| 00:00.03 | NirS | use pastbin.com or pastebin.ca to do so |
| 00:00.10 | Scar-G | ok |
| 00:01.01 | NirS | btw, how's the weather in Karachi these days ? |
| 00:01.02 | Scar-G | by the dialplan, you mean the extensions_additional.conf ? |
| 00:01.06 | NirS | yupp |
| 00:01.16 | Scar-G | it's a bit cold |
| 00:01.31 | Scar-G | cold for us but i think normal for you guys |
| 00:01.38 | NirS | well, where I am, we have shit storm running outside |
| 00:01.46 | Scar-G | ooo |
| 00:02.56 | robl^laptop | shit storm?!?! that brings up very unpleasant imagery, |
| 00:03.28 | Scar-G | :D |
| 00:05.28 | Scar-G | http://pastebin.com/Ckp1Z51A |
| 00:07.56 | Scar-G | and |
| 00:07.56 | Scar-G | http://pastebin.com/0ST8Qiwt |
| 00:08.34 | NirS | brb |
| 00:08.39 | Scar-G | okiedokie |
| 00:08.40 | *** join/#asterisk TimeRider (~steve@204.93.201.17) |
| 00:10.19 | NirS | well |
| 00:10.30 | NirS | it's never too cold in Israel, but this winter is really cold |
| 00:10.41 | Scar-G | hmmmmmm |
| 00:10.46 | NirS | btw, scar, what context should I be looking at ? |
| 00:10.54 | Scar-G | macro-dialout-trunk |
| 00:11.53 | NirS | ok, where's the bit that you are referring to exactly ? |
| 00:11.55 | NirS | which line ? |
| 00:12.05 | Scar-G | let me paste |
| 00:12.30 | Scar-G | exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM},300,${DIAL_TRUNK_OPTIONS}) |
| 00:12.40 | NirS | I guess you are referring to the confirm thing that I'm looking at ? |
| 00:12.48 | Scar-G | DIAL_TRUNK_OPTIONS = D(12345:12345) |
| 00:13.14 | Scar-G | perhaps you need to look at extensions.conf too ? |
| 00:13.34 | NirS | no, I can see the globals section, that's finer |
| 00:13.43 | Scar-G | okay. |
| 00:14.10 | NirS | so, what you say is this, you want to dial the call, once the call is answered and bridged, you want to issue a set of DTMF sequences, right ? |
| 00:14.20 | Scar-G | YES |
| 00:14.24 | Scar-G | Answered AND BRIDGEd |
| 00:14.41 | Scar-G | but we are only able to send when it's answered AND NOT bridged |
| 00:14.49 | NirS | I need to understand something |
| 00:14.52 | Scar-G | After bridged. |
| 00:14.54 | Scar-G | ? |
| 00:15.13 | NirS | you pick up the phone, dial a number and once both sides are talking you want to issue a DTMF sequence ? |
| 00:15.18 | Scar-G | yes |
| 00:15.23 | Scar-G | exactly |
| 00:15.33 | NirS | I fail to see the logic in that, can you please elaborate ? |
| 00:15.49 | Scar-G | I just want to do it using asterisk. |
| 00:16.16 | Scar-G | Explaining the logic would be a bit complex. |
| 00:16.20 | Scar-G | :D |
| 00:17.26 | NirS | well, I'm not really sure you can do that |
| 00:17.31 | Scar-G | :( |
| 00:17.49 | *** join/#asterisk haroldp (~Digger@99-46-24-87.lightspeed.renonv.sbcglobal.net) |
| 00:17.56 | haroldp | hello |
| 00:18.01 | NirS | as far as I know, once a call is bridged, in order to send DTMF's on the bridge, it needs to originate from one of the sides |
| 00:18.05 | NirS | hi haroldp |
| 00:18.20 | Scar-G | Hello haroldp |
| 00:18.48 | Scar-G | This guys here says that it can be done VIA AMI : http://forums.digium.com/viewtopic.php?f=13&t=79213&sid=fbe9cd270deb291716e05ed75de3de1c |
| 00:19.13 | Scar-G | to put it in one line : |
| 00:19.14 | Scar-G | http://forums.digium.com/viewtopic.php?f=13&t=79213&sid=fbe9cd270deb291716e05ed75de3de1c |
| 00:20.12 | Scar-G | bit.ly/zRoa53 |
| 00:20.25 | haroldp | I just got a new SIP trunk (new voip provider? jargon?) and updated extensions.conf and sip.conf with the info. I can call out, but when calling in askerisk logs the error, "chan_sip.c:14205 check_auth: username mismatch, have <MYUSER>, digest has <s>" |
| 00:20.26 | NirS | "Provided that DTMF sending is complete before the speech path is established, there should be many ways." this means - prior to the bridge being established |
| 00:20.46 | haroldp | what obvious thing am I screwing up? :) |
| 00:20.52 | Scar-G | the last line says : If you want to do it after the speech path is established (bridged), I think you will need to use AMI. |
| 00:21.03 | NirS | harold is your provider setup with a static IP number? |
| 00:21.41 | haroldp | um... I dunno. as in, did I configure my IP with them? or? |
| 00:22.19 | NirS | there IP number |
| 00:22.24 | NirS | their IP number I mean |
| 00:22.34 | NirS | Scar, actually, there may be a way to do so |
| 00:22.46 | Scar-G | I'm listening. |
| 00:23.06 | Scar-G | ( or rather seeing ) |
| 00:23.31 | haroldp | ...I have a hostname for their server. |
| 00:23.35 | NirS | but, in order for this to work, you'll need to originate a call to a Local channel, pass to it the originating channel, then use the SendDTMF application to sent the signals to the original channel |
| 00:23.58 | NirS | and, looks like it will only work on 1.8.X and upwards |
| 00:24.03 | Scar-G | hmmmmm |
| 00:24.23 | NirS | harold, do you have insecure=port,invite as part of your trunk configuraiton ? |
| 00:24.39 | NirS | Scar, let's put it this way - this won't be easy |
| 00:24.47 | haroldp | I have insecure=very |
| 00:24.56 | Scar-G | hmmmmm hmmmmm |
| 00:26.11 | NirS | well, that should be enough in general |
| 00:26.25 | Scar-G | thanks mate |
| 00:26.26 | NirS | can you paste your trunk configuration to pastebin.com |
| 00:26.54 | NirS | you're welcome |
| 00:27.12 | NirS | harold, what kind of switch are trying to connect to exactly? who is this provider ? |
| 00:28.25 | NirS | BRB |
| 00:28.34 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 00:36.34 | NirS | I'm back |
| 00:37.11 | haroldp | NieS: my provider is voip.ms |
| 00:38.02 | NirS | let's take a look |
| 00:38.30 | NirS | and you are using SIP ? |
| 00:38.33 | haroldp | ok, sorry to be slow. everything wants to break at the same time :) |
| 00:38.42 | haroldp | yes, SIP |
| 00:39.45 | NirS | can you show me your configuration ? |
| 00:40.52 | NirS | are you using their username based authetication or IP based authentication ? |
| 00:41.10 | haroldp | http://pastebin.ca/2119190 |
| 00:41.17 | haroldp | that is sip.conf |
| 00:41.46 | haroldp | I replaced my voip.ms username with MYUSER and my voip.ms password with **s |
| 00:44.25 | NirS | well, you should remove the [authentication] part of the config, it's not required for this trunk |
| 00:44.47 | haroldp | ok. I'll try that. |
| 00:44.57 | NirS | take a look here - taken from their website |
| 00:44.57 | NirS | http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 |
| 00:46.04 | haroldp | yeah. |
| 00:46.25 | haroldp | I was hoping to replace old auth info with new, but oh well :) |
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| 00:59.25 | haroldp | I'm making progress. I have all new errors :) |
| 01:00.15 | NirS | put on pastebin |
| 01:03.32 | haroldp | chan_sip.c:22461 handle_request_invite: Call from 'MYUSER' (67.215.241.250:5060) to extension 'MYVOIPDID' rejected because extension not found in context 'mycontext'. |
| 01:03.47 | NirS | that's good |
| 01:03.54 | NirS | that means that the call was accepted by Asterisk |
| 01:04.12 | NirS | however, the DID you are dialing isn't configured in the associated context |
| 01:05.18 | haroldp | yet there it is. obviously i am still doing something wrong |
| 01:09.10 | NirS | that means that your inbound context is incomplete |
| 01:09.13 | haroldp | here's a excerpt from my extensions.conf: http://pastebin.ca/2119222 |
| 01:09.57 | NirS | try doing a 'sip set debug on' then see what DID they are sending the call to |
| 01:10.10 | NirS | maybe you have to tweak it a bit on your side |
| 01:10.17 | haroldp | ok |
| 01:11.52 | haroldp | looks right |
| 01:13.37 | NirS | please put on pastebin |
| 01:13.55 | NirS | in addition, please put the output of your console on pastebin |
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| 01:29.48 | [TK]D-Fender | haroldp: Doesn't exist... just like it says |
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| 01:58.16 | dijib | anyone in need of an FXO? |
| 01:58.18 | *** part/#asterisk wesphillips (~wphill04@c-76-30-173-12.hsd1.tx.comcast.net) |
| 01:58.44 | dijib | one of these times someones going to say yeh! right here, pay you $150 for your sangoma card dijib |
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| 02:26.13 | dijib | ok guys problem with startx on centos |
| 02:26.19 | dijib | all text is blocks. |
| 02:26.21 | dijib | what do? |
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| 02:52.08 | p3nguin | dijib: Do you have an xorg.conf? |
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| 03:31.17 | SeRi | wow is been a while. whats going on guys. |
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| 03:45.52 | haroldp | failed to set the 'context' right in sip.conf |
| 03:45.58 | haroldp | fixed and working now. |
| 03:48.51 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
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| 05:25.40 | dijib | p3nguin: no i dont have an xorg.conf |
| 05:25.55 | dijib | install xorg? |
| 05:26.39 | dijib | SeRi: you in here? |
| 05:28.29 | p3nguin | dijib: Xorg can work without one, usually, but try to create one and see if it succeeds. As root, run "xorg -configure" |
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| 06:04.39 | dijib | 4~i lost |
| 06:04.44 | dijib | work on it tomorrow |
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| 08:24.32 | *** join/#asterisk shihan (shihan@2600:3c01::f03c:91ff:fe96:8f5c) |
| 08:26.31 | shihan | howdy all, im a bit stuck... i have a simple setup, a few internal sip phones and an upstream voip provider, calls between the sip phones work, calls inbound from the voip provider work, but i cant get calls outbound working... it appears to try and send the call to the provider, but end up with this "chan_sip.c: Failed to authenticate on INVITE to...." and i dont know what im missing really |
| 08:26.57 | shihan | asterisk 1.8.4 btw |
| 08:34.46 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-90.telkomadsl.co.za) |
| 08:42.50 | ChrisInSydney | shihan: Pastebin your relevant sip.conf. Change the passwords !!! |
| 08:43.20 | ChrisInSydney | ill have a look |
| 08:43.53 | p3nguin | If it happens in the next few minutes, I might look. |
| 08:44.28 | ChrisInSydney | p3nguin: got one for you |
| 08:45.20 | ChrisInSydney | the nat security nag that comes up if you have name=yes in general and nat=no in the peer definition |
| 08:46.32 | ChrisInSydney | whats now best practice if you have your box behind nat and handsets routable, and vice versa, where you have your handsets behind NAT and a public IP |
| 08:46.34 | ChrisInSydney | ?? |
| 08:47.31 | shihan | thanks, here ya go: http://pastebin.com/gqMvSPSu |
| 08:47.39 | ChrisInSydney | Previously, I have been using different nat=yes / no depending on where the box was. with 1.8.9 it now complains |
| 08:48.14 | ChrisInSydney | shihan: iinet ? an aussie too ? |
| 08:49.04 | shihan | yeah :) |
| 08:49.12 | ChrisInSydney | where ? |
| 08:49.20 | shihan | same place, sydney :) |
| 08:49.58 | ChrisInSydney | cool, I'm in the eastern beaches area :-) |
| 08:50.22 | shihan | heh, im more out west, chatswood :) |
| 08:50.33 | ChrisInSydney | Chastwood is north |
| 08:51.03 | ChrisInSydney | used to live in Chatswood west, down near the park on lady game drive |
| 08:51.05 | ChrisInSydney | nice area |
| 08:51.16 | ChrisInSydney | close to stuff |
| 08:51.28 | shihan | oh right, i was thinking manly area when you said eastern beaches.. doh |
| 08:51.41 | shihan | actually im really close to there |
| 08:51.42 | ChrisInSydney | nah, randwick / coogee area |
| 08:52.08 | ChrisInSydney | anyways, |
| 08:52.46 | shihan | up on johnson street |
| 08:53.00 | ChrisInSydney | ahh |
| 08:54.05 | ChrisInSydney | Is that your entire sip.conf, or just the relevant bits ?? |
| 08:55.01 | shihan | the entire thing |
| 08:55.14 | shihan | i tried to simplify it cause i was getting rather confused |
| 08:55.23 | ChrisInSydney | K, I'm going to throw some stuff at it |
| 08:56.04 | ChrisInSydney | put the [general] at the top. most people expect it there. Small thing |
| 08:56.14 | ChrisInSydney | disallow=all |
| 08:56.15 | ChrisInSydney | allow=alaw |
| 08:56.25 | ChrisInSydney | You should only need alaw. |
| 08:56.38 | ChrisInSydney | gsm is a bit crap and I dont think iinet support it directly |
| 08:56.50 | shihan | k |
| 08:56.52 | ChrisInSydney | alaw is standard for aus |
| 08:57.32 | ChrisInSydney | put the same in your extension setting |
| 08:57.37 | ChrisInSydney | so you match codecs |
| 08:58.16 | ChrisInSydney | Now, I am assuming, like sensible people, your asterisk box is behind NAT, so you will have to put a local net and externip setting |
| 08:58.29 | ChrisInSydney | that helps the audio get through |
| 08:58.43 | shihan | well, actually i moved it into my router, poor choice? |
| 08:58.56 | ChrisInSydney | ahh, what router ?? |
| 08:59.11 | shihan | its a generic x86 box running openwrt |
| 08:59.45 | ChrisInSydney | thats cool, you will just need a secure dialplan, and it wouldn't hurt to run fail2ban |
| 09:00.15 | ChrisInSydney | so you dont need the externip stuff as you have probably bound to 0.0.0.0 |
| 09:00.33 | shihan | yeah, currently is bound to 0.0.0.0 |
| 09:00.55 | ChrisInSydney | so its directly addressable from inside and outside the lan |
| 09:01.07 | shihan | yep |
| 09:02.06 | ChrisInSydney | http://whirlpool.net.au/wiki/iinetphone_asterisk |
| 09:02.10 | ChrisInSydney | Whirlpool rocks |
| 09:02.29 | shihan | actually, i tried that very config and mine just kept timing out registering |
| 09:03.03 | ChrisInSydney | but there are only some bits that are relevant in that post |
| 09:03.18 | shihan | k |
| 09:04.06 | ChrisInSydney | Did you see: Note:' IP Address of your state SIP server' is the ip address in dotted notation. Asterisk does not currently operate with iiNet if you use a host name here. A side effect is you will get the 'number disconnected message' when you dial a known good number if you use a name and not a dotted address. |
| 09:05.07 | ChrisInSydney | That could be your "challenge" |
| 09:05.17 | shihan | ahhh, no, i missed that bit |
| 09:06.05 | shihan | oh, no, i did read that bit, but ended up going "oh, its 1.6" |
| 09:06.33 | WIMPy | Asterisk has always supported hostnames. |
| 09:06.38 | ChrisInSydney | I havent got any clients on iinet for calls, hang on, I may have one..... |
| 09:06.45 | WIMPy | They must have some sort of DNS issue if that doesn;t work. |
| 09:06.50 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 09:07.18 | ChrisInSydney | that one was on internode |
| 09:07.47 | ChrisInSydney | https://iihelp.iinet.net.au/VoIP_settings |
| 09:07.50 | shihan | oky doky, changed it to ip address.. still the same thing with oubound calls |
| 09:07.51 | ChrisInSydney | just looking at this |
| 09:08.15 | ChrisInSydney | pastebin the error in the cli |
| 09:09.09 | ChrisInSydney | duh, just found it !!! |
| 09:09.16 | ChrisInSydney | no username= |
| 09:09.36 | shihan | http://pastebin.com/vPkm4aCQ |
| 09:09.37 | ChrisInSydney | dns shoudl be OK |
| 09:10.20 | shihan | oh wow, thats embarresiing |
| 09:10.21 | shihan | :) |
| 09:10.26 | shihan | working now |
| 09:10.36 | ChrisInSydney | put a username=02987654342 in there |
| 09:11.16 | ChrisInSydney | thats cool. Whats embarrasing is hacking asterisk on a saturday night in Sydney !!! |
| 09:11.32 | shihan | sad, but true :) |
| 09:11.42 | ChrisInSydney | so what sort of handsets you got ? |
| 09:12.18 | shihan | acutally, all software, 2 android sipdroid's, a mythtv box (eventually) and a desktop client here (linphone) |
| 09:12.39 | *** join/#asterisk nzkiwi1 (~michaelnz@ip210-48-102-29.nettrust.net.nz) |
| 09:12.58 | nzkiwi1 | hi |
| 09:13.07 | nzkiwi1 | is this the best place to be for support? |
| 09:13.24 | ChrisInSydney | cool. Been using Snom mostly. Nice, but a little exe. They come up on ebay reasonably often and they are easy to set up |
| 09:13.41 | ChrisInSydney | this, the forums, the wiki |
| 09:13.44 | ChrisInSydney | ~book |
| 09:13.44 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 09:13.49 | nzkiwi1 | tried the forums |
| 09:13.59 | nzkiwi1 | spent hours on Google |
| 09:14.26 | nzkiwi1 | My phone is unreachable when using TLS |
| 09:14.38 | nzkiwi1 | with UDP is fine |
| 09:14.42 | shihan | i have one of these as well, http://www.telephonemagic.com/nortel-i2007-voip-phone.htm, but i really have no intention of using it i dont think... requires poe or an adapter and i just cant quite be bothered with it atm |
| 09:15.11 | shihan | i havent seen snom before tho |
| 09:15.17 | ChrisInSydney | shihan, you will need to run it with unistim I think |
| 09:15.23 | ChrisInSydney | they aren't exactly SIP |
| 09:15.44 | shihan | yeah, i did get it running once... wasnt hard... but that was a pretty simple setup at work on a poe switch |
| 09:15.53 | nzkiwi1 | someone must have seen this before and solved this? |
| 09:16.02 | shihan | i was really quite annoyed when i realised the phone didnt come with a power adapter |
| 09:16.09 | ChrisInSydney | nzkiwi1: Ive never done TLS, but it sounds like a certificate issue. You getting any authentication attempts ?? |
| 09:16.23 | ChrisInSydney | shihan: from ebay ? |
| 09:16.37 | nzkiwi1 | asterisk works fine when talking TLS to upstream provider |
| 09:16.55 | ChrisInSydney | I think you can pick up a tplink 4+4 PoE switch for around $60 |
| 09:17.11 | shihan | chris, naah, this was "here, have this" from a guy i was working with at the time who was doing a network rollout and replacing nortel eq with cisco |
| 09:17.16 | nzkiwi1 | with tls can make calls from phone but not receive them (asterisk thinks phone is unavailable) it does register tho |
| 09:17.37 | nzkiwi1 | I have a 16 port level 1 switch with PoE they are ok |
| 09:18.07 | ChrisInSydney | Level 1. They work. They're cheap too |
| 09:18.21 | shihan | i havent had a good reason to bother plugging it in at home yet... so far im hoping the sipdroid software phones on android will perform ok |
| 09:18.46 | ChrisInSydney | shihan: Media5 is free, if you pay for it, you get g722 ! |
| 09:19.08 | nzkiwi1 | with TLS does the phone need a server cert? |
| 09:19.27 | shihan | oh, its on android as well, i didnt realise |
| 09:20.11 | ChrisInSydney | nzkiwi1: From my verrrry vague understanding, it needs the CA and plus its own public and private |
| 09:20.29 | ChrisInSydney | I may be talking (typing) crape here though |
| 09:21.11 | ChrisInSydney | nzkiwi1 So if you do a sip show peers, unavailable ? |
| 09:21.29 | nzkiwi1 | I have installed the CA cert and even tried a server cert |
| 09:21.42 | nzkiwi1 | sip show peers = unavail |
| 09:22.07 | nzkiwi1 | installing the server cert makes no diff |
| 09:22.21 | ChrisInSydney | nzkiwi1: what is the handset ? |
| 09:22.30 | nzkiwi1 | yealink T28p |
| 09:22.58 | ChrisInSydney | good, cheap phone, looks awefully like a Cisco, but with buttons |
| 09:23.02 | ChrisInSydney | latest firmware ? |
| 09:23.10 | nzkiwi1 | v61 f/w |
| 09:23.15 | nzkiwi1 | yes |
| 09:23.54 | ChrisInSydney | do you need TLS on the phone ? (easy fix / lets me avoid the question) |
| 09:24.18 | nzkiwi1 | right now no but I need to get it working so I am not stuffing around when I need it |
| 09:24.31 | ChrisInSydney | Could you just get away with [S|Z]RTP ? |
| 09:24.40 | ChrisInSydney | fair enough |
| 09:25.02 | nzkiwi1 | plus I am just about to buy more of these (unless I cant get it working) |
| 09:25.04 | ChrisInSydney | I'll have a quick look at my Snom |
| 09:25.39 | nzkiwi1 | SRTP works (even though asterisk shows errors in the cli) but I am led to believe is ineffective without tls |
| 09:27.18 | ChrisInSydney | you are stretchng me with this. I have just reset my Snom870. I'll have a lok there. I also have a yealink t38 but its not plugged in anywhere meaningful |
| 09:27.50 | nzkiwi1 | I just ordered a T38 though will cancel if I cant get the T28 working |
| 09:28.14 | nzkiwi1 | not spending $380 for something that doesnt 100% work |
| 09:28.44 | ChrisInSydney | Cost me Aus $190 for two |
| 09:28.51 | nzkiwi1 | which model? |
| 09:28.56 | ChrisInSydney | t38s |
| 09:29.03 | ChrisInSydney | special last year |
| 09:29.07 | nzkiwi1 | how the hell you manage that? |
| 09:29.16 | nzkiwi1 | that is a major special |
| 09:29.20 | ChrisInSydney | dealer XP |
| 09:29.33 | nzkiwi1 | even in ozzy I would pay over $200 AU$ |
| 09:29.41 | ChrisInSydney | not any more |
| 09:29.44 | ChrisInSydney | yup |
| 09:29.49 | nzkiwi1 | dealer price here $330 + 15 % GST |
| 09:29.54 | nzkiwi1 | NZ$ |
| 09:30.03 | ChrisInSydney | forgot the 10% tax |
| 09:30.08 | ChrisInSydney | ouch |
| 09:30.21 | nzkiwi1 | it went up from 12.5 - 15% |
| 09:30.28 | ChrisInSydney | how much for a Snom 870, they must hurt |
| 09:30.33 | nzkiwi1 | under a tory govt too |
| 09:30.50 | shihan | chris, thanks for that, thats working really well now, thats awesome |
| 09:31.14 | nzkiwi1 | snom 821 is 375+gst |
| 09:31.27 | nzkiwi1 | snom 870 not listed |
| 09:31.42 | ChrisInSydney | shihan: Just make sure that your inbound context only allows inbound, and you ip limit your handset registrations |
| 09:31.53 | ChrisInSydney | who is the wholesaler ?? |
| 09:32.09 | nzkiwi1 | snappernet.co.nz |
| 09:33.00 | nzkiwi1 | I considered getting it from ozzy but if I buy local I can trial it for a few days 1st |
| 09:33.09 | ChrisInSydney | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 09:33.23 | nzkiwi1 | I have gone thru all of that and more... |
| 09:33.28 | ChrisInSydney | I thjink Alloy are in NZ too, bnut ythey can be expensive |
| 09:34.12 | nzkiwi1 | yealink.co.nz -> an iozzy site |
| 09:34.17 | nzkiwi1 | ozzy |
| 09:34.37 | ChrisInSydney | I think I have found some cheap Cisco 7971G-GEs, but they are a bit of a dog of a phone |
| 09:35.32 | ChrisInSydney | does it need a .pem, or a crt/key combination ?? |
| 09:35.52 | nzkiwi1 | one file. .pem works |
| 09:36.19 | ChrisInSydney | have you tried with a softphone ?? |
| 09:36.40 | nzkiwi1 | no because I wont be using a soft phone anyway |
| 09:36.44 | ChrisInSydney | shihan: Cool.Gald to have helped |
| 09:37.09 | ChrisInSydney | I guess, get it working somehow, then make sure the Yeakink isnt screwed |
| 09:37.15 | ChrisInSydney | firmware wise |
| 09:37.26 | nzkiwi1 | I produced a .pem file from the CA cert and imported it to the yealink. It imports fine but doesnt work |
| 09:37.40 | ChrisInSydney | Lets you check that the certs are good |
| 09:38.24 | WIMPy | Great. My Line has one-way audio. |
| 09:38.33 | nzkiwi1 | i used the script included with asterisk and the by hand method. either way with same results |
| 09:38.53 | ChrisInSydney | WIMPy: depends on which way, you could give it to my ex wife |
| 09:39.03 | nzkiwi1 | f/w rules? |
| 09:39.09 | ChrisInSydney | hmmm |
| 09:39.15 | ChrisInSydney | you behind NAT ?? |
| 09:39.20 | nzkiwi1 | me? |
| 09:39.24 | ChrisInSydney | yes |
| 09:39.38 | ChrisInSydney | handset <-> NAT <-> Ast |
| 09:39.41 | nzkiwi1 | no. asterisk has public IP and phones are on LAN to asterisk |
| 09:39.56 | ChrisInSydney | dual home |
| 09:39.59 | nzkiwi1 | yes |
| 09:40.14 | nzkiwi1 | phones are on 192 IPs but no nat |
| 09:40.41 | nzkiwi1 | pbx is directly on public IP out of our range |
| 09:40.51 | ChrisInSydney | when you boot the phone, do you see anything in the CLI ?? |
| 09:41.10 | nzkiwi1 | nothing unusual |
| 09:41.17 | nzkiwi1 | do u want em to try? |
| 09:41.23 | ChrisInSydney | no authentiaton attempts ? |
| 09:41.33 | nzkiwi1 | auth attempts come after boot |
| 09:41.38 | ChrisInSydney | yep. Reboot the phone and watch |
| 09:41.54 | nzkiwi1 | welcome inititalising |
| 09:41.59 | nzkiwi1 | pls wait |
| 09:42.16 | p3nguin | chrisinsydney: You either enable nat or disable nat in the general section only. |
| 09:42.41 | nzkiwi1 | normal screen |
| 09:42.45 | nzkiwi1 | auth attempst now |
| 09:42.50 | nzkiwi1 | all done |
| 09:42.54 | p3nguin | chrisinsydney: The nat setting doesn't force anything, really, it just makes it nat-capable. |
| 09:43.17 | p3nguin | So if you have asterisk behind nat, nat=yes in the general section. |
| 09:43.29 | ChrisInSydney | p3nguin: So, if I need some NATed handsets and not NATted trunks or vice versa, I would use nat=yes ?? |
| 09:43.33 | p3nguin | So if you have phones behind a different nat, nat=yes in the general section. |
| 09:44.00 | p3nguin | If you do not have asterisk behind a nat and no phones behind other nat, nat=no. |
| 09:44.14 | ChrisInSydney | if I need NAt then nat=yes and it wont break anything for routable connections |
| 09:44.26 | p3nguin | Think of it like "make this system nat friendly" by setting nat=yes. |
| 09:45.39 | p3nguin | I'm not sure what you mean by routable connections. |
| 09:46.09 | ChrisInSydney | p3nguin: ahh, because I am playing with direct rtp things, using OpenSer on the router and an asterisk on a public IP. I would like to route the SIP through asterisk, but the media to be direct between the handset via OpenSER and the ITSP |
| 09:46.36 | ChrisInSydney | routable = no nat between and pingable |
| 09:46.41 | ChrisInSydney | in both directions |
| 09:47.05 | p3nguin | For example, my phone is connected directly to a cable modem? |
| 09:47.21 | ChrisInSydney | K |
| 09:47.46 | p3nguin | Having nat=yes will not break it. nat=yes just means it will be nat friendly if necessary. |
| 09:48.04 | p3nguin | But you can force it to not be nat friendly by saying nat=no. |
| 09:49.03 | ChrisInSydney | OK then: |
| 09:49.18 | p3nguin | I always forced my providers to nat=no because they are never behind a nat. Now everything is nat=yes because my asterisk is behind a nat and I have other phones on the internet behind other nats. |
| 09:49.18 | ChrisInSydney | ;directmedia=yes |
| 09:49.18 | ChrisInSydney | ;directrtpsetup=yes |
| 09:49.30 | ChrisInSydney | should work even when nat=yes |
| 09:49.58 | ChrisInSydney | p3nguin: Same here |
| 09:50.06 | p3nguin | I have no idea how things work with nat and a proxy together. |
| 09:50.11 | ChrisInSydney | handsets are behind NAT, providers are not. |
| 09:50.19 | ChrisInSydney | set nat=no you get a nag |
| 09:50.41 | p3nguin | But with the changes, I have put nat=yes in general and removed all other instances of the setting from all peers. |
| 09:50.44 | ChrisInSydney | playing with dd-wrt and mikfish / openser |
| 09:51.22 | ChrisInSydney | so best practice, set nat=yes in [general] and forget about the trunks |
| 09:51.22 | p3nguin | # grep ^nat /etc/asterisk/sip.conf |wc -l |
| 09:51.23 | p3nguin | 1 |
| 09:52.20 | ChrisInSydney | nzkiwi1: So what happened, you could see the auth ? does it work now ?? |
| 09:52.58 | p3nguin | Comparing against an old sip.conf backup I have, it has 18 lines of nat settings in it. |
| 09:53.09 | p3nguin | phones, providers, etc. |
| 09:53.27 | ChrisInSydney | standard on every peer / user |
| 09:53.28 | p3nguin | Huge difference. |
| 09:53.31 | ChrisInSydney | for me anyway |
| 09:53.42 | ChrisInSydney | maybe no longer |
| 09:53.55 | nzkiwi1 | chris - it registers ok, it is unreachable |
| 09:54.00 | p3nguin | Yes, I have a nat setting on almost every peer entry in that old file. But now I have only the one in the general section. |
| 09:54.07 | nzkiwi1 | just tested and is unreachable with TCP |
| 09:54.08 | nzkiwi1 | mode |
| 09:54.17 | nzkiwi1 | as well |
| 09:54.24 | p3nguin | TCP? SIP is UDP. |
| 09:56.11 | nzkiwi1 | it is usually udp |
| 09:56.21 | nzkiwi1 | can be set to TCP |
| 09:59.11 | ChrisInSydney | p3nguin: Im in the process of putting our first production system on 1.8, so I'm doing a clean up while I have the opportunity. |
| 09:59.25 | ChrisInSydney | that was one of the cleanups |
| 10:00.54 | ChrisInSydney | I'll stick to te nat=yes, failling that, im going to code up an option nonatnag=yes for the conf in chan_sip.c |
| 10:01.42 | p3nguin | Just set it to nat=yes and go on with life as usual. |
| 10:01.53 | ChrisInSydney | easier that way |
| 10:02.14 | ChrisInSydney | nzkiwi1: Maybe check it against a softphone |
| 10:02.38 | ChrisInSydney | see what it does. Yeakink are ok but.... |
| 10:03.14 | nzkiwi1 | everything I find on Google suggests that is an ongoing problem with asterisk |
| 10:03.25 | nzkiwi1 | there is reports of this issue from 2009 |
| 10:03.49 | ChrisInSydney | nzkiwi1: which ver u runnin' |
| 10:03.54 | nzkiwi1 | I know the PBX works with TLS |
| 10:03.58 | nzkiwi1 | 10.1.12 |
| 10:04.01 | nzkiwi1 | 10.1.2 |
| 10:04.11 | nzkiwi1 | cos our upline uses TLS |
| 10:04.26 | nzkiwi1 | it's the phone I need to get working |
| 10:04.36 | ChrisInSydney | Now you are stretchng me, like the wallabies, I'm throwing in the towel ;-) |
| 10:04.59 | ChannelZ | I don't know what that means but it sounds dirty |
| 10:05.09 | ChrisInSydney | can you get the phone to talk directly to the ITSP ? |
| 10:05.23 | nzkiwi1 | the ITSP uses asterisk as well |
| 10:05.50 | ChrisInSydney | good test then ;-) |
| 10:06.11 | nzkiwi1 | no it isnt |
| 10:06.20 | *** join/#asterisk TimeRider (~steve@02de05ed.bb.sky.com) |
| 10:06.34 | nzkiwi1 | then I will have no info on what is happening at the other end |
| 10:06.58 | ChrisInSydney | but you can test if the handset receives a call = the handset works |
| 10:08.13 | nzkiwi1 | I have spent hours and hours on red herrings. |
| 10:08.42 | nzkiwi1 | I want it working with my setup which is no different to the ITSP, and is probably newer software |
| 10:11.38 | nzkiwi1 | I have followed all the OFFICIAL documentation, used a phone that is certified with asterisk and the latest software. Everyone says go to the forums. I have posted up 3 threads and no answer |
| 10:11.44 | ChrisInSydney | divide and conquor: find something tat works and build from there. |
| 10:12.30 | ChrisInSydney | I have posted more lonely threads than I have had answered, but most of the issues I post are SoB issues |
| 10:12.57 | ChrisInSydney | BTW, does anyone know how to stop: setup_srtp: No SRTP module loaded, can't setup SRTP session messages |
| 10:13.01 | nzkiwi1 | I have done everything by the book |
| 10:13.45 | ChrisInSydney | Wireshark, sip debug, have a look at the chatter |
| 10:13.58 | ChrisInSydney | see if you can make it work on somethng else |
| 10:14.09 | ChrisInSydney | see if you can get something else to work. |
| 10:14.12 | nzkiwi1 | the yealink phones are what I have |
| 10:14.26 | nzkiwi1 | I do not want to use a softphone |
| 10:14.28 | ChrisInSydney | Once you get something working, build from there, one step ata time |
| 10:14.32 | ChrisInSydney | at a |
| 10:14.52 | nzkiwi1 | this has worked since 2008 when we switched to voiup until now |
| 10:15.02 | ChrisInSydney | its not a permanent solution, its a debugging tool |
| 10:17.40 | ChrisInSydney | nzkiwi1: You are not usig real time I assume |
| 10:17.53 | nzkiwi1 | no |
| 10:18.25 | nzkiwi1 | asterisk RT or RT kernel? |
| 10:18.59 | ChrisInSydney | ast RT |
| 10:19.03 | nzkiwi1 | no |
| 10:20.53 | ChrisInSydney | If I was in your position, I would.... |
| 10:21.01 | ChrisInSydney | 1: Test the phone direct to the ITSP |
| 10:21.25 | ChrisInSydney | 2: test another phone / softphone / smartphone SIP client against asterisk |
| 10:21.29 | ChrisInSydney | try and get something to work |
| 10:21.45 | nzkiwi1 | Give your customers a rock solid guarantee that your product works with Asterisk. Digiums product and service certification program offers advanced alignment with Asterisk, the open source communications market leader. |
| 10:21.53 | nzkiwi1 | from asterisk exchange |
| 10:21.57 | nzkiwi1 | (a digium website |
| 10:22.03 | ChrisInSydney | Then work out what has changed and go from there |
| 10:22.09 | nzkiwi1 | the t28p is listed as certified |
| 10:22.19 | nzkiwi1 | I say bollocks |
| 10:22.32 | nzkiwi1 | it clearly does not work |
| 10:23.14 | ChrisInSydney | I may be inclined to agree. Then again, I have Cisco SPA525G2 handsets, on Cisco catalyst switches playing up and locking up |
| 10:23.25 | *** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net) |
| 10:23.46 | WIMPy | The days of things working are clearly over. |
| 10:24.02 | nzkiwi1 | I am going to write an email to digium |
| 10:25.04 | ChrisInSydney | At least with the stuff I try to work with, you can always edit the source, even if it is to dump some debug stuff to stderr or a file |
| 10:25.11 | nzkiwi1 | I am not testing this with the itsp as their TLS support is in beta and it does nothing to help me make it work in my system |
| 10:25.26 | nzkiwi1 | that's nice if you know how to code in C |
| 10:25.28 | ChrisInSydney | it will tell you that the phone works with TLS, |
| 10:26.10 | ChrisInSydney | you just need to know how to printf / fprintf. If you know php, you can navigate c enough |
| 10:26.22 | nzkiwi1 | it is of no use because I cant see their CLI |
| 10:26.53 | ChrisInSydney | you dont have to. you register the phone, them try to call it |
| 10:26.56 | ChrisInSydney | tehn |
| 10:26.59 | ChrisInSydney | then |
| 10:27.02 | nzkiwi1 | to state my case I need hard evidence - files, logs etc |
| 10:27.20 | ChrisInSydney | if it rings, then its reistered, if it doesn;t then it probably never will |
| 10:27.21 | ChrisInSydney | see |
| 10:27.32 | nzkiwi1 | I said before it does register |
| 10:27.42 | nzkiwi1 | it shows as uneachable |
| 10:27.48 | nzkiwi1 | but does register |
| 10:27.57 | ChrisInSydney | Then, try a soft phone against asteris, if it works, then there is ain issue with the t28 |
| 10:28.25 | nzkiwi1 | I have already tested the asterisk with TLS |
| 10:28.29 | nzkiwi1 | and it works |
| 10:28.39 | ChrisInSydney | between what and what ? |
| 10:28.44 | nzkiwi1 | I use TLS to talk to our ITSP |
| 10:28.53 | nzkiwi1 | TLS and SRTP |
| 10:28.58 | ChrisInSydney | ok so you register to the ITSP, |
| 10:29.10 | ChrisInSydney | your handset registers to Asterisk |
| 10:29.13 | nzkiwi1 | trunked |
| 10:29.18 | ChrisInSydney | thats backwasrds |
| 10:29.22 | ChrisInSydney | ahhh |
| 10:29.33 | ChrisInSydney | what about a fixed IP on the handset |
| 10:29.35 | nzkiwi1 | ITSP with TLS shows as "REACHABLE" |
| 10:29.46 | nzkiwi1 | it is a fixed IP issued with DHCP |
| 10:29.53 | nzkiwi1 | on the handset |
| 10:30.06 | ChrisInSydney | set asterisk to send to the IP of the phone |
| 10:30.15 | ChrisInSydney | as opposed to a register |
| 10:30.19 | ChrisInSydney | does that work ? |
| 10:30.26 | ChrisInSydney | break the problem down |
| 10:30.26 | nzkiwi1 | using host = 192.168.0.1 ? |
| 10:30.49 | nzkiwi1 | it knows the IP on the because it shows when I do "show sip peer 100" |
| 10:31.00 | ChrisInSydney | Do you have host=dynamic ? |
| 10:31.04 | nzkiwi1 | yes |
| 10:31.20 | ChrisInSydney | change it to a static, whatever the syntax for that is |
| 10:31.32 | ChrisInSydney | see what happens |
| 10:32.57 | nzkiwi1 | it says 'peer is not supposed to register' |
| 10:33.45 | nzkiwi1 | it is set as type=peer |
| 10:33.54 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 10:34.00 | nzkiwi1 | opps |
| 10:34.04 | nzkiwi1 | type=friend |
| 10:34.12 | ChrisInSydney | then turn off registration and turn on accept direct IP calls on the handset. |
| 10:34.24 | ChrisInSydney | ahhhhhhh |
| 10:34.32 | ChrisInSydney | he he he he he |
| 10:34.55 | nzkiwi1 | by removing the "register name"? |
| 10:34.59 | ChrisInSydney | now try |
| 10:35.34 | ChrisInSydney | did you have type=peer ? |
| 10:41.05 | nzkiwi1 | should I change to type=peer? |
| 10:42.03 | nzkiwi1 | either way does not work |
| 10:42.46 | nzkiwi1 | also getting a lot of |
| 10:42.49 | nzkiwi1 | tcptls.c:235 handle_tcptls_connection: FILE * open failed |
| 10:43.00 | nzkiwi1 | which seems to happen when the phone has a server cert installed |
| 10:44.41 | ChrisInSydney | should be a friend |
| 10:45.04 | nzkiwi1 | thats what I had to start with |
| 10:45.17 | ChrisInSydney | have you look at the SIP debug / pcap stuff on the phone ? |
| 10:45.49 | nzkiwi1 | on the phone? |
| 10:46.07 | nzkiwi1 | type=friend must be set as host=dynamic apparently |
| 10:48.08 | ChrisInSydney | OK then split it to peer and user entries and test it with host=ipaddr |
| 10:48.27 | nzkiwi1 | type=peer does not work |
| 10:48.36 | nzkiwi1 | that has been tested |
| 10:48.40 | ChrisInSydney | peer is for outgoing only |
| 10:48.48 | ChrisInSydney | user is incomming |
| 10:49.09 | ChrisInSydney | so have yoiu tried host=dynamic / type=friend ? |
| 10:50.50 | nzkiwi1 | host = dynamic and type = friend is what it was |
| 10:50.58 | nzkiwi1 | type = user Purely numeric hostname (100), and not a peer--rejecting! |
| 10:51.09 | nzkiwi1 | on a call to the phone |
| 10:51.57 | nzkiwi1 | dynamic friend works fine for UDP and it's what I use |
| 10:52.36 | ChrisInSydney | ok |
| 10:53.04 | shihan | lol, its my lucky day, app_voicemail.so wont load... such is life |
| 10:53.26 | nzkiwi1 | as soon as I select TCP or TLS calls to the phone fail cos it is "unreachable" even tho it is registered |
| 10:53.37 | nzkiwi1 | calls from the phone work |
| 10:59.59 | ChrisInSydney | have you look at the SIP debug / pcap stuff on the phone ? |
| 11:01.04 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 11:02.31 | nzkiwi1 | be back soon |
| 11:03.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:14.28 | *** join/#asterisk jkroon (~jkroon@dsl-244-31-102.telkomadsl.co.za) |
| 11:20.00 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 11:24.35 | nzkiwi1 | you there Chris? |
| 11:28.09 | ChrisInSydney | yup. Fighting with isphone. Cant get caller ID working outbound |
| 11:30.30 | nzkiwi1 | TLS does not work with ITSP. |
| 11:30.37 | nzkiwi1 | it does with asterisk |
| 11:30.44 | nzkiwi1 | not with the yealink |
| 11:30.48 | nzkiwi1 | outgoing calls work |
| 11:30.51 | nzkiwi1 | incoming not |
| 11:30.59 | nzkiwi1 | just like on my ast system |
| 11:32.16 | ChrisInSydney | ahh, now try the softphone on asterisk and ITSP. If that works then t28 is not up to the job....like our rugby side |
| 11:37.19 | nzkiwi1 | which s/w you recommend? |
| 11:37.53 | ChrisInSydney | no idea. Not sure whats out there. Can anyone else make a suggestion ?? |
| 11:51.19 | nzkiwi1 | tes |
| 11:56.07 | nzkiwi1 | is that thru a ITSP the CID doesnt work? |
| 12:01.47 | ChrisInSydney | fixed it, sort |
| 12:01.48 | ChrisInSydney | a |
| 12:02.42 | ChrisInSydney | sendrpid=no |
| 12:11.07 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 12:12.46 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 12:16.13 | nzkiwi1 | I'm just running a batch of windows updates on my laptop (havent used it awhile) before using it with a softph to test |
| 12:27.10 | nzkiwi1 | bye chris |
| 12:27.27 | nzkiwi1 | I will come back at some stage |
| 12:27.36 | nzkiwi1 | thanks |
| 12:31.58 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 13:41.43 | Srini | If on a 2 span PRI card, both lines connected, how will asterisk choose to send the calls out? Where we will define this? |
| 13:44.08 | WIMPy | You choose. |
| 13:44.30 | WIMPy | Sensibly via One to three groups. |
| 13:44.47 | WIMPy | deflates the O a little |
| 13:49.27 | *** join/#asterisk Srini (Srinivasa@116.202.103.172) |
| 13:50.52 | Srini | On a PRI card with 2 spans, both connected to 2 different service providers, How do I choose to send a call out on a particular span? |
| 13:58.24 | *** join/#asterisk atekinalp (~alper@85.105.28.77) |
| 14:06.19 | WIMPy | You choose. |
| 14:06.23 | WIMPy | Sensibly via one to three groups. |
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| 14:50.17 | Srini | WIMPy, thanks! Yes I got it going that way |
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| 15:26.42 | saxa | hi iz Zapateller() still used in * 1.8.x ? |
| 15:27.00 | saxa | or is it renamed to maybe Dahditeller() ? :) |
| 15:31.55 | kaldemar | still Zapateller. |
| 15:35.35 | WIMPy | If someone needs a quad BRI or two, there are two on ebay in a few moments. |
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| 16:29.30 | saxa | thx kaldemar |
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| 16:54.19 | nkuttler | hi. i've never used asterisk and wanted to know if it's possible to forward a call and to play some audio file to the recipient before he gets the caller. how complex would this be to set up? experienced linux admin. |
| 16:54.46 | nkuttler | or would i use a different tool for this? |
| 16:55.14 | WIMPy | That's easy. |
| 16:55.21 | WIMPy | Once you've got it running. |
| 16:55.26 | nkuttler | ;) nice. thanks |
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| 17:13.14 | saxa | simple question with rtp ports. If I limit the range of rtp ports on my * server from 10000 to 10200 UDP, do I need to configure something on the phones outside the nat also to make them work , or do asterisk inform the phones that the range is the one in rtp.conf ? |
| 17:15.00 | WIMPy | The later |
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| 17:24.22 | saxa | WIMPy: so nothing is needed to set in the phones, correct ? |
| 17:25.35 | saxa | WIMPy: last time you have been able to connect from your phone on my asterisk and hear the voicemail and echo test and so on, I'm trying to do that from my italian office too, but no success, it calls I see the activity on the console and no audio |
| 17:26.00 | saxa | and I tried that with 2 different phones |
| 17:27.27 | saxa | I can hear the moh if I call thru my italian asterisk box connected to the one in brasil |
| 17:28.01 | saxa | but if I call directly via my hardphone connected to a SIP account in Brasil I don't hear anything |
| 17:28.07 | WIMPy | Each device tells the other one where it expects RTP. |
| 17:28.27 | saxa | ok |
| 17:28.56 | saxa | so why your phone was able to work ? |
| 17:29.09 | WIMPy | I assume your phone is behind NAT? And the Asterisk you used? |
| 17:29.15 | saxa | yes |
| 17:29.28 | saxa | both are behind the nat |
| 17:29.39 | saxa | also the asterisk in IT is behind a nat |
| 17:29.40 | WIMPy | The same net? |
| 17:29.52 | saxa | * to * I connect via IAX2 |
| 17:30.06 | saxa | so there is no problems at all |
| 17:30.11 | saxa | yes |
| 17:30.16 | WIMPy | Double NAT. Good idea to use IAX. |
| 17:30.25 | saxa | yes |
| 17:31.12 | saxa | but I want to understand why my other phone I have in the Brasilian home doesnt work anymore |
| 17:31.35 | saxa | so thats why I'm trying to connect directly to the * box |
| 17:31.39 | saxa | via SIP |
| 17:31.45 | [TK]D-Fender | No need for IAX2 there... |
| 17:32.11 | saxa | In fact I set up 2 accounts on my phone |
| 17:32.22 | saxa | one is to my local * box |
| 17:32.32 | saxa | another one connects to * in BR |
| 17:33.47 | saxa | <PROTECTED> |
| 17:34.01 | *** join/#asterisk luckman212_ (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 17:34.02 | WIMPy | I don't see why there should be a difference between using Asterisk and a phone. |
| 17:34.10 | saxa | I see this on the console on the * server in Brasil |
| 17:34.32 | saxa | WIMPy: * to * goes via IAX2 |
| 17:34.33 | *** join/#asterisk pdtpatrick_ (~pdtpatric@12.249.4.226) |
| 17:34.47 | saxa | phone to * goes via SIP |
| 17:35.15 | saxa | but this is just for my testing purpose |
| 17:35.31 | WIMPy | Ok, I thought you tried * to * via SIP as well. |
| 17:35.33 | saxa | I want to understand why phone to * via SIP doesnt work |
| 17:35.42 | saxa | no * to * is ok |
| 17:35.47 | saxa | forget about it |
| 17:35.54 | saxa | it works and no problem at all |
| 17:36.32 | saxa | since I limited the ports to 10000-10200 I tought it need to be set up also in the phone |
| 17:36.43 | saxa | but you said, its no needed |
| 17:37.03 | *** join/#asterisk caveat-_ (hoax@gateway/shell/bshellz.net/x-ssxptmhcfltgofzx) |
| 17:37.04 | saxa | so now I'm asking myself, why i have no audio. |
| 17:37.39 | saxa | I will try to put my phone directly to the network |
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| 17:51.11 | autofsckk | hello, is there a way to uninstall freepbx and just keep asterisk? |
| 17:53.16 | [TK]D-Fender | man rm |
| 17:53.42 | autofsckk | but what about the symbolic links on /etc/asterisk? just delete those tho? |
| 17:53.46 | autofsckk | *to? |
| 17:53.52 | WIMPy | was about to write the same :-) |
| 17:53.55 | autofsckk | ha too |
| 17:56.29 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 17:56.35 | autofsckk | thanks |
| 17:56.39 | tompaw | Hey. |
| 17:56.49 | tompaw | Anyone from China here? :> |
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| 19:01.37 | saxa | ok setting my phone in IT to a static public IP , registers to my box in BR and it also works |
| 19:01.59 | saxa | so definetly there is something wrong with the NAT on the phone side I have |
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| 19:38.14 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 19:38.36 | WIMPy | Or your Asterisk configuration? |
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| 20:05.37 | afink | anybody know what would cause all phones to lose registration all at once? |
| 20:13.21 | afink | looks like a sip DOS attack |
| 20:13.46 | ChannelZ | yay! |
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| 20:26.07 | oneadvent_ | can someone in the united states send me a fax please? |
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| 20:54.38 | Kobaz | anyone have an example of corporate directory ldap on polycom |
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| 21:13.29 | digitalirony | sup |
| 21:13.49 | digitalirony | so, I am having an odd issue |
| 21:14.23 | digitalirony | Audio issues over SIP, no nat. I tried asterisk 1.6 and 1.10 |
| 21:15.39 | digitalirony | its almost like one way audio, but is not. I did a record test.gsm, then playback. it does record me, jusy either it only gets a small amount of audio, or it plays it back very quickly so I can only hear a 5sec recording in 1sec |
| 21:16.27 | digitalirony | when I call between peers, I can hear audio for just a split second, but then it just dies |
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| 21:27.43 | digitalirony | So, to add to this. I notice on my sip client when I first make the call, it show it as using pcma, then as soon as I send audio, it changes to 'no data'. Its as if asterisk isn't accepting audio after a few seconds |
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| 21:41.35 | AviMarcus | Anyone got a lead on Switzerland DIDs that would be good for a calling card? |
| 21:49.40 | afink | I did a reload and got a sip response bad event from IP that I do not know and isn't present in sip.conf. I should be worried about this correct? |
| 21:51.03 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 21:53.03 | [TK]D-Fender | Depends... perhaps you should be really looking at what that is in response to |
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| 22:16.37 | tmrhmdv | For over a week I had problems installing asterisk on Amazon EC2 and almost gave up learning it. Today, I gave it another shot and installed it to a VirtualBox server. Now everything's working. I'm reading the "Asterisk: the Definitive Guide". I'm so excited and thrilled and wanted to thank the creators/community of asterisk and that book's authors. You guys are awesome! |
| 22:17.29 | ChrisInSydney | tmrhmdv: And partially OCD |
| 22:17.36 | ChrisInSydney | :p |
| 22:17.49 | tmrhmdv | IYes :) |
| 22:19.18 | ChrisInSydney | Question for you people: I am trying to change the outgoing caller ID on a new provider I am testing. I am assuming that if Set(CALLERID(num)=0987654321) and SIPAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM}>) dont work, it cant be done |
| 22:19.48 | ChrisInSydney | sendrpid=rpid and sendrpid=yes have also been tried |
| 22:20.47 | ChrisInSydney | The web interface on their management console works, I can manually put a different number in there (They block mobile / cell numbers), |
| 22:21.31 | ChrisInSydney | did a sip trace, all looks OK. |
| 22:21.46 | ChrisInSydney | BTW SIP provider obviously |
| 22:22.09 | [TK]D-Fender | ChrisInSydney: ${CALLERIDNUM} <- this is an Asterisk 1.0 variable. Please fast forward 6 years to today.... |
| 22:22.37 | ChrisInSydney | Sorry, been listing to my woodstock bootlegs |
| 22:23.10 | ChrisInSydney | About the same era of voip-info |
| 22:23.58 | ChrisInSydney | where I cut and pasted from :-/ |
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| 22:24.24 | *** join/#asterisk NirS (~Nir@62.128.39.228.cable.012.net.il) |
| 22:24.43 | [TK]D-Fender | I recommend referencing that function you seemed to want to set |
| 22:26.23 | ChrisInSydney | fixed, and no it doesnt work |
| 22:26.46 | ChrisInSydney | header looks ok |
| 22:27.29 | ChrisInSydney | Even set it in sip.conf, so looks like they have locked it down. Pity :-/ |
| 22:27.58 | *** join/#asterisk k6b (~k6b@2001:41d0:1:70e3::d00b:1e) |
| 22:28.46 | ChrisInSydney | [TK]D-Fender: Thats the only things I can really change it, CALLERID(num) and an explicit P-Asserted-Identity: ? |
| 22:32.36 | [TK]D-Fender | Those are the usuals |
| 22:33.01 | ChrisInSydney | It is possible to set the following options for handling identity information: |
| 22:33.01 | ChrisInSydney | <PROTECTED> |
| 22:33.07 | [TK]D-Fender | Do you sewe it going out right now? |
| 22:33.12 | ChrisInSydney | Help screen from the UI |
| 22:33.21 | ChrisInSydney | Yup |
| 22:33.43 | [TK]D-Fender | Well then, that's that |
| 22:34.05 | ChrisInSydney | Remote-Party-ID: "0299999999" <sip:0212345678@sip2.newprovider.net>;party=calling;privacy=off;screen=no |
| 22:35.37 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 22:35.41 | dym | hi everyone |
| 22:35.44 | dym | (: |
| 22:36.28 | ChrisInSydney | hey dym |
| 22:36.38 | dym | Hah! Chris - long time no "read" (: |
| 22:36.41 | dym | How are things? |
| 22:36.44 | ChrisInSydney | good |
| 22:36.48 | dym | Great to hear |
| 22:37.03 | ChrisInSydney | hang on, 2 year old is demanding my attention |
| 22:37.08 | dym | np |
| 22:45.36 | ChrisInSydney | must go. Time for domestics |
| 22:45.42 | ChrisInSydney | Sunday morning here |
| 22:46.18 | dym | okay |
| 22:46.21 | dym | take care buddy |
| 22:46.47 | oneadvent_ | what is domestics? |
| 22:47.52 | dym | well - things you do at home |
| 22:47.55 | dym | homestuff |
| 22:47.59 | dym | cleaning up, etc (: |
| 22:48.32 | oneadvent_ | oh i see ok thanks |
| 22:48.59 | dym | np |
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