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| 00:43.18 | [T]ank | ok... so for nat... i have enabled port 5060-5061 udp and 10000-200000 udp and forwarded it all to the ATA's address. On the sip.conf I have nat=yes |
| 00:43.22 | [T]ank | what else do i need to consider? |
| 00:43.34 | [T]ank | never had this problem before. |
| 00:43.42 | [T]ank | in fact, I had this thing working the other day. |
| 00:44.03 | [T]ank | it is on only on the side of the ATA that I hear the sound cutting in and out. The sound is 100% on my cell phone |
| 00:44.46 | Chainsaw | [T]ank: Spastic echo canceller, perhaps? |
| 00:45.06 | *** join/#asterisk ctp (n=ctp@brsg-d9beeb78.pool.mediaWays.net) |
| 00:45.08 | [T]ank | well... tried it on several ATAs. just tried a linksys also. |
| 00:45.19 | Chainsaw | [T]ank: (Or something like automatic gain control on both sides of a call, that would also get messy) |
| 00:45.23 | [T]ank | and it is perfectly 1 second appart... sounds more like port stuff to me |
| 00:45.27 | [T]ank | im sure its nat related |
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| 01:08.35 | Olobola | do you think switching from gsm to speex will help with bandwidth issues? I'm on a slow connection.. could be a jitter issue though. |
| 01:12.10 | carrar | Whats slow |
| 01:12.36 | carrar | usr 729, 723.1 or lpc |
| 01:12.50 | carrar | speex 5.95 |
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| 01:59.34 | acxty | Hi guys, I am using a softphone. I want to make a outgoing call. I am using DIAL(SIP/5032312/${EXTEND},20) |
| 01:59.47 | acxty | but it says that extension is not found |
| 02:00.15 | acxty | is that the correct way? |
| 02:02.40 | acxty | someone out there? |
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| 02:11.32 | tAnk | It's 4.11 am here in central Europe |
| 02:11.36 | tAnk | Dunno :P |
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| 03:26.18 | Micc | it seems like I can make my own call parking with RedirectChannel to an extension that just plays music on hold for a while then calls back or redirects back if they call the right extension. |
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| 03:31.36 | Borai | hello |
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| 04:06.07 | Borai | 1. Added support for 64-bit platform. b19 lumenvox but i cant get the new one to work |
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| 04:29.47 | [T]ank | ok... so I am still working on troubleshooting this issue where a phone behind nat, connected to an ATA device connected via sip to a server across the internet is getting the sound cut out every 1 second. If I call my cell phone from my asterisk phone i only hear the issue on the asterisk phone side. not the cell side. Does not matter inbound or outbound. |
| 04:30.02 | [T]ank | here are the outputs from the sip set debug ip <ipaddress> |
| 04:30.02 | [T]ank | http://pastebin.ca/1484424 |
| 04:30.24 | [T]ank | i am on asterisk 1.6.11 |
| 04:30.27 | [T]ank | any ideas? |
| 04:31.31 | [T]ank | that also includes the sip.conf entries. |
| 04:33.16 | [T]ank | im thinking its a nat issue, but I cannot figure it out. |
| 04:33.38 | [T]ank | i have ports 5060-5061 udp forwarded to my ata device as well as on the other end forwarded to my server |
| 04:33.56 | [T]ank | i also have ports 10000-20000 udp on both sides forwarded to the appropriate devices. |
| 04:34.06 | [T]ank | can anyone see where I might be running into an issue here? |
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| 05:24.49 | [T]ank | anyone here a pro with NAT? |
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| 05:54.27 | beit | newbie q. using x-lite on windows, i'm trying to register, but i don't see asterisk sending OK after seeing REGISTER on ngrep..why? |
| 05:55.25 | beit | i have the simplest sip.conf. taken from the book on asterisk |
| 05:55.29 | beit | but it doesn't seem to work |
| 05:55.32 | beit | i can't figure it out. |
| 05:55.40 | beit | any hints? |
| 06:01.10 | Borai | f. |
| 06:05.17 | Borai | res_snmp |
| 06:05.19 | Borai | doesnt want to compile |
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| 06:09.41 | Micc | Tank, I've had that problem because the timing device wasn't compiled properly. |
| 06:10.01 | Micc | Tank, when I upgraded to 1.6 it had that problem because I didn't compile dahdi first. |
| 06:17.36 | Borai | fuck |
| 06:19.06 | *** join/#asterisk aplund (n=aplund@220-244-113-164.static.tpgi.com.au) |
| 06:19.31 | aplund | Does doing Set(_CDR(userfield)=blah) work? |
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| 06:24.04 | beit | anybody have any ideas? |
| 06:24.17 | beit | why doesn't asterisk send OK after receiving REGISTER? |
| 06:24.36 | carrar | remove the _ |
| 06:25.03 | aplund | err |
| 06:25.26 | aplund | the _ is there explicitly so that it is inherited |
| 06:25.33 | aplund | I guess nobody here knows and I'll just have to try |
| 06:25.40 | beit | aplund: can you help me? |
| 06:25.44 | carrar | other way around |
| 06:25.54 | aplund | ? |
| 06:25.55 | carrar | _ lets a var pass to the next context |
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| 06:26.02 | aplund | that's what I want |
| 06:26.16 | carrar | But you are setting a CDR variable |
| 06:26.20 | aplund | I know |
| 06:26.38 | carrar | Once you set it, it stays |
| 06:26.51 | aplund | not if you are using Dial(Local/) |
| 06:27.34 | carrar | so then use _ |
| 06:27.35 | carrar | or __ |
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| 06:28.19 | carrar | it will still be in the CDR |
| 06:28.25 | carrar | just on a different line |
| 06:28.36 | beit | i see REGISTER coming in, but no OK for response? |
| 06:29.06 | carrar | actually I don;t have that issue |
| 06:29.51 | aplund | nope |
| 06:29.56 | aplund | prepending __ doesn't work |
| 06:30.07 | aplund | Function __CDR not registered |
| 06:30.58 | carrar | so use none reserved variables |
| 06:31.11 | aplund | ugh |
| 06:32.09 | aplund | then patch it up on the other side of the dial? |
| 06:32.22 | aplund | seems pretty ugly |
| 06:32.41 | carrar | why |
| 06:33.36 | aplund | cause I'll have to do it for every extension that is dialled? |
| 06:33.48 | carrar | why |
| 06:34.03 | carrar | look at dstchannel |
| 06:34.46 | carrar | that wouldbe where the call ended up in a multiple Local/ dial statment |
| 06:34.56 | carrar | with the userfield and times |
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| 06:53.07 | beit | is soo confused |
| 07:05.03 | [TK]D-Fender | [02:24]<beit>why doesn't asterisk send OK after receiving REGISTER? |
| 07:05.15 | [TK]D-Fender | beit: PASTEBIN is your friend... |
| 07:05.16 | [TK]D-Fender | ~pb |
| 07:05.17 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 07:13.46 | ramindia | ~pb |
| 07:13.46 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 07:23.02 | beit | [TK]D-Fender: what do you want me to paste? |
| 07:23.21 | [TK]D-Fender | beit: SIP debug from your syetem of course. |
| 07:23.41 | beit | ok here's one issue. |
| 07:23.48 | beit | when i do asterisk -rvvv |
| 07:24.13 | beit | and then try to register an x-lite softphone, i don't see any info. i only see it when i do ngrep |
| 07:24.22 | beit | do you consider ngrep the sip debug? |
| 07:24.34 | beit | i see the REGISTER msgs, but i don't see OK for them |
| 07:24.40 | beit | so the x-lite times out |
| 07:24.58 | beit | this is all on same subnet, so i don't think NAT is the issue |
| 07:25.52 | beit | i can paste you my sip.conf if you want |
| 07:27.27 | ramindia | beit: paste that ngrep report when you intiate that Registering from X-lite |
| 07:27.59 | [TK]D-Fender | beit: Go to * CLI and "sip set debug on" |
| 07:28.26 | beit | ok one sec |
| 07:28.53 | [TK]D-Fender | beit: So far I'm inclined to believe the packets aren't even making it to * |
| 07:29.28 | beit | well maybe.. its windows machine.. i turned off firewall. i can ping the windows client/xlite machine....but in the ngrep i don't see an OK sent from PBX |
| 07:29.36 | beit | one sec, i'll paste the ngrep output |
| 07:32.52 | [TK]D-Fender | beit: I just said that you need to look at SIP DEBUG from * CLI <- |
| 07:33.01 | beit | ok let me try that |
| 07:34.17 | beit | ok i set debug on, i tried to register, but how do i view the debug now? |
| 07:35.01 | [TK]D-Fender | beit: * tells you that it has enabled SIP debug? |
| 07:35.44 | b14ck | happy 4th to all u americans ^^ |
| 07:35.46 | beit | yes |
| 07:35.51 | aplund | carrar: Thank you! I think I'm on top of it now. |
| 07:36.08 | beit | localhost*CLI> sip set debug on |
| 07:36.10 | beit | SIP Debugging enabled |
| 07:36.12 | [TK]D-Fender | beit: Then if you don't see anything, then nothing is reaching * |
| 07:36.22 | beit | but ngrep is showing it coming in |
| 07:37.05 | [TK]D-Fender | beit: Go prove * has bound the port <- |
| 07:37.20 | beit | http://pastebin.com/m1f78829d |
| 07:37.53 | beit | that's my ngrep output |
| 07:38.33 | beit | shoudn't i see an OK from asterisk going to .102? |
| 07:38.33 | [TK]D-Fender | beit: Go prove * has bound the port <- |
| 07:38.43 | beit | [TK]D-Fender: how do you mena? |
| 07:38.49 | beit | the port 5060 you mean? |
| 07:38.53 | [TK]D-Fender | DUH |
| 07:39.01 | beit | how do i prove that? |
| 07:39.13 | [TK]D-Fender | netstat <- |
| 07:39.18 | beit | ok one sec |
| 07:39.50 | [TK]D-Fender | beit: And I'm not sure what part of the IP stack ngrep works on, if that's pre or post-filter |
| 07:40.18 | beit | netstat|grep 5060 - doesn't output anything, am i doing it right? |
| 07:40.33 | beit | appreciates this help |
| 07:40.40 | [TK]D-Fender | beit: "netstat -an" |
| 07:40.50 | [TK]D-Fender | beit: Look ebfore you grep |
| 07:42.15 | beit | ok my netstat -na shows 583 lines. nothing seems relevant. if i do netstat -na |grep 5060 i see this: |
| 07:42.26 | beit | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 07:42.33 | beit | so its running on 0.0.0.0 |
| 07:43.51 | [TK]D-Fender | beit: Something is anywhay, no go prove your firewall isn't in the way |
| 07:44.31 | beit | ok one thing though. when i do ngrep, and i see the REGISTER coming in, in there, should i see OK going out? |
| 07:44.40 | [TK]D-Fender | FUCK NGREP |
| 07:44.42 | beit | or would that only show coming in, not going out? |
| 07:44.57 | [TK]D-Fender | * isn't getting the damn packet. |
| 07:44.57 | beit | ok so how do i prove firewall is not in the way |
| 07:45.01 | [TK]D-Fender | ..... |
| 07:45.08 | [TK]D-Fender | "iptables --list" |
| 07:45.14 | beit | ok let me try that |
| 07:46.16 | beit | firefox crashing, sorry man, give me 10 seconds. |
| 07:46.39 | beit | http://pastebin.com/m71143d68 |
| 07:46.41 | beit | that's my iptables |
| 07:47.02 | beit | doesn't even care if that's not secure to do, i just want my * to work |
| 07:47.34 | [TK]D-Fender | beit: "iptables --flush" |
| 07:47.36 | ramindia | beit: try flush Iptables and see |
| 07:47.58 | beit | i will. right now, but can you elaborate why i need to? |
| 07:48.01 | beit | so i understand |
| 07:49.06 | beit | ok now i flushed it, now when i do --list it shows nothign |
| 07:49.14 | beit | should i try register again? |
| 07:49.23 | [TK]D-Fender | yes |
| 07:49.33 | beit | ok one sec |
| 07:50.25 | beit | lol |
| 07:50.26 | beit | it worked |
| 07:50.39 | beit | omg, so PLEASE!!!! why the hell wasn't it working |
| 07:50.48 | [TK]D-Fender | FIREWALL |
| 07:50.48 | beit | i can see 'sip show peers' now it shows up |
| 07:51.00 | beit | firewall on the linux side? on the * server side? |
| 07:51.39 | [TK]D-Fender | beit: Thats all you modified... do you really have to ask? |
| 07:52.16 | beit | [TK]D-Fender: listen man, i'm just learning, i know linux and solaris, but for some reason, i still don't understand this issue |
| 07:52.56 | [TK]D-Fender | beit: Well work with the empirical evidence. Yes, iptables was in the way |
| 07:53.23 | beit | ya i gotta read up on that a bit more. the asterisk book didn't mention anything about this. |
| 07:54.14 | [TK]D-Fender | beit: I'd lay bets the book tells you what ports the different protocols rquire.. |
| 07:54.45 | beit | true, i guess it's not its job to figure out my FC10 config for me. |
| 07:55.09 | beit | but still, iptables --flush, did that clear the firewall? i dont get it. |
| 07:55.11 | [TK]D-Fender | beit: Correct. |
| 07:55.49 | [TK]D-Fender | beit: Before it had entries. You issued the flush command I gave you. After it doesn't. Empirical evidence wins again |
| 07:56.19 | beit | [TK]D-Fender: listen man, i agree, you know your shit, but to someone who doesn't konw this shit, it seems strange |
| 07:57.00 | [TK]D-Fender | beit: No, these things you are asking are things that answer themselves. You don't actually have to know anything to judge the result |
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| 07:57.32 | [TK]D-Fender | beit: Before = full, Action taken. After = empty. Guess we know that action did it. |
| 07:57.57 | [TK]D-Fender | beit: Keeps your eyes open to cause & effect |
| 07:57.57 | beit | ya, i don't know what iptables --flush does, so i have to readup on |
| 07:58.00 | beit | it* |
| 07:58.11 | [TK]D-Fender | beit: think "toilet" :) |
| 07:59.14 | beit | in terms of...? |
| 07:59.23 | beit | which is the toilet in this case? |
| 07:59.34 | [TK]D-Fender | beit: Fulsh all your filtering rules down the toilet. As in GONE. |
| 07:59.41 | [TK]D-Fender | Flush* |
| 07:59.52 | beit | you mean set by default by FC? |
| 08:00.25 | [TK]D-Fender | beit: No as in "doesn't matter when they got set, just TRASH THEM" |
| 08:00.49 | beit | but doesn't that comporise my other shit on the system? |
| 08:00.52 | [TK]D-Fender | beit: This is a live action, and will no impact your next boot. |
| 08:01.15 | [TK]D-Fender | beit: Depending on what was being filtered, what running processes you have, etc. |
| 08:01.28 | beit | ok |
| 08:01.42 | beit | well thanks ....at least i nkow its not a * problem |
| 08:02.44 | [TK]D-Fender | beit: 1 step down... |
| 08:02.55 | beit | i still can't phone to phone..lol... but at least my phones register |
| 08:05.49 | [TK]D-Fender | beit: Yup, now at least * has the oppotunity to tell your phone to go to hell personally :) |
| 08:06.58 | beit | thanks |
| 08:07.18 | beit | and throughout, thanks for the sarcasm and smartass-ism |
| 08:07.19 | beit | <PROTECTED> |
| 08:07.45 | [TK]D-Fender | beit: No sarcasm anywhere there. |
| 08:08.37 | [TK]D-Fender | beit: and only my "DUH" really rang of anything other than "really, look at whats happening right in front of you" in a lighter light |
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| 08:34.15 | [TK]D-Fender | checkout time. later all |
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| 12:45.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 13:04.40 | *** join/#asterisk ikevin (n=kevin_fr@serveur6.illux.org) |
| 13:04.45 | ikevin | hello |
| 13:05.12 | ikevin | i've a problem with asterisk and sip peer account |
| 13:05.39 | ikevin | asterisk always timeout when registering to a peer |
| 13:06.18 | ikevin | i'm sure that my peer is availlable |
| 13:06.35 | ikevin | i've the problem on 2 of my 3 server |
| 13:07.03 | ikevin | anyone know what can block tha connexion? |
| 13:07.11 | ikevin | (sorry for my bad english) |
| 13:23.14 | errotan | ikevin: are your servers behind a router or firewall ? |
| 13:23.36 | ikevin | one of the twice yes, the second no |
| 13:24.06 | errotan | ikevin: and have you opened and forwarded the ports ? |
| 13:24.23 | ikevin | yes, i use a dmz on the natted server |
| 13:25.34 | errotan | i once tried dmz and failed don't use it |
| 13:25.55 | errotan | open port 5060 and set port triggering |
| 13:26.18 | ikevin | port 5060 is open, external client can connect on |
| 13:26.38 | errotan | but disable dmz and try again |
| 13:26.41 | ikevin | this server make some strange things, like: |
| 13:26.45 | ikevin | [Jul 5 15:25:32] NOTICE[28341]: chan_sip.c:15851 sip_poke_noanswer: Peer 'kevin' is now UNREACHABLE! Last qualify: 0 |
| 13:26.56 | ikevin | [kevin] |
| 13:26.56 | ikevin | <PROTECTED> |
| 13:27.23 | ikevin | errotan, it's not a real dmz, router don't support it, i use port range redirect (1-65535) |
| 13:27.31 | ikevin | in both tcp/udp |
| 13:28.58 | errotan | i never heard of "port range redirect" but if you say so :) anyway maybe you have to set externip= option is sip.conf |
| 13:29.08 | errotan | in sip.conf* |
| 13:29.12 | ikevin | i try |
| 13:29.32 | errotan | what are you peers ? |
| 13:30.05 | ikevin | same with externip |
| 13:30.50 | ikevin | i try to connect to a sip provider for getting a gateway for external call |
| 13:31.19 | ikevin | it's strange because ~ 2 week ago this working perfectly |
| 13:31.38 | errotan | have you asked the service provider ? |
| 13:31.53 | errotan | maybe they have some problem |
| 13:31.59 | ikevin | service provider work, my first asterisk server work fine on |
| 13:32.30 | ikevin | i have 3 different account on this provider, one per server, just one is working |
| 13:32.57 | ikevin | i have make a try to connect natted server to the "working" server |
| 13:33.00 | errotan | and all of them are behind the same public ip address ? |
| 13:33.01 | ikevin | i've the same error |
| 13:33.06 | ikevin | nop |
| 13:33.22 | ikevin | 2 are on a diffrent data center, one is at home |
| 13:33.26 | errotan | asterisk version ? |
| 13:33.36 | ikevin | 1.4.21.2 |
| 13:33.52 | ikevin | i use debian lenny's package |
| 13:34.48 | errotan | try turning on sip debug for the ip of the provider |
| 13:35.08 | errotan | try to sniff the packet with some program like wireshark |
| 13:35.18 | ikevin | how can i filter the ip in debug mode? |
| 13:35.27 | errotan | sip debug ip 10.1.2.3 |
| 13:35.43 | ikevin | thx |
| 13:35.57 | errotan | i be back in 10 minutes if have some problem |
| 13:37.09 | ikevin | ok |
| 13:37.12 | ikevin | thx for your help |
| 13:37.15 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:37.24 | ikevin | i try to check it with debug & tcpdump |
| 13:40.37 | ikevin | asterisk try to send packet to a non peer account :o |
| 13:49.43 | ikevin | packet are not received by the peer :x |
| 13:52.36 | *** join/#asterisk [netman] (n=netman@158.Red-88-24-108.staticIP.rima-tde.net) |
| 13:56.24 | errotan | re |
| 13:56.37 | errotan | non peer account ? :) |
| 13:57.14 | ikevin | i have a sip account with "type=friend" |
| 13:57.22 | ikevin | asterisk try to make a connexion on |
| 13:57.57 | ikevin | [Jul 5 15:57:44] NOTICE[29344]: chan_sip.c:15851 sip_poke_noanswer: Peer 'kevin' is now UNREACHABLE! Last qualify: 0 |
| 13:58.21 | ikevin | i have check with tcpdump, asterisk try to sent packets, so they are never received |
| 13:59.02 | ikevin | if i use ekiga to connect on the provider, it working perfectly (without stun server) |
| 13:59.17 | errotan | but where are they sending the packets and what's in the packet VIA: header ? |
| 13:59.53 | ikevin | REGISTER & OPTION |
| 14:00.57 | ikevin | http://pastebin.com/m2a0246ee |
| 14:01.57 | errotan | this information is not enough... |
| 14:02.01 | errotan | Via: SIP/2.0/UDP 192.168.56.3:33548;branch=z9hG4bK-d8754z-22295d483271131c-1---d8754z-;rport |
| 14:02.09 | errotan | this is a via header :) |
| 14:02.48 | ikevin | Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK46b9e1dc;rport |
| 14:03.16 | errotan | that seems to be good for a lan server |
| 14:03.38 | ikevin | i think to |
| 14:03.52 | ikevin | he has already working fine with this config file :x |
| 14:04.14 | ikevin | i've checked if port 5060 is close by my isp |
| 14:04.42 | *** join/#asterisk [netman] (n=netman@158.Red-88-24-108.staticIP.rima-tde.net) |
| 14:04.51 | ikevin | so a "dig bouh.com @frontier.icedslash.net -p 5060" is received by the remote host |
| 14:05.47 | ikevin | if i set a "port=XXX" on the peer config, does it the port used for outgoing connexion? |
| 14:06.36 | errotan | in sip.conf port= refers to the target peer port |
| 14:06.46 | ikevin | ok |
| 14:07.25 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 14:08.09 | errotan | but if you send the packet to the right ip address with the right via header and you don't get any reply you should recheck your routers configuration |
| 14:08.13 | ikevin | if i set "port=5061", the server receive packets |
| 14:10.04 | errotan | that is strange... comment out the port= option in your sip peer because the sip service provider uses the default 5060 port |
| 14:11.16 | ikevin | yep |
| 14:11.39 | ikevin | i think there are a problem on my connexion on port 5060 |
| 14:12.26 | ikevin | i've tryed to open port 5061 on one of my server |
| 14:12.36 | ikevin | it working on, so port 5060 don't work |
| 14:13.00 | errotan | you set the bindport option to 5061 ? |
| 14:13.13 | ikevin | yep |
| 14:13.19 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:13.41 | errotan | your isp is blocking the packets on 5060 ? |
| 14:14.11 | ikevin | i don't find any infos about that on my home connexion |
| 14:14.22 | ikevin | so, for the other server in dc, no |
| 14:14.35 | errotan | or do you have some other voip stuff on lan that listens on 5060 ? maybe that is the problem |
| 14:14.57 | ikevin | i've voip line by my isp, i search if it use sip |
| 14:15.16 | ikevin | pc no, so it's possible that the router use it |
| 14:15.27 | ikevin | so i don't know how can i check that |
| 14:15.32 | errotan | that is a possiblility |
| 14:16.20 | errotan | use nmap to find out that is listening on port 5060 |
| 14:16.40 | errotan | have to go now again for some time |
| 14:16.41 | errotan | brb |
| 14:17.06 | ikevin | nmap never show my port 5060 is open (on server where he's open too) |
| 14:17.08 | ikevin | cya |
| 14:17.10 | ikevin | cu* |
| 14:17.15 | ikevin | thx a lot for your help |
| 14:23.24 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:30.55 | errotan | np |
| 14:36.53 | ramindia | ikevin: check with netstat |
| 14:37.24 | ikevin | i don't have access to the router |
| 14:37.53 | ikevin | i've found the option to scan udp with nmap |
| 14:37.59 | ikevin | so scan is long |
| 14:44.23 | pa | best sip client free for windows? |
| 14:44.49 | ikevin | idefisk, ekiga, xlite |
| 14:45.11 | ikevin | -idefisk+zoiper (he have changed of name) |
| 14:45.28 | pa | thanks! |
| 14:45.33 | *** join/#asterisk coolthreads (n=coolthre@203-97-238-71.cable.telstraclear.net) |
| 14:47.00 | coolthreads | anyone got any quick solutions to get asterisk to load at start up, currently root starts asterisk, but doesnt have permission to load as it shows |
| 14:47.27 | coolthreads | have to always load it by hand |
| 14:47.31 | ramindia | coolthreads: what distro ? |
| 14:47.35 | coolthreads | ubuntu |
| 14:47.53 | ramindia | coolthreads: you can put the /etc/rc.local |
| 14:48.02 | ramindia | or make as service in /etc/init.d/ |
| 14:49.03 | coolthreads | ok is that those scripts from the src folder just copy the debian over to that folder? |
| 14:49.09 | pa | lol |
| 14:49.11 | pa | cant believe |
| 14:49.24 | pa | my extension to exit on some zap channel didnt work |
| 14:49.25 | pa | now it works |
| 14:49.28 | ramindia | check in contrib |
| 14:49.31 | pa | didnt do anything.. |
| 14:49.33 | ramindia | you see startup files |
| 14:49.34 | [TK]D-Fender | coolthreads: Root may start *, but its starting it as another user and you have failed to set permissions on something. A common mplace is the PID it writes. |
| 14:49.37 | coolthreads | yup |
| 14:50.26 | coolthreads | I usually just start it from the terminal as sudo asterisk or asterisk -vvvvvvvc if Im working on it |
| 14:50.57 | *** join/#asterisk HorizonXP (n=xitij@69-196-163-234.dsl.yaknet.ca) |
| 14:51.02 | coolthreads | and so got to the point where I would like it to start auto |
| 14:51.05 | ramindia | coolthreads: check in the source folder of *.. in the contrib u see some startup files |
| 14:51.49 | [TK]D-Fender | ramindia: It IS loading on startup, only FAILING. |
| 14:52.40 | coolthreads | yup im under init.d |
| 14:53.58 | coolthreads | well contrib/init.d in the asterisk source, I find one thats debian plus others |
| 14:54.32 | coolthreads | /usr/src/asterisk-1.4.25.1/contrib/init.d$ |
| 14:55.10 | coolthreads | is it a matter of just copying the debian to /etc/rc.local |
| 14:55.38 | ramindia | you need to copy that to /etc/init.d/ and make service on |
| 14:56.14 | coolthreads | thought it was that folder, just read earlier message |
| 14:56.21 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:57.22 | coolthreads | yup, I cp it to /etc/init.d |
| 14:58.05 | ramindia | run /etc/init.d/asterisk ( with options) see is everything ok |
| 15:03.33 | coolthreads | doesnt do anything, only if I run asterisk by its self or -vvvvvc as I usually do before trying the auto. |
| 15:07.06 | coolthreads | it tries to start asterisk when booting up but comes up as permission denied |
| 15:08.26 | [TK]D-Fender | ... |
| 15:08.31 | [TK]D-Fender | FIX THE PERMISSIONS <- |
| 15:08.39 | [TK]D-Fender | ~asterisk-non-root |
| 15:08.40 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 15:09.32 | coolthreads | ok |
| 15:10.14 | *** join/#asterisk ingenius (n=alektro@host117.190-138-52.telecom.net.ar) |
| 15:17.54 | *** join/#asterisk JackTheNipple (n=JackTheN@static-87-79-237-194.netcologne.de) |
| 15:18.23 | *** join/#asterisk hi365 (n=hi365@94.159.178.118) |
| 15:34.12 | coolthreads | [TK]D-Fender, thanks. Sorted now |
| 15:34.21 | coolthreads | ramindia, thanks |
| 15:35.04 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 15:35.47 | ramindia | coolthreads: YW |
| 15:41.26 | *** join/#asterisk klochan (n=Klochan@195.222.70.1) |
| 16:07.08 | *** join/#asterisk jicksta (n=jicksta@c-67-169-183-106.hsd1.ca.comcast.net) |
| 16:10.45 | *** join/#asterisk hi365 (n=hi365@95.86.105.206) |
| 16:25.19 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:31.04 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:31.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:58.58 | *** join/#asterisk tobias (n=tobias@6.sub-97-1-61.myvzw.com) |
| 16:59.33 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 17:04.04 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 17:04.14 | *** join/#asterisk apeiron (i=apeiron@isuckatdomains.net) |
| 17:04.18 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 17:23.10 | *** join/#asterisk NickRios05 (n=uniccall@static-200-105-140-234.acelerate.net) |
| 17:31.33 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 17:43.05 | *** join/#asterisk phurl (n=mdupont@82.114.94.18) |
| 17:44.25 | *** join/#asterisk vegbox (n=kevinle@24-205-54-7.dhcp.gldl.ca.charter.com) |
| 17:45.02 | *** part/#asterisk JackTheNipple (n=JackTheN@static-87-79-237-194.netcologne.de) |
| 17:45.34 | *** join/#asterisk JackTheNipple (n=JackTheN@static-87-79-237-194.netcologne.de) |
| 17:46.01 | JackTheNipple | Back again |
| 17:46.12 | *** join/#asterisk QaDeS (n=mklaus@dslb-084-056-246-236.pools.arcor-ip.net) |
| 17:59.29 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com) |
| 18:07.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:10.47 | *** join/#asterisk Loki (i=loki@unaffiliated/loki) |
| 18:11.53 | Loki | Howdy. I am using asterisk on Ubuntu 9.04 and for some odd reason when trying to connect to it in commandline, i am getting the error nable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 18:12.21 | JackTheNipple | does it? |
| 18:12.33 | Loki | I have checked, asterisk, is in deed running, and I have tailed my logs of /var/log/asterisk/full and there are no errors, and there is /var/run/asterisk/asterisk.ctl |
| 18:12.44 | Loki | as is there the .pid file |
| 18:13.12 | JackTheNipple | you did asterisk -r ? |
| 18:13.22 | Loki | I did asterisk -vvvr |
| 18:13.51 | JackTheNipple | firewall? |
| 18:14.05 | JackTheNipple | is there any iptabkes running? |
| 18:14.08 | Loki | interal box |
| 18:14.13 | Loki | same machine |
| 18:14.37 | JackTheNipple | maybe iptables on the localhost? do a iptables --flush |
| 18:14.51 | Loki | same error |
| 18:15.09 | Loki | root@karma:/var/run/asterisk# ls /var/run/asterisk/ |
| 18:15.09 | Loki | asterisk.ctl asterisk.pid |
| 18:15.10 | Loki | root@karma:/var/run/asterisk# |
| 18:15.52 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 18:16.14 | JackTheNipple | ah, aterisk is using socket in this case, not tcp |
| 18:16.34 | JackTheNipple | okay. Access-rights not sockets are given? |
| 18:16.51 | Loki | What do you mean access-rights? |
| 18:17.33 | JackTheNipple | ls -l /var/run/asterisk/asterisk.ctl |
| 18:17.39 | JackTheNipple | what does it say? |
| 18:18.06 | ikevin | check permission on the socket |
| 18:18.21 | Loki | srwxrwxrwx 1 root root 0 2009-07-05 14:08 asterisk.ct |
| 18:18.28 | Loki | srwxrwxrwx 1 root root 0 2009-07-05 14:08 asterisk.ctl |
| 18:18.39 | JackTheNipple | pretty optimistic |
| 18:18.40 | Loki | I then changed it to the asterisk user |
| 18:18.41 | JackTheNipple | ;-) |
| 18:18.42 | *** part/#asterisk phurl (n=mdupont@82.114.94.18) |
| 18:18.45 | Loki | same issue |
| 18:19.13 | JackTheNipple | maybe asterisk* is locking for socket at the wrong place |
| 18:19.45 | JackTheNipple | let find out how to tell* were to find the socket (should be in /etc/asterisk/asterisk.conf) |
| 18:20.59 | JackTheNipple | what does /etc/asterisk/asterisk.conf look like? |
| 18:21.33 | JackTheNipple | especially rundir.... |
| 18:22.41 | Loki | http://pastebin.com/d269379c5 |
| 18:22.53 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 18:24.22 | JackTheNipple | loki: but asterisk is running? |
| 18:24.27 | Loki | yes |
| 18:24.53 | JackTheNipple | netstat -apn | grep asterisk |
| 18:25.06 | JackTheNipple | loki: does this show you the socket? |
| 18:26.12 | Loki | well it is running, it does not show the socket, i can still see admin in console |
| 18:27.05 | JackTheNipple | loki: this is odd. You should see the socket in the list as "listening" |
| 18:28.01 | JackTheNipple | you're running * as its own user? |
| 18:28.32 | JackTheNipple | or running it as root? |
| 18:28.43 | Loki | hrm |
| 18:28.50 | Loki | *? |
| 18:28.59 | JackTheNipple | asterisk = * |
| 18:29.05 | JackTheNipple | ;-) |
| 18:29.20 | JackTheNipple | asterisk == * (would be more correct ) |
| 18:29.57 | Loki | as it |
| 18:30.00 | Loki | 's own user |
| 18:31.09 | JackTheNipple | okay, so I guess in /etc/asterisk/asterisk.conf you use runuser = asterisk & rungroup = asterisk |
| 18:31.38 | JackTheNipple | check if this users have full access to the directory & path were the socket is located |
| 18:32.05 | [TK]D-Fender | Why isn't there an asterisk.pid <------ |
| 18:32.26 | Loki | there is one |
| 18:33.03 | [TK]D-Fender | Loki: show it already... |
| 18:35.13 | JackTheNipple | for testing you could change the user/group to root in asterisk.conf |
| 18:36.07 | JackTheNipple | Loki: if this works, you know to check your access-rights on folder/files for the given user |
| 18:36.52 | JackTheNipple | Loki: asterisk -r will look for the path were to find the socket in asterisk.conf - and the path config you pasted looks good |
| 18:36.59 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 18:38.06 | Loki | JackTheNipple: whats odd is it shows up in my console |
| 18:38.12 | Loki | in console-8 |
| 18:38.18 | JackTheNipple | Loki: maybe there is a cli parameter for asterisk to give pass a socket to use. |
| 18:38.34 | JackTheNipple | Loki: and its really the same box ;-) |
| 18:39.54 | JackTheNipple | asterisk -s YOURSOCKETPATH -r |
| 18:40.28 | JackTheNipple | Loki: sorry (asterisk: invalid option -- s) |
| 18:41.38 | Loki | .ctl is gone :o |
| 18:42.48 | JackTheNipple | console still up? |
| 18:43.49 | Loki | ya i just did stop gracefully |
| 18:44.39 | Loki | it isn't saving the pid for some reason anymore |
| 18:44.50 | JackTheNipple | Loki: wtf? |
| 18:44.54 | JackTheNipple | :-) |
| 18:44.57 | Loki | But it is running |
| 18:45.25 | JackTheNipple | without complaining at startup? do you start with -vvvvvvvvvvvc? |
| 18:45.50 | JackTheNipple | Loki: should complain about not beeing able to write PID and stuff |
| 18:46.59 | JackTheNipple | Loki: what version? 1.4. does not seem to have "-s" option, |
| 18:47.46 | JackTheNipple | Loki: I don't have 1.6 close to me, but if you're on 1.6, check you man-page for socket-parameter |
| 18:48.47 | Loki | ARG no ia m in 1.4 |
| 18:49.52 | ikevin | i don't have socket option on 1.4.21 too |
| 18:50.09 | ikevin | i think ubuntu use a renamed debian's package |
| 18:51.32 | JackTheNipple | get asterisk running as root for testing |
| 18:51.50 | JackTheNipple | Loki: change the user in asterisk.conf to root |
| 18:52.13 | JackTheNipple | Loki: if this works, you know were to go deeper in |
| 18:52.43 | JackTheNipple | Loki: is this not solving your problem - you also know it does not have to do with permissions |
| 18:52.49 | NickRios05 | just a question i have a DID pointing to my asterisk and whenever a call comes in i get the following any thoughts http://pastebin.com/m386a579e |
| 18:53.11 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 18:54.11 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 18:54.23 | JackTheNipple | NickRios05: did you check whats written there? I mean, wring password does sound quite informative to me |
| 18:56.02 | NickRios05 | right i understand that, but am not sure were to config the string cuz calls get to my server like XXXXXXXXXXX@myvoipprovider |
| 18:57.29 | Loki | JackTheNipple: got it working |
| 18:58.21 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 18:58.31 | NickRios05 | JackTheNipple: cuz am using asterisk for outbound calls right now and it really works fine, am sorry am really a newby to asterisk |
| 18:58.32 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 18:59.13 | JackTheNipple | Loki: what did you do? |
| 18:59.37 | Loki | JackTheNipple: Seems it was a permission issue |
| 19:00.17 | JackTheNipple | NickRios05: does your provider hit you from fixed addresses? |
| 19:00.57 | tzafrir_laptop | ikevin, it indeed does |
| 19:01.17 | NickRios05 | JackTheNipple: Yes, from only one address actually, I guess I have to setup asterisk to allow calls from that ip address right?? |
| 19:01.28 | [TK]D-Fender | [14:52]<NickRios05>just a question i have a DID pointing to my asterisk and whenever a call comes in i get the following any thoughts http://pastebin.com/m386a579e <- that isn't a call coming in, thats a call going OUT |
| 19:03.16 | NickRios05 | [TK]D-Fender: Yes I dialed from the console to check what error does asterisk give, but the first part of the notice appears everytime i get an incoming call |
| 19:06.44 | [TK]D-Fender | NickRios05: that is a OUTBOUND call. Show an actual inbound call and enable SIP DEBUG |
| 19:08.46 | *** join/#asterisk Zuchmir2 (n=Zuchmir-@ool-18bd3bfc.dyn.optonline.net) |
| 19:09.06 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 19:13.24 | *** join/#asterisk CyberPony (n=CyberPon@cpe-071-075-195-085.carolina.res.rr.com) |
| 19:13.46 | Zuchmir2 | having trouble setting up asterisk over VPN |
| 19:14.19 | Zuchmir2 | it rings, but the voice does not work |
| 19:14.42 | NickRios05 | [TK]D-Fender: This is what i get http://pastebin.com/m4cce584b |
| 19:15.26 | [TK]D-Fender | NickRios05: Coplete call from beginning to end |
| 19:15.30 | [TK]D-Fender | Complete* |
| 19:17.39 | [TK]D-Fender | Zuchmir2: pastebin a caomplete failed call attempt from CLI with SIP debug enabled |
| 19:17.40 | [TK]D-Fender | ~pb |
| 19:17.41 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 19:21.25 | NickRios05 | [TK]D-Fender: I think this is the complete call http://pastebin.com/m253ded6c |
| 19:22.59 | [TK]D-Fender | NickRios05: What is sending you that call? |
| 19:23.30 | NickRios05 | [TK]D-Fender: My providers softswitch |
| 19:23.47 | [TK]D-Fender | NickRios05: For your peer entry do "insecure=port,invite" |
| 19:26.57 | NickRios05 | [TK]D-Fender: Thank you so much it works now |
| 19:32.53 | ikevin | anyone know why i receive this log: |
| 19:32.54 | ikevin | [Jul 5 21:31:01] NOTICE[4919]: chan_sip.c:15851 sip_poke_noanswer: Peer 'kevin' is now UNREACHABLE! Last qualify: 0 |
| 19:33.14 | ikevin | 'kevin' is not a peer account (type=friend) |
| 19:33.57 | [TK]D-Fender | ikevin: Same thing |
| 19:34.13 | ikevin | ? |
| 19:34.39 | [TK]D-Fender | ikevin: friend = peer + user |
| 19:34.54 | ikevin | mmm |
| 19:35.07 | ikevin | why i don't have a similare line on other sip account? |
| 19:35.45 | [TK]D-Fender | ikevin: "Friend" was largely phased out from 1.4 |
| 19:36.02 | [TK]D-Fender | ikevin: And its your setup, don't ask us why you did what you did |
| 19:36.55 | ikevin | i can remove "type=XXX" from my config? |
| 19:37.36 | [TK]D-Fender | ikevin: No, you do need to define a type |
| 19:38.49 | ikevin | mmm, ok, i've never see that "friend" is for both, maybe "user" is better in my case |
| 19:39.37 | [TK]D-Fender | Highly doubt that. |
| 19:39.51 | [TK]D-Fender | ikevin: user is only for inbound from providers |
| 19:41.41 | ikevin | i'm not sure to perfectly understand :x |
| 19:41.54 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 19:43.26 | *** join/#asterisk juanIMP (n=juan@201.245.253.116) |
| 19:44.41 | [TK]D-Fender | ikevin: most of the time all you need is "peer" for providers, SIP phones, etc alike. |
| 19:45.05 | [TK]D-Fender | ikevin: Some providers will require separate auth methods requiring a user entry to handle inbound from them. |
| 19:45.20 | ikevin | ok |
| 19:47.16 | ikevin | i thought peer was only for provider :x |
| 19:48.03 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:51.57 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 20:04.09 | *** join/#asterisk dieno (i=773f8846@gateway/web/freenode/x-4bbd685482c57ed1) |
| 20:04.16 | dieno | hello everyone |
| 20:04.29 | dieno | does any one of you have experience reducing the PDD in Dahdi |
| 20:06.24 | *** part/#asterisk Loki (i=loki@unaffiliated/loki) |
| 20:07.16 | tzafrir_laptop | PDD? |
| 20:07.19 | JackTheNipple | dieno: PDD? |
| 20:08.13 | JackTheNipple | dieno: ss7? |
| 20:09.29 | JackTheNipple | ...PDD (post dialing delay) |
| 20:10.08 | [TK]D-Fender | What "delay" |
| 20:11.41 | JackTheNipple | found this on google (not my request - but I'm interested ;-) : |
| 20:11.53 | JackTheNipple | PDD is one of the indicators which is commonly used to measure the quality of a route in wholesale telecommunications industry. |
| 20:11.57 | rob0 | ADI: Anticipatory Dialing Interface: Sees the user sit down at the phone, determines the number most likely to be dialed, dials it when the phone goes offhook. |
| 20:12.44 | rob0 | That will be included in Asterisk 3.14159 |
| 20:13.02 | JackTheNipple | I did'nt think about this as relevant, but actually worth to think about: |
| 20:13.02 | [TK]D-Fender | mmmmmm pi |
| 20:13.03 | JackTheNipple | http://lists.digium.com/pipermail/asterisk-dev/2006-September/022788.html |
| 20:13.42 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 20:13.59 | carrar | moo |
| 20:14.21 | JackTheNipple | I think Asterisk3 will be able to cook eggs, wash my car and produce TV programms better than we have them today ^^ |
| 20:14.32 | [TK]D-Fender | JackTheNipple: Dial vs "answered" means lite in most cases. If you call me and I let my phone ring, what useful measurement does that give you?> |
| 20:14.50 | [TK]D-Fender | JackTheNipple: So becasue I took 10s to get off my ass to answer the phone, you have a "low quality route"? |
| 20:15.32 | JackTheNipple | [TK]D-Fender: maybe its more the idea of "until it starts to ring", like setup time |
| 20:16.04 | JackTheNipple | [TK]D-Fender: callsetup deducting the ringtime? |
| 20:16.10 | *** join/#asterisk dshap (n=dshap@216-165-39-50.DYNAPOOL.NYU.EDU) |
| 20:16.16 | [TK]D-Fender | JackTheNipple: Except of course that information does not exist |
| 20:16.41 | JackTheNipple | [TK]D-Fender: maybe indeed this sounds too theoretically ;-) |
| 20:19.20 | JackTheNipple | [TK]D-Fender: the "calculation" Rushowr gives as workarround in the above digium-post sounds in deed interesting |
| 20:21.27 | vegbox | I need some help with my x100p card. I got it working, ie, it picks up calls on the default call plan. But I want i to do a bit more. The line coming in to the x100p is extension 250. But I want to be able to make a dial plan that when you dial from another extension (example 120) it will send you directly to your voicemail prompt password |
| 20:21.30 | JackTheNipple | [TK]D-Fender: but I don't get the idea of the (already gone) dieno - Asterisk will not really be able to reduce this setup. Depends too much on the peering-partners... |
| 20:21.53 | vegbox | Can anyone help me set up a sample dialplan that I can follow |
| 20:22.38 | dshap | vegbox: have you read the asterisk book? |
| 20:22.42 | JackTheNipple | vegbox: did you have a look at some dialplans? there are tons of examples for nearly everything out there |
| 20:22.49 | vegbox | Yes |
| 20:22.51 | dshap | vegbox: it walks you through a very simple dialplan and explains it in the most clear of terms |
| 20:22.59 | vegbox | But most of the dialplans are for SIP |
| 20:23.04 | [TK]D-Fender | vegbox: All calls coming in on it land on "s" in the context you specified. Do what you want from there |
| 20:23.05 | vegbox | I am still using analog phone lines |
| 20:23.37 | dshap | vegbox: i'm pretty sure nothing in my dialplan has to do with the fact that i'm using SIP |
| 20:23.51 | JackTheNipple | word... |
| 20:23.55 | vegbox | Well I think my problem is that when I dial from another phone |
| 20:24.07 | vegbox | the x100p/asterisk does not see the ext number calling in |
| 20:24.48 | [TK]D-Fender | vegbox: what "ext number"? |
| 20:24.56 | vegbox | extension number |
| 20:25.13 | vegbox | heres the scenario, i have an old dos voicemail system with an nec elekra phone system |
| 20:25.26 | vegbox | I want to replace this dos voicemail with asterisk, so i picked up an x100p card |
| 20:26.01 | [TK]D-Fender | vegbox: All you've got to work with is caller-id. |
| 20:26.12 | [TK]D-Fender | vegbox: Does your PBX send it? |
| 20:26.13 | vegbox | not the $EXT variable? |
| 20:26.32 | vegbox | it should i mean the old do system knows if you are calling from an extension |
| 20:26.45 | vegbox | wait now that i tink about it |
| 20:26.47 | vegbox | it does not |
| 20:27.10 | vegbox | because currently we need to dial, 250, then 9+three digit extension then vmail box password |
| 20:28.44 | vegbox | so i guess i need to know how to capture input and do voicemail($input) |
| 20:29.36 | *** join/#asterisk ingenius (n=alektro@host117.190-138-52.telecom.net.ar) |
| 20:32.14 | dshap | can anyone help me out with this timezone nightmare i'm having? if i use the PHP function putenv() in a PHP/AGI script, will that affect the timezone asterisk uses for SayUnixTime? |
| 20:32.26 | dshap | err, it seems that it DOES affect this |
| 20:32.30 | dshap | i'm just trying to figure out how |
| 20:33.06 | dshap | i've got my AGI debugs printing out 5:00 PM and asterisk is saying 2:00 PM |
| 20:33.48 | dshap | then when i take out putenv("TZ=US/Eastern") into my script, the output is still 5:00 PM (because of some functionality specific to my application) but asterisk says 5:00 PM |
| 20:33.54 | dshap | out of my script* |
| 20:34.27 | dshap | damn that probably was not clear enough |
| 20:41.22 | dshap | oh damn |
| 20:41.23 | dshap | figured it out |
| 20:41.30 | dshap | i was using the wrong timezone codes |
| 20:41.31 | dshap | *sigh |
| 20:41.33 | dshap | sorry |
| 20:58.03 | *** join/#asterisk Olobola (i=Olobola@86.sub-75-208-107.myvzw.com) |
| 21:23.12 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 21:24.27 | *** join/#asterisk XiXaQ (n=jes@135.137.34.95.customer.cdi.no) |
| 21:31.15 | *** join/#asterisk Loki (i=loki@unaffiliated/loki) |
| 21:36.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:46.45 | Loki | Hello, I am having an issue with an installation of Asterisk 1.4.21.2 with FreePBX2.5.1 and for some reason the system is saying that it can not register with my trunks. They are all reporting as Request Sent in the sip show registry, and then Unreachable in the sip show peers section in CLI. All three of my trunks reply to standard ping requests. |
| 21:51.27 | *** join/#asterisk fernandojdk (n=fernando@189.24.43.12) |
| 21:51.34 | fernandojdk | hi all |
| 21:51.55 | fernandojdk | i have one problem |
| 21:53.07 | fernandojdk | scenario: Calling part makes a call, called party answer. Problem: If called party hangups a call, i able to continue with my prioritys in my extensions.conf |
| 21:54.22 | fernandojdk | but, when the calling party hangups the call, i'm not unable to continue the priority on my extensions |
| 21:55.01 | fernandojdk | i like to continue execution of my extensions after the calling party hangup the call |
| 21:55.07 | fernandojdk | some idea? |
| 21:55.56 | [TK]D-Fender | fernandojdk: Go read about the "h" Asterisk Standard Extension |
| 21:56.38 | fernandojdk | Fender: I'm read about the 'h' entension |
| 21:57.09 | fernandojdk | but, i'm need to execute some functions in the extension before goto h extension |
| 21:58.22 | Loki | aa31 |
| 21:58.24 | Loki | .31 |
| 21:58.49 | Loki | err sorry about that. |
| 21:58.59 | [TK]D-Fender | fernandojdk: Not possible. |
| 21:59.15 | [TK]D-Fender | fernandojdk: Caller hangs up, call goes right to "h". |
| 21:59.21 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:59.53 | fernandojdk | right. Thanks for your help Fender |
| 22:00.06 | fernandojdk | I'm trying another ways. |
| 22:00.08 | fernandojdk | ;) |
| 22:03.29 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 22:05.44 | *** part/#asterisk Loki (i=loki@unaffiliated/loki) |
| 22:13.17 | *** join/#asterisk serph (n=serph@70.49.145.76) |
| 22:47.03 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.216) |
| 23:16.49 | *** join/#asterisk vegbox (n=kevinle@24-205-54-7.dhcp.gldl.ca.charter.com) |
| 23:21.13 | *** join/#asterisk XiXaQ (n=jes@132.127.34.95.customer.cdi.no) |
| 23:22.56 | *** join/#asterisk LeddyHM (n=NONE@you.cant.hack.thisbox.org) |
| 23:25.00 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088800059.dsl.bell.ca) |
| 23:29.22 | JimVanM | wanted to ask a question about the function MEETME_INFO |
| 23:29.39 | JimVanM | I want to use it prior to joining a conference (to limit the number of participants to 2) |
| 23:30.02 | JimVanM | problem is, if I'm the first party into the conference, I get this error when running the function |
| 23:30.11 | JimVanM | [Jul 5 19:21:09] ERROR[8251]: /home/murf/asterisk/1.6.1/main/ast_expr2.y:1068 op_func: Error! 'conference' is not possibly a function name![Jul 5 19:21:09] WARNING[8251]: ast_expr2.fl:441 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '<token>', expecting $end; Input: |
| 23:30.11 | JimVanM | Error: conference (550001) not found > 1 |
| 23:30.30 | JimVanM | it says the conference is not found, which kinda makes sense since there's nobody yet in it |
| 23:30.47 | JimVanM | just not sure if there's a better way to check if anybody is in a room before joining |
| 23:30.53 | JimVanM | any advice would be much appreciated |
| 23:31.32 | JimVanM | (the function works great as long as somebody is in the conference to begin with) |
| 23:32.05 | *** join/#asterisk ahat (n=antonis@cust-201-74.on2.ontelecoms.gr) |
| 23:33.39 | ahat | hey guys, first time in irc, first asterisk installation too, and I would like to ask for some advice |
| 23:35.17 | ahat | I actually managed to complete an asterisk@home installation (asterisk 1.4 - freepbx) on a friends old server, and now I am trying to install asterisk 1.6 and freepbx at my ubuntu desktop |
| 23:37.08 | ahat | after 2 days work it seems to be ok (at least that's how it looks from inside freepbx) but I am trying to register an ekiga softphone running also on my ubuntu desktop and I keep getting a "No matching peer found" |
| 23:37.16 | ahat | anyone been in the same situation? |
| 23:40.48 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 23:44.55 | *** join/#asterisk QaDeS (n=mklaus@dslb-084-056-246-236.pools.arcor-ip.net) |
| 23:45.38 | *** part/#asterisk mgarfias (n=mike@scio-fe2-67-43-74-192.smt-net.com) |
| 23:48.00 | jaytee | ahat, you'll need to make Ekiga use port 5061 instead of 5060 since Asterisk is using that. |
| 23:53.50 | [TK]D-Fender | JimVanM: The function isn't your problem, your EXPRESSION is. |
| 23:54.05 | [TK]D-Fender | JimVanM: You know there is this nifty book I heard about.... |
| 23:54.22 | JimVanM | [TK]D-Fender: LOL! Everybody keeps bugging me about that! |
| 23:55.42 | JimVanM | [TK]D-Fender: I am ashamed to admit that I have been spoiled by having Leif as a friend. I hardly have to look anything up anymore. Just bring over some beers, and I've got my Asterisk Encyclopedia! |
| 23:56.09 | [TK]D-Fender | JimVanM: PB your code adn I'll point you in the right direction... |
| 23:56.25 | JimVanM | exten => s,n,Set(THEIRCONFERENCE=550001) ; normally get this from ${ClientID} |
| 23:56.25 | JimVanM | <PROTECTED> |
| 23:59.47 | [TK]D-Fender | exten => s,n,GotoIf($[0${MEETME_INFO(parties,${THEIRCONFERENCE})} > 1]?conferencefull:joinconference) |