| 00:09.30 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:09.55 | *** join/#asterisk CrazyTux (~brandon@wsip-174-76-16-181.oc.oc.cox.net) |
| 00:10.29 | CrazyTux | Hey guys - by default asterisk voicemail when listening to a voicemail will it move it into the 'Old' messages, it seems this behavior is not being observed. Is there a specific setting to accomplish this? |
| 00:13.48 | Kobaz | not that i know of |
| 00:14.52 | JunK-Y | Kobaz: read of which function? |
| 00:14.57 | *** join/#asterisk CrazyTux (~brandon@wsip-174-76-16-181.oc.oc.cox.net) |
| 00:15.00 | Kobaz | never mind, i think it's my code |
| 00:15.01 | Kobaz | heh |
| 00:15.30 | JunK-Y | so i can go to the training room :) |
| 00:15.33 | Kobaz | sure |
| 00:15.41 | Kobaz | where's that? |
| 00:15.56 | JunK-Y | close to the pool |
| 00:16.05 | Kobaz | what sort of workout is planned? |
| 00:16.14 | Kobaz | i should do my ab workout |
| 00:16.45 | JunK-Y | there's really cheap carpet, no bench-press. |
| 00:16.50 | JunK-Y | only cute machines. |
| 00:16.56 | Kobaz | i should meet you down there... what are you wearing |
| 00:16.57 | Kobaz | hah |
| 00:17.13 | JunK-Y | is too sexy for Kobaz ! :P |
| 00:17.41 | adnc | What is this error, it occures since update from 1.4 to 1.6: "Call rejected, CallToken Support required. If unexpected, resolve by placing address" |
| 00:17.44 | Kobaz | well, before we have some sort of epic |
| 00:17.53 | Kobaz | are you at the gaylord? |
| 00:18.12 | JunK-Y | ya |
| 00:18.14 | Kobaz | heh |
| 00:18.33 | Kobaz | some other people are at the westin or comfort in... wouldn't be wanting to wander around looking for someone that's at a different place |
| 00:18.41 | JunK-Y | look in the elevator, the last button is identified as fitness/pool |
| 00:18.50 | Kobaz | yeah, i saw that |
| 00:19.02 | CrazyTux | Hey guys - by default asterisk voicemail when listening to a voicemail will it move it into the 'Old' messages, it seems this behavior is not being observed. Is there a specific setting to accomplish this? (repost, sorry I got disconnected) |
| 00:19.03 | Kobaz | so, i'll just wander around yelling... juuuunnkkyyyy???!! |
| 00:23.32 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 00:24.16 | Kobaz | i suppose so... okay... going to the fitness room |
| 00:31.15 | adnc | is anyone using asterisk 1.6 with kabeldeutschland? |
| 00:32.10 | WIMPy | adnc: I used to. |
| 00:33.14 | adnc | WIMPy, i just did an update from asterisk 1.4 to 1.6. my kabeldeutschland sip was working fine. now the sip numbers are registered but i can not get calls |
| 00:33.36 | WIMPy | What happens? |
| 00:33.37 | *** join/#asterisk CrazyTux (~brandon@wsip-174-76-16-181.oc.oc.cox.net) |
| 00:33.58 | adnc | sip show peers shows for these UNREACHABLE |
| 00:34.10 | adnc | Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 00:35.12 | *** join/#asterisk jbg_ (~jbg@194.96.124.202.static.snap.net.nz) |
| 00:35.22 | jbg_ | just set up asterisk 1.8 and trying to set up SIP TLS |
| 00:35.39 | jbg_ | when my phone tries to register i see "tcptls.c: FILE * open failed!" in the logs and the phone reports 503 |
| 00:35.46 | WIMPy | adnc: Well, thats a clear statement. |
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| 00:37.02 | adnc | WIMPy, ??? |
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| 00:38.23 | adnc | i don't understand |
| 00:39.17 | WIMPy | adnc: the host is not reachable. |
| 00:39.25 | adnc | WIMPy, no, it is reachable |
| 00:39.44 | adnc | it was also working some minutes ago. and before the update |
| 00:40.10 | WIMPy | resolver problem? |
| 00:40.20 | WIMPy | Does it find the correct IP to connect? |
| 00:41.21 | adnc | WIMPy, resolving is a matter of dns resolver and that is not an asterisk issue |
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| 01:25.57 | Juggie | can chan_gtalk dial free longdistance as google talk can do from within gmail? |
| 01:25.59 | Juggie | i'm guessing no. |
| 01:26.18 | adnc | does anyone know of a freefax service? |
| 01:33.17 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup) |
| 01:33.45 | pabelanger | Juggie: Yup, with 1.8 :) |
| 01:33.56 | Juggie | really? |
| 01:33.57 | Juggie | hot |
| 01:34.09 | Juggie | i'm ashamed to say it but i'm resetting up asterisk for the first time since i moved |
| 01:34.17 | Juggie | which was concidentally 1 year ago last week :P |
| 01:34.18 | dzup2 | anyone recomend me a good channel bank with 20 channels to be use with asterisk? |
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| 01:34.44 | Juggie | dzup2, check voip-info |
| 01:34.55 | pabelanger | Juggie: Ya, it's pretty simple. I use it all the time for outbound |
| 01:35.00 | Juggie | hawt |
| 01:35.03 | Juggie | free LD |
| 01:35.12 | dzup2 | Juggie: there is plenty of interfaces, but i need a expert opinion, thank you |
| 01:35.25 | pabelanger | I finally setup a GV number last night in the coding zone. So, next week I'll be playing with it |
| 01:35.36 | Juggie | dzup2, there arnt that many over 16 ports |
| 01:35.44 | Juggie | and you'll find comments etc on the wiki |
| 01:35.49 | *** join/#asterisk rossand (~aross@207.219.49.68) |
| 01:35.55 | Juggie | if that doesnt satisfy your question, post to asterisk-users. |
| 01:36.19 | *** part/#asterisk rossand (~aross@207.219.49.68) |
| 01:36.33 | dzup2 | Juggie: is for a hotel with 20 rooms |
| 01:37.14 | dzup2 | i guess one with 24 be ok |
| 01:37.20 | adnc | how can i find out if a line is busy and if so with what number is he calling? |
| 01:39.54 | Kobaz | because JunK-Y said 10:00 -5 |
| 01:39.56 | Kobaz | er |
| 01:40.22 | Juggie | hmmm |
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| 01:40.46 | Juggie | dahdi trunk is failing to compile for me |
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| 02:26.54 | moltar_net | I know this might be "out there", but is it possible to forward SMS messages via DID to cellphone number? |
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| 04:40.34 | mikejf | Hi, I am currently setting up an asterisk SIP server, it will eventually be connected to an upstream phone line but not yet, I have configured it with a couple of sip extensions and from both of them I can call the test number and get the "congratulattions" message, but I haven't been able to get them to call each other. |
| 04:42.49 | mikejf | both extensions have an entry in the correct context in extensions.conf, and when I dial the extension from the SIP client it says "Remote user rejected the call" |
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| 04:52.59 | mikejf | Damnit, I spent an hour and a half trying to figure that out then less than 10 minutes after I ask for help I figure out my problem and it was sooo simple |
| 04:53.01 | mikejf | grrrrrr |
| 04:53.57 | mikejf | in extensions.conf I had used Macro(page.... when I should have been using Macro(standard-extension |
| 04:58.42 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
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| 05:11.53 | ChannelZ | snap |
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| 05:46.47 | kaldemar | mikejf: any particular reason you're using macros instead of the Dial application directly? |
| 05:47.22 | mikejf | It was because I didn't know about the Dial application, I have since learned about it and am now fixing that |
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| 05:48.06 | mikejf | although now I can't figure out how to put a wildcard into the extension so that I don't need to specify a new extension everytime there is a new user |
| 05:48.30 | mikejf | I have seen ${EXTEN} get used in Macro however that doesn't seem to work in the Dial application |
| 05:49.09 | mikejf | And Macro strangely stopped working after one call either from each client so I haven't been able to fallback to that |
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| 05:57.11 | mikejf | hmm, the ${EXTEN} is working fine my problem is actually in the client not working |
| 06:01.10 | kaldemar | mikejf: extension patterns are what you're looking for. |
| 06:01.31 | kaldemar | mikejf: EXTEN is a variable that holds the current extension. |
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| 06:02.20 | mikejf | ok, thanks |
| 06:02.56 | kaldemar | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 06:05.47 | shamelessn00b | kaldemar: can you look at my extensions.conf I can't understand the 'G' option available in the dial command |
| 06:06.38 | shamelessn00b | what I want to do is dial to a local channel bridge it to my sip channel keep the SIP channel executing the dialplan but the local channel should block execution on EAGI |
| 06:08.18 | shamelessn00b | http://pastebin.com/1qm6nvts |
| 06:08.42 | shamelessn00b | I dial on the extension 1114 from the SIP phone |
| 06:09.09 | shamelessn00b | G(context^exten^pri): If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. This allows the dialplan to distinguish between the calling and called legs of the call (new in v1.2). You cannot use any options that would affect the post-answer state if this option is used. |
| 06:09.30 | shamelessn00b | Thats the description of the 'G' option for Dial command from voip-info.org |
| 06:10.40 | kaldemar | btw, you should always read the documentation for _your_ version, which can be found in the CLI. core show application Dial. |
| 06:12.21 | kaldemar | shamelessn00b: what is the real purpose of this mindfuck? :) just executing the AGI and some dialplan simultaneously? |
| 06:13.00 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 06:17.49 | shamelessn00b | :P |
| 06:18.00 | shamelessn00b | I need to stream audio to Sphinx 4 |
| 06:18.11 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 06:18.17 | shamelessn00b | while the dialplan executes normally |
| 06:18.32 | shamelessn00b | for the user who dialed extension 1114 |
| 06:18.47 | kaldemar | what version of asterisk are you using? |
| 06:18.57 | shamelessn00b | latest trunk |
| 06:19.12 | kaldemar | i'd take a look at the originate application. |
| 06:19.43 | shamelessn00b | I need to be able to have control in the dialplan when to stream audio |
| 06:19.50 | kaldemar | or do you want the streaming to end when the caller hangs up? |
| 06:20.10 | shamelessn00b | the streaming would end when I close the socket connection in sphinx 4 |
| 06:20.16 | shamelessn00b | the EAGI is opening a TCP socket |
| 06:21.03 | kaldemar | well, try adding /n to the end of your local channel. that makes the local channel behave just like any other, and asterisk won't try to bridge the channels, bypassing the local one. that might make a difference. |
| 06:21.42 | shamelessn00b | if it doesn't bridge them, I wont get any audio on my local channel |
| 06:21.44 | shamelessn00b | :/ |
| 06:23.25 | shamelessn00b | this should iin theory work |
| 06:23.33 | shamelessn00b | thats what the description says |
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| 06:48.37 | bn-7bc | sorry to repaet my self egain but tgis druves me a bit crazy. Is there a way to make Asterisk 1.8.0 nind to ipv4 and ipc6 at the same time (for sip), I have tried evereyring I can tthink of withot luck, any tips? |
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| 06:55.12 | kaldemar | bn-7bc: at least the :: wildcard works. are you trying to bind to specific addresses? |
| 06:56.35 | bn-7bc | kaldmar: no but holdom a sec |
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| 06:57.47 | shamelessn00b | how would I do this, dial to a user and playback an IVR as opposed to the user calling into * |
| 06:58.47 | kaldemar | shamelessn00b: originate somehow. AMI, app Originate or a callfile. |
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| 07:00.09 | bn-7bc | kaødemar: hmm no luck here is the bindaddr lin from zip comf bindaddr = 0.0.0.0 :: |
| 07:00.14 | shamelessn00b | hmm |
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| 07:01.35 | kaldemar | bn-7bc: is that what the actual line looks like? hopefully zip comf is not the name of the file. :P |
| 07:02.00 | shamelessn00b | f this, I modify mixmonitor :D |
| 07:02.29 | bn-7bc | kaldemar: upps tyår filnema_sip.conf whar is the line supposed to look like |
| 07:02.49 | kaldemar | bn-7bc: the correct line would be "udpbindaddr=::" |
| 07:03.13 | bn-7bc | tnx |
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| 07:03.56 | asterisk-learner | hello |
| 07:06.03 | shamelessn00b | hi |
| 07:06.45 | bn-7bc | kaldemar: still the sameptoblem astreisk omly binds to eiter ipv4 or ipc6 npr both at the same time |
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| 07:06.53 | asterisk-learner | Does anyone knows if we can whisper on a (IAX2) call, while MOH is turned on ? |
| 07:07.26 | bn-7bc | udpbindaddr = 0.0.0.0 |
| 07:07.27 | bn-7bc | udpbindaddr = :: |
| 07:07.51 | bn-7bc | these are the to lines from sip.conf |
| 07:07.59 | kaldemar | bn-7bc: don't use two lines. |
| 07:08.05 | kaldemar | and ditch the spaces. |
| 07:08.18 | kaldemar | bn-7bc: did you try it? as in make a call? |
| 07:09.02 | bn-7bc | no the client wil notregoster |
| 07:09.47 | kaldemar | netstat will only show you one listening socket but there should be an ipv4-mapped address. |
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| 07:15.22 | kaldemar | if /proc/sys/net/ipv6/bindv6only has 1 in it, then you might have a binding issue in the stack. |
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| 07:33.24 | bn-7bc | worksnow |
| 07:33.41 | shamelessn00b | originate Sip/3001 application dial local/1115@default&local/1114@default why would this not work |
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| 07:34.18 | shamelessn00b | the channel which I dial to first gets bridged, the other channel hangs up |
| 07:36.32 | shamelessn00b | http://pastebin.com/Y8eNC1hZ |
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| 07:45.35 | kaldemar | shamelessn00b: with local/1115@default&local/1114@default, whatever channel answers first, gets the call and the other channel is hung up. |
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| 07:47.08 | shamelessn00b | so I can't dial to multiple channels ? |
| 07:47.41 | shamelessn00b | http://www.russellbryant.net/blog/2008/06/18/how-to-use-dial-and-another-application-at-the-same-time/ |
| 07:48.36 | kaldemar | sure you can dial them, but once a channel answers, others are dropped. |
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| 07:50.09 | shamelessn00b | so I can't connect 2 registered SIP users unless I use conference? |
| 07:50.32 | kaldemar | use originate. |
| 07:51.07 | shamelessn00b | inside the dialplan |
| 07:51.33 | shamelessn00b | change caller id call first number, change caller id call second number |
| 07:51.39 | kaldemar | app originate |
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| 10:15.35 | eMBee | good evening |
| 10:15.55 | Chainsaw | Good morning eMBee. |
| 10:19.22 | *** join/#asterisk emc (~cburgdorf@109.231.205.3) |
| 10:19.47 | emc | hi, can anyone recommend a company which can provide land line numbers (premium and non premium) to dial into an asterisk system hooked on the internet? (Gateway) ? ... or can someone give me a hint what I should google for? |
| 10:20.22 | *** join/#asterisk Montys (~dmartinez@nat/digium/x-nlnkjurajfieclns) |
| 10:20.25 | Montys | elo |
| 10:20.58 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 10:22.42 | ectospasm | emc: you probably want a standard telco, ala Bell/AT&T/Verizon (in the US). You'll need a PSTN adapter (analog or digital) |
| 10:22.57 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 10:23.00 | ectospasm | Digium sells a whole line of PSTN adapters |
| 10:23.19 | ectospasm | analog (POTS), digital (T1/E1 PRI) |
| 10:24.11 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:24.18 | Montys | elo, any1 has an idea what this means "Codecs : 0x9fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|g726aal2)" |
| 10:24.29 | ectospasm | emc: or, there are numerous SIP/VoIP providers who provide DIDs ("landline" numbers) |
| 10:24.50 | ectospasm | Montys: that means that side of the VoIP conversation speaks ALL of those codecs |
| 10:25.20 | ectospasm | Montys: during negotiation, the channels will pick which one is best. |
| 10:25.23 | emc | ectospasm: ah, interesting, thanks for the info... can you name one or two? Are there some who do it internationally |
| 10:25.26 | emc | ? |
| 10:25.47 | ectospasm | emc: for a local DID you'll probably need a local or regional provider |
| 10:25.55 | ectospasm | emc: I can't recommend anything specific |
| 10:26.38 | Montys | ectospasm: what I want to know on specific is the "0x9fe" |
| 10:26.38 | ectospasm | Montys: you probably forgot to put "disallow=all" in sip.conf |
| 10:26.53 | ectospasm | Montys: that's the bitmap of all those codecs. |
| 10:27.18 | ectospasm | Montys: each codec is one bit, like G.729 is 0x100 (*I think*, not sure atm) |
| 10:27.41 | ectospasm | all the bits are bitwise and'ed together to get 0x9fe |
| 10:28.11 | Montys | ectospasm: I purposely enabled all the codecs. I just trying to figure out how the codec negotiation works, |
| 10:28.21 | ectospasm | Montys: try one codec at a time |
| 10:28.34 | ectospasm | ...you'll notice the hex value is really only one bit |
| 10:28.45 | ectospasm | When you have two codecs, two bits, etc. |
| 10:29.07 | Montys | as an example, if you have an * system that supports ulaw, alaw and gsm and is connected to another asterisk system that supports gam, alaw and ulaw |
| 10:29.32 | Montys | if I place a call, how those two system determine which is the codec that they going to use |
| 10:29.55 | ectospasm | Montys: the order that they're listed in sip.conf governs which one is preferred over the others. |
| 10:29.57 | Montys | s/gam/gsm* |
| 10:30.04 | emc | ectospasm: ok, thanks, that helped a lot! |
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| 10:40.48 | fauxalliance | hmm... it's about that time.... |
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| 10:55.48 | joel_oliveira | hello |
| 10:55.57 | joel_oliveira | still fighting with this since yesterday morning |
| 10:56.26 | joel_oliveira | is it possible to get the IP address in the logs from someone that tries to make a call without being registered? |
| 10:57.03 | joel_oliveira | something like this guy was facing : http://forums.digium.com/viewtopic.php?f=1&t=74947&p=147355 |
| 10:59.03 | *** join/#asterisk coldsteal (~Antonni@unaffiliated/coldsteal) |
| 11:03.00 | coldsteal | could someone help me? I cant get the voicemail to send me my vm messages. Im getting the following error: WARNING[31939] app_voicemail.c: Unable to launch '/usr/sbin/sendmail -t' (can't create temporary file), the recording are being made successfully in /var/spool/asterisk/voicemail/default/86/tmp/ |
| 11:11.22 | fauxalliance | coldsteal, what version asterisk? |
| 11:13.43 | ectospasm | coldsteal: can you send mail using sendmail -t at the terminal console? |
| 11:13.51 | ectospasm | coldsteal: that's usually what that means |
| 11:14.49 | fauxalliance | sendmail has a 'headquarters'? http://en.wikipedia.org/wiki/File:6475christieave.jpg |
| 11:16.13 | fauxalliance | ectospasm, I think we lost him. |
| 11:17.33 | ectospasm | s'pose so |
| 11:19.33 | coldsteal | i actually use postfix |
| 11:19.34 | coldsteal | asterisk 1.4 |
| 11:19.50 | fauxalliance | coldsteal, the script CALLS SENDMAIL! |
| 11:20.02 | fauxalliance | <ectospasm> coldsteal: can you send mail using sendmail -t at the terminal console? |
| 11:20.03 | jpmcallister | postfix istalls a sendmail wraper |
| 11:20.13 | jpmcallister | should be no problem |
| 11:20.34 | jpmcallister | coldsteal: do as fauxalliance suggest and try to send an email with sendmail -t |
| 11:20.55 | coldsteal | i can send mail fine using the mail cmd |
| 11:20.59 | jpmcallister | coldsteal: what is the output of the command which sendmail |
| 11:21.18 | ectospasm | coldsteal: upon closer inspection, I see the temporary file can't be created |
| 11:21.24 | ectospasm | check to make sure /tmp isn't full |
| 11:22.08 | fauxalliance | (and postfix _actually_ works) |
| 11:22.30 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 11:23.19 | coldsteal | i have plenty of space in my /tmp |
| 11:23.28 | fauxalliance | coldsteal, is it world writable? |
| 11:23.36 | coldsteal | yes |
| 11:23.39 | E-bola | Is 1.8 going to be re-released because of the huge bug for transfers? |
| 11:23.40 | fauxalliance | sendmail doesn't think so |
| 11:23.44 | E-bola | https://issues.asterisk.org/view.php?id=18185 |
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| 11:24.43 | fauxalliance | coldsteal, for the third time... did you try to use sendmail at the console? |
| 11:25.01 | *** join/#asterisk hatrix76 (~ray@213.201.24.127.static.user.ono.com) |
| 11:25.36 | coldsteal | im looking up how to use the sendmail command to send emails now. i have only used mail |
| 11:25.38 | fauxalliance | E-bola, patch from 18192. It resolves the problem. |
| 11:26.41 | fauxalliance | coldsteal, sendmail -t foo@bar.ca |
| 11:26.43 | E-bola | Ya counting on it, but its a brutally serious bug. So would asume a new version woudl be release asap |
| 11:27.02 | ectospasm | E-bola: the devs probably need to get back from Astricon |
| 11:27.11 | fauxalliance | E-bola, or it just gets fixed... plus, all transfers should be 'warm' |
| 11:27.21 | E-bola | warm? |
| 11:27.40 | fauxalliance | patch from 18192. It resolves the problem |
| 11:27.44 | fauxalliance | oops |
| 11:27.48 | fauxalliance | http://en.wikipedia.org/wiki/Call_transfer |
| 11:27.59 | fauxalliance | Other terms commonly used for an announced transfer include attended,consult, full-consult, supervised, conference and warm transfer. |
| 11:28.15 | E-bola | Why on earth should all transfers be attended/warm? |
| 11:28.25 | coldsteal | i did that and now it looks like its waiting for input |
| 11:28.28 | fauxalliance | so calls dont get dropped E-bola |
| 11:28.45 | E-bola | fauxalliance: Umm calls dont get dropped unless there are bugs in the software? |
| 11:29.13 | hatrix76 | Hello, I have a strange problem, normally the system works fine, but a few times a day, in an outgoing call the user hears ringing all the time until the timeout. The other party picks up, but does not hear anything, and for the caller it just keeps ringing. it is worse for some local numbers, calls to another country for example are not affected, some local numbers have a 50/50 chance of hitting this problem, some numbers are even wors |
| 11:29.13 | hatrix76 | have an asterisk and dahdi from ubuntu 10.04.1 LTS installed, a junghanns quadbri in the server, NO irq-sharing, no irq balancing, etc. any idea? |
| 11:29.23 | fauxalliance | E-bola, with a working patch. |
| 11:33.17 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 11:35.06 | coldsteal | sendmail -t email@domain just hangs and looks like its waiting for something |
| 11:35.27 | coldsteal | curser is on lext like and nothing is happening |
| 11:35.33 | jpmcallister | coldsteal: ps fax | sendmail -t email@domain |
| 11:35.57 | jpmcallister | coldsteal: it is waiting for input |
| 11:36.08 | jpmcallister | coldsteal: try this: ps fax | sendmail -t email@domain |
| 11:38.48 | coldsteal | okay |
| 11:39.25 | coldsteal | yeah usually by now i would have an email |
| 11:39.34 | coldsteal | if i were using the mail cmd |
| 11:42.22 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 11:42.57 | coldsteal | yeah i dont get it because fail2ban uses sendmail and works just fine. i just restarted it and recieved emails which were sent via sendmail |
| 11:43.22 | coldsteal | when i did the ps fax|sendmail -t emailaddr i didnt recieve anything |
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| 11:48.30 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 11:48.34 | adnc | sorry, but where was the language setup for the voicemail? i could not find anything in voicemail.conf |
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| 11:53.34 | coldsteal | okay so this works: echo testinf | sendmail -f email@domain -t email@domain |
| 11:53.57 | coldsteal | but if i remove the -f email@domain then it doesnt |
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| 11:54.14 | coldsteal | any ideas? |
| 11:56.27 | coldsteal | okay sendmail -t is working now |
| 11:56.59 | fauxalliance | called it as a ``trusted'' user coldsteal ? |
| 11:58.20 | coldsteal | no i forgot that i have gmail archiving all mail from my server right away so it was skipping my inbox |
| 11:59.27 | coldsteal | okay so how do i allow the asterisk user controll over sendmail? |
| 12:06.23 | angryuser | Good day, i am facing a strange problem, i have converted some wav files 16bit pcm 8000hz mono, and whatever i do asterisk cuts the end, i have tryed to convert with sox/audacity it is still no go, they play well in any player |
| 12:06.45 | angryuser | Can someone give me a 100% way to convert files as needed ? |
| 12:07.43 | *** join/#asterisk pabelanger (~pabelange@12.182.24.2) |
| 12:09.17 | adnc | i'm using asterisk 1.6 unfortunately it doesn't use my german language prompts |
| 12:09.28 | adnc | <PROTECTED> |
| 12:09.36 | adnc | but plays the english one |
| 12:10.20 | jpmcallister | adnc: that means * doesn't located your version of the file |
| 12:10.30 | jpmcallister | adnc: where is the file located? |
| 12:11.05 | adnc | it is debian and i placed them to /var/lib/asterisk/sounds |
| 12:11.23 | adnc | before i tried /usr/share/asterisk/sounds/de and /var/lib/asterisk/sounds/de |
| 12:11.30 | jpmcallister | adnc: you need to put in /var/lib/asterisk/sounds/de/ |
| 12:11.55 | adnc | well there they are |
| 12:12.07 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 12:12.08 | jpmcallister | adnc: ls -d /var/lib/asterisk/sounds/de |
| 12:12.32 | adnc | /var/lib/asterisk/sounds/de |
| 12:12.47 | jpmcallister | adnc: sorry. ls -ld /var/lib/asterisk/sounds/de |
| 12:13.12 | adnc | drw-r--r-- 7 asterisk asterisk 4200 2008-10-31 15:08 /var/lib/asterisk/sounds/de |
| 12:13.22 | adnc | permissions are ok |
| 12:13.30 | adnc | they are gsm files |
| 12:13.31 | jpmcallister | adnc: doesn't look ok |
| 12:13.41 | adnc | no? how does it need to look |
| 12:14.06 | adnc | ahh x bit |
| 12:14.08 | jpmcallister | adnc: drwxrx-rx- |
| 12:14.17 | jpmcallister | yep |
| 12:14.35 | adnc | thats 755 |
| 12:14.44 | jpmcallister | exactly |
| 12:15.10 | adnc | haha, it works. just the 19 cant be pronounced in german |
| 12:15.34 | adnc | the rest is ok |
| 12:15.46 | jpmcallister | adnc: do you have all the digits files? |
| 12:16.06 | adnc | i do actually |
| 12:16.13 | drmessano | I should change all my prompts to german and my MoH to Rammstein |
| 12:16.29 | drmessano | Most. Hardcore. PBX. Ever. |
| 12:16.33 | adnc | in the digits directory there is 19.gsm |
| 12:16.39 | adnc | i remember i had this before aswell |
| 12:16.50 | jpmcallister | adnc: the digits files need to be at /var/lib/asterisk/sounds/de/digits make sure the permissions are for this folder too |
| 12:17.44 | adnc | jpmcallister, thanks again permission problem |
| 12:17.45 | jpmcallister | adnc: check the permissions at digits and letters folders as well |
| 12:17.49 | adnc | ;) |
| 12:18.07 | adnc | can ls only display directory names? |
| 12:19.01 | jpmcallister | <PROTECTED> |
| 12:19.08 | adnc | ;) |
| 12:19.17 | adnc | i think ls can also handle t |
| 12:19.36 | adnc | jpmcallister, vielen dank! |
| 12:19.36 | jpmcallister | ? |
| 12:19.51 | adnc | i think there is a switch for ls aswell |
| 12:20.02 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:20.23 | *** join/#asterisk kslater (~kslater@static-acs-24-154-228-19.zoominternet.net) |
| 12:20.27 | jpmcallister | adnc: welcome |
| 12:21.00 | *** join/#asterisk dkk (~areski@12.182.24.2) |
| 12:21.28 | binbash_ | ls -hal |
| 12:22.47 | coldsteal | i am nolonger getting the warning about sendmail not being able to write the tmp file but im not getting any emails either |
| 12:23.06 | binbash_ | coldsteal what is /var/log/syslog saying :-)? |
| 12:24.43 | jpmcallister | coldsteal: tail -f /var/log/maillog |
| 12:24.44 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:24.53 | kslater | I don't understand something about ARA with MySQL. Usually when you have a package that uses a database backend, there are some steps to create the database and the schema, but I can't seem to find any details about doing this. I'm puzzled. |
| 12:25.12 | kslater | sorry, using 1.8.0 built from source. |
| 12:27.15 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 12:27.32 | coldsteal | well syslog is saying status sent from postfix |
| 12:28.10 | hatrix76 | Hi, a few times a day, in an outgoing call the user hears ringing all the time until the timeout. The other party picks up, but does not hear anything, and for the caller it just keeps ringing. |
| 12:28.27 | hatrix76 | this does not seem to be normal ... anybody any idea? |
| 12:28.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:29.50 | binbash_ | Yeah, are they both in the same network? |
| 12:30.02 | *** join/#asterisk ukine_droid (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 12:31.29 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 12:31.45 | hatrix76 | no, one is SIP the other PSTN (over a junghanns quadbri card) |
| 12:32.24 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 12:32.30 | binbash_ | Okay, and the SIP one doesn't hear anything? |
| 12:33.41 | coldsteal | mail.log http://pastebin.com/P3d6cQdW |
| 12:34.16 | binbash_ | uhh |
| 12:34.20 | binbash_ | it's mailing to user@gmail.com |
| 12:34.23 | binbash_ | or did you edit that |
| 12:34.24 | binbash_ | :P? |
| 12:34.33 | coldsteal | its an edit |
| 12:34.36 | binbash_ | okay :D |
| 12:34.48 | binbash_ | Seems fine, you sure it's not somewhere in your spam box |
| 12:34.54 | binbash_ | or autoarchived? |
| 12:35.16 | coldsteal | yeah i looked every where |
| 12:35.31 | hatrix76 | binbash_: no, the other way round, the SIP is the caller, calling a PSTN number, the SIP hears ringing, the PSTN picks up and answers, SIP keeps hear ringing, PSTN hears nothing, SIP goes on ringing until time-out. only sometimes, and some numbers way worse than others |
| 12:35.42 | binbash_ | hmm |
| 12:35.44 | binbash_ | strange |
| 12:36.09 | binbash_ | I would try to trace it with whireshark or ngrep :) check out what happens with the audio stream. |
| 12:37.32 | hatrix76 | it's driving our receptionists crazy, because it happens most of the time with the taxi number, it's really terrible. I do not believe it's network issue, there's gigabit network all over the place, and because it is so unconsistent .... I fear it might have to do with some sort of signalling issue .... but I am not a specialist regarding this stuff .... |
| 12:37.52 | binbash_ | understandable |
| 12:38.05 | binbash_ | like i said, try to trace it, if it's traceable for you |
| 12:38.22 | WIMPy | hatrix76: Have you tried to collect some more information? |
| 12:39.25 | hatrix76 | well, i can try to create a wireshark trace, but, WIMPy I looked at the logs, but I really did not find out that much, if someone can tell me which logs and which loglevel would show interesting results I'l get them, but I still would need help figuring out what's the real issue |
| 12:40.32 | WIMPy | Well, the normal console putput with debug and verbose turned up should already tell you quite a bit. |
| 12:41.18 | WIMPy | If that doesn't reveal enough a sip debug and pri debug will tell you the whole story. |
| 12:41.27 | hatrix76 | Yes, but it doesn't (at least for me) as I did not see any difference between such a call and a normal call ... but I'l get them again |
| 12:41.44 | WIMPy | What channel driver are you using for the BRI? |
| 12:41.56 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:42.12 | hatrix76 | with channeldriver you mean dahdi? |
| 12:42.56 | hatrix76 | I have the ubuntu packages, 2.2.1-0ubuntu2 |
| 12:43.01 | hatrix76 | for dahdi |
| 12:43.07 | coldsteal | whats the difference between the pager and user email addresses in voicemail.conf? |
| 12:44.01 | hatrix76 | The first thing I thought I will try this weekend is recompiling everything with the newest sources, but I would feel so much better if I knew what's wrong and how to fix it exactly ... |
| 12:44.41 | hatrix76 | the other problem it's a golf-course so they work every day and there is no day off where i can play around, only the nights are available to me :-( |
| 12:45.58 | [TK]D-Fender | coldsteal: Mail can have the ercording attached to it. |
| 12:46.02 | [TK]D-Fender | coldsteal: Main* |
| 12:46.41 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 12:47.01 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 12:48.00 | asterisk-learner | hello |
| 12:50.36 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 12:50.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 12:51.08 | hatrix76 | one very interesting fact may be that it does not affect calls to a different country, there are a lot of outgoing calls cross-border, and these have no problems whatsoever, the only problems I have is with local calls ... |
| 12:52.18 | coldsteal | [TK]D-Fender: thanks |
| 12:53.03 | coldsteal | okay i finally got it emailing correctly, i had to restart the service |
| 12:53.44 | WIMPy | hatrix76: Try to reproduce the problem and show us some logs. Everything else is going to be guesswork. |
| 12:53.52 | coldsteal | but now the from address is the same as the to address |
| 12:54.11 | kslater | Juggie: it's funny that I seem to run into the same people that I see on other irc channels |
| 12:54.18 | hatrix76 | ok, i'l do that, I just hoped someone recognizes the problem ... :-) |
| 12:55.15 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:55.20 | coldsteal | whats the serveremail setting? |
| 12:55.46 | [TK]D-Fender | coldsteal: "From:" <- |
| 12:56.25 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:56.31 | *** join/#asterisk ukine_droid (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 12:56.32 | asterisk-learner | can anyone check this bug and see if he /she faced a similar problem already ? |
| 12:56.37 | asterisk-learner | http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Asterisk_/Q_26576035.html |
| 12:58.35 | *** join/#asterisk iulhk (~adil@119.73.48.150) |
| 12:59.01 | *** part/#asterisk Montys (~dmartinez@nat/digium/x-nlnkjurajfieclns) |
| 12:59.20 | coldsteal | YES its perfect now :) |
| 12:59.49 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 13:00.01 | t_dot_zilla | asterisk SCF? |
| 13:00.02 | [TK]D-Fender | asterisk-learner: No dates on that, not versions, no REAL debug. Trash. |
| 13:00.19 | [TK]D-Fender | asterisk-learner: We don't see the CALL. |
| 13:00.40 | asterisk-learner | asterisk 1.4.10 |
| 13:01.21 | asterisk-learner | both asterisk same version |
| 13:01.49 | [TK]D-Fender | asterisk-learner: That is TWENTY-FIVI releases and 4 BRANCHES OLD. |
| 13:02.08 | [TK]D-Fender | asterisk-learner: Go upgrade to something the maggots havn't finished eating yet... |
| 13:02.23 | [TK]D-Fender | Sorry... TWENTY-SIX |
| 13:02.35 | coldsteal | is it possible to get asterisk working with SMS through any sip provider or do i need to look for something spacific ? |
| 13:03.26 | [TK]D-Fender | coldsteal: * only supports sending SMS on E1. Nothng more. * is not an SMS platform |
| 13:05.19 | t_dot_zilla | is asterisk SCF a complete rewrite of asterisk ? |
| 13:06.54 | pabelanger | t_dot_zilla: no |
| 13:07.40 | t_dot_zilla | is it an addon to asterisk? |
| 13:08.23 | [TK]D-Fender | t_dot_zilla: go read the WIKI |
| 13:08.28 | [TK]D-Fender | t_dot_zilla: and NO. |
| 13:09.17 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 13:09.54 | pabelanger | t_dot_zilla: parallel project |
| 13:15.18 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:15.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:17.39 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 13:22.18 | *** join/#asterisk QN (~mIRC@175.156.199.171) |
| 13:22.31 | QN | hi all. would like to check if there's any way to append a prefix via the Dial Patterns under the outbound route? |
| 13:22.57 | QN | thanks. |
| 13:23.03 | [TK]D-Fender | QbWrong channell.... two doors to the left... |
| 13:23.08 | [TK]D-Fender | ~freepbx |
| 13:23.08 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 13:23.18 | [TK]D-Fender | QN: rather... |
| 13:23.51 | QN | oh. thanks. sorry. |
| 13:24.10 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:25.42 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:34.19 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 13:34.23 | Katty | GOOD MORNING |
| 13:36.30 | robl^laptop | MEDIOCRE MORNING |
| 13:36.54 | Katty | :< |
| 13:36.59 | Katty | applies hugging to robl^laptop |
| 13:37.05 | jpmcallister | Interesting problem. I have hat setup Phone1 --- LEGACYPBX --- E1 --- * SITE1 -----------IAX-------------* SITE2 --- E1 --- LEGACYPBX --- Phone 2 |
| 13:37.06 | jpmcallister | If a make a call from phone1 to phone2 and phone2 is busy I hear two ringtones and them the busy tone. If a make a call from a sipphone at *site1 to phone 2 the call is hangup immediately |
| 13:37.13 | Katty | jpmcallister: ohai |
| 13:37.40 | robl^laptop | Katty: hehe.. thanks! I was expect just a cup of coffee there, but a hug is always welcome |
| 13:38.30 | Katty | :>> |
| 13:38.41 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 13:38.47 | Katty | i only like coffee with milk and creamer. |
| 13:38.53 | Katty | and then iced. |
| 13:41.25 | Katty | crittercam is back up after 4 months |
| 13:41.30 | Katty | infobot: crittercam |
| 13:41.30 | infobot | crittercam is, like, Katty's live broadcast of The Nut House at http://www.ustream.tv/channel-popup/squirrel-critter-cam |
| 13:41.38 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:41.44 | Katty | sadly, no squirrels yet |
| 13:41.49 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 13:43.37 | robl^laptop | sends all 3 of his cats to the crittercam filming location to assist |
| 13:43.48 | telnettech | good morning all.....I am looking for a reliable open source network monitoring software....i want to be able to monitor multiple routers, port utilization, and as a bonus SIP channels of an Asterisk box if possible.....anyone have a good idea for this? |
| 13:44.16 | Katty | robl^laptop: :<< |
| 13:44.27 | Katty | robl^laptop: why would you do that |
| 13:45.19 | jpmcallister | telnettech: nagios.org |
| 13:45.19 | Nugget | telnet is eeeeeeevil! |
| 13:45.43 | robl^laptop | Katty: my cats are "odd". the wouldn't harm any critters.. they would actually be running from the squirrels. last night they found a gecko and all 3 hid under the couch |
| 13:46.06 | telnettech | thanks jpm |
| 13:46.50 | Katty | gosh |
| 13:47.31 | jpmcallister | telnettech: http://www.voip-info.org/wiki/view/Asterisk+monitoring You will find plugins for nagios |
| 13:48.10 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 13:52.03 | *** join/#asterisk Polysics (~Polysics@host113-41-static.25-87-b.business.telecomitalia.it) |
| 13:52.06 | Polysics | hello |
| 13:52.17 | Polysics | i might be about to say an heresy |
| 13:52.24 | Polysics | but is a macro also a context? |
| 13:52.39 | Polysics | ducks under teh table |
| 13:52.46 | [TK]D-Fender | Polysics: Sort-of |
| 13:53.05 | [TK]D-Fender | Polysics: You can "gog" it, but it is also functionally merged with the calling context |
| 13:53.07 | [TK]D-Fender | goto* |
| 13:53.31 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:53.34 | Polysics | so i would be better off putting the functionality in a context, and having the macro Goto that? |
| 13:53.59 | [TK]D-Fender | Polysics: Depends on the functionality of course |
| 13:54.12 | [TK]D-Fender | Polysics: We'd have to see what you're doing to give a better imporession |
| 13:54.16 | [TK]D-Fender | impression* |
| 13:54.31 | Polysics | basically doing FollowMe |
| 13:54.48 | [TK]D-Fender | Polysics: Would need to see the specifics of what you're doing in it |
| 13:54.50 | Polysics | but i was curious as to how to reuse macro content in the eventuality |
| 13:55.06 | kslater | can anyone tell me about what needs to be done to prep mysql to be used with asterisk 1.8.0? |
| 13:55.10 | Polysics | i might be better off not worrying for now and making followem work :-) |
| 13:55.11 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-pktgdknsgjpqhnra) |
| 13:55.50 | [TK]D-Fender | kslater: in what capacity? |
| 13:56.25 | kslater | well, I would like the use the ARA stuff with mysql, but I don't see instructions on how I should create the user / db / schema |
| 13:57.12 | [TK]D-Fender | kslater: There is a scripts folder that should show a sample structure being created |
| 13:58.11 | kslater | right off the main folder? |
| 13:58.15 | kslater | I don't see scripts |
| 13:58.37 | kslater | contrib/scripts is there though |
| 13:58.41 | Qwell | JunK-Y says hi. |
| 13:58.42 | Qwell | to everyone |
| 13:59.02 | [TK]D-Fender | Qwell: Pass on a "salut mon ostie!" |
| 14:01.23 | kslater | I guess maybe I should look for a simple - get Asterisk 1.8 running tutorial first and crawl before I try to walk. |
| 14:03.05 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 14:04.12 | *** join/#asterisk cparker (~cparker@fsf/member/cparker) |
| 14:04.15 | Katty | HAI JUNKY |
| 14:05.27 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:05.31 | [TK]D-Fender | kslater: Actually SQL integration tends to be in the jogging phase... |
| 14:06.01 | kslater | [TK]D-Fender: well said! |
| 14:06.02 | kslater | lol |
| 14:06.25 | kslater | * is like a candy store - so many tasty modules to add |
| 14:07.38 | [TK]D-Fender | kslater: Actually.. they all tend to be "already there", just need you to fill int he parms to have it use them |
| 14:11.46 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
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| 14:28.20 | *** part/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 14:34.48 | *** join/#asterisk suvir (~suvir@ppp-124-120-128-62.revip2.asianet.co.th) |
| 14:34.53 | telnettech | all......looking to find out what the recommended number of simulataneous calls per minute of an asterisk system? |
| 14:35.38 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:36.23 | [TK]D-Fender | telnettech: "depends" |
| 14:36.46 | telnettech | TK: depends on what? |
| 14:37.06 | [TK]D-Fender | telnettech: Machine-spec, what you ar calling out of, etc |
| 14:37.19 | Qwell | 6 |
| 14:37.19 | Qwell | exactly 6 |
| 14:38.06 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:38.33 | Katty | so. |
| 14:38.36 | Katty | i looked at crittercam. |
| 14:38.43 | Katty | and there was a sign in front of it that said i love you. |
| 14:38.57 | Katty | cutest. thing. ever. |
| 14:39.04 | telnettech | TK: I understand that it depends on the machine specs as I have done small pbx style systems. What I am wanting to do something like 8k to 10k calls per minutes with the average call time to be 5 minutes. |
| 14:39.48 | [TK]D-Fender | telnettech: Do you realize that is an incremental-only sample? :) |
| 14:40.16 | [TK]D-Fender | telnettech: And I'm rather sure you'll need multiple large boxes for that |
| 14:40.48 | [TK]D-Fender | telnettech: Or relegate * to being a back-end-only app server and use something more robust at the front end... like everybody else |
| 14:41.36 | telnettech | I understand but that is the requirements that have been given to me to figure out what it wll take to do it... |
| 14:42.12 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 14:42.18 | [TK]D-Fender | telnettech: probably 2-3 servers + SER |
| 14:42.19 | telnettech | so I need to know if Asterisk can do it with the right hardware and be a reliable software or if I need to look other ways |
| 14:48.55 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:50.24 | *** join/#asterisk ukine_droid (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 14:53.29 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
| 14:54.02 | jkroon | hi guys, on asterisk 1.6.2.13 I have a very weird IAX/2 handoff problem. |
| 14:55.34 | jkroon | three * boxes, SIP phone on A, passes to B via IAX/2 passes to C via IAX/2, C sends to SIP phone, IAX/2 tries to handoff connection direct between A and C, I see traffic but there comes in a channel discrepency between A and C (B no longer knows about the call). |
| 14:56.27 | jkroon | C thinks the call (both legs) is Up, whereas A thinks the IAX/2 channel to C is Ringing and the SIP caller is in Ring. |
| 14:56.37 | jkroon | This is obviously wrong, how do I go about trouble shooting? |
| 14:59.35 | [TK]D-Fender | jkroon: Go post a bug on the tracker |
| 15:02.46 | psilikon | I keep getting the following PRI Error: We think we're the CPE, but they think they're the CPE too yet I can still send traffic without a problem. I am using a sangoma a108 with two PRIs and one zhone channel bank. |
| 15:03.41 | WIMPy | You obviousely should configure it to be the network side then. |
| 15:04.50 | psilikon | I tried adding pri_net in zapata.conf but nothing changed. Why would this occur just out of the blue? |
| 15:06.37 | [TK]D-Fender | psilikon: Maybe the Zhone changed. And jsut because you changed a config doesn't make the change effective |
| 15:06.54 | [TK]D-Fender | psilikon: You have wanpipe configs for this as well and need to reload chan_zap too |
| 15:07.27 | psilikon | [TK]D-Fender, yeah I stopped wanpipe1-4 then issed a wanrouter restart after stopping asterisk |
| 15:09.41 | psilikon | Now I see : PRI Error: We think we're the network, but they think they're the network, too. since I changed pri_cpe to pri_net |
| 15:10.44 | leifmadsen | My AstriCon presentations are available now on my website under the Presentations tab: http://www.leifmadsen.com |
| 15:15.08 | WIMPy | psilikon: This time on the other onterface? |
| 15:15.17 | WIMPy | s/ont/int/ |
| 15:15.44 | psilikon | [TK]D-Fender, currently I am using wanpipe drivers 3.2.1, 3.5.x is current. Is there any reason I should not upgrade the drivers? I figure it can only help. |
| 15:16.00 | psilikon | WIMPy, nope same interface |
| 15:16.24 | [TK]D-Fender | psilikon: Well you've stated you're still on zaptel which is a concern on its own... |
| 15:17.20 | psilikon | [TK]D-Fender, yes, and asterisk 1.2.24 |
| 15:17.29 | [TK]D-Fender | psilikon: Ancient.... |
| 15:24.04 | jkroon | [TK]D-Fender, will do. |
| 15:24.55 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 15:27.02 | *** join/#asterisk upb (cmpxchg@preteam.org) |
| 15:27.47 | upb | hi, when using asterisk spool files and MaxRetries, is there anyway in the dialplan to know that the current call is retry x out of y ? |
| 15:31.39 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 15:32.26 | [TK]D-Fender | upb: Not unless you use dialplan code in the Channel: to log its progress. |
| 15:34.55 | upb | [TK]D-Fender: what do you mean by 'Channel:' ? |
| 15:35.16 | [TK]D-Fender | upb: SPOOL FILE <----------- |
| 15:35.52 | upb | sorry i'm not following |
| 15:36.23 | [TK]D-Fender | upb: How is it you're working with spool files and don't even know the parameters you are passing it? |
| 15:36.39 | upb | doesnt the channel specify where the call is placed? |
| 15:36.58 | [TK]D-Fender | upb: You have to be using CHAN_LOCAL for your Channel: to be able to add any kind of tracking code and not dialing out some other tech directly |
| 15:37.00 | upb | how could i put 'dialplan code' in the channel variable |
| 15:37.32 | p3nguin | Yay for local channels. |
| 15:37.40 | upb | i'm using it like this Channel: WOOMERA/g1/53429833 |
| 15:38.12 | p3nguin | If only someone would give some sort of clue how it could be done. |
| 15:38.52 | [TK]D-Fender | upb: Local/53429833@somecontext/n |
| 15:39.07 | [TK]D-Fender | upb: use the DIALPLAQN to add in your progress tracking and have IT Dial() out |
| 15:39.40 | upb | ohhhh |
| 15:40.01 | upb | but why cant i do it without the local channel ? |
| 15:40.14 | upb | since when the call fails, it goes to the failed extension in the context |
| 15:40.56 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:41.34 | WIMPy | psilikon: Did Asterisk already support PRI that time? |
| 15:42.25 | psilikon | WIMPy, yep |
| 15:42.34 | upb | [TK]D-Fender: when i Set() a variable in the failed extension, its not preserved for the next retry right ? |
| 15:43.02 | upb | [TK]D-Fender: but when using chan local to redirect it to another context, it is ? |
| 15:43.13 | [TK]D-Fender | upb: Of course not, it is a channel variable and dies with that channel. Use something else persistent like AstDB, etc |
| 15:43.23 | upb | blaaaaaah |
| 15:44.03 | upb | so what you're suggesting, is to build the retry code myself in the dialplan ? |
| 15:44.11 | upb | and not use MaxRetries ? |
| 15:45.58 | [TK]D-Fender | upb: Both are options |
| 15:46.12 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 15:46.16 | [TK]D-Fender | upb: Either add tracking to the code or do all of the retry attempts in your own code |
| 15:46.25 | [TK]D-Fender | upb: I would probably do the latter |
| 15:46.37 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 15:47.49 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 15:49.44 | upb | okay, thanks |
| 15:50.36 | *** part/#asterisk upb (cmpxchg@preteam.org) |
| 15:54.09 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:07.42 | *** join/#asterisk Bladerunner05 (~Bladerunn@81.174.56.54) |
| 16:14.24 | *** join/#asterisk Jasnejac (kvirc@81.91.107.235) |
| 16:14.33 | Bladerunner05 | hello on asterisk 1.4.17 with my short dialplan http://pastebin.ca/1976426 if no one answer it don't return to menù |
| 16:14.39 | Bladerunner05 | please take a look |
| 16:15.46 | *** join/#asterisk ukine_droid (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 16:16.59 | Gugge | Bladerunner05: show the verbose log from CLI while you make a call |
| 16:17.36 | Gugge | and i assume the 4 last lines are in another context than daytime |
| 16:18.35 | p3nguin | add: exten => s,5,Goto(1) |
| 16:19.12 | p3nguin | or Goto() where ever you want the call to go when there is no answer. |
| 16:19.22 | Bladerunner05 | Gugge: the last lines are in daytime context |
| 16:19.45 | Gugge | Bladerunner05: why do you have two t,1 in your daytime context ? |
| 16:19.47 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:19.54 | Gugge | and two i,1 |
| 16:20.59 | p3nguin | I know, I know! |
| 16:21.25 | p3nguin | I'll wait for his answer, though. |
| 16:21.41 | Gugge | :) |
| 16:22.42 | *** join/#asterisk abel408 (429863dc@gateway/web/freenode/ip.66.152.99.220) |
| 16:22.57 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
| 16:24.21 | abel408 | Hey everyone. I'm having ringback problems. Incomming calls to my asterisk system do not hear any sort of ringing, just silence. I have played around with progressinband with no success. The only thing I see strange in sip debug is the order in which 180 ringing and 183 session progress is called. 183 session progress is sent before 180 ringing. Is this normal? |
| 16:25.35 | *** join/#asterisk ukine_droid (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 16:28.04 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
| 16:28.21 | *** join/#asterisk pabelanger (~pabelange@12.182.24.2) |
| 16:30.49 | Bladerunner05 | sorry, i'm back |
| 16:30.55 | Bladerunner05 | -- Nobody picked up in 15000 ms |
| 16:30.56 | Bladerunner05 | <PROTECTED> |
| 16:30.56 | Bladerunner05 | <PROTECTED> |
| 16:31.12 | Bladerunner05 | I remove the two i,1 and leave only 1 |
| 16:31.22 | Bladerunner05 | but it hang up instead of go to menu again |
| 16:31.50 | *** join/#asterisk oryxtec (~aa@119.152.53.40) |
| 16:32.29 | p3nguin | You don't have any dialplan instructions to go to a menu. |
| 16:33.07 | p3nguin | Don't expect magic; create appropriate dialplan for what you want. |
| 16:33.36 | *** join/#asterisk ukine_droid (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 16:33.37 | Bladerunner05 | p3nguin: I believe that the last 2 lines do that ? |
| 16:33.54 | Bladerunner05 | may u suggest me how to go to menu again ? |
| 16:33.57 | p3nguin | I don't see any instructions to go to any menu. |
| 16:34.06 | Bladerunner05 | sure :) |
| 16:34.22 | p3nguin | If you've changed your dialplan, change your paste as well. |
| 16:34.44 | Bladerunner05 | ok |
| 16:34.46 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 16:34.58 | Katty | i'm having a blonde moment. |
| 16:35.04 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:35.15 | Katty | [Oct 29 11:32:46] NOTICE[2977]: chan_sip.c:20039 handle_request_invite: Call from '110' to extension '500' rejected because extension not found. <- i'm getting that and staring at my extensions.conf |
| 16:35.20 | Katty | which was copied from another machine |
| 16:35.31 | Katty | dialplan show doesn't look right either |
| 16:35.50 | p3nguin | 500 does not exist in the context where the call was sent. |
| 16:36.17 | Bladerunner05 | p3nguin: is here http://pastebin.ca/1976447 |
| 16:36.24 | Katty | p3nguin: yeahi see that. |
| 16:36.30 | Katty | double checks sip.con |
| 16:36.59 | p3nguin | bladerunner05: i,1 is still duplicated. |
| 16:37.11 | Katty | ok. sip show peers shows me in the right context |
| 16:37.23 | Katty | but dialplan show is all snickerdoodled up |
| 16:37.28 | p3nguin | The sip debug will show wheret he call is being sent. |
| 16:37.30 | tzanger | snickerdoodled |
| 16:37.30 | Bladerunner05 | p3nguin: please correct it !!! |
| 16:37.59 | Bladerunner05 | and let * go to background again if no answer |
| 16:38.19 | oryxtec | guys i am having a small issue. i have a sip extension when i dial from a softphone .. i can make call.. but when i register same extension on a server and make call it simply hangup.. please can some 1 have a look to my config file thanks http://pastebin.com/E1KnUWrQ |
| 16:38.35 | Katty | ooh debug. that's a good idea. |
| 16:38.48 | p3nguin | bladerunner05: http://pastebin.ca/1976449 |
| 16:39.09 | Bladerunner05 | p3nguin: thanks I try.. |
| 16:39.50 | p3nguin | I should have changed i,2 to i,n but I missed it. |
| 16:41.14 | Katty | does it show the context at the cli |
| 16:41.16 | Katty | or in the log |
| 16:41.36 | p3nguin | The sip debug will show it. |
| 16:41.45 | p3nguin | (that's on CLI) |
| 16:42.17 | p3nguin | It'll say something like "looking for 500 in your-context." |
| 16:42.41 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:43.00 | p3nguin | My bet is that the peer sending the call in is not matching the peer definition, thereby sending the call to the default context. |
| 16:43.40 | oryxtec | guys plz can any one look on my config and guide me plz ? |
| 16:43.49 | oryxtec | http://pastebin.com/E1KnUWrQ |
| 16:44.24 | Katty | http://pastebin.ca/1976453 <- i'm apparently blind. where's my context. |
| 16:44.30 | ManxPower | Does anyone have recommendations for specific models of PLANTRONICS headsets for use with Polycom 550 (RJ12) phones? I've tried one or two in the past off the polycom compatible headset list and they REALLY SUCKED. |
| 16:45.07 | ManxPower | Katty, your context is in sip.conf |
| 16:45.32 | ManxPower | you would use NORMAL debug, not sip debug to see where the matched incoming call is going to |
| 16:45.43 | Katty | ...oh. |
| 16:45.55 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 16:46.33 | p3nguin | Looking for 500 in TRG-Outgoing |
| 16:46.44 | p3nguin | That broke my theory. |
| 16:47.04 | ManxPower | Katty, the idea is that once the call has come and been authenticated, it is a core asterisk / dialplan thing not a sip thing |
| 16:47.37 | ManxPower | Looking for 500 in TRG-Outgoing (domain 192.168.0.11) |
| 16:47.42 | ManxPower | that is what you are looking for |
| 16:47.43 | Katty | yeah. that's correct |
| 16:47.48 | Katty | but dialplan show... |
| 16:47.51 | Katty | doesn't look right at all |
| 16:48.00 | p3nguin | dialplan show 500@TRG-Outgoing |
| 16:48.13 | Katty | failed |
| 16:48.21 | Katty | extensions.conf looks okay |
| 16:48.24 | Katty | thoughts? |
| 16:48.28 | p3nguin | dialplan reload |
| 16:49.13 | Katty | i don't think that's in 1.6 |
| 16:49.45 | Kobaz | katty |
| 16:49.51 | Kobaz | you're missing astricon |
| 16:50.11 | Katty | dialplan debug doesn't look right either |
| 16:50.28 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 16:50.29 | tompaw | Hello. |
| 16:50.31 | p3nguin | Never heard of that one. |
| 16:50.52 | tompaw | Running 1.6.0.22 here. Blind transfer works, but on attended transfer I'm getting a one-way audio + a lot of noise. |
| 16:50.57 | tompaw | Any idea what could be causing that issue? |
| 16:51.29 | Katty | let me pastebin this |
| 16:51.35 | Katty | someone will recognize what i'm staring at |
| 16:51.42 | Katty | http://pastebin.ca/1976463 |
| 16:52.14 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 16:52.32 | *** join/#asterisk ccooperz28 (~ccooperz2@68.188.5.142) |
| 16:53.06 | p3nguin | This looks like an example dialplan. It doesn't have your TRG-Outgoing context in it. |
| 16:53.22 | Katty | so what would cause asterisk to not read my extensions.conf properly |
| 16:53.37 | Katty | deletes it and remakes it |
| 16:54.30 | ccooperz28 | does anyone know where to get the libical dependancy for res_calendar_exchange on centos 5.5 |
| 16:54.56 | [TK]D-Fender | Katty: your PBX core isn't loaded at all |
| 16:55.11 | p3nguin | hates when that happens |
| 16:55.20 | *** join/#asterisk _pepo_ (c837e07d@gateway/web/freenode/ip.200.55.224.125) |
| 16:55.28 | _pepo_ | hi friends |
| 16:55.48 | [TK]D-Fender | Katty: module load pbx_config.so |
| 16:55.54 | tompaw | Are configs compatible between 1.6.0.x and 1.6.2.x? |
| 16:56.03 | [TK]D-Fender | tompaw: Most for most things |
| 16:56.10 | Katty | OH |
| 16:56.11 | Katty | there we go |
| 16:56.13 | [TK]D-Fender | tompaw: go read the UPGRADE docs included in the tarball |
| 16:56.13 | Katty | how'd that happen |
| 16:56.20 | Katty | or what causes it |
| 16:56.20 | Katty | etc |
| 16:56.24 | tompaw | [TK]D-Fender: thx. Looks like my atxfer issue is asterisk-related, so will try an update... |
| 16:56.26 | [TK]D-Fender | Katty: Check if your modules.conf has autoload enabled or not |
| 16:56.54 | [TK]D-Fender | tompaw: If you say so... so far you've shown us nothing and told us almost as much... |
| 16:57.04 | *** join/#asterisk EndEng (~epierce@mail.endeavoreng.com) |
| 16:57.38 | Katty | weird. |
| 16:57.41 | Katty | autoload=yes |
| 16:57.55 | [TK]D-Fender | Katty: Hrm... ok, is it all good for now? |
| 16:57.59 | tompaw | [TK]D-Fender: to be honest, I don't know what to show... traces, error logs, everything looks normal. My blind transfer works perfectly, but on attended transfer, after I press * and talk to the 2nd number and hang up, and these two legs are connected, all they can hear is noise + one way audio. |
| 16:58.12 | [TK]D-Fender | tompaw: still no useful details. |
| 16:58.32 | Katty | all except for dahdi hicckuping |
| 16:58.33 | Katty | brb |
| 16:59.25 | tompaw | [TK]D-Fender: ok... is atxfer anyhow different from blindxfer? I mean the moment, when the two remote legs are connected to each other. |
| 17:00.04 | [TK]D-Fender | tompaw: Maybe after you get a clue and start properly describing your situation... |
| 17:01.29 | tompaw | Ok, another try: when I place a call to an extension (let's call it 1) and then I perform a blind transfer to another extension (2), it works perfectly fine, and 1 and 2 can talk to each other with the voice flowing through the pbx. |
| 17:02.05 | tompaw | Now, when I try the same with attended transfer, there is a media issue described above. I compared the SIP traces for both scenarios, and found no differences. SIP/SDP headers are exactly the same. |
| 17:02.19 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 17:02.28 | tompaw | sip show channels shows the same stuff, there are no errors reported, and yet - during attended transfer, there is only one-way audio. |
| 17:02.31 | [TK]D-Fender | moves on to more productive things... |
| 17:02.38 | Katty | k well modules are coming up on their own now, but dahdi is still spazing |
| 17:03.00 | Katty | the typical unable to create channel type dahdi cause unkwnon stuffs |
| 17:04.12 | Katty | what does State In Service mean? |
| 17:04.49 | Katty | http://pastebin.ca/1976478 <- |
| 17:07.39 | ManxPower | Katty, pastebin the output of "cat /proc/dahdi/1" assuming you only have one dahdi card |
| 17:07.54 | ManxPower | Does anyone have recommendations for specific models of PLANTRONICS headsets for use with Polycom 550 (RJ12) phones? I've tried one or two in the past off the polycom compatible headset list and they REALLY SUCKED. |
| 17:08.13 | Naikrovek | no experience with headsets really |
| 17:08.14 | Katty | says it's all in use |
| 17:08.15 | Katty | that's odd |
| 17:08.25 | Naikrovek | wish i could help with that one, i may need to get some |
| 17:08.31 | Katty | i'm guessing it's a config |
| 17:08.41 | Katty | http://pastebin.ca/1976485 |
| 17:08.43 | ManxPower | Katty, "in use" means "asterisk is loaded" |
| 17:09.18 | Katty | ohah, k'then |
| 17:09.26 | ManxPower | Katty, ask on #sangoma how to look at the sangoma card status (I assume you are on sangoma) |
| 17:09.39 | ManxPower | sangoma can hide the alarms on the T-1 |
| 17:09.43 | ManxPower | from DAHDI |
| 17:10.06 | ManxPower | I think Sangoma does official support on #sangoma |
| 17:10.15 | tompaw | https://issues.asterisk.org/view.php?id=16287 |
| 17:10.22 | [TK]D-Fender | ManxPower: H261N Binaural on an M22 amp |
| 17:10.23 | Katty | oh i think i found it |
| 17:10.30 | tompaw | That pretty much describes my situation... |
| 17:10.44 | [TK]D-Fender | ManxPower: Forget about headsets without an amp |
| 17:10.53 | Katty | yeah |
| 17:10.55 | Katty | wrong group number |
| 17:10.59 | ManxPower | [TK]D-Fender, does that support pickup/hangup via the headset? |
| 17:11.00 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 17:11.04 | Katty | horay! |
| 17:11.17 | WIMPy | You mean forget about phones without an amp? |
| 17:12.18 | [TK]D-Fender | ManxPower: Sorry, no... not sure what base you'd use for that.. |
| 17:12.27 | [TK]D-Fender | ManxPower: My guys use the HS button to answer. |
| 17:12.35 | [TK]D-Fender | (on the phone) |
| 17:13.01 | ManxPower | [TK]D-Fender, any reason to get the binaural .vs mono? |
| 17:13.57 | WIMPy | ManxPower: Less processor power for noise canelling :-) |
| 17:14.06 | [TK]D-Fender | ManxPower: Helps block out the loud fucker whom cubicles can barely supress at all |
| 17:14.24 | ManxPower | [TK]D-Fender, not a major issue for me, but still an advantage. |
| 17:14.34 | ManxPower | I'm a Devout fan of the M175 |
| 17:14.45 | *** join/#asterisk ukine_droid (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 17:14.56 | ManxPower | Been using them for over 10 years |
| 17:19.17 | [TK]D-Fender | ManxPower: then by all means. |
| 17:19.31 | ManxPower | heh, the M175 is a 2.5mm style. |
| 17:19.44 | [TK]D-Fender | ManxPower: EW |
| 17:19.57 | ManxPower | they work perfect for use with analog and cell phones |
| 17:20.16 | ManxPower | which, until now, is the only time I ever used a headset. 8-0 |
| 17:25.00 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 17:25.36 | EndEng | building asterisk from scratch on ubuntu, what version should i build, latest in svn or specific version? |
| 17:25.40 | *** join/#asterisk mandragor (~ergudicsu@70.158.116.62) |
| 17:25.48 | EndEng | only for voip, no hardware |
| 17:25.50 | mandragor | is there a way to know the state of a device? |
| 17:26.20 | Katty | does dialplan reload include musiconhold.conf? |
| 17:26.34 | p3nguin | no |
| 17:26.41 | Katty | module reload... |
| 17:26.42 | Katty | moh? |
| 17:26.47 | p3nguin | moh reload |
| 17:26.54 | Katty | ty dear |
| 17:27.02 | *** part/#asterisk moltar_net (~Roman@CPE940c6dac4ffb-CM001ade8cc08e.cpe.net.cable.rogers.com) |
| 17:27.15 | *** join/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 17:27.28 | Katty | yay |
| 17:28.59 | *** join/#asterisk jkroon (~jkroon@dsl-241-245-110.telkomadsl.co.za) |
| 17:32.51 | Katty | so who wants to help me import a pgdump |
| 17:33.12 | Katty | :> |
| 17:34.53 | ManxPower | gets the garlic and wooden stake and edges away from Katty |
| 17:34.58 | Katty | :< |
| 17:35.15 | Katty | sniffle. |
| 17:35.24 | Katty | tugs on ManxPower's sleeve |
| 17:35.26 | Katty | pouts |
| 17:45.10 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
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| 18:16.45 | *** join/#asterisk Arnoz (~Arnoz@91.183.59.105) |
| 18:23.34 | *** join/#asterisk powerunits (~aa@119.152.53.40) |
| 18:23.53 | powerunits | hello world... |
| 18:23.55 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:25.02 | powerunits | there is issue on my asterisk box. |
| 18:25.15 | powerunits | i am using voippro.com as my sip career.. |
| 18:25.29 | Qwell | they sound like they are voip professionals. |
| 18:25.33 | tompaw | [TK]D-Fender: I know you're probably dying to know how I solved my xfer problem. Well, I upgraded from 1.6.0 to 1.6.2 and it's gone :-) |
| 18:25.41 | powerunits | when i dial from sip phone eyebeam.. |
| 18:25.49 | Qwell | ~enter |
| 18:25.49 | infobot | the enter key is not a substitute for punctuation. Use a period '.', exclamation mark '!', question mark '?', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 18:25.53 | Qwell | powerunits: ^^^ |
| 18:26.08 | powerunits | it shows trying but on other end it start ringing... but no ring or bell sound on my headset |
| 18:26.16 | powerunits | some time it do like this.. |
| 18:26.25 | powerunits | and some time it connect the cal |
| 18:26.40 | powerunits | any known reason? |
| 18:26.44 | powerunits | i m using asterisk 1.4 |
| 18:27.00 | [TK]D-Fender | tompaw: You took this long without even telling what you were RUNNING. And those aren't even VERSIONS, they aer BRANCHES |
| 18:28.09 | tompaw | 18:50 < tompaw> Running 1.6.0.22 here. Blind transfer works [...] |
| 18:29.07 | powerunits | ?? |
| 18:29.18 | powerunits | some help on my issue? |
| 18:31.37 | jdoe | Qwell: I don't think he understood. |
| 18:32.09 | p3nguin | Type more, press Enter less. |
| 18:33.19 | powerunits | some 1 alive in this room ? :) |
| 18:35.02 | jpmcallister | I've stalled asterisk18 from digium yum repositories. Doesn't it come with jabber and gtalk? |
| 18:36.12 | Qwell | jpmcallister: not yet. it will Soon(TM) |
| 18:36.14 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:36.37 | jpmcallister | :( |
| 18:36.39 | [TK]D-Fender | "Stalled" indeed |
| 18:42.08 | powerunits | please can some 1 guide me? |
| 18:44.05 | *** join/#asterisk capitan (~A@ppp-71-140-67-195.dsl.irvnca.pacbell.net) |
| 18:46.46 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 18:56.10 | ManxPower | jpmcallister, are you willing to help debug Asterisk 1.8? If not, don't get the SVN version, get the released tarball |
| 18:57.58 | jpmcallister | ManxPower: What are you talking about. I get the rpm packages via yum from digium repositories |
| 18:59.09 | jpmcallister | ManxPower: I'm to lazy to compile. I'll just wait for digium to release the binaries with jabber/gtalk |
| 19:00.18 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 19:01.52 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 19:02.08 | ManxPower | jpmcallister, I miss-read, I read that as "svn" not "yum". Only crazy people use Asterisk packages. |
| 19:02.23 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:03.54 | p3nguin | [tk]d-fender: I went to Radio Shack yesterday... told the guy I needed a modular jack for a DSL circuit I was installing. He asks, "Do you need any batteries?" Ugh. |
| 19:04.13 | Qwell | p3nguin: "You've got questions. We've got batteries." |
| 19:04.19 | p3nguin | exactly |
| 19:04.38 | ManxPower | your problem is that you went to radio shack |
| 19:04.48 | ManxPower | find a nearby Granger if you are in the USA |
| 19:04.50 | jpmcallister | ManxPower: then I'm right to use them |
| 19:05.09 | ManxPower | jpmcallister, as long as you don't expect support from people on this channel |
| 19:05.22 | p3nguin | I doubt there's a Grainger within 300 miles of here. |
| 19:05.34 | ManxPower | p3nguin, where are you? |
| 19:05.42 | p3nguin | Southern IL |
| 19:06.10 | ManxPower | I didn't think there would be any near me, but there is one about a 2 hr drive from me. |
| 19:06.17 | p3nguin | Oh, there's one in Saint Louis. Never knew that. |
| 19:06.47 | p3nguin | ~75 miles |
| 19:06.55 | ManxPower | Does walmart still carry such things? |
| 19:07.11 | p3nguin | I'm sure they do. |
| 19:07.19 | ManxPower | They have the incredible ability to stop stocking something, shortly before I want to buy it. |
| 19:07.21 | p3nguin | Radio Shack was about 30 feet away, so I went there. |
| 19:07.45 | p3nguin | I actually went to Sears first, which was about 20 feet away. |
| 19:10.08 | ManxPower | p3nguin, next time ask the guy to show you how to install the batteries in the DSL jack |
| 19:10.47 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:10.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:11.39 | *** part/#asterisk jpmcallister (~EC06113@200.242.28.231) |
| 19:14.21 | Jasnejac | p3nguin: small stores near you |
| 19:14.24 | *** join/#asterisk BMJ (~bmj@12.182.24.2) |
| 19:15.24 | mandragor | I am trying to write a script that sends the incoming call to an attendant that is available for answering the phone. Is this possible? I don't even know how to start. |
| 19:16.08 | p3nguin | It's a small strip mall. |
| 19:18.12 | p3nguin | And my measurements weren't exact. |
| 19:20.11 | ManxPower | mandragor, read The Book |
| 19:20.13 | ManxPower | ~book |
| 19:20.13 | 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, or http://ofps.oreilly.com/ |
| 19:28.02 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 19:28.56 | mandragor | I've read bits and pieces of the book but I didn't see how it could be possible to find out who is available from a queue |
| 19:30.16 | mandragor | I stubled unto the show hints command in the CLI which shows the status of the channel as InUse,Idle,Unavailable but I'm not sure if that is what I need |
| 19:33.38 | ManxPower | an auto-attendant has nothing to do with in use, idle, unavailable |
| 19:34.02 | ManxPower | route your call to a context with your autoattendant in it. |
| 19:34.17 | ManxPower | or a macro, or a subrouting, or an external IVR server. |
| 19:36.47 | mandragor | essentially I want a queue but the call is not answered until a human picks up the phone (it costs us money otherwise) |
| 19:37.13 | p3nguin | How are you going to queue calls if you don't answer them? |
| 19:37.30 | ManxPower | mandragor, http://www.voip-info.org/wiki/view/Asterisk+tips+IVR+menu |
| 19:37.35 | ManxPower | Didn't you say IVR? |
| 19:38.37 | mandragor | p3nguin, I don't want to queue them but I don't want to answer them |
| 19:39.37 | *** join/#asterisk BMJ (~bmj@12.182.24.2) |
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| 19:41.27 | *** part/#asterisk iulhk (~adil@119.73.48.150) |
| 19:45.27 | *** join/#asterisk emc (~cburgdorf@78-105-21-182.zone3.bethere.co.uk) |
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| 19:49.29 | [TK]D-Fender | mandragor: So you're happy having them "ringing" indefinitely until someone answers? |
| 19:50.04 | [TK]D-Fender | mandragor: You know many telcos may cut them off from waiting forever and many people will hang up aft 30s of no answer expecting that it will never get answered |
| 19:50.40 | p3nguin | I hate when someone calls me and they hang up after a mere 3 rings. |
| 19:50.55 | p3nguin | Sometimes I don't have time to get to the phone in just 18 seconds. |
| 19:51.00 | [TK]D-Fender | ~10 sec |
| 19:54.25 | *** join/#asterisk ritzt3ch (~ritzt3ch@ip70-189-221-21.lv.lv.cox.net) |
| 19:55.22 | mandragor | We are paying for an automated sales lead service, so they send the customer to us, if we don't pick after some time they send the call to another company, if we pick up the phone we get charged for the lead, at $30 a lead it's adding up. If we don't answer the call we don't get charge and somenoe else can have that lead. |
| 19:55.33 | ritzt3ch | dam gerbils |
| 19:57.02 | telnettech | mandragor: sounds like you need to hire more sales people to take those leads..... or find another service to provide the leads |
| 19:58.31 | mandragor | we are hiring more people but we also don't want to have a lot of people sitting around waiting for a call |
| 19:59.06 | mandragor | so it would be great if we could have some sort of group and we select someone in that group to send the call to |
| 20:00.04 | telnettech | mandragor: sounds like you need to get with the service provider and tell them how many calls you want per hour and if more than that, to go to other companies that are paying for the same leads |
| 20:00.14 | telnettech | im sure that they can accomodate that request |
| 20:01.14 | ritzt3ch | whats a good tcpdump to get the 5060 Sip Most info too see like 200 300 Etc error messages of not registering correctly |
| 20:01.23 | *** join/#asterisk ukine_droid (~ukine@64.134.182.143) |
| 20:01.26 | mandragor | we do have that, but there are still times when everyone is busy and then a call comes in |
| 20:01.32 | [TK]D-Fender | mandragor:Solution : Originate a Local channel exten passing the current channel name as a target for pickup and have it loop asking to "press 1 to pickup". Then dump that channel into your Queue. |
| 20:03.51 | [TK]D-Fender | checkout time, later |
| 20:04.33 | *** join/#asterisk seanthegeek (~Sean@cpe-65-31-22-4.insight.res.rr.com) |
| 20:04.36 | mandragor | [TK]D-Fender: thanks, it's going to take sometime to understand all that |
| 20:05.09 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 20:07.06 | seanthegeek | Hi, I'd like to replace my parents magic jack system with an Asterisk box. Is there a reliable SIP carrier that would allow them to port their number? They want to keep the magic jack number, otherwise I'd go with Google Voice. |
| 20:07.31 | p3nguin | Most ITSPs I know of allow number porting. |
| 20:07.46 | Qwell | seanthegeek: it's incredibly unlikely that magicjack would allow you to port their number away |
| 20:07.49 | Qwell | hint: you dont own it |
| 20:08.23 | p3nguin | Does anyone ever "own" a number? |
| 20:08.30 | Qwell | yes |
| 20:08.39 | p3nguin | Under what circumstances? |
| 20:08.49 | Qwell | when one owns it |
| 20:09.36 | seanthegeek | Figured as much. I dispise it so. |
| 20:10.45 | ritzt3ch | tcpdump -i eth1 -n -s0 -v udp port 5060 is that a good one to use to get most of the sip info |
| 20:11.51 | SaiSoma | hey guys, I've just upgraded to 1.8.0 and am attempting to get called party indicators to work. Can you check this to see if my syntax is correct? http://pastebin.com/qALzmkUp |
| 20:13.56 | SaiSoma | ahh . think i have it wrong on my fourth read . .should be using redirecting |
| 20:18.20 | *** join/#asterisk jpmcallister (~EC06113@187.59.101.217) |
| 20:27.28 | kn0x | anyone have a good name for Sip digest users (as opposed to a IP authenticated user being called a Trunk).. I'd say SIP UAs, but really applies to the UAC software and not the 'user' |
| 20:36.07 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 20:36.42 | vinhdizzo | do u have to do anything special to get .wav and .mp3 files working with Background() in asterisk 1.8? I get silence when i point it to a .wav file. |
| 20:36.59 | *** join/#asterisk mcargile (~mikec@office2.vicidial.com) |
| 20:37.27 | mcargile | Anyone have experience with Cisco phones? |
| 20:37.33 | p3nguin | Sure. |
| 20:37.38 | p3nguin | Next question. |
| 20:37.55 | thehar | haha |
| 20:38.00 | mcargile | I got a Cisco 7975g which is not registering properly to my PBX |
| 20:38.11 | Qwell | they do that |
| 20:38.18 | p3nguin | What channel tech are you using on the phone? on the PBX? |
| 20:38.38 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
| 20:38.38 | mcargile | it sends the register. the PBX sends a Trying then Unauthorize, and the phone never follows up |
| 20:38.45 | mcargile | SIP |
| 20:40.29 | mcargile | It is not sending the follow up Register with the authorization header |
| 20:41.10 | ritzt3ch | is there a way to not send AUTHentication OUTBOUND on asterisk |
| 20:42.04 | ritzt3ch | with a sip trunk |
| 20:43.12 | mcargile | you mean do host based authentication? |
| 20:46.10 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 20:47.38 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 20:48.14 | ritzt3ch | im trying to register to a PBX and im getting a |
| 20:48.15 | ritzt3ch | Reason: SIP;text="Username not recognized by registrar" |
| 20:53.52 | *** join/#asterisk jpmcallister (~EC06113@187.59.225.136) |
| 20:57.29 | ccooperz28 | ser |
| 20:58.02 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 20:58.04 | QubeZ | hello all |
| 20:58.45 | QubeZ | how would i construct a dialplan line if i needed something to be set (vTO=30) if time is between some range AND variable vQT=1. I can do each one separate but not sure how to AND them. |
| 21:04.52 | *** join/#asterisk lesouvage (~lesouvage@524947AA.cm-4-2b.dynamic.ziggo.nl) |
| 21:06.00 | lesouvage | . |
| 21:06.51 | lesouvage | sorry, testing my new irc client on my iphone. i |
| 21:11.05 | *** join/#asterisk ukine_droid (~ukine@pool-173-78-3-252.tampfl.fios.verizon.net) |
| 21:11.35 | *** join/#asterisk lesouvage (~lesouvage@524947AA.cm-4-2b.dynamic.ziggo.nl) |
| 21:19.38 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 21:20.28 | *** join/#asterisk MrWork (~Mr@host216.ezlinx.net) |
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| 21:29.38 | p3nguin | |
| 21:32.29 | vinhdizzo | can anyone help in getting a wav or mp3 file to work with asterisk? |
| 21:33.20 | p3nguin | Playback(some-file-name-goes-here) |
| 21:34.24 | p3nguin | If it doesn't work, show me that it doesn't work. |
| 21:34.59 | vinhdizzo | p3nguin: it works, but i get silence. i think it has to do with the formatting of the wav file |
| 21:36.17 | p3nguin | Show me that the file is being played. |
| 21:37.25 | vinhdizzo | i just converted to gsm and it plays |
| 21:37.39 | vinhdizzo | p3nguin: u want a full debug? |
| 21:39.41 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 21:44.44 | *** join/#asterisk ukine_droid (~ukine@pool-173-78-3-252.tampfl.fios.verizon.net) |
| 21:45.32 | vinhdizzo | p3nguin: Ground("Gtalk/+1714My10Digit-0605", "/home/vinh/Documents/Church/TNTT/Phone/welcome.wav") in new stack |
| 21:45.32 | vinhdizzo | [Oct 29 14:43:56] WARNING[10472] file.c: File /home/vinh/Documents/Church/TNTT/Phone/welcome.wav does not exist in any format |
| 21:46.18 | p3nguin | "does not exist in any format" That could be a problem. |
| 21:46.38 | vinhdizzo | what does that mean? |
| 21:46.54 | vinhdizzo | I tried it with .wav and without |
| 21:46.55 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 21:47.06 | p3nguin | What is the message without? |
| 21:47.24 | p3nguin | Playback(/home/vinh/Documents/Church/TNTT/Phone/welcome) |
| 21:47.44 | vinhdizzo | one sec |
| 21:48.00 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 21:48.17 | p3nguin | Also, see what "ls -l /home/vinh/Documents/Church/TNTT/Phone/welcome.*" says. |
| 21:49.21 | vinhdizzo | -rw-r--r-- 1 vinh vinh 3260986 2010-10-28 23:59 /home/vinh/Documents/Church/TNTT/Phone/welcome.wav |
| 21:50.05 | vinhdizzo | for without .wav, http://pastebin.com/b43wGsgE |
| 21:52.15 | *** join/#asterisk twanny796 (~twanny@80.77.205.35) |
| 21:52.21 | vinhdizzo | welcome.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz |
| 21:52.46 | vinhdizzo | I converted to 8 pit, 8000 Hz and it still didn't play |
| 21:52.55 | vinhdizzo | but converting to .gsm does work |
| 21:54.45 | vinhdizzo | any thoughts? |
| 22:03.40 | vinhdizzo | p3nguin??? |
| 22:10.44 | *** join/#asterisk ukine_droid (~ukine@pool-173-78-3-252.tampfl.fios.verizon.net) |
| 22:12.32 | *** join/#asterisk ccomp5950 (~ccomp5950@24.204.47.5) |
| 22:13.46 | vinhdizzo | p3nguin: http://pastebin.com/HedVVEvL actually .gsm doesn't work either for me now...very weird |
| 22:14.52 | Gianlu | hello everybody |
| 22:15.34 | Gianlu | does anyone have any idea why the ExtensionState action, which I call by AMI, is throwing a timeout exception? |
| 22:16.16 | twanny796 | cannot see my ISDN card?? |
| 22:20.12 | *** join/#asterisk Skeeter- (~Skeeter@173.182.155.155) |
| 22:20.23 | Skeeter- | is it possible to reg a polycom via sip without a boot server? |
| 22:24.31 | *** join/#asterisk MCIML (~MCIML@c-68-44-177-83.hsd1.nj.comcast.net) |
| 22:25.45 | MCIML | hey guys i'm having an issue when installing the asterisknow iso, everything seemed to install normally but whne i go to boot up i get a kernel panic - not syncing: vfs: unable to mount root fs on unknown-block(0,0) |
| 22:33.14 | *** join/#asterisk ukine_droid (~ukine@pool-173-78-3-252.tampfl.fios.verizon.net) |
| 22:39.13 | *** join/#asterisk ArchGT (~archgt@unaffiliated/archgt) |
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| 22:55.09 | *** join/#asterisk NirS (~NirS@74.11.214.130) |
| 22:55.19 | NirS | evening all |
| 22:59.24 | MCIML | ey |
| 23:05.11 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 23:12.18 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 23:14.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:21.01 | ritzt3ch | SIP response 480 "Temporarily Unavailable where can i find why |
