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| 04:36.21 | joelsolanki | Hello. |
| 04:37.04 | joelsolanki | is there any limit in asterisk for registerations ? because we are planning to deploy it but we have about 600 users so 600 registerations. any suggestions |
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| 05:01.01 | pabelanger | joelsolanki: load test, but you should be fine |
| 05:06.29 | p3nguin | Yes, there is a limit, and your 600 is nearly 10,000 below the limit. |
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| 05:35.06 | joelsolanki | oh ok got you |
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| 05:45.18 | bcrisp | is skype for asterisk worth the money? |
| 05:52.20 | florz | skype is not worth anything, if that is the question |
| 05:52.27 | ChannelZ | I suppose that depends on your own point of view |
| 05:52.46 | ChannelZ | It works |
| 05:55.52 | bcrisp | we're using ulaw codec and wanted to try something that sounds clearer ... small/medium setup |
| 05:56.24 | bcrisp | im a dummy when it comes to codecs and telephony in general so suggestions / advice welcome |
| 05:58.40 | bcrisp | skype uses G729? - were considering this but I read somewhere that it isn't good at transmitting DTMF which we need for IVR |
| 06:02.39 | ChannelZ | It can but I run it in ulaw |
| 06:04.24 | ChannelZ | To be honest I'm not sure but the DTMF might be decoded by the channel driver internally |
| 06:04.25 | bcrisp | in your implementations have you found jittberbuffer useful to enable? |
| 06:11.43 | florz | bcrisp: you do understand that the protocol is (kindof) orthogonal to the codec used, and that you can't improve quality by recoding to a different codec? |
| 06:12.54 | bcrisp | florz, nope |
| 06:14.21 | bcrisp | florz, but would be interested in understanding how I can maximize quality obviously |
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| 06:20.04 | florz | bcrisp: by buying good telephones and using µlaw/alaw only |
| 06:20.14 | florz | bcrisp: if that's not enough, use g.722 |
| 06:20.26 | florz | bcrisp: as in: buy telephones that support it |
| 06:20.46 | florz | bcrisp: for everyone involved in the call, that is |
| 06:23.30 | bcrisp | ok |
| 06:24.08 | bcrisp | florz, thanks for the info |
| 06:24.20 | bcrisp | how are you typing the mu character by the way? |
| 06:24.44 | florz | by pressing altgr-m |
| 06:25.05 | bcrisp | hehe was it worth all that extra work? |
| 06:26.41 | bcrisp | we're experimenting with using Linksys® PAPT2 devices rather than buying expensive voip phones - seems to be working out pretty well so far |
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| 08:51.38 | nik_ | hi |
| 08:52.17 | nik_ | having an issue with outgoing calls on a new asterisk now installation |
| 08:54.08 | nik_ | here is the log http://pastebin.com/ZJdqV9ac |
| 08:54.30 | nik_ | any help with debugging the issue is very much appreciated\ |
| 08:57.22 | WIMPy | Looks like the called party doesn't exist. |
| 09:01.00 | nik_ | called party being the number I am dialing? |
| 09:01.13 | WIMPy | yes |
| 09:01.37 | WIMPy | It's valid, but noone seems to care. |
| 09:01.45 | nik_ | I know for certain that if I dial from my mobile 0117826163 it does work |
| 09:02.30 | nik_ | so it must be something surrounding the rest of the parameters that makes it "not exist" |
| 09:03.00 | nik_ | DAHDI/G2/0117826163 |
| 09:03.10 | nik_ | DAHDI is the driver, right? |
| 09:03.13 | nik_ | what is G2? |
| 09:03.37 | nik_ | and will 0117826163 be the number that is dialed exactly as it is? |
| 09:04.16 | nik_ | I put an extra 0 in front to dial out, but it seems that it gets stripped correctly |
| 09:04.17 | WIMPy | Yes, a group of channels (trunk), yes. |
| 09:04.19 | nik_ | Executing [00117826163@DLPN_local:1] |
| 09:04.25 | nik_ | here it shows with extra 0 in front |
| 09:05.07 | nik_ | can you suggest how should I approach the problem, given that I know that the number is valid and working? |
| 09:05.40 | nik_ | what are the possible reasons for asterisk (or DAHDI) to think that the line got hangup, if dialing an existing number |
| 09:07.17 | WIMPy | Because the switch to it. |
| 09:08.57 | WIMPy | told |
| 09:10.30 | nik_ | sorry, I am not sure what you mean |
| 09:10.32 | tzafrir | nik_: this is ISDN? |
| 09:10.37 | nik_ | yes |
| 09:11.15 | tzafrir | One possible thing to check the "TON" (Type Of Number) |
| 09:11.22 | nik_ | and digium b410p |
| 09:11.31 | WIMPy | You told your elco, you want a connection to that number and tehy told you that noone (nothing) cared. |
| 09:11.39 | tzafrir | If set (not "unknown"), the provider may assume some prefix |
| 09:11.46 | tzafrir | What version of Asterisk is it? |
| 09:11.59 | nik_ | it is the current asterisknow |
| 09:12.20 | tzafrir | The version of asterisk is? 'core show version' ? |
| 09:12.27 | nik_ | Asterisk 1.6.2.11 |
| 09:12.30 | tzafrir | 1.6.2.<something>? |
| 09:12.32 | tzafrir | ok |
| 09:12.55 | nik_ | which file is the TON in? |
| 09:13.12 | tzafrir | So if you didn't set 'pridialplan' or some '*prefix' setting in /etc/asterisk/chan_dahdi.conf, the default is unknown |
| 09:13.42 | tzafrir | (*prefix: nationalprefix, internationalprefix, and such) |
| 09:14.01 | nik_ | pridialplan=unknown |
| 09:14.07 | nik_ | prilocaldialplan=unknown |
| 09:14.25 | nik_ | so I have to set to "national"? |
| 09:15.18 | WIMPy | Unless you have a reason to change it, you'd best stay with 'unknown'/ |
| 09:17.23 | nik_ | ok |
| 09:22.40 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:24.42 | nik_ | so any other tips? |
| 09:24.48 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:33.11 | nik_ | is it possible to be some setting for the hardware perhaps? |
| 09:33.22 | nik_ | I can receive calls |
| 09:33.26 | nik_ | but not dial out |
| 09:34.31 | WIMPy | Did you try different numbers? What were the cause codes? |
| 09:36.42 | nik_ | I tried two numbers |
| 09:36.47 | nik_ | one land line and one mobile |
| 09:36.55 | nik_ | exactly the same thing happens |
| 09:36.57 | nik_ | same log |
| 09:37.04 | nik_ | just the number is obviously different |
| 09:37.10 | nik_ | the cause code is |
| 09:37.21 | WIMPy | Turn on pri intensive debug and paste the output. |
| 09:37.25 | nik_ | Channel 0/2, span 2 got hangup, cause 18 |
| 09:37.29 | nik_ | I believe it is 18 |
| 09:37.49 | nik_ | I did -vvvvvv |
| 09:38.01 | nik_ | what do you mean "on pri intensive" |
| 09:38.05 | nik_ | how do I do that |
| 09:38.08 | WIMPy | I wonder if I havewn't seen chan_dahdi (or is ot libpri?) wrongly giving code 18 on a timeout. |
| 09:38.17 | nik_ | sorry I this is my first asterisk install |
| 09:38.56 | WIMPy | enter 'pri intensive debig span 1' |
| 09:39.20 | WIMPy | Or as you're diallong out group 2, maybe it's not span 1 for you. |
| 09:39.30 | nik_ | ok |
| 09:39.32 | nik_ | let me try |
| 09:40.23 | nik_ | No such command 'pri intensive debug span 2' |
| 09:40.57 | nik_ | what is "pri" for |
| 09:40.58 | WIMPy | Right. New syntax... pri set debug 2 span 2 |
| 09:41.23 | nik_ | better |
| 09:41.24 | nik_ | got |
| 09:41.25 | WIMPy | Primary Rate Interface. Applies to BRI as well. |
| 09:41.25 | nik_ | Enabled debugging on span 2 |
| 09:42.34 | nik_ | ok will put in pastebin |
| 09:43.33 | nik_ | http://pastebin.com/8CUVMnfn |
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| 09:45.40 | WIMPy | 1. your Type Of Numbers are set to national, not unknown. |
| 09:46.06 | WIMPy | 2. You don't receice a reply from your telco. |
| 09:46.59 | WIMPy | That means the reported cause 18 is a bug. |
| 09:47.23 | WIMPy | You can receive calls on that line? Is the audio quality ok? |
| 09:47.46 | nik_ | I can receive calls |
| 09:48.13 | nik_ | quality of voice is great on the receiving side, but it seems on the caller side is very soft |
| 09:48.21 | nik_ | I will investigate this later |
| 09:48.34 | nik_ | let me investigate the TON |
| 09:48.48 | nik_ | <PROTECTED> |
| 09:48.53 | nik_ | is this the one? |
| 09:48.56 | WIMPy | You might have an interrupt problem that causes this. |
| 09:49.01 | WIMPy | yes |
| 09:50.00 | nik_ | hmm |
| 09:50.06 | nik_ | how do I check that |
| 09:50.16 | nik_ | the interrupt problem |
| 09:50.37 | nik_ | also, the TON, I just double checked the chan_dahdi.conf and it is set to unknown |
| 09:50.46 | nik_ | I also realoded asterisk with asterisk reload |
| 09:50.57 | nik_ | where else can I check re the TON? |
| 09:52.08 | WIMPy | You might have to reload chan_dahdi, or just stop and start Asterisk. |
| 09:52.16 | nik_ | ok |
| 09:52.58 | WIMPy | reload does not reload everything. Only most things. |
| 09:54.32 | nik_ | I did now restart of both asterisk and dahdi |
| 09:54.38 | nik_ | it looks a bit different now |
| 09:54.39 | nik_ | http://pastebin.com/Ph4TppZD |
| 09:54.45 | nik_ | but still says National though |
| 09:56.26 | WIMPy | But still no replies, what so ever. That makes it quite astounding that yu can receive calls. |
| 09:57.15 | nik_ | let me try quickly to confirm :) |
| 09:57.50 | WIMPy | It looks more like there's nothing on the other end of that line. |
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| 09:58.59 | WIMPy | See line 114: "link is DOWN" |
| 10:00.53 | nik_ | ops |
| 10:00.54 | *** join/#asterisk datalay (~datalay@85.105.107.83) |
| 10:01.07 | nik_ | it seems that after the restart cannot receive calls either |
| 10:01.58 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:04.43 | nik_ | lights are green |
| 10:05.36 | WIMPy | Light_s_? Do you have multiple lines? |
| 10:07.08 | nik_ | 2 lines |
| 10:07.13 | nik_ | the card has 4 ports |
| 10:07.16 | nik_ | 2 are connected |
| 10:08.30 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:09.08 | WIMPy | Two independant lines or do they come as one logical interface? |
| 10:09.55 | *** join/#asterisk kamanashisroy (~kamanashi@180.234.58.175) |
| 10:10.08 | nik_ | two independent |
| 10:10.12 | kamanashisroy | #firefox |
| 10:10.22 | *** part/#asterisk kamanashisroy (~kamanashi@180.234.58.175) |
| 10:10.30 | nik_ | they even have two different numbers at the moment |
| 10:13.29 | *** join/#asterisk TobSnyder (~Tobias@91-64-185-67-dynip.superkabel.de) |
| 10:15.03 | WIMPy | Ok. And they're both the same? |
| 10:15.46 | WIMPy | What does 'dahdi show status' give? |
| 10:16.36 | nik_ | B4XXP (PCI) Card 0 Span 1 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) |
| 10:16.45 | nik_ | B4XXP (PCI) Card 0 Span 2 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) |
| 10:17.03 | nik_ | not sure what do you mean by the lines being the same |
| 10:17.11 | nik_ | they are two independent ISDN2a lines |
| 10:17.16 | nik_ | each can carry 2 voice calls apparently |
| 10:17.29 | nik_ | and they have two different numbers |
| 10:17.32 | WIMPy | Do they both react the same? |
| 10:17.37 | nik_ | yes |
| 10:17.48 | nik_ | both can be used for incoming and outgoing calls |
| 10:18.18 | WIMPy | No, I mean: Do you get the same results on both? |
| 10:18.38 | nik_ | oh sorry |
| 10:18.42 | nik_ | let me try |
| 10:20.31 | nik_ | sorry, I think after a restart I am getting incoming calls again |
| 10:20.43 | nik_ | let me test that quickly and will run outgoing tests again |
| 10:23.28 | nik_ | ok receiving calls works on both lines |
| 10:23.41 | nik_ | I am gonna try now outgoing again |
| 10:24.12 | nik_ | but the voice on the external side is VERY soft |
| 10:24.18 | nik_ | anyway, one problem at a time |
| 10:25.12 | nik_ | it is different now ... |
| 10:25.19 | WIMPy | That might be te same problem actually. |
| 10:25.37 | nik_ | now I got a voice "the number you have dialed does not exist" |
| 10:25.43 | nik_ | which I think might be coming from the telco |
| 10:25.57 | nik_ | so it seems that it is dialing, but it dialing the number wrongly |
| 10:25.59 | WIMPy | That's probably because of the TON. |
| 10:26.03 | nik_ | probably because of the TON yes |
| 10:26.04 | nik_ | but why |
| 10:26.13 | nik_ | it is set to unknown in the config |
| 10:26.32 | WIMPy | Maybe there's another config. |
| 10:26.53 | WIMPy | Try to grep in /etc/dahdi and /etc asterisk. |
| 10:27.16 | WIMPy | still wonders where that cause 18 is generated... |
| 10:28.38 | WIMPy | Don't know the numbering plan in your place, but you could try to dial without the leading 0. |
| 10:29.16 | tzafrir | TON is not realted to anything in /etc/dahdi |
| 10:29.38 | tzafrir | Is is basically chan_dahdi.conf |
| 10:29.59 | WIMPy | Who knows what's included where... |
| 10:30.23 | WIMPy | tzafrir: Do you have an idea, where to search for that cause 18? |
| 10:30.35 | tzafrir | But you can easily tell what TON you send from the SETUP message in the 'pri debug' trace |
| 10:31.27 | WIMPy | The trace shows national. That's the issue. |
| 10:33.51 | nik_ | I am unable to find much |
| 10:34.46 | nik_ | http://pastebin.com/p5vVzN8x |
| 10:34.53 | nik_ | here is the chan_dahdi.conf |
| 10:35.06 | nik_ | I removed the lines starting with ; |
| 10:37.47 | nik_ | wait |
| 10:37.49 | nik_ | found something |
| 10:37.52 | nik_ | in ./asterisk/users.conf |
| 10:39.30 | WIMPy | sees no channels. |
| 10:40.05 | nik_ | yupeeeeee, it seems to work, after I changed it and restarted everything!!! |
| 10:40.31 | nik_ | THANKS a million for this! |
| 10:42.29 | nik_ | now have to attend to several other less pressing porblems |
| 10:42.57 | nik_ | like the softness of the voice at the external party |
| 10:46.51 | nik_ | something else |
| 10:47.21 | nik_ | when dialing, I have set it up so I dial 0<the number I want to dial> |
| 10:47.29 | nik_ | 0 meaning "get an external line" |
| 10:47.34 | nik_ | I think it is pretty standard |
| 10:47.36 | nik_ | but |
| 10:48.00 | nik_ | at other places, when you dial 0 alone, you get a "dial tone" in the handset |
| 10:48.39 | nik_ | currently, I get no dial tone, I just enter the whole number, wait for a bit, and I see it dials and rings the phone I dialed |
| 10:48.56 | WIMPy | What kind of phones are you using? |
| 10:48.59 | nik_ | so, how do I get a standard dial tone right after keying in 0, to dial external line |
| 10:49.34 | nik_ | at the moment I have only one VoIP phone |
| 10:49.41 | nik_ | let me try find out the model |
| 10:49.52 | nik_ | but I am also using x-lite |
| 10:49.54 | nik_ | SIP phone |
| 10:49.56 | WIMPy | That was enough information :-) |
| 10:49.56 | nik_ | on the PCs |
| 10:50.13 | WIMPy | SIP doesn't support overlap dialling. |
| 10:50.42 | WIMPy | So you can only fake a 2nd dial tone from within the phone. |
| 10:51.01 | nik_ | sorry, Atcom AT-530P |
| 10:51.34 | WIMPy | Or you could just dial 0 and press send and send the rest to your dialplan afterwards. |
| 10:51.50 | WIMPy | But you can't get the usual behaviour. |
| 10:52.07 | nik_ | what is the "recommended way" |
| 10:52.13 | nik_ | set it up in the phone, if phone supports it? |
| 10:52.30 | *** join/#asterisk rethus (~suther@p5087DE5C.dip.t-dialin.net) |
| 10:53.14 | WIMPy | Get used to the new way? |
| 10:53.29 | nik_ | ok, I am in the phone settings |
| 10:54.04 | WIMPy | Some SIP phones can fake overlap dialling, like Snom. But usually SIP does behave differently to a normal phone. |
| 10:54.25 | WIMPy | That's just the way it is. |
| 10:54.32 | nik_ | I see, ok |
| 10:55.24 | nik_ | btw do you know which is the best code, I have g711, g722, g723, g729 |
| 10:55.26 | WIMPy | It's just like with mobile phones. |
| 10:55.48 | nik_ | ok, I understand |
| 10:55.52 | nik_ | anyway it is not too important |
| 10:55.55 | WIMPy | Your line will use G.711. |
| 10:56.29 | WIMPy | You could use normal ISDN phones connected to the other two ports of your card. They can be made to behave the usual way. |
| 10:57.29 | nik_ | no, it is fine, I actually prefer to get everybody to use soft phone |
| 10:58.00 | nik_ | takes less desk space, it is free, and you don't need expensive phone headset |
| 10:58.46 | nik_ | the softness of the incoming voice at the external party, what could be the cause? |
| 10:59.20 | WIMPy | Can you describe that more? |
| 11:01.00 | rethus | i use agi to do a pinrequest, and if nothing is enterd, * should paly another sound. But problem is, * didn't ask for the pin, he jump directly to the second message: |
| 11:01.10 | rethus | http://pastebin.com/kkA4MzJD |
| 11:02.11 | WIMPy | Why do you use an agi to request a pin? |
| 11:02.30 | nik_ | WIMPy: I will do shortly, just want to do few more tests, so I can give you more precise info |
| 11:04.27 | rethus | <PROTECTED> |
| 11:04.31 | rethus | with phpagi |
| 11:04.54 | *** join/#asterisk datalay (~datalay@85.105.107.83) |
| 11:05.11 | WIMPy | rethus: Whatever, but why don't you requst the pin first and one you got it call your script to validate it? |
| 11:05.21 | WIMPy | once |
| 11:05.34 | rethus | WIMPy: its a long storry |
| 11:05.43 | rethus | but it's the only way it works |
| 11:05.51 | WIMPy | It works? |
| 11:07.22 | rethus | yes it does. but i rewrite the script to run a server-application instead of a standalone... and now the request for the pin didn't work anymore... |
| 11:07.28 | rethus | so what xould be the problem? |
| 11:07.42 | rethus | in my pastebin u see the result of agi debug |
| 11:07.58 | nik_ | WIMPy: the softness problem seems to be related to the handset, I tested with x-lite now and the quality is perfect |
| 11:08.03 | rethus | he call the conf-getpin.gsm, but i don't hear it at phone... |
| 11:08.23 | rethus | result for caller is, that if u call u directly hear PIN iinvalid message |
| 11:09.05 | WIMPy | Did you answer() the call before? |
| 11:12.14 | *** join/#asterisk Jamie27 (541925e6@gateway/web/freenode/ip.84.25.37.230) |
| 11:12.23 | rethus | no |
| 11:12.31 | rethus | have i to do this? |
| 11:13.06 | Jamie27 | Hi, I'm new to asterisk and would like to install it but don't know what to choose, freepbx or asteriskGUI? Can anyone advise me? |
| 11:13.09 | WIMPy | Not sure if it happens implicitly whan you play a file via agi. it surely won;t do any harm. |
| 11:13.34 | WIMPy | Jamie27: What dou you want to do with Asterisk? |
| 11:14.02 | Jamie27 | for now testing purposes only but we would like to use it in our callcenter, about 80 users |
| 11:14.21 | rethus | WIMPy: playing files work well... the "pin wrong-message" is played. |
| 11:14.31 | rethus | question is, why i don't be asked for insert pin |
| 11:15.02 | WIMPy | Jamie27: The GUIs limit your possibilities. And if you want to learn about Asterisk you should try without either. |
| 11:15.05 | rethus | even if agi set, he execute my request : <SIP/dev1-00000042>AGI Rx << GET DATA conf-getpin 10000 5 |
| 11:17.10 | Jamie27 | Well to be honest I'm a linux noob and allthough I'm still learning, I would prefer a gui to better understand it, thereby we have end-users (managers) who we would like to give access to some kind of gui |
| 11:17.52 | Jamie27 | what are the main differences between the two? I cannot find this on the asterisk website, maybe you have an url for me? |
| 11:18.26 | WIMPy | Jamie27: A GUI will not help you understand. In fact the opposite may be true. |
| 11:19.02 | WIMPy | Not sure if anyone in here have much experience with the GUIs. They're not supported here. |
| 11:19.20 | Jamie27 | ow ok i didn't know that |
| 11:19.51 | Jamie27 | is there any tutorial on how to use it, config it without gui? |
| 11:20.38 | WIMPy | ~book |
| 11:20.38 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 11:20.52 | rethus | Jamie27: if u try to setup a callcenter with more than 80 Users, it's better to ask a professional to do this, There are manny SIP-Hackers outside, and iff you know nothing about linux AND asterisk, it can be very very expansive if you've ben hacked. Or does the callcenter only for intranet? |
| 11:21.35 | Jamie27 | only intranet environment |
| 11:21.42 | Jamie27 | but I understand what you mean |
| 11:22.26 | Jamie27 | problem is I must first test and see it Asterisk is suitable for us, cannot spend any money on someone install/configuring it. |
| 11:22.47 | rethus | whats your native language |
| 11:22.49 | Jamie27 | at the moment we use SipX and I am curious which one is the beter of the two |
| 11:23.02 | Jamie27 | Dutch |
| 11:23.37 | rethus | this is german, but maybe this book are also in dutch aviable: http://www.das-asterisk-buch.de/2.1/ |
| 11:23.39 | Kevin` | you are unlikely to do a very good job of setting it up or evaluating it if you know nothing about either |
| 11:23.40 | rethus | give it a try |
| 11:24.25 | Kevin` | now, learning will work for that, but I have never seen someone willing to spend time learning for an evaluation comparing products |
| 11:24.28 | Kevin` | this would be the first |
| 11:24.48 | WIMPy | Well, if you can get what you want from a GUI it will surely bee good for you, but if you can't that may just be because of the GUI, not because Asterisk can't. |
| 11:24.56 | rethus | anyone can help with my "GET DATA conf-getpin" problem? |
| 11:25.00 | Jamie27 | I have much time to learn and the most important thing for now is to see if it is a better product than SipX |
| 11:25.29 | Jamie27 | if i really get stuck, i can ask a C programmer we have in our company so I have some options |
| 11:26.48 | rethus | so i use a gui (webapplication) which i created on my own... but i remember freepbx was betterone - and have a larger community |
| 11:27.00 | rethus | maybe try it |
| 11:27.20 | Jamie27 | thanx I think that will be a good thing to start with |
| 11:27.58 | rethus | good luck |
| 11:28.46 | Jamie27 | thanx! |
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| 11:35.49 | rethus | noone? |
| 11:36.09 | WIMPy | rethus: I guess you will have to paste your script. |
| 11:41.14 | rethus | normaly, GET DATA conf-getpin should wait for an input... but seems, the whole script run without any stop |
| 11:48.17 | nik_ | WIMPy: do you know by any chance if there is a way to do a call transfer in x-lite the free version |
| 11:48.49 | nik_ | or do you know of any good softphone which supports call transfers |
| 11:49.03 | WIMPy | You can do it via Asterisk's features. See features.conf. |
| 11:49.33 | WIMPy | Zoiper will do, but only blind transfers in the free version. |
| 11:50.07 | WIMPy | Attended transfers seems to be one of the main thing used as argument for paid versions. |
| 11:50.41 | nik_ | I need attended transfer, e.g. it righs at reception and want to transfer to another user |
| 11:51.14 | nik_ | but I see people referring to ##<extension> to do an attended transfer with x-lite |
| 11:51.33 | WIMPy | ^^ features.conf |
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| 11:56.27 | nik_ | WIMPy: thanks, looking at it |
| 12:04.55 | *** part/#asterisk rethus (~suther@p5087DE5C.dip.t-dialin.net) |
| 12:17.00 | tzafrir | /home/tzafrir/.config/awesome/rc.lua:59: attempt to index field '?' (a nil value) |
| 12:17.23 | tzafrir | line 59 is the first line of the remmed-out lines |
| 12:17.37 | tzafrir | sorry, wrong window |
| 12:27.23 | datalay | <PROTECTED> |
| 12:27.28 | datalay | this screen but i cant see anylink like FreePBX |
| 12:33.55 | *** join/#asterisk rethus (~suther@p5087DE5C.dip.t-dialin.net) |
| 12:34.39 | rethus | extendions.conf.... i do READ to get some data from the caller... how can i check, if nothing is enterd... then goto x ? |
| 12:38.28 | nik_ | WIMPy: something strange though, aftrer a while, I got the same problem as earlier, cannot dial, then restarted dahdi and asterisk, and it worked again |
| 12:39.18 | WIMPy | Does it only affect outbound calls? |
| 12:39.51 | rethus | how can i check if no valid pin is enterd into "READ()?" |
| 12:40.33 | nik_ | WIMPy: I forgot to check before the restart, now everything works fine |
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| 12:40.51 | WIMPy | rethus: ExecIf? GotoIf? |
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| 12:42.28 | rethus | exten => auth,1,GotoIf( (Read(CONG,conf-getchannel,3,skip,3)) ? getPin,1,1: auth,1,8; |
| 12:42.30 | rethus | would this work? |
| 12:42.58 | WIMPy | nik_: Some telcos take teh line down after some time without any activity to save power. The has been a bug that sometimes prevented Asterisk from re-activation it. |
| 12:43.36 | WIMPy | rethus: Didn't you recommend that book earlier on? |
| 12:44.13 | rethus | in the book, there is not info abotu the READ response between the atempt |
| 12:44.27 | WIMPy | rethus: Maybe you have to replace libpri or just try to install Asterisk from souce. |
| 12:45.00 | rethus | why installing? cause in didn't know the response of comand READ ? |
| 12:45.36 | WIMPy | Ups. That one was for nik_. |
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| 12:46.06 | WIMPy | rethus: What does "between the atempt" mean? |
| 12:47.02 | nik_ | WIMPy: will try collect more info before attempting anything, thanks |
| 12:48.15 | rethus | <PROTECTED> |
| 12:49.29 | rethus | i try the following: |
| 12:50.23 | WIMPy | Yes, but I'm not sure, how usefull that is. |
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| 12:50.37 | rethus | user should insert PIN1 (maybe 3 atemt) after that PIN2 (also till 3 times), after that, the script switch to agi to check all these stuff... if right, goto xy, if wrong increment a counter-var and start at beginning again. |
| 12:51.12 | rethus | if the global counter is 3, caller should be loged out |
| 12:52.45 | datalay | "rejected because extension not found" |
| 12:52.49 | datalay | how can handle with it |
| 12:52.52 | rethus | exten => 1,1,GotoIf($["${PINENTRYS}" != "3"]?getPin,1,1:auth,1,8) |
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| 13:19.19 | nik_ | WIMPy: it affects only outgoing calls |
| 13:19.25 | nik_ | it happened again |
| 13:19.33 | nik_ | I will put logs in pastebin |
| 13:20.08 | WIMPy | nik_: That would be the deactivation thing then. |
| 13:20.49 | WIMPy | After a successfull inbound call, outbound calls will work again for a while. |
| 13:23.18 | nik_ | I did such test, but outgoing still fail |
| 13:23.26 | nik_ | http://pastebin.com/zXDQCEGm |
| 13:23.46 | WIMPy | Oh, interesting. even while the inbound call is active? |
| 13:26.44 | nik_ | hmm didn't try that |
| 13:26.53 | nik_ | but after I hang up the inbound |
| 13:26.56 | nik_ | it failed |
| 13:27.09 | nik_ | can you see anything in the logs? what do you think it is? |
| 13:29.00 | WIMPy | I see an alleged reject in line 85. But I can't see any reject. |
| 13:29.32 | WIMPy | I think the real error happens at line 51. |
| 13:29.45 | nik_ | I did that test too, and still no outgoing call |
| 13:30.02 | WIMPy | But sseing all those RRs I'm unsure, what really happened. |
| 13:31.03 | WIMPy | No, got it. I transmits with TEI 127. So it's the TEI management going wrong. |
| 13:31.26 | WIMPy | And as far as I remember TEI management was wart of that deactivation bug. |
| 13:31.41 | WIMPy | You should update. |
| 13:32.12 | WIMPy | ... was part ... |
| 13:34.07 | nik_ | what do I need to update |
| 13:35.33 | WIMPy | libpri. But you need to recompile asterisk (chan_dahdi) with it. So you'll have to do it from source. |
| 13:36.58 | *** join/#asterisk student_1995 (~d@41-134-22-10.dsl.mweb.co.za) |
| 13:37.33 | WIMPy | Or you could try an older verison. It didn't exist so long, I think. |
| 13:40.02 | student_1995 | hi there i am receiving a number that looks like this 1000_bill_home i cut the 3 dif strings and add them to variables ${usernumber} ${username} ${userdest} i then pass them on to the context to be dialed when call ends and i call the h extension i lost some of the variale info |
| 13:40.38 | rethus | GotoIf($["${PINENTRY}" != "3"] |
| 13:40.44 | rethus | can i also use < 3 ? |
| 13:40.51 | rethus | means handle this as integer? |
| 13:40.55 | nik_ | WIMPy: isn't libpri to do with primary rate ISDN only? |
| 13:41.02 | rethus | GotoIf($["${PINENTRY}" < 3] ? |
| 13:41.19 | nik_ | it says on the asterisk site that rethus "... encapsulates the protocols used to communicate over ISDN Primary Rate Interfaces (T1, E1, J1) ..." |
| 13:41.25 | nik_ | I have BRI, not PRI |
| 13:41.28 | WIMPy | rethus: Use the power of the books. Or jut try. :-) |
| 13:42.19 | WIMPy | nik_: Must be something historic. libpri does all the ISDN stuff. In fact there's little difference between BRI and PRI, except for the bandwidth. |
| 13:42.41 | WIMPy | Remember you used _pri_ debug? |
| 13:42.42 | nik_ | ok, just wanted to make sur |
| 13:42.44 | nik_ | sure |
| 13:43.02 | nik_ | what is _pri_ debug? |
| 13:43.17 | rethus | WIMPy: found nothing in the books |
| 13:43.21 | WIMPy | isdn debug. |
| 13:45.09 | rethus | other question: if i use the n operator for priority... can i jump to the last entry of the context with some special char (maybe -1) |
| 13:45.13 | Jasnejac | student_1995: are the variables inheritable? |
| 13:45.25 | nik_ | ok, how can I check the precise version of libpri to make sure I don't have the latest aleady |
| 13:45.43 | nik_ | I have libpri.so.1.4 but the one I downloaded from asterisk is libpri-1.4.11.4 |
| 13:45.45 | WIMPy | rethus you can use labels. |
| 13:46.42 | rethus | mens (mylabel)READ(blahblah) ? |
| 13:46.59 | WIMPy | n(label) |
| 13:47.27 | rethus | such way: exten => auth,n(end),Hangup() ? |
| 13:47.51 | WIMPy | exactely |
| 13:47.59 | rethus | great, thanks |
| 13:48.46 | nik_ | WIMPy: I've build libpri, I just did straight make though, I don't see any installation or configuration instructions |
| 13:49.07 | rethus | andt to jumpt the label i can also use: auth,auth,hangup or only Goto()-statement? |
| 13:49.20 | rethus | (iff label is hangup) |
| 13:50.22 | WIMPy | nik: in libpri just make;make install then the same for asterisk. |
| 13:50.39 | WIMPy | rethus: You use the label as priority. |
| 13:52.24 | rethus | great, works fine. thanks |
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| 14:17.23 | datalay | in extensions.conf |
| 14:17.48 | datalay | default routing: how should be set.. it is: exten => _X.,2,Dial(sip1,20,rt) |
| 14:17.52 | datalay | but it s wrong i think |
| 14:18.01 | datalay | i want to forward all outbound calling to my sip |
| 14:19.08 | nik_ | WIMPy: done the reinstall from source |
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| 14:19.38 | nik_ | outgoing is working for the moment, but will see for how long |
| 14:20.42 | datalay | [Oct 10 17:20:19] NOTICE[12265]: chan_sip.c:15150 handle_request_invite: Call from '6000' to extension '02324121563' rejected because extension not found. |
| 14:21.06 | nik_ | WIMPy: btw, I installed the latest 1.6.2 release, as I am unsure if it is safe to go 1.8 |
| 14:21.51 | WIMPy | I think 1.8 is ok, but it's libpri that's important, not asterisk itself. |
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| 14:25.49 | p3nguin | datalay: extension '02324121563' is not found in the context where you sent the call. |
| 14:26.21 | wwalker | ]/part |
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| 15:18.57 | datalay | how can display call logs |
| 15:19.22 | datalay | i couldnt find it asterisk now |
| 15:19.33 | datalay | on asteriskgui |
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| 15:24.37 | datalay | how can configure asterisk for call logs |
| 15:24.39 | datalay | manager.conf? |
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| 16:53.23 | as001 | Hello is it possible to pass option like in dial command to call file in channel ? I want to limit duration of call. I know I can do that with L(x:y:z) with dial but how can I do that in call file ? |
| 16:54.02 | as001 | Channel=SIP/EXTEN,,L(x:y:z) does not work |
| 16:54.20 | as001 | I mean Channel: ... |
| 17:00.46 | as001 | Solution might be to connect call channel to extension wich limit duration but trouble is that on that extension sits Meetme and I can't see how I can set maximum duration of conference |
| 17:03.19 | ManxPower | use Local/ |
| 17:03.25 | ManxPower | i.e. chan_local |
| 17:04.19 | as001 | what is chan_local ? |
| 17:05.25 | ManxPower | read the localchannel doc in the asterisk source, see the local channel info in voip-info.org, or the Asterisk book |
| 17:05.28 | ManxPower | ~doc |
| 17:05.28 | infobot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 17:05.35 | ManxPower | ~answers |
| 17:05.35 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 17:06.22 | as001 | ok |
| 17:10.23 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 17:10.41 | as001 | thanks I am reading.. |
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| 17:55.30 | giacomo1989 | hi all . is there anyone? |
| 17:55.51 | giacomo1989 | i ve a problem with iax2 |
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| 18:21.50 | KingDavidNYC | Hello everyone |
| 18:22.44 | KingDavidNYC | can somebody please help me with an issue I can't play a file and it is a nating issue |
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| 18:25.22 | Sstudent_1995 | any one here used the MYSQL application for asterisk before? |
| 18:27.49 | Sstudent_1995 | i have used it intensively for a project query databases left right and center, can not test the looad with more that 8 concurrent calls, wanna roll it out as a production system with 80 users, i need someone to set my mind at ease that i it will not crash, if it does say 80 mysql request per minute? |
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| 18:57.22 | DelphiWorld | hi friends |
| 18:57.34 | DelphiWorld | why i get error 483 while registering asterisk to a provider? |
| 18:57.53 | DelphiWorld | to many hop |
| 19:00.14 | p3nguin | Try a closer server. |
| 19:02.55 | DelphiWorld | p3nguin: no choice |
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| 19:17.19 | KingDavidNYC | Hello |
| 19:20.10 | lesouvage | Good evening |
| 19:22.24 | DelphiWorld | hi lesouvage |
| 19:22.38 | DelphiWorld | lesouvage: speak french i guess right? |
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| 19:25.17 | rethus | [general] |
| 19:25.17 | rethus | exten => 1,1,Set(CHANNEL(language)=de); |
| 19:25.18 | rethus | exten => 1,n,Set(CALLER_ID=${CALLERID(num)}); |
| 19:25.39 | rethus | the variable CALLER_ID is not set. any idea why |
| 19:25.53 | DelphiWorld | rethus: later please use pastebin |
| 19:26.17 | rethus | normaly context general should be execute for each following context ?! |
| 19:26.21 | rethus | ok |
| 19:26.59 | ChannelZ | DelphiWorld: what the heck are you connecting TO? the default max hops is something pretty high like 70 |
| 19:27.01 | p3nguin | rethus: You should never have extens in the general context. |
| 19:27.35 | rethus | what u mean? not set a var in general? |
| 19:27.43 | DelphiWorld | ChannelZ: i am connecting to voicenetwork.ca |
| 19:27.47 | rethus | or ever leave general empty at all |
| 19:28.37 | ChannelZ | DelphiWorld: Have you looked at your sip debug? It seems like something else is going on |
| 19:28.39 | p3nguin | rethus: What I meant is: do not configure extensions in the [general] context -- put them in more appropriately named contexts and only use general for its intended purposes. |
| 19:29.08 | DelphiWorld | ChannelZ: that's it... shell reading isn't easy for me;) |
| 19:30.13 | p3nguin | delphiworld: Can you pastebin a debug so someone else can look? |
| 19:30.43 | DelphiWorld | p3nguin: should by very dificult to do it, because no pastebin utility is here and unable to coppy it because is only a small box |
| 19:31.26 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 19:31.55 | p3nguin | delphiworld: Other people manage in similar conditions. |
| 19:32.12 | p3nguin | delphiworld: Maybe you can install a pastebin upload script. |
| 19:32.17 | DelphiWorld | p3nguin: ? |
| 19:32.21 | DelphiWorld | p3nguin: this is uclinux |
| 19:32.26 | rethus | p3nguin: thanks thats helped me |
| 19:32.28 | DelphiWorld | p3nguin: no centos or debian |
| 19:32.49 | p3nguin | rethus: All fixed now? |
| 19:33.02 | *** join/#asterisk Alagar (~Administr@122.164.30.104) |
| 19:33.58 | rethus | nearly :) |
| 19:34.33 | rethus | have still a problem with STREAM FILE privacy-unident... out of my agi-script |
| 19:34.44 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 19:34.49 | [sr] | hellou my friends |
| 19:34.59 | rethus | it doesn't play the sound, but response this on agi debug mode: |
| 19:35.10 | rethus | <SIP/dev1-00000002>AGI Rx << STREAM FILE privacy-unident "" 0 |
| 19:35.11 | rethus | <PROTECTED> |
| 19:37.26 | DelphiWorld | [sr]: lol, [sr]===sip-router project? lol http://www.sip-router.org |
| 19:37.57 | [sr] | DelphiWorld: no, it has no meaning, just [sr] |
| 19:38.12 | DelphiWorld | [ssronly joking with my friend;) |
| 19:38.31 | [sr] | no problemo |
| 19:39.26 | *** join/#asterisk luckman212 (~quassel@pool-96-246-172-233.nwrknj.fios.verizon.net) |
| 19:41.54 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 19:41.55 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 19:42.10 | *** join/#asterisk [netman] (~netman@83.54.225.227) |
| 19:44.43 | rethus | can somebody help to debug my error, that agi don't play my audiofiles? |
| 19:44.57 | rethus | which logs could i check for error? |
| 19:44.59 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 19:45.05 | ChannelZ | the console for starters |
| 19:45.49 | rethus | /var/log/asterisk/messages throw no error |
| 19:46.23 | rethus | /var/log/messages throw no error |
| 19:46.26 | ChannelZ | asterisk -rvvv <-- the console |
| 19:47.28 | rethus | mhh, no errors, only this while doing the call in cli: http://pastebin.com/G8UM0VYV |
| 19:47.54 | rethus | Line 25 say Playing sound... but nothing is played |
| 19:48.25 | rethus | the file exist |
| 19:49.35 | rethus | if i do exten => auth,n(pinrequest),Read(CONG,conf-getchannel,3); in extensions.conf, the sound is played |
| 19:49.44 | rethus | any idea, where i can catch the error? |
| 19:51.29 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:03.03 | DelphiWorld | iveryone have a good night! |
| 20:03.11 | *** part/#asterisk DelphiWorld (~Delphi@41.200.14.45) |
| 20:12.12 | DrDigital | i need to find an asterisk guru that once i get a system up that can come in and configure everything for my clients... I had someone doing elastix systems that seemed to have disappered after doing work with me for nearly a year and a half |
| 20:12.34 | DrDigital | so where would be a good place to find someone? |
| 20:15.27 | p3nguin | If you're talking about just Asterisk without a GUI, I might be willing to do it. It really depends on what you have planned for the system and how much my time is worth to you. |
| 20:21.29 | DrDigital | ive never used straight asterisk |
| 20:21.43 | DrDigital | some needs would ne |
| 20:22.03 | p3nguin | Are you looking for an elastix admin? *snicker* |
| 20:22.23 | DrDigital | id need a gui on some and others not |
| 20:22.38 | DrDigital | and i want to go back to all the elastix installs and replace with asterisk |
| 20:23.08 | DrDigital | the guy i had working with me, he use to work for palosantos |
| 20:23.26 | DrDigital | i have used car dealerships as clients |
| 20:24.14 | DrDigital | 12 extensions with phones, 20 cell phones, time conditions and they use like 20K minutes a month easily |
| 20:25.12 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:26.52 | DrDigital | I have a C7 license, so im able to pull CAT5 cabling and I have a friend with a contractors license so Im able to go into locations and totally rewire and upgrade from really old analog lines, I want someone who can help plan a system give me a price, i add up everything i have to do to be able to hand you the IP, username and password to come in and configure all the hardware and DIDs and trunks etc |
| 20:27.22 | DrDigital | sometimes i have to much work that i turn jobs down or tell them can i get back to them next week/month |
| 20:34.47 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:36.41 | lesouvage | DrDigital: at the Asterisk forum there is a kind of marketplace. I think it is wise to post your request and wait for offers to roll in. |
| 20:37.33 | DrDigital | lesouvage, thanks thats why i asked... was hoping for some suggestions |
| 20:37.37 | lesouvage | Where is your office? |
| 20:37.43 | DrDigital | California |
| 20:38.53 | lesouvage | Well I live in Holland so that would be kind of complicated to do the job. |
| 20:39.41 | lesouvage | But setting up a straight Asterisk system for 12 extensions and 20 cellphones integrated into the system isn't that hard if you stick to the basics. |
| 20:40.17 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:45.17 | DrDigital | my last guy was in ecuador |
| 20:49.24 | p3nguin | What kind of channel tech do you use for integrating cell phones? |
| 20:50.56 | DrDigital | we just used the cell number in the call queue |
| 20:51.19 | DrDigital | however theirs a slight issue, we need to come up with a like check in deal |
| 20:51.36 | DrDigital | if a sales guy cell phone goes dead, his voice mail answers first ring |
| 20:51.44 | DrDigital | keeping anyone else from getting a sales cal |
| 20:51.46 | DrDigital | call* |
| 20:52.32 | DrDigital | and every day one of the girls in the office updates the queue list because the schedule for work days always changes |
| 20:52.46 | DrDigital | and some guys even though its their day off, they still want calls |
| 20:53.05 | DrDigital | and every hour she calls and checks to see whos voice mail she gets to remove them manually |
| 20:53.19 | *** join/#asterisk Hband (~Hband@ip68-2-140-46.ph.ph.cox.net) |
| 20:53.35 | DrDigital | we would like to see where when they get to work or want to be put in call, they call and with CID, they press 1 and the system knows that cell phone/employee is on duty |
| 20:53.56 | DrDigital | and like every 20 minutes the system calls and if they dont answer removes them... or something like that |
| 20:54.06 | DrDigital | open to suggestions and ideas |
| 20:54.57 | DrDigital | im waiting on the email for my asterisk account |
| 20:57.04 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 21:03.18 | rethus | how can i activate eagi in asterisk? |
| 21:08.55 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 21:11.13 | *** join/#asterisk logicwrath (~no@adsl-99-56-133-2.dsl.sfldmi.sbcglobal.net) |
| 21:17.39 | logicwrath | "VERBOSE[6981] chan_sip.c: Retransmitting #4" - What does the term "retransmitting" mean? Does it indicate that Asterisk did not properly read a response from the ITSP and is sending multiple registrations for the same expiry period? |
| 21:20.33 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 21:20.53 | ChannelZ | retransmitting can mean a few different things |
| 21:21.12 | ChannelZ | If it didn't get a reply to a SIP packet, it will retransmit |
| 21:21.42 | ChannelZ | Whether or not that's because the remote end didn't reply, couldn't reply, or the message never made it out of your system in the first place... |
| 21:24.27 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 21:26.27 | logicwrath | either way the term identifies that it is retrying something because of a problem |
| 21:27.08 | logicwrath | is that correct? |
| 21:30.02 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 21:38.49 | p3nguin | I would think yes. |
| 21:39.55 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 21:42.53 | *** join/#asterisk fofware (~Fabian@host199.190-31-51.telecom.net.ar) |
| 21:45.57 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 21:49.35 | *** join/#asterisk [cannibalera] (~cannibale@200-180-6-131.fnsce703.dsl.brasiltelecom.net.br) |
| 21:49.52 | *** part/#asterisk [cannibalera] (~cannibale@200-180-6-131.fnsce703.dsl.brasiltelecom.net.br) |
| 21:52.01 | *** join/#asterisk [netman] (~netman@83.54.226.99) |
| 21:57.33 | logicwrath | when i grep my asterisk log for retransmitting i get: http://pastebin.com/t7YnEzs2 - I think this might be why my ITSP is blocking my connection multiple times a day. I just started having this problem when I upgraded to asterisk 1.6.2.12 - could this be a bug? |
| 21:58.23 | logicwrath | i think it might be sending the registration too many times and not honoring or parsing the responses correctly |
| 22:02.11 | logicwrath | I see the 200 - OK's coming in from the ITSP in this log: http://pastebin.com/awSCDczq |
| 22:02.40 | logicwrath | well actually right now im blocked |
| 22:02.48 | logicwrath | i need to switch proxies so that log is no good |
| 22:13.01 | logicwrath | I switched over the new proxy and I am still re-transmitting over and over: http://pastebin.com/cXCJvmnt & http://pastebin.com/b7fae7p7 |
| 22:13.28 | logicwrath | it seems like some registrations are still trying the old server but I am registered on all 7 trunks with the new proxy |
| 22:18.26 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176003193.dsl.bell.ca) |
| 22:35.14 | ChannelZ | Are you behind a firewall? |
| 22:35.29 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:38.48 | logicwrath | ive tried 2 firewalls |
| 22:38.58 | logicwrath | and i currently have it DMZ'd |
| 22:39.16 | logicwrath | but yes i am behind NAT |
| 22:39.26 | logicwrath | i have it configured for NAT too |
| 22:42.13 | *** join/#asterisk Hband (~Hband@ip68-2-140-46.ph.ph.cox.net) |
| 22:42.21 | ChannelZ | And you've set your externip and localnet in sip.conf? |
| 22:42.44 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 22:43.19 | logicwrath | yes |
| 22:43.46 | ChannelZ | well assuming your network is functioning as it should in spite of your firewall then your ITSP is ignoring you or something else is happening and you'd have to ask them |
| 22:44.28 | logicwrath | did you review the logs? I see SIP 200 responses coming in |
| 22:44.35 | logicwrath | should i be expecting something else |
| 22:45.36 | logicwrath | http://pastebin.com/cXCJvmnt - the first 11 lines of that include a SIP 200 OK |
| 22:47.38 | ChannelZ | it's replying SOMETIMES which is why I doubt your network is setup right |
| 22:48.02 | ChannelZ | (or that the remote site is insane) |
| 22:49.56 | ChannelZ | I don't know what all you have going on but I'm seeing failures to totally different places which is why I suspect your network |
| 22:51.05 | logicwrath | can you give me an example line so i can review? i am new to sip debugging |
| 22:53.24 | ChannelZ | Retransmitting #2 (NAT) to 147.135.32.221:5060 Reliably Transmitting (NAT) to 147.135.0.128:5060 Reliably Transmitting (NAT) to 68.62.25.95:25082 |
| 22:53.29 | ChannelZ | All different IPs |
| 22:54.40 | logicwrath | the other 147 address is the proxy i was using earlier today |
| 22:54.46 | logicwrath | before they blocked me because i register too much |
| 22:54.52 | logicwrath | i had to switch proxy hosts |
| 22:54.58 | logicwrath | the other IP is my softphone at home |
| 22:56.03 | ChannelZ | well if you are "registering too much" or your ITSP is otherwise angry about what you're doing and blocking your IP, there's nothing I can help do about it |
| 22:56.20 | ChannelZ | clearly they are playing some games if that's what you're saying, so you should be calling them |
| 22:57.50 | logicwrath | they are blocking me because my * box is registering too many times. i have seen more than one issues with asterisk 1.6 not properly reading the UDP datagrams |
| 22:58.02 | logicwrath | i am first trying to make sure asterisk is not at fault |
| 22:58.32 | logicwrath | i have called them more than once and they dont like to help asterisk people |
| 22:59.40 | ChannelZ | Then perhaps you should find someone who does |
| 23:00.00 | ChannelZ | I'm running .11 just fine, I'll build .12 for shits and giggles |
| 23:00.01 | logicwrath | yes, i am in the process of doing that. i was thinking about using iCall |
| 23:10.06 | ChannelZ | well .13 is working fine here |
| 23:10.54 | logicwrath | meh, i believe you |
| 23:11.51 | ChannelZ | softphone is regging fine, calls are flowing |
| 23:11.58 | ChannelZ | no retransmissions |
| 23:12.08 | logicwrath | i can send and recieve calls i just retransmit over and over |
| 23:12.14 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 23:18.45 | logicwrath | http://pastebin.com/PJJC5DDr - i am seeing the 200 OK's in this log yet i still see the retransmitting incrementing to the proxy 147.135.0.128 |
| 23:18.47 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 23:19.40 | logicwrath | lines 83-97 is a retransmit |
| 23:19.59 | logicwrath | then 300-314 |
| 23:20.28 | logicwrath | lines 101-111 show the SIP 200 |
| 23:21.00 | *** join/#asterisk fofware (~Fabian@host199.190-31-51.telecom.net.ar) |
| 23:21.54 | logicwrath | am i reading this incorrectly? |
| 23:26.11 | *** join/#asterisk logicwrath-droid (~logicwrat@173-110-142-100.pools.spcsdns.net) |
| 23:26.40 | logicwrath-droid | I got to get pick up some takeout switching to my droid |
| 23:34.43 | *** join/#asterisk Hband (~Hband@ip68-2-140-46.ph.ph.cox.net) |
| 23:46.09 | *** join/#asterisk [cannibalera] (~cannibale@200-180-6-131.fnsce703.dsl.brasiltelecom.net.br) |
| 23:46.22 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 23:52.06 | jdoe | er... |
| 23:52.26 | jdoe | wtf, why is rc3's init script launching me into some pseudo-asterisk console |
| 23:54.52 | ChannelZ | logic: I think your config is screwed that you're registering to the same place multiple times or something.. you register, get a response, and then immediately register again with another call ID from what I can tell |
| 23:56.03 | ChannelZ | oh these are different DIDs I guess |
| 23:58.04 | ChannelZ | Is there a reason you register a different peer for every single one? |