din3sh | hello all |
| 09:02.38 | x1user | kaldemar:thanks |
| 09:02.43 | x1user | the truth is always obious |
| 09:06.03 | din3sh | wct4xxp 0000:0b:08.0: TE210P: RECEIVE slip NEGATIVE on span 2------>1st span connected to E1, 2nd span connected to nortel |
| 09:06.15 | din3sh | Kaldemar what does this mean? |
| 09:09.59 | kaldemar | thats a timing slip on span 2. |
| 09:11.34 | din3sh | my system.conf ---> http://pastebin.com/bhP1nKDM |
| 09:12.28 | kaldemar | timing parameters look ok. |
| 09:15.04 | din3sh | the timing source for span 1 is the E1 provider, and the timing source for the nortel is asterisk? |
| 09:17.54 | kaldemar | well, not asterisk but DAHDI. |
| 09:20.03 | din3sh | ok dahdi |
| 09:20.23 | din3sh | what if the E1 line providing the timing is faulty |
| 09:20.30 | din3sh | it will pass on the fault to span 2? |
| 09:23.56 | kaldemar | DAHDI takes one clock which in your case is now the E1 line. it uses it to clock span2. |
| 09:24.43 | kaldemar | the provider link is not the first place i'd check first. what's the clock setting on the nortel? |
| 09:26.12 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-147-114.pools.arcor-ip.net) |
| 09:27.06 | hegars | din3sh, just out of interest is it a DMS100? |
| 09:40.19 | *** part/#asterisk rox (~rox@212.30.81.3) |
| 09:40.55 | *** join/#asterisk _zoom_ (~zoom@196.1.219.122) |
| 09:41.42 | _zoom_ | weired thing, I installed asterisk 1.8 from epel repo, it doesn't have meetme module, any idea? |
| 09:43.07 | din3sh | I have no control on the nortel |
| 09:43.10 | din3sh | :/ |
| 09:43.26 | din3sh | what if its DMS100 |
| 09:43.27 | din3sh | ? |
| 09:43.38 | din3sh | its a 10yr old nortel meridien system |
| 09:45.05 | kaldemar | _zoom_: meetme requires DAHDI. maybe it's in a separate package. |
| 09:46.02 | kaldemar | _zoom_: so it seems. install asterisk-dahdi. |
| 09:49.42 | _zoom_ | cheers kaldemar i thought that dahdi-tools is enough :p |
| 09:52.58 | nunne_ | din3sh, can you update the DMS100? I worked with nortel a couple of years ago and I know that many of their firmwares regarding PRI signalling is buggy as hell. loss of frame that it couldn't recover from.. only way was to update the firmware |
| 09:55.37 | *** join/#asterisk studybot_ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
| 09:57.44 | din3sh | I have no prior knowledge of nortel systems, its another provider's system, i have only hooked my asterisk in between |
| 09:57.45 | din3sh | :s |
| 09:57.55 | *** join/#asterisk chendy (~alexc@113.116.60.114) |
| 09:58.34 | din3sh | the funny thing is that when i did some tests with asterisk 1.4.x and zaptel, i didnt have such problems |
| 09:58.53 | din3sh | now with asterisk 1.8 and dahdi am having timing slips |
| 09:58.59 | kaldemar | din3sh: are there any issues with the connectivity or is it just the slip print your worried about? |
| 09:59.18 | kaldemar | if you see a lot of those, then it will be an issue of course. |
| 10:00.50 | hegars | did you change kernels when you did that? |
| 10:02.30 | nunne_ | din3sh, is crc set or not set? i have had nortel systems which didn't give the correct error when crc was not set/unset accordingly.. because I would get all the link up |
| 10:04.16 | kaldemar | din3sh: beware that zaptel had slip debugging off by default. it did not print those messages unless you had debugslips module parameter set. |
| 10:05.16 | kaldemar | so you might have had slips but did not see them. |
| 10:07.37 | din3sh | [Feb 6 20:28:40] NOTICE[23866] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 2 |
| 10:07.37 | din3sh | [Feb 6 20:29:14] NOTICE[23865] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 |
| 10:09.24 | din3sh | <PROTECTED> |
| 10:10.26 | din3sh | I get the HDLC Bad FCS on span 1 only |
| 10:11.03 | din3sh | dahdi test for more than 3 hours: --- Results after 27968 passes --- |
| 10:11.03 | din3sh | Best: 100.000% -- Worst: 99.862% -- Average: 99.995729% |
| 10:11.03 | din3sh | Cummulative Accuracy (not per pass): 99.996 |
| 10:13.37 | din3sh | server: HP DL380 g6, 8gb memory, quad proc, TE220 with echo cancel module, centos 6.2 with latest kernel |
| 10:18.26 | din3sh | CRC is not set on any span |
| 10:19.51 | *** join/#asterisk af_ (~getsmart@5.157.110.131) |
| 10:24.07 | din3sh | ?? |
| 10:29.56 | *** join/#asterisk timahvo1 (~rogue@41.90.29.152) |
| 10:32.48 | Rico29 | hi all |
| 10:32.57 | Rico29 | is there an easy way to handle 302 redirects in asterisk 1.8? |
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| 11:22.43 | zamba | we want to do video through asterisk.. is this possible? our endpoints are cisco/tandberg C40s |
| 11:22.51 | zamba | i guess this will be over sip |
| 11:23.04 | zamba | but also h323 |
| 11:34.10 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 11:37.56 | zamba | can asterisk do a forward in the pstn network? |
| 11:38.37 | zamba | meaning.. it receives a call.. and then handles it back to the pstn network telling it to terminate to another number, thus freeing the channel? |
| 11:40.38 | zamba | is this at all possible? |
| 11:43.31 | kaldemar | asterisk does video over SIP. the latter depends on the used technology. |
| 11:43.59 | zamba | i guess it's called call transfer? |
| 11:44.05 | zamba | kaldemar: what do i need to know? |
| 11:44.31 | kaldemar | how you connect to PSTN. |
| 11:45.18 | zamba | analogue gateway |
| 11:46.34 | zamba | either a audiocode or a tenor |
| 11:46.43 | zamba | according to my colleague this works perfectly here in norway |
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| 11:48.13 | kaldemar | i guess you're out of luck then. |
| 11:48.21 | zamba | hm? why? |
| 11:50.43 | kaldemar | because it is not up to asterisk. you can try sending the gateway a SIP 302 with app Transfer and see what it does. but only if you don't answer the call first. |
| 11:59.28 | zamba | can asterisk do video over h323 as well? |
| 12:00.36 | ectospasm | h323 isn't well supported, IIRC |
| 12:01.02 | ectospasm | there's no reason it can't work, but I dunno what the state of the driver is in |
| 12:01.40 | zamba | ectospasm: but most endpoints today generally support both protocols, i guess? |
| 12:02.04 | ectospasm | depends on the endpoints. Digium phones do not. |
| 12:02.21 | kaldemar | most physical do not. |
| 12:19.21 | zamba | so h323 is the safest bet between endpoints? |
| 12:21.41 | kaldemar | because it isn't well supported? |
| 12:22.25 | kaldemar | safest bet is something that is best supported in asterisk, and on VoIP that is SIP. |
| 12:25.55 | din3sh | when dtmf is keyed in rapidly, asterisk does not read the dtmf well or not at all, when there is a 1sec pause between keying the numbers, asterisk reads the dtmf properly, why is it so? |
| 12:41.11 | leifmadsen | asterisk has h323 support, but SIP is far and away the more supported protocol |
| 12:56.51 | ectospasm | din3sh: what version of Asterisk? |
| 12:57.33 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 12:58.16 | ectospasm | din3sh: if you're using a version prior to Asterisk 1.8.18.0, or 10.10.0, see this: https://issues.asterisk.org/jira/browse/ASTERISK-19610 |
| 12:59.00 | ectospasm | It appears that 11 is unaffected. |
| 13:00.03 | din3sh | i upgraded from 1.8.13 to 1.18.20 last night |
| 13:00.15 | din3sh | has it been fixed in 1.18.20? |
| 13:07.40 | *** join/#asterisk blee (~blee@68.204.217.123) |
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| 13:12.12 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:13.40 | zamba | leifmadsen: i just need to make sure that our endpoints still are able to reach remote endpoints |
| 13:13.46 | zamba | leifmadsen: which currently is using h323 |
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| 13:49.12 | ghost75 | is stun used to not use port forwardings on router? |
| 13:49.58 | WIMPy | If by "use" you mean "detect", yes. |
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| 13:52.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:52.54 | Kobaz | ugh, why do customers insist on calling it 'Que' and not Queue |
| 13:53.31 | ghost75 | too much bond |
| 13:54.25 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
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| 13:57.27 | leifmadsen | Que == that |
| 13:57.58 | leifmadsen | so just read their sentence like, "When Johnny joined that he wasn't able to get any calls" |
| 13:58.04 | leifmadsen | then just response with, "that what?" |
| 13:58.18 | leifmadsen | the que |
| 13:58.21 | leifmadsen | the that? |
| 13:58.42 | Kobaz | the what? |
| 13:58.49 | leifmadsen | the thtat |
| 13:58.56 | Kobaz | the hat? |
| 13:59.01 | leifmadsen | cat in the hat? |
| 13:59.07 | Kobaz | sure |
| 13:59.13 | leifmadsen | I like cats. |
| 13:59.28 | Kobaz | i'm allergic |
| 13:59.30 | ghost75 | schrödingers cat |
| 13:59.31 | Kobaz | i like them |
| 13:59.38 | Kobaz | but they carry eviiiil allergies |
| 13:59.48 | ghost75 | i am allergic too |
| 14:01.34 | leifmadsen | I haz not teh allergies |
| 14:01.40 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:02.28 | ghost75 | u haz been lucky |
| 14:02.53 | leifmadsen | indeed |
| 14:04.27 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
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| 14:09.43 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
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| 14:11.49 | ghost75 | do you use any windows clients with asterisk? |
| 14:12.21 | bulkorok | phonerlite |
| 14:12.24 | bulkorok | xlite |
| 14:12.56 | ghost75 | found none with addresses over mysql |
| 14:13.37 | ghost75 | always just outlook bah |
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| 14:27.25 | [TK]D-Fender | ghost75, What percentage of Windows-based apps are likely to be built for using your OSS DB server vs things normal windows users have and use? |
| 14:27.51 | [TK]D-Fender | ghost75, Common sense. Got a windows app? Expect integration with Windows-based solutions. |
| 14:28.06 | greenwolf | yup |
| 14:28.19 | greenwolf | sup [TK]D-Fender |
| 14:29.08 | ghost75 | outlook sucks |
| 14:30.27 | Katty | hi lads. |
| 14:30.34 | Katty | i have coffee AND mt. dew. |
| 14:30.38 | Katty | be very, very afraid. |
| 14:30.55 | ghost75 | strangely everybody wants to have outlook |
| 14:31.08 | Katty | it's a business standard these days. |
| 14:31.13 | Katty | so of course they want to have it. |
| 14:31.28 | ghost75 | they are victims |
| 14:31.31 | greenwolf | yea outlook sucks but they make us use it at work...terrible product if you ask me |
| 14:31.45 | Katty | same as everyone wants Louis Vuitton purses and Jessica Simpson shoes. |
| 14:31.54 | ghost75 | even notes is ways better |
| 14:32.11 | Katty | it may be a terrible product to /you/ |
| 14:32.18 | Katty | but it may make other people far more productive. |
| 14:32.24 | Katty | it's easy to understand. |
| 14:32.32 | Katty | and supporting the masses is what it does. |
| 14:32.33 | ghost75 | lol easy to understand |
| 14:32.41 | Katty | so perk up buttercup. |
| 14:33.03 | Katty | everything is relative. |
| 14:35.42 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
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| 14:47.40 | zamba | anyone here familiar with polycom's phones? |
| 14:47.51 | zamba | right now i'm talking about IP 650 |
| 14:47.57 | zamba | i want to enable vad |
| 14:48.06 | zamba | and i have added some options to sip.cfg on the provisioning server |
| 14:48.11 | zamba | but how can i confirm that it's working? |
| 14:49.03 | SuperNull | uhg. When did using underscores in variable names stop being .. a 'thing' ? 1.8 ? |
| 14:49.07 | [TK]D-Fender | Use it and watch * complain as it doesn't support VAD |
| 14:49.13 | *** join/#asterisk hegars (~hegars@203186072235.static.ctinets.com) |
| 14:49.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:49.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:49.30 | [TK]D-Fender | SuperNull, Example? |
| 14:49.56 | Katty | SuperNull: about the same time that marketing based on Christian Values became a thing. Do you need a christian realtor?! |
| 14:50.06 | SuperNull | old school it used to signifiy 'local' allocation or per-channel vs global variable |
| 14:50.07 | zamba | [TK]D-Fender: oh.. asterisk doesn't support it? |
| 14:50.30 | [TK]D-Fender | zamba, Correct |
| 14:50.30 | SuperNull | i have dialplan that was created in possibly 1.2 |
| 14:50.42 | SuperNull | just auditing to see which versions need __ removed.. |
| 14:50.51 | [TK]D-Fender | SuperNull, Location counts and details are scarce |
| 14:50.53 | SuperNull | and/or which old ass boxes need it added. |
| 14:51.07 | [TK]D-Fender | SuperNull, and underscores in front are for INHERITANCE and always have been... |
| 14:51.24 | [TK]D-Fender | SuperNull, You should be reviewing your dialplan basics for this... |
| 14:51.39 | zamba | [TK]D-Fender: so it isn't the phone itself that generates the comfort noise? |
| 14:52.08 | [TK]D-Fender | zamba, * kills endpoints for having no RTP |
| 14:55.25 | SuperNull | TK when is _ required.. exactly.. |
| 14:58.15 | [TK]D-Fender | SuperNull, Inheritance..... read up... |
| 14:58.47 | SuperNull | i did its a 1 line description of allowing it on a spawn .. is that a dial ? |
| 14:59.14 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
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| 15:05.04 | carl0s- | Hi. Asterisk 1.4. Is there no way at all of having a wildcard at the beginning or middle of a dial pattern? I want to match the last 8 digits of a number, not the fist 8, to cater for when the international prefix is present or not. this doesn't seem to be possible. X requires a digit to be present, and . cannot be followed by anything. |
| 15:07.44 | WIMPy | carl0s-: That's the way it is. |
| 15:08.13 | WIMPy | Either you create lots of patterns with different lengths or you can use loopback switches. |
| 15:08.35 | WIMPy | If you don't care about dialing a simple goto might be good enough. |
| 15:10.23 | psykon | Anyone in the UK want to help me test out mu UK freephone number? :) |
| 15:10.42 | bchia | carl0s WIMPy - could you use REGEX() function for that? |
| 15:11.02 | [TK]D-Fender | carl0s- is using FreePBX. Toss most real processing out the door |
| 15:11.20 | WIMPy | N/me doesn't see any agvantage over a simple Goto(If). |
| 15:11.55 | [TK]D-Fender | carl0s-, Not with FreePBX, and not without matching more than you want then validating in the actual dialplan. |
| 15:12.11 | carl0s- | sorry.. was in the wrong window there |
| 15:12.16 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 15:12.27 | carl0s- | [TK]D-Fender, ;) |
| 15:12.29 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 15:13.24 | carl0s- | OK thanks. For now we will have to make sure the users enter the number in the "as dialled" format into the database. |
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| 16:42.50 | Kyosh | i am using asterisk in realtime with mysql. my extensions.conf has contexts setup as [context-name] and underneath it i have a line such as switch => Realtime/@ ... is there a way in extensions.conf to have a "catch-all" for the contexts so that any context will redirect to switch => Realtime/@ ? thanks. |
| 16:44.50 | [TK]D-Fender | Kyosh, No |
| 16:45.32 | Kyosh | [TK]D-Fender: btw, thanks last night. i debugged for a few hours and found out that this was the problem. regardless of realtime, the context wasnt defined in extensions.conf |
| 16:45.47 | Kyosh | but now, i am hoping that i can have a catchall for the contexts to redirect to the database. |
| 16:46.04 | [TK]D-Fender | Kyosh, Not happening unfortunately. |
| 16:46.25 | Kyosh | cause if a new context is created dynamically via database, its a pain to manually log in to the asterisk box just to manually update the extensions.conf file |
| 16:46.32 | Kyosh | :( |
| 16:46.35 | Kyosh | dag nabbit |
| 16:46.51 | Qwell | I'd suggest using templates, but I think your use of realtime prevents that. |
| 16:46.56 | Kyosh | that REALLY puts a strangle hold on things |
| 16:47.52 | Kyosh | i dont even know what templates are :( |
| 16:48.48 | Kyosh | which .c file loads the extensions.conf contexts? |
| 16:48.48 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:48.59 | Qwell | pbx/pbx_config.c |
| 16:49.06 | [TK]D-Fender | Qwell, That work with 1.4? |
| 16:49.11 | Kyosh | awesome, maybe i'll get stupid and break mine |
| 16:49.15 | [TK]D-Fender | waits for the expected.... |
| 16:49.16 | Qwell | [TK]D-Fender: Who cares? :) |
| 16:49.21 | [TK]D-Fender | Qwell, HE will ;) |
| 16:49.25 | Qwell | ~upgrade asterisk |
| 16:49.26 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 16:49.30 | Qwell | *WE* won't. |
| 16:49.33 | Kyosh | nah. if i break it, i break it |
| 16:49.43 | igcewieling | Qwell: I'm AKA ManxPower in case you did not know. |
| 16:49.50 | Qwell | igcewieling: yes, I'm aware |
| 16:49.59 | Qwell | also, that was random |
| 16:50.04 | igcewieling | you didn't know I'm in huntsville so I wasn't sure. |
| 16:50.11 | Qwell | I knew you were in AL. |
| 16:50.21 | Kyosh | oh here's a question ...maybe. can the AMI create a new contexxt in extensions.conf? |
| 16:50.25 | Qwell | and I knew you were in the area at one point. I just didn't realize you lived here. |
| 16:50.33 | Qwell | Kyosh: sure |
| 16:50.34 | igcewieling | 5 more days until palm trees and warm weather! |
| 16:55.31 | _Corey_ | looks out his window to see if the latest snowstorm has arrived yet... |
| 16:58.46 | *** join/#asterisk ra21vi (~ravi@122.177.161.195) |
| 16:59.06 | ra21vi | Is it possible to run an AGI script inside queue? |
| 17:00.02 | ra21vi | I wrote an agi which asks caller about some options and send him to queue. Now I want to let user opt for some options if he is waiting for too much in queue. |
| 17:00.26 | ra21vi | I search documentations and examples, but could not find any way in which an agi can be run in queue |
| 17:00.47 | leifmadsen | you can accept a DTMF I think to execute a script or dialplan (which could run an AGI) |
| 17:02.06 | ra21vi | leifmadsen: when caller in queue presses key, there DTMF event is fired, that i can capture using AMI. but how that will invoke a dialplan? |
| 17:02.52 | leifmadsen | ra21vi: check the queue application for the flag I was trying to think might exist |
| 17:02.58 | leifmadsen | ignore the AMI part for now |
| 17:04.06 | *** part/#asterisk wwalker (~wwalker@208.92.232.27) |
| 17:04.28 | ra21vi | leifmadsen: Queue(queuename,options,URL,announceoverride,timeout,AGI,macro,gosub,rule,position) |
| 17:04.42 | leifmadsen | ya look at the available options |
| 17:04.44 | ra21vi | leifmadsen: AGI is there. I will look into it what it does |
| 17:04.58 | leifmadsen | that's not exactly what I meant, but ya, start with reading that documention |
| 17:05.03 | leifmadsen | documentation* |
| 17:05.06 | leifmadsen | that's all I would do heh |
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| 17:07.24 | ra21vi | :) thank you. Actually i saw some docs online and there was no such option. I think AGI queue does not have it. |
| 17:07.42 | ra21vi | But still I can use Agi Exec to run Queue application |
| 17:11.18 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 17:12.24 | [TK]D-Fender | ra21vi, No point.... Queue is blocking.... |
| 17:12.44 | [TK]D-Fender | ravIf you want options while in the queue, that's what the exit context is for... but you will leave the queue and your place in line |
| 17:15.35 | ra21vi | [TK]D-Fender: oh. |
| 17:16.20 | ra21vi | leifmadsen: that AGI in queue will run when the agent will pick the call.. the agi script runs on caller's channel.. so that will not help me. |
| 17:16.45 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 17:18.28 | ra21vi | [TK]D-Fender and leifmadsen : I think features would have something to do. Since I cannot execute agi while caller is in queue, but i can always play custom announcements for some DTMF if they want to register for "Schedule a call".. and that special DTMF sequence can be sort of configured in features. This is my very vague idea, the direction i am thinking off. May be completely wrong path. :) |
| 17:20.10 | [TK]D-Fender | ra21vi, Wouldn't this action imply they were also goint to leave the queue? |
| 17:21.36 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:44c9:c90b:a530:9b49) |
| 17:21.41 | ra21vi | [TK]D-Fender: example, announcement will say if you want us to schedule a call, press * now.. If the user press * then caller will be off the queue and in other IVR which will ask for some inputs and then caller channel hangup. |
| 17:22.50 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:22.50 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:24.47 | [TK]D-Fender | ra21vi, Yes, this is certainly easy |
| 17:24.57 | [TK]D-Fender | ra21vi, just use the exit context and you're done... |
| 17:25.28 | ra21vi | [TK]D-Fender: what is exit context and how can I use it? Sorry I dont know about it |
| 17:25.54 | [TK]D-Fender | ra21vi, context=justanothercontextwithsingledigitexteninittomatch |
| 17:26.01 | [TK]D-Fender | ra21vi, in your queue definition |
| 17:27.02 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 17:27.27 | Kyosh | how would i create a new context in extensions.conf via AMI? |
| 17:27.39 | ra21vi | [TK]D-Fender: and in my dialplan I will write IVR login inside justanothercontextwithsingledigitexteninittomatch |
| 17:28.27 | rdm | yawns |
| 17:31.18 | SuperNull | hey guys.. why would an older 1.4 not load nearly ANY modules on start yet a preload => xxx.so makes it work.. |
| 17:34.21 | ra21vi | [TK]D-Fender: thank you. After reading the documentation, I think context is what I was looking for. |
| 17:36.02 | [TK]D-Fender | SuperNull, missing autoload |
| 17:36.26 | [TK]D-Fender | Kyosh, did you read the the complete AMI command list? |
| 17:38.02 | *** join/#asterisk SparFux (~rli@e182025185.adsl.alicedsl.de) |
| 17:38.03 | Kyosh | [TK]D-Fender: i believe i did, but i havent found anything to create/modify the dialplan |
| 17:38.28 | SparFux | Hi all. Short question: does anybody successfully use adapters with HFC ISDN BRI chipset on USB ports? |
| 17:38.35 | SparFux | With DAHDI, I mean. |
| 17:39.23 | *** join/#asterisk deo (~deo@112.198.79.214) |
| 17:45.10 | [TK]D-Fender | Kyosh, You should focus on things you CAN get to via AMI that themselves have such an ability... |
| 17:46.45 | Kyosh | ok so, "can" the extensions.conf be modified via AMI or can it "not" ? |
| 17:47.02 | Kyosh | that last remark really threw me for a loop |
| 17:47.15 | leifmadsen | no |
| 17:47.28 | SparFux | Will the Eicon Diva USB ISDN adapter work with DAHDI? |
| 17:47.36 | leifmadsen | you can modify the dialplan with vim, or some other script that updates the file (or better, the includes) |
| 17:49.37 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 17:49.42 | WIMPy | SparFux: The USB dongles work very well, but the only USB device supported by dahdi is the Astribank. |
| 17:50.30 | tzafrir_laptop | Eicon should be supported, or not (no idea) with chan_capi, right? |
| 17:50.35 | Kyosh | so there is no way to programmatically change the extensions.conf file. hmmm. this is bad. |
| 17:50.55 | Kyosh | cant have a wildcard for catch-all on extensions.conf contexts in real-time either |
| 17:51.01 | tzafrir_laptop | and the HFC-USB is sadly not supported in DAHDI. Maybe in misdn |
| 17:51.25 | WIMPy | CAPI is an option, but chan_capi doesn't seem too alive. MISDN definitely works. |
| 17:51.29 | SparFux | WIMPy: Hi there. What are USB dongles? |
| 17:51.49 | WIMPy | Stuff you connect via USB. |
| 17:52.04 | SparFux | WIMPy: Ok, but what USB dongle will work with DAHDI? |
| 17:52.13 | WIMPy | None. |
| 17:52.16 | tzafrir_laptop | An Astribank is no USB dongle :-) |
| 17:52.34 | tzafrir_laptop | But it's a "stuff connected through USB" |
| 17:52.39 | WIMPy | Either get an Astribank or use mISDN. |
| 17:53.06 | SparFux | tzafrir_laptop: ok :-P Well, I am looking for a way to connect my freeswitch box to the ISDN BRI. I have the HFC support patched into DAHDI and for PCI cards it works really well. So now I am looking for a USB solution. |
| 17:53.30 | WIMPy | Which reminds me that I disn;t get a final answer to the usual timing issues and how far Astibank is affected. |
| 17:53.45 | tzafrir_laptop | WIMPy, what issue? |
| 17:53.54 | WIMPy | SparFux: How often do you need the answer? |
| 17:54.22 | SparFux | WIMPy: Ok, it's not gonna work. :-( |
| 17:54.30 | WIMPy | tzafrir_laptop: The one that but Digium and Sangoma cards only support one timing source per card. |
| 17:54.42 | WIMPy | s/but/both/ |
| 17:55.05 | [TK]D-Fender | <Kyosh> ok so, "can" the extensions.conf be modified via AMI or can it "not" ? <- yes |
| 17:55.37 | WIMPy | I asked Xorcom sales, but the answer wasn;t clear to me and I somehow forgot to ask again. |
| 17:56.33 | tzafrir_laptop | I guess it's basically the same for the AB. |
| 17:57.08 | Kyosh | [TK]D-Fender: i was starting to get the impression that it wasnt possible. can you tell me the ami command and i will go find the references? |
| 17:57.29 | [TK]D-Fender | Kyosh, command |
| 17:57.34 | *** part/#asterisk SparFux (~rli@e182025185.adsl.alicedsl.de) |
| 17:57.35 | tzafrir_laptop | This is because the respective chips that implement the low-level signalling won't handle separate timing for the different ports, basically. |
| 17:57.37 | WIMPy | tzafrir_laptop: It might actually be a dahdi issue. If I understood it right what I read here a while ago that's even an issue with the B410P, but in that case it's not a hardware issue. |
| 17:58.22 | WIMPy | I understood it's a hardware issue for the PRI cards. |
| 17:58.53 | tzafrir_laptop | (disclaimer: I don't speak for Sangoma and Digium and maybe not familiar with their hardware well enough) |
| 17:58.59 | Kyosh | doesnt command just execute a CLI command? |
| 17:59.00 | igcewieling | WIMPy: If I understand it correctly, Sangoma cards can have a different timing source for each port. |
| 17:59.14 | igcewieling | Digium cards, unless they were changed in the last 5 years don't. |
| 17:59.42 | WIMPy | igcewieling: Do you have any further information? |
| 17:59.58 | igcewieling | I'm only referring to T-1 cards, of course. |
| 18:00.05 | igcewieling | WIMPy: on digium or on sangoma? |
| 18:00.20 | tzafrir_laptop | WIMPy, it's basically the same hardware issue on BRI: the Cologne HFC / XHFC chips handle the on-wire "bit transfer" and hence the synchronization |
| 18:00.37 | WIMPy | Sangoma |
| 18:00.45 | tzafrir_laptop | (Not well familiar with the Sangoma BRI card) |
| 18:01.23 | WIMPy | tzafrir_laptop: As far as I understand the HFC4/8-S can handle asynchronous ports. I certainly didn't have any issues so far. |
| 18:02.15 | WIMPy | What I'm not entierely sure about is if that needs to be set up some way. So as it would explain this not working when using dahdi. |
| 18:03.02 | *** join/#asterisk alami (~alami@unaffiliated/alami) |
| 18:03.13 | WIMPy | It's nice for vendors that sell line syncronizers, but tht's not really a solution. |
| 18:05.43 | cj | hey folks |
| 18:05.50 | cj | any of you used a grandstream ATA? |
| 18:06.16 | *** join/#asterisk b0ot (~DynamicFa@147.177.61.147) |
| 18:06.25 | igcewieling | I wouldn't use a grandstream ATA to killa roach, let along let one on my network |
| 18:06.31 | cj | I'm not hearing the DTMF when I press the keys |
| 18:06.32 | b0ot | Anyone know of a good place to get free ringtones, tones, etc? |
| 18:08.02 | cj | meh. changed phones and now it works. |
| 18:08.44 | [TK]D-Fender | <Kyosh> doesnt command just execute a CLI command? <- You might be catching on.... |
| 18:09.25 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-190-103.red.bezeqint.net) |
| 18:11.08 | Kyosh | ok so what would the CLI command be to create a new context and add an extension? |
| 18:12.20 | WIMPy | Kyosh: Why do you want to do it via CLI? |
| 18:13.52 | Kyosh | wimpy i actually wanted to do it via AMI but i am being lead to use the AMI 'command' to do it via CLI |
| 18:14.30 | igcewieling | Kyosh: generally if you want to dynamically add and remove extensions you want to use Realtime and a database. |
| 18:15.11 | WIMPy | I don't know if it automatically creates contexts when you try to add an extension to an unknown context. Try it. |
| 18:15.20 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-245-234.pools.arcor-ip.net) |
| 18:15.30 | WIMPy | But .. what igcewieling said. Or anything else :-) |
| 18:16.18 | igcewieling | I don't trust Asterisk to write out a new extensions.conf without messing it up and if you don't write out extensions.conf your shiny new extensions will be lost when you restart Asterisk |
| 18:17.58 | Kyosh | igcewieling, i am using realtime. problem is i need to be able to add new contexts. that has to be done via extensions.conf and have the [context] added with switch => Realtime/@ |
| 18:18.51 | Kyosh | [TK]D-Fender tried to *hint* me in a direction of using the ami "command" command, but i feel thats actually pulling me away from the goal without undertanding more, where UpdateConfig may be more of what I need. |
| 18:25.47 | igcewieling | whoo! whoo! we are finally doing interop testing with a new million-mins-a-month customer |
| 18:26.20 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 18:29.42 | [TK]D-Fender | Kyosh, there is an Asterisk ****CLI**** command set for making changes to the dialplan.... |
| 18:30.16 | [TK]D-Fender | Kyosh, dialplan <tab> |
| 18:30.21 | [TK]D-Fender | Kyosh, Start drilling.... |
| 18:31.43 | igcewieling | drill baby drill! |
| 18:36.08 | Kyosh | [TK]D-Fender i appreciate you nudging me to look into it but its not helping much as i have looked into 'dialplan add extension' and cannot seem to get it working, which is why i came here for help. i do try (unlike many) to do research myself and get things going and even work with others before coming here to ask, but sadly i dont feel i'm receiving actual help. if i knew the command for something that someone was specifically as |
| 18:37.43 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 18:45.15 | ChannelZ | dialplan add extension 1111,1,Dial,SIP/Bob into newcontext |
| 18:45.29 | ChannelZ | Context 'newcontext' did not exist prior to add extension - the context will be created. |
| 18:45.52 | ChannelZ | -- Registered extension context 'newcontext'; registrar: pbx_config |
| 18:46.55 | ChannelZ | But with the realtime thing I have no idea.. |
| 18:48.08 | cj | hurm... this is getting printed literally... how do I get asterisk to interpolate the variable's value into this? |
| 18:48.11 | cj | same => n,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(num)}>) |
| 18:48.31 | cj | (I get the following on the console) |
| 18:48.31 | cj | <PROTECTED> |
| 18:48.49 | WIMPy | num? all? |
| 18:49.21 | cj | er, sorry |
| 18:49.22 | WIMPy | Might be something missing in SipAddHeader. |
| 18:49.40 | navaismo | Which file i need to copy to avoid run make menuselect from previous asterisk install? |
| 18:49.51 | cj | oh, I see. that's odd. |
| 18:49.58 | WIMPy | menuselect.makeopts |
| 18:50.19 | cj | this is also not good: [Feb 7 10:49:00] WARNING[7537]: pbx.c:4218 pbx_extension_helper: No application '1,SipAddHeader' for extension (thresh-ser, 12062265809, 5) |
| 18:50.52 | igcewieling | cj: you have a syntax error, maybe an extra spacce where it doesn't belong? |
| 18:51.13 | navaismo | thanks WIMPy |
| 18:51.20 | WIMPy | double priority |
| 18:51.28 | WIMPy | ? |
| 18:52.24 | navaismo | another question its possible to compile dahdi against the new kernel without rebooting the machine? |
| 18:52.28 | cj | yeah, double priority. thanks. |
| 18:53.03 | cj | yeah, that's much better ;-) |
| 18:53.23 | cj | navaismo: if the kernel you are compiling it against is the currently running kernel, it should not need a reboot |
| 18:55.30 | navaismo | nope thats not the newest kernel, after a kernel upgrade i need to reboot and compile again |
| 18:55.46 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 18:56.08 | navaismo | there is a directive like KSRC=newkernelpath can I use in the make cmd? |
| 18:57.12 | cj | yeah, that should probably do. |
| 18:58.02 | navaismo | let me see |
| 19:04.51 | [TK]D-Fender | Kyosh, Those should work, and also there is a "save" option to export out. If those don't work for you for some reason I would change the approach and have your AMI call an update script that will poll your DB and rebuild the contexts accordingly. |
| 19:05.03 | [TK]D-Fender | Kyosh, You could also do this via a DB trigger instead. |
| 19:05.21 | [TK]D-Fender | Kyosh, There are many ways to achieve the end you have describe. |
| 19:06.24 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 19:06.50 | navaismo | cj, the option is KVERS |
| 19:06.59 | navaismo | make KVERS=newkernel |
| 19:07.09 | navaismo | thnaks |
| 19:07.28 | *** join/#asterisk jakent (~jakent@c-71-63-6-140.hsd1.va.comcast.net) |
| 19:08.41 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:08.41 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:09.58 | cj | okay, is there a way to set a variable in friend entries in sip.conf so that the CALLERID(num) will be set to this value? |
| 19:13.09 | [TK]D-Fender | cj, You set the "callerid" |
| 19:15.31 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:15.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:16.45 | cj | [TK]D-Fender: that's the CALLERID(name)... I am looking to set the num |
| 19:18.12 | cj | heya leifmadsen! |
| 19:18.35 | igcewieling | cj: no, that is BOTH |
| 19:18.43 | igcewieling | callerid=The Name <thenum> |
| 19:18.56 | cj | is there a place in the book I can look for all k/v options for entries in sip.conf ? |
| 19:19.03 | cj | igcewieling: oh? neat. |
| 19:19.12 | igcewieling | if you want to set a variable for all calls from that peer you can put setvar=MyVar=My Value in the peer entry. |
| 19:19.35 | *** join/#asterisk rolandow (~roland@546BB29B.cm-12-4c.dynamic.ziggo.nl) |
| 19:19.47 | leifmadsen | ohai :) |
| 19:20.10 | rolandow | hi guys... i switched from asterisk 1.8 to 11.1.. i have directmedia=no in my config, but still it says "Locally bridging" after the call is being setup. |
| 19:20.23 | rolandow | now i don't have audio .. how do i stop it from locally bridging? |
| 19:20.52 | igcewieling | rolandow: either you have an Answer in your dialplan, the T t W w or r option on your dial or the codecs for the two legs of the calls do not match |
| 19:21.09 | igcewieling | oh! nevermind. I read that wrong. |
| 19:21.15 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 19:21.56 | rolandow | is the nevermind for me? :) |
| 19:22.23 | cj | or me?! ohnoes! ambiguity! I can't deal with ambiguity! |
| 19:23.21 | rolandow | actually i don't have any options to my dial . |
| 19:28.16 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 19:28.53 | rolandow | so why doesn't asterisk respect the directmedia=no ? |
| 19:29.33 | [TK]D-Fender | rolandow, show your configs & call |
| 19:29.36 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 19:30.14 | igcewieling | rolandow: it does. Locally Bridging indicates the call is NOT reinvited |
| 19:30.31 | igcewieling | Remotely Bridging would indicate directmedia / reinvites are happening |
| 19:31.06 | rolandow | ok |
| 19:32.17 | *** join/#asterisk andy09usa (~Andrey@audotov.com) |
| 19:37.27 | rolandow | so that's what happens, creating port forwards right before you have to leave :) |
| 19:37.39 | rolandow | wrong ip .. hehe |
| 19:50.40 | cj | carrar: ping |
| 19:52.11 | ketas | hahaha, having 5060 open in public ip is fun |
| 19:52.26 | ketas | lot of calls collecting |
| 19:52.54 | zamba | oh? i haven't got that problem |
| 19:52.55 | *** join/#asterisk igcewieling (~igcewieli@38.sub-70-193-65.myvzw.com) |
| 19:53.16 | zamba | "problem" |
| 19:53.19 | ketas | sadly this is my linphone |
| 19:53.27 | WIMPy | I find it much more interestin to see where they are trying to call. |
| 19:53.35 | ketas | should have asterisk there, recording stuff maybe |
| 19:54.03 | WIMPy | But it would be good if it wasn't that easy to get DOSed that way. |
| 19:54.09 | ketas | voip spammers |
| 19:54.12 | zamba | but you have to have allowguest=yes, right? |
| 19:54.54 | WIMPy | yes |
| 20:01.57 | *** part/#asterisk ra21vi (~ravi@122.177.161.195) |
| 20:02.01 | *** join/#asterisk mcrane (~mcrane@24-117-99-123.cpe.cableone.net) |
| 20:13.44 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 20:21.25 | chris_n | how does one get rid of this log entry: http://pastebin.com/B6FEGMmW |
| 20:21.44 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:21.44 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:23.15 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:23.15 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:23.54 | ChannelZ | That's from PHP it looks like |
| 20:26.33 | chris_n | yeah, it looks like somebody is passing incorrect tz foo to a log function in phpagi-asmanager.php |
| 20:26.53 | chris_n | is not sure what that script does |
| 20:27.10 | ChannelZ | you might be able to mute it by calling @date |
| 20:28.24 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 20:29.26 | *** join/#asterisk igcewieling (~igcewieli@38.sub-70-193-65.myvzw.com) |
| 20:33.02 | *** part/#asterisk mcrane (~mcrane@24-117-99-123.cpe.cableone.net) |
| 20:35.49 | SeRi | man I love the Polycom 670. |
| 20:36.25 | zamba | SeRi: poh? |
| 20:36.27 | zamba | oh* |
| 20:36.36 | SeRi | lol :) |
| 20:37.17 | SeRi | zamba: yeap is nice. not to crazy about the web interface but is ok. |
| 20:37.26 | SeRi | a whole new learning curve there but not too bad. |
| 20:37.44 | *** join/#asterisk din3sh (~din3sh@196.20.246.82) |
| 20:37.51 | SeRi | migrated my configs over from 3.x to 4.x |
| 20:37.55 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:38.12 | zamba | i'm trying to work out the whole provisioning stuff.. using tftp for it now |
| 20:38.21 | zamba | it kind of works, but if something breaks, i have no idea what to do :) |
| 20:41.54 | SeRi | move over to ftp |
| 20:42.04 | SeRi | I had issues with tfp and moved over to ftp |
| 20:42.16 | zamba | well, i have no problems, really |
| 20:42.18 | SeRi | s/tfp/tftp/ |
| 20:42.22 | zamba | but i'm not sure about the whole process |
| 20:42.44 | SeRi | read the admin guide |
| 20:43.01 | SeRi | is the best admin guide I have ever read. |
| 20:43.06 | SeRi | complete |
| 20:43.08 | zamba | SeRi: link? |
| 20:43.13 | SeRi | one sec |
| 20:43.57 | zamba | because i'm also curious about upgrading the firmware on the phones |
| 20:44.46 | SeRi | zamba: http://support.polycom.com/PolycomService/support/latinamerica/support/voice/soundpoint_ip/soundpoint_ip670.html |
| 20:45.27 | zamba | which one of the PDFs? |
| 20:45.39 | SeRi | go to the bottom |
| 20:45.42 | SeRi | the admin guide |
| 20:45.50 | zamba | different ones there |
| 20:45.56 | zamba | UCS? what's that? |
| 20:46.00 | zamba | vs SIP? |
| 20:46.25 | SeRi | SoundPoint IP, SoundStation IP and Polycom VVX Administrator?s Guide - UCS 3.3.0 |
| 20:46.55 | SeRi | sip is the old firmwares |
| 20:46.58 | SeRi | now is UC |
| 20:48.05 | leifmadsen | UC ftw |
| 20:48.06 | zamba | aha, ok |
| 20:48.13 | leifmadsen | 4.x is the new hotness |
| 20:49.18 | zamba | which devices can run 4.x? |
| 20:49.28 | Kobaz | only new new ones |
| 20:49.31 | Kobaz | 331 550 650 |
| 20:49.35 | Kobaz | etc |
| 20:49.45 | zamba | can i list the different devices currently registered at my asterisk? |
| 20:49.56 | Kobaz | sip show peers iax2 show peers |
| 20:50.11 | zamba | yeah, but what they're running |
| 20:50.21 | Kobaz | sip show peer xxxx |
| 20:50.36 | zamba | PolycomSoundPointIP-SPIP_650-UA/4.0.2.11307 |
| 20:50.41 | zamba | i guess i'm already running 4.x :) |
| 20:50.41 | Kobaz | <PROTECTED> |
| 20:50.42 | SeRi | leifmadsen: +1 |
| 20:50.42 | Kobaz | etc |
| 20:52.07 | zamba | but the phones will only pick the software they're able to run, right? |
| 20:52.33 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:53.39 | SeRi | zamba: Yes. |
| 20:54.00 | zamba | cool |
| 20:54.02 | SeRi | I use split |
| 20:54.16 | SeRi | clean out the uneccesary stuff |
| 20:54.22 | SeRi | and let it rip! |
| 21:01.37 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.225.166) |
| 21:05.25 | *** join/#asterisk DarthExpeditor (~DarthExpe@rrcs-71-43-76-226.se.biz.rr.com) |
| 21:13.01 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 21:23.33 | zamba | hehe |
| 21:23.45 | zamba | split or combined, right? |
| 21:30.43 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-176-240.red.bezeqint.net) |
| 21:34.29 | cj | is there a way to get asterisk to show me the raw SIP traffic? |
| 21:34.43 | cj | I'm using ipsec, and can't see the outbound SIP traffic in the capture |
| 21:40.08 | cj | ah, sip set debug peer <foo> |
| 21:40.42 | *** join/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 21:41.29 | polysics | hi there |
| 21:41.41 | polysics | I am trying to use UniMRCP with Cepstral nad Asterisk |
| 21:41.52 | polysics | but it seems mrcp-cepstral has simply disappeared |
| 21:42.01 | polysics | does anyone have information about that, please? |
| 21:42.16 | polysics | it is actually missing from the tree |
| 21:43.34 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:54.30 | *** join/#asterisk BrokenArrow (~Lp@unaffiliated/brokenarrow) |
| 21:55.24 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 22:00.11 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 22:03.19 | *** join/#asterisk rdm (rdm@unaffiliated/qubix) |
| 22:07.58 | *** join/#asterisk darkdrgn2k (~DarkPhoni@69-165-131-20.dsl.teksavvy.com) |
| 22:08.00 | darkdrgn2k | Hi all |
| 22:08.08 | darkdrgn2k | can T.38 be used over G729? |
| 22:09.14 | igcewieling | darkdrgn2k: T38 is its own protocol and has nothing whatsoever to do with your voice codec. |
| 22:09.30 | darkdrgn2k | igcewieling: ok so im not crazy :) |
| 22:09.37 | polysics | it does look like mrcp-cepstral is dead. Is that the case, if anyone knows? |
| 22:09.44 | igcewieling | darkdrgn2k: that has not been established yet. |
| 22:09.55 | darkdrgn2k | igcewieling: so why is it that everywhere i see people saying you NEED to set it to g711 for t38 to work? |
| 22:10.36 | igcewieling | because they are wrong. Some people try (and fail) to get FAX over G711 working. |
| 22:10.57 | igcewieling | if you are not using T38 then you do need g711, but it won't work reliably anyway so why even bother with g711 |
| 22:11.12 | darkdrgn2k | ok so my sip trunk provides support t38 and g729 but not g711 |
| 22:11.43 | igcewieling | change providers. Any provider which does not support g711ulaw is not a provider you want to use. |
| 22:12.03 | darkdrgn2k | i thought g729 was "better" |
| 22:12.27 | igcewieling | darkdrgn2k: there is no "better" there is only "better at X" or "better in situation X" |
| 22:12.45 | igcewieling | g729 is "better" (lower) bandwidth usage. |
| 22:12.54 | leifmadsen | g729 is better at being a compressed codec than a fax transport |
| 22:13.07 | igcewieling | g711 ulaw and alaw are "better" audio quality. |
| 22:13.24 | darkdrgn2k | ok anyway back to the original question |
| 22:13.31 | leifmadsen | asked and answered |
| 22:13.33 | darkdrgn2k | <PROTECTED> |
| 22:13.42 | darkdrgn2k | are they smoking something? |
| 22:13.43 | leifmadsen | because they aren't using T.38 |
| 22:13.47 | darkdrgn2k | yes they are |
| 22:13.54 | igcewieling | darkdrgn2k: because when you are using ANALOG or T-1 it must be ulaw |
| 22:14.01 | igcewieling | but we are talking VoIP here right? |
| 22:14.07 | darkdrgn2k | yes |
| 22:14.36 | igcewieling | darkdrgn2k: if what they say is true it is entirely a limitation of Dialogic. |
| 22:14.50 | darkdrgn2k | could be just a dumb support guy |
| 22:14.54 | darkdrgn2k | time to make some more tests... |
| 22:15.15 | WIMPy | Sounds like a non voip related statement. |
| 22:15.22 | darkdrgn2k | yep |
| 22:15.24 | igcewieling | darkdrgn2k: the Dialogic device has no analog ports or T-1 ports? |
| 22:15.40 | darkdrgn2k | nop its their breakout(?) sip channels |
| 22:16.02 | igcewieling | so no fax machines connected to it? |
| 22:16.06 | darkdrgn2k | nop |
| 22:16.18 | darkdrgn2k | SR140 |
| 22:16.32 | igcewieling | Dialogic doesn't have much of a reputation for playing nice with...welll... anything |
| 22:16.45 | darkdrgn2k | http://www.dialogic.com/en/products/fax-boards-and-software/foip/sr140.aspx |
| 22:16.58 | darkdrgn2k | Fax transmission ITU T.38; G.711 pass-through T.38 |
| 22:17.18 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 22:17.45 | igcewieling | PASSTHROUGH |
| 22:17.52 | igcewieling | passthru means "when not using T.38" |
| 22:18.10 | darkdrgn2k | but they also have ITU T.38 |
| 22:18.27 | leifmadsen | those are used separately, not at the same time |
| 22:18.28 | igcewieling | right. So if you are not using t.38 then the call must be ulaw. |
| 22:18.33 | leifmadsen | the latter is the failover transport |
| 22:18.37 | darkdrgn2k | so as long as im using ulaw |
| 22:18.45 | leifmadsen | or alaw |
| 22:18.53 | darkdrgn2k | ok goo let me do some testing ,... the do some wireshark caps.. then bang my head against the wall soem more |
| 22:18.55 | igcewieling | as long as you are using ulaw or alaw then you can expect about a 60% success rate. |
| 22:19.05 | darkdrgn2k | and t.38? |
| 22:19.22 | igcewieling | if you are using T.38 in theory success rate might be ah high as 90% according to some guy who wrote an asterisk book. |
| 22:19.31 | darkdrgn2k | i hear 96% |
| 22:19.39 | darkdrgn2k | but 90 is about what i expect from a regular fax machine! |
| 22:20.08 | igcewieling | darkdrgn2k: they are on some really good drugs unless they are counting fail, redial, success as a "success" |
| 22:20.46 | darkdrgn2k | LOL |
| 22:20.56 | darkdrgn2k | i heard incomming is better then outgoing??? |
| 22:21.59 | *** join/#asterisk jsjc (~Adium@164.Red-2-136-102.dynamicIP.rima-tde.net) |
| 22:23.53 | igcewieling | darkdrgn2k: I've never successfully ever gotten a T.38 fax call to go through asterisk. i.e. provider -> asterisk -> t.38 endpoint. We normally use provider -> sip proxy -> t.38 endpoint |
| 22:24.16 | igcewieling | apparently lots of people do, but not us. very vexing |
| 22:24.29 | darkdrgn2k | Well im trying to do everytihng as right as i can |
| 22:24.38 | darkdrgn2k | i got the fax server in a datacenter, same datacetner as our provider |
| 22:24.40 | darkdrgn2k | so ouir RTT is 5ms |
| 22:25.07 | igcewieling | direct provider -> endpoint keeps things simple. |
| 22:25.29 | darkdrgn2k | yeh.. sadly right now its provider -> SBC -> PBX -> Fax SErver |
| 22:25.34 | darkdrgn2k | which sux.. but meh.. |
| 22:27.59 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 22:27.59 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:29.59 | zamba | we have an issue where we need to see if a channel is reachable before deciding which way to send a call.. is this possible with asterisk? |
| 22:30.19 | zamba | if a SIP channel is down, then the call should be routed somewhere else |
| 22:30.20 | SeRi | Yes |
| 22:30.26 | SeRi | I do that |
| 22:30.26 | zamba | how can this be done in the dialplan? |
| 22:30.34 | SeRi | one sec. |
| 22:30.37 | zamba | thanks :) |
| 22:30.47 | ghost75 | manager access is by default md5 capable? |
| 22:30.52 | *** join/#asterisk ponyofdeath (~vladi@cpe-75-80-173-129.san.res.rr.com) |
| 22:31.14 | ponyofdeath | hi, anyone know when the change to fix the google voice reconnect issue will hit v11 ? |
| 22:31.54 | SeRi | zamba: http://pastebin.com/scFEFbKB |
| 22:31.57 | zamba | related to the same issue, we also want to do call transfer in PSTN.. meaning that the call should be forwarded and the call isn't routed through asterisk no more.. anyone familiar with this? |
| 22:32.09 | zamba | oh |
| 22:32.12 | zamba | interesting |
| 22:32.58 | zamba | what is voipms? |
| 22:33.08 | SeRi | my itsp |
| 22:33.15 | zamba | itsp? |
| 22:33.18 | zamba | never heard about :) |
| 22:33.20 | SeRi | ~itsp |
| 22:33.21 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 22:33.25 | zamba | aha |
| 22:33.44 | zamba | so what is really doing the magic here? |
| 22:34.32 | zamba | what's the command that actually checks if stuff is up? "-- voipms status: ..."? |
| 22:35.11 | zamba | that's just the println, right? |
| 22:35.49 | SeRi | zamba: you new to asterisk? |
| 22:36.06 | zamba | well.. kind of :) |
| 22:36.17 | zamba | i'm new to more advanced dialplans :) |
| 22:38.02 | zamba | same => n,ExecIf($["${SIPPEER(${PEERCHECK1},status):0:2}"="OK"]?Set(TRUNKCHECK=1)) |
| 22:38.10 | zamba | this is the entry that actually does the checking, right? |
| 22:39.16 | zamba | http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail |
| 22:39.19 | zamba | can't this be used instead? |
| 22:39.23 | leifmadsen | no one uses that |
| 22:39.27 | leifmadsen | because it's not that reliable |
| 22:40.08 | zamba | http://www.voip-info.org/wiki/view/Asterisk+func+sippeer |
| 22:40.10 | zamba | aha |
| 22:40.28 | leifmadsen | please don't use voip-info either |
| 22:40.28 | zamba | this is the function actually used? |
| 22:40.30 | SeRi | lol |
| 22:40.30 | leifmadsen | it's usually old and outdated, check wiki.asterisk.org instead |
| 22:41.02 | leifmadsen | wiki.asterisk.org actually imports the documentation from the current source code, so it'll always be up to date (within reason) |
| 22:41.25 | zamba | status - Status (if qualify=yes). |
| 22:41.35 | zamba | which means i have to have qualify turned on for this to be reliable, yeaH? |
| 22:41.54 | zamba | but i guess i should have that anyway for the sip trunk to lync |
| 22:46.14 | darkdrgn2k | hmm |
| 22:46.41 | zamba | ok.. i'm not sure if i like wiki.asterisk.org.. i want to find the documentation for sip.conf? |
| 22:47.09 | zamba | checked under both 1.8 and "configuration and operation" |
| 22:47.23 | WIMPy | No, you need qulaify to get any result. |
| 22:47.38 | SeRi | zamba: listen to leifmadsen. I can say that it has worked for me 100% but leifmadsen must have his reason and he is the expert not me. |
| 22:48.14 | leifmadsen | sip.conf is in the source within the configs/sip.conf.sample file |
| 22:48.15 | zamba | SeRi: you're using sippeer.. and that should work fine, shouldn't it? |
| 22:48.48 | zamba | WIMPy: yeah, so i need qualify for the lync trunk.. isn't that recommended anyway? |
| 22:48.48 | SeRi | yes. works fine for me. |
| 22:49.30 | WIMPy | Only if you want to know the current status. |
| 22:49.39 | zamba | i definitely need to know that |
| 22:49.44 | darkdrgn2k | correct me if im wrong but is this an issue with coded negotiations ? http://pastebin.ca/2311652 |
| 22:50.52 | Kyosh | [TK]D-Fender: add extension in the CLI does not work in realtime. it takes the command fine, but does not create the new extension. also there is no way from here to create a new context as it gives an error if the context does not exist. also, AMI UpdateConfig is simply not working well under 1.4.21 as you can see here: http://pastebin.ca/2311653 i am wondering if this is an effect of realtime. |
| 22:52.03 | zamba | is it possible to use dialplan functions without actually writing a dialplan for them? |
| 22:52.14 | zamba | let's say i'm in the CLI and just want to test the SIPPEER function? |
| 22:58.23 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 23:00.26 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.22.5) |
| 23:01.01 | zamba | how do i disable the loading of extensions.ael? |
| 23:01.14 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:02.07 | navaismo | hmm IIR is : noload => pbx_ael.so |
| 23:02.15 | navaismo | in the modules.conf |
| 23:04.27 | zamba | ah! thanks |
| 23:15.19 | darkdrgn2k | what does "telephone-event/8000" mean? |
| 23:16.45 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 23:30.41 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 23:36.45 | *** join/#asterisk darkdrgn2k (~DarkPhoni@69-165-131-20.dsl.teksavvy.com) |
| 23:44.21 | blizzow | Is there a way to up the verbosity while leaving asterisk running? |
| 23:44.31 | blizzow | log verbosity that is. |
| 23:47.18 | blizzow | slaps forehead. |
| 23:47.27 | blizzow | lmgtfy.com |
| 23:49.46 | darkdrgn2k | correct me if im wrong but doesnt this show that 16 rejected t38 http://imagebin.org/245869 |
| 23:50.04 | darkdrgn2k | <blizzow>: asterix -r then sip set debug on ?????? |
| 23:51.14 | blizzow | darkdrgn2k: core set verbose X (where X is the level of verbosity) |
| 23:51.23 | darkdrgn2k | well that too :-P |
| 23:51.39 | blizzow | or for debug messages (core set debug X) |
| 23:51.47 | mjordan | darkdrgn2k: 170 rejected the T38 offer. |
| 23:52.06 | darkdrgn2k | 70 did or 16 ? |
| 23:52.15 | darkdrgn2k | oo yeh 70 did |
| 23:52.16 | darkdrgn2k | !@#%@%@%# |
| 23:52.16 | mjordan | sorry, 70 |
| 23:52.17 | mjordan | :-) |
| 23:52.22 | mjordan | 16 did the re-INVITE |
| 23:52.30 | darkdrgn2k | 70 supports t38 damit |
| 23:52.40 | darkdrgn2k | have i told you how much i HATE sip! |
| 23:52.49 | mjordan | don't hate SIP, hate fax |
| 23:52.56 | darkdrgn2k | nop... itps sip |
| 23:53.22 | mjordan | hm... if it's a blamefest of SIP versus Fax, I'm going to vote for fax, but that's just me |
| 23:53.26 | darkdrgn2k | had a 2 hour confence call arguing whos fault it is that rtp ports are changing and now being dealt with properly! |
| 23:53.53 | darkdrgn2k | fax is the faulout from that.. cause G711 was turned off.. but t38 SHOULD be working.. |
| 23:55.43 | igcewieling | heh, our initial failures on t.38 were because I forgot to open the udptl ports (4000-4999) in iptables |
| 23:55.58 | igcewieling | then it went downhill from there. |
| 23:56.26 | darkdrgn2k | im just ready to cry! |
| 23:56.38 | darkdrgn2k | so 70 is saying "i dont speak 38" right? |
| 23:59.48 | darkdrgn2k | ? |
| 23:59.55 | igcewieling | Not many people know this, but the ancient Buddhist monks used T.38 to train novices in "infinite patience" |